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ignore_display_updates blocks display name for IVR
by mehmet ozisik 13 Feb '14

13 Feb '14

Hi, I am using FreeSWITCH-mod_sofia/1.5.8b+git~20140204T065630Z~5228e02b19~64bit. To block to get display update after connected to IVR , I have added <X-PRE-PROCESS cmd="set" data="ignore_display_updates=false"/> to vars.xml and FS does not send update anymore. But this time on the phone screen, "Outgoing Call" is written instead of IVR. I have checked the code find the place what causes this, but I wonder this is related with this variable somehow. I would like to ask you that is there any way to send IVR string as displayname? Remote-Party-ID: "Outgoing Call" <sip:IVR@192.168.169.169 >;party=calling;privacy=off;screen=no regards Mehmet

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libsofia_sip_ua_static won't compile
by Andre 13 Feb '14

13 Feb '14

When trying to compile the latest stable branch in VS 2012, release, 64 bit libsofia_sip_ua_static has over 807 Errors. So many it's hard to know what is wrong. Anyone have a clue what's wrong? Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory ..\libs\sofia-sip\libsofia-sip-ua\sip\sip_extra.c 45 1 libsofia_sip_ua_static Error 1845 error C2059: syntax error : ')' ..\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h 188 1 libsofia_sip_ua_static Error 758 error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory ..\libs\sofia-sip\libsofia-sip-ua\sip\sip_extra.c 45 1 libsofia_sip_ua_static

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xml cdr problems
by Jason Holden 12 Feb '14

12 Feb '14

Hi, I have a fs that is running an older master build of 1.3.0 and I'm noticing speratic problems with cdr postings where it doesn't post, or fields are missing. Is anyone awhere of issues with xmlcdr around that release? Is their a way I can debug this in the fs cli? Currently I've looked through my logs and am not seing any xmlcdr errors. Thanks.

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FYI - sql quoting issue with mod_skinny configs
by Nathan Neulinger 12 Feb '14

12 Feb '14

FS-6220 fixed with c65f2884747bfd4c0a97d540620a58596eb8f595 If you were taking any special steps to double single quotes inside of mod_skinny configs (such as for speed dials) that is now corrected in the code. None of the mod_skinny queries were protecting sql quotes. All of those cases have been corrected in both v1.2stable and master. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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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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