[asterisk-bugs] [JIRA] (ASTERISK-29327) Issue related to Transfer of call to external number : Constant poor audio and frequent drop of call

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29327?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=254064#comment-254064 ] 

Asterisk Team commented on ASTERISK-29327:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.

Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/].

> Issue related to Transfer of call to external number  : Constant poor audio and frequent drop of call
> -----------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-29327
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29327
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip, Channels/chan_sip/Interoperability, Channels/chan_sip/Transfers, PBX/pbx_config
>    Affects Versions: 16.8.0
>            Reporter: Prune
>            Severity: Major
>
> Architecture:
> We are interconnected in SS7
> Our "class 5" ISwitch translated into SIP
> The traffic is sent to our Asterisk server.
> We use:
> - PJSIP channels
>  - 711u
> - The 711a and 711u codecs are used in the ISwitch.
> We have no problem with incoming and outgoing calls to an external number.
> Call forwarding to an external number works well;
> But, any TRANSFER to an external number creates an audio problem and sometimes an interruption of the call.
> 1 / Audio problem when transferring to an external number:
> - No sound at all after a few moments,
> then the conversation is audible again.
> - White noise / cracking
> We noticed an audio problem with the transfer to a landline;
> but, the transfer to the cell phone no problem noticed.
> We don't have a lot of traffic because we are still testing the call manager so we don't think it's coming from bandwidth.
> 2 / Call stopping after some time after a transfer:
> From time to time we have the loss of the transferred call and the callback of the transferred customer.
> Example: we transfer a call to an agent, the caller and the agent are both talking; then the call is terminated and returned to the call manager. The call manager should take the call back and transfer another time. The agent is not on the line after the call is disconnected.
> The customer did not decide to hang up and was interrupted in the middle of the conversation.
> Thanks for your precious help.



--
This message was sent by Atlassian JIRA
(v6.2#6252)