[asterisk-bugs] [JIRA] (ASTERISK-26142) chan_mobile: Audio issues when connected to chan_sip or chan_pjsip

[Asterisk Team (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26142?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Asterisk Team updated ASTERISK-26142:
-------------------------------------

    Assignee: Asterisk Team  (was: Nilesh G)
      Status: Triage  (was: Waiting for Feedback)

> chan_mobile: Audio issues when connected to chan_sip or chan_pjsip
> ------------------------------------------------------------------
>
>                 Key: ASTERISK-26142
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26142
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Addons/chan_mobile
>    Affects Versions: 13.9.1
>         Environment: Ubuntu 16.04, Debian 8
>            Reporter: Nilesh G
>            Assignee: Asterisk Team
>            Severity: Minor
>
> I configured 2 asterisk systems - one on Debian 8 (with asterisk 11) and another on Ubuntu 16.04 (with asterisk 13) and configured chan_mobile, res_pjsip and chan_sip accordingly.
> The same usb bluetooth dongles with latest firmware were used.
> When a call is received and bridged to SIP phone in asterisk 11, sound is perfectly clear on both sides, but when the same test is done on asterisk 13, sound coming from the other end (ie. from person connected to the mobile phone) is not audible in SIP (either chan_sip or res_pjsip, doesn't matter). There's just some noise in the SIP receiver.
> Also, it's not about the difference between the Linux distributions. I have experienced the same when I was trying asterisk 13 on Debian 8 too.
> Devices used:
> 1. Nokia C1-01
> 2. Cisco 7940
> 3. Kinivo BTD-400 Bluetooth dongle with proper Broadcom firmware loaded in /lib/firmware.
> Steps to reproduce:
> 1. Configure chan_mobile using documentation provided at https://wiki.asterisk.org/wiki/display/AST/Mobile+Channel
> 2. Configure res_pjsip using documentation at https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Sections+and+Relationships & https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip
> 3. Make a call from SIP to some other number. Whatever is spoken in the mic of the SIP phone is audible at the other end, but whatever other end speaks comes out as noise from the SIP phone's speaker / handset.
> 4. Make a call from outside to the mobile phone connected to asterisk and after it is connected to the SIP phone by dialing extension, same behaviour as in 3 is observed.
> 5. Configure chan_sip using documentation, and disable res_pjsip and repeat the tests 3 and 4 - again same behaviour.
> This is happening on asterisk 13, but follow the same steps and documentation for asterisk 11 and it works flawlessly!



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