Disconnecting channel for lack of audio RTP activity in 30 seconds

Asterisk Asterisk SIP
1

Hello guy’s, i make a call to an external number but neither party can hear the other. after 30s the call is automatically disconnected.

I have this in the logs:
Disconnecting channel for lack of audio RTP activity in 30 seconds

//rtp_custum.conf

[general]
rtpstart=10000
rtpend=20000
icesupport=true
stunaddr=stun.l.google.com:19302

//pjsip.transport.conf
[0.0.0.0-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
allow_reload=no
tos=cs3
cos=3
local_net=10.38.43.0/24

[0.0.0.0-wss]
type=transport
protocol=wss
bind=0.0.0.0
allow_reload=no
tos=cs3
cos=3
local_net=10.38.43.0/24

//pjsip.endpoint.conf
[004058]
type=endpoint
aors=004058
auth=004058-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=ulaw,alaw,g729
context=from-internal
callerid=dialerTEST004058 <004058>

dtmf_mode=rfc4733
direct_media=yes
outbound_auth=004058-auth
mailboxes=004058@default

mwi_subscribe_replaces_unsolicited=yes
aggregate_mwi=yes
use_avpf=no
rtcp_mux=no
max_audio_streams=1
max_video_streams=1
bundle=no
ice_support=no
media_use_received_transport=no
trust_id_inbound=yes
user_eq_phone=no
send_connected_line=yes
media_encryption=no
timers=yes
timers_min_se=90
media_encryption_optimistic=no
refer_blind_progress=yes
refer_blind_progress=yes
rtp_timeout=30
rtp_timeout_hold=300
rtp_keepalive=0
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=fr
one_touch_recording=on
record_on_feature=apprecord
record_off_feature=apprecord




//log outputs




freepbx*CLI> sngrep
No such command 'sngrep' (type 'core show help sngrep' for other possible comman                                                                                                                                                             ds)
<--- Received SIP request (686 bytes) from UDP:10.38.2.233:53143 --->
REGISTER sip:pbx.com:5060 SIP/2.0
Via: SIP/2.0/UDP 10.38.2.233:53143;rport;branch=z9hG4bKPj0ca0bf98c4fa42c793cfa13                                                                                                                                                             f8eda0a61
Route: <sip:pbx.com:5060;lr>
Max-Forwards: 70
From: "dialer-test-004055" <sip:004055@pbx.com>;tag=c80da8                                                                                                                                                             57a2e94c778ebed1362db4ebd9
To: "dialer-test-004055" <sip:004055@pbx.com>
Call-ID: bd0137f24cb9492db91abfcc5a5bfe11
CSeq: 26795 REGISTER
User-Agent: MicroSIP/3.21.3
Contact: "dialer-test-004055" <sip:004055@10.38.2.233:53143;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER,                                                                                                                                                              MESSAGE, OPTIONS
Content-Length:  0


<--- Transmitting SIP response (633 bytes) to UDP:10.38.2.233:53143 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.38.2.233:53143;rport=53143;received=10.38.2.233;branch=z9hG4                                                                                                                                                             bKPj0ca0bf98c4fa42c793cfa13f8eda0a61
Call-ID: bd0137f24cb9492db91abfcc5a5bfe11
From: "dialer-test-004055" <sip:004055@pbx.com>;tag=c80da8                                                                                                                                                             57a2e94c778ebed1362db4ebd9
To: "dialer-test-004055" <sip:004055@pbx.com>;tag=z9hG4bKP                                                                                                                                                             j0ca0bf98c4fa42c793cfa13f8eda0a61
CSeq: 26795 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1715881394/ef0a29d7a6d6cff3bb47                                                                                                                                                             f14791db2a8e",opaque="0dd6f5776cf88cfc",algorithm=MD5,qop="auth"
Server: FPBX-16.0.40.7(18.16.0)
Content-Length:  0


<--- Received SIP request (995 bytes) from UDP:10.38.2.233:53143 --->
REGISTER sip:pbx.com:5060 SIP/2.0
Via: SIP/2.0/UDP 10.38.2.233:53143;rport;branch=z9hG4bKPj25f3a6ee98a74a1ca5c9f10                                                                                                                                                             27b96c008
Route: <sip:pbx.com:5060;lr>
Max-Forwards: 70
From: "dialer-test-004055" <sip:004055@pbx.com>;tag=c80da8                                                                                                                                                             57a2e94c778ebed1362db4ebd9
To: "dialer-test-004055" <sip:004055@pbx.com>
Call-ID: bd0137f24cb9492db91abfcc5a5bfe11
CSeq: 26796 REGISTER
User-Agent: MicroSIP/3.21.3
Contact: "dialer-test-004055" <sip:004055@10.38.2.233:53143;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER,                                                                                                                                                              MESSAGE, OPTIONS
Authorization: Digest username="004055", realm="asterisk", nonce="1715881394/ef0                                                                                                                                                             a29d7a6d6cff3bb47f14791db2a8e", uri="sip:pbx.com:5060", resp                                                                                                                                                             onse="52943c7716598c7a274376f7b7873b21", algorithm=MD5, cnonce="759a31a49309477a                                                                                                                                                             a66e50311813ca55", opaque="0dd6f5776cf88cfc", qop=auth, nc=00000001
Content-Length:  0


    -- Added contact 'sip:004055@10.38.2.233:53143;ob' to AOR '004055' with expi                                                                                                                                                             ration of 300 seconds
    -- Removed contact 'sip:004055@10.38.2.217:63664;ob' from AOR '004055' due t                                                                                                                                                             o remove existing
<--- Transmitting SIP response (584 bytes) to UDP:10.38.2.233:53143 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.38.2.233:53143;rport=53143;received=10.38.2.233;branch=z9hG4                                                                                                                                                             bKPj25f3a6ee98a74a1ca5c9f1027b96c008
Call-ID: bd0137f24cb9492db91abfcc5a5bfe11
From: "dialer-test-004055" <sip:004055@pbx.com>;tag=c80da8                                                                                                                                                             57a2e94c778ebed1362db4ebd9
To: "dialer-test-004055" <sip:004055@pbx.com>;tag=z9hG4bKP                                                                                                                                                             j25f3a6ee98a74a1ca5c9f1027b96c008
CSeq: 26796 REGISTER
Date: Thu, 16 May 2024 17:43:14 GMT
Contact: <sip:004055@10.38.2.233:53143;ob>;expires=299
Expires: 300
Server: FPBX-16.0.40.7(18.16.0)
Content-Length:  0


<--- Received SIP request (1140 bytes) from UDP:10.38.2.233:5060 --->
PUBLISH sip:004055@pbx.com:5060 SIP/2.0
Via: SIP/2.0/UDP 10.38.2.233:5060;rport;branch=z9hG4bKPj7b3c44ddbbd742b18b46eba7                                                                                                                                                             9ff4ea6e
Route: <sip:pbx.com:5060;lr>
Max-Forwards: 70
From: "dialer-test-004055" <sip:004055@pbx.com>;tag=0bfd45                                                                                                                                                             bc8b2344d8850cf7799a2a4e3c
To: "dialer-test-004055" <sip:004055@pbx.com>
Call-ID: 2d5a2e8792da4a6087646488a269a213
CSeq: 36564 PUBLISH
Event: presence
User-Agent: MicroSIP/3.21.3
Content-Type: application/pidf+xml
Content-Length:   573

<?xml version="1.0" encoding="UTF-8"?>
<presence entity="sip:004055@pbx.com:5060"                                                                                                                                                              arams:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid=                                                                                                                                                             "urn:ietf:params:xml:ns:pidf:rpid">
 <tuple id="pje4e0aa4c04f4443d91cb552b932cf5bb">
  <status>
   <basic>open</basic>
  </status>
  <timestamp>2024-05-16T17:43:14.419Z</timestamp>
  <note>Away</note>
 </tuple>
 <dm:person id="pj30e7ec55d0ec4698b3a56633e75dc867">
  <rpid:activities>
   <rpid:away />
  </rpid:activities>
  <dm:note>Away</dm:note>
 </dm:person>
</presence>

<--- Transmitting SIP response (630 bytes) to UDP:10.38.2.233:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.38.2.233:5060;rport=5060;received=10.38.2.233;branch=z9hG4bK                                                                                                                                                             Pj7b3c44ddbbd742b18b46eba79ff4ea6e
Call-ID: 2d5a2e8792da4a6087646488a269a213
From: "dialer-test-004055" <sip:004055@pbx.com>;tag=0bfd45                                                                                                                                                             bc8b2344d8850cf7799a2a4e3c
To: "dialer-test-004055" <sip:004055@pbx.com>;tag=z9hG4bKP                                                                                                                                                             j7b3c44ddbbd742b18b46eba79ff4ea6e
CSeq: 36564 PUBLISH
WWW-Authenticate: Digest realm="asterisk",nonce="1715881394/ef0a29d7a6d6cff3bb47                                                                                                                                                             f14791db2a8e",opaque="74e1b41b3532c6b0",algorithm=MD5,qop="auth"
Server: FPBX-16.0.40.7(18.16.0)
Content-Length:  0


  == Contact 004055/sip:004055@10.38.2.217:63664;ob has been deleted
  == Endpoint 004055 is now Unreachable
<--- Transmitting SIP request (433 bytes) to UDP:10.38.2.233:53143 --->
OPTIONS sip:004055@10.38.2.233:53143;ob SIP/2.0
Via: SIP/2.0/UDP 10.38.43.75:5060;rport;branch=z9hG4bKPjd07b1d40-e0b8-45c6-96c9-                                                                                                                                                             c2291d0caa0d
From: <sip:004055@10.38.43.75>;tag=96e50f2c-c3ac-47c7-870b-732c74916552
To: <sip:004055@10.38.2.233;ob>
Contact: <sip:004055@10.38.43.75:5060>
Call-ID: 1ca6c4dd-87bc-4aae-b62d-e5ff3a83a25b
CSeq: 1366 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.40.7(18.16.0)
Content-Length:  0


<--- Received SIP response (796 bytes) from UDP:10.38.2.233:53143 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.38.43.75:5060;rport=5060;received=10.38.43.75;branch=z9hG4bK                                                                                                                                                             Pjd07b1d40-e0b8-45c6-96c9-c2291d0caa0d
Call-ID: 1ca6c4dd-87bc-4aae-b62d-e5ff3a83a25b
From: <sip:004055@10.38.43.75>;tag=96e50f2c-c3ac-47c7-870b-732c74916552
To: <sip:004055@10.38.2.233;ob>;tag=z9hG4bKPjd07b1d40-e0b8-45c6-96c9-c2291d0caa0                                                                                                                                                             d
CSeq: 1366 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER,                                                                                                                                                              MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, applicatio                                                                                                                                                             n/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposin                                                                                                                                                             g+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


  == Endpoint 004055 is now Reachable
    -- Contact 004055/sip:004055@10.38.2.233:53143;ob is now Reachable.  RTT: 12                                                                                                                                                             .115 msec
freepbx*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
[root@freepbx ~]# sngrep
[root@freepbx ~]# ti<--- Received SIP request (686 bytes) from UDP:10.38.2.233:53143 --->

-bash: syntax error near unexpected token `('
[root@freepbx ~]# REGISTER sip:pbx.com:5060 SIP/2.0
5
-bash: REGISTER: command not found
[root@freepbx ~]# Via: SIP/2.0/UDP 10.38.2.233:53143;rport;branch=z9hG4bKPj0ca0bf98c4fa42c793cfa13                                                                                                                                                             f8eda0a61

-bash: Via:: command not found
-bash: rport: command not found
-bash: f8eda0a61: command not found
[root@freepbx ~]# Route: <sip:pbx.com:5060;lr>
-bash: syntax error near unexpected token `newline'
[root@freepbx ~]# Max-Forwards: 70
1
-bash: Max-Forwards:: command not found
[root@freepbx ~]# From: "dialer-test-004055" <sip:004055@pbx.com>;tag=c80da8                                                                                                                                                             57a2e94c778ebed1362db4ebd9
-bash: syntax error near unexpected token `;'
[root@freepbx ~]# To: "dialer-test-004055" <sip:004055@pbx.com>
-bash: syntax error near unexpected token `newline'
[root@freepbx ~]# Call-ID: bd0137f24cb9492db91abfcc5a5bfe11
p
-bash: Call-ID:: command not found
[root@freepbx ~]# CSeq: 26795 REGISTER
0
-bash: CSeq:: command not found
[root@freepbx ~]# User-Agent: MicroSIP/3.21.3

-bash: User-Agent:: command not found
[root@freepbx ~]# Contact: "dialer-test-004055" <sip:004055@10.38.2.233:53143;ob>
-bash: syntax error near unexpected token `newline'
[root@freepbx ~]# Expires: 300
P
-bash: Expires:: command not found
[root@freepbx ~]# Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER,                                                                                                                                                              MESSAGE, OPTIONS

-bash: Allow:: command not found
[root@freepbx ~]# Content-Length:  0
-
-bash: Content-Length:: command not found
[root@freepbx ~]#
[root@freepbx ~]#
[root@freepbx ~]# <--- Transmitting SIP response (633 bytes) to UDP:10.38.2.233:53143 --->
-bash: syntax error near unexpected token `('
[root@freepbx ~]# SIP/2.0 401 Unauthorized

-bash: SIP/2.0: No such file or directory
[root@freepbx ~]# Via: SIP/2.0/UDP 10.38.2.233:53143;rport=53143;received=10.38.2.233;branch=z9hG4                                                                                                                                                             bKPj0ca0bf98c4fa42c793cfa13f8eda0a61
p
-bash: Via:: command not found
-bash: bKPj0ca0bf98c4fa42c793cfa13f8eda0a61: command not found
[root@freepbx ~]# Call-ID: bd0137f24cb9492db91abfcc5a5bfe11
2
-bash: Call-ID:: command not found
[root@freepbx ~]# From: "dialer-test-004055" <sip:004055@pbx.com>;tag=c80da8                                                                                                                                                             57a2e94c778ebed1362db4ebd9
-bash: syntax error near unexpected token `;'
[root@freepbx ~]# To: "dialer-test-004055" <sip:004055@pbx.com>;tag=z9hG4bKP                                                                                                                                                             j0ca0bf98c4fa42c793cfa13f8eda0a61
-bash: syntax error near unexpected token `;'
[root@freepbx ~]# CSeq: 26795 REGISTER

-bash: CSeq:: command not found
[root@freepbx ~]# WWW-Authenticate: Digest realm="asterisk",nonce="1715881394/ef0a29d7a6d6cff3bb47                                                                                                                                                             f14791db2a8e",opaque="0dd6f5776cf88cfc",algorithm=MD5,qop="auth"
a
-bash: WWW-Authenticate:: command not found
[root@freepbx ~]# Server: FPBX-16.0.40.7(18.16.0)
-bash: syntax error near unexpected token `('
[root@freepbx ~]# Content-Length:  0
v
-bash: Content-Length:: command not found
[root@freepbx ~]#
[root@freepbx ~]#
[root@freepbx ~]# <--- Received SIP request (995 bytes) from UDP:10.38.2.233:53143 --->
-bash: syntax error near unexpected token `('
[root@freepbx ~]# REGISTER sip:pbx.com:5060 SIP/2.0

-bash: REGISTER: command not found
[root@freepbx ~]# Via: SIP/2.0/UDP 10.38.2.233:53143;rport;branch=z9hG4bKPj25f3a6ee98a74a1ca5c9f10                                                                                                                                                             27b96c008
:
-bash: Via:: command not found
-bash: rport: command not found
-bash: 27b96c008: command not found
[root@freepbx ~]# Route: <sip:pbx.com:5060;lr>
-bash: syntax error near unexpected token `newline'
[root@freepbx ~]# Max-Forwards: 70
-
-bash: Max-Forwards:: command not found
[root@freepbx ~]# From: "dialer-test-004055" <sip:004055@pbx.com>;tag=c80da8                                                                                                                                                             57a2e94c778ebed1362db4ebd9
-bash: syntax error near unexpected token `;'
[root@freepbx ~]# To: "dialer-test-004055" <sip:004055@pbx.com>
-bash: syntax error near unexpected token `newline'
[root@freepbx ~]# Call-ID: bd0137f24cb9492db91abfcc5a5bfe11
 -bash: Call-ID:: command not found
[root@freepbx ~]# CSeq: 26796 REGISTER
-bash: CSeq:: command not found
[root@freepbx ~]# User-Agent: MicroSIP/3.21.3
-bash: User-Agent:: command not found
[root@freepbx ~]# Contact: "dialer-test-004055" <sip:004055@10.38.2.233:53143;ob>
-bash: syntax error near unexpected token `newline'
[root@freepbx ~]# Expires: 300
-bash: Expires:: command not found
[root@freepbx ~]# Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER,                                                                                                                                                              MESSAGE, OPTIONS
-bash: Allow:: command not found
[root@freepbx ~]# Authorization: Digest username="004055", realm="asterisk", nonce="1715881394/ef0                                                                                                                                                             a29d7a6d6cff3bb47f14791db2a8e", uri="sip:pbx.com:5060", resp
-bash: Authorization:: command not found
[root@freepbx ~]# Content-Length:  0
-bash: Content-Length:: command not found
[root@freepbx ~]#
[root@freepbx ~]#
[root@freepbx ~]#     -- Added contact 'sip:004055
>     -- Removed contact 'sip:004055@10.38.2.217:63664;ob' from AOR '004055' due t                                                                                                                                                             o remove
> <--- Transmitting SIP response (584 bytes) to UDP:10.38.2.233:531
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.38.2.233:53143;rport=53143;received=10.38.2.233;branch=z9hG4                                                                                                                                                             bKPj25f3a6ee98a74a1ca5c9f1027b96c008
> Call-ID: bd0137f24cb9492db91abfcc5a5bfe11
> From: "dialer-test-004055" <sip:004055@p
> To: "dialer-test-0
> CSeq: 26796 REGISTER
> Date:
> Contact: <sip:004055@10.38.2.233:53143;ob>;expires=299
> Expires: 300
> Server: FP
> Content-Length:  0
>
>
> <--- Received SIP request (1140 bytes) from UDP:10.38.2.233:5060 --->
> PUBLISH sip:004055@pbx.com:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.38.2.233:5060;rport;branch=z9hG4bKPj7b3c44ddbbd742b18b46eba7
>`Preformatted text`

Can someone help me understand why please?

2

On Thursday 16 May 2024 at 20:10:55, dereck22dev via Asterisk Community wrote:

Hello guy’s, i make a call to an external number but neither party can hear
the other. after 30s the call is automatically disconnected.

I recommend that you ask on the FreePBX forums, since that is what you are
using.

Asterisk is simply one small(ish) part of FreePBX, and the configuration they
have built around it is far more important in dealing with problems like this
than the way Asterisk itself would be used on its own.

Antony.

–
Is it venison for dinner again? Oh deer.

                                               Please reply to the list;
                                                     please *don't* CC me.
1 Like
3

What type of firewall are you using?
Do you have proper ports opened for traffic?

4

hello @FrancoSmash ,
i was using iptable and i opened ports 5*** for signaling and 10000:20000 for rtp transport. since that didn’t work, i authorized the whole ports and stopped iptable.

5

change your target from 10000-20000 to 8800-49990
& I’d change iptables for firewalld.service

6

On Thursday 16 May 2024 at 22:42:39, FrancoSmash via Asterisk Community wrote:

change your target from 10000-20000 to 8800-49990

Why?

On Thursday 16 May 2024 at 20:10:55, dereck22dev via Asterisk Community wrote:

//rtp_custum.conf

[general]
rtpstart=10000
rtpend=20000

So why set the firewall rules with a wider range?

& I’d change iptables for firewalld.service

Is that likely to make a functional difference, or is this just a personal
preference?

Antony.

–
“Why do you keep on writing such awful software?”
“No comment.”

                                               Please reply to the list;
                                                     please *don't* CC me.
7

For me it’s a “personal preference” I’ve gotten quite comfortable with managing firewalld.service

As for RTP port range for my agents on sip phones it works better for us to have a wider range of ports.

1 Like
8

I’ve do it but it doesn’t work for me. anyway, 10000:20000 is already assigned for rtp stream. To make sure that the firewall was the problem, I stop it, but it still doesn’t work.
I’m still trying to find out what’s wrong.

9

Your log contains no INVITE transactions, so it is not possible to see where the media should be going.

Your logs do not have RTP logging enabled, so it is not possible to see what media is arriving, from where, and to where it is being sent.

The entity encoding of the log, and the spurious initial “>”, make it difficult to read.

The above is the wrong file for FreePBX.

10

Add “,!all” (without the quotes) to that line.

1 Like
11

I believe the !all needs to be at the beginning, not the end of the line.

1 Like
12

Most likely it’s a firewall or NAT issue. You’ll need to look at what happens to the RTP packets, as the SIP packets won’t directly help you if this is the case.

13

The SIP packets will show where you are being told to send RTP.

15

This is what I get after enabling rtp debug. this case is a call between two extensions

  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Audio CoS mark 5
    -- Executing [004059@from-internal:1] GotoIf("PJSIP/004058-00000063", "0?ext-local,*004059,1") in new stack
    -- Executing [004059@from-internal:2] GotoIf("PJSIP/004058-00000063", "0?ext-local,004059,1:followme-check,004059,1") in new stack
    -- Goto (followme-check,004059,1)
    -- Executing [004059@followme-check:1] Gosub("PJSIP/004058-00000063", "followme-sub,004059,1()") in new stack
    -- Executing [004059@followme-sub:1] Set("PJSIP/004058-00000063", "__FMFM=TRUE") in new stack
    -- Executing [004059@followme-sub:2] GotoIf("PJSIP/004058-00000063", "0?skipclid") in new stack
    -- Executing [004059@followme-sub:3] Macro("PJSIP/004058-00000063", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("PJSIP/004058-00000063", "TOUCH_MONITOR=1715945563.99") in new stack
    -- Executing [s@macro-user-callerid:2] Set("PJSIP/004058-00000063", "CHANCONTEXT=") in new stack
    -- Executing [s@macro-user-callerid:3] Progress("PJSIP/004058-00000063", "") in new stack
    -- Executing [s@macro-user-callerid:4] Set("PJSIP/004058-00000063", "CHANCONTEXT=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("PJSIP/004058-00000063", "CHANEXTENCONTEXT=004058-00000063") in new stack
    -- Executing [s@macro-user-callerid:6] Set("PJSIP/004058-00000063", "CHANEXTEN=004058-00000063") in new stack
    -- Executing [s@macro-user-callerid:7] Set("PJSIP/004058-00000063", "CALLERID(number)=004058") in new stack
    -- Executing [s@macro-user-callerid:8] Set("PJSIP/004058-00000063", "AMPUSER=004058") in new stack
    -- Executing [s@macro-user-callerid:9] Set("PJSIP/004058-00000063", "HOTDESCKCHAN=004058-00000063") in new stack
    -- Executing [s@macro-user-callerid:10] Set("PJSIP/004058-00000063", "HOTDESKEXTEN=004058") in new stack
    -- Executing [s@macro-user-callerid:11] Set("PJSIP/004058-00000063", "HOTDESKCALL=0") in new stack
    -- Executing [s@macro-user-callerid:12] ExecIf("PJSIP/004058-00000063", "0?Set(HOTDESKCALL=1)") in new stack
    -- Executing [s@macro-user-callerid:13] ExecIf("PJSIP/004058-00000063", "0?Set(CALLERID(name)=)") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("PJSIP/004058-00000063", "0?report") in new stack
    -- Executing [s@macro-user-callerid:15] ExecIf("PJSIP/004058-00000063", "1?Set(REALCALLERIDNUM=004058)") in new stack
    -- Executing [s@macro-user-callerid:16] Set("PJSIP/004058-00000063", "AMPUSER=004058") in new stack
    -- Executing [s@macro-user-callerid:17] GotoIf("PJSIP/004058-00000063", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:18] Set("PJSIP/004058-00000063", "AMPUSERCIDNAME=CATARINATEST004058") in new stack
    -- Executing [s@macro-user-callerid:19] ExecIf("PJSIP/004058-00000063", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
    -- Executing [s@macro-user-callerid:20] GotoIf("PJSIP/004058-00000063", "0?report") in new stack
    -- Executing [s@macro-user-callerid:21] Set("PJSIP/004058-00000063", "AMPUSERCID=004058") in new stack
    -- Executing [s@macro-user-callerid:22] Set("PJSIP/004058-00000063", "__DIAL_OPTIONS=HhTtr") in new stack
    -- Executing [s@macro-user-callerid:23] Set("PJSIP/004058-00000063", "CALLERID(all)="CATARINATEST004058" <004058>") in new stack
    -- Executing [s@macro-user-callerid:24] ExecIf("PJSIP/004058-00000063", "0?Set(CUSDIAL=)") in new stack
    -- Executing [s@macro-user-callerid:25] ExecIf("PJSIP/004058-00000063", "0?Set(CALLERID(all)="CATARINATEST004058" <004058>)") in new stack
    -- Executing [s@macro-user-callerid:26] GotoIf("PJSIP/004058-00000063", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:27] ExecIf("PJSIP/004058-00000063", "0?Set(GROUP(concurrency_limit)=004058)") in new stack
    -- Executing [s@macro-user-callerid:28] ExecIf("PJSIP/004058-00000063", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:29] NoOp("PJSIP/004058-00000063", "Macro Depth is 1") in new stack
    -- Executing [s@macro-user-callerid:30] GotoIf("PJSIP/004058-00000063", "1?report2:macroerror") in new stack
    -- Goto (macro-user-callerid,s,31)
    -- Executing [s@macro-user-callerid:31] GotoIf("PJSIP/004058-00000063", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:32] ExecIf("PJSIP/004058-00000063", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
    -- Executing [s@macro-user-callerid:33] Set("PJSIP/004058-00000063", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:34] GotoIf("PJSIP/004058-00000063", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,50)
    -- Executing [s@macro-user-callerid:50] Set("PJSIP/004058-00000063", "CALLERID(number)=004058") in new stack
    -- Executing [s@macro-user-callerid:51] Set("PJSIP/004058-00000063", "CALLERID(name)=CATARINATEST004058") in new stack
    -- Executing [s@macro-user-callerid:52] GotoIf("PJSIP/004058-00000063", "0?cnum") in new stack
    -- Executing [s@macro-user-callerid:53] Set("PJSIP/004058-00000063", "__MCNUM=004058") in new stack
    -- Executing [s@macro-user-callerid:54] Set("PJSIP/004058-00000063", "__MCNAME=CATARINATEST004058") in new stack
    -- Executing [s@macro-user-callerid:55] Set("PJSIP/004058-00000063", "__MCEXTEN=004058") in new stack
    -- Executing [s@macro-user-callerid:56] Set("PJSIP/004058-00000063", "__MCORGCHAN=PJSIP/004058-00000063") in new stack
    -- Executing [s@macro-user-callerid:57] Set("PJSIP/004058-00000063", "CDR(cnam)=CATARINATEST004058") in new stack
    -- Executing [s@macro-user-callerid:58] Set("PJSIP/004058-00000063", "CDR(cnum)=004058") in new stack
    -- Executing [s@macro-user-callerid:59] Set("PJSIP/004058-00000063", "CHANNEL(language)=fr") in new stack
    -- Executing [004059@followme-sub:4] Set("PJSIP/004058-00000063", "DIAL_OPTIONS=HhTtrI") in new stack
    -- Executing [004059@followme-sub:5] Set("PJSIP/004058-00000063", "CONNECTEDLINE(num,i)=004059") in new stack
    -- Executing [004059@followme-sub:6] Gosub("PJSIP/004058-00000063", "sub-presencestate-display,s,1(004059)") in new stack
    -- Executing [s@sub-presencestate-display:1] Goto("PJSIP/004058-00000063", "state-available,1") in new stack
    -- Goto (sub-presencestate-display,state-available,1)
    -- Executing [state-available@sub-presencestate-display:1] Set("PJSIP/004058-00000063", "PRESENCESTATE_DISPLAY=(Disponible)") in new stack
    -- Executing [state-available@sub-presencestate-display:2] Return("PJSIP/004058-00000063", "") in new stack
    -- Executing [004059@followme-sub:7] Set("PJSIP/004058-00000063", "CONNECTEDLINE(name)=CATARINATEST004059(Disponible)") in new stack
    -- Executing [004059@followme-sub:8] Set("PJSIP/004058-00000063", "FM_DIALSTATUS=NOT_INUSE") in new stack
    -- Executing [004059@followme-sub:9] Set("PJSIP/004058-00000063", "__EXTTOCALL=004059") in new stack
    -- Executing [004059@followme-sub:10] Set("PJSIP/004058-00000063", "__PICKUPMARK=004059") in new stack
    -- Executing [004059@followme-sub:11] Macro("PJSIP/004058-00000063", "blkvm-setifempty,") in new stack
    -- Executing [s@macro-blkvm-setifempty:1] GotoIf("PJSIP/004058-00000063", "1?init") in new stack
    -- Goto (macro-blkvm-setifempty,s,4)
    -- Executing [s@macro-blkvm-setifempty:4] Set("PJSIP/004058-00000063", "__BLKVM_CHANNEL=PJSIP/004058-00000063") in new stack
    -- Executing [s@macro-blkvm-setifempty:5] Set("PJSIP/004058-00000063", "SHARED(BLKVM,PJSIP/004058-00000063)=TRUE") in new stack
    -- Executing [s@macro-blkvm-setifempty:6] Set("PJSIP/004058-00000063", "GOSUB_RETVAL=TRUE") in new stack
    -- Executing [s@macro-blkvm-setifempty:7] MacroExit("PJSIP/004058-00000063", "") in new stack
    -- Executing [004059@followme-sub:12] GotoIf("PJSIP/004058-00000063", "1?skipov") in new stack
    -- Goto (followme-sub,004059,15)
    -- Executing [004059@followme-sub:15] Set("PJSIP/004058-00000063", "RRNODEST=") in new stack
    -- Executing [004059@followme-sub:16] Set("PJSIP/004058-00000063", "__NODEST=004059") in new stack
    -- Executing [004059@followme-sub:17] GosubIf("PJSIP/004058-00000063", "0?sub-fmsetcid,s,1()") in new stack
    -- Executing [004059@followme-sub:18] GotoIf("PJSIP/004058-00000063", "1?skipprepend") in new stack
    -- Goto (followme-sub,004059,20)
    -- Executing [004059@followme-sub:20] Set("PJSIP/004058-00000063", "RecordMethod=Group") in new stack
    -- Executing [004059@followme-sub:21] Gosub("PJSIP/004058-00000063", "sub-record-check,s,1(exten,004059,)") in new stack
    -- Executing [s@sub-record-check:1] GotoIf("PJSIP/004058-00000063", "0?initialized") in new stack
    -- Executing [s@sub-record-check:2] Set("PJSIP/004058-00000063", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:3] Set("PJSIP/004058-00000063", "NOW=1715945563") in new stack
    -- Executing [s@sub-record-check:4] Set("PJSIP/004058-00000063", "__DAY=17") in new stack
    -- Executing [s@sub-record-check:5] Set("PJSIP/004058-00000063", "__MONTH=05") in new stack
    -- Executing [s@sub-record-check:6] Set("PJSIP/004058-00000063", "__YEAR=2024") in new stack
    -- Executing [s@sub-record-check:7] Set("PJSIP/004058-00000063", "__TIMESTR=20240517-113243") in new stack
    -- Executing [s@sub-record-check:8] Set("PJSIP/004058-00000063", "__FROMEXTEN=004058") in new stack
    -- Executing [s@sub-record-check:9] Set("PJSIP/004058-00000063", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:10] NoOp("PJSIP/004058-00000063", "Recordings initialized") in new stack
    -- Executing [s@sub-record-check:11] ExecIf("PJSIP/004058-00000063", "1?Set(ARG3=dontcare)") in new stack
    -- Executing [s@sub-record-check:12] Set("PJSIP/004058-00000063", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:13] ExecIf("PJSIP/004058-00000063", "0?Set(REC_STATUS=NO)") in new stack
    -- Executing [s@sub-record-check:14] GotoIf("PJSIP/004058-00000063", "5?checkaction") in new stack
    -- Goto (sub-record-check,s,17)
    -- Executing [s@sub-record-check:17] GotoIf("PJSIP/004058-00000063", "1?sub-record-check,exten,1") in new stack
    -- Goto (sub-record-check,exten,1)
    -- Executing [exten@sub-record-check:1] NoOp("PJSIP/004058-00000063", "Exten Recording Check between 004058 and 004059") in new stack
    -- Executing [exten@sub-record-check:2] Set("PJSIP/004058-00000063", "CALLTYPE=internal") in new stack
    -- Executing [exten@sub-record-check:3] ExecIf("PJSIP/004058-00000063", "0?Set(CALLTYPE=)") in new stack
    -- Executing [exten@sub-record-check:4] Set("PJSIP/004058-00000063", "CALLEE=dontcare") in new stack
    -- Executing [exten@sub-record-check:5] ExecIf("PJSIP/004058-00000063", "0?Set(CALLEE=dontcare)") in new stack
    -- Executing [exten@sub-record-check:6] GotoIf("PJSIP/004058-00000063", "0?callee") in new stack
    -- Executing [exten@sub-record-check:7] GotoIf("PJSIP/004058-00000063", "1?caller") in new stack
    -- Goto (sub-record-check,exten,13)
    -- Executing [exten@sub-record-check:13] Set("PJSIP/004058-00000063", "RECMODE=dontcare") in new stack
    -- Executing [exten@sub-record-check:14] Set("PJSIP/004058-00000063", "CALLERRECMODE=dontcare") in new stack
    -- Executing [exten@sub-record-check:15] Set("PJSIP/004058-00000063", "CALEERECMODE=dontcare") in new stack
    -- Executing [exten@sub-record-check:16] GotoIf("PJSIP/004058-00000063", "0?processnormal") in new stack
    -- Executing [exten@sub-record-check:17] ExecIf("PJSIP/004058-00000063", "0?Set(RECMODE=dontcare)") in new stack
    -- Executing [exten@sub-record-check:18] ExecIf("PJSIP/004058-00000063", "0?Set(RECMODE=dontcare)") in new stack
    -- Executing [exten@sub-record-check:19] ExecIf("PJSIP/004058-00000063", "0?Set(RECMODE=dontcare)") in new stack
    -- Executing [exten@sub-record-check:20] ExecIf("PJSIP/004058-00000063", "0?Set(RECMODE=dontcare)") in new stack
    -- Executing [exten@sub-record-check:21] ExecIf("PJSIP/004058-00000063", "0?Set(RECMODE=dontcare)") in new stack
    -- Executing [exten@sub-record-check:22] ExecIf("PJSIP/004058-00000063", "1?Set(RECMODE=dontcare)") in new stack
    -- Executing [exten@sub-record-check:23] Gosub("PJSIP/004058-00000063", "recordcheck,1(dontcare,internal,004059)") in new stack
    -- Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/004058-00000063", "Starting recording check against dontcare") in new stack
    -- Executing [recordcheck@sub-record-check:2] Goto("PJSIP/004058-00000063", "dontcare") in new stack
    -- Goto (sub-record-check,recordcheck,3)
    -- Executing [recordcheck@sub-record-check:3] Return("PJSIP/004058-00000063", "") in new stack
    -- Executing [exten@sub-record-check:24] Return("PJSIP/004058-00000063", "") in new stack
    -- Executing [004059@followme-sub:22] GotoIf("PJSIP/004058-00000063", "1?skipdring") in new stack
    -- Goto (followme-sub,004059,25)
    -- Executing [004059@followme-sub:25] Set("PJSIP/004058-00000063", "STRATEGY=ringallv2-prim") in new stack
    -- Executing [004059@followme-sub:26] Set("PJSIP/004058-00000063", "__RVOL=") in new stack
    -- Executing [004059@followme-sub:27] GotoIf("PJSIP/004058-00000063", "1?skipsimple") in new stack
    -- Goto (followme-sub,004059,30)
    -- Executing [004059@followme-sub:30] Set("PJSIP/004058-00000063", "RingGroupMethod=ringallv2-prim") in new stack
    -- Executing [004059@followme-sub:31] Set("PJSIP/004058-00000063", "_FMGRP=004059") in new stack
    -- Executing [004059@followme-sub:32] GotoIf("PJSIP/004058-00000063", "1?DIALGRP") in new stack
    -- Goto (followme-sub,004059,36)
    -- Executing [004059@followme-sub:36] ExecIf("PJSIP/004058-00000063", "1?Set(DOPTS=HhTtrI):Set(DOPTS=m(Ring)HhTtI)") in new stack
    -- Executing [004059@followme-sub:37] Set("PJSIP/004058-00000063", "__ALT_CONFIRM_MSG=") in new stack
    -- Executing [004059@followme-sub:38] GotoIf("PJSIP/004058-00000063", "0?doconfirm") in new stack
    -- Executing [004059@followme-sub:39] GotoIf("PJSIP/004058-00000063", "1?ringallv21") in new stack
    -- Goto (followme-sub,004059,42)
    -- Executing [004059@followme-sub:42] Macro("PJSIP/004058-00000063", "dial,27,HhTtrI,004059") in new stack
    -- Executing [s@macro-dial:1] NoOp("PJSIP/004058-00000063", "Blind Transfer: , Attended Transfer: , User: 004058, Alert Info: ") in new stack
    -- Executing [s@macro-dial:2] Set("PJSIP/004058-00000063", "__CRM_SOURCE=004058") in new stack
    -- Executing [s@macro-dial:3] ExecIf("PJSIP/004058-00000063", "1?Set(ALERT_INFO=)") in new stack
    -- Executing [s@macro-dial:4] ExecIf("PJSIP/004058-00000063", "0?Set(ALERT_INFO=)") in new stack
    -- Executing [s@macro-dial:5] ExecIf("PJSIP/004058-00000063", "0?Set(ALERT_INFO=)") in new stack
    -- Executing [s@macro-dial:6] ExecIf("PJSIP/004058-00000063", "0?Set(CHANNEL(musicclass)=)") in new stack
    -- Executing [s@macro-dial:7] AGI("PJSIP/004058-00000063", "agi://127.0.0.1/dialparties.agi") in new stack
 agi://127.0.0.1/dialparties.agi: Starting New Dialparties.agi
 agi://127.0.0.1/dialparties.agi: Caller ID name is 'CATARINATEST004058' number is '004058'
 agi://127.0.0.1/dialparties.agi: CW Ignore is:
 agi://127.0.0.1/dialparties.agi: CF Ignore is:
 agi://127.0.0.1/dialparties.agi: CW IN_USE/BUSY is: 1
 agi://127.0.0.1/dialparties.agi: Ringgroup confirm is  :
 agi://127.0.0.1/dialparties.agi: Methodology of ring is  'ringallv2-prim'
    -- agi://127.0.0.1/dialparties.agi: Added extension 004059 to extension map
    -- agi://127.0.0.1/dialparties.agi: Extension 004059 cf is disabled
    -- agi://127.0.0.1/dialparties.agi: Extension 004059 do not disturb is disabled
 agi://127.0.0.1/dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
 agi://127.0.0.1/dialparties.agi: Extension 004059 has ExtensionState: 0
  == agi://127.0.0.1/dialparties.agi: Discovered PJSIP Endpoint PJSIP/004059
    -- agi://127.0.0.1/dialparties.agi: Ended up with real PJSIP Dial string PJSIP/004059/sip:004059@10.38.2.219:65275;ob
    -- agi://127.0.0.1/dialparties.agi: dbset CALLTRACE/004059 to 004058
    -- agi://127.0.0.1/dialparties.agi: Filtered ARG3: 004059
    -- agi://127.0.0.1/dialparties.agi: RING ALL V2 :
 agi://127.0.0.1/dialparties.agi: RVOL_MODE ''
 agi://127.0.0.1/dialparties.agi: RVOL is:
 agi://127.0.0.1/dialparties.agi: RVOLPARENT is:
    -- <PJSIP/004058-00000063>AGI Script agi://127.0.0.1/dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:8] GotoIf("PJSIP/004058-00000063", "1?normdial") in new stack
    -- Goto (macro-dial,s,11)
    -- Executing [s@macro-dial:11] NoOp("PJSIP/004058-00000063", "Returned from dialparties with groups to dial") in new stack
    -- Executing [s@macro-dial:12] NoOp("PJSIP/004058-00000063", "ringall array  ") in new stack
    -- Executing [s@macro-dial:13] NoOp("PJSIP/004058-00000063", "ds= PJSIP/004059/sip:004059@10.38.2.219:65275;ob,27,HhTtrIM(auto-blkvm) ") in new stack
    -- Executing [s@macro-dial:14] NoOp("PJSIP/004058-00000063", "dsextra= ") in new stack
    -- Executing [s@macro-dial:15] Set("PJSIP/004058-00000063", "ds=PJSIP/004059/sip:004059@10.38.2.219:65275;ob,27,HhTtrIM(auto-blkvm)") in new stack
    -- Executing [s@macro-dial:16] NoOp("PJSIP/004058-00000063", "ds= PJSIP/004059/sip:004059@10.38.2.219:65275;ob,27,HhTtrIM(auto-blkvm)") in new stack
    -- Executing [s@macro-dial:17] Set("PJSIP/004058-00000063", "__FMGL_DIAL=") in new stack
    -- Executing [s@macro-dial:18] Set("PJSIP/004058-00000063", "LOOPCNT=1") in new stack
    -- Executing [s@macro-dial:19] Set("PJSIP/004058-00000063", "ITER=1") in new stack
    -- Executing [s@macro-dial:20] Set("PJSIP/004058-00000063", "__EXTTOCALL=004059") in new stack
    -- Executing [s@macro-dial:21] Set("PJSIP/004058-00000063", "__MCEXTTOCALL=004059") in new stack
    -- Executing [s@macro-dial:22] NoOp("PJSIP/004058-00000063", "Working with 004059") in new stack
    -- Executing [s@macro-dial:23] ExecIf("PJSIP/004058-00000063", "0?Set(HASH(__SIPHEADERS,Alert-Info)=Normal;volume=)") in new stack
    -- Executing [s@macro-dial:24] ExecIf("PJSIP/004058-00000063", "0?Set(HASH(__SIPHEADERS,Alert-Info)=Normal;volume=)") in new stack
    -- Executing [s@macro-dial:25] Set("PJSIP/004058-00000063", "ITER=2") in new stack
    -- Executing [s@macro-dial:26] GotoIf("PJSIP/004058-00000063", "0?ndloopbegin") in new stack
    -- Executing [s@macro-dial:27] Macro("PJSIP/004058-00000063", "dial-ringall-predial-hook,") in new stack
    -- Executing [s@macro-dial-ringall-predial-hook:1] MacroExit("PJSIP/004058-00000063", "") in new stack
    -- Executing [s@macro-dial:28] ExecIf("PJSIP/004058-00000063", "0?Set(CWRING=r(callwaiting)):Set(CWRING=)") in new stack
    -- Executing [s@macro-dial:29] ExecIf("PJSIP/004058-00000063", "0?Set(ds=PJSIP/004059/sip:004059@10.38.2.219:65275;ob,27,HhTtrIM(auto-blkvm)g)") in new stack
    -- Executing [s@macro-dial:30] Dial("PJSIP/004058-00000063", "PJSIP/004059/sip:004059@10.38.2.219:65275;ob,27,HhTtrIM(auto-blkvm)b(func-apply-sipheaders^s^1),") in new stack
    -- PJSIP/004059-00000064 Internal Gosub(func-apply-sipheaders,s,1) start
    -- Executing [s@func-apply-sipheaders:1] ExecIf("PJSIP/004059-00000064", "0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
    -- Executing [s@func-apply-sipheaders:2] NoOp("PJSIP/004059-00000064", "Applying SIP Headers to channel PJSIP/004059-00000064") in new stack
    -- Executing [s@func-apply-sipheaders:3] Set("PJSIP/004059-00000064", "localchan=004059-00000064") in new stack
    -- Executing [s@func-apply-sipheaders:4] Set("PJSIP/004059-00000064", "DialMCEXT=004059") in new stack
    -- Executing [s@func-apply-sipheaders:5] Set("PJSIP/004059-00000064", "CHANNEL(hangup_handler_push)=app-missedcall-hangup,004059,1") in new stack
    -- Executing [s@func-apply-sipheaders:6] Set("PJSIP/004059-00000064", "TECH=PJSIP") in new stack
    -- Executing [s@func-apply-sipheaders:7] Set("PJSIP/004059-00000064", "SIPHEADERKEYS=") in new stack
    -- Executing [s@func-apply-sipheaders:8] While("PJSIP/004059-00000064", "0") in new stack
    -- Jumping to priority 14
    -- Executing [s@func-apply-sipheaders:15] Return("PJSIP/004059-00000064", "") in new stack
  == Spawn extension (func-apply-sipheaders, s, 15) exited non-zero on 'PJSIP/004059-00000064'
    -- PJSIP/004059-00000064 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
    -- Called PJSIP/004059/sip:004059@10.38.2.219:65275;ob
  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Audio CoS mark 5
    -- Connected line update to PJSIP/004058-00000063 prevented.
    -- PJSIP/004059-00000064 is ringing
Sent RTP packet to      10.38.2.206:4018 (type 00, seq 018553, ts 000160, len 000160)
Sent RTP packet to      10.38.2.206:4018 (type 00, seq 018554, ts 000320, len 000160)
Sent RTP packet to      10.38.2.206:4018 (type 00, seq 018555, ts 000480, len 000160)
Sent RTP packet to      10.38.2.206:4018 (type 00, seq 018759, ts 033120, len 000160)
Sent RTP packet to      10.38.2.206:4018 (type 00, seq 018760, ts 033280, len 000160)
Sent RTP packet to      10.38.2.206:4018 (type 00, seq 018761, ts 033440, len 000160)
Sent RTP packet to      10.38.2.206:4018 (type 00, seq 018762, ts 033600, len 000160)
Sent RTP packet to      10.38.2.206:4018 (type 00, seq 018763, ts 033760, len 000160)
Sent RTP packet to      10.38.2.206:4018 (type 00, seq 018764, ts 033920, len 000160)
Sent RTP packet to      10.38.2.206:4018 (type 00, seq 018765, ts 034080, len 000160)
Sent RTP packet to      10.38.2.206:4018 (type 00, seq 018766, ts 034240, len 000160)
Sent RTP packet to      10.38.2.206:4018 (type 00, seq 018767, ts 034400, len 000160)
Sent RTP packet to      10.38.2.206:4018 (type 00, seq 018768, ts 034560, len 000160)
    -- PJSIP/004059-00000064 answered PJSIP/004058-00000063
    -- Executing [s@macro-auto-blkvm:1] ExecIf("PJSIP/004059-00000064", "0?Set(CDR(recordingfile)=.wav)") in new stack
Sent RTP packet to      10.38.2.206:4018 (type 00, seq 018769, ts 034720, len 000160)
    -- Executing [s@macro-auto-blkvm:2] Set("PJSIP/004059-00000064", "__MACRO_RESULT=") in new stack
    -- Executing [s@macro-auto-blkvm:3] Set("PJSIP/004059-00000064", "CFIGNORE=") in new stack
    -- Executing [s@macro-auto-blkvm:4] Set("PJSIP/004059-00000064", "MASTER_CHANNEL(CFIGNORE)=") in new stack
    -- Executing [s@macro-auto-blkvm:5] Set("PJSIP/004059-00000064", "FORWARD_CONTEXT=from-internal") in new stack
    -- Executing [s@macro-auto-blkvm:6] Set("PJSIP/004059-00000064", "MASTER_CHANNEL(FORWARD_CONTEXT)=from-internal") in new stack
    -- Executing [s@macro-auto-blkvm:7] Macro("PJSIP/004059-00000064", "blkvm-clr,") in new stack
    -- Executing [s@macro-blkvm-clr:1] Set("PJSIP/004059-00000064", "SHARED(BLKVM,PJSIP/004058-00000063)=") in new stack
    -- Executing [s@macro-blkvm-clr:2] Set("PJSIP/004059-00000064", "GOSUB_RETVAL=") in new stack
    -- Executing [s@macro-blkvm-clr:3] MacroExit("PJSIP/004059-00000064", "") in new stack
    -- Executing [s@macro-auto-blkvm:8] ExecIf("PJSIP/004059-00000064", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=004059/sip:004059@10.38.2.219:65275;ob)") in new stack
    -- Executing [s@macro-auto-blkvm:9] ExecIf("PJSIP/004059-00000064", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=)") in new stack
    -- Channel PJSIP/004059-00000064 joined 'simple_bridge' basic-bridge <e91b7cc2-a233-4bcf-b6b9-a59ea4bcac15>
    -- Channel PJSIP/004058-00000063 joined 'simple_bridge' basic-bridge <e91b7cc2-a233-4bcf-b6b9-a59ea4bcac15>
[2024-05-17 11:33:06] WARNING[26768]: res_pjsip_outbound_registration.c:1364 handle_registration_response: 403 Forbidden fatal response received from 'sip:185.101.180.252:5060' on registration attempt to 'sip:TRUNK_CATARINA_TEST_IN@185.101.180.252:5060', retrying in '30' seconds
[2024-05-17 11:33:06] ERROR[23176]: iostream.c:647 ast_iostream_start_tls: Problem setting up ssl connection: error:00000005:lib(0):func(0):DH lib, System call EOF
[2024-05-17 11:33:06] ERROR[23176]: tcptls.c:179 handle_tcptls_connection: Unable to set up ssl connection with peer '10.41.2.20:40372'
[2024-05-17 11:33:06] ERROR[23176]: iostream.c:552 ast_iostream_close: SSL_shutdown() failed: error:00000001:lib(0):func(0):reason(1), Internal SSL error
[2024-05-17 11:33:13] NOTICE[17722]: res_pjsip_sdp_rtp.c:146 rtp_check_timeout: Disconnecting channel 'PJSIP/004059-00000064' for lack of audio RTP activity in 30 seconds
    -- Channel PJSIP/004059-00000064 left 'simple_bridge' basic-bridge <e91b7cc2-a233-4bcf-b6b9-a59ea4bcac15>
    -- PJSIP/004059-00000064 Internal Gosub(app-missedcall-hangup,004059,1) start
    -- Executing [004059@app-missedcall-hangup:1] NoOp("PJSIP/004059-00000064", "Dialed: 004059") in new stack
    -- Executing [004059@app-missedcall-hangup:2] NoOp("PJSIP/004059-00000064", "Caller: 004058") in new stack
    -- Executing [004059@app-missedcall-hangup:3] GotoIf("PJSIP/004059-00000064", "0?exit") in new stack
    -- Executing [004059@app-missedcall-hangup:4] Set("PJSIP/004059-00000064", "EXTENNUM=004059") in new stack
    -- Executing [004059@app-missedcall-hangup:5] Set("PJSIP/004059-00000064", "FEXTENNUM=004059") in new stack
    -- Executing [004059@app-missedcall-hangup:6] GotoIf("PJSIP/004059-00000064", "0?exit") in new stack
    -- Executing [004059@app-missedcall-hangup:7] AGI("PJSIP/004059-00000064", "agi://127.0.0.1/missedcallnotify.php,004059,,004059,0,,PJSIP/004059-00000064,,,,TRUE") in new stack
    -- Channel PJSIP/004058-00000063 left 'simple_bridge' basic-bridge <e91b7cc2-a233-4bcf-b6b9-a59ea4bcac15>
  == Spawn extension (macro-dial, s, 30) exited non-zero on 'PJSIP/004058-00000063' in macro 'dial'
  == Spawn extension (followme-sub, 004059, 42) exited non-zero on 'PJSIP/004058-00000063'
    -- <PJSIP/004059-00000064>AGI Script agi://127.0.0.1/missedcallnotify.php completed, returning 0
    -- Executing [004059@app-missedcall-hangup:8] Return("PJSIP/004059-00000064", "") in new stack
  == Spawn extension (app-missedcall-hangup, 004059, 8) exited non-zero on 'PJSIP/004059-00000064'
    -- PJSIP/004059-00000064 Internal Gosub(app-missedcall-hangup,004059,1) complete GOSUB_RETVAL=
16

like this ??


  Current Mode: Online [any]                 Dialogs: 8
  Match Expression:                          BPF Filter:
  Display Filter:
      ^Idx Method     SIP From                  SIP To                    Msgs  Source                 Destination            Call State
  [ ] 1    INVITE     004058@pbx 004059@pbx 14    10.38.2.206:63748      10.38.43.75:5060       COMPLETED
  [ ] 2    INVITE     004058@10.38.43.75        004059@10.38.2.219        7     10.38.43.75:5060       10.38.2.219:65275      COMPLETED
  [ ] 3    OPTIONS    004059@10.38.43.75        004059@10.38.2.219        2     10.38.43.75:5060       10.38.2.219:65275
  [ ] 4    REGISTER   TRUNK_dialer_TEST_IN@18 TRUNK_dialer_TEST_IN@18 2     10.38.43.75:5060       185.101.180.252:5060
  [ ] 5    REGISTER   004058@pbx 004058@pbx 4     10.38.2.206:63748      10.38.43.75:5060
  [ ] 6    NOTIFY     004058@10.38.43.75        004058@10.38.2.206        2     10.38.43.75:5060       10.38.2.206:63748
  [ ] 7    PUBLISH    004058@pbx 004058@pbx 4     10.38.2.206:5060       10.38.43.75:5060
  [ ] 8    OPTIONS    TRUNK_dialer_TEST@10.38 TRUNK_dialer_TEST@sbc.m 2     10.38.43.75:5060       185.101.180.190:5060
17

The full contents of the packets relating to items 1 and 2, or, at least, all those that have a non-zero content length. I’d generally prefer them as logged by setting “pjsip set logger on”.

19 

<--- Received SIP request (1104 bytes) from UDP:10.38.2.206:63748 --->

INVITE sip:004059@pbx.com:5060 SIP/2.0

Via: SIP/2.0/UDP 10.38.2.206:63748;rport;branch=z9hG4bKPj206762ce3b9d45e799352c0459731178

Max-Forwards: 70

From: "004058" <sip:004058@pbx.com>;tag=2142053c76ed48acbab1bcb354c01f64

To: <sip:004059@pbx.com>

Contact: "004058" <sip:004058@10.38.2.206:63748;ob>

Call-ID: 13785677a1a24e2a86943843de941a0e

CSeq: 10236 INVITE

Route: <sip:pbx.com:5060;lr>

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: 363

v=0

o=- 3924944128 3924944128 IN IP4 10.38.2.206

s=pjmedia

b=AS:84

t=0 0

a=X-nat:0

m=audio 4004 RTP/AVP 8 0 18 101

c=IN IP4 10.38.2.206

b=TIAS:64000

a=rtcp:4005 IN IP4 10.38.2.206

a=sendrecv

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ssrc:1510811297 cname:74a804292fbd3aae

<--- Transmitting SIP response (594 bytes) to UDP:10.38.2.206:63748 --->

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 10.38.2.206:63748;rport=63748;received=10.38.2.206;branch=z9hG4bKPj206762ce3b9d45e799352c0459731178

Call-ID: 13785677a1a24e2a86943843de941a0e

From: "004058" <sip:004058@pbx.com>;tag=2142053c76ed48acbab1bcb354c01f64

To: <sip:004059@pbx.com>;tag=z9hG4bKPj206762ce3b9d45e799352c0459731178

CSeq: 10236 INVITE

WWW-Authenticate: Digest realm="asterisk",nonce="1715948128/6a70a57a2e38e3590f70f31d5725acef",opaque="17f24a707a8ba3f4",algorithm=MD5,qop="auth"

Server: FPBX-16.0.40.7(18.16.0)

Content-Length: 0

<--- Received SIP response (1184 bytes) from UDP:10.38.2.219:53690 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.38.43.75:5060;rport=5060;received=10.38.43.75;branch=z9hG4bKPje09af701-f0d8-4dae-868c-0d86fda21a2a

Call-ID: df72a9ac-7c3b-4e99-a3c1-7c1014f97ee2

From: "dialerTEST004058" <sip:004058@10.38.43.75>;tag=12cbf4af-b9ce-4f8b-bf14-96259e23a940

To: <sip:004059@10.38.2.219;ob>;tag=cbe24da6e9d64632bdf754ab9412d8dd

CSeq: 3169 INVITE

Contact: "004059" <sip:004059@10.38.2.219:53690;ob>;+sip.ice

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Supported: replaces, 100rel, timer, norefersub

Session-Expires: 1800;refresher=uac

Require: timer

Content-Type: application/sdp

Content-Length: 501

v=0

o=- 3924944128 3924944129 IN IP4 10.38.2.219

s=pjmedia

b=AS:84

t=0 0

a=X-nat:0

m=audio 4027 RTP/AVP 0 101

c=IN IP4 10.38.2.219

b=TIAS:64000

a=rtcp:4024 IN IP4 10.38.2.219

a=sendrecv

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ssrc:1881103619 cname:3e121a495f323bf6

a=ice-ufrag:139d7049

a=ice-pwd:692c4a80187e16c568993cd5

a=candidate:Ha2602db 1 UDP 2130706431 10.38.2.219 4027 typ host

a=candidate:Ha2602db 2 UDP 2130706430 10.38.2.219 4024 typ host

> 0x7f69c004db10 -- Strict RTP learning after remote address set to: 10.38.2.219:4027

<--- Transmitting SIP request (443 bytes) to UDP:10.38.2.219:53690 --->

ACK sip:004059@10.38.2.219:53690;ob SIP/2.0

Via: SIP/2.0/UDP 10.38.43.75:5060;rport;branch=z9hG4bKPjf7e63a64-bbc7-4849-ae7f-a8c9cb62e95b

From: "dialerTEST004058" <sip:004058@10.38.43.75>;tag=12cbf4af-b9ce-4f8b-bf14-96259e23a940

To: <sip:004059@10.38.2.219;ob>;tag=cbe24da6e9d64632bdf754ab9412d8dd

Call-ID: df72a9ac-7c3b-4e99-a3c1-7c1014f97ee2

CSeq: 3169 ACK

Max-Forwards: 70

User-Agent: FPBX-16.0.40.7(18.16.0)

Content-Length: 0

-- PJSIP/004059-0000006c answered PJSIP/004058-0000006b

-- Executing [s@macro-auto-blkvm:1] ExecIf("PJSIP/004059-0000006c", "0?Set(CDR(recordingfile)=.wav)") in new stack

-- Executing [s@macro-auto-blkvm:2] Set("PJSIP/004059-0000006c", "__MACRO_RESULT=") in new stack

-- Executing [s@macro-auto-blkvm:3] Set("PJSIP/004059-0000006c", "CFIGNORE=") in new stack

-- Executing [s@macro-auto-blkvm:4] Set("PJSIP/004059-0000006c", "MASTER_CHANNEL(CFIGNORE)=") in new stack

-- Executing [s@macro-auto-blkvm:5] Set("PJSIP/004059-0000006c", "FORWARD_CONTEXT=from-internal") in new stack

-- Executing [s@macro-auto-blkvm:6] Set("PJSIP/004059-0000006c", "MASTER_CHANNEL(FORWARD_CONTEXT)=from-internal") in new stack

-- Executing [s@macro-auto-blkvm:7] Macro("PJSIP/004059-0000006c", "blkvm-clr,") in new stack

-- Executing [s@macro-blkvm-clr:1] Set("PJSIP/004059-0000006c", "SHARED(BLKVM,PJSIP/004058-0000006b)=") in new stack

-- Executing [s@macro-blkvm-clr:2] Set("PJSIP/004059-0000006c", "GOSUB_RETVAL=") in new stack

-- Executing [s@macro-blkvm-clr:3] MacroExit("PJSIP/004059-0000006c", "") in new stack

-- Executing [s@macro-auto-blkvm:8] ExecIf("PJSIP/004059-0000006c", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=004059/sip:004059@10.38.2.219:53690;ob)") in new stack

-- Executing [s@macro-auto-blkvm:9] ExecIf("PJSIP/004059-0000006c", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=)") in new stack

Sent RTP packet to 10.38.2.206:4004 (type 00, seq 002599, ts 002400, len 000160)

<--- Transmitting SIP response (1114 bytes) to UDP:10.38.2.206:63748 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.38.2.206:63748;rport=63748;received=10.38.2.206;branch=z9hG4bKPj789d7f90d19b4b4a8dc71bc2f814a814

Call-ID: 13785677a1a24e2a86943843de941a0e

From: "004058" <sip:004058@pbx.com>;tag=2142053c76ed48acbab1bcb354c01f64

To: <sip:004059@pbx.com>;tag=9afccdff-431a-4954-960f-737b86bf3802

CSeq: 10237 INVITE

Server: FPBX-16.0.40.7(18.16.0)

Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER

Contact: <sip:10.38.43.75:5060>

Supported: 100rel, timer, replaces, norefersub

Session-Expires: 1800;refresher=uac

Require: timer

P-Asserted-Identity: "dialerTEST004059(Disponible)" <sip:004059@pbx.com>

Content-Type: application/sdp

Content-Length: 306

v=0

o=- 3924944128 3924944130 IN IP4 10.38.43.75

s=Asterisk

c=IN IP4 10.38.43.75

t=0 0

m=audio 10782 RTP/AVP 0 8 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=maxptime:150

a=sendrecv

-- Channel PJSIP/004059-0000006c joined 'simple_bridge' basic-bridge <2ee938d3-5d81-4a10-980f-352963c0029e>

-- Channel PJSIP/004058-0000006b joined 'simple_bridge' basic-bridge <2ee938d3-5d81-4a10-980f-352963c0029e>

<--- Received SIP request (406 bytes) from UDP:10.38.2.206:63748 --->

ACK sip:10.38.43.75:5060 SIP/2.0

Via: SIP/2.0/UDP 10.38.2.206:63748;rport;branch=z9hG4bKPj927c5a8234e840e5b2a7c23479345d6f

Max-Forwards: 70

From: "004058" <sip:004058@pbx.com>;tag=2142053c76ed48acbab1bcb354c01f64

To: <sip:004059@pbx.com>;tag=9afccdff-431a-4954-960f-737b86bf3802

Call-ID: 13785677a1a24e2a86943843de941a0e

CSeq: 10237 ACK

Content-Length: 0

<--- Received SIP request (909 bytes) from UDP:10.38.2.206:63748 --->

UPDATE sip:10.38.43.75:5060 SIP/2.0

Via: SIP/2.0/UDP 10.38.2.206:63748;rport;branch=z9hG4bKPjb697a075a4954a10b170041a24f0350f

Max-Forwards: 70

From: "004058" <sip:004058@pbx.com>;tag=2142053c76ed48acbab1bcb354c01f64

To: <sip:004059@pbx.com>;tag=9afccdff-431a-4954-960f-737b86bf3802

Contact: "004058" <sip:004058@10.38.2.206:63748;ob>

Call-ID: 13785677a1a24e2a86943843de941a0e

CSeq: 10240 UPDATE

Supported: replaces, 100rel, timer, norefersub

Session-Expires: 1800;refresher=uac

Min-SE: 90

Content-Type: application/sdp

Content-Length: 313

v=0

o=- 3924944128 3924944133 IN IP4 10.38.2.206

s=pjmedia

b=AS:84

t=0 0

a=X-nat:0

m=audio 4004 RTP/AVP 0 101

c=IN IP4 10.38.2.206

b=TIAS:64000

a=rtcp:4005 IN IP4 10.38.2.206

a=ssrc:1510811297 cname:74a804292fbd3aae

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

> 0x7f69c4015190 -- Strict RTP learning after remote address set to: 10.38.2.206:4004

<--- Transmitting SIP response (1043 bytes) to UDP:10.38.2.206:63748 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.38.2.206:63748;rport=63748;received=10.38.2.206;branch=z9hG4bKPjb697a075a4954a10b170041a24f0350f

Call-ID: 13785677a1a24e2a86943843de941a0e

From: "004058" <sip:004058@pbx.com>;tag=2142053c76ed48acbab1bcb354c01f64

To: <sip:004059@pbx.com>;tag=9afccdff-431a-4954-960f-737b86bf3802

CSeq: 10240 UPDATE

Session-Expires: 1800;refresher=uac

Require: timer

Contact: <sip:10.38.43.75:5060>

Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER

Supported: 100rel, timer, replaces, norefersub

P-Asserted-Identity: "dialerTEST004059(Disponible)" <sip:004059@pbx.com>

Server: FPBX-16.0.40.7(18.16.0)

Content-Type: application/sdp

Content-Length: 235

v=0

o=- 3924944128 3924944133 IN IP4 10.38.43.75

s=Asterisk

c=IN IP4 10.38.43.75

t=0 0

m=audio 10782 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 request (433 bytes) to UDP:10.38.2.206:63748 --->

OPTIONS sip:004058@10.38.2.206:63748;ob SIP/2.0

Via: SIP/2.0/UDP 10.38.43.75:5060;rport;branch=z9hG4bKPj7685b970-8479-4bea-b082-7bbc4026376d

From: <sip:004058@10.38.43.75>;tag=c2bf19dd-1f1d-4728-b052-9dbc7832d30e

To: <sip:004058@10.38.2.206;ob>

Contact: <sip:004058@10.38.43.75:5060>

Call-ID: c2a62701-cd88-4ec9-b7e9-c172885546db

CSeq: 9489 OPTIONS

Max-Forwards: 70

User-Agent: FPBX-16.0.40.7(18.16.0)

Content-Length: 0

<--- Received SIP response (796 bytes) from UDP:10.38.2.206:63748 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.38.43.75:5060;rport=5060;received=10.38.43.75;branch=z9hG4bKPj7685b970-8479-4bea-b082-7bbc4026376d

Call-ID: c2a62701-cd88-4ec9-b7e9-c172885546db

From: <sip:004058@10.38.43.75>;tag=c2bf19dd-1f1d-4728-b052-9dbc7832d30e

To: <sip:004058@10.38.2.206;ob>;tag=z9hG4bKPj7685b970-8479-4bea-b082-7bbc4026376d

CSeq: 9489 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

<--- Transmitting SIP request (488 bytes) to UDP:185.101.180.190:5060 --->

OPTIONS sip:TRUNK_dialer_TEST@sbc.maniterm.com:5060 SIP/2.0

Via: SIP/2.0/UDP 10.38.43.75:5060;rport;branch=z9hG4bKPj34389770-c520-4d4f-beb3-3c2e3287480f

From: <sip:TRUNK_dialer_TEST@10.38.43.75>;tag=2bb8c6ed-c32e-4f76-9234-9f84d53efdd0

To: <sip:TRUNK_dialer_TEST@sbc.maniterm.com>

Contact: <sip:TRUNK_dialer_TEST@10.38.43.75:5060>

Call-ID: 2d1d5307-e80e-4846-ba55-08ccfee169de

CSeq: 9780 OPTIONS

Max-Forwards: 70

User-Agent: FPBX-16.0.40.7(18.16.0)

Content-Length: 0

<--- Received SIP response (403 bytes) from UDP:185.101.180.190:5060 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.38.43.75:5060;received=41.86.249.180;rport=33988;branch=z9hG4bKPj34389770-c520-4d4f-beb3-3c2e3287480f

To: <sip:TRUNK_dialer_TEST@sbc.maniterm.com>;tag=23ca.eb8711feec03e332c591fdb7378f9c1a

From: <sip:TRUNK_dialer_TEST@10.38.43.75>;tag=2bb8c6ed-c32e-4f76-9234-9f84d53efdd0

Call-ID: 2d1d5307-e80e-4846-ba55-08ccfee169de

CSeq: 9780 OPTIONS

Content-Length: 0

> 0x7f69c004db10 -- Strict RTP learning after ICE completion

<--- Transmitting SIP request (600 bytes) to UDP:185.101.180.252:5060 --->

REGISTER sip:185.101.180.252:5060 SIP/2.0

Via: SIP/2.0/UDP 10.38.43.75:5060;rport;branch=z9hG4bKPje3578b7e-bce1-4872-a35c-956b17a99add

From: <sip:TRUNK_dialer_TEST_IN@185.101.180.252>;tag=e1a8e3c0-bc05-40d7-9d68-5224975a657c

To: <sip:TRUNK_dialer_TEST_IN@185.101.180.252>

Call-ID: 276acb85-5b70-4798-910d-497e1380892b

CSeq: 65055 REGISTER

Contact: <sip:s@10.38.43.75:5060;line=czpjdvu>

Expires: 3600

Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER

Max-Forwards: 70

User-Agent: FPBX-16.0.40.7(18.16.0)

Content-Length: 0

<--- Received SIP response (442 bytes) from UDP:185.101.180.252:5060 --->

SIP/2.0 403 Forbidden

Via: SIP/2.0/UDP 10.38.43.75:5060;received=41.86.249.180;rport=5060;branch=z9hG4bKPje3578b7e-bce1-4872-a35c-956b17a99add

To: <sip:TRUNK_dialer_TEST_IN@185.101.180.252>;tag=0232.a6741c44eab4d95bdce57f2bf0ed97a3

From: <sip:TRUNK_dialer_TEST_IN@185.101.180.252>;tag=e1a8e3c0-bc05-40d7-9d68-5224975a657c

Call-ID: 276acb85-5b70-4798-910d-497e1380892b

CSeq: 65055 REGISTER

Server: Manivox SBC

Content-Length: 0

[2024-05-17 12:15:46] WARNING[28726]: res_pjsip_outbound_registration.c:1364 handle_registration_response: 403 Forbidden fatal response received from 'sip:185.101.180.252:5060' on registration attempt to 'sip:TRUNK_dialer_TEST_IN@185.101.180.252:5060', retrying in '30' seconds

<--- Transmitting SIP request (499 bytes) to UDP:185.101.180.252:5060 --->

OPTIONS sip:TRUNK_dialer_TEST_IN@185.101.180.252:5060 SIP/2.0

Via: SIP/2.0/UDP 10.38.43.75:5060;rport;branch=z9hG4bKPj54332fd7-2b83-448e-9daf-50daaa98f36d

From: <sip:TRUNK_dialer_TEST_IN@10.38.43.75>;tag=a7f89708-6427-4b1e-9a0b-1d3922d81ee3

To: <sip:TRUNK_dialer_TEST_IN@185.101.180.252>

Contact: <sip:TRUNK_dialer_TEST_IN@10.38.43.75:5060>

Call-ID: 71553ca4-0b57-4999-a1a8-3a186698535f

CSeq: 54279 OPTIONS

Max-Forwards: 70

User-Agent: FPBX-16.0.40.7(18.16.0)

Content-Length: 0

<--- Received SIP response (430 bytes) from UDP:185.101.180.252:5060 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.38.43.75:5060;received=41.86.249.180;rport=5060;branch=z9hG4bKPj54332fd7-2b83-448e-9daf-50daaa98f36d

To: <sip:TRUNK_dialer_TEST_IN@185.101.180.252>;tag=0232.76b0333c30f4e783f8e9c88f52e626c8

From: <sip:TRUNK_dialer_TEST_IN@10.38.43.75>;tag=a7f89708-6427-4b1e-9a0b-1d3922d81ee3

Call-ID: 71553ca4-0b57-4999-a1a8-3a186698535f

CSeq: 54279 OPTIONS

Server: Manivox SBC

Content-Length: 0

[2024-05-17 12:15:58] NOTICE[17722]: res_pjsip_sdp_rtp.c:146 rtp_check_timeout: Disconnecting channel 'PJSIP/004059-0000006c' for lack of audio RTP activity in 30 seconds

-- Channel PJSIP/004059-0000006c left 'simple_bridge' basic-bridge <2ee938d3-5d81-4a10-980f-352963c0029e>

-- PJSIP/004059-0000006c Internal Gosub(app-missedcall-hangup,004059,1) start

-- Executing [004059@app-missedcall-hangup:1] NoOp("PJSIP/004059-0000006c", "Dialed: 004059") in new stack

-- Executing [004059@app-missedcall-hangup:2] NoOp("PJSIP/004059-0000006c", "Caller: 004058") in new stack

-- Executing [004059@app-missedcall-hangup:3] GotoIf("PJSIP/004059-0000006c", "0?exit") in new stack

-- Executing [004059@app-missedcall-hangup:4] Set("PJSIP/004059-0000006c", "EXTENNUM=004059") in new stack

-- Executing [004059@app-missedcall-hangup:5] Set("PJSIP/004059-0000006c", "FEXTENNUM=004059") in new stack

-- Executing [004059@app-missedcall-hangup:6] GotoIf("PJSIP/004059-0000006c", "0?exit") in new stack

-- Executing [004059@app-missedcall-hangup:7] AGI("PJSIP/004059-0000006c", "agi://127.0.0.1/missedcallnotify.php,004059,,004059,0,,PJSIP/004059-0000006c,,,,TRUE") in new stack

-- Channel PJSIP/004058-0000006b left 'simple_bridge' basic-bridge <2ee938d3-5d81-4a10-980f-352963c0029e>

== Spawn extension (macro-dial, s, 30) exited non-zero on 'PJSIP/004058-0000006b' in macro 'dial'

== Spawn extension (followme-sub, 004059, 42) exited non-zero on 'PJSIP/004058-0000006b'

<--- Transmitting SIP request (481 bytes) to UDP:10.38.2.206:63748 --->

BYE sip:004058@10.38.2.206:63748;ob SIP/2.0

Via: SIP/2.0/UDP 10.38.43.75:5060;rport;branch=z9hG4bKPj83413120-d28d-465d-84e8-0e9269c9488e

From: <sip:004059@pbx.com>;tag=9afccdff-431a-4954-960f-737b86bf3802

To: "004058" <sip:004058@pbx.com>;tag=2142053c76ed48acbab1bcb354c01f64

Call-ID: 13785677a1a24e2a86943843de941a0e

CSeq: 18503 BYE

Reason: Q.850;cause=44

Max-Forwards: 70

User-Agent: FPBX-16.0.40.7(18.16.0)

Content-Length: 0

@david551

20

You’ve sent media to the A side, presumably ring back, but it has sent nothing to you, and your media stops when the call connects. The B side has had no media in either direction.

Media and signalling addresses seem to match.

You need to trace the media through the network and find out where it is stopping.

1 Like
21

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