Intermittent failure of set to stop ringing after group call is answered

Asterisk Asterisk Support
1

This configuration worked for years on 1.6
The upgrade to the latest version, including PJSIP, has introduced this new anomaly.

Here are the relevant bits of the dialplan:

[globals]
OPERATORS-DMA = PJSIP/7105&PJSIP/7106&PJSIP/7107&PJSIP/7108&PJSIP/7109

[dma]
exten => s,1,Answer()
exten => s,n(callops),Dial(${OPERATORS-DMA},9,ctT)
exten => s,n(auto),Background(dma-auto-attendant-greeting)
exten => s,n,WaitExten(3)
exten => s,n,Voicemail(199@dma)
exten => s,n,Hangup()

Here’s a bit of the logging showing a regular call:

[2025-01-13 17:38:59] VERBOSE[22527][C-000000a8] pbx_builtins.c: Goto (dma,s,1)
[2025-01-13 17:38:59] VERBOSE[22527][C-000000a8] pbx.c: Executing [s@dma:1] Answer("IAX2/voipms-1680", "") in new stack
[2025-01-13 17:39:00] VERBOSE[22527][C-000000a8] pbx.c: Executing [s@dma:2] Dial("IAX2/voipms-1680", "PJSIP/7105&PJSIP/7106&PJSIP/7107&PJSIP/7108&PJSIP/7109,9,ctT") in new stack
[2025-01-13 17:39:00] VERBOSE[22527][C-000000a8] app_dial.c: Called PJSIP/7105
[2025-01-13 17:39:00] VERBOSE[22527][C-000000a8] app_dial.c: Called PJSIP/7106
[2025-01-13 17:39:00] VERBOSE[22527][C-000000a8] app_dial.c: Called PJSIP/7107
[2025-01-13 17:39:00] VERBOSE[22527][C-000000a8] app_dial.c: Called PJSIP/7108
[2025-01-13 17:39:00] VERBOSE[22527][C-000000a8] app_dial.c: Called PJSIP/7109
[2025-01-13 17:39:00] VERBOSE[22527][C-000000a8] app_dial.c: PJSIP/7108-000000d2 connected line has changed. Saving it until answer for IAX2/voipms-1680
[2025-01-13 17:39:00] VERBOSE[22527][C-000000a8] app_dial.c: PJSIP/7107-000000d1 connected line has changed. Saving it until answer for IAX2/voipms-1680
[2025-01-13 17:39:00] VERBOSE[22527][C-000000a8] app_dial.c: PJSIP/7106-000000d0 connected line has changed. Saving it until answer for IAX2/voipms-1680
[2025-01-13 17:39:00] VERBOSE[22527][C-000000a8] app_dial.c: PJSIP/7105-000000cf connected line has changed. Saving it until answer for IAX2/voipms-1680
[2025-01-13 17:39:00] VERBOSE[22527][C-000000a8] app_dial.c: PJSIP/7109-000000d3 connected line has changed. Saving it until answer for IAX2/voipms-1680
[2025-01-13 17:39:00] VERBOSE[22527][C-000000a8] app_dial.c: PJSIP/7105-000000cf is ringing
[2025-01-13 17:39:00] VERBOSE[22527][C-000000a8] app_dial.c: PJSIP/7109-000000d3 is ringing
[2025-01-13 17:39:00] VERBOSE[22527][C-000000a8] app_dial.c: PJSIP/7108-000000d2 is ringing
[2025-01-13 17:39:00] VERBOSE[22527][C-000000a8] app_dial.c: PJSIP/7106-000000d0 is ringing
[2025-01-13 17:39:00] VERBOSE[22527][C-000000a8] app_dial.c: PJSIP/7107-000000d1 is ringing
[2025-01-13 17:39:03] VERBOSE[2391] res_rtp_asterisk.c: 0x7f176c06b750 -- Strict RTP learning after remote address set to: [192.1.1.109:2252](http://192.1.1.109:2252)
[2025-01-13 17:39:03] VERBOSE[22527][C-000000a8] app_dial.c: PJSIP/7109-000000d3 answered IAX2/voipms-1680

After the user at 7109 picked up the call, the other sets stopped ringing.

About one in ten calls goes like this:

[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] pbx_builtins.c: Goto (dma,s,1)
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] pbx.c: Executing [s@dma:1] Answer("IAX2/voipms-9139", "") in new stack
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] pbx.c: Executing [s@dma:2] Dial("IAX2/voipms-9139", "PJSIP/7105&PJSIP/7106&PJSIP/7107&PJSIP/7108&PJSIP/7109,9,ctT") in new stack
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: Called PJSIP/7105
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: Called PJSIP/7106
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: Called PJSIP/7107
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: Called PJSIP/7108
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: Called PJSIP/7109
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7109-00000090 connected line has changed. Saving it until answer for IAX2/voipms-9139
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7108-0000008f connected line has changed. Saving it until answer for IAX2/voipms-9139
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7107-0000008e connected line has changed. Saving it until answer for IAX2/voipms-9139
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7105-0000008c connected line has changed. Saving it until answer for IAX2/voipms-9139
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7109-00000090 is ringing
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7105-0000008c is ringing
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7108-0000008f is ringing
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7107-0000008e is ringing
[2025-01-21 12:58:25] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7109-00000090 answered IAX2/voipms-9139

When the user at 7109 picked up the call the set at 7106 continued to ring.

Its not the sets. We swapped out Linksys/Cisco 942s for Polycom VVX 250 sets and had the same behavior.

That last output was parsed out of the full console debug.

I was going to post the whole thing here but there’s a character limit.

Then I was going to upload it but FNGs cannot upload files.
(FNG in these forums – which is a testament to the reliability of the (verison 1.6) product that I have been using for 14 years and haven’t had to need to reach out until now)

Until I can share all the details, here’s the timestamped lines out of that detailed file; you can see something happens later on in the call with the anomalous set, but its hard to say what.

[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] pbx_builtins.c: Goto (dma,s,1)
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] pbx.c: Executing [s@dma:1] Answer("IAX2/voipms-9139", "") in new stack
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] pbx.c: Executing [s@dma:2] Dial("IAX2/voipms-9139", "PJSIP/7105&PJSIP/7106&PJSIP/7107&PJSIP/7108&PJSIP/7109,9,ctT") in new stack
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: Called PJSIP/7105
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: Called PJSIP/7106
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: Called PJSIP/7107
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: Called PJSIP/7108
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: Called PJSIP/7109
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7109-00000090 connected line has changed. Saving it until answer for IAX2/voipms-9139
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7108-0000008f connected line has changed. Saving it until answer for IAX2/voipms-9139
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7107-0000008e connected line has changed. Saving it until answer for IAX2/voipms-9139
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7105-0000008c connected line has changed. Saving it until answer for IAX2/voipms-9139
[2025-01-21 12:58:21] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (957 bytes) to UDP:192.1.1.105:5060 --->
[2025-01-21 12:58:21] VERBOSE[6862] res_pjsip_logger.c: <--- Transmitting SIP request (958 bytes) to UDP:192.1.1.108:5060 --->
[2025-01-21 12:58:21] VERBOSE[28888] res_pjsip_logger.c: <--- Transmitting SIP request (958 bytes) to UDP:192.1.1.107:5060 --->
[2025-01-21 12:58:21] VERBOSE[6863] res_pjsip_logger.c: <--- Transmitting SIP request (954 bytes) to UDP:192.1.1.109:5060 --->
[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (460 bytes) from UDP:192.1.1.107:5060 --->
[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (457 bytes) from UDP:192.1.1.105:5060 --->
[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (459 bytes) from UDP:192.1.1.108:5060 --->
[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (460 bytes) from UDP:192.1.1.109:5060 --->
[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (526 bytes) from UDP:192.1.1.109:5060 --->
[2025-01-21 12:58:21] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (435 bytes) to UDP:192.1.1.109:5060 --->
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7109-00000090 is ringing
[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (523 bytes) from UDP:192.1.1.105:5060 --->
[2025-01-21 12:58:21] VERBOSE[28888] res_pjsip_logger.c: <--- Transmitting SIP request (432 bytes) to UDP:192.1.1.105:5060 --->
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7105-0000008c is ringing
[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (525 bytes) from UDP:192.1.1.108:5060 --->
[2025-01-21 12:58:21] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (434 bytes) to UDP:192.1.1.108:5060 --->
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7108-0000008f is ringing
[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (526 bytes) from UDP:192.1.1.107:5060 --->
[2025-01-21 12:58:21] VERBOSE[28888] res_pjsip_logger.c: <--- Transmitting SIP request (435 bytes) to UDP:192.1.1.107:5060 --->
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7107-0000008e is ringing
[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (455 bytes) from UDP:192.1.1.109:5060 --->
[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (452 bytes) from UDP:192.1.1.105:5060 --->
[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (454 bytes) from UDP:192.1.1.108:5060 --->
[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (455 bytes) from UDP:192.1.1.107:5060 --->
[2025-01-21 12:58:21] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (407 bytes) to UDP:192.1.1.109:5060 --->
[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (702 bytes) from UDP:192.1.1.109:5060 --->
[2025-01-21 12:58:25] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (849 bytes) from UDP:192.1.1.109:5060 --->
[2025-01-21 12:58:25] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (406 bytes) to UDP:192.1.1.109:5060 --->
[2025-01-21 12:58:25] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7109-00000090 answered IAX2/voipms-9139
[2025-01-21 12:58:25] VERBOSE[28888] res_pjsip_logger.c: <--- Transmitting SIP request (469 bytes) to UDP:192.1.1.107:5060 --->
[2025-01-21 12:58:25] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (469 bytes) to UDP:192.1.1.108:5060 --->
[2025-01-21 12:58:25] VERBOSE[6870][C-00000069] bridge_channel.c: Channel PJSIP/7109-00000090 joined 'simple_bridge' basic-bridge <c36fce7b-75cc-412b-9daf-490cdeffd9d2>
[2025-01-21 12:58:25] VERBOSE[6861][C-00000069] bridge_channel.c: Channel IAX2/voipms-9139 joined 'simple_bridge' basic-bridge <c36fce7b-75cc-412b-9daf-490cdeffd9d2>
[2025-01-21 12:58:25] VERBOSE[6862] res_pjsip_logger.c: <--- Transmitting SIP request (468 bytes) to UDP:192.1.1.105:5060 --->
[2025-01-21 12:58:25] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (434 bytes) from UDP:192.1.1.107:5060 --->
[2025-01-21 12:58:25] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (432 bytes) from UDP:192.1.1.105:5060 --->
[2025-01-21 12:58:25] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (434 bytes) from UDP:192.1.1.108:5060 --->
[2025-01-21 12:58:25] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (468 bytes) from UDP:192.1.1.105:5060 --->
[2025-01-21 12:58:25] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (404 bytes) to UDP:192.1.1.105:5060 --->
[2025-01-21 12:58:25] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (471 bytes) from UDP:192.1.1.107:5060 --->
[2025-01-21 12:58:25] VERBOSE[28888] res_pjsip_logger.c: <--- Transmitting SIP request (406 bytes) to UDP:192.1.1.107:5060 --->
[2025-01-21 12:58:25] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (470 bytes) from UDP:192.1.1.108:5060 --->
[2025-01-21 12:58:25] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (405 bytes) to UDP:192.1.1.108:5060 --->
[2025-01-21 12:58:26] VERBOSE[705] res_pjsip_logger.c: <--- Transmitting SIP request (955 bytes) to UDP:192.1.1.106:5060 --->
[2025-01-21 12:58:26] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (460 bytes) from UDP:192.1.1.106:5060 --->
[2025-01-21 12:58:26] VERBOSE[28888] res_pjsip_logger.c: <--- Transmitting SIP request (390 bytes) to UDP:192.1.1.106:5060 --->
[2025-01-21 12:58:26] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (434 bytes) from UDP:192.1.1.106:5060 --->
[2025-01-21 12:58:26] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (471 bytes) from UDP:192.1.1.106:5060 --->
[2025-01-21 12:58:26] VERBOSE[705] res_pjsip_logger.c: <--- Transmitting SIP request (406 bytes) to UDP:192.1.1.106:5060 --->
2

Debug output part 1

[2025-01-21 12:58:21] VERBOSE[27307] chan_iax2.c: Accepting AUTHENTICATED call from 208.100.60.53:4569:
       > requested auth methods = (MD5),
       > actual auth method = MD5,
       > encrypted = no,
       > requested format = ulaw,
       > requested prefs = (ulaw|g729|g722|gsm),
       > actual format = ulaw,
       > host prefs = (ulaw|gsm|g729),
       > priority = mine
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] pbx.c: Executing [4165551212@default:1] NoOp("IAX2/voipms-9139", "20250121-12:58:21") in new stack
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] pbx.c: Executing [4165551212@default:2] NoOp("IAX2/voipms-9139", "CallerID Name: 4168083200") in new stack
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] pbx.c: Executing [4165551212@default:3] NoOp("IAX2/voipms-9139", "CallerID Number: 4168083200") in new stack
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] pbx.c: Executing [4165551212@default:4] Goto("IAX2/voipms-9139", "dma,s,1") in new stack
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] pbx_builtins.c: Goto (dma,s,1)
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] pbx.c: Executing [s@dma:1] Answer("IAX2/voipms-9139", "") in new stack
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] pbx.c: Executing [s@dma:2] Dial("IAX2/voipms-9139", "PJSIP/7105&PJSIP/7106&PJSIP/7107&PJSIP/7108&PJSIP/7109,9,ctT") in new stack
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: Called PJSIP/7105
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: Called PJSIP/7106
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: Called PJSIP/7107
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: Called PJSIP/7108
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: Called PJSIP/7109
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7109-00000090 connected line has changed. Saving it until answer for IAX2/voipms-9139
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7108-0000008f connected line has changed. Saving it until answer for IAX2/voipms-9139
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7107-0000008e connected line has changed. Saving it until answer for IAX2/voipms-9139
[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7105-0000008c connected line has changed. Saving it until answer for IAX2/voipms-9139
[2025-01-21 12:58:21] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (957 bytes) to UDP:192.1.1.105:5060 --->
INVITE sip:7105@192.1.1.105 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj68e5d7b5-940f-4ece-a252-30d7c8a8b6ec
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=bc481dae-2437-40ac-9dcb-8dbd3a03c1ec
To: <sip:7105@192.1.1.105>
Contact: <sip:asterisk@192.1.1.253:5060>
Call-ID: fcdfac91-6a51-45ad-9481-0fcf31a8b9a1
CSeq: 2122 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Type: application/sdp
Content-Length:   282

v=0
o=- 1282926992 1282926992 IN IP4 192.1.1.253
s=Asterisk
c=IN IP4 192.1.1.253
t=0 0
m=audio 13966 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

[2025-01-21 12:58:21] VERBOSE[6862] res_pjsip_logger.c: <--- Transmitting SIP request (958 bytes) to UDP:192.1.1.108:5060 --->
INVITE sip:7108@192.1.1.108 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj3a861656-86ec-4316-b593-3d8e0acaead2
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=ea02f4cd-e075-4ab6-a064-103d66f9df6d
To: <sip:7108@192.1.1.108>
Contact: <sip:asterisk@192.1.1.253:5060>
Call-ID: 16bdab29-74cc-41fb-bda6-fa14594361a9
CSeq: 31944 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Type: application/sdp
Content-Length:   282

v=0
o=- 1604429432 1604429432 IN IP4 192.1.1.253
s=Asterisk
c=IN IP4 192.1.1.253
t=0 0
m=audio 26006 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

[2025-01-21 12:58:21] VERBOSE[28888] res_pjsip_logger.c: <--- Transmitting SIP request (958 bytes) to UDP:192.1.1.107:5060 --->
INVITE sip:7107@192.1.1.107 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj30927b05-da8a-4e91-8311-3aa1ee6dc0a8
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=eff0c57a-f25d-4e72-9096-d5d4d0d1311f
To: <sip:7107@192.1.1.107>
Contact: <sip:asterisk@192.1.1.253:5060>
Call-ID: 12e8051e-452e-4ae0-91c2-d9b8abe82e55
CSeq: 16290 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Type: application/sdp
Content-Length:   282

v=0
o=- 1686459799 1686459799 IN IP4 192.1.1.253
s=Asterisk
c=IN IP4 192.1.1.253
t=0 0
m=audio 25072 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

[2025-01-21 12:58:21] VERBOSE[6863] res_pjsip_logger.c: <--- Transmitting SIP request (954 bytes) to UDP:192.1.1.109:5060 --->
INVITE sip:7109@192.1.1.109 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj71e97358-b864-4687-977c-6a61d731c3a3
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=6de4296f-4eac-421b-a0f1-5f512a770710
To: <sip:7109@192.1.1.109>
Contact: <sip:asterisk@192.1.1.253:5060>
Call-ID: 7c285257-ed00-483d-a3dd-97145146aed9
CSeq: 13399 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Type: application/sdp
Content-Length:   278

v=0
o=- 12597244 12597244 IN IP4 192.1.1.253
s=Asterisk
c=IN IP4 192.1.1.253
t=0 0
m=audio 24768 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (460 bytes) from UDP:192.1.1.107:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj30927b05-da8a-4e91-8311-3aa1ee6dc0a8
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=eff0c57a-f25d-4e72-9096-d5d4d0d1311f
To: "7107 - Incoming" <sip:7107@192.1.1.107>;tag=7CABB6FC-CDF072FF
CSeq: 16290 INVITE
Call-ID: 12e8051e-452e-4ae0-91c2-d9b8abe82e55
Contact: <sip:7107@192.1.1.107>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Content-Length: 0


[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (457 bytes) from UDP:192.1.1.105:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj68e5d7b5-940f-4ece-a252-30d7c8a8b6ec
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=bc481dae-2437-40ac-9dcb-8dbd3a03c1ec
To: "7105- Incoming" <sip:7105@192.1.1.105>;tag=597FD601-250F173
CSeq: 2122 INVITE
Call-ID: fcdfac91-6a51-45ad-9481-0fcf31a8b9a1
Contact: <sip:7105@192.1.1.105>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Content-Length: 0


[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (459 bytes) from UDP:192.1.1.108:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj3a861656-86ec-4316-b593-3d8e0acaead2
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=ea02f4cd-e075-4ab6-a064-103d66f9df6d
To: "7108 - Incoming" <sip:7108@192.1.1.108>;tag=26DA56D-8F5BB885
CSeq: 31944 INVITE
Call-ID: 16bdab29-74cc-41fb-bda6-fa14594361a9
Contact: <sip:7108@192.1.1.108>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Content-Length: 0


[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (460 bytes) from UDP:192.1.1.109:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj71e97358-b864-4687-977c-6a61d731c3a3
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=6de4296f-4eac-421b-a0f1-5f512a770710
To: "7109 - Incoming" <sip:7109@192.1.1.109>;tag=FBC3A33B-4055FD68
CSeq: 13399 INVITE
Call-ID: 7c285257-ed00-483d-a3dd-97145146aed9
Contact: <sip:7109@192.1.1.109>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Content-Length: 0


[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (526 bytes) from UDP:192.1.1.109:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj71e97358-b864-4687-977c-6a61d731c3a3
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=6de4296f-4eac-421b-a0f1-5f512a770710
To: "7109 - Incoming" <sip:7109@192.1.1.109>;tag=FBC3A33B-4055FD68
CSeq: 13399 INVITE
Call-ID: 7c285257-ed00-483d-a3dd-97145146aed9
Contact: <sip:7109@192.1.1.109>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Allow-Events: conference,talk,hold
Accept-Language: en
Require: 100rel
RSeq: 8193
Content-Length: 0


[2025-01-21 12:58:21] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (435 bytes) to UDP:192.1.1.109:5060 --->
PRACK sip:7109@192.1.1.109 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj91e3cdfd-a3a5-4caf-9d02-5974cf5d9f7e
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=6de4296f-4eac-421b-a0f1-5f512a770710
To: <sip:7109@192.1.1.109>;tag=FBC3A33B-4055FD68
Call-ID: 7c285257-ed00-483d-a3dd-97145146aed9
CSeq: 13400 PRACK
RAck: 8193 13399 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7109-00000090 is ringing
[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (523 bytes) from UDP:192.1.1.105:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj68e5d7b5-940f-4ece-a252-30d7c8a8b6ec
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=bc481dae-2437-40ac-9dcb-8dbd3a03c1ec
To: "7105- Incoming" <sip:7105@192.1.1.105>;tag=597FD601-250F173
CSeq: 2122 INVITE
Call-ID: fcdfac91-6a51-45ad-9481-0fcf31a8b9a1
Contact: <sip:7105@192.1.1.105>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Allow-Events: conference,talk,hold
Accept-Language: en
Require: 100rel
RSeq: 8193
Content-Length: 0


[2025-01-21 12:58:21] VERBOSE[28888] res_pjsip_logger.c: <--- Transmitting SIP request (432 bytes) to UDP:192.1.1.105:5060 --->
PRACK sip:7105@192.1.1.105 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj792f6bbc-9fab-4a6d-93cc-2b467b66e525
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=bc481dae-2437-40ac-9dcb-8dbd3a03c1ec
To: <sip:7105@192.1.1.105>;tag=597FD601-250F173
Call-ID: fcdfac91-6a51-45ad-9481-0fcf31a8b9a1
CSeq: 2123 PRACK
RAck: 8193 2122 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7105-0000008c is ringing
[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (525 bytes) from UDP:192.1.1.108:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj3a861656-86ec-4316-b593-3d8e0acaead2
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=ea02f4cd-e075-4ab6-a064-103d66f9df6d
To: "7108 - Incoming" <sip:7108@192.1.1.108>;tag=26DA56D-8F5BB885
CSeq: 31944 INVITE
Call-ID: 16bdab29-74cc-41fb-bda6-fa14594361a9
Contact: <sip:7108@192.1.1.108>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Allow-Events: conference,talk,hold
Accept-Language: en
Require: 100rel
RSeq: 8193
Content-Length: 0


[2025-01-21 12:58:21] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (434 bytes) to UDP:192.1.1.108:5060 --->
PRACK sip:7108@192.1.1.108 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj3d0b76ba-0d8a-4fc7-a531-caadeb4740df
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=ea02f4cd-e075-4ab6-a064-103d66f9df6d
To: <sip:7108@192.1.1.108>;tag=26DA56D-8F5BB885
Call-ID: 16bdab29-74cc-41fb-bda6-fa14594361a9
CSeq: 31945 PRACK
RAck: 8193 31944 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7108-0000008f is ringing
[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (526 bytes) from UDP:192.1.1.107:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj30927b05-da8a-4e91-8311-3aa1ee6dc0a8
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=eff0c57a-f25d-4e72-9096-d5d4d0d1311f
To: "7107 - Incoming" <sip:7107@192.1.1.107>;tag=7CABB6FC-CDF072FF
CSeq: 16290 INVITE
Call-ID: 12e8051e-452e-4ae0-91c2-d9b8abe82e55
Contact: <sip:7107@192.1.1.107>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Allow-Events: conference,talk,hold
Accept-Language: en
Require: 100rel
RSeq: 8193
Content-Length: 0


[2025-01-21 12:58:21] VERBOSE[28888] res_pjsip_logger.c: <--- Transmitting SIP request (435 bytes) to UDP:192.1.1.107:5060 --->
PRACK sip:7107@192.1.1.107 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj4a64f462-27f6-4eaf-8606-5ec77f59c8b7
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=eff0c57a-f25d-4e72-9096-d5d4d0d1311f
To: <sip:7107@192.1.1.107>;tag=7CABB6FC-CDF072FF
Call-ID: 12e8051e-452e-4ae0-91c2-d9b8abe82e55
CSeq: 16291 PRACK
RAck: 8193 16290 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:58:21] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7107-0000008e is ringing
[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (455 bytes) from UDP:192.1.1.109:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj91e3cdfd-a3a5-4caf-9d02-5974cf5d9f7e
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=6de4296f-4eac-421b-a0f1-5f512a770710
To: "7109 - Incoming" <sip:7109@192.1.1.109>;tag=FBC3A33B-4055FD68
CSeq: 13400 PRACK
Call-ID: 7c285257-ed00-483d-a3dd-97145146aed9
Contact: <sip:7109@192.1.1.109>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Content-Length: 0


[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (452 bytes) from UDP:192.1.1.105:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj792f6bbc-9fab-4a6d-93cc-2b467b66e525
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=bc481dae-2437-40ac-9dcb-8dbd3a03c1ec
To: "7105- Incoming" <sip:7105@192.1.1.105>;tag=597FD601-250F173
CSeq: 2123 PRACK
Call-ID: fcdfac91-6a51-45ad-9481-0fcf31a8b9a1
Contact: <sip:7105@192.1.1.105>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Content-Length: 0


[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (454 bytes) from UDP:192.1.1.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj3d0b76ba-0d8a-4fc7-a531-caadeb4740df
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=ea02f4cd-e075-4ab6-a064-103d66f9df6d
To: "7108 - Incoming" <sip:7108@192.1.1.108>;tag=26DA56D-8F5BB885
CSeq: 31945 PRACK
Call-ID: 16bdab29-74cc-41fb-bda6-fa14594361a9
Contact: <sip:7108@192.1.1.108>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Content-Length: 0


[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (455 bytes) from UDP:192.1.1.107:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj4a64f462-27f6-4eaf-8606-5ec77f59c8b7
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=eff0c57a-f25d-4e72-9096-d5d4d0d1311f
To: "7107 - Incoming" <sip:7107@192.1.1.107>;tag=7CABB6FC-CDF072FF
CSeq: 16291 PRACK
Call-ID: 12e8051e-452e-4ae0-91c2-d9b8abe82e55
Contact: <sip:7107@192.1.1.107>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Content-Length: 0


[2025-01-21 12:58:21] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (407 bytes) to UDP:192.1.1.109:5060 --->
OPTIONS sip:109@192.1.1.109 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPjd851e1b7-4daf-428a-9410-59c02c8be3c6
From: <sip:109@192.1.1.253>;tag=d62a3100-8601-4041-be44-b8b56476c3c3
To: <sip:109@192.1.1.109>
Contact: <sip:109@192.1.1.253:5060>
Call-ID: 6327bc69-59f0-4cc0-b923-946368d8cb39
CSeq: 42967 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:58:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (702 bytes) from UDP:192.1.1.109:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPjd851e1b7-4daf-428a-9410-59c02c8be3c6
From: <sip:109@192.1.1.253>;tag=d62a3100-8601-4041-be44-b8b56476c3c3
To: "109 - User109" <sip:109@192.1.1.109>;tag=909E45BD-657C43AD
CSeq: 42967 OPTIONS
Call-ID: 6327bc69-59f0-4cc0-b923-946368d8cb39
Contact: <sip:109@192.1.1.109>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Supported: replaces,100rel
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Supported: 100rel,replaces,norefersub,sdp-anat
Content-Length: 0


[2025-01-21 12:58:25] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (849 bytes) from UDP:192.1.1.109:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj71e97358-b864-4687-977c-6a61d731c3a3
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=6de4296f-4eac-421b-a0f1-5f512a770710
To: "7109 - Incoming" <sip:7109@192.1.1.109>;tag=FBC3A33B-4055FD68
CSeq: 13399 INVITE
Call-ID: 7c285257-ed00-483d-a3dd-97145146aed9
Contact: <sip:7109@192.1.1.109>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Supported: replaces,100rel
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Allow-Events: conference,talk,hold
Accept-Language: en
Content-Type: application/sdp
Content-Length: 209

v=0
o=- 1737482309 1737482309 IN IP4 192.1.1.109
s=Polycom IP Phone
c=IN IP4 192.1.1.109
t=0 0
a=sendrecv
m=audio 2234 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

[2025-01-21 12:58:25] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (406 bytes) to UDP:192.1.1.109:5060 --->
ACK sip:7109@192.1.1.109 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj05d0dfeb-a0d5-4729-9e82-5fbb06cffc34
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=6de4296f-4eac-421b-a0f1-5f512a770710
To: <sip:7109@192.1.1.109>;tag=FBC3A33B-4055FD68
Call-ID: 7c285257-ed00-483d-a3dd-97145146aed9
CSeq: 13399 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0
3

Debug output part 2
(starts with the call being answered)

[2025-01-21 12:58:25] VERBOSE[6861][C-00000069] app_dial.c: PJSIP/7109-00000090 answered IAX2/voipms-9139
[2025-01-21 12:58:25] VERBOSE[28888] res_pjsip_logger.c: <--- Transmitting SIP request (469 bytes) to UDP:192.1.1.107:5060 --->
CANCEL sip:7107@192.1.1.107 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj30927b05-da8a-4e91-8311-3aa1ee6dc0a8
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=eff0c57a-f25d-4e72-9096-d5d4d0d1311f
To: <sip:7107@192.1.1.107>
Call-ID: 12e8051e-452e-4ae0-91c2-d9b8abe82e55
CSeq: 16290 CANCEL
Reason: SIP;cause=200;text="Call completed elsewhere"
Reason: Q.850;cause=26
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:58:25] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (469 bytes) to UDP:192.1.1.108:5060 --->
CANCEL sip:7108@192.1.1.108 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj3a861656-86ec-4316-b593-3d8e0acaead2
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=ea02f4cd-e075-4ab6-a064-103d66f9df6d
To: <sip:7108@192.1.1.108>
Call-ID: 16bdab29-74cc-41fb-bda6-fa14594361a9
CSeq: 31944 CANCEL
Reason: SIP;cause=200;text="Call completed elsewhere"
Reason: Q.850;cause=26
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:58:25] VERBOSE[6870][C-00000069] bridge_channel.c: Channel PJSIP/7109-00000090 joined 'simple_bridge' basic-bridge <c36fce7b-75cc-412b-9daf-490cdeffd9d2>
[2025-01-21 12:58:25] VERBOSE[6861][C-00000069] bridge_channel.c: Channel IAX2/voipms-9139 joined 'simple_bridge' basic-bridge <c36fce7b-75cc-412b-9daf-490cdeffd9d2>
[2025-01-21 12:58:25] VERBOSE[6862] res_pjsip_logger.c: <--- Transmitting SIP request (468 bytes) to UDP:192.1.1.105:5060 --->
CANCEL sip:7105@192.1.1.105 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj68e5d7b5-940f-4ece-a252-30d7c8a8b6ec
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=bc481dae-2437-40ac-9dcb-8dbd3a03c1ec
To: <sip:7105@192.1.1.105>
Call-ID: fcdfac91-6a51-45ad-9481-0fcf31a8b9a1
CSeq: 2122 CANCEL
Reason: SIP;cause=200;text="Call completed elsewhere"
Reason: Q.850;cause=26
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:58:25] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (434 bytes) from UDP:192.1.1.107:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj30927b05-da8a-4e91-8311-3aa1ee6dc0a8
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=eff0c57a-f25d-4e72-9096-d5d4d0d1311f
To: "7107 - Incoming" <sip:7107@192.1.1.107>
CSeq: 16290 CANCEL
Call-ID: 12e8051e-452e-4ae0-91c2-d9b8abe82e55
Contact: <sip:7107@192.1.1.107>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Content-Length: 0


[2025-01-21 12:58:25] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (432 bytes) from UDP:192.1.1.105:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj68e5d7b5-940f-4ece-a252-30d7c8a8b6ec
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=bc481dae-2437-40ac-9dcb-8dbd3a03c1ec
To: "7105- Incoming" <sip:7105@192.1.1.105>
CSeq: 2122 CANCEL
Call-ID: fcdfac91-6a51-45ad-9481-0fcf31a8b9a1
Contact: <sip:7105@192.1.1.105>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Content-Length: 0


[2025-01-21 12:58:25] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (434 bytes) from UDP:192.1.1.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj3a861656-86ec-4316-b593-3d8e0acaead2
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=ea02f4cd-e075-4ab6-a064-103d66f9df6d
To: "7108 - Incoming" <sip:7108@192.1.1.108>
CSeq: 31944 CANCEL
Call-ID: 16bdab29-74cc-41fb-bda6-fa14594361a9
Contact: <sip:7108@192.1.1.108>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Content-Length: 0


[2025-01-21 12:58:25] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (468 bytes) from UDP:192.1.1.105:5060 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj68e5d7b5-940f-4ece-a252-30d7c8a8b6ec
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=bc481dae-2437-40ac-9dcb-8dbd3a03c1ec
To: "7105- Incoming" <sip:7105@192.1.1.105>;tag=597FD601-250F173
CSeq: 2122 INVITE
Call-ID: fcdfac91-6a51-45ad-9481-0fcf31a8b9a1
Contact: <sip:7105@192.1.1.105>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Content-Length: 0


[2025-01-21 12:58:25] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (404 bytes) to UDP:192.1.1.105:5060 --->
ACK sip:7105@192.1.1.105 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj68e5d7b5-940f-4ece-a252-30d7c8a8b6ec
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=bc481dae-2437-40ac-9dcb-8dbd3a03c1ec
To: <sip:7105@192.1.1.105>;tag=597FD601-250F173
Call-ID: fcdfac91-6a51-45ad-9481-0fcf31a8b9a1
CSeq: 2122 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:58:25] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (471 bytes) from UDP:192.1.1.107:5060 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj30927b05-da8a-4e91-8311-3aa1ee6dc0a8
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=eff0c57a-f25d-4e72-9096-d5d4d0d1311f
To: "7107 - Incoming" <sip:7107@192.1.1.107>;tag=7CABB6FC-CDF072FF
CSeq: 16290 INVITE
Call-ID: 12e8051e-452e-4ae0-91c2-d9b8abe82e55
Contact: <sip:7107@192.1.1.107>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Content-Length: 0


[2025-01-21 12:58:25] VERBOSE[28888] res_pjsip_logger.c: <--- Transmitting SIP request (406 bytes) to UDP:192.1.1.107:5060 --->
ACK sip:7107@192.1.1.107 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj30927b05-da8a-4e91-8311-3aa1ee6dc0a8
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=eff0c57a-f25d-4e72-9096-d5d4d0d1311f
To: <sip:7107@192.1.1.107>;tag=7CABB6FC-CDF072FF
Call-ID: 12e8051e-452e-4ae0-91c2-d9b8abe82e55
CSeq: 16290 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:58:25] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (470 bytes) from UDP:192.1.1.108:5060 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj3a861656-86ec-4316-b593-3d8e0acaead2
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=ea02f4cd-e075-4ab6-a064-103d66f9df6d
To: "7108 - Incoming" <sip:7108@192.1.1.108>;tag=26DA56D-8F5BB885
CSeq: 31944 INVITE
Call-ID: 16bdab29-74cc-41fb-bda6-fa14594361a9
Contact: <sip:7108@192.1.1.108>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Content-Length: 0


[2025-01-21 12:58:25] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (405 bytes) to UDP:192.1.1.108:5060 --->
ACK sip:7108@192.1.1.108 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj3a861656-86ec-4316-b593-3d8e0acaead2
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=ea02f4cd-e075-4ab6-a064-103d66f9df6d
To: <sip:7108@192.1.1.108>;tag=26DA56D-8F5BB885
Call-ID: 16bdab29-74cc-41fb-bda6-fa14594361a9
CSeq: 31944 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:58:26] VERBOSE[705] res_pjsip_logger.c: <--- Transmitting SIP request (955 bytes) to UDP:192.1.1.106:5060 --->
INVITE sip:7106@192.1.1.106 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj7b4be001-9033-4052-80f2-716609dcc0ee
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=51289dc3-359c-4798-9a0e-a4f31c530d94
To: <sip:7106@192.1.1.106>
Contact: <sip:asterisk@192.1.1.253:5060>
Call-ID: 1f8d6f6b-5389-4549-b8b5-938be63cf16f
CSeq: 18307 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Type: application/sdp
Content-Length:   279

v=0
o=- 633314109 633314109 IN IP4 192.1.1.253
s=Asterisk
c=IN IP4 192.1.1.253
t=0 0
m=audio 6822 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

[2025-01-21 12:58:26] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (460 bytes) from UDP:192.1.1.106:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj7b4be001-9033-4052-80f2-716609dcc0ee
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=51289dc3-359c-4798-9a0e-a4f31c530d94
To: "7106 - Incoming" <sip:7106@192.1.1.106>;tag=4B333E06-A6481068
CSeq: 18307 INVITE
Call-ID: 1f8d6f6b-5389-4549-b8b5-938be63cf16f
Contact: <sip:7106@192.1.1.106>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Content-Length: 0


[2025-01-21 12:58:26] VERBOSE[28888] res_pjsip_logger.c: <--- Transmitting SIP request (390 bytes) to UDP:192.1.1.106:5060 --->
CANCEL sip:7106@192.1.1.106 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj7b4be001-9033-4052-80f2-716609dcc0ee
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=51289dc3-359c-4798-9a0e-a4f31c530d94
To: <sip:7106@192.1.1.106>
Call-ID: 1f8d6f6b-5389-4549-b8b5-938be63cf16f
CSeq: 18307 CANCEL
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:58:26] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (434 bytes) from UDP:192.1.1.106:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj7b4be001-9033-4052-80f2-716609dcc0ee
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=51289dc3-359c-4798-9a0e-a4f31c530d94
To: "7106 - Incoming" <sip:7106@192.1.1.106>
CSeq: 18307 CANCEL
Call-ID: 1f8d6f6b-5389-4549-b8b5-938be63cf16f
Contact: <sip:7106@192.1.1.106>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Content-Length: 0


[2025-01-21 12:58:26] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (471 bytes) from UDP:192.1.1.106:5060 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj7b4be001-9033-4052-80f2-716609dcc0ee
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=51289dc3-359c-4798-9a0e-a4f31c530d94
To: "7106 - Incoming" <sip:7106@192.1.1.106>;tag=4B333E06-A6481068
CSeq: 18307 INVITE
Call-ID: 1f8d6f6b-5389-4549-b8b5-938be63cf16f
Contact: <sip:7106@192.1.1.106>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Content-Length: 0


[2025-01-21 12:58:26] VERBOSE[705] res_pjsip_logger.c: <--- Transmitting SIP request (406 bytes) to UDP:192.1.1.106:5060 --->
ACK sip:7106@192.1.1.106 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj7b4be001-9033-4052-80f2-716609dcc0ee
From: "4168083200" <sip:4168083200@192.1.1.253>;tag=51289dc3-359c-4798-9a0e-a4f31c530d94
To: <sip:7106@192.1.1.106>;tag=4B333E06-A6481068
Call-ID: 1f8d6f6b-5389-4549-b8b5-938be63cf16f
CSeq: 18307 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:58:36] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (407 bytes) to UDP:192.1.1.106:5060 --->
OPTIONS sip:106@192.1.1.106 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj47eb81a7-30f3-464f-82eb-55dbb484e3a7
From: <sip:106@192.1.1.253>;tag=838b9682-dd5e-4704-a393-630de3f50c81
To: <sip:106@192.1.1.106>
Contact: <sip:106@192.1.1.253:5060>
Call-ID: 73468184-6b98-427c-91f5-b74ee8f3460b
CSeq: 44091 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:58:36] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (700 bytes) from UDP:192.1.1.106:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj47eb81a7-30f3-464f-82eb-55dbb484e3a7
From: <sip:106@192.1.1.253>;tag=838b9682-dd5e-4704-a393-630de3f50c81
To: "106 - User106" <sip:106@192.1.1.106>;tag=F33A1CF1-467CBE85
CSeq: 44091 OPTIONS
Call-ID: 73468184-6b98-427c-91f5-b74ee8f3460b
Contact: <sip:106@192.1.1.106>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Supported: replaces,100rel
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Supported: 100rel,replaces,norefersub,sdp-anat
Content-Length: 0



[2025-01-21 12:58:53] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (407 bytes) to UDP:192.1.1.108:5060 --->
OPTIONS sip:108@192.1.1.108 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPjccaf7599-4757-4fcc-a978-d67dd47d54ee
From: <sip:108@192.1.1.253>;tag=eaa27ad3-750c-4ca3-b9d7-b058c6df7061
To: <sip:108@192.1.1.108>
Contact: <sip:108@192.1.1.253:5060>
Call-ID: d7e7a547-3e24-4fae-a1f3-af3270f57633
CSeq: 40434 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:58:53] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (702 bytes) from UDP:192.1.1.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPjccaf7599-4757-4fcc-a978-d67dd47d54ee
From: <sip:108@192.1.1.253>;tag=eaa27ad3-750c-4ca3-b9d7-b058c6df7061
To: "108 - User108" <sip:108@192.1.1.108>;tag=72DBC0EF-D17D6890
CSeq: 40434 OPTIONS
Call-ID: d7e7a547-3e24-4fae-a1f3-af3270f57633
Contact: <sip:108@192.1.1.108>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Supported: replaces,100rel
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Supported: 100rel,replaces,norefersub,sdp-anat
Content-Length: 0


[2025-01-21 12:59:01] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (411 bytes) to UDP:192.1.1.107:5060 --->
OPTIONS sip:7107@192.1.1.107 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj82bae012-0169-460c-b1e8-0b40dfe04df4
From: <sip:7107@192.1.1.253>;tag=70d177c6-287c-4dff-ab95-07b53a192d0f
To: <sip:7107@192.1.1.107>
Contact: <sip:7107@192.1.1.253:5060>
Call-ID: 56857537-d1fb-44f6-b0a7-e2430610e345
CSeq: 42020 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:59:01] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (708 bytes) from UDP:192.1.1.107:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj82bae012-0169-460c-b1e8-0b40dfe04df4
From: <sip:7107@192.1.1.253>;tag=70d177c6-287c-4dff-ab95-07b53a192d0f
To: "7107 - Incoming" <sip:7107@192.1.1.107>;tag=F4C9B015-FD0C4992
CSeq: 42020 OPTIONS
Call-ID: 56857537-d1fb-44f6-b0a7-e2430610e345
Contact: <sip:7107@192.1.1.107>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Supported: replaces,100rel
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Supported: 100rel,replaces,norefersub,sdp-anat
Content-Length: 0


[2025-01-21 12:59:03] VERBOSE[28888] res_pjsip_logger.c: <--- Transmitting SIP request (411 bytes) to UDP:192.1.1.105:5060 --->
OPTIONS sip:7105@192.1.1.105 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj974b6840-556b-4f7b-8d9b-7fc633242f14
From: <sip:7105@192.1.1.253>;tag=3da64a32-fd7e-4754-8061-4e0ffb84401a
To: <sip:7105@192.1.1.105>
Contact: <sip:7105@192.1.1.253:5060>
Call-ID: 04098381-5892-42b7-ae2e-66d13eccf090
CSeq: 57527 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:59:03] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (707 bytes) from UDP:192.1.1.105:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj974b6840-556b-4f7b-8d9b-7fc633242f14
From: <sip:7105@192.1.1.253>;tag=3da64a32-fd7e-4754-8061-4e0ffb84401a
To: "7105- Incoming" <sip:7105@192.1.1.105>;tag=253CD62D-E4299141
CSeq: 57527 OPTIONS
Call-ID: 04098381-5892-42b7-ae2e-66d13eccf090
Contact: <sip:7105@192.1.1.105>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Supported: replaces,100rel
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Supported: 100rel,replaces,norefersub,sdp-anat
Content-Length: 0


[2025-01-21 12:59:03] VERBOSE[705] res_pjsip_logger.c: <--- Transmitting SIP request (410 bytes) to UDP:192.1.1.106:5060 --->
OPTIONS sip:7106@192.1.1.106 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj5084014c-f0d0-4c97-ab3c-8f3df9cdd82a
From: <sip:7106@192.1.1.253>;tag=853eb089-ee50-4da8-8dc1-1b52e1965680
To: <sip:7106@192.1.1.106>
Contact: <sip:7106@192.1.1.253:5060>
Call-ID: d8109693-6ef1-4729-bc6d-82622f006673
CSeq: 3685 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:59:03] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (707 bytes) from UDP:192.1.1.106:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj5084014c-f0d0-4c97-ab3c-8f3df9cdd82a
From: <sip:7106@192.1.1.253>;tag=853eb089-ee50-4da8-8dc1-1b52e1965680
To: "7106 - Incoming" <sip:7106@192.1.1.106>;tag=2C888433-6CEA32A1
CSeq: 3685 OPTIONS
Call-ID: d8109693-6ef1-4729-bc6d-82622f006673
Contact: <sip:7106@192.1.1.106>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Supported: replaces,100rel
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Supported: 100rel,replaces,norefersub,sdp-anat
Content-Length: 0


[2025-01-21 12:59:06] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP request (410 bytes) to UDP:192.1.1.109:5060 --->
OPTIONS sip:7109@192.1.1.109 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPja9491b25-a44f-4a01-82db-62e71732cf9d
From: <sip:7109@192.1.1.253>;tag=b401567d-9962-4790-80cf-95a1ba27595c
To: <sip:7109@192.1.1.109>
Contact: <sip:7109@192.1.1.253:5060>
Call-ID: ba0a2228-b513-4807-9364-ad3fc958315d
CSeq: 1607 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:59:06] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (707 bytes) from UDP:192.1.1.109:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPja9491b25-a44f-4a01-82db-62e71732cf9d
From: <sip:7109@192.1.1.253>;tag=b401567d-9962-4790-80cf-95a1ba27595c
To: "7109 - Incoming" <sip:7109@192.1.1.109>;tag=B0364371-345E513F
CSeq: 1607 OPTIONS
Call-ID: ba0a2228-b513-4807-9364-ad3fc958315d
Contact: <sip:7109@192.1.1.109>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Supported: replaces,100rel
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Supported: 100rel,replaces,norefersub,sdp-anat
Content-Length: 0


[2025-01-21 12:59:09] VERBOSE[28888] res_pjsip_logger.c: <--- Transmitting SIP request (407 bytes) to UDP:192.1.1.107:5060 --->
OPTIONS sip:107@192.1.1.107 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj66d39198-6d14-4336-baf1-c07f9c8a85d5
From: <sip:107@192.1.1.253>;tag=43e5461b-fead-4da4-99db-3829bc00202d
To: <sip:107@192.1.1.107>
Contact: <sip:107@192.1.1.253:5060>
Call-ID: 0d980643-ab8c-402c-9ba1-b9ecb400f325
CSeq: 34459 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:59:09] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (698 bytes) from UDP:192.1.1.107:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj66d39198-6d14-4336-baf1-c07f9c8a85d5
From: <sip:107@192.1.1.253>;tag=43e5461b-fead-4da4-99db-3829bc00202d
To: "107 - User107" <sip:107@192.1.1.107>;tag=A83C7ECF-75CD37E1
CSeq: 34459 OPTIONS
Call-ID: 0d980643-ab8c-402c-9ba1-b9ecb400f325
Contact: <sip:107@192.1.1.107>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Supported: replaces,100rel
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Supported: 100rel,replaces,norefersub,sdp-anat
Content-Length: 0


[2025-01-21 12:59:18] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (707 bytes) from UDP:192.1.1.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj2cf54650-ad21-4585-a8ac-c318a67c3574
From: <sip:7108@192.1.1.253>;tag=8da10fba-74ec-484e-be7e-1502c9f78399
To: "7108 - Incoming" <sip:7108@192.1.1.108>;tag=BEDC7DDD-51335473
CSeq: 2706 OPTIONS
Call-ID: a4746e59-35ba-4a8d-a1ec-0402a080163b
Contact: <sip:7108@192.1.1.108>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Supported: replaces,100rel
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Supported: 100rel,replaces,norefersub,sdp-anat
Content-Length: 0


[2025-01-21 12:59:21] VERBOSE[705] res_pjsip_logger.c: <--- Transmitting SIP request (407 bytes) to UDP:192.1.1.109:5060 --->
OPTIONS sip:109@192.1.1.109 SIP/2.0
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj569ee9ab-3cc2-4979-b018-e2338c756302
From: <sip:109@192.1.1.253>;tag=7b2c349e-64d7-4748-b27b-fd9479c180d8
To: <sip:109@192.1.1.109>
Contact: <sip:109@192.1.1.253:5060>
Call-ID: 80f17238-4891-4119-a62b-bd5b99551fa5
CSeq: 59357 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:59:21] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP response (702 bytes) from UDP:192.1.1.109:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.253:5060;rport;branch=z9hG4bKPj569ee9ab-3cc2-4979-b018-e2338c756302
From: <sip:109@192.1.1.253>;tag=7b2c349e-64d7-4748-b27b-fd9479c180d8
To: "109 - User109" <sip:109@192.1.1.109>;tag=412EC3B7-BBDB7027
CSeq: 59357 OPTIONS
Call-ID: 80f17238-4891-4119-a62b-bd5b99551fa5
Contact: <sip:109@192.1.1.109>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Supported: replaces,100rel
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Supported: 100rel,replaces,norefersub,sdp-anat
Content-Length: 0


[2025-01-21 12:59:24] VERBOSE[27288] res_pjsip_logger.c: <--- Received SIP request (475 bytes) from UDP:192.1.1.109:5060 --->
BYE sip:asterisk@192.1.1.253:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.1.1.109;branch=z9hG4bK24a7a910AEBFF754
From: "7109 - Incoming" <sip:7109@192.1.1.109>;tag=FBC3A33B-4055FD68
To: "4168083200" <sip:4168083200@192.1.1.253>;tag=6de4296f-4eac-421b-a0f1-5f512a770710
CSeq: 1 BYE
Call-ID: 7c285257-ed00-483d-a3dd-97145146aed9
Contact: <sip:7109@192.1.1.109>
User-Agent: PolycomVVX-VVX_250-UA/6.4.7.4477
Accept-Language: en
Max-Forwards: 70
Content-Length: 0


[2025-01-21 12:59:24] VERBOSE[31278] res_pjsip_logger.c: <--- Transmitting SIP response (379 bytes) to UDP:192.1.1.109:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.1.109;rport=5060;received=192.1.1.109;branch=z9hG4bK24a7a910AEBFF754
Call-ID: 7c285257-ed00-483d-a3dd-97145146aed9
From: "7109 - Incoming" <sip:7109@192.1.1.109>;tag=FBC3A33B-4055FD68
To: "4168083200" <sip:4168083200@192.1.1.253>;tag=6de4296f-4eac-421b-a0f1-5f512a770710
CSeq: 1 BYE
Server: Asterisk PBX 18.23.1
Content-Length:  0


[2025-01-21 12:59:24] VERBOSE[6870][C-00000069] bridge_channel.c: Channel PJSIP/7109-00000090 left 'simple_bridge' basic-bridge <c36fce7b-75cc-412b-9daf-490cdeffd9d2>
[2025-01-21 12:59:24] VERBOSE[6861][C-00000069] bridge_channel.c: Channel IAX2/voipms-9139 left 'simple_bridge' basic-bridge <c36fce7b-75cc-412b-9daf-490cdeffd9d2>
[2025-01-21 12:59:24] VERBOSE[6861][C-00000069] pbx.c: Spawn extension (dma, s, 2) exited non-zero on 'IAX2/voipms-9139'
[2025-01-21 12:59:24] VERBOSE[6861][C-00000069] pbx.c: Executing [h@dma:1] NoOp("IAX2/voipms-9139", "") in new stack
[2025-01-21 12:59:24] VERBOSE[6861][C-00000069] pbx.c: Executing [h@dma:2] System("IAX2/voipms-9139", "sh /root/alert-frinklabs-sendlog.sh dmacall") in new stack
4

You would need a debug level log[1]. I will say that calling is an asynchronous action, so it is possible something inside the PJSIP stack or us was slightly delayed so it called for a second and then cancelled. Additionally Asterisk 18 is now security fix only, so if an issue is filed it would never be resolved in that version.

[1] Collecting Debug Information - Asterisk Documentation

5

Thanks for the quick response!

I have moved the Linksys sets to a testbed with a server running 22.1 so I will try to recreate the issue over there, with the appropriate debugging level set.

If that’s stable then I should probably upgrade the production server.

Tangentially, they both run the same version of PJSIP 2.14.1 so if it turns out to be an issue within that I could possibly keep the v18 server.

(It was a challenge to go from 1.6.10 to v18 so I am reticent to upgrade)

Meanwhile, in the time it takes for me to do all that, I hope that my account here will have been updated to allow file uploads so that I don’t have to post a thousand replies?

6

It’s set to that of a basic user which if I recall should allow it.

7

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