[SR-Users] Wrong ACK packet sent from Cisco gw
Klaus Darilion
klaus.mailinglists at pernau.at
Mon Jan 17 18:17:17 CET 2011
Nawfel, the trace looks OK until the poart where the ACK hits the proxy.
I guess you have a bug in your config. Are you sure that the ACK is
routed by loose_route()?
regards
klaus
Am 17.01.2011 17:35, schrieb Nawfel Oujdi:
> Hello!
>
>
> 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.
>
>
>
> ACK sip:79.125.120.12:5060;lr=on;did=ce.3716ea02 SIP/2.0
> Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange at cisco_gw"
> From: <sip:911873699 at cisco_gw>;tag=65FB8-B18
>
> Route: <sip:911111500 at PBX:5060>
>
> To: <sip:911111500 at sip_proxy>;tag=as7f388e3f
>
> Date: Mon, 17 Jan 2011 09:26:36 GMT
> Call-ID: B6F61A2E-215211E0-802BD462-C4432B89 at cisco_gw
>
>
> To work fine , the content of Route header should be in ACK header and
> viceversa.
>
>
>
> 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.
>
>
>
> INVITE sip:911111500 at sip_proxy:5060 SIP/2.0
> Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange at cisco_gw"
> From: <sip:911873699 at cisco_gw>;tag=65FB8-B18
> To: <sip:911111500 at sip_proxy>
>
>
> i m using c5300-is-mz.123-26.bin ios version.
>
>
> Anybody understand what is happening in there?? is there any
> solution?? i ll send more information if it s requested.
>
> Thanks in advance.
>
> Nawfel Oujdi
>
>
>
> here is the result of ngrep:
> U 2011/01/13 15:14:43.791514 cisco_gw:51703 -> sip_server:5060
> INVITE sip:911111500 at sip_server:5060 SIP/2.0.
> Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange at cisco_gw".
> From: <sip:911873699 at cisco_gw>;tag=4F226C8-2DC.
> To: <sip:911111500 at sip_server>.
> Date: Thu, 13 Jan 2011 14:14:43 GMT.
> Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787 at cisco_gw.
> Supported: timer,100rel.
> Min-SE: 1800.
> Cisco-Guid: 1295951687-508957152-2608788105-28919687.
> User-Agent: Cisco-SIPGateway/IOS-12.x.
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
> SUBSCRIBE, NOTIFY, INFO.
> CSeq: 101 INVITE.
> Max-Forwards: 6.
> Remote-Party-ID:
> <sip:911873699 at cisco_gw>;party=calling;screen=yes;privacy=off.
> Timestamp: 1294928083.
> Contact: <sip:911873699 at cisco_gw:5060>.
> Expires: 180.
> Allow-Events: telephone-event.
> Content-Type: application/sdp.
> Content-Length: 270.
> .
> v=0.
> o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw.
> s=SIP Call.
> c=IN IP4 cisco_gw.
> t=0 0.
> m=audio 16924 RTP/AVP 18 101.
> c=IN IP4 cisco_gw.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
>
> U 2011/01/13 15:14:43.791893 sip_server:5060 -> cisco_gw:5060
> SIP/2.0 100 Giving a try.
> Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange at cisco_gw".
> From: <sip:911873699 at cisco_gw>;tag=4F226C8-2DC.
> To: <sip:911111500 at sip_server>.
> Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787 at cisco_gw.
> CSeq: 101 INVITE.
> Server: OpenSIPS (1.6.3-notls (i386/linux)).
> Content-Length: 0.
> .
>
>
> U 2011/01/13 15:14:43.791957 sip_server:5060 -> asterisk_server:5060
> INVITE sip:911111500 at sip_server:5060 SIP/2.0.
> Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.
> Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0.
> Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange at cisco_gw".
> From: <sip:911873699 at cisco_gw>;tag=4F226C8-2DC.
> To: <sip:911111500 at sip_server>.
> Date: Thu, 13 Jan 2011 14:14:43 GMT.
> Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787 at cisco_gw.
> Supported: timer,100rel.
> Min-SE: 1800.
> Cisco-Guid: 1295951687-508957152-2608788105-28919687.
> User-Agent: Cisco-SIPGateway/IOS-12.x.
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
> SUBSCRIBE, NOTIFY, INFO.
> CSeq: 101 INVITE.
> Max-Forwards: 5.
> Remote-Party-ID:
> <sip:911873699 at cisco_gw>;party=calling;screen=yes;privacy=off.
> Timestamp: 1294928083.
> Contact: <sip:911873699 at cisco_gw:5060>.
> Expires: 180.
> Allow-Events: telephone-event.
> Content-Type: application/sdp.
> Content-Length: 270.
> .
> v=0.
> o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw.
> s=SIP Call.
> c=IN IP4 cisco_gw.
> t=0 0.
> m=audio 16924 RTP/AVP 18 101.
> c=IN IP4 cisco_gw.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
>
> U 2011/01/13 15:14:43.792775 asterisk_server:5060 -> sip_server:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP
> sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server.
> Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange at cisco_gw".
> Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.
> From: <sip:911873699 at cisco_gw>;tag=4F226C8-2DC.
> To: <sip:911111500 at sip_server>.
> Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787 at cisco_gw.
> CSeq: 101 INVITE.
> Server: Asterisk PBX 1.6.2.13.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
> Supported: replaces, timer.
> Require: timer.
> Session-Expires: 1800;refresher=uas.
> Contact: <sip:911111500 at asterisk_server>.
> Content-Length: 0.
> .
>
>
> U 2011/01/13 15:14:43.793770 asterisk_server:5060 -> sip_server:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server.
> Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange at cisco_gw".
> Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.
> From: <sip:911873699 at cisco_gw>;tag=4F226C8-2DC.
> To: <sip:911111500 at sip_server>;tag=as19e8a82f.
> Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787 at cisco_gw.
> CSeq: 101 INVITE.
> Server: Asterisk PBX 1.6.2.13.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
> Supported: replaces, timer.
> Require: timer.
> Session-Expires: 1800;refresher=uas.
> Contact: <sip:911111500 at asterisk_server>.
> Content-Type: application/sdp.
> Content-Length: 260.
> .
> v=0.
> o=root 1750021131 1750021131 IN IP4 asterisk_server.
> s=Asterisk PBX 1.6.2.13.
> c=IN IP4 asterisk_server.
> t=0 0.
> m=audio 10798 RTP/AVP 18 101.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> a=sendrecv.
>
>
> U 2011/01/13 15:14:43.794688 sip_server:5060 -> cisco_gw:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange at cisco_gw".
> Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.
> From: <sip:911873699 at cisco_gw>;tag=4F226C8-2DC.
> To: <sip:911111500 at sip_server>;tag=as19e8a82f.
> Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787 at cisco_gw.
> CSeq: 101 INVITE.
> Server: Asterisk PBX 1.6.2.13.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
> Supported: replaces, timer.
> Require: timer.
> Session-Expires: 1800;refresher=uas.
> Contact: <sip:911111500 at asterisk_server>.
> Content-Type: application/sdp.
> Content-Length: 316.
> .
> v=0.
> o=root 1750021131 1750021131 IN IP4 79.125.41.121.
> s=Asterisk PBX 1.6.2.13.
> c=IN IP4 79.125.41.121.
> t=0 0.
> m=audio 10798 RTP/AVP 18 101.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> a=sendrecv.
> a=oldmediaip:asterisk_server.
> a=oldmediaip:asterisk_server.
>
>
> U 2011/01/13 15:14:43.856520 cisco_gw:57947 -> sip_server:5060
> ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0.
> Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange at cisco_gw".
> From: <sip:911873699 at cisco_gw>;tag=4F226C8-2DC.
> To: <sip:911111500 at sip_server>;tag=as19e8a82f.
> Date: Thu, 13 Jan 2011 14:14:43 GMT.
> Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787 at cisco_gw.
> Route: <sip:911111500 at asterisk_server:5060>.
> Max-Forwards: 6.
> Content-Length: 0.
> CSeq: 101 ACK.
> .
>
>
> U 2011/01/13 15:14:43.993417 asterisk_server:5060 -> sip_server:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server.
> Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange at cisco_gw".
> Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.
> From: <sip:911873699 at cisco_gw>;tag=4F226C8-2DC.
> To: <sip:911111500 at sip_server>;tag=as19e8a82f.
> Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787 at cisco_gw.
> CSeq: 101 INVITE.
> Server: Asterisk PBX 1.6.2.13.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
> Supported: replaces, timer.
> Require: timer.
> Session-Expires: 1800;refresher=uas.
> Contact: <sip:911111500 at asterisk_server>.
> Content-Type: application/sdp.
> Content-Length: 260.
> .
> v=0.
> o=root 1750021131 1750021131 IN IP4 asterisk_server.
> s=Asterisk PBX 1.6.2.13.
> c=IN IP4 asterisk_server.
> t=0 0.
> m=audio 10798 RTP/AVP 18 101.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> a=sendrecv.
>
>
> U 2011/01/13 15:14:43.993613 sip_server:5060 -> cisco_gw:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange at cisco_gw".
> Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.
> From: <sip:911873699 at cisco_gw>;tag=4F226C8-2DC.
> To: <sip:911111500 at sip_server>;tag=as19e8a82f.
> Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787 at cisco_gw.
> CSeq: 101 INVITE.
> Server: Asterisk PBX 1.6.2.13.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
> Supported: replaces, timer.
> Require: timer.
> Session-Expires: 1800;refresher=uas.
> Contact: <sip:911111500 at asterisk_server>.
> Content-Type: application/sdp.
> Content-Length: 316.
> .
> v=0.
> o=root 1750021131 1750021131 IN IP4 79.125.41.121.
> s=Asterisk PBX 1.6.2.13.
> c=IN IP4 79.125.41.121.
> t=0 0.
> m=audio 10798 RTP/AVP 18 101.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> a=sendrecv.
> a=oldmediaip:asterisk_server.
> a=oldmediaip:asterisk_server.
>
>
> U 2011/01/13 15:14:44.038774 cisco_gw:57947 -> sip_server:5060
> ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0.
> Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange at cisco_gw".
> From: <sip:911873699 at cisco_gw>;tag=4F226C8-2DC.
> To: <sip:911111500 at sip_server>;tag=as19e8a82f.
> Date: Thu, 13 Jan 2011 14:14:43 GMT.
> Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787 at cisco_gw.
> Route: <sip:911111500 at asterisk_server:5060>.
> Max-Forwards: 6.
> Content-Length: 0.
> CSeq: 101 ACK.
> .
>
>
> U 2011/01/13 15:14:44.193431 asterisk_server:5060 -> sip_server:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server.
> Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange at cisco_gw".
> Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.
> From: <sip:911873699 at cisco_gw>;tag=4F226C8-2DC.
> To: <sip:911111500 at sip_server>;tag=as19e8a82f.
> Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787 at cisco_gw.
> CSeq: 101 INVITE.
> Server: Asterisk PBX 1.6.2.13.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
> Supported: replaces, timer.
> Require: timer.
> Session-Expires: 1800;refresher=uas.
> Contact: <sip:911111500 at asterisk_server>.
> Content-Type: application/sdp.
> Content-Length: 260.
> .
> v=0.
> o=root 1750021131 1750021131 IN IP4 asterisk_server.
> s=Asterisk PBX 1.6.2.13.
> c=IN IP4 asterisk_server.
> t=0 0.
> m=audio 10798 RTP/AVP 18 101.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> a=sendrecv.
>
> --
>
>
>
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