July 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Gateway setup
by Ilan Perez 31 Jul '08

31 Jul '08

Dear All, I am not sure who is an expert in PBX/ FS and who isnt but I dont know that I certainly am not Jin light of that I really need some help getting things moving with this system. Just before I start, I want to say that the wiki isnt the be-all and end all in terms of answers. I have read the wiki many times and steps for setup are left out. So people like myself struggle and fiddle without actually understanding what they are doing. Here is what I have and what I need help with I have 3 computers. 1. Server @ 192.168.43.111 2. Softphone @ 192.168.43.5 ext 1207 3. Softphone @ 192.168.43.6 ext 1208 4. And a Linksys SPA-3120 @ 192.168.43.3 5. My router/ default gateway is at 192.168.43.1 I have setup the router and the server fine. The two softphones have x-lite installed and are registering correctly and I can make call between them. I have also configure the public.xml file in the dialplan directory to accept my custom extension of 1208, 1207 <extension name="public_extensions"> <condition field="destination_number" expression="^([1-9][0-9][0-9][0-9])$"> <action application="transfer" data="$1 XML default"/> </condition> </extension> The next step for my project is to get setup incoming and outgoing calls for the Linksys SPA 3120 Here is what I have done To the directory I have created a custom extension 1207, 1208 and sippura Sippura looks like this <include> <user id="1200"> <gateway name="sippura"> <param name="username" value="1200"/> <param name="password" value="1200"/> <param name="proxy" value="192.168.43.3"/> </gateway> </user> </include> To the default.xml file in the dialplan directory I have added this <extension name="Outbound - Linksys"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="set" data="effective_caller_id_number=12223334444"/>   <action application="bridge" data="sofia/gateway/sippura/$1(a)192.168.43.3"/> </condition> </extension> To the public.xml file in the dialplan directory I have added this <extension name="sippura"> <condition field="destination_number" expression="^(0283472006)$"> <action application="transfer" data="$1 XML default"/> </condition> </extension> To the sip_profiles directory I have added a file call sippura.xml This file contains <include> <gateway name="sippura"> <param name="username" value="1200"/> <param name="password" value="1200"/> <param name="extension" value="sippura"/> <param name="proxy" value="192.168.43.3"/> </gateway> </include> è I know I have setup my gateway correctly as I know how to do thisfrom previous attempts with other PBX software, but just for your information the Username, password and User ID is all 1200 What I need is someone to comment on my steps by sayingdont need this or you left out line xand please add the line in. I promise once this is sorted I will submit a compelte walkthrough to the wiki for all hopeless newbies like myself J Looking forward to your help Ilan Perez Sydney Australia

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Turn on call recording
by Ilan Perez 31 Jul '08

31 Jul '08

I was wondering if someone can tell me the command to turn on call recording. Ilan Perez Webmaster

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Some little help with regexp and extensions
by Diego Viola 31 Jul '08

31 Jul '08

Hi everyone, I have this in my default dialplan for dialing out: <extension name="outbound"> <condition field="destination_number" expression="^9(\d+)$"> <action application="bridge" data="sofia/gateway/teliax/$1"/> </condition> </extension> It works fairly well... but when I try to dial other extensions that start with "9", like 9996 (echo extension) I get this: http://pastebin.freeswitch.org/5026 and in my softphone I get "service or option unavailable". If I remove the "outbound" extension works. I know it might be something simple... but I'm not very familiar with regular expressions yet, so if you can help me with this I will be really thankful. Thanks everyone... FreeSWITCH rocks =D Diego

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Calling card billing for FS
by Alex Kinch 30 Jul '08

30 Jul '08

Hi gang, Are there any calling card billing packages out there that are compatible with Freeswitch? Obviously I've heard of a2billing, but a recent forum thread suggests that it's going to be a while until it's ported to work with FS (if ever). Thanks, Alex

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Having FreeSwitch using UDP only
by afro.systems＠gmail.com 30 Jul '08

30 Jul '08

Hi all, A newbie question, Is it possible to configure FreeSwitch to be using UDP only even fore the signaling? -- Tzury Bar Yochay

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ANNOUNCEMENT: wiki documentation conf call today, 1PM CST
by Michael Collins 30 Jul '08

30 Jul '08

Here's the agenda: http://wiki.freeswitch.org/wiki/Wiki_meet_2008_07_30 All are welcome. -MC

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Trunk to 3rd party PBX
by Simon Shaw 30 Jul '08

30 Jul '08

Is it possible to configure FS such that it will only handle SIP Register messages internally and will pass all other messages via a SIP trunk to an existing PBX. This would be very useful to gradually move existing IP phones over to FS. The stage after that would be to define that certain phones have all their call processing done on the FS box. Thanks, Simon

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Re: [Freeswitch-users] How to compile and load mod_cdr
by Shehzad Pankhawala 30 Jul '08

30 Jul '08

Hi, thanks mike, I want to get CDR into MySql. organizing CSV and XML files for each user is some how combursome.. thanks Shehzad

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How to compile and load mod_cdr
by Shehzad Pankhawala 30 Jul '08

30 Jul '08

Hi every body, I have downloaded mod_cdr from svn as shown in http://wiki.freeswitch.org/wiki/Mod_cdr I tried to use 'automake' to make the module but it gives the following error: automake: `configure.ac' or `configure.in' is required Anybody have solution then please reply.. thanks. Shehzad Pankhawala

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Gafachi Again
by Klaus Teller 30 Jul '08

30 Jul '08

Hi, i have a javscript script that was working with Freeswitch 1.0. The only issue i had is that outgoing dialing via Gafachi was not transmitting the caller ID. Then I upgraded to the current trunk version (FreeSwitch Version 1.0.trunk (9205)). But now I'm not able to place calls anymore. The error message i get is the following: ===== 2008-07-29 23:21:11 [ERR] switch_core_session.c:249 switch_core_session_outgoing_channel() Could not locate channel type 2008-07-29 23:21:11 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [] cause: [CHAN_NOT_IMPLEMENTED] 2008-07-29 23:21:11 [WARNING] mod_spidermonkey.c:2923 session_originate() Cannot Create Outgoing Channel! [/sofia/gateway/sip.gafachi.com/14155465584] === The exact same script is able to place outbound calls via Les.net. And here is what i do in the script: ====== new_session2 = new Session(); new_session2.setCallerData("caller_number_number", caller_number); new_session2.originate(null, '/sofia/gateway/sip.gafachi.com/1'+callee_number, 15); new_session2.setAutoHangup(true); ======= Any idea? Thanks, Klaus. -- GMX Kostenlose Spiele: Einfach online spielen und Spaß haben mit Pastry Passion! http://games.entertainment.gmx.net/de/entertainment/games/free/puzzle/61691…

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