Hi All
I am using OpenSER as a proxy to make outbound calls my config is very simple if the number dialled is not an openser account route it to the PSTN gw.
if(does_uri_exist()){ # local uri does exist, is probably a user. # lookup location if(lookup("location")){ route(1); return; } } else { # probably a call to pstn.... route(2); return; } and route[2] { # pstn handling, simply route out to pstn. sethostport("xx.xx.xx.xx:5060"); route(1); }
The problem is that once the SIP packet arrives at the PSTN GW it does NOT have the correct TO: set. Therefore the call does not get routed .
In the example below TO: is sip:calledNumber@myOpenserDomain instead of sip:calledNumber@PSTN.GATEWAY.IP.IP
Caller: ali [!at] jabber.splendor.net (replace the [!at] with a @) Callee: 009613041708 OpenSerDomain: jabber.splendor.net
U +0.289348 PSTN.GW.IP.IP:5060 -> 193.237.226.252:5060 SIP/2.0 404 Not Found . Via: SIP/2.0/UDP 193.237.226.252;rport;branch=z9hG4bK6828.10c0315.0. Via: SIP/2.0/UDP 192.168.0.176:65068;received=193.227.186.146;branch=z9hG4bK-d87543-be62c 55d821be10d-1--d87543-;rport=65068. Record-Route: sip:193.237.226.252;lr=on. From: "ssafass" sip:ali@jabber.splendor.net;tag=f36d6608. To: "009613041705" sip:009613041705@jabber.splendor.net;tag=GR52RWG346-34. Call-ID: 0942e159a72eab40ZmViZWY4YTVlOTRlOGJmZTM5ZDdkZGJiZjFmMTlmMjk.. CSeq: 1 INVITE. Contact: "0000" sip:PSTN.GW.IP.IP:5060. User-Agent: eyeBeam release 1003s stamp 31159. Content-Length: 0.
I did a siptrace on the interface of the SIP proxy
http://pastebin.com/d56426d63 http://www.voipuser.org/ship_to.php?url=http://pastebin.com/d56426d63
This is my config:
http://pastebin.com/m128ca16e http://www.voipuser.org/ship_to.php?url=http://pastebin.com/m128ca16e
Hi Ali,
SIP routing (RFC3261) is done based on RURI and not To URI - TO does not change in the message. I would say your GW is outdated.
Regards, Bogdan
Ali Jawad wrote:
Hi All
I am using OpenSER as a proxy to make outbound calls my config is very simple if the number dialled is not an openser account route it to the PSTN gw.
if(does_uri_exist()){ # local uri does exist, is probably a user. # lookup location if(lookup("location")){ route(1); return; } *} else { # probably a call to pstn.... route(2); return; }* and route[2] { # pstn handling, simply route out to pstn. *sethostport("xx.xx.xx.xx:5060");* route(1); }
The problem is that once the SIP packet arrives at the PSTN GW it does NOT have the correct TO: set. Therefore the call does not get routed .
In the example below TO: is sip:calledNumber@myOpenserDomain instead of sip:calledNumber@PSTN.GATEWAY.IP.IP
Caller: ali [!at] jabber.splendor.net (replace the [!at] with a @) Callee: 009613041708 OpenSerDomain: jabber.splendor.net
U +0.289348 PSTN.GW.IP.IP:5060 -> 193.237.226.252:5060 SIP/2.0 404 Not Found . Via: SIP/2.0/UDP 193.237.226.252;rport;branch=z9hG4bK6828.10c0315.0. Via: SIP/2.0/UDP 192.168.0.176:65068;received=193.227.186.146;branch=z9hG4bK-d87543-be62c55d821be10d-1--d87543-;rport=65068.
Record-Route: sip:193.237.226.252;lr=on. From: "ssafass" sip:ali@jabber.splendor.net;tag=f36d6608. To: "009613041705" *sip:009613041705@jabber.splendor.net;tag=GR52RWG346-34.* Call-ID: 0942e159a72eab40ZmViZWY4YTVlOTRlOGJmZTM5ZDdkZGJiZjFmMTlmMjk.. CSeq: 1 INVITE. Contact: "0000" sip:PSTN.GW.IP.IP:5060. User-Agent: eyeBeam release 1003s stamp 31159. Content-Length: 0.
I did a siptrace on the interface of the SIP proxy
http://pastebin.com/d56426d63 http://www.voipuser.org/ship_to.php?url=http://pastebin.com/d56426d63
This is my config:
http://pastebin.com/m128ca16e http://www.voipuser.org/ship_to.php?url=http://pastebin.com/m128ca16e
Users mailing list Users@lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users
The error message I am getting is Call Failed: Not Found. The thing is that the GW is working with Asterisk, Linksys phones and other 3rd Party SIP proxies. Is there something I can do using OpenSer ? I have even contacted the GW company and they said that they do have clients using OpenSer. Thanks for your reply.
-----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Friday, February 01, 2008 3:06 PM To: Ali Jawad Cc: users@lists.openser.org Subject: Re: [OpenSER-Users] Wrong To: Field In SIP Packet when Sending to PSTN GW
Hi Ali,
SIP routing (RFC3261) is done based on RURI and not To URI - TO does not
change in the message. I would say your GW is outdated.
Regards, Bogdan
Ali Jawad wrote:
Hi All
I am using OpenSER as a proxy to make outbound calls my config is very
simple if the number dialled is not an openser account route it to the
PSTN gw.
if(does_uri_exist()){ # local uri does exist, is probably a user. # lookup location if(lookup("location")){ route(1); return; } *} else { # probably a call to pstn.... route(2); return; }* and route[2] { # pstn handling, simply route out to pstn. *sethostport("xx.xx.xx.xx:5060");* route(1); }
The problem is that once the SIP packet arrives at the PSTN GW it does
NOT have the correct TO: set. Therefore the call does not get routed .
In the example below TO: is sip:calledNumber@myOpenserDomain instead of sip:calledNumber@PSTN.GATEWAY.IP.IP
Caller: ali [!at] jabber.splendor.net (replace the [!at] with a @) Callee: 009613041708 OpenSerDomain: jabber.splendor.net
U +0.289348 PSTN.GW.IP.IP:5060 -> 193.237.226.252:5060 SIP/2.0 404 Not Found . Via: SIP/2.0/UDP 193.237.226.252;rport;branch=z9hG4bK6828.10c0315.0. Via: SIP/2.0/UDP
192.168.0.176:65068;received=193.227.186.146;branch=z9hG4bK-d87543-be62c 55d821be10d-1--d87543-;rport=65068.
Record-Route: sip:193.237.226.252;lr=on. From: "ssafass" sip:ali@jabber.splendor.net;tag=f36d6608. To: "009613041705" *sip:009613041705@jabber.splendor.net;tag=GR52RWG346-34.* Call-ID: 0942e159a72eab40ZmViZWY4YTVlOTRlOGJmZTM5ZDdkZGJiZjFmMTlmMjk.. CSeq: 1 INVITE. Contact: "0000" sip:PSTN.GW.IP.IP:5060. User-Agent: eyeBeam release 1003s stamp 31159. Content-Length: 0.
I did a siptrace on the interface of the SIP proxy
http://www.voipuser.org/ship_to.php?url=http://pastebin.com/d56426d63
This is my config:
http://www.voipuser.org/ship_to.php?url=http://pastebin.com/m128ca16e
------------------------------------------------------------------------
Users mailing list Users@lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users
I cannot pin point the problem, but as you originally said that the GW is looking for number in TO I just pointed you that this is not correct.
Of course, as an ultimate solution you can abuse the subst() function from textops module and to brutally change the TO header, but it is not something you should do at all because (a) it it against SIP protocol and (b) the problem is in other place.
Regards, Bogdan
Ali Jawad wrote:
The error message I am getting is Call Failed: Not Found. The thing is that the GW is working with Asterisk, Linksys phones and other 3rd Party SIP proxies. Is there something I can do using OpenSer ? I have even contacted the GW company and they said that they do have clients using OpenSer. Thanks for your reply.
-----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Friday, February 01, 2008 3:06 PM To: Ali Jawad Cc: users@lists.openser.org Subject: Re: [OpenSER-Users] Wrong To: Field In SIP Packet when Sending to PSTN GW
Hi Ali,
SIP routing (RFC3261) is done based on RURI and not To URI - TO does not
change in the message. I would say your GW is outdated.
Regards, Bogdan
Ali Jawad wrote:
Hi All
I am using OpenSER as a proxy to make outbound calls my config is very
simple if the number dialled is not an openser account route it to the
PSTN gw.
if(does_uri_exist()){ # local uri does exist, is probably a user. # lookup location if(lookup("location")){ route(1); return; } *} else { # probably a call to pstn.... route(2); return; }* and route[2] { # pstn handling, simply route out to pstn. *sethostport("xx.xx.xx.xx:5060");* route(1); }
The problem is that once the SIP packet arrives at the PSTN GW it does
NOT have the correct TO: set. Therefore the call does not get routed .
In the example below TO: is sip:calledNumber@myOpenserDomain instead of sip:calledNumber@PSTN.GATEWAY.IP.IP
Caller: ali [!at] jabber.splendor.net (replace the [!at] with a @) Callee: 009613041708 OpenSerDomain: jabber.splendor.net
U +0.289348 PSTN.GW.IP.IP:5060 -> 193.237.226.252:5060 SIP/2.0 404 Not Found . Via: SIP/2.0/UDP 193.237.226.252;rport;branch=z9hG4bK6828.10c0315.0. Via: SIP/2.0/UDP
192.168.0.176:65068;received=193.227.186.146;branch=z9hG4bK-d87543-be62c 55d821be10d-1--d87543-;rport=65068.
Record-Route: sip:193.237.226.252;lr=on. From: "ssafass" sip:ali@jabber.splendor.net;tag=f36d6608. To: "009613041705" *sip:009613041705@jabber.splendor.net;tag=GR52RWG346-34.* Call-ID: 0942e159a72eab40ZmViZWY4YTVlOTRlOGJmZTM5ZDdkZGJiZjFmMTlmMjk.. CSeq: 1 INVITE. Contact: "0000" sip:PSTN.GW.IP.IP:5060. User-Agent: eyeBeam release 1003s stamp 31159. Content-Length: 0.
I did a siptrace on the interface of the SIP proxy
http://www.voipuser.org/ship_to.php?url=http://pastebin.com/d56426d63
This is my config:
http://www.voipuser.org/ship_to.php?url=http://pastebin.com/m128ca16e
Users mailing list Users@lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users
Thanks for your reply I have used the subst function and as you said it did not work and that is not what is wrong..could it be different interpretation of the SIP stack..the GW software is supposed to follow RFC standards.
-----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Fri 2/1/2008 9:04 PM To: Ali Jawad Cc: users@lists.openser.org Subject: Re: [OpenSER-Users] Wrong To: Field In SIP Packet when Sending to PSTN GW
I cannot pin point the problem, but as you originally said that the GW is looking for number in TO I just pointed you that this is not correct.
Of course, as an ultimate solution you can abuse the subst() function from textops module and to brutally change the TO header, but it is not something you should do at all because (a) it it against SIP protocol and (b) the problem is in other place.
Regards, Bogdan
Ali Jawad wrote:
The error message I am getting is Call Failed: Not Found. The thing is that the GW is working with Asterisk, Linksys phones and other 3rd Party SIP proxies. Is there something I can do using OpenSer ? I have even contacted the GW company and they said that they do have clients using OpenSer. Thanks for your reply.
-----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Friday, February 01, 2008 3:06 PM To: Ali Jawad Cc: users@lists.openser.org Subject: Re: [OpenSER-Users] Wrong To: Field In SIP Packet when Sending to PSTN GW
Hi Ali,
SIP routing (RFC3261) is done based on RURI and not To URI - TO does not
change in the message. I would say your GW is outdated.
Regards, Bogdan
Ali Jawad wrote:
Hi All
I am using OpenSER as a proxy to make outbound calls my config is very
simple if the number dialled is not an openser account route it to the
PSTN gw.
if(does_uri_exist()){ # local uri does exist, is probably a user. # lookup location if(lookup("location")){ route(1); return; } *} else { # probably a call to pstn.... route(2); return; }* and route[2] { # pstn handling, simply route out to pstn. *sethostport("xx.xx.xx.xx:5060");* route(1); }
The problem is that once the SIP packet arrives at the PSTN GW it does
NOT have the correct TO: set. Therefore the call does not get routed .
In the example below TO: is sip:calledNumber@myOpenserDomain instead of sip:calledNumber@PSTN.GATEWAY.IP.IP
Caller: ali [!at] jabber.splendor.net (replace the [!at] with a @) Callee: 009613041708 OpenSerDomain: jabber.splendor.net
U +0.289348 PSTN.GW.IP.IP:5060 -> 193.237.226.252:5060 SIP/2.0 404 Not Found . Via: SIP/2.0/UDP 193.237.226.252;rport;branch=z9hG4bK6828.10c0315.0. Via: SIP/2.0/UDP
192.168.0.176:65068;received=193.227.186.146;branch=z9hG4bK-d87543-be62c 55d821be10d-1--d87543-;rport=65068.
Record-Route: sip:193.237.226.252;lr=on. From: "ssafass" sip:ali@jabber.splendor.net;tag=f36d6608. To: "009613041705" *sip:009613041705@jabber.splendor.net;tag=GR52RWG346-34.* Call-ID: 0942e159a72eab40ZmViZWY4YTVlOTRlOGJmZTM5ZDdkZGJiZjFmMTlmMjk.. CSeq: 1 INVITE. Contact: "0000" sip:PSTN.GW.IP.IP:5060. User-Agent: eyeBeam release 1003s stamp 31159. Content-Length: 0.
I did a siptrace on the interface of the SIP proxy
http://www.voipuser.org/ship_to.php?url=http://pastebin.com/d56426d63
This is my config:
http://www.voipuser.org/ship_to.php?url=http://pastebin.com/m128ca16e
Users mailing list Users@lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users
-
Ali Jawad -
Bogdan-Andrei Iancu