have a Cisco 2800 like a voice GW, making a Trunk SIP with a Asterisk PBX. I don’t know the Asterisk configuration. My GW is used only to external calls, using a PRI port. We have problems with calls since 10 days ago: I can make a call but the called can’t hear me and I hear him.
The communication between the Asterisk Box and my 2800 is Internet, both have publics IPs, so there’s not firewall between them. I should say that my 2800 have been working till now like GW from differents ASterisk’s box, and we don’t have problems till now, but this “Asterisk’s cluster” have differents VoIP GW from differents partner and they use them on demand. We gave them support over our GW only when they have a problem.
So we are checking actually if this Asterisk Box is new or if it’s one of the oldes’t (from de Asterisk’s cluster).
I need some help about this problem. I attach logs with differents debugs from the asterisk pbx.
Asterisk’s owner tell me that their asterisk box don’t accept g711 codec but I don’t think the same looking their traces.
Coud you please help me? I need to know if the asterisk box is misconfigured or if the problem is my cisco 2800 gw (which work with another pbxs).
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Creating new subscription
Sending to 129.168.1.104 : 52980 (NAT)
Found peer ‘1713’ no
pbxa*CLI> sip no
<— Transmitting (no NAT) to 129.168.1.104:52980 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 129.168.1.104:52980;branch=z9hG4bK-dxxx54z-2f34490f07682f67-1—dxxx54z-;received=129.168.1.104;rport=52980
From: "1713"sip:1713@129.168.1.200;tag=716a5b17
To: "1713"sip:1713@129.168.1.200;tag=as04178e06
Call-ID: NDVhZjJjZWUxYzhmNzdiY2MzNjc5YmM3OGE2ZDMxMjY.
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“pbxa”, nonce="16d8597b"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘NDVhZjJjZWUxYzhmNzdiY2MzNjc5YmM3OGE2ZDMxMjY.’ in 6528 ms (Method: SUBSCRIBE)
pbxa*CLI> sip no
<— SIP read from xxx.230.32.69:5060 —>
SIP/2.0 4xxx Hacked response generated by Kamailio to avoid the softswitch’s bug (just one Via)
Via: SIP/2.0/UDP xxx.111.78.170:5060;branch=z9hG4bK31f6bd6b;rport=1024;received=xxx.111.78.170
From: “Ext.1621” sip:1621@xxx.111.78.170;tag=as6d640294
To: sip:090250189@xxx.230.32.69;tag=057dd06935a5249a03503725a8fbfd3b-b018
Call-ID: 2395dcxxx2294ff2a3280d010534fcf04@xxx.111.78.170
CSeq: 102 INVITE
Server: Kamailio SIP Proxy (CallerID Loco)
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Transmitting (no NAT) to xxx.230.32.69:5060:
ACK sip:090250189@xxx.230.32.69 SIP/2.0
Via: SIP/2.0/UDP xxx.111.78.170:5060;branch=z9hG4bK31f6bd6b;rport
From: “Ext.1621” sip:1621@xxx.111.78.170;tag=as6d640294
To: sip:090250189@xxx.230.32.69;tag=057dd06935a5249a03503725a8fbfd3b-b018
Contact: sip:1621@xxx.111.78.170
Call-ID: 2395dcxxx2294ff2a3280d010534fcf04@xxx.111.78.170
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
Really destroying SIP dialog ‘243ac1ac77cfa5a4770fcdfd4912fbef@129.168.1.200’ Method: OPTIONS
Really destroying SIP dialog ‘2395dcxxx2294ff2a3280d010534fcf04@xxx.111.78.170’ Method: INVITE
pbxa*CLI> sip no
<— SIP read from 129.168.1.104:52980 —>
SUBSCRIBE sip:1713@129.168.1.200 SIP/2.0
Via: SIP/2.0/UDP 129.168.1.104:52980;branch=z9hG4bK-dxxx54z-d21af03c9321eb63-1—dxxx54z-;rport
Max-Forwards: 70
Contact: sip:1713@129.168.1.104:52980
To: "1713"sip:1713@129.168.1.200
From: “1713"sip:1713@129.168.1.200;tag=716a5b17
Call-ID: NDVhZjJjZWUxYzhmNzdiY2MzNjc5YmM3OGE2ZDMxMjY.
CSeq: 2 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 1100z stamp 47739
Authorization: Digest username=“1713”,realm=“pbxa”,nonce=“16d8597b”,uri="sip:1713@129.168.1.200”,response=“f7e237b925e6129a1019474470478cc8”,algorithm=MD5
Event: message-summary
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Creating new subscription
Sending to 129.168.1.104 : 52980 (NAT)
Found peer ‘1713’ no
Looking for 1713 in pbx-a (domain 129.168.1.200)
pbxa*CLI> sip no
<— Transmitting (no NAT) to 129.168.1.104:52980 —>
SIP/2.0 404 Not found (no mailbox)
Via: SIP/2.0/UDP 129.168.1.104:52980;branch=z9hG4bK-dxxx54z-d21af03c9321eb63-1—dxxx54z-;received=129.168.1.104;rport=52980
From: "1713"sip:1713@129.168.1.200;tag=716a5b17
To: "1713"sip:1713@129.168.1.200;tag=as04178e06
Call-ID: NDVhZjJjZWUxYzhmNzdiY2MzNjc5YmM3OGE2ZDMxMjY.
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
[Oct 13 17:45:07] NOTICE[18371]: chan_sip.c:15094 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1713
Really destroying SIP dialog ‘NDVhZjJjZWUxYzhmNzdiY2MzNjc5YmM3OGE2ZDMxMjY.’ Method: SUBSCRIBE
Reliably Transmitting (no NAT) to 129.168.1.129:1369:
OPTIONS sip:2001@129.168.1.129:1369 SIP/2.0
Via: SIP/2.0/UDP 129.168.1.200:5060;branch=z9hG4bK5aefb3b7;rport
From: “asterisk” sip:asterisk@129.168.1.200;tag=as719e48b1
To: sip:2001@129.168.1.129:1369
Contact: sip:asterisk@129.168.1.200
Call-ID: 2ee01f646cdxxx00e083f0848492ce765@129.168.1.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 13 Oct 2010 15:45:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
pbxa*CLI> sip no
<— SIP read from 129.168.1.129:1369 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 129.168.1.200:5060;rport=5060;received=129.168.1.200;branch=z9hG4bK5aefb3b7
Call-ID: 2ee01f646cdxxx00e083f0848492ce765@129.168.1.200
From: “asterisk” sip:asterisk@129.168.1.200;tag=as719e48b1
To: sip:2001@129.168.1.129
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, norefersub
Allow-Events: presence, refer
User-Agent: TatinSIP v1.1/win32
Content-Type: application/sdp
Content-Length: 451
v=0
o=- 3495980707 3495980707 IN IP4 129.168.1.129
s=pjmedia
c=IN IP4 129.168.1.129
t=0 0
m=audio 4000 RTP/AVP 103 102 104 113 3 0 8 9 101
a=rtcp:4001 IN IP4 129.168.1.129
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:113 iLBC/8000
a=fmtp:113 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (13 headers 19 lines) —
Really destroying SIP dialog ‘2ee01f646cdxxx00e083f0848492ce765@129.168.1.200’ Method: OPTIONS
pbxa*CLI> sip no de
<— SIP read from 129.168.1.129:1369 —>
<------------->
pbxa*CLI> sip no debu
<— SIP read from 129.168.1.104:52980 —>
<------------->
Reliably Transmitting (no NAT) to 129.168.1.130:10084:
OPTIONS sip:2003@129.168.1.130:10084;rinstance=ab53a51b3f88edd1 SIP/2.0
Via: SIP/2.0/UDP 129.168.1.200:5060;branch=z9hG4bK7f17d4dd;rport
From: “asterisk” sip:asterisk@129.168.1.200;tag=as5c5b454d
To: sip:2003@129.168.1.130:10084;rinstance=ab53a51b3f88edd1
Contact: sip:asterisk@129.168.1.200
Call-ID: 4f7affc80d4ae621190ff1207722c561@129.168.1.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 13 Oct 2010 15:45:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
pbxa*CLI> sip no debug
<— SIP read from 129.168.1.111:1591 —>
<------------->
pbxa*CLI> sip no debug
SIP Debugging Disabled
– Executing [YYYYYYYY@pbx-a:1] NoOp(“SIP/1622-cc002f80”, " >>>> Llamada saliente XtraTelecom <<<< ") in new stack
– Executing [YYYYYYYY@pbx-a:2] Set(“SIP/1622-cc002f80”, “DIR=/var/spool/asterisk/monitor/20101013”) in new stack
– Executing [YYYYYYYY@pbx-a:3] System(“SIP/1622-cc002f80”, “/bin/mkdir -p /var/spool/asterisk/monitor/20101013”) in new stack
– Executing [YYYYYYYY@pbx-a:4] Set(“SIP/1622-cc002f80”, “FILENAME=Salientes-174511-1286984711.25522-684485009-1622”) in new stack
– Executing [YYYYYYYY@pbx-a:5] Set(“SIP/1622-cc002f80”, “MONITOR_FILENAME=/var/spool/asterisk/monitor/20101013/Salientes-174511-1286984711.25522-684485009-1622”) in new stack
– Executing [YYYYYYYY@pbx-a:6] MixMonitor(“SIP/1622-cc002f80”, “/var/spool/asterisk/monitor/20101013/Salientes-174511-1286984711.25522-684485009-1622.wav”) in new stack
– Executing [YYYYYYYY@pbx-a:7] Dial(“SIP/1622-cc002f80”, “SIP/proveedor-xtra/684485009”) in new stack
– Called proveedor-xtra/684485009
== Begin MixMonitor Recording SIP/1622-cc002f80
– SIP/proveedor-xtra-1a836a20 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [YYYYYYYY@pbx-a:8] Hangup(“SIP/1622-cc002f80”, “”) in new stack
== Spawn extension (pbx-a, YYYYYYYY, 8) exited non-zero on ‘SIP/1622-cc002f80’
== End MixMonitor Recording SIP/1622-cc002f80
– Executing [YYYYYYYY@pbx-a:1] NoOp(“SIP/1622-cc044390”, " >>>> Llamada saliente XtraTelecom <<<< ") in new stack
– Executing [YYYYYYYY@pbx-a:2] Set(“SIP/1622-cc044390”, “DIR=/var/spool/asterisk/monitor/20101013”) in new stack
– Executing [YYYYYYYY@pbx-a:3] System(“SIP/1622-cc044390”, “/bin/mkdir -p /var/spool/asterisk/monitor/20101013”) in new stack
– Executing [YYYYYYYY@pbx-a:4] Set(“SIP/1622-cc044390”, “FILENAME=Salientes-174519-1286984719.25524-684485009-1622”) in new stack
– Executing [YYYYYYYY@pbx-a:5] Set(“SIP/1622-cc044390”, “MONITOR_FILENAME=/var/spool/asterisk/monitor/20101013/Salientes-174519-1286984719.25524-684485009-1622”) in new stack
– Executing [YYYYYYYY@pbx-a:6] MixMonitor(“SIP/1622-cc044390”, “/var/spool/asterisk/monitor/20101013/Salientes-174519-1286984719.25524-684485009-1622.wav”) in new stack
– Executing [YYYYYYYY@pbx-a:7] Dial(“SIP/1622-cc044390”, “SIP/proveedor-xtra/684485009”) in new stack
– Called proveedor-xtra/684485009
== Begin MixMonitor Recording SIP/1622-cc044390
– SIP/proveedor-xtra-cc09fe90 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [YYYYYYYY@pbx-a:8] Hangup(“SIP/1622-cc044390”, “”) in new stack
== Spawn extension (pbx-a, YYYYYYYY, 8) exited non-zero on ‘SIP/1622-cc044390’
== End MixMonitor Recording SIP/1622-cc044390
– Executing [9690250180@pbx-a:1] NoOp(“SIP/1621-cc09fe90”, " >>>> Llamada saliente XtraTelecom <<<< ") in new stack
– Executing [9690250180@pbx-a:2] Set(“SIP/1621-cc09fe90”, “DIR=/var/spool/asterisk/monitor/20101013”) in new stack
– Executing [9690250180@pbx-a:3] System(“SIP/1621-cc09fe90”, “/bin/mkdir -p /var/spool/asterisk/monitor/20101013”) in new stack
– Executing [9690250180@pbx-a:4] Set(“SIP/1621-cc09fe90”, “FILENAME=Salientes-174522-1286984722.25526-690250180-1621”) in new stack
– Executing [9690250180@pbx-a:5] Set(“SIP/1621-cc09fe90”, “MONITOR_FILENAME=/var/spool/asterisk/monitor/20101013/Salientes-174522-1286984722.25526-690250180-1621”) in new stack
– Executing [9690250180@pbx-a:6] MixMonitor(“SIP/1621-cc09fe90”, “/var/spool/asterisk/monitor/20101013/Salientes-174522-1286984722.25526-690250180-1621.wav”) in new stack
– Executing [9690250180@pbx-a:7] Dial(“SIP/1621-cc09fe90”, “SIP/proveedor-xtra/690250180”) in new stack
– Called proveedor-xtra/690250180
== Begin MixMonitor Recording SIP/1621-cc09fe90
– SIP/proveedor-xtra-1a869770 is making progress passing it to SIP/1621-cc09fe90
– SIP/proveedor-xtra-1a869770 is making progress passing it to SIP/1621-cc09fe90
== Spawn extension (pbx-a, 9690250180, 7) exited non-zero on ‘SIP/1621-cc09fe90’
== End MixMonitor Recording SIP/1621-cc09fe90
Reliably Transmitting (no NAT) to xxx.230.32.69:5060:
OPTIONS sip:xxx.230.32.69 SIP/2.0
Via: SIP/2.0/UDP xxx.111.78.170:5060;branch=z9hG4bK11db0c89;rport
From: “asterisk” sip:asterisk@xxx.111.78.170;tag=as1bdd0124
To: sip:xxx.230.32.69
Contact: sip:asterisk@xxx.111.78.170
Call-ID: 6e04c2a25b34cc8b1e9c62865c1412d6@xxx.111.78.170
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 13 Oct 2010 15:59:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
pbxa*CLI> sip
<— SIP read from xxx.230.32.69:5060 —>
SIP/2.0 200 PONG
Via: SIP/2.0/UDP xxx.111.78.170:5060;branch=z9hG4bK11db0c89;rport=1024;received=xxx.111.78.170
From: “asterisk” sip:asterisk@xxx.111.78.170;tag=as1bdd0124
To: sip:xxx.230.32.69;tag=d6d56cd7e4c8012831499a1ee11682b3.36cc
Call-ID: 6e04c2a25b34cc8b1e9c62865c1412d6@xxx.111.78.170
CSeq: 102 OPTIONS
Server: Kamailio SIP Proxy (CallerID Loco)
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘6e04c2a25b34cc8b1e9c62865c1412d6@xxx.111.78.170’ Method: OPTIONS
Reliably Transmitting (NAT) to 213.179.96.22:5060:
OPTIONS sip:213.179.96.22 SIP/2.0
Via: SIP/2.0/UDP xxx.111.78.170:5060;branch=z9hG4bK300d9961;rport
From: “asterisk” sip:asterisk@xxx.111.78.170;tag=as0138ebec
To: sip:213.179.96.22
Contact: sip:asterisk@xxx.111.78.170
Call-ID: 73540d0f5f8d7f0742ee8c4b27dbf1d6@xxx.111.78.170
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 13 Oct 2010 15:59:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
pbxa*CLI> sip
<— SIP read from 213.179.96.22:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.111.78.170:5060;branch=z9hG4bK300d9961;rport
From: “asterisk” sip:asterisk@xxx.111.78.170;tag=as0138ebec
To: sip:213.179.96.22;tag=F6B72140-1D3C
Date: Wed, 13 Oct 2010 16:03:33 GMT
Call-ID: 73540d0f5f8d7f0742ee8c4b27dbf1d6@xxx.111.78.170
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 170
v=0
o=CiscoSystemsSIP-GW-UserAgent 42xxx 8005 IN IP4 213.179.96.22
s=SIP Call
c=IN IP4 213.179.96.22
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15 3
c=IN IP4 213.179.96.22
<------------->
— (14 headers 7 lines) —
Really destroying SIP dialog ‘73540d0f5f8d7f0742ee8c4b27dbf1d6@xxx.111.78.170’ Method: OPTIONS
pbxa*CLI> sip
<— SIP read from 129.168.1.106:24428 —>
SUBSCRIBE sip:1724@129.168.1.200 SIP/2.0
Via: SIP/2.0/UDP 129.168.1.106:24428;branch=z9hG4bK-dxxx54z-f572652f24183145-1—dxxx54z-;rport
Max-Forwards: 70
Contact: sip:1724@129.168.1.106:24428
To: "1724"sip:1724@129.168.1.200
From: "1724"sip:1724@129.168.1.200;tag=a3326b6b
Call-ID: ZTVlYzc0NTI1MDgwNDM0OThhYjRlZGY5MmMxOTNlNWI.
CSeq: 1 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 1100z stamp 47739
Event: message-summary
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Creating new subscription
Sending to 129.168.1.106 : 24428 (NAT)
Found peer '1724’
pbxa*CLI> sip
<— Transmitting (no NAT) to 129.168.1.106:24428 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 129.168.1.106:24428;branch=z9hG4bK-dxxx54z-f572652f24183145-1—dxxx54z-;received=129.168.1.106;rport=24428
From: "1724"sip:1724@129.168.1.200;tag=a3326b6b
To: "1724"sip:1724@129.168.1.200;tag=as5684a895
Call-ID: ZTVlYzc0NTI1MDgwNDM0OThhYjRlZGY5MmMxOTNlNWI.
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“pbxa”, nonce="4b4730d7"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘ZTVlYzc0NTI1MDgwNDM0OThhYjRlZGY5MmMxOTNlNWI.’ in 6592 ms (Method: SUBSCRIBE)
pbxa*CLI> sip
<— SIP read from 129.168.1.106:24428 —>
SUBSCRIBE sip:1724@129.168.1.200 SIP/2.0
Via: SIP/2.0/UDP 129.168.1.106:24428;branch=z9hG4bK-dxxx54z-38189a065006df18-1—dxxx54z-;rport
Max-Forwards: 70
Contact: sip:1724@129.168.1.106:24428
To: "1724"sip:1724@129.168.1.200
From: “1724"sip:1724@129.168.1.200;tag=a3326b6b
Call-ID: ZTVlYzc0NTI1MDgwNDM0OThhYjRlZGY5MmMxOTNlNWI.
CSeq: 2 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 1100z stamp 47739
Authorization: Digest username=“1724”,realm=“pbxa”,nonce=“4b4730d7”,uri="sip:1724@129.168.1.200”,response=“8d3de5386f18bd68bce2a838c9f73154”,algorithm=MD5
Event: message-summary
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Creating new subscription
Sending to 129.168.1.106 : 24428 (NAT)
Found peer '1724’
Looking for 1724 in pbx-a (domain 129.168.1.200)
pbxa*CLI> sip
<— Transmitting (no NAT) to 129.168.1.106:24428 —>
SIP/2.0 404 Not found (no mailbox)
Via: SIP/2.0/UDP 129.168.1.106:24428;branch=z9hG4bK-dxxx54z-38189a065006df18-1—dxxx54z-;received=129.168.1.106;rport=24428
From: "1724"sip:1724@129.168.1.200;tag=a3326b6b
To: "1724"sip:1724@129.168.1.200;tag=as5684a895
Call-ID: ZTVlYzc0NTI1MDgwNDM0OThhYjRlZGY5MmMxOTNlNWI.
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
[Oct 13 17:59:50] NOTICE[18371]: chan_sip.c:15094 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1724
Really destroying SIP dialog ‘ZTVlYzc0NTI1MDgwNDM0OThhYjRlZGY5MmMxOTNlNWI.’ Method: SUBSCRIBE
Reliably Transmitting (no NAT) to 129.168.1.128:1146:
OPTIONS sip:1600@129.168.1.128:1146 SIP/2.0
Via: SIP/2.0/UDP 129.168.1.200:5060;branch=z9hG4bK0d34c592;rport
From: “asterisk” sip:asterisk@129.168.1.200;tag=as6121b882
To: sip:1600@129.168.1.128:1146
Contact: sip:asterisk@129.168.1.200
Call-ID: 60c4813439b06b2258e58a311c03481b@129.168.1.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 13 Oct 2010 15:59:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
Retransmitting #1 (no NAT) to 129.168.1.128:1146:
OPTIONS sip:1600@129.168.1.128:1146 SIP/2.0
Via: SIP/2.0/UDP 129.168.1.200:5060;branch=z9hG4bK0d34c592;rport
From: “asterisk” sip:asterisk@129.168.1.200;tag=as6121b882
To: sip:1600@129.168.1.128:1146
Contact: sip:asterisk@129.168.1.200
Call-ID: 60c4813439b06b2258e58a311c03481b@129.168.1.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 13 Oct 2010 15:59:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------->
— (13 headers 19 lines) —
Really destroying SIP dialog ‘09dfde452525a1f55da4f898168b315e@129.168.1.200’ Method: OPTIONS
Retransmitting #4 (no NAT) to 89.140.7.221:5060:
OPTIONS sip:89.140.7.221 SIP/2.0
Via: SIP/2.0/UDP xxx.111.78.170:5060;branch=z9hG4bK0b13e80f;rport
From: “asterisk” sip:asterisk@xxx.111.78.170;tag=as46602673
To: sip:89.140.7.221
Contact: sip:asterisk@xxx.111.78.170
Call-ID: 746585155618a7c63677558d3c0c70e5@xxx.111.78.170
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 13 Oct 2010 16:05:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
Really destroying SIP dialog ‘746585155618a7c63677558d3c0c70e5@xxx.111.78.170’ Method: OPTIONS
Reliably Transmitting (no NAT) to 129.168.1.129:1369:
OPTIONS sip:2001@129.168.1.129:1369 SIP/2.0
Via: SIP/2.0/UDP 129.168.1.200:5060;branch=z9hG4bK2de92aa1;rport
From: “asterisk” sip:asterisk@129.168.1.200;tag=as123452a0
To: sip:2001@129.168.1.129:1369
Contact: sip:asterisk@129.168.1.200
Call-ID: 0a396b8e7c28560a5c4c8292526c9f1f@129.168.1.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 13 Oct 2010 16:05:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
pbxa*CLI> sip debu
<— SIP read from 129.168.1.129:1369 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 129.168.1.200:5060;rport=5060;received=129.168.1.200;branch=z9hG4bK2de92aa1
Call-ID: 0a396b8e7c28560a5c4c8292526c9f1f@129.168.1.200
From: “asterisk” sip:asterisk@129.168.1.200;tag=as123452a0
To: sip:2001@129.168.1.129
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, norefersub
Allow-Events: presence, refer
User-Agent: TatinSIP v1.1/win32
Content-Type: application/sdp
Content-Length: 451
v=0
o=- 3495981907 3495981907 IN IP4 129.168.1.129
s=pjmedia
c=IN IP4 129.168.1.129
t=0 0
m=audio 4000 RTP/AVP 103 102 104 113 3 0 8 9 101
a=rtcp:4001 IN IP4 129.168.1.129
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:113 iLBC/8000
a=fmtp:113 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (13 headers 19 lines) —
Really destroying SIP dialog ‘0a396b8e7c28560a5c4c8292526c9f1f@129.168.1.200’ Method: OPTIONS
Reliably Transmitting (no NAT) to 129.168.1.130:10084:
OPTIONS sip:2003@129.168.1.130:10084;rinstance=ab53a51b3f88edd1 SIP/2.0
Via: SIP/2.0/UDP 129.168.1.200:5060;branch=z9hG4bK62bd273e;rport
From: “asterisk” sip:asterisk@129.168.1.200;tag=as21d34c70
To: sip:2003@129.168.1.130:10084;rinstance=ab53a51b3f88edd1
Contact: sip:asterisk@129.168.1.200
Call-ID: 7baedb9b588441e03ddd3e525a655054@129.168.1.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 13 Oct 2010 16:05:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
pbxa*CLI> sip d
<— SIP read from 129.168.1.129:1369 —>
<------------->
pbxa*CLI> sip d
<— SIP read from 129.168.1.130:10084 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 129.168.1.200:5060;branch=z9hG4bK62bd273e;rport=5060
Contact: sip:129.168.1.130:10084
To: sip:2003@129.168.1.130:10084;rinstance=ab53a51b3f88edd1;tag=6f246f37
From: "asterisk"sip:asterisk@129.168.1.200;tag=as21d34c70
Call-ID: 7baedb9b588441e03ddd3e525a655054@129.168.1.200
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 1100z stamp 47739
Content-Length: 0
<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘7baedb9b588441e03ddd3e525a655054@129.168.1.200’ Method: OPTIONS
pbxa*CLI> sip
<— SIP read from 129.168.1.131:1128 —>
<------------->
pbxa*CLI> sip n
<— SIP read from 129.168.1.136:1142 —>
<------------->
pbxa*CLI> sip no
<— SIP read from xxx.230.32.69:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.111.78.170:5060;received=xxx.111.78.170;branch=z9hG4bK09297ba1;rport=1024
Record-Route: sip:xxx.230.32.69;lr;request-type=sip;did=2e8.1488ea83;rtpproxy=yes
Contact: sip:606791717@xxx.230.5.228:5060
To: sip:606791717@xxx.230.32.69;tag=5823a15-co1809-INS001
From: "911331825"sip:911331825@callerid-loco.xtratelecom.es;tag=as20918ebb
Call-ID: 3f68549b2b91172103a31c9946fc25d5@xxx.111.78.170
CSeq: 102 INVITE
Content-Type: application/sdp
User-Agent: ENSR2.5.48.12-IS1-RMRG482-RG4070-CPO55xxx
Content-Length: 237
v=0
o=- 92420629 92420629 IN IP4 xxx.230.5.228
s=ENSResip
c=IN IP4 xxx.230.32.79
t=0 0
m=audio 38688 RTP/AVP 0 101
a=fmtp:101 0-16
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=nortpproxy:yes
<------------->
— (11 headers 12 lines) —
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port xxx.230.32.79:38688
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port xxx.230.32.79:38688
list_route: hop: sip:xxx.230.32.69;lr;request-type=sip;did=2e8.1488ea83;rtpproxy=yes
set_destination: Parsing sip:xxx.230.32.69;lr;request-type=sip;did=2e8.1488ea83;rtpproxy=yes for address/port to send to
set_destination: set destination to xxx.230.32.69, port 5060
Transmitting (no NAT) to xxx.230.32.69:5060:
ACK sip:606791717@xxx.230.5.228:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.111.78.170:5060;branch=z9hG4bK7c073e10;rport
Route: sip:xxx.230.32.69;lr;request-type=sip;did=2e8.1488ea83;rtpproxy=yes
From: “Ext.1622” sip:1622@xxx.111.78.170;tag=as20918ebb
To: sip:606791717@xxx.230.32.69;tag=5823a15-co1809-INS001
Contact: sip:1622@xxx.111.78.170
Call-ID: 3f68549b2b91172103a31c9946fc25d5@xxx.111.78.170
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
-- SIP/proveedor-xtra-1a836a20 answered SIP/1622-c41adca0
Audio is at 129.168.1.200 port 16254
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
pbxa*CLI> sip no
<— Reliably Transmitting (no NAT) to 129.168.1.105:3153 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 129.168.1.105:3153;branch=z9hG4bKPjaf485d7d1a204954acd1ecxxxca05495d;received=129.168.1.105;rport=3153
From: sip:1622@129.168.1.200;tag=0d3be1c7bbca42aa923a882a90e7fa3e
To: sip:9606791717@129.168.1.200;tag=as2fdbd3b3
Call-ID: c55663fe6ec44edca7cxxx4f842e95154
CSeq: 13734 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:9606791717@129.168.1.200
Content-Type: application/sdp
Content-Length: 2xxx
v=0
o=root 3008 3009 IN IP4 129.168.1.200
s=session
c=IN IP4 129.168.1.200
t=0 0
m=audio 16254 RTP/AVP 3 0 8 101
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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
pbxa*CLI> sip no
<— SIP read from 129.168.1.105:3153 —>
ACK sip:9606791717@129.168.1.200 SIP/2.0
Via: SIP/2.0/UDP 129.168.1.105:3153;rport;branch=z9hG4bKPj4a3b9e8a46284f76a423e6fd2ae037f7
Max-Forwards: 70
From: sip:1622@129.168.1.200;tag=0d3be1c7bbca42aa923a882a90e7fa3e
To: sip:9606791717@129.168.1.200;tag=as2fdbd3b3
Call-ID: c55663fe6ec44edca7cxxx4f842e95154
CSeq: 13734 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
pbxa*CLI> sip no debu
<— SIP read from 129.168.1.104:52980 —>