Displaying 20 results from an estimated 1100 matches similar to: "IAX calling number"
2003 Jul 18
16
Call Transfer
hi,
Can anybody pls tell me, how to increase the time gap between 2 digits when you transfer a call.
ie, the operator answers the call, and presses hash key to transfer, and then enters the extension
number, some times, it timeouts too quickly before the operator enters the whole extension number
(may be bcos the operator is slow).
I tried the following, but it doesn't seems to be helping
2003 Oct 13
6
Asterisk Manager
Hello all,
Can I execute linux command like(ls, mkdir) through the Manager interface?
I can't seem to access the manual at digium.com. I keep getting 'Forbidden'
error. Looks like they are upgrading or something.
CF
2006 Feb 09
1
Problems with gnugk, asterisk, and ooh323
Greetings to All,
I hope someone has already gotten this working. I spent all day today trying
to get ooh323 and gnugk to run on the same box. After a lot of tweaking to
get everything compiled, I got both up and running.
I can make calls IAX to H323, but cannot make calls in the reverse
direction. I have tried many different configs on the GK, but always come up
with the same error. It appears
2003 Aug 21
3
Conference + time limit
Hello
Conference again. Meetme can now limit number of users in a room. Can it also limit how long a conference session? Someone ask the same question (from achive) but doesn't have a solid answer.
Foong
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2005 Feb 24
2
asterisk supports VXML?
Hello,
Does asterisk supports VXML?
Couldn't find much resource on that on google and wiki.
Thanks
Foong
2003 Aug 06
1
chan_oh323 + dtmf
Hello all,
I have a cisco AS5300 which is register with a gatekeeper and a Asterisk server also register with the gatekeeper.
PSTN ---->AS5300 ---->Gatekeeper ---->Asterisk
I set up a conference room on the Asterisk sever (Room No 1234).
I try to call from PSTN to AS5300, The AS5300 will call the Asterisk server through the gatekeeper.
I manage to get to the start of the conference
2003 Aug 12
1
Conference + E100P + H323
Hello,
I have a E100P card from digium and I try to implement a conference bridge in asterisk.
I wonder since I got the E100P card do I still need to load ztdummy for caller from h323 endpoints to work with Meetme?
I load the E100P driver but i did not load the ztdummy driver. My h323 caller does not hear any voice play by Meetme.
Looks like ztdummy is required as long as h323 is concern and
2003 Sep 09
1
Dial + disconnect
Hello,
When I have the following extension:
exten => 900,1,dial(Zap/0122740900)
can I know whether 'dial' actually gets through or the called party is busy at the moment. I want to perform different action based on whether the 'dail' success or not.
Foong
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2003 Sep 22
2
Meetme Admin menu
Hello,
Is there a asterisk developer guide/source code doc or something like that?
I want to see if I can implement the admin menu function for meetme.
Foong
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2003 Oct 22
1
IAX with multiple NIC
Hello,
I have been using IAX to serve clients endpoints for a while with no
problem.
But recently, to increase the bandwidth to the Asterisk server, I add
another network interface card to my Asterisk server which connected to a
different service provider that I currently have. Both of my nic is assigned
different public ip. the client will actually choose one of these ip and
authenticate itself.
2003 Sep 22
2
G.729A + Cisco AS5300
Hello,
I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected.
The codec list show on my cisco AS5300 for g.729 are:
g729r8
g729br8
I suspect that
2003 Aug 05
4
SendDtmf
Hello all,
I am trying to use asterisk to call a local access gateway by dialing a fix number, after getting connected, the is a IVR prompt for pin number and finally the real destination number. I manage to use asterisk to dial to the gateway but have no idea how to send the pin number and destination number. This is due to asterisk only process the next ext only if dial app has terminated. My
2003 Apr 24
1
GnuGK -> Asterisk problem
Hi, i'm trying to setup Asterisk to work with GnuGK using the Openh323
channel driver.
I have a Gatekeeper that gets H.323 calls from a Cisco GW.
To this Gatekeeper I've registered some endpoints, Cisco ATA186, Snom
100, etc.
Now i want send the numbers 083xxx into Asterisk.
Easy, i'll just enter something like this into oh323.conf:
gwprefix=083
And all my calls starting with 083
2003 Aug 13
2
reload
Hello All,
I wonder is there a way where I reload asterisk on CLI without disconnect any call that is currently taken place.
Foong
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2005 Jul 14
1
LED went off after loading wct4xxp
Hello,
I have a Digium TE410P card.
I get the "knight rider" lights before the module (wct4xxp) loads, but after
the
modules are loaded I don't get any lights.
I have found the following 2 posts but still could not solve the problem
http://lists.digium.com/pipermail/asterisk-users/2004-November/075277.html
http://www.voip-info.org/tiki-index.php?page=Asterisk+TE410p+No+Interrupts
2003 Dec 07
3
FARFON lives!
Some of you have been following our progress on
http://farfon.convergence.com.pk as we blundered our way through the
development of a low-cost ethernet IP phone that does IAX and augments the
client options currently available for the kick-assterisk server.
With help from the denizens of #asterisk and kind words of advice from Mr.
Spencer and the rest of the gang ... we're proud to have
2009 Dec 22
2
getent passwd problem
Hi,
I am having a weird issue with samba where once a week approximately at the
same time users will lose connectivity,
if i run
wbinfo -u all users are displayed
wbinfo -g all groups are displayed
However running getent passwd only shows local-users, no remote users are
shown..
To fix the issue I have to change the name of my idmap config and restart
samba and winbind and everything works
2009 Aug 08
4
how to get id of other table
Hi All,
I have a doubt regarding join tables
I''m having 2 models
1)Fac
2)Cont
and both models have " has and belong to many" relationships
so there are 3 tables
1)facs
2)conts
3)conts_facs
then i''m fetching the data in controller as
@conts=Cont.find(:all])
@cfacs=Fac.all(:joins=>:conts, :select=>"facs.name")
but i dont know how to get the
2004 Sep 21
0
Asterisk + GnuGK :::: Unreachable Destination.
Hello again.
I'm stll struggling trying to terminate calls from SIP through Asterisk
and throught my H323 gateways...
Basically the call is accepted by GnuGK but then dropped with
*reason = unreachableDestination <<null>>*
I did a *debug trc 10* on GnuGK and looked at the sessions... one from
X-Lite through Asterisk... and one from OpenPhone... The one from
OpenPhone works
2004 Jun 29
5
SIP->Asterisk->GnuGK->Cisco 5300
Hi all,
I would like to call from SIP client to Asterisk then GnuGk, then Cisco 5300
to PSTN phone. Is this possible? I need simple config asterisk and gnugk.Can
somebody help me?
Ganbaa
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