- FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP profile not loading
by Chris Young 01 Sep '15

01 Sep '15

Hi all, Recently, we've begun experiencing a strange problem whereby the first SIP profile to be loaded gets 'stuck' and never actually completes its initialisation. This always seems to affect the first profile only, so if I have profiles named (for example): dummy.xml external.xml internal.xml then 'dummy' would fail to load but 'external' and 'internal' would be fine. No error messages are output to the logs but preliminary investigation suggests that the profile thread is becoming blocked for some reason. The specified IP address is valid and available and there are no other processes using the requested port. FreeSWITCH comes up successfully but 'sofia status' shows only 'external' and 'internal'. At this point, I can use 'sofia profile dummy start' and the profile loads correctly so it appears to be valid. Has anybody else seen this kind of behaviour or know what could be causing it? Many thanks, Chris

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 111, Issue 3
by Gary Foreman 01 Sep '15

01 Sep '15

I'm getting the issue on a physical server with no load unfortunately. I'm trying to build the latest FS to test but its proving difficult on CentOS 6, looking like a Debian VM is the next step. On Tue, Sep 1, 2015 at 12:04 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Broken silence with webrtc (Stanislav Sinyagin) > > > ---------- Forwarded message ---------- > From: Stanislav Sinyagin <ssinyagin(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 1 Sep 2015 13:03:25 +0200 > Subject: Re: [Freeswitch-users] Broken silence with webrtc > is it running on a virtual machine? > > I found a strange effect that I could only reproduce in a VM, and never on > physical hardware: > https://freeswitch.org/jira/browse/FS-7805 > under certain load, an originate command triggers a continuous distortion > in another, running and unrelated, channel. > > It seems to be triggered by insufficient CPU resource at the moment of the > origination. > > > > > > > On Tue, Sep 1, 2015 at 12:49 PM, Gary Foreman <gaz.foreman(a)gmail.com> > wrote: > >> I've found that it occurs after any bridge, its not specific to the >> originate command. >> >> Would you require a wireshark trace or the output of the freeswitch >> console? >> >> The scenario below reproduces the issue ... >> >> Test extension >> >> <extension name="Inbound_Routing"> >> <condition field="destination_number" expression="^2003$"> >> <action application="export" data="dialed_extension=$1"/> >> <action application="export" data="transfer_ringback=$${uk-ring}"/> >> <action application="set" data="RECORD_TITLE=Title goes here"/> >> <action application="set" >> data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/> >> <action application="export" data="send_silence_when_idle=false"/> >> <action application="export" data="bridge_generate_comfort_noise=false"/> >> <action application="export" data="rtp_enable_vad_in=false"/> >> <action application="export" data="rtp_enable_vad_out=false"/> >> <action application="export" data="fire_talk_events=true"/> >> <action application="export" data="fire_not_talk_events=true"/> >> <action application="export" data="recording_id=${uuid}"/> >> <action application="set" data="media_bug_answer_req=true"/> >> <action application="answer" /> >> <action application="playback" >> data="tone_stream://L=100;%(400,200,400,450);%(400,2000,400,450)"/> >> </condition> >> </extension> >> >> I originate a call from a polycom handset using g722 to the extension >> above. >> I originate a call using the verto client to the extension above. >> >> I get the uuids of the channels using show channels and use uuid_bridge >> [uuid1] [uuid2] to merge the channels. >> >> On Tue, Sep 1, 2015 at 11:30 AM, < >> freeswitch-users-request(a)lists.freeswitch.org> wrote: >> >>> Send FreeSWITCH-users mailing list submissions to >>> freeswitch-users(a)lists.freeswitch.org >>> >>> To subscribe or unsubscribe via the World Wide Web, visit >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> or, via email, send a message with subject or body 'help' to >>> freeswitch-users-request(a)lists.freeswitch.org >>> >>> You can reach the person managing the list at >>> freeswitch-users-owner(a)lists.freeswitch.org >>> >>> When replying, please edit your Subject line so it is more specific >>> than "Re: Contents of FreeSWITCH-users digest..." >>> >>> Today's Topics: >>> >>> 1. How send in To header anonymous and in P-Asserted-Identity >>> caller-id-number (Alex Polischuk) >>> 2. Loop play while wait DTMF digit (Dmitry Mordovin) >>> 3. Broken silence with webrtc (Gary Foreman) >>> 4. Re: Broken silence with webrtc (Brian West) >>> 5. SIP profile not loading (Chris Young) >>> 6. Playing with conditions (Dmitry Mordovin) >>> 7. Re: Playing with conditions (Avi Marcus) >>> >>> >>> ---------- Forwarded message ---------- >>> From: Alex Polischuk <alxpol(a)gmail.com> >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Mon, 31 Aug 2015 17:02:25 +0300 >>> Subject: [Freeswitch-users] How send in To header anonymous and in >>> P-Asserted-Identity caller-id-number >>> Hi all, >>> >>> How I can define different users and domains in To and P-Asserted-Identity >>> headers? >>> >>> Thanks, >>> Alex >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Dmitry Mordovin <d.mordovin(a)dwide.com> >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Mon, 31 Aug 2015 18:24:08 +0400 >>> Subject: [Freeswitch-users] Loop play while wait DTMF digit >>> Hello >>> >>> This example play conf-pin.wav and wait DTMF. >>> >>> <extension name="play_and_get_digits example"> >>> <condition field="destination_number" expression="^(1888)$"> >>> <action application="play_and_get_digits" data="2 5 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/> >>> <action application="log" data="CRIT ${foobar}"/> >>> </condition> >>> </extension> >>> >>> >>> Is it possible to play WAV file in infinity loop and wait user DTMF? >>> >>> And how can I check DTMF input after user entered DTMF? >>> >>> In dialpeer, like. >>> If ${DTMF} = 1 then bridge to XXX >>> If ${DTMF} = 2 then play file and finish session >>> >>> >>> >>> Thank you. >>> Dmitry >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Gary Foreman <gaz.foreman(a)gmail.com> >>> To: freeswitch-users(a)lists.freeswitch.org >>> Cc: >>> Date: Mon, 31 Aug 2015 20:36:59 +0100 >>> Subject: [Freeswitch-users] Broken silence with webrtc >>> Hi all, >>> >>> Hoping someone can point me in the right direction because after several >>> hours I'm out of ideas. >>> >>> I'm having an issue where the 2nd leg of a call that is bridged to a >>> verto endpoint has broken silence. When there is no sound on the line the >>> audio goes completely silent (all background noise is dropped) but around >>> every third of a second it repeatedly cuts back in for a fraction, then >>> goes completely silent again. This only happens when the call is created >>> using the originate command. >>> >>> Can anyone give me an idea of where to look next? I have a wireshark >>> trace that during playback shows the audio cutting out periodically on the >>> 2nd leg during periods of silence. The 1st leg is using webrtc encryption >>> and I cant decode the stream. >>> >>> Thanks in advance! >>> >>> >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Brian West <brian(a)freeswitch.org> >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Mon, 31 Aug 2015 14:40:17 -0500 >>> Subject: Re: [Freeswitch-users] Broken silence with webrtc >>> Please provide logs and samples of how you originate this, sounds like >>> Voice Activity Detection possibly. >>> >>> On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman <gaz.foreman(a)gmail.com> >>> wrote: >>> >>>> Hi all, >>>> >>>> Hoping someone can point me in the right direction because after >>>> several hours I'm out of ideas. >>>> >>>> I'm having an issue where the 2nd leg of a call that is bridged to a >>>> verto endpoint has broken silence. When there is no sound on the line the >>>> audio goes completely silent (all background noise is dropped) but around >>>> every third of a second it repeatedly cuts back in for a fraction, then >>>> goes completely silent again. This only happens when the call is created >>>> using the originate command. >>>> >>>> Can anyone give me an idea of where to look next? I have a wireshark >>>> trace that during playback shows the audio cutting out periodically on the >>>> 2nd leg during periods of silence. The 1st leg is using webrtc encryption >>>> and I cant decode the stream. >>>> >>>> Thanks in advance! >>>> >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)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(a)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(a)freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> 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) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> >>> ---------- Forwarded message ---------- >>> From: Chris Young <Chris.Young(a)enghouse.com> >>> To: "freeswitch-users(a)lists.freeswitch.org" < >>> freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Tue, 1 Sep 2015 07:31:00 +0000 >>> Subject: [Freeswitch-users] SIP profile not loading >>> >>> Hi all, >>> >>> >>> >>> Recently, we've begun experiencing a strange problem whereby the first >>> SIP profile to be loaded gets 'stuck' and never actually completes its >>> initialisation. This always seems to affect the first profile only, so if I >>> have profiles named (for example): >>> >>> >>> >>> dummy.xml >>> >>> external.xml >>> >>> internal.xml >>> >>> >>> >>> then 'dummy' would fail to load but 'external' and 'internal' would be >>> fine. No error messages are output to the logs but preliminary >>> investigation suggests that the profile thread is becoming blocked for some >>> reason. The specified IP address is valid and available and there are no >>> other processes using the requested port. FreeSWITCH comes up successfully >>> but 'sofia status' shows only 'external' and 'internal'. At this point, I >>> can use 'sofia profile dummy start' and the profile loads correctly so it >>> appears to be valid. >>> >>> >>> >>> Has anybody else seen this kind of behaviour or know what could be >>> causing it? >>> >>> >>> >>> Many thanks, >>> >>> Chris >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Dmitry Mordovin <d.mordovin(a)dwide.com> >>> To: "freeswitch-users(a)lists.freeswitch.org" < >>> freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Tue, 01 Sep 2015 12:56:08 +0400 >>> Subject: [Freeswitch-users] Playing with conditions >>> Hello >>> >>> <section name="dialplan" description="Local call"> >>> <context name="public"> >>> <extension name="local_1"> >>> >>> <condition field="destination_number" expression="^(\d{4})$"> >>> <action application="answer"/> >>> >>> <action application="play_and_get_digits" data="1 1 3 7000 # >>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav >>> foobar \d+"/> >>> <action application="log" data="CRIT ${foobar}"/> >>> >>> <condition field="${foobar}" expression="1$"> >>> <action application="log" data="CRIT >>> --------------1------------------"/> >>> <anti-action application="log" data="CRIT --------------not >>> 1------------------"/> >>> </condition> >>> >>> <action application="log" data="CRIT ---- last ${foobar}"/> >>> </condition> >>> >>> </extension> >>> </context> >>> </section> >>> >>> Why inner condition not works? >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Avi Marcus <avi(a)avimarcus.net> >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Tue, 1 Sep 2015 10:30:00 +0000 >>> Subject: Re: [Freeswitch-users] Playing with conditions >>> Short answer: Each extension only has 1 set of conditions. >>> The condition evaluating foobar is run *before* it gets set. >>> After the play_and_get_digits you should transfer/execute_extension to >>> a new extension that will evaluate ${foobar} >>> >>> -Avi Marcus >>> BestFone >>> >>> On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin <d.mordovin(a)dwide.com> >>> wrote: >>> >>>> Hello >>>> >>>> <section name="dialplan" description="Local call"> >>>> <context name="public"> >>>> <extension name="local_1"> >>>> >>>> <condition field="destination_number" expression="^(\d{4})$"> >>>> <action application="answer"/> >>>> >>>> <action application="play_and_get_digits" data="1 1 3 7000 # >>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav >>>> /invalid.wav foobar \d+"/> >>>> <action application="log" data="CRIT ${foobar}"/> >>>> >>>> <condition field="${foobar}" expression="1$"> >>>> <action application="log" data="CRIT >>>> --------------1------------------"/> >>>> <anti-action application="log" data="CRIT --------------not >>>> 1------------------"/> >>>> </condition> >>>> >>>> <action application="log" data="CRIT ---- last ${foobar}"/> >>>> </condition> >>>> >>>> </extension> >>>> </context> >>>> </section> >>>> >>>> Why inner condition not works? >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)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(a)lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)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(a)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(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Loop play while wait DTMF digit
by Dmitry Mordovin 01 Sep '15

01 Sep '15

Hello This example play conf-pin.wav and wait DTMF. <extension name="play_and_get_digits example"> <condition field="destination_number" expression="^(1888)$"> <action application="play_and_get_digits" data="2 5 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/> <action application="log" data="CRIT ${foobar}"/> </condition> </extension> Is it possible to play WAV file in infinity loop and wait user DTMF? And how can I check DTMF input after user entered DTMF? In dialpeer, like. If ${DTMF} = 1 then bridge to XXX If ${DTMF} = 2 then play file and finish session Thank you. Dmitry

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Re: [Freeswitch-users] Broken silence with webrtc
by Gary Foreman 01 Sep '15

01 Sep '15

I've found that it occurs after any bridge, its not specific to the originate command. Would you require a wireshark trace or the output of the freeswitch console? The scenario below reproduces the issue ... Test extension <extension name="Inbound_Routing"> <condition field="destination_number" expression="^2003$"> <action application="export" data="dialed_extension=$1"/> <action application="export" data="transfer_ringback=$${uk-ring}"/> <action application="set" data="RECORD_TITLE=Title goes here"/> <action application="set" data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/> <action application="export" data="send_silence_when_idle=false"/> <action application="export" data="bridge_generate_comfort_noise=false"/> <action application="export" data="rtp_enable_vad_in=false"/> <action application="export" data="rtp_enable_vad_out=false"/> <action application="export" data="fire_talk_events=true"/> <action application="export" data="fire_not_talk_events=true"/> <action application="export" data="recording_id=${uuid}"/> <action application="set" data="media_bug_answer_req=true"/> <action application="answer" /> <action application="playback" data="tone_stream://L=100;%(400,200,400,450);%(400,2000,400,450)"/> </condition> </extension> I originate a call from a polycom handset using g722 to the extension above. I originate a call using the verto client to the extension above. I get the uuids of the channels using show channels and use uuid_bridge [uuid1] [uuid2] to merge the channels. On Tue, Sep 1, 2015 at 11:30 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. How send in To header anonymous and in P-Asserted-Identity > caller-id-number (Alex Polischuk) > 2. Loop play while wait DTMF digit (Dmitry Mordovin) > 3. Broken silence with webrtc (Gary Foreman) > 4. Re: Broken silence with webrtc (Brian West) > 5. SIP profile not loading (Chris Young) > 6. Playing with conditions (Dmitry Mordovin) > 7. Re: Playing with conditions (Avi Marcus) > > > ---------- Forwarded message ---------- > From: Alex Polischuk <alxpol(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 31 Aug 2015 17:02:25 +0300 > Subject: [Freeswitch-users] How send in To header anonymous and in > P-Asserted-Identity caller-id-number > Hi all, > > How I can define different users and domains in To and P-Asserted-Identity > headers? > > Thanks, > Alex > > > > ---------- Forwarded message ---------- > From: Dmitry Mordovin <d.mordovin(a)dwide.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 31 Aug 2015 18:24:08 +0400 > Subject: [Freeswitch-users] Loop play while wait DTMF digit > Hello > > This example play conf-pin.wav and wait DTMF. > > <extension name="play_and_get_digits example"> > <condition field="destination_number" expression="^(1888)$"> > <action application="play_and_get_digits" data="2 5 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/> > <action application="log" data="CRIT ${foobar}"/> > </condition> > </extension> > > > Is it possible to play WAV file in infinity loop and wait user DTMF? > > And how can I check DTMF input after user entered DTMF? > > In dialpeer, like. > If ${DTMF} = 1 then bridge to XXX > If ${DTMF} = 2 then play file and finish session > > > > Thank you. > Dmitry > > > > ---------- Forwarded message ---------- > From: Gary Foreman <gaz.foreman(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Mon, 31 Aug 2015 20:36:59 +0100 > Subject: [Freeswitch-users] Broken silence with webrtc > Hi all, > > Hoping someone can point me in the right direction because after several > hours I'm out of ideas. > > I'm having an issue where the 2nd leg of a call that is bridged to a verto > endpoint has broken silence. When there is no sound on the line the audio > goes completely silent (all background noise is dropped) but around every > third of a second it repeatedly cuts back in for a fraction, then goes > completely silent again. This only happens when the call is created using > the originate command. > > Can anyone give me an idea of where to look next? I have a wireshark trace > that during playback shows the audio cutting out periodically on the 2nd > leg during periods of silence. The 1st leg is using webrtc encryption and I > cant decode the stream. > > Thanks in advance! > > > > > > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 31 Aug 2015 14:40:17 -0500 > Subject: Re: [Freeswitch-users] Broken silence with webrtc > Please provide logs and samples of how you originate this, sounds like > Voice Activity Detection possibly. > > On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman <gaz.foreman(a)gmail.com> > wrote: > >> Hi all, >> >> Hoping someone can point me in the right direction because after several >> hours I'm out of ideas. >> >> I'm having an issue where the 2nd leg of a call that is bridged to a >> verto endpoint has broken silence. When there is no sound on the line the >> audio goes completely silent (all background noise is dropped) but around >> every third of a second it repeatedly cuts back in for a fraction, then >> goes completely silent again. This only happens when the call is created >> using the originate command. >> >> Can anyone give me an idea of where to look next? I have a wireshark >> trace that during playback shows the audio cutting out periodically on the >> 2nd leg during periods of silence. The 1st leg is using webrtc encryption >> and I cant decode the stream. >> >> Thanks in advance! >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)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(a)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(a)freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > 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) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > ---------- Forwarded message ---------- > From: Chris Young <Chris.Young(a)enghouse.com> > To: "freeswitch-users(a)lists.freeswitch.org" < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 1 Sep 2015 07:31:00 +0000 > Subject: [Freeswitch-users] SIP profile not loading > > Hi all, > > > > Recently, we've begun experiencing a strange problem whereby the first SIP > profile to be loaded gets 'stuck' and never actually completes its > initialisation. This always seems to affect the first profile only, so if I > have profiles named (for example): > > > > dummy.xml > > external.xml > > internal.xml > > > > then 'dummy' would fail to load but 'external' and 'internal' would be > fine. No error messages are output to the logs but preliminary > investigation suggests that the profile thread is becoming blocked for some > reason. The specified IP address is valid and available and there are no > other processes using the requested port. FreeSWITCH comes up successfully > but 'sofia status' shows only 'external' and 'internal'. At this point, I > can use 'sofia profile dummy start' and the profile loads correctly so it > appears to be valid. > > > > Has anybody else seen this kind of behaviour or know what could be causing > it? > > > > Many thanks, > > Chris > > > > > ---------- Forwarded message ---------- > From: Dmitry Mordovin <d.mordovin(a)dwide.com> > To: "freeswitch-users(a)lists.freeswitch.org" < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 01 Sep 2015 12:56:08 +0400 > Subject: [Freeswitch-users] Playing with conditions > Hello > > <section name="dialplan" description="Local call"> > <context name="public"> > <extension name="local_1"> > > <condition field="destination_number" expression="^(\d{4})$"> > <action application="answer"/> > > <action application="play_and_get_digits" data="1 1 3 7000 # > $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav > foobar \d+"/> > <action application="log" data="CRIT ${foobar}"/> > > <condition field="${foobar}" expression="1$"> > <action application="log" data="CRIT > --------------1------------------"/> > <anti-action application="log" data="CRIT --------------not > 1------------------"/> > </condition> > > <action application="log" data="CRIT ---- last ${foobar}"/> > </condition> > > </extension> > </context> > </section> > > Why inner condition not works? > > > > > ---------- Forwarded message ---------- > From: Avi Marcus <avi(a)avimarcus.net> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 1 Sep 2015 10:30:00 +0000 > Subject: Re: [Freeswitch-users] Playing with conditions > Short answer: Each extension only has 1 set of conditions. > The condition evaluating foobar is run *before* it gets set. > After the play_and_get_digits you should transfer/execute_extension to a > new extension that will evaluate ${foobar} > > -Avi Marcus > BestFone > > On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin <d.mordovin(a)dwide.com> > wrote: > >> Hello >> >> <section name="dialplan" description="Local call"> >> <context name="public"> >> <extension name="local_1"> >> >> <condition field="destination_number" expression="^(\d{4})$"> >> <action application="answer"/> >> >> <action application="play_and_get_digits" data="1 1 3 7000 # >> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav >> /invalid.wav foobar \d+"/> >> <action application="log" data="CRIT ${foobar}"/> >> >> <condition field="${foobar}" expression="1$"> >> <action application="log" data="CRIT >> --------------1------------------"/> >> <anti-action application="log" data="CRIT --------------not >> 1------------------"/> >> </condition> >> >> <action application="log" data="CRIT ---- last ${foobar}"/> >> </condition> >> >> </extension> >> </context> >> </section> >> >> Why inner condition not works? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)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(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Broken silence with webrtc
by Gary Foreman 31 Aug '15

31 Aug '15

Hi all, Hoping someone can point me in the right direction because after several hours I'm out of ideas. I'm having an issue where the 2nd leg of a call that is bridged to a verto endpoint has broken silence. When there is no sound on the line the audio goes completely silent (all background noise is dropped) but around every third of a second it repeatedly cuts back in for a fraction, then goes completely silent again. This only happens when the call is created using the originate command. Can anyone give me an idea of where to look next? I have a wireshark trace that during playback shows the audio cutting out periodically on the 2nd leg during periods of silence. The 1st leg is using webrtc encryption and I cant decode the stream. Thanks in advance!

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How send in To header anonymous and in P-Asserted-Identity caller-id-number
by Alex Polischuk 31 Aug '15

31 Aug '15

Hi all, How I can define different users and domains in To and P-Asserted-Identity headers? Thanks, Alex

1 0

problem with "play_and_get_digits" command (DTMF delay time)
by Manish Talwar 31 Aug '15

31 Aug '15

Hi, I have found one small problem with "play_and_get_digits" command. We have a IVR application and I am trying to get either "16 digit" card number or "0" for move to customer support from the user. Please find my dialplan as follows: 2015-08-25 18:05:37.535806 [CRIT] mod_httapi.c:1148 Debugging Return Data: <document type="xml/freeswitch-httapi"><params></params><work><execute applicati on="log" data="card number inputed: 2"/><execute application="flush_dtmf"/><exec ute application="sleep" data="2000"/><execute application="clear_digit_action" data="my_digit" /><execute application="bind_digit_action" data="my_digit,~\d{1 6}|^[0]{1},exec:transfer,dummy_transfer" /><execute action="http:/ /localhost:8080/ivr/fsentercardnumber/response/" application="play_and_get_digi ts" data="0 16 1 1000 # en-US/enter_card_number.mp3 '' Digits '' 1000 '' " /><ge tVariable name="Digits"/><execute action="http://localhost:8080/ivr/fsentercardnumber/response/" a pplication="play_and_get_digits" data="0 16 1 20000 # en-US/menu_speak_csr.mp3 ' ' Digits '' 20000 '' " /><getVariable name="Digits"/><execute application="clea r_digit_action" data="my_digit" /></work></document> It is working fine as expected if user input 16 digit number continuously without any delay. After getting full 16 digit number it will match the "bind_digit_action" and move to other dialplan. But my problem is if a user enter card number slowly then its trying to match already inputed DTMF values with matching binding and received next inputed value as different DTMF values. As a example, if I tried to input 55 and then wait for 1-2 seconds then Freeswitch tried to match 55 with binding. 2015-08-25 18:05:39.535920 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L1 6@8000hz 1 channels 20ms 2015-08-25 18:05:39.755933 [DEBUG] switch_rtp.c:5819 RTP RECV DTMF 5:2080 2015-08-25 18:05:40.015948 [DEBUG] switch_rtp.c:5819 RTP RECV DTMF 5:2080 2015-08-25 18:05:41.536035 [DEBUG] mod_dptools.c:132 sofia/internal/18188535351@ 192.168.1.112 Digit NOT match binding [55] 2015-08-25 18:05:41.536035 [DEBUG] switch_channel.c:486 RECV DTMF 5:2000 2015-08-25 18:05:41.536035 [DEBUG] switch_channel.c:582 sofia/internal/181885353 51(a)192.168.1.112 Queue dtmf digit=5 ms=250 samples=2000 2015-08-25 18:05:41.536035 [DEBUG] switch_channel.c:486 RECV DTMF 5:2000 2015-08-25 18:05:41.536035 [DEBUG] switch_channel.c:582 sofia/internal/181885353 51(a)192.168.1.112 Queue dtmf digit=5 ms=250 samples=2000 I don't want fresswtich tried to match the binding if there is any pause time while inputed the number as user can type slowly. I want to reset this binding delay time so that user can type slowly and we will get full 16 digits number for matching inputed value in certain time interval. Please suggest me how can I resolve this problem and increase the DTMF delay time. Thanks, Regards, Manish Talwar

3 5

XML RPC
by Abdul Hakeem 31 Aug '15

31 Aug '15

Hello Experts, Is there any possibility of using a different XML-RPC engine that does not involve HTTP ? The reason being, there are a few XML engines out there that avoids the HTTP overhead, TCP setup-teardown, & also uses very minimal resources Best regards, Abdul Hakeem

3 2

Python implementation of Event Socket Library doesn't work on Mac
by Carlos Fererro 30 Aug '15

30 Aug '15

All, to be more specific (if needed) - I am running on OS X 10.10.5 Yosemite. Not only it was a challenge to build the library on OS X (had to remove some flags some invalid flag(s) to avoid obscure "atom not found in symbolIndex ..." linker error) ... But!!! No client connection to the FreeSWITCH is getting established - even if client program and FreeSWITCH are running on the same machine and "host" parameter to the ESLconnection is "localhost" (which seems to be working fine on all flavors of Linux). Will be more than happy to provide any additional information is needed. Any help/advise is greatly appreciated. Thanks Carlos

1 0

Using hunt, original extension is busy while other one answered
by Yehavi Bourvine 30 Aug '15

30 Aug '15

Hi, We are using extension hunting (multiple destinations separated by | ) in order to allow the call to jump to the next extension if the first one is busy or does not answer. We are also using the limit() function in order to know whether an extension is in a call or not (people here do not like waiting calls). I also set limit_ignore_transfer=false. The problem we have is: If the destination is A|B, A does not answer and the call jumps to B and answered by him. At this state, limit shows that A has one call and B has 0 calls; That way, B cannot transfer the call back to A. Is this something wrong I am doing, an intended behavior, or a bug (for which I have to open a JIRA)? Thanks! __Yehavi:

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