nik600 wrote:
i'm sorry but due to some problem the software will be released not
first than Wednesday 7/02/2007. i'll post a message .
This should be Wednesday 7/3/2007. right?
--
Tomislav Parcina
[EMAIL PROTECTED]
___
--Bandwidth and Colocation
Hello Steve,
thanks for your anwer.
Yes, you are right we want to do VoIP telephone system capable also of
public address (overhead paging) service.
So synchronization is a key issue if we want to avoid unpleasant effects.
We are designing our phones and they will have also onboard amplifiers.
Hi Zoa,
yes the phone are design by ourselves and they will be application
specific.
Thanks,
Stefano
Zoa [EMAIL PROTECTED]
Inviato da:
Dear all,
I've implemented BLF for use with some Grandstream GXP-2000 phones and
it works fine in 1.2.x versions of Asterisk, although I tested it with
version 1.4.0 and it doesn't work! Has the needed syntax changed for
configure BLF for this version of Asterisk? It it a bug of this version?
Hi everybody :)
Can i configure My E1 line te recive send Fax?
Thank You
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hello Benito,
I suggest to specify which span to be used as the clock source (check
http://lists.digium.com/pipermail/svn-commits/2005-October/007955.html)
span=1,1,0,ccs,hdb3,crc4
bchan=25-39,41-55
dchan=40
span=2,0,0,ccs,hdb3,crc4
bchan=56-70,72-86
dchan=71
HTH
Best regards,
## nini @
On Fri, Mar 02, 2007 at 02:55:57PM +1100, Devraj Mukherjee wrote:
Hi Axel,
Everything installed and working well. Thanks very much. Quick
question, do you have MySQL support compiled into the rpms?
Yes, but not in these rpm, due to mysql's GPL licensing, read
Chris,
Here is how I might use this, I have a context called inside, is where each
of my extensions is dialed from. On my home box it looks like this.
[inside]
exten = 1000,1,Dial(SIP/1000,20,t)
What I would probably do is add the Notify command to each of my extensions
before my Dial, like so
Supa wrote:
Try this:
http://www.bayhamsystems.com/asterisk.html
Works for me just fine, and it is very easy to get up and running, even
with older version 1.2.3
I don't see a point of using providers as Bayhamsystems. First, it's
unpractical to send SMS from phone. If I'm going to use web
Ok, I've made some tests and resovled one of these problems.
PAP2 was configured to send a 100ms long DTMF tone and that is too
low. I changed it to 250ms and everything works fine now.
Now to the second problems, about Asterisk misinterpreting DTMF tone.
I put my sip peer into debug mode and
On 13:36, Fri 02 Mar 07, Tomislav Parcina wrote:
Supa wrote:
Try this:
http://www.bayhamsystems.com/asterisk.html
Works for me just fine, and it is very easy to get up and running, even
with older version 1.2.3
I don't see a point of using providers as Bayhamsystems. First, it's
Hi Trevor,
Thanks for your suggestion, it works by adding a Answer() in between!
However it will make everycall in the CDR become Answered.
Later on I found that setting progressinband=no in sip.conf finally fixed
this problem!
Best Regards,
Vincent
Message: 12
Date: Thu, 01 Mar 2007
On 2/28/07, Webster, Andrew [EMAIL PROTECTED] wrote:
Outbound calls on my Telus PRI aren't taking the Name portion of the
callerID. I've looked at the logs, and it is being set (see below),
but
the
PRI debug output doesn't show the name being sent anywhere. As a
result,
received
What do you mean by outbound CallerID Name? So that when calling a
POTS with CallerID service from telco the Name should show up as you
send it?
If the answer to the above is yes, then stop trying to do that. It
won't work, as the name that the POTS subscriber sees is NOT the one
you send,
younss azzayani wrote:
Hi everybody :)
Can i configure My E1 line te recive send Fax?
Yes, via iaxmodem and HylaFAX+
http://iaxmodem.sourceforgenet
http://hylafax.sourceforge.net
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
Stefano Totaro,
Off topic. I just noticed your name and was a little surprised!? ;-)
Are you in Italy / Sicily?
Anyways, you can achieve overhead paging through a sound card hooked to
an Amp and speakers from your PBX. I have yet to do it but have read
about it. I think this may be the
Tomislav Parcina wrote:
Supa wrote:
Try this:
http://www.bayhamsystems.com/asterisk.html
Works for me just fine, and it is very easy to get up and running,
even with older version 1.2.3
I don't see a point of using providers as Bayhamsystems. First, it's
unpractical to send SMS from phone.
Vincent Tam wrote:
Hi Trevor,
Thanks for your suggestion, it works by adding a Answer() in between!
However it will make everycall in the CDR become Answered.
Later on I found that setting progressinband=no in sip.conf finally
fixed this problem!
Best Regards,
Vincent
So does that mean that
Hi
The same is for me. BLF doesn't work with 1.4. I've added notifyringing
= yes and this doesn't help.
Show hints doesn't show any status changes so asterisk doesn't send any
NOTIFY messages to grandstream. Message is only sent when extension
unregisters.
Andrew.
Ricardo Carvalho:
Dear
Looks like it's a problem already logged on Mantis:
http://bugs.digium.com/view.php?id=8800
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrey Solovjov
Sent: 02 March 2007 14:01
To: Asterisk Users Mailing List - Non-Commercial Discussion
Telco Switching could be waiting for a ten digit number. I know that
Sprint and some others expect ten digit local calls
On 2/28/07, Mark Engelhardt [EMAIL PROTECTED] wrote:
Hello,
I just installed a PRI and when I make a local (seven digit) call, I
get Code 28 back from the telco, (I
Group
I'm having some trouble with asterisk and the page cmd.
Any help would be great!
This is what's in my extensions.conf
exten = _**2,1,SIPAddHeader(Call-Info: answer-after=0)
exten = _**2,2,Page(SIP/36651)|d
exten = _**2,3,Hangup
CLI output
Dears
Any one know how to let t38 works on asterisk 1.2 or an distribution like
trixbox have asterisk 1.4
Regards
Khaled Chehab
System Integration Engineer
Xplorium Offshore.
Sakiet Al Janzir
Postal Code: 1102-2080
Tel: (961) 1- 868 686
Fax :(961) 1-808 810
GSM: (961) 3-979
thanks Ioan
I've try this too but problem continues
I put :
span=1,1,0,ccs,hdb3,crc4
span=2,2,0,ccs,hdb3,crc4
span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
span=1,0,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
In all this configurations the problem is the same
I'm desperate
Jason,
If you do test if JR's tip works, please share your finding with us. I am
interested in this as well.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson
Sent: Thursday, March 01, 2007 21:11
To: asterisk-users@lists.digium.com
Quite surprising, yes! :-)
I am from north east Italy, now I live in Verona (Romeo and Juliet's city :).
I cannot do it connecting amp to the PBX. I have quite a long distance to cover
and a network is already there.
My phones are have a quite smart processor so we may probably run the ices2
Mike wrote:
Jason,
If you do test if JR's tip works, please share your finding with us. I am
interested in this as well.
It'll work fine, the Polycom responds with BUSY when the DND button is
pressed. Using DIALSTATUS, it'll drop to voicemail and play the busy
message if recorded if
On 3/2/07, Webster, Andrew [EMAIL PROTECTED] wrote:
What do you mean by outbound CallerID Name? So that when calling a
POTS with CallerID service from telco the Name should show up as you
send it?
If the answer to the above is yes, then stop trying to do that. It
won't work, as the name
Hi, I'm the admin on the list your test didin't work you should resend it.
Well I am not the admin, just wanted you to realized that you
shouldn't annoy thousands of people just
Hall, Eric M. wrote:
Group
I’m having some trouble with asterisk and the page cmd.
Any help would be great!
This is what’s in my extensions.conf
exten = _**2,1,SIPAddHeader(Call-Info: answer-after=0)
exten = _**2,2,Page(SIP/36651)|d
exten = _**2,3,Hangup
Looks like you have at least a
Thanks Steve!
What are usually the best approaches in troubleshooting the audio
quality issues and QoS related stuff when putting two Asterisk boxes
together via IAX?
Have you ever tried connecting Asterisk boxes in the same VPN (but still
in different countries)?
Regards,
Alex
-Original
Hi,
I have a requirement for sending and receiving faxes and was wondering the best
way to achieve it with Asterisk as I only have one phone line.
I currently have a TDM11B in my server (1 x FXO, 1 x FXS), so I was thinking
that I would need to get a additional FXS module, connect that to a
Interesting read in Alec Saunders blog today.
http://saunderslog.com/2007/03/01/mashable-telcos/
Thought it may interest some people on this list.
As food for thought, why it is that ITSP's haven't come up with more
'interesting' voice applications? When asterisk first became available I
For an all electronic solution, use fax2mail and mail2fax (from
www.generationd.com). For a fancier all VOIP solution consider hylafax.
For analog only you can plug your fax machine in as you suggest. For a step
up, buy an ATA with T.38 capability and plug your fax machine into that.
MD
You will likely have latency issues - causing choppiness. Start with a
traceroute to validate latency.
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support
Visit us at www.generationd.com
-Original Message-
From: [EMAIL
I am using Asterisk 1.2 with a TE410P connected to E1 PRI trunks.
The application relies on a DTMF digit string sent by the phone
after the call has connected. This DTMF is detected by Asterisk
under the control of WAIT FOR DIGIT commands send from an AGI
processor over a FastAGI connection.
On Fri, Mar 02, 2007 at 03:34:19PM +, --[ UxBoD ]-- wrote:
Hi,
I have a requirement for sending and receiving faxes and was
wondering the best way to achieve it with Asterisk as I only have one
phone line.
I currently have a TDM11B in my server (1 x FXO, 1 x FXS), so I was
I would just put the phone line into the fax machine and the phone jack
from it into the fxs card. It is a simple solution.
Bob
--[ UxBoD ]-- wrote:
Hi,
I have a requirement for sending and receiving faxes and was wondering the best
way to achieve it with Asterisk as I only have one phone
Asterisk wrote:
Thanks Steve!
What are usually the best approaches in troubleshooting the audio
quality issues and QoS related stuff when putting two Asterisk boxes
together via IAX?
Have you ever tried connecting Asterisk boxes in the same VPN (but still
in different countries)?
Regards,
--[ UxBoD ]-- wrote:
Hi,
I have a requirement for sending and receiving faxes and was wondering the best
way to achieve it with Asterisk as I only have one phone line.
I currently have a TDM11B in my server (1 x FXO, 1 x FXS), so I was thinking
that I would need to get a additional FXS
You will likely have latency issues - causing choppiness. Start with a
traceroute to validate latency.
Anybody tried IAX trunking on G.729 with jitter buffer internationaly?
-HJC
___
--Bandwidth and Colocation provided by Easynews.com --
Dean Collins wrote:
Interesting read in Alec Saunders blog today.
http://saunderslog.com/2007/03/01/mashable-telcos/
Thought it may interest some people on this list.
As food for thought, why it is that ITSP’s haven’t come up with more
‘interesting’ voice applications? When asterisk first
Hi all,
We want to put an Asterisk Voicemail Server behind a Cisco Call Manager.
The idea is to have Cisco Phones (SCCP) registred to the CCM and the voicemail
in the Asterisk Box.
The trunk inter PBX is in SIP.
My question is:
Is it possible to activate MWI LED from the Asterisk to the Cisco
Sounds like the DTMF tones are too far from spec, or noisy. Is the DTMF
being transcoded somewhere along the way?
If you have time to killtry to separate the two frequencies in your
software (I don't know goldwave) - are both present and clean and same
amplitude and on freq? Remove the two
Thanks for the tip.
Would that apply to connections via VPN, or is this likely to happen in
any scenario where two Asterisk boxes are far away from each other?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michelle
Dupuis
Sent: Friday, March 02, 2007
exten = 111,1,Wait(1)
exten = 111,2,Playback(Randy)
exten = 111,3,Dial(Sip/Randy,20)
exten = 111,4,Goto(111-${DIALSTATUS},1)
exten = 111-BUSY,1,Voicemail([EMAIL PROTECTED],u)
exten = 111-NOANSWER,1,Dial(IAX2/${TELIAX_OUT}/212551212)
works GREAT
Thanks a lot
Jason
Doug Lytle wrote:
Mike
Some tricks:
If you have a high latency link, watch out i've experienced problems
with it in the past. (high latency = 300ms)
Stay away from trunking unless you have a lot of time to spend, if you
do use trunks, do not use a jitter buffer, make sure it works in both
directions and don't
I don't think it is a matter of GeeWhizz, for me it is about simple
things that make my life easier.
1) I get an SMS from meetme when the first joiner arrives , the sms
has the DID of the conference room in it.
2) I get calls from asterisk when one of our servers are down. The
calls are
Henry J. Cobb wrote:
You will likely have latency issues - causing choppiness. Start with a
traceroute to validate latency.
Anybody tried IAX trunking on G.729 with jitter buffer internationaly?
-HJC
Not G.729 but SPEEX 8 variable. Sounded great but lots of delay where
GSM was
Lol - I didn't mean it that way besides there is a big difference
between discussing an application and replicating it.
Besides if you are out there selling something the general public is
going to know about it.
Or should I be checking your public website for the super duper cool
ones :)
Hello all,
I will be glad, if someone can throw light on Voicemail to SMS using
asterisk.
1. I want my users to dial certain number.
2. Record a voicemail with destination number.
3. Convert this Voicemail to Text.
4. Send the text with sms apps.
and I wish i connect my
On Fri, 02 Mar 2007 10:59:25 -0500
Robert A. Rawlinson [EMAIL PROTECTED] wrote:
I would just put the phone line into the fax machine and the phone jack
from it into the fxs card. It is a simple solution.
Bob
--[ UxBoD ]-- wrote:
Hi,
I have a requirement for sending and receiving
I don't think it works. I tried calling my own yahoo messenger ID with
no success: it rings a number of times and then it goes to some sort of
voice mail.
And I did invite the user they specified to my yahoo list, I also
entered my yahoo id into the registration form on the site.
I used a
Hello all,
I will be glad, if someone can throw light on Voicemail to SMS using
asterisk.
1. I want my users to dial certain number.
2. Record a voicemail with destination number.
3. Convert this Voicemail to Text.
4. Send the text with sms apps.
and I wish i connect my
On 08:42, Fri 02 Mar 07, Steve Totaro wrote:
Do they let you specify what number the SMS is coming from or does it
just come from one of their number pool? Is it possible for the person
reply to the SMS and have it come straight to my phone?
Yes. Both CIDname and CIDnumber can be specified
Hi,
I have two asterisk servers one is connected to the PSTN and the other
one is connected to SIP users. The two servers connect with each other
using IAX. When I have an incoming call from PSTN to the asterisk
servers and have a forward to go back out to the PSTN the two IAX
channel bridge
With one IVR payment system, I noticed quite a difference in DTMF
transmission between these two cards. The IVR missed nearly all digits from
X100P, while receiving digits from TDM fine.
Since neither card process or synthesize audio, what can the difference be?
(This particular IVR has
Nerd Vittles has some pretty cool apps for news and having your email
read to you. I am downloading now, hopefully it is halfway decent.
Thanks,
Steve
Dean Collins wrote:
Lol - I didn't mean it that way besides there is a big difference
between discussing an application and replicating it.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tim Panton
Sent: Friday, 2 March 2007 11:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Alec Saunders post about Mashable
Telco's
On
Goke Aruna wrote:
Hello all,
I will be glad, if someone can throw light on Voicemail to SMS using
asterisk.
1. I want my users to dial certain number.
2. Record a voicemail with destination number.
3. Convert this Voicemail to Text.
4. Send the text with sms apps.
and
On 2 Mar 2007, at 08:04, Steve Totaro wrote:
Asterisk wrote:
Thanks Steve!
What are usually the best approaches in troubleshooting the audio
quality issues and QoS related stuff when putting two Asterisk boxes
together via IAX?
Have you ever tried connecting Asterisk boxes in the same VPN
Bruce Reeves wrote:
There was talk last week that SLA in 1.4 was not working correctly and
was being rewritten for a 1.4.1 release.
The re-write of SLA support in Asterisk 1.4 is pretty much complete.
Asterisk 1.4.1 will be released very soon with it included. I don't
expect to make any
Remi Quezada wrote:
I have two asterisk servers one is connected to the PSTN and the other
one is connected to SIP users. The two servers connect with each other
using IAX. When I have an incoming call from PSTN to the asterisk
servers and have a forward to go back out to the PSTN the two IAX
Somebody make an application where I can browse to a web front end, set
up some RSS feeds, and the system assigns me an access number or PIN.
Later, when I am stuck in the airport, I call the app, and a decent TTS
engine reads me my RSS feeds.
Cory Andrews
-Original Message-
From:
How about mashing text to speech and Avantgo together.
Using dtmf you could have your avantgo selections read to you while you
drive to work in the morning? (or even better use lumenvox/tell me
speech recognition)
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
Anyone know how to let asterisk deal with the progress codes coming
from the carrier? The problem I am having is when a customer calls an
invalid number the carrier tells me the call is invalid via a progress
code but doesn't route me to a recording (this number is invalid).
Instead they hang up
I DID receive it. Please don't re-send it.
C F wrote:
Hi, I'm the admin on the list your test didin't work you should resend it.
Well I am not the admin, just wanted you to realized
I am trying to have asterisk update the system name in my realtime
peers, but it does not seem to be working. Here is what I've done so
far.
- added systemname = mysystemname in asterisk.conf
- set rtsavesysname=yes in sip.conf.
- created a table called sysname in my peers table in mysql
-
Another option is to have the user hit the forward button on their phone
and manually type in their cellphone number when they're going to be out
of the office.
Jason Walker wrote:
exten = 111,1,Wait(1)
exten = 111,2,Playback(Randy)
exten = 111,3,Dial(Sip/Randy,20)
exten =
Possibly the called party is not sending their DTMF properly? maybe
experiment with inband/rfc2833/etc in the CALLED party's peer definition
Denis V. Gudtsov wrote:
Hello!
I'm using asterisk 1.2.13, in extension.ael is set Dial(SIP/${EXTEN},12,tT)
but only calling party can do forward. How
On Fri, Mar 02, 2007 at 09:36:28AM -0800, Yuan LIU wrote:
With one IVR payment system, I noticed quite a difference in DTMF
transmission between these two cards. The IVR missed nearly all digits
from X100P, while receiving digits from TDM fine.
Since neither card process or synthesize
On Fri, Mar 02, 2007 at 10:20:07AM -0800, Tim Panton wrote:
On 2 Mar 2007, at 08:04, Steve Totaro wrote:
Asterisk wrote:
Thanks Steve!
What are usually the best approaches in troubleshooting the audio
quality issues and QoS related stuff when putting two Asterisk boxes
together via
On 3/2/07, Russell Bryant [EMAIL PROTECTED] wrote:
If you are interested in beginning to look at it now, just pull the code
from the 1.4 branch.
Russell, I don't have any specifics at this time. I need to dig a
little further. I'm thinking the autocontext is what is giving me
fits. I can
Did that. No change
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee
Jenkins
Sent: Friday, March 02, 2007 10:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Spam? Re: [asterisk-users] cmd page crashes Asterisk
Now if only Tellme would agree to a pre-paid sip gateway for the
asterisk community you could have pretty much everything you wanted :)
http://www.voip-info.org/wiki/view/tellme
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642
Hello Benito,
From http://www.beronet.com/download/card_installation_guide.pdf we
could find that:
/After loading the driver and the executing ztcfg, status LEDs for
each port should be flashing red, unless the port is connected to a
device. If the LED does not light up, the driver did not
Then install it out in Iowa and get your money back at 2c a minute :)
http://gigaom.com/2007/02/26/iowa-telcos-att-owes-12-million
http://saunderslog.com/2006/10/11/whats-with-the-712-area-code
I posted this question the other day but didn't get an answer, are there
any other toll zones
No. Asterisk does not have fax support and there are no plans to add it.
You can however send the faxes as voice calls, however there is no
assurance as to its reliability. Maybe most will work but some will
without a doubt be predestine to fail.
On 3/2/07, --[ UxBoD ]-- [EMAIL PROTECTED]
No there is no fax support in Asterisk.
On 3/2/07, Khaled [EMAIL PROTECTED] wrote:
Dears
Any one know how to let t38 works on asterisk 1.2 or an distribution like
trixbox have asterisk 1.4
Regards
Khaled Chehab
System Integration Engineer
Xplorium Offshore.
Sakiet Al Janzir
If the remote end isnt up that could explain why the correct settings
dont work. If they told you to use those settings thats how it should
be provisioned, either:
1) You think your configuration is correct but you are wrong and it
indeed is incorrect
2) Your telco didn't correctly provision or
That's like saying a pinto is fast when you upgrade the engine. Well
Asterisk also supports T.38 for free... if you backport OpenPBX.org
fixes.
But realisticly ASTERISK DOES NOT HAVE FAX SUPPORT STOP CLUTTERING MY
INBOX WITH DISCUSSION OF FEATURE THAT DOES NOT EXSIST.
On 3/1/07, Zoa [EMAIL
Maybe.. if you dont expect to recieve calls to any device, then I just
wouldnt bother to register.
On 2/28/07, Ricardo Carvalho [EMAIL PROTECTED] wrote:
Can't I register multiple phones with the same user/password? That's
what I pretend to do, not ring groups...
Thanks,
Ricardo.
John Bittner wrote:
Anyone know how to let asterisk deal with the progress codes coming
from the carrier? The problem I am having is when a customer calls an
invalid number the carrier tells me the call is invalid via a progress
code but doesn't route me to a recording (this number is invalid).
Please note that we are available to fix the current REMOTE crash that
affects Asterisk/openpbx/trixbox and crashes these systems via a malformed
packet
please contacts use if you need a hand to patch your systems.
--
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP
Try renaming you column in the peers table to regserver
On 3/2/07, David Thomas [EMAIL PROTECTED] wrote:
I am trying to have asterisk update the system name in my realtime
peers, but it does not seem to be working. Here is what I've done so
far.
- added systemname = mysystemname in
Steve Totaro wrote:
Goke Aruna wrote:
Hello all,
I will be glad, if someone can throw light on Voicemail to SMS using
asterisk.
1. I want my users to dial certain number.
2. Record a voicemail with destination number.
3. Convert this Voicemail to Text.
4. Send the
Has anyone used talked to astmanproxy with the Asterisk Java Manager
interface? First suspiscions are that it will not work.
Astmanproxy sends a connection banner of 'Asterisk Call Manager
Proxy/1.21' which is not what Asterisk Java is expecting. Also,
astmanproxy preprends the name of the
Hi ALL!!
This is my first time with asterisk and my first post:)
Please be so kind and give me few clues for the good beginning:)
I am using version 1.4 from svn on the Debian etch OS kernel 2.6.
I have four question:
1. In many docs on the web there is an info to make asterisk by invoking
I have multiple IP addresses on my box. My provider just changed my eth0 IP
off to another interface (lo:9) and a new IP on eth0. Nothing works anymore
because calls to the old IP address are being answered by the new IP
address. How do I straighten this out?
From: Tzafrir Cohen [EMAIL PROTECTED]
Date: Fri, 2 Mar 2007 22:14:09 +0200
On Fri, Mar 02, 2007 at 09:36:28AM -0800, Yuan LIU wrote:
With one IVR payment system, I noticed quite a difference in DTMF
transmission between these two cards. The IVR missed nearly all digits
from X100P, while
Ok, so I ain't much of a Java programmer, but...
Can the Asterisk Java API be written with threads? Ie, I need to connect
to multiple Asterisk systems from the one java application. I tried to
make my class which implements ManagerEventListener, also implement
Runnable, but got errors
On 3/2/07, Mike Lynchfield [EMAIL PROTECTED] wrote:
Please note that we are available to fix the current REMOTE crash that
affects Asterisk/openpbx/trixbox and crashes these systems via a malformed
packet
please contacts use if you need a hand to patch your systems.
And you'll do it for
Mike Lynchfield wrote:
Please note that we are available to fix the current REMOTE crash that
affects Asterisk/openpbx/trixbox and crashes these systems via a
malformed packet
please contacts use if you need a hand to patch your systems.
This list is for non-commercial discussion, as is
1) You think your configuration is correct but you are wrong and it
indeed is incorrect
evenif with my config i get a green led and with the telco config the
lid is red ?
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On Fri, Mar 02, 2007 at 01:48:21PM -0800, Yuan LIU wrote:
From: Tzafrir Cohen [EMAIL PROTECTED]
Date: Fri, 2 Mar 2007 22:14:09 +0200
On Fri, Mar 02, 2007 at 09:36:28AM -0800, Yuan LIU wrote:
With one IVR payment system, I noticed quite a difference in DTMF
transmission between these two
Doug Garstang wrote:
Can the Asterisk Java API be written with threads?
sure.
Ie, I need to connect
to multiple Asterisk systems from the one java application. I tried to
make my class which implements ManagerEventListener, also implement
Runnable, but got errors because the Runnable
I would like to log a verbose statement in my 900/976 extens to a
special file called 'attacks'.
These are not standard messages like debug, notice, warning, error,
vebose or dtmf that could be logged to /var/log/asterisk/messages.
Does the 'verbose' in VERBOSE commands have anything to do
nope
http://svn.digium.com/view/asterisk/branches/1.2/channels/chan_sip.c?r1=56230r2=57475
is avail for free..
6574a6575,6579
if (uri == NULL) {
ast_log(LOG_WARNING, register_verify: URI is NULL!\n);
transmit_response_with_date(p, 503 Bad Request, req);
Lacy Moore - Aspendora wrote:
Russell, I don't have any specifics at this time. I need to dig a
little further. I'm thinking the autocontext is what is giving me
fits. I can receive calls and place calls, but the hint status is not
working. It currently registers as a hint showing not in
The Asterisk and Zaptel development teams have released Zaptel 1.2.15.
This release contains a significant Astribank (XPP) driver update,
support for Digium's TE120P card, and various bug fixes.
Thanks for your support of Asterisk and Zaptel!
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