Please advise how to assure that originating SIP-phone call timer waits for answer.
On my FreePBX-13 installation, when a SIP phone dials the 10 digit number of a PSTN phone, the originating phone timer waits for the call to be answered, as expected. Also
when placing a call from one internal extension to another, the originating phone timer starts timing the call when it is answered. However, if an extension dials a 10 digit phone number that is routed out on a SIP trunk, back in on a SIP trunk routed to a target extension, the originating phone timer starts while the called extension starts is ringing.
Additional observations: 1) when the experiment is conducted on an old Trixbox 2.4.2 platform, the originating phone timer waits for the call to be answered, as desired.
Thanks for any advice.
Possibly, your incoming trunk provider is giving FAS (False Answer Supervision). Google Voice and some other ‘free’ providers do this, though it shouldn’t be a problem with reputable paid VoSPs.
If the Incoming Route does not go directly to the extension, e.g. it’s routed via an IVR, announcement, etc., then Asterisk will immediately answer the call to play the prompts. If routed via a Ring Group, some options require Asterisk to answer the call. Also, Detect Faxes requires Asterisk to answer.
If the above does not explain your situation, make a test call from your mobile to the DID in question, to determine whether the incoming leg shows the trouble, regardless of source. Then post the Asterisk log for a failing call.
Thanks for your suggestions.
I had ruled out provider FAS, IVR, announcement, ring group and other Asterisk options that might answer the call. I would not have thought of “detect faxes” as the source of my problem. Disabling this feature in the inbound route settings resolved it!!
Asterisk has to answer the call to listen to the incoming tones to determine if this is a FAX call or not.
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.