I am using FreePBX 22.214.171.124. Works fantastic with a very complex routing system, except for one small problem I experience.
We have two service providers, one SIP and one ISDN. We route all calls via the SIP supplier (cheaper) and fail over to the the ISDN service provider.
Sometimes the SIP provider takes a second or two too long to connect the call and then the call automatically routes via the ISDN supplier resulting in a more expensive call.
In our test we have seen, that as the number we are calling start to ring, the call gets routed via the second provider.
According to the log files there is “NO ANSWER”:
"",“106”,“90833873873”,“from-internal”,“106”,“SIP/106-00000053”,“SIP/Neotel-00000054”,“Dial”,“SIP/Neotel/0833873873,300,wW”,“2013-08-19 05:22:59”,“2013-08-19 05:23:04”,5,0,“NO ANSWER”,“DOCUMENTATION”,“1376889779.212”,""
The phone only rang once. I spoke to our SIP Supplier and according to them they did not send the “NO ANSWER” back, which means somewhere in FreePBX there might be a time limit set to go to the next route if no response.
If such a limit, please let me know where to find it.
Ask Neotel if they support “progress” - If they are as backwards as I have find some telco’s, then they will only support progress “inband” or “progress tones”
In a nutshell, FreePBX should not try the other route if it is getting a response from Neotel. Given that the number you are dialing is a mobile, I would strongly suggest that Neotel is not updating you until it gets an update from the mobile network (or they are, but in progress tones).
Open “asterisk -r” and enter “sip set debug peer neotel” and try the calls again. You are looking for a 183 response from neotel in the sip headers. If you don’t see this at all, then you should investigate progresstones and Asterisk.
Thanks for the quick response. Test with the debug on as suggested, resulted in no “183” response from Neotel.
I will follow up with them about progress tones and see how I can resolve this.
Thanks for the suggestions and help.