Can't Make or Receive Calls - SIP Tunk

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]
raspbx
CLI>

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.

A 404 error typically means that the called number is not valid or not formatted correctly, though some trunking providers also give this error for an incorrect calling account/number or an invalid domain/IP address.

Does “3787959” represent a phone number, or an account number of another user on the same service? If the former, it’s likely that the provider requires area code or even country code information to be included. For example, if this were a number in New York they may want 2123787959, 12123787959 or +12123787959.

What country are you in? Who is the ISP? Are they also the trunk supplier? What kind of modem do you have? Is it configured as a router? Separate router/firewall, if any?

Do you have another device (IP phone, ATA, softphone, SIP app) that has worked on this trunk? If so, please post settings. Also, please post whatever documentation you have for the trunk settings (I’m sure that you didn’t just guess the 10.64.0.31 address).

What happens on incoming call attempts? If nothing hits your Pi, check that you have forwarded UDP port 5060 (in your modem or router, whatever has the 10.64.0.16 WAN address) to 192.1168.2.3

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