Outbound caller ID always different and not my own

SparkyRih · January 20, 2024, 12:38pm

I setup a complete fresh FreePBX distro, used latest stable version.

I’ve setup a SIP Trunk (caller ID = <3188*******>), and I do force it.
I’ve setup an outbound route, left the caller ID blank
I’ve setup an extension, also leaving the caller ID blank.
I also configured inbound routes, this works perfectly fine.

Now for testing I use MicroSIP for a softphone. But whenever I try to make an outbound call, I receive it on my mobile phone with a completely different Caller ID (+44 20 709783**). But that’s not even the strange part, the last 2 digits which I left out, are always random…

What am I doing wrong?

ashcortech · January 20, 2024, 1:55pm

Most likely that’s on your provider side. Check with your provider.

SparkyRih · January 20, 2024, 2:09pm

I’ve created a ticket with my provicder, so waiting on that.

I did call back a couple of those +44 20 709783** numbers, they all answer with the same message, and then “waiting music” is playing. Actually, the same waiting music we have by default in Asterisk…

When I call my own number, I get the speaking clock (which I indeed did setup for my inbound route).

Stewart1 · January 20, 2024, 3:05pm

Most likely, the caller ID you are presenting is not in the format the provider expects (for example, they may need +3188******* or 088*******, or you are not using the expected headers (they may require P-Asserted-Identity or Remote-Party-ID).

If they have not documented this well, a good bet is to send them what they send you. At the Asterisk command prompt, type
pjsip set logger on
or if for some strange reason you have a chan_sip trunk
sip set debug on
Then call in and look at the Asterisk log or the console for the From header. Set up your caller ID in that format. Also, look for the presence of the headers mentioned above and on the Advanced tab for your pjsip trunk, set Send RPID/PAI appropriately. For chan_sip, set
sendrpid=yes
or
sendrpid=pai
in your PEER Details.

When you make a test outbound call, look at the INVITE you are sending, to confirm that the caller ID is correctly set up.

If you still have trouble, check the caller ID sent on a call to a landline. You can try calling
+18004377950
or call +44 20 3026 4621, press # to get pass the record/playback test, then press 7 to hear your caller ID.

The above all assumes that you obtained the +3188 number from a Netherlands provider and you are using that provider to make the test call, to a +316 number obtained from a Netherlands MNO or MVNO. If not the case, please provide details.

SparkyRih · January 21, 2024, 2:04pm

I did try all the suggestions on a trial-and-error basis, but no luck.

This is my debug output

Last login: Sun Jan 21 14:49:55 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 = 2395)
<--- Transmitting SIP request (450 bytes) to UDP:185.118.63.77:5060 --->
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 185.52.***.***:5060;rport;branch=z9hG4bKPj55930a52-e075-4076-99c3-d88af23cc138
From: <sip:[email protected]>;tag=58b90414-06ba-4422-944f-86fec3041b8b
To: <sip:[email protected]>
Contact: <sip:[email protected].***.***:5060>
Call-ID: f2e7ab0b-8615-48ac-92a4-966cef61b53d
CSeq: 20565 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.33(18.16.0)
Content-Length:  0


<--- Received SIP response (391 bytes) from UDP:185.118.63.77:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 185.52.***.***:5060;rport=32015;branch=z9hG4bKPj55930a52-e075-4076-99c3-d88af23cc138;received=185.52.***.***
From: <sip:[email protected]>;tag=58b90414-06ba-4422-944f-86fec3041b8b
To: <sip:[email protected]>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.854d446b
Call-ID: f2e7ab0b-8615-48ac-92a4-966cef61b53d
CSeq: 20565 OPTIONS
Content-Length: 0


<--- Received SIP request (964 bytes) from UDP:10.100.0.1:60400 --->
INVITE sip:316********@10.2.0.20 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.1:60400;rport;branch=z9hG4bKPjd8e1bca83dd94d43806b6fb1b17ab21c
Max-Forwards: 70
From: "Marc" <sip:[email protected]>;tag=7ad1867f5eb84c0c9ef3d1b7a206201d
To: <sip:316********@10.2.0.20>
Contact: "Marc" <sip:[email protected]:60400;ob>
Call-ID: d8bf1b6debf24631b13aa19802aac736
CSeq: 29368 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:   334

v=0
o=- 3914879804 3914879804 IN IP4 10.100.0.1
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4022 RTP/AVP 8 0 101
c=IN IP4 10.100.0.1
b=TIAS:64000
a=rtcp:4023 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:2084714661 cname:30f9506477b00c55

<--- Transmitting SIP response (554 bytes) to UDP:10.100.0.1:60400 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.100.0.1:60400;rport=60400;received=10.100.0.1;branch=z9hG4bKPjd8e1bca83dd94d43806b6fb1b17ab21c
Call-ID: d8bf1b6debf24631b13aa19802aac736
From: "Marc" <sip:[email protected]>;tag=7ad1867f5eb84c0c9ef3d1b7a206201d
To: <sip:316********@10.2.0.20>;tag=z9hG4bKPjd8e1bca83dd94d43806b6fb1b17ab21c
CSeq: 29368 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1705847804/22832631bf86cbce27f4e391936a3fa7",opaque="4dbc9e09256feb4c",algorithm=MD5,qop="auth"
Server: FPBX-16.0.33(18.16.0)
Content-Length:  0


<--- Received SIP request (379 bytes) from UDP:10.100.0.1:60400 --->
ACK sip:316********@10.2.0.20 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.1:60400;rport;branch=z9hG4bKPjd8e1bca83dd94d43806b6fb1b17ab21c
Max-Forwards: 70
From: "Marc" <sip:[email protected]>;tag=7ad1867f5eb84c0c9ef3d1b7a206201d
To: <sip:316********@10.2.0.20>;tag=z9hG4bKPjd8e1bca83dd94d43806b6fb1b17ab21c
Call-ID: d8bf1b6debf24631b13aa19802aac736
CSeq: 29368 ACK
Content-Length:  0


<--- Received SIP request (1259 bytes) from UDP:10.100.0.1:60400 --->
INVITE sip:316********@10.2.0.20 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.1:60400;rport;branch=z9hG4bKPj08b08c02af70447d9f29566e01365d47
Max-Forwards: 70
From: "Marc" <sip:[email protected]>;tag=7ad1867f5eb84c0c9ef3d1b7a206201d
To: <sip:316********@10.2.0.20>
Contact: "Marc" <sip:[email protected]:60400;ob>
Call-ID: d8bf1b6debf24631b13aa19802aac736
CSeq: 29369 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="1705847804/22832631bf86cbce27f4e391936a3fa7", uri="sip:316********@10.2.0.20", response="232b1b5f34f5d31bde95be1b69ece1c4", algorithm=MD5, cnonce="a6b79cef38064a6eb7824e4195076c83", opaque="4dbc9e09256feb4c", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   334

v=0
o=- 3914879804 3914879804 IN IP4 10.100.0.1
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4022 RTP/AVP 8 0 101
c=IN IP4 10.100.0.1
b=TIAS:64000
a=rtcp:4023 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:2084714661 cname:30f9506477b00c55

<--- Transmitting SIP response (356 bytes) to UDP:10.100.0.1:60400 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.100.0.1:60400;rport=60400;received=10.100.0.1;branch=z9hG4bKPj08b08c02af70447d9f29566e01365d47
Call-ID: d8bf1b6debf24631b13aa19802aac736
From: "Marc" <sip:[email protected]>;tag=7ad1867f5eb84c0c9ef3d1b7a206201d
To: <sip:316********@10.2.0.20>
CSeq: 29369 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:10.100.0.1:60400 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.100.0.1:60400;rport=60400;received=10.100.0.1;branch=z9hG4bKPj08b08c02af70447d9f29566e01365d47
Call-ID: d8bf1b6debf24631b13aa19802aac736
From: "Marc" <sip:[email protected]>;tag=7ad1867f5eb84c0c9ef3d1b7a206201d
To: <sip:316********@10.2.0.20>;tag=b2d9eee1-3613-428f-bf26-5a7aeab35c50
CSeq: 29369 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=- 3914879804 3914879806 IN IP4 10.2.0.20
s=Asterisk
c=IN IP4 10.2.0.20
t=0 0
m=audio 15232 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 (847 bytes) from UDP:10.100.0.1:60400 --->
UPDATE sip:10.2.0.20:5060 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.1:60400;rport;branch=z9hG4bKPj902a00e0964f41f983b148b8a977b780
Max-Forwards: 70
From: "Marc" <sip:[email protected]>;tag=7ad1867f5eb84c0c9ef3d1b7a206201d
To: <sip:316********@10.2.0.20>;tag=b2d9eee1-3613-428f-bf26-5a7aeab35c50
Contact: "Marc" <sip:[email protected]:60400;ob>
Call-ID: d8bf1b6debf24631b13aa19802aac736
CSeq: 29370 UPDATE
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length:   310

v=0
o=- 3914879804 3914879805 IN IP4 10.100.0.1
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4022 RTP/AVP 0 101
c=IN IP4 10.100.0.1
b=TIAS:64000
a=rtcp:4023 IN IP4 10.100.0.1
a=ssrc:2084714661 cname:30f9506477b00c55
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:10.100.0.1:60400 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.100.0.1:60400;rport=60400;received=10.100.0.1;branch=z9hG4bKPj902a00e0964f41f983b148b8a977b780
Call-ID: d8bf1b6debf24631b13aa19802aac736
From: "Marc" <sip:[email protected]>;tag=7ad1867f5eb84c0c9ef3d1b7a206201d
To: <sip:316********@10.2.0.20>;tag=b2d9eee1-3613-428f-bf26-5a7aeab35c50
CSeq: 29370 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=- 3914879804 3914879807 IN IP4 10.2.0.20
s=Asterisk
c=IN IP4 10.2.0.20
t=0 0
m=audio 15232 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

[2024-01-21 14:36:44] ERROR[24202]: res_pjsip_header_funcs.c:717 remove_header: No headers had been previously added to this session.
  == Spawn extension (func-apply-sipheaders, s, 13) exited non-zero on 'PJSIP/DIDLogicTrunk-0000005d'
<--- Transmitting SIP request (1164 bytes) to UDP:185.118.63.77:5060 --->
INVITE sip:316********@sip.nl.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 185.52.***.***:5060;rport;branch=z9hG4bKPj6ad98f14-63d8-4604-8fde-95768cd4127d
From: <sip:3188*******@10.100.0.18>;tag=00b39610-6a82-4482-8561-e336e3b060ce
To: <sip:316********@sip.nl.didlogic.net>
Contact: <sip:[email protected].***.***:5060>
Call-ID: 96d617ad-bd5e-4719-8a3f-cae340a3ef44
CSeq: 31083 INVITE
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:3188*******@10.100.0.18>
Remote-Party-ID: <sip:3188*******@10.100.0.18>;party=calling;privacy=off;screen=no
Max-Forwards: 70
User-Agent: FPBX-16.0.33(18.16.0)
Content-Type: application/sdp
Content-Length:   337

v=0
o=- 856107292 856107292 IN IP4 185.52.***.***
s=Asterisk
c=IN IP4 185.52.***.***
t=0 0
m=audio 19690 RTP/AVP 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (539 bytes) from UDP:185.118.63.77:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 185.52.***.***:5060;rport=32015;branch=z9hG4bKPj6ad98f14-63d8-4604-8fde-95768cd4127d;received=185.52.***.***
From: <sip:3188*******@10.100.0.18>;tag=00b39610-6a82-4482-8561-e336e3b060ce
To: <sip:316********@sip.nl.didlogic.net>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.0e02c88a
Call-ID: 96d617ad-bd5e-4719-8a3f-cae340a3ef44
CSeq: 31083 INVITE
Proxy-Authenticate: Digest realm="sip.nl.didlogic.net", nonce="Za0tKGWtK/zNUUwZlLCxqYKgvjI+FA8reW3S94A=", qop="auth"
Content-Length: 0


<--- Transmitting SIP request (450 bytes) to UDP:185.118.63.77:5060 --->
ACK sip:316********@sip.nl.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 185.52.***.***:5060;rport;branch=z9hG4bKPj6ad98f14-63d8-4604-8fde-95768cd4127d
From: <sip:3188*******@10.100.0.18>;tag=00b39610-6a82-4482-8561-e336e3b060ce
To: <sip:316********@sip.nl.didlogic.net>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.0e02c88a
Call-ID: 96d617ad-bd5e-4719-8a3f-cae340a3ef44
CSeq: 31083 ACK
Max-Forwards: 70
User-Agent: FPBX-16.0.33(18.16.0)
Content-Length:  0


<--- Transmitting SIP request (1443 bytes) to UDP:185.118.63.77:5060 --->
INVITE sip:316********@sip.nl.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 185.52.***.***:5060;rport;branch=z9hG4bKPj5d5866c7-3c35-4200-ad83-6d944cce7cb6
From: <sip:3188*******@10.100.0.18>;tag=00b39610-6a82-4482-8561-e336e3b060ce
To: <sip:316********@sip.nl.didlogic.net>
Contact: <sip:[email protected].***.***:5060>
Call-ID: 96d617ad-bd5e-4719-8a3f-cae340a3ef44
CSeq: 31084 INVITE
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-16.0.33(18.16.0)
Proxy-Authorization: Digest username="86346", realm="sip.nl.didlogic.net", nonce="Za0tKGWtK/zNUUwZlLCxqYKgvjI+FA8reW3S94A=", uri="sip:316********@sip.nl.didlogic.net", response="2a206020df0253f466d7d6439a667ece", cnonce="6be52cded77f41f09d7267d185778838", qop=auth, nc=00000001
P-Asserted-Identity: <sip:3188*******@10.100.0.18>
Remote-Party-ID: <sip:3188*******@10.100.0.18>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length:   337

v=0
o=- 856107292 856107292 IN IP4 185.52.***.***
s=Asterisk
c=IN IP4 185.52.***.***
t=0 0
m=audio 19690 RTP/AVP 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (384 bytes) from UDP:185.118.63.77:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 185.52.***.***:5060;rport=32015;branch=z9hG4bKPj5d5866c7-3c35-4200-ad83-6d944cce7cb6;received=185.52.***.***
From: <sip:3188*******@10.100.0.18>;tag=00b39610-6a82-4482-8561-e336e3b060ce
To: <sip:316********@sip.nl.didlogic.net>
Call-ID: 96d617ad-bd5e-4719-8a3f-cae340a3ef44
CSeq: 31084 INVITE
Content-Length: 0


<--- Received SIP response (1047 bytes) from UDP:185.118.63.77:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 185.52.***.***:5060;received=185.52.***.***;rport=32015;branch=z9hG4bKPj5d5866c7-3c35-4200-ad83-6d944cce7cb6
Record-Route: <sip:185.118.63.77;lr=on;ftag=00b39610-6a82-4482-8561-e336e3b060ce;did=7d.4e11;nat=yes;vsf=AAAAAAcFCggCAAgBAQMDd3EBHh8BAB4eHh8JOA-->
From: <sip:3188*******@10.100.0.18>;tag=00b39610-6a82-4482-8561-e336e3b060ce
To: <sip:316********@sip.nl.didlogic.net>;tag=as2c7ce0c7
Call-ID: 96d617ad-bd5e-4719-8a3f-cae340a3ef44
CSeq: 31084 INVITE
Server: DID Logic MGW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:316********@81.171.28.137:5080>
Content-Type: application/sdp
Content-Length: 306

v=0
o=didlogic 95118517 95118517 IN IP4 81.171.28.137
s=DID Logic MGW
c=IN IP4 81.171.28.137
t=0 0
m=audio 11748 RTP/AVP 0 8 9 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (906 bytes) to UDP:10.100.0.1:60400 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.100.0.1:60400;rport=60400;received=10.100.0.1;branch=z9hG4bKPj08b08c02af70447d9f29566e01365d47
Call-ID: d8bf1b6debf24631b13aa19802aac736
From: "Marc" <sip:[email protected]>;tag=7ad1867f5eb84c0c9ef3d1b7a206201d
To: <sip:316********@10.2.0.20>;tag=b2d9eee1-3613-428f-bf26-5a7aeab35c50
CSeq: 29369 INVITE
Server: FPBX-16.0.33(18.16.0)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Contact: <sip:10.2.0.20:5060>
P-Asserted-Identity: "CID:3188*******" <sip:316********@10.2.0.20>
Content-Type: application/sdp
Content-Length:   255

v=0
o=- 3914879804 3914879806 IN IP4 10.2.0.20
s=Asterisk
c=IN IP4 10.2.0.20
t=0 0
m=audio 15232 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 (861 bytes) from UDP:10.100.0.1:60400 --->
UPDATE sip:10.2.0.20:5060 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.1:60400;rport;branch=z9hG4bKPja2703bcbe6434105b2c78a29192df5aa
Max-Forwards: 70
From: "Marc" <sip:[email protected]>;tag=7ad1867f5eb84c0c9ef3d1b7a206201d
To: <sip:316********@10.2.0.20>;tag=b2d9eee1-3613-428f-bf26-5a7aeab35c50
Contact: "Marc" <sip:[email protected]:60400;ob>
Call-ID: d8bf1b6debf24631b13aa19802aac736
CSeq: 29371 UPDATE
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Min-SE: 90
Content-Type: application/sdp
Content-Length:   310

v=0
o=- 3914879804 3914879807 IN IP4 10.100.0.1
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4022 RTP/AVP 0 101
c=IN IP4 10.100.0.1
b=TIAS:64000
a=rtcp:4023 IN IP4 10.100.0.1
a=ssrc:2084714661 cname:30f9506477b00c55
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

<--- Transmitting SIP response (969 bytes) to UDP:10.100.0.1:60400 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.100.0.1:60400;rport=60400;received=10.100.0.1;branch=z9hG4bKPja2703bcbe6434105b2c78a29192df5aa
Call-ID: d8bf1b6debf24631b13aa19802aac736
From: "Marc" <sip:[email protected]>;tag=7ad1867f5eb84c0c9ef3d1b7a206201d
To: <sip:316********@10.2.0.20>;tag=b2d9eee1-3613-428f-bf26-5a7aeab35c50
CSeq: 29371 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
P-Asserted-Identity: "CID:3188*******" <sip:316********@10.2.0.20>
Server: FPBX-16.0.33(18.16.0)
Content-Type: application/sdp
Content-Length:   231

v=0
o=- 3914879804 3914879808 IN IP4 10.2.0.20
s=Asterisk
c=IN IP4 10.2.0.20
t=0 0
m=audio 15232 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

<--- Received SIP request (368 bytes) from UDP:10.100.0.1:60400 --->
CANCEL sip:316********@10.2.0.20 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.1:60400;rport;branch=z9hG4bKPj08b08c02af70447d9f29566e01365d47
Max-Forwards: 70
From: "Marc" <sip:[email protected]>;tag=7ad1867f5eb84c0c9ef3d1b7a206201d
To: <sip:316********@10.2.0.20>
Call-ID: d8bf1b6debf24631b13aa19802aac736
CSeq: 29369 CANCEL
User-Agent: MicroSIP/3.21.3
Content-Length:  0


<--- Transmitting SIP response (393 bytes) to UDP:10.100.0.1:60400 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.100.0.1:60400;rport=60400;received=10.100.0.1;branch=z9hG4bKPj08b08c02af70447d9f29566e01365d47
Call-ID: d8bf1b6debf24631b13aa19802aac736
From: "Marc" <sip:[email protected]>;tag=7ad1867f5eb84c0c9ef3d1b7a206201d
To: <sip:316********@10.2.0.20>;tag=b2d9eee1-3613-428f-bf26-5a7aeab35c50
CSeq: 29369 CANCEL
Server: FPBX-16.0.33(18.16.0)
Content-Length:  0


<--- Transmitting SIP response (588 bytes) to UDP:10.100.0.1:60400 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.100.0.1:60400;rport=60400;received=10.100.0.1;branch=z9hG4bKPj08b08c02af70447d9f29566e01365d47
Call-ID: d8bf1b6debf24631b13aa19802aac736
From: "Marc" <sip:[email protected]>;tag=7ad1867f5eb84c0c9ef3d1b7a206201d
To: <sip:316********@10.2.0.20>;tag=b2d9eee1-3613-428f-bf26-5a7aeab35c50
CSeq: 29369 INVITE
Server: FPBX-16.0.33(18.16.0)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
P-Asserted-Identity: "CID:3188*******" <sip:316********@10.2.0.20>
Content-Length:  0


<--- Received SIP request (374 bytes) from UDP:10.100.0.1:60400 --->
ACK sip:316********@10.2.0.20 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.1:60400;rport;branch=z9hG4bKPj08b08c02af70447d9f29566e01365d47
Max-Forwards: 70
From: "Marc" <sip:[email protected]>;tag=7ad1867f5eb84c0c9ef3d1b7a206201d
To: <sip:316********@10.2.0.20>;tag=b2d9eee1-3613-428f-bf26-5a7aeab35c50
Call-ID: d8bf1b6debf24631b13aa19802aac736
CSeq: 29369 ACK
Content-Length:  0


  == Spawn extension (app-missedcall-hangup, s, 4) exited non-zero on 'PJSIP/DIDLogicTrunk-0000005d'
  == Spawn extension (macro-dialout-trunk, s, 36) exited non-zero on 'PJSIP/100-0000005c' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 316********, 11) exited non-zero on 'PJSIP/100-0000005c'
<--- Transmitting SIP request (435 bytes) to UDP:185.118.63.77:5060 --->
CANCEL sip:316********@sip.nl.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 185.52.***.***:5060;rport;branch=z9hG4bKPj5d5866c7-3c35-4200-ad83-6d944cce7cb6
From: <sip:3188*******@10.100.0.18>;tag=00b39610-6a82-4482-8561-e336e3b060ce
To: <sip:316********@sip.nl.didlogic.net>
Call-ID: 96d617ad-bd5e-4719-8a3f-cae340a3ef44
CSeq: 31084 CANCEL
Reason: Q.850;cause=127
Max-Forwards: 70
User-Agent: FPBX-16.0.33(18.16.0)
Content-Length:  0


<--- Received SIP response (401 bytes) from UDP:185.118.63.77:5060 --->
SIP/2.0 200 canceling
Via: SIP/2.0/UDP 185.52.***.***:5060;rport=32015;branch=z9hG4bKPj5d5866c7-3c35-4200-ad83-6d944cce7cb6;received=185.52.***.***
From: <sip:3188*******@10.100.0.18>;tag=00b39610-6a82-4482-8561-e336e3b060ce
To: <sip:316********@sip.nl.didlogic.net>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-f921c88a
Call-ID: 96d617ad-bd5e-4719-8a3f-cae340a3ef44
CSeq: 31084 CANCEL
Content-Length: 0


<--- Received SIP response (515 bytes) from UDP:185.118.63.77:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 185.52.***.***:5060;received=185.52.***.***;rport=32015;branch=z9hG4bKPj5d5866c7-3c35-4200-ad83-6d944cce7cb6
From: <sip:3188*******@10.100.0.18>;tag=00b39610-6a82-4482-8561-e336e3b060ce
To: <sip:316********@sip.nl.didlogic.net>;tag=as2c7ce0c7
Call-ID: 96d617ad-bd5e-4719-8a3f-cae340a3ef44
CSeq: 31084 INVITE
Server: DID Logic MGW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


<--- Transmitting SIP request (419 bytes) to UDP:185.118.63.77:5060 --->
ACK sip:316********@sip.nl.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 185.52.***.***:5060;rport;branch=z9hG4bKPj5d5866c7-3c35-4200-ad83-6d944cce7cb6
From: <sip:3188*******@10.100.0.18>;tag=00b39610-6a82-4482-8561-e336e3b060ce
To: <sip:316********@sip.nl.didlogic.net>;tag=as2c7ce0c7
Call-ID: 96d617ad-bd5e-4719-8a3f-cae340a3ef44
CSeq: 31084 ACK
Max-Forwards: 70
User-Agent: FPBX-16.0.33(18.16.0)
Content-Length:  0


  == Spawn extension (macro-hangupcall, s, 5) exited non-zero on 'PJSIP/100-0000005c' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/100-0000005c'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/100-0000005c'
<--- Transmitting SIP request (414 bytes) to UDP:10.100.0.1:60400 --->
OPTIONS sip:[email protected]:60400;ob SIP/2.0
Via: SIP/2.0/UDP 10.2.0.20:5060;rport;branch=z9hG4bKPj2819f416-eb71-49df-ae12-4cb14a2e6fdc
From: <sip:[email protected]>;tag=3922e9c6-c34d-4d56-8ffc-fe91c1a03b84
To: <sip:[email protected];ob>
Contact: <sip:[email protected]:5060>
Call-ID: 656711e5-be13-4bb1-90f4-e207ccb9a790
CSeq: 18617 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.33(18.16.0)
Content-Length:  0


<--- Received SIP response (786 bytes) from UDP:10.100.0.1:60400 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.0.20:5060;rport=5060;received=10.2.0.20;branch=z9hG4bKPj2819f416-eb71-49df-ae12-4cb14a2e6fdc
Call-ID: 656711e5-be13-4bb1-90f4-e207ccb9a790
From: <sip:[email protected]>;tag=3922e9c6-c34d-4d56-8ffc-fe91c1a03b84
To: <sip:[email protected];ob>;tag=z9hG4bKPj2819f416-eb71-49df-ae12-4cb14a2e6fdc
CSeq: 18617 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub, trickle-ice
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.21.3
Content-Length:  0

The only documentation I have from my provider seems to be pretty outdated (at least 4 years old): (I can’t post the link, but I use DID Logic)

Stewart1 · January 21, 2024, 4:08pm

I’m puzzled, because it all looks good, except possibly the From header. When you called +18004377950 and +44 20 3026 4621, did you hear the same incorrect caller IDs?

I know nothing about didlogic, but the page at Asterisk SIP Trunk Configuration Guide has (as an example)
fromuser=50841
where 50841 is the username at didlogic. If your username is other than the 3188 number (which I assume is your DID), try setting fromuser accordingly. For a pjsip trunk, the setting is From User

Stewart1 · January 21, 2024, 4:26pm

I was playing around on the didlogic site and found this:
https://didlogic.com/account-levels
It appears that you need at least a Business level account to send your own caller ID???
Do you have that?
Of course, you can use a different provider for outbound calling.
I have trunking with AnveoDirect, and tried to see whether they would better meet your needs. Unfortunately, they only offer +3185 for NL national, not +3188. Does that make them unsuitable?

Edit: I don’t understand who this nonsense applies to.
Sometimes, I see

and other times see that sending your own caller ID is included in Basic.

SparkyRih · January 21, 2024, 6:31pm

Oh damn, I missed the “send your own caller id” in the account types. I do in fact have a basic account. The business account would be way too expensive for my use case.

I think I’ll be looking for a new provider than. I do prefer a “0475” prefix, since that’s the area code where my business is located. But 088 and 085 are both “non regional”, so they are fine as well. I will look into your suggestion first, thanks so much!

Edit: DID Logic “made some changes” after I sent them a question about this matter. I did make a test call, and I do indeed receive it with my own caller ID now!

system · February 21, 2024, 6:31pm

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