Brian Capouch wrote:
What's the status of SRTP?
I remember seeing things floating around about it being under
development, but various sotto voce conversations I've had around over
the past few days would indicate that it hasn't gained much/any traction.
I'd be glad to be disabused of
El Thursday 25 October 2007 11:31:37 Tilghman Lesher escribió:
On Thursday 25 October 2007 07:40:06 Pepo wrote:
I am trying to use Asterisk as the voicemail system of the TELCO where I
work. I wanna test with 2 mail boxes ( and later with a better
machine/server I hope try with 7 ).
El Thursday 25 October 2007 11:31:37 Tilghman Lesher escribió:
On Thursday 25 October 2007 07:40:06 Pepo wrote:
I am trying to use Asterisk as the voicemail system of the TELCO where
I
work. I wanna test with 2 mail boxes ( and later with a better
machine/server I hope try with
On Saturday 27 October 2007 14:35:55 Benny Amorsen wrote:
DL == Doug Lytle [EMAIL PROTECTED] writes:
DL Michelle Dupuis wrote:
Ok - that's great. I see how the destination number will match to
the exten value, but how do I access the from number '248xxx'?
DL exten =
Lyle Giese wrote:
I had a working 1.0.x Asterisk setup using:
SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2)
Which used the short quick rings.
In Asterisk 1.4, I have tried several things, but I think the correct
syntax is:
Set(_ALERT_INFO=http://127.0.0.1/Bellcore-dr2)
On Sun, 2007-10-28 at 02:27 -0500, Pepo wrote:
El Thursday 25 October 2007 11:31:37 Tilghman Lesher escribió:
On Thursday 25 October 2007 07:40:06 Pepo wrote:
I am trying to use Asterisk as the voicemail system of the TELCO where I
work. I wanna test with 2 mail boxes ( and later with
Philipp Kempgen wrote:
Lyle Giese wrote:
I had a working 1.0.x Asterisk setup using:
SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2)
Which used the short quick rings.
In Asterisk 1.4, I have tried several things, but I think the correct
syntax is:
Ah jeez. All I wanted to do was connect to a carrier and then perform fail over
logic based on their SIP response.
Not supposed to be difficult. This is what Asterisk is supposed to be good at.
We have a SIP module, why not have SIP responses available to the module.
Now, I have to look at the
Richard Lyman wrote:
Steve Totaro wrote:
Richard Lyman wrote:
Steve Totaro wrote:
I need to create a couple of tie lines between a legacy system and an
Asterisk system. I was told that the tie lines are E4 Superframe EM.
I have done EM wink but have no idea
We are planning a very large Asterisk deployment, using Wifi SIP phones.
We've done installs using Spectralink and the SVP to manage congestion at
the access points, but we have a client that doesn't want Spectralinks.
Anyone have experience with an alternative congestion management (AP
Steve Totaro wrote:
Richard Lyman wrote:
Steve Totaro wrote:
Richard Lyman wrote:
Steve Totaro wrote:
I need to create a couple of tie lines between a legacy system and an
Asterisk system. I was told that the tie lines are E4
On Sun, 28 Oct 2007, arkda wrote:
I've been looking around for an example of a method of reading back a caller
ID value, but I haven't found anything that doesn't use Festival. I'd rather
not resort to the Mr. Roboto voice if I can avoid it.
Playback of the numbers one at a time is perfectly
Knew there must have been an easier way and something I was missing. Thanks
Anthony.
On 10/28/07, Anthony Messina [EMAIL PROTECTED] wrote:
On Saturday 27 October 2007 11:19:05 pm arkda wrote:
I've been looking around for an example of a method of reading back a
caller ID value, but I
Hi all,
I was trying to change some of the sound files for the meet me conference
application, the one where the user is waiting in the conference with the
users waiting in to join (the M option-- enable music on hold when the
conference has a single caller) Also what is the name of this sound
Steve Totaro wrote:
Richard Lyman wrote:
Steve Totaro wrote:
Richard Lyman wrote:
Steve Totaro wrote:
I need to create a couple of tie lines between a legacy system and an
Asterisk system. I was told that the tie lines are E4
*snipped
Just for the record. I had to reconfigure the Sangoma startup scripts
even though my zap files were correct with d4,ami.
The real gotcha was in the cabling. One crossover was 1-5, 2-4, the
other 1-4, 2-5 and then straight through all required for the same
dialer. I have seen
Hi
Is there a way to program #700 to one of the speed dial buttons. I
put #700 in the config file for the last button but it's like pressing a
dead button. Is there some trick to using the #key to transfer in speed
dial buttons.
Thanks
Kelly
___
Hi,
I'm trying to have a SER machine send calls to an Asterisk server
working as an IVR. I was able to do this part just fine. Also, when
the caller makes certain options in the IVR, the call is then
transferred to an extension via SER. This part is also just fine.
However, I'm trying to
On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
I'm trying to have a SER machine send calls to an Asterisk server
working as an IVR. I was able to do this part just fine. Also, when
the caller makes certain options in the IVR, the call is then
transferred to an extension via
I am usesing this stanz for channel monitoring
;---call monitoring-
exten = 996,1,Answer()
exten = 996,2,Wait(1)
exten = 996,3,ChanSpy(SIP/,q)
exten = 996,4,Hangup
This is working fine in my case I have asterisk 1.4.x
Sanspareils Greenlans [EMAIL PROTECTED] wrote: Sir,
I have
On 10/29/07, Arpit Mehta [EMAIL PROTECTED] wrote:
Hi all,
I was trying to change some of the sound files for the meet me conference
application, the one where the user is waiting in the conference with the
users waiting in to join (the M option-- enable music on hold when the
conference has
On 10/27/07, wassim darwish [EMAIL PROTECTED] wrote:
Hi:
Iam using an asterisk server with astcc ,iam facing a problem with astcc
that when the call is hangup sometimes astcc doesnt calculate the call cost
and the call time and without writing the call status on cdrs table .
I tried to
On 10/27/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi Pablo;
How the IP address will be wrong, and asterisk able to
do registeration on the destination?
If the IP address wrong, so I will not be able to
register on that IP address.
Hi
i see 2 causes
1. it could be Dialplan issue (
search , firewall, and confiration the software. the configuration the user
is bad ???
use asterisk2billing it is good
On 10/28/07, ram [EMAIL PROTECTED] wrote:
On 10/27/07, wassim darwish [EMAIL PROTECTED] wrote:
Hi:
Iam using an asterisk server with astcc ,iam facing a problem
Here are more details:
The phone and the Asterisk box are behind the same router (the Asterisk
machine is 192.168.0.2 and the phone is 192.168.0.4).
A ping command works:
[EMAIL PROTECTED]:~$ ping -c 10 192.168.0.4
PING 192.168.0.4 (192.168.0.4) 56(84) bytes of data.
64 bytes from 192.168.0.4:
Hello all,
i am Presently working on integration of
asterisk and openser
i have a doubt regarding the asterisk .
if you take openser when users register it stores the users
in location table whether the users running behind NAT or on global ips
and when comes to asterisk where does it store
On 10/29/07, srinivas Antarvedi [EMAIL PROTECTED] wrote:
Hello all,
i am Presently working on integration of
asterisk and openser
i have a doubt regarding the asterisk .
if you take openser when users register it stores the users
in location table whether the users running behind NAT
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