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
