1 Feb
2008
1 Feb
'08
2 p.m.
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(a)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(a)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 I would say your GW is outdated.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:
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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>
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