DIDWW outbound not working

SparkyRih · February 6, 2024, 10:16am

I switched SIP provider, I did use DIDLogic before, but now switched to DIDWW. My inbound SIP trunk and route are working fine. But for DIDWW I need to setup a separate outbound trunk (I’m still using the same route, which did work with my previous provider).

However, I can’t get it to work. I’ve tried asking DIDWW support, the only suggestion they could come up with after reviewing my config was setting SIP qualifying to no. But that does not resolve the issue.

have the following debug information:

Last login: Tue Feb  6 08:37:42 2024 from 192.168.20.2
______                   ______ ______ __   __
|  ___|                  | ___ \| ___ \\ \ / /
| |_    _ __   ___   ___ | |_/ /| |_/ / \ V /
|  _|  | '__| / _ \ / _ \|  __/ | ___ \ /   \
| |    | |   |  __/|  __/| |    | |_/ // /^\ \
\_|    |_|    \___| \___|\_|    \____/ \/   \/


NOTICE! You have 4 notifications! Please log into the UI to see them!
Current Network Configuration
+-----------+-------------------+--------------------------+
| Interface | MAC Address       | IP Addresses             |
+-----------+-------------------+--------------------------+
| eth0      | 00:0C:29:1B:7E:9B | 10.2.0.20                |
|           |                   | fe80::20c:29ff:fe1b:7e9b |
+-----------+-------------------+--------------------------+

Please note most tasks should be handled through the GUI.
You can access the GUI by typing one of the above IPs in to your web browser.
For support please visit:
    http://www.freepbx.org/support-and-professional-services

[root@freepbx ~]# asterisk -rvv
Asterisk 18.16.0, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 18.16.0 currently running on freepbx (pid = 2425)
<--- Received SIP request (963 bytes) from UDP:10.100.0.1:59923 --->
INVITE sip:316********@10.2.0.20 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.1:59923;rport;branch=z9hG4bKPj32bd55c04c7244638f61b04b2effdff5
Max-Forwards: 70
From: "Marc" <sip:[email protected]>;tag=7f780b779fa546a98edce084c8da5263
To: <sip:316********@10.2.0.20>
Contact: "Marc" <sip:[email protected]:59923;ob>
Call-ID: 1c2b0d30286342fb8d8b3bf7bdf8bfe1
CSeq: 17292 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.21.3
Content-Type: application/sdp
Content-Length:   333

v=0
o=- 3916240435 3916240435 IN IP4 10.100.0.1
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4018 RTP/AVP 8 0 101
c=IN IP4 10.100.0.1
b=TIAS:64000
a=rtcp:4019 IN IP4 10.100.0.1
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:877933626 cname:05276dd402a16e30

<--- Transmitting SIP response (554 bytes) to UDP:10.100.0.1:59923 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.100.0.1:59923;rport=59923;received=10.100.0.1;branch=z9hG4bKPj32bd55c04c7244638f61b04b2effdff5
Call-ID: 1c2b0d30286342fb8d8b3bf7bdf8bfe1
From: "Marc" <sip:[email protected]>;tag=7f780b779fa546a98edce084c8da5263
To: <sip:316********@10.2.0.20>;tag=z9hG4bKPj32bd55c04c7244638f61b04b2effdff5
CSeq: 17292 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1707208435/c2545aa5071fbfbb7dffb2e70a37b040",opaque="149e43db650c700c",algorithm=MD5,qop="auth"
Server: FPBX-16.0.33(18.16.0)
Content-Length:  0


<--- Received SIP request (379 bytes) from UDP:10.100.0.1:59923 --->
ACK sip:316********@10.2.0.20 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.1:59923;rport;branch=z9hG4bKPj32bd55c04c7244638f61b04b2effdff5
Max-Forwards: 70
From: "Marc" <sip:[email protected]>;tag=7f780b779fa546a98edce084c8da5263
To: <sip:316********@10.2.0.20>;tag=z9hG4bKPj32bd55c04c7244638f61b04b2effdff5
Call-ID: 1c2b0d30286342fb8d8b3bf7bdf8bfe1
CSeq: 17292 ACK
Content-Length:  0


<--- Received SIP request (1258 bytes) from UDP:10.100.0.1:59923 --->
INVITE sip:316********@10.2.0.20 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.1:59923;rport;branch=z9hG4bKPj09fc50a07b1d433b9a96082b9a19a6b6
Max-Forwards: 70
From: "Marc" <sip:[email protected]>;tag=7f780b779fa546a98edce084c8da5263
To: <sip:316********@10.2.0.20>
Contact: "Marc" <sip:[email protected]:59923;ob>
Call-ID: 1c2b0d30286342fb8d8b3bf7bdf8bfe1
CSeq: 17293 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.21.3
Authorization: Digest username="100", realm="asterisk", nonce="1707208435/c2545aa5071fbfbb7dffb2e70a37b040", uri="sip:316********@10.2.0.20", response="7d11127f585563cfe09fffc530b6bcde", algorithm=MD5, cnonce="fdc7d27f03ea4042acd7945cfa60f5e3", opaque="149e43db650c700c", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   333

v=0
o=- 3916240435 3916240435 IN IP4 10.100.0.1
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4018 RTP/AVP 8 0 101
c=IN IP4 10.100.0.1
b=TIAS:64000
a=rtcp:4019 IN IP4 10.100.0.1
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:877933626 cname:05276dd402a16e30

<--- Transmitting SIP response (356 bytes) to UDP:10.100.0.1:59923 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.100.0.1:59923;rport=59923;received=10.100.0.1;branch=z9hG4bKPj09fc50a07b1d433b9a96082b9a19a6b6
Call-ID: 1c2b0d30286342fb8d8b3bf7bdf8bfe1
From: "Marc" <sip:[email protected]>;tag=7f780b779fa546a98edce084c8da5263
To: <sip:316********@10.2.0.20>
CSeq: 17293 INVITE
Server: FPBX-16.0.33(18.16.0)
Content-Length:  0


  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Audio CoS mark 5
<--- Transmitting SIP response (838 bytes) to UDP:10.100.0.1:59923 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.100.0.1:59923;rport=59923;received=10.100.0.1;branch=z9hG4bKPj09fc50a07b1d433b9a96082b9a19a6b6
Call-ID: 1c2b0d30286342fb8d8b3bf7bdf8bfe1
From: "Marc" <sip:[email protected]>;tag=7f780b779fa546a98edce084c8da5263
To: <sip:316********@10.2.0.20>;tag=ac87e290-d62c-4c16-a634-c4940da4117e
CSeq: 17293 INVITE
Server: FPBX-16.0.33(18.16.0)
Contact: <sip:10.2.0.20:5060>
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Content-Type: application/sdp
Content-Length:   255

v=0
o=- 3916240435 3916240437 IN IP4 10.2.0.20
s=Asterisk
c=IN IP4 10.2.0.20
t=0 0
m=audio 11158 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (846 bytes) from UDP:10.100.0.1:59923 --->
UPDATE sip:10.2.0.20:5060 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.1:59923;rport;branch=z9hG4bKPj1d2ea1b296b04a9aa06a8014e218a592
Max-Forwards: 70
From: "Marc" <sip:[email protected]>;tag=7f780b779fa546a98edce084c8da5263
To: <sip:316********@10.2.0.20>;tag=ac87e290-d62c-4c16-a634-c4940da4117e
Contact: "Marc" <sip:[email protected]:59923;ob>
Call-ID: 1c2b0d30286342fb8d8b3bf7bdf8bfe1
CSeq: 17294 UPDATE
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length:   309

v=0
o=- 3916240435 3916240436 IN IP4 10.100.0.1
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4018 RTP/AVP 0 101
c=IN IP4 10.100.0.1
b=TIAS:64000
a=rtcp:4019 IN IP4 10.100.0.1
a=ssrc:877933626 cname:05276dd402a16e30
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

<--- Transmitting SIP response (901 bytes) to UDP:10.100.0.1:59923 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.100.0.1:59923;rport=59923;received=10.100.0.1;branch=z9hG4bKPj1d2ea1b296b04a9aa06a8014e218a592
Call-ID: 1c2b0d30286342fb8d8b3bf7bdf8bfe1
From: "Marc" <sip:[email protected]>;tag=7f780b779fa546a98edce084c8da5263
To: <sip:316********@10.2.0.20>;tag=ac87e290-d62c-4c16-a634-c4940da4117e
CSeq: 17294 UPDATE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: <sip:10.2.0.20:5060>
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: FPBX-16.0.33(18.16.0)
Content-Type: application/sdp
Content-Length:   231

v=0
o=- 3916240435 3916240438 IN IP4 10.2.0.20
s=Asterisk
c=IN IP4 10.2.0.20
t=0 0
m=audio 11158 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (545 bytes) to UDP:10.100.0.1:59923 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.100.0.1:59923;rport=59923;received=10.100.0.1;branch=z9hG4bKPj09fc50a07b1d433b9a96082b9a19a6b6
Call-ID: 1c2b0d30286342fb8d8b3bf7bdf8bfe1
From: "Marc" <sip:[email protected]>;tag=7f780b779fa546a98edce084c8da5263
To: <sip:316********@10.2.0.20>;tag=ac87e290-d62c-4c16-a634-c4940da4117e
CSeq: 17293 INVITE
Server: FPBX-16.0.33(18.16.0)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Reason: Q.850;cause=34
Content-Length:  0


<--- Received SIP request (374 bytes) from UDP:10.100.0.1:59923 --->
ACK sip:316********@10.2.0.20 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.1:59923;rport;branch=z9hG4bKPj09fc50a07b1d433b9a96082b9a19a6b6
Max-Forwards: 70
From: "Marc" <sip:[email protected]>;tag=7f780b779fa546a98edce084c8da5263
To: <sip:316********@10.2.0.20>;tag=ac87e290-d62c-4c16-a634-c4940da4117e
Call-ID: 1c2b0d30286342fb8d8b3bf7bdf8bfe1
CSeq: 17293 ACK
Content-Length:  0


  == Spawn extension (from-internal, 316********, 7) exited non-zero on 'PJSIP/100-0000007e'
  == Spawn extension (macro-hangupcall, s, 5) exited non-zero on 'PJSIP/100-0000007e' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/100-0000007e'
<--- Received SIP request (509 bytes) from UDP:46.19.210.19:5060 --->
OPTIONS sip:[email protected].***.***:5060;line=trlzcyn SIP/2.0
Via: SIP/2.0/UDP 46.19.210.19;branch=z9hG4bK1be5.0e3306cddd697a634db88d9fd011d28c.0
Via: SIP/2.0/UDP 46.19.210.17;received=46.19.210.17;branch=z9hG4bKz5P4Ha0p;rport=5060
From: sip:[email protected].***.***:5060;line=trlzcyn;tag=registrar
To: sip:[email protected].***.***:5060;line=trlzcyn
CSeq: 10 OPTIONS
Call-ID: 13-61E01B8B-65C1EF01000F3A3D-881FB6C0
Max-Forwards: 69
User-Agent: DIDWW Y SBC node
Contact: <sip:46.19.210.17:5060;transport=udp>
Content-Length: 0


<--- Transmitting SIP response (970 bytes) to UDP:46.19.210.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 46.19.210.19;rport=5060;received=46.19.210.19;branch=z9hG4bK1be5.0e3306cddd697a634db88d9fd011d28c.0
Via: SIP/2.0/UDP 46.19.210.17;rport=5060;received=46.19.210.17;branch=z9hG4bKz5P4Ha0p
Call-ID: 13-61E01B8B-65C1EF01000F3A3D-881FB6C0
From: <sip:[email protected].***.***>;tag=registrar;line=trlzcyn
To: <sip:[email protected].***.***>;tag=z9hG4bK1be5.0e3306cddd697a634db88d9fd011d28c.0;line=trlzcyn
CSeq: 10 OPTIONS
Accept: application/sdp, application/simple-message-summary, application/dialog-info+xml, application/xpidf+xml, application/cpim-pidf+xml, application/pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, message/sipfrag;version=2.0
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: FPBX-16.0.33(18.16.0)
Content-Length:  0

danishhafeez · February 6, 2024, 12:02pm

SIp/2.0 401 unauthorized i think this error indicating that your DID number is not accepted by your trunk provider as caller id. so contact to your provider to add fixed did to your trunk for making outbound calls.
i hope it will be useful for you.

SparkyRih · February 6, 2024, 12:10pm

DIDWW allows me to change those settings in their control panel, and the trunk on their side has been setujp to accept any DIDS. Also, I’ve provided them whit this exact log, but they are referring me to FreePBX support now…

BlazeStudios · February 6, 2024, 1:24pm

This capture is between endpoint 100 and the PBX. The INVITE is from a MicroSIP client to the PBX, the 401 challenge (which is part of the user/pass auth process) is from the PBX back to MicroSIP and the 503 error is the PBX telling MicroSIP that the service is unavailable.

There is nothing in this capture that shows FreePBX to DIDWW or what response/error was presented by DIDWW for the call. The only things from DIDWW in this capture is them sending OPTIONS to the PBX and the PBX sending a 200 OK to said OPTIONS.

There needs to be a capture that shows the call between the PBX and DIDWW in order to see what is actually happening.

Stewart1 · February 6, 2024, 1:37pm

As @BlazeStudios says. But also, please paste the relevant data from the Asterisk log (/var/log/asterisk/full, not the console output) at pastebin.com and post the link here. If you are too new to post links, just post the last eight characters of the URL.

jcolp · February 6, 2024, 1:49pm

I adjusted the settings so new users can post a reasonable number of links which should cover common usage.

SparkyRih · February 6, 2024, 2:49pm

I’ve pasted the last part of my full log file: [2024-02-06 14:41:25] WARNING[719] res_pjsip_outbound_registration.c: No respo - Pastebin.com
I made a phone call with my mobile phone to my Asterisk server, which has an IVR, if I press 2 it should forward the call back to my mobile phone.
I also made another call via MicroSIP after that.

BlazeStudios · February 6, 2024, 3:03pm

You need to contact DIDWW support and find out why they are giving you

[2024-02-06 14:44:05] VERBOSE[2464] res_pjsip_logger.c: <--- Received SIP response (658 bytes) from UDP:46.19.210.19:5060 --->
SIP/2.0 403 Forbidden

They challenged you for auth, you sent back a new INVITE with auth details and then they replied with 403 Forbidden. There could be a few reasons why they did that, all of which they would be able to tell you.

SparkyRih · February 6, 2024, 4:27pm

Turns out something went wrong in their whitelisting of my IP address. It’s all fixed now…

system · March 8, 2024, 4:28pm

This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.