Asterisk 11 : Audio Issue : Codec Translation

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Asterisk 11 : Audio Issue : Codec Translation

Postby jaredlam » Sat Nov 24, 2012 7:14 pm

I just installed Asterisk 11.0.1 on a Centos 6.3. I complied the Asterisk from source with the default configuration settings.

I'm having audio issue if using different codec with phone and sip trunk provider. Here is my setup:

Grandstream gxp1450 (NAT) <-> Cisco router <-> Asterisk (No Nat: public facing IP) <-> SIP Trunk provider.

On the gxp1450, I have the codec set to g722 and on the SIP Trunk, the codec is ulaw. The problem with audio is either there is only one-way audio or no audio at all. If I call from the gpx1450 phone to my cell phone, I can hear audio from the cell phone; however, from the cell phone there is no audio. Sometimes, no audio audio on both ends.

If if set the codec to ulaw on gxp1450, audio works fine.

Is this by design? Or, is there a configuration setting or settings that I missed.

Here is my sip.conf:

[2600L1]
type=friend
host=dynamic
secret=*******
disallow=all
allow=ulaw
;allow=all
context=home_context
nat=yes
qualify=yes
canreinvite=no

[voipms]
canreinvite=no
context=voipms_trunk
host=losangeles.voip.ms
type=peer
disallow=all
allow=ulaw
;allow=all
nat=yes
secret=**************
username=******
fromuser=******
trustrpid=yes
sendrpid=yes
insecure=invite


Thanks for any information.

Jared.
jaredlam
Newsterisk
 
Posts: 3
Joined: Sat Nov 24, 2012 6:56 pm

Re: Asterisk 11 : Audio Issue : Codec Translation

Postby david55 » Sun Nov 25, 2012 4:16 am

It is generally easier to understand such issues if you provide the standard debugging information for SIP problems, i.e. sip set debug on, core set verbose 5, core set debug 5 chan_sip, with logging configured to include debug output.

canreinvite has been deprecated (in favour of directmedia) for some time.

Incidentally, nat=yes should serve no useful purpose on the trunk side. It may be needed on the phone side, but only if the NAT support is broken. Also, I believe that the value "yes", for this parameter, is also deprecated.

Unless you have supplied an incomplete sip.conf, you have allowguest defaulting to yes, which is usually not a good idea.

Why did you post two complete copies of your question (this is becoming quite common)?
david55
Moves Like Spencer
 
Posts: 10549
Joined: Fri Sep 26, 2008 5:03 am

Re: Asterisk 11 : Audio Issue : Codec Translation

Postby jaredlam » Sun Nov 25, 2012 6:56 pm

Thanks for the allowguest setting; I have updated the setting to no. With the current setup, either it works or no audio. Here is the latest sip.conf follow by debug information:

[2600L1]
type=friend
host=dynamic
secret=****************
disallow=all
allow=g722
allow=ulaw
;allow=all
context=home_context
nat=force_rport,comedia
qualify=yes
directmedia=nonat

[voipms]
directmedia=no
context=voipms_trunk
host=losangeles.voip.ms
type=peer
disallow=all
allow=ulaw
;allow=all
;nat=yes
secret=********************
username=*******
fromuser=*******
trustrpid=yes
sendrpid=yes
insecure=invite

DEBUG INFORMATION (no audio):

[Nov 25 16:48:25] Asterisk 11.0.1 built by root @ defiant on a x86_64 running Linux on 2012-11-14 07:27:49 UTC
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 0 [ 38]: INVITE sip:7168124@64.5.99.180 SIP/2.0
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 76.14.161.117:17329;branch=z9hG4bK1752883052;rport
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 2 [ 57]: From: "Jared Lam" <sip:2600L1@64.5.99.180>;tag=2128823278
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 3 [ 29]: To: <sip:7168124@64.5.99.180>
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 4 [ 41]: Call-ID: 374146249-17329-28@HG.BE.BGB.BBH
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 5 [ 16]: CSeq: 270 INVITE
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 6 [ 53]: Contact: "Jared Lam" <sip:2600L1@76.14.161.117:17329>
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 8 [ 40]: User-Agent: Grandstream GXP1450 1.0.4.23
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 9 [ 13]: Privacy: none
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 10 [ 58]: P-Preferred-Identity: "Jared Lam" <sip:2600L1@64.5.99.180>
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 11 [ 32]: Supported: replaces, path, timer
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 12 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 14 [ 47]: Accept: application/sdp, application/dtmf-relay
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 15 [ 19]: Content-Length: 410
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 16 [ 0]:
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 0 [ 3]: v=0
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 1 [ 39]: o=2600L1 8000 8000 IN IP4 76.14.161.117
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 2 [ 10]: s=SIP Call
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 3 [ 22]: c=IN IP4 76.14.161.117
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 4 [ 5]: t=0 0
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 5 [ 44]: m=audio 59530 RTP/AVP 0 13 9 8 18 4 97 2 101
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 6 [ 10]: a=sendrecv
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 8 [ 10]: a=ptime:20
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 9 [ 19]: a=rtpmap:13 CN/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 10 [ 20]: a=rtpmap:9 G722/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 11 [ 20]: a=rtpmap:8 PCMA/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 12 [ 21]: a=rtpmap:18 G729/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 13 [ 20]: a=rtpmap:4 G723/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 14 [ 21]: a=rtpmap:97 iLBC/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 15 [ 17]: a=fmtp:97 mode=30
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 16 [ 23]: a=rtpmap:2 G726-32/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 17 [ 33]: a=rtpmap:101 telephone-event/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 18 [ 15]: a=fmtp:101 0-15
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: = Looking for Call ID: 374146249-17329-28@HG.BE.BGB.BBH (Checking From) --From tag 2128823278 --To-tag
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.5.99.180:5060
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Allocating new SIP dialog for 374146249-17329-28@HG.BE.BGB.BBH - INVITE (No RTP)
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Nov 25 16:51:14] DEBUG[2065][C-00000003] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, path, timer"
[Nov 25 16:51:14] DEBUG[2065][C-00000003] sip/reqresp_parser.c: Found SIP option: -replaces-
[Nov 25 16:51:14] DEBUG[2065][C-00000003] sip/reqresp_parser.c: Matched SIP option: replaces
[Nov 25 16:51:14] DEBUG[2065][C-00000003] sip/reqresp_parser.c: Found SIP option: -path-
[Nov 25 16:51:14] DEBUG[2065][C-00000003] sip/reqresp_parser.c: Matched SIP option: path
[Nov 25 16:51:14] DEBUG[2065][C-00000003] sip/reqresp_parser.c: Found SIP option: -timer-
[Nov 25 16:51:14] DEBUG[2065][C-00000003] sip/reqresp_parser.c: Matched SIP option: timer
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Initializing initreq for method INVITE - callid 374146249-17329-28@HG.BE.BGB.BBH
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #64
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 76.14.161.117:17329
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 0 [ 35]: ACK sip:7168124@64.5.99.180 SIP/2.0
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 76.14.161.117:17329;branch=z9hG4bK1752883052;rport
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 2 [ 57]: From: "Jared Lam" <sip:2600L1@64.5.99.180>;tag=2128823278
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 3 [ 44]: To: <sip:7168124@64.5.99.180>;tag=as102b84be
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 4 [ 41]: Call-ID: 374146249-17329-28@HG.BE.BGB.BBH
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 5 [ 13]: CSeq: 270 ACK
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 6 [ 17]: Content-Length: 0
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: = Looking for Call ID: 374146249-17329-28@HG.BE.BGB.BBH (Checking From) --From tag 2128823278 --To-tag as102b84be
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #64
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Stopping retransmission on '374146249-17329-28@HG.BE.BGB.BBH' of Response 270: Match Found
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 0 [ 38]: INVITE sip:7168124@64.5.99.180 SIP/2.0
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 1 [ 66]: Via: SIP/2.0/UDP 76.14.161.117:17329;branch=z9hG4bK149596414;rport
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 2 [ 57]: From: "Jared Lam" <sip:2600L1@64.5.99.180>;tag=2128823278
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 3 [ 29]: To: <sip:7168124@64.5.99.180>
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 4 [ 41]: Call-ID: 374146249-17329-28@HG.BE.BGB.BBH
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 5 [ 16]: CSeq: 271 INVITE
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 6 [ 53]: Contact: "Jared Lam" <sip:2600L1@76.14.161.117:17329>
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 7 [165]: Authorization: Digest username="2600L1", realm="defiant", nonce="02936234", uri="sip:7168124@64.5.99.180", response="e059dc2fb203dba10c66f8f1679c17db", algorithm=MD5
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 9 [ 40]: User-Agent: Grandstream GXP1450 1.0.4.23
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 10 [ 13]: Privacy: none
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 11 [ 58]: P-Preferred-Identity: "Jared Lam" <sip:2600L1@64.5.99.180>
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 12 [ 32]: Supported: replaces, path, timer
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 13 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 15 [ 47]: Accept: application/sdp, application/dtmf-relay
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 16 [ 19]: Content-Length: 410
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 17 [ 0]:
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 0 [ 3]: v=0
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 1 [ 39]: o=2600L1 8000 8000 IN IP4 76.14.161.117
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 2 [ 10]: s=SIP Call
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 3 [ 22]: c=IN IP4 76.14.161.117
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 4 [ 5]: t=0 0
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 5 [ 44]: m=audio 59530 RTP/AVP 0 13 9 8 18 4 97 2 101
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 6 [ 10]: a=sendrecv
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 8 [ 10]: a=ptime:20
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 9 [ 19]: a=rtpmap:13 CN/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 10 [ 20]: a=rtpmap:9 G722/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 11 [ 20]: a=rtpmap:8 PCMA/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 12 [ 21]: a=rtpmap:18 G729/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 13 [ 20]: a=rtpmap:4 G723/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 14 [ 21]: a=rtpmap:97 iLBC/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 15 [ 17]: a=fmtp:97 mode=30
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 16 [ 23]: a=rtpmap:2 G726-32/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 17 [ 33]: a=rtpmap:101 telephone-event/8000
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Body 18 [ 15]: a=fmtp:101 0-15
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: = Looking for Call ID: 374146249-17329-28@HG.BE.BGB.BBH (Checking From) --From tag 2128823278 --To-tag
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Initializing initreq for method INVITE - callid 374146249-17329-28@HG.BE.BGB.BBH
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Setting NAT on RTP to On
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing session-level SDP o=2600L1 8000 8000 IN IP4 76.14.161.117... UNSUPPORTED OR FAILED.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED OR FAILED.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing session-level SDP c=IN IP4 76.14.161.117... OK.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:13 CN/8000... OK.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:4 G723/8000... OK.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 iLBC/8000... OK.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=fmtp:97 mode=30... OK.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:2 G726-32/8000... OK.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: We're settling with these formats: (ulaw|g722)
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Checking SIP call limits for device 2600L1
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Updating call counter for incoming call
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: *** Our native formats are (g722)
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: *** Joint capabilities are (ulaw|g722)
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: *** Our capabilities are (ulaw|g722)
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: *** AST_CODEC_CHOOSE formats are g722
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: This channel will not be able to handle video.
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: build_route: Contact hop: "Jared Lam" <sip:2600L1@76.14.161.117:17329>
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: SIP/2600L1-00000006: New call is still down.... Trying...
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 76.14.161.117:17329
[Nov 25 16:51:14] DEBUG[2057] chan_sip.c: Checking device state for peer 2600L1
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Asked to create a SIP channel with formats: (g722)
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Allocating new SIP dialog for 34240f704b660bb853c1e18949c156d5@64.5.99.180:5060 - INVITE (No RTP)
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Setting NAT on RTP to On
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: OBPROXY: Not applying OBproxy to this call
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.5.99.180:5060
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: *** Our native formats are (ulaw)
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: *** Joint capabilities are (nothing)
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: *** Our capabilities are (ulaw)
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: *** Our preferred formats from the incoming channel are (g722)
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: This channel will not be able to handle video.
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Outgoing Call for 19167168124
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Updating call counter for outgoing call
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: ** Our capability: (ulaw) Video flag: False Text flag: False
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: ** Our prefcodec: (g722)
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: -- Done with adding codecs to SDP
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Done building SDP. Settling with this capability: (ulaw)
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Initializing initreq for method INVITE - callid 4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Header 0 [ 49]: INVITE sip:19167168124@losangeles.voip.ms SIP/2.0
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Header 1 [ 56]: Via: SIP/2.0/UDP 64.5.99.180:5060;branch=z9hG4bK13e9a605
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Header 3 [ 62]: From: "Jared Lam" <sip:106011_jls3@64.5.99.180>;tag=as73bf10ff
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Header 4 [ 40]: To: <sip:19167168124@losangeles.voip.ms>
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Header 5 [ 43]: Contact: <sip:106011_jls3@64.5.99.180:5060>
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Header 6 [ 58]: Call-ID: 4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 11.0.1
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Header 9 [ 35]: Date: Mon, 26 Nov 2012 00:51:14 GMT
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Header 12 [ 93]: Remote-Party-ID: "Jared Lam" <sip:9162824726@64.5.99.180>;party=calling;privacy=off;screen=no
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #67
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 67.215.241.250:5060
[Nov 25 16:51:14] DEBUG[2090][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 76.14.161.117:17329
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 0 [ 41]: SIP/2.0 407 Proxy Authentication Required
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 1 [ 77]: Via: SIP/2.0/UDP 64.5.99.180:5060;branch=z9hG4bK13e9a605;received=64.5.99.180
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 2 [ 62]: From: "Jared Lam" <sip:106011_jls3@64.5.99.180>;tag=as73bf10ff
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 3 [ 55]: To: <sip:19167168124@losangeles.voip.ms>;tag=as37624bff
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 4 [ 58]: Call-ID: 4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 6 [ 25]: User-Agent: VoIPMS/SERAST
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 8 [ 19]: Supported: replaces
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 9 [ 86]: Proxy-Authenticate: Digest algorithm=MD5, realm="losangeles.voip.ms", nonce="5d88b8b1"
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 10 [ 17]: Content-Length: 0
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: = Looking for Call ID: 4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060 (Checking To) --From tag as73bf10ff --To-tag as37624bff
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Acked pending invite 102
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #67
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Stopping retransmission on '4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060' of Request 102: Match Found
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: SIP response 407 to standard invite
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Trying to put 'ACK sip:191' onto UDP socket destined for 67.215.241.250:5060
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Auth attempt 1 on INVITE
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: ** Our capability: (ulaw) Video flag: False Text flag: False
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: ** Our prefcodec: (g722)
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: -- Done with adding codecs to SDP
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Done building SDP. Settling with this capability: (ulaw)
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #70
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 67.215.241.250:5060
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 1 [ 77]: Via: SIP/2.0/UDP 64.5.99.180:5060;branch=z9hG4bK23debecc;received=64.5.99.180
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 2 [ 62]: From: "Jared Lam" <sip:106011_jls3@64.5.99.180>;tag=as73bf10ff
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 3 [ 40]: To: <sip:19167168124@losangeles.voip.ms>
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 4 [ 58]: Call-ID: 4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 6 [ 25]: User-Agent: VoIPMS/SERAST
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 8 [ 19]: Supported: replaces
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 9 [ 41]: Contact: <sip:19167168124@67.215.241.250>
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: Header 10 [ 17]: Content-Length: 0
[Nov 25 16:51:14] DEBUG[2065] chan_sip.c: = Looking for Call ID: 4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060 (Checking To) --From tag as73bf10ff --To-tag
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: *** SIP TIMER: Cancelling retransmission #70 - INVITE (got response)
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060' Request 103: Found
[Nov 25 16:51:14] DEBUG[2065][C-00000003] chan_sip.c: SIP response 100 to standard invite
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 0 [ 32]: REGISTER sip:64.5.99.180 SIP/2.0
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 1 [ 66]: Via: SIP/2.0/UDP 76.14.161.117:17329;branch=z9hG4bK357538498;rport
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 2 [ 45]: From: <sip:2600L1@64.5.99.180>;tag=1744819414
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 3 [ 28]: To: <sip:2600L1@64.5.99.180>
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 4 [ 41]: Call-ID: 1189973717-45417-1@HG.BE.BGB.BBH
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 5 [ 19]: CSeq: 2055 REGISTER
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 6 [114]: Contact: <sip:2600L1@76.14.161.117:17329>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B822D7D40>"
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 7 [157]: Authorization: Digest username="2600L1", realm="defiant", nonce="307bfa5c", uri="sip:64.5.99.180", response="6da13efed2da22f4c7ba8b26b45dbe91", algorithm=MD5
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 9 [ 40]: User-Agent: Grandstream GXP1450 1.0.4.23
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 10 [ 15]: Supported: path
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 11 [ 13]: Expires: 3600
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 12 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 13 [ 17]: Content-Length: 0
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: = Looking for Call ID: 1189973717-45417-1@HG.BE.BGB.BBH (Checking From) --From tag 1744819414 --To-tag
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.5.99.180:5060
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Allocating new SIP dialog for 1189973717-45417-1@HG.BE.BGB.BBH - REGISTER (No RTP)
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Initializing initreq for method REGISTER - callid 1189973717-45417-1@HG.BE.BGB.BBH
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 76.14.161.117:17329
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 0 [ 32]: REGISTER sip:64.5.99.180 SIP/2.0
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 76.14.161.117:17329;branch=z9hG4bK1127759780;rport
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 2 [ 45]: From: <sip:2600L1@64.5.99.180>;tag=1744819414
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 3 [ 28]: To: <sip:2600L1@64.5.99.180>
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 4 [ 41]: Call-ID: 1189973717-45417-1@HG.BE.BGB.BBH
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 5 [ 19]: CSeq: 2056 REGISTER
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 6 [114]: Contact: <sip:2600L1@76.14.161.117:17329>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B822D7D40>"
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 7 [157]: Authorization: Digest username="2600L1", realm="defiant", nonce="2f59072e", uri="sip:64.5.99.180", response="7a472fb7ae6f21393f7f9d2cad79a9af", algorithm=MD5
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 9 [ 40]: User-Agent: Grandstream GXP1450 1.0.4.23
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 10 [ 15]: Supported: path
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 11 [ 13]: Expires: 3600
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 12 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 13 [ 17]: Content-Length: 0
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: = Looking for Call ID: 1189973717-45417-1@HG.BE.BGB.BBH (Checking From) --From tag 1744819414 --To-tag
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Initializing initreq for method REGISTER - callid 1189973717-45417-1@HG.BE.BGB.BBH
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Store REGISTER's src-IP:port for call routing.
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Allocating new SIP dialog for 6082627c031cf546759f6da66b94fea4@64.5.99.180:5060 - OPTIONS (No RTP)
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.5.99.180:5060
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Initializing initreq for method OPTIONS - callid 1557895d4009b927690028fb602d9296@64.5.99.180:5060
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 0 [ 46]: OPTIONS sip:2600L1@76.14.161.117:17329 SIP/2.0
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 64.5.99.180:5060;branch=z9hG4bK7c755eae;rport
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 3 [ 58]: From: "asterisk" <sip:asterisk@64.5.99.180>;tag=as02dfc9ff
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 4 [ 36]: To: <sip:2600L1@76.14.161.117:17329>
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 5 [ 40]: Contact: <sip:asterisk@64.5.99.180:5060>
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 6 [ 58]: Call-ID: 1557895d4009b927690028fb602d9296@64.5.99.180:5060
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 11.0.1
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 9 [ 35]: Date: Mon, 26 Nov 2012 00:51:18 GMT
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #73
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 76.14.161.117:17329
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 76.14.161.117:17329
[Nov 25 16:51:18] DEBUG[2057] chan_sip.c: Checking device state for peer 2600L1
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 64.5.99.180:5060;branch=z9hG4bK7c755eae;rport=5060
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 2 [ 58]: From: "asterisk" <sip:asterisk@64.5.99.180>;tag=as02dfc9ff
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 3 [ 50]: To: <sip:2600L1@76.14.161.117:17329>;tag=234829099
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 4 [ 58]: Call-ID: 1557895d4009b927690028fb602d9296@64.5.99.180:5060
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 6 [ 32]: Supported: replaces, path, timer
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 7 [ 40]: User-Agent: Grandstream GXP1450 1.0.4.23
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 8 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Header 9 [ 17]: Content-Length: 0
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: = Looking for Call ID: 1557895d4009b927690028fb602d9296@64.5.99.180:5060 (Checking To) --From tag as02dfc9ff --To-tag 234829099
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #73
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Stopping retransmission on '1557895d4009b927690028fb602d9296@64.5.99.180:5060' of Request 102: Match Found
[Nov 25 16:51:18] DEBUG[2065] chan_sip.c: Destroying SIP dialog 1557895d4009b927690028fb602d9296@64.5.99.180:5060
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Header 0 [ 28]: SIP/2.0 183 Session Progress
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Header 1 [ 77]: Via: SIP/2.0/UDP 64.5.99.180:5060;branch=z9hG4bK23debecc;received=64.5.99.180
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Header 2 [ 62]: From: "Jared Lam" <sip:106011_jls3@64.5.99.180>;tag=as73bf10ff
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Header 3 [ 55]: To: <sip:19167168124@losangeles.voip.ms>;tag=as79478509
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Header 4 [ 58]: Call-ID: 4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Header 6 [ 25]: User-Agent: VoIPMS/SERAST
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Header 8 [ 19]: Supported: replaces
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Header 9 [ 41]: Contact: <sip:19167168124@67.215.241.250>
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Header 11 [ 19]: Content-Length: 244
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Header 12 [ 0]:
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Body 0 [ 3]: v=0
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Body 1 [ 40]: o=root 16887 16887 IN IP4 67.215.241.250
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Body 2 [ 9]: s=session
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Body 3 [ 23]: c=IN IP4 67.215.241.250
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Body 4 [ 5]: t=0 0
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Body 5 [ 27]: m=audio 14172 RTP/AVP 0 101
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - -
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Body 10 [ 10]: a=ptime:20
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: Body 11 [ 10]: a=sendrecv
[Nov 25 16:51:19] DEBUG[2065] chan_sip.c: = Looking for Call ID: 4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060 (Checking To) --From tag as73bf10ff --To-tag as79478509
[Nov 25 16:51:19] DEBUG[2065][C-00000003] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060' Request 103: Found
[Nov 25 16:51:19] DEBUG[2065][C-00000003] chan_sip.c: SIP response 183 to standard invite
[Nov 25 16:51:19] DEBUG[2065][C-00000003] chan_sip.c: build_route: Contact hop: <sip:19167168124@67.215.241.250>
[Nov 25 16:51:19] DEBUG[2065][C-00000003] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Nov 25 16:51:19] DEBUG[2065][C-00000003] chan_sip.c: Processing session-level SDP o=root 16887 16887 IN IP4 67.215.241.250... UNSUPPORTED OR FAILED.
[Nov 25 16:51:19] DEBUG[2065][C-00000003] chan_sip.c: Processing session-level SDP s=session... UNSUPPORTED OR FAILED.
[Nov 25 16:51:19] DEBUG[2065][C-00000003] chan_sip.c: Processing session-level SDP c=IN IP4 67.215.241.250... OK.
[Nov 25 16:51:19] DEBUG[2065][C-00000003] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[Nov 25 16:51:19] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Nov 25 16:51:19] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Nov 25 16:51:19] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED.
[Nov 25 16:51:19] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED OR FAILED.
[Nov 25 16:51:19] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[Nov 25 16:51:19] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Nov 25 16:51:19] DEBUG[2065][C-00000003] chan_sip.c: We're settling with these formats: (ulaw)
[Nov 25 16:51:19] DEBUG[2065][C-00000003] chan_sip.c: We have an owner, now see if we need to change this call
[Nov 25 16:51:19] DEBUG[2065][C-00000003] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw)
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 1 [ 77]: Via: SIP/2.0/UDP 64.5.99.180:5060;branch=z9hG4bK23debecc;received=64.5.99.180
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 2 [ 62]: From: "Jared Lam" <sip:106011_jls3@64.5.99.180>;tag=as73bf10ff
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 3 [ 55]: To: <sip:19167168124@losangeles.voip.ms>;tag=as79478509
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 4 [ 58]: Call-ID: 4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 6 [ 25]: User-Agent: VoIPMS/SERAST
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 8 [ 19]: Supported: replaces
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 9 [ 41]: Contact: <sip:19167168124@67.215.241.250>
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 11 [ 19]: Content-Length: 244
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 12 [ 0]:
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Body 0 [ 3]: v=0
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Body 1 [ 40]: o=root 16887 16888 IN IP4 67.215.241.250
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Body 2 [ 9]: s=session
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Body 3 [ 23]: c=IN IP4 67.215.241.250
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Body 4 [ 5]: t=0 0
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Body 5 [ 27]: m=audio 14172 RTP/AVP 0 101
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - -
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Body 10 [ 10]: a=ptime:20
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Body 11 [ 10]: a=sendrecv
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: = Looking for Call ID: 4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060 (Checking To) --From tag as73bf10ff --To-tag as79478509
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Acked pending invite 103
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Stopping retransmission on '4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060' of Request 103: Match Found
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: SIP response 200 to standard invite
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Processing session-level SDP o=root 16887 16888 IN IP4 67.215.241.250... UNSUPPORTED OR FAILED.
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Processing session-level SDP s=session... UNSUPPORTED OR FAILED.
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Processing session-level SDP c=IN IP4 67.215.241.250... OK.
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED.
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED OR FAILED.
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: We're settling with these formats: (ulaw)
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: We have an owner, now see if we need to change this call
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw)
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Updating call counter for outgoing call
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: build_route: Contact hop: <sip:19167168124@67.215.241.250>
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Strict routing enforced for session 4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Trying to put 'ACK sip:191' onto UDP socket destined for 67.215.241.250:5060
[Nov 25 16:51:23] DEBUG[2090][C-00000003] chan_sip.c: SIP answering channel: SIP/2600L1-00000006
[Nov 25 16:51:23] DEBUG[2090][C-00000003] chan_sip.c: Setting framing from config on incoming call
[Nov 25 16:51:23] DEBUG[2090][C-00000003] chan_sip.c: ** Our capability: (ulaw|g722) Video flag: True Text flag: True
[Nov 25 16:51:23] DEBUG[2090][C-00000003] chan_sip.c: ** Our prefcodec: (nothing)
[Nov 25 16:51:23] DEBUG[2090][C-00000003] chan_sip.c: -- Done with adding codecs to SDP
[Nov 25 16:51:23] DEBUG[2090][C-00000003] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|g722)
[Nov 25 16:51:23] DEBUG[2090][C-00000003] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #77
[Nov 25 16:51:23] DEBUG[2090][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 76.14.161.117:17329
[Nov 25 16:51:23] DEBUG[2057] chan_sip.c: Checking device state for peer voipms
[Nov 25 16:51:23] DEBUG[2057] chan_sip.c: Checking device state for peer 2600L1
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 0 [ 40]: ACK sip:7168124@64.5.99.180:5060 SIP/2.0
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 76.14.161.117:17329;branch=z9hG4bK1571678258;rport
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 2 [ 57]: From: "Jared Lam" <sip:2600L1@64.5.99.180>;tag=2128823278
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 3 [ 44]: To: <sip:7168124@64.5.99.180>;tag=as10c51fd8
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 4 [ 41]: Call-ID: 374146249-17329-28@HG.BE.BGB.BBH
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 5 [ 13]: CSeq: 271 ACK
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 6 [ 41]: Contact: <sip:2600L1@76.14.161.117:17329>
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 8 [ 32]: Supported: replaces, path, timer
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 9 [ 40]: User-Agent: Grandstream GXP1450 1.0.4.23
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 10 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: Header 11 [ 17]: Content-Length: 0
[Nov 25 16:51:23] DEBUG[2065] chan_sip.c: = Looking for Call ID: 374146249-17329-28@HG.BE.BGB.BBH (Checking From) --From tag 2128823278 --To-tag as10c51fd8
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #77
[Nov 25 16:51:23] DEBUG[2065][C-00000003] chan_sip.c: Stopping retransmission on '374146249-17329-28@HG.BE.BGB.BBH' of Response 271: Match Found
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 0 [ 40]: BYE sip:7168124@64.5.99.180:5060 SIP/2.0
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 1 [ 66]: Via: SIP/2.0/UDP 76.14.161.117:17329;branch=z9hG4bK119980366;rport
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 2 [ 57]: From: "Jared Lam" <sip:2600L1@64.5.99.180>;tag=2128823278
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 3 [ 44]: To: <sip:7168124@64.5.99.180>;tag=as10c51fd8
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 4 [ 41]: Call-ID: 374146249-17329-28@HG.BE.BGB.BBH
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 5 [ 13]: CSeq: 272 BYE
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 6 [ 41]: Contact: <sip:2600L1@76.14.161.117:17329>
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 8 [ 32]: Supported: replaces, path, timer
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 9 [ 40]: User-Agent: Grandstream GXP1450 1.0.4.23
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 10 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 11 [ 17]: Content-Length: 0
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: = Looking for Call ID: 374146249-17329-28@HG.BE.BGB.BBH (Checking From) --From tag 2128823278 --To-tag as10c51fd8
[Nov 25 16:51:41] DEBUG[2065][C-00000003] chan_sip.c: **** Received BYE (8) - Command in SIP BYE
[Nov 25 16:51:41] DEBUG[2065][C-00000003] chan_sip.c: Initializing initreq for method BYE - callid 374146249-17329-28@HG.BE.BGB.BBH
[Nov 25 16:51:41] DEBUG[2065][C-00000003] chan_sip.c: Setting SIP_ALREADYGONE on dialog 374146249-17329-28@HG.BE.BGB.BBH
[Nov 25 16:51:41] DEBUG[2065][C-00000003] chan_sip.c: Received bye, issuing owner hangup
[Nov 25 16:51:41] DEBUG[2065][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 76.14.161.117:17329
[Nov 25 16:51:41] DEBUG[2090][C-00000003] chan_sip.c: Hangup call SIP/voipms-00000007, SIP callid 4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060
[Nov 25 16:51:41] DEBUG[2090][C-00000003] chan_sip.c: Strict routing enforced for session 4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060
[Nov 25 16:51:41] DEBUG[2090][C-00000003] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #80
[Nov 25 16:51:41] DEBUG[2090][C-00000003] chan_sip.c: Trying to put 'BYE sip:191' onto UDP socket destined for 67.215.241.250:5060
[Nov 25 16:51:41] DEBUG[2057] chan_sip.c: Checking device state for peer voipms
[Nov 25 16:51:41] DEBUG[2090][C-00000003] chan_sip.c: Hangup call SIP/2600L1-00000006, SIP callid 374146249-17329-28@HG.BE.BGB.BBH
[Nov 25 16:51:41] DEBUG[2057] chan_sip.c: Checking device state for peer 2600L1
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 1 [ 77]: Via: SIP/2.0/UDP 64.5.99.180:5060;branch=z9hG4bK573a6d1a;received=64.5.99.180
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 2 [ 62]: From: "Jared Lam" <sip:106011_jls3@64.5.99.180>;tag=as73bf10ff
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 3 [ 55]: To: <sip:19167168124@losangeles.voip.ms>;tag=as79478509
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 4 [ 58]: Call-ID: 4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 5 [ 13]: CSeq: 104 BYE
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 6 [ 25]: User-Agent: VoIPMS/SERAST
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 8 [ 19]: Supported: replaces
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Header 9 [ 17]: Content-Length: 0
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: = Looking for Call ID: 4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060 (Checking To) --From tag as73bf10ff --To-tag as79478509
[Nov 25 16:51:41] DEBUG[2065][C-00000003] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #80
[Nov 25 16:51:41] DEBUG[2065][C-00000003] chan_sip.c: Stopping retransmission on '4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060' of Request 104: Match Found
[Nov 25 16:51:41] DEBUG[2065] chan_sip.c: Destroying SIP dialog 4fe321e63eb8dff31817f13273e07945@64.5.99.180:5060
[Nov 25 16:51:47] DEBUG[2065] chan_sip.c: Auto destroying SIP dialog '374146249-17329-28@HG.BE.BGB.BBH'
[Nov 25 16:51:47] DEBUG[2065] chan_sip.c: Destroying SIP dialog 374146249-17329-28@HG.BE.BGB.BBH
[Nov 25 16:51:50] DEBUG[2065] chan_sip.c: Auto destroying SIP dialog '1189973717-45417-1@HG.BE.BGB.BBH'
[Nov 25 16:51:50] DEBUG[2065] chan_sip.c: Destroying SIP dialog 1189973717-45417-1@HG.BE.BGB.BBH
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: OBPROXY: Not applying OBproxy to this call
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: SIP Registry losangeles.voip.ms: refcount now 3
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: SIP Registry losangeles.voip.ms: refcount now 2
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Allocating new SIP dialog for 2f5141983d4bb1a8129737235f256f1d@64.5.99.180 - REGISTER (No RTP)
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: OBPROXY: Not applying OBproxy to this call
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: SIP Registry losangeles.voip.ms: refcount now 3
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 64.5.99.180:5060
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: SIP Registry losangeles.voip.ms: refcount now 4
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Scheduled a registration timeout for losangeles.voip.ms id #81
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: >>> Re-using Auth data for 106011_jls3@losangeles.voip.ms
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Initializing initreq for method REGISTER - callid 2f5141983d4bb1a8129737235f256f1d@64.5.99.180
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 0 [ 39]: REGISTER sip:losangeles.voip.ms SIP/2.0
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 1 [ 56]: Via: SIP/2.0/UDP 64.5.99.180:5060;branch=z9hG4bK54d9da7c
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 3 [ 57]: From: <sip:106011_jls3@losangeles.voip.ms>;tag=as20e68865
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 4 [ 40]: To: <sip:106011_jls3@losangeles.voip.ms>
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 5 [ 53]: Call-ID: 2f5141983d4bb1a8129737235f256f1d@64.5.99.180
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 6 [ 18]: CSeq: 106 REGISTER
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 7 [ 31]: User-Agent: Asterisk PBX 11.0.1
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 8 [180]: Authorization: Digest username="106011_jls3", realm="losangeles.voip.ms", algorithm=MD5, uri="sip:losangeles.voip.ms", nonce="7b33b1bc", response="90fdffe6c0ef0d004547c656eb99be8a"
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 9 [ 12]: Expires: 120
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 10 [ 33]: Contact: <sip:s@64.5.99.180:5060>
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: REGISTER attempt 1 to 106011_jls3@losangeles.voip.ms
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #82
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 67.215.241.250:5060
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: SIP Registry losangeles.voip.ms: refcount now 3
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 1 [ 77]: Via: SIP/2.0/UDP 64.5.99.180:5060;branch=z9hG4bK54d9da7c;received=64.5.99.180
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 2 [ 57]: From: <sip:106011_jls3@losangeles.voip.ms>;tag=as20e68865
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 3 [ 40]: To: <sip:106011_jls3@losangeles.voip.ms>
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 4 [ 53]: Call-ID: 2f5141983d4bb1a8129737235f256f1d@64.5.99.180
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 5 [ 18]: CSeq: 106 REGISTER
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 6 [ 25]: User-Agent: VoIPMS/SERAST
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 8 [ 19]: Supported: replaces
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 9 [ 17]: Content-Length: 0
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: = Looking for Call ID: 2f5141983d4bb1a8129737235f256f1d@64.5.99.180 (Checking To) --From tag as20e68865 --To-tag
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 1 [ 77]: Via: SIP/2.0/UDP 64.5.99.180:5060;branch=z9hG4bK54d9da7c;received=64.5.99.180
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 2 [ 57]: From: <sip:106011_jls3@losangeles.voip.ms>;tag=as20e68865
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 3 [ 55]: To: <sip:106011_jls3@losangeles.voip.ms>;tag=as2c799403
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 4 [ 53]: Call-ID: 2f5141983d4bb1a8129737235f256f1d@64.5.99.180
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 5 [ 18]: CSeq: 106 REGISTER
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 6 [ 25]: User-Agent: VoIPMS/SERAST
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 8 [ 19]: Supported: replaces
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 9 [ 84]: WWW-Authenticate: Digest algorithm=MD5, realm="losangeles.voip.ms", nonce="6ed11ec7"
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 10 [ 17]: Content-Length: 0
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: = Looking for Call ID: 2f5141983d4bb1a8129737235f256f1d@64.5.99.180 (Checking To) --From tag as20e68865 --To-tag as2c799403
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #82
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Stopping retransmission on '2f5141983d4bb1a8129737235f256f1d@64.5.99.180' of Request 106: Match Found
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: OBPROXY: Not applying OBproxy to this call
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: SIP Registry losangeles.voip.ms: refcount now 4
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: SIP Registry losangeles.voip.ms: refcount now 3
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Initializing already initialized SIP dialog 2f5141983d4bb1a8129737235f256f1d@64.5.99.180 (presumably reinvite)
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 0 [ 39]: REGISTER sip:losangeles.voip.ms SIP/2.0
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 1 [ 56]: Via: SIP/2.0/UDP 64.5.99.180:5060;branch=z9hG4bK10460463
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 3 [ 57]: From: <sip:106011_jls3@losangeles.voip.ms>;tag=as50b9ef49
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 4 [ 40]: To: <sip:106011_jls3@losangeles.voip.ms>
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 5 [ 53]: Call-ID: 2f5141983d4bb1a8129737235f256f1d@64.5.99.180
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 6 [ 18]: CSeq: 107 REGISTER
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 7 [ 31]: User-Agent: Asterisk PBX 11.0.1
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 8 [180]: Authorization: Digest username="106011_jls3", realm="losangeles.voip.ms", algorithm=MD5, uri="sip:losangeles.voip.ms", nonce="6ed11ec7", response="763fb60be7562d40f05c72d66f4308e9"
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 9 [ 12]: Expires: 120
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 10 [ 33]: Contact: <sip:s@64.5.99.180:5060>
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: REGISTER attempt 2 to 106011_jls3@losangeles.voip.ms
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #83
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 67.215.241.250:5060
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 1 [ 77]: Via: SIP/2.0/UDP 64.5.99.180:5060;branch=z9hG4bK10460463;received=64.5.99.180
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 2 [ 57]: From: <sip:106011_jls3@losangeles.voip.ms>;tag=as50b9ef49
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 3 [ 40]: To: <sip:106011_jls3@losangeles.voip.ms>
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 4 [ 53]: Call-ID: 2f5141983d4bb1a8129737235f256f1d@64.5.99.180
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 5 [ 18]: CSeq: 107 REGISTER
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 6 [ 25]: User-Agent: VoIPMS/SERAST
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 8 [ 19]: Supported: replaces
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 9 [ 17]: Content-Length: 0
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: = Looking for Call ID: 2f5141983d4bb1a8129737235f256f1d@64.5.99.180 (Checking To) --From tag as50b9ef49 --To-tag
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 1 [ 77]: Via: SIP/2.0/UDP 64.5.99.180:5060;branch=z9hG4bK10460463;received=64.5.99.180
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 2 [ 57]: From: <sip:106011_jls3@losangeles.voip.ms>;tag=as50b9ef49
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 3 [ 55]: To: <sip:106011_jls3@losangeles.voip.ms>;tag=as2c799403
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 4 [ 53]: Call-ID: 2f5141983d4bb1a8129737235f256f1d@64.5.99.180
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 5 [ 18]: CSeq: 107 REGISTER
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 6 [ 25]: User-Agent: VoIPMS/SERAST
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 8 [ 19]: Supported: replaces
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 9 [ 12]: Expires: 120
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 10 [ 45]: Contact: <sip:s@64.5.99.180:5060>;expires=120
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 11 [ 35]: Date: Mon, 26 Nov 2012 00:51:56 GMT
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Header 12 [ 17]: Content-Length: 0
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: = Looking for Call ID: 2f5141983d4bb1a8129737235f256f1d@64.5.99.180 (Checking To) --From tag as50b9ef49 --To-tag as2c799403
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #83
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Stopping retransmission on '2f5141983d4bb1a8129737235f256f1d@64.5.99.180' of Request 107: Match Found
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Registration successful
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: Cancelling timeout 81
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: SIP Registry losangeles.voip.ms: refcount now 2
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: SIP Registry losangeles.voip.ms: refcount now 1
[Nov 25 16:51:56] DEBUG[2065] chan_sip.c: SIP Registry losangeles.voip.ms: refcount now 2
jaredlam
Newsterisk
 
Posts: 3
Joined: Sat Nov 24, 2012 6:56 pm

Re: Asterisk 11 : Audio Issue : Codec Translation

Postby jaredlam » Sun Nov 25, 2012 8:00 pm

With new sip setttings (listed from the previous post under sip.conf), I don't get the one-way audio any more; it is either works or no audio at all. I have tested with g722 codec and ulaw codec for probably over 50 times calling my cell phone. With the ulaw codec, it works a lot more consistently than g722 codec.

RTP traffic not getting through from either direction? The Asterisk is on a public network with UDP port 5060 and UDP port range from 10000 - 20000 opened on the firewall. If figured if the NAT is the problem, I should get one-way audio.

Confused...

Thanks,
jaredlam
Newsterisk
 
Posts: 3
Joined: Sat Nov 24, 2012 6:56 pm

Re: Asterisk 11 : Audio Issue : Codec Translation

Postby david55 » Mon Nov 26, 2012 3:50 am

You didn't do sip set debug on, so one can't see the outgoing SDP.

Does the firewall have any restrictions on remote port numbers, as the 10000 to 20000 only relates to the local ones.
david55
Moves Like Spencer
 
Posts: 10549
Joined: Fri Sep 26, 2008 5:03 am


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