Re: [asterisk-users] How to do the Call Snooping

2006-11-15 Thread Vij
chanspy

see: http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy

-VijOn 11/15/06, raviprakash sunkara [EMAIL PROTECTED] wrote:
Hello Users,I googled Call Snooping, its shows only the it means, But i didn't find How to dialplan the Call Snooping,
I seen that  What is Trixbox  in Asterisk I Use only some Feature in Asterisk (20), 
Is it need Asterisk to install the TrixBox in that same System where i installed the Asterisk ServerHelp me please :P-- Thanks and RegardsRavi Prakash Sunkara		

[EMAIL PROTECTED] 	M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 		
[EMAIL PROTECTED]
www.hyperion-tech.com

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Re: [asterisk-users] Desktop integration

2006-11-13 Thread Vij
Basically here is what the application you need has to do:1. take the number you paste2. make a call file 3. drop it in the outgoing spool directory of asteriskThis could be easily done in php - just one page. Donno if any app already exists (have heard of many, but not sure if they come alone or as part of other apps). But writing one should not take much time.
-VijOn 11/13/06, Ondrej Valousek [EMAIL PROTECTED] wrote:



  
  


Hi all,

I am interested in integrating my telephone system (I am using
hardphones and Asterisk) with my desktop - something like this:

1. someone sends me his/her phone number via email/icq
2. I cut/paste the number in some application/web page (?)
3. my phone starts ringing and when I pick it up I will get connected
with the remote party.

Now I know I have read some discussion about this possibility but I can
not recall where.
Many thanks for any point.

Ondrej




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Re: [Asterisk-Users] Memory-leak 1.2.7.1

2006-05-29 Thread Vij
May be updatedb or some other such heavy application, which runs at
night is causing heavy load on the system and spoils the working of
asterisk.

See if this phenomenon happens at the same time of the day everyday. Also, see what processes run at *that time*.
Cheers,
Vij

On 5/29/06, Attilla de Groot [EMAIL PROTECTED] wrote:
Marco Mouta wrote: I'm also not an expert, but could it as any relationship with your Telephony card drivers?? Which Telephony boards do u use?None. :)I only use Asterisk as an VoIP pbx. Only the zaptel drivers installed
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Re: [Asterisk-Users] asterisk silence suppression?

2006-05-28 Thread Vij
Hi,
 Has there been any improvements to this
patch?, what is its state now?. Has anybody tested this?. Any results?

I tried the link, seems the site is not up. Where can I download the patch from?

-Vij


On 3/3/06, Juan Salas [EMAIL PROTECTED] wrote:







I will 
try to test your adaptation.
How I 
congfigureto enable VAD?

Regards

Jsalas

  -Mensaje original-De: Moises Silva 
  [mailto:[EMAIL PROTECTED]]Enviado el: Friday, February 17, 
  2006 11:26 AMPara: Asterisk Users Mailing List - Non-Commercial 
  DiscussionAsunto:  Re: [Asterisk-Users] asterisk silence 
  suppression?
   The patch you saw is 
  not for the stable branch.
  Salu2
  Jsalas
  Right, but try using this, i adapted it, no 
  guarantees, i have not made tests, just modified it to apply properly, it 
  would be great if some one can test it:http://chewbacca.ivsol.net/asterisk-1.2.1-
silence-suppression-4.patchRegards
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Re: [Asterisk-Users] Packetization configuration of IAX channels

2006-05-24 Thread Vij
Does this mean sending multiple frames per packet not possible with IAX currently?

Thanks,
VijOn 5/24/06, Dan Austin [EMAIL PROTECTED] wrote:
Vij wrote:I have seen a few threads where people have applied packetization patch and have varied the packetizing rate of RTP/SIP and hence reducing the bandwidth required for the call.
 Is there a way to do the same with IAX?.Not at this time.There was some discussion about making thepacketizationinfrastructure be frame-based so any that IAX could make use of it, butsofar no one with the knwoledge and skills to make it happen has stepped
up. Will the tos=0x08 (highthroughput), or using the bandwidth directivehelp?That will set the ToS/DiffServ flags to give the IAX packets a priorityboost, but will not reduce bandwidth requirements.
 Thanks in advance, VijDan___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list
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[Asterisk-Users] Packetization configuration of IAX channels

2006-05-23 Thread Vij

Hi,
   I have seen a few threads where people have applied packetization
patch and have varied the packetizing rate of RTP/SIP and hence
reducing the bandwidth required for the call.

Is there a way to do the same with IAX?.
Will the tos=0x08 (highthroughput), or using the bandwidth directive help?

Thanks in advance,
Vij
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Re: [Asterisk-Users] does Jitter calculation in chan_iax2.c work???

2005-05-27 Thread Vij
SteveK and Andrew,
 Thanks a
lot for the suggestion. It helped. We didnt know that jitterbuffer wont
be enabled with sip endpoints. forcejitterbuffer=true solved the
problem.

Thanks again,
Vijay  AshishOn 5/27/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On May 27, 2005 01:47 am, Vij wrote: The above command always shows zero value for jitter. (Actually, only rtt
 and kpkts are non-zero). The behaviour is the same even for cross-continental calls.Post your iax.conf without passwords.Also, are there any native bridges going on on either side?There will be no
jitter buffer used if so.Also, there will be no jitter buffer enabled ifthe endpoints just go to another VOIP technology (e.g. to another IAX phoneor to a SIP phone).You can force it with forcejitterbuffer=yes in 
iax.conf. Is this a bug in the implementation or a configuration problem?.Honestly, you haven't even begun to give us enough information to determinethat.-A.___
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[Asterisk-Users] does Jitter calculation in chan_iax2.c work???

2005-05-26 Thread Vij
Hi,
 We are trying to get the jitter of a channel for iax channels.

iax2 show netstats

The above command always shows zero value for jitter. (Actually, only
rtt and kpkts are non-zero). The behaviour is the same even for
cross-continental calls.

Is this a bug in the implementation or a configuration problem?. 

Thanks,
Vijay  Ashish

PS:We have enabled jitterbuffer at both ends.
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