Hi all,
First time poster new to FreePBX (running Asterisk v1.8.20.0), so please be gentle if my question is basic… we’ve all got to start somewhere and your assistance is appreciated!
First, what is working:
- I have 2 SIP extensions and call between them
- I have an Outbound Trunk with VoIPtalk and call SIP to PSTN
- I have an Inbound Trunk with 3C.co.uk and can call PSTN to SIP
However, I want to be able to use my VoIPtalk SIP trunk also for Inbound calls, as I have DDI number assigned.
The VoIPtalk Trunk is configured with Outbound Settings and the Registration String (it is showing connected under sip show peers and sip show registry). There is no configuration in the Inbound Settings section for this Trunk. Should there be?
I have an Inbound Route configured with the ID of my VoIPtalk number to route calls to one of the SIP extensions.
If I debug SIP, this is what I see in the log file when calling internationally from GSM 00971561797574 to PSTN 08438490944 (my VoIPtalk Incoming number):
<— SIP read from UDP:77.240.54.13:5063 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 77.240.54.13:5063;branch=z9hG4bK09fafa2d;rport
Max-Forwards: 70
From: “00971561797574” sip:[email protected]:5063;tag=as53cef0d4
To: sip:[email protected]
Contact: sip:[email protected]:5063
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: voip
Date: Thu, 05 Dec 2013 19:54:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Privacy: none
P-Asserted-Identity: “00971561797574” sip:[email protected]
Content-Type: application/sdp
Content-Length: 386
v=0
o=voip 2121146035 2121146035 IN IP4 77.240.54.13
s=voip
c=IN IP4 77.240.54.13
t=0 0
m=audio 16208 RTP/AVP 8 0 3 97 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
— (16 headers 18 lines) —
Sending to 77.240.54.13:5063 (no NAT)
Using INVITE request as basis request - [email protected]
No matching peer for ‘00971561797574’ from ‘77.240.54.13:5063’
<— Reliably Transmitting (no NAT) to 77.240.54.13:5063 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 77.240.54.13:5063;branch=z9hG4bK09fafa2d;received=77.240.54.13;rport=5063
From: “00971561797574” sip:[email protected]:5063;tag=as53cef0d4
To: sip:[email protected];tag=as2e83b177
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.11.0(1.8.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="45ccd8f0"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)
<— SIP read from UDP:77.240.54.13:5063 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 77.240.54.13:5063;branch=z9hG4bK09fafa2d;rport
Max-Forwards: 70
From: “00971561797574” sip:[email protected]:5063;tag=as53cef0d4
To: sip:[email protected];tag=as2e83b177
Contact: sip:[email protected]:5063
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: voip
Content-Length: 0
<------------->
— (10 headers 0 lines) —
Hope this is enough information for some initial thoughts from the community?
Thanks!