This works when I change the host to non-dynamic and
insecure=port,invite for the peer, but does not work when
host=dynamic.
Also my sip peers are realtime. If I remove the realtime peer and
create a peer in sip.conf this works !!
On Tue, May 3, 2011 at 11:15 AM, Justin Case
Hi,
As per your Dialplan MixMonitor will work after call bridge, In you case
still call is not bridge. That's why MixMonitor is waiting of call bridge...
*MixMonitor(Q-${QDIALER_QUEUE}-${UNIQUEID}.WAV,b,)
option b=** A bridge flag allows recording to only take place when the
channel is bridged.*
This will help you start:
http://www.757.org/~joat/wiki/index.php?n=Main.HomebrewAsteriskConferenceManager
On Sun, May 1, 2011 at 12:41 PM, Alec Taylor alec.tayl...@gmail.com wrote:
Good Afternoon,
I'm working on an audio conferencing web-frontend.
It'd be helpful if I could know:
•
Am 02.05.2011 15:59, schrieb A E [Gmail]:
On Mon, May 2, 2011 at 3:15 AM, A E
[Gmail] all.efor...@gmail.com
wrote:
Hello All,
Probably a silly question, but we're
On Tue, May 3, 2011 at 4:41 AM, Thorsten Göllner t...@ovm-group.com wrote:
Am 02.05.2011 15:59, schrieb A E [Gmail]:
On Mon, May 2, 2011 at 3:15 AM, A E [Gmail] all.efor...@gmail.comwrote:
Hello All,
Probably a silly question, but we're wondering if people have had any
experience and
On Tue, May 03, 2011 at 01:09:14AM -0400, A E [Gmail] wrote:
On Mon, May 2, 2011 at 9:45 PM, C F shma...@gmail.com wrote:
Just from my experience with different DBs, stay away from BLOB data
types as much as possible.
Hi CF,
any particular reason why? I've had a good experience with
Thanks, looks really helpful for managing connected users (half my problem).
On the web-interface question, how do I create a website with a
[call-in] button?
I'm using Drupal, so will make it a members only page. Basically they
click the [call-in] button, and straight away they're in the
hello List
i need to be able to record the call transferred from iax extension to sip
extension
when i call the sip extension from the IAX extension i can record the call
without any issue
but when i receive a call from customer in IAX and i transfer this call to
SIP client
the
2011/5/3 Steven Howes steve-li...@geekinter.net
On 3 May 2011, at 13:34, Olivier wrote:
It seems that Asterisk DEVICE_STATE function can't be mapped with this
bicolor feature.
So how could this 4-states BLF be implemented ?
Any suggestion ?
The handset is what maps the colours to the
I have a couple of questions about asterisk 1.6:
Can you change codecs mid-call upon re-invite?
Can you handle the SDP offer-answer in the 200-ACK instead of the usual
INVITE-200?
Thanks in advance for any insight.
Gary
--
On 05/03/2011 12:43 PM, Gary Graves wrote:
Can you change codecs mid-call upon re-invite?
Do you mean: can Asterisk be configured to _initiate_ such a change
at some point, mid-call? Or do you mean: Will Asterisk properly
react to such a re-INVITE and change codecs if asked to do so by
Can you answer both?
Can Asterisk be configured to _initiate_ such a change at some point,
mid-call?
and
Will Asterisk properly react to such a re-INVITE and change codecs if asked
to do so by the dialog counterparty?
On Tue, May 3, 2011 at 12:56 PM, Alex Balashov
On 11-04-30 03:10 PM, Alec Taylor wrote:
Good Evening,
I'm setting up an Internet Radio website with call-in functionality,
and need to know the kinds of FOSS tools I should install to get the
job done.
Here's an example of what I'm looking for: http://i56.tinypic.com/aafz4k.png
Call
Hi
Is it possible to dial from within voicemail to reach another extension.
I would like my customers to have a choice of dialing 1 to get my cell
phone while in voicemail or to just leave a message at the tone.
Thanks
Kelly
--
Hmm... it's just that I've seen implementation of this done already,
in Google Voice, BlindSide, BigBlueButton and others, however none
provide a simple interface for voice-only broadcast from the browser.
I'm sure there's a way to do it using Asterisk, I just don't know of it!
Please suggest
Hey Guys!
Anybody have basic and simple call forwarding dialplan ? I search on google and
i found many but those are pretty complicated and most are for trixbox and GUI.
-S
--
_
--
From: Kelly Opal
Sent: Tue 5/3/2011 1:57 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] dial from voicemail
Hi
Is it possible to dial from within voicemail to reach another extension.
I would like my customers to have a choice of dialing 1 to get my cell
phone while in
Kelly Opal wrote:
I would like my customers to have a choice of dialing 1 to get my cell
phone while in voicemail
I use a combination of the follow me feature, along with the 'a' extension:
[s-NOANSWER]
exten = s,1,Gosub(mailbox_exist,s,1)
exten =
I found following dialplan on net but somehow its not going to set CFIM in
asterisk database (asterisk 1.8.3.3). Any idea ?
exten = *72,1,Answer
exten = *72,2,Wait(1)
exten = *72,3,BackGround(please-enter-your)
exten = *72,4,Playback(extension)
exten = *72,5,Read(fromext,then-press-pound)
On Fri, Apr 29, 2011 at 01:04:42AM +0200, Gilles wrote:
On Wed, 27 Apr 2011 14:15:10 +0300, Ashik Ali
beaasteriskg...@gmail.com wrote:
Anybody can explain me why asterisk is unable to detect ringback tone from
PSTN telco ? .
I guess it was a lot of work, and nobody bothered adding this to
does anybody know a good tutorial on how to setup asterisk to receive faxes
(no need to send them) ? i've tried using app_fax.so with T38 but i keep
getting Transmission failed
--
_
-- Bandwidth and Colocation Provided by
You need spandsp i guess following is my dialplan is working example
[fax]
exten = 9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
exten = 9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
exten = 9000,n,ReceiveFax(${FAXFILE})
exten = 9000,n,Hangup()
Date: Tue, 3 May 2011
i have spandsp and app_fax.so is loaded but i get:
app_fax.c:820 transmit: Transmission failed
when trying to fax from a POTS line...
On Tue, May 3, 2011 at 3:27 PM, satish patel satish...@hotmail.com wrote:
You need spandsp i guess following is my dialplan is working example
[fax]
exten =
did you set faxdetect=both or incoming
and faxbuffer=?
-S
Date: Tue, 3 May 2011 15:28:36 -0400
From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
i have spandsp and app_fax.so is loaded but i get:
app_fax.c:820 transmit: Transmission
add this line at the end of the IAX account definition and try again
requirecalltoken=no
On Wed, Apr 27, 2011 at 2:40 PM, John Alexis kasteris...@gmail.com wrote:
Unfortunatelly that doesn't change anything. I got exactly the same error
(Everyone is busy/congested at this time (1:0/0/1) ...
I tried with those settings and without... same error:
WARNING[18090]: app_fax.c:820 transmit: Transmission failed
On Tue, May 3, 2011 at 3:32 PM, satish patel satish...@hotmail.com wrote:
did you set faxdetect=both or incoming
and faxbuffer=?
-S
--
Date:
I'd enable full debug at logger.conf and try to find issue.
-S
Date: Tue, 3 May 2011 15:55:51 -0400
From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
I tried with those settings and without... same error:
WARNING[18090]: app_fax.c:820
On 05/03/2011 01:16 PM, Gary Graves wrote:
Can you answer both?
Can Asterisk be configured to _initiate_ such a change at some point,
mid-call?
I don't know of a way to do that. I suppose it might be possible if a
call were asynchronously transferred to a SIP peer that had different
codec
i have full log.. only thing that stands out are two warnings:
[May 3 16:10:40] WARNING[18176] app_fax.c: Error transmitting fax.
result=13: Unexpected message received.
[May 3 16:10:40] WARNING[18176] app_fax.c: Transmission failed
On Tue, May 3, 2011 at 4:05 PM, satish patel
As I'm reading this, libpri thinks that the SV8300 is complaining that
a mandatory IE is missing, in this case time/date. However, the
field is
THERE. But when I go back to a working libpri (r1878), I see that the
time/date is NOT sent on the CONNECT.
If I'm reading Q.931 correctly, 5.1.8
Please create a mantis issue describing this problem.
Pardon my ignorance, but what does mantis refer to?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Please create a mantis issue describing this problem.
Pardon my ignorance, but what does mantis refer to?
Mantis is the issue tracker at:
https://issues.asterisk.org
Richard
--
_
-- Bandwidth and Colocation Provided by
Enable debug and verbose on CLI ?
Did you enable and also at logger.conf
full = notice,warning,error,debug,verbose,dtmf,fax
Date: Tue, 3 May 2011 16:12:06 -0400
From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
i have full log.. only
Hi All;
I need to configure the SIP account so if first IP address failed then to send
for the second IP address. How to do this?
While configuring the sip account, at the host parameter, can I give two IP
addresses separated by comma? Or what should I do to have such redundancy?
Regards
On Tue, May 3, 2011 at 5:31 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
I need to configure the SIP account so if first IP address failed then to
send for the second IP address. How to do this?
While configuring the sip account, at the host parameter, can I give two IP
addresses
On Tue, 3 May 2011 18:45:32 +
satish patel satish...@hotmail.com wrote:
I found following dialplan on net but somehow its not going to set
CFIM in asterisk database (asterisk 1.8.3.3). Any idea ?
exten = *72,1,Answer
exten = *72,2,Wait(1)
exten =
Thank you so much that solved my database issue. Now how asterisk will
forward call ?
Or I need to specify gotoif statment in my stdexten to check database
key and take action?
--
Sent from my iPhone
On May 3, 2011, at 5:41 PM, Chad Wallace cwall...@lodgingcompany.com
wrote:
On Tue,
On Tue, May 3, 2011 at 1:09 AM, A E [Gmail] all.efor...@gmail.com wrote:
On Mon, May 2, 2011 at 9:45 PM, C F shma...@gmail.com wrote:
Just from my experience with different DBs, stay away from BLOB data
types as much as possible.
Hi CF,
any particular reason why? I've had a good experience
On Mon, May 2, 2011 at 11:46 PM, || dave cantera Mobile
david.cant...@ibsonecall.com wrote:
I've been away from asterisk for a while since 1.4.16 and only installed 1.6
once to run a test... can someone recommend what the best version to install
is and the recommended CPU/motherboard for an *
C F wrote:
model name : AMD-K6(tm) 3D processor
*shudder*
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
_
--
On Tue, 3 May 2011 18:10:55 -0400
Satish Patel satish...@hotmail.com wrote:
Thank you so much that solved my database issue. Now how asterisk
will forward call ?
Or I need to specify gotoif statment in my stdexten to check
database key and take action?
Yes, you need to write the dialplan
Thanks for the input.
Yes, I did call out many times, but the recording doesn't happen even after
the call is bridged and there is two way audio. I also took out the b
option and so it should recording the ringing right (even before call is
bridged) but it doesn't do that or any recording at all.
Under 1.4.35 I get this message printed MANY times
[May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame
type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000
(g722)(4096)/0x1000 (g722)(4096)
[May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame
On 3/05/11 10:16 PM, Eduardo Leones wrote:
Guys,
I'm having problems in the fading voice calls, receptive and active,
that in SIP accounts. While few people using the system, calls are
perfect, but it beats the normal use of connections (average 30
concurrent), the voice begins to fade from
It should work when host is dynamic. Is this a bug ?
On Tue, May 3, 2011 at 9:03 AM, Deepesh D deep.d2...@gmail.com wrote:
This works when I change the host to non-dynamic and
insecure=port,invite for the peer, but does not work when
host=dynamic.
Also my sip peers are realtime. If I remove
45 matches
Mail list logo
