Copy the list!
I don't understand your problem. The BYE will be caught by the loose route section of the config and relayed. The standard getting started should work fine. g-)
------- Original message ------- From: flavio flavio.patria@gmail.com Sent: 12.5.'07, 15:09
2007/5/12, Greger V. Teigre greger@teigre.com:
Flavio, SER doesn't reply 200 OK to BYE message, the SIP UA does that. So, you are fine, the BYE reaches the UA. g-)
That's right.
If my gateway send INVITE message to SER, it does proxy it to UAC reached! So UAC (SIP Phone) and Gateway (UAC in SIP scenario) are involved in a media session. If my gateway send BYE message to SER, it does not proxy it to UAC. So I could send reply (200 OK) to Gateway (It works fine, but is not solution!), but I cannot proxy BYE to UAC.
How can proxy this?
Thanks,
Flavio
2007/5/12, Greger Viken Teigre greger@teigre.com:
Copy the list!
Greg, all, first of all thank you for support and patience. I follow the getting started cfg example to edit my ser.cfg. I would use my gateway in order to handle calls from and to pstn. Here is VoIP Calls graph grep from Ethereal, to explain clearly my problem:
Time | 10.28.52.107 | 10.28.19.202 | 10.28.19.124 | |21,624 | INVITE SDP ( g729 g711A g711U) | |SIP From: sip:0672028405@10.28.52.107 To:sip:0660522014@10.28.19.202 | |(5060) ------------------> (5060) | | |21,624 | 100 trying -- your call is important to us | |SIP Status | |(5060) <------------------ (5060) | | |21,624 | | INVITE SDP ( g729 g711A g711U) |SIP Request | | |(5060) ------------------> (5060) | |21,631 | | 100 Trying| |SIP Status | | |(5060) <------------------ (5060) | |21,633 | | 180 Ringing |SIP Status | | |(5060) <------------------ (5060) | |21,634 | 180 Ringing | |SIP Status | |(5060) <------------------ (5060) | | |22,712 | | 200 OK SDP ( g729) |SIP Status | | |(5060) <------------------ (5060) | |22,712 | 200 OK SDP ( g729) | |SIP Status | |(5060) <------------------ (5060) | | |22,717 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |22,717 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |22,717 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |26,037 | BYE | | |SIP Request | |(5060) ------------------> (5060) | | |26,037 | 404 User Not Found | |SIP Status | |(5060) <------------------ (5060) | | |26,040 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |26,041 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |26,041 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |29,570 | | BYE | |SIP Request | | |(5060) <------------------ (5060) | |29,570 | BYE | | |SIP Request | |(5060) <------------------ (5060) | | |29,573 | 481 Transaction Does Not Exist | |SIP Status | |(5060) ------------------> (5060) | | |29,574 | | 481 Transaction Does Not Exist |SIP Status | | |(5060) ------------------> (5060) |
10.28.52.107 is IP for my gateway 10.28.19.202 is IP for SER 10.28.19.124 is IP for my SIP Phone.
As you can see when Gateway send BYE message, SER does not relay it to IP Phone, but reply with 404 Not Found SIP Message. Have you any suggestions about this?
I don't understand your problem. The BYE will be caught by the loose route section of the config and relayed. The standard getting started should work fine. g-)
------- Original message ------- From: flavio flavio.patria@gmail.com Sent: 12.5.'07, 15:09
2007/5/12, Greger V. Teigre greger@teigre.com:
Flavio, SER doesn't reply 200 OK to BYE message, the SIP UA does that. So, you are fine, the BYE reaches the UA. g-)
Hello, The call signalling exchange is a good start but it's not enough.
What some people here asked you is a capture of the SIP BYE mesg which GW sends to SER.
something like that, for example :
BYE sip:user1@1.2.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK3E01596 From: sip:1234567890@5.6.7.8;tag=12FB3150-15EC To: sip:12341234567890@mydomain.com;tag=sei-6416 Date: Tue, 15 May 2007 11:42:43 GMT Call-ID: 3A6DE4DC-21011DC-998FA089-7CE33D49@5.6.7.8 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Route: sip:3.3.3.3;ftag=12FB3150-15EC;lr=on Timestamp: 1179229379 CSeq: 102 BYE Reason: Q.850;cause=16 Content-Length: 0
1.2.3.4 : SIP Phone 5.6.7.8 : GW 3.3.3.3 : SER
1234567890 : PSTN number of the caller (@ PSTN side) 12341234567890 : PSTN number of the callee (@ IP side)
Try to find a way (ngrep/tcpdump) to capture this mesg or all mesgs for this session .
Kostas
flavio wrote:
2007/5/12, Greger Viken Teigre greger@teigre.com:
Copy the list!
Greg, all, first of all thank you for support and patience. I follow the getting started cfg example to edit my ser.cfg. I would use my gateway in order to handle calls from and to pstn. Here is VoIP Calls graph grep from Ethereal, to explain clearly my problem:
Time | 10.28.52.107 | 10.28.19.202 | 10.28.19.124 | |21,624 | INVITE SDP ( g729 g711A g711U) | |SIP From: sip:0672028405@10.28.52.107 To:sip:0660522014@10.28.19.202 | |(5060) ------------------> (5060) | | |21,624 | 100 trying -- your call is important to us | |SIP Status | |(5060) <------------------ (5060) | | |21,624 | | INVITE SDP ( g729 g711A g711U) |SIP Request | | |(5060) ------------------> (5060) | |21,631 | | 100 Trying| |SIP Status | | |(5060) <------------------ (5060) | |21,633 | | 180 Ringing |SIP Status | | |(5060) <------------------ (5060) | |21,634 | 180 Ringing | |SIP Status | |(5060) <------------------ (5060) | | |22,712 | | 200 OK SDP ( g729) |SIP Status | | |(5060) <------------------ (5060) | |22,712 | 200 OK SDP ( g729) | |SIP Status | |(5060) <------------------ (5060) | | |22,717 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |22,717 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |22,717 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |26,037 | BYE | | |SIP Request | |(5060) ------------------> (5060) | | |26,037 | 404 User Not Found | |SIP Status | |(5060) <------------------ (5060) | | |26,040 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |26,041 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |26,041 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |29,570 | | BYE | |SIP Request | | |(5060) <------------------ (5060) | |29,570 | BYE | | |SIP Request | |(5060) <------------------ (5060) | | |29,573 | 481 Transaction Does Not Exist | |SIP Status | |(5060) ------------------> (5060) | | |29,574 | | 481 Transaction Does Not Exist |SIP Status | | |(5060) ------------------> (5060) |
10.28.52.107 is IP for my gateway 10.28.19.202 is IP for SER 10.28.19.124 is IP for my SIP Phone.
As you can see when Gateway send BYE message, SER does not relay it to IP Phone, but reply with 404 Not Found SIP Message. Have you any suggestions about this?
I don't understand your problem. The BYE will be caught by the loose route section of the config and relayed. The standard getting started should work fine. g-)
------- Original message ------- From: flavio flavio.patria@gmail.com Sent: 12.5.'07, 15:09
2007/5/12, Greger V. Teigre greger@teigre.com:
Flavio, SER doesn't reply 200 OK to BYE message, the SIP UA does that. So,
you
are fine, the BYE reaches the UA. g-)
Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
2007/5/15, Kostas Marneris K.Marneris@otenet.gr:
Hello, The call signalling exchange is a good start but it's not enough.
What some people here asked you is a capture of the SIP BYE mesg which GW sends to SER.
Kostas, Thank you for suggestion. I've already posted SIP BYE message found with ngrep :-D. However here is: U 2007/05/15 14:26:15.072602 10.28.52.105:5060 -> 10.28.19.202:5060 BYE sip:10.28.19.202 SIP/2.0. To: sip:06605XXXXX@10.28.19.202;tag=23d09968cf17c929. From: "06720XXXXX" sip:06720XXXXX@10.28.52.105;tag=DXsf22384254031Yaz079220. Via: SIP/2.0/UDP 10.28.52.105:5060;branch=z9hG4bK2384256106429488807479222. Contact: sip:10.28.52.105:5060. Call-ID: 238425403179219@10.28.52.105. Max-Forwards: 70. User-Agent: Netsynt-Gateway/SIP_AC_SRVlkup_3.1.12_4. CSeq: 23334 BYE. Content-Length: 0.
where: 06605XXXXX is the called 06720XXXXX is the callee 10.28.52.105 is IP for GW 10.28.19.202 is IP for SER
Thanks for support.
******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ********************************
As far as I know and understand it seems that the R-URI of the BYE mesg (1st line) which GW sends to SER is wrong.
I expected to see : BYE sip:user@SIP_Phone_IP_Address SIP/2.0
or the problem is that the 'Route:' header is missing from the BYE mesg so it's not loose_routed.
Do you Record_Route the original INVITE ?
Check also : rfc3261 / 16.12.1
Kostas
flavio wrote:
2007/5/15, Kostas Marneris K.Marneris@otenet.gr:
Hello, The call signalling exchange is a good start but it's not enough.
What some people here asked you is a capture of the SIP BYE mesg which GW sends to SER.
Kostas, Thank you for suggestion. I've already posted SIP BYE message found with ngrep :-D. However here is: U 2007/05/15 14:26:15.072602 10.28.52.105:5060 -> 10.28.19.202:5060 BYE sip:10.28.19.202 SIP/2.0. To: sip:06605XXXXX@10.28.19.202;tag=23d09968cf17c929. From: "06720XXXXX" sip:06720XXXXX@10.28.52.105;tag=DXsf22384254031Yaz079220. Via: SIP/2.0/UDP 10.28.52.105:5060;branch=z9hG4bK2384256106429488807479222. Contact: sip:10.28.52.105:5060. Call-ID: 238425403179219@10.28.52.105. Max-Forwards: 70. User-Agent: Netsynt-Gateway/SIP_AC_SRVlkup_3.1.12_4. CSeq: 23334 BYE. Content-Length: 0.
where: 06605XXXXX is the called 06720XXXXX is the callee 10.28.52.105 is IP for GW 10.28.19.202 is IP for SER
Thanks for support.
- (o< ing. Patria Flavio
- //\ phone 0823451358
- V_/_ mobile 3407873357
Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Yes, if you look at the BYE there is NO WAY SER can find out where to send it. The To header cannot (by RFC) be used for routing and no other header contains either location destination (as in Route) or aor (as in request uri).
To check your record routing, look at the INVITE outgoing from your SER towards the gw.
If you do proper record routing, take the RFC snippets below and hit the vendor hard on the head. (it may seem that the GW is doing strict routnig, i.e. old-old RFC). g-)
The relevant snippet from RFC3261, 12.2.1.1:
If the route set is empty, the UAC MUST place the remote target URI into the Request-URI. The UAC MUST NOT add a Route header field to the request.
If the route set is not empty, and the first URI in the route set contains the lr parameter (see Section 19.1.1), the UAC MUST place the remote target URI into the Request-URI and MUST include a Route header field containing the route set values in order, including all parameters.
If the route set is not empty, and its first URI does not contain the lr parameter, the UAC MUST place the first URI from the route set into the Request-URI, stripping any parameters that are not allowed in a Request-URI. The UAC MUST add a Route header field containing the remainder of the route set values in order, including all parameters. The UAC MUST then place the remote target URI into the Route header field as the last value.
Kostas Marneris wrote:
As far as I know and understand it seems that the R-URI of the BYE mesg (1st line) which GW sends to SER is wrong.
I expected to see : BYE sip:user@SIP_Phone_IP_Address SIP/2.0
or the problem is that the 'Route:' header is missing from the BYE mesg so it's not loose_routed.
Do you Record_Route the original INVITE ?
Check also : rfc3261 / 16.12.1
Kostas
flavio wrote:
2007/5/15, Kostas Marneris K.Marneris@otenet.gr:
Hello, The call signalling exchange is a good start but it's not enough.
What some people here asked you is a capture of the SIP BYE mesg which GW sends to SER.
Kostas, Thank you for suggestion. I've already posted SIP BYE message found with ngrep :-D. However here is: U 2007/05/15 14:26:15.072602 10.28.52.105:5060 -> 10.28.19.202:5060 BYE sip:10.28.19.202 SIP/2.0. To: sip:06605XXXXX@10.28.19.202;tag=23d09968cf17c929. From: "06720XXXXX" sip:06720XXXXX@10.28.52.105;tag=DXsf22384254031Yaz079220. Via: SIP/2.0/UDP 10.28.52.105:5060;branch=z9hG4bK2384256106429488807479222. Contact: sip:10.28.52.105:5060. Call-ID: 238425403179219@10.28.52.105. Max-Forwards: 70. User-Agent: Netsynt-Gateway/SIP_AC_SRVlkup_3.1.12_4. CSeq: 23334 BYE. Content-Length: 0.
where: 06605XXXXX is the called 06720XXXXX is the callee 10.28.52.105 is IP for GW 10.28.19.202 is IP for SER
Thanks for support.
- (o< ing. Patria Flavio
- //\ phone 0823451358
- V_/_ mobile 3407873357
Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
2007/5/15, Greger V. Teigre greger@teigre.com:
Yes, if you look at the BYE there is NO WAY SER can find out where to send it. The To header cannot (by RFC) be used for routing and no other header contains either location destination (as in Route) or aor (as in request uri).
Greger, Kostas, All,
Thank you very much for support and patience :-) Today I've tested my ser.cfg with another gateway (AudiocCodes) that insert userinfo in request uri so all works fine.
For Your information I use my NetSynt gateway with another platform that also parses "To" header, out of RFC :P Could I try it with SER? Which file should I modify to? Any reference will be well accepted ^_^
txs,
flavio
http://www.iptel.org/FAQ_To_From_change g-)
flavio wrote:
2007/5/15, Greger V. Teigre greger@teigre.com:
Yes, if you look at the BYE there is NO WAY SER can find out where to send it. The To header cannot (by RFC) be used for routing and no other header contains either location destination (as in Route) or aor (as in request uri).
Greger, Kostas, All,
Thank you very much for support and patience :-) Today I've tested my ser.cfg with another gateway (AudiocCodes) that insert userinfo in request uri so all works fine.
For Your information I use my NetSynt gateway with another platform that also parses "To" header, out of RFC :P Could I try it with SER? Which file should I modify to? Any reference will be well accepted ^_^
txs,
flavio
-
flavio -
Greger V. Teigre -
Greger Viken Teigre
-
Kostas Marneris