Hi all,
I'm testing chan_mobile for a couple of months and I'm facing some
instability problems.
I would appreciate if somebody could help me with these issues:
- after a call the bluetooth connection disconnects;
- when I make a outgoing call and the other side answer, sometimes
asterisk is
Hi,
I've a DID number that gets passed to three internal phones and a cell
phone via my outbound IAX trunk. If the cell phone is off or out of
coverage, its voice mail captures the call.
What's the best way to avoid this? Is there a recommended way to force
the cell phone user to press 1 before
You can dial the cell like this
Dial(DAHDI/1c/w5551212) instead of
Dial(DAHDI/1/w5551212)
The 'c' makes the other end press 1 to start the call.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell Horn
Hi all,
I would like to ask please how to configure asterisk in order to unforce
rtp traffic to pass through it and send them to a separate RTp proxy?
Regards
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asterisk-users
On Thu, Nov 5, 2009 at 3:51 PM, Danny Nicholas da...@debsinc.com wrote:
You can dial the cell like this
Dial(DAHDI/1c/w5551212) instead of
Dial(DAHDI/1/w5551212)
Danny - thanks, however I think that's a feature of DAHDI. My outbound
trunk is IAX.
I don't think that's a standard feature of
Hi All,
I was actually trying to use the dialplan application that uses 'Dial' and
when the: Dial(SIP/xxx...@|20|) command is executed and the
destination number rings for 20 sec after which I receive as 503 Service
Unavailable, but not 480 Temporarily unavailable.
Hi,
I've noticed that my MeetMe install seems to think chan_dahdi is missing:
app_meetme.c: No DAHDI channel available for conference, user
introduction disabled (is chan_dahdi loaded?)
However, it definitely is since I have 3 PRIs functioning normally :)
Is there anything I should check before
On Thu, Nov 5, 2009 at 6:27 AM, Vieri rentor...@yahoo.com wrote:
Despite the simpler setup, the faxes don't come in.
From the logs I can see that Asterisk receives fax calls and dials the
iaxmodem (on localhost). However, no data is transmitted according to Hylafax.
Modify your dialplan to
can you define are not working?
I just tried it on my cell phone and doesn't work either. Probably
because ATT didn't define them.
2009/11/5 Torintino T torinti...@hotmail.com:
I found and *65 are not working.
Please how can i re-enable them again.
Thanks
Dear list,
I have problems with DISA on an specific server with Asterisk 1.4.26.2.
After starting DISA I can only press one key and DISA is jumping direct
into the context without waiting for further digits.
In dtmf.log I found this:
[Nov 6 00:09:28] DTMF[2413] channel.c: DTMF begin '7'
Ott Rose wrote:
I have question thats not really about astrisk but I figure you guys are
doing this sort of thing.
We use Aastra 6757i phones. there is some support for XML. the question
is how would i go about learning to customize these phones?
Read the manual on the Aastra website.
Thank you Tarek!
I now know is possible.. i was a little confuse about it. Your
response gave some guidance.. all i have to do now is get the details.
I will be using A2Billing for this.
Regards,
Carlos C
On Nov 5, 2009, at 6:48 AM, Tarek Sawah wrote:
If you are a FreePBX user then i
Hello,
Yeah i will be using asterisk and will be getting a Core 2 Due for the
production server.
Thanks,
Carlos C.
On Nov 5, 2009, at 8:21 AM, B.Masoud @ SH wrote:
Hello,
I am doing termination for about a year, I have used quintum 24
ports for termination, compared to asterisk with
Russell Horn wrote:
Hi,
I've a DID number that gets passed to three internal phones and a cell
phone via my outbound IAX trunk. If the cell phone is off or out of
coverage, its voice mail captures the call.
What's the best way to avoid this? Is there a recommended way to force
the cell
hi all,
i have started installing asterisk on CENTOS 5.3. i have to
install these three under /usr/src
asterisk-current.tar.gz
libpri-current.tar.gz
zaptel-current.tar.gz
now i had installed
ASTERISK-1.6.1.9.TAR.GZ
LIBPRI-1.4.10.2.TAR.GZ
ZAPTEL IS PENDING
guys plz suggest me
Hello again Asterisk people.
I am running Asterisk 1.42 on an old PowerPC ibook. I have had this
deployed for several years now, with pretty good results.
Recently I added a callerid service to my landline (qwest).
I am using the audiocodes MP114 2fxo/2fxs gateway, which is an
outstanding
On 6/11/09 3:25 PM, Darrick Hartman wrote:
Russell Horn wrote:
Hi,
I've a DID number that gets passed to three internal phones and a cell
phone via my outbound IAX trunk. If the cell phone is off or out of
coverage, its voice mail captures the call.
What's the best way to avoid this? Is
On 6/11/09 3:37 AM, Antony Stone wrote:
On Thursday 05 November 2009 14:28, Danny Nicholas wrote:
Hi.
I have several Asterisk 1.4.21 machines, each with ISDN cards in them, and
Polycom SIP phones on people's desks.
I'm trying to work out how to provide a remote pickup facility along the
On 26/10/09 3:47 AM, Lukasz Pakula wrote:
Dear all,
I'm trying to install Asterisk-stat (ASTERISK CDR ANALYSER) following:
http://www.voip-info.org/wiki/index.php?page=Asterisk+CDR+Areski+GUI
however it fails to run properly - lots of lines like:
*Notice*: Undefined variable: s in
On 23/10/09 6:11 AM, jonas kellens wrote:
On Thu, 2009-10-22 at 13:45 +1300, Matt Riddell wrote:
It's really simple you just read from standard input and write to
standard output.
If you tell us a programming language you'd like to use (i.e.
php/c/perl/bash etc) we can give you a link to
On 27/10/09 2:07 AM, Steve Davies wrote:
Hi,
First a confession - The box in question is a 1.2.35 box, so this may
be solved in a newer version as I know the JB code is all hugely
changed, but... It may be worth checking into.
Scenario:
- IAX outbound call from Asterisk, which rings okay.
On 25/10/09 11:52 AM, Paul Hales wrote:
I have used both misdn and dahdi_bri over the last year, and would happy
take dahdi if for no other reason that it's much easier to install.
A patch is available to allow dahdi_bri to work with Asterisk 1.4, and I
have used that successfully.
Ooh
On 25/10/09 11:52 AM, Paul Hales wrote:
I have used both misdn and dahdi_bri over the last year, and would happy
take dahdi if for no other reason that it's much easier to install.
A patch is available to allow dahdi_bri to work with Asterisk 1.4, and I
have used that successfully.
Which
On 5/11/09 9:14 PM, Stefan Schmidt wrote:
Hello,
i use sendjabber notifications when a call is answered to send the
answering user information about the caller also with links to our CRM
or ticket system.
My problem is that i dont know how i can make a link like CRM and not
have to use
Hi all,
I'm trying to set up an odbc connection to a ms-sql server from an
asterisk 1.6.1 install
My problem is that I cannot get asterisk to build func_odbc
res_odbc.so
I installed yum -y install unixODBC unixODBC-devel libtool-ltdl
libtool-ltdl-devel
And then went on to reconfigure /
2009/11/6 Neeraj Chand neeraj.ch...@ocis.com.au
Hi all,
I'm trying to set up an odbc connection to a ms-sql server from an
asterisk 1.6.1 install
My problem is that I cannot get asterisk to build func_odbc
res_odbc.so
I installed yum -y install unixODBC unixODBC-devel libtool-ltdl
hello,
I'm using Asterisk 1.6.0.5 . And I'm creating IVR, and I need that Read
application accepts # sign,
So is it possible? And maybe there is a workaround?
Thanks
--
Pagarbiai / Best Regards,
Giedrius
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-- Bandwidth and Colocation Provided by
Gotcha! Missed libtool! :)
-Original Message-
From: Neeraj Chand
Sent: Friday, 6 November 2009 6:43 PM
To: 'asterisk-users@lists.digium.com'
Subject: RE: odbc to ms-sql server
Hi all,
I'm trying to set up an odbc connection to a ms-sql server from an
asterisk 1.6.1 install
My
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