January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

CDR and reporting.
by Dome Charoenyost 13 Jan '10

13 Jan '10

Dear All, I have 200k cdr record daily. what's good solution to record and report ? Now I'm thinking about CDRTOOL from ag project and areski stat (now part of a2billing). My idea use mod_xml_cdr -> http (may be php, or perlembded on nginx) -> mysql i want to report ASR, ACD , any comment ? Dome C.

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Inbound DTMF Not Recognized In Latest Version
by Jerry Richards 13 Jan '10

13 Jan '10

I am having a problem where in inbound call from the PSTN going to the demo IVR does not recognize DTMF digits. I posted a log at http://pastebin.freeswitch.org/11769. I am running Freeswitch (freeswitch-1.0.5-20100112-0400.tar.gz) with an A101D Sangoma Card and wanpipe driver: wanpipe-3.5.10.tgz. I am not using libpri. It appears that each DTMF digit might be getting detected twice. I tried to dial "5401" and it thought I dialed "5544". I am following the instructions at http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf, as shown below. Also, I tried moving the "start_dtmf" statement in between all the statements within the condition tag, but it didn't make any difference. <extension name="ivr_demo"> <condition field="destination_number" expression="5000"> <action application="answer"/> <action application="start_dtmf" /> <action application="sleep" data="2000"/> <action application="ivr" data="demo_ivr"/> </condition> </extension> Also, I noticed at line #293, the log is saying: [ERR] mod_sndfile.c:194 Error Opening File [/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System error : No such file or directory.] Has anyone else encountered this issue? Best Regards, Jerry

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Sip provider Using IP authenication
by John 13 Jan '10

13 Jan '10

I'm am attempting to use Voipinnovations.com (M2Tel) as a SIP provider. They do not utilize username/password; how would I go about creating a gateway entry for them without username/password? Thanks

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GSM modem for Skype
by 大泥人 13 Jan '10

13 Jan '10

Hello All, I had a GSM VoIP Gateway which can be connected to service providers who use SIP protocal. I remember somebody in Freeswitch community mentioned there is a software which can connect GSM network with Skype network. I'm not sure if this software can work with my GSM VOIP gateway or not. If somebody can send me such software that would be much appriciated and even license fee required would be accepted. I hope I can have a testing on this or I can send this gateway to those guys who have this kind of software for testing. Thanks and Merry Christmas. Daniel Zeng _________________________________________________________________ 约会说不清地方？来试试微软地图最新msn互动功能！ http://ditu.live.com/?form=TL&swm=1

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playback and play_and_get_digits strange misunderstanding
by François Legal 13 Jan '10

13 Jan '10

Hello, trying to make so dialplan extensions that use the playbvack and play_and_get_digits applications, but I'm having trouble with the file name specification. The files I want to play are in the french language (fr/fr/julie as configured in lang/fr/fr.xml) My extension is as follows : The channel is using a bit rate of 8000 Hz, so by the set default_language=fr I would expect freeswitch to playback the file at ${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/8000/ringing_disabled.wav whereas it tries to playback the file at ${FREESWITCH_PATH}/sounds/en/us/callie/misc/ringing_disabled.wav I have the same with play_and_get_digits application. What am I doing wrong ? François

1 0

Re: [Freeswitch-users] sangoma echo cancellation software
by Jeff Lenk 13 Jan '10

13 Jan '10

No such luck - I dont know any solution other than get the other card -----Original Message----- From: Diego Toro Sent: 1/13/2010 2:44:13 PM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] sangoma echo cancellation software greetings, I have sangoma analog card without echo cancellation hardware, I have problems with echo on calls, does FreeSwitch support echo cancellation software on windows ? Thanks Diego Toro http://lacarretade.blogspot.com

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Help rewriting hangup causes
by Ahmed Naji 13 Jan '10

13 Jan '10

Dear All, I posted a thread re the subject but didn't get any joy, so perhaps second time lucky. I need to rewrite a couple of hangup causes to mean NORMAL_CONGESTION and no matter which variables I set, this isn't happening. The idea is to control what codes are returned to an end point after a successful bridge, as well as deal with what codes are returned if the bridge is unsuccessful (e.g. user_busy, originator_cancel ...etc). I've had limited success by setting hangup_after_bridge=false then bridging to error/<required_code>. This, however only works when the B-leg terminates the call after a successful answer. Any other codes are not rewritten. I've also tried playing with the bridge_hangup_code and hangup_code variables prior and after bridging, still no joy. I have also set sip_ignore_remote_cause=true prior to entering the bridge, as well explicitly in vars.xml. By the way, I'm running in proxy-media mode, but I did try it with bypass-media as well. Same symptoms, same behaviour. Any help with this would be highly appreciated. Regards, Ahmed.

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Hardware echo cancellation.
by Tim Uckun 13 Jan '10

13 Jan '10

I am about to build a new machine as a VOIP server. I am going to get either a quad core intel or a six core AMD processor with at least eight gigabytes of RAM in it. Given that much horsepower I am wondering if there is any need to purchase hardware with echo cancellation (I am thinking about redfone devices).. I can save some money by not getting the echo cancellation. So is it worth saving that money? Is it always better to have hardware echo cancellation? Is a quad core capable of dealing with echo cancellation needs of an IVR which is going to take lots of simultaneous calls? Thanks.

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need help with switch_ivr_originate
by Juan Backson 13 Jan '10

13 Jan '10

Hi, I tried to use the function switch_ivr_originate to originate a call instead of using the "bridge" app. However, I am getting "ORIGINATOR CANCEL" error after switch_ivr_originate is executed. Does anyone know why would freeswitch cancel the call ? I have checked all sample usage of switch_ivr_originate in the code but no luck. thx, jb

1 0

multiple leg and multiple rtp
by David Villasmil 13 Jan '10

13 Jan '10

is it possible? can i bridge to multiple b-sides and múltiple

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