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codec transcodation not working
by neeraj.p 24 Oct '13

24 Oct '13

Hey, I tried to establish audio call between two clients having different codec supports . I expected that transcoding would take place. But I am getting a codec negotiation error . Here is the information about client I am using leg A : suppoerted codec : speex leg B: supported codec : G722,PCMU,PCMA,GSM,OPUS Here is some lines from my freeswitch logs 2013-10-22 10:31:12.795786 [INFO] switch_ivr_originate.c:1190 Sending early media 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[G722:9:8000:20:64000] 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[PCMU:0:8000:20:64000] 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[PCMA:8:8000:20:64000] 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[GSM:3:8000:20:13200] 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:3064 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2013-10-22 10:31:12.795786 [ERR] mod_sofia.c:2122 CODEC NEGOTIATION ERROR. SDP: v=0 o=Zoiper 0 0 IN IP4 120.63.38.94 s=Zoiper c=IN IP4 120.63.38.94 t=0 0 m=audio 11525 RTP/AVP 110 a=rtpmap:110 speex/8000 Here is some info about my freeswitch version - 1.4 beta codecs supported - codec,ADPCM (IMA),mod_spandsp codec,AMR,mod_amr codec,B64 (STANDARD),mod_b64 codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.722,mod_spandsp codec,G.723.1 6.3k,mod_g723_1 codec,G.726 16k,mod_spandsp codec,G.726 16k (AAL2),mod_spandsp codec,G.726 24k,mod_spandsp codec,G.726 24k (AAL2),mod_spandsp codec,G.726 32k,mod_spandsp codec,G.726 32k (AAL2),mod_spandsp codec,G.726 40k,mod_spandsp codec,G.726 40k (AAL2),mod_spandsp codec,G.729,mod_g729 codec,GSM,mod_spandsp codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,LPC-10,mod_spandsp codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,VP8 Video (passthru),mod_vp8 My sip_profile configurations <param name="disable-transcoding" value="false"/> <param name="inbound-late-negotiation" value="true"/> I also tried with inbound-late-negotiaion=false . But still getting codec negotiation error. Please help . Regards, Neeraj

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Out of band DTMF in Javascript
by Murthy Gandikota 24 Oct '13

24 Oct '13

Hi Can anyone please confirm that the following template is RFC2833 compliant out-of-band DTMF for the #3#? "%(200,0,941.0,1477.0);%(200,0,697.0, 1477.0);%(200,0,941.0,1477.0)"; I am trying to use TeleTone to generate the #3# response to a caller as follows: var pound3pound ="v=-7;%(200,0,941.0,1477.0);%(200,0,697.0, 1477.0);%(200,0,941.0,1477.0)"; var tts = new TeleTone(session); tts.generate(pound3pound); I can hear the digits play, but don't know if they are RFC2833 compliant out-of-band DTMF. Thanks Murthy

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disconnect after 30-60 with freeswitch and polycom phones on same network
by Allen Schultz 23 Oct '13

23 Oct '13

I'm having an issue with the 30-60 second disconnect on both calls (from outside sip provider) and local voicemail (seperate incident, but related). Below I have some pastebin logs of some information with some preliminary steps with the #freeswitch irc group. Event: with sofia siptrace on <http://pastebin.com/cvhjAJpE> sofia status <http://pastebin.com/C6uFZTTC> sofia status profile internal <http://pastebin.com/Km059vhC> sofia status profile external <http://pastebin.com/eVwZTNJy> Polycom soundpoint ip 430 rev a configured polycoms to go directly to 192.168.123.6 Any help will be appreciated. Allen

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Audio quality issues
by Grant Bagdasarian 23 Oct '13

23 Oct '13

Hello, I was wondering what the maximum concurrent calls for FS before audio quality becomes an issue? I assume the specs of the machine would also affect this. We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality at these rates is still fair, but we do notice some quality issue's. Going above these numbers screws up the audio quality: choppy sound, audio drops etc. We aren't doing any heavy media processing, just simply playing a file (G711-Alaw) which lasts about 2 minutes during the load test. These numbers are for one way audio, where Sipp doesn't echo the RTP back. These numbers get lower once Sipp echo's the RTP. I've tried FS on a physical box and also on a virtual box (ESXi 5.1), but the performance gain on physical vs virtual isn't that much. I disabled all the modules we don't need, like CDR's, conferencing, etc. Are there any parameters(config files)/modules that can affect the quality of the audio stream? Regards, Grant

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FreeSwitchConfig GUI Demo Server
by Roger Castaldo 23 Oct '13

23 Oct '13

Hi everyone, I have finally had the time and ability to get up a bit of a demo server for the FreeSwitchConfig gui that I have been working on. https://wiki.freeswitch.org/wiki/Freeswitch_Gui#FreeSWITCHConfig It is on a single CPU, low memory, free VPS, so its not going to respond as fast as it could on real hardware. As well as the server is kind of locked down for connecting to the freeswitch instance because its purpose is to demo the site, and hopefully get some feedback on the interface itself, as well as maybe get some interest in people using it, or helping develop it. site: http://freeswitchconfig.dalnet.ca:8080 user name: admin password: admin As per the wiki entry, every night it rebuilds to the current trunk as well as wipes out all the database data.

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Re: [Freeswitch-users] Limit help
by Andre Demattia 23 Oct '13

23 Oct '13

Can I use limit_execute for a outbound gateway in place of the loop back option? -----Original Message----- From: "Matt Williams" <matt(a)williamsmjw.com> Sent: ‎10/‎22/‎2013 12:01 PM To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Limit help You need to set your CPS and port limits separately. You can set limits per gateway the same way you would limit any application. It is not bound to any specific application. The values you use are completely arbitrary. That said here is an example from the Wiki http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-u… On Tue, Oct 22, 2013 at 10:40 AM, Andre Demattia <andretodd(a)verizon.net> wrote: Hi, how do I set the CPS and Port limits on a bridge Can I set both on one application or do I need to set 2 limits? 5 ports <action application="limit" data="hash outbound customerxxx 5 !NORMAL_TEMPORARY_FAILURE" /> 5 cps <action application="limit" data="hash outbound customerxxx 5/1 !NORMAL_TEMPORARY_FAILURE" /> Can I do this for both 5 ports and 5 cps? <action application="limit" data="hash outbound customerxxx 5 5/1 !NORMAL_TEMPORARY_FAILURE" /> Also how do I do the same on a gateway? I need to make sure I don't send too many calls to the provider. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Phones registered to internal profile hit external profile when calling
by hcoin 23 Oct '13

23 Oct '13

This has been a really frustrating problem, I'm sure the answer is simple but I just can't see it. I had several extensions registered to the internal profile, sending calls out the external profile to a sip-pstn gateway, all seemed fine. Then created another internal profile, using a different sip port on the same lan address, because of 'no device left behind' and NAT issues. All seemed well, all the phones register normally. Looking at the databases in FS they all show the proper ports, the proper domains, etc. However, every single call gets picked up as a new call via sophia/external/... and it hits the public dialplan normally -- except that's the wrong plan, it should hit the default plan and be identified as sofia/internal/.... and so forth. 2013-10-22 00:31:11.001600 [NOTICE] switch_channel.c:1034 New Channel sofia/external/hcoin(a)pbx.foobar.com [28ed125a-3adb-11e3-9cc1-cbb8efb09b83] What could possibly be the reason phones registered on the internal profile have their new calls identified as sophia/external and don't hit the correct plan? Both the phones and the freeswitch are on the same subnet. This should be so vanilla. What am I missing?

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FreeTDM tones.conf Disconnect Supervision
by Nandy Dagondon 23 Oct '13

23 Oct '13

Hello all, Our telco recently changed their call progress tones: Dialtone: 450Hz continuous Ringback tone: 450Hz/1 sec ON 4 sec OFF Busy tone: 450Hz/360ms ON 360 OFF Attn: 950Hz/continuous (0db) The way I understand it, FXO listens for the dialtone then sends DTMF tones. After this, it expects either the Ringback or Busy tone. I read Michael Collin's post that FreeTDM doesn't detect (as of now) cadence. http://permalink.gmane.org/gmane.comp.telephony.freeswitch.user/36531 Questions: 1. What state will FreeTDM go when it detects 450Hz? Ringing or Busy? I presume it would be Ringing or it would hangup. 2. Can I use tone_detect to distinguish them? Can I use a series of tone_detects? Any suggestion where to go? If not ... My last recourse will be - wait for the Attn tone. But ... will FreeTDM jump to hangup the call upon detecting this? Thanks, /Nandy

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How to configure a Gateway that registers on my FS
by David Villasmil 23 Oct '13

23 Oct '13

Hello, There are many examples for gateways on which FS registers... but how about the other way around? I have a gateway that needs to register on my FS... I know I can configure it as a client, but then how do I use it as a gateway? Thanks David

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Configurable Hold Music while conference in freeswitch
by Harsimran singh 23 Oct '13

23 Oct '13

Hi, I want play configurable hold sound while conference using freeswitch. I am using "moh-sound" variable in conference.conf.xml as follows. <param name="moh-sound" value="/srv/sounds/${uuid}/knowlus_hold_music.mp3"/> The uuid is the variable "variable_uuid" in the events comes from freeswitch,but the freeswitch is unable to expand this variable and error comes as follows : [ERR] mod_shout.c:683 Error opening /srv/sounds/${uuid}/knowlus_hold_music.mp3 [ERR] mod_shout.c:862 Error from mpg123: File access error. (code 22) Freeswitch is not able to expand the variable as variable exists and the file which i want to play also exists. Can anybody help me out in this ? With Regards Harsimran Singh +91-9711271158

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