Thanks for all the replies and help - For future reference I eventually
decided to go for a func_odbc solution and use a database sequence that
*is* atomic and gives me what I needed.
Julian.
Julian Lyndon-Smith wrote:
Sorry if this appears twice - I originally sent it nearly 18 hours ago
On Mon, Jul 30, 2007 at 05:31:51PM +0300, Tzafrir Cohen wrote:
On Mon, Jul 30, 2007 at 09:45:10PM +0800, GNUbie wrote:
Hello all,
Where can I find a list of description for each sound files provided by the
asterisk-sounds-main Debian package? You can find the contents of my
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ahoy
Jared Smith wrote:
On Mon, 2007-07-30 at 16:46 +0200, Florian Arthofer wrote:
Shouldn't i see _something_ on the console, even if
the DID which is dialed isn't configured yet?
Unfortunately, I don't think so. You might want to add a
Alex Balashov schrieb:
On Mon, 30 Jul 2007, Knud Müller wrote:
what does your modules directory contain? Can you find a file
/usr/lib/asterisk/modules/app_meetme.so after make install?
No. I know it needs to be compiled, but it is not being compiled no
matter what I seem to do in the
I found the solution by myself
http://www.voip-info.org/wiki/index.php?page=Asterisk+IAX+channels
you have to surround the key filename with square brackets
Tzafrir Cohen schrieb:
On Mon, Jul 30, 2007 at 02:01:49PM +0200, Jack wrote:
Hi all,
I have a Wildcard TE110P connected to a E1 line an I want to reserve
channels in the following way:
channels 1-15 and 17-21 for incoming calls
channels 22-28 for outgoing calls
channels 29-31 for
Eric ManxPower Wieling schrieb:
Jack wrote:
Hi all,
I have a Wildcard TE110P connected to a E1 line an I want to reserve
channels in the following way:
channels 1-15 and 17-21 for incoming calls
channels 22-28 for outgoing calls
channels 29-31 for emergency calls
My zaptel.conf
C F schrieb:
Why would you want to do that? let Asterisk (using zap/g in app_dial)
take care of which channel are used for outbound but assign all the
channels to that g, reject any incoming calls if there are already 7
incoming active calls with a congestion PRI_CAUSE. Do the same for 20
On Tue, Jul 31, 2007 at 10:06:30AM +0200, Jack wrote:
Tzafrir Cohen schrieb:
On Mon, Jul 30, 2007 at 02:01:49PM +0200, Jack wrote:
signalling=pri_cpe
channel = 29-31
and then in extensions.conf:
[hangup-calls]
; not sure that this is precisly the right thing to do:
exten
Hello ppl,
Searched all over, but couldn't find anything conclusive.
Does an off-the-shelf version of Asterisk run without any issues on a
64-bit machine?
Does anyone have any 'conclusive' figures?
Apologies if this is a repeat question. Would appreciate if I could be
redirected to the
Your Grandstream device is using a UDP message of 8 x 0x00 bytes as it's
RTP keepalive message.
When Asterisk tries to read this it logs a warning because it is
expecting a valid RTP packet.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
On 7/31/07, Benjamin Jacob [EMAIL PROTECTED] wrote:
Searched all over, but couldn't find anything conclusive.
Does an off-the-shelf version of Asterisk run without any issues on a
64-bit machine?
Does anyone have any 'conclusive' figures?
I've run 1.2.14 - 1.2.18 and 1.4.4 - 1.4.9 on CentOS
Am Montag, den 30.07.2007, 14:29 -0700 schrieb Lee Howard:
http://www.asterisk.org/node/48327
I mean, really... you're kidding me, right?
It is not at all April 1st... however, I see the point in having a
simple demo app. Wether you call it helloworld or hellomarc, the
difference is not too
Tzafrir Cohen schrieb:
On Tue, Jul 31, 2007 at 10:06:30AM +0200, Jack wrote:
Tzafrir Cohen schrieb:
On Mon, Jul 30, 2007 at 02:01:49PM +0200, Jack wrote:
signalling=pri_cpe
channel = 29-31
and then in extensions.conf:
[hangup-calls]
; not sure that
Benjamin Jacob ha scritto:
Hello ppl,
Searched all over, but couldn't find anything conclusive.
Does an off-the-shelf version of Asterisk run without any issues on a
64-bit machine?
Does anyone have any 'conclusive' figures?
Apologies if this is a repeat question. Would appreciate if I
On Tue, 2007-07-31 at 09:00 +0200, Florian Arthofer wrote:
So, if my ISDN-number is for example 1234567, then i should, if i dial
1234567-123, see _something_ on the console and at least i should hear
it ringing on the phone i place the call with. Am i right?
OK, ordinarily this would be true,
Nhadie Ramos wrote:
Hi john,
Thank you for your reply, i finally stumbled on google what the problem is.
The driver does not compile on kernel newer than 2.6.19.
you can try beta wanpipe 3.xx which compiles on 2.6.20. If you want to
try 2.6.22 and newer, there is no official patches. I've
On Mon, 2007-07-30 at 09:31 -0400, Jared Smith wrote:
The second major difference between the cards is echo cancellation. The
TE212P comes with an echo cancellation module installed, while the TE220
card comes without one. (You can always add a VPMOCT064 module to it,
but it doesn't come
On Tue, 2007-07-31 at 06:36 +0100, Deepak Naidu wrote:
I think we need to pay for the later, but I am not sure if we need to
pay for the inbuilt asterisk(freepbx) on hold music.
I'm no lawyer, but here's what I understand. (Please consult with an
attorney in your area, and don't consider this
Ciao Florian,
I'm trying to setup Asterisk on debian etch (with the debian packages)
with a Fritz!Card PCI ISDN card and chan_capi.
Why don't you use mISDN drivers?
Bye,
--
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
Hi,
I have asterisk 1.2.18.
I am trying to get asterisk to react to an (out of dialog) REFER ...
see below. I get a 603 (no dialog) ... and in the code (sip.conf:3277)
a comment being able but not supporting it??
Any pointers would be great ... is it a configuration option?
REFER sip:[EMAIL
1and1 dedicated server's service has been down for a few hours , unable
to reach them by phone or email. do anyone know what is going on there ?
Mario
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing
[Resent due to non-descriptive subject line.]
Hi folks
When connecting two SIP users, is there any way to stop Asterisk from
sending SIP 183 Session Progress messages, either globally or
per-peer?
Scenario as follows:
Call from UA1 to Asterisk (UA2) to UA3.
UA3 sends RTP before SIP OK to
Hi folks
When connecting two SIP users, is there any way to stop Asterisk from
sending SIP 183 Session Progress messages, either globally or
per-peer?
Call from UA1 to Asterisk (UA2) to UA3
UA3 sends RTP before SIP OK to Asterisk (UA2)
Asterisk (UA2) detects early audio from UA3 and sends 183
Jack wrote:
Actually I preferred to reserve the channels in asterisk, but this seems
to be the easiest way.
Does anybody know if the mapping from telco channels to zap channels is
fixed? Is the first telco channel always mapped to the first zap channel
or is this mapping dynamic?
You
Hello,
After upgrading to 1.4.9 the above function does not work anymore; it claims
that child went away while the child is probably not born at all...
Before I open a bug on it, anyone has a clue?
Thanks! __Yehavi:
Please start new threads for new messages (don't reply and just wipe out
the body). The headers still exist so you wind up with screwy threading in
the list archives (ditto for those of us who have e-mail software that
supports threading).
AR
On 7/31/07, Richard Brady [EMAIL PROTECTED] wrote:
Thanks Steve,
am I correct in assuming that Asterisk will ignore this packet? It will
not have any harmful effect if the caller is on hold for an hour (120
packets)?
In order to feed back to Grandstream, is there a correct format for a
keepalive message or is this one adequate?
regards,
On Tue, Jul 31, 2007 at 07:25:00AM -0400, Jared Smith wrote:
On Tue, 2007-07-31 at 06:36 +0100, Deepak Naidu wrote:
I think we need to pay for the later, but I am not sure if we need to
pay for the inbuilt asterisk(freepbx) on hold music.
I'm no lawyer, but here's what I understand.
Hello All,
Can anyone help me with this... This is what my program does: -
1) At certain time the system generates a .call and make a call to User A.
2) When User A picks up the phone call, system will play a menu select
option.
a) Press 1 to call your supervisor.
b) Press 2 to
Does anyone know the algorithm that Asterisk uses to figure out when
you'll be speaking with an agent?I've heard it say such bizarre
things as '2 minutes' when there area 2 calls waiting ahead of a
person and 1 agent logged in.
___
--Bandwidth and
Sorry Drew - I don't know enough about Asterisk to comment on the
effects of this on the RTP stack.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Drew
Gibson
Sent: 31 July 2007 15:02
To: Asterisk Users Mailing List -
Hi Alex
Apologies for that, I noticed this immediately after I sent the email
and resent it with fresh headers and a descriptive subject line. I
will be more careful in future.
Regards,
Richard
--
Richard Brady
T: +44 (0)7771 623 348
E: [EMAIL PROTECTED]
I meant to send that just to you, not the list - My apologies. I wasn't
trying to be the public list cop.
AR
On 7/31/07, Richard Brady [EMAIL PROTECTED] wrote:
Hi Alex
Apologies for that, I noticed this immediately after I sent the email
and resent it with fresh headers and a descriptive
On Tue, 2007-07-31 at 10:16 -0400, Matt wrote:
Does anyone know the algorithm that Asterisk uses to figure out when
you'll be speaking with an agent?I've heard it say such bizarre
things as '2 minutes' when there area 2 calls waiting ahead of a
person and 1 agent logged in.
I don't know
Hi
I am trying to make this setup work
phone1---g729---asterisk1---sip---asterisk2---g729---phone2
I have tried several configurations but none worked
I keep getting transcoding errors
I have installed one g729 licence on each asterisk, but I can't verifiy
because the show g729 command is
Hi all,
When I try to update zaptel to 1.4.3 or 1.4.4 it seems to hang the
server at loading zap hardware modules. I can install 1.4.2.1 and it
will load OK at startup although I have to run
..zaptel-source/xpp/utils/genzaptelconf -u before rebooting the server
or I run into trouble on
Can you verify the correctness of that URL? 404 from here.
Thanks!
Jared Smith wrote:
On Tue, 2007-07-31 at 10:16 -0400, Matt wrote:
Does anyone know the algorithm that Asterisk uses to figure out when
you'll be speaking with an agent?I've heard it say such bizarre
things as '2 minutes'
Hi all,
There is a patch in the Mantis that provides a function to reboot GS phones.
In order to use it, I need to know how the digest for SIP Notify messages is
calculated.
If anyone knows that, please let me now
Regards,
Yann
___
--Bandwidth
Hi Nitesh,
you are missing Extension
try with
$call = $asm-send_request('Originate',
array('Channel'=SIP/xo-out/$supervisor_num,
'Context'='default',
'Exten'= your_extensions_here,
'Priority'=1,
'Callerid'=$cid));
or you
On Tue, 31 Jul 2007, Yann JOUANIN wrote:
Hi all,
There is a patch in the Mantis that provides a function to reboot GS phones.
In order to use it, I need to know how the digest for SIP Notify messages is
calculated.
If anyone knows that, please let me now
I don't know about the SIP
Thanks Nasir,
By putting 'Exten'= your_extensions_here it will create a new channel
to that extension, correct?
What I want to do is to join two channels... Join the User A channel
which is active with supervisor.
Cheers,
Nitesh
Nasir Iqbal wrote:
Hi Nitesh,
you are missing Extension
On Tue, Jul 31, 2007 at 10:42:15AM -0500, Nick Whitaker wrote:
Hi all,
When I try to update zaptel to 1.4.3 or 1.4.4 it seems to hang the
server at loading zap hardware modules. I can install 1.4.2.1 and it
will load OK at startup although I have to run
Is this a web hosting forum or mailing list ?
On 31/07/07, Asterisk guy [EMAIL PROTECTED] wrote:
1and1 dedicated server's service has been down for a few hours , unable
to reach them by phone or email. do anyone know what is going on there ?
Mario
Hi guys,
Any one know about any php or other GUI to manage Polycom Directory XML
contacts ?
I would like to setup a web page were I manage Directory contacts to provide
XML file to feed my polycom phones on Boot with all the contacts.
Does any one has already done this? I notice a bash script
Do you have proper version of zaptel installed corresponding to your
asterisk version ?
On 31/07/07, Knud Müller [EMAIL PROTECTED] wrote:
Alex Balashov schrieb:
On Mon, 30 Jul 2007, Knud Müller wrote:
what does your modules directory contain? Can you find a file
On Tue, 31 Jul 2007, Jaswinder Singh wrote:
Do you have proper version of zaptel installed corresponding to your
asterisk version ?
At the time that I downloaded the source I got the latest version of
zaptel as well.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Matt wrote:
I'll take either
Actually now that I have had a chance to think about what I did (sorry
bad week here). Yes, I will admit I did patrionize the users list...
sorry if I offended anyone. I just figured I'd try to help any
SunRocket users out that may not be on the biz list.
Thanks Jared, Yes I am using with Asterisk only. So I am using the inbuilt
music from Asterisk for onhold.
--
Deepak
Jared Smith [EMAIL PROTECTED] wrote: On Tue, 2007-07-31 at 06:36 +0100,
Deepak Naidu wrote:
I think we need to pay for the later, but I am not sure if we need to
pay for the
Patrick,
Make sure you have install the g729 modules correctly as per the
instructions and restarted Asterisk.
Other method is you can configure your Asterisk to do Pass-thru g729
which you don't require to install any g729 license on Asterisk. As far
as both of your phones has g729 installed
We're using a TE212P card with one PRI connection.
Thanks,
Nick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Tuesday, July 31, 2007 11:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Zaptel update trouble
On Tue, 2007-07-31 at 07:51 -0800, Mojo with Horan Company, LLC wrote:
Can you verify the correctness of that URL? 404 from here.
Thanks!
Strange... it works for me here. Maybe one of the web mirrors is out of
sync? Anyway, if you're got the source code to Asterisk 1.4 handy, it
should be
Hello.
We are running Asterisk 1.2.23 iaxmodem-0.2.1 and hylafax-4.3.3
When we send faxes the people who receive the faxes complain that they
look wrinkled or smashed up. Sometimes they are missing random lines.
Has anyone seen this happen, or know how to fix it?
We're getting close to wrapping up the final list of speakers for
AstriCon, but wanted to give the Asterisk community one more chance to
speak up and be heard. If you're interested in presenting at AstriCon,
please go to http://www.astricon.net/ and click on the Speak at
AstriCon link on the
Hi All,
I have a telephony project for which I need
to build a prototype to demo for management.
The prototype must work on a GSM phone network.
In the demo system, a call from GSM phone comes
into the demo box. The demo box runs CallWeaver.
Callweaver picks up the GSM call, answers it and
Hello all
Sorry for the repost, but I believe this is essential for Aastra phones
to use directed pickup with BLF in 1.2 (and it's still experimental in
1.4 I think).
pickup-mgernoth-2006-07-28.patch.txt
If anyone knows otherwise than that this is still required in latest 1.2
or 1.4 please
Guys,
I've downloaded AsteriskNOW few days ago so I'm new to this product.
The first issue is on service provider area.
I've already used a VoIP account already configured with my ISP, it
works fine!
This configuration has been used until now with the client SJphone,
Now I would use this
So is there a simple way to license decent, up to date music? Can I
just go to a website, click a buy button, pay my money and download
the song?
It seems idiotic that you need 15 lawyers and a million bucks use
decent on hold music.
Maybe I just don't know the procedure.
I am all for paying
Victor Toofic wrote:
El Sun, Jul 29 de 2007 a las 20:04 +0800, Steve Underwood comentaba:
What versions of software did you use to get a screwed up result like
that? The message Don't know how to handle signalling event Accepted
is printed at the end of a case statement which does handle
Michael Rice wrote:
Hello.
We are running Asterisk 1.2.23 iaxmodem-0.2.1 and hylafax-4.3.3
When we send faxes the people who receive the faxes complain that they
look wrinkled or smashed up. Sometimes they are missing random lines.
Has anyone seen this happen, or know how to fix it?
Well,
Oh,
you need Dial application instead of origination.
so no need to AGI Script simply add
the dialplan called for .call should look like this
exten = yourexten,1,BackGround(your_menu_ivr)
exten = yourexten,n,WaitExten()
exten = 1,1,Dial(SIP/xo-out/$supervisor_num) ;for Supervisor
exten =
Just Google for: royalty free music, and will find plenty of sites that will
serve your needs.
John Beaman
Telecom Specialist
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331
[EMAIL PROTECTED] 7/31/2007 12:49:45 PM
So is there a simple way to license
Anyone know what they are going to be announcing?
Any relevance to Asterisk? When are we going to see a decent video
conferencing implementation into Asterisk?
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+61-2-9016-5642
On Tue, Jul 31, 2007 at 12:49:45PM -0500, voiplist wrote:
So is there a simple way to license decent, up to date music? Can I
just go to a website, click a buy button, pay my money and download
the song?
In some cases, yes.
It seems idiotic that you need 15 lawyers and a million bucks use
I have done this in the past and I don't recall ever finding any
popular music by popular artist.
For example, if I wanted to play oh I don't know an original song
performed by the original artist such as Nora Jones or The Beatles
will I find this sort of thing at a Royalty Free Site?
On
We have a PRI and use a sangoma a101d to a PRI. The Asterisk and
IAXModem are on the same box.
Here is a link to the output from cat /proc/interrupts
http://fluxbox.pastebin.ca/640841
I put it here since you recommended putting this question on the
IAXModem list.
Thanks for any help
Lee
No, you will not. According to the music industry those artists are all are
entitled to compensation for every time their song is broadcast, which includes
MoH. AFAIK, there are no popular songs by popular artists that are
royalty-free.
John Beaman
Telecom Specialist
Voice
On Tue, Jul 31, 2007 at 01:37:00PM -0500, voiplist wrote:
I have done this in the past and I don't recall ever finding any
popular music by popular artist.
For example, if I wanted to play oh I don't know an original song
performed by the original artist such as Nora Jones or The Beatles
There's no royalty free popular songs by popular artists. Not only
would you have to pay royalties, you would have to secure the rights
from the artists' representatives just to get the permission to play
the songs in the first place.
Darrell S. Long
BestWeb Corporation
john beaman wrote:
Ok,
This tools directly log on the web interface, so if anyone knows about the
SIP NOTIFY message ...
Thanks
yann
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Gordon
Henderson
Envoyé : mardi 31 juillet 2007 18:17
À : Asterisk Users Mailing List -
Thanks Nasir,
That helped alot...
Cheers,
Nitesh
Nasir Iqbal wrote:
Oh,
you need Dial application instead of origination.
so no need to AGI Script simply add
the dialplan called for .call should look like this
exten = yourexten,1,BackGround(your_menu_ivr)
exten =
Tzafrir Cohen wrote:
It seems that a bunch of 1.2 users answered this guy and wondered why
this text file is not there. It's there in 1.2 and not there in 1.4, and
hence also removed from the Debian package.
I didn't see any open bug about this but I vaguely recall some folks
asking here
Tzafrir Cohen wrote:
OTOH: /usr/sbin/asterisk depends on libopenh323 . Now this *is* bloat.
Did you bother to check into why this is the case before spreading this
unfounded information on the mailing list?
First of all, this is *NOT* true unless the chan_h323 module was built,
which the vast
On Tue, Jul 31, 2007 at 03:05:32PM -0400, Jay R. Ashworth wrote:
On Tue, Jul 31, 2007 at 01:37:00PM -0500, voiplist wrote:
I have done this in the past and I don't recall ever finding any
popular music by popular artist.
For example, if I wanted to play oh I don't know an original song
Hello All,
Something strange I found that my .call file is running twice...
Just after 60 sec it will run again, without any application invoking it.
This is my .call file: -
=
Channel: SIP/xo-out/19097773456
Callerid: 9097773456
MaxRetries: 3
RetryTime: 30
WaitTime: 15
Context:
Another day, another apparant unexplained hardware incompatibility.
I have a TE412P and a TDM400B living quite happily in a whitebox using an
Intel motherboard:
http://www.intel.com/design/servers/boards/se7230nh1-e/index.htm
I tried to move to an IBM x3650 system. It uses a slightly newer
Is your .call file writable by asterisk?
$ chmod 777 sample.call
On 7/31/07, Nitesh Divecha [EMAIL PROTECTED] wrote:
Hello All,
Something strange I found that my .call file is running twice...
Just after 60 sec it will run again, without any application invoking it.
This is my .call file:
Thanks Steve,
I'll keep digging,
regards,
Drew
Steve Langstaff wrote:
Sorry Drew - I don't know enough about Asterisk to comment on the
effects of this on the RTP stack.
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of
*Drew Gibson
*Sent:* 31 July 2007
On Tue, Jul 31, 2007 at 02:46:15PM -0500, Kevin P. Fleming wrote:
Tzafrir Cohen wrote:
It seems that a bunch of 1.2 users answered this guy and wondered why
this text file is not there. It's there in 1.2 and not there in 1.4, and
hence also removed from the Debian package.
I didn't
Thanks Atis,
Yes and the .call executes fine... but after 60 seconds it executes
again automatically without any application executing it.
Cheers,
Nitesh
Atis wrote:
Is your .call file writable by asterisk?
$ chmod 777 sample.call
On 7/31/07, Nitesh Divecha [EMAIL PROTECTED] wrote:
Make sure you have a blank line at the end of your .call file.
Nitesh Divecha wrote:
Hello All,
Something strange I found that my .call file is running twice...
Just after 60 sec it will run again, without any application invoking it.
This is my .call file: -
=
Channel:
. . .
Even if you can find non-original-artist recordings of such music, the
*compositions* are registered with BMI and ASCAP, and you'll need
blanket licenses to play them. (Well, if you only wanted one or two
tracks, you might negotiate specific licenses, but I'm not sure it
would be
my shared webhosting is going strong...
daveC
Asterisk guy wrote:
1and1 dedicated server's service has been down for a few hours ,
unable to reach them by phone or email. do anyone know what is going
on there ?
Mario
On Tuesday July 31 2007 4:44 pm, Joe acquisto wrote:
. . .
Even if you can find non-original-artist recordings of such music, the
*compositions* are registered with BMI and ASCAP, and you'll need
blanket licenses to play them. (Well, if you only wanted one or two
tracks, you might
Quoting John Millican [EMAIL PROTECTED]:
there are plenty of radio stations with internet feeds of their audio,
piping that in would not change any coverage area since anyone with
internet could listen anywhere already, you're only providing that to
the listener through a phone handset
I have a polycom 501 phone,
when I call into the dialplan and do the following I do not hear the
DTMF digit.
exten = 51,1,playback(invalid)
exten = 51,2,SendDTMF(1)
This is on the 1.2.23 asterisk.
I am trying to re-create calling sendDTMF in an agi and not hearing the
digit either. The above
Yes and No
The D500 is a terrible thing
First problem: it sends some horrible DTMF, so if your voicemail is
configured to send #H and *H, it will not work, configure it to send
numbers, like 8H and 9H (H is a placeholder for the extension).
I also managed to use the MWI (message light), it's a
Hi all,
We have recently implemented an Asterisk system using Trixbox
(asterisk v1.4.4 at the moment, yet to move to 1.4.9) but are getting
pressure to switch back to our old key system unless we fix two major
issues. So please help me avoid switching back!
An overview: We have about 12
On Tue, Jul 31, 2007 at 05:22:20PM -0400, Jon Pounder wrote:
Quoting John Millican [EMAIL PROTECTED]:
there are plenty of radio stations with internet feeds of their audio,
piping that in would not change any coverage area since anyone with
internet could listen anywhere already, you're
Hi,
when I pick up a call with *8, the number of the caller isn't show on
the phone that picked up the call. Is there a way/chance to keep or
transfer the number of the caller?
We are currently using Asterisk version 1.4.1.
Thanks for any hints,
Stefan
--
Well, what i described it's called follow id
first you hook up , and then the pbx sends you the digits, after that
the call is connected until you hang up; this is the most common
method for connecting voicemails to pbx (the analogic way).
I think for the caller id you need an add-on card in the
On 7/31/07, Jeng Yu [EMAIL PROTECTED] wrote:
Hi All,
I have a telephony project for which I need
to build a prototype to demo for management.
The prototype must work on a GSM phone network.
In the demo system, a call from GSM phone comes
into the demo box. The demo box runs CallWeaver.
On 7/31/07, Benjamin Jacob [EMAIL PROTECTED] wrote:
Hello ppl,
Searched all over, but couldn't find anything conclusive.
Does an off-the-shelf version of Asterisk run without any issues on a
64-bit machine?
Does anyone have any 'conclusive' figures?
Apologies if this is a repeat question.
I have not had a chance to look at this, but if it is a fully
functional and threaded iax load balancer, thats extremely cool.
-bk
Stanisław Pitucha wrote:
Very low chances for that module if any. I haven't been using OpenSER much
and I don't think I'll be using it soon - but who knows.
Chan_bluetooth is now chan_mobile and included in
trunk/asterisk-addons. That would be my suggestion. It works very well.
Thanks,
Steve Totaro
Andrew Joakimsen wrote:
On 7/31/07, *Jeng Yu* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi All,
I have a telephony project
On 7/31/07, Richard Brady [EMAIL PROTECTED] wrote:
[Resent due to non-descriptive subject line.]
Hi folks
When connecting two SIP users, is there any way to stop Asterisk from
sending SIP 183 Session Progress messages, either globally or
per-peer?
Yes, the option is progressinband in
On 7/30/07, Vieri [EMAIL PROTECTED] wrote:
I would like to know if one can set the outgoing
caller ID within Asterisk when calls are going out
through:
1) an analog POTS line (I suppose not)
No.
2) a telco BRI line (I don't think so)
3) a telco PRI line (maybe)
Both are the same
Turn OFF CDP on the phones. I don't know if those phones support CDP,
but since CDP is the Cisco Discovery Protocol and those Linksys is owned
by Cisco As for Echo Canceling, that is the job of the device that
does VoIP/PSTN gateway functions.
Tom Lanyon wrote:
Hi all,
We have
Tom Wrote:
Hi all,
We have recently implemented an Asterisk system using Trixbox
(asterisk v1.4.4 at the moment, yet to move to 1.4.9) but are
getting pressure to switch back to our old key system unless
we fix two major issues. So please help me avoid switching back!
Have you tried
But the OP stated this was SIP - SIP calls -- As Dan mentioned, check
environmental issues -- hard walls, poor handset quality, noisy desks,
volume levels too high?
Eric ManxPower Wieling wrote:
by Cisco As for Echo Canceling, that is the job of the device that
does VoIP/PSTN gateway
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