Hi all,
Thanks to everyone in advance. I seem to be having issues regarding making calls and receive calls through my sip trunk. I’m not very experienced on Freepbx so i don’t know where’s the problem with my configuration. Here´s my sip.conf:
host=10.64.0.31
context=default
type=friend
careinvite=no
disallow=all
allow=alaw&ulaw&gsm
dtmfmode=rfc2833
qualify=yes
The peers is working, as far as i know:
raspbxCLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
SOC 10.64.0.31 Yes Yes 5060 OK (3 ms)
1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline]
raspbxCLI>
Here’s the SIP debug of a failed attemp to make a call:
SIP Debugging enabled
Audio is at 42520
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.64.0.31:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK4fb4c907;rport
Max-Forwards: 70
From: “Digno” <sip:[email protected]>;tag=as3f924a5d
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-13.0.195.22(13.24.1)
Date: Fri, 04 Jan 2019 11:32:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 297
v=0
o=root 1622748209 1622748209 IN IP4 192.168.2.3
s=Asterisk PBX 13.24.1
c=IN IP4 192.168.2.3
t=0 0
m=audio 42520 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<— SIP read from UDP:10.64.0.31:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.3:5060;rport=34076;received=10.64.0.16;branch=z9hG4bK4fb4c907
To: <sip:[email protected]>
From: “Digno” <sip:[email protected]>;tag=as3f924a5d
CSeq: 102 INVITE
Call-ID: [email protected]:5060
Content-Length: 0
<------------->
— (7 headers 0 lines) —
<— SIP read from UDP:10.64.0.31:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.2.3:5060;rport=34076;received=10.64.0.16;branch=z9hG4bK4fb4c907
To: <sip:[email protected]>;tag=946684840154d1392-f81a-472e-a2b1-01fa896e0e5b
From: “Digno” <sip:[email protected]>;tag=as3f924a5d
CSeq: 102 INVITE
Call-ID: [email protected]:5060
Server: Epygi Quadro SIP User Agent/v6.1.6 (QX-E1T1)
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Transmitting (NAT) to 10.64.0.31:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK4fb4c907;rport
Max-Forwards: 70
From: “Digno” <sip:[email protected]>;tag=as3f924a5d
To: <sip:[email protected]>;tag=946684840154d1392-f81a-472e-a2b1-01fa896e0e5b
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-13.0.195.22(13.24.1)
Content-Length: 0
Scheduling destruction of SIP dialog ‘[email protected]:5060’ in 6400 ms (Method: INVITE)
Really destroying SIP dialog ‘[email protected]:5060’ Method: INVITE
And the SIP channel, when the call is active:
raspbx*CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
10.64.0.31 3787959 6f4d892f5fdfed9 (nothing) No Tx: ACK SOC
1 active SIP dialog
Please let me know if you find the issue here.