On Thursday 28 April 2005 12:57 pm, Nicolás Gudiño wrote:
It almost sounds like there needs to me a new manager action:
Action: Bridge
ChannelA: SIP/199testfone-1f3c
ChannelB: Zap/6-1
It sounds like the intrinsic functionality for 'bridging' is already
there in Asterisk (duh!), it
Hi folks,
I am having a problem with MusicOnHold. Right now I have the following
configuration:
Default = mp3:/var/lib/asterisk/mohmp3
The problem is that I can't hear the music or sometimes the music seems to
skip like a scratched record...
Robson
Matt Roth wrote:
Asterisk Users / Asterisk Biz List Members,
Have you decided which PSTN-VoIP gateway you'll use?
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Chrisopher Jacob wrote,
What I am trying to do is track down a rough idea of how many Asterisk
systems are in production right now. Ideally as this information was
gathered it could be sorted by country, state, industry, etc.
Does anyone have any information, or any idea of where to start?
What version of Asterisk are you using?
How were the music files transferred to *?
Some FTP programs default to ASCII and if you don't tell them to use Binary,
the file will transfer over with errors.
Done that several times and my MP3 files soudned horrible if they played at all.
Do you have
Robson Ribeiro wrote:
I am having a problem with MusicOnHold. Right now I have the following
configuration:
Default = mp3:/var/lib/asterisk/mohmp3
The problem is that I can't hear the music or sometimes the music seems to
skip like a scratched record...
You are not running mpg123 0.59r. Uninstall
Title: Polycom IP500 - Phone TIme
Does anyoe know where I can set the timezone in the configuration files?
I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter this into the gmt fields in ipmid.cfg nothing seems to happen.
Here are the fields...
tcpIpApp.sntp.address=
I would suggest opening up a bug on the tracker, if it hasn't been done
already, for all of these discussions to be logged to.
From my glancing at the code, I think there would be a little more cleanup
involved in it than just the ast_bridge_chan function. But either way being
part of the Manager
On Thursday 28 April 2005 11:07 am, barney wrote:
Hi there,
I`m trying to add some prefix before my local extensions, when my calls are
routed to ZAP trunk.
(i.e.: my local extension is , and i would like to send request to my
telco provider with source phone number 55)
Is
Wiley Siler wrote:
Does anyoe know where I can set the timezone in the configuration files?
I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter
this into the gmt fields in ipmid.cfg nothing seems to happen.
Here are the fields...
tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset=
I've only programmed my IP500 through the web interface. On the web
interface, I set the GMT Offset to -6 (for Central). It
works for me. BTW, I'm using pool.ntg.org as my sntp server.
Now what doesn't work, is that when I change something in the web
interface and the phone reboots, the time
Hello,
After upgrading from 1.0.1 to 1.0.7 I get no audio playback with the
playback and background commands. I'm running a 100% voip system over
an IAX channel. I've removed the bandwidth and disabled jitterbuffer in
the iax.conf file. Regular calls work fine.
My iax.conf file:
Title: Polycom IP500 - Phone TIme
Hi Everyone,
I am feeling sad :-(
I am sure some of you * Guru's will have some tips
for me.
Basic problem is my SIP phones work and talk to
each other no problem. I have a single 7920 WiFi phone which can receive calls
no prob at all. However when you
Hello everyone,
I don't know if it is just me, but I can never get a connection out of
Digium's FTP server. I can connect and login just fine, but both active
and passive ftp timeout before I can even get a directory listing...
This is from multiple OS's, networks, machines etc. It is
H my phones are static IP. Sooo...
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling aka ManxPower
Sent: Thursday, April 28, 2005 11:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hi all,
I need some help on a potential project. I am an open-source advocate,
and have been working with Linux for a very long time. I am not a VoIP
expert, even though I have been installing a few C*sco call managers. I
have been doing some development in the past, but I am now better at
Problem found
Polycom field below is in seconds not in hours like in the web
interface...
So for my -7 hour offset have to multiply 3600 (num of secs in 1 hour)
times the number of hours (7)
Value = -25200
That gives me Mountain time accurately.
For anyone else who needs time on their
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Wiley Siler
Sent: Thursday, April 28, 2005 12:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom IP500 - Phone TIme
H my phones are
Given that many providers offer IAX termination, I would think that would
indicate there were a lot of systems in use. I would ask the providers for
an estimate of the number of systems they are seeing.
Chris Mason
www.anguillaguide.com
-Original Message-
From: [EMAIL PROTECTED]
Hi,
i patched *-CVS-HEAD-04/28/05-19:20:40 with bristuff-0.2.0-RC8a-CVS and
now no callerid is transmitted to my SIP-phones. With 1.0.7 everything
works fine!
I want to use bristuff-0.2.0-RC8a-CVS - i have to set the
bearer-capabilities for fax-calls.
-- Accepting voice call from '16375733257'
My understanding of the T-1 TDM and the PSTN side is pretty solid, as it
is mainly based off of Intel Corporation's T1/E1 Technology Primer (see
Sources), but the CPE side is largely deduced from what I knew about the
PSTN side. There may be holes or mistakes, so I would appreciate any
Has anyone come across any software that can control adding/editing
SIP extension properties and perhaps dial plan properties on a context
basis. What I mean is I would like it so an admin user from Company A
can manipulate
properties for extensions in his context but not in another
On Thursday 28 April 2005 19:49, Kristian Kielhofner wrote:
Hello everyone,
I don't know if it is just me, but I can never get a connection out of
Digium's FTP server. I can connect and login just fine, but both active
and passive ftp timeout before I can even get a directory
We would gladly host an HTTP mirror. I can't seem to find the proper
contact at digium for this. Any suggestions?
On Thu, 2005-04-28 at 13:49 -0500, Kristian Kielhofner wrote:
Hello everyone,
I don't know if it is just me, but I can never get a connection out of
Digium's FTP server.
On Thu, 28 Apr 2005, Bob Goddard wrote:
The ftp server has been broken for months. If you keep trying
you will eventually get a listing or a file.
It has problems with passive mode, turn it off and it works fine.
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On Tue, Apr 26, 2005 at 04:31:59PM +0200, flavio patria wrote:
I have just installed a release version 1.0.7 of asterisk: I already
installed in past asterisk and in my previous installation I may find
the dial command on CLI that now I haven't found: it is possible?
The lack of dial CLI
Hi All,
I am replacing Cisco Call Manager with Asterisk. As you know CCM
is on a MCS 7835 Server which comes with a custom version of
Windows. Does any one know how to install Linux on that H/W. My
guess is that someone must have tried the same thing before. I
know how to install Linux however I
Eric Wieling aka ManxPower wrote:
Wiley Siler wrote:
Does anyoe know where I can set the timezone in the configuration files?
I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter
this into the gmt fields in ipmid.cfg nothing seems to happen.
Here are the fields...
Thanks, Don't know how I could have missed that, This works on incoming
calls to the station and calls from station to station. How do I make it
work if I dial out over a zap channel and then want to transfer to another
extension the # doesn't do anything except generate tone on the line. N0
Would some kind soul reply with the zap*.conf snippets needed to configure
2 PRI's with NFAS on a te410p?
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline [EMAIL
Can asterisk send AT commands to a modem?
If so, how?
I have two ISDN cards (with i4l - capi4linux doesn't work with them),
and would like to specify which card to choose for dialing out (without
it, i4l uses first free /dev/ttyI device).
Tomek
___
On Thursday 28 April 2005 1:52 pm, Greg Eaton wrote:
...
In terms of finding out true factual data, in terms of reference perhaps
digium would share market data (but then they might not!) and we could
extrapolate from there, but given how much of a community is building up
around the product I
Hi Everyone,
I have bought a Digium TDM400P and am using asterisk on RedHat 9.0. I was
able to configure Asterisk and SJPhone, so I have been able to call from IP
to IP and also from IP to a analog phone which is attached to the digium
card.
My problem now is to dial from an IP phone to an
We have lots of customers who want to be able to look at their Cisco 79XX
phone and see lines are in use.
Do hints work with Cisco phones?
Perhaps someone can clarify this: We have many 7960's. Which means 6 lines
right? Not really, cause each line must have a SIP username/password and
must login
On Tue, Apr 26, 2005 at 09:04:48AM -0500, Matthew Boehm wrote:
Trying to make a call via our PRI: (CVS everything,
CVS-NHEAD-04/23/05-16:08:12)
-- Executing Dial(IAX2/[EMAIL PROTECTED],
Zap/R2/2815699900|30) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called
Patrick Worte:
The problem is indeed unique to the TDM400 FXO daughter board. I can
confirm that the X100P and clones do correctly detect hangup on the BT
network, but are plagued by echo problems due the impedance mismatch
with the UK phone network.
Ok I might be beingd really dumb here, but if
Title: Asterisk Home .9 with TDM11B
Hi all,
I am new to Linux so sorry for asking stupid questions. I have the TDM11B(TDM 400) card and I cant get it to work at all. Can someone point me to the right direction?
Thanks
___
Asterisk-Users
Hello evrybody,
trying to compile spandsp 0.0.2pre15 and pre16 I got this
if /bin/sh ../libtool --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I.
-I-g -O2 -MT t31.lo -MD -MP -MF .deps/t31.Tpo -c -o t31.lo t31.c; \
then mv -f .deps/t31.Tpo .deps/t31.Plo; else rm -f .deps/t31.Tpo;
exit 1; fi
gcc
Amit Singla wrote:
Hi Everyone,
I have bought a Digium TDM400P and am using asterisk on RedHat 9.0. I
was able to configure Asterisk and SJPhone, so I have been able to call
from IP to IP and also from IP to a analog phone which is attached to
the digium card.
My problem now is to dial from an
Actually,I just setup the server to update its tiem and DHCP gets set
automatically on my domain controller/DHCP Server.
Then I set the IP statically in my phones and tell them the SNTP server
is my domain controller.
http://www.google.com/search?hl=enlr=q=set+time+net+windowsbtnG=Searc
h
On
I am almost positive that these are just HP/Compaq servers
- Original Message -
From: Walid Azab [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, April 28, 2005 3:07 PM
Subject: [Asterisk-Users] Install Asterisk on CCM MCS-7835 Server
Hi All,
I am replacing Cisco
What specific Dell Servers are you having trouble with, and what specific
TDM card?
I'm running dell 2650 and the TE410P and haven't had any problems... yet...
Duane Cox
- Original Message -
From: Matt Schulte [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday,
Hi
Everybody,
I have installed [EMAIL PROTECTED] 0.9 recently.
Since the very
beginning I was unable to listen to prompts or MoH. After doing 'yum update'
everything went back to normal except for the fact of having problems while
dialing to conference rooms: conference room was always
I have an asterisk box build from cvs stable that I am trying to use with 5
IP500 Polycom SIP phones. I can receive calls in through a digium wctdm
line. I can call out from the SIP IP500 phones to a PSTN number through the
same card. In other words, incoming and outgoing calls work just fine. It
On Wed, 2005-04-27 at 23:22 -0400, Steven Kalcevich wrote:
I think its a win win situation. Cisco has tons of money to throw at
them to get a better product with more features. I dont believe they
would aquire them and not put money in them to make a better product.
I guess the
http://lists.digium.com/pipermail/asterisk-users/2005-April/103136.html
On 4/28/05, Daniel Salama [EMAIL PROTECTED] wrote:
Does anyone know what this mean?
Apr 28 11:53:51 WARNING[907]: chan_iax2.c:6039 socket_read: Received
mini frame before first full voice frame
Thanks,
Daniel
Thank you,
I have been watching with interest your postings.
While I have not read everything, I have stored your messages.
I think your contributions will inspire a new VoIP soft switch movement.
Race The Tyrant Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Thanks,
Daniel
On Apr 28, 2005, at 5:31 PM, Andy Hamilton wrote:
http://lists.digium.com/pipermail/asterisk-users/2005-April/103136.html
On 4/28/05, Daniel Salama [EMAIL PROTECTED] wrote:
Does anyone know what this mean?
Apr 28 11:53:51 WARNING[907]: chan_iax2.c:6039 socket_read: Received
mini
Oh boy I am getting crazy...
I installed an asterisk with J4BRI ( 3 BRI point to point ) , Snom phones
and everything works fine. Where's
the problem? Well I can not get music on hold.. Well really MusicOnHold
works, works on Queue, works on #
transfer , but hell it does not work when I press the
In 1998 Cisco purchased a company called Summa Four for $116 Million,
and left them to die on the vine. It all depends on what they (Cisco )
want from the transaction. If Sipura has a part in causing a drop in
Cisco revenue due to adoption by the Open source community, then they
may well buy the
Andres wrote:
In 1998 Cisco purchased a company called Summa Four for $116 Million,
and left them to die on the vine. It all depends on what they (Cisco )
want from the transaction. If Sipura has a part in causing a drop in
Cisco revenue due to adoption by the Open source community, then they
may
Hi,
I've been reading on the wiki as well as on this list, different
suggestions of what to look for when designing an asterisk server with
a lot of traffic. By a lot of traffic, I mean a box with a a TE4XXP,
that will be hit to full capacity (96 simultaneous calls). This box
will also deliver
when you have asterisk install first log into the console and type
help-aah for some more info. Look also at the [EMAIL PROTECTED] handbook
online.
Then for the TDM11B card do a genzaptelconf -s -d.
Then you need to setup a zap trunk via your web interface. you might also
want to look at the
Perhaps someone else can make sense of this but I can't figure out why
asterisk is not paying attention to the NAT setting. Asterisk keeps trying
to use the UA's internal address to respond too.
We have 10 of these phones in another situation and they work fine.
The originating IP address (the
David Boyd wrote:
In 1998 Cisco purchased a company called Summa Four for $116
Million, and left them to die on the vine. It all depends on
what they (Cisco ) want from the transaction. If Sipura has
a part in causing a drop in Cisco revenue due to adoption by
the Open source
Brian Leyton wrote:
Well, I tend to think that Sipura's products will come out ahead here. For
example, when Cisco bought Linksys, one might have expected them to cripple
or shut them down - after all, Linksys makes routers, and Cisco makes
routers. Obviously that didn't happen, because Cisco
There are no firewalls involved. I've got a simple 10/100 switch running and
all connections involved are 100bT FD. switch configuration, nothing
spectacular. Both phones I am testing and the asterisk server are on the
same subnet, there are no routers or firewalls between any of the devices
I have an asterisk server and 4 Sipura phones behind a Linksys WRT54G
router. I have set the DMZ to the Asterisk server's IP so that it can be
seen from outside. I have a Sipura SPA-841 phone outside the router and set
to proxy to the public IP of the router. The outside phone registers fine,
Hey guys,
Got a customer who wants to integrate some video conferencing into his
company. He hasn't purchased it yet so I'm looking for some suggestions,
brand names specifically. I'm looking at Polycom and Tandberg. He really
only needs the camera/unit not a tv (already has one). If the camera
I just read a great paper that said turn off anything that won't be
used. Serial, USB , Printer ports, ETC. No Xwindows!
Daniel Salama wrote:
Hi,
I've been reading on the wiki as well as on this list, different
suggestions of what to look for when designing an asterisk server with
a lot of
I have never been able to do more than 50 concurrent recordings with Zap -
SIP phone calls without the audio skipping and/or breaking up. Also, if you
are using Digium TE4XXP and want to do a lot of recording I would recommend
against a SCSI RAID card because of the interrupt conflicts that you
Michael,
Have you decided which PSTN-VoIP gateway you'll use?
Not yet, but our preference is a Cisco gateway. Lucent, Quintum, and
AudioCodes also make TDM-VoIP gateways.
Prior to purchasing any hardware, our entire layout will be posted to
this list in detail for review.
Matthew Roth
Greetings:
Can anyone point me in the redirection in troubleshooting a problem of
garbled voice audio
that three of my remote client on our Asterisk network are having.
We have the following configuration: I have two remote asterisk clients
going through a cisco routet to our
external
Could you point me in the direction where you read that? Maybe there is
more there to read.
Thanks,
Daniel
On Apr 28, 2005, at 6:31 PM, Michael D Schelin wrote:
I just read a great paper that said turn off anything that won't be
used. Serial, USB , Printer ports, ETC. No Xwindows!
Daniel
Hi, I have an asterisk server with a quad E1 card, Span 1 to PSTN, and calls
flowing fine to and from the PSTN.
I would now like to connect a legacy PBX to span 2 of the E1 card, I made a
crossover, configured zapta.conf and zaptel.conf.
Zttools shows both spans as up and active, but I am now
Thank you again. I will definitely do that. By cheaper asterisk
servers, do you mean single-CPU machines that can handle Quad T1s and
still do the call monitoring?
BTW, I tried the monitoring without the 'm' option and mounted the
audio directory via NFS. Big NO NO for everyone. Just do what
Daniel Salama wrote:
Thank you again. I will definitely do that. By cheaper asterisk
servers, do you mean single-CPU machines that can handle Quad T1s and
still do the call monitoring?
BTW, I tried the monitoring without the 'm' option and mounted the
audio directory via NFS. Big NO NO for
I was having problems with voicepulse about a week or two ago...
Incoming calls would fail, and one incoming call, would block all outgoing
calls.
Then one day, the DTMF tones stopped working.
I could call into Asterisk, but I could not navigate because my tones were
being ignored.
I ran
Daniel,
Could you expand upon your experience recording to an NFS mounted drive.
We are looking to use a TDM-VoIP gateway to route 16+ spans to a single
Asterisk server. We were hoping to Monitor using the following scheme:
- Monitor application executed on Asterisk server (no 'm' flag)
- Pick
George Pajari wrote:
In capacity planning for production Asterisk servers it is essential to
have an accurate statistical picture of the utilisation of finite
resources such as disk space, CPU utilisation, B channels on PRIs, and
G.729 codec licences.
The first two have well defined
The ftp server has been broken for months. If you keep trying
you will eventually get a listing or a file.
i'm using
ftp://ftp.ipex.cz/pub/mirrors/ftp.asterisk.org/
---
Marek Cervenka
Centrum Vypocetni Techniky
CVT - http://cvt.fpf.slu.cz
FPF SLU
Hello,
I can´t get chan_capi to work with any version of asterisk. I tried several
versions, all with the same effect: the phone rings, as soon as the call gets
answerd, asterisk crashes. This only happens with chan_capi, chan_sip and
chan_zap work flawlessly.
Unfortunately, I dont have a
George Pajari wrote:
In capacity planning for production Asterisk servers it is essential
to have an accurate statistical picture of the utilisation of finite
resources such as disk space, CPU utilisation, B channels on PRIs, and
G.729 codec licences.
The first two have well defined
It is my understanding that TDM is circuit switched and VoIP is packet
switched. It would seem to me that at some point in a TDM-VoIP gateway,
a change from circuit switching to packet switching is happening, and
vice versa depending on the direction of the signals. I was just
wondering if
Guys
I have a problem getting a TDM400P card to go.
It has 4 FXS ports (green modules) and I get this error:
[EMAIL PROTECTED] root]# ztcfg -v
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default)
Max TNT's are pretty cheap they'll need to price it accordingly.
On Thu, 28 Apr 2005, David Josephson wrote:
Your reference picture is fine ... but note that Asterisk can be the TDM/VoIP
gateway, particularly when Digium releases their DS3 card (644 voice
channels!) working, a lot more
Guys, is there any way to generate simulated traffic via sip or IAX2 for
testing cpu load and asterisk? (sip client simulation, etc)?
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
Anton Krall
Sent: Thursday, April 28, 2005 6:07 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Traffic Testing
Guys, is there any way to generate
Hi, just wanted to ask if there is anyone here ever tried
creating an extension, then get a did from a pstn network, route the call to
that extension. The catch is, the extension and did is of similar value.
Example: extension = 6945192
did = 6945192
Please help me find out how I
Thx!
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|[EMAIL PROTECTED]
|Sent: Jueves, 28 de Abril de 2005 07:05 p.m.
|To: asterisk-users@lists.digium.com
|Subject: RE: [Asterisk-Users] Traffic Testing
|
|
| -Original Message-
| From: [EMAIL
--- Tim Burt [EMAIL PROTECTED] wrote:
I was having problems with voicepulse about a week
or two ago...
Incoming calls would fail, and one incoming call,
would block all outgoing
calls.
Then one day, the DTMF tones stopped working.
I could call into Asterisk, but I could not
On April 28, 2005 08:02 pm, Matt Klein wrote:
Max TNT's are pretty cheap they'll need to price it accordingly.
Be careful.
MaxTNTs are fine, but you need a fully unlocked shelf controller to do
SIP/h323. You also need their fancy ethernet card; for whatever reason their
standard ones
Can you send some command line examples on how to use it?
Thx!
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|[EMAIL PROTECTED]
|Sent: Jueves, 28 de Abril de 2005 07:05 p.m.
|To: asterisk-users@lists.digium.com
|Subject: RE: [Asterisk-Users]
You can throw together a single P4 3GHz with 1GB RAM and 2 x 80GB SATA HD
for about $600. One of those can easily handle a Sangoma dual T1 card($900)
or a Digium quad T1 card($1400). For that you can have a system for about
$1500-$2000 that will be able to fully record 2 T1s(48 channels) worth of
I use Opexld.com for my 800 service, then send the number to a did.
More reliable this way in case a provider has problems.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean
Kennedy
Sent: Thursday, April 28, 2005 9:29 AM
To:
To comment on all the different voip connection problems that are
popping up, this does not suprise me, a couple weeks ago the FCC
(america) ruled that all voip providers that connect to the PSTN
(vonage, broadvoice, voicepulse, etc) have to have CALEA support
(wiretap equipment for law
I'm using dtmfmode=inband with Sipura=3000 when I dial an internal
extension most of the time the first digit is missing and I get an
invalid extension message.
Could it be dtmf problem or SIP?
--
#Joseph
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Asterisk-Users mailing list
This works from-pstn just fine.
exten = 10 digit Inbound PhoneNum from the
pstn,1,Dial(sip/[EMAIL PROTECTED] IP address,,r,)
How can I add the Variable exten with the proxie IP address. I want the
exten to call my proxie.
exten = _.,1,Dial(sip/[EMAIL PROTECTED] IP address,,r,)
Hello Tomek,
I suspect I may have located my problem. The Hisax driver supports
NI-1, but I believe my CO is sending signalling in 5ESS, which does not
seem to be supported by the hisax driver
Looks like it's back to the drawing board...
Greg
-Original Message-
From: [EMAIL
This is great information. I have the following questions based on a
hypothetical scenario and some assumptions:
Based on the price of these configurations, I wouldn't even mind
putting two servers each with 2 T1s just so that I could get all calls
recorded and distribute the risk of failure.
Wtihout seeing any conf files...it's a bit hard to say
Are you sure you haven't don't something strage with your dial plans?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Friday, 29 April 2005 11:53 AM
To: Asterisk Users Mailing List
Wiley Siler wrote:
Does anyoe know where I can set the timezone in the configuration
files?
I am in Phoenix, AZ which has a GMT offset of -7 hours but when I
enter
this into the gmt fields in ipmid.cfg nothing seems to happen.
Here are the fields...
tcpIpApp.sntp.address=
On Fri, 2005-04-29 at 12:38 +1000, David Phelan wrote:
Wtihout seeing any conf files...it's a bit hard to say
Are you sure you haven't don't something strage with your dial plans?
Here are the sip.conf and extension.conf; when I dial for example 218
command line display 18 (missing first
All,
I have the following config problem with dtmfmode
I use ANTEK gw which only support dtmfmode=info but it is not
supported in Asterisk voicemail. I wonder if it is possilbe to setup
config that is runtime determined. I mean say, if I dial to voicemail then
the asterisk can choose
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Morin
Sent: Thursday, April 28, 2005 11:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom IP500 - Phone TIme
Wiley Siler wrote:
Does anyoe know
You ar making this a lot harder than it is.
If your incomming trunk is in the same context and there is an extension that
matches the DID the did will route to the correct place. Matter of fact I
always try to match DID to extension. Usually I have the telco only send
me the last 4 of the
On 4/28/05, Joseph [EMAIL PROTECTED] wrote:
I'm using dtmfmode=inband with Sipura=3000 when I dial an internal
extension most of the time the first digit is missing and I get an
invalid extension message.
Could it be dtmf problem or SIP?
This is covered in the sipura SPA-3000 faq. Can't
Hello,
I am facing difficulty in restricting the number on agents
logged onto a SIP extension to one. Can someone help me with this.
Thanx.
Kashif.
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snip
I think it would be really easy to make a PHP page that just list the
monitor files. All files are stored in /var/spool/asterisk/monitor/.
You could then click the link to listen to it.
I wrote some Mono ASP.NET pages for such a purpose. I can make them
available if there's enough
The voicemail is there. When a user hits hit message button and enters
in his password the voicemail will play but sometimes will stop and go
back to the menu. The user will hit 5 (replay the current message) and
the voicemail will either play all the way through or stop again. The
user will
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