Hello!<br><br><br> I m facing the same strange behaviour with my AS5300 voice gateway. When the gw is connected directly to PBX everythings works well but when i put a sip proxy forwarding calls between gw and PBX all the calls hangs up after 5 sec (+or -). Looking into the trace sip i realize that gw send a wrong ACK in reply of INVITE , then sip proxy discard it and PBX hangs the call cause he never receive the ACK.<br>
<br><br><br>ACK sip:79.125.120.12:5060;lr=on;did=ce.3716ea02 SIP/2.0<br> Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw"<br> From: <sip:911873699@cisco_gw>;tag=65FB8-B18<br><br> Route: <sip:911111500@PBX:5060><br>
<br> To: <sip:911111500@sip_proxy>;tag=as7f388e3f<br><br> Date: Mon, 17 Jan 2011 09:26:36 GMT<br> Call-ID: B6F61A2E-215211E0-802BD462-C4432B89@cisco_gw<br><br><br>To work fine , the content of Route header should be in ACK header and viceversa.<br>
<br><br><br> I tried to compare between the sip trace of a wrong call and a good one (using other cisco gw AS5350 who works well with sip proxy in the same escenario) and i realize that the only difference is the INVITE of wrong case doesn' t send branch number in the via header.<br>
<br><br><br>INVITE sip:911111500@sip_proxy:5060 SIP/2.0<br> Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw"<br> From: <sip:911873699@cisco_gw>;tag=65FB8-B18<br> To: <sip:911111500@sip_proxy><br>
<br><br>i m using c5300-is-mz.123-26.bin ios version.<br><br><br>Anybody understand what is happening in there?? is there any solution?? i ll send more information if it s requested.<br><br>Thanks in advance.<br><br>Nawfel Oujdi<br>
<br><br><br>here is the result of ngrep:<br>U 2011/01/13 15:14:43.791514 cisco_gw:51703 -> sip_server:5060<br>INVITE sip:911111500@sip_server:5060 SIP/2.0.<br>Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw".<br>
From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.<br>To: <sip:911111500@sip_server>.<br>Date: Thu, 13 Jan 2011 14:14:43 GMT.<br>Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw.<br>Supported: timer,100rel.<br>
Min-SE: 1800.<br>Cisco-Guid: 1295951687-508957152-2608788105-28919687.<br>User-Agent: Cisco-SIPGateway/IOS-12.x.<br>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO.<br>CSeq: 101 INVITE.<br>
Max-Forwards: 6.<br>Remote-Party-ID: <sip:911873699@cisco_gw>;party=calling;screen=yes;privacy=off.<br>Timestamp: 1294928083.<br>Contact: <sip:911873699@cisco_gw:5060>.<br>Expires: 180.<br>Allow-Events: telephone-event.<br>
Content-Type: application/sdp.<br>Content-Length: 270.<br>.<br>v=0.<br>o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw.<br>s=SIP Call.<br>c=IN IP4 cisco_gw.<br>t=0 0.<br>m=audio 16924 RTP/AVP 18 101.<br>c=IN IP4 cisco_gw.<br>
a=rtpmap:18 G729/8000.<br>a=fmtp:18 annexb=no.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br><br><br>U 2011/01/13 15:14:43.791893 sip_server:5060 -> cisco_gw:5060<br>SIP/2.0 100 Giving a try.<br>
Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw".<br>From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.<br>To: <sip:911111500@sip_server>.<br>Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw.<br>
CSeq: 101 INVITE.<br>Server: OpenSIPS (1.6.3-notls (i386/linux)).<br>Content-Length: 0.<br>.<br><br><br>U 2011/01/13 15:14:43.791957 sip_server:5060 -> asterisk_server:5060<br>INVITE sip:911111500@sip_server:5060 SIP/2.0.<br>
Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.<br>Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0.<br>Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw".<br>From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.<br>
To: <sip:911111500@sip_server>.<br>Date: Thu, 13 Jan 2011 14:14:43 GMT.<br>Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw.<br>Supported: timer,100rel.<br>Min-SE: 1800.<br>Cisco-Guid: 1295951687-508957152-2608788105-28919687.<br>
User-Agent: Cisco-SIPGateway/IOS-12.x.<br>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO.<br>CSeq: 101 INVITE.<br>Max-Forwards: 5.<br>Remote-Party-ID: <sip:911873699@cisco_gw>;party=calling;screen=yes;privacy=off.<br>
Timestamp: 1294928083.<br>Contact: <sip:911873699@cisco_gw:5060>.<br>Expires: 180.<br>Allow-Events: telephone-event.<br>Content-Type: application/sdp.<br>Content-Length: 270.<br>.<br>v=0.<br>o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw.<br>
s=SIP Call.<br>c=IN IP4 cisco_gw.<br>t=0 0.<br>m=audio 16924 RTP/AVP 18 101.<br>c=IN IP4 cisco_gw.<br>a=rtpmap:18 G729/8000.<br>a=fmtp:18 annexb=no.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br>
<br><br>U 2011/01/13 15:14:43.792775 asterisk_server:5060 -> sip_server:5060<br>SIP/2.0 100 Trying.<br>Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server.<br>Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw".<br>
Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.<br>From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.<br>To: <sip:911111500@sip_server>.<br>Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw.<br>CSeq: 101 INVITE.<br>
Server: Asterisk PBX 1.6.2.13.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.<br>Supported: replaces, timer.<br>Require: timer.<br>Session-Expires: 1800;refresher=uas.<br>Contact: <sip:911111500@asterisk_server>.<br>
Content-Length: 0.<br>.<br><br><br>U 2011/01/13 15:14:43.793770 asterisk_server:5060 -> sip_server:5060<br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server.<br>Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw".<br>
Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.<br>From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.<br>To: <sip:911111500@sip_server>;tag=as19e8a82f.<br>Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw.<br>
CSeq: 101 INVITE.<br>Server: Asterisk PBX 1.6.2.13.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.<br>Supported: replaces, timer.<br>Require: timer.<br>Session-Expires: 1800;refresher=uas.<br>
Contact: <sip:911111500@asterisk_server>.<br>Content-Type: application/sdp.<br>Content-Length: 260.<br>.<br>v=0.<br>o=root 1750021131 1750021131 IN IP4 asterisk_server.<br>s=Asterisk PBX 1.6.2.13.<br>c=IN IP4 asterisk_server.<br>
t=0 0.<br>m=audio 10798 RTP/AVP 18 101.<br>a=rtpmap:18 G729/8000.<br>a=fmtp:18 annexb=no.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br>a=sendrecv.<br><br><br>U 2011/01/13 15:14:43.794688 sip_server:5060 -> cisco_gw:5060<br>
SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw".<br>Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.<br>From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.<br>To: <sip:911111500@sip_server>;tag=as19e8a82f.<br>
Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw.<br>CSeq: 101 INVITE.<br>Server: Asterisk PBX 1.6.2.13.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.<br>Supported: replaces, timer.<br>Require: timer.<br>
Session-Expires: 1800;refresher=uas.<br>Contact: <sip:911111500@asterisk_server>.<br>Content-Type: application/sdp.<br>Content-Length: 316.<br>.<br>v=0.<br>o=root 1750021131 1750021131 IN IP4 79.125.41.121.<br>s=Asterisk PBX 1.6.2.13.<br>
c=IN IP4 79.125.41.121.<br>t=0 0.<br>m=audio 10798 RTP/AVP 18 101.<br>a=rtpmap:18 G729/8000.<br>a=fmtp:18 annexb=no.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br>a=sendrecv.<br>a=oldmediaip:asterisk_server.<br>
a=oldmediaip:asterisk_server.<br><br><br>U 2011/01/13 15:14:43.856520 cisco_gw:57947 -> sip_server:5060<br>ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0.<br>Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw".<br>
From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.<br>To: <sip:911111500@sip_server>;tag=as19e8a82f.<br>Date: Thu, 13 Jan 2011 14:14:43 GMT.<br>Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw.<br>Route: <sip:911111500@asterisk_server:5060>.<br>
Max-Forwards: 6.<br>Content-Length: 0.<br>CSeq: 101 ACK.<br>.<br><br><br>U 2011/01/13 15:14:43.993417 asterisk_server:5060 -> sip_server:5060<br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server.<br>
Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw".<br>Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.<br>From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.<br>To: <sip:911111500@sip_server>;tag=as19e8a82f.<br>
Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw.<br>CSeq: 101 INVITE.<br>Server: Asterisk PBX 1.6.2.13.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.<br>Supported: replaces, timer.<br>Require: timer.<br>
Session-Expires: 1800;refresher=uas.<br>Contact: <sip:911111500@asterisk_server>.<br>Content-Type: application/sdp.<br>Content-Length: 260.<br>.<br>v=0.<br>o=root 1750021131 1750021131 IN IP4 asterisk_server.<br>s=Asterisk PBX 1.6.2.13.<br>
c=IN IP4 asterisk_server.<br>t=0 0.<br>m=audio 10798 RTP/AVP 18 101.<br>a=rtpmap:18 G729/8000.<br>a=fmtp:18 annexb=no.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br>a=sendrecv.<br><br><br>U 2011/01/13 15:14:43.993613 sip_server:5060 -> cisco_gw:5060<br>
SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw".<br>Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.<br>From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.<br>To: <sip:911111500@sip_server>;tag=as19e8a82f.<br>
Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw.<br>CSeq: 101 INVITE.<br>Server: Asterisk PBX 1.6.2.13.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.<br>Supported: replaces, timer.<br>Require: timer.<br>
Session-Expires: 1800;refresher=uas.<br>Contact: <sip:911111500@asterisk_server>.<br>Content-Type: application/sdp.<br>Content-Length: 316.<br>.<br>v=0.<br>o=root 1750021131 1750021131 IN IP4 79.125.41.121.<br>s=Asterisk PBX 1.6.2.13.<br>
c=IN IP4 79.125.41.121.<br>t=0 0.<br>m=audio 10798 RTP/AVP 18 101.<br>a=rtpmap:18 G729/8000.<br>a=fmtp:18 annexb=no.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br>a=sendrecv.<br>a=oldmediaip:asterisk_server.<br>
a=oldmediaip:asterisk_server.<br><br><br>U 2011/01/13 15:14:44.038774 cisco_gw:57947 -> sip_server:5060<br>ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0.<br>Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw".<br>
From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.<br>To: <sip:911111500@sip_server>;tag=as19e8a82f.<br>Date: Thu, 13 Jan 2011 14:14:43 GMT.<br>Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw.<br>Route: <sip:911111500@asterisk_server:5060>.<br>
Max-Forwards: 6.<br>Content-Length: 0.<br>CSeq: 101 ACK.<br>.<br><br><br>U 2011/01/13 15:14:44.193431 asterisk_server:5060 -> sip_server:5060<br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server.<br>
Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw".<br>Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.<br>From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.<br>To: <sip:911111500@sip_server>;tag=as19e8a82f.<br>
Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw.<br>CSeq: 101 INVITE.<br>Server: Asterisk PBX 1.6.2.13.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.<br>Supported: replaces, timer.<br>Require: timer.<br>
Session-Expires: 1800;refresher=uas.<br>Contact: <sip:911111500@asterisk_server>.<br>Content-Type: application/sdp.<br>Content-Length: 260.<br>.<br>v=0.<br>o=root 1750021131 1750021131 IN IP4 asterisk_server.<br>s=Asterisk PBX 1.6.2.13.<br>
c=IN IP4 asterisk_server.<br>t=0 0.<br>m=audio 10798 RTP/AVP 18 101.<br>a=rtpmap:18 G729/8000.<br>a=fmtp:18 annexb=no.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br>a=sendrecv.<br clear="all">
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