July 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Which FS version to use?
by Miha 20 Jul '12

20 Jul '12

Hi, on my production server I have version: freeswitch@default> version FreeSWITCH Version 1.0.head (git-00de8e6 2011-11-01 17-27-13 -0600) which is in my case very stable, I can say perfect. As I have prepare other, more powerfull server for FS I have installed FS from latest git. With version 1.2 rc1 I was having problems with load and all users did not regisered (loosing connection). I do no know if this was FS problem but I do not have this problems on production server. Few day ago I did a new git pull. Now I have version 1.2 rc2 which enum do not work. I traced with wireshark but afer I put in fs_cli enum 123123123, Fs sends nothing. So on git.freeswitch.com are versions 1.0.6, 1.2 rc2 and 1.2 rc1. Which version to use? Is 1.0.6 most stable for now as 1.2 is rc? Where can I find version of FS which I am running on production server? Thanks for all your help! miha

10 12

ENUM do not work
by Miha 20 Jul '12

20 Jul '12

Hi, is there any problem with enum on new git? When I do nslookup in linux, I get result but if I do it from fs_cli (enum 1231231231) I can see with wireshark that FS do not send lookup. p.s.: I have same configuration on different older FS and works. Thanks! miha

5 8

Update On Donations For H.323
by Michael Collins 19 Jul '12

19 Jul '12

We have received several donations already - thank you very much! Just a reminder: please be sure to leave a note with your donation so that we know what your donation is for. Our target for getting Robert Jongbloed to Chicago from Australia is US $3000. We will let you know when we've met our goal and we'll give you status updates along the way. Thanks again for your generosity! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

1 0

mod_opus difficulties
by Shaunak Shah 19 Jul '12

19 Jul '12

I am trying to get mod_opus to work with freeswitch. I am able to load mod_opus alright but to be able to use OPUS there needs to be something related to OPUS in vars.xml. What do I add is vars.xml. Currently my vars.xml has following: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,PCMU"/> I want to test audio quality with OPUS so I need to put some string in vars.xml ? I tried "Opus-0.9.0" and "OPUS (BETA 0.9.0)" but none of them seem to work and I get a 488 - Not Accepted here while negotiating codecs. Thanks in advance. -- Shaunak Shah

1 0

Looking for dial plan examples for FS and SPA3102 router
by Todd Bailey 19 Jul '12

19 Jul '12

Hi All, I'm having issues getting a dial plan to work on FS and a Cisco SPA 3102 router. when I dial 0, 1 or 9 plus a 10 number, I get to the router's dial tone but I have to reenter the number I want to connect to. The expected action is to only need to enter the number to dial one Can some one provide dial plan and/or other config file example on how to resolve this issue? here is what I have so far: /usr/local/freeswitch/conf/dialplan/default.xml  <extension name="external"> <condition field="destination_number" expression="^(1{0,1,9}\d{10})$"> <action application="set" data="effective_caller_id_number=12223334444"/>     <action application="bridge" data="sofia/internal/$1@192.168.1.5:5061" /> </condition> </extension> /usr/local/freeswitch/conf/dialplan/default/00_spa3102.xml <include> <extension name="To PSTN"> <condition field="destination_number" expression="(.*)"> <action application="bridge" data="sofia/internal/${destination_number}@192.168.1.5:5061" /> </condition> </extension> </include>

2 2

Lua outbound executing another lua script
by Alpha Niner 19 Jul '12

19 Jul '12

Hi, Firstly apologies for the newbie question. I've tried searching the various documentation and trying a few different things but to no avail, so I've come here to ask for help. I think my problem relates to sessions, but I'm not totally sure. What I'm trying to do is: 1. A lua script is run by someone calling into FS (the initiator) 2. They select an option, which originates and makes an outbound call to another phone number 3. The 3rd party (hopefully) answers their phone and receives a prompt that they're going to be connected (the receiver) 4. Both parties join a conference and talk to each other (a conference as there may be other parties in future) To do this I'm creating a session and using execute_on_answer to run the second script and provide the prompt to the receiving party. However the receiving party never hears a prompt, comes into the conference, immediately drops out and hangs up. Here are the scripts, they're pretty simple; The initiator: session:answer() session:streamFile("WaitForOutdial.wav") new_session = freeswitch.Session("[execute_on_answer=lua /usr/local/freeswitch/scripts/receivewithwhisper.lua]sofia/external/12345@mydest"); session:execute("conference", "CONFTEST@default+flags{endconf, moderator}") session:hangup(); The above works fine and the outbound call is placed. The script below is then executed but the receiving phone rings but nothing is heard, the conference is joined/exited and it terminates again; The receiver (receivewithwhisper.lua): session:answer() session:streamFile("ConnectingToCaller.wav") session:execute("conference", "CONFTEST@default+flags{endconf, moderator}") session:hangup(); The receiving scripts runs when the receiver answers their phone, and executes (quickly) and terminates. I was expecting it to drop into the conference and I'd be able to talk to the other person. It does join the conference, but then exits right away and falls through to the hangup section. As I say, I'm not exactly sure if this is the best way (maybe there's another - suggestions are welcome) and not exactly sure what the problem is although I suspect it's something to do with the sessions? Any help, advice, corrections or examples would be appreciated. It's all done in lua and I've tried to avoid session.originate as I understand this is deprecated. Thanks Adam

2 1

Skype (+connect) Freeswitch/SIPfoundry
by Todd Bailey 19 Jul '12

19 Jul '12

I'll be installing Fedora F17 over the next few days and hope to take my successes and er well, failures on FreeSwitch (FS) and move them into the new install. In looking at Skype it states to use SIP approved pbx's and SIPFoundry (SF) is one of them. In looking at SF I see a lot of FS rpms etc. Any one care to help minimize my confusion on what the differences are and what might be the better product to install? I'd like to use skype for outbound long distance calls, an added plus would be to have bi-directional calls, but outbound calling is key. I'm still using a analog line for local bidirectional calling but it has it's own set of issues. thanks

2 1

Re. trigger custom freeswitch application to notify users in a conference
by A.S. Shaja 19 Jul '12

19 Jul '12

Hi, Is it possible to create an application in FreeSWITCH that when a user presses a series of keys, it could trigger a custom FreeSWITCH application to notify all moderators via an audio message? Can you please help me on how to achieve this? Regards, Shaja. -- "TajMahal would not have been so beautiful if shajahan had asked for three quotations and had chosen the least one!!!"

5 5

How You Can Help With H.323 In FreeSWITCH
by Michael Collins 19 Jul '12

19 Jul '12

Greetings! We would like to let you know about an opportunity we all have to assist with H.323 support in FreeSWITCH. We are working on getting Robert Jongbloed, author of mod_opal and all around cool guy, to ClueCon 2012. Robert does a lot of work with H.323 and his presence with us at ClueCon would go a long way in helping us get H.323 fully implemented in FreeSWITCH. The challenge is that Robert lives in Australia! To that end we are taking up a collection to help defray the substantial travel costs. Here's what you can do: visit FreeSWITCH.org and click the Donate button. In your Paypal donation, click "Add special instructions to the seller" and make a note that this donation is to assist with getting Robert to ClueCon. We'll take care of the rest. Thanks again for all of your wonderful support! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

1 0

Rooms Added
by Brian West 19 Jul '12

19 Jul '12

I have just sent the paperwork in to get more rooms on the dates everyone was having problems with. Please give them an hour or so to get them into the system. -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410

1 0