Hello,
I’ve recently deployed a pbx with multiple sip accounts. I followed the instructions that my sip provider provided me (http://www.belcentrale.nl/include/dl.php?dir=websitedownloads%2Fhandleidingen%2F&download=Handleiding+Trixbox+PBX+Belcentrale.pdf) to set up the extensions/routes/trunks. Almost everything works (I can place and receive calls to all my sip phonenumbers) but I am experiencing very strange behaviour regarding missed calls.
When someone misses a call. The phones history shows that I missed a call from 1234003 (where 1234003 is the username I entered in the last Trunk I created, if I would delete the 3d trunk it would show a missed call from 1234*002)
I than run asterisk -vrr with sip debug to see what was going on and noticed the following:
Sip debug info:
<------------->
— (8 headers 0 lines) —
Transmitting (NAT) to 192.168.10.18:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.5:5060;branch=xxx;rport
From: “1234567890” sip:1234*[email protected];tag=xxx
To: sip:[email protected]:5060;tag=xxx
Contact: sip:1234*[email protected]
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
where 192.168.10.18 is the ip of the phone and 192.168.10.5 is the ip of the pbx.
dialparties.agi: Caller ID name is ‘1234567890’ number is ‘1234*001’
where 1234567890 the callerid and 1234*001 is the sip username from my sip provider.
Somehow instead of the number the last trunk username is being showed as if it where the phonenumber. I also contacted my voip provider (Belcentrale) about this but they told me they tested it and it had to be a freepbx configuration issue and that they couldn’t help me with that any further.
Does anyone had similar issues or maybe any ideas on how to fix this problem because it’s very annoying not to know whos calls you missed and not being able to call them back.
Cheers,
Stan