Audio cut off at the begin of the verto call to sip external voicemail
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
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.orghttp://www.freeswitchsolutions.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon....
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest
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 <http://www.itcenter.com.pt/> <http://www.itcenter.com.pt/> <http://www.itcenter.com.pt/> www.itcenter.com.pt [image: ITCENTER Store] <http://store.itcenter.com.pt/> [image: ITCENTER Helpdesk] <http://support.itcenter.com.pt/otrs/customer.pl> [image: ITCENTER Facebook] <https://www.facebook.com/itcenterpt> [image: ITCENTER Linkedin] <https://www.linkedin.com/company/itcenter> [image: ITCENTER Twitter] <https://twitter.com/ITCenter_PT> 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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.orghttp://www.freeswitchsolutions.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon....
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com https://www.gofundme.com/freeswitch_ubuntu
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> | *M:*+1918424WEST (9378) *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
<http://www.itcenter.com.pt/> <http://www.itcenter.com.pt/> <http://www.itcenter.com.pt/> www.itcenter.com.pt [image: ITCENTER Store] <http://store.itcenter.com.pt/> [image: ITCENTER Helpdesk] <http://support.itcenter.com.pt/otrs/customer.pl> [image: ITCENTER Facebook] <https://www.facebook.com/itcenterpt> [image: ITCENTER Linkedin] <https://www.linkedin.com/company/itcenter> [image: ITCENTER Twitter] <https://twitter.com/ITCenter_PT>
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.orghttp://www.freeswitchsolutions.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon....
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com https://www.gofundme.com/freeswitch_ubuntu
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> | *M:*+1918424WEST (9378) *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
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
Os melhores cumprimentos / Best regards,
José Lopes Research and Development Phone: +351 256 370 980 Email: jose.lopes@itcenter.com.pt
<http://www.itcenter.com.pt/> <http://www.itcenter.com.pt/> <http://www.itcenter.com.pt/> www.itcenter.com.pt [image: ITCENTER Store] <http://store.itcenter.com.pt/> [image: ITCENTER Helpdesk] <http://support.itcenter.com.pt/otrs/customer.pl> [image: ITCENTER Facebook] <https://www.facebook.com/itcenterpt> [image: ITCENTER Linkedin] <https://www.linkedin.com/company/itcenter> [image: ITCENTER Twitter] <https://twitter.com/ITCenter_PT>
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.orghttp://www.freeswitchsolutions.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon....
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com https://www.gofundme.com/freeswitch_ubuntu
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> | *M:*+1918424WEST (9378) *Skype:*briankwest
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.orghttp://www.freeswitchsolutions.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon....
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com https://www.gofundme.com/freeswitch_ubuntu
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> | *M:*+1918424WEST (9378) *Skype:*briankwest
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.orghttp://www.freeswitchsolutions.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon....
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com https://www.gofundme.com/freeswitch_ubuntu
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> | *M:*+1918424WEST (9378) *Skype:*briankwest
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
Hello Anthony, At this replicated scenario, the box I am calling on SIP is FS. But on real scenario, the box I am calling on SIP is not Freeswitch, it is an external voicemail server and the initial message have audio cut off. Thanks for the information about variable ringback, I am already using on real scenario. One strange thing is if I use the codec OPUS at verto, this issue doesn't happen. But I need to use codec PCMU to avoid audio transcoding. Let me know if you need more information to debug this issue. Best Regards, Jose Lopes 2017-01-09 18:25 GMT+00:00 Anthony Minessale <anthony.minessale@gmail.com>:
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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting@freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.... > > FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees... > > > > ____________________________________________________________ > _____________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free > switch-users > http://www.freeswitch.org >
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com https://www.gofundme.com/freeswitch_ubuntu
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> | *M:*+1918424WEST (9378) *Skype:*briankwest
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
The minute you call it an Issue you should be filing it on JIRA. We get countless emails a day to the list so I don't always read them all so you are lucky I have managed to follow this thread. https://freeswitch.org/jira We have a small team and dealing with the mailing list is a volunteer effort. Here is also a tip. Just provide the info to questions asked. I asked you to do a diagnostic test for me by adding sleep to the other FS. Regardless if you can change the production or not, its still relevant to me what happens when you change it. On Tue, Jan 10, 2017 at 3:58 AM, José Lopes <jose.lopes@itcenter.com.pt> wrote:
Hello Anthony,
At this replicated scenario, the box I am calling on SIP is FS. But on real scenario, the box I am calling on SIP is not Freeswitch, it is an external voicemail server and the initial message have audio cut off.
Thanks for the information about variable ringback, I am already using on real scenario.
One strange thing is if I use the codec OPUS at verto, this issue doesn't happen. But I need to use codec PCMU to avoid audio transcoding.
Let me know if you need more information to debug this issue.
Best Regards, Jose Lopes
2017-01-09 18:25 GMT+00:00 Anthony Minessale <anthony.minessale@gmail.com> :
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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting@freeswitch.orghttp://www.freeswitchsolutions.com >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.... >> >> FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees... >> >> >> >> ____________________________________________________________ >> _____________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >> switch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian@freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: > /r/freeswitch <https://www.reddit.com/r/freeswitch> > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 > <(918)%20420-9002> | *M:*+1918424WEST (9378) > *Skype:*briankwest > > ____________________________________________________________ > _____________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free > switch-users > http://www.freeswitch.org >
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
Hello Anthony Thanks for your reply. I am sorry, I didn't notice that it was a diagnose test. I am available to make the tests that you need to analyse this situation and i put bellow information about the test you ask. If you see it is better to create a issue on JIRA, i will do it. Thanks for your and FreeSwitch Team effort. If I add a sleep of 2000 at voicemail server, after the answer there is no audio cut off (I put bellow the change on dialplan). But there is silence of 3/4 seconds between the ivr-say_name and the initial message from voicemail without audio cut off. Let me know if you need more tests or information. Dialplan on FreeSwitch Test: <include> <context name="default"> <extension name="itsp_send_call"> <condition field="destination_number" expression="^.*$"> <action application="answer"/> <action application="playback" data="/usr/share/freeswitch/sounds/en/us/callie/ivr/8000/ivr-say_name.wav"/> <action application="set" data="ringback=$${us-ring}"/> <action application="bridge" data="{absolute_codec_string='$${sip_codec_prefs}'}sofia/gateway/${caller_id_number}/${destination_number}"/> </condition> </extension> </context> </include> Extract of Dialplan on FreeSwitch External Voicemail Server <!-- voicemail main extension --> <extension name="vmain"> <condition field="destination_number" expression="^vmain$|^4000$|^\*98$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="sleep" data="2000"/> <!-- line added --> <action application="voicemail" data="check default ${domain_name}"/> </condition> </extension> Best Regards, Jose Lopes 2017-01-10 17:22 GMT+00:00 Anthony Minessale <anthony.minessale@gmail.com>:
The minute you call it an Issue you should be filing it on JIRA. We get countless emails a day to the list so I don't always read them all so you are lucky I have managed to follow this thread.
We have a small team and dealing with the mailing list is a volunteer effort.
Here is also a tip. Just provide the info to questions asked. I asked you to do a diagnostic test for me by adding sleep to the other FS. Regardless if you can change the production or not, its still relevant to me what happens when you change it.
On Tue, Jan 10, 2017 at 3:58 AM, José Lopes <jose.lopes@itcenter.com.pt> wrote:
Hello Anthony,
At this replicated scenario, the box I am calling on SIP is FS. But on real scenario, the box I am calling on SIP is not Freeswitch, it is an external voicemail server and the initial message have audio cut off.
Thanks for the information about variable ringback, I am already using on real scenario.
One strange thing is if I use the codec OPUS at verto, this issue doesn't happen. But I need to use codec PCMU to avoid audio transcoding.
Let me know if you need more information to debug this issue.
Best Regards, Jose Lopes
2017-01-09 18:25 GMT+00:00 Anthony Minessale <anthony.minessale@gmail.com
:
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 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: consulting@freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.... >>> >>> FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees... >>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>> switch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian@freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> https://www.gofundme.com/freeswitch_ubuntu >> >> Got Bugs? Report them here <https://freeswitch.org/jira>! | >> Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> >> >> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 >> <(918)%20420-9002> | *M:*+1918424WEST (9378) >> *Skype:*briankwest >> >> ____________________________________________________________ >> _____________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >> switch-users >> http://www.freeswitch.org >> > > > ____________________________________________________________ > _____________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free > switch-users > http://www.freeswitch.org >
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Yes, please file a JIRA because that was my main concern. Its very easy for me to forget about this thread or not notice it was updated. Once it is in JIRA it serves a place holder for all the info. On Tue, Jan 10, 2017 at 12:27 PM, José Lopes <jose.lopes@itcenter.com.pt> wrote:
Hello Anthony
Thanks for your reply. I am sorry, I didn't notice that it was a diagnose test. I am available to make the tests that you need to analyse this situation and i put bellow information about the test you ask. If you see it is better to create a issue on JIRA, i will do it. Thanks for your and FreeSwitch Team effort.
If I add a sleep of 2000 at voicemail server, after the answer there is no audio cut off (I put bellow the change on dialplan). But there is silence of 3/4 seconds between the ivr-say_name and the initial message from voicemail without audio cut off.
Let me know if you need more tests or information.
Dialplan on FreeSwitch Test: <include> <context name="default"> <extension name="itsp_send_call"> <condition field="destination_number" expression="^.*$"> <action application="answer"/> <action application="playback" data="/usr/share/freeswitch/ sounds/en/us/callie/ivr/8000/ivr-say_name.wav"/> <action application="set" data="ringback=$${us-ring}"/> <action application="bridge" data="{absolute_codec_string=' $${sip_codec_prefs}'}sofia/gateway/${caller_id_number}/${ destination_number}"/> </condition> </extension> </context> </include>
Extract of Dialplan on FreeSwitch External Voicemail Server
<!-- voicemail main extension --> <extension name="vmain"> <condition field="destination_number" expression="^vmain$|^4000$|^\* 98$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="sleep" data="2000"/> <!-- line added --> <action application="voicemail" data="check default ${domain_name}"/> </condition> </extension>
Best Regards, Jose Lopes
2017-01-10 17:22 GMT+00:00 Anthony Minessale <anthony.minessale@gmail.com> :
The minute you call it an Issue you should be filing it on JIRA. We get countless emails a day to the list so I don't always read them all so you are lucky I have managed to follow this thread.
We have a small team and dealing with the mailing list is a volunteer effort.
Here is also a tip. Just provide the info to questions asked. I asked you to do a diagnostic test for me by adding sleep to the other FS. Regardless if you can change the production or not, its still relevant to me what happens when you change it.
On Tue, Jan 10, 2017 at 3:58 AM, José Lopes <jose.lopes@itcenter.com.pt> wrote:
Hello Anthony,
At this replicated scenario, the box I am calling on SIP is FS. But on real scenario, the box I am calling on SIP is not Freeswitch, it is an external voicemail server and the initial message have audio cut off.
Thanks for the information about variable ringback, I am already using on real scenario.
One strange thing is if I use the codec OPUS at verto, this issue doesn't happen. But I need to use codec PCMU to avoid audio transcoding.
Let me know if you need more information to debug this issue.
Best Regards, Jose Lopes
2017-01-09 18:25 GMT+00:00 Anthony Minessale < anthony.minessale@gmail.com>:
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 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: consulting@freeswitch.orghttp://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.... >>>> >>>> FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees... >>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting@freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian@freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> https://www.gofundme.com/freeswitch_ubuntu >>> >>> Got Bugs? Report them here <https://freeswitch.org/jira>! | >>> Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> >>> >>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 >>> <(918)%20420-9002> | *M:*+1918424WEST (9378) >>> *Skype:*briankwest >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>> switch-users >>> http://www.freeswitch.org >>> >> >> >> ____________________________________________________________ >> _____________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >> switch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > <http://freeswitch.org/g+>* > > ClueCon Weekly Development Call > ☎ sip:888@conference.freeswitch.org ☎ +19193869900 > <(919)%20386-9900> > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA >
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
I created the issue at JIRA: https://freeswitch.org/jira/browse/FS-9939 . I tried to put all the information discussed here. Thanks again for your help. 2017-01-10 18:59 GMT+00:00 Anthony Minessale <anthony.minessale@gmail.com>:
Yes, please file a JIRA because that was my main concern. Its very easy for me to forget about this thread or not notice it was updated. Once it is in JIRA it serves a place holder for all the info.
On Tue, Jan 10, 2017 at 12:27 PM, José Lopes <jose.lopes@itcenter.com.pt> wrote:
Hello Anthony
Thanks for your reply. I am sorry, I didn't notice that it was a diagnose test. I am available to make the tests that you need to analyse this situation and i put bellow information about the test you ask. If you see it is better to create a issue on JIRA, i will do it. Thanks for your and FreeSwitch Team effort.
If I add a sleep of 2000 at voicemail server, after the answer there is no audio cut off (I put bellow the change on dialplan). But there is silence of 3/4 seconds between the ivr-say_name and the initial message from voicemail without audio cut off.
Let me know if you need more tests or information.
Dialplan on FreeSwitch Test: <include> <context name="default"> <extension name="itsp_send_call"> <condition field="destination_number" expression="^.*$"> <action application="answer"/> <action application="playback" data="/usr/share/freeswitch/so unds/en/us/callie/ivr/8000/ivr-say_name.wav"/> <action application="set" data="ringback=$${us-ring}"/> <action application="bridge" data="{absolute_codec_string=' $${sip_codec_prefs}'}sofia/gateway/${caller_id_number}/${des tination_number}"/> </condition> </extension> </context> </include>
Extract of Dialplan on FreeSwitch External Voicemail Server
<!-- voicemail main extension --> <extension name="vmain"> <condition field="destination_number" expression="^vmain$|^4000$|^\* 98$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="sleep" data="2000"/> <!-- line added --> <action application="voicemail" data="check default ${domain_name}"/> </condition> </extension>
Best Regards, Jose Lopes
2017-01-10 17:22 GMT+00:00 Anthony Minessale <anthony.minessale@gmail.com
:
The minute you call it an Issue you should be filing it on JIRA. We get countless emails a day to the list so I don't always read them all so you are lucky I have managed to follow this thread.
We have a small team and dealing with the mailing list is a volunteer effort.
Here is also a tip. Just provide the info to questions asked. I asked you to do a diagnostic test for me by adding sleep to the other FS. Regardless if you can change the production or not, its still relevant to me what happens when you change it.
On Tue, Jan 10, 2017 at 3:58 AM, José Lopes <jose.lopes@itcenter.com.pt> wrote:
Hello Anthony,
At this replicated scenario, the box I am calling on SIP is FS. But on real scenario, the box I am calling on SIP is not Freeswitch, it is an external voicemail server and the initial message have audio cut off.
Thanks for the information about variable ringback, I am already using on real scenario.
One strange thing is if I use the codec OPUS at verto, this issue doesn't happen. But I need to use codec PCMU to avoid audio transcoding.
Let me know if you need more information to debug this issue.
Best Regards, Jose Lopes
2017-01-09 18:25 GMT+00:00 Anthony Minessale < anthony.minessale@gmail.com>:
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.freesw >>>>> itch.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 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: consulting@freeswitch.orghttp://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.... >>>>> >>>>> FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees... >>>>> >>>>> >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting@freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users@lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian@freeswitch.org >>>> >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> https://www.gofundme.com/freeswitch_ubuntu >>>> >>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | >>>> Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> >>>> >>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 >>>> <(918)%20420-9002> | *M:*+1918424WEST (9378) >>>> *Skype:*briankwest >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting@freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>> switch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >> >> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >> http://twitter.com/FreeSWITCH >> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >> <http://freeswitch.org/g+>* >> >> ClueCon Weekly Development Call >> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >> <(919)%20386-9900> >> >> https://www.youtube.com/watch?v=9XXgW34t40s >> https://www.youtube.com/watch?v=NLaDpGQuZDA >> > > > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > <http://freeswitch.org/g+>* > > ClueCon Weekly Development Call > ☎ sip:888@conference.freeswitch.org ☎ +19193869900 > <(919)%20386-9900> > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > ____________________________________________________________ > _____________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free > switch-users > http://www.freeswitch.org >
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free switch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900>
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (4)
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Anthony Minessale
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Bipin Patel
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Brian West
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José Lopes