- FreeSWITCH-users - FreeSWITCH Mailing Lists

Receive chat
by Sameer Khan 05 May '15

05 May '15

Hi, Is there any command which can receive chat from sip channel so I can put that received message in to a variable.

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407 Proxy authentication
by bhavik patel 05 May '15

05 May '15

Hi, We need 407 Proxy authentication require sip dialog while calling between sip to sip extension. Currently when we are calling, we never got that sip dialog. I am sure there must be some configuration to achieve this. Can anyone please help us to make it working? -- Thanks, Bhavik Patel

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FreeSWITCH and the VUC
by Ken Rice 04 May '15

04 May '15

New Post on freeswitch.org from Kathleen check it out at http://ift.tt/1KHXdS4 FreeSWITCH and the VUC Go check out all the new, awesome features going into FreeSWITCH! The core developers made an appearance on the VUC call this past Friday to talk about and demonstrate some of the fantastic, new features in FreeSWITCH.

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FreeSWITCH Week in Review (Master Branch) April 25th-May 1st
by Ken Rice 04 May '15

04 May '15

New Post on freeswitch.org from Kathleen check it out at http://ift.tt/1Ia2Ngj FreeSWITCH Week in Review (Master Branch) April 25th-May 1st Hello, again. This passed week in the FreeSWITCH master branch we had 14 commits. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. The following bugs were squashed: FS-7472 [mod_sofia] Fix for a bug where the rtp-digit-delay profile param was being ignored FS-7488 [mod_managed] Fixed a build error FS-7490 [mod_rayo] Fixed a bug with the format of mod_rayo generated regex not working with newer libpcre FS-7491 [mod_graylog2] Send timestamp with millisecond precision instead of microsecond as required by GELF FS-7466 Fixed a bug causing audio issues by repeated log lines printing when rtp_manual_rtp_bugs is set to ALWAYS_AUTO_ADJUST FS-7496 [mod_http_cache] Fixed an issue with the URL args being included in the cache file name and causing problems opening the files later

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Help with FSV record.
by dex_Cz 04 May '15

04 May '15

Hi guys! I have a question. Somebody tried to record in the FSV file when i bridge call? I hame a simple context from example. But it is not working. <include> <context name="centos"> <extension name="from-centos"> <condition field="destination_number" expression="^(onops)$"> <action application="record_fsv" data="/usr/local/freeswitch/recordings/testrecord.fsv"/> <action application="sleep" data="100"/> <action application="bridge" data="user/1004(a)172.16.0.2"/> <action application="hangup"/> </condition> </extension> </context> </include> I tried to change the context <include> <context name="centos"> <extension name="from-centos"> <condition field="destination_number" expression="^(onops)$"> <action application="export" data="execute_on_answer=uuid_record ${uuid} start /usr/local/freeswitch/recordings/testrecord.fsv"/> <action application="bridge" data="{ignore_early_media=true}user/1004(a)172.16.0.2"/> <action application="hangup"/> </condition> </extension> </context> </include> It also did not help. In the end I try to enter these commands from the console uuid_setvar <uuid> enable_file_write_buffering false (its says ok) and uuid_record <uuid> start /usr/local/freeswitch/recordings/testrecord.fsv But i have a error 2015-04-30 12:27:16.635463 [ERR] mod_fsv.c:702 You are asking to write 342 bytes of data which is not supported. Please set enable_file_write_buffering=false to use .fsv format 5e9ce2b0-ef23-11e4-b181-d1f037dabce9 2015-04-30 12:27:16.635463 [ERR] switch_ivr_async.c:1155 Error writing /usr/local/freeswitch/recordings/testrecord.fsv Anyone can help me, What I'm doing wrong? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Help-with-FSV-record-tp759615… Sent from the freeswitch-users mailing list archive at Nabble.com.

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core dumped. PANIC: unprotected error in call to Lua API (multiple Lua VMs detected)
by Aqs Younas 04 May '15

04 May '15

Hi, users. I have installed the freeswitch(FreeSWITCH Version 1.4.18~64bit ( 64bit) on fresh installation of ubuntu(14.04). When i try to load mod_vlc i see this error and core dumped. PANIC: unprotected error in call to Lua API (multiple Lua VMs detected) Your help would be much appreciated. Thanks

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restrict gateway to domain
by Roman Dissauer 04 May '15

04 May '15

Hey! I’m having troubles restricting a gateway to a single domain in a multi-tenant freeswitch setup. It is clear for me that if the gateway is defined in sip_profiles everyone on the switch can bridge to "sofia/gateway/<gateway id>/<number>“. So I defined the gateway in directory/<domain>/<gateway>.xml. Now the gateway is wrapped in <domain name="<domain>"> tag, but users from other domains can still bridge to "sofia/gateway/<gateway id>/<number>“. What do I miss here? Thanks! Roman Dissauer

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problems using systemd freeswitch unit
by covici＠ccs.covici.com 04 May '15

04 May '15

Hi. In the freeswitch tree freeswitch-new/debian/freeswitch-systemd.freeswitch.service was there and I tried to get it to work in my systemd. The problem seems to be that systemd is getting very confused when fs backgrounds the process. I got things to work by adding -nf and changing the type from forking to simple, so fs never backgrounds. My question is, has anyone gotten the original systemd unit to work and if so, what did you do? The only things I changed was the directories and I added some ExecStartPre commands that I needed. Thanks in advance for any ideas. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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sip_to_user and destination number
by Tanguy 03 May '15

03 May '15

Hello, My provider did not send correct DID number in the INVITE packet but i can use "To" argument INVITE sip:gw+0a96c3d3-0b0e-4864-b9ec-759fa4422429@92.222.18.xxx:5080;transport=udp;gw=0a96c3d3-0b0e-4864-b9ec-759fa4422429 SIP/2.0. Call-ID: 25016-VB-188fd96b-526e3dbd4(a)sip.ovh.fr. Contact: <sip:10.7.1.60:5060>. Content-Type: application/sdp. CSeq: 403749831 INVITE. From: "0967212xxx" <sip:0967212xxx@sip.ovh.fr;user=phone>;tag=25016-VE-188fd96c-18bb43586. Max-Forwards: 27. Record-Route: <sip:91.121.129.20:5060;lr>. *To: <sip:0557590xxx@10.7.1.60;user=phone>.* Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-WGZO-1fe73949-2df58378. Using asterisk i can bypass the issue using something like exten => s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) but i am unable to do the same under freeswitch. My trunk configuration seems correct, as you can see i used auto_to_user, but the destination number remains 0033972480xxx when i call 0557590xxx. 2015-05-01 18:02:12.235827 [INFO] mod_dialplan_xml.c:635 Processing 0967212xxx <0967212xxx>->0033972480xxx in context public <include> <gateway name="0a96c3d3-0b0e-4864-b9ec-759fa4422429"> <param name="username" value="0033972480xxx"/> <param name="password" value="xxxxxxx"/> <param name="proxy" value="sip.ovh.fr"/> <param name="expire-seconds" value="800"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> <param name="extension" value="auto_to_user"/> <param name="context" value="public"/> </gateway> </include> I tried to edit my inbound dialplan manually, it works using <condition field="${sip_to_user}" expression="0557590xxx" > but i prefer a proper way to do this because i will also use telcos with normal invite packets I how i can copy $sip_to_header to destination for this specific trunk ? Please note that i use fusionpbx. Best regards, sorry for my bad English

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Fail at save voicemail message
by Alejandro 03 May '15

03 May '15

Hi all, I has an issue with my voicemail setup. I'm running this: FXO -> FreeSwitch -> Voicemail When I record the message all is ok, just that this fail to save. I see this error: 2015-05-02 21:18:05.539885 [DEBUG] switch_core_session.c:2893 sofia/internal/6303XXXX(a)fs-01.domain.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) My Dialplan be:  <extension name="public_did"> <condition field="destination_number" expression="^(4862XXXX)$"> <action application="answer"/> <action application="playback" data="/usr/storage/voicemail/default/ domain.com/1000/msg_06c6288e-ee11-11e4-b862-1dea27edc68b.wav"/>  <action application="set" data="skip_greeting=true"/> <action application="set" data="skip_instructions=true"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> Some idea, what I'm doing wrong? Thanks in advance. Alejandro

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