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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 20 Mar '15

20 Mar '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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XML and HTTP engines or libraries
by Abdul Hakeem 20 Mar '15

20 Mar '15

Hello, Can someone point me to the XML/HTTP engines or libraries used in Freeswitch ? Thanks, Abdul Hakeem

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Multilanguages in dialplan and voicemail
by Victor Medina 20 Mar '15

20 Mar '15

Hi guys! Im having troubles trying to support multilanguages destinations in freeswitch dialplan and voicemail. I have modules correctly loaded in modules as: <load module="mod_say_en"/> <load module="mod_say_es"/> In vars I only have the following... <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/> And two internal destinations defined as... Spanish exts.... <extension name="Local_Extension1"> <condition field="destination_number" expression="^([0-9][0-9][0-9])$"> ....................... <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> English..... <extension name="Local_Extension2"> <condition field="destination_number" expression="^(1[0-9][0-9][0-9])$"> ................. <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> I need that Local_Extension1 have a spanish VoiceMail Voice and Local_Extension2 have a english VoiceMails. I have tried explicitly setting languages in each of the dialplan sections like.. <action application="set" data="sound_prefix=$${sounds_dir}/es/mx/maria"/> <action application="set" data="default_language=es"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> I'm even seeing correctly processed in console: Dialplan: sofia/internal/163(a)cibersys.com Action sleep(1000) Dialplan: sofia/internal/163(a)cibersys.com Action set(sound_prefix=/opt/CloudVoice-vPBX/fsw14/sounds/es/mx/maria) Dialplan: sofia/internal/163(a)cibersys.com Action set(default_language=es) Dialplan: sofia/internal/163(a)cibersys.com Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) Any ideas on what could I do? Are things setted in vars.xml taking precedence ? For example should I remove sound_prefix from vars? Also... any way to reload vars.xml? -- Víctor E. Medina M. Software [image: Zoiper Click2Dial]+58424 291 4561 <zoiper://+584242914561>[image: ve] BB #79A8AFA2 /@VMCibersys

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ClueCon is Coming. Start Getting Ready!
by Ken Rice 20 Mar '15

20 Mar '15

New Post on freeswitch.org from anthm check it out at http://ift.tt/1x6SG9o ClueCon is Coming. Start Getting Ready! ClueCon was a big hit last year but don’t take it from us. Here are some testimonials from 2014. Don’t forget the call for speakers is now live. Visit http://ift.tt/1FBM7gN and reserve a slot!

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start_dtmf when no RFC 2833
by Phillip Jones 19 Mar '15

19 Mar '15

Hi All, The following dial plan detects RFC 2833 and if it is not present, runs the "start_dtmf" application. <condition field="${switch_r_sdp}" expression="a=rtpmap:(\d+)\stelephone-event/8000" break="never"> <action application="set" data="rtp_payload_number=$1"/> <anti-action application="start_dtmf"/> </condition> I need to do this on the outbound leg - using Lua. Can anyone point me to some code example that would achieve this or get me started? Any help appreciated. Thanks Phil

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Problem with make current
by Guillermo Ruiz Camauer 19 Mar '15

19 Mar '15

I am having problems trying to update my Freeswitch instances. MAKE CURRENT ends with: make[1]: Entering directory `/usr/src/freeswitch' Pulling updates... error: Failed connect to freeswitch.org:443; Connection timed out while accessing https://freeswitch.org/stash/scm/fs/freeswitch.git/info/refs fatal: HTTP request failed make[1]: *** [update] Error 1 make[1]: Leaving directory `/usr/src/freeswitch' Are the stash servers down? -- Guillermo Ruiz Camauer

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variable_sip_to_tag not set in Leg A for bridged call
by Luis Azedo 19 Mar '15

19 Mar '15

Hi, making a bridge call and listening to all events in both legs with esl. i was expecting that CHANNEL_PROGRESS_MEDIA / CHANNEL_ANSWER (on Leg A) would include variable_sip_to_tag since there is already this info (seen in message sent) but we never get this variable to be set in any event. we only get the to_tag on leg A when the call is disconnected or if we put onhold/unhold to force the creation of event. during the call (before forcing hold/unhold) i do a uuid_dump of leg A and i can see the to_tag variable. the channel variable seems to be set only on nua_i_ack and there seems to be no event fired after that. thoughts ? Best

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Automated testing
by Stanislav Sinyagin 19 Mar '15

19 Mar '15

I need to build a test suite for a service provider's SIP service infrastructure. The box will initiate and accept calls and verify that SIP messages are valid, audio media is two-way, and call control features are also working. My plan for the moment is to use FreeSWITCH and to trigger the outbound calls via ESL, and then use packet capture and Perl modules for parsing and verifying the SIP messages. What are other options? I see that SIPp doesn't look too bad, but I never worked with it. This page lists mostly outdated and closed-source packages, so not very helpful: http://www.voip-info.org/wiki/view/Protocol+Verification+and+Testing I'm planning to publish as much of my work as possible as open source. Any input will be appreciated. thanks, stan

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Any whay to reload vars.xml?
by Victor Medina 19 Mar '15

19 Mar '15

Hi guys! Is there any way to reload vars.xml? -- Víctor E. Medina M. Software [image: Zoiper Click2Dial]+58424 291 4561 <zoiper://+584242914561>[image: ve] BB #79A8AFA2 /@VMCibersys

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bug with events?
by Денис 19 Mar '15

19 Mar '15

Hi. Possible, we found a bug. If we call to freeswitch with redirect to mobile phone and hangup before state "ringing" - ESL events show leg B without Call-ID (i.e. CHANNEL_HANGUP, CHANNEL_HANGUP_COMPLETE...) example (json view) https://pastebin.freeswitch.org/24022

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