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> <jose.lopes@itcenter.com.pt> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> <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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