January 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

call getting hangup after git update
by ovvenkat 12 Feb '11

12 Feb '11

Hi, Today, I have updated freeSwitch to latest git. After that, When I am trying to do outbound call, Call is getting hangup. When I check the fs_cli logs, its showing that, *mod_dptools.c:2610 Originate Failed. Cause: NO_ROUTE_DESTINATION* Here is the logs for the same http://pastebin.freeswitch.org/14951 I have added STD code , before the number, Still no luck. I am getting error like *Originate Failed. Cause: NORMAL_UNSPECIFIED* Here is the log after adding STD code http://pastebin.freeswitch.org/14952 Regards, Venkat. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV.

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Yet another DTMF question
by Travis Stutsman 11 Feb '11

11 Feb '11

I've looked through a lot of threads regarding DTMF troubles, but none of them seem to relate to me. I've just pulled a fresh copy of the freeswitch source and compiled it and I cannot get my digits recognized - neither in-band nor out-of-band. I haven't messed with INFO yet as my phones don't support it. I've done packet captures on my local interface and on the server's interface and I can see the RTP rfc2833 digits showing up, but freeswitch completely ignores them. I feel like I must be overlooking something horribly obvious. How can this be? Any hints guys? Thanks in advance!

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Language files - French, German, Spanish, etc.
by Michael Collins 10 Feb '11

10 Feb '11

Hello all! I have heard about several of our intrepid community members creating sound sets in various languages. If you have created a sound set in any language other than US English please contact me off list. I want to help get this sound prompts organized so that the global community can use them. Thanks! -Michael

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Polycom and registering with domains (user@domain.tld@domain.tld)
by Gabriel Gunderson 10 Feb '11

10 Feb '11

All, I have a multi-tenant FreeSWITCH server up and running. We have all the proper SVR records setup for each domain and everything works great. Most SIP clients Just Work (tm), however, the new Polycom phones I have don't seem to auth properly with domains. I've tried every combination of settings (via the web config, I haven't setup a TFTP server yet). The issue is that auth has the domain twice, like so: user@domain.tld@domain.tld. It seems like the Internet has threads of discussion where others express frustration with this, but I can't seem to find a documented solution. It's maddening because every other User Agent Client works swimmingly and no matter what I've tried, I can't get the right combination of settings on the Polycom. I love the phones, but this is getting old :) Anyway, any pointers for me? Thanks, Gabe sofia.conf: ************************************************************* [SNIP] <domains> <domain name="all" alias="true" parse="false" /> </domains> <settings> <param name="log-auth-failures" value="true" /> <param name="forward-unsolicited-mwi-notify" value="false" /> <param name="context" value="public" /> <param name="sip-port" value="$${internal_sip_port}" /> <param name="dialplan" value="XML" /> <param name="inbound-codec-prefs" value="$${global_codec_prefs}" /> <param name="outbound-codec-prefs" value="$${global_codec_prefs}" /> <param name="rtp-ip" value="$${internal_ip}" /> <param name="sip-ip" value="$${internal_ip}" /> <param name="hold-music" value="$${hold_music}" /> <param name="record-path" value="$${recordings_dir}" /> <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav" /> <param name="auth-calls" value="true" /> <param name="ext-rtp-ip" value="auto-nat" /> <param name="ext-sip-ip" value="auto-nat" /> </settings> [/SNIP]

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Error Loading mod_java.so
by srinivasula reddy 08 Feb '11

08 Feb '11

Hi All, Currently i am using JDK1.06.0_23 and jre also same version for using mod_java module in freeswith, i can able to generate mod_java.so and freeswitch.jar. and building and installation had succussful without any proble. when i am trying to run the freeswitch, i am gettig the following error. 2010-12-13 16:43:56.988097 [NOTICE] modjava.c:342 Java Framework Loading... 2010-12-13 16:43:57.042698 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** 2010-12-13 16:43:57.043603 [CONSOLE] switch_loadable_module.c:946 Successfully Loaded [mod_lua] and please find the following java.conf.xml file. <configuration name="java.conf" description="Java Plug-Ins">   <javavm path="/usr/lib/jvm/java-1.6.0-openjdk-1.6.0.0.x86_64/jre/lib/amd64/server/libjvm.so"/>  <options>  <option value="-Djava.class.path=/usr/local/freeswitch/scripts/freeswitch.jar "/>   </options> <startup class="net/cog/fs/system/Control" method="startup" arg="start up arg"/> <shutdown class="net/cog/fs/system/Control" method="shutdown" arg="shutdown arg"/> </configuration> Any Idea why i am getting the error? Thanks Srinivasula Reddy K

4 4

Doxygen help
by Yossi Neiman 08 Feb '11

08 Feb '11

I am not sure how many folks make use of the Doxygen documentation. I know I do all the time. I find that it provides a nice conceptual view of FreeSWITCH's API, which can make it very easy to find the functions and data types needed for writing modules for FreeSWITCH. A person by the name of Mohammad Shahzad (apologies if I misspelled it) started to do a lot of work on revamping the Doxygen configuration for FreeSWITCH a few months back. The problem with the work that he did was that he used Doxygen 1.6.x specific configuration parameters that are not understood by the Doxygen 1.4.x tree (which is what we currently use in FreeSWITCH) or even the 1.5.x tree which is what is in Slackware. In my discussions with Michael Collins we have decided that it might not hurt to go ahead and move the Doxygen version forward to match the 1.6.x tree. We are looking for somebody who has interest in working with the Doxygen configuration to continue the work that Mohammad had started and to help the project out. If you are such a person, please contact me off-list and I can provide further information. Thanks! Yossi Neiman

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blocking 183 w/o sdp
by Sam 07 Feb '11

07 Feb '11

Hi, how can i ignore 183 without sdp, what happens is the provider sends 183 without sdp and by applying "<action application="set" data="sip_ignore_183nosdp=true"/>" the FS sends 180 to the leg a. Here i want to block the 183 with SDP just like router as b2bua and send nothing to leg a, and when actual 183 with sdp comes it should send . Its because, providers are sending false signaling by sending 183 without sdp,and it hampers while @ production, Although by cisco sbc i have done this but i want to do it by FS, Take a scenario, when call is send 183 without sdp for 10 secs and then followed by 183 with sdp ( actual signal), but when some one dials invalid number it rings for 10 secs and then gives SIP cause 404, which is bad from the providers. So this is the reason i want to block it. Most of the providers do this, the way out is blocking. I have got an advice from Tihomir to do "execute_on_ring and parse your 180 / 183 messages in search of SDP, once you get 183 without SDP do not send it back to leg a and send signal only when you got 183 with sdp or 180 " And this could be valid call flow. This happens in many cases where the provider is having nextone as a sbc and that too tier 1 ! Regards Sam

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Freeswitch and mysql
by Antonio 06 Feb '11

06 Feb '11

Hello, I asked and more I searched, I found nothing, how to integrate FreeSwitch with mysql. Some time ago I found Asterisk-realtime and wanted to know if there is something similar in FreeSwitch. Is to avoid double configurations, for example if there is a conference room that can be accessed from multiple servers, etc. I'm new in FreeSwitch and I have much knowledge ;-) Does anyone have any url or book on how to do? Lua?. Thanks. Greetings

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deflect
by Sam 04 Feb '11

04 Feb '11

Hi, When call comes on 1 server and plays an application and after execution of the application the call is bridge to the other server ,but here after bridging the call should refer/deflect to other server, how this can be done ? Here just using the deflect variable is not recommended as there is proxy in between, so once the call is bridge the next step would be deflect the leg totally to another server via proxy. Regards Sam

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Portaudio Improvements
by Mitch Capper 04 Feb '11

04 Feb '11

I have submitted a ticket with a patch for portaudio and my improvements. This includes the improvements I had discussed previously during my call for input on the mailing list and is available at: http://jira.freeswitch.org/browse/FS-3006 It was only tested in windows, however most of the changes should not effect default behavior (but please test if you can). To test you will want to enable some things in the portaudio config including: <param name="live-stream-switch" value="true" /> <param name="no-auto-resume-call" value="true" /> <param name="no-ring-during-call" value="true" /> you will then be able to use the new features fully. Many of these changes were made to add better support for softphone's using freeswitch, I will be releasing my embedded freeswitch phone (FSClient) for the conference call in a week and a half it is .net/c# but WPF so sadly windows only currently. jlink and drk are helping so we should end up with a nice installer also for it. If you would like to test it please let me know. ~Mitch

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