- FreeSWITCH-users - FreeSWITCH Mailing Lists

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

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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

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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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unable to dial out troug siemens Hie9200 softswitch
by Meftah Tayeb 13 Jan '10

13 Jan '10

hi dear all, we have a siemens HIE 9200 softswitch that support SIP, SIP-T and H.248 we are trying to interconnect freeswitch with it if HIE9200 dial out troug freeswitch, is passing the call, but without RTP but if the freeswitch dial out troug the HIE9200, the call is unable to pass with error 500 the trace is here: http://siplabs.net/tracebin/fs-siemens-500.pcap http://siplabs.net/tracebin/fs-siemens-rtp.pcap unfortunatly the freeswitch don't support H.248 otherwise i will control the media gateway (HIG 1100) thanks for any help

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Hello , about calculate numbers
by Chia-Yen Wu 13 Jan '10

13 Jan '10

Hello everyone, Is there anyway to calculate number in dialplan.xml? like example: <action application="set" data="var=1"/> and with an action , I can do : var=var+1 I have tried mod_expr, but failed to catch the variable, like below: <action application="set" data="var=${expr($var+1)}"/>

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Bypass_media mode
by Mark Campbell-Smith 13 Jan '10

13 Jan '10

Hi! Hi! I am calling from extension 2010 to extension 1000. Both have ip addesses 192.168.1.x. In the 2000 series dialplan (a separate context) I have the following to try to enable bypass_media. <extension name="Local_1000_9xxx"> <condition field="destination_number" expression="^(10[01][0-9]|9\d{3})$"> <action application="set" data="dialed_extension=$1"/> <action application="set" data="proxy_media=false"/> <action application="set" data="bypass_media=true"/> <action application="transfer" data="${dialed_extension} XML default"/> </condition> </extension> Is this how bypass media should be enabled? This fails fo me (the calls hang up and no audio). The debug trace is in http://pastebin.freeswitch.org/11737 What have I done wrong? Thanks

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