Incoming SIP issue

FreePBX
1

I am trying to determine why my incoming calls are not working. I have tried almost everything I can think of. Currently all firewalls are off and my calls do not get through.

Current Asterisk Version: 13.15.0
Outgoing calls work fine
Internal calls work fine
Verbose logging is set to 3 nothing registers in the log
SIP provider Twilio

Chan_Sip Peers

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
444 (Unspecified) D No No A 0 UNKNOWN
997000 (Unspecified) D No No A 0 UNKNOWN
997001 (Unspecified) D No No A 0 UNKNOWN
997003 (Unspecified) D No No A 0 UNKNOWN
997004 (Unspecified) D No No A 0 UNKNOWN
997005 (Unspecified) D No No A 0 UNKNOWN
997006 (Unspecified) D No No A 0 UNKNOWN
997007 (Unspecified) D No No A 0 UNKNOWN
997008 (Unspecified) D No No A 0 UNKNOWN
997009 (Unspecified) D No No A 0 UNKNOWN
997010 (Unspecified) D No No A 0 UNKNOWN
997011 (Unspecified) D No No A 0 UNKNOWN
997012 (Unspecified) D No No A 0 UNKNOWN
997013 (Unspecified) D No No A 0 UNKNOWN
Twilio-IN 54.172.60.1 Yes Yes 5060 Unmonitored
Twilio-IN_copy_2 54.172.60.2 Yes Yes 5060 Unmonitored
Twilio-OUT/icsipuser 54.172.60.1 Yes Yes 5060 Unmonitored
Twilio-OUT_copy_2/icsipus 54.172.60.1 Yes Yes 5060 Unmonitored
18 sip peers [Monitored: 0 online, 14 offline Unmonitored: 4 online, 0 offline]

2

I would like to add my pcap file but since I am a new user it looks like the forums prevents me from doing so.

3

hi,
check you port in new version port 5160 for sip
tycons

4

Thanks Tycoons, I am new to this. Where would I check port version?

5

my freepx is 13.51 the GUI say me sip port i have change 5160 and 5060 is for other driver i forget the name

6

I am a little confused should I be using chan_sip or chan_pjsip for the trunk? I am currently using chan_sip.

Also

for the extensions I am using pjsip. Is this where I made my mistake?

7

not confused is here for help comunauty
me i used sip port 5160 i don t use chan_pjsip port 5060
i am not a specialist yet but I work there :smile:
I think that it is necessary to use a single driver either sip or chan_pjsip so the port changes

8

I think I am closer, still no ringing but at least I have something in the log. not sure why its not ringing though:

<------------->
[2017-05-24 11:23:55] VERBOSE[2001] chan_sip.c:
<— SIP read from UDP:192.168.1.2:60339 —>

<------------->
[2017-05-24 11:24:19] VERBOSE[2001] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.2:60339:
OPTIONS sip:[email protected]:60339;rinstance=1802457e666b29a1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5160;branch=z9hG4bK1b79f66b
Max-Forwards: 70
From: “Unknown” sip:[email protected]:5160;tag=as7ff3b3a5
To: sip:[email protected]:60339;rinstance=1802457e666b29a1
Contact: sip:[email protected]:5160
Call-ID: [email protected]:5160
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.191.11(13.15.0)
Date: Wed, 24 May 2017 15:24:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

[2017-05-24 11:24:19] VERBOSE[2001] chan_sip.c:
<— SIP read from UDP:192.168.1.2:60339 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.15:5160;branch=z9hG4bK1b79f66b
Contact: sip:192.168.1.2:60339
To: sip:[email protected]:60339;rinstance=1802457e666b29a1;tag=802b6713
From: “Unknown” sip:[email protected]:5160;tag=as7ff3b3a5
Call-ID: [email protected]:5160
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Supported: replaces
User-Agent: X-Lite release 4.9.8 stamp 84253
Allow-Events: talk, hold
Content-Length: 0

<------------->
[2017-05-24 11:24:19] VERBOSE[2001] chan_sip.c: — (14 headers 0 lines) —
[2017-05-24 11:24:19] VERBOSE[2001] chan_sip.c: Really destroying SIP dialog ‘[email protected]:5160’ Method: OPTIONS
FreePBX® FreePBX is a registered trademark of
Sangoma Technologies Inc.
FreePBX 13.0.191.11 is licensed under the GPL
Copyrigh

9

show me you sip conf trunk
i think forget one parametre

10

I ended up reinstalling FreePBX 10.13.66 with Asterisk 11 and not 13. It was easier than trying to uncoil sip vs pjsip port issues. You did help me make that decision and I am very pleased that you helped me.

11

there is no issues with FreePBX 13. You can config your sip channel on what ever port you like.