Hi list,
I have bought some Cisco 7941G-GE IP phones and want to use them with
asterisk. Before bying I tested the whole setup with three different
models of the old 79X0 series (a 7912, 7940 and a 7960). Flashing the
formerly provided SCCP-Image to SIP was no problem, but now it complains
about
Umm... create your dial plan then use a softphone? You'll have to work
out the audio connections in and out of your computer that feeds a audio
channel and outputs the monitor back to the computer but it can be done
pretty easily.
Shane D wrote:
Hello Asterisc-Users List,
I am new to the
Hello Dean,
2008/1/3, Dean Collins [EMAIL PROTECTED]:
Can you provide more details on what you are trying to do. Your
explanation is a bit confusing – sounds interesting but just want to make
sure I have your idea right.
Please, apologize for not being very clear (side effects of new year
2008/1/4, BJ Weschke [EMAIL PROTECTED]:
MatsK wrote:
Olivier wrote:
Hi,
I'm wondering whether or not it is achievable to build a web based
click2dial application that could automatically detect that a user is
connected from office or home.
Another option is to directly ask user
On Fri, Jan 04, 2008 at 01:21:13PM +0530, Rajkumar S wrote:
Hi,
I have callcenter running with v 1.2 with AgentCallbackLogin and now
trying to move to 1.4 using the example doc,
doc/queues-with-callback-members.txt.
From what I understand the basic idea in the example is to
1.
[queue-caller]
exten = s,1,Set(TEST=a)
exten = s,n,Queue(various args)
The error was that I didn't make the variable inheritable, prepending it with
one or two underscores.
Now it works, thanks!
--
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it -
On Friday 04 January 2008 11:58:25 Tomasz Zieleniewski wrote:
Hi,
I have the following problem that when asterisk receives SIP response 302
it cannot forward the call
I get such debug:
[Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel
type registered for 'Local'
[Jan
TODAY, Friday January 4th at 12 Noon EST, 11 AM Central, 9AM Pacific,
Mountain figure it out, 17:00 UTC
Mark joins us to talk about IAX, the appliance, what's new in the
asterisk worldwide communities and answer any questions you may have.
Why not take this opportunity to ask questions or make
I had thought of that. And sorry for the misspelling of asterisk. For
some reason, I thought C.
Anyway, I would prefer not to use a softphone, but I could if need be.
I just want a phone number, located who cares where, that will ring my
asterisk box. My friend has done this successfully for
d tbsky wrote:
hi:
thanks for the information. you are the second one who mentioned
atcom. so i think this phone has basic quality.
i don't have atcom in hand. but i have other china brand(fanvil)
phone which seems the same as atcom: infeneon based, sip, iax, good
sound quality.
but
On Fri, 2008-01-04 at 09:11 +0100, Christophorus Laube wrote:
Hi list,
I have bought some Cisco 7941G-GE IP phones and want to use them with
asterisk. Before bying I tested the whole setup with three different
models of the old 79X0 series (a 7912, 7940 and a 7960). Flashing the
formerly
Very cool. I may have to listen to it recorded, though. I'm not
available at that time.
So you are using talkshoe for the show? What inspired that decision?
On 1/4/08, randulo [EMAIL PROTECTED] wrote:
TODAY, Friday January 4th at 12 Noon EST, 11 AM Central, 9AM Pacific,
Mountain figure it out,
Hi,
I have the following problem that when asterisk receives SIP response 302 it
cannot forward the call
I get such debug:
[Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel
type registered for 'Local'
[Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable
Snapanumber does this but with only certain browsers. (it doesn't work with ie
which is what I use).
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
I've upgraded from SCCP to SIP 8.x.x branch on 7961g and 7911g without
any problems.
As far as the CTLSEP File (Straight from Cisco):
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/i
pp7960/addprot/mgcp/frmwrup.htm#wp1047292
The CTLSEP MAC file is a certificate trust
Lars Bensmann [EMAIL PROTECTED] writes:
Does anybody have an idea where I can start looking to fix this?
Or is this regular behaviour of asterisk that it does not show an
extension as busy when it initiated the call?
It isn't how Asterisk 1.2 behaved. Can you reproduce on a vanilla 1.4?
I
4 jan 2008 kl. 11.50 skrev Benchev:
On Friday 04 January 2008 11:58:25 Tomasz Zieleniewski wrote:
Hi,
I have the following problem that when asterisk receives SIP
response 302
it cannot forward the call
I get such debug:
[Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No
On Fri, Jan 04, 2008 at 10:09:06AM +0100, Olivier wrote:
Hello Dean,
2008/1/3, Dean Collins [EMAIL PROTECTED]:
Can you provide more details on what you are trying to do. Your
explanation is a bit confusing – sounds interesting but just want to make
sure I have your idea right.
Rajkumar S wrote:
Hi,
I have callcenter running with v 1.2 with AgentCallbackLogin and now
trying to move to 1.4 using the example doc,
doc/queues-with-callback-members.txt. From what I understand the basic
idea in the example is to
1. Authenticate a caller with VMAuthenticate
2. Get his
Hey folks,
Is anyone working on Asterisk (or other) presentation proposals for
OSCON 2008 in Portland, OR? Here's the link, in case:
http://en.oreilly.com/oscon2008/public/cfp/13
I'd love to see more Asterisk content there!
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic
Have you looked into
http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html
-E
On Jan 4, 2008 8:43 AM, Remco Barendse [EMAIL PROTECTED] wrote:
You can use the D option with the Dial command.
Something like this should work:
exten =
On Thu, Jan 03, 2008 at 11:17:24PM +0200, Dovid B wrote:
How do you have BLF set up ?
sip.conf:
[general]
context=from-internal
language=de
qualify=yes
subscribecontext=blf
[pio]
type=friend
username=pio
secret=X
host=dynamic
callerid=User1 13
limitonpeers = yes
call-limit=100
vmexten=902
On 12/31/07, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Gregory Malsack wrote:
-- Digium --
That's news to me as well as the rest of the sales team. We were told that
users cannot change the 1gb flash card.
I just spoke with one of the Sales Engineers
Thanks. I will be e-mailing ou off list. I want more instructions on
hos to use the soundcard. I am an asterisk novice.
Shane
On 1/4/08, dave cantera [EMAIL PROTECTED] wrote:
shane, et. al.
shouldn't the console/dsp work? I have a handset, let me get it no
markings on it, has jacks to
Dear Rickygm,
Am Donnerstag, den 03.01.2008, 20:19 -0600 schrieb troxlinux:
hello list, happy new year to all, also to digium for their great work
with asterisk .
I want to make an automatic call marking an extension from my dial
plan , an example that when marking the extension 100, tell
4 jan 2008 kl. 13.06 skrev Johansson Olle E:
4 jan 2008 kl. 11.50 skrev Benchev:
On Friday 04 January 2008 11:58:25 Tomasz Zieleniewski wrote:
Hi,
I have the following problem that when asterisk receives SIP
response 302
it cannot forward the call
I get such debug:
[Jan 4 10:43:27]
Roy,
I just attempted to use a blank CTLSEP(mac).tlv file with my Cisco 7971G-GE
and it would not complete booting. The message log shows that the when the
phone finds the blank CTLSEP(mac).tlv file it says CTL update failed,
retries two more times, then the status line shows Configuring IP then
Hi lan:
thanks for your reply. i already discussed with atcom engineer. they
are sorry that they
can not satisfy any of my request. they will release an advanced model
this year and hope
it can catch up others.
fanvil is really poor. we have dozens of fanvil FV6050 and now we
have to give up
As I see it this matrix is only for the 79X0 generation, right? Every
howto I found said that it would be no problem to have the CTLSEP file
empty. I just tried to build up an empty file on Windows but that did
not help. So my problem is that every howto is proposing that this will
work with an
On Jan 4, 2008 4:21 PM, BJ Weschke [EMAIL PROTECTED] wrote:
AddQueueMember(queuename[|interface[|penalty[|options[|membername):
Thanks BJ Weschke and Alexandre Snarskii. Your mails together gives
complete solution to my problem!
raj
___
to all,
I had a similar thought... what I came up with was, not my idea just saw
it done somewhere else, a small windows binary that was exectued on
login. registered your login name with a server (content filter in that
case)... any browser requests were logged for filtering and tracking
Ok yep now I understand. (in reply to your other email).
As per cookie below - no need.
From your description below your explanation of the application is you
don't have a public facing web page but you are looking for people to
click on but a personalized list of numbers. In order for
shane, et. al.
shouldn't the console/dsp work? I have a handset, let me get it no
markings on it, has jacks to plug into your sound card for audio in and
audio out.. it worked on my laptop with asterisk or maybe a softphone...
if your sound board has an audio in/out that might work
I must say I don't follow,
On Fri, Jan 04, 2008 at 07:57:19AM -0500, dave cantera wrote:
to all,
I had a similar thought... what I came up with was, not my idea just saw
it done somewhere else, a small windows binary that was exectued on
login.
At the client? You assume that this is a
You can use the D option with the Dial command.
Something like this should work:
exten = _06,1,Dial(SIP/gsm_gateway,45,D(${EXTEN})
It worked
Here is how i did it in FreePBX :
1) Setup a SIP extension for the ATA device, in my case i give it
extension number 298. Edit the
In your SEPmac.cnf.xml file look for the setting below and set it to
0:
deviceSecurityMode0/deviceSecurityMode
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Glenn Cobb
Sent: Friday, January 04, 2008 9:37 AM
To: 'Asterisk Users Mailing List -
Christophorus Laube wrote:
Hi list,
I have bought some Cisco 7941G-GE IP phones and want to use them with
asterisk. Before bying I tested the whole setup with three different
models of the old 79X0 series (a 7912, 7940 and a 7960). Flashing the
formerly provided SCCP-Image to SIP was no
I've always found it best to run with no CTLSEPMAC.tlv file in the
tftp server directory; it will ignore that and move on.
With the 7961's you'll be best in the 8-3-3SR2 leading edge, the DND
button is where it should be, and transfers work the way they should
again.
The XML configuration file
On Fri, 4 Jan 2008, EdPimentl wrote:
Have you looked into
http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html
-E
Yes i did, looks like an excellent product with many, many features and
of outstanding quality.
However, given the cost of that unit i would
Thanks for the hint. I just tried that although I only see my worries
coming true: the CTLSEPmac.tlv file is the first one the phone
requests when booting, no possibility to set something different as the
SEPmac.cnf.xml should be loaded after the successful load of the CTL
file. And thus the phone
Hi,
Im getting mad with this error, I have a x100p installed with wcfxo
module loaded perfectly, I can receive incoming calls and detect very
good the hangup for incoming calls. But for outgoing calls its a mess.
When I place a call for outgoing, i heard the ringing, my cell or
phone
Thanks it helped, I had the noload in modules.conf.
But now I have another problem:
When 302 response is received by asterisk it falls in to some context.
according to rfc 3261 uac which receives 302 should retry the request at the
address given by the contact header filed.
I am not able to make
Here is a little more info...
I hooked up the 7971G-GE to my pc and grabbed this with tera-term. Its the
console output during the CTL update process. I am using SIP70.8-3-3.
NOT 09:28:45.969295 DHCP: Restart - delay = 1
NOT 09:28:45.981198 DHCP: Sending Release...
NOT 09:28:49.000449 DHCP:
You're right. That was my mistake.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Novack
Sent: Friday, January 04, 2008 11:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
What is the output of:
pri show spans
PRI span 1/0: Provisioned, Down, Active
PRI span 2/0: Provisioned, Down, Active
Do incoming calls work?
Negative, and nothing shows up on the CLI. And that's after creating
separate contexts
I have a SIP trunk to Broadvoice. My Asterisk box (1.4.13) is on public
addresses behind a firewall.
Originally it was behind a Linksys WRT54G running sveasoft. Sveasoft
really can't NOT do NAT even when you turn it off. My Asterisk box is
defined as the DMZ box to Sveasoft and it seemed
This should result in the same problem. The CTLSEPmac file is the
first that is requested on the TFTP server. But I am going to try that.
Regards and thanks,
Christophorus
Try naming the empty file:
SEP0019E7D16CD6.tlv
Not
CTLSEP0019E7D16CD6.tlv
-Original Message-
From: [EMAIL
Try naming the empty file:
SEP0019E7D16CD6.tlv
Not
CTLSEP0019E7D16CD6.tlv
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Christophorus Laube
Sent: Friday, January 04, 2008 10:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
That is not the name the phone requests
When uping my 7960, the empty file did the trick
I so far am unable to go beyond 7.1 however, as Asterisk rejects
anything I dial with 7.3
Anyone have SIMPLE sample config files?
John Novack
Anciso, Roy wrote:
Try naming the empty file:
The official URI for this is
tel:
see http://www.ietf.org/rfc/rfc3966.txt
It isn't implemented everywhere, but most cellphone browsers seem to.
Tim
On 4 Jan 2008, at 12:48, Dean Collins wrote:
Snapanumber does this but with only certain browsers. (it doesn't
work with ie which is what I
On Fri, Jan 04, 2008 at 04:00:30PM +0100, Miguel A Felipe Rodríguez wrote:
Hi,
Im getting mad with this error, I have a x100p installed with wcfxo
module loaded perfectly, I can receive incoming calls and detect very
good the hangup for incoming calls. But for outgoing calls its a
On Fri, Jan 04, 2008 at 04:29:19PM +0100, Jaap Winius wrote:
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
What is the output of:
pri show spans
PRI span 1/0: Provisioned, Down, Active
PRI span 2/0: Provisioned, Down, Active
Do incoming calls work?
Negative, and
Here is the info on the Cisco console cable I use. There are a couple ways
to make the connection depending on what you have available. I used a piece
of 6 wire flat satin (like the 4 wire telephone station cord only with 6
wires) and put an RJ-12 on one end then put an RJ-24 on the other with
Your nomenclature on the modular plugs are incorrect.
RJxx references specific JACK wiring schemes originally defined in FCC
part 68, but removed in year 2000
There are:
4position 4 contact plugs - used for handsets and Sangoma A200
6 position plugs, with either 2,4 or 6 contacts - commonly and
On Fri, Jan 04, 2008 at 03:08:59PM +0100, Benny Amorsen wrote:
Lars Bensmann [EMAIL PROTECTED] writes:
Does anybody have an idea where I can start looking to fix this?
Or is this regular behaviour of asterisk that it does not show an
extension as busy when it initiated the call?
It
I was thinking something same, but the problem this in the agi, where
I get an example, the idea is that when programming the call, she
calls to the specified extension and reproduce the voice message
greetingss
rickygm
2008/1/4, Anselm Martin Hoffmeister [EMAIL PROTECTED]:
Am Donnerstag, den
I am trying to configure a Polycom IP4000 for use with Asterisk 1.4.5 on
a flat local network.
I followed the provisioning guides that I found on the Web, and I have
the phone downloading bootrom.ld, sip.ld, and a bunch of configuration
files. This all works properly.
However, I receive the
On Thu, 2008-01-03 at 18:25 +, Russell Brown wrote:
Why-o-why setting DYNAMIC_FEATURES causes the PPP hookup from my old PBX
to fail I really can't imagine. Any developers care to comment? (I'm
happy to insert debug and send info)... or should I file a bug report?
I can't imagine why
You are correct, I made some corrections.
Here is the info on the Cisco console cable I use. There are a couple ways
to make the connection depending on what you have available. I used a piece
of 6 wire flat satin (like the 4 wire telephone station cord only with 6
wires) and put a 6 pin modular
On Fri, Jan 04, 2008 at 03:08:59PM +0100, Benny Amorsen wrote:
It isn't how Asterisk 1.2 behaved. Can you reproduce on a vanilla 1.4?
This is also not on a vanilla 1.4. It's the BRIstuffed version from the
xorcom archive.
I can see if I can install a vanilla 1.4 off-hours and just test the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jaap Winius wrote:
switchtype = euroisdn
signalling = bri_cpe_ptmp
I don't know about NL but in the UK, multiple ISDN2e lines have to be
configured as bri_cpe_ptp not bri_cpe_ptmp. Have you tried this mode?
HTH
- --
Ron Wellsted
[EMAIL
A more recent one is,
http://www.iana.org/assignments/uri-schemes.html
http://www.ietf.org/rfc/rfc4395.txt
But I have seen more web pages with a callto: tag than with the tel:
tag. But from what I have seen have the callto: tag been disregarded
since there is a tel: tag.
/Mats
Tim Panton
Does anyone know of a good VOIP dialtone provider in the northern Chicago area.
My client has tried Broadvoice and Mix and is having problems with latency in
the middle of the traceroute between him and the provider.
Thanks,
Greg
No virus found in this outgoing message.
Checked by AVG
Shane D wrote:
I had thought of that. And sorry for the misspelling of asterisk. For
some reason, I thought C.
Anyway, I would prefer not to use a softphone, but I could if need be.
I just want a phone number, located who cares where, that will ring my
asterisk box. My friend has done
Hi Senad,
Did you happen to find out if it was indeed anywhere in the US48?
Thanks!
- sf
On 1/2/08, Senad Jordanovic [EMAIL PROTECTED] wrote:
Dovid B wrote:
Senad,
You can get unlimited as in FREE to any where in US48 or just local ?
As far I know it is anywhere to US48. I will find
hi guys.
i'm writing some simple applications for the cisco 7970
services button. i read the asterisk wiki and it mention
there's a CMXML_App_Guide.pdf file but there's nowhere
can i find a link for it. does anybody know where can
i find it?
regards.
--
Edwin Lam [EMAIL PROTECTED]
Systems
I discovered the following in this reguard:
http://freeworlddialup.com
Sign up for that, and then you will use:
http://ipcall.com (US Numbers)
http://uknumber.co.uk (UK Numbers)
They both provide free phone numbers for a box.
Shane
On 1/4/08, Troy Ayers [EMAIL PROTECTED] wrote:
Shane D wrote:
Steve Finkelstein wrote:
Hi Senad,
Did you happen to find out if it was indeed anywhere in the US48?
Thanks!
- sf
Hi Steve,
Yes it is. Contact me of the list if you need to.
Regards,
Senad
On 1/2/08, *Senad Jordanovic* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
I have changed the signalling of the x100p to a fxsls, now i can make
outgoing calls, but now I have another problem, cant detect hangup. I
post the zapara.conf and the zaptel.conf so if any has idea of waht to
change y have tested changing busydetect, busypattern, callprogress,
etc.. but no
Hi everybody
We have a strange problem with several asterisk servers (Version
1.4.11) using PRI cards (tied to telco here in Belgium).
Indeed we noticed that whenever a local user places an outgoing call
through the PRI (and telco) to another IPBX (tied to telco using BRI
or PRI), if the
Gregory Malsack wrote:
In terms of ease, what is actually stored on the card, is it possible to
simply place the card in a cf reader connected to a usb port on a linux box,
tar up the contents and untar the contents on a larger cf card? Or is this
something that would require a dd?
Yes,
On Wed, 2007-12-26 at 17:47 +0200, Tzafrir Cohen wrote:
On Wed, Dec 26, 2007 at 08:38:31AM -0600, Bob Smither wrote:
On Wed, 2007-12-26 at 07:27 +0200, Tzafrir Cohen wrote:
Just to stress the point: you mention that you don't see the card on
lspci. If this is so: the problem is not
On Fri, Jan 04, 2008 at 04:39:44PM -0600, Bob Smither wrote:
One last question if someone has an answer - Using Kwelstart signaling *
can detect incoming calls and can detect hang ups, but doesn't seem to
be able to detect when an outgoing (over a Zap channel) call is
answered. Is this
On 16:10, Fri 04 Jan 08, Remco Barendse wrote:
On Fri, 4 Jan 2008, EdPimentl wrote:
Have you looked into
http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html
-E
Yes i did, looks like an excellent product with many, many features and
of outstanding
Is there a way to disable the b2bua feature in asterisk.
I would like asterisk to work as a sip server and not be involved in the RTP
path between phones.___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing
On 20:33, Fri 04 Jan 08, Ron Wellsted wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jaap Winius wrote:
switchtype = euroisdn
signalling = bri_cpe_ptmp
I don't know about NL but in the UK, multiple ISDN2e lines have to be
configured as bri_cpe_ptp not bri_cpe_ptmp.
I am trying to setup asterisk as a registrar and sip server only.
Currently When I make calls all my rtp traffic is going through the asterisk
server as a B2BUA.
Is it possible to turn off this feature and have all my calls RTP traffic going
directly to the SIP
Hi All;
Is there a command that can let me execute the
Dial(.) if {CALLERIDNUM}= ..? Without using
GotoIf?
Any help?
Regards
Bilal
Be a better friend, newshound, and
know-it-all with Yahoo!
Hi Shane,
On Jan 4, 2008 12:18 PM, Shane D [EMAIL PROTECTED] wrote:
Very cool. I may have to listen to it recorded, though. I'm not
available at that time.
Too bad, but you can listen to the roecording. Allison makes a guest
appearance, I'll upload that tomorrow. Just getting back from the
Hi List;
Is there any possibility to let asterisk support G723?
Regards
Bilal
Looking for last minute shopping deals?
Find them fast with Yahoo! Search.
Martin Smith wrote:
Hey folks,
Is anyone working on Asterisk (or other) presentation proposals for
OSCON 2008 in Portland, OR? Here's the link, in case:
http://en.oreilly.com/oscon2008/public/cfp/13
I'm working on a proposal to do a tutorial there on Asterisk. I have
done such the past
Hi All;
Is there a method to use OR and AND operator with
GotoIf, so I can make better logical expression?
Regards
Bilal
Looking for last minute shopping deals?
Find them fast with Yahoo! Search.
Quoting Michiel van Baak [EMAIL PROTECTED]:
I don't know about NL but in the UK, multiple ISDN2e lines have to be
configured as bri_cpe_ptp not bri_cpe_ptmp. Have you tried this mode?
It's the same here in .nl
Interesting, but I would think this to be unnecessary in my case,
since I have
On Sat, Jan 05, 2008 at 01:18:18AM +0100, Jaap Winius wrote:
Quoting Michiel van Baak [EMAIL PROTECTED]:
I don't know about NL but in the UK, multiple ISDN2e lines have to be
configured as bri_cpe_ptp not bri_cpe_ptmp. Have you tried this mode?
It's the same here in .nl
Interesting,
Thank you very much.
On 1/4/08, randulo [EMAIL PROTECTED] wrote:
Hi Shane,
On Jan 4, 2008 12:18 PM, Shane D [EMAIL PROTECTED] wrote:
Very cool. I may have to listen to it recorded, though. I'm not
available at that time.
Too bad, but you can listen to the roecording. Allison makes a
ah. I was afraid of that. None of the phones I have seem to support it
either. Supposedly Grandstream does (from what I've seen randomly online),
but you can't set it to only use that through the web config.
Thanks anyway guys. I'll go bug someone else, but you may see me around
occasionally
Gregory Malsack wrote on 1/4/08 4:48 PM:
Does anyone know of a good VOIP dialtone provider in the northern
Chicago area. My client has tried Broadvoice and Mix and is having
problems with latency in the middle of the traceroute between him and
the provider.
I use Broadvoice and haven't had
brian,
cool, I attended one of you tutorials in baltimore... it was Great!
would go again because I know I would learn even more this time after
being exposed to it in greater detail... I could absorb more this time...
daveC
Brian Capouch wrote:
Martin Smith wrote:
Hey folks,
Is
On Friday 04 January 2008 16:45:00 ameel wrote:
Is there a way to disable the b2bua feature in asterisk.
I would like asterisk to work as a sip server and not be involved in the
RTP path between phones.
No. And by the way, b2bua is not a feature. It's is literally what Asterisk
is.
--
On Friday 04 January 2008 17:52:52 bilal ghayyad wrote:
Is there a command that can let me execute the
Dial(.) if {CALLERIDNUM}= ..? Without using
GotoIf?
Exec(${IF($[${CALLERID(num)} = 12345]?Dial(foo):NoOp)}
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Tilghman
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On Friday 04 January 2008 18:03:30 bilal ghayyad wrote:
Is there a method to use OR and AND operator with
GotoIf, so I can make better logical expression?
$[(${a} = 2) | (${b} = 3)]
$[(${a} = 3) (${b} = 4)]
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Tilghman
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On Friday 04 January 2008 18:06:08 bilal ghayyad wrote:
Is there any possibility to let asterisk support G723?
Buy the TC400P transcoder card from Digium.
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Tilghman
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Hi
I am setting up a Calling card Plat form
I have incoming toll number, the provider charges incoming calls
I see some spammers( competetors) keep calling my toll. so iam getting huge
invoices
how can i identify those kind of spammers and block the callerID for some
time
any suggestions or
ram wrote:
Hi
I am setting up a Calling card Plat form
I have incoming toll number, the provider charges incoming calls
I see some spammers( competetors) keep calling my toll. so iam getting
huge invoices
how can i identify those kind of spammers and block the callerID for
On Saturday 05 January 2008 00:45:00 ameel wrote:
Is there a way to disable the b2bua feature in asterisk.
I would like asterisk to work as a sip server and not be involved in the
RTP path between phones.
You can do it by setting BOTH peers
canreinvite=yes
On Jan 5, 2008 4:40 AM, ameel [EMAIL PROTECTED] wrote:
I am trying to setup asterisk as a registrar and sip server only.
Currently When I make calls all my rtp traffic is going through the
asterisk server as a B2BUA.
Is it possible to turn off this feature and have all my calls RTP traffic
On Friday 04 January 2008 16:45:00 ameel wrote:
Is there a way to disable the b2bua feature in asterisk.
I would like asterisk to work as a sip server and not be involved in the
RTP path between phones.
No. And by the way, b2bua is not a feature. It's is literally what Asterisk
is.
If I
On Saturday 05 January 2008 00:06:00 Yehavi Bourvine +972-8-9489444 wrote:
On Friday 04 January 2008 16:45:00 ameel wrote:
Is there a way to disable the b2bua feature in asterisk.
I would like asterisk to work as a sip server and not be involved in the
RTP path between phones.
No. And
On Jan 5, 2008 11:26 AM, Trevor Peirce [EMAIL PROTECTED] wrote:
ram wrote:
Hi
I am setting up a Calling card Plat form
I have incoming toll number, the provider charges incoming calls
I see some spammers( competetors) keep calling my toll. so iam getting
huge invoices
how
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Dovid B wrote:
Do you mind sharing that script with us ?
Thomas Balsfulland wrote:
this is not the right way, because it takes 30 sec before asterisk try
dialout over peer2. the dial-timer ([EMAIL PROTECTED],30) is normaly set
for
the time to
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