Tzafrir Cohen wrote:
On Tue, Aug 22, 2006 at 08:42:36AM -0500, Rich Adamson wrote:
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
I did yum update last week and here is my current kernel:
I had no problem at all with zaptel. I am only using TDM400P though, in
Which version of Web-MeetMe did you download? The
process up to 2.0.1 is, well, annoying.
Copy app_cbmysql.c to ./asterisk/apps and modify the
Makefile to include the application.
The project is now hosted on SourceForge and has a much
improved build process, but I have
not built a release
-Original Message-
From: Matthew Crocker [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 22, 2006 10:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime Extensions -- Comments?
Add a boolean field to the table then create a
What version of asterisk ?
Julian
Steve Hanselman wrote:
Is there any reason why I can’t use the xxx/callerid format in an
include section?
It doesn’t seem to work, but if I paste the lines into the main section
where I include the block it does?
E.g. this doesn’t work
note http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions
comment from Philipp Dunkel.
On 22 Aug 2006, at 17:13, Douglas Garstang wrote:
-Original Message-
From: Jeremy McNamara [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 22, 2006 9:45 AM
To: Asterisk Users Mailing
He meant he added another column in the database table...
In the asterisk source everything database related for the default database
stuff is explicitly named...
Like INSERT INTO blah(col1,col2,col3) Values(foo,foo,foo)
- Original Message -
From: Douglas Garstang [EMAIL PROTECTED]
Thanks Kevin,
This is very exciting news. Let me know what if anything I can do to help
test this. I have a Grandstream GXP2000 and Aastra 9133i phone. What
sort's of phones and or features are required to support this? The Aastra
supports Broadworks SLA and standard SIP BLF but not alternate
On Tue, 2006-08-22 at 10:14 -0600, Douglas Garstang wrote:
-Original Message-
From: Jason Parker [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 22, 2006 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime Extensions --
Hi Julian,
Ah, a very good point, I put that in my first cut but had completely
forgotten in this one!
1.2.10
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 22 August 2006 17:30
To: Asterisk Users Mailing List -
- Douglas Garstang [EMAIL PROTECTED] wrote:
-Original Message-
From: Jason Parker [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 22, 2006 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime Extensions -- Comments?
On 8/20/06, Rich Adamson [EMAIL PROTECTED] wrote:
voiplist wrote:
Any ideas? Would be nice to use this feature as we have with other
Sipura products in the past.
Sounds like a config problem with asterisk. The 941/942's here worked
just fine right out of the box.
941 message waiting is
Personally I will build multi-thread server to work as AGI server.
Apache is an option if you know how to implement programs on web server,
but know how to build your server. You can hardcode the HTTP header into
a global variable in your dialplan, and send the header to Apache first,
then follow
Hi,
I'm trying to get my asterisk recieve faxes via rxfax and spandsp. It creates
the connection, recieves some info about the sender and dies. Result is not
valid tif file of size 330 or 334 B. Debug messages are shown only sometimes
and it ends with something like FLOW Fast carrier
Hi,
I'm trying to get my asterisk recieve faxes via rxfax and spandsp. It
creates
the connection, recieves some info about the sender and dies. Result is
not
valid tif file of size 330 or 334 B. Debug messages are shown only
sometimes
and it ends with something like FLOW Fast carrier
I suspect that your dialplan is more than you show ;)
It works just fine for me with svn trunk
[from-sip]
include = common
[common]
exten = 1234,1,NoOp(Hmm ${CALLERID(num)})
exten = 1234/7708,1,NoOp(Here)
If I dial 1234 from my 7708 extension, I get the NoOp(Here)
If I dial 1234 from my
Yes, but that doesn't stop Asterisk from still treating that row as a valid
extension.
-Original Message-
From: Don [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 22, 2006 10:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime
How do you lower the volume of MP3 hold music?
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or
I have the opposite problem. I can hardly hear the hold music at all.On 8/22/06, Dennis P. Clark [EMAIL PROTECTED]
wrote:How do you lower the volume of MP3 hold music?Dennis Clark
DENPROWRK 207.618.1998CEL 443.415.0527FAX 1.888.811.8809[EMAIL
David Freeman wrote:
I have the opposite problem. I can hardly hear the hold music at all.
On 8/22/06, *Dennis P. Clark* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
How do you lower the volume of MP3 hold music?
I'm certainly not an expert on MOH, but I don't believe there are
Hi friends,I did music on hold. How can we implement music on call transfer? I am unable to find any tutorial about setting up music on call transfer, when call is transferring from one extension to another extension. So, I have two doubts. They are:1) How can I put music on call transfer (Not in
Hello,
Have someone implemented * like as a CTI
platform with IVR, VoiceMail, Fax to tiff files, etc etc using Digium/Sangoma
Dual T1/E1 interface cards? Does it work ok? Is the audio quality good when all the
ports are in use? Is there any issue regarding on faxes on digital trunks?
How do I enter a trunk with multiple IPs.
xyz voip provider has 4 IPs and I want to allow incoming from any of
them: 1.1.1.1, 1.1.1.2, 1.1.1.3 and 1.1.1.4
Do I put 4 separate host= lines, do I put a single host=line that is
comma separated or do I have to set up 4 separate incoming trunks?
I was able to get it working with a meetme.
The D and M options seemed to lock the audio channel for too long.
This is what worked:
exten = 5481,1,NoOp(${TIMESTAMP} paging Group 1 Office Page)
exten = 5481,2,DIAL(Zap/g2/5110,,G(ext-local-custom^5485^1))
exten = 5482,1,NoOp(${TIMESTAMP} paging
If MP3s are too loud then their should be an internal function that
modifies all MP3s in the folder to be at one consistent volume
(normalization). Anybody know of a way to do this?
Here is my musiconhold.conf just in case I missed something.
[classes]
Random = quietmp3:/var/lib/asterisk/mohmp3
On Tue, Aug 22, 2006 at 11:16:38AM -0500, Rich Adamson wrote:
Tzafrir Cohen wrote:
On Tue, Aug 22, 2006 at 08:42:36AM -0500, Rich Adamson wrote:
Tomislav Parčina wrote:
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
I did yum update last week and here is my current kernel:
I had
Im not sure, but there is a commented column that could have 0(not
commented) or 1(commented) as values.
Is this right?
P.S.: I got it from voip--info.org on the realtime Static page...
D e n i s G a l v ã o
iSolve - Solve Is Our Business
Av. Candido de Abreu, 526 1610A
CEP: 80530-000 -
Hey David,
Yes, it can, you just have to play around with the logic and what you
are comparing and when you can do the comparison.
Try something like this:
exten = _18XXNXX,1, NoOP()
exten = _18XXNXX,n,gotoif(${EXTEN}:2:2 = 00 | ${EXTEN}:2:2 =
66 | ${EXTEN}:2:2 = 77 | ${EXTEN}:2:2 =
I know I am responding to an old post but dont think you would want to
change the title of your site from osCommerce to your name ?
- Original Message -
From: Sam Tam [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent:
Hi Everyone,
I am looking into implementing a PRI with our asterisk system. Does anyone recommend a certain PRI to T1 ethernet bridge like red-phone? Are their any installation hiccups I should look out for?
Thanks
Julian
___
--Bandwidth and
Assuming you use Perl for AGI scripting, which you should be doing
anyways ;-)
*cough* You made a typo... you really meant to say 'python'. :P
flame
Python is to Perl what Pascal was to C. A nice toy ;-)
/flame
___
--Bandwidth and
I am trying to configure Eicon's SIP version of SoftIP to work with Asterisk.
The documentation is poor and my general knowledge of SIP communications is
limited. I am getting a circuits busy message when trying to call the IP
address of the server with SoftIP. If anybody has gotten this to
I am running Asterisk 1.2.7 and SpanDSP0.0.2 with Kernel version
2.6.17-1.2141_FC4. I am getting an error after compiling the SpanDSP
and putting the .c files in the correct place and then patching the
make file. All goes well but then when asterisk is opened, I get this
message:
I'm using a Windows software call mp3gain. It can normalize directory.
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Dennis P.
Clark
Envoyé : 22 août 2006 15:35
À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Objet
Most of the time, the
sample provided by Polycom is the same than the other version. So you dont
need to use the new xml. If they add something, its written in the
release (pdf) so you just need to add what they say.
David
De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la
.
*
-- next part --
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060822/245f726d/attachment-0001.htm
--
Message: 5
Date: Tue, 22 Aug 2006 10:19:55 -0600
From: Douglas Garstang [EMAIL
Hi,
I'm thinking on setting up an asterisk server with two providers. One
will let us to make international calls and provide to us a TollFree
number. The other will provide local numbers (i'm from Argentina). The
problem is that the local number provider requires a dedicated
connection and the
If you're looking for alternatives to Zetafax why not look at AsterFax
(http://asterfax.sourceforge.net)? Your clients can use their existing
e-mail client to send faxes.
Warrick
___
--Bandwidth and Colocation provided by Easynews.com --
Is there a way to have asterisk send an email when a extension disappears or is
disconnected?
Roger Workman
Business Development
Upperclassman/Universal Holdings LLC
Voice: 304.324.3800
Fax: 304.324.3801
ICQ: 4447584
FWD Network: 56505
Website: http://www.upperclassman.net
Billing Questions:
Hi,
I am getting the following message on the CLI:
-- Got SIP response 480 Temporarily Unavailable back from 192.168.1.60
-- SIP/EXT23-d910 is circuit-busy
and the call hangs up.
The peer is correctly registered and I'm not getting unavailable messages.
I really need help with this error.
Greetings!
Asterisk 1.2.10, Zaptel 1.2.7
For AGES have been using a box setup with 1.0.9 on it and an clone X100P
FXO card. Right now, it's only set up to listen to the PSTN line and
grab the CLID and shove that to an AGI so it can IM me who is calling.
However, reinstalled the box from
Have you run SIP DEBUG PEER 192.168.1.60? It may help...tcpdump is also one
of my personal favorites
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Diego Andres Asenjo G.
Sent: Tuesday, August 22, 2006 6:50 PM
To: asterisk-users@lists.digium.com
Hi Everyone
Any opinions on this?
Thanks
Julian
From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Tue, 18 Jul 2006 01:29:57 +Subject: [asterisk-users] PRI and Asterisk
Hi All,I am planning to order a PRI and would like to know your opinions on a devices like the Redfone
Thanks Kevin! That's what is great about these forums. I never thought
of using gotoif() inside ... one of those Doh! moments.
I included your concept in my standard [dial-ld] context with
${EXTEN}:1:3=800, etc. rather than by 2's, (so it doesn't overlap with
8XX area codes) and select my
Christian Jensen wrote:
I am running Asterisk 1.2.7 and SpanDSP0.0.2 with Kernel version
2.6.17-1.2141_FC4. I am getting an error after compiling the SpanDSP
and putting the .c files in the correct place and then patching the
make file. All goes well but then when asterisk is opened, I get
Hi, I'm using the hint extension to monitoring the status of some
extensions. If the extension is defined as a friend, the monitoring
doesn't work any more. It only work if I define it as a peer. Is that
right ? I mean, I supposed that an extension defined as a friend should
have all the
Rushowr wrote:
Have you run SIP DEBUG PEER 192.168.1.60? It may help...tcpdump is also one
of my personal favorites
Yes, I have used it. The lines are extracted from a sip debug on the
CLI. I'm going to paste more lines:
Sip read:
SIP/2.0 480 Temporarily Unavailable
To: sip:[EMAIL
Hi,
this question may sound a little dumb, but I need opinion of who are
already using asterisk.
questions are :
1) which format is best suited for asterisk (.gsm, .wav etc, also what
sampling rate and bit size)
2) What are the best sources (cost effective) to get prompts recorded.
thanks in
Is there anyway to set the Contact header on outbound INVITEs
such as there is for the REGISTER? I would also like to be able to set the
Contact header on responses.
Thanks,
Michael
This email may contain confidential information. If you are not the intended recipient, please
Hello,
I'm trying to do click-to-dial using a script from
* http://www.azxws.com/asterisk/
It is a perl script that talks to the Asterisk Manager.
I have my asterisk box setup and registering with the provider but when
I execute the script, I can see from my ngrep dumps that asterisk is
Im thinking of taking another run at www.Tellme.com to set up an open access
Pay-As-You-Go SIP gateway for their Speech Recognition services.
I tried to do this about a year ago http://www.voip-info.org/wiki/view/Tellme
and whilst the initial enthusiasm was good they ended up more or
What would the recommended timing source be other than zaptel cards for
Asterisk svn trunk on Linux 2.6.x?
I have a server that was running with 2 te410p's. I've converted to using
SIP so I don't need the cards anymore -- other than to provide a timing
source for our applications -- mostly
Never tested Redfone box.
Digium and Sangoma cards works fine for me.
Jorge Mendoza
Julian Varanini wrote:
Hi Everyone
Any opinions on this?
Thanks
Julian
From: [EMAIL PROTECTED]
To:
Hello -I'm searching for a simple php or perl script to parse Asterisk's CDR csv into a formatted webpage - anyone have any suggestions?-- --Christopher T Aloi--
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
Hi
Im using asterisk 1.0.9 with Linux Ubuntu 5.10 and
I have a problem with the time in the cdr-csv. The time registered for the calls
is 5 hours earlier than he actual time, it seems respond to the COT and not to
the UCT that represents the actual time
#date -u
Tue Aug 22 19:52:16 UCT 2006
Are you having this problem with the trunk?
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Lucas Alvarez
Envoyé : 22 août 2006 18:23
À : Asterisk Developers Mailing List; Asterisk Users Mailing List -
Non-Commercial Discussion
Objet : [asterisk-users]
JOSE MANUEL CORTES DAVID wrote:
Does anybody know how to fix this?
You can start by using a recent version of Asterisk. Current
development work is heading toward v1.4.
v1.0 is stupid old.
Then ensure your Timezone is set properly.
Jeremy McNamara
The Asterisk Development Team is pleased to announce new releases of
three of our projects:
Asterisk 1.2.11 includes a number of bug fixes, along with an update to
the chan_misdn driver for mISDN devices, including Digium's new B410P
quad BRI interface card.
Asterisk-Addons 1.2.4 includes quite
I need to give Asterisk access to my external IP address to prevent
the NAT problem where caller cannot hear the callee's voice.
According to Asterisk - The Future of Telephony page 92 Environment
Variables:
Environment variables are a way of accessing Unix environment
variables from
Hello
I'm having a problem with the Linksys 3102: With incoming PSTN calls, I
can hear the caller through the X-Ten softphone, but he can't hear me. The
problem is worse with Sjphone and the GrandStream 100 hardphone, as I get
no sound in either direction.
FWIW...
- the SIP client, the
Diego Andres Asenjo G. wrote:
Hi,
I am getting the following message on the CLI:
-- Got SIP response 480 Temporarily Unavailable back from 192.168.1.60
-- SIP/EXT23-d910 is circuit-busy
and the call hangs up.
The peer is correctly registered and I'm not getting unavailable messages.
I
On 8/21/06, Noah Miller [EMAIL PROTECTED] wrote:
Hi Doug -
One thing you could try that may or may not help: a different FTP
server. I've been using ProFTPd (particularly because you can
configure it to use the Polycom default username and password, so you
don't have to manually type in those
DM wrote:
Why do you think the problem may be with the FTP server? I've been
running vsftpd on several different systems, all with Polycom's.
There were reports that the Polycoms preferred some FTP servers over
others, but I also use vsftpd (using the default PlcmsSpIp
username/password
I had the same problem.
The problem was another sip extensions whit the same ip.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Rich Adamson
Enviado el: Martes, 22 de Agosto de 2006 11:21 p.m.
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Is anybody making calls over VPN? If so what is the penalty as
encryption is involved.
I was planning to use VPN to register Sipura units to my local asterisk
this way I don't have to deal with NAT issues.
--
#Joseph
___
--Bandwidth and Colocation
We did a setup of 70 sites connected back to a central Asterisk box, and
it worked very well over an MPLS VPN.
regards,
PaulH
AsteriskIT
www.asteriskit.com.au
On Tue, 2006-08-22 at 20:43 -0600, Joseph wrote:
Is anybody making calls over VPN? If so what is the penalty as
encryption is
We did a setup of 70 sites connected back to a central Asterisk box, and
it worked very well over an MPLS VPN.
regards,
PaulH
AsteriskIT
www.asteriskit.com.au
The best part about VPN is that it makes it harder for the ISPs to track
and mess with. ;-)
--
Henry J. Cobb
MPLS is a VPN, but it doesn't use encryption in most cases.-brandonOn 8/22/06, Paul Hales [EMAIL PROTECTED]
wrote:We did a setup of 70 sites connected back to a central Asterisk box, and
it worked very well over an MPLS VPN.regards,PaulHAsteriskITwww.asteriskit.com.auOn Tue, 2006-08-22 at 20:43
Funnily enough, in this case the ISP supported our choice of running
Asterisk...
PaulH
AsteriskIT
www.asteriskit.com.au
On Tue, 2006-08-22 at 22:04 -0500, Henry J. Cobb wrote:
We did a setup of 70 sites connected back to a central Asterisk box, and
it worked very well over an MPLS VPN.
I can login on the web http://myasterisk.com/cgi-bin/vmail.cgi without
problems but i cant see the messages on any folder.
Thanks, Sergio.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
Larry, am I missing something but you seem to be putting the externip
into the MYIP variable but reading some EXTERNIP variable through
$ENV{}. Shouldn't you be doing something like externip=${ENV{MYIP}}?
The other issue is also the use of curly brackets as opposed to
paranthesis. The snip from
Hi List,
I want to do a layout like this:
Corporate-Asterisk01
Site A-Asterisk02Site B-Asterisk03
I will have phones register to each server at each location, and also want to store the users voicemail there.
Now here is my question.
Can I setup the phones (Polycom I was hoping) to
I was thinking of using openVPN
--
#Joseph
On Tue, 2006-08-22 at 22:06 -0500, Brandon Galbraith wrote:
MPLS is a VPN, but it doesn't use encryption in most cases.
-brandon
On 8/22/06, Paul Hales [EMAIL PROTECTED] wrote:
We did a setup of 70 sites connected back to a
Nitin,
I'm sure others have better advice but there's no best format per
se. Whatever makes asterisk and more importantly the CPU work less in
playing those prompts is probably best. from what I understand (*)
picks up the best suited format based on the capabilities of the
channel and endpoint.
RR wrote:
Larry, am I missing something but you seem to be putting the externip
into the MYIP variable but reading some EXTERNIP variable through
$ENV{}. Shouldn't you be doing something like externip=${ENV{MYIP}}?
The other issue is also the use of curly brackets as opposed to
paranthesis. The
Can you explain what you mean by disappears? or by disconnected?
On 8/22/06, Roger Workman [EMAIL PROTECTED] wrote:
Is there a way to have asterisk send an email when a extension disappears or is
disconnected?
Roger Workman
Business Development
Upperclassman/Universal Holdings LLC
Voice:
Mate, I'm beginning to think that it can't be done. As in, maybe
you're not allowed to put anything into externip other a valid IP
address and the $ENV{} variable doesn't really work there. You might
want to decipher your externip by registering your server with a
dynamic dns service and then
Hi friends,I did music on hold. How can we implement music on call transfer? I am unable to find any tutorial about setting up music on call transfer, when call is transferring from one extension to another extension. So, I have two doubts. They are:1) How can I put music on call transfer (Not
Wait a minutewhy are you putting 227 into the CALLERID function? You
should read this:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid
The (number) portion is the argument to CALLERID telling it what to give
you, not what to insert/write
-Original Message-
From:
On Tue, Aug 22, 2006 at 03:27:57PM +0200, Jan Fousek wrote:
Hi,
I'm trying to get my asterisk recieve faxes via rxfax and spandsp. It creates
the connection, recieves some info about the sender and dies. Result is not
valid tif file of size 330 or 334 B. Debug messages are shown only
On Wed, Aug 23, 2006 at 08:28:51AM +1000, Warrick Zedi wrote:
If you're looking for alternatives to Zetafax why not look at AsterFax
(http://asterfax.sourceforge.net)? Your clients can use their existing
e-mail client to send faxes.
AsterFax is a rather comlicated setup. Hylafax+IAXModem
Tzafrir,
When last did you look at AsterFax? What do you believe is required to
set it up? In what way are there wheel reinventings in either HylaFax
or AsterFax?
Tzafrir Cohen wrote:
On Wed, Aug 23, 2006 at 08:28:51AM +1000, Warrick Zedi wrote:
If you're looking for alternatives to
Today I had a brief power outage which caused the Asterisk server and
DSL modem to reboot. The Asterisk server came up before the internet
connection was working, so it failed when try to look up some of the
hosts for my outbound voip providers in sip.conf.
Asterisk never recovered from that,
Updating Asterisk is worth a go - I know of someone else who contacted
us with a distorted music on hold problem, and an Asterisk updated fixed
it.
PaulH
AsteriskIT
www.asteriskit.com.au
On Mon, 2006-08-21 at 16:14 +0800, Nathan Alberti wrote:
On 20/08/2006, at 8:38 PM, Paul Hales wrote:
Today I had a brief power outage which caused the Asterisk server and
DSL modem to reboot. The Asterisk server came up before the internet
connection was working, so it failed when try to look up some of the
hosts for my outbound voip providers in sip.conf.
Asterisk never recovered from
I have the similar problem but with IAX.I have two servers. First is primary with dynamic IP and open 4569 port. Second is behind firewall.If first server is being disconnected for some time, the second server cannot reconnect. I have to either restart asterisk or stop it, wait for some time and
Hello good people,
I'm sure this has been brought up previously but I basically wanted to
wait to resurrect this topic till 1.2.10 has been out for a little
while, like a cpl of mths. Now I think it has and I just wanted to
request for peope who've chosen to upgrade their systems to 1.2.10 to
James Harper wrote:
Are you sure that it was Asterisk? Did you try an nslookup after the
network came up while Asterisk wasn't working? How long did you leave it
before taking matters into your own hands?
Fairly sure. I didn't realize there was a problem until about 6 hours
after the power
Hello everyone,
I have a question that seems simple, but I am stumped. Basically, I have
several incoming SIP trunk from gateways connected to physical E1 lines
around the world. Everything works, but the DNIS that comes in are non
standard and sometimes conflict. It varies not only by
Hi
all,
I'm using a Snom360
with bristuffed asterisk and iwant to known if is possibile
realizesomthing of this: I receive an incoming call andthen
answeredI want to transfer it to a cell phone (or another pubblic number),
so press "transfer" on the phone, call the number and only if the
Hello
Ive been trying to use the lbProxy SIP load balancing proxy.
After I actually got it compiled, using the CMSOFAZ.COM version, I began
experimenting.
I quickly ran into a problem. Heres my setup:
wan--|lan
Phone - lbProxy - Asterisk
lbProxy sends all of the sip packets
Hi,all:
in /etc/asterisk/h323.conf
I setting
gatekeeper=192.168.0.19
secret=3001
and on server 192.168.0.19 I running a openh323gk
and add a user
3001 and password is 3001 too, but when I booting asterisk, I
got
messages :
Error registering with gatekeeper
Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5 over
Grandstream HT488 ATA. I have plugged gateway to FXO port of ATA. Incoming
calls work the way I call GSM number and then I get DISA to call inside
company. Outgoing call work well when I call VoIP number of
In the above, Jean-Michel puts it right on the table: of what
possible use is AEL? Why am I bothering to waste my time with it? It's a
valid question! It deserves some discussion!
First of all I'd like to thank for all the good answers and valid points
people have made to this question.
What you need is something like:
exten = _456.,1,Dial(SIP/[EMAIL PROTECTED],30,tTD(${EXTEN:3}))
regards,
PaulH
AsteriskIT
www.asteriskit.com.au
On Tue, 2006-08-22 at 10:59 +0200, Tomislav Parčina wrote:
Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5 over
Assuming the 456 is the ATA number and the outside number is always 10
digits.
exten = _456.,1,Dial(SIP/${EXTEN:0:3}/${EXTEN:-10},tT)
but then it might as well be
exten = _456.,1,Dial(SIP/456/${EXTEN:-10},tT)
; to dial outside thru GSM gateway
exten =
RR wrote:
I'd read a lot of mails about people having upgraded to 1.2.9.1. only
to realise that they were better off with 1.2.7 or 1.2.6. Has this
been the case with 1.2.10 or is this definately a more stable release
I've had good results with 1.2.10. But, I'm only using queues lightly.
John Marvin wrote:
Today I had a brief power outage which caused the Asterisk server and
DSL modem to reboot. The Asterisk server came up before the internet
connection was working, so it failed when try to look up some of the
hosts for my outbound voip providers in sip.conf.
Asterisk never
There use to be forum on this site http://forum.globalvoicenet.com/ and you
could read mails from this list on Forum-like fashion. Does anybody know what
happened? Have they moved somewhere else?
Thank you for info.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.:
Rich Adamson wrote:
If memory serves correctly, most of the above has been raised as issues
in the past and the suggested work around has been to run a dns caching
server on the asterisk box.
That's exactly what I am doing, unless you mean that the dns caching
server caches results over a
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I did yum update last week and here is my current kernel:
I had no problem at all with zaptel. I am only using TDM400P though, in
case that matters.
Hi Anto!
The thing is that I can't rely on yum update for asterisk installation. I
1 - 100 of 165 matches
Mail list logo
