[Freeswitch-users] Audio cut off at the begin of the verto call to sip external voicemail

[Brian West]

Prefix them with silence_stream://2000 or 3000 and it should go away.

/b


On Thu, Jan 5, 2017 at 9:29 AM, Bipin Patel <bipin at xbipin.com> wrote:

> hi,
>
> i have the same issue, i think its related to slow audio setup during the
> call
>
>
> Regards,
> Bipin
>
>
> ------------------------------
> -------- Original Message --------
> Subject: [Freeswitch-users] Audio cut off at the begin of the verto call
> to sip external voicemail
> From: José Lopes <jose.lopes at itcenter.com.pt> <jose.lopes at itcenter.com.pt>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> <freeswitch-users at lists.freeswitch.org>
> Date: 1/5/2017, 6:35:45 PM
>
> Hello Guys,
>
> I have audio cut off at the begin of the verto call to  FreeSwitch that
> redirect to sip external voicemail (Access voicemail mailbox) .
>
> This happen when I use PCMU at verto codecs and sip codecs (if i use opus
> at verto codecs, there is no issue, but this causes audio transcoding) .
>
> At dialplan i used the example "Bridging from WebRTC (mod_verto) to
> PSTN/ITSPs" from https://freeswitch.org/confluence/display/FREESWITCH/
> mod_verto.
> I notice if i remove the playback action, there is no issue. But I need
> the playback action to send rtp packets to verto client.
>
> I simulate this using another FreeSwitch as external voicemail server and
> I only listen "id followed by pound" from the initial message of voicemail
> ("Please enter your id followed by pound").
> The log of this call is at https://pastebin.freeswitch.org/view/507fa115
>
> What I can do to use PCMU at verto codecs and sip codecs on type of call?
> Should i open a issue on FreeSwitch JIRA ?
>
>
> Best regards,
> Jose Lopes
>
>
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>
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-- 

*Brian West*
brian at freeswitch.org


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