- FreeSWITCH-users - FreeSWITCH Mailing Lists

debugging no audio I/O
by Neil Patel 31 Mar '10

31 Mar '10

Hi all, I'm stuck trying to figure out why my lua app does not output any audio or detect any DTMF input. This app has been installed and running successfully on another server. The lua code in question is: read(absolute_path_to_wav, 0); Logs show: 2010-03-29 20:12:15.596856 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16@8000hz 1 channels 20ms 2010-03-29 20:12:15.596856 [DEBUG] switch_ivr_play_say.c:1444 done playing file DTMF input is not being recognized, nothing is showing up in the logs. App is being accessed with SIP from a remote client. I'm not really sure how to debug this, so any pointers are appreciated. This is FS from trunk. Thanks, Neil

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Calling Phone Behind Netgear Router (with NAT)
by Jerry Richards 31 Mar '10

31 Mar '10

We are trying call a phone that resides behind a Netgear router (with NAT). The Bria softphone registers and we can make calls from the softphone, but we cannot make calls to the softphone. WAN ------------ public IP -------- 10.10.10.xx ---------------- |Freeswitch|<---------->|Router|<----------->|Bria softphone| ------------ -------- ---------------- Does anyone know of good documentation/example showing this configuration? Best Regards, Jerry

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Re: [Freeswitch-users] Some question about mod_fifo ??
by Nguyễn Mạnh Hùng 31 Mar '10

31 Mar '10

Hi Mike and Seven Du. Thanks to yours help. I known the mechanism of mod_fifo. >> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175… . What a pity, It can't solve this problem. I can't use freeswitch for my call center. Hope new version can solve this !!!

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I need some comment for Priority Call
by Dome Charoenyost 31 Mar '10

31 Mar '10

Dear All, I'm planing to make priority call in dial plan. it's mean if we have maximum 30 concurrent port for dial out. and use all port. when some extention make call (VIP call) dial plan hangup some longest call and give port for VIP. any idea ? Dome C.

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Regular Expression for match ip address
by Dome Charoenyost 31 Mar '10

31 Mar '10

Dear All, I want to autorize incoming call to public context by IP by <condition field="network_addr" expression="^192\.168\.1\.1$"/> but some time i want to allow 1 Class C or some range of IP. How can do that ? BG Dome C.

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Variable "default_language" has no effect on javascript
by eliimap 31 Mar '10

31 Mar '10

Hi I set the variable "default_language" to be "ru" and then I run a js to play a file .... session.streamFile("conference/ronf-kicked.wav", ""); .... I got an error message 2010-03-25 10:23:46.430391 [ERR] mod_sndfile.c:194 Error Opening File [/freeswitch-1.0.5-20100314-0400/sounds/en/us/callie/conference/ronf-kicked.wav] [System error : No such file or directory.] that indicates that mod_sndfile is not checking this value but always using en Please check Eli

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Radius/Authentication/Authorization
by Abid Saleem 31 Mar '10

31 Mar '10

Hi Neal and other Contributors to FS, I recieved an answer on the list that mod_rad_auth is ready. I upgraded FS to download and install it by "make current" and it is successfully built and installed. Could you please outline the detailed steps to do all this configuration. If it is still not ready, is there any other method that somebody has already implemented like perl scripts etc. Any help in this regard is badly required. THanks for your cooperation. Regards----------Abid SaleemSr. Product Manager _________________________________________________________________ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969

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Newbie MWI question
by lloyd thomas 31 Mar '10

31 Mar '10

I am more or less using the default config files with FS. I have a Mitel 5212 phone, which I leave a message for but the MWI light never illuminates Any Ideas?

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DTMF
by Sean Holt 31 Mar '10

31 Mar '10

Hello, I¹ve posted on the question on this topic before but I¹ll try again with more detail. I have this svn version FreeSWITCH Version 1.0.trunk (16952) SIP provider: Bandwidth.com I¹m assuming that DTMF rc2833 is the default, but I can¹t get my ivr to recognize the incoming call tones. Bandwidth can set up either inband or RC2833. They require these codecs, G.711Ulaw and G.729A. The tech at Bandwidth wanted me to setup inband to test if that works better, but wasn¹t sure where to set it up. I added this to my dialplan <extension name="main_number_23" continue="true"> <condition field="destination_number" expression="^3000$"> <action application="answer"/> <action application="start_dtmf"/> <action application="sleep" data="1000"/> <action application="conference" data="conference_1@conference_1"/> </condition> </extension> Should I add it to my sip trunk? Not sure how to troubleshoot this issue Thanks Sean

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Favorite SIP Providers? Australian providers?
by Fraser Redmond 30 Mar '10

30 Mar '10

I was looking through the list of Providers with example configs in the wiki: http://wiki.freeswitch.org/wiki/SIP_Provider_Examples I was wondering if anyone had a provider that they really loved? And why? I've got inbound 1800/0800 in UK & USA already, but might change. Also interested in outbound. Also, I want to get a 1800 number in Australia - any suggestions? Sorry in advance if this is one of those questions that does the rounds a couple of times a year (I did have a look through the archives via Google and didn't see anything), or if I'm treading on dangerous ground asking for recommendations of commercial services - so I guess only recommend services you use, not services you provide ;-) Cheers, Fraser

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