[asterisk-bugs] [JIRA] (ASTERISK-30428) asterisk continues to play music on hold after INVITE with replaces

David Middleton (JIRA) noreply at issues.asterisk.org
Tue Feb 21 10:19:03 CST 2023


    [ https://issues.asterisk.org/jira/browse/ASTERISK-30428?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=261405#comment-261405 ] 

David Middleton commented on ASTERISK-30428:
--------------------------------------------

Apologies. I hadn't set the log levels.

This is more helpful - 2023022100_debug_log_ASTERISK-30428.txt
Party A calling Party B
ingress: PJSIP/219-astpjsip-00000014
egress: PJSIP/proxy-00000015

Party A calling Party C
ingress: PJSIP/219-astpjsip-00000016
egress: PJSIP/proxy-00000017

> asterisk continues to play music on hold after INVITE with replaces
> -------------------------------------------------------------------
>
>                 Key: ASTERISK-30428
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-30428
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Bridges/bridge_simple, Bridges/bridge_softmix
>    Affects Versions: 18.14.0
>            Reporter: David Middleton
>            Assignee: Unassigned
>         Attachments: 20230130_teams-pstn_teams-pstn_merge_pjsip_nok_asterisk_anon.pcap, 20230214 pcap breakdown.txt, 2023022000_debug_log_ASTERISK-30428.txt, 2023022100_debug_log_ASTERISK-30428.txt
>
>
> Party A calls Party B.
> Party A puts Party B on hold and Asterisk plays moh to Party B.
> Party A then calls Party C.
> Party A then merges the two calls (to Party B and Party C) - a new INVITE is received by asterisk replacing the original held call to Party B.
> Asterisk continues to send music on hold to Party B, as well as the media from the new replacement call, all using the same SSRC and UDP ports, but with different timestamps and sequence numbers (causing Party B to hear broken moh and audio). Media from party B is ok, as well as media to/from Parties A and C.
> I've attached an anonymised pcap that shows all the SIP signalling, but for simplification, just the egress RTP (to/from Party B and C).
> I've also attached a pcap breakdown text file.
> I did try unloading 'bridge_simple' as suggested in ASTERISK-29273 but the issue remained.



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