September 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

OpenSIPs + Freeswitch + Registrations
by Chris B. Ware 16 Sep '13

16 Sep '13

Hi, I'm a bit confused (as usual) on call flow between FS boxes behind an OpenSIPs balancer. All servers are on the same LAN, so the phones. Registrations are stored on common DB. PhoneA sends a register and Opensips balancer forward it to FS1. PhoneB sends a register and Opensips balancer forward it to FS2. PhoneA send an INVITE for PhoneB and OpenSIPs forward it to FS1. FS1 answers "Originate Failed. Cause: USER_NOT_REGISTERED" An IRC friend says to use path (RFC 3327). Is there a config solution or should I follow the his advice? Thanks, Chris B.

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Issue with JSSIP + Freeswitch
by Shahrzad A. 16 Sep '13

16 Sep '13

Hi everyone I'm having latest version of Freeswitch installed on Ubuntu 12.04.2 LTS with the latest version of Openssl ('OpenSSL 1.0.1e 11 Feb 2013') I'm using the default configuration and just uncommentated the ' <param name="ws-binding" value=":5066"/> ' in internal.xml in order to have the support for webrtc. As the client I'm having JSSIP, the latest version with the adjustment to have (DtlsSrtpKeyAgreement:true). Both my clients and server located in the same network without having any firewalls therefore I'm not using any STUN server. I can make call but there is no sound! and after setting debig_level 10 and dial from extension 1003 to 5000 I'm having the following Alert in console and no sound: [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!) its the same case if I dial from extension 1002 to 1005! (Both are JSSIP clients) If I call from extension 1000 which is a SIP client set on my iPhone and call 1003 I'm having the following alerts but voice on both JSSIP client and SIP client: 2013-08-05 14:17:05.222446 [ALERT] switch_rtp.c:4563 sofia/internal/ 1000@10.0.14.16:5060 timer while HOT 2013-08-05 14:17:05.242449 [ALERT] switch_rtp.c:4546 sofia/internal/ 1000@10.0.14.16:5060 Hot Hit 1 And from extension 1003 (JSSIP) to extension 1000 ( SIP on Iphone) No voice and I'm getting the following alerts: 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4546 sofia/internal/ sip:1000@10.0.14.182:5065 Hot Hit 4 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4563 sofia/internal/ sip:1000@10.0.14.182:5065 timer while HOT 2013-08-05 14:19:57.132441 [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!) If I use bypass media or proxy media I will have voice on both JSSIP clients but cant ring any SIP clients and again no voice if I call 5000! Wonder to know if there is any special setting required on FreeSwitch or its and issue from JSSIP? Thanks in advanced! Sherry

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Commit message usefullness
by Florian Lohoff 15 Sep '13

15 Sep '13

Hi, just a quick note - For a customer of us i was looking throught the commit messages of FreeSwitch because we suspected a bug in freeswitch. I immediatly stumbled upon commit messages like "oops", "doh" or "bump". Basically most of the commit logs were useless. One could track messages like "FS-5731 --resolve" by looking at the Jira but that would be very time consuming to look up every single bug. Finally i went for grepping through commits touching certain files and using "git blame" to find commits. This message is just for highlighting that commit messages are important and IMHO should contain more informations than a "Bug ID" which may be transitional - one day FreeSwitch might use Jira+1 and all bugs are gone and those references will be just as useless as a random number. Flo -- Florian Lohoff f(a)zz.de

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How to play G729 recording ?
by Yuri Ritvin 15 Sep '13

15 Sep '13

Hi, I'm trying to work with commercial FS G729 module and want to make recordings into G729 format, but looks like a playback of such files is not as straight forward as expected. If there are FS users who have already achieved this functionality - what is a best/good practice to playback such FS created G729 recordings ? Thank you in advance. Yuri

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mod_managed trouble
by Ed Graves 15 Sep '13

15 Sep '13

Hi. I've been working for days now trying to get a simple inbound call to work using vb.net on Windows using FreeSWITCH version 1.2.12 and Visual Studio 2012. I'm tired and frustrated, but I think I'm almost there. Please help. I'm using the sample script below, but when I copy the .exe into place, FreeSWITCH tries to execute it immediately. When this happens, I get the output below, with lots of "session is not initialized" errors, which makes sense, because there is no session. But then when I place an inbound call using my soft-phone, the script runs and I still get the "session not initialized" errors. So, I don't understand why A) FreeSWITCH tries to run my app immediately when I copy it into place when I don't ask it to, and B) why I get these "session not initialized" errors when it runs in response to my inbound softphone call. ********** My Sample App ********** Module inbound Sub Main() Dim fs As New FreeSWITCH.Native.ManagedSession() fs.Answer() fs.StreamFile("C:\Genesis\freeswitch\freeswitch1.2.12\Debug\sounds\en\us\callie\ivr\8000\ivr-congratulations_you_pressed_star.wav", 0) FreeSWITCH.Log.WriteLine(FreeSWITCH.LogLevel.Error, "I'm working!!!") fs.StreamFile("C:\Genesis\freeswitch\freeswitch-1.2.12\Debug\sounds\en\us\callie\ivr\8000\ivr-congratulations_you_pressed_star.wav", 0) fs.Hangup("end") End Sub End Module ******** My output ******** 2013-09-14 01:55:08.541500 [INFO] mod_dialplan_xml.c:558 Processing egg <egg>->12345 in context public 2013-09-14 01:55:08.541500 [ERR] switch_cpp.cpp:632 session is not initalized 2013-09-14 01:55:08.541500 [ERR] switch_cpp.cpp:951 session is not initalized 2013-09-14 01:55:08.541500 [ERR] switch_cpp.cpp:1287 I'm working!!! 2013-09-14 01:55:08.541500 [ERR] switch_cpp.cpp:951 session is not initalized 2013-09-14 01:55:08.541500 [ERR] switch_cpp.cpp:668 session is not initalized Again, I'd really appreciate any help. Thanks in adance, Ed

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wiki cookies problem
by Karl Schmidt 15 Sep '13

15 Sep '13

Says: > > Login error > FreeSWITCH Wiki uses cookies to log in users. You have cookies disabled. Please enable them and try again. Except this is not true - tested in two browsers... -------------------------------------------------------------------------------- Karl Schmidt EMail Karl(a)xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 Opportunity is missed by most people because it is dressed in overalls and looks like work. Thomas Alva Edison --------------------------------------------------------------------------------

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sip simple chat forward
by pankajanand18＠gmail.com 14 Sep '13

14 Sep '13

Hi, Is it possible to forward/receive the chat messages to other sip servers using gateways ? Thanks, Pankaj

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answer the call after park_timeout
by Jayesh Nambiar 14 Sep '13

14 Sep '13

Hello, I have a following scenario: 1) A call comes into freeswitch and gets parked with a park timeout of 25 seconds. 2) It is expected that another call will come and intercept into this UUID to start a conversation. 3) But if that doesn't happen, I need to answer this call from freeswitch and take into somebody's voicemail box. Is this possible?? Generally freeswitch just hangs up after the park timeout. I also tried doing this: 1) A call comes in, I call ring_ready application and sleep for 25 seconds. 2) After this I answer the call from freeswitch and take to voicemail. 3) But here when another call intercepts this UUID, the call gets established but the media from A-Leg starts only after 25 seconds since the channel on A-Leg is asked to sleep. So for this to work, I need some application to wake up the channel from sleep forcefully. I don't see how this can be done either. I would really appreciate if someone has done anything similar or point me to some relevant direction to start looking at. Thanks, --- Jayesh

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Updating TLS
by privs 14 Sep '13

14 Sep '13

Hi, Given the recent revelations regarding the state of the net's (in)security, and as a longtime user of TLS with SRTP in FS, I think it would be prudent to update the TLS version and ciphers that FS can negotiate. Currently FS only supports TLS 1.0 (correct me if I'm wrong), which has some worrying security issues <http://www.wired.co.uk/news/archive/2013-02/05/weakness-in-tsl-protocol> . Is it possible to upgrade to TLS 1.1 (or even better, 1.2)? I'd like to use AES in the 256 key length instead of the standard 128. Anyone know if this is on the agenda and/or where I can have a closer look at the cipher order presented by FS? Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Updating-TLS-tp7594774.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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when originate fails
by Valter Nogueira 13 Sep '13

13 Sep '13

I am testing originate command. *originate user/1000 extension xml context* When phone answer it jumps to the extension xml context But if phone don't answer or is busy how do I catch this? Thanks, Valter * *

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