[Freeswitch-users] Choosing a Codec.

Hi, I am playing a file to a landline number. the format of the file is as follows: [r...@static-host var]# file message.wav message.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 8000 Hz In my vars.xml file I have used the following codec prefs: X-PRE-PROCESS cmd=set

Re: [Freeswitch-users] Choosing a Codec.

...@freeswitch.org wrote: Why? You don't have to avoid it... why bother? /b On Dec 22, 2009, at 4:28 PM, Vinuth Madinur wrote: My basic intent is to avoid on-the-fly transcoding, while having a high quality audio playing on PSTN. ___ FreeSWITCH-users

Re: [Freeswitch-users] Choosing a Codec.

. Can you elaborate on your setup a bit more? /b On Dec 22, 2009, at 4:54 PM, Vinuth Madinur wrote: The audio quality is a lot different when it plays on the landline. And the quality degrades a bit when the message played is lengthy 30s. So I thought it would be better if I have the file

Re: [Freeswitch-users] Choosing a Codec.

On Wed, Dec 23, 2009 at 12:17 PM, David Knell d...@3c.co.uk wrote: On Tue, 2009-12-22 at 23:52 -0500, Michael Jerris wrote: That being said, ulaw l16 alaw will cause degredation and any other modifications such as volume adjustment in this path will make it worse. Indeed. Storing

Re: [Freeswitch-users] Performance Tuning

1. http://wiki.freeswitch.org/wiki/Modules.conf.xml http://wiki.freeswitch.org/wiki/Modules.conf.xml2. http://wiki.freeswitch.org/wiki/Sofia.conf.xml#manage-presence http://wiki.freeswitch.org/wiki/Sofia.conf.xml#manage-presence3. http://wiki.freeswitch.org/wiki/Getting_Started_Guide#SIP_Profiles

Re: [Freeswitch-users] Estimating Call Capacity

Here are a few benchmarks that I had stumbled upon. http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2 http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2 Thanks, Vinuth. On Tue, Oct 27, 2009 at 1:43 AM, Brian West br...@freeswitch.org

[Freeswitch-users] SIT tones and SIP Trunk provider.

Hi, Does Freeswitch detect all of these hangup cases mentioned here [ http://wiki.freeswitch.org/wiki/Hangup_causes] when using it through a SIP Trunk provider? If not, should I put in tone_detect application in the dialplan for detecting the SITs? Won't freeswitch have to depend on the SIP

Re: [Freeswitch-users] SIT tones and SIP Trunk provider.

Thanks Michael. I'll go through the resources you mentioned. Thanks, Vinuth. On Tue, Oct 13, 2009 at 2:15 AM, Michael Collins m...@freeswitch.org wrote: On Mon, Oct 12, 2009 at 12:58 PM, Vinuth Madinur vinuth.madi...@gmail.com wrote: Hi, Does Freeswitch detect all of these hangup cases

Re: [Freeswitch-users] Need Help in Getting DTMF

You can use play_and_get_digits command or the read command. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read Thanks, Vinuth. On Sat, Oct 3, 2009 at 9:54

Re: [Freeswitch-users] Listening to a connected call [barge in]

Hi, Use the eavesdrop command. Just supply it with the call UUID and the extension of B. Wiki has more details. Thanks, Vinuth. On Thu, Oct 1, 2009 at 7:20 PM, Nagalenoj nagale...@gmail.com wrote: In ES outbound, I need to do the following, * A calls 2000(FS ES outbound extension) * In the

[Freeswitch-users] Mod_vmd and Session Locking.

Hello All, I'm trying to do a simple dialer, where I am: 1. Initiating mod_vmd on channel answer. 2. Staying quiet until there is a beep. 3. Leave a message on beep. 4. Hangup. (in my scenario it's guaranteed to hit voicemail.) But the session isn't exiting properly after hangup, with last

Re: [Freeswitch-users] Mod_vmd and Session Locking.

Yeah. Anthm fixed it. Was talking over on IRC yesterday. Thanks for the fix. On Sun, Sep 27, 2009 at 7:49 AM, Seven Du dujinf...@gmail.com wrote: what's your rev? I think rev14494 might related to you. 2009/9/27 Vinuth Madinur vinuth.madi...@gmail.com Hello All, I'm trying to do a simple