Hello again!
So I tried again, experimented a bit more and got this:
channel originate sip/e...@iptel.org
[Jun 5 12:14:29] WARNING[8537]: chan_sip.c:17882 handle_response_invite:
Re-invite to non-existing call leg on other UA. SIP dialog
'3b39b40240b6126a61c7ad16108be...@91.58.24.59'. Giving
Depending upon what you want to do, you could add ** to features.conf e.g.:
[featuremap]
disconnect = **
Hugh
2010/6/4 Anahi Ludueña a_ludu...@hotmail.com
Hi people, I need to detect when the user presses twice *...
In the dialplan I added the following, but it doesn't work.
Could you help
Asterisk 1.6
CentOS 5.0
All -
I'd like to offer my users the ability to hangup a call by pressing **. I'm
using an attendant, so when ** is dialled I'd like processing to return to
the attendant so the user can place a subsequent call. I have setup
features.conf to include:
[featuremap]
Asterisk 1.6
CentOS 5.0
All -
I'd like to offer my users the ability to hangup a call by pressing **. I'm
using an attendant, so when ** is dialled I'd like processing to return to
the attendant so the user can place a subsequent call. I have setup
features.conf to include:
[featuremap]
Hello list,
using asterisk 1.4.30 and trying GROUP() and GROUP_COUNT() for the first
time... Having some troubles.
This the dialplan (using a sub) :
exten = s,n,Set(_custID=${custID})
exten = s,n,GROUP(${custID})
exten = s,n,NoOp(grouppcount = GROUP_COUNT(${custID}))
exten = s,n,GoToIf($[
On Thu, 03 Jun 2010 23:04:45 +0200, Hans Witvliet wrote:
Just curious,
Any chance of using amr for asterisk?
http://en.wikipedia.org/wiki/Adaptive_Multi-Rate_audio_codec
The codecs (both wb and nb) seems to be available at packman:
On 6/5/2010 10:13 AM, Jonas Kellens wrote:
Hello list,
using asterisk 1.4.30 and trying GROUP() and GROUP_COUNT() for the
first time... Having some troubles.
This the dialplan (using a sub) :
exten = s,n,Set(_custID=${custID})
exten = s,n,GROUP(${custID})
exten = s,n,NoOp(grouppcount =
I made your adjustments, but still the same result .
dialplan :
exten = s,n,Set(GROUP()=${custID})
exten = s,n,NoOp(This channel is member of group: ${GROUP()})
exten = s,n,GROUP()
exten = s,n,NoOp(groepcount = GROUP_COUNT(${custID}))
The CLI shows :
[Jun 5 16:50:04] -- Executing
I am sorry I did not get back to you last night - I was so tired!
I don't have such a file, instead I have cdr_mysql.conf which does
connect. same thing I think - perhaps a different flavour of the module.
If you type asterisk -rx show modules | grep mysql at the command prompt, not
the CLI,
I'm getting a crash relating to this field and I'm missing something.
It seems to be initialized to zero, then used in memmove, then
DECREMENTED. Where is it ever incremented?
--
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Hello list.
The VoiceMailMain()-application has an advanced menu. Can I get a
Voicemail-application that has less functionality ?
I only want the user to listen to new voicemail-messages (and delete
them), not the functionality with the folders and redirecting messages
to other mailboxes...
I installed a queue for a client with 10 officers so far so good. Now the
client wants an agent when making a call to this will leave any customer
information using the phone as a key.
I'm trying to do this app using delphi obviously I will need to connect to the
AMI, but do not quite
I hope that this simple example will clarify things a bit, so is an example of
a test; trying to count incoming and outgoing in one trunk.
exten = _9.,n,Set(GROUP(${TRUNK})=${EXTEN)
Example:
exten = _9.,n,Set(GROUP(COMP1)=${EXTEN:1})
exten = _9.,n,Set(GROUP(COMP2)=${CALLERID(num)})
exten =
Hello everyone!
I still am not much further along with my sip calling. I changed my sip.conf
taking suggestions from the net (voip-info.org in particular). I changed
iptel's position from friend to peer. I turned on and off nat, I chose
different codecs in first place, entered my outward IP
On Sat, Jun 5, 2010 at 6:26 PM, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello list.
The VoiceMailMain()-application has an advanced menu. Can I get a
Voicemail-application that has less functionality ?
I only want the user to listen to new voicemail-messages (and delete them),
not the
Julien Claassen wrote:
Hello everyone!
I still am not much further along with my sip calling. I changed my
sip.conf
taking suggestions from the net (voip-info.org in particular). I changed
iptel's position from friend to peer. I turned on and off nat, I chose
different codecs in first
At 01:16 PM 6/5/2010, you wrote:
Please can someone help me clear up this mess. I'm completely
frustrated and
don't know what else to do, where else to look.
I've always forwarded port 5060 and all the RTP ports, in my case
16000-16100, directly to my Asterisk box and I've never had
Hello Lyle!
Thanks for your answer!
I don't know, if the server sees me at the local-ip or not. I only know,
that I'm able to register at iptel.org successfully. So asterisk tells me.
I believe my router is a Samsung router 3010 phone SL. Samsung it tells me,
the rest I had to search on
Hello Ira!
I will have a look at my rtp.conf and change the rtp-port range there. As to
forwarding: Well it remains to be seen - pardon the pun - if I can find
someone willing and patient enough to be my pair of eyes. :-)
Kindly yours
Julien
Music was my first love
Håkon Nessjøen haa...@avelia.no writes:
But for a few years ago, I did some testing with Local/ channels, and they
seemed somewhat unstable in large quantity.
Are they more safe now? Is it safe to use local channels with the /n
modifier as queue members? (i need the n modifier to be able to
Hello folks,
I want to write an AGI script doing this:
1-user call a number.
2-asterisk call the agi script
3-the script dial the peer
4-if the call is answered, let the call up for 1min
5-then the script hangs up the channel.
I tried either in php or in java but no success.
In java i did
Is this bug alive in 1.6.2.9-rc1? I'm getting segfaults from chan_dahdi.
If it does effect 1.6.2.8-rc1, I'll just wait for rc2 to see if this
is my problem, instead of filing.
sean
Alec Davis wrote:
I filed the following bug on the 28th of May.
0017371: [patch] [regression] DAHDI analog
On 6/5/2010 11:37 AM, Jonas Kellens wrote:
I made your adjustments, but still the same result .
dialplan :
exten = s,n,Set(GROUP()=${custID})
exten = s,n,NoOp(This channel is member of group: ${GROUP()})
exten = s,n,GROUP()
GROUP is not an application. You cannot call it like that. Try to
I am trying to help a guy out with his Atcom IP04. He has set it up like this.
He has a handful of IP phones all connecting via SIP. He has two phone
lines connected to the FXO ports one from telecom, another from
vodaphone. He has set up the dialplan so that one of the trunks fails
over to the
On Sat, 5 Jun 2010, Adil Zaaraoui wrote:
I want to write an AGI script doing this:
1-user call a number.
2-asterisk call the agi script
3-the script dial the peer
4-if the call is answered, let the call up for 1min
5-then the script hangs up the channel.
I tried either in php or in
Is this bug alive in 1.6.2.9-rc1? I'm getting segfaults from chan_dahdi.
If it does effect 1.6.2.8-rc1, I'll just wait for rc2 to see if this
is my problem, instead of filing.
I reported another instance of this today and it was fixed in the SVN a few
hours later.
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