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Cell phone to SIP phone configuration.

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Cell phone to SIP phone configuration.

by Femto » Tue Jul 17, 2012 4:32 am

Hi,
I'm attempting to get a cell phone to connect to a SIP phone. Luckily I have access to the complete system so I'm not limited to any other service provider

I would appreciate some guidance on how to make the connection between them. My setup is the following...

CellPhone <--> BTS <--> BSC <--> MSC <-NETWORK-> ASTERISK-SIP-GATEWAY <--> SIP SOFTPHONE.

Thanks
Femto

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Re: Cell phone to SIP phone configuration.

by david55 » Tue Jul 17, 2012 5:05 am

What is the protocol on NETWORK? How is NETWORK connected to Asterisk?

Some more background would be useful as very few people would be licensed ot operate a BTS and want to do this.

david55: Moves Like Spencer; Posts: 7885; Joined: Fri Sep 26, 2008 5:03 am

Re: Cell phone to SIP phone configuration.

by Femto » Tue Jul 17, 2012 8:29 am

Hi,
I'm fairly new to all this so please forgive the naive answers. The network is the office one. As far as I understand it, all the connections are via Ethernet onto the network. The BSC, MSC and Asterisk are ‘pingable’ from my PC.

The Wireshark log of the connection between my PC and SIP softphone shows the TCP, IPv4 and SIP protocols.

I agree that the BTS element is unusual as I work for a vendor. It’s the general principal of how a cell phone is able to communicate with an internal SIP phone I’m after.

Regards
Femto

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Re: Cell phone to SIP phone configuration.

by Femto » Wed Jul 18, 2012 8:44 am

Hi,
I'm getting closer! My soft SIP phone is now registered to Asterisk. I can ping my MSC (Mobile Switching Center) from Asterisk. I now need to route the SIP phone to the cell phone. Any ideas/hints on how to do this?

Cheers
Femto.

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Posts: 16

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Re: Cell phone to SIP phone configuration.

by david55 » Wed Jul 18, 2012 8:52 am

An MSC would normally use ISDN with the the SS7 mobile user part. You will need a suitable ISDN card, and cabling. You will probably need additional drivers or to support MUP, as I doubt that Asterisk currently supports MUP. The MSC might also use standard SS7, telephone user part, to interface to the fixed telephone network, in which case you would install the same hardware, but configure Asterisk for use with a normal ISDN backbone trunk. (I'm not sufficiently familiar with end user SS7 to know whether Asterisk supports this. I don't believe that TUP is normally extended to the end user in the UK.)

Your reference to pings suggests that your MSC may have some, non-standard, VoIP landline side interface, in which case you will need to read its documentation in detail.

I am still concerned that you seem to have been let loose with equipment that has a potential for disrupting cellular communications in range of it without any understanding of how it works.

david55: Moves Like Spencer; Posts: 7885; Joined: Fri Sep 26, 2008 5:03 am

Re: Cell phone to SIP phone configuration.

by Femto » Wed Jul 18, 2012 9:10 am

Hi,
Thanks for the detailed reply, much appreciated. Hopefully I can alleviate any fears as far as RF goes. In a previous life I was working within the DSP team and have written several device drivers for RF synthesiser chips. To me, that is the easy bit!

It’s true that a have a free reign over the rest of the equipment but it’s for me to setup and fix a bug which only occurs when a cell phone communicates with a SIP phone. My current difficulties are on how to get the various bits of equipment to communicate with each other. My test mobile has been camped on the BTS for over a week!

Most, if not all of the Asterisk examples appear to relate to intra SIP calls.

Cheers
Femto

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Posts: 16

Joined: Tue Jul 17, 2012 4:22 am

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Re: Cell phone to SIP phone configuration.

by david55 » Wed Jul 18, 2012 9:27 am

Using the 1990s, UK, System X architecture, there would normally be a fixed network digital main switching unit and a DSSS concentrator between any mobile switching centre. and where one would normally put Asterisk. I'm not familiar with the current UK backbone architecture.

david55: Moves Like Spencer; Posts: 7885; Joined: Fri Sep 26, 2008 5:03 am

Re: Cell phone to SIP phone configuration.

by Femto » Thu Jul 19, 2012 8:08 am

Hi,
I'm getting closer still! The cell phone is registered on the MSC and is aware of its presence. The SIP soft phone appears to be attempting to contact the MSC. Below is a section of the log as taken from the CLI. Any nudges gratefully received.

Cheers
Femto

Code: Select all: =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2012.07.19 14:18:53 =~=~=~=~=~=~=~=~=~=~=~= asterisk -r Asterisk 1.6.2.11, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 1.6.2.11 currently running on localhost (pid = 3489) localhost*CLI> Verbosity is at least 3 localhost*CLI> <--- SIP read from UDP:172.28.3.149:5060 ---> <-------------> localhost*CLI> <--- SIP read from UDP:172.28.3.149:5060 ---> SUBSCRIBE sip:4412490018@172.28.3.211 SIP/2.0 Via: SIP/2.0/UDP 172.28.3.149:5060;rport;branch=z9hG4bKPjc554f632e97244438a3d0800a88cee43 Max-Forwards: 70 From: "Femto" <sip:pg5@172.28.3.211>;tag=28aa7a96d406496fbc26c848e2627d05 To: <sip:4412490018@172.28.3.211> Contact: "Femto" <sip:pg5@172.28.3.149:5060;ob> Call-ID: 0335c078e21a4fd89e9d3ac053690900 CSeq: 3625 SUBSCRIBE Event: presence Expires: 600 Supported: replaces, 100rel, timer, norefersub Accept: application/pidf+xml, application/xpidf+xml Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.1.7 Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Creating new subscription Sending to 172.28.3.149 : 5060 (no NAT) list_route: hop: <sip:pg5@172.28.3.149:5060;ob> Found peer 'pg5' for 'pg5' from 172.28.3.149:5060 <--- Transmitting (NAT) to 172.28.3.149:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.28.3.149:5060;branch=z9hG4bKPjc554f632e97244438a3d0800a88cee43;received=172.28.3.149;rport=5060 From: "Femto" <sip:pg5@172.28.3.211>;tag=28aa7a96d406496fbc26c848e2627d05 To: <sip:4412490018@172.28.3.211>;tag=as3fe876e3 Call-ID: 0335c078e21a4fd89e9d3ac053690900 CSeq: 3625 SUBSCRIBE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="032dad75" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0335c078e21a4fd89e9d3ac053690900' in 6400 ms (Method: SUBSCRIBE) localhost*CLI> <--- SIP read from UDP:172.28.3.149:5060 ---> SUBSCRIBE sip:4412490018@172.28.3.211 SIP/2.0 Via: SIP/2.0/UDP 172.28.3.149:5060;rport;branch=z9hG4bKPj88a12c9217484021866f300df85d6d77 Max-Forwards: 70 From: "Femto" <sip:pg5@172.28.3.211>;tag=28aa7a96d406496fbc26c848e2627d05 To: <sip:4412490018@172.28.3.211> Contact: "Femto" <sip:pg5@172.28.3.149:5060;ob> Call-ID: 0335c078e21a4fd89e9d3ac053690900 CSeq: 3626 SUBSCRIBE Event: presence Expires: 600 Supported: replaces, 100rel, timer, norefersub Accept: application/pidf+xml, application/xpidf+xml Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.1.7 Authorization: Digest username="pg5", realm="asterisk", nonce="032dad75", uri="sip:4412490018@172.28.3.211", response="22cb8b8a90c398c4c4d17324cc98c151", algorithm=MD5 Content-Length: 0 <-------------> --- (16 headers 0 lines) --- Creating new subscription Sending to 172.28.3.149 : 5060 (NAT) Found peer 'pg5' for 'pg5' from 172.28.3.149:5060 Looking for 4412490018 in device-hints (domain 172.28.3.211) <--- Transmitting (NAT) to 172.28.3.149:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 172.28.3.149:5060;branch=z9hG4bKPj88a12c9217484021866f300df85d6d77;received=172.28.3.149;rport=5060 From: "Femto" <sip:pg5@172.28.3.211>;tag=28aa7a96d406496fbc26c848e2627d05 To: <sip:4412490018@172.28.3.211>;tag=as3fe876e3 Call-ID: 0335c078e21a4fd89e9d3ac053690900 CSeq: 3626 SUBSCRIBE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog '0335c078e21a4fd89e9d3ac053690900' Method: SUBSCRIBE localhost*CLI> <--- SIP read from UDP:172.28.3.149:5060 ---> PUBLISH sip:pg5@172.28.3.211 SIP/2.0 Via: SIP/2.0/UDP 172.28.3.149:5060;rport;branch=z9hG4bKPj67098175c1b5422781a697a74a9649c7 Max-Forwards: 70 From: "Femto" <sip:pg5@172.28.3.211>;tag=d6633b6b426a4663842c9681538c1b7a To: "Femto" <sip:pg5@172.28.3.211> Call-ID: 04326f0385c64517b8c9c3796f2a7b52 CSeq: 35139 PUBLISH Event: presence User-Agent: MicroSIP/3.1.7 Content-Type: application/pidf+xml Content-Length: 283 <?xml version="1.0" encoding="UTF-8"?> <presence entity="sip:pg5@172.28.3.211" xmlns="urn:ietf:params:xml:ns:pidf"> <tuple id="pj3f3c03ab123e44c78393bff8253559bd"> <status> <basic>open</basic> </status> <timestamp>2012-07-19T14:19:01.085Z</timestamp> </tuple> </presence> <-------------> --- (11 headers 9 lines) --- <--- Transmitting (no NAT) to 172.28.3.149:5060 ---> SIP/2.0 501 Method Not Implemented Via: SIP/2.0/UDP 172.28.3.149:5060;branch=z9hG4bKPj67098175c1b5422781a697a74a9649c7;received=172.28.3.149;rport=5060 From: "Femto" <sip:pg5@172.28.3.211>;tag=d6633b6b426a4663842c9681538c1b7a To: "Femto" <sip:pg5@172.28.3.211>;tag=as2b35a8a2 Call-ID: 04326f0385c64517b8c9c3796f2a7b52 CSeq: 35139 PUBLISH Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> localhost*CLI> Reliably Transmitting (NAT) to 172.28.7.26:5075: OPTIONS sip:pg@172.28.7.26:5075;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.28.3.211:5060;branch=z9hG4bK54469788;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@172.28.3.211>;tag=as5a122507 To: <sip:pg@172.28.7.26:5075;transport=udp> Contact: <sip:asterisk@172.28.3.211> Call-ID: 3ba73c7537b62c786d7d895e47a9daff@172.28.3.211 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.11 Date: Thu, 19 Jul 2012 12:12:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- localhost*CLI> <--- SIP read from UDP:172.28.7.26:5075 ---> SIP/2.0 200 OK CSeq: 102 OPTIONS Via: SIP/2.0/UDP 172.28.3.211:5060;branch=z9hG4bK54469788;rport=5060;received=172.28.3.211 From: "asterisk" <sip:asterisk@172.28.3.211>;tag=as5a122507 Call-ID: 3ba73c7537b62c786d7d895e47a9daff@172.28.3.211 To: <sip:pg@172.28.7.26:5075;transport=udp> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '3ba73c7537b62c786d7d895e47a9daff@172.28.3.211' Method: OPTIONS localhost*CLI> Reliably Transmitting (NAT) to 172.28.3.149:5060: OPTIONS sip:pg5@172.28.3.149:5060;ob SIP/2.0 Via: SIP/2.0/UDP 172.28.3.211:5060;branch=z9hG4bK08d003b0;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@172.28.3.211>;tag=as47f87f5c To: <sip:pg5@172.28.3.149:5060;ob> Contact: <sip:asterisk@172.28.3.211> Call-ID: 0272d24d5f05c9e74748cdba30abe3f0@172.28.3.211 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.11 Date: Thu, 19 Jul 2012 12:12:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- localhost*CLI> <--- SIP read from UDP:172.28.3.149:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.28.3.211:5060;rport=5060;received=172.28.3.211;branch=z9hG4bK08d003b0 Call-ID: 0272d24d5f05c9e74748cdba30abe3f0@172.28.3.211 From: "asterisk" <sip:asterisk@172.28.3.211>;tag=as47f87f5c To: <sip:pg5@172.28.3.149;ob>;tag=z9hG4bK08d003b0 CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.1.7 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '0272d24d5f05c9e74748cdba30abe3f0@172.28.3.211' Method: OPTIONS localhost*CLI> Reliably Transmitting (no NAT) to 172.29.1.189:5060: OPTIONS sip:172.29.1.189 SIP/2.0 Via: SIP/2.0/UDP 172.28.3.211:5060;branch=z9hG4bK3eacba63;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@172.28.3.211>;tag=as21cb39c4 To: <sip:172.29.1.189> Contact: <sip:asterisk@172.28.3.211> Call-ID: 6c7c6f5125b56f8b11628b02444fd797@172.28.3.211 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.11 Date: Thu, 19 Jul 2012 12:12:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- localhost*CLI> <--- SIP read from UDP:172.29.1.189:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.28.3.211:5060;branch=z9hG4bK3eacba63;rport To: <sip:172.29.1.189> From: "asterisk" <sip:asterisk@172.28.3.211>;tag=as21cb39c4 Call-ID: 6c7c6f5125b56f8b11628b02444fd797@172.28.3.211 CSeq: 102 OPTIONS Allow: INVITE,ACK,BYE,CANCEL,UPDATE,OPTIONS Content-Length: 0 <-------------> --- (8 headers 0 lines) --- localhost*CLI> Really destroying SIP dialog '6c7c6f5125b56f8b11628b02444fd797@172.28.3.211' Method: OPTIONS localhost*CLI> <--- SIP read from UDP:172.28.3.149:5060 ---> INVITE sip:4412490018@172.28.3.211 SIP/2.0 Via: SIP/2.0/UDP 172.28.3.149:5060;rport;branch=z9hG4bKPjcede2b44e31c491dac1d9250c31c1d87 Max-Forwards: 70 From: "Femto" <sip:pg5@172.28.3.211>;tag=b0061bb240b843218a995512d0d18948 To: <sip:4412490018@172.28.3.211> Contact: "Femto" <sip:pg5@172.28.3.149:5060;ob> Call-ID: 1319651346d74c5e865572a27677cc6a CSeq: 21957 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: MicroSIP/3.1.7 Content-Type: application/sdp Content-Length: 660 v=0 o=- 3551696345 3551696345 IN IP4 172.28.3.149 s=pjmedia c=IN IP4 172.28.3.149 b=AS:84 t=0 0 a=X-nat:0 m=audio 40016 RTP/AVP 98 97 99 104 3 0 8 9 18 120 119 118 117 96 c=IN IP4 172.28.3.149 b=TIAS:64000 a=rtcp:40017 IN IP4 172.28.3.149 a=sendrecv a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:120 SILK/24000 a=rtpmap:119 SILK/16000 a=rtpmap:118 SILK/12000 a=rtpmap:117 SILK/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> --- (15 headers 29 lines) --- localhost*CLI> == Using SIP RTP CoS mark 5 localhost*CLI> Sending to 172.28.3.149 : 5060 (no NAT) localhost*CLI> Using INVITE request as basis request - 1319651346d74c5e865572a27677cc6a Found peer 'pg5' for 'pg5' from 172.28.3.149:5060 localhost*CLI> <--- Reliably Transmitting (NAT) to 172.28.3.149:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.28.3.149:5060;branch=z9hG4bKPjcede2b44e31c491dac1d9250c31c1d87;received=172.28.3.149;rport=5060 From: "Femto" <sip:pg5@172.28.3.211>;tag=b0061bb240b843218a995512d0d18948 To: <sip:4412490018@172.28.3.211>;tag=as1d637358 Call-ID: 1319651346d74c5e865572a27677cc6a CSeq: 21957 INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="42e10ca0" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1319651346d74c5e865572a27677cc6a' in 6400 ms (Method: INVITE) localhost*CLI> <--- SIP read from UDP:172.28.3.149:5060 ---> ACK sip:4412490018@172.28.3.211 SIP/2.0 Via: SIP/2.0/UDP 172.28.3.149:5060;rport;branch=z9hG4bKPjcede2b44e31c491dac1d9250c31c1d87 Max-Forwards: 70 From: "Femto" <sip:pg5@172.28.3.211>;tag=b0061bb240b843218a995512d0d18948 To: <sip:4412490018@172.28.3.211>;tag=as1d637358 Call-ID: 1319651346d74c5e865572a27677cc6a CSeq: 21957 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- localhost*CLI> <--- SIP read from UDP:172.28.3.149:5060 ---> INVITE sip:4412490018@172.28.3.211 SIP/2.0 Via: SIP/2.0/UDP 172.28.3.149:5060;rport;branch=z9hG4bKPj0805e5280288478a91051586f86a1aa5 Max-Forwards: 70 From: "Femto" <sip:pg5@172.28.3.211>;tag=b0061bb240b843218a995512d0d18948 To: <sip:4412490018@172.28.3.211> Contact: "Femto" <sip:pg5@172.28.3.149:5060;ob> Call-ID: 1319651346d74c5e865572a27677cc6a CSeq: 21958 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: MicroSIP/3.1.7 Authorization: Digest username="pg5", realm="asterisk", nonce="42e10ca0", uri="sip:4412490018@172.28.3.211", response="4744ceb438b535ba3b9764d7d6dd8c36", algorithm=MD5 Content-Type: application/sdp Content-Length: 660 v=0 o=- 3551696345 3551696345 IN IP4 172.28.3.149 s=pjmedia c=IN IP4 172.28.3.149 b=AS:84 t=0 0 a=X-nat:0 m=audio 40016 RTP/AVP 98 97 99 104 3 0 8 9 18 120 119 118 117 96 c=IN IP4 172.28.3.149 b=TIAS:64000 a=rtcp:40017 IN IP4 172.28.3.149 a=sendrecv a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:120 SILK/24000 a=rtpmap:119 SILK/16000 a=rtpmap:118 SILK/12000 a=rtpmap:117 SILK/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> --- (16 headers 29 lines) --- Sending to 172.28.3.149 : 5060 (NAT) Using INVITE request as basis request - 1319651346d74c5e865572a27677cc6a Found peer 'pg5' for 'pg5' from 172.28.3.149:5060 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 99 Found RTP audio format 104 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 18 Found RTP audio format 120 Found RTP audio format 119 Found RTP audio format 118 Found RTP audio format 117 Found RTP audio format 96 Found audio description format speex for ID 98 Found audio description format speex for ID 97 Found audio description format speex for ID 99 Found audio description format iLBC for ID 104 Found audio description format GSM for ID 3 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G722 for ID 9 Found audio description format G729 for ID 18 Found audio description format SILK for ID 120 Found audio description format SILK for ID 119 Found audio description format SILK for ID 118 Found audio description format SILK for ID 117 Found audio description format telephone-event for ID 96 Capabilities: us - 0x1402 (gsm|ilbc|g722), peer - audio=0x30170e (gsm|ulaw|alaw|g729|speex|ilbc|g722|h263p|h264)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1402 (gsm|ilbc|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.28.3.149:40016 Looking for 4412490018 in test (domain 172.28.3.211) list_route: hop: <sip:pg5@172.28.3.149:5060;ob> <--- Transmitting (NAT) to 172.28.3.149:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.3.149:5060;branch=z9hG4bKPj0805e5280288478a91051586f86a1aa5;received=172.28.3.149;rport=5060 From: "Femto" <sip:pg5@172.28.3.211>;tag=b0061bb240b843218a995512d0d18948 To: <sip:4412490018@172.28.3.211> Call-ID: 1319651346d74c5e865572a27677cc6a CSeq: 21958 INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: <sip:4412490018@172.28.3.211> Content-Length: 0 <------------> -- Executing [4412490018@test:1] Verbose("SIP/pg5-0000004b", "3,Hello") in new stack -- Hello -- Executing [4412490018@test:2] Set("SIP/pg5-0000004b", "CALLERID(num)=6018") in new stack -- Executing [4412490018@test:3] Verbose("SIP/pg5-0000004b", "3,6018") in new stack -- 6018 -- Executing [4412490018@test:4] Set("SIP/pg5-0000004b", "CALLERID(name)=Femto") in new stack -- Executing [4412490018@test:5] Verbose("SIP/pg5-0000004b", "3,Femto") in new stack -- Femto -- Executing [4412490018@test:6] Dial("SIP/pg5-0000004b", "SIP/4412490018@ADC") in new stack localhost*CLI> == Using SIP RTP CoS mark 5 localhost*CLI> Audio is at 172.28.3.211 port 11680 Adding codec 0x1000 (g722) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP localhost*CLI> Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 172.29.1.189:5060: INVITE sip:4412490018@172.29.1.189 SIP/2.0 Via: SIP/2.0/UDP 172.28.3.211:5060;branch=z9hG4bK16259a4e;rport Max-Forwards: 70 From: "Femto" <sip:6018@172.28.3.211>;tag=as4d875199 To: <sip:4412490018@172.29.1.189> Contact: <sip:6018@172.28.3.211> Call-ID: 3b6e9a501be0a08a1a8cf2587fbe6f8b@172.28.3.211 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.11 Date: Thu, 19 Jul 2012 12:12:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 308 v=0 o=root 2057561930 2057561930 IN IP4 172.28.3.211 s=Asterisk PBX 1.6.2.11 c=IN IP4 172.28.3.211 t=0 0 m=audio 11680 RTP/AVP 9 3 0 8 101 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called 4412490018@ADC localhost*CLI> <--- SIP read from UDP:172.29.1.189:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 172.28.3.211:5060;branch=z9hG4bK16259a4e;rport To: <sip:4412490018@172.29.1.189> From: "Femto" <sip:6018@172.28.3.211>;tag=as4d875199 Call-ID: 3b6e9a501be0a08a1a8cf2587fbe6f8b@172.28.3.211 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Transmitting (no NAT) to 172.29.1.189:5060: ACK sip:4412490018@172.29.1.189 SIP/2.0 Via: SIP/2.0/UDP 172.28.3.211:5060;branch=z9hG4bK16259a4e;rport Max-Forwards: 70 From: "Femto" <sip:6018@172.28.3.211>;tag=as4d875199 To: <sip:4412490018@172.29.1.189> Contact: <sip:6018@172.28.3.211> Call-ID: 3b6e9a501be0a08a1a8cf2587fbe6f8b@172.28.3.211 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.11 Content-Length: 0 --- localhost*CLI> -- SIP/ADC-0000004c is circuit-busy localhost*CLI> == Everyone is busy/congested at this time (1:0/1/0) localhost*CLI> -- Auto fallthrough, channel 'SIP/pg5-0000004b' status is 'CONGESTION' localhost*CLI> <--- Reliably Transmitting (NAT) to 172.28.3.149:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 172.28.3.149:5060;branch=z9hG4bKPj0805e5280288478a91051586f86a1aa5;received=172.28.3.149;rport=5060 From: "Femto" <sip:pg5@172.28.3.211>;tag=b0061bb240b843218a995512d0d18948 To: <sip:4412490018@172.28.3.211>;tag=as0edc7e8b Call-ID: 1319651346d74c5e865572a27677cc6a CSeq: 21958 INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 Content-Length: 0 <------------> localhost*CLI> <--- SIP read from UDP:172.28.3.149:5060 ---> ACK sip:4412490018@172.28.3.211 SIP/2.0 Via: SIP/2.0/UDP 172.28.3.149:5060;rport;branch=z9hG4bKPj0805e5280288478a91051586f86a1aa5 Max-Forwards: 70 From: "Femto" <sip:pg5@172.28.3.211>;tag=b0061bb240b843218a995512d0d18948 To: <sip:4412490018@172.28.3.211>;tag=as0edc7e8b Call-ID: 1319651346d74c5e865572a27677cc6a CSeq: 21958 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '3b6e9a501be0a08a1a8cf2587fbe6f8b@172.28.3.211' Method: INVITE localhost*CLI> <--- SIP read from UDP:172.29.1.189:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 172.28.3.211:5060;branch=z9hG4bK16259a4e;rport To: <sip:4412490018@172.29.1.189> From: "Femto" <sip:6018@172.28.3.211>;tag=as4d875199 Call-ID: 3b6e9a501be0a08a1a8cf2587fbe6f8b@172.28.3.211 CSeq: 102 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- localhost*CLI> e <--- SIP read from UDP:172.28.3.149:5060 ---> <-------------> localhost*CLI> exit [root@localhost ~]#

<------------>
-- Executing [4412490018@test:1] Verbose("SIP/pg5-0000004b", "3,Hello") in new stack
-- Hello
-- Executing [4412490018@test:2] Set("SIP/pg5-0000004b", "CALLERID(num)=6018") in new stack
-- Executing [4412490018@test:3] Verbose("SIP/pg5-0000004b", "3,6018") in new stack
-- 6018
-- Executing [4412490018@test:4] Set("SIP/pg5-0000004b", "CALLERID(name)=Femto") in new stack
-- Executing [4412490018@test:5] Verbose("SIP/pg5-0000004b", "3,Femto") in new stack
-- Femto
-- Executing [4412490018@test:6] Dial("SIP/pg5-0000004b", "SIP/4412490018@ADC") in new stack
localhost*CLI>
== Using SIP RTP CoS mark 5
localhost*CLI>
Audio is at 172.28.3.211 port 11680
Adding codec 0x1000 (g722) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
localhost*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.29.1.189:5060:
INVITE sip:4412490018@172.29.1.189 SIP/2.0
Via: SIP/2.0/UDP 172.28.3.211:5060;branch=z9hG4bK16259a4e;rport
Max-Forwards: 70
From: "Femto" <sip:6018@172.28.3.211>;tag=as4d875199
To: <sip:4412490018@172.29.1.189>
Contact: <sip:6018@172.28.3.211>
Call-ID: 3b6e9a501be0a08a1a8cf2587fbe6f8b@172.28.3.211
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Date: Thu, 19 Jul 2012 12:12:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 308
v=0
o=root 2057561930 2057561930 IN IP4 172.28.3.211
s=Asterisk PBX 1.6.2.11
c=IN IP4 172.28.3.211
t=0 0
m=audio 11680 RTP/AVP 9 3 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called 4412490018@ADC
localhost*CLI>
<--- SIP read from UDP:172.29.1.189:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.28.3.211:5060;branch=z9hG4bK16259a4e;rport
To: <sip:4412490018@172.29.1.189>
From: "Femto" <sip:6018@172.28.3.211>;tag=as4d875199
Call-ID: 3b6e9a501be0a08a1a8cf2587fbe6f8b@172.28.3.211
CSeq: 102 INVITE
Content-Length: 0

Newsterisk

Posts: 16

Joined: Tue Jul 17, 2012 4:22 am

E-mail Femto

Re: Cell phone to SIP phone configuration.

by david55 » Thu Jul 19, 2012 8:28 am

The MSC and the mobile phone are largely irrelevant. Also, MSCs predate SIP by one or two decades, so, if the MSC were relevant, it would be the particular model.

It looks like this is actually a SIP phone to cell phone call, not a cell to SIP one.

There is nothing that obviously distinguishes the SIP user agent in the MSC from any other SIP user agent.

The number you are sending is simply not a valid phone number, as far as the destination is concerned. It is too short for a UK number, in national format, so I doubt that it is looking for a +. It might be looking for + and a country code, maybe +1?

I doubt there is an international standard for combined MSC and SIP user agent devices, so you will have to experiment, or read the documentation.

david55: Moves Like Spencer; Posts: 7885; Joined: Fri Sep 26, 2008 5:03 am

Re: Cell phone to SIP phone configuration.

by Femto » Thu Jul 19, 2012 9:26 am

Hi,
In this case, all the equipment is internal but unfortunately for me no one quite knows how to set it up

It's true that in this case, it was SIP to cell but (I take these things for granted) I would like to call in both directions to test the link.

The phone number in this case I think is irrelevant as all the equipment is internal (unless international dialling conventions are mandated in the equipment). I need to understand the generic rules on how to set-up the connections. SIP phones appear to identify themselves as <user>@<address>, mobiles as <number>. So, a SIP user can just tap the <number> in and the mobile rings. Similarly, the mobile needs to identify the SIP via a number rather than <user>@<address>. I'm expecting some translation, presumably in Asterisk, which converts <number> to <user>@<address>.

Unfortunately for me if I find the documentation, I'll read it :cry:

:cry:

The previous user of this system has set up the ADC MSC in sip.conf. The IP address is that of the BSC

[ADC]
type=peer
host=172.29.1.189
dtmfmode=rfc2833
qualify=yes
context=test
allow=g722
allow=gsm
allow=ilbc

I must be missing the connection on how Asterisk talks to the MSC and visa-versa.

Cheers
Femto.

Newsterisk

Posts: 16

Joined: Tue Jul 17, 2012 4:22 am

E-mail Femto

Re: Cell phone to SIP phone configuration.

by david55 » Thu Jul 19, 2012 9:34 am

The phone number is very relevant, because the call failed 404 Not Found, which normally means that the phone number was not in a recognizable format, or is unassigned.

(It is just possible that the destination is fussy about the exact form of the IP address, or domain name. Asterisk wouldn't normally check that but your remote device might be different.)

david55: Moves Like Spencer; Posts: 7885; Joined: Fri Sep 26, 2008 5:03 am

Re: Cell phone to SIP phone configuration.

by Femto » Thu Jul 19, 2012 10:03 am

Hi,
I meant that the format of the number was (probably) irrelevant but clearly the mapping/routing of numbers is not. If I could set up the system as '1' for the mobile and '2' for the SIP then that would be fine. I don't even need a generic solution as only one mobile will talk to the SIP. I'm simply trying to get the two to communicate with each other.

Thank you david55 for your help.

Cheers
Femto.

Newsterisk

Posts: 16

Joined: Tue Jul 17, 2012 4:22 am

E-mail Femto

Re: Cell phone to SIP phone configuration.

by Femto » Mon Jul 23, 2012 8:49 am

Hi,
Getting agonisingly close! Firstly the Asterisk IP address had changed not once but twice and the MSC was pointing to the incorrect place :roll:

:roll:

Now, on dialling 123453000 from a cell phone (4412490018) the SIP phone now rings. However, there is no audio from cell to SIP.

sip.conf
[123453000]
type=friend
host=dynamic
canreinvite=no
username=123453000
secret=MySecretCode
nat=yes
qualify=yes
mailbox=1000
context=test
allow=all

extensions.conf
[test]
exten=123453000,1,Log(NOTICE, Incoming call from ${CALLERID(all)})
exten=123453000,n,Dial(SIP/123453000)
exten=123453000,n,Answer

Any ideas gratefully received!

Cheers
Femto.

Newsterisk

Posts: 16

Joined: Tue Jul 17, 2012 4:22 am

E-mail Femto

Re: Cell phone to SIP phone configuration.

by david55 » Mon Jul 23, 2012 8:53 am

NAT and frewalls are the most common causes. It might also happen if one side negotiated G.729 but there were no licences.

For any more detailed debugging, you need to run at least debug level 5 on chan_sip and also have "sip set debug on" set. You need a log enabled for DEBUG class output.

david55: Moves Like Spencer; Posts: 7885; Joined: Fri Sep 26, 2008 5:03 am

Re: Cell phone to SIP phone configuration.

by Femto » Tue Jul 24, 2012 4:20 am

Hi,
I've been assured by our IT that in this case, the firewall is allowing communications.

How do I set a debug level of 5 on chan_sip?
How do I get log enabled for DEBUG class output?

It seems that the only codec which will at least allow signalling to go through and keep the call up despite no audio is the iLBC one.

Cheers
Femto

Newsterisk

Posts: 16

Joined: Tue Jul 17, 2012 4:22 am

E-mail Femto

Re: Cell phone to SIP phone configuration.

by david55 » Tue Jul 24, 2012 4:30 am

https://wiki.asterisk.org/wiki/display/ ... nformation

(This sets debugging for all modules. Use the built-in help if you want to limit it to just one.)

david55: Moves Like Spencer; Posts: 7885; Joined: Fri Sep 26, 2008 5:03 am

Re: Cell phone to SIP phone configuration.

by Femto » Tue Jul 24, 2012 7:00 am

Hi,
I managed to get a log and it appears to go wrong from line 1412 (where the data starts below as the log is too long) onwards. Unfortunately I cannot translate it

Cheers
Femto

p.s. Thank you for your help david55

Code: Select all: <-------------> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 0 [ 33]: REGISTER sip:172.29.7.100 SIP/2.0 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 172.28.3.149:5060;rport;branch=z9hG4bKPjf7025e01c54d4cb3808c67a75945c724 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 3 [ 87]: From: "" <sip:123453000@172.29.7.100>;tag=1851b2b901bd4ae297bf356fde62bcdb [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 4 [ 48]: To: "" <sip:123453000@172.29.7.100> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 5 [ 41]: Call-ID: 381a170e36f846d4b1fdfd8d6eba6dfa [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 6 [ 20]: CSeq: 40869 REGISTER [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 7 [ 26]: User-Agent: MicroSIP/3.1.7 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 8 [ 61]: Contact: "" <sip:123453000@172.28.3.149:5060;ob> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 9 [ 12]: Expires: 300 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 10 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 11 [ 18]: Content-Length: 0 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 12 [ 0]: [Jul 24 10:44:14] VERBOSE[3555] chan_sip.c: --- (12 headers 0 lines) --- [Jul 24 10:44:14] DEBUG[3555] acl.c: Found IP address for this socket [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 172.29.7.100:5060 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Allocating new SIP dialog for 381a170e36f846d4b1fdfd8d6eba6dfa - REGISTER (No RTP) [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Initializing initreq for method REGISTER - callid 381a170e36f846d4b1fdfd8d6eba6dfa [Jul 24 10:44:14] VERBOSE[3555] chan_sip.c: Sending to 172.28.3.149 : 5060 (no NAT) [Jul 24 10:44:14] VERBOSE[3555] chan_sip.c: <--- Transmitting (NAT) to 172.28.3.149:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.28.3.149:5060;branch=z9hG4bKPjf7025e01c54d4cb3808c67a75945c724;received=172.28.3.149;rport=5060 From: "" <sip:123453000@172.29.7.100>;tag=1851b2b901bd4ae297bf356fde62bcdb To: "" <sip:123453000@172.29.7.100>;tag=as3a079e7a Call-ID: 381a170e36f846d4b1fdfd8d6eba6dfa CSeq: 40869 REGISTER Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="450d64a5" Content-Length: 0 <------------> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 172.28.3.149:5060 [Jul 24 10:44:14] VERBOSE[3555] chan_sip.c: Scheduling destruction of SIP dialog '381a170e36f846d4b1fdfd8d6eba6dfa' in 32000 ms (Method: REGISTER) [Jul 24 10:44:14] VERBOSE[3555] chan_sip.c: <--- SIP read from UDP:172.28.3.149:5060 ---> REGISTER sip:172.29.7.100 SIP/2.0 Via: SIP/2.0/UDP 172.28.3.149:5060;rport;branch=z9hG4bKPjfd70c6ea232d48858ecafbc0a5737716 Max-Forwards: 70 From: "" <sip:123453000@172.29.7.100>;tag=1851b2b901bd4ae297bf356fde62bcdb To: "" <sip:123453000@172.29.7.100> Call-ID: 381a170e36f846d4b1fdfd8d6eba6dfa CSeq: 40870 REGISTER User-Agent: MicroSIP/3.1.7 Contact: "" <sip:123453000@172.28.3.149:5060;ob> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="123453000", realm="asterisk", nonce="450d64a5", uri="sip:172.29.7.100", response="478fee0c6077d14db2f0c9d6dcaa066d", algorithm=MD5 Content-Length: 0 <-------------> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 0 [ 33]: REGISTER sip:172.29.7.100 SIP/2.0 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 172.28.3.149:5060;rport;branch=z9hG4bKPjfd70c6ea232d48858ecafbc0a5737716 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 3 [ 87]: From: "" <sip:123453000@172.29.7.100>;tag=1851b2b901bd4ae297bf356fde62bcdb [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 4 [ 48]: To: "" <sip:123453000@172.29.7.100> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 5 [ 41]: Call-ID: 381a170e36f846d4b1fdfd8d6eba6dfa [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 6 [ 20]: CSeq: 40870 REGISTER [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 7 [ 26]: User-Agent: MicroSIP/3.1.7 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 8 [ 61]: Contact: "" <sip:123453000@172.28.3.149:5060;ob> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 9 [ 12]: Expires: 300 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 10 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 11 [162]: Authorization: Digest username="123453000", realm="asterisk", nonce="450d64a5", uri="sip:172.29.7.100", response="478fee0c6077d14db2f0c9d6dcaa066d", algorithm=MD5 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 12 [ 18]: Content-Length: 0 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 13 [ 0]: [Jul 24 10:44:14] VERBOSE[3555] chan_sip.c: --- (13 headers 0 lines) --- [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Initializing initreq for method REGISTER - callid 381a170e36f846d4b1fdfd8d6eba6dfa [Jul 24 10:44:14] VERBOSE[3555] chan_sip.c: Sending to 172.28.3.149 : 5060 (NAT) [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Allocating new SIP dialog for 51d330957e7a915950f7ae96359d4a37@127.0.0.1 - OPTIONS (No RTP) [Jul 24 10:44:14] DEBUG[3555] acl.c: Found IP address for this socket [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 172.29.7.100:5060 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Initializing initreq for method OPTIONS - callid 23771251109c204c1b66f526137965e1@172.29.7.100 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 0 [ 50]: OPTIONS sip:123453000@172.28.3.149:5060;ob SIP/2.0 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 172.29.7.100:5060;branch=z9hG4bK56677287;rport [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 3 [ 59]: From: "asterisk" <sip:asterisk@172.29.7.100>;tag=as1dcd69e7 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 4 [ 40]: To: <sip:123453000@172.28.3.149:5060;ob> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 5 [ 36]: Contact: <sip:asterisk@172.29.7.100> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 6 [ 54]: Call-ID: 23771251109c204c1b66f526137965e1@172.29.7.100 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.6.2.11 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 9 [ 35]: Date: Tue, 24 Jul 2012 09:44:14 GMT [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jul 24 10:44:14] VERBOSE[3555] chan_sip.c: Reliably Transmitting (NAT) to 172.28.3.149:5060: OPTIONS sip:123453000@172.28.3.149:5060;ob SIP/2.0 Via: SIP/2.0/UDP 172.29.7.100:5060;branch=z9hG4bK56677287;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@172.29.7.100>;tag=as1dcd69e7 To: <sip:123453000@172.28.3.149:5060;ob> Contact: <sip:asterisk@172.29.7.100> Call-ID: 23771251109c204c1b66f526137965e1@172.29.7.100 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.11 Date: Tue, 24 Jul 2012 09:44:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5637 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.28.3.149:5060 [Jul 24 10:44:14] VERBOSE[3555] chan_sip.c: <--- Transmitting (NAT) to 172.28.3.149:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.28.3.149:5060;branch=z9hG4bKPjfd70c6ea232d48858ecafbc0a5737716;received=172.28.3.149;rport=5060 From: "" <sip:123453000@172.29.7.100>;tag=1851b2b901bd4ae297bf356fde62bcdb To: "" <sip:123453000@172.29.7.100>;tag=as3a079e7a Call-ID: 381a170e36f846d4b1fdfd8d6eba6dfa CSeq: 40870 REGISTER Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 300 Contact: <sip:123453000@172.28.3.149:5060;ob>;expires=300 Date: Tue, 24 Jul 2012 09:44:14 GMT Content-Length: 0 <------------> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 172.28.3.149:5060 [Jul 24 10:44:14] VERBOSE[3555] chan_sip.c: Scheduling destruction of SIP dialog '381a170e36f846d4b1fdfd8d6eba6dfa' in 32000 ms (Method: REGISTER) [Jul 24 10:44:14] DEBUG[3382] devicestate.c: No provider found, checking channel drivers for SIP - 123453000 [Jul 24 10:44:14] DEBUG[3382] chan_sip.c: Checking device state for peer 123453000 [Jul 24 10:44:14] DEBUG[3382] devicestate.c: Changing state for SIP/123453000 - state 1 (Not in use) [Jul 24 10:44:14] DEBUG[3382] devicestate.c: device 'SIP/123453000' state '1' [Jul 24 10:44:14] DEBUG[3542] app_queue.c: Device 'SIP/123453000' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 24 10:44:14] VERBOSE[3555] chan_sip.c: <--- SIP read from UDP:172.28.3.149:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.29.7.100:5060;rport=5060;received=172.29.7.100;branch=z9hG4bK56677287 Call-ID: 23771251109c204c1b66f526137965e1@172.29.7.100 From: "asterisk" <sip:asterisk@172.29.7.100>;tag=as1dcd69e7 To: <sip:123453000@172.28.3.149;ob>;tag=z9hG4bK56677287 CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer User-Agent: MicroSIP/3.1.7 Content-Length: 0 <-------------> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 172.29.7.100:5060;rport=5060;received=172.29.7.100;branch=z9hG4bK56677287 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 2 [ 54]: Call-ID: 23771251109c204c1b66f526137965e1@172.29.7.100 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 3 [ 59]: From: "asterisk" <sip:asterisk@172.29.7.100>;tag=as1dcd69e7 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 4 [ 55]: To: <sip:123453000@172.28.3.149;ob>;tag=z9hG4bK56677287 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 7 [177]: Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 9 [ 46]: Allow-Events: presence, message-summary, refer [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 10 [ 26]: User-Agent: MicroSIP/3.1.7 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 11 [ 18]: Content-Length: 0 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 12 [ 0]: [Jul 24 10:44:14] VERBOSE[3555] chan_sip.c: --- (12 headers 0 lines) --- [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #5637 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Stopping retransmission on '23771251109c204c1b66f526137965e1@172.29.7.100' of Request 102: Match Found [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Destroying SIP dialog 23771251109c204c1b66f526137965e1@172.29.7.100 [Jul 24 10:44:14] VERBOSE[3555] chan_sip.c: Really destroying SIP dialog '23771251109c204c1b66f526137965e1@172.29.7.100' Method: OPTIONS [Jul 24 10:44:14] VERBOSE[3555] chan_sip.c: <--- SIP read from UDP:172.28.3.149:5060 ---> PUBLISH sip:123453000@172.29.7.100 SIP/2.0 Via: SIP/2.0/UDP 172.28.3.149:5060;rport;branch=z9hG4bKPj4bbc74e4e30b47559eeef9f2dc9b91a2 Max-Forwards: 70 From: "" <sip:123453000@172.29.7.100>;tag=372469981df14d3fbe1542ad9e8d382a To: "" <sip:123453000@172.29.7.100> Call-ID: 85c9ce12ae474f04ba04893d59d31b9b CSeq: 8070 PUBLISH Event: presence User-Agent: MicroSIP/3.1.7 Content-Type: application/pidf+xml Content-Length: 289 <?xml version="1.0" encoding="UTF-8"?> <presence entity="sip:123453000@172.29.7.100" xmlns="urn:ietf:params:xml:ns:pidf"> <tuple id="pj191e14ec466c4dc3b2da7b26d3e45b44"> <status> <basic>open</basic> </status> <timestamp>2012-07-24T11:51:23.828Z</timestamp> </tuple> </presence> <-------------> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 0 [ 42]: PUBLISH sip:123453000@172.29.7.100 SIP/2.0 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 172.28.3.149:5060;rport;branch=z9hG4bKPj4bbc74e4e30b47559eeef9f2dc9b91a2 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 3 [ 87]: From: "" <sip:123453000@172.29.7.100>;tag=372469981df14d3fbe1542ad9e8d382a [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 4 [ 48]: To: "" <sip:123453000@172.29.7.100> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 5 [ 41]: Call-ID: 85c9ce12ae474f04ba04893d59d31b9b [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 6 [ 18]: CSeq: 8070 PUBLISH [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 7 [ 15]: Event: presence [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 8 [ 26]: User-Agent: MicroSIP/3.1.7 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 9 [ 34]: Content-Type: application/pidf+xml [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 10 [ 21]: Content-Length: 289 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Header 11 [ 0]: [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Body 0 [ 38]: <?xml version="1.0" encoding="UTF-8"?> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Body 1 [ 82]: <presence entity="sip:123453000@172.29.7.100" xmlns="urn:ietf:params:xml:ns:pidf"> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Body 2 [ 48]: <tuple id="pj191e14ec466c4dc3b2da7b26d3e45b44"> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Body 3 [ 10]: <status> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Body 4 [ 22]: <basic>open</basic> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Body 5 [ 11]: </status> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Body 6 [ 49]: <timestamp>2012-07-24T11:51:23.828Z</timestamp> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Body 7 [ 9]: </tuple> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Body 8 [ 11]: </presence> [Jul 24 10:44:14] VERBOSE[3555] chan_sip.c: --- (11 headers 9 lines) --- [Jul 24 10:44:14] DEBUG[3555] acl.c: Found IP address for this socket [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 172.29.7.100:5060 [Jul 24 10:44:14] VERBOSE[3555] chan_sip.c: <--- Transmitting (no NAT) to 172.28.3.149:5060 ---> SIP/2.0 501 Method Not Implemented Via: SIP/2.0/UDP 172.28.3.149:5060;branch=z9hG4bKPj4bbc74e4e30b47559eeef9f2dc9b91a2;received=172.28.3.149;rport=5060 From: "" <sip:123453000@172.29.7.100>;tag=372469981df14d3fbe1542ad9e8d382a To: "" <sip:123453000@172.29.7.100>;tag=as5c25c81c Call-ID: 85c9ce12ae474f04ba04893d59d31b9b CSeq: 8070 PUBLISH Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Trying to put 'SIP/2.0 501' onto UDP socket destined for 172.28.3.149:5060 [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Got a request with unsupported SIP method. [Jul 24 10:44:14] DEBUG[3555] chan_sip.c: Invalid SIP message - rejected , no callid, len 750 [Jul 24 10:44:14] DEBUG[11411] rtp.c: Got RTCP report of 52 bytes [Jul 24 10:44:16] VERBOSE[3555] chan_sip.c: <--- SIP read from UDP:172.29.1.189:5060 ---> BYE sip:123453000@172.29.7.100 SIP/2.0 Via: SIP/2.0/UDP 172.29.1.189:5060;branch=z9hG4bK-10322-1343133824 To: <sip:123453000@172.29.7.100;user=phone>;tag=as5b51fce7 From: <sip:+4412490018@172.29.1.189;user=phone>;tag=90500E9872133D2 Call-ID: 169133D4500E9872@172.29.1.189 CSeq: 2 BYE Allow: INVITE,ACK,BYE,CANCEL,UPDATE,REFER,NOTIFY,OPTIONS Reason: Q.850;cause=16 Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 0 [ 38]: BYE sip:123453000@172.29.7.100 SIP/2.0 [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 1 [ 66]: Via: SIP/2.0/UDP 172.29.1.189:5060;branch=z9hG4bK-10322-1343133824 [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 2 [ 58]: To: <sip:123453000@172.29.7.100;user=phone>;tag=as5b51fce7 [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 3 [ 67]: From: <sip:+4412490018@172.29.1.189;user=phone>;tag=90500E9872133D2 [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 4 [ 38]: Call-ID: 169133D4500E9872@172.29.1.189 [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 5 [ 11]: CSeq: 2 BYE [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 6 [ 56]: Allow: INVITE,ACK,BYE,CANCEL,UPDATE,REFER,NOTIFY,OPTIONS [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 7 [ 22]: Reason: Q.850;cause=16 [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70 [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 10 [ 0]: [Jul 24 10:44:16] VERBOSE[3555] chan_sip.c: --- (10 headers 0 lines) --- [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Initializing initreq for method BYE - callid 169133D4500E9872@172.29.1.189 [Jul 24 10:44:16] VERBOSE[3555] chan_sip.c: Sending to 172.29.1.189 : 5060 (no NAT) [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Setting SIP_ALREADYGONE on dialog 169133D4500E9872@172.29.1.189 [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Received bye, issuing owner hangup [Jul 24 10:44:16] VERBOSE[3555] chan_sip.c: <--- Transmitting (no NAT) to 172.29.1.189:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.29.1.189:5060;branch=z9hG4bK-10322-1343133824;received=172.29.1.189 From: <sip:+4412490018@172.29.1.189;user=phone>;tag=90500E9872133D2 To: <sip:123453000@172.29.7.100;user=phone>;tag=as5b51fce7 Call-ID: 169133D4500E9872@172.29.1.189 CSeq: 2 BYE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 172.29.1.189:5060 [Jul 24 10:44:16] DEBUG[11411] rtp.c: p2p-rtp-bridge: Ooh, got a hangup [Jul 24 10:44:16] DEBUG[11411] channel.c: Returning from native bridge, channels: SIP/ADC-0000006e, SIP/123453000-0000006f [Jul 24 10:44:16] DEBUG[11411] channel.c: Hanging up channel 'SIP/123453000-0000006f' [Jul 24 10:44:16] DEBUG[11411] chan_sip.c: Hangup call SIP/123453000-0000006f, SIP callid 1ad0747a1f712ce17d90b9f548f3bfab@172.29.7.100 [Jul 24 10:44:16] VERBOSE[11411] chan_sip.c: Scheduling destruction of SIP dialog '1ad0747a1f712ce17d90b9f548f3bfab@172.29.7.100' in 6400 ms (Method: INVITE) [Jul 24 10:44:16] DEBUG[11411] chan_sip.c: Strict routing enforced for session 1ad0747a1f712ce17d90b9f548f3bfab@172.29.7.100 [Jul 24 10:44:16] VERBOSE[11411] chan_sip.c: set_destination: Parsing <sip:172.28.3.149:5060> for address/port to send to [Jul 24 10:44:16] VERBOSE[11411] chan_sip.c: set_destination: set destination to 172.28.3.149, port 5060 [Jul 24 10:44:16] VERBOSE[11411] chan_sip.c: Reliably Transmitting (NAT) to 172.28.3.149:5060: BYE sip:172.28.3.149:5060 SIP/2.0 Via: SIP/2.0/UDP 172.29.7.100:5060;branch=z9hG4bK03c96c6e;rport Max-Forwards: 70 From: "" <sip:+4412490018@172.29.7.100>;tag=as75b7e129 To: <sip:123453000@172.28.3.149:5060;ob>;tag=f373f181419e4687bf11a1f2a9dfafe0 Call-ID: 1ad0747a1f712ce17d90b9f548f3bfab@172.29.7.100 CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.2.11 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jul 24 10:44:16] DEBUG[11411] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5642 [Jul 24 10:44:16] DEBUG[11411] chan_sip.c: Trying to put 'BYE sip:172' onto UDP socket destined for 172.28.3.149:5060 [Jul 24 10:44:16] DEBUG[11411] rtp.c: Channel '<unspecified>' has no RTP, not doing anything [Jul 24 10:44:16] DEBUG[3382] devicestate.c: No provider found, checking channel drivers for SIP - 123453000 [Jul 24 10:44:16] DEBUG[11411] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jul 24 10:44:16] DEBUG[3382] chan_sip.c: Checking device state for peer 123453000 [Jul 24 10:44:16] DEBUG[11411] pbx.c: Spawn extension (test,123453000,3) exited non-zero on 'SIP/ADC-0000006e' [Jul 24 10:44:16] DEBUG[3382] devicestate.c: Changing state for SIP/123453000 - state 1 (Not in use) [Jul 24 10:44:16] DEBUG[3382] devicestate.c: device 'SIP/123453000' state '1' [Jul 24 10:44:16] DEBUG[3542] app_queue.c: Device 'SIP/123453000' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 24 10:44:16] VERBOSE[11411] pbx.c: == Spawn extension (test, 123453000, 3) exited non-zero on 'SIP/ADC-0000006e' [Jul 24 10:44:16] DEBUG[11411] channel.c: Soft-Hanging up channel 'SIP/ADC-0000006e' [Jul 24 10:44:16] DEBUG[11411] channel.c: Hanging up channel 'SIP/ADC-0000006e' [Jul 24 10:44:16] DEBUG[11411] chan_sip.c: Hangup call SIP/ADC-0000006e, SIP callid 169133D4500E9872@172.29.1.189 [Jul 24 10:44:16] DEBUG[3382] devicestate.c: No provider found, checking channel drivers for SIP - ADC [Jul 24 10:44:16] DEBUG[3382] chan_sip.c: Checking device state for peer ADC [Jul 24 10:44:16] DEBUG[3382] devicestate.c: Changing state for SIP/ADC - state 1 (Not in use) [Jul 24 10:44:16] DEBUG[3382] devicestate.c: device 'SIP/ADC' state '1' [Jul 24 10:44:16] DEBUG[3542] app_queue.c: Device 'SIP/ADC' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 24 10:44:16] VERBOSE[3555] chan_sip.c: <--- SIP read from UDP:172.28.3.149:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.29.7.100:5060;rport=5060;received=172.29.7.100;branch=z9hG4bK03c96c6e Call-ID: 1ad0747a1f712ce17d90b9f548f3bfab@172.29.7.100 From: "" <sip:+4412490018@172.29.7.100>;tag=as75b7e129 To: <sip:123453000@172.28.3.149;ob>;tag=f373f181419e4687bf11a1f2a9dfafe0 CSeq: 103 BYE Content-Length: 0 <-------------> [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 172.29.7.100:5060;rport=5060;received=172.29.7.100;branch=z9hG4bK03c96c6e [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 2 [ 54]: Call-ID: 1ad0747a1f712ce17d90b9f548f3bfab@172.29.7.100 [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 3 [ 67]: From: "" <sip:+4412490018@172.29.7.100>;tag=as75b7e129 [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 4 [ 72]: To: <sip:123453000@172.28.3.149;ob>;tag=f373f181419e4687bf11a1f2a9dfafe0 [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 5 [ 13]: CSeq: 103 BYE [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 6 [ 18]: Content-Length: 0 [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Header 7 [ 0]: [Jul 24 10:44:16] VERBOSE[3555] chan_sip.c: --- (7 headers 0 lines) --- [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #5642 [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Stopping retransmission on '1ad0747a1f712ce17d90b9f548f3bfab@172.29.7.100' of Request 103: Match Found [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Destroying SIP dialog 1ad0747a1f712ce17d90b9f548f3bfab@172.29.7.100 [Jul 24 10:44:16] VERBOSE[3555] chan_sip.c: Really destroying SIP dialog '1ad0747a1f712ce17d90b9f548f3bfab@172.29.7.100' Method: INVITE [Jul 24 10:44:16] DEBUG[3555] chan_sip.c: Destroying SIP dialog 169133D4500E9872@172.29.1.189 [Jul 24 10:44:16] VERBOSE[3555] chan_sip.c: Really destroying SIP dialog '169133D4500E9872@172.29.1.189' Method: BYE [Jul 24 10:44:17] DEBUG[11414] http.c: mmm ... cookie! Name: 'mansession_id' Value: '25e1786b' [Jul 24 10:44:17] DEBUG[11414] http.c: mmm ... cookie! Name: 'username' Value: 'admin' [Jul 24 10:44:17] DEBUG[11414] http.c: mmm ... cookie! Name: 'rwaccess' Value: 'yes' [Jul 24 10:44:17] DEBUG[11414] http.c: mmm ... cookie! Name: 'advancedmode' Value: 'yes' [Jul 24 10:44:17] DEBUG[11414] http.c: match request [rawman] with handler [httpstatus] len 0 [Jul 24 10:44:17] DEBUG[11414] http.c: match request [rawman] with handler [phoneprov] len 10 [Jul 24 10:44:17] DEBUG[11414] http.c: match request [rawman] with handler [manager] len 9 [Jul 24 10:44:17] DEBUG[11414] http.c: match request [rawman] with handler [rawman] len 7 [Jul 24 10:44:17] VERBOSE[11414] manager.c: > HTTP Manager add header action: ping [Jul 24 10:44:17] VERBOSE[11414] manager.c: > HTTP Manager add header advancedmode: yes [Jul 24 10:44:17] VERBOSE[11414] manager.c: > HTTP Manager add header rwaccess: yes [Jul 24 10:44:17] VERBOSE[11414] manager.c: > HTTP Manager add header username: admin [Jul 24 10:44:17] VERBOSE[11414] manager.c: > HTTP Manager add header mansession_id: 25e1786b [Jul 24 10:44:17] DEBUG[11414] manager.c: Manager received command 'ping' [Jul 24 10:44:22] DEBUG[11415] http.c: mmm ... cookie! Name: 'mansession_id' Value: '25e1786b' [Jul 24 10:44:22] DEBUG[11415] http.c: mmm ... cookie! Name: 'username' Value: 'admin' [Jul 24 10:44:22] DEBUG[11415] http.c: mmm ... cookie! Name: 'rwaccess' Value: 'yes' [Jul 24 10:44:22] DEBUG[11415] http.c: mmm ... cookie! Name: 'advancedmode' Value: 'yes' [Jul 24 10:44:22] DEBUG[11415] http.c: match request [rawman] with handler [httpstatus] len 0 [Jul 24 10:44:22] DEBUG[11415] http.c: match request [rawman] with handler [phoneprov] len 10 [Jul 24 10:44:22] DEBUG[11415] http.c: match request [rawman] with handler [manager] len 9 [Jul 24 10:44:22] DEBUG[11415] http.c: match request [rawman] with handler [rawman] len 7 [Jul 24 10:44:22] VERBOSE[11415] manager.c: > HTTP Manager add header action: ping [Jul 24 10:44:22] VERBOSE[11415] manager.c: > HTTP Manager add header advancedmode: yes [Jul 24 10:44:22] VERBOSE[11415] manager.c: > HTTP Manager add header rwaccess: yes [Jul 24 10:44:22] VERBOSE[11415] manager.c: > HTTP Manager add header username: admin [Jul 24 10:44:22] VERBOSE[11415] manager.c: > HTTP Manager add header mansession_id: 25e1786b [Jul 24 10:44:22] DEBUG[11415] manager.c: Manager received command 'ping' [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Allocating new SIP dialog for 3ce4d1297e72e5e257e1407b55184e8f@127.0.0.1 - OPTIONS (No RTP) [Jul 24 10:44:25] DEBUG[3555] acl.c: Found IP address for this socket [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 172.29.7.100:5060 [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Initializing initreq for method OPTIONS - callid 70b207e7237702c56865ba8a302ffad6@172.29.7.100 [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 0 [ 32]: OPTIONS sip:172.29.1.189 SIP/2.0 [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 172.29.7.100:5060;branch=z9hG4bK43e69085;rport [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 3 [ 59]: From: "asterisk" <sip:asterisk@172.29.7.100>;tag=as5dc7b460 [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 4 [ 22]: To: <sip:172.29.1.189> [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 5 [ 36]: Contact: <sip:asterisk@172.29.7.100> [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 6 [ 54]: Call-ID: 70b207e7237702c56865ba8a302ffad6@172.29.7.100 [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 8 [ 33]: User-Agent: Asterisk PBX 1.6.2.11 [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 9 [ 35]: Date: Tue, 24 Jul 2012 09:44:25 GMT [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jul 24 10:44:25] VERBOSE[3555] chan_sip.c: Reliably Transmitting (no NAT) to 172.29.1.189:5060: OPTIONS sip:172.29.1.189 SIP/2.0 Via: SIP/2.0/UDP 172.29.7.100:5060;branch=z9hG4bK43e69085;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@172.29.7.100>;tag=as5dc7b460 To: <sip:172.29.1.189> Contact: <sip:asterisk@172.29.7.100> Call-ID: 70b207e7237702c56865ba8a302ffad6@172.29.7.100 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.11 Date: Tue, 24 Jul 2012 09:44:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5643 [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.29.1.189:5060 [Jul 24 10:44:25] VERBOSE[3555] chan_sip.c: <--- SIP read from UDP:172.29.1.189:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.29.7.100:5060;branch=z9hG4bK43e69085;rport To: <sip:172.29.1.189> From: "asterisk" <sip:asterisk@172.29.7.100>;tag=as5dc7b460 Call-ID: 70b207e7237702c56865ba8a302ffad6@172.29.7.100 CSeq: 102 OPTIONS Allow: INVITE,ACK,BYE,CANCEL,UPDATE,OPTIONS Content-Length: 0 <-------------> [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 172.29.7.100:5060;branch=z9hG4bK43e69085;rport [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 2 [ 22]: To: <sip:172.29.1.189> [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 3 [ 59]: From: "asterisk" <sip:asterisk@172.29.7.100>;tag=as5dc7b460 [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 4 [ 54]: Call-ID: 70b207e7237702c56865ba8a302ffad6@172.29.7.100 [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 6 [ 43]: Allow: INVITE,ACK,BYE,CANCEL,UPDATE,OPTIONS [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Header 8 [ 0]: [Jul 24 10:44:25] VERBOSE[3555] chan_sip.c: --- (8 headers 0 lines) --- [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #5643 [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Stopping retransmission on '70b207e7237702c56865ba8a302ffad6@172.29.7.100' of Request 102: Match Found [Jul 24 10:44:25] DEBUG[3555] chan_sip.c: Destroying SIP dialog 70b207e7237702c56865ba8a302ffad6@172.29.7.100 [Jul 24 10:44:25] VERBOSE[3555] chan_sip.c: Really destroying SIP dialog '70b207e7237702c56865ba8a302ffad6@172.29.7.100' Method: OPTIONS [Jul 24 10:44:27] DEBUG[11416] http.c: mmm ... cookie! Name: 'mansession_id' Value: '25e1786b' [Jul 24 10:44:27] DEBUG[11416] http.c: mmm ... cookie! Name: 'username' Value: 'admin' [Jul 24 10:44:27] DEBUG[11416] http.c: mmm ... cookie! Name: 'rwaccess' Value: 'yes' [Jul 24 10:44:27] DEBUG[11416] http.c: mmm ... cookie! Name: 'advancedmode' Value: 'yes' [Jul 24 10:44:27] DEBUG[11416] http.c: match request [rawman] with handler [httpstatus] len 0 [Jul 24 10:44:27] DEBUG[11416] http.c: match request [rawman] with handler [phoneprov] len 10 [Jul 24 10:44:27] DEBUG[11416] http.c: match request [rawman] with handler [manager] len 9 [Jul 24 10:44:27] DEBUG[11416] http.c: match request [rawman] with handler [rawman] len 7 [Jul 24 10:44:27] VERBOSE[11416] manager.c: > HTTP Manager add header action: ping [Jul 24 10:44:27] VERBOSE[11416] manager.c: > HTTP Manager add header advancedmode: yes [Jul 24 10:44:27] VERBOSE[11416] manager.c: > HTTP Manager add header rwaccess: yes [Jul 24 10:44:27] VERBOSE[11416] manager.c: > HTTP Manager add header username: admin [Jul 24 10:44:27] VERBOSE[11416] manager.c: > HTTP Manager add header mansession_id: 25e1786b [Jul 24 10:44:27] DEBUG[11416] manager.c: Manager received command 'ping' [Jul 24 10:44:29] VERBOSE[3555] chan_sip.c: <--- SIP read from UDP:172.28.3.149:5060 ---> <-------------> [Jul 24 10:44:29] DEBUG[3555] chan_sip.c: Header 0 [ 0]: [Jul 24 10:44:32] DEBUG[11417] http.c: mmm ... cookie! Name: 'mansession_id' Value: '25e1786b' [Jul 24 10:44:32] DEBUG[11417] http.c: mmm ... cookie! Name: 'username' Value: 'admin' [Jul 24 10:44:32] DEBUG[11417] http.c: mmm ... cookie! Name: 'rwaccess' Value: 'yes' [Jul 24 10:44:32] DEBUG[11417] http.c: mmm ... cookie! Name: 'advancedmode' Value: 'yes' [Jul 24 10:44:32] DEBUG[11417] http.c: match request [rawman] with handler [httpstatus] len 0 [Jul 24 10:44:32] DEBUG[11417] http.c: match request [rawman] with handler [phoneprov] len 10 [Jul 24 10:44:32] DEBUG[11417] http.c: match request [rawman] with handler [manager] len 9 [Jul 24 10:44:32] DEBUG[11417] http.c: match request [rawman] with handler [rawman] len 7 [Jul 24 10:44:32] VERBOSE[11417] manager.c: > HTTP Manager add header action: ping [Jul 24 10:44:32] VERBOSE[11417] manager.c: > HTTP Manager add header advancedmode: yes [Jul 24 10:44:32] VERBOSE[11417] manager.c: > HTTP Manager add header rwaccess: yes [Jul 24 10:44:32] VERBOSE[11417] manager.c: > HTTP Manager add header username: admin [Jul 24 10:44:32] VERBOSE[11417] manager.c: > HTTP Manager add header mansession_id: 25e1786b [Jul 24 10:44:32] DEBUG[11417] manager.c: Manager received command 'ping' [Jul 24 10:44:33] DEBUG[11418] http.c: mmm ... cookie! Name: 'mansession_id' Value: '25e1786b' [Jul 24 10:44:33] DEBUG[11418] http.c: mmm ... cookie! Name: 'username' Value: 'admin' [Jul 24 10:44:33] DEBUG[11418] http.c: mmm ... cookie! Name: 'rwaccess' Value: 'yes' [Jul 24 10:44:33] DEBUG[11418] http.c: mmm ... cookie! Name: 'advancedmode' Value: 'yes' [Jul 24 10:44:33] DEBUG[11418] http.c: match request [rawman] with handler [httpstatus] len 0 [Jul 24 10:44:33] DEBUG[11418] http.c: match request [rawman] with handler [phoneprov] len 10 [Jul 24 10:44:33] DEBUG[11418] http.c: match request [rawman] with handler [manager] len 9 [Jul 24 10:44:33] DEBUG[11418] http.c: match request [rawman] with handler [rawman] len 7 [Jul 24 10:44:33] VERBOSE[11418] manager.c: > HTTP Manager add header action: command [Jul 24 10:44:33] VERBOSE[11418] manager.c: > HTTP Manager add header command: core set debug 0 [Jul 24 10:44:33] VERBOSE[11418] manager.c: > HTTP Manager add header advancedmode: yes [Jul 24 10:44:33] VERBOSE[11418] manager.c: > HTTP Manager add header rwaccess: yes [Jul 24 10:44:33] VERBOSE[11418] manager.c: > HTTP Manager add header username: admin [Jul 24 10:44:33] VERBOSE[11418] manager.c: > HTTP Manager add header mansession_id: 25e1786b [Jul 24 10:44:33] DEBUG[11418] manager.c: Manager received command 'command' [Jul 24 10:44:37] VERBOSE[11419] manager.c: > HTTP Manager add header action: command [Jul 24 10:44:37] VERBOSE[11419] manager.c: > HTTP Manager add header command: core set verbose 0 [Jul 24 10:44:37] VERBOSE[11419] manager.c: > HTTP Manager add header advancedmode: yes [Jul 24 10:44:37] VERBOSE[11419] manager.c: > HTTP Manager add header rwaccess: yes [Jul 24 10:44:37] VERBOSE[11419] manager.c: > HTTP Manager add header username: admin [Jul 24 10:44:37] VERBOSE[11419] manager.c: > HTTP Manager add header mansession_id: 25e1786b [Jul 24 10:44:44] VERBOSE[3555] chan_sip.c: <--- SIP read from UDP:172.28.3.149:5060 ---> <-------------> [Jul 24 10:44:46] VERBOSE[3555] chan_sip.c: Really destroying SIP dialog '381a170e36f846d4b1fdfd8d6eba6dfa' Method: REGISTER [Jul 24 10:44:59] VERBOSE[3555] chan_sip.c: <--- SIP read from UDP:172.28.3.149:5060 ---> <------------->

Newsterisk

Posts: 16

Joined: Tue Jul 17, 2012 4:22 am

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Re: Cell phone to SIP phone configuration.

by Femto » Tue Jul 24, 2012 10:29 am

Hi,
It seems odd that the only codec I can get to negotiate is 'ILBC'. So as an aside, had took several logs at the various system elements to see the codec type translation. The cell phone originates GSM-EFR. I took a log at the MSC and there is a...

X-Asterisk-HangupCauseCode: 58
[Expert Info (Note/Undecoded): Unrecognised SIP header (X-Asterisk-HangupCauseCode)]
[Message: Unrecognised SIP header (X-Asterisk-HangupCauseCode)]
[Severity level: Note]
[Group: Undecoded]
Content-Length: 0

error.

Code: Select all: No. Time Source Destination Protocol Length Info 812 10.546108 172.29.1.189 172.29.7.100 SIP/SDP 705 Request: INVITE sip:123453000@172.29.7.100;user=phone, with session description Frame 812: 705 bytes on wire (5640 bits), 705 bytes captured (5640 bits) Linux cooked capture Internet Protocol Version 4, Src: 172.29.1.189 (172.29.1.189), Dst: 172.29.7.100 (172.29.7.100) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:123453000@172.29.7.100;user=phone SIP/2.0 Method: INVITE Request-URI: sip:123453000@172.29.7.100;user=phone [Resent Packet: False] Message Header Via: SIP/2.0/UDP 172.29.1.189:5060;branch=z9hG4bK-10419-1343152253 Transport: UDP Sent-by Address: 172.29.1.189 Sent-by port: 5060 Branch: z9hG4bK-10419-1343152253 To: <sip:123453000@172.29.7.100;user=phone> SIP to address: sip:123453000@172.29.7.100;user=phone SIP to address User Part: 123453000 SIP to address Host Part: 172.29.7.100 From: <sip:+4412490018@172.29.1.189;user=phone>;tag=BA500EE07D97299 SIP from address: sip:+4412490018@172.29.1.189;user=phone SIP tag: BA500EE07D97299 Call-ID: 1C39729B500EE07D@172.29.1.189 CSeq: 1 INVITE Sequence Number: 1 Method: INVITE Contact: <sip:+4412490018@172.29.1.189> Contact-URI: sip:+4412490018@172.29.1.189 Contactt-URI User Part: +4412490018 Contact-URI Host Part: 172.29.1.189 Allow: INVITE,ACK,BYE,CANCEL,UPDATE,REFER,NOTIFY,OPTIONS P-Asserted-Identity: sip:+4412490018@172.29.1.189 SIP PAI Address: sip:+4412490018@172.29.1.189 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 147 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 1343145504184 1343145504184 IN IP4 172.29.7.13 Session Name (s): - Connection Information (c): IN IP4 172.29.7.13 Media Description, name and address (m): audio 4184 RTP/AVP 97 Media Attribute (a): rtpmap:97 GSM-EFR/8000 Media Attribute Fieldname: rtpmap Media Format: 97 MIME Type: GSM-EFR Sample Rate: 8000 Media Attribute (a): sendrecv No. Time Source Destination Protocol Length Info 813 10.546988 172.29.7.100 172.29.1.189 SIP 517 Status: 100 Trying Frame 813: 517 bytes on wire (4136 bits), 517 bytes captured (4136 bits) Linux cooked capture Internet Protocol Version 4, Src: 172.29.7.100 (172.29.7.100), Dst: 172.29.1.189 (172.29.1.189) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Status-Code: 100 [Resent Packet: False] [Request Frame: 812] [Response Time (ms): 0] Message Header Via: SIP/2.0/UDP 172.29.1.189:5060;branch=z9hG4bK-10419-1343152253;received=172.29.1.189 Transport: UDP Sent-by Address: 172.29.1.189 Sent-by port: 5060 Branch: z9hG4bK-10419-1343152253 Received: 172.29.1.189 From: <sip:+4412490018@172.29.1.189;user=phone>;tag=BA500EE07D97299 SIP from address: sip:+4412490018@172.29.1.189;user=phone SIP tag: BA500EE07D97299 To: <sip:123453000@172.29.7.100;user=phone> SIP to address: sip:123453000@172.29.7.100;user=phone SIP to address User Part: 123453000 SIP to address Host Part: 172.29.7.100 Call-ID: 1C39729B500EE07D@172.29.1.189 CSeq: 1 INVITE Sequence Number: 1 Method: INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:123453000@172.29.7.100> Contact-URI: sip:123453000@172.29.7.100 Contactt-URI User Part: 123453000 Contact-URI Host Part: 172.29.7.100 Content-Length: 0 No. Time Source Destination Protocol Length Info 817 10.563183 172.29.7.100 172.29.1.189 SIP 533 Status: 180 Ringing Frame 817: 533 bytes on wire (4264 bits), 533 bytes captured (4264 bits) Linux cooked capture Internet Protocol Version 4, Src: 172.29.7.100 (172.29.7.100), Dst: 172.29.1.189 (172.29.1.189) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Status-Code: 180 [Resent Packet: False] [Request Frame: 812] [Response Time (ms): 17] Message Header Via: SIP/2.0/UDP 172.29.1.189:5060;branch=z9hG4bK-10419-1343152253;received=172.29.1.189 Transport: UDP Sent-by Address: 172.29.1.189 Sent-by port: 5060 Branch: z9hG4bK-10419-1343152253 Received: 172.29.1.189 From: <sip:+4412490018@172.29.1.189;user=phone>;tag=BA500EE07D97299 SIP from address: sip:+4412490018@172.29.1.189;user=phone SIP tag: BA500EE07D97299 To: <sip:123453000@172.29.7.100;user=phone>;tag=as08e64d27 SIP to address: sip:123453000@172.29.7.100;user=phone SIP to address User Part: 123453000 SIP to address Host Part: 172.29.7.100 SIP tag: as08e64d27 Call-ID: 1C39729B500EE07D@172.29.1.189 CSeq: 1 INVITE Sequence Number: 1 Method: INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:123453000@172.29.7.100> Contact-URI: sip:123453000@172.29.7.100 Contactt-URI User Part: 123453000 Contact-URI Host Part: 172.29.7.100 Content-Length: 0 No. Time Source Destination Protocol Length Info 1042 13.407618 172.29.7.100 172.29.1.189 SIP 595 Status: 503 Service Unavailable Frame 1042: 595 bytes on wire (4760 bits), 595 bytes captured (4760 bits) Linux cooked capture Internet Protocol Version 4, Src: 172.29.7.100 (172.29.7.100), Dst: 172.29.1.189 (172.29.1.189) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 503 Service Unavailable Status-Code: 503 [Resent Packet: False] [Request Frame: 812] [Response Time (ms): 2862] Message Header Via: SIP/2.0/UDP 172.29.1.189:5060;branch=z9hG4bK-10419-1343152253;received=172.29.1.189 Transport: UDP Sent-by Address: 172.29.1.189 Sent-by port: 5060 Branch: z9hG4bK-10419-1343152253 Received: 172.29.1.189 From: <sip:+4412490018@172.29.1.189;user=phone>;tag=BA500EE07D97299 SIP from address: sip:+4412490018@172.29.1.189;user=phone SIP tag: BA500EE07D97299 To: <sip:123453000@172.29.7.100;user=phone>;tag=as08e64d27 SIP to address: sip:123453000@172.29.7.100;user=phone SIP to address User Part: 123453000 SIP to address Host Part: 172.29.7.100 SIP tag: as08e64d27 Call-ID: 1C39729B500EE07D@172.29.1.189 CSeq: 1 INVITE Sequence Number: 1 Method: INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-Asterisk-HangupCause: Bearer capability not available [Expert Info (Note/Undecoded): Unrecognised SIP header (X-Asterisk-HangupCause)] [Message: Unrecognised SIP header (X-Asterisk-HangupCause)] [Severity level: Note] [Group: Undecoded] X-Asterisk-HangupCauseCode: 58 [Expert Info (Note/Undecoded): Unrecognised SIP header (X-Asterisk-HangupCauseCode)] [Message: Unrecognised SIP header (X-Asterisk-HangupCauseCode)] [Severity level: Note] [Group: Undecoded] Content-Length: 0 No. Time Source Destination Protocol Length Info 1043 13.415258 172.29.1.189 172.29.7.100 SIP 384 Request: ACK sip:123453000@172.29.7.100;user=phone Frame 1043: 384 bytes on wire (3072 bits), 384 bytes captured (3072 bits) Linux cooked capture Internet Protocol Version 4, Src: 172.29.1.189 (172.29.1.189), Dst: 172.29.7.100 (172.29.7.100) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: ACK sip:123453000@172.29.7.100;user=phone SIP/2.0 Method: ACK Request-URI: sip:123453000@172.29.7.100;user=phone [Resent Packet: False] [Request Frame: 812] [Response Time (ms): 2870] Message Header Via: SIP/2.0/UDP 172.29.1.189:5060;branch=z9hG4bK-10419-1343152253 Transport: UDP Sent-by Address: 172.29.1.189 Sent-by port: 5060 Branch: z9hG4bK-10419-1343152253 To: <sip:123453000@172.29.7.100;user=phone>;tag=as08e64d27 SIP to address: sip:123453000@172.29.7.100;user=phone SIP to address User Part: 123453000 SIP to address Host Part: 172.29.7.100 SIP tag: as08e64d27 From: <sip:+4412490018@172.29.1.189;user=phone>;tag=BA500EE07D97299 SIP from address: sip:+4412490018@172.29.1.189;user=phone SIP tag: BA500EE07D97299 Call-ID: 1C39729B500EE07D@172.29.1.189 CSeq: 1 ACK Sequence Number: 1 Method: ACK Max-Forwards: 70 Content-Length: 0 No. Time Source Destination Protocol Length Info 1327 17.293223 172.29.7.100 172.29.1.189 SIP 554 Request: OPTIONS sip:172.29.1.189 Frame 1327: 554 bytes on wire (4432 bits), 554 bytes captured (4432 bits) Linux cooked capture Internet Protocol Version 4, Src: 172.29.7.100 (172.29.7.100), Dst: 172.29.1.189 (172.29.1.189) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: OPTIONS sip:172.29.1.189 SIP/2.0 Method: OPTIONS Request-URI: sip:172.29.1.189 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 172.29.7.100:5060;branch=z9hG4bK4ba05ac9;rport Transport: UDP Sent-by Address: 172.29.7.100 Sent-by port: 5060 Branch: z9hG4bK4ba05ac9 RPort: rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@172.29.7.100>;tag=as5d7ae94f SIP Display info: "asterisk" SIP from address: sip:asterisk@172.29.7.100 SIP tag: as5d7ae94f To: <sip:172.29.1.189> SIP to address: sip:172.29.1.189 SIP to address Host Part: 172.29.1.189 Contact: <sip:asterisk@172.29.7.100> Contact-URI: sip:asterisk@172.29.7.100 Contactt-URI User Part: asterisk Contact-URI Host Part: 172.29.7.100 Call-ID: 3c3480607af8b62374be1ce040d2185d@172.29.7.100 CSeq: 102 OPTIONS Sequence Number: 102 Method: OPTIONS User-Agent: Asterisk PBX 1.6.2.11 Date: Tue, 24 Jul 2012 14:51:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 No. Time Source Destination Protocol Length Info 1328 17.295657 172.29.1.189 172.29.7.100 SIP 351 Status: 200 OK Frame 1328: 351 bytes on wire (2808 bits), 351 bytes captured (2808 bits) Linux cooked capture Internet Protocol Version 4, Src: 172.29.1.189 (172.29.1.189), Dst: 172.29.7.100 (172.29.7.100) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] [Request Frame: 1327] [Response Time (ms): 2] Message Header Via: SIP/2.0/UDP 172.29.7.100:5060;branch=z9hG4bK4ba05ac9;rport Transport: UDP Sent-by Address: 172.29.7.100 Sent-by port: 5060 Branch: z9hG4bK4ba05ac9 RPort: rport To: <sip:172.29.1.189> SIP to address: sip:172.29.1.189 SIP to address Host Part: 172.29.1.189 From: "asterisk" <sip:asterisk@172.29.7.100>;tag=as5d7ae94f SIP Display info: "asterisk" SIP from address: sip:asterisk@172.29.7.100 SIP tag: as5d7ae94f Call-ID: 3c3480607af8b62374be1ce040d2185d@172.29.7.100 CSeq: 102 OPTIONS Sequence Number: 102 Method: OPTIONS Allow: INVITE,ACK,BYE,CANCEL,UPDATE,OPTIONS Content-Length: 0

Cheers
Femto

Newsterisk

Posts: 16

Joined: Tue Jul 17, 2012 4:22 am

E-mail Femto

Re: Cell phone to SIP phone configuration.

by david55 » Tue Jul 24, 2012 10:56 am

Asterisk only supports standard, full rate, GSM coding, not enhanced full rate coding. The call is being rejected because there are no codecs in common. GSM-EFR was the only one offered.

I think you will find that restriction fairly common in the VoIP world.

david55: Moves Like Spencer; Posts: 7885; Joined: Fri Sep 26, 2008 5:03 am

Re: Cell phone to SIP phone configuration.

by Femto » Wed Jul 25, 2012 3:24 am

OK. I'm speculating here but I suspect most 2G phones will originate the call with GSM-EFR (as seen in my example). Indeed if iLBC is enabled on the SIP phone, then at least signalling appears to work. There seems to be some translation going on.

sip.conf seems to 'allow' codecs and in that case it must act as a pipe to allow traffic through the gateway. Does the Asterisk codec restriction only apply when it is required to answer a call?

I have logged the 'INVITE' from X-Lite to see what codec types it offers in SDP.

X-Lite
SIP/SDP protocol
Media format...
GSM Codec 0x33 '3' -> GSM 06.10 == FR/GSM-FR
G711 aLaw 0x38 '8' -> ITU-T G.711 PCMA
G711 uLaw 0x30 '0' -> ITU-T G.711 PCMU
iLBC Codec 0x39 0x38 '98' -> iLBC/8000

Cheers
Femto

Newsterisk

Posts: 16

Joined: Tue Jul 17, 2012 4:22 am

E-mail Femto

Re: Cell phone to SIP phone configuration.

by david55 » Wed Jul 25, 2012 3:55 am

Asterisk has format and codec modules. If a format module exists, and both parties can handle the codec, the call will succeed but you may lose things like in-band tones and voice announcements. If the codec module also exists, Asterisk may be able to transcode.

I am not sure why Asterisk sent ringing before rejecting the codec. Maybe that version of Asterisk does know about GSM-EFR, but the phone doesn't.

david55: Moves Like Spencer; Posts: 7885; Joined: Fri Sep 26, 2008 5:03 am

Re: Cell phone to SIP phone configuration.

by Femto » Wed Jul 25, 2012 4:21 am

Hi,
I have logged from both ends of the system. The phone is a Nokia C2. I tried to interrogate its codec type via AT commands but it looks like this is not available. Wireshark logs show that the GSM phone will offer GSM-EFR. The X-Lite Softphone with GSM codec enabled is logged as GSM 06.10 (GSM-FR). The common denominator appears to be iLBC.

I get the feeling that to route a call, the signalling and voice channels are done separately. In that case, perhaps the SIP phone can ring but there is no voice available.

Cheers
Femto.

Newsterisk

Posts: 16

Joined: Tue Jul 17, 2012 4:22 am

E-mail Femto

Re: Cell phone to SIP phone configuration.

by david55 » Wed Jul 25, 2012 4:33 am

The call is being rejected, but only after the ringing.

Normally, for SIP calls, you don't get to find the destination codec capabilities until it answers, athough you might expect it to reject the call immediately if it cannot send in one of the allowed codecs.

iLBC may be common, but iLBC is not being offered on the incoming call.

You really need the Asterisk debug and the SIP from both sides, to work out exactly what is going on codec wise, but my guess is that that version of Asterisk does recognize EFR, and the X-Lite is failing to reject the call at the earliest opportunity.

Note, there is the possibility of delayed offer in SIP. That is not happening inbound, and, unless it has changed, I don't think that Asterisk uses it outbound, but it is possible to send INVITE without SDP and therefore codecs. If Asterisk did do late offer, Asterisk would not know there was a codec problem until the call was answered, and the X-Lite would not know until the call was fully up, and would have to immedaitely BYE it. I don't think this is the case here.

david55: Moves Like Spencer; Posts: 7885; Joined: Fri Sep 26, 2008 5:03 am

Re: Cell phone to SIP phone configuration.

by Femto » Wed Jul 25, 2012 5:23 am

Bingo!

I now have a call from cell to SIP! At the moment it's a bit of a workaround but it gives me a base to work from. So X-Lite has GSM codec selected (GSM-FR/GSM 06.10). The workaround was to restrict the BTS to only support (GSM-FR). The negotiation then settles at GSM-FR and the voice is established.

SIP to cell does not work yet .... but that's another story :roll:

:roll:

Cheers
Femto

Newsterisk

Posts: 16

Joined: Tue Jul 17, 2012 4:22 am

E-mail Femto

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