October 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH - Segmentation fault in libcrypto.so.1.0.0
by Shaun Stokes 15 Oct '15

15 Oct '15

Hi, We use FreeSWITCH in a production enviroment and this morning experienced an outage caused by a segmentation fault (error 4 in libcrypto.so.1.0.0). Based on this error I'm geussing this is linked to the OpenSSL module? We have TLS disabled in all of our SIP profiles, how can we stop FreeSWITCH from using libcrypto, or is anyone able to advise on another method to fix this? Our system: Ubuntu 12.04 LTS FreeSwitch 1.4.15 PostgreSQL 9.3.6 OpenSSL 1.0.1 Please see below for a copy of the kernal log, I've replaced server names and IP addresses with xxxx for security purposes. Sep 19 00:21:13 xxxx kernel: [1276167.422451] show_signal_msg: 57 callbacks suppressed Sep 19 00:21:13 xxxx kernel: [1276167.422456] freeswitch[53622]: segfault at 7f4590c6b000 ip 00007f45908fb6e3 sp 00007f453f50dfc0 error 4 in libcrypto.so.1.0.0[7f4590890000+1b2000] This isn't the first time we've seen this error, but it has been many months since this last occured. The last time we expeirenced this type of segmentation fault (libcrypto) in FreeSWITCH we installed all available updates. On this occasion the segmentation fault happened overnight when we only have the occasional call. This happened approx 37 seconds after the last call had ended (see below). d89ff492-5e5b-11e5-90c5-d74017d9130d 2015-09-19 00:20:36.601890 [NOTICE] sofia.c:952 Hangup sofia/internal/3303@xxxx [CS_EXECUTE] [NORMAL_CLEARING] da61a97e-5e5b-11e5-90f6-d74017d9130d 2015-09-19 00:20:36.621841 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/+xxxx@xxxx [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] There doesn't seem to be any connection with the time or any events on the server, there were no authentication attempts or logs that we can find which could be linked to segmentation fault. Hope someone here can advise. Many Thanks, Shaun ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________

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Create table in PostgreSQL for CDRs
by Michael Nielsen 15 Oct '15

15 Oct '15

I'm trying to use the mod_cdr_pg_csv module for storing CDRs in PostgreSQL. https://wiki.freeswitch.org/wiki/Mod_cdr_pg_csv I can see the schema/fields in the .xml config, but are there a .sql-script for creating the table in PostgreSQL with the correct column types etc.?

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Export all user vars to the b leg?
by Victor Chukalovskiy 14 Oct '15

14 Oct '15

Good day, wondering if there is a one-liner method to export all user-specific variables to the b-leg of the call. I'm referring to all the vars I set within <variables> section of <user> in the directory. I could've done it in the dialplan, but then I need to do it once for each var: <action application="export" data="foo1=${foo1}"/> ..... <action application="export" data="fooN=${fooN}"/> Many thanks! -Victor

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Re: [Freeswitch-users] Stereo Playback
by Sam Machin 14 Oct '15

14 Oct '15

Hi Everyone, Is it possible to make the playback command play a file in stereo, I have a 2 channel media session open from my client but I get a message in the log "File has 2 channels, muxing to 1 channel will occur." The codec is reported as : a=rtpmap:120 opus/48000/2 so I'm assuming the /2 is 2 channels right? I got the impression that mod_stream would playback stereo MP3's too but I couldn't seem to get that working either, any suggestions welcome the box is purely to allow SIP clients to call in and hear a high quality stereo stream so pretty open to options or hacking something like conference even!

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Developing with FreeSWITCH - getting started
by rick.jarvis＠magicmail.mooo.com 14 Oct '15

14 Oct '15

I want to create an app for call handling on FreeSWITCH, in particular stacking calls, connecting to audio channels on a sound card in response to mouse clicks etc. What’s the best way to get started? In particular: What are my options, API / Language wise? I’m familiar with Python, Bash, and Basic. Are these options? Any tips, or pointers to useful ‘idiots guides’ / ‘howtos’ …? Thanks! RICK

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Freeswitch ignoring BYE sent by LegA when ignore_early_media=True
by Sushil Tripathi 14 Oct '15

14 Oct '15

I need to play a hold music file on LegA while Freeswitch tries to bridge call to one of the listed numbers. Here in dial plan I have used only one number. If I use following dialplan without ignore_early_media=true in bridge, Leg A is able to disconnect the call when Freeswitch is trying to bridge the call to one of the number in legB, but if ignore_early_media is true in that case BYE sent by Leg A is ignored by freeswitch causing call to keep ringing at LegB. <extension name="SIP-TO-SIP"> <condition field="destination_number" expression="^91(\d+)$"> <action application="answer"/> <action application="set" data="hold_music=/srv/sounds/123.mp3"/> <action application="bridge" data="{caller_id=+2000,ignore_early_media=true,RECORD_STEREO=true,originate_timeout=150,hangup_after_bridge=true,legA_UUID=${uuid}}sofia/gateway/anveo/${destination_number}"/> <action application="hangup"/> </condition> </extension> While debugging for the root cause of this issue we found that this bug is because of side-effect of commit-id "3e00d270d7477a63b3c2e845b5473e6e7e3f1cfc" which was added to fix bug https://freeswitch.org/jira/browse/FS-6006 . Now we are not sure whether to remove this patch or not. Can you please suggest how to proceed further. Thanks & Regards, Sushil Tripathi +91-9910700668 -- SuperReceptionist is now available on Android mobiles. Track your business on the go with call analytics, recordings, insights and more: Download the app here <https://play.google.com/store/apps/details?id=com.superreceptionist>

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No SIP UPDATE will send
by Alexander Haugg 14 Oct '15

14 Oct '15

Hi all, I have the problem, that the freeSwitch don't send a SIP UPDATE to the PBX. Scenario: PBX (Siemens H3K) is connected via SIP trunking to the FreeSwitch. If I have two Legs from the H3K to the Freeswitch I'm using the uuid_bridge command to connect this two legs (works fine), but the call legs will not updated via SIP UPDATE. The PBX have no information to update the displays on the telephones. The target number (toHeader) was the routing number from the H3K to the freeswitch. Now I need a SIP UPDATE from FreeSwitch to PBX to actualize the toHeader with the new partner number (the partner number is the fromHeader of the other leg). In the sofia profile I was try it with the parameter <param name="send-display-update" value="true"/> and without this parameter (in the source code the default is "true") What can I do to force an UPDATE to the call legs? Version of the FreeSwitch FreeSWITCH Version 1.5.15b+git~20150121T080154Z~90ab1d16f5~64bit (git 90ab1d1 2015-01-21 08:01:54Z 64bit) In the INVITE from the PBX the ALLOW header includes the UPDATE Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, INFO, PRACK, UPDATE Many thanks!

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Re: [Freeswitch-users] Help with compiling libyuv 32bits
by Luis Daniel Lucio Quiroz 13 Oct '15

13 Oct '15

BTW, has anybody been able compile it in 32 bits? Le 2015-10-13 12:28 PM, "Mario" <mario_fs(a)mgtech.com> a écrit : > Thank you, a question: How does this differ from the one in the support > lib? The URL will go into the wiki and possibly a homebrew formula. > > I was able to make install it into /usr/local instead of /usr but > freeswitch still does not find it. Will look into it further. > > > On Oct 12, 2015, at 10:48 AM, Michael Jerris <mike(a)jerris.com> wrote: > > 509 git clone https://mikej@freeswitch.org/stash/scm/sd/libyuv.git > 510 cd libyuv > 512 make > 514 sudo make install > > just did on my machine and it worked fine? > > On Oct 12, 2015, at 1:39 PM, Mario <mario_fs(a)mgtech.com> wrote: > > I got nowhere this weekend trying to compile libyuv on OS X so will wait > for Michael’s magic. I am also looking into making the new prereqs homebrew > formulas and want to see if it’s possible for libyuv. If so, I was thinking > of doing a homebrew formula called “freeswitch-support/prereqs” with > everything in it. > > > On Oct 8, 2015, at 6:31 PM, Michael Jerris <mike(a)jerris.com> wrote: > > what does this have to do with this thread? > > On Thursday, October 8, 2015, Thomas Quintana <thomas(a)bettervoice.com> > wrote: > >> https://hub.docker.com/r/bettervoice/freeswitch-container/ >> >> Thomas Quintana >> Chief Technology Officer >> Phone: 1 (512) 677-6200 >> Website: http://www.bettervoice.com >> >> >> >> On Thu, Oct 8, 2015 at 3:54 PM, Mario <mario_fs(a)mgtech.com> wrote: >> >>> Thanks I look forward to it since I am closer to 1.6! >>> Mario G >>> >>> > On Oct 8, 2015, at 12:50 PM, Michael Jerris <mike(a)jerris.com> wrote: >>> > >>> > You don't need the utilities, I'm going to have to go through and >>> figure out what all i needed to do manually and how I did it and write it >>> up for you... I'll try to do this soon. >>> > >>> >> On Oct 8, 2015, at 2:44 PM, Mario <mario_fs(a)mgtech.com> wrote: >>> >> >>> >> I git cloned libyuv from the FreeSWITCH repos into OS X. The official >>> libyuv google doc states I need the google chromium utilities. But >>> searching the web I see FreeSWITCH folks are just using “make”. There is no >>> configure file but make seems to work. But make install fails since it does >>> not know the target directories. The question is: >>> >> >>> >> Do I need the chromium utilities to build the FreeSWITCH libyuv >>> version or can I just use make/make install and how to specify targets. I >>> know it’s OS X (BSD) but Linux should be close enough. Thanks, >>> >> Mario G >>> >> >>> _________________________________________________________________________ >>> >> 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 >>> > >>> > >>> > >>> _________________________________________________________________________ >>> > 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 >>> >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > 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 >

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How to originate a SIP call without SDP
by Gustavo Garcia 13 Oct '15

13 Oct '15

Is it possible to originate a SIP call from Freeswitch without SDP in the INVITE? (SDP offer/answer will happen in the 200 / ACK) I have tried to play with the enable-3pcc variable without luck. Any help is very welcomed. Thx a lot, G.

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Gateway State
by Leonardo Lima Ribeiro 13 Oct '15

13 Oct '15

Hello all! I need to know how many calls are running for each gateway (in my case outbound calls). When I do sofia status I have the running number of calls for the external profile like RUNNING (1), but at the gateway line of the result, the state remains REGED. I tried too sofia status gateway, but no success. Here’s my Sofia status: freeswitch@internal> sofia status Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external-ipv6::example.com gateway sip:joeuser@example.com NOREG 192.168.10.72 alias internal ALIASED external profile sip:mod_sofia@192.168.10.72:5080 RUNNING (1) external::example.com gateway sip:joeuser@example.com NOREG external::example.com gateway sip:joeuser@example.com NOREG external::5_JIVE gateway sip:**********@domain.jive.rtcfront.net REGED external::4_NEXTIVA gateway sip:*******@sbc.voipdnsservers.com REGED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@192.168.10.72:5060 RUNNING (1) ================================================================================================= 4 profiles 1 alias And my sofia status gateway, here I can’t see even the number of calls that is running: freeswitch@internal> sofia status gateway Profile::Gateway-Name Data State Ping Time IB Calls(F/T) OB Calls(F/T) ================================================================================================= external-ipv6::example.com sip:joeuser@example.com NOREG 0.00 0/0 0/0 external::example.com sip:joeuser@example.com NOREG 0.00 0/0 0/0 external::example.com sip:joeuser@example.com NOREG 0.00 0/0 0/0 external::5_JIVE sip:**********@domain.jive.rtcfront.net REGED 0.00 0/0 0/0 external::4_NEXTIVA sip:*****@sbc.voipdnsservers.com REGED 0.00 0/0 0/2 ================================================================================================= 5 gateways: Inbound(Failed/Total): 0/0,Outbound(Failed/Total):0/2 So, I just want to know if my current call is using 5_JIVE or 4_NEXTIVA. In this actual scenario I can know that my calls are being made by Nextiva because I have 0/2 outbound calls there and for Jive I have 0/0… But if I have any number for Jive, I will not know anymore... Any idea? Thank you, Leonardo Ribeiro

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