Displaying 20 results from an estimated 400 matches similar to: "SIP group management"
2006 Nov 20
2
Recording g729
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<font face="Helvetica, Arial, sans-serif">Before ordering I want to be
2003 Dec 24
5
Sip phones on the same extension?
Hello. I'm a new Asterisk user, but I'm impressed with the
flexibility and versatility of Asterisk, and am moving quickly to adopt
it's main-line use in our company. Hopefully, you'll be hearing more from
me as the project moves forward.
Right now, though, I have a question about SIP peer registration.
Right now, for our SIP-based phone,s, we're using the Sip Express Router
2004 Jul 17
1
Using a group variable for a group ofextension to dial
That could be it. What I want to do is set a group of callers and have
the event cause the phone to ring them in order. I will tie it to my
IVR portion and thus I can make sure peole in sales get calls based on
our hierarchy in the office. So if I am reading your example right the
syntax is....
Exten => 501,1,Dial(SIP/PHONE1&SIP/PHONE2&SIP/PHONE3), rtf)
Is that a valid way to cause
2006 Jan 20
1
2400P Pinouts
Hi,
this is probably a stupid question but can someone point me to where I
can find the pinouts for the connector at the back of the TDM2400P card
from Digium.
I am assuming this is some kind of telecomms standard but I need to
connect to 8 existing analogue lines and therefor need to wire this
connector to a patch panel or R11 jacks.
Also most of the docs do not mention this card - am I
2006 Jun 07
1
Delay on calls
Hi,
I have a 1.2.4 * box with two HFC modems using chan_modem_i4l and
several SIP phones and ATA's.
We have a terrible delay on calls between the PSTN (isdn BRI) and the
SIP phones. All internal calls are fine. My first thought was that the
transcoding could cause the delay but all of the SIP phones default to
ulaw so there should not be any transcoding needed.
I also checked the load
2005 Mar 28
3
can a sip.conf stanza be shared by several phones?
Hi,
If several phones register to the same sip.conf section what will happen
with a "Dial SIP/shared" in asterisk?
All phones ringing and the first one to answer gets the call?
Undefined behavior?
Thanks,
--
Jesus is coming! Everyone look busy!
2008 Dec 29
3
Join empty queue property
I want the callers don't join in a queue when the agents are busy.
I suposse it is easy but i can't get the solution for this.
Can you suggest me something?
Thanks.
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2006 Jan 19
3
Processor Size
Can someone give me an idea of the processor power I will need for 1 x
TDM240 with 2xquad FXO's and 8 sip phones/ATA's on a quite 100Mbit LAN.
The machine we have available of hand is a P4 1GHz with 768MB RAM.
Tx
Marnus van Niekerk
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas Alva Edison - Inventor of 1093
2004 Jul 17
1
Using a group variable for a groupofextension to dial
Actually doing both sounds good to me. Can you explain further about
ringing them all at once?
Here is how I tried to make mine work and failed...
{global}
PHONES0=SIP/2000
PHONES1=SIP/2001
[local]
exten => 6001,1,Dial(${PHONES0&PHONES1),20,trf)
When I dial 6001 I see my debugger tell me that I am using the wrong
syntax.
Do you know the correct syntax for ringing them all at once?
I
2006 Nov 17
5
spc.exe
Does anyone have a copy of spc.exe they could send me?
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2006 Apr 24
1
1.2.4/7 and chan_modem
Hi,
I am currently running several * boxes on 1.0.9 with HFC chipset ISDN
modems using i4l's hisax driver and chan_modem.
Will I be able to use my existing chan_modem setup with 1.2.4 or 1.2.7
or will I need to change it to use bristuff or chan_capi?
I want to do the upgrade with as little changes as possible.
Thank you
Marnus van Niekerk
--
"Opportunity is missed by most people
2006 Dec 06
1
Ping
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<font face="Helvetica, Arial, sans-serif">Sorry to do this but I sent a
couple of posts and I do not
2007 Feb 22
1
GotoIf DURATION
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<font face="Helvetica, Arial, sans-serif">Hi,<br>
<br>
I am
2006 Jan 31
1
Leftover sound on isdn modem channel
Hi,
I have a strange problem on some isdn modem channels. (* 1.0.9 /
chan_modem / 2xHFC-S cards).
Everything works fine but when the 2nd (and 3rd etc..) call comes in and
* answers and there is about a 1/2 second of sound from the previous
call (ivr) before the sound from the new call is heard. It just sounds
bad and is quite annoying.
I am assuming this is sound that is still in a buffer
2006 Nov 28
2
Symbian Softphone
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<font face="Helvetica, Arial, sans-serif">Anybody know of a SIP/IAX
2006 Jun 01
2
Large Asterisk System
Hey Guys
I've been browsing the list looking for more information on asterisk
behavior for large system.
As for now I've got a project with
300 SIP Extensions to start, with future growth (scalability)
Capability of recording all extensions simultaneously during peak time.
And keeping the call recording for 30 days.
SIP Calls being terminated over a Cisco 5400 Gateway
Extra 100(+) FXO
2006 Mar 28
2
Problems Configuring Cisco 12SP+
Hi,
After reading this valuable forum and the voip-info wiki and follow
all the steps , but my Cisco 12SP+ remains unregistered.
These are my config files:
skinny.conf
[general]
port = 2000 ; Port to bind to, default tcp/2000
bindaddr = 172.20.1.1 ; Address to bind to
dateFormat = D-M-Y ; M,D,Y in any order (5 chars max)
keepAlive = 120
languaje=es
allow = all
; disallow
2018 Dec 07
4
how to use a database
On 12/07/2018 03:36 PM, Administrator TOOTAI wrote:
> Le 07/12/2018 à 14:32, hw a écrit :
>
> [...]
>>
>> Queues seem to be the only way to have several phones ring at once, or
>> are there other ways?
>
> Dial(SIP/Phone1&SIP/Phone2&...&SIP/Phonex,,)
>
Good to know, thanks!
What are the entries needed in the queue_members table when using
2005 Sep 13
1
wctdm, issue w/outbound calls
Hi all,
I've been running Asterisk with a TDM400P for about 6months, no problems.
2 in/outgoing analog lines, one analog phone. Recently I was messing with
the XTEN client, got to finagling with things, and not knowing what was
wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was
testing various things, and found everything worked except outgoing calls.
So I checked
2007 Nov 19
5
shared memory and event channel
Hi,
For each domUs there is unique shared memory(2-way circular queue) and event-channel(one shared memory and event-channel per domU)
or there is only one shared memory and interdomain event-channel(for every DomU)?
regards:
Amit
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