I cannot get Voxbone outbound trunk to work with our 3CX PBX

[gcnltd]

This is this the error messages I get:

07/23/2019 12:38:13 PM - Endpoint Extn:xx has refreshed contact <sip:[email protected]:5063>
07/23/2019 12:37:19 PM - Endpoint Extn:xx has refreshed contact <sip:[email protected]:5063>
07/23/2019 12:37:12 PM - Endpoint Extn:xx has refreshed contact <sip:[email protected]:5060>
07/23/2019 12:36:24 PM - Endpoint Extn:xx has refreshed contact <sip:[email protected]:5063>
07/23/2019 12:35:30 PM - Endpoint Extn:xx has refreshed contact <sip:[email protected]:5063>
07/23/2019 12:35:16 PM - Endpoint Extn:xx has refreshed contact <sip:[email protected]:5060>
07/23/2019 12:35:12 PM - Stop call record for leg L:385.1[Extn:xx]
07/23/2019 12:35:12 PM - Removing leg L:385.1[Extn:xx]
07/23/2019 12:35:12 PM - L:385.1[Extn:00]: Terminating targets, reason: SIP ;cause=200 ;text="User hangup"
07/23/2019 12:35:12 PM - Leg L:385.1[Extn:xx] is terminated: Cause: BYE from 127.0.0.1:5080
07/23/2019 12:35:12 PM - Call(C:385), Extn:xx on entry: DlgInfo(385-4687/Terminated / I)
07/23/2019 12:35:12 PM - Notify dialog-info: Extn:xx: sip:[email protected]:5060;rinstance=1-2019a091258b4396b9c4232c71af0c72, Call(C:385)
07/23/2019 12:35:12 PM - L:385.1[Extn:xx] got Terminated Recv Req BYE from 127.0.0.1:5080 tid=Pj1fa1c1a850f64d40a3eb8785b95ea7f7 Call-ID=8d2fd2cd32104dfa80c895e1adfe608b:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-524287-2---Pj1fa1c1a850f64d40a3eb8785b95ea7f7;rport=5080
Via: SIP/2.0/UDP [::ffff:xx.xx.xx.xx]:61627;branch=z9hG4bK-524287-1---tunneltid;rport;tnlid=clnt.1-2019a091258b4396b9c4232c71af0c72
Via: SIP/2.0/TCP 1xx.1xx.x.xxx:50195;rport;branch=z9hG4bKPj1fa1c1a850f64d40a3eb8785b95ea7f7;alias
Max-Forwards: 69
Record-Route: <sip:[email protected]:5080;user=proxy;uri=clnt.1-2019a091258b4396b9c4232c71af0c72>
To: <sip:[email protected];tag3cx=52e3277d0d47444c9df41aadf3a139bc>;tag=7c013a12
From: "User Name" <sip:[email protected]>;tag=e60365e1db5341a9b8be5a6d15d61d0e
Call-ID: 8d2fd2cd32104dfa80c895e1adfe608b
CSeq: 476 BYE
User-Agent: 3CXPhone for Windows 16.0.1.81
Content-Length: 0

 

[JohnS_3CX]

Hi gcnltd

Have you contacted your provider first to ask their requirements before setting up the trunk?

You will have to provide some basic info

 

[gcnltd]

Hi gcnltd

Have you contacted your provider first to ask their requirements before setting up the trunk?

You will have to provide some basic info

Hello John,
Yes I did and this was their response: "We do not have a specific template or guide for how to configure 3CX PBX to work with Voxbone outbound "
This is an example of their config:

An example of the first INVITE (The X's are your IP)

U XXX.XXX.XXX.XXX:5060 -> 81.201.82.250:5060
INVITE sip:[email protected] SIP/2.0.
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK58f69613.
Max-Forwards: 70.
From: "+3228081151" <sip:[email protected]>;tag=as6243bb32.
To: <sip:[email protected]>.
Contact: <sip:[email protected]>.
Call-ID: [email protected].
CSeq: 102 INVITE.
User-Agent: PBX.
Date: Wed, 15 Jan 2014 12:58:06 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 247.

 

[JohnS_3CX]

And does your plan include Outbound Calls?

 

[gcnltd]

Yes, it does, I have inbound/outbound calls.

 

[JohnS_3CX]

Ok great, in this case your provider gave the format they expect to see when you send an invite for an outbound call.

In case you they don't have other guidelines, what you can do is run a capture while making a call and see what format your PBX is sending. If there are any fields that don't agree, you need to make some changes (i.e number formatting or other) so they see what they expect in order to allow the call through.

Then simply make a new call and see that the format appears as they expect. Usually if you contact a provider their support can tell you why they reject a call and can help you make the necessary adjustments by highlighting the changes needed in the invite

 

[gcnltd]

This is what I got from the capture:

2019/07/23 15:33:09.092|694|0016|Verb|CHR: Adding event: 2019-07-23 15:33:01.277|IncomingCall|0000016C1F3FF65C_4549|3dd4e949e033/Participant DN.00/Extension dn-name='User' epname='Ext.00'

3dd4e949e033/Participant DN.00/Extension dn-name='User' epname='Ext.00'

disp_name=User
number=00
sip.contact="User" <sip:[email protected]:5060;rinstance=1-2019a091258b4396b9c4232c71af0c72>
sip.from="User"<sip:[email protected]>;tag=38e9403ce5f646b8a38abf05e2c7fd15
sip.rl_uri=sip:[email protected]:5060;transport=UDP;tag3cx=51f37fa39da14c55a6e8f6d6d8e50542
sip.src_addr=127.0.0.1:5080
sip.to=<sip:[email protected];tag3cx=51f37fa39da14c55a6e8f6d6d8e50542>
sip.user_ag=3CXPhone for Windows 16.0.1.81
target=+xxxxxxxxxxx

2019/07/23 15:33:09.093|694|0016|Verb|CHR: Adding event: 2019-07-23 15:33:01.278|TargetAdded|0000016C1F3FF65C_4549|6038cd56a416/Target DN.+xxxxxxxxxxx/LineSet dn-name='' epname='IL_Outbound Route_Conversion'

6038cd56a416/Target DN.+xxxxxxxxxxx/LineSet dn-name='' epname='IL_Outbound Route_Conversion'

ep.dialing=+xxxxxxxxxxx
ep.originator.caller_num=
ep.originator.dial=+xxxxxxxxxxx
orig.dest=+xxxxxxxxxxx
orig.party.id=3dd4e949e033

2019/07/23 15:33:09.093|694|0016|Verb|CHR: Adding event: 2019-07-23 15:33:01.558|RouteConnected|0000016C1F3FF65C_4549|01e1835d0d66/Route DN.EndCall/EndCall dn-name='' epname='EndCall'

01e1835d0d66/Route DN.EndCall/EndCall dn-name='' epname='EndCall'

ep.contact=sip:[email protected]:5483;rinstance=0ec4ec859ca4e18c
ep.dialing=EndCall
ep.disp_name=User
ep.originator.caller_num=
ep.originator.dial=EndCall
ep.to=
orig.party.id=3dd4e949e033
target.party.id=01e1835d0d66
target.party.type=Target

2019/07/23 15:33:09.093|694|0016|Verb|CHR: Adding event: 2019-07-23 15:33:01.559|Recording|0000016C1F3FF65C_4549|3dd4e949e033/None

3dd4e949e033/None

rec.filename=00/[User]_00-+xxxxxxxxxxx_20190723143301(4549).wav
rec.ison.int=1

2019/07/23 15:33:09.093|694|0016|Verb|CHR: Adding event: 2019-07-23 15:33:08.622|PartyRemoved|0000016C1F3FF65C_4549|01e1835d0d66/None

01e1835d0d66/None

2019/07/23 15:33:09.093|694|0016|Verb|CHR: Adding event: 2019-07-23 15:33:08.623|Recording|0000016C1F3FF65C_4549|3dd4e949e033/None

3dd4e949e033/None

rec.filename=00/[User]_00-+xxxxxxxxxxx_20190723143301(4549).wav
rec.ison.int=0

2019/07/23 15:33:09.093|694|0016|Verb|CHR: Adding event: 2019-07-23 15:33:08.625|PartyRemoved|0000016C1F3FF65C_4549|3dd4e949e033/None

3dd4e949e033/None

2019/07/23 15:33:09.093|694|0016|Verb|CHR: Adding event: 2019-07-23 15:33:09.092|Disconnected|0000016C1F3FF65C_4549|Party.???

Party.???

replaced_by.chid=

2019/07/23 15:33:09.093|694|0016|Trac|CHR: Target has changed: Tgt 6038cd56a416{+xxxxxxxxxxx}[Ok/Unknown]
2019/07/23 15:33:09.093|694|0016|Erro|CHR: Targeting segment is not found for target 01e1835d0d66
2019/07/23 15:33:09.093|694|0016|Trac|CHR: Saving joined call:
2019/07/23 15:33:09.093|694|0016|Trac|CHR: Executing SQL: INSERT into cl_calls (id,start_time,end_time,is_answered,ringing_dur,talking_dur)
VALUES (DEFAULT,'2019-07-23 14:33:01','2019-07-23 14:33:09',FALSE,'00:00:00.281382','00:00:07.063757')
RETURNING id
2019/07/23 15:33:09.094|694|0029|Trac|CHR: * Participant 3dd4e949e033/Participant DN.00/Extension dn-name='User' epname='Ext.00'; answered at 7/23/19 3:33:01 PM
2019/07/23 15:33:09.094|694|0029|Trac|CHR: Executing SQL: INSERT into cl_party_info (id,dn_type,dn,caller_number,display_name,dn_class,firstlastname,did_number)
VALUES (DEFAULT,0,'00','Ext.00','User',0,'User','')
RETURNING id
2019/07/23 15:33:09.094|694|0029|Trac|CHR: Executing SQL: INSERT into cl_participants (id,call_id,info_id,role,is_inbound,end_status,forward_reason,failure_reason,start_time,answer_time,end_time,recording_url,billing_group)
VALUES (DEFAULT,71036,187812,1,TRUE,1,0,1,'2019-07-23 14:33:01','2019-07-23 14:33:01','2019-07-23 14:33:08','00/[User]_00-+xxxxxxxxxxx_20190723143301(4549).wav','')
RETURNING id
2019/07/23 15:33:09.094|694|0029|Trac|CHR: * Target 6038cd56a416/Target DN.+xxxxxxxxxxx/LineSet dn-name='+xxxxxxxxxxx' epname='+xxxxxxxxxxx'; originated by 3dd4e949e033
2019/07/23 15:33:09.094|694|0029|Trac|CHR: Executing SQL: INSERT into cl_party_info (id,dn_type,dn,caller_number,display_name,dn_class,firstlastname,did_number)
VALUES (DEFAULT,13,'+xxxxxxxxxxx','+xxxxxxxxxxx','+xxxxxxxxxxx',0,'','')
RETURNING id
2019/07/23 15:33:09.095|694|0029|Trac|CHR: Executing SQL: INSERT into cl_participants (id,call_id,info_id,role,is_inbound,end_status,forward_reason,failure_reason,start_time,end_time)
VALUES (DEFAULT,71036,187813,2,FALSE,1,0,1,'2019-07-23 14:33:01','2019-07-23 14:33:01')
RETURNING id
2019/07/23 15:33:09.095|694|0029|Trac|CHR: * Participant 01e1835d0d66/Participant DN.EndCall/EndCall dn-name='+xxxxxxxxxxx' epname='EndCall'; answered at 7/23/19 3:33:01 PM
2019/07/23 15:33:09.095|694|0029|Trac|CHR: Executing SQL: INSERT into cl_party_info (id,dn_type,dn,caller_number,display_name,dn_class,firstlastname,did_number)
VALUES (DEFAULT,12,'EndCall','EndCall','+xxxxxxxxxxx',0,'','')
RETURNING id
2019/07/23 15:33:09.095|694|0029|Trac|CHR: Executing SQL: INSERT into cl_participants (id,call_id,info_id,role,is_inbound,end_status,forward_reason,failure_reason,start_time,answer_time,end_time,billing_group)
VALUES (DEFAULT,71036,187814,1,FALSE,0,0,1,'2019-07-23 14:33:01','2019-07-23 14:33:01','2019-07-23 14:33:08','')
RETURNING id
2019/07/23 15:33:09.096|694|0029|Trac|CHR: Saving Pbx call 0000016C1F3FF65C_4549
2019/07/23 15:33:09.096|694|0029|Info|CHR: #1: Seg[1/#1] 3dd4e949e033(Ext.00)=>6038cd56a416(+xxxxxxxxxxx) (15:33:01.277..15:33:01.558) Ringing Connected
2019/07/23 15:33:09.096|694|0029|Trac|CHR: Executing SQL: INSERT into cl_segments (id,call_id,seq_order,seq_group,src_part_id,dst_part_id,start_time,end_time,type,action_id)
VALUES (DEFAULT,71036,1,1,187816,187817,'2019-07-23 14:33:01','2019-07-23 14:33:01',1,1)
RETURNING id
2019/07/23 15:33:09.096|694|0029|Info|CHR: #2: Seg[1/#2] 3dd4e949e033(Ext.00)=>01e1835d0d66(EndCall) (15:33:01.558..15:33:08.622) Talking TerminatedByDst
2019/07/23 15:33:09.096|694|0029|Trac|CHR: Executing SQL: INSERT into cl_segments (id,call_id,seq_order,seq_group,src_part_id,dst_part_id,start_time,end_time,type,action_id)
VALUES (DEFAULT,71036,2,1,187816,187818,'2019-07-23 14:33:01','2019-07-23 14:33:08',2,6)
RETURNING id
2019/07/23 15:33:09.097|694|0029|Trac|CHR: Executing SQL: INSERT into myphone_callhistory_v14 (idmpch14,call_id,calltype,dnowner,party_dn,party_dntype,party_name,party_callerid,via_role,via_dn,via_dntype,via_dnname,via_callerid,start_time,established_time,end_time,end_status,end_status_hint)
VALUES (DEFAULT,71036,3,'00','EndCall',0,'+xxxxxxxxxxx','EndCall',3,'+xxxxxxxxxxx',128,'+xxxxxxxxxxx','+xxxxxxxxxxx','2019-07-23 14:33:01','2019-07-23 14:33:01','2019-07-23 14:33:08',0,0)
RETURNING idmpch14

 

[leejor]

Looking quickly through the log, I didn't see a specific error response from the provider, I might have missed it. that is going to give a clue as to what they don't like.

Is the format of the number you are sending, correct (a leading +, reather than all digits)?
Do you receive incoming calls?
Have they told you what they see when you attempt a call?

 

[gcnltd]

The format is:
Format: E.164 CLI
Prefix: +{E164} [email protected]:5060

They have a sample config for Asterisk

Asterisk
Add the following to /etc/asterisk/sip.conf:

[
"#{server.machine_name}"
]
"host="/>
"type=friend"
"insecure=port,invite"
"context=your-context"
"canreinvite=no"
"qualify=no"

Replacing 'your-context' with an existing context in /etc/asterisk/extensions.conf.
The 'insecure = very' allows all calls from this IP address.
In extensions.conf, then add a line to route the DIDs as you wish. For example, to route 1234567890 to SIP phone 100:

exten => 1234567890, 1, Dial(SIP/100)

Once you have edited sip.conf and extensions.conf, load the Asterisk console, and type the following commands:

sip reload
extensions reload

Users of Asterisk behind a NAT must add the 'externip' line in their "sip.conf" file :

externip=<External IP Address> ; e.g. externip=11.22.33.44 where 11.22.33.44 is the external IP address of the router.
If the external IP address is dynamic (e.g. a DSL provider handing out a different IP address every day) you will need to use a Dynamic DNS client.
Asterisk servers with both internal and external SIP clients will need to add the 'localnet' line in their "sip.conf" file :

localnet=<internal range with mask> ; e.g. localnet=192.168.1.0/255.255.255.0 where 192.168.1.0 is the local internal range

 

[leejor]

What you have posted, not only deals with Asterisk, (which if you haven't used before, isn't all that helpful), but much of it is on how to route incoming calls.

I'm completely not clear if the number format with the leading + is what they send, or expect. If you are certain that they expect that, then so be it.

There should still be a distinct error/failure message when you sent a call to them. call to (IP of provider or URL)...Failed

If you do a google search (3CX Voxbone), you will come across a few older posts about the provider, in the forum. Unfortunately, some never did get a response. Some are quite old but may be of use.

https://www.3cx.com/community/threads/voxbone-trunk-anyone-got-this-working.21503/

 

[gcnltd]

I configured inbound route and tested it and I am receiving calls just fine on the same trunk, I just can't seem to get the outbound route to work. You can see from the capture below.

07/24/2019 10:32:32 AM - L:6057.2[Extn:xx] got Connected.UAC Recv 200/INVITE from 127.0.0.1:5080 tid=b43e664b1685152b Call-ID=4sZTnexfBjZ5Sqb9p04XMw..:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;rport=5060;received=xx.xxx.xxx.xxx;branch=z9hG4bK-524287-1---b43e664b1685152b
Record-Route: <sip:[email protected]:5080;user=proxy;uri=clnt.1-c7ded83a489c4fabacd018f9ca57f53b>
Contact: "User" <sip:[email protected]:5060;rinstance=1-c7ded83a489c4fabacd018f9ca57f53b>
To: <sip:[email protected]>;tag=a3deccbec81843769f6cb106af041d8a
From: "+xxxxxxxxxx" <sip:[email protected]:5060;nf=e>;tag=bf6b8337
Call-ID: 4sZTnexfBjZ5Sqb9p04XMw..
CSeq: 1 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Type: application/sdp
Supported: replaces, 100rel, timer, norefersub
Content-Length: 269

v=0
o=- 3772953146 3772953147 IN IP4 xxx.xxx.x.xxx
s=pjmedia
c=IN IP4 127.0.0.1
b=AS:84
t=0 0
a=X-nat:0
m=audio 14978 RTP/AVP 0 101
c=IN IP4 127.0.0.1
b=TIAS:64000
b=AS:84
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
07/24/2019 10:32:32 AM - Call(C:6057), Extn:xx on entry: DlgInfo(6057-1782/Confirmed / R)
07/24/2019 10:32:32 AM - Notify dialog-info: Extn:x: sip:[email protected]:5060;rinstance=1-c7ded83a489c4fabacd018f9ca57f53b, Call(C:6057)
07/24/2019 10:32:32 AM - Remote SDP is set for leg L:6057.2[Extn:xx]
07/24/2019 10:32:32 AM - Answer SDP arrived on Leg L:6057.2[Extn:xx];
OnAnswer Recv 200/INVITE from 127.0.0.1:5080 tid=b43e664b1685152b Call-ID=4sZTnexfBjZ5Sqb9p04XMw..:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;rport=5060;received=xx.xxx.xxx.xxx;branch=z9hG4bK-524287-1---b43e664b1685152b
Record-Route: <sip:[email protected]:5080;user=proxy;uri=clnt.1-c7ded83a489c4fabacd018f9ca57f53b>
Contact: "User" <sip:[email protected]:5060;rinstance=1-c7ded83a489c4fabacd018f9ca57f53b>
To: <sip:[email protected]>;tag=a3deccbec81843769f6cb106af041d8a
From: "+xxxxxxxxxx" <sip:[email protected]:5060;nf=e>;tag=bf6b8337
Call-ID: 4sZTnexfBjZ5Sqb9p04XMw..
CSeq: 1 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Type: application/sdp
Supported: replaces, 100rel, timer, norefersub
Content-Length: 269

v=0
o=- 3772953146 3772953147 IN IP4 xxx.xxx.x.xxx
s=pjmedia
c=IN IP4 127.0.0.1
b=AS:84
t=0 0
a=X-nat:0
m=audio 14978 RTP/AVP 0 101
c=IN IP4 127.0.0.1
b=TIAS:64000
b=AS:84
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

07/24/2019 10:32:32 AM - L:6057.2[Extn:xx] Sending: OnSendReq Send Req ACK from 0.0.0.0:0 tid=e241543c1c269d17 Call-ID=4sZTnexfBjZ5Sqb9p04XMw..:
ACK sip:[email protected]:5060;rinstance=1-c7ded83a489c4fabacd018f9ca57f53b SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---e241543c1c269d17;rport
Max-Forwards: 70
Route: <sip:[email protected]:5080;user=proxy;uri=clnt.1-c7ded83a489c4fabacd018f9ca57f53b>
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>;tag=a3deccbec81843769f6cb106af041d8a
From: "+xxxxxxxxxx" <sip:[email protected]:5060;nf=e>;tag=bf6b8337
Call-ID: 4sZTnexfBjZ5Sqb9p04XMw..
CSeq: 1 ACK
Content-Length: 0

 

[leejor]

Route: <sip:[email protected]:5080;user=proxy;uri=clnt.1-c7ded83a489c4fabacd018f9ca57f53b>

How is a Bridge involed?

 

[gcnltd]

I do not know, the only bridge on the system is the WebMeeting bridge.

 

[Saqqara]

Can you show us your outbound rules

 

[leejor]

It really does not appears that the call is going to your provider. As I said before, you should be seeing a rejection, of some sort from them, if the call is failing becuase of something they don't like.

 

[gcnltd]

This in my outbound routes I am currently using 2 different trunk providers but only one work.
Voxbone is the one I have issues with the other is working fine no issues

 

[gcnltd]

This is from my 1st provider that is working fine with no issues.

07/24/2019 4:59:59 PM - L:10510.1[Extn:31] got Confirmed Recv Req ACK from 127.0.0.1:5080 tid=Pj6862248aedf0490aa7aebeabc211b64a Call-ID=da5bf9b9a79c45f9818ebd18210e3348:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-524287-2---Pj6862248aedf0490aa7aebeabc211b64a;rport=5080
Via: SIP/2.0/UDP [::ffff:18.130.38.145]:61280;branch=z9hG4bK-524287-1---tunneltid;rport;tnlid=clnt.1-f11efbddde5a4107a6af1b917fd563a8
Via: SIP/2.0/TCP 192.168.1.180:50195;rport;branch=z9hG4bKPj6862248aedf0490aa7aebeabc211b64a;alias
Max-Forwards: 69
Record-Route: <sip:[email protected]:5080;user=proxy;uri=clnt.1-f11efbddde5a4107a6af1b917fd563a8>
To: <sip:[email protected];tag3cx=748ed708c8b94313b7956b381a21f259>;tag=a3bf3a38
From: "User" <sip:[email protected]>;tag=fdb0c23bc0d84e96b0b23c57d20bcf1d
Call-ID: da5bf9b9a79c45f9818ebd18210e3348
CSeq: 19925 ACK
Content-Length: 0

07/24/2019 4:59:59 PM - ConnCfmd from "User"<sip:[email protected]>;tag=fdb0c23bc0d84e96b0b23c57d20bcf1d to <sip:[email protected];tag3cx=748ed708c8b94313b7956b381a21f259>;tag=a3bf3a38
07/24/2019 4:59:59 PM - L:10510.1[Extn:31] Sending: OnSendResp Send 200/INVITE from 0.0.0.0:0 tid=Pj6bd3a6a37dde421a837338b7895f03ef Call-ID=da5bf9b9a79c45f9818ebd18210e3348:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-524287-2---Pj6bd3a6a37dde421a837338b7895f03ef;rport=5080
Via: SIP/2.0/UDP [::ffff:18.130.38.145]:61280;branch=z9hG4bK-524287-1---tunneltid;rport;tnlid=clnt.1-f11efbddde5a4107a6af1b917fd563a8
Via: SIP/2.0/TCP 192.168.1.180:50195;rport;branch=z9hG4bKPj6bd3a6a37dde421a837338b7895f03ef;alias
Record-Route: <sip:[email protected]:5080;user=proxy;uri=clnt.1-f11efbddde5a4107a6af1b917fd563a8>
Require: timer
Contact: <sip:[email protected]:5060>
To: <sip:[email protected];tag3cx=748ed708c8b94313b7956b381a21f259>;tag=a3bf3a38
From: "User"<sip:[email protected]>;tag=fdb0c23bc0d84e96b0b23c57d20bcf1d
Call-ID: da5bf9b9a79c45f9818ebd18210e3348
CSeq: 19925 INVITE
Session-Expires: 1800;refresher=uac
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE
Content-Type: application/sdp
Supported: replaces, timer
User-Agent: 3CXPhoneSystem 16.0.2.910 (910)
Content-Length: 382

v=0
o=3cxPS 768159481069568 3315191638917121 IN IP4 127.0.0.1
s=3cxPS Audio call
c=IN IP4 127.0.0.1
t=0 0
m=audio 7150 RTP/AVP 3 18 0 8 9 96
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
m=video 0 RTP/AVP 31
c=IN IP4 127.0.0.1

07/24/2019 4:59:59 PM - SendMsg from <sip:[email protected];tag3cx=748ed708c8b94313b7956b381a21f259>;tag=a3bf3a38 to "User"<sip:[email protected]>;tag=fdb0c23bc0d84e96b0b23c57d20bcf1d
07/24/2019 4:59:59 PM - Replaced Via with xxx.xxx.xx.xx:5060
07/24/2019 4:59:59 PM - Replaced Via with xxx.xxx.xx.xx:5060
07/24/2019 4:59:58 PM - [CM503007]: Call(C:10510): Line:10003>>xxxxxxxxxxxx has joined, contact <sip:[email protected]:5060>
07/24/2019 4:59:58 PM - [CM503007]: Call(C:10510): Extn:31 has joined, contact <sip:[email protected]:5060>
07/24/2019 4:59:58 PM - Leg L:10506.1[Extn:40] is terminated: Cause: BYE from local
07/24/2019 4:59:58 PM - L:10506.1[Extn:40] got Terminated Send Req BYE from 0.0.0.0:0 tid=ec7a5a25f2e4557e Call-ID=3c1b75fe246e44a0b6537752664da5c4:
BYE sip:[email protected]:5060;rinstance=1-17a08f13ff484773b5b7c903c69548f7 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---ec7a5a25f2e4557e;rport
Max-Forwards: 70

 

[gcnltd]

It really does not appears that the call is going to your provider. As I said before, you should be seeing a rejection, of some sort from them, if the call is failing because of something they don't like.

I do not think it does either because as soon as I attempt to make a call with this trunk I get a message saying "The destination is not available, registered or there are no routes available. " which all last only 7 seconds.

 

[leejor]

The Activity Log should show a match of the number dialled to the name of the rule being used. There should then be a log showing the call to the provider, and a response from them. the fact that the logs show something to do with a 3CX Bridge, seems suspicious.
As mentioned previously review, or post your outbound rules. Perhaps a rule, above the one you are hoping to use (they are read from top to bottom), is intercepting the calls.

 

[AWS2P]

May be this