Try making the call with fsctl debug_level 10 and observe the logs, answer followed by silence_stream should send audio to the client. Also try playing an audio file instead of silence stream to see if you hear it. On Thu, Jan 5, 2017 at 11:58 AM, José Lopes <jose.lopes@itcenter.com.pt> wrote:
Hello Brian,
Thanks for your reply.
I tried the dialplan bellow with silence_stream://2000, and i have that issue. I tried with silence_stream://3000 and the audio cut off is greater. Without the playback, there is no audio cut off, but FreeSwitch doesn't send any rtp packets to verto client before the bridge.
There is any thing more that i can do?
<include> <context name="default"> <extension name="call_debug" continue="true"> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> </condition> </extension> <extension name="itsp_send_call"> <condition field="destination_number" expression="^.*$"> <action application="answer"/> <action application="playback" data="silence_stream://2000"/> <action application="bridge" data="{absolute_codec_string=' PCMU'}sofia/gateway/1002/${destination_number}"/> </condition> </extension> </context> </include>
Best Regards, Jose Lopes
Os melhores cumprimentos / Best regards,
José Lopes Research and Development Phone: +351 256 370 980 Email: jose.lopes@itcenter.com.pt
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2017-01-05 15:47 GMT+00:00 Brian West <brian@freeswitch.org>:
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/conflue nce/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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