Intermittant call problems with Sipgate SIP Trunk
Submitted by ateece on Wed, 08/06/2008 - 3:11am.
Hi
We are having out-bound calling problems with a SipGate.co.uk SIP trunk.
We can 100% reliably call 01xxx xxxxxx numbers (ie UK numbers).
If we call a UK mobile 07xxx xxxxxx we only get audio 50% of the time.
Now I know that this sounds like a typical NAT problem, but both 07 and 01 calls are going over the exact same outbound route, using the exact same trunk in asterisk. So I cannot see why/how there would be a difference at our end.
Problem is, I started speaking to SipGate support who say they wont help with Asterisk. So I tried using their x-lite client and don't experiance the problem at all, directing me back towards our Trixbox.
We are behind a firewall, but I have the following ports directed to our Trixbox;
5060 TCP/UDP
10000 - 20000 TCP/UDP
To go with this, we have the localnet, externip and externrefresh settings in the sip_general_custom.conf and rtpstart=10000 and rtpend=20000 in the rtp.conf.
We are using ulaw and alaw codecs. When testing x-lite I limited it to G711u and a also.
Does anyone have any idea where I should start debugging this? How could a trunk behave differently when calling different numbers, but the problem be internal?
Member Since:
2006-06-27