Unable to hear any sound from

[sampath11111]

Hi,
When i call for conference i unable to hear any sound from asterisk which is installed in ubuntu at amazon instance. i have installed dahdi, mysql module in asterisk. please help me to fix this. Following log printed in console when i call for conference.

Code: Select all

  == Parsing '/etc/asterisk/cli.conf':   == Found
*CLI> [Aug  6 13:56:52] NOTICE[12734]: chan_sip.c:27424 build_peer: The 'username' field for sip peers has been deprecated in favor of the term 'defaultuser'
       > Saved useragent "X-Lite release 5.0.0 stamp 67284" for peer 99966666
[Aug  6 13:56:52] NOTICE[12734]: chan_sip.c:20764 handle_response_peerpoke: Peer '99966666' is now Reachable. (276ms / 2000ms)
  == Using SIP RTP CoS mark 5
    -- Executing [10@test:1] Answer("SIP/99966666-00000000", "") in new stack
    -- Executing [10@test:2] Playback("SIP/99966666-00000000", "conference-theme") in new stack
    -- <SIP/99966666-00000000> Playing 'conference-theme.gsm' (language 'en')
    -- Executing [10@test:3] MeetMe("SIP/99966666-00000000", "1000") in new stack
  == Parsing '/etc/asterisk/meetme.conf':   == Found
    -- Created MeetMe conference 1023 for conference '1000'
    -- <SIP/99966666-00000000> Playing 'conf-getpin.slin' (language 'en')
[Aug  6 13:57:16] WARNING[12734]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission MzAwYjVlOTE5ODkwZWIwOTc3MDA1MTFlYjI1ZTJiOTA. for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 17663ms with no response
[Aug  6 13:57:16] WARNING[12734]: chan_sip.c:3670 retrans_pkt: Hanging up call MzAwYjVlOTE5ODkwZWIwOTc3MDA1MTFlYjI1ZTJiOTA. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
    -- Hungup 'DAHDI/pseudo-1900234286'
  == Spawn extension (test, 10, 3) exited non-zero on 'SIP/99966666-00000000'

OS:
Linux ip-10-168-190-222 3.2.0-25-virtual #40-Ubuntu SMP Wed May 23 22:20:17 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux

lscpu retuns
Architecture: x86_64
CPU op-mode(s): 32-bit, 64-bit
Byte Order: Little Endian
CPU(s): 2
On-line CPU(s) list: 0,1
Thread(s) per core: 2
Core(s) per socket: 1
Socket(s): 1
NUMA node(s): 1
Vendor ID: GenuineIntel
CPU family: 6
Model: 26
Stepping: 5
CPU MHz: 2266.746
BogoMIPS: 4533.49
Hypervisor vendor: Xen
Virtualization type: para
L1d cache: 32K
L1i cache: 32K
L2 cache: 256K
L3 cache: 4096K
NUMA node0 CPU(s): 0,1

modprobe -l | grep dahdi returns
updates/dkms/dahdi_echocan_kb1.ko
updates/dkms/dahdi_transcode.ko
updates/dkms/dahdi_dynamic.ko
updates/dkms/dahdi_echocan_oslec.ko
updates/dkms/dahdi_vpmadt032_loader.ko
updates/dkms/dahdi_echocan_sec2.ko
updates/dkms/dahdi_dynamic_loc.ko
updates/dkms/dahdi_echocan_sec.ko
updates/dkms/dahdi_echocan_jpah.ko
updates/dkms/dahdi_dummy.ko
updates/dkms/dahdi_voicebus.ko
updates/dkms/dahdi.ko
updates/dkms/dahdi_echocan_mg2.ko
updates/dkms/dahdi_dynamic_eth.ko

ls /dev/dahdi returns
channel ctl pseudo timer transcode

dahdi show status returns
Description Alarms IRQ bpviol CRC Fra Codi Options LBO
DAHDI_DUMMY/1 (source: HRtimer) 1 UNCONFI 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1)

when i do it in my local machine, everything works fine . But i dont know why it makes problem in amazon instance
Thanks in advance

[david55]

Probably a firewall problem. Also some versions of X-Lite have broken re-invite handling, and will drop a call after about 20s.

You need to get a SIP trace to find out which packet is failing to get through.

I would never recommend running Asterisk on a cloud VM.

[azhar117]

Set NAT =yes
it might work for u as it worked for me

[david55]

If this is a NAT issue, X-Lite supports STUN, so the correct solution would be to configure that under Accounts | Topology.

nat=yes is a hack in Asterisk that causes it to use the actual source IP address and port, rather than the ones that the protocol is telling it to use. It is for working round misconfigured or broken clients.

[sampath11111]

Thanks for your reply..

Problem identified and fixed!!

Once i changed dsn name to static ip in externaddr varible in sip.conf conference, i could hear the sound plays on the asterisk server as well as i can chat with other people in the conference without any problem. But in asterisk 1.6 dont have this problem. i mean it works fine with externip=domain_name in 1.6.

Now i have a problem in one to one chat. when we make a call to other user, both user unable to hear any voice from both end. Here is the log from asterisk console.

Code: Select all

  == Using SIP RTP CoS mark 5
    -- Executing [2002@default:1] Dial("SIP/2001-00000002", "SIP/2002") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/2002
    -- SIP/2002-00000003 is ringing
    -- SIP/2002-00000003 is ringing
    -- SIP/2002-00000003 is ringing
    -- SIP/2002-00000003 is ringing
    -- SIP/2002-00000003 is ringing
    -- SIP/2002-00000003 is ringing
    -- SIP/2002-00000003 is ringing
    -- SIP/2002-00000003 is ringing
    -- SIP/2002-00000003 is ringing
    -- SIP/2002-00000003 answered SIP/2001-00000002
    -- Locally bridging SIP/2001-00000002 and SIP/2002-00000003


sip.conf
[2001]
type=friend
secret=pass1
host=dynamic
nat=yes
qualify=yes
canreinvite=no
sipreinvite=no


[2002]
type=friend
secret=pass1
host=dynamic
nat=yes
canreinvite=no
sipreinvite=no

my dialplan:

[default]
exten => 2001,1,Dial(SIP/2001)
exten => 2001,2,Playback(beep)
exten => 2001,3,Hangup


exten => 2002,1,Dial(SIP/2002)
exten => 2002,2,Playback(beep)
exten => 2002,3,Hangup

dialplan debug default displays

Code: Select all

     In-mem exten Trie for Fast Extension Pattern Matching:


           Explanation: Node Contents Format = <char(s) to match>:<pattern?>:<specif>:[matched extension]
                        Where <char(s) to match> is a set of chars, any one of which should match the current character
                              <pattern?>: Y if this a pattern match (eg. _XZN[5-7]), N otherwise
                              <specif>: an assigned 'exactness' number for this matching char. The lower the number, the more exact the match
                              [matched exten]: If all chars matched to this point, which extension this matches. In form: EXTEN:<exten string>
                        In general, you match a trie node to a string character, from left to right. All possible matching chars
                        are in a string vertically, separated by an unbroken string of '+' characters.

[ Context 'default' created by 'pbx_config' ]
2:N:-:1:
+       0:N:-:1:
+       +       0:N:-:1:
+       +       +       2:N:-:1:EXTEN:2002(0x2e97250)
+       +       +       1:N:-:1:EXTEN:2001(0x2e95c20)

-= 1 context. =-
*CLI>

sip set debug peer for 2001 & 2002

Code: Select all

*CLI> sip set debug peer 2001
SIP Debugging Enabled for IP: 121.242.232.129
*CLI> sip set debug peer 2002
SIP Debugging Enabled for IP: 121.242.232.129
*CLI> Reliably Transmitting (NAT) to 121.242.232.129:51584:
OPTIONS sip:2002@184.169.169.99:5060 SIP/2.0
Via: SIP/2.0/UDP 184.169.169.99:5060;branch=z9hG4bK1af65bc7;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@184.169.169.99>;tag=as7fa81888
To: <sip:2002@184.169.169.99:5060>
Contact: <sip:asterisk@184.169.169.99:5060>
Call-ID: 2cc0cef9335ffa1e469810316b173c82@184.169.169.99:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 08 Aug 2012 14:28:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:121.242.232.129:51584 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.169.169.99:5060;branch=z9hG4bK1af65bc7;rport
To: <sip:2002@184.169.169.99:5060>
From: "asterisk" <sip:asterisk@184.169.169.99>;tag=as7fa81888
Call-ID: 2cc0cef9335ffa1e469810316b173c82@184.169.169.99:5060
CSeq: 102 OPTIONS
User-Agent: NCH Software Express Talk 3.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Accept: application/sdp
Supported: replaces
Content-Type: application/sdp
Content-Length: 305

v=0
o=NCHSoftware-Talk 1344435883 1344435883 IN IP4
s=Express Talk Call
c=IN IP4
t=0 0
m=audio 0 RTP/AVP 0 8 96 3 13 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (12 headers 14 lines) ---
Really destroying SIP dialog '2cc0cef9335ffa1e469810316b173c82@184.169.169.99:5060' Method: OPTIONS
  == Using SIP RTP CoS mark 5
[Aug  8 14:28:51] NOTICE[5161]: chan_sip.c:22622 handle_request_invite: Call from '2001' (121.242.232.129:52102) to extension '200' rejected because extension not found in context 'default'.
Reliably Transmitting (NAT) to 121.242.232.129:51584:
OPTIONS sip:2002@184.169.169.99:5060 SIP/2.0
Via: SIP/2.0/UDP 184.169.169.99:5060;branch=z9hG4bK4b47ef0f;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@184.169.169.99>;tag=as717bb627
To: <sip:2002@184.169.169.99:5060>
Contact: <sip:asterisk@184.169.169.99:5060>
Call-ID: 73d337c8592f1b5a13d925e776cf26a7@184.169.169.99:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 08 Aug 2012 14:28:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:121.242.232.129:51584 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.169.169.99:5060;branch=z9hG4bK4b47ef0f;rport
To: <sip:2002@184.169.169.99:5060>
From: "asterisk" <sip:asterisk@184.169.169.99>;tag=as717bb627
Call-ID: 73d337c8592f1b5a13d925e776cf26a7@184.169.169.99:5060
CSeq: 102 OPTIONS
User-Agent: NCH Software Express Talk 3.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Accept: application/sdp
Supported: replaces
Content-Type: application/sdp
Content-Length: 305

v=0
o=NCHSoftware-Talk 1344435884 1344435884 IN IP4
s=Express Talk Call
c=IN IP4
t=0 0
m=audio 0 RTP/AVP 0 8 96 3 13 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (12 headers 14 lines) ---
Really destroying SIP dialog '73d337c8592f1b5a13d925e776cf26a7@184.169.169.99:5060' Method: OPTIONS
  == Using SIP RTP CoS mark 5
    -- Executing [2002@default:1] Dial("SIP/2001-00000000", "SIP/2002") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 14858
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 121.242.232.129:51584:
INVITE sip:2002@184.169.169.99:5060 SIP/2.0
Via: SIP/2.0/UDP 184.169.169.99:5060;branch=z9hG4bK052a7a25;rport
Max-Forwards: 70
From: "Peacock" <sip:2001@184.169.169.99>;tag=as6e236871
To: <sip:2002@184.169.169.99:5060>
Contact: <sip:2001@184.169.169.99:5060>
Call-ID: 43500f4a7e70ccd451fa1c8f3e89e40d@184.169.169.99:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 08 Aug 2012 14:29:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 300

v=0
o=root 521233269 521233269 IN IP4 184.169.169.99
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 184.169.169.99
t=0 0
m=audio 14858 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/2002

<--- SIP read from UDP:121.242.232.129:51584 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 184.169.169.99:5060;branch=z9hG4bK052a7a25;rport
To: <sip:2002@184.169.169.99:5060>;tag=5736
From: "Peacock" <sip:2001@184.169.169.99>;tag=as6e236871
Call-ID: 43500f4a7e70ccd451fa1c8f3e89e40d@184.169.169.99:5060
CSeq: 102 INVITE
User-Agent: NCH Software Express Talk 3.10
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
list_route: no route
    -- SIP/2002-00000001 is ringing

<--- SIP read from UDP:121.242.232.129:51584 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 184.169.169.99:5060;branch=z9hG4bK052a7a25;rport
To: <sip:2002@184.169.169.99:5060>;tag=5736
From: "Peacock" <sip:2001@184.169.169.99>;tag=as6e236871
Call-ID: 43500f4a7e70ccd451fa1c8f3e89e40d@184.169.169.99:5060
CSeq: 102 INVITE
User-Agent: NCH Software Express Talk 3.10
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
list_route: no route
    -- SIP/2002-00000001 is ringing

<--- SIP read from UDP:121.242.232.129:51584 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 184.169.169.99:5060;branch=z9hG4bK052a7a25;rport
To: <sip:2002@184.169.169.99:5060>;tag=5736
From: "Peacock" <sip:2001@184.169.169.99>;tag=as6e236871
Call-ID: 43500f4a7e70ccd451fa1c8f3e89e40d@184.169.169.99:5060
CSeq: 102 INVITE
User-Agent: NCH Software Express Talk 3.10
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
list_route: no route
    -- SIP/2002-00000001 is ringing

<--- SIP read from UDP:121.242.232.129:51584 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.169.169.99:5060;branch=z9hG4bK052a7a25;rport
To: <sip:2002@184.169.169.99:5060>;tag=5736
From: "Peacock" <sip:2001@184.169.169.99>;tag=as6e236871
Call-ID: 43500f4a7e70ccd451fa1c8f3e89e40d@184.169.169.99:5060
CSeq: 102 INVITE
User-Agent: NCH Software Express Talk 3.10
Contact: <sip:2002@184.169.169.99:5060>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Accept: application/sdp
Supported: replaces
Content-Type: application/sdp
Content-Length: 281

v=0
o=NCHSoftware-Talk 1344435862 1344435864 IN IP4 184.169.169.99
s=Express Talk Call
c=IN IP4 192.168.1.50
t=0 0
m=audio 8000 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (13 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.50:8000
list_route: hop: <sip:2002@184.169.169.99:5060>
set_destination: Parsing <sip:2002@184.169.169.99:5060> for address/port to send to
set_destination: set destination to 184.169.169.99:5060
Transmitting (NAT) to 121.242.232.129:51584:
ACK sip:2002@184.169.169.99:5060 SIP/2.0
Via: SIP/2.0/UDP 184.169.169.99:5060;branch=z9hG4bK6a912cf3;rport
Max-Forwards: 70
From: "Peacock" <sip:2001@184.169.169.99>;tag=as6e236871
To: <sip:2002@184.169.169.99:5060>;tag=5736
Contact: <sip:2001@184.169.169.99:5060>
Call-ID: 43500f4a7e70ccd451fa1c8f3e89e40d@184.169.169.99:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---
    -- SIP/2002-00000001 answered SIP/2001-00000000
    -- Locally bridging SIP/2001-00000000 and SIP/2002-00000001
Reliably Transmitting (NAT) to 121.242.232.129:51584:
OPTIONS sip:2002@184.169.169.99:5060 SIP/2.0
Via: SIP/2.0/UDP 184.169.169.99:5060;branch=z9hG4bK0b030128;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@184.169.169.99>;tag=as1ded94ed
To: <sip:2002@184.169.169.99:5060>
Contact: <sip:asterisk@184.169.169.99:5060>
Call-ID: 436b5472346d803b6fcdeb847f9e350c@184.169.169.99:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 08 Aug 2012 14:29:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:121.242.232.129:51584 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.169.169.99:5060;branch=z9hG4bK0b030128;rport
To: <sip:2002@184.169.169.99:5060>
From: "asterisk" <sip:asterisk@184.169.169.99>;tag=as1ded94ed
Call-ID: 436b5472346d803b6fcdeb847f9e350c@184.169.169.99:5060
CSeq: 102 OPTIONS
User-Agent: NCH Software Express Talk 3.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Accept: application/sdp
Supported: replaces
Content-Type: application/sdp
Content-Length: 305

v=0
o=NCHSoftware-Talk 1344435885 1344435885 IN IP4
s=Express Talk Call
c=IN IP4
t=0 0
m=audio 0 RTP/AVP 0 8 96 3 13 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (12 headers 14 lines) ---
Really destroying SIP dialog '436b5472346d803b6fcdeb847f9e350c@184.169.169.99:5060' Method: OPTIONS

<--- SIP read from UDP:121.242.232.129:51584 --->
BYE sip:2001@184.169.169.99:5060 SIP/2.0
Via: SIP/2.0/UDP 184.169.169.99:5060;rport;branch=z9hG4bK22688
To: "Peacock" <sip:2001@184.169.169.99>;tag=as6e236871
From: <sip:2002@184.169.169.99:5060>;tag=5736
Call-ID: 43500f4a7e70ccd451fa1c8f3e89e40d@184.169.169.99:5060
CSeq: 626 BYE
Max-Forwards: 20
User-Agent: NCH Software Express Talk 3.10
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 121.242.232.129:51584 (NAT)
Scheduling destruction of SIP dialog '43500f4a7e70ccd451fa1c8f3e89e40d@184.169.169.99:5060' in 19392 ms (Method: BYE)

<--- Transmitting (NAT) to 121.242.232.129:51584 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.169.169.99:5060;branch=z9hG4bK22688;received=121.242.232.129;rport=51584
From: <sip:2002@184.169.169.99:5060>;tag=5736
To: "Peacock" <sip:2001@184.169.169.99>;tag=as6e236871
Call-ID: 43500f4a7e70ccd451fa1c8f3e89e40d@184.169.169.99:5060
CSeq: 626 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (default, 2002, 1) exited non-zero on 'SIP/2001-00000000'
Reliably Transmitting (NAT) to 121.242.232.129:51584:
OPTIONS sip:2002@184.169.169.99:5060 SIP/2.0
Via: SIP/2.0/UDP 184.169.169.99:5060;branch=z9hG4bK4f69a1b9;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@184.169.169.99>;tag=as62637ff8
To: <sip:2002@184.169.169.99:5060>
Contact: <sip:asterisk@184.169.169.99:5060>
Call-ID: 35162e1f121b27f639c825bd512fc5c4@184.169.169.99:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 08 Aug 2012 14:29:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:121.242.232.129:51584 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.169.169.99:5060;branch=z9hG4bK4f69a1b9;rport
To: <sip:2002@184.169.169.99:5060>
From: "asterisk" <sip:asterisk@184.169.169.99>;tag=as62637ff8
Call-ID: 35162e1f121b27f639c825bd512fc5c4@184.169.169.99:5060
CSeq: 102 OPTIONS
User-Agent: NCH Software Express Talk 3.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Accept: application/sdp
Supported: replaces
Content-Type: application/sdp
Content-Length: 305

v=0
o=NCHSoftware-Talk 1344435886 1344435886 IN IP4
s=Express Talk Call
c=IN IP4
t=0 0
m=audio 0 RTP/AVP 0 8 96 3 13 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->

Please help me fix this problem,
Thanks in advance

[david55]

You need to fix NCH Software Express Talk 3.10, or its router, so that it sends its public IP address rather than a private one, here:

c=IN IP4 192.168.1.50

Basically, you either need to make it aware that it is behind NAT, or you need a router that is sufficiently SIP aware to fix up the consequences of its not being aware.

Incidentally, dialplan debug is for dealing with rather technical implementation errors, not for run of the mill configuration problems.

Also, I see no evidence of an X-Lite in that trace.

[sampath11111]

Thanks again...

Once i shifted to X-Lite client everyting works fine!!.
Thanks a lot guys!!