February 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Mod_Manage won't build switch_core_port_allocator_new
by Andre Demattia 12 Feb '14

12 Feb '14

I'm not good at git. Can you tell me the entire syntax? git pull git://git.freeswitch.org/freeswitch/freeswitch.git -----Original Message----- From: "Peter Olsson" <peter(a)olssononline.se> Sent: ‎2/‎10/‎2014 2:28 AM To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Mod_Manage won't build switch_core_port_allocator_new Just "git pull" the updates and it will be there. 2014-02-09 22:12 GMT+01:00 Andre <andretodd(a)verizon.net>: How do I get that? git clone -b d6dfab2955df6fdb40e0da11c06bd5bc2647fb1f git://git.freeswitch.org/freeswitch.git ??? Also, does this new stable have this fix too? http://jira.freeswitch.org/browse/FS-5994 Thanks so much Brian, Andre -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Saturday, February 8, 2014 7:51 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Mod_Manage won't build switch_core_port_allocator_new commit d6dfab2955df6fdb40e0da11c06bd5bc2647fb1f -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Feb 8, 2014, at 3:46 PM, Andre <andretodd(a)verizon.net> wrote: > Hi, I see the master has the fix. > I used this > > git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git > > and this still has the wrong code. How do I get the stable version > with the fixes? > Thanks > Andre _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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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Missing <domain name="... failed to configure rayo server
by ik 12 Feb '14

12 Feb '14

Hello, I'ved updated today my development server for fs 1.2.21, and suddenly I've started getting this error message from mod_rayo (that I'm using). I'm using the default rayo.conf.xml file, that worked well for me up until now. At line 86, there is a <domain name="$${rayo_domain_name"} So what am I missing here ? Thanks, Ido

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play_and_get_digits disconnect after input
by Grant Bagdasarian 12 Feb '14

12 Feb '14

Hello, I have the following in my dialplan. <context name="1000"> <extension name="start"> <condition field="destination_number" expression="^(.*)$" require-nested="false" break="never">  <action application="answer"/> <action application="play_and_get_digits" data="1 1 2 5000 # ${prompts_directory}/reprompt.wav ${prompts_directory}/invalid.wav dtmfInput \d+ 2500"/> <condition field="${dtmfInput}" expression="1" break="never">  <action application="javascript" data="$${scripts_directory}/say_characters.js ${security_code} ${characters_directory} 500"/> </condition>  <action application="hangup"/> </condition> </extension> </context> As soon as I enter 1, the condition block isn't executed and the channel is hungup immediately. |--- Dialplan: Processing recursive conditions level:1 [start_recur_1] require-nested=FALSE |--- Dialplan: sofia/inbound/1002(a)192.168.1.1 Regex (FAIL) [start_recur_1] ${dtmfInput}() =~ /1/ break=never Dialplan: sofia/inbound/1002(a)192.168.1.1 Regex (PASS) [start] destination_number(start) =~ /^(.*)$/ break=never ................. EXECUTE sofia/inbound/1002(a)192.168.1.1 play_and_get_digits(1 1 2 5000 # /usr/src/freeswitch/sounds//1/prompts//reprompt.wav /usr/src/freeswitch/sounds//1/prompts//invalid.wav dtmfInput \d+ 2500) 2014-02-12 13:25:53.440479 [DEBUG] switch_ivr_play_say.c:1315 Codec Activated L16@8000hz 1 channels 20ms 2014-02-12 13:25:59.720479 [DEBUG] switch_rtp.c:3901 RTP RECV DTMF 1:1280 2014-02-12 13:25:59.720479 [DEBUG] switch_channel.c:471 RECV DTMF 1:1280 2014-02-12 13:25:59.720479 [DEBUG] switch_ivr_play_say.c:1703 done playing file /usr/src/freeswitch/sounds//1/prompts//reprompt.wav EXECUTE sofia/inbound/1002(a)192.168.1.1 hangup() Could anyone tell me what I'm doing wrong? Don't mind the variables, I removed some pieces of the dialplan.

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rtmp session timeouit
by Javier Menendez 12 Feb '14

12 Feb '14

Hi, I think it is a simple question but still I can't find the answer. Do the rtmp sessions expire? Sometimes I get the USER_NOT_REGISTERED error when calling a previous registered user, so I guess it is expiring but I can not find where to configure the expiration time.

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FreeSWITCH 1.2.20...err umm 1.2.21 Released
by Ken Rice 12 Feb '14

12 Feb '14

Hey FreeSWITCHers! The FreeSWITCH Team is Proud to Announce FreeSWITCH 1.2.21. Early this week Anthony tracked down a slight problem with Sofia that could lead to memory corruption. This lead to us cutting and getting 1.2.20 ready to go on Monday. Debian and Centos6 packages we moving along just fine, however Centos5 was being a bit problematic. Took us a little bit to track that down, but a Chris Rienzo contributed a patch that took care of a small bit of code that gcc on Centos5 didn¹t like. Other updates from 1.2.19 include: Updates to TLS/SSL code for better dection of the libs and allow some better options for selecting cipher suites Updates to mod_rayo Updates to Sofia TCP handling deletion retired project support code for svn that has not been used in ages Along with several other issues... Source tarballs are available at http://files.freeswitch.org/freeswitch-1.2.21.tar.bz2 (gz, xz, and zip can be found here also) Debian and Centos packages are available via the FreeSWITCH project repos as usual. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Fax T.38 reinvite drops most of the SDP
by Peter Steinbach 11 Feb '14

11 Feb '14

Hallo, we have the following fax scenario: Asterisk(SIP provider) => Freeswitch => Auerswald COMmander 6000R PBX IPS 10.10.10.10 => 20.20.20.20 => 30.30.30.30 Fax is initiated by Asterisk with PCMA codec Auerswald connects with PCMA and then does a T.38 reinvite In the T.38 reinvite Auserwald PBX sends a m=audio 0 RTP/AVP 19 m=image 49376 udptl t38. Freeswitch then finally drops most of the SDP and answers with m=image 0 udptl t38. m=image 0 udptl t38. and Auerwald PBX hangs up Anybody has an idea what is causing this? Why is Freeswitch not forwarding the complete SDP message. Maybe the "m=audio 0" is causing the problem? Dialplan is just sip_copy_multipart=false and bridge In internal and external profile <param name="t38-passthru" value="true"/> is set. Freeswitch log shows in between: 2014-02-10 13:23:10.869112 [INFO] sofia.c:5543 Activating T38 Passthru ========= Here are the SDPs================== SDP from Reinvite Auerswald to Freeswitch v=0. o=- 5045970593938686503 4940450077875186463 IN IP4 30.30.30.30. s=-. c=IN IP4 30.30.30.30. t=0 0. m=audio 0 RTP/AVP 19. m=image 49376 udptl t38. a=T38FaxVersion:0. a=T38MaxBitRate:14400. a=T38FaxUdpEC:t38UDPRedundancy. a=T38FaxUdpEC:t38UDPFEC. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxDatagram:512. a=T38FaxMaxBuffer:1472. SDP from Invite Freeswich to Asterisk: v=0. o=FreeSWITCH 1392015640 1392015642 IN IP4 20.20.20.20. s=FreeSWITCH. c=IN IP4 20.20.20.20. t=0 0. m=image 19342 udptl t38. a=T38FaxVersion:0. a=T38MaxBitRate:14400. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:1472. a=T38FaxMaxDatagram:512. a=T38FaxUdpEC:t38UDPFEC. SDP from Ok from Asterisk to Freeswitch: v=0. o=root 1238355194 1238355195 IN IP4 10.10.10.10. s=Asterisk PBX 11.0.1. c=IN IP4 10.10.10.10. t=0 0. m=image 4609 udptl t38. a=T38FaxVersion:0. a=T38MaxBitRate:14400. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxDatagram:702. a=T38FaxUdpEC:t38UDPFEC. SDP from 200 OK from Freeswitch to Auerswald: v=0. o=FreeSWITCH 1392020936 1392020938 IN IP4 20.20.20.20. s=FreeSWITCH. c=IN IP4 20.20.20.20. t=0 0. m=image 0 udptl t38. m=image 0 udptl t38. Auerswald sends a BYE with Reason: SIP;cause=488;text="Not Acceptable Here". -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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Opus error: Invalid SDP for opus. Don't ask.. but it needs a /2
by Donny Hardyanto 11 Feb '14

11 Feb '14

Hi, I just testing the Opus in FS talking with Linphone. There is error in FS log: Invalid SDP for opus. Don't ask.. but it needs a /2 What is that mean? The call is success full connected but one side voice. Donny

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Freeswitch Week in Review(Master Branch) Feb 2nd through Feb 8th
by Kathleen King 11 Feb '14

11 Feb '14

This past week was productive for the Freeswitch master branch with many new changes and bug fixes. On the packaging front, .zip joined tar.xz in the default source release archives. Located here: http://files.freeswitch.org/ In terms of stability there were several use cases that were fixed: --Fixed mod_lua crashes related to build symbols introduced during upgrade from lua 5.1 to lua 5.2 --Fixed a crash in mod_conference related to whisper tones --Fixed a crash in mod_shell_stream --Fixed a crash involving T-38 in mod_sofia when the call in bridged between TDM and SIP Several new features were added: – New command 'gethost' allows for a DNS lookup from the Freeswitch console --Added bypass-media-after-hold to complement the existing parameter resume-media-on-hold with more information located at: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#bypass-media-after-hold --Added capability to record conference with a single participant --Added support for Sofia to configure accepted TLS versions and enabled support for perfect forward secrecy. --Expanded Sofia profile status to include websockets Improvements in cross platform build supports: --Spandsp improvements for BSD --Some initial build fixes for Ubuntu 14.04 --Fixing some compiler optimization snafus Feedback welcome. The referenced commits are in the attached text file. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king(a)quentustech.com

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ZRTP proxying
by Dahlberg, David 11 Feb '14

11 Feb '14

Hello all, I have some problems with proxying ZRTP streams trough Freeswitch. My setup is as follows: [FS-A]----[FS-B]----[FS-C] For End-to-end security, "A" should initiate a ZRTP session and "C" shall terminate it. What works: * With late negotiation, two ZRTP security associations (A,B) and (B,C) are established. Of course in that case there is no E2E encryption, so no real advantage over the usage of TLS/SAVP . According to the Wiki, it should be possible to use "proxy_media" with "inbound-zrtp-passthru". Only this does not work and the problem is this: ZRTP uses a slightly modified RTP header format with a CRC checksum that is calculated over the whole ZRTP header and payload (excluding the CRC field). Freeswitch ("FS-B" in my use-case) on the other hand exchanges the RTP "sequence numbers" and "syncronization source identifiers" (SSRC) (which in ZRTP become "Source Identifiers"). Which leads to the case that the checksums are valid on call leg (A,B) but invalid on leg (B,C). Accordingly, "FS-C" receives ZRTP hello packets from "FS-A", calculates the checksums, finds a mismatch and discards the packets. I have found a thread on [freeswitch-users] dated January 2013 and with a subject "problems with freeswitch + zrtp in proxy-media mode" which discusses exactly this problem. Within this thread was mentioned, that FS would change the proxy behaviour, if it encounters "zrtp-hash" attribute in the SDP and a quick test with SFLphone and w/o proxy_media seems to confirm this. So is there any possibility to originate this attribute, when FS originates the ZRTP? Regards, David -- David Dahlberg Fraunhofer FKIE, Dept. Communication Systems (KOM) | Tel: +49-228-9435-845 Fraunhoferstr. 20, 53343 Wachtberg, Germany | Fax: +49-228-856277

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Controlling B-Leg on A-Leg hangup
by Alex Lake 11 Feb '14

11 Feb '14

I've got a situation where I think I want to prevent the A-Leg hangup resulting in an immediate B-Leg termination. It's like a form of attended transfer where A calls B, B calls C and then hangs up, with the desired outcome that A is talking to C. Suggestions?

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