it shows empty string
On 4/3/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
You can use the following to display what you receive from user (dtmf):
exten= 1,1,Read(test)
exten= 1,2,NoOp(DTMF Received: $test)
exten= 1,3,Hangup
On 4/3/07, Nitin Gupta [EMAIL PROTECTED] wrote:
I upgraded to 1.4.1
I upgraded to 1.4.1 and my DTMF has stopped working, I tried
rfc2833compensate=yes and relaxdtmf=yes etc but none working.
Everything seems to work fine with 1.2.10
Is there any way I dump the dtmf data packets received by asterisk on
console?
Any idea or pointers to debug the issue will be
Hi,
Is there any way asterisk can detect if the outgoing call is being received by a user or it has been forwarded to his voicemail.
Actuallymy requirement is to proceed only if user picks up the phone otherwiseto hangup as soon as the call goes to voicemail.
Is there anyway to do this?
Thanks
thanks Dovid, infact I just got things recorded from her.
Nitin
On 8/27/06, Dovid Bender [EMAIL PROTECTED] wrote:
snip
2) What are the best sources (cost effective) to get prompts recorded.
/snip
I would go with allison. She is the one that did all the voice files that
you currently have on
Hi,
this question may sound a little dumb, but I need opinion of who are
already using asterisk.
questions are :
1) which format is best suited for asterisk (.gsm, .wav etc, also what
sampling rate and bit size)
2) What are the best sources (cost effective) to get prompts recorded.
thanks in
thanks a lot David, its really useful
after googling I found one more link on ramfs http://www.linuxfocus.org/English/July2001/article210.shtml
thought this can be useful for others.
Nitin
On 8/18/06, David Gagnon [EMAIL PROTECTED] wrote:
Hi, Take a look at ramfs
Hi,
Is there any option in asterisk to load all the prompt files into
memory on startup, so that it doesn;t have to hit the disk to read
prompts for any call.
Or any plugin / suggestion to avoid hitting the disk for prompt files?
Thanks in advance.
Nitin
Hi,
did anyone try do load-testing on asterisk,for sip channel calls?
I want to have a rough estimate about - how many calls, an asterisk server, running on say dual 240 opteron with 1 GB memory, can handle?
Also how much internet bandwidth does a typical call requires? I heard around 20Kbps with
Hi,
I need to monitoran asterisk server, so planning to use some tools which can initiate call to a number (for asterisk server)periodically and can interpret the response, is anything as such already available?? or any pointer??
thanks in advance
Nitin
Hi,
Is there anyway in Asterisk to know that outgoing call has been forwarded to voicemail by the callee system?
Someof my users don't want to connect the call if its forwarded to callee voicemail, so I am wondering if theres anyway to identify this in Asterisk and drop the call.
Thanks
Nitin
Thanks for the information, I will surely look into it!
Nitin
On 5/10/06, Kerry Garrison [EMAIL PROTECTED] wrote:
Have you looked at CBeyond? I like their T1 SIPConnect product.
From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] On Behalf Of Nitin GuptaSent: Wednesday, May 10, 2006 7:04 PM
Hi,
I am looking for voip providers in bay area, any suggestions?
My requirements are:
handling around 2000 calls a day (incoming) and around 1000 calls a day outgoing. I have a Asterisk PBX server to take care of routing calls to appropriate deparment. So I am looking mainly for IAX2 or SIP
thanks for the information Peter, its really helpful. Also Ihave one more question - do you have any idea how many such simultaneous calls can an asterisk server handle (say running od 2.6Ghz, 1GB Ram, fedora machine)?
Thanks,
Nitin
On 2/13/06, Peter Fern [EMAIL PROTECTED] wrote:
You can enable
Hi I was wondering if its possible to make Dial commandbridge two channels and after bridgingbypass asterisk, so that the voice doesn't need to pass through my asterisk server.
For e.g., I have a user dialed in and he verifies himself and then dials an international extension, after the call
Sorry for re-posting this message -
Iam trying to run the latest stable Asterix version 1.2.4. on 64 bit amd procesor.
Things are working but the playback sounds that I hear when tring to connect over IAX are of very high frequency.
i.e a sentence which should finish in 4 secs finishes in much
sorry can you elaborate a little, what exactly is timing issue?
Thanks
On 2/12/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Feb 12, 2006, at 1:05 AM, Nitin Gupta wrote: Sorry for re-posting this message - Iam trying to run the latest stable Asterix version
1.2.4. on 64 bit amd procesor. Things
willing to debug things further??
Nitin
On 2/12/06, Nitin Gupta [EMAIL PROTECTED] wrote:
sorry can you elaborate a little, what exactly is timing issue?
Thanks
On 2/12/06, Martin Joseph [EMAIL PROTECTED]
wrote:
On Feb 12, 2006, at 1:05 AM, Nitin Gupta wrote: Sorry for re-posting this message -
Iam
got the answer the gettimeofday() is twice as fast as the one in older box, problem with system clock.
Nitin
On 2/12/06, Nitin Gupta [EMAIL PROTECTED] wrote:
well if I pass the parameter aswhennext/4 instead of whennext/8 in file.c = ast_readaudio_callback()= ast_sched_add(.,whennext/4
:
Hi,how did you come to that? :)How did you fix it?Regards, TamasNitin Gupta wrote:
got the answer the gettimeofday() is twice as fast as the one in older box, problem with system clock. Nitin On 2/12/06, *Nitin Gupta*
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: well if I pass
Iam trying to run the latest stable Asterix version 1.2.4. on 64 bit amd procesor.
Things are working but the playback sounds that I hear when tring to connect over IAX are of very high frequency.
i.e a sentece which shoudl finish in 4 secs finishes in much lesser time. Where can be the problem?
20 matches
Mail list logo
