So that concludes that media is already established on the webrtc end and there is no problem with that. The box you are calling on SIP is also FS, you may want to add a sleep 2000 in that dialplan before the voicemail. Also since webrtc has no ringing indication you may want to set the variable ringback to get some audible feedback when making calls. On Fri, Jan 6, 2017 at 5:08 AM, José Lopes <jose.lopes@itcenter.com.pt> wrote:
Hello Anthony,
Thanks for your reply.
I tried to use an audio file (sounds/en/us/callie/ivr/8000/ivr-say_name.wav with ~2 seconds) instead of silence_stream. When i make the call from verto client, i ear the audio file, then no audio for ~2/3 seconds and then i ear "id followed by pound" (audio cut off from voicemail initial message "Please enter your id followed by pound").
I checked if i have the variable answer_delay and i don't have it.
The log of this call is at https://pastebin.freeswitch.org/view/e130e172 .
There is any thing more that i can do?
Best Regards, Jose Lopes
2017-01-05 18:14 GMT+00:00 Anthony Minessale <anthony.minessale@gmail.com> :
Also make sure you don't have answer_delay set in your vars.xml
On Thu, Jan 5, 2017 at 12:13 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
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
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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