Outgoing calls do not work, dialplan not correct?

int_ar · March 3, 2023, 2:13pm

Hi,

I have the following setup

PBX Station Aastra MXOne as main phone station. IP is 172.23.1.233
There I created an extension 200.

Set up a freepbx box (172.23.1.42)
Created a pjsip trunk using credentials from MXOne Extension called “MXOne200”

Then I created 2 extensions in freepbx (88888 and 77777) for testing.
I use softphones (phoner.exe) on 88888 and 77777 and did some internal testing → works fine

Created an inbound route any to 88888.
I call from my normal phone (MXOne extension 172) to 200 → softphone 88888 rings and speech is working fine in all directions.

Then I created an outbound route “Outbound 200”
trunk sequence MXOne200 and that’s it. The dial patterns are I think my problem… I want to call the 172 from my 88888 phone, so dial pattern ZXX (als the first digit for internal callings is 1-9, all others 0-9. three digits in total, nothing else)
I always get “The number you have dialed is not in service, please check the number and try again later”
A dial pattern of . is not changing anything
phoner.exe IP is 172.23.1.67

I am missing something, but I cannot see it.
Any hint would be appreciated

thanks in advance from Germany
André

Here’s the pjsip log of the call

PJSIP Logging enabled
<--- Received SIP request (1145 bytes) from UDP:172.23.1.67:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.23.1.67:5060;branch=z9hG4bK80f434ba3ab8ed11896694d592ba3f90;rport
From: <sip:[email protected]>;tag=2436471351
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060;gr=005075B2-3AB8-ED11-895C-94D592BA3F90>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Supported: 100rel, replaces, from-change, gruu
User-Agent: phoner 3.23
P-Preferred-Identity: <sip:[email protected]>
Content-Length:   513

v=0
o=- 2335364209 1 IN IP4 172.23.1.67
s=phoner 3.23
c=IN IP4 172.23.1.67
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 9 18 111 112 113 114 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:111 speex/16000
a=rtpmap:112 G726-16/8000
a=rtpmap:113 G726-24/8000
a=rtpmap:114 G726-40/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:320634361
a=sendrecv

<--- Transmitting SIP response (531 bytes) to UDP:172.23.1.67:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.23.1.67:5060;rport=5060;received=172.23.1.67;branch=z9hG4bK80f434ba3ab8ed11896694d592ba3f90
Call-ID: [email protected]
From: <sip:[email protected]>;tag=2436471351
To: <sip:[email protected]>;tag=z9hG4bK80f434ba3ab8ed11896694d592ba3f90
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1677852589/468832aa37ffd92587c8966aa365cb52",opaque="1fa699a209e33b61",algorithm=MD5,qop="auth"
Server: FPBX-16.0.33(16.30.0)
Content-Length:  0


<--- Received SIP request (331 bytes) from UDP:172.23.1.67:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.23.1.67:5060;branch=z9hG4bK80f434ba3ab8ed11896694d592ba3f90;rport
From: <sip:[email protected]>;tag=2436471351
To: <sip:[email protected]>;tag=z9hG4bK80f434ba3ab8ed11896694d592ba3f90
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0


<--- Received SIP request (1436 bytes) from UDP:172.23.1.67:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.23.1.67:5060;branch=z9hG4bK80f434ba3ab8ed11896894d592ba3f90;rport
From: <sip:[email protected]>;tag=2436471351
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]:5060;gr=005075B2-3AB8-ED11-895C-94D592BA3F90>
Authorization: Digest username="88888", realm="asterisk", nonce="1677852589/468832aa37ffd92587c8966aa365cb52", uri="sip:[email protected]", response="d2da75aaffbbd3153cda9fd9bb70da73", algorithm=MD5, cnonce="80f434ba3ab8ed11896794d592ba3f90", opaque="1fa699a209e33b61", qop=auth, nc=00000001
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Supported: 100rel, replaces, from-change, gruu
User-Agent: phoner 3.23
P-Preferred-Identity: <sip:[email protected]>
Content-Length:   513

v=0
o=- 2335364209 1 IN IP4 172.23.1.67
s=phoner 3.23
c=IN IP4 172.23.1.67
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 9 18 111 112 113 114 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:111 speex/16000
a=rtpmap:112 G726-16/8000
a=rtpmap:113 G726-24/8000
a=rtpmap:114 G726-40/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:320634361
a=sendrecv

<--- Transmitting SIP response (335 bytes) to UDP:172.23.1.67:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.23.1.67:5060;rport=5060;received=172.23.1.67;branch=z9hG4bK80f434ba3ab8ed11896894d592ba3f90
Call-ID: [email protected]
From: <sip:[email protected]>;tag=2436471351
To: <sip:[email protected]>
CSeq: 2 INVITE
Server: FPBX-16.0.33(16.30.0)
Content-Length:  0


<--- Transmitting SIP response (888 bytes) to UDP:172.23.1.67:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.23.1.67:5060;rport=5060;received=172.23.1.67;branch=z9hG4bK80f434ba3ab8ed11896894d592ba3f90
Call-ID: [email protected]
From: <sip:[email protected]>;tag=2436471351
To: <sip:[email protected]>;tag=e093cf34-e19f-4cbb-a159-1cf55f917240
CSeq: 2 INVITE
Server: FPBX-16.0.33(16.30.0)
Contact: <sip:172.23.1.42:5060>
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Content-Type: application/sdp
Content-Length:   324

v=0
o=- 2335364209 3 IN IP4 172.23.1.42
s=Asterisk
c=IN IP4 172.23.1.42
t=0 0
m=audio 14976 RTP/AVP 0 8 3 2 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 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

[2023-03-03 14:09:51] ERROR[72189]: res_pjsip_header_funcs.c:670 remove_header: No headers had been previously added to this session.
<--- Transmitting SIP request (1000 bytes) to UDP:172.23.1.233:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.1.42:5060;rport;branch=z9hG4bKPjfaed3553-d38c-4127-bbed-c756db902c0e
From: <sip:[email protected]>;tag=e4f4a92b-6691-4b30-bce7-d0b4fc512535
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: ac769527-08fa-4dc3-a920-67d3e49a5dc2
CSeq: 2410 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(16.30.0)
Content-Type: application/sdp
Content-Length:   337

v=0
o=- 2097015622 2097015622 IN IP4 172.23.1.42
s=Asterisk
c=IN IP4 172.23.1.42
t=0 0
m=audio 10764 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 (369 bytes) from UDP:172.23.1.233:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.23.1.42:5060;rport=5060;branch=z9hG4bKPjfaed3553-d38c-4127-bbed-c756db902c0e
To: <sip:[email protected]>;tag=3c7cf35c
From: <sip:[email protected]>;tag=e4f4a92b-6691-4b30-bce7-d0b4fc512535
Call-ID: ac769527-08fa-4dc3-a920-67d3e49a5dc2
CSeq: 2410 INVITE
User-Agent: Aastra MX-ONE SN/17.4.0.1.5
Content-Length: 0


<--- Transmitting SIP request (390 bytes) to UDP:172.23.1.233:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.1.42:5060;rport;branch=z9hG4bKPjfaed3553-d38c-4127-bbed-c756db902c0e
From: <sip:[email protected]>;tag=e4f4a92b-6691-4b30-bce7-d0b4fc512535
To: <sip:[email protected]>;tag=3c7cf35c
Call-ID: ac769527-08fa-4dc3-a920-67d3e49a5dc2
CSeq: 2410 ACK
Max-Forwards: 70
User-Agent: FPBX-16.0.33(16.30.0)
Content-Length:  0


<--- Transmitting SIP response (950 bytes) to UDP:172.23.1.67:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.23.1.67:5060;rport=5060;received=172.23.1.67;branch=z9hG4bK80f434ba3ab8ed11896894d592ba3f90
Call-ID: [email protected]
From: <sip:[email protected]>;tag=2436471351
To: <sip:[email protected]>;tag=e093cf34-e19f-4cbb-a159-1cf55f917240
CSeq: 2 INVITE
Server: FPBX-16.0.33(16.30.0)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Contact: <sip:172.23.1.42:5060>
P-Asserted-Identity: "CID:03841254200" <sip:[email protected]>
Content-Type: application/sdp
Content-Length:   324

v=0
o=- 2335364209 3 IN IP4 172.23.1.42
s=Asterisk
c=IN IP4 172.23.1.42
t=0 0
m=audio 14976 RTP/AVP 0 8 3 2 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 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

<--- Transmitting SIP response (576 bytes) to UDP:172.23.1.67:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 172.23.1.67:5060;rport=5060;received=172.23.1.67;branch=z9hG4bK80f434ba3ab8ed11896894d592ba3f90
Call-ID: [email protected]
From: <sip:[email protected]>;tag=2436471351
To: <sip:[email protected]>;tag=e093cf34-e19f-4cbb-a159-1cf55f917240
CSeq: 2 INVITE
Server: FPBX-16.0.33(16.30.0)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Reason: Q.850;cause=17
P-Asserted-Identity: "CID:03841254200" <sip:[email protected]>
Content-Length:  0


<--- Received SIP request (619 bytes) from UDP:172.23.1.67:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.23.1.67:5060;branch=z9hG4bK80f434ba3ab8ed11896894d592ba3f90;rport
From: <sip:[email protected]>;tag=2436471351
To: <sip:[email protected]>;tag=e093cf34-e19f-4cbb-a159-1cf55f917240
Call-ID: [email protected]
CSeq: 2 ACK
Authorization: Digest username="88888", realm="asterisk", nonce="1677852589/468832aa37ffd92587c8966aa365cb52", uri="sip:[email protected]", response="d2da75aaffbbd3153cda9fd9bb70da73", algorithm=MD5, cnonce="80f434ba3ab8ed11896794d592ba3f90", opaque="1fa699a209e33b61", qop=auth, nc=00000001
Content-Length: 0


pbx*CLI> pjsip set logger off
PJSIP Logging disabled
pbx*CLI>

david55 · March 3, 2023, 3:24pm

The Mx-one is telling you that 172 is not a number that it provides.

dobrosavljevic · March 3, 2023, 3:26pm

I am not understanding this section. What do you mean from your normal phone extension 172? Is this an extension on a completely separate PBX?

david55 · March 3, 2023, 3:28pm

It looks like Mx-one is a PBX from its web page. There used to be an Mx-one pbx decades ago, but I suspect someone has recycled the name, rather than this being related. In fact I think the original product got rebranded.

int_ar · March 3, 2023, 3:30pm

I have 2 PBX, an Astra MXOne and a freePBX
The freepbx is using a sip extension from the MX One.

Main goal at the end: I want to provide dial in via an freePBX IVR to Jitsi meet.

172 is my digital extension on the MXOne, 200 is the SIP extension on the MXOne, registered client is the freepbx (200 extension from MXOne is the trunk for freepbx)

if I connect phoner as 200 using the same credentials as the freepbx to register as extension on the MXOne, calling MXOne extension 172 works.
as stated, incoming calls work fine

int_ar · March 3, 2023, 3:33pm

@David55 yes, it is an older PBX providing, analogue, digital and SIP extensions,
it was originally realeased by aastra, which was bought by Mitel.

I am using ~20 SIP Clients on the MXOne, mostly snom or mitel phones. ANd phoner for testing purposes. There are no problems with incoming/outgoing calls, but I admit,a full pbx is “a bit more” than a regular softphone

Stewart1 · March 3, 2023, 3:58pm

int_ar:

INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.1.42:5060;rport;branch=z9hG4bKPjfaed3553-d38c-4127-bbed-c756db902c0e
From: <sip:[email protected]>;tag=e4f4a92b-6691-4b30-bce7-d0b4fc512535

Setting up the link between the systems as an extension on the Aastra side has many limitations; normally it would be a trunk on both ends. Please explain why you did not (or cannot) set up a trunk on the Aastra.

But given that it is an extension, try setting From User in your FreePBX trunk to 200. With luck, that will cause the Aastra to recognize the call as ‘local’ and route it to ext. 172. If this doesn’t help, paste another log with the change. It would also be useful to include a log of a successful incoming call, because what the Aastra sends is a clue to what it expects to receive.

int_ar · March 7, 2023, 12:17pm

Sorry for the delay.
I routed incoming calls from 200 to my phoner test client 77777

calling from my MXOne Extension 172 to MXOne Extension 200

pbx*CLI> pjsip set logger on
PJSIP Logging enabled
<--- Received SIP request (1236 bytes) from UDP:172.23.1.233:5060 --->
INVITE sip:[email protected]:5060;user=phone;transport=UDP;line=ceoclsq SIP/2.0
Via: SIP/2.0/UDP 172.23.1.233:5060;branch=z9hG4bK-524287-1---643e7551f8520439;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=UDP>
To: <sip:[email protected];user=phone>
From: "Andre Rode"<sip:[email protected];user=phone>;tag=7ec11f68
Call-ID: RhKm7JXLTtUEKO0Gx7ZeQQ..
CSeq: 1 INVITE
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, OPTIONS, BYE, ACK, CANCEL, INFO, REGISTER, REFER, PRACK, SUBSCRIBE, NOTIFY, MESSAGE, UPDATE, PUBLISH
Content-Type: application/sdp
Supported: timer, replaces, 100rel
User-Agent: Aastra MX-ONE SN/17.4.0.1.5
Content-Length: 543

v=0
o=- 7207777007629986538 1 IN IP4 172.23.1.233
s=MX-ONE
c=IN IP4 172.23.1.237
t=0 0
m=audio 21092 RTP/AVP 8 0 18 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=rtpmap:13 CN/8000
a=ptime:20
a=silenceSupp:on - - - -
a=sendrecv
a=sqn: 0
a=cdsc:1 image udptl t38
a=cpar:a=T38FaxVersion:0
a=cpar:a=T38MaxBitRate:14400
a=cpar:a=T38FaxRateManagement:transferredTCF
a=cpar:a=T38FaxMaxBuffer:9772
a=cpar:a=T38FaxMaxDatagram:1472
a=cpar:a=T38FaxUdpEC:t38UDPRedundancy

<--- Transmitting SIP response (364 bytes) to UDP:172.23.1.233:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.23.1.233:5060;rport=5060;received=172.23.1.233;branch=z9hG4bK-524287-1---643e7551f8520439
Call-ID: RhKm7JXLTtUEKO0Gx7ZeQQ..
From: "Andre Rode" <sip:[email protected];user=phone>;tag=7ec11f68
To: <sip:[email protected];user=phone>
CSeq: 1 INVITE
Server: FPBX-16.0.33(16.30.0)
Content-Length:  0

<--- Transmitting SIP response (839 bytes) to UDP:172.23.1.233:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.23.1.233:5060;rport=5060;received=172.23.1.233;branch=z9hG4bK-524287-1---643e7551f8520439
Call-ID: RhKm7JXLTtUEKO0Gx7ZeQQ..
From: "Andre Rode" <sip:[email protected];user=phone>;tag=7ec11f68
To: <sip:[email protected];user=phone>;tag=af5a478b-ef1b-4b8b-9b78-dcfb05ec985e
CSeq: 1 INVITE
Server: FPBX-16.0.33(16.30.0)
Contact: <sip:172.23.1.42:5060>
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Content-Type: application/sdp
Content-Length:   246

v=0
o=- 223978 3 IN IP4 172.23.1.42
s=Asterisk
c=IN IP4 172.23.1.42
t=0 0
m=audio 11464 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

[2023-03-07 12:16:00] WARNING[225825][C-00000069]: chan_sip.c:23281 func_header_read: This function can only be used on SIP channels.
[2023-03-07 12:16:00] WARNING[225825][C-00000069]: chan_sip.c:23281 func_header_read: This function can only be used on SIP channels.
[2023-03-07 12:16:00] WARNING[225825][C-00000069]: chan_sip.c:23281 func_header_read: This function can only be used on SIP channels.
[2023-03-07 12:16:00] WARNING[225825][C-00000069]: chan_sip.c:23281 func_header_read: This function can only be used on SIP channels.
[2023-03-07 12:16:00] WARNING[225825][C-00000069]: chan_sip.c:23281 func_header_read: This function can only be used on SIP channels.
[2023-03-07 12:16:00] WARNING[225825][C-00000069]: chan_sip.c:23281 func_header_read: This function can only be used on SIP channels.
[2023-03-07 12:16:00] WARNING[225825][C-00000069]: chan_sip.c:23281 func_header_read: This function can only be used on SIP channels.
[2023-03-07 12:16:00] WARNING[225825][C-00000069]: chan_sip.c:23281 func_header_read: This function can only be used on SIP channels.
[2023-03-07 12:16:00] WARNING[225825][C-00000069]: chan_sip.c:23281 func_header_read: This function can only be used on SIP channels.
[2023-03-07 12:16:00] WARNING[225825][C-00000069]: chan_sip.c:23281 func_header_read: This function can only be used on SIP channels.
[2023-03-07 12:16:00] WARNING[225825][C-00000069]: chan_sip.c:23281 func_header_read: This function can only be used on SIP channels.
[2023-03-07 12:16:00] WARNING[225825][C-00000069]: chan_sip.c:23281 func_header_read: This function can only be used on SIP channels.
<--- Transmitting SIP response (839 bytes) to UDP:172.23.1.233:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.23.1.233:5060;rport=5060;received=172.23.1.233;branch=z9hG4bK-524287-1---643e7551f8520439
Call-ID: RhKm7JXLTtUEKO0Gx7ZeQQ..
From: "Andre Rode" <sip:[email protected];user=phone>;tag=7ec11f68
To: <sip:[email protected];user=phone>;tag=af5a478b-ef1b-4b8b-9b78-dcfb05ec985e
CSeq: 1 INVITE
Server: FPBX-16.0.33(16.30.0)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Contact: <sip:172.23.1.42:5060>
Content-Type: application/sdp
Content-Length:   246

v=0
o=- 223978 3 IN IP4 172.23.1.42
s=Asterisk
c=IN IP4 172.23.1.42
t=0 0
m=audio 11464 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

<--- Transmitting SIP request (1065 bytes) to UDP:172.23.1.67:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.1.42:5060;rport;branch=z9hG4bKPjb7e99b46-a69f-4eaf-a72b-b844ee69a3e3
From: "Andre Rode" <sip:[email protected]>;tag=9ae8ba27-60ce-48bf-bc40-3f98173861b4
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: 3bb1b08c-a230-4951-8043-23f4d26c2765
CSeq: 27357 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: "Andre Rode" <sip:[email protected]>
Max-Forwards: 70
User-Agent: FPBX-16.0.33(16.30.0)
Content-Type: application/sdp
Content-Length:   337

v=0
o=- 1580025981 1580025981 IN IP4 172.23.1.42
s=Asterisk
c=IN IP4 172.23.1.42
t=0 0
m=audio 11046 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 (427 bytes) from UDP:172.23.1.67:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.23.1.42:5060;rport=5060;branch=z9hG4bKPjb7e99b46-a69f-4eaf-a72b-b844ee69a3e3
From: "Andre Rode" <sip:[email protected]>;tag=9ae8ba27-60ce-48bf-bc40-3f98173861b4
To: <sip:[email protected]>
Call-ID: 3bb1b08c-a230-4951-8043-23f4d26c2765
CSeq: 27357 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: phoner 3.23
Content-Length: 0

<--- Received SIP response (592 bytes) from UDP:172.23.1.67:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.23.1.42:5060;rport=5060;branch=z9hG4bKPjb7e99b46-a69f-4eaf-a72b-b844ee69a3e3
From: "Andre Rode" <sip:[email protected]>;tag=9ae8ba27-60ce-48bf-bc40-3f98173861b4
To: <sip:[email protected]>;tag=0040757d4fbbed118c1394d592ba3f90
Call-ID: 3bb1b08c-a230-4951-8043-23f4d26c2765
CSeq: 27357 INVITE
Contact: <sip:[email protected]:5060;gr=005F7F77-4FBB-ED11-8C0A-94D592BA3F90>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Supported: 100rel, replaces, from-change, gruu
Server: phoner 3.23
Content-Length: 0

<--- Transmitting SIP response (839 bytes) to UDP:172.23.1.233:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.23.1.233:5060;rport=5060;received=172.23.1.233;branch=z9hG4bK-524287-1---643e7551f8520439
Call-ID: RhKm7JXLTtUEKO0Gx7ZeQQ..
From: "Andre Rode" <sip:[email protected];user=phone>;tag=7ec11f68
To: <sip:[email protected];user=phone>;tag=af5a478b-ef1b-4b8b-9b78-dcfb05ec985e
CSeq: 1 INVITE
Server: FPBX-16.0.33(16.30.0)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Contact: <sip:172.23.1.42:5060>
Content-Type: application/sdp
Content-Length:   246

v=0
o=- 223978 3 IN IP4 172.23.1.42
s=Asterisk
c=IN IP4 172.23.1.42
t=0 0
m=audio 11464 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 response (938 bytes) from UDP:172.23.1.67:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.1.42:5060;rport=5060;branch=z9hG4bKPjb7e99b46-a69f-4eaf-a72b-b844ee69a3e3
From: "Andre Rode" <sip:[email protected]>;tag=9ae8ba27-60ce-48bf-bc40-3f98173861b4
To: <sip:[email protected]>;tag=0040757d4fbbed118c1394d592ba3f90
Call-ID: 3bb1b08c-a230-4951-8043-23f4d26c2765
CSeq: 27357 INVITE
Contact: <sip:[email protected]:5060;gr=005F7F77-4FBB-ED11-8C0A-94D592BA3F90>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Supported: 100rel, replaces, from-change, gruu
Server: phoner 3.23
Content-Length:   316

v=0
o=- 672356453 1 IN IP4 172.23.1.67
s=phoner 3.23
c=IN IP4 172.23.1.67
t=0 0
m=audio 5062 RTP/AVP 0 8 3 2 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1485448949
a=sendrecv

<--- Transmitting SIP request (462 bytes) to UDP:172.23.1.67:5060 --->
ACK sip:[email protected]:5060;gr=005F7F77-4FBB-ED11-8C0A-94D592BA3F90 SIP/2.0
Via: SIP/2.0/UDP 172.23.1.42:5060;rport;branch=z9hG4bKPjf9187e7b-cc05-483a-bc19-26402bc60a75
From: "Andre Rode" <sip:[email protected]>;tag=9ae8ba27-60ce-48bf-bc40-3f98173861b4
To: <sip:[email protected]>;tag=0040757d4fbbed118c1394d592ba3f90
Call-ID: 3bb1b08c-a230-4951-8043-23f4d26c2765
CSeq: 27357 ACK
Max-Forwards: 70
User-Agent: FPBX-16.0.33(16.30.0)
Content-Length:  0

<--- Transmitting SIP response (926 bytes) to UDP:172.23.1.233:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.1.233:5060;rport=5060;received=172.23.1.233;branch=z9hG4bK-524287-1---643e7551f8520439
Call-ID: RhKm7JXLTtUEKO0Gx7ZeQQ..
From: "Andre Rode" <sip:[email protected];user=phone>;tag=7ec11f68
To: <sip:[email protected];user=phone>;tag=af5a478b-ef1b-4b8b-9b78-dcfb05ec985e
CSeq: 1 INVITE
Server: FPBX-16.0.33(16.30.0)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Contact: <sip:172.23.1.42:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   246

v=0
o=- 223978 3 IN IP4 172.23.1.42
s=Asterisk
c=IN IP4 172.23.1.42
t=0 0
m=audio 11464 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 (482 bytes) from UDP:172.23.1.233:5060 --->
ACK sip:172.23.1.42:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.23.1.233:5060;branch=z9hG4bK-524287-1---49e0775b569c1609;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=UDP>
To: <sip:[email protected];user=phone>;tag=af5a478b-ef1b-4b8b-9b78-dcfb05ec985e
From: "Andre Rode" <sip:[email protected];user=phone>;tag=7ec11f68
Call-ID: RhKm7JXLTtUEKO0Gx7ZeQQ..
CSeq: 1 ACK
User-Agent: Aastra MX-ONE SN/17.4.0.1.5
Content-Length: 0

<--- Received SIP request (1177 bytes) from UDP:172.23.1.233:5060 --->
INVITE sip:172.23.1.42:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.23.1.233:5060;branch=z9hG4bK-524287-1---ad0b5e4b34da593f;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=UDP>
To: <sip:[email protected];user=phone>;tag=af5a478b-ef1b-4b8b-9b78-dcfb05ec985e
From: "Andre Rode" <sip:[email protected];user=phone>;tag=7ec11f68
Call-ID: RhKm7JXLTtUEKO0Gx7ZeQQ..
CSeq: 2 INVITE
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, OPTIONS, BYE, ACK, CANCEL, INFO, REGISTER, REFER, PRACK, SUBSCRIBE, NOTIFY, MESSAGE, UPDATE, PUBLISH
Content-Type: application/sdp
Supported: timer, replaces, resource-priority
User-Agent: Aastra MX-ONE SN/17.4.0.1.5
Content-Length: 443

v=0
o=- 7207777007629986538 2 IN IP4 172.23.1.233
s=MX-ONE
c=IN IP4 172.23.1.237
t=0 0
m=audio 21092 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
a=sqn: 0
a=cdsc:1 image udptl t38
a=cpar:a=T38FaxVersion:0
a=cpar:a=T38MaxBitRate:14400
a=cpar:a=T38FaxRateManagement:transferredTCF
a=cpar:a=T38FaxMaxBuffer:9772
a=cpar:a=T38FaxMaxDatagram:1472
a=cpar:a=T38FaxUdpEC:t38UDPRedundancy

<--- Transmitting SIP response (902 bytes) to UDP:172.23.1.233:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.1.233:5060;rport=5060;received=172.23.1.233;branch=z9hG4bK-524287-1---ad0b5e4b34da593f
Call-ID: RhKm7JXLTtUEKO0Gx7ZeQQ..
From: "Andre Rode" <sip:[email protected];user=phone>;tag=7ec11f68
To: <sip:[email protected];user=phone>;tag=af5a478b-ef1b-4b8b-9b78-dcfb05ec985e
CSeq: 2 INVITE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: <sip:172.23.1.42:5060>
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: FPBX-16.0.33(16.30.0)
Content-Type: application/sdp
Content-Length:   222

v=0
o=- 223978 4 IN IP4 172.23.1.42
s=Asterisk
c=IN IP4 172.23.1.42
t=0 0
m=audio 11464 RTP/AVP 8 101
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 (482 bytes) from UDP:172.23.1.233:5060 --->
ACK sip:172.23.1.42:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.23.1.233:5060;branch=z9hG4bK-524287-1---bd02266ef505a864;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=UDP>
To: <sip:[email protected];user=phone>;tag=af5a478b-ef1b-4b8b-9b78-dcfb05ec985e
From: "Andre Rode" <sip:[email protected];user=phone>;tag=7ec11f68
Call-ID: RhKm7JXLTtUEKO0Gx7ZeQQ..
CSeq: 2 ACK
User-Agent: Aastra MX-ONE SN/17.4.0.1.5
Content-Length: 0

<--- Transmitting SIP request (414 bytes) to UDP:172.23.1.233:5060 --->
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.1.42:5060;rport;branch=z9hG4bKPj354ddca2-2767-492c-9f81-3a6c8814d583
From: <sip:[email protected]>;tag=5d3b449d-88d7-4665-904c-9aebc9bb937c
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: 72532e31-ac2c-4905-8241-58ff0611722c
CSeq: 6508 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.33(16.30.0)
Content-Length:  0

<--- Received SIP response (740 bytes) from UDP:172.23.1.233:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.1.42:5060;rport=5060;branch=z9hG4bKPj354ddca2-2767-492c-9f81-3a6c8814d583
Contact: <sip:172.23.1.233:5060>
To: <sip:[email protected]>;tag=f4aaa121
From: <sip:[email protected]>;tag=5d3b449d-88d7-4665-904c-9aebc9bb937c
Call-ID: 72532e31-ac2c-4905-8241-58ff0611722c
CSeq: 6508 OPTIONS
Accept: application/sdp, multipart/mixed
Accept-Language: en
Allow: INVITE, OPTIONS, BYE, ACK, CANCEL, INFO, REGISTER, REFER, PRACK, SUBSCRIBE, NOTIFY, MESSAGE, UPDATE, PUBLISH
Supported: timer, replaces, resource-priority
User-Agent: Aastra MX-ONE SN/17.4.0.1.5
Allow-Events: message-summary, reg, dialog, server-side-terminal-control, call-info, line-seize, call-completion, presence
Content-Length: 0

<--- Received SIP request (487 bytes) from UDP:172.23.1.67:5060 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.1.67:5060;branch=z9hG4bK00c708814fbbed118c1394d592ba3f90;rport
From: <sip:[email protected]>;tag=0040757d4fbbed118c1394d592ba3f90
To: "Andre Rode" <sip:[email protected]>;tag=9ae8ba27-60ce-48bf-bc40-3f98173861b4
Call-ID: 3bb1b08c-a230-4951-8043-23f4d26c2765
CSeq: 27358 BYE
Contact: <sip:[email protected]:5060;gr=005F7F77-4FBB-ED11-8C0A-94D592BA3F90>
Max-Forwards: 70
User-Agent: phoner 3.23
Content-Length: 0

<--- Transmitting SIP response (396 bytes) to UDP:172.23.1.67:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.1.67:5060;rport=5060;received=172.23.1.67;branch=z9hG4bK00c708814fbbed118c1394d592ba3f90
Call-ID: 3bb1b08c-a230-4951-8043-23f4d26c2765
From: <sip:[email protected]>;tag=0040757d4fbbed118c1394d592ba3f90
To: "Andre Rode" <sip:[email protected]>;tag=9ae8ba27-60ce-48bf-bc40-3f98173861b4
CSeq: 27358 BYE
Server: FPBX-16.0.33(16.30.0)
Content-Length:  0

<--- Transmitting SIP request (467 bytes) to UDP:172.23.1.233:5060 --->
BYE sip:[email protected]:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.23.1.42:5060;rport;branch=z9hG4bKPj988867f1-2b15-4af8-b1db-78302eb26238
From: <sip:[email protected];user=phone>;tag=af5a478b-ef1b-4b8b-9b78-dcfb05ec985e
To: "Andre Rode" <sip:[email protected];user=phone>;tag=7ec11f68
Call-ID: RhKm7JXLTtUEKO0Gx7ZeQQ..
CSeq: 29520 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: FPBX-16.0.33(16.30.0)
Content-Length:  0

<--- Received SIP response (449 bytes) from UDP:172.23.1.233:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.1.42:5060;rport=5060;branch=z9hG4bKPj988867f1-2b15-4af8-b1db-78302eb26238
Contact: <sip:[email protected]:5060;transport=UDP>
To: "Andre Rode"<sip:[email protected];user=phone>;tag=7ec11f68
From: <sip:[email protected];user=phone>;tag=af5a478b-ef1b-4b8b-9b78-dcfb05ec985e
Call-ID: RhKm7JXLTtUEKO0Gx7ZeQQ..
CSeq: 29520 BYE
User-Agent: Aastra MX-ONE SN/17.4.0.1.5
Content-Length: 0

int_ar · March 7, 2023, 12:21pm

I changed the From User setting to 200
doesn’t change the problem sadly.

I cannot use a trunk at the mxone… because… actually I don’t intend to use one?
As I stated in the end freepbx should provide a IVR for a Jitsi meet server, no additional functions.
It works already (the IVR), I can call into meetings. But I do wish to make a call from the meeting room

system · April 7, 2023, 12:21pm

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