December 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Record sound quality not good but pcap is quite good
by Ali Pey 12 Jan '15

12 Jan '15

Hello, I use record to record part of a call and sometimes I get some static noises and distortion; however when I listen to the call on pcap capture, it's quite clear and no noise. I record to a wav file and then use sox to convert it to ulaw. Any suggestions? Any help would be greatly appreciated. Thanks, Ali Pey

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Call back through SIP trunk
by Sina Owolabi 10 Jan '15

10 Jan '15

Hi List! FreeSWITCHNewbie here. Please can I have some guidance on how to setup call back? I would like to be able to dial the DID attached to the SIP trunk Freeswitch is registered to, and then have freeSWITCH hang up the call and dial the caller id number back through any other SIP trunk FreeSWITCH Is registered with, but with the origination number set to the DID that the first call came through in the first place. Please is this possible just through the dial plan? Thanks for any help!

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Fwd: OverTime RAM usage keep inceasing
by indra sena 08 Jan '15

08 Jan '15

Hi Team, I have observed that overtime RAM utilization/usage is keep increasing maybe due to memory leaks ? when I put load initially it starts with 5GB/12GM RAM, I have kept for 15 hours load test and oberserved now it is using almost 11GB/12GB starts swap also increasing now. Do you have config changes to fix or to take as work around ? How can we reduce RAM utilization overtime ? Could you please provide some solution for this ? Thanks & Regards, Indra.

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How to play a stream other than mp3 with mod_shout.
by Aqs Younas 06 Jan '15

06 Jan '15

Hi, All How can i play a live stream other than mp3 with mod_shout or any module.? Is there any way to buffer the stream before playing it with mod_shout. Currently i have a list of streams and when i play them with mod_shout some work fine but others give (time out) error. How can i play mostly stream in freeswitch? Thanks

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Unbridge and send to queue
by David Villasmil Govea 03 Jan '15

03 Jan '15

Hello Guys, I receive a call from side A and send it out to side B. I need to unbridge this call once it is answered and send the B side to a queue to give it moh. How do I do that? I'm doing everything with lua and I've tried: on my dialplan I have: <action application="set" data="execute_on_answer=lua /usr/local/freeswitch/scripts/check_answered.lua"/> in check_answered-lua, after checking the call was in fact answered I do: session:execute("transfer", "-bleg 9999 XML default"); and on my dialplan I have: <extension name="9999"> <action application="log" data="INFO SENDING CALL TO QUEUE >>>>>>>>>>>>>>>>>>"/> <action application="lua" data="/usr/local/freeswitch/scripts/queue.lua"/> <action application="hangup" data="NORMAL_CLEARING"/> </extension> queue.lua: session:execute("sched_hangup","+50 alloted_timeout"); session:execute("callcenter","agents_queue"); session:execute("sleep",my_dur); But this doesn't seem to work at all... Can anyone give me a hand? thanks! -- DVG

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SSD Drives vs 15k drives
by Andrew 02 Jan '15

02 Jan '15

Hi, We are going to be buying new servers to run FS as a sip server. I know some people say use SSD drives but I wanted to ask. Performance is a big deal so I like to do anything I can to even get one more CPS. We will be running the FS with no SQL and no RTP(media bypass) and in a ram drive. Will the SSD Drives even be hit in this case? Andrew

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SIP/2.0 480 Temporarily Unavailable
by George F. Phelps 02 Jan '15

02 Jan '15

I am debugging a new/initial Freeswitch configuration. I believe that I have successfully registered with my VoIP provider - "State=REGED". I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1. When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of "SIP/2.0 480 Temporarily Unavailable". Problem #2. Do I have to do anything to enable calling to x9196? And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3. I am getting this critical error on startup. Problem #4. 2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so **/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory** Any suggestions as to what configuration might be wrong? Or how I can get additional debug information? Version info: FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit) Thanks!

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mod_verto and FireFox 34
by Igor Olhovskiy 02 Jan '15

02 Jan '15

Hi! Seems to be mod_verto have no sound with Firefox 34. Even on demo site....

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Polycom horror story
by Luis Daniel Lucio Quiroz 31 Dec '14

31 Dec '14

Hello, Using Freeswitch 1.4.14 with multitenant, I can not make this @#$@%# telephone to register: model: Soundpoint IP501 firmware: 3.1.8.0070 I have done next: NDLB-force-rport with true and safe values No nat involved (switch and phone on same segment) I have tried multiple combinations of configurations on the phone (using web interface). with or without proxy, different sip extensions, different passwords, different ports. I always get 403 error code. however, i can register only if i do next: after telephone is booted, i edit sip password using the phone keypad. And telephone register. This makes me thing it is a bug on the polycom and has nothing to do with Freeswitch. It works on all other phones I have, snom, bria, linphone. But here I am, maybe someone has workarround this somehow. LD

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Flash Hook to an FXO port (ata)
by davy van de moere 31 Dec '14

31 Dec '14

Gents, Does any one have a good setup to send a flash hook to an analogue line? Preferably with an ATA of some sort. So far I've been trying with an Linksys SPA3102, which can do a flash hook with the specific mime type : application/hook-flash , with Freeswitch I the send_dtmf didn't get me any further than sending the special flash key F in application/dtmf-relay. Which is not interpreted by the SPA3102 as a flash hook. Hence, does anyone have a good strategy from a Freeswitch pov to send a flash hook signal (do I need to think sipsak?), or an ATA/firmware which does support the F key in dtmf? Season grtz! Davy

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