- FreeSWITCH-users - FreeSWITCH Mailing Lists

Verto & Sofia Registrations
by Abdul Hakeem 21 Feb '16

21 Feb '16

Hello, My understanding is that Sofia can only do one registration at a time, does Veryo exhibit this limit ? Is there any way to process multiple registrations in the background or to configure FS to use the database of another REGISTRAR ?. Much obliged, AH

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Freeswitch extended with ZRTP
by Assaf Dahary 21 Feb '16

21 Feb '16

Hi, I would like to setup FS1 as a 'ZRTP gateway' to a none-zrtp sip client. Here is my net setup: Client1 (without ZRTP)-> LAN -> FS1 (Gateway: register to FS2) -> NAT -> Internet-> Public IP (not NAT) FS2 -> CSipSimple (ZRTP enabled). Client1: Ø Ex#1000, Registered over LAN to FS1. FS1: Ø Being NAT with dynamic IP address Ø Registered as a Gateway to FS2 (in Internal profile). So FS1 is extended on FS2. Ø Setup as 'Proxy-Media = false' && zrtp_enrollment=true (trusted MITM). FS2: Ø Connected with static public IP address (not behind NAT) Ø Setup as 'Proxy-Media = true' && inbound-late-negotiation=true. CSipSimple: Ø Behind NAT (remote WiFi/3G) and is registered on FS2 (Internal profile). FS2 is successfully serving multiple CSipSimple ZRTP clients with end-to-end ZRTP secure calls. The problem: When calling from Client1/FS1 to FS2/CSipSimple, then FS1 shows ' WARNING! Incoming ZRTP CRC validation fails' and FS2 shows ' ZRTP not negotiated on both sides; disabling ZRTP passthr'. I have tested FS1 locally with CSipSimple and it manages to connect with ZRTP/MITM so it is capable of ZRTP. I follow up what have been recommended on the forum to extend FS1 and to verify matching codecs (I forced PCMU only on all devices) and to check RTP/UDP ports flow (SIP trace). I would appreciate any help on how to setup end-to-end ZRTP calls between FS1 and remote CsipSimple (FS2). Regards Assaf

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recording problems with mod_shout debian/master
by Luis Azedo 20 Feb '16

20 Feb '16

Hi, anyone having problems recording in mp3 format ? 2016-02-01 13:41:47.108078 [INFO] kazoo_node.c:625 exec: *uuid_record(96722133-5060-508(a)BJC.BGI.CG.BD <96722133-5060-508(a)BJC.BGI.CG.BD> start /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 600)* 2016-02-01 13:41:47.108078 [DEBUG]* switch_core_file.c:323 File /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 sample rate 8000 doesn't match requested rate 16000* 2016-02-01 13:41:47.128429 [DEBUG] *switch_core_media_bug.c:828 Attaching BUG to sofia/sipinterface_1/995582142(a)teste.sip.90e9.com <995582142(a)teste.sip.90e9.com>* 2016-02-01 13:41:47.148020 [DEBUG] *switch_ivr_async.c:1491 No silence detection configured; assuming start of speech* 2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 LAME 3.99.5 64bits (http://lame.sf.net <http://lame.sf.net>)* 2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 polyphase lowpass filter disabled* 2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!* 2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* 2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!* 2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!* 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!* 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!* 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* Thanks

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one way audio - another report related to auto-jitter-buffer-msec possible behavior change with 1.6
by Nathan Neulinger 20 Feb '16

20 Feb '16

Just wanted to add an additional data point on the one way audio issue (as it turns out - also with grandstreams) with 1.6. AnalogDevice <-> Grandstream <-> FS <-> SIP-Provider One my most recent test/capture/etc, the behavior I saw was that it looked to me like audio stopped being sent from freeswitch to outside/external leg once early media finished. Note that the grandstream was continuing to send RTP to FS, but FS wasn't passing any of it on to the provider. Once I saw the thread from the other user having this problem, tried turning it off on my profiles that had it enabled, and the problem appears to have gone away. I can provide logs/debug/captures/etc. if requested from a test environment. I did not have any reports of this issue except related to grandstreams (ht-701) - in my case, almost all related to analog faxes (bulk of our ATA usage). Also have not been able to verify the fix yet on a call originating from the grandstream (testing from home). -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Skypopen Debian Jessie install issues
by Jason Bedward 20 Feb '16

20 Feb '16

Hi, I have tried to install this on Debian Jessie but had some issues with dependencies which are not matching what the install guide says. When I got to the CLI and load the module I get the error below. I have libjpeg62-turbo & libjpeg62-turbo-dev installed which I think it the current version for Debian. Please can someone help guide me with this version of Debian. Thanks +OK log level 9 [9] +OK console log level set to DEBUG freeswitch@192.168.2.233@internal> load mod_skypopen +OK Reloading XML -ERR [module load file routine returned an error] 2015-12-15 12:01:55.996236 [INFO] mod_enum.c:880 ENUM Reloaded 2015-12-15 12:01:55.996236 [INFO] switch_time.c:1415 Timezone reloaded 1781 definitions 2015-12-15 12:01:55.996236 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_skypopen.so **/usr/local/freeswitch/mod/mod_skypopen.so: undefined symbol: jpeg_resync_to_restart**

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[mod_skyopen][Debian Jessie][Error while loading]
by Bilal Abbasi 20 Feb '16

20 Feb '16

Hi Users, I am getting error while loading mod_skyopen on debian jessie 8 64bit, i am using freeswitch Version 1.7.0 git 09b4156 2016-02-04 17:08:30Z 64bit Logs: https://pastebin.freeswitch.org/24569 I could not find any official link for installation on Debian 8. Regards Abbasi

1 0

new version of vpx needed for those tracking master.
by Michael Jerris 20 Feb '16

20 Feb '16

Please note, we have updated code in master that will now require the newer version of libvpx (and anything such as libav that is linked against vpx will also need to be updated) in order to use video features. The tarball for this newer version is located at http://files.freeswitch.org/downloads/libs/libvpx2-1.5.0.tar.gz . If you are using our debian packages for dependencies, these packages are already available in the unstable repo.

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UNALLOCATED_NUMBER handling
by Jerry Kendall 20 Feb '16

20 Feb '16

Hello all.... I'm trying to add code to the dialplan to handle the UNALLOCATED_NUMBER error. I'm in Canada, when I call a US based 800 number I should, and now am getting UNALLOCATED_NUMBER error from the Canadian based ITSP. However, I want to handle it in the DialPlan. I want to tell the users, via audio file playback, that the number is not assigned/allocated. I have looked in lots of places and found references to transfer_on_fail and a few other ideas and have tried many but, I just can't seem to make this work. Not sure if this is the best way to handle this. my 'default' dialplan has this------------- <include>  <extension name="public_did_11234567890" continue="true"> <condition field="${caller_id_number}" expression="^11234567890$"/> <condition field="${destination_number}" expression="^1?([2-9]\d{2}[2-9]\d{2}\d{4})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="transfer_on_fail=UNALLOCATED_NUMBER UNALLOCATED_NUMBER xml error"/> <action application="bridge" data="sofia/gateway/GW-0/1$1"/> </condition> </extension> </include> and I have the following in for UNALLOCATED_NUMBER extension---- <include> <extension name="UNALLOCATED_NUMBER" continue="true"> <condition field="destination_number" expression="UNALLOCATED_NUMBER" continue="false" break="on-true"> <action application="sleep" data="1000"/> <action application="playback" data="$${conf_dir}/sounds/unallocated_number.wav"/> <action application="hangup" data="NORMAL_CLEARING"/> </condition> </extension> </include> Jerry

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Library linking problem in ESL
by Deepika Yadav 19 Feb '16

19 Feb '16

Hi, I am using java ESL, and getting the following error : Exception in thread "main" java.lang.UnsatisfiedLinkError:org.freeswitch.swig.freeswitchJNI.new_JavaSession_SWIG_0()J at org.freeswitch.swig.freeswitchJNI.new_JavaSession__SWIG_0(Native Method) at org.freeswitch.swig.JavaSession.<init>(JavaSession.java:37) at MyESLTest.main(MyESLTest.java:465) and I also activates another similar esl program from the dialplan that runs fine. would be thankful if any clue regarding resolving it can be suggested. Regards, Deepika

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Selective Forwarding Unit (SFU)
by martin.hoole＠emailn.de 19 Feb '16

19 Feb '16

Are there plans to implement support for Selective Forwarding Unit (SFU) for video or audio rather than mixing video together? https://tools.ietf.org/html/draft-ietf-avtcore-rtp-topologies-update-01

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