7 Mar
2013
7 Mar
'13
11:04 a.m.
apparently the cisco device is a pre-3261 implementation, i.e.
it behaves correctly in the historical RFC2543 terms.
normal SER configuration should actually handle it in code
like this...
if (loose_route()) {
route(RECORD_ROUTE);
route(FORWARD); // t_relay
jiri
this is a message digest:
INVITE sip:911111500@sip_server:5060
Contact: <sip:911873699@cisco_gw:5060>
200
Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.
Contact: <sip:911111500@asterisk_server>.
ACK sip:sip_server:5060;lr=on;did=015.864b8107
Route: <sip:911111500@asterisk_server:5060>.
On 1/17/11 5:33 PM, Nawfel Oujdi wrote:
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@cisco_gw"
From: <sip:911873699@cisco_gw>;tag=65FB8-B18
Route: <sip:911111500@PBX:5060>
To: <sip:911111500@sip_proxy>;tag=as7f388e3f
Date: Mon, 17 Jan 2011 09:26:36 GMT
Call-ID: B6F61A2E-215211E0-802BD462-C4432B89@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@sip_proxy:5060 SIP/2.0
Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw"
From: <sip:911873699@cisco_gw>;tag=65FB8-B18
To: <sip:911111500@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@sip_server:5060 SIP/2.0.
Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw".
From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.
To: <sip:911111500@sip_server>.
Date: Thu, 13 Jan 2011 14:14:43 GMT.
Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@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@cisco_gw>;party=calling;screen=yes;privacy=off.
Timestamp: 1294928083.
Contact: <sip:911873699@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@cisco_gw".
From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.
To: <sip:911111500@sip_server>.
Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@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@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@cisco_gw".
From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.
To: <sip:911111500@sip_server>.
Date: Thu, 13 Jan 2011 14:14:43 GMT.
Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@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@cisco_gw>;party=calling;screen=yes;privacy=off.
Timestamp: 1294928083.
Contact: <sip:911873699@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@cisco_gw".
Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.
From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.
To: <sip:911111500@sip_server>.
Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@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@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@cisco_gw".
Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.
From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.
To: <sip:911111500@sip_server>;tag=as19e8a82f.
Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@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@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@cisco_gw".
Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.
From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.
To: <sip:911111500@sip_server>;tag=as19e8a82f.
Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@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@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@cisco_gw".
From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.
To: <sip:911111500@sip_server>;tag=as19e8a82f.
Date: Thu, 13 Jan 2011 14:14:43 GMT.
Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw.
Route: <sip:911111500@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@cisco_gw".
Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.
From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.
To: <sip:911111500@sip_server>;tag=as19e8a82f.
Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@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@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@cisco_gw".
Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.
From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.
To: <sip:911111500@sip_server>;tag=as19e8a82f.
Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@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@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@cisco_gw".
From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.
To: <sip:911111500@sip_server>;tag=as19e8a82f.
Date: Thu, 13 Jan 2011 14:14:43 GMT.
Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw.
Route: <sip:911111500@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@cisco_gw".
Record-Route: <sip:sip_server;lr=on;did=015.864b8107>.
From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC.
To: <sip:911111500@sip_server>;tag=as19e8a82f.
Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@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@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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