Unable to make call or receive them by SIP Trunk

@DJ26 wrote:

Hi all,

I opened the topic once again because i went on vacation and it got closed Sorry to everyone.

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:

raspbx CLI> 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:3787959@10.64.0.31](mailto:sip:3787959@10.64.0.31) SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK4fb4c907;rport
Max-Forwards: 70
From: “Digno” <[sip:3780201@192.168.2.3](mailto:sip:3780201@192.168.2.3)>;tag=as3f924a5d
To: <[sip:3787959@10.64.0.31](mailto:sip:3787959@10.64.0.31)>
Contact: <sip:3780201@192.168.2.3:5060>
Call-ID: 58f5038313d4fe8d2e4b15853ba5bd9a@192.168.2.3:5060
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:3787959@10.64.0.31](mailto:sip:3787959@10.64.0.31)>
From: “Digno” <[sip:3780201@192.168.2.3](mailto:sip:3780201@192.168.2.3)>;tag=as3f924a5d
CSeq: 102 INVITE
Call-ID: 58f5038313d4fe8d2e4b15853ba5bd9a@192.168.2.3:5060
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:3787959@10.64.0.31](mailto:sip:3787959@10.64.0.31)>;tag=946684840154d1392-f81a-472e-a2b1-01fa896e0e5b
From: “Digno” <[sip:3780201@192.168.2.3](mailto:sip:3780201@192.168.2.3)>;tag=as3f924a5d
CSeq: 102 INVITE
Call-ID: 58f5038313d4fe8d2e4b15853ba5bd9a@192.168.2.3:5060
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:3787959@10.64.0.31](mailto:sip:3787959@10.64.0.31) SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK4fb4c907;rport
Max-Forwards: 70
From: “Digno” <[sip:3780201@192.168.2.3](mailto:sip:3780201@192.168.2.3)>;tag=as3f924a5d
To: <[sip:3787959@10.64.0.31](mailto:sip:3787959@10.64.0.31)>;tag=946684840154d1392-f81a-472e-a2b1-01fa896e0e5b
Contact: <sip:3780201@192.168.2.3:5060>
Call-ID: 58f5038313d4fe8d2e4b15853ba5bd9a@192.168.2.3:5060
CSeq: 102 ACK
User-Agent: FPBX-13.0.195.22(13.24.1)
Content-Length: 0

Scheduling destruction of SIP dialog ‘58f5038313d4fe8d2e4b15853ba5bd9a@192.168.2.3:5060’ in 6400 ms (Method: INVITE)
Really destroying SIP dialog ‘58f5038313d4fe8d2e4b15853ba5bd9a@192.168.2.3:5060’ 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

·······································································································

I’m from Panama, since it’s a small country, it only has the country code (+507). Fixed numbers are 7 digits long (like 378-7959 wich is my office number) and mobile numbers are 8 digits long (all of them starts with 6, for example 66163230) so my 2 only Outbound Routes are:

NXXXXXX – for Fixed Numbers
6XXXXXXX – For Mobile Numbers

My ISP is the same that gives me the SIP Trunk, it gives me a E1 and I convert that to SIP trough an Epygi Module. That’s where the 10.64.0.31, it’s an IP i gave to the Epygi so it can work as my Sip server. As to how I connect everything.

E1 --> Epygi --> Router (Nat between 10.64.0.31 and the subnet the RPi is, which is 192.168.2.0/24) Teldat M1 --> RPi and Softphone.

Edit: My Inbound Calls doesn’t work either, the call doesn’t get to the Asterisk.

Please let me know if you find the issue here.

Posts: 1

Participants: 1

Read full topic