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@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@itcenter.com.pt>
To: FreeSWITCH Users Help <freeswitch-users@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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