Hello Friend,
i have a problem with my sip trunk in a2billing and i hope that you could help me. Iam using centos 6.3 64 bits, asterisk 1.8 and a2billing 2.0.1. I have a gsm gateway system ( sim server and gsm gateways), i have 80 gsm ports in this system, i have connceted this system to my a2billing system using the sip trunk so i have create a sip trunk in a2billing and i have set max use= -1 but when my asterisk server send calls throw this trunk i see that it pass few calls and reject more calls inspite i have ab lot of gsm ports free but the trunk for example if i send 10 calls it pass 3 and reject 7 calls i don't know the problem exactly this is the trunk configuration:
[trunkName] ACCOUNTCODE=.... REGEXTEN=.... disallow=all allow=g729,g723 qualify=no port=5060 type=friend nat=yes dtmfmode=rfc2833 canreinvite=no rtptimeout=600 rtpholdtimeout=600 insecure=port,invite cancallforward=no allowtransfer=no
Best Regards Mettichi Bassem
|