May 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

recording to ogg vorbis
by Kees Varekamp 30 May '10

30 May '10

Hello, Can I record to ogg vorbis? I've read that anything other than wav is not recommended, but the good thing about ogg is that it is the only format that can be played in all the html5 compliant browsers. If it can't be done, is there any way to build a conversion hook into the record command? (convert after finished recording) Thanks, Kees

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nat problem!
by babak yakhchali 30 May '10

30 May '10

Hi all I googled the problem(RECOVERY_ON_TIMER_EXPIRE) and it seems it is related to NAT, but how can I fix it? I've changed: <param name="enable-timer" value="false"/> but it is not working thanx

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how to get session of bridged Bleg
by Frank ＠ Impact 29 May '10

29 May '10

FreeSWITCH Version 1.0.head (git-) from 5/28/10 I have a lua script called from the dialplan that bridges the call with session:execute("bridge","{route=sofia/gateway/US/15555551212} sofia/gateway/US/15555551212") after the call ends, I am trying to get some variables out of the Bleg session, the bridged leg. But I do not have a session handle to get the variables. I know what the uuid is from global variable_bridge_uuid. But how can I get the session of the B-leg so that I can do a getVariable()? The variables I want are not available in any of the global variables I can see with info.

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4 WEEKS LEFT! FreeSWITCH Official Training Course - June 28th-30th in San Francisco, CA
by Darren Schreiber 29 May '10

29 May '10

Hi folks, In case you haven't heard, we're pleased to offer the first ever Official FreeSWITCH Training course, held June 28th-30th in San Francisco, CA. The class has been filling up (and airline tickets will go up, too) so if you are thinking of attending we *strongly* encourage you to sign-up now. The training is different then a basic PBX training - we'll be diving a bit deeper into cool features within FreeSWITCH. Since we're designing the course from scratch we can also customize it so you learn what you want to learn. So far, topics requested have focused more along the lines of specific modules and carrier-style switching services, so we're going to try and focus on those topics. The training is also hands-on. You'll be working with FreeSWITCH on individual workstations during the course - doing labs, etc. We'll try and keep it as interactive as we can. And of course, if you like saving money... We have two deals going right now: 1. If you send two or more people we are now offering a good discount. It will vary based on the number of people you send. Please contact me directly if you are interested in this option. 2. If you sign up for the Training + ClueCon option you save ~$300! Flood yourself with FreeSWITCH knowledge and contacts by attending both events. This is a really good deal! For more information, please visit the website at http://www.voipkb.com/ or drop me a note. Thanks much and I look forward to seeing you at the training! Sincerely, Darren Schreiber

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streamFile() problem
by babak yakhchali 29 May '10

29 May '10

Hi using session.streamfile() , caller hears the playback from the sec 5 or 6 not from the start. it seems the file was playing but the media is not sent to caller till sec 5 6

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Register On-Line For ClueCon!
by Michael Collins 29 May '10

29 May '10

Hey there! We just wanted to let everyone know that our Web site has been updated: http://www.cluecon.com You may now click the link to get registered. Remember that you need to create a user for the ClueCon Web site or use the same username/password that you used last year. Feel free to contact us if you have any concerns or questions. Looking forward to Chicago! The ClueCon Team http://www.cluecon.com 877.742.CLUE

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Roadwarrier setup on external profile.
by Joakim Gissberg 28 May '10

28 May '10

Hi. I have set up my freeswitch server so that the default profile binds to my internal interface port 5060, and the external profile is bind to my external interface on the same port. What I want to accomplis is that each incomming call also rings the "roadwarrior" connected sip-clients, ie the ones authenticated to the external profile. I have created a ringgroup, and each incoming call is ringed in the public context (dialplan/public/00_XXX_incoming.xml) with > <action application="transfer" data="1020 XML default"/> and every user is in that group. Works if I authenticate with the internal profile. I have uncomment the transfer-authentificated-sip-clients-to-default-context part in dialplan/public.xml and I can call and use those extensions from the external profile (for example call a local extensions authenticated against the default internal profile from the roadwarrior). The problem firstly I guess is that an internal client can't call the externaly connected client, the call gets directly to voicemail with error "[ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]" The configuration is pretty much default other than the changes explained above and added outgoing SIP-trunks. Any pointers to how I could solve this? - J

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Installing OpenZap?
by GillesToo 28 May '10

28 May '10

Hello I read the documents in the wiki, but my Dahdi + OpenZap + FXO card combo doesn't work, so I'd like to check if I made a mistake, overlooked something, or maybe some things have changed since Asterisk moved from Zaptel to Dahdi and the wiki hasn't been updated accordingly: 1. Compiled Dahdi 2.3.0, and successfully loaded it: # /etc/init.d/dahdi status ### Span 1: WCFXO/0 "Wildcard X100P Board 1" (MASTER) RED 1 FXO FXSKS (SWEC: MG2) RED 2. vi /usr/local/freeswitch/conf/zt.conf: Nothing to do here, apparently 3. vi /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml: Uncommented <load module ="mod_openzap"/> 4. vi /usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml: Left default settings 5. vi /usr/local/freeswitch/conf/openzap.conf: [span zt] name => OpenZAP number => 1 fxo-channel => 1 6. The wiki says to then run fs_cli, followed by "oz list", but: "oz list: Command not found!" Did I miss something? Thank you for any hint. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Installing-OpenZap-tp5111050p… Sent from the freeswitch-users mailing list archive at Nabble.com.

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sip_cid_type settings
by Frank ＠ Impact 28 May '10

28 May '10

I can only find documentation on setting this value at the bridge level. Is there a way to control this setting in the gateway profile instead of having to do it in the bridge setup?

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Q931_Pcap debug - Error
by lakshmanan ganapathy 28 May '10

28 May '10

Hi all, I'm trying to configure digium with freeswitch. I configued zaptel and then I installed freeswitch with openzap. Here is my openzap.conf [span zt PRI_1] trunk_type => e1 b-channel => 1-15 d-channel => 16 b-channel => 17-31 [span zt PRI_2] trunk_type => e1 b-channel => 32-46 d-channel => 47 b-channel => 48-62 openzap.conf.xml: <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="10"/> </settings> <pri_spans> <span id="1" name="PRI_1"> <param name="q921loglevel" value="debug"/> <param name="q931loglevel" value="debug"/> <param name="mode" value="user"/> <param name="dialect" value="q931"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> <span id="2" name="PRI_2"> <param name="q921loglevel" value="debug"/> <param name="q931loglevel" value="debug"/> <param name="mode" value="net"/> <param name="dialect" value="q931"/> <param name="dialplan" value="XML"/> <param name="context" value="outgoing"/> </span> </pri_spans> </configuration> I started freeswitch. When I said, "oz q931_pcap 1 on" in fs_cli, it tells the following error 2010-05-28 17:14:08.122324 [WARNING] mod_openzap.c:3070 Error couldn't (re-)enable Q931-To-Pcap! I've installed libpcap and its devel package Can any one please say how to solve this issue?

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