December 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

python in freeswitch
by Joli Martinez 04 Dec '13

04 Dec '13

Hello, I would like to be able to mute/unmute users in a conference using a python script. Not sure how to do that. Please help, Thanks

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FreeSWITCH and Raspberry.
by Fernando Villarroel 04 Dec '13

04 Dec '13

Dear All. Someone tested FS and Raspberry as a SOHO PBX. I need to know their experiences, i want to run in Raspberry : USBfxo Sangoma. FreeSWITCH PostreSQL Nginx Django What do you think? I wait your feedback or advice. Enviado desde mi iPhone

7 14

Getting FreeSWITCH to use RTP port + 1 for RTCP
by James Mortensen 04 Dec '13

04 Dec '13

Hello, I've searched the list and documentation, and it's not yet clear if FreeSWITCH is able to send candidates in the same manner as we were doing with Asterisk 1.5 where it sends the RTCP to the RTP port +1. In other words, I'd see candidates that looked like this: a=candidate:3441596188 1 udp 659136 10.3.1.1 20622 typ host generation 0 a=candidate:3441596188 1 udp 659136 50.18.X.X 20622 typ srflx generation 0 a=candidate:3441596188 2 udp 659136 10.3.1.1 20623 typ host generation 0 a=candidate:3441596188 2 udp 659136 50.18.X.X 20623 typ srflx generation 0 For instance, if the RTP port was 20622, the RTCP port would be 20623. We're getting no audio when trying to make Chrome to Android and Android to Android WebRTC calls, and we thought we'd try a different RTCP method as we did have it working with Asterisk but want to use FreeSWITCH instead. I've checked out the latest FreeSWITCH code from the master branch. Not sure if FreeSWITCH can do this yet, but if anyone knows that would be helpful. Thank you! James

3 13

Caller ID Name = Outbound Call
by Roger Castaldo 04 Dec '13

04 Dec '13

Hi everyone, I am having a difficult time with an idea that I thought made sense in my head but apparently isn't working how I think it should and need to know what I am doing wrong. I originate a call using the event socket originate {userid=1234,execute_on_answer='transfer call_user_into_conference_123456 XML public',ignore_early_media=true,origination_caller_id_name='1234',originate_timeout=120,origination_caller_id_number='1234' }user/1000 &park() This should and does execute the extension <extension name="call_user_into_conference"> <condition field="destination_number" expression="^call_user_into_conference_(\\d+)$"> <action application="playback" data="conference/8000/welcome.wav"/> <action application="conference" data="$1@wideband"/> </condition> </extension> when I answer the phone, testing the call, it plays the file then moves me into the conference which is perfect. The problem is I have code monitoring Caller-Caller-ID-Name and that gets moved to Outbound Caller some time between me answering the call and conferencing it, I have tried setting effective_caller_id_name and such but it still changes my caller id name to "Outbound Call". Any advice on how to fix this would be awesome.

2 2

FS codec negotiation...
by rumodigital . 04 Dec '13

04 Dec '13

hello guys! I'm migrating from Asterisk to FS... after a few hours studying and am very pleased with what I have seen. I take the opportunity to ask a question about the codecs: if I have a softphone or hardphone with different codecs no way to communicate? eg. Doug (A) and Bob (B) (A) codecs: G722,PCMU (B) Codecs: GSM,Speex FS list codecs: G722,PCMU,PCMA,GSM,Speex profile: ... <param name="inbound-codec-negotiation" value="generous"/> <param name="inbound-late-negotiation" value="true"/> ... codecs FS: codec, G.722, mod_spandsp codec, GSM, mod_spandsp codec, Speex, mod_speex Doug dials Bob is possible transcoding by FS? here does not work, I'm missing some detail? Regards to all! Doug

1 0

freeswitch SLA and Yealink phones
by Jason Holden 04 Dec '13

04 Dec '13

I am currently testing a t32g and t46g and am experiencing the following problems. The t32g is set under account / advanced for shared line as broadsoft SCA and under DSC the line1 is set to shared. When I press the line key to place a call I receive the message shared line unavailable. On a polycom phone with shared line I am able to place an outbound call using a shared line can anyone provide some advise on this? As well I have a t46G but when I select account / advanced it does not allow me to change the share type from disabled and I can not figure out why. I've already updated the firmware as for Yealink support's advise. Any help on the above would be much appreciated. Thanks.

2 1

Recording Calls
by Hélder Pinheiro 04 Dec '13

04 Dec '13

Hello all, I just recently got to know FreeSwitch and I am very well surprised with it. I am trying to implement a solution using Freeswitch, where a call is established, and in case of the other end caller digits a specific key/set of keys(DTMF), the call starts being recorded. Do you know if FreeSwitch supports this? and if so, could you give me an overview how to do it? Thank you. Regards, Hélder Pinheiro

3 2

undefined Pizza Demo Grammar in Mod pocketsphinx
by Abdullah 04 Dec '13

04 Dec '13

hi every one , i have successfully compile free-switch for Mod pocketsphinx . but that asr is not recognizing the voice , i have see log where grammar is undefined .. please help me where is i getting wrong ? i have used latest pizza_js script and install also its grammar and sound files.. Many Thanks , iifi -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/undefined-Pizza-Demo-Grammar-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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16khz support in mod_opus
by Jan Sverre Andreassen 04 Dec '13

04 Dec '13

Great to see that we now have 8khz and 48khz support in Opus. Is 16khz support also around the corner? Best Regards, Jan Sverre Andreassen ________________________________ This email and any files contained therein is confidential and may contain privileged information. If you are not the named addressee(s) or you have otherwise received this in error, you should not distribute or copy this e-mail or use any of its content for any purpose. Please notify the sender immediately by e-mail if you have received this e-mail in error and delete it from your system.

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On-conference distortion for FreeTDM channels, but not SIP channels
by Melanie Treitinger 04 Dec '13

04 Dec '13

Hi Bob, we have the same problem as you with FreeTDM and distorted audio in a conference. Did you find out what is causing this distortion? Did you find a solution? Thanks! Melanie

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