At 08:40 PM 11/29/2006, you wrote:
Either write what you want, or learn to use what we have and hope
that SLA when it appears is better. Parking is not the best solution,
I think that's the problem with the Asterisk community right
now. Anytime something is suggested, the response is either
On Wed, 2006-11-29 at 22:57 -0500, Cory Andrews wrote:
Andrew - I have been told they have no plans to introduce US
distribution or availability on these products in the foreseeable
future. I was told this by one of the channel managers from Siemens.
I received some eval units of some of the
Hi,
I have the most stupid problem in my dialplan.
I need to do something as trivial as splitting a string, with a semicolon as
separator.
I was thinking the 'CUT' function would be perfect for this.
But the problem is the semicolon. In the dialplan it is always understood as
a separator for
- Original Message -
From: Douglas Garstang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, November 30, 2006 12:16 AM
Subject: RE: [asterisk-users] What's up with the Manager Interface?!?!
-Original
- Original Message -
From: Douglas Garstang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, November 30, 2006 12:19 AM
Subject: RE: [asterisk-users] What's up with the Manager Interface?!?!
-Original
Hi Koen,
Try:
exten = s,n,NoOp(CUT(${v},${sep},1))
Cheers
Koen Van Impe wrote:
Hi,
I have the most stupid problem in my dialplan.
I need to do something as trivial as splitting a string, with a
semicolon as separator.
I was thinking the 'CUT' function would be perfect for this.
But the
Hi list!
I need help with distinctive ring on Cisco 7940 phone. I'm using Asterisk 1.2.5
(I know, I should upgrade) and in dial plan I have:
exten = _64X,n,Set(_ALERT_INFO=Chirp2)
exten = _64X,n,Dial(SIP/${EXTEN},30,wWtT)
On Cisco in Settings = Ring type I have Chirp1 and Chirp2. By default
Hi for all
I've a problem. I'm trying to detect the progress of an invalid call. For
example, if I phone to a busy number (or invalid number), my misdn always
detect ring. Have you got any suggestion?
2006/11/29, Patrick [EMAIL PROTECTED]:
On Wed, 2006-11-29 at 16:38 +0100, Timothy Parez
Hi Group,
I have an asterisk running as media gateway with a Digium TE405P 2nd Gen
rev 2 with echo cancellation. It is interconnected to a telco carrier via ISDN
Pri. The voice quality is clear except that sometimes a hear a beep sound that
occure around 5 to 10 secs in the middle of the
Can anyone help with the use of regexten? (* 1.4.3)
I've got Asterisk creating extensions for my SIP phones using regexten
but I can't seem to figure out how to make use of them once they're
registered.
Here's my dialplan for from-sip (the SIP's default context):
asterisk*CLI dialplan show
I am trying to find out the best way to replace one of
our hardware PBXs. It currently has 200+ analog phones
connected to it. The idea is to take advantage of the
already installed phone cables (big building) so I'm
trying to avoid the use of ethernet adapters (if
possible). However, I'm
Take a look at Channel Banks
On 11/30/06, Vieri [EMAIL PROTECTED] wrote:
I am trying to find out the best way to replace one of
our hardware PBXs. It currently has 200+ analog phones
connected to it. The idea is to take advantage of the
already installed phone cables (big building) so I'm
On nov/30/2006, Vieri wrote:
Is there another way of doing this (hopefully cheaper
and more convenient)?
VoIP Gateways with 48 FXS ports.
--
Paco Brufal[EMAIL PROTECTED]
ServiTux Servicios Informáticos S.L.
Tel. 966 160 600 / Fax. 966 160 601
Peter,
Thanks for your reply!
It didn't work though.
There's actually already a problem setting the semicolon as value for the
'sep' variable.
*The functions:*
exten = s,n,Set(sep=';')
exten = s,n,NoOp(${CUT(v,${sep},1)})
*The output:*
-- Executing Set(SIP/1649-09ca84f0, sep=) in new stack
--
I think It would be cheaper to use dedicated VoIP PSTN Gateways, such as
audiocodes or similar (audiocodes is actually a bad example, as their not that
cheap). But dedicated ATA hardware with 24 or more ports.
Jon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
Sorry to re-post this but I'm sure it's something simple that someone has
found before.
To summarise:
Dial plan answers the phone
AGI script executes
AGI debug in console show phonetics ABC - However no audio at all on the
phone and this step is less than 1 second.
Dial plan Busy
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Greetings,
I am cutting my teeth with SIP phones and my first issue is getting a
Cisco 7940 to Authenticate with my VoIP provider (BBTelsys).
I did read some notes on the vo-ip website about 7.5 being the better
firmware version.
Try using set without ' or . I mean:
exten = s,n,Set(sep=;)
And next step try using CUT with and without ${..}.
exten = s,n,Noop(${CUT(v,sep,1)})
or
exten = s,n,Noop(${CUT(v,${sep},1)})
First parameter is using variable without surrounding ${..}.
bye,
a
On 11/30/06, Koen Van Impe [EMAIL
On Thu, Nov 30, 2006 at 12:03:24AM -0600, Lacy Moore - Aspendora wrote:
The question is what is the best interface? On our old system, we put the
caller on hold, went to another phone, pressed pickup and then entered the
extension where the call is on hold. I never liked that, especially if I
_The functions:_
exten = s,n,Set(sep=';')
exten = s,n,NoOp(${CUT(v,${sep},1)})
Have you tried to put a '\' in front of the ';': Set(sep='\;')?
Peter
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asterisk-users mailing list
To
On Tue, 28 Nov 2006 17:57:04 -0600
Octavio Ruiz (Ta^3) [EMAIL PROTECTED] wrote:
Is there an isolated example somewhere of how to use existing
dialplan logic and dynamic queue membership to simulate the current
behaviour?
Hi all,
I get the following message in the CLI after enabling video
function. I have searched about the codec 126 but nothing found.
Anybody can tell me how to fix the problem?
Nov 30 15:54:27 NOTICE[16508]: rtp.c:576 ast_rtp_read: Unknown RTP
codec 126 received
Nov 30 15:54:27 NOTICE[16508]:
All,
The last Peter got it right! :-)
The final solution:
exten = s,n,Set(sep='\;')
exten = s,n,NoOp(${CUT(v,${sep},1)})
Thanks for you input and have a very nice day!
Koen
On 11/30/06, Peter Boehm [EMAIL PROTECTED] wrote:
_The functions:_
exten = s,n,Set(sep=';')
exten =
You could put at least two Rhino quad t1 cards and that would give you
8 times 24 ports and I heard of one with those cards plus a dual t1
card which is 240 extensions on one server.
this would take up 3 pci slots.
on Thursday 11/30/2006 Jon Schøpzinsky([EMAIL PROTECTED]) wrote
I think It would
I would just guess that the PCI bus would be pretty busy, with 3 T1 cards.
Couldn't that be a problem?
Jon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici
Sent: 30. november 2006 12:07
To: Asterisk Users Mailing List - Non-Commercial
You could go for 2 quad pri cards + channel banks or for TDMoE or usb
channel banks.
The last option would be the cheaper and more scalable one imho
www.spidermux.org
www.xorcom.com
Joachim
John covici wrote:
You could put at least two Rhino quad t1 cards and that would give you
8 times 24
rhino tells me no, they have a computer you can buy on which they have
tested such things. I don't have this myself, however.
on Thursday 11/30/2006 Jon Schøpzinsky([EMAIL PROTECTED]) wrote
I would just guess that the PCI bus would be pretty busy, with 3 T1 cards.
Couldn't that be a
Vincent Delporte wrote:
Hello
When I make calls from home to the PSTN by going through the Net -
Asterisk - the Net - VoIP provider - PSTN, I get no sound either way.
I assume it's because I must tell Asterisk to use fixed ranges of UDP
ports and map ports accordingly on the NAT firewall
Does anyone on list have experience with Digium hardware in the following
servers:
Dell poweredge SC440
IBM xSeries x226
Have just had major hassles getting TE205P ISDN cards going in these boxes.
No joy so far.
Anyone managed to do it yet?
Thanks.
Mark
Hi Steve,
Ok Playback could be used here, indeed.
But if you are using automonitor - by default activated by (*1) - I
think there is no way how to implement this.
Am I right?
Thanks,
Ondrej
Steve Totaro wrote:
[EMAIL PROTECTED]
wrote:
Hello,
I'm discovering asterisk, it seem
RR wrote:
Norbert, mate, I don't know why you're having so much problems. Do you
wanna post your extconfig.conf here? just to humour us? I have it
running with MSSQLServer a more complicated prospect than mySQL which
has a dedicated driver for it, and it still works.
RR, mate, I don't think
Probably find you have less hassle ditching the proprietary PBX's altogether
and just use the * boxes at each end of an IAX trunk. Probably be a cheaper
solution in the long run.
On 11/30/06, asterisk-robert [EMAIL PROTECTED] wrote:
Inital setup for testing will be 2-4 channels in order to
Norbert Zawodsky wrote:
RR wrote:
Norbert, mate, I don't know why you're having so much problems. Do you
wanna post your extconfig.conf here? just to humour us? I have it
running with MSSQLServer a more complicated prospect than mySQL which
has a dedicated driver for it, and it still works.
you can change the configs to have multiple beeps, and adjust the
timing of them, but when we tried the problem then is the beep is not
added to the incoming audio, but replaces it, so you lose the far end
speaking, went back to default.
On Nov 29, 2006, at 3:34 PM, Dovid B wrote:
Hi
On Wed, Nov 29, 2006 at 11:34:41PM +0200, Dovid B said:
I have a Polycom 601 that when the user is on the phone they only hear
one beep and the CID of the second incoming call is not shown. Is
there a way to have the CID show up for the second call ? And a way to
configure the phone to beep
On Tue, 28 Nov 2006 17:57:04 -0600
Octavio Ruiz (Ta^3) [EMAIL PROTECTED] wrote:
Why? Seems that reinventing the well was the agentcallbacklogin
implementation, when it could be happend in dialplan logic.
Hm, now that I have examined this in more depth, I still seem to be
missing one vital
I've got a Dell SC440 running just fine with a Digium TDM-400 card in
it. It's running CentOS-64bit.
Mark Edwards wrote:
Does anyone on list have experience with Digium hardware in the following
servers:
Dell poweredge SC440
IBM xSeries x226
Have just had major hassles getting TE205P ISDN
I have seen a couple of posts related to this, but no workaround.
Setup;
Asterisk 1.2.13 with Polycom IP501 phones
Caller is sent to the queue with the t option
Agent is logged in via AgentCallbackLogin on an extension that is in a
context that includes exclusively agent extensions.
Thanks Joe. Although youre card isn't quite the same as the one I am trying
to use you've given me a possible idea to play around with - to try and get
the 64 bit stuff going and see if that has some sort of positive effect...
Still out there looking for someone with a 205, 207, 405 or 407 in
I've got a simple set up with 1 fxo port and 1 fxs port in a Digium card
connected to a POTS line and a phone set (physical extension). I've got
all incoming calls launching directly into an AGI script. I'd like to do
the same for the physical extension. In other words, when picking up the
hand
RR, mate, I don't think that I have so many problems.
1.) I asked a simple question:
Is it (still not) possible to connect Asterisk directly (= without ODBC)
to mySQL for the purpose of storing voicemail data?
Now, some posts later I've got a simple answer:
No!
Oh, haha sorry about that, I
Creating a context in your extensions.conf with the same name as your
regcontext will cause all kinds of oddness to happen, among them this.
What you need to do is have a differently-named context in
extensions.conf with your 2-n priorities and include sip_autoreg in
that.
Regards,
- Brad
Asterisk 1.2.7
Redhat 9
I have DiDs from two different ITSP both set up as IAX2. Each one
works when it's the only one in my iax.conf, but when I have them both
defined in iax.conf at the same time, only one will work. My iax.conf
is provided below.
Any ideas how to fix? I'd like to use both
Setup:
Office A:
router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv
Asterisk: v.1.2.4
static IP
Office B:
router: Linksys WRT54GL running Linksys firmware v4.30.2
Asterisk: v.1.2.7.1
dynamic IP (using dyndns name)
Office A is set up with refresh dns and cron job for iax2 reload
The linksys firmware on the WRT54G's on hardware versions 5 and above
are notorious for layer 2 problems. Can you swap out that router?
DM wrote:
Setup:
Office A:
router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv
Asterisk: v.1.2.4
static IP
Office B:
router: Linksys WRT54GL
I have an asterisk server with TDM2402 card that has about 10 extensions
on it.
Both video phones and just audio phones.
Normal calls coming in are received on the TDM lines and routed to an
extension.
If someone wants to call me based on my servers IP address and reach an
extension
on my
Hi Ed -
With Asterisk BE I am trying to record calls coming to a queue,. I am getting
the call to record, however the file name that the file saves to, is not the
correct one.
In my extensions.conf, I have the following entry to set the file name.
exten=
I think you are right or i didn't find how to to it without using a
conference.
And even with conference didn't find a smart way to make it.
Ondrej Valousek a écrit :
Hi Steve,
Ok Playback could be used here, indeed.
But if you are using automonitor - by default activated by (*1) - I
think
Can anyone suggest a reason why these channels might end up zombies?
The process is;
Call comes in via SIP into a context that appends the caller ID name as
follows;
[cnam-lookup]
exten =
_[2-9]X,1,set(CALLERID(name)=${CURL(http://cnam.provider.com/?co
Time Bandit wrote:
I've got a simple set up with 1 fxo port and 1 fxs port in a Digium card
connected to a POTS line and a phone set (physical extension). I've got
all incoming calls launching directly into an AGI script. I'd like to do
the same for the physical extension. In other words, when
That is a huge question, but the short answer is;
They sent you s SIP invite to the [EMAIL PROTECTED] including
whatever credentials are required to authenticate them based on how you
have them defined in your sip.conf.
You could allow anonymous, but be careful that the context it comes into
Or you could use a couple of these boxes:
http://www.xorcom.com/astribank/features-32.html
On 11/30/06, Vieri [EMAIL PROTECTED] wrote:
I am trying to find out the best way to replace one of
our hardware PBXs. It currently has 200+ analog phones
connected to it. The idea is to take advantage of
Hello,
I've a monitoring problem with app_meetme,
I'd like to record a zap channel, which goes to a meetme conference
Monitor doesn't record the voice of another members in the conference.
Thanks any help
Tamas
___
--Bandwidth and Colocation provided
We are looking for an offline billing solution. We have a couple of
particular requirements:
1) Since it's offline, we need to be able to import the CDR.
2) A way to support account credits based on referrals. Meaning, that if a
member refers a new account, that member would get a free month of
Hi
Better late than ever, I guess,
On Mon, Nov 27, 2006 at 10:18:56PM +, Julian Varanini wrote:
Hi all
For some dumb reason I decided to upgrade from Mandriva 2006 to 2007,
thinking I could install asterisk all over again. Anyway I did install
asterisk, zaptel and libpri. After
In article [EMAIL PROTECTED], Mike [EMAIL PROTECTED] wrote:
Just curious if anyone knows of any hacks to enable announce entry/exit
in MeetMe conferences with SIP (as opposed to ZAP) channels since the |i
option will not work with SIP.
The |i option does indeed work with SIP. You do have to
In article [EMAIL PROTECTED],
Chris Blunt [EMAIL PROTECTED] wrote:
Sorry to re-post this but I'm sure it's something simple that someone has
found before.
To summarise:
Dial plan answers the phone
AGI script executes
AGI debug in console show phonetics ABC - However no audio at all
When using autoreg, is there any way to extract the userid somehow? IE:
SIP.com
regcontext=registrations
[123]
regexten=2125551212
extensions.conf
[phones]
include = registrations
exten = _212NXX,2,Dial(SIP/${VARIABLE}))
exten = _212NXX,3,VoiceMail(u${EXTEN})
Honestly I dont see the
I assume you are referring to the 54GL in Office B?
What about replacing the firmware with SVEASOFT or DDWRT? Would this fix it?
On 11/30/06, jason [EMAIL PROTECTED] wrote:
The linksys firmware on the WRT54G's on hardware versions 5 and above
are notorious for layer 2 problems. Can you swap
Hello!
I have problems compiling zaptel (tried 1.2.11, 1.2.10 and 1.4.0-beta2 -- all
give the same error) with 2.6.19 kernel
CC [M] /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.o
In file included
from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xpd.h:26,
from
I'm compiling from downloded source:
http://soft-switch.org/downloads/spandsp/spandsp-0.0.3pre26.tgz
and
http://soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/*
on a slackware 11 system with asterisk 1.4 beta3
Note in the message below I've added information about another
Does anyone on this list know of a reputable T1/PRI provider in St. Lucia?
If so, what monthly costs am I looking at? I do know that Cable and
Wireless are the biggest Telco.
Steve
___
--Bandwidth and Colocation provided by Easynews.com --
Hi,
I am trying to use Asterisk cmd ZapRAS
(http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ZapRAS),
I pathed the ppp daemon ftp://ftp.digium.com/pub/zaptel/misc/, but when I
try to use it, I obtain the following error:
Connected to Asterisk 1.2.4 currently running on TSU-R1 (pid =
I've noticed that some products, like Fonality's HUD, allow live
monitoring of a VoIP call (not just Zap Barge). The Asterisk {client |
manager} command set only seems to allow recording to a file without the
use of a meetme room. Does anyone have a good solution for this?
What I'd like to
On Thursday 30 November 2006 06:13, Zoa wrote:
You could go for 2 quad pri cards + channel banks or for TDMoE or usb
channel banks.
The last option would be the cheaper and more scalable one imho
The scale here is already bordering on unrealistic. I wouldn't expect them to
want to make this
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Joakimsen
Sent: Thursday, November 30, 2006 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trouble with
On Thursday 23 November 2006 11:44, Heidi Mendoza wrote:
We're looking at using 4 or 8 port T1 cards with echo cancellation and are
evaluating brands to go with. We know that Sangoma has excellent solutions
especially when it comes to echo. But we still have to hear about actual
performance
Hi All,
Recent discussions on app_cepstral on the list have led me to believe
there's some issues with Asterisk 1.4 I set about creating a small
howto for people to get cepstral, with app_swift working.
Check it out:
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Joakimsen
Sent: Thursday, November 30, 2006 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trouble with
DM wrote:
snip
Office A is set up with refresh dns and cron job for iax2 reload every
5 minutes. It rarely looses connection to Office B.
Surprisingly, Office B is the one loosing connection with Office A.
I'm surprised because Office A is the one with the static IP address.
When I do a IAX2
What I'd like to implement, ideally, is that once an incoming call is
transferred to a particular operator, the system also calls a manager
who can monitor silently.
I think you are looking for this :
http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy
Interesting product, I didn't know about this one until just now. I've heard
that TDMoE is more trouble than it's worth, though, and may eventually be
phased out of Asterisk. Can anyone from Digium give some more information or
suggestions?
-A.
I'm not from digium but am the proud
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Chris Blunt [EMAIL PROTECTED] wrote:
Sorry to re-post this but I'm sure it's something simple that someone has
found before.
To summarise:
Dial plan answers the phone
AGI script executes
AGI debug in console show phonetics
Great link. After I all you said I get this error loading the module in
asterisk via load app_swift
The 'load' command is deprecated and will be removed in a future
release. Please use 'module load' instead.
[Nov 30 13:54:08] WARNING[7825]: loader.c:362 load_dynamic_module: Error
loading
Forum wrote:
Does anyone on this list know of a reputable T1/PRI provider in St.
Lucia? If so, what monthly costs am I looking at? I do know that
Cable and Wireless are the biggest Telco.
I think you will find they are the only telco and the cost will be enormous.
--
Chris Mason
(264)
Great link. After I all you said I get this error loading the module
in
asterisk via load app_swift
The 'load' command is deprecated and will be removed in a future
release. Please use 'module load' instead.
[Nov 30 13:54:08] WARNING[7825]: loader.c:362 load_dynamic_module:
Error
if its a version 5 or higher, that wont be an option, but if its not,
give openwrt or ddwrt a try.
DM wrote:
I assume you are referring to the 54GL in Office B?
What about replacing the firmware with SVEASOFT or DDWRT? Would this
fix it?
On 11/30/06, jason [EMAIL PROTECTED] wrote:
The
Fixed my problem!
Note to self... READ EVERYTHING in the instructions!
Again thanks for the information!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M.
Sent: Thursday, November 30, 2006 1:56 PM
To: Asterisk Users Mailing List -
I'm having problems installing ztdummy on my
CentOS-2.6.16-rc6/Asterisk-1.2.11 datacenter server (no ZAP HW, SIP only
to PSTN). I unpacked the kernel sources and headers in a directory, made
(but not re/installed) the kernel, unpacked the zaptel-1.2.11 tarball,
then went thru the make
On Thu, Nov 30, 2006 at 07:19:14PM +0200, Roman Yeryomin wrote:
Hello!
I have problems compiling zaptel (tried 1.2.11, 1.2.10 and 1.4.0-beta2 -- all
give the same error) with 2.6.19 kernel
CC [M] /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.o
In file included
from
On Nov 29, 2006, at 11:40 PM, Lacy Moore - Aspendora wrote:
[snip]
I went from a Lucent Merlin Legend system to Asterisk. For me,
it's a tradeoff for features. To my users, it was a step
backward. I also upgraded an office from a Partner system to
Asterisk. To the users, it is a huge
Which version?
Similar issues parsing callback number in 1.2.12
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner
Sent: Thursday, September 28, 2006 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial
I'm having problems installing ztdummy on my
CentOS-2.6.16-rc6/Asterisk-1.2.11 datacenter server (no ZAP HW, SIP only
to PSTN). I unpacked the kernel sources and headers in a directory, made
(but not re/installed) the kernel, unpacked the zaptel-1.2.11 tarball,
then went thru the make
I'm having problems installing ztdummy on my
CentOS-2.6.16-rc6/Asterisk-1.2.11 datacenter server (no ZAP HW, SIP only
to PSTN). I unpacked the kernel sources and headers in a directory, made
(but not re/installed) the kernel, unpacked the zaptel-1.2.11 tarball,
then went thru the make
if its a version 5 or higher, that wont be an option, but if its not,
give openwrt or ddwrt a try.
Actually, this is no longer true (at least for WRT54G), see
http://en.wikipedia.org/wiki/DD-WRT for the official list of supported
models
___
--Bandwidth
On 11/30/06, Dave Fullerton [EMAIL PROTECTED] wrote:
DM wrote:
snip
Office A is set up with refresh dns and cron job for iax2 reload every
5 minutes. It rarely looses connection to Office B.
Surprisingly, Office B is the one loosing connection with Office A.
I'm surprised because Office A
Hi
On Thu, Nov 30, 2006 at 12:59:13PM -0500, Andrew Kohlsmith wrote:
On Thursday 30 November 2006 06:13, Zoa wrote:
You could go for 2 quad pri cards + channel banks or for TDMoE or usb
channel banks.
[ disclaimer: I work for the company that makes the USB channel bank
which was mentioned
Can anyone give me some insight on this? If I am not making myself clear
please let me know.
At voip-info.org they show the following example
exten = s,1,Set(foo=${STAT(s,/var/t3)})
which I guess is suppose to work and make foo = size of t3
I did the following
exten =
Nov 30 00:19:06 ERROR[23493]: pbx.c:1382 ast_func_read: Function STAT not
registered
from http://voip-info.org/wiki/view/Asterisk+functions :
Functions in the below list are marked in red if they are only
available in version 1.4 and higher.
And STAT is marked in red so I guess you're not
Hi list !
I implemented *8 to pickup any call on my asterisk system. But after the
pickup callerid is missing, so there is no way to see from where the call
originated. How can this callerid be passed on.
Nik
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When I call to VoicemailMain it just sits.
; Retrieve Voice Mail
exten = 2500,1,Wait(2)
exten = 2500,2,VoicemailMain(s100)
exten = 2500,3,Macro(endcall)
1.4.3 latest SVN.
voicemail(100) works and the mwi systems works. I am not using ODBC or SQL.
Voice mail to email works ok.
I just cannot
Asterisk 1.2.7
Redhat 9
I have DiDs from two different ITSP both set up as IAX2. Each one
works when it's the only one in my iax.conf, but when I have them both
defined in iax.conf at the same time, only one will work. My iax.conf
is provided below.
Any ideas how to fix? I'd like to use both
You can get a basic VOIP phone for the same price that it will cost you for
a FXS port. As far as wiring you can go with a bit more expensive phone and
get a dual port with POE (if they have an existing computer network).
- Original Message -
From: Vieri [EMAIL PROTECTED]
To:
Hi !
as the wiki says there is only the possibility to set incominglimit
and outgoinglimit to type peer, how can I accomplish this with the
type friend?
nik
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I have a Linksys PAP2 connected to Asterisk. Have one of the FXS ports
working fine. I am unable to get the other to work. Does anybody have an
example configuration to make both work. Both are registering fine but
there's just no dialtone on the non working port.
TIA
I looked at the voicemail.c code and you must have the res.adsi module
loaded. I was not loading it.
Now it works.
Something to remember.
Thanks
--john
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We do have @cough VoIP GSM Gateway for sell as well @ cough
Try to search on ebay for gsm voip gateway and you will see some in there
As far as I am concern it is cheaper than 2n.
And if you are looking for multi ports then it will come off as RJ11 ports
rather than voip and they are £100 per
Where are you looking for the caller id at?
On 11/30/06, Nik Engel [EMAIL PROTECTED] wrote:
Hi list !
I implemented *8 to pickup any call on my asterisk system. But after the
pickup callerid is missing, so there is no way to see from where the call
originated. How can this callerid be passed
What do you get in the CLI ?
- Original Message -
From: John Hill [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, November 30, 2006 11:24 PM
Subject: [asterisk-users] voicemailmain
When I call to VoicemailMain it just sits.
; Retrieve Voice Mail
exten =
Hi
Thanks for the advice but it really is more fundamental.
I have an old (v5) sccp phone. I need to upgrade it to v7 sccpbefore I can
load the Sip image. I downloaded the V7 sccp file from the cisco website but
it seems to want call manager to load.
Does anyone have any experience of
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