July 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Multiple gateways: Priority the first bridge with prefix
by Edmar Cruz 27 Jul '09

27 Jul '09

Hi FS Users, I just want to try multiple gateways. It works actually like this... <action application= "bridge" data="sofia/sip/51(a)222.333.444.555 | sofia/sip/63(a)111.222.333.3333"/> But I test call like 5133333 at 222.333.444.555, it also calls the second bridge 111.222.333.333. It there any way to determine which prefix will call to a bridge specified. E.g. for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not at the second bridge and vice versa. Please help.. -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-p… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Sofia SIP Subscription For External Events
by afshin afzali 26 Jul '09

26 Jul '09

Hi Guys, I'm going to use OpenSER as SIP Platform (Registrar, Proxy, Presence) and FreeSWITCH in my application. So I need to subscribe the FreeSWITCH for presence information of users who involve in the application. After some looking of Sofia.conf.xml , it seems there is not support to doing so, Is it right ? appreciate all comments, -- afshin

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A possible bug in FS causing Linux Kernel crash
by Muhammad Shahzad 26 Jul '09

26 Jul '09

Hi, I am having random Linux Kernel crash problems while running FreeSWITCH as Skype to/from SIP gateway on one of our production servers. This machine is running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE with FS svn revision number 13754. At time of Kernel crash i could find following crash messages which point to some source code file in FS source tree. --------------------- Kernel Begin ------------------------ 3 Time(s): ======================= 3 Time(s): [<c0404eff>] syscall_call+0x7/0xb 3 Time(s): [<c043ed22>] sys_delete_module+0x192/0x1b8 3 Time(s): [<c0449011>] audit_syscall_entry+0x14b/0x17d 3 Time(s): [<c049f4fe>] remove_proc_entry+0x139/0x18c 3 Time(s): [<f8d96281>] alsa_sound_exit+0xa/0x30 [snd] 3 Time(s): [<f8d96304>] snd_info_done+0x46/0x49 [snd] 3 Time(s): BUG: warning at fs/proc/generic.c:732/remove_proc_entry() (Not tainted) 1 Time(s): snd-malloc: Memory leak? pages not freed = 1 ---------------------- Kernel End ------------------------- While the problem seems to arise from ALSA kernel module but it blames FS file fs/proc/generic.c:732 for this. The only FS module that is using ALSA is mod_skypiax but as far as i remember that module is using FS internal routines to allocate and de-allocate sound driver services for Skype client. Please suggest a solution. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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Mod_Limit - Limiting different destinations on same IP
by Saeed Ahmad 26 Jul '09

26 Jul '09

Dear All, Is it possible to apply different limits on a same IP by different destinations. Example: IP: 1.2.3.4 Destination: Germany Mobile (491) => max-channels=10 Destination: Germany (49) => max-channel=20 Sometimes the supplier provides limited capacity on different destinations; so in this case its necessary to apply the limit so that after limit exceeds the call can go to next endpoint. Thanks Saeed.

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Regarding IVR
by Thangappan.M 25 Jul '09

25 Jul '09

I am learning freeswitch for implementing IVR in this software. In our organization we are using Perl language.So I decided to implement a IVR in Perl on Freeswitch. What are the steps I need to do here. I am also new to IVR.Can you little bit explain about IVR (not basic) how to define a menu like that.. -- Regards, Thangappan.M

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FS and Nokia E71
by bakko 25 Jul '09

25 Jul '09

Hi all, i have some problems tu make calls from a nokia E71 connect to FS. This is the scenario: Nokia E71 -> NAT - Internet - FS (public IP) I'm using the 5059 UDP port in FS. A can receive calls from others phones to nokia E71 but i can't make calls from nokia E71. If a change the UDP port to 5060 in vars.xml all work fine. I think the problem is the VoIP cllient in the nokia E71 but i dont´t know how resolve the problem. Thank's in advance. BR

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Application to Record Calls - Out of Band
by Matthew Fong 24 Jul '09

24 Jul '09

Hi, I'm trying to build an application that provides statistics of calls and call recording. Someone told me this could be done out of band with a SPAN (?) port that would replicate SIP and media packets to a separate NIC without having to actually pass the real-calls thru FreeSWITCH. It was explained that this SPAN port would in the SBC would replicate data received. If this is done, is there a way I can utilize FreeSWITCH to interpret these packets without actually having any control of the calls? If so how? Sorry, I'm new to telco, so hopefully this post makes sense to someone. --matt

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freeswitch-1.0.4pre10 ?
by Dome Charoenyost 24 Jul '09

24 Jul '09

I found freeswitch-1.0.4pre10 in http://files.freeswitch.org/freeswitch-1.0.4pre10.tar.bz2 But no news about this version Dome C.

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IAX Transfer support
by Gu Sh 24 Jul '09

24 Jul '09

I have been using freeswitch for over a year and I love all of the features, extensibility etc. Recently one of the clients wanted to use a IAX client and call from the IAX client works fine but there was one feature requested by my client that did not work. The feature is the "IAX Transfer" and I see the Transfer message come through by turning up debugging in the iax.conf file but freeswitch does not do anything with it. What is the current status of IAX support on freeswitch?

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setInputCallback not working with Javascript?
by mszlazak＠aol.com 24 Jul '09

24 Jul '09

I'm getting in build 12653M: [ERR] notify.js:130 mod_spidermonkey()? TypeError: session.setInputCallback is not a function The wiki says this function should work in Javascript. http://wiki.freeswitch.org/wiki/CoreSession_Constructor#session:setInputCal… Also, has there been changes to session.collectInput with type="event"? I get dtmf type events with my callback function but can't seem to get type="event" with speech events. Mark.

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