-Original Message-
From: Stephen R. Besch [mailto:[EMAIL PROTECTED]
Sent: 04 June 2003 20:09
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Budgettone 100 phone Configuration
snip
mailbox=100 ;Set to use MWI on phone
Stephen, does your MWI work? It doesn't seem to work for
Eduardo Goncalves wrote:
Hi,
I have an ansterisk and a cisco 827-4v registered to a Gatekeeper.
asterisk has two extensions:
exten = 223,1,Dial,OH323/[EMAIL PROTECTED]
Is 223 a registered alias of 827?
Have you configured this destination pattern in 827?
exten =
I'm using * as a Call Agent for two DL102S but I have some problems, like the tones not being sending from the phone to the *. I have not changed the configuration of the DL, except the IP and the Notify Entity (*). Must I change another thing in * or in the device? Thanks very much michelle
Tu
Hi,
We recently installed asterisk too replace our office PABX, however we
are finding it hard to get documentation on the way Agents login.
In the agents.conf we have setup a user of agent = 1003,,Test. In
the extentions.conf file we have added in exten = 9,1,AgentLogin.
We can dial 9 and
I have a 7960 phone I'm trying to upgrade to SIP mode. Unfortunately there
is an error in the firmware and I need to upgrade to a newer skinny first.
I'm looking for somthing like version
P00303020209.bin
I have the latest SIP, but the phone times out trying to upgrade. According
to Cisco this
Some hardware modems are able to do full duplex, and some don`t.
They
do
predomenantly however have drivers for linux available that allow
voice-modem use.
Either way - you would then need an asterisk channel for full-duplex
voice
modem.
This is the point Have someone
In h323.conf
did you add sections like
[eanders]
type=friend
context=default
incominglimit=6
outgoinglimit=4
You need the context=default
Erik
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Simon J Mudd
Sent: Wednesday, June 04, 2003 6:46 AM
To:
On Wednesday, June 04, 2003 1:25 AM, Jeremy McNamara
[SMTP:[EMAIL PROTECTED] wrote:
There is a little discussion going on comp.dcom.voice-over-ip and I
could not answer these two questions... The author has sorta
clarified his two questions, but they still mean little to me.
6) A redundant
Hi TC,
I was stating my opinion which is why I didn't provide any examples. If
you need examples here you go. I have to administer equipment remotely.
Since these devices have no network interface this is not possible. I am
forced to use a handset and key sequences whenever I need to
Actually Dan, you are mistaken here, many serial fax/data/voice modems
come with answering machine programs which tell the modem to
send the voice data through the serial line. Also the vgetty
program in linux does the same.
The problem is using the AT command set, not whether or not
I've played with modifying the extensions.conf and h323.conf but don't
have things right. I keep getting a message on the
console:
ERROR[376849]: File chan_h323.c, Line 974 (setup_incoming_call): Call from
user 'Simon' rejected due to no default context
However I am unsure what this really
Hi,
I have OH323 installed and all you need to do is to define the context in
oh323.conf file, a set of codecs and then in NM to set the gateway (in
advanced settings) to point to the Asterisk server.
Then call the extension directly.
For the rest, just define an extension like:
exten =
Steve Underwood wrote:
Mark Spencer wrote:
You could, of course, contribute the changes needed to make libpri do
Q.SIG, and everyone will benefit. :-)
What all is required for Q.SIG?
Another bunch of Q.931 type messages, basically. It builds on Q.931 to
add PBX (i.e. private network)
Don Pobanz wrote:
When I was looking at timing before this is the conclusion that I have
come to.
The T400P card has an internal clock that all four T1s of that card
will be timed off of. This internal clock can be free running (not
referenced to any other clock) or reference to another clock
Hi Andy,
Thanks for responding.
[EMAIL PROTECTED] (Andy Powell) writes:
ERROR[376849]: File chan_h323.c, Line 974 (setup_incoming_call): Call from
user 'Simon' rejected due to no default context
However I am unsure what this really means and how to configure the
extensions to allow
Hi Dan,[and thanks for bearing with me]
I'm going to top post partly as I may be making a mistake in my
assumptions. I've been mentioning using NM with h323 and editing the
/etc/asterisk/h323.conf and you mention oh323. Perhaps I've been
doing the wrong thing or am trying to configure the
Hi,
I have an ansterisk and a cisco 827-4v registered to a Gatekeeper.
asterisk has two extensions:
exten = 223,1,Dial,OH323/[EMAIL PROTECTED]
exten = 730,1,Dial(IAX/[EMAIL PROTECTED]) (IAX are working well)
When I try to call each other, gnugk shows a ARJ:
I'm using a 827-4v and specifying codec g711ulaw on the cisco device
seems to work okay
-Original Message-
From: Eduardo Goncalves [mailto:[EMAIL PROTECTED]
Sent: 04 June 2003 16:19
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] h323 and g729
Hi,
I have an ansterisk and a
On Wed, 4 Jun 2003 at 17:09, Michael Labuschke wrote:
wasn't the capi driver for the avm only for one card per pc ?
That only applies to their passive cards. The c2 is an active card
with two ISDN ports, so it should be able to handle them both at once.
cu
Reinhard
Hi All,
I'm using the Call Queue without the Agent Login to provide a company
with a Call Queue for their tech support Staff. Basically they have
several techs the work from home. I dump the calls back out the PSTN
via a SIP gateway.
With the Queue application is there a way to define an
Hi,
If Asterisk is registered to that gateway too, I think that you must dial
using only :
exten = 223,1,Dial,OH323/BYEXTENSION
If you have not installed G.729 on Asterisk, then G.729 is just passed
through Asterisk
BR,
Dan
- Original Message -
From: Eduardo Goncalves [EMAIL
Hi,
Does anyone know anything about the callerid format that NTL uses here
in Cambridge, UK. This is the former Cambridge Cable, who is
sometimes different from the rest of NTL.
They did say that only some equipment works with their switch. I
hoped that they might use US-style CID, which would
Does anyone have any suggestions on how to send different rings to a Zap
channel depending on if the call is from an internal extension or from
an outside line?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
My problem is that I cannot get call waiting to work when somebody calls
from the PSTN.
My hardware:
X100P
TDM400P
Working Scenario:
Phone A, Phone B, and Phone C are all connected to my Asterisk
box
Phone A calls Phone B
Phone B picks up and they have a
put the same extension in different contexts accessible to inside/outside
and use r2, r3, etc to differentiate.
At 01:49 PM 6/4/2003 -0500, you wrote:
Does anyone have any suggestions on how to send different rings to a Zap
channel depending on if the call is from an internal extension or from
It might be done using the chan_local channel driver,
You could add this member in queue.conf
member = local/[EMAIL PROTECTED]
and in extensions.conf
[timeout]
exten = s,1,Wait,600
exten = s,2,Voicemail,b1000
I don't know if that'll work but it's worth checking.
regards
Martin
On 4 Jun
Hello,
I created a new application for myself called Dialtone() by modifing
res/res_indications.c file. It can be used as such:
exten = s,4,Dialtone(30|${CALLERIDNUM})
exten = s,5,Playback(time-exceeded)
exten = s,6,Goto(s|1)
It will stutter if you have new voicemail and you have
I am sorry there is a mistake. The return from handle_dialtone() should be
'return res;' and NOT 'return 0;' It will not work properly without this
detail.
Jason
- Original Message -
From: Surfer Dude [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, June 04, 2003 4:16 PM
Just wanted to say that this is very encouraging, and I'd like to see
more! I've got MacOS-X, and I've been desparately waiting for a SIP
or IAX client for this system for a while. I've got your test code
running, which looks extremely promising. Alas, I am not a
programmer, and couldn't
Simon J Mudd wrote:
Hi Dan,[and thanks for bearing with me]
I'm going to top post partly as I may be making a mistake in my
assumptions. I've been mentioning using NM with h323 and editing the
/etc/asterisk/h323.conf and you mention oh323. Perhaps I've been
doing the wrong thing or am
Good evening.
I have a problem with my Xpressa phone, when i dialed from/to it i don't
get audio, my other UA are a SJPhone and XLite, i already debug it with
ethereal and tcpdump, i dialed the echo test extension from the demo
files of asterisk and is the same result, no audio/rtp coming from
Hello All.
Does anyone have experience with the Valiant Comms vcl30 channel
and the Digium E100P in asterisk ? We have the vcl30 channel bank,
loaded with FXO interfaces. We have set up * for fxs in zaptel.conf
and in zapata.conf, but are not able to get any incoming calls.
The vcl fxs
Let me be a bit more specific about my meaning:
Using a PRI interface, it is possible to get good call progress
supervision, where one can see if the far end has answered. This is
also possible with standard Zap interfaces, and certainly with SIP (I
assume H323 is also possible, but my
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