March 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

how to use " remote_media_ip, remote_media_port" in dialplan
by hjz51 05 Mar '12

05 Mar '12

Hi, I tried to use the variable remote_media_ip, remote_media_port within dialplan, but it is not returning anything. Does anyone know when this variable gets set and how to have this variable to be set as soon as an INVITE hit freeswitch? Thanks,

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Using mod_shout over ssl, curl issue with curl-ca-bundle.crt file location
by Matt Stockton 05 Mar '12

05 Mar '12

Hi all, I just upgraded to the latest git, and I'm trying to dive into an issue I'm having. I am using mod_shout and in some instances am playing files that are hosted on web servers protected by https. This seemed to be working fine before I upgraded, but now I am getting the following issues, which is preventing the streaming of the files: 12-03-02 19:06:57.926919 [WARNING] mod_shout.c:468 CURL returned error:[77] problem with the SSL CA cert (path? access rights?) : error setting certificate verify locations: CAfile: /usr/local/freeswitch/share/curl/curl-ca-bundle.crt CApath: none I looked at the code and the git history in mod_shout.c where it is setting all the curl options, nothing seems to have changed there since I last updated FS (12/07), however, the curl-ca-bundle file is certainly not located at /usr/local/freeswitch/share/curl/curl-ca-bundle.crt and never has been as far as I know. I also looked at other mods that are using curl and where they are calling switch_curl_easy_setopt (mod_xml_curl, mod_httapi), and noticed that those mods are setting options that might be related to what I need? CURLOPT_SSLCERT I am confused as to what is causing the breakage, since mod_shout hasn't changed since I last updated, yet none of the ssl curl options are set in mod_shout..and I never had any problems with the mod_shout curl usage finding the certificate verify locations by default. Is there some other default that used to be set in the freeswitch configuration that I need to set manually? Any help is appreciated!!! Thanks! Matt

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mod_managed
by Gregor Nanger 04 Mar '12

04 Mar '12

I'm so close to anounce Freeswitch as the best voip software ever created :-) I am using mod_managed with c# and trying Test Visual studio project that is included. I can connect inbound and outbound and I can control call. But I cannot read header. If I use method getHeader("EVENT-NAME") , response is null. Whichever header I read, I get null. Serialize method prints all headers.. Any suggestion? I am using Freeswitch with Windows OS.

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ManagedESL Run Issues
by John Rose 04 Mar '12

04 Mar '12

Has anyone tried to run the latest ManagedEslTest.2010 and run into esl.dll issues such as: "An attempt was made to load a program with an incorrect format. (Exception from HRESULT: 0x8007000B)" at ESLPINVOKE.SWIGExceptionHelper.SWIGRegisterExceptionCallbacks_ESL(ExceptionD elegate applicationDelegate, ExceptionDelegate arithmeticDelegate, ExceptionDelegate divideByZeroDelegate, ExceptionDelegate indexOutOfRangeDelegate, ExceptionDelegate invalidCastDelegate, ExceptionDelegate invalidOperationDelegate, ExceptionDelegate ioDelegate, ExceptionDelegate nullReferenceDelegate, ExceptionDelegate outOfMemoryDelegate, ExceptionDelegate overflowDelegate, ExceptionDelegate systemExceptionDelegate) at ESLPINVOKE.SWIGExceptionHelper..cctor() in D:\FreeSWITCH\freeswitch\libs\esl\managed\ESLPINVOKE.cs:line 105 John

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how to alter the SDP information in freeswitch
by 逆水行舟 04 Mar '12

04 Mar '12

Hi, I'd like to know how to alter the SDP information in freeswitch ? Any help is appreciated! Thanks!

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Freeswitch + MSSQL
by Gregor Nanger 04 Mar '12

04 Mar '12

Hi! I am using Freeswitch with Windows. I hope I am not the only one. I also want to make mod_sofia to talk via ODBC with MSSQL. I can connect, but on call I get error, when sofia wants to write to mssql: [STATE: 42000 CODE 102 ERROR: [Microsoft][ODBC SQL Server Driver][SQL Server]Incorrect syntax near '|'. ] Is this know issue? Any suggestions? On MySQL it is working ok..

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For all
by 逆水行舟 04 Mar '12

04 Mar '12

Hi, I would like to use my own equipment for the endpoints transcoding not by freeswitch. So I want to modify the sdp. The media stream to point to the transcoding device. Is there anyway of modifying o= and c= in sdp protocol? (modify ip and port of rtp server in sdp) How to modify the sdp with in C code. Other methods to achieve it? Any idea? Thanks ! best regards!

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Freeswitch-users] spandsp error reading frame
by Jean-Marc Hyppolite 04 Mar '12

04 Mar '12

Hello, I am trying to use mod_spandsp to detect call progress. I haven t been successful so far. I am getting the following error messages. ================================================ 2012-03-03 23:55:08.398205 [DEBUG] mod_spandsp_dsp.c:379 (sofia/outbound/.@...) Starting tone detection for '1' 2012-03-03 23:55:08.398205 [INFO] mod_spandsp_dsp.c:411 (sofia/outbound/.@...) initializing tone detector 2012-03-03 23:55:08.398205 [DEBUG] switch_core_media_bug.c:462 Attaching BUG to sofia/outbound/.@... 2012-03-03 23:55:10.218175 [INFO] mod_spandsp_dsp.c:422 (sofia/outbound/.@...) error reading frame 2012-03-03 23:55:10.218175 [INFO] mod_spandsp_dsp.c:447 (sofia/outbound/.@...) destroying tone detector ================================================ Any help would be appreciated. Thanks Jean-Marc.

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TCAPI_User in Contact
by Andrew Melton 04 Mar '12

04 Mar '12

I know this has been asked before, but I can't seem to put my finger on the answer, any suggestions of where to look are appreciated. Freeswitch (Sofia) is rewriting the Conact header on calls originating on a trunk ('inbound' context), routing to another trunk ('outbound' context). None of the calls are from registered users or extensions, this is more of an SBC configuration (LCR). For example, incoming to Freeswitch: Contact: <sip:5122046107@74.37.201.198:5060;user=phone> Outbound from Freeswitch: Contact: <sip:TCAPI_User@84.19.135.13:5060;transport=udp;gw=trunk_2> I really don't want to modify the from userpary, nor do I want to include gw=trunk_2. I just can't figure out the right syntax or parameter to change it. Is it a parameter in the dialplan, or should I be using a regex on the outbound interface? Thanks.

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send message via gateway from mod_sms
by Seven Du 04 Mar '12

04 Mar '12

Hi, I'd like to know is there a way to send message to a gateway like bridge for sound? Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com)

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