[bug]: MixMonitor doesn't record after Blind Transfer

[umerov1999]

Severity

Trivial

Versions

18.20.1

Components/Modules

MixMonitor

Operating Environment

Debian 12, Freepbx 16

Frequency of Occurrence

None

Issue Description

After a blind transfer on the call recording, I only hear the voice from the call initiator on PJSIP. There are no errors in the logs. Sometimes it happens that both voices are recorded. Codec alaw, recording format wav

Relevant log output

No response

Asterisk Issue Guidelines

• Yes, I have read the Asterisk Issue Guidelines

[jcolp]

We require additional debug to continue with triage of your issue. Please follow the instructions on the wiki [1] for how to collect debugging information from Asterisk. For expediency, where possible, attach the debug with a '.txt' file extension so that the debug will be usable for further analysis.

Thanks!

[1] https://docs.asterisk.org/Operation/Logging/Collecting-Debug-Information/?h=collecting+debug

[umerov1999]

We require additional debug to continue with triage of your issue. Please follow the instructions on the wiki [1] for how to collect debugging information from Asterisk. For expediency, where possible, attach the debug with a '.txt' file extension so that the debug will be usable for further analysis.

Thanks!

[1] https://docs.asterisk.org/Operation/Logging/Collecting-Debug-Information/?h=collecting+debug

log.txt
Call Recoed_1703189518.65_From_79859844541.zip

I am sending you the logs and recording. On the 2nd blind transfer, the voice of the extension disappears
External caller says "Odin" "Dva" "Tri" Chetyre". Internal caller - playing music And Transfering from 101 to 102, and 102 Transfer 101

[jcolp]

And audio flows as expected between them despite the recording? The debug makes it seem as though it is not getting audio from one side at the expected interval to satisfy recording.

[umerov1999]

And audio flows as expected between them despite the recording? The debug makes it seem as though it is not getting audio from one side at the expected interval to satisfy recording.

The sound on phones works fine, but disappears on recordings.

[jcolp]

What type of device or endpoint is on each side? Are they encrypted? Can you do a packet capture of both streams?

[umerov1999]

What type of device or endpoint is on each side? Are they encrypted? Can you do a packet capture of both streams?

Good evening. We used a mobile phone on one side and a Fanvil X303w on the other. At the time of taking the logs, I was using the MicroSip application instead of Fanvil. Traffic is not encrypted.

I can only record packet traces on weekends.

[jcolp]

Does it happen with only those involved, or with other types of endpoints?

[umerov1999]

Does it happen with only those involved, or with other types of endpoints?

The type of endpoints is not important. We tried different devices, and even tried an E1 DAHDI trunk with pjsip endpoints, also pure calls between endpoints - the result was the same

[umerov1999]

Does it happen with only those involved, or with other types of endpoints?

In general, the problem remains if g711(a-law, u-law) codecs are used, on others everything works. Audio record format - wav slin

[github-actions]

This issue is stale because it has been open 7 days with no activity. Remove stale label or comment or this will be closed in 14 days.

[github-actions]

This issue was closed because it has been stalled for 14 days with no activity.

[seanbright]

I might be experiencing this too. Can we re-open?

[jcolp]

Request completed successfully.

[umerov1999]

I dropped debian patch "2015_opus.patch and 2016_opus_plc.patch", and recording work fine with codec alaw/ulaw