January 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Hello!
by J. G. 28 Jan '09

28 Jan '09

FS users, Just joined the growing list of FS users and found this list. I spend a lot of time in Asterisk (own a 48 seat call center that uses an Asterisk distro as our main switch, etc) and am playing with my first integration between FS and Asterisk, so far so good - looking forward to Anders' release of Phonebooth for the masses. Anyway, glad to be on the list! J. -- ----- Jason Gehman General Manager North Voice Communications www.NorthVC.com

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portaudio error
by e schmidbauer 28 Jan '09

28 Jan '09

hey guys im getting an error when trying to use a USB mic with portaudio. wondering if anyone could help me solve this problem. im using ubuntu intrepid studio. portaudio is loaded correctly and sees the all the devices. when i do a "pa call" using the built in sound card, i am able to connect to the call. but when i set the "pa indev #7" (#7 is the USB mic) and then do a "pa call" i get the following error >pa call 4800 2009-01-28 12:33:09 [NOTICE] switch_channel.c:566 switch_channel_set_name() New Channel portaudio/4800 [bae25c96-ed61-11dd-a1c3-d19a8f52ba15] 2009-01-28 12:33:09 [ERR] mod_portaudio.c:1350 engage_device() Error opening audio device retrying 2009-01-28 12:33:11 [ERR] mod_portaudio.c:1359 engage_device() Can't open audio device 2009-01-28 12:33:11 [NOTICE] mod_portaudio.c:1739 place_call() Close Channel portaudio/4800 [CS_NEW] API CALL [pa(call 4800)] output: FAIL:Device Error! here is the output from "pa devlist" > pa devlist API CALL [pa(devlist)] output: 0;/dev/dsp;16;16;r,o 1;/dev/dsp1;16;0; 2;Intel ICH6: Intel ICH6 (hw:0,0);2;2; 3;Intel ICH6: Intel ICH6 - MIC ADC (hw:0,1);2;0; 4;Intel ICH6: Intel ICH6 - MIC2 ADC (hw:0,2);2;0; 5;Intel ICH6: Intel ICH6 - ADC2 (hw:0,3);2;0; 6;Intel ICH6: Intel ICH6 - IEC958 (hw:0,4);0;2; 7;Samson C01U : USB Audio (hw:1,0);2;0;i 8;front;0;2; 9;default;128;128; 10;dmix;0;2; any input would be much appreciated. thanks again for developing such a fine piece of software!

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DTMF with Early Media Disabled
by Klaus Teller 28 Jan '09

28 Jan '09

Hi, My settings does not allow me to test the following right now. So I'm wondering if somebody knowledgeable could help me answer the following question. I do know that if i call Freeswitch, i can use Javascript to read DTMF even without answering the call. My question is can i do this even if early media is disabled on the inbound call? Thanks, Klaus. -- NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a

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Sending SIP calls via outbound proxy
by Erick Johnson 28 Jan '09

28 Jan '09

i forgot to give you the pastebin URL http://pastebin.freeswitch.org/6379 > > I'm running latest trunk - Revision: 10682 > > I've been doing an ngrep on my external freeswitch SIP port and FS > is not sending any SIP packets anywhere when I run the following command. > Bumping up TPORT_LOG to 9 also confirms this, as no SIP packets are > logged. > > originate > 'sofia/external/erick at ejjohnson.org;fs_path=proxybeta.jnctn.net' > &echo() > > Also, just to be clear, when I remove ";fs_path=..." from the command > above a call > is set up normally to erick at ejjohnson.org and the SIP packets are > logged > to console. > > Thanks guys. > > > What SVN rev are you running? Also, could you do a SIP trace? > > TPORT_LOG=1 && /usr/local/freeswitch/bin/freeswitch > > Pastebin the output of that and we'll take it from there. > > -MC > > > > On Tue, Dec 9, 2008 at 11:41 AM, Erick Johnson > > <erick at junctionnetworks.com> wrote: > > > Both: > > > > > > originate sofia/external/'erick at > > ejjohnson.org;fs_path=proxybeta.foo.net' > > > &echo() > > > originate sofia/external/erick at > > ejjohnson.org;fs_path=proxybeta.foo.net > > > &echo() > > > > > > produce the exact same result & log > > > > > > :( > > > > > >> * I think you need to '' the sofia uri /b > > > > > > > > -- Erick Johnson sip/email erick(a)junctionnetworks.com 1-800-801-3381 x7006 Software Engineer Junction Networks

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Another way to set CallerID?
by Ron McCarthy 28 Jan '09

28 Jan '09

Hi, I am having a weird issue with setting the callerID number for outbound calls, I have this: <action application="set" data="effective_caller_id_number=17025551234"/> <action application="bridge" data="sofia/gateway/carrierX/17025559988"/> Ive set the callerID correcly on a Asterisk box and the carrier sees it and passed it correct, but for some reason any calls from Freeswitch won't work, it either shows up as private or another number, so they are not seeing the CID for whatever reason. This method works for other carrirers just fine, but not them. From what I read this sets the from in the SIP message, maybe for some reason that is goofing it up. Any other way to set the callerID in Freeswitch, that's the only way I have found. FS shows it's connecting to a Sonus, does it need to be set different because of some Sonus issue that im not aware of? Any help would be great. Thanks!

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mod_radius_cdr questions and thoughts
by Apostolos Pantsiopoulos 28 Jan '09

28 Jan '09

I am trying to implement a radius based solution using FS. I have seen that the mod_radius_cdr module is actively maintained. so I have a few questions/remarks : 1) When I place a call and my radius server is down, the call blocks forever instead of just radius_timeout * radius_retries seconds (I have declared only one server). I would expect that FS would stop trying to send an Acc Start packet after some time and get on with the call. 2) I have also noticed that FS sends only 1 packet (I waited for a minute) instead of 3 (default in the config) since the first (and second) attempt failed. If my server was up (the port was responding) but it returned a req. failed answer would the above time-out be valid? 3) When I tried to load the dictionary.freeswitch to my freeradius server, it complained : "Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary.freeswitch[33]: dict_addattr: Duplicate attribute name NAS-Port-Id" How can I overcome this? When I "instructed" my freeradius to only load the freesitch dictionary I got this : "Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary.freeswitch[257]: unknown option "Freeswitch"" How can I use the dictionary with the freeradius? (or other radius servers?) 4) The radius attributes included in the current requests are a) hard-coded, b) limited in number. I think many of us would like to use more attributes. Or even better define what to include (and what to put in them) using a config file (the same maybe?) 5) Does the module send accounting packets only for the a-leg of a call or for both legs? (Maybe that could be configurable too). If anyone is interested in the above questions/remarks please post a reply. I would really like to know how many of the mailing list users are also interested in FS radius support and your opinions on the matter. Regards, -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

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Method getVariable cause error on FS 1.0.2 in javascript
by shehzad p 28 Jan '09

28 Jan '09

Hi all, I am using the javascript for call termination, and after call complete, I use getVariable mehtod to get some values like PDD. Below is my javascript that perform this: ============================================================ dialstr="sofia/outbound/1111(a)xxx.xxx.xx.xx"; newsession.originate(session, dialstr); if(session.ready() && newsession.ready()) bridge(session, newsession); if(newsession.getVariable("progressmsec") != false) //in FS 1.0.2 and SVN version SCRIPT TERMINATES on above line pdd = newsession.getVariable("progressmsec"); else pdd = 0; .................<other code> ========================================================== Above script works fine in FS 1.0.1. Now I am testing FS 1.0.2, It also works fine when call is being ANSWERED, PDD is correctly collected, BUT when Call is not routed successfully due to some network problem, or User No-Answer / Busy, then below error occurs: ------------------------------------------------------------------------------------------------------------------------------------------------------- 2009-01-19 19:40:43 [ERR] inline:1 mod_spidermonkey() You must call the session.originate method before calling this method! -------------------------------------------------------------------------------------------------------------------------------------------------------- And after this error, script DOES NOT continue execution and TERMINATES. I also tested SVN version (revision 11266), but same error occurs there too. What should be the cause here, so this behavior of FS is changed from 1.0.1 to 1.0.2. Thanks. msp -- View this message in context: http://www.nabble.com/Method-getVariable-cause-error-on-FS-1.0.2-in-javascr… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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javascript setHangupHook .. looking for a working example
by Stephen Crosby 27 Jan '09

27 Jan '09

I'm running rev 11131, and I cannot make these hangup events work in javascript... does anyone have a working example? I tried this script from the wiki: http://wiki.freeswitch.org/wiki/Example_Hangup_hook --Stephen

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Call out to gtalk, asterisk and skype
by EdPimentl 27 Jan '09

27 Jan '09

Hello Everyone, Has anyone done a scalable deployment of calling to Skype/Gtalk/Asterisk Say many 3-25/50 users with various endpoint (skype/gtalk/asterisk)platforms. http://blog.tmcnet.com/blog/tom-keating/asterisk/skype-for-asterisk-launche… http://www.mhspot.com/siptheeskype_skype_trunk_howto.html Thanks in advance, -E

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[CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock
by Chav Paskov 27 Jan '09

27 Jan '09

Hi everybody, i got a strange message today: [CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock i'm running 1.0.trunk (10803) as a VM on ESXi i have a similar setup of 1.0.1 as VM on ESXi but i've never seen this message. Does anybody have an idea what is wrong here? All tips / hints appreciated. Best Regards Chav

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