Hi,
From: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] LipZ4 Sip Soft Phone
Date: Sat, 20 Mar 2004 09:33:42 -0500 (EST)
Thanks a lot I might give it a try. Any specific instructions for running
it with asterisk?
AJ
Checkout these urls, these might be of your interest:
http://www.zultys.com/pro
At 09:34 PM 3/21/2004 -0500, you wrote:
>On Sun, 2004-03-21 at 20:48, Jon Myers wrote:
>
>> "Online" since 1985 (I know, not longer than alot of prople, but more
>> than a couple years).
>
>But apparently not long enough to know that top posting and not trimming
>quotes are both just as bad as repl
If you have your IVR under context [mainmenu] and your extensions under
context [default]. Then make sure you include context default under
context mainmenu...
Because your mainmenu context does not know about any other extensions
if you don't.
-gcc
-Original Message-
From: [EMAIL PROT
David Krider [EMAIL PROTECTED] wrote:
> On Sun, 2004-03-21 at 20:48, Jon Myers wrote:
> > "Online" since 1985 (I know, not longer than alot of prople, but more
> > than a couple years).
> >
> But apparently not long enough to know that top posting and not trimming
> quotes are both just as bad as r
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Damian Dicks
> Sent: Sunday, March 21, 2004 9:18 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Home users
>
>
> I am trying to setup the following scenario.
[...]
> there is nothing. I do
> >From the 7960 at my home I get connected. I can then call any other
> phone in the office and call outside calls. The problem is as soon as
> someone picks up their office phone there is dead silence. The office
> phone can call my home phone and it rings and again when I pick up the
> home p
Hello,
I own 2 7905G phones running sip. You can download the image from CCO.
It is installed by TFTP, you just specify the server inside the menu,
reboot the phone and if you have the image and the config file with the
image in the root directory it will install the new OS.
Regards,
Matt
You're positive the 7905G supports SIP? How did you upgrade it? Just a
TFTP server?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Sunday, March 21, 2004 9:08 PM
To: Asterisk Users
Subject: Re: [Asterisk-Users] Cisco 7960 vs 7905
I s
On Sun, 2004-03-21 at 20:48, Jon Myers wrote:
> "Online" since 1985 (I know, not longer than alot of prople, but more
> than a couple years).
But apparently not long enough to know that top posting and not trimming
quotes are both just as bad as reply-to-sender.
;-)
dk
___
I am trying to setup the following scenario.
7960 --- Linksys firewall Internet Firewall Linux server
7960
Home Running
Office
Asterisk
>From the 7960 at my home I get connected. I can then call any other
On Sun, 2004-03-21 at 16:05, Jerry Geis wrote:
> All,
>
> I am looking for a way to have my AGI startup on a channel
> automatically when asterisk starts. Is this possible?
>
> I have my AGI working for when a call comes in - however I
> would like the AGI started up automatically with asterisk o
I stand corrected. I assume that the info at
http://www.cisco.com/en/US/products/hw/phones/ps379/products_qanda_item09186a008010a826.shtml
is referring to First Customer Ship, rather than current, since it lists
the 7905G as not supporting SIP. And I KNOW the 7905G supports SIP. I
was using one l
Strongly Agree :)
WW
- Original Message Follows -
> The only thing I hate more than not having a proper
> reply-to on a mailing list (one that replies to the LIST)
> is the people who havn't been on the net long enough to
> know how mailing lists work, and their whole function.
> Mailing l
The only thing I hate more than not having a proper reply-to on a mailing list (one
that replies to the LIST) is the people who havn't been on the net long enough to know
how mailing lists work, and their whole function. Mailing lists are communities. The
primary function is to share procedure
On Fri, 2004-03-19 at 11:34, Victor Perez wrote:
> Is there a way to use an X100P as a trunk with DID numbers and all?
>
> We just bought one of these and want to create some VoIP extensions
> connected to our PBX as a trial. The PBX does not have capacity for
> any more T1 cards so it is the only
FYI...
Cisco released v6.3 sip firmware around March 12th. Resolved caveats:
# definitions within dialplan file are not functional
Wrong SDP message for a G711 codec negotiation
79x0 Config is not saved on upgrade to LA/BA as LA is named P003
___
A
On Fri, 2004-03-19 at 22:23, Darren Nickerson wrote:
> Folks,
>
> I strongly support removing the current reply-to-list setting, and you
> should too.
>
> Like many new list admins, I once thought the reply-to was kewel. Requests
> to remove it kept coming up, ... usually around the same time som
Eric Wieling wrote:
The 7905G (but not the non-G) supports SIP. It does NOT support XML.
So only the non-SIP phone can use the XML functionality?
That s*cks.
B.
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On Sun, Mar 21, 2004 at 05:53:00PM -0600, Rich Adamson wrote:
>
> The 7940/7960 have been around for a long time while the 7905 is a
> rather recent addition to their product line. I believe the 7905 only
> supports the Cisco proprietary firmware (not sip) while the 7960
> supports either Cisco o
The 7905G (but not the non-G) supports SIP. It does NOT support XML.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
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To UNSUBSCR
On Sun, Mar 21, 2004 at 10:09:33PM +0100, Thomas Schroeter wrote:
> OK, I solved the problem by myself:
>
> openssl-devel was not installed.
>
> Unfortunately, there's not a deb-package, so I had to convert the
> RPM.
Here's the one I use:
http://packages.debian.org/stable/devel/libssl-dev
--
> I'm interested in picking up a Cisco SIP phone, but I don't have enough
> information to decide between the 7940/60 family and the 7905/12
> family. Between the wiki and Cisco's web site, it seems clean that the
> 7905/12 don't have a speakerphone, and that the 7905 doesn't have a
> built-in
Title: RE: [Asterisk-Users] If you know your party's extension # please dial it now ...
Simply define your local
extensions as well as your virtual extensions...
exten 1,1,Play...
exten 2,1,Play...
exten 3,1,Play...
exten 333,1,Play...
When they press 1 the system will
immediately Play,
Hi
Robert Boardman wrote:
In the UK we have a service that if you dial 1471, the last 6 calls
are read out to you and you can pick which one you want by pressing
3, this means that 1471 shows in the cdr, has anyone created a script
or an application that will read out the last callers and th
In the UK we have a service that if you dial 1471, the last 6 calls are
read out to you and you can pick which one you want by pressing 3,
this means that 1471 shows in the cdr, has anyone created a script or an
application that will read out the last callers and then dial the
number? ( that
Hi
Dee Lowndes wrote:
Hi,
I use telewest my self and i have it set up to use Kewlstart, it does
disconnect the call, but its only after the teleewest line plays a
ringing noise, and then the telewest woman says "the other person has
cleared".
That is exactly what happens with mine by any
I'm interested in picking up a Cisco SIP phone, but I don't have enough
information to decide between the 7940/60 family and the 7905/12
family. Between the wiki and Cisco's web site, it seems clean that the
7905/12 don't have a speakerphone, and that the 7905 doesn't have a
built-in Ethernet
Oh, it also seems to crash my Asterisk. (0.7.2).
On Sun, 2004-03-21 at 16:27, Eric Wieling wrote:
> My Cisco 7910 works fine with chan_skinny.
>
> I'm now trying to use the 7910 with chan_sccp. The phone hangs with a
> message "Requesting Server List".
>
> Has anyone seen this problem. Happe
My Cisco 7910 works fine with chan_skinny.
I'm now trying to use the 7910 with chan_sccp. The phone hangs with a
message "Requesting Server List".
Has anyone seen this problem. Happens with both chan_sccp CVS and with
0.02.
--Eric
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2
Please elaborate ...
- Original Message Follows -
> You don't have to avoid using an option 3 when even if
> extensions are 3XXX
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf
> > Of Rich Adamson
> > Sent: Sunday, March 21, 2004 4:19
All,
I am looking for a way to have my AGI startup on a channel
automatically when asterisk starts. Is this possible?
I have my AGI working for when a call comes in - however I
would like the AGI started up automatically with asterisk on
a couple channels as I want to monitor my database and when
You don't have to avoid using an option 3 when even if extensions are
3XXX
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Rich Adamson
> Sent: Sunday, March 21, 2004 4:19 PM
> To: Asterisk Users
> Subject: Re: [Asterisk-Users] If you know your
> I've built the usual "press one for sales, 2 for support" IVR which works
> fine but I'm having difficulty in allowing callers to type in whole
> extension numbers.
>
> My internal extn ranges are 3xxx and 4xxx. I have pasted the IVR below
> (just in case someone wants one). The welcome message
> I've tried all the other methods of dispelling the x100p echo mystery such
> as echo training rx tx gain through ztmon and swapping POTS lines etc etc.
> Can someone mail me a step by step guide to changing the echo cancellation
> algorithms such as Mark,Mark2, Steve etc.
I spent a fair amount
>- apparently when I call from ISDN to an IAX gnophone, I get a very
> short ring then an error: (XXX are mine)
This doesn't happen when gnophone is configured as `Use Asterisk'
apparently. So this is now solved.
___
Asterisk-Users mailing list
OK, I solved the problem by myself:
openssl-devel was not installed.
Unfortunately, there's not a deb-package, so I had to convert the
RPM.
Regards,
thomas
On 21 Mar 2004 at 21:29, Thomas Schroeter wrote:
> Hello,
> I have the following problem installing Asterias on Debian woody:
>
> Inst
> > The wiki indicates Alert_Info can be set to a number, and implies that
> > number is the ringer type listed on the phone. Is there a way to select
> > one of the internal ringer types via Alert_Info?
>
> My understanding is that:
>
> 1. 7940/7960 pre version 6 may support numeric values 1-5 (
Carey Jung wrote:
We're using Asterisk CVS 3-19-04 with four polycom IP600s. The work
very well and we're quite happy with them. They register fine and all
four are able to place and receive calls, BUT two of them are behind NAT
routers and when they place a call on hold, the call is dropped wit
Hello,
I have the following problem installing Asterias on Debian woody:
Installation of zaptel and libpri works find, after "make clean; make install;" for
asterisk, it exits with
make: *** [ast_expr.c] Error 1
Before there were several errors, starting with:
cli.c:31: build.h: No such file o
Hi David,
Thanks, yes that was the problem.
Really appreciate your tip.
Cheers
Sathya
From: David Croft <[EMAIL PROTECTED]>
Organization: Sargasso Networks
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] can't get the full callerid php/agi
Reply-To: [EMAIL PROTECTED]
Your script is receiv
Hi,
> I use telewest my self and i have it set up to use Kewlstart, it does
> disconnect the call, but its only after the teleewest line plays a
> ringing noise, and then the telewest woman says "the other person has
> cleared".
>
That is exactly what happens with mine by any chance did you get c
On Sun, Mar 21, 2004 at 07:42:05AM -0600, Eric Wieling wrote:
> You are correct, IAXtel does not send the called number. Calls from
> both IAXTel accounts will fall into the "s" extension.
Oh, I see.
So I have now implemented a menu. If you call 1-700-895-5211
you can now dial 0800 numbers in S
On Sun, 21 Mar 2004 09:11:33 -0800 (PST), suresh kumar <[EMAIL PROTECTED]>
wrote:
>I would like to get some help from you.
>My server ip is 192.168.1.1 and i would like to
>connect to another ip 192.168.1.2. So how can i
>specify the ip 192.168.1.2 so that make a call from
>192.168.1.1?
Barry Fawthrop wrote:
Thanks Willy and others
It seems I am not able to make myself clear on my two problems
I wish to try again, as I'm sure it is not the phones, but
some stupid config problem on my part. I repeat alot of
what I have said in order to try make myself clear.
That is all problem
If Asterisk can't determine whether you want 3 or 3XXX, it will wait for
DigitTimeout. So if someone dials 3 for echo test, it will take 3
seconds in your case before it jumps to that extension.
David
Mark Phillips wrote:
Hi all,
I've built the usual "press one for sales, 2 for support" IVR w
On Sun, 21 Mar 2004 07:38:17 -0800 (PST), suresh kumar <[EMAIL PROTECTED]>
wrote:
>Hi,
>
>Yes.. i installed iaxComm in the same machine. Hope that was a wrong method.
>How can i uninstall iaxComm so that i can get the CLI prompt?
>Please help me to provide a solution for this.
>
>Thanks & Regar
>
> We're using Asterisk CVS 3-19-04 with four polycom IP600s. The work
> very well and we're quite happy with them. They register fine and all
> four are able to place and receive calls, BUT two of them are behind NAT
> routers and when they place a call on hold, the call is dropped within 5
> s
Hi, all
We're using Asterisk CVS 3-19-04 with four polycom IP600s. The work
very well and we're quite happy with them. They register fine and all
four are able to place and receive calls, BUT two of them are behind NAT
routers and when they place a call on hold, the call is dropped within 5
Does anybody know of a utility that can convert voice prompts from one
codec to another? I'm trying to convert some prompts stored as .gsm to
.g729
- Khan
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Hi Iain
I use telewest my self and i have it set up to use Kewlstart, it does
disconnect the call, but its only after the teleewest line plays a
ringing noise, and then the telewest woman says "the other person has
cleared".
HTH
Welby
Iain Stevenson wrote:
Well. if the Telewest line signall
Hi,
Yes... finally i solved that problem. I am getting CLI
prompt.
When i type asterisk -r command, Now i got display as
[EMAIL PROTECTED] asterisk]# asterisk -r
Asterisk CVS-03/18/04-18:01:45, Copyright (C)
1999-2004 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
===
Well. if the Telewest line signalling is the same as BT uses it "should"
work. When the call ends the Telewest switch should signal this with a
change in the line power which the X100P relies on to disconnect. the call.
You'll probably need to measure the line voltage to sort this out. If you
To All,
Several months (2003) ago there was a discussion regarding
overhead paging & intercom functionality with SIP /
Asterisk. Jerry Gibson, John Todd and various others
participated (from checking the archives). One person even
responded that they had the stuff working with the snom
200s.
Voic
Ok,
so I've re-reported a feature request
http://bugs.digium.com/bug_view_page.php?bug_id=0001265
because
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9
was closed for no apparent reason. Is it now policy to simply close off feature
requests when they haven't been added? If it is
Thanks Willy and others
It seems I am not able to make myself clear on my two problems
I wish to try again, as I'm sure it is not the phones, but
some stupid config problem on my part. I repeat alot of
what I have said in order to try make myself clear.
I have a 4401, 4403 and 4405 in sip.conf a
Hello Azher-
I have a similar setup in hardware, ie: TE410P running on dual-xeon system,
however I'm running IVR only. I start getting the I-frame errors above
about 80 simultaneous calls. I do not get IRQ misses at all. Also I do not
get the startup error messages. The errors I get the most u
Yeah,
as in my reply to yoru earlier message, I don't see '4405'
in your sip.conf
WW
- Original Message Follows -
> Here's another funny
> * CLI puts put
> "-- Registered SIP '4405' at IP.address Port 5060 Expires
> 3600 " and within seconds the snomm 200 beeps the MWI goes
> on the LCD and
Barry,
My snom are on the same LAN as asterisk hence ...
Now, you can set parameters etc. through the web interface.
On the LAN where the snon is/are type in teh IP address in a
browser,
e.g: http://192.168.1.101
This opens the Web Interface
Look in SIP Lines You will get an indication whether the
Hi,
I am having some problems mentioned below, the box is in production live
environment with traffic around 30 - 100 calls.
I am running T/E410P in a Dual P4 xeon with HT disabled. I am using
zaptel 0.9.0 and asterisk stable 1 release. There is no gui, just mysql,
perl (small script) and asteris
Michael Welter wrote:
Actually, please leave this thread on the list.
Question: since this is a local connection between the Definity and
Asterisk on a crossover cable, could E1 PRI be used, even though we're
in the US, to realize another 8 channels? I have TN464F cards that I
will be using to
Hi,
Yes.. i installed iaxComm in the same machine. Hope that was a wrong method.
How can i uninstall iaxComm so that i can get the CLI prompt?
Please help me to provide a solution for this.
Thanks & Regards,
SurMichael Van Donselaar <[EMAIL PROTECTED]> wrote:
On Sun, 21 Mar 2004 04:00:39 -0800
Hey All,
I am using an x100p on a UK Telewest phone line and appear to be having
problems with end user hang ups.
If I call my * from and phone line and let * pick it up when I hang up the
mobile or whatever I am calling from * continues with the call as if I
haven't hung up.
Was wondering if a
Thanks to All who replied
I have tried all the steps above. and from the website given
I have two snom 200 next to each other
4403 and 4405
when I dial 4405 -> 4403 nothing rings and
* CLI reports voicemail/default/4403/busy
when I dial 4403 -> 4405 nothing rings and
* CLI reports vm-
Yes.
Just because US carriers don't offer E1 doesn't mean we can't between our
equipment in our setups. For hardware to hardware it is beneficial to
utilize E1 for that exact reason. As long as your hardware on both ends
supports it, no problem.
James-
- Original Message -
From: "Mic
On Sun, 2004-03-21 at 08:56, Azher Amin wrote:
> But if you r using AGI scripting then u can the DTMF during the
> Playbacks.
>
> e.g. $ret=$AGI->stream_file("$file","12*");
>
> here it will return 0 if nothing out of 12* pressed duringthe playback,
> otherwise it will stop playing and return ei
But if you r using AGI scripting then u can the DTMF during the
Playbacks.
e.g. $ret=$AGI->stream_file("$file","12*");
here it will return 0 if nothing out of 12* pressed duringthe playback,
otherwise it will stop playing and return either 1 2 *
Regards
Azher
---
htt
Actually, please leave this thread on the list.
Question: since this is a local connection between the Definity and
Asterisk on a crossover cable, could E1 PRI be used, even though we're
in the US, to realize another 8 channels? I have TN464F cards that I
will be using to connect with Asterisk
Asterisk doesn't accept keystrokes during playback,
use BackGround to play while waiting for keystrokes.
/Stig
At 08:37 2004-03-21 -0500, you wrote:
>Hi all,
>
>I've built the usual "press one for sales, 2 for support" IVR which works
>fine but I'm having difficulty in allowing callers to type
> I've tried all the other methods of dispelling the x100p echo mystery
> such as echo training rx tx gain through ztmon and swapping POTS lines
> etc etc. Can someone mail me a step by step guide to changing the echo
> cancellation algorithms such as Mark,Mark2, Steve etc.
There isn't much to the
You say no one can dial your extensions? Well no one should be able to,
your extensions aren't listed in the IVR.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Mark Phillips
> Sent: Sunday, March 21, 2004 8:37 AM
> To: Asterisk Users
> Subject
On Sun, Mar 21, 2004 at 08:37:25AM -0500, Mark Phillips wrote:
>
> I'm starting to think that if my extn's are 3xxx and 4xxx I can't have any
> menu choices beginning with 3 or 4. Would this be correct? If so how does
> the received DTMF break out of the IVR and get matched to the relevant
> dialp
Jeb,
Do you know what slot it is in? Carrier A (top) or B (bottom)? We should
take this off list though and reply to me directly from this point, since
this is not really * related now.
There are 2 ways to do this:
At the system propmt type: list configuration ds1 (will list all DS boards
in
On Sun, 2004-03-21 at 07:27, [EMAIL PROTECTED] wrote:
> unfortunately it doesn't seem seem to work easily, maybe because IAXTEL
> doesn't send me the called ID ?
You are correct, IAXtel does not send the called number. Calls from
both IAXTel accounts will fall into the "s" extension.
--Eric
--
Hi all,
I've built the usual "press one for sales, 2 for support" IVR which works
fine but I'm having difficulty in allowing callers to type in whole
extension numbers.
My internal extn ranges are 3xxx and 4xxx. I have pasted the IVR below
(just in case someone wants one). The welcome message sta
On Sun, 21 Mar 2004 04:00:39 -0800 (PST), suresh kumar <[EMAIL PROTECTED]>
wrote:
>Hi,
>
>Thanks a lot for your help.
>
>After installing iaxComm, When I test Asterisk typing
># asterisk c
Are you running iaxComm on the same machine as asterisk? You can't do that.
>
Hi,
I have two IAXTEL accounts, which I activate with:
register => alphanet:[EMAIL PROTECTED] ; 1-700-895-5211
register => cril:[EMAIL PROTECTED] ; 1-700-669-1152
when someone dial this number, it goes through the iaxtel-user
context.
In extensions.conf, I tried:
exten => 17008955211,
Hi,
I have received more excellent problem report information, and I have
resolved a number of issues affecting my soft FAX machine when working
with various models of real FAX machine. The code now seems to be
working with a much greater range of fax machines. A problem affecting
the reliabil
On Sat, 20 Mar 2004, Zac Amsler wrote:
> I know this issue has been address before, but I can not find someone who
> has the answer.
> I am trying to get my * server to authenticate directly to packet8.
> I was very close to them actually giving me the information and possibly
> using them for m
You are right to suspect codec issues here. What codec are you using at
the various endpoints?
Make sure that the Asterisk box is set up with the correct codecs in
the conf files, otherwise it will try to transcode and this will often
cause bad audio quality like you mentioned. If you're using
Hi Girish,
Thanks a lot for your help.
I am also made an attempt to install Linphone, but i
got an error. According to your suggestion, i
installed softphone from zultys.com. That's fine.
I had gone through the
"http://www.automated.it/guidetoasterisk.htm"; link and
got more information from thi
Hi,
Thanks a lot for your help.
After installing iaxComm, When I test Asterisk typing
# asterisk c
I got a display like this (Not getting any CLI prompt)
Hi Jeb,
Have a look on:
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20Avaya
I think it's what you need.
Marc
>-- Message original --
>To: [EMAIL PROTECTED]
>From: Jeb Campbell <[EMAIL PROTECTED]>
>Subject: [Asterisk-Users] T100P T1 problem (Avaya -> asterisk IVR)
>Reply-To: [
On Sun, 2004-03-21 at 04:29, Chris Jones wrote:
> In my opinion just dump firefly and use something
> that works. I did.
Works for me, receives calls makes calls, doesn't make the coffee.
HP Omnibook, W2KPro via Wifi using IAX2
--
Dave Cotton <[EMAIL PROTECTED]>
_
Hi,
I've tried all the other methods of dispelling the x100p echo mystery such as echo training rx tx gain through ztmon and swapping POTS lines etc etc.
Can someone mail me a step by step guide to changing the echo cancellation algorithms such as Mark,Mark2, Steve etc.
Muchos!
Taff.
Yaho
James,
Thanks so much for taking the taking the time to help me figure this
out and learn something.
It is a Definity, but I'm not sure about the card -- just a basic t1
card is all I know (on Monday I could get more info).
Is there a command to find out which card is installed? or if that is
Barry Fawthrop wrote:
For some reason MWI, wants to dial [EMAIL PROTECTED], I have not exten
or
account "asterisk" ???, can't even find where this is set ?
http://www.voip-info.org/wiki-Asterisk+phone+snom
/O
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[EMAIL PROT
Barry Fawthrop wrote:
From: "Olle E. Johansson" <[EMAIL PROTECTED]>
snip
Check the CDRuserfield - it's a free field in the CDR you set in the
dialplan or from a script.
How would you set the CDRuserfield from the dialplan
exten => ?
http://www.voip-info.org/tiki-index.php?page=Asterisk+c
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