Asterisk config - failover to another trunk if all GSM lines busy

Hi,

Sorry to make trouble to you.

Please contact me directly,so that i can check your issues remotly.

My contact info be shown as below:
Skype: rick.zhu3
Gtalk: This email address is being protected from spambots. You need JavaScript enabled to view it.
e-mail: This email address is being protected from spambots. You need JavaScript enabled to view it.

Best Regards.

Hi,

Now the gateway can support your needs.

I give you a sample manual,you can refer it and configure it.

Please download it from attachment.

If you have any issues please contact me,My contact info be shown as up.

Best Regards.

This is attachment.

Yes, I am aware that this feature is present in the gateway itself. But if I use it, the PBX billing system won't be able to distinguish whether the calls were made through GSM or SIP, that's why I need to configure that in the Asterisk system that connects to the GMS gateway as a SIP peer, so that it could make a proper record to the billing data. So, the question is: does the gateway provide the necessary information (and what kind of information) to the Asterisk on the PBX, that connects to the gateway as a SIP client?

Hi alayer,

Please contact me directly,so that i can know more details about your question.

My contact info be shown as up.

Best Regards.

Finally, I managed to configure it.
But one question remains. If I save SIP Endpoint settings, a line "insecure=invite,port' appears in the corresponding section of sip_endpoints.conf. But I don't need it. Can you tell me how to turn it off?

i] Last edited by aleyer at 2014-2-11 17:10 [/i