[OpenSER-Users] Wrong To: Field In SIP Packet when Sending to PSTN GW

Ali Jawad ali.jawad at splendor.net
Fri Feb 1 14:09:00 CET 2008


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 at voice-system.ro] 
Sent: Friday, February 01, 2008 3:06 PM
To: Ali Jawad
Cc: users at 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 I would say your GW is outdated.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 at myOpenserDomain instead 
> of sip:calledNumber at 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 at jabber.splendor.net>;tag=f36d6608.
> To: "009613041705" 
> *<sip:009613041705 at 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>
>
>  
>
>
------------------------------------------------------------------------
>
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> Users mailing list
> Users at lists.openser.org
> http://lists.openser.org/cgi-bin/mailman/listinfo/users
>   


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