I'll talk to your boss if he has a problem! ;)
/b
On Jul 23, 2005, at 11:03 PM, Terry Moore-Read wrote:
Mine did.
[EMAIL PROTECTED] 7/21/2005 2:54 PM
Brian West wrote:
ClueCon is coming in 2 weeks so we urge everyone who plans on
attending to register today so we get a proper
Joseph wrote:
[snip]
exten = _6XXX,2,Busy
exten = _6XXX,3,Hangup
But the whole point is that I don't want the caller to hear a busy
signal or get hung up, I want the Queue to try the next available
agent. Which it does at the moment, just with the errors mentioned in
the error log
PLEASE FOR THE LOVE OF GOD put a NAME in your email program.. I'm
sure it makes going back and finding stuff in the archives when you
and about 100 other people use Asterisk in their names This
goes for anyone that uses Asterisk, Asterisk PBX or any form
there of .. lets put a name in
Adam Goryachev wrote:
[snip]
This busy means, tell the queue app that the agent is busy. The queue
app willl go try someone else. The caller will keep hearing music. :)
Julian, and others,
If someone offers you a suggestion towards solving your problem, you
might at least try it before
Brian West wrote:
PLEASE FOR THE LOVE OF GOD put a NAME in your email program.. I'm sure
it makes going back and finding stuff in the archives when you and
about 100 other people use Asterisk in their names This goes for
anyone that uses Asterisk, Asterisk PBX or any form there of ..
On Sat, 2005-07-23 at 17:51 -0400, Julio Arruda wrote:
OpenVPN can use TCP, and really, I would expect that many users using
openvpn to bypass firewall rules, would be using TCP not UDP.
Yes OpenVPN _can_ be _configured_ to use TCP, just shows what a powerful
tool it really is.
--
Dave
Where can I find the documentation or an overview of everything that is new
in Asterisk 1.2 ?
Thanks.
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does astcc support h323 because i didnt find in the
trunks any technology named by h323.
__
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Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
___
TWV wrote:
Where can I find the documentation or an overview of everything that is new
in Asterisk 1.2 ?
There's no good documentation on that out there... yet.
Read all the sample configuration files, the READMEs and especially the
updating file.
Any documentation people with time to write
in etc/asterisk/extensions.conf
where you define the desteny you should put the time. example:
exten=_009[13456789].,1,Dial(SIP/operador/${EXTEN},60,tr)
in this example I am giving 60 seconds waiting for someone to pickup the
phone.
From: chawki hammoud [EMAIL PROTECTED]
Reply-To: Asterisk
Olle E. Johansson wrote:
There's no good documentation on that out there... yet.
Read all the sample configuration files, the READMEs and especially the
updating file.
Any documentation people with time to write out there?
What would be the best place for (this kind of) documentation? On the
On Thursday 21 July 2005 17:58, Kristian Kielhofner wrote:
They are not the fastest - running hdparm -tT on them reveals a speed of
2Mb/s which is about a third of the speed of 100Mbits ethernet. For call
recording I usually add an IDE hard drive and make sure that most
filesystems (e.g.
Keep in mind that Cisco Phones with the SIP firmware do not support
XML Pushing in theire native format. Using the HTTP refresh is how
most do it there. Just like ADSI most of the vendors are not sharing
everything about their phones in a hope to profit.
Andrew Latham
On 7/24/05, Anton Krall
Darren Nickerson wrote:
"Brian West" [EMAIL PROTECTED]:
Aidan isn't a troll he does raise a very
valid point.
Which was, I presume, that companies that once collaborated on Asterisk
development such as Sangoma don't find themselves on friendly terms
with Digium now that
Hello,
after a while that it has been forcibly down, the ASTERISK-ITA mailing
list, dedicated to general Asterisk discussion in Italian, is now back
online.
The list is located at http://groups.yahoo.com/group/asterisk-ita
Thanks for your interest
l.
--
Assum est, versa et manduca.
Anton Krall wrote:
Also, both asterisks have notransfer?yes and I get this
-- Attempting native bridge of IAX2/[EMAIL PROTECTED] and
IAX2/voipjet-9
Why? Seems its not taking the notransfer into account.
As was just covered exhaustively on asterisk-dev, 'native bridge' and
'native
Darko Sundek wrote:
SORRY FOR MY LONG PRELUDE ( we respect kbps)
Molim, pozdravi mi Podgoricu i celu Crnu Goru :)
1. What we need to know about our LOCAL PSTN telco (digital) lines to
be shure in our hardware choise (voltage, current etc.)?
*IF* they are EuroISDN, then they are
Hello
I am using a Junghanns QuadBRI ISDN card - the
module name is qozap. If I like at my interrupt assignment, qozap is
sharing interrupt 10 with libata and uhci_hcd.
I think libata is the IDE hard drive module and
uhci_hcd is a USB module.
linux:~ # modprobe qozaplinux:~ # cat
Sorry about the late post, but you can also use a console cable to reset the
password.
- Original Message -
From: Shaun Ewing [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Thursday, July 21,
chan_sccp2 is pretty usable as far as i can tell. So you can take advantage
of full feature set of cisco's and asterisk
On 7/24/05, Andrew Latham [EMAIL PROTECTED] wrote:
Keep in mind that Cisco Phones with the SIP firmware do not support
XML Pushing in theire native format. Using the HTTP
I'm trying to get a patch tested for inclusion in CVS.
Anyone that is running EM on a T1 and had to fool around with
emdigitwait could you please try this. This patch removes the need
for the emdigitwait parameter and speeds up dialing.
This situation is mostly interfacing a legacy
I've wikied and googled, but could not find any appropriate scripts - :
I was wanting to stress-test my new server, and as I have a TE410p card
(but only using 2 ports), I was going to connect ports 3 4 with a
cross-over cable so that I could make a number of outbound calls on port
3 and
On Fri, 22 Jul 2005, Thomas Christie wrote:
* If you can get the song from this flash animation converted to MP3, then
it might be good (bad):
http://www.ebaumsworld.com/flash/spacepeople.html .
http://damin.umlcoop.net/spacepeople.mp3
--
Vice President of N2Net, a New Age Consulting
Maybe we can have a wiki section with success stories using Asterisk
CVS HEAD. Some new features tested and succefully used.
It could be a point to start a 1.2 documentation.
I'm available to do it, or better, to put some success stories on it.
Denis.
On 23 de jul de 2005, at 09:52, Olle
For example, ADSI, how does it work and how can it be integrated into
asterisk?
I think vendors should release that info in order for us to develop more
asterisk based apps for their phones = more sales :)
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On
Again, the idea is to see which other phones besides Cisco, can use this or
any other method to server information apps on them, like ADSI, etc.. Any
ideas? I just cant believe Cisco are the only ones capable of this...
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL
Joseph wrote:
[EMAIL PROTECTED] wrote:
I am using firefly as my iax client, and it does not seems to work when
I use 1001,1,Dial(IAX2/1001) instead of 1001,1,Dial(IAX/1001)
Change the lines below from IAX to IAX2
Thanks a lot Joseph for your reply.
As you can see from my mail, I had
Anton Krall wrote:
Again, the idea is to see which other phones besides Cisco, can use this or
any other method to server information apps on them, like ADSI, etc.. Any
ideas? I just cant believe Cisco are the only ones capable of this...
Polycom Soundpoint IP 600 can. The 300 and 500
I have 2 sip accounts setup - 200 and 202. If
I do sip show peers I get:
sip show peersName/username
Host Dyn Nat
ACL Mask
Port
Status202/202
192.168.0.6
D 255.255.255.255
5060
Unmonitored201/201
(Unspecified)
D 255.255.255.255
5060
Unmonitored200/200
192.168.0.3
D
With your current setup, your IDE channels,
ethernet card and BRI card are all on seeperate interrupts, so you shouldn't
have problems unless you will be making heavy use of USB devices.
It appears that you are using an
Intel ICH5 based motherboard. The Intel ICH5 chipset supports IO-APIC.
Does anyone have a collection of stupid hold music? Y'know, the sort of
thing that would drive a person mad? Silly songs, repetative tunes etc?
You should be flogged pubically for bringing up this subject - that
last space people song almost made me wash out my ears with
sulphuric acid!!!
73
Ok,
Here's a bad story about using CVS head:
I have a client using Asterisk as a predictive dialer. The dialer
originates calls to people via a Zap channel,
and awaits input from the users. Problem is: the DTMF's are never picked
up. Now, when I tried using
the stable branch - it worked.
I have 2 sip accounts setup - 200 and 202. If I do sip show peers I get:
sip show peers
Name/usernameHostDyn Nat ACL Mask Port Status
202/202 192.168.0.6 D 255.255.255.255 5060
Unmonitored
201/201 (Unspecified)D
It appears from the debug that extension 200 is
trying to call 777, not 202. Your Asterisk server can't find an extension 777
and returns "404 not found". That will explain why you can't call extension 777
from extension 200. If you want to call extension 202, you will need to dial 202
on
I just wanted to post here and let everyone know that the TE406P (quadspan
T1/E1 with hardware echo can) kicks some serious ass.
We've been running a PRI now for over a year with Asterisk (every single call
in and out is through two Asterisk boxes, including faxes) and while the
software based
Derek: you reply is uncorrect. If Angus has the extension 777 in his
dialplan/extensions.conf which will dial 202. The name of the peer has
absolutely nothing to do with which number/name he would have to dial.
Without dialplan he will be unable to call any extension even 202, as
202 is only
It is actually found in astcc.agi
Find the line that fits the channel like:
$dialstr = Zap/$res-{path}/$phone|30|HL( . ($maxtime * 60 * 1000) .
:6:3);
Change the |30| to |60| or whatever you want.
Thanks,
Steve
- Original Message -
From: Patricio Ku [EMAIL PROTECTED]
To:
Nir Simionovich wrote:
I have a client using Asterisk as a predictive dialer. The dialer
originates calls to people via a Zap channel,
and awaits input from the users. Problem is: the DTMF's are never picked
up. Now, when I tried using
the stable branch - it worked.
This problem was fixed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darren Wiebe
Sent: Saturday, July 23, 2005 3:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ASTCC gives me only the time,
but no cost
Thank you
Depends how many. For a few you can go with a TDM40b. For large numbers
you can get a quad T1/E1 board and hook one port to the PSTN E1 and another
port configure at T1 and attach it to a channel bank such as the Adtran 600
with up to 24 FXS ports.
- Original Message -
From: Denis
But I guess I'm wondering ... does the present licensing model discourage other vendors from contributing to *? I'm not sure Sangoma developers could sign the disclaimers even if they wanted to ... but then again I don't know if there's anyone there with anything to offer. I would think that that
Hello,
I'm looking for US DID and US50/CA 800# Providers.
I found voiceconduits.com 8 month ago, there interface looks good, but
there are still not live, I believe they won't be any time soon.
I found sixtel, but order take eternities, they probably won't get my
orders right any soon.
So
Hello everyone,
Well here is my initial posting to the list, and I will admit Asterisk is new
to me. I just got everything running here a couple days ago, so still learning
the ropes for sure.
OK, here is my problem. Currently I have it setup talking to a couple Cisco
IP phones, and some
Marc: My answer is not incorrect... it is incomplete.
The OP stipulated 2 extensions 200 and 202... and provided a sip debug
indicating a call from 200 to 777.
I pointed out the obvious.
If the OP is dialing 202 on the phone, and the phone is dialing 777, then he
needs to look at the dialplan
The easiest way to reset a Cisco phone is via tftp
hands down
Download the image from Cisco (I can get it for you if you like.
And re provision it via TFTP
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I'm trying to get inbound calls from a TNT working but get 407 errors from
the TNT. This is what I have in sip.conf:
[maxtnt]
type=friend
host=x.x.x.x
dtmfmode=rfc2833
callerid=MaxTNT maxtnt
context=demo
qualify=yes
disallow=all
allow=g729
allow=ulaw
insecure=very
This is what the TNT is
Your [frombroadvoice] context is incorrect. You have set a global variable
BVNUMBER and used it as the extension match in the context. The problem is
that the extension match syntax does not support variable substitution
unless you are using a relatively current CVS HEAD. As [EMAIL PROTECTED] is
Hi,
I have a mint, like-new (used only once) Zhone Z-Plex channel bank for
sale. It has 16 FXS and 8 FXO (part number: Z-PLEX-10-24S/O). It
functions as a channel bank, router, and CSU/DSU, combining T1, analog
voice, and data. D4, SF, and ESF line formats are supported with AMI
or B8ZS
OK, I think I understood what you were saying, but let me type this in here as
like I said I am for sure trying to figure this sucker out still..
I just tried the following:
exten = s,1,Macro(exten-vm,[EMAIL PROTECTED],${BVRINGS})
I also tried this:
exten = 240524,1,Macro(exten-vm,[EMAIL
Man I almost passed from laughing when I read this, that is the best
description of bad echo I have ever heard:)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Sunday, July 24, 2005 12:28 PM
To: Asterisk Users Mailing List -
Not to continue or feed any flame wars, because flame wars and holy
wars only create hurt feelings. However, this is my personal
experience:
To the contrary I have found Digiums support to be exceptional.
They have helped support me with several minor and small problems, and
all I did was buy
I think the 777 may be a bit of a Red Herring. I dialed 777 as a test. I
can't dial 202 from 200 if I actually dial 202!
My extensions.conf file:
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in
Oh GOSH
That was awesome!
I have an even better one in store but gotta capture it from it's
rather obscure source..
Stay tuned!!!
Steve (N8LBV)
Does anyone have a collection of stupid hold music? Y'know, the sort of
thing that would drive a person mad? Silly songs,
Ok.. I screwed up...
You have a register statement:
[EMAIL PROTECTED]:123abc:[EMAIL PROTECTED]/
201
so, the incoming call from broadvoice will be sent to extension 201 in the
frombroadvoice context.
To ensure what is going on, use this as your context.
exten = _X.,1,Noop(Incoming call for
Ok your extensions.conf doesn't mention anything about an
extension/number equal to 202 or 200. You must know that the name of a
SIP and IAX2 peer is only an address, you will have to assign a number
via extensions.conf to this address.
Have a look at www.voip-info.org and of course
The extensions.conf file you provided looks suspiciously like the asterisk
configs/extensions.conf.sample file.
Did you create a dialplan for your specific configuration or did you just
copy the sample file?
- Original Message -
From: Angus Comber [EMAIL PROTECTED]
To: Asterisk Users
On Sunday 24 July 2005 16:30, Chris Modesitt wrote:
Man I almost passed from laughing when I read this, that is the best
description of bad echo I have ever heard:)
:-) Well the bad bad echo I described I am almost positive occurs because the
echo canceller either mistakenly turns off (false
OK Angus
just start here
mv extensions.conf extensions.conf.old
and create a new extensions.conf
[default]
exten = _2XX,1,Dial(SIP/${EXTEN},20,Ttm)
exten = _2XX,2,Hangup
just those 3 lines
do an 'extensions reload' in the CLI or just restart Asterisk
and see if it works
regards,
PS you would be better seeing this debugged with set verbose 5 in the CLI
regards,
Mark
On 7/25/05, Mark Edwards [EMAIL PROTECTED] wrote:
OK Angus
just start here
mv extensions.conf extensions.conf.old
and create a new extensions.conf
[default]
exten =
Would this do it:
exten = _2XX,1,Dial(${ARG1},30)
Then I would fallback to voicemail (or something else) after the 30 seconds?
Angus
- Original Message -
From: Marc Storck [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi all, i have a weird problem, i need to know why
this 'simple' macro works Only with a * STABLE version (1.0.0.x to 1.0.0.9) , if
i try it with a CVS HEAD version.. don't works ;-(
Thanks in advance.
[local]
include = default
[macro-tono_simulato]exten =
s,1,Set(DIALED=${ARG1})exten =
You observed correctly. Yes I just copied the sample file, hoping it would
work.
I didn't realise I had to do anything special with the dialplan just for
dialing internal extensions.
Can I use something fairly generic like this (assuming all my extensions are
three digit starting with
OK was actually able to pull it out of the archives!
It's now at http://stuff.michiganbroadband.com/asterisk
I'll leave it there for about a week or two then remove it.
T othe best of my knowledge it's public domain, if anyone needs more info
please contact me offlist.
This is right up there in
No please use ${EXTEN}, ${ARG1} is for macros.
And of course you will use the protocol in front of ${EXTEN}
So for SIP use:
exten = _2XX,1,Dial(SIP/${EXTEN},30)
and for IAX2 use:
exten = _2XX,1,Dial(IAX2/${EXTEN},30)
Regards,
Marc
Angus Comber wrote:
Would this do it:
exten =
Interesting, I've been on this list for almost a year now, and I
didn't recieve this spam. Are you sure you didn't download any sip
softphones and gave your email address?
In which case it is NOT spam, you gave your email address to them.
On 7/22/05, Jay Milk [EMAIL PROTECTED] wrote:
Try VIM and Apache. If you know what you doing you will have one that
at least you like.
On 7/22/05, Dante Renda [EMAIL PROTECTED] wrote:
what is the best web based managment aplication for asterisk ???
Dante
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xAD wrote:
Hi all, i have a weird problem, i need to know why this 'simple' macro works
Only with a * STABLE version (1.0.0.x to 1.0.0.9) , if i try it with a CVS HEAD
version.. don't works ;-(
Thanks in advance.
[local]
include = default
[macro-tono_simulato]
exten =
Sorry to be a pain... but
I restarted my * and now when I launch * get this:
== Parsing '/etc/asterisk/zapata.conf': Found
Jul 24 18:52:45 WARNING[6817]: chan_zap.c:932 zt_open: Unable to specify
channel 1: No such device or address
Jul 24 18:52:45 ERROR[6817]: chan_zap.c:6473 mkintf: Unable
I am having a problem with your my nufone service.
I'm trying to setup incoming calls and I'm having no
success. Outgoing works fine though. The message I'm
getting is the person you are call is not currently
reachable. I'm going to give you as much info as I
can. I'm also an asterisk newb!
Hello,
I added queuing support (based on SQLite database to store the queue)
for my SIP Messaging patch. Works with eyeBeam, probably lots of bugs,
but it's at least something.
I created page about installation on the tips and tricks of voip-info.org:
OK, I added in the rule you gave me below, and yep it said 201, so your right
on with that one.
So with the following extension rules:
;setup SIP extension for BroadVoice
[globals]
BVNUMBER=240524 ; your calling number
BVRINGS=201 ; the phone to ring
BVVMBOX=201 ; the VM box for this user
Thanks Brian:
Sorry for the late reply. That was helpful. Problem solved.
Scott
At 10:43 PM 7/10/2005, you wrote:
You might want to recompile the res_config_mysql or configure
res_config_odbc which works via myodbc and is just as good!
/b
---
Anakin: You're either with me, or you're my
Just because you give your email address does not allow for unsolicited
emails unless you agree in the EULA or terms and conditions.
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday,
That was another problem - now fixed.
Thanks for all your help on extensions.conf
Angus
- Original Message -
From: Angus Comber [EMAIL PROTECTED]
To: Mark Edwards [EMAIL PROTECTED]; Asterisk Users Mailing
List - Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Sunday,
Hello
I am sure this is a very basic Linux
question.
But every time I reboot my * I need to
modprobe module
and then
ztcfg
After doing this I can then run * without it
complaining about not loading a channel. The module being loaded is qozap
- a ISDN card.
What do I need to do to
Does anyone have a recommendation for a good SIP phone with
a busy lamp field? I need my operator to be able to see extension status for
about 20 extensions and transfer via HOLD + extension button. Ive got a
pair of SNOM 360s with the sidecar, but Im very disappointed with them.
The
What is happening is that the _X. extension is catching the call... you need
to take it out... it was only meant as a test to make sure which extension
it was actually being sent to...
It seems that it is then executing your vm macro without any parameters...
don't know why that would happen...
The Cisco 7960 SIP firmware does not support the
7914 "sidecar"... To use the "sidecar" you need to use the SCCP (Call Manager)
Firmware. There is chan_skinny in Asterisk but I don't recall seing support in
it for the 7914. You may want to look at the chan_sccp driver.. it is supposed
to
=RECORD-IN, key=
Jul 24 19:55:41 DEBUG[1078]: Unable to find key '' in family 'RECORD-IN'
Jul 24 19:55:41 VERBOSE[1078]: -- DBget: Value not found in database.
Jul 24 19:55:41 VERBOSE[1078]: -- Executing SetVar(SIP/240524-7457,
CALLFILENAME=20050724-195541-1122249341.21) in new stack
Jul 24 19
Mine had no problem sending me. I can't wait!
--
Tom
On 7/24/05, Brian West [EMAIL PROTECTED] wrote:
I'll talk to your boss if he has a problem! ;)
/b
On Jul 23, 2005, at 11:03 PM, Terry Moore-Read wrote:
Mine did.
[EMAIL PROTECTED] 7/21/2005 2:54 PM
Brian West wrote:
dbruce wrote:
The Cisco 7960 SIP firmware does not support the 7914 sidecar... To
use the sidecar you need to use the SCCP (Call Manager) Firmware.
There is chan_skinny in Asterisk but I don't recall seing support in it
for the 7914. You may want to look at the chan_sccp driver.. it is
This may not be the answer you're looking for ... but why
not whip up a little program or web application for the operator's PC that shows
the extension, name, busy status, new voice-mail count, etc? Or you could
have it done by a consultant. In the long run, it will probably be less
Add your module to your module startup method for your distro. See
http://voip-info.org and search for startup or boot, then read.
On 7/24/05, Angus Comber [EMAIL PROTECTED] wrote:
Hello
I am sure this is a very basic Linux question.
But every time I reboot my * I need to
Thomas Christie wrote:
This may not be the answer you're looking for ... but why not whip up
a little program or web application for the operator's PC that shows
the extension, name, busy status, new voice-mail count, etc? Or you
could have it done by a consultant. In the long run, it will
I thought there would already be one or eight hundred of them out there
already. I was joking about doing the work for ... Um ... What's-his-name
(no offense, your email is already deleted).
Have you used this one or others?
What's your opinion of this gentleman using this application instead
Dear all,
I want to create a php web front end to disconnect a SIP call (from a
particular sip phone) which is in progress. Any ideas how to do so?
Thanks in advance.
peiyin
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Yes your are right, this is the modified version for CVS HEAD ..but don't
works ;-D you have tried it?
this version works only on STABLE:
[macro-tono_simulato]
exten = s,1,SetVar(DIALED=${ARG1})
exten = s,2,SetVar(TOCONTEXT=${ARG2})
exten = s,3,SetVar(STRIP=${ARG3})
exten =
I have a bunch of Polycom Soundpoint 600 phones and they are working
great. The only thing I can't seem to get them to do is to ring-answer
without the ring.
This is what I have in my sip.cfg file on the boot server:
alertInfo voIpProt.SIP.alertInfo.2.value=RA
How come nobody has mentioned The Girl From Ipanema as performed by
Herb Alpert and Co.?
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To UNSUBSCRIBE or update options visit:
]: -- Executing
SetVar(SIP/240524-7457,
CALLFILENAME=20050724-195541-1122249341.21) in new stack
Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
Jul 24 19:55:41 VERBOSE[1078]: -- Executing
GotoIf(SIP/240524-7457,
0?15:99) in new stack
Jul 24 19:55:41 VERBOSE[1078]: -- Goto (macro-record-enable
xAD wrote:
Yes your are right, this is the modified version for CVS HEAD ..but don't
works ;-D you have tried it?
No, because you haven't bothered to tell us what is wrong. This doesn't
work is not much to go on, I'm sure nobody is going to try your macro
without you first describing what
Thomas Christie wrote:
(no offense, your email is already deleted).
None taken.
Have you used this one or others?
What's your opinion of this gentleman using this application instead of a
hardware solution?
In the process of setting it up for a August 6th install. I won't know
Brian West [EMAIL PROTECTED] wrote:
Or better yet.. modify the disclaimer like I and a few others did to
say that the only thing you will disclaim are things you post on the
bug tracker! NO UPDATES, NO CHANGES, NO NOTHING! If its not posted
under your user on mantis IT IS NOT DISCLAIMED!
I am having a problem with your my nufone service.
I'm trying to setup incoming calls and I'm having no
success. Outgoing works fine though. The message I'm
getting is the person you are call is not currently
reachable. I'm going to give you as much info as I
can. I'm also an asterisk
Billy Dunn wrote:
I have a bunch of Polycom Soundpoint 600 phones and they are working
great. The only thing I can't seem to get them to do is to ring-answer
without the ring.
This is what I have in my sip.cfg file on the boot server:
alertInfo voIpProt.SIP.alertInfo.2.value=RA
I'm looking for US DID and US50/CA 800# Providers.
I found voiceconduits.com 8 month ago, there interface looks good, but
there are still not live, I believe they won't be any time soon.
I found sixtel, but order take eternities, they probably won't get my
orders right any soon.
So
Is anyone aware of a ISDN HFC-S card available in Australia that is
A-tick certified?
Thanks,
Stephen Allan.
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what is the best web based managment aplication for asterisk ???
This topic as been discussed many times. Search the archives :
http://www.google.ca/search?hl=enq=site%3Alists.digium.com+web+management
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turn off the ringer. Put it in silent mode!
Brad
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If you use the polycom provided config files, the default ring_answer class
is 4 and the auto_answer class is 3. So for your RANR alertinfo entry,
change the class to 3 and it will work as you expect. ie:
alertInfo voIpProt.SIP.alertInfo.3.value=RANR
voIpProt.SIP.alertInfo.3.class=3/
The
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