It has worked ok for me. Do you have the 'dtmfmode=rfc2833' line in the sip
config for the 9133i extension?
-Original Message-
From: Karl S. Katzke [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 30, 2005 5:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
I can't comment on anything but AMP. I've found it very easy to use.
Most recently, I've switched to [EMAIL PROTECTED] 1.5 which includes AMP +
other tools and found this package to be very interesting and intuitive.
I believe an SMB could manage Asterisk with a little guidance using the
[EMAIL
Has anyone made this work? For me everything is fine until I switch
canreinvite form no to yes. What happens is that asterisk hangs on
attempting native bridge ... from what I understand attempting native
bridge means that the RTP is routed through asterisk (just without any
codec translation)
Can someone tell me how to do this...Given the following line:
exten = *97,3,VoicemailMain([EMAIL PROTECTED])
Is it possible to add some logic to manipulate the CALLERIDNUM to send
back 801 even if the extension is 601 and 901 even if the extension is
701? I have 2 branch offices where users
Let me know the cost.
regards,
Umair bari
On 8/31/05, Chris A. Icide [EMAIL PROTECTED] wrote:
In the next week to two weeks I'll be posting some informationconcerning a system I've been designing.It currently does three layer
hosted VoIP pbx services as well as hosted ITSP services (the model
Hi,
Try SetCIDNum application before VoiceMail application
regards,
srsergio
-Mensaje original-
De: Chad Brown [mailto:[EMAIL PROTECTED]
Enviado el: miércoles, 31 de agosto de 2005 8:48
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [Asterisk-Users] Manipulate
This is probably part of the equation. However, I really need a dev to suggest
some logic that first determines if the CALLERIDNUM starts with a 6 and if so
increase the CALLERIDNUM by 200. (Probably suing the SetCIDNum)
I just need the logic. ;-) Thanks for your help.
-Original
Hi,
Few days ago I bought a Sipura SPA-3000 Gateway. Outgoing calls works fine
but incomming calls behave very strange. When I dial my Sipura from outside and
cancel befor picking up, the phone still rings for about one minut. What is
wrong - the Sipura Gateway or I did something wrong with
On Tuesday 30 August 2005 17:01, Braz wrote:
Your kernel has to be compile with CONFIG_CRC_CCITT=y or m.
I couldn't find that option in the kernel, but inserting the zaptel module
before ztdummy works of course.
___
--Bandwidth and Colocation
This option is under Library routines in your kernel configuration.
Regards,
srsergio
-Mensaje original-
De: Christoph Eicke [mailto:[EMAIL PROTECTED]
Enviado el: miércoles, 31 de agosto de 2005 10:59
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re:
Am I reading the data below incorrectly, or does it appear that even
though I have the directive canreinvite=no set for the two asterisk
boxes, they are trying to do a reinvite (which fails) anyway?
Is this expected behaviour in this situation? If so, how can I prevent
this?
Lots of
On Wednesday 31 August 2005 11:11, Sergio Serrano wrote:
This option is under Library routines in your kernel configuration.
ah, yes. In kernel 2.6.* it is. not in 2.4.26 ;-)
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users
voipreach.net - are they functioning, I sent them a few emails and they did not
reply.
are they operational?
This message was sent using IMP, the Internet Messaging Program.
___
Florian Overkamp wrote:
Hi,
Daniel Grad wrote:
I am writing a script (php script that runs via fastAGI) that takes
incoming calls and processes them in various ways depending on
settings from a database.
At some point, I need the script to receive an incoming fax. But the
problem is that
On Tue, Aug 30, 2005 at 03:01:07PM +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Christoph Eicke [EMAIL PROTECTED] wrote:
Hi!
When I try to load the ztdummy driver via insmod ztdummy, I get the
following errors:
/lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol
Karl -
Try setting the parameter"Live Dialpad" to
"on" to fix this issue. This is option 6 on the physical menu on the
telephone or you can set it through the mac.cfg file on your TFTP
server.
I'm using the 9133i's without that issue regardless
of the live dialpad setting. I have a
Nobody else needing this feature?
On Mon, 29 Aug 2005 17:43:07 +0200
ChB [EMAIL PROTECTED] wrote:
hello!
i'm looking for a feature to play a sound-file containing a text until the
called party picks up the phone. i've already tried with the 'special'
musiconhold-feature by adding the
Yes ever since the hurricane hit, I have had crackling on the line and MAJOR
delays and even some echo. Some odd pings to Teliax have been noted as
well. I have had no problems with Telasip (my backup). But on a similar
note, I have tried to dial many texas, mississipi and florida phone and
Hi!
I have a strange problem with Asterisk (Asterisk 1.0.8-BRIstuffed-0.2.0-RC7k)
on a VIA Samuel x86. When I make a call from the CLI either over IAX2 or SIP,
my first call after the initial start of Asterisk works fine, even though
upons starting Asterisk tells me Read error on sound device:
[EMAIL PROTECTED] wrote:
Probably not Geoff. It is still digital at that point I think.
It should be coming to you as a four wire balanced circuit.
It depends on which legacy PBX you are using (tho it is pretty
standard) And if it wasn't right - it probably wouldn't work at all.
Brett
hi frank,
i appologize but, i dont know what to change, which makefile to change , and
what will be the changes.
i dont have any idea how to make scripts and codes.
pls. help.
thnks.
- Original Message -
From: Frank Tarczynski [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
Stephen wrote:
Hi All,
I have configure my Asterisk as follow (using [EMAIL PROTECTED]):
[zaptel.conf]
span=1,1,0,ccs,hdb3,yellow
bchan=1-15,17-31
dchan=16
loadzone = uk
defaultzone=uk
try this in your zaptel.conf:
span=1,0,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
CRC required for
Soner Tari escreveu:
I use HT488, and I can make and receive FXO calls. It's actually quite
simple, you create a SIP acount in sip.conf. On the FXO section of
HT488 web admin page you enter these registration values. When you
reboot the HT488 you should see it registering on Asterisk CLI.
Hi,
Anyone can point me to a way to get the SIP phones status information
(off-hook, on-hook,...). Either through Asterisk or directly from the
phone (standard API?).
I'm working with the Aastra 9133i.
Thanks for any pointers.
--
Andre Courchesne
I just recompiled my version from this morning's CVS Head.
My systems voice files (voicemail, time etc) were playing nicely. Until
that is I added an extension and now the files won't play.
Worse than that, * thinks the files have played and goes to the next
step in the dial plan.
What
Hi everybody
After setting up trunk with FWD, all I get on my Asterisk
box is message saying that all circuits are busy now, try your call later. Even
612 (time) says the same thing. Why is it that and how can I fix it.
Zeeshan
___
Joseph wrote:
Is it possible to do nested dial() command on one line,
Dial number, wait new seconds, dial another number etc.
or dial number and jump to another line to continue dialing.
D(ww) doesn't work as it sends DTMF but before the call is bridged, and
I need to send numbers after the
I use HT488, and I can make and receive FXO calls. It's actually quite
simple, you create a SIP acount in sip.conf. On the FXO section of HT488
web admin page you enter these registration values. When you reboot the
HT488 you should see it registering on Asterisk CLI.
What's left is a
Sounds like you lost timing, either because a zaptel device driver did
not load or ztdummy did not load if you have no zap hardware.
* needs a cock to play sounds and keep timing, the clock comes from the
PSTN, zap hardware, or ztdummy depending on how you are set up.
-Original
I am planning to sign up for a VoIP service in the U.S. Can anyone
recommend anything cheap, reliable and good quality? I want to use it
for my primary house phone (I also own a cell phone).
I also want the service to be asterisk friendly so I can play with it :-)
Thanks in advance.
On Wed, 2005-08-31 at 09:54 -0300, Keith Yoder wrote:
Soner Tari escreveu:
I use HT488, and I can make and receive FXO calls. It's actually quite
simple, you create a SIP acount in sip.conf. On the FXO section of
HT488 web admin page you enter these registration values. When you
Here, have an 'l' - I've go a couple spare on my keyboard :)
I guess it needs a clock to play sounds...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Damon Estep
Sent: 31 August 2005 14:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Asterisk only needs a cLock to play MOH afaik, on 2.6.x kernels you
don't need any timing help, on 2.4.x you can use ztdummy on the USB drivers
Damon Estep wrote:
Sounds like you lost timing, either because a zaptel device driver did
not load or ztdummy did not load if you have no zap
I use BINK to burn ISO Images and it works great.
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Tuesday, August 30, 2005 11:09 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] (no subject)
On Tue,
Anybody have any clue whats up with them?
Bruce can you contact your friend over at simpletelecom?
It has been a while they had enough time to move the stand why are
they not up yet?
http://lists.digium.com/pipermail/asterisk-users/2005-July/115137.html
Peter Svensson wrote:
ICD has its own mailinglist at [EMAIL PROTECTED] There is
close to zero traffic there as well. I think the authors read it though.
Peter, thank you very much for the response (which I snipped), and for
the pointer to these (very quiet) lists as well. I just subscribed to
Yeah,
When the content warning came back from the mail filter if figured
typing messages on my pocket pc was not a great idea!
I guess that is where ztdummy comes into play.
A little comedy to start the day...
Damon
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
Not so (at least in HEAD), no prompts, including voicemail, will play
without clock at least in my experience with FC4 2.6.12.
Timing is also important in other areas, meetme for one.
You really can not or should not run without a reliable timing source or
you will not have a reliable system.
Perhaps this is all related.
I'm on a 2.4 kernel. I have a an x100p clone installed which the drivers
find but * doesn't.
Mark
Damon Estep wrote:
Sounds like you lost timing, either because a zaptel device driver did
not load or ztdummy did not load if you have no zap hardware.
* needs a
I'm on Broadvoice and I think its fine.
They don't support it though. It is however well documented.
There's not too many providers that'll let you play with your service.
Most of the Broadvoice types won't even allow you to set your own CID.
Mark
Mag Gam wrote:
I am planning to sign up for
You don't say how you are connecting to FWD? I use IAX and it works like
a charm.
Mark
Zeeshan Zakaria wrote:
Hi everybody
After setting up trunk with FWD, all I get on my Asterisk box is message
saying that all circuits are busy now, try your call later. Even 612
(time) says the same
Here, have a couple of my s ;)
Steve Langstaff wrote:
Here, have an 'l' - I've go a couple spare on my keyboard :)
I guess it needs a clock to play sounds...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Damon Estep
Sent: 31
B. J. Bomar wrote:
I am also having the same issue from the ftp tarball.
I've tested the tarball on a bunch of different systems and it worked
properly.
Please post the contents of the include/asterisk/version.h file from
your source tree after the build (and check if there is a .version
Hi Mag,
I use Packet 8, you have to use their ata
but there is an unlock software that allows you to modify the codecs J (makes a huge
difference)
They also allow you to forward all calls
to a second number so your incoming asterisk line could also be permanently
forwarded to your cell
I just setup VoicePulse (http://connect.voicepulse.com/), and it was
very easy (good setup docs in [EMAIL PROTECTED] guide, which is what I'm
using).
Limited experience yet, but it was very easy and quick, and pretty
cheap too. IAX support, multiple phone #'s, etc.
On an semi-unrelated note,
Scott's SpanDSP FAQ mentions that the senders' TSID (20 bytes sent by
sending fax machine) is made available, but doesn't mention where or
how. Is there a variable or something that's set? I didn't see it in
http://www.voip-info.org/tiki-index.php?page=Asterisk+variables
Thanks,
Woody
Thanks everyone for their replies!
I am going for something with no activation charges.
Brian of
PlainVoip Support do you guys have any such charges? I want unlimited long distance, how much should I expect to pay?
I am faily new to this, can I buy myself a VoIP phone, like Cisco or 3com
On Wed, 31 Aug 2005, Hadar Pedhazur wrote:
My only real problem with my current setup is that because I use Call
Files to contact the Agents, I have no direct way to cancel ringing
phones when the call has been bridged to another channel.
You can use the Manager interface with the Originate
Des anyone is using telextreme and asterisk?
Can I just take the parameters out the ata and place it to * and works?
ANy experienice?
I have been googglin about it, and there is no further information...
___
--Bandwidth and Colocation sponsored by
Hi all,
We've got a department that has 5 phones using a * 1.0.9 box. They need
to have an extension that rings all 5 phones at the same time. Getting
all of the phones to ring isn't a problem, but they are running into a
problem with the phones ringing in their ears when they are already
Woody Sturges wrote:
Scott's SpanDSP FAQ mentions that the senders' TSID (20 bytes sent by
sending fax machine) is made available, but doesn't mention where or
how. Is there a variable or something that's set? I didn't see it in
We did quite some work on ICD for a customer implementation and will
be passing that to bruce et all during the next few days.
On 8/31/05, Hadar Pedhazur [EMAIL PROTECTED] wrote:
Peter Svensson wrote:
ICD has its own mailinglist at [EMAIL PROTECTED] There is
close to zero traffic there as
You can give www.broadvoice.com a try. Only been using
them for a few weeks, but their prices are decent.
Bjorn
Fra:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
På vegne av Mag Gam
Sendt: 31. august 2005 15:17
Til:
asterisk-users@lists.digium.com
Emne: [Asterisk-Users] VoIP
1) I dial the following number
02111203012
And I get a no answer.
2) But dialing
2111203012 does answer
I have two routes up:
^2.* ALCATEL-OMNI 500 10 500
^021.* ALCATEL-OMNI 500 10 2500
Can someone tell me where the problem is ?
thanks
--
No virus found in
After setting up trunk with FWD, all I get on my Asterisk box is message
saying that all
circuits are busy now, try your call later. Even 612
(time) says the same thing. Why is it that and how can I fix it.
If you're using iax with FWD, it would appear they have a problem with
it. I've
If you can do this is a dial plan have you tried chanisavail
application? Seems like that should work for you.
on Wednesday 08/31/2005 Eric \Skippy\ Hope([EMAIL PROTECTED]) wrote
Hi all,
We've got a department that has 5 phones using a * 1.0.9 box. They need
to have an extension that
Asterisk Supporter wrote:
Additional info:
This is on the 1.0.9 CVS. I failed to include that info on my original
post. I have not tried the tarball.
It can't be, because there is no ast_expr2 _anything_ in CVS v1-0. I
suspect you may have a source directory with mixed checkouts from
I installed/ran both MozPhone and DIAX but did not see in the debug any
information of the URL I sent. Perhaps the real question is: if
optionalurl is used, how is the url sent to the device(s)?
Has anyone applied this within a solution and is willing to share their
experience?
Thanks!
Jason
Okay! What do you guys think about SunRocket
(https://www.sunrocket.com/sign_up/linkshare_entry.do?plan=ypartner=lssiteID=ydmf4rFDNTw-WD0UFulSHtqwjMnOUau4yg
)
Should I go for this?
On 8/31/05, Bjørn Ove Kristiansen [EMAIL PROTECTED] wrote:
You can give
www.broadvoice.com a try.
Peter Svensson wrote:
On Wed, 31 Aug 2005, Hadar Pedhazur wrote:
My only real problem with my current setup is that because I use Call
Files to contact the Agents, I have no direct way to cancel ringing
phones when the call has been bridged to another channel.
You can use the Manager
You could do that and terminate directly
to a sip provider but then you lose the benefits of asterisk.
If you are only new to asterisk then check
out [EMAIL PROTECTED] http://asteriskathome.sourceforge.net
its by far the easiest way to get started.
Cheers,
Dean
From:
Thanks Steve. That's exactly what I was looking for.
At the risk of embarrasing myself (I know, too late), where is this
'help'? I've searched the Wiki, I've been on the Digium forums, I've
looked for it in the Asterisk CLI, etc. I'm missing something that
seems to be pretty obvious. I'm
Some people on another forum have been
complaining about them, they also dont allow you to port your number out
if you want to leave.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mag Gam
Sent: Wednesday, 31 August 2005
10:57 AM
To: Asterisk Users Mailing List
On 10:22, Wed 31 Aug 05, Eric Skippy Hope wrote:
Hi all,
We've got a department that has 5 phones using a * 1.0.9 box. They need
to have an extension that rings all 5 phones at the same time. Getting
all of the phones to ring isn't a problem, but they are running into a
problem with
The sea is getting bigger.. (sea of providers.. sea of emails on this topic)
voip-info.org has a list of quite a few. I use broadvoice and have no issues, but if when you say asterisk friendly you mean supports iax, then you'll have to look eslewhere. I have both SIP and IAX providers and I
I have a small phone system built around Asterisk stable utilizing a PRI
trunk and approximately 25 Uniden UIP 200 sip phones. I have two call
queues, nothing exotic, serviced by up to three call agents. Whenever the
agents transfer a call, the queues do not register a call transfer or
Hey,
I am getting a dedicated server 2Ghz, 1GB RAM, 60GB (RAID 01) from
EV1Servers and getting the Unlimited Residential + 9.95/month fax line
from Voice Pulse Connect. I have a 5 phone setup.
Does anyone have any comments on this setup or recommendations ?
I will not use any VOIP service that requires a large
upfront payment, in this case a 1 year service charge
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mag
GamSent: Wednesday, August 31, 2005 10:57 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
I thought that I would try this on iax.conf as well however I still get
asterisk asterisk as the callerid name and num.
I have the latest CVS as of 8/17/05
Has anyone have this working with iax incoming?
Thanks
Ben
That worked. The following line also got rid of asterisk without
entering
Dear list,
i am using the manager.conf account for ruuning with
the asterisk mrtg perl file but it does not
authenticates my account the error messages is as
followed
./a.out -h localhost -u vrk -p vrk -P 6080
Constant subroutine POLLIN redefined at
for the particular configuration of software/hardware that connects to
my asterisk pstn gateway I need to do something like the following :
[...]
exten = _X,3,Dial(CAPI/02xxx.b${EXTEN},60,M(senddtmf))
[...]
[macro-senddtmf]
exten = s,1,SendDTMF(*)
but the DTMF must be sended to the caller
Hi all,
I've been told that part of best practices is to noload all modules that
you don't use. Adsi being one of them (are there actually people using
adsi asterisk?) I noloaded it in /etc/asterisk/modules.conf:
noload = res_adsi.so
noload = app_adsiprog.so
When I start * (this is cvs head of
Jason Walker a écrit :
I installed/ran both MozPhone and DIAX but did not see in the debug any
information of the URL I sent. Perhaps the real question is: if
optionalurl is used, how is the url sent to the device(s)?
Has anyone applied this within a solution and is willing to share their
andrutto wrote:
Hi,
Few days ago I bought a Sipura SPA-3000 Gateway. Outgoing calls works fine
but incomming calls behave very strange. When I dial my Sipura from outside
and cancel befor picking up, the phone still rings for about one minut. What
is wrong - the Sipura Gateway or I did
I have used iaxComm successfully (http://iaxclient.sourceforge.net/
iaxcomm/).
We worked with the author, Michael Van Donselaar, to enhance some of
the features of this software, particularly the handling of URLs, for
a fee, with the condition that any changes we financially supported
Now I don't feel so inadequate ;)
This is exactly what I am doing. Perhaps there is more to this particular
option.
Here is more information -
I am testing this on * ver. 1.0.7 (I have another box with 1.0.9 and another
one with CVS HEAD). Is 1.0.7 too old? Is this command not applicable to
Need help!
I install the Asterisk and I´m doing some tests with X-lite, I configured 2
extension, and try to call each other, but in both aways the X-lite always
says Call Failed 486 Busy Here,but the extensions are not busy.
How can I fix it?
Thanks,
Ed.
On Wed, 2005-08-31 at 08:33 -0800, Matthew Schumacher wrote:
andrutto wrote:
Hi,
Few days ago I bought a Sipura SPA-3000 Gateway. Outgoing calls works
fine but incomming calls behave very strange. When I dial my Sipura from
outside and cancel befor picking up, the phone still rings
I could not find a .version file at the top level of the tarball. Below is
what my include/asterisk/version.h file contains.
/*
* version.h
* Automatically generated
*/
#define ASTERISK_VERSION
#define ASTERISK_VERSION_NUM 00
It's not outside of reason to think that I have screwed
From the asterisk cli 'show application txfax' and 'show application rxfax'
Craig
- Original Message -
From: Woody Sturges [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, August 31, 2005 11:15 PM
Subject:
Few days ago I bought a Sipura SPA-3000 Gateway. Outgoing calls works
fine but
incomming calls behave very strange. When I dial my Sipura from outside and
cancel befor
picking up, the phone still rings for about one minut. What is wrong - the
Sipura Gateway or I
did something
Are you using the Queue(queue-name,options,URL) syntax to send a URL to
the client? Do you have to configure any options on the iaxComm side for
this to work properly? Or is the URL option interpreted and executed with
the default browser on the PC?
Thanks!
-Original Message-
From:
Jason Walker wrote:
I am testing this on * ver. 1.0.7 (I have another box with 1.0.9 and another
one with CVS HEAD). Is 1.0.7 too old? Is this command not applicable to ver
1.0.7.
That's probably your problem there. I know most newer versions of DIAX
will do this. There is one of the later
On 30/08/2005, at 11:19 AM, Andrew Thrift wrote:
I will also post them onto the wiki.
Regards,
Not yet... :(
My sangoma cards just arrived and i'm waiting for the E1 circuit to
be provisioned, i'd be interested in your findings.
Regards,
Nathan.
Patrick wrote:
So appareantly app_voicemail depends on this adsi stuff. Is there a way
to disable adsi in app_voicemail? I looked through app_voicemail.c but
don't have enough knowledge of C how I would go about it.
Not right now, no. It would be nice if someone could figure out an
B. J. Bomar wrote:
I could not find a .version file at the top level of the tarball. Below is
what my include/asterisk/version.h file contains.
Please re-download the tarball, making a note of the IP address of the
server you get it from. If it still doesn't contain a .version file,
email
I'm experimenting with realtime (CVS HEAD), but using odbc to a
third-party database (progress) instead of mysql.
Following the instructions on voip-info, I created a table for voicemail
called rtvm with the following fields:
CREATE TABLE `rtvm` (
`uniqueid` int(11) NOT NULL auto_increment,
Jason Walker a écrit :
Now I don't feel so inadequate ;)
This is exactly what I am doing. Perhaps there is more to this particular
option.
Here is more information -
I am testing this on * ver. 1.0.7 (I have another box with 1.0.9 and another
one with CVS HEAD). Is 1.0.7 too old? Is this
Hello,
is it possible to use a eicon diva server 2M as FXO ?
Regards
Andreas Moroder
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Is there a specific version of DIAX that I should use? I grabbed the latest
release...Looking at the DIAX site, 910g has the URL feature fixed. Is it
broken again in 915a?
Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Bockman
Sent:
I
was wondering what peoples thoughts are about this. It seem that * works
just as well on Linux 2.6 as 2.4. Maybe a few small issues here and there
but generally itseems to me that * is just as stable on either
platform. 2.4 isthe obvious choice forthe highest possiblility
of astable well
Queue + URL and Dial + URL have been in asterisk for a long time (well
before 1.0) so that is not your problem.
Yes, but I'm pretty sure that Queue URL was broken in one of the
previous releases and not fixed until a few months ago.
Kevin
___
On 30/08/05, Tim Dodge [EMAIL PROTECTED] wrote:
I've tried using DEBUG level logging, and when there's an incoming
call I get the Using history buffer to to extract UK Caller ID
message (chan_zap.c:5172), but then nothing else before all the
extensions.conf logging starts.
I'm guessing
On Wed, 2005-08-31 at 10:42 -0700, canuck15 wrote:
I was wondering what peoples thoughts are about this. It seem that *
works just as well on Linux 2.6 as 2.4. Maybe a few small issues here
and there but generally it seems to me that * is just as stable on
either platform. 2.4 is the
Julian Lyndon-Smith wrote:
After an afternoon of chasing all sorts of dead-ends (permissions etc) I
finally changed the uniqueid from an int to a character field, and it
all updates ok now.
Now, is this a problem with res_odbc, the linux odbc client or the sql
server itself ?
Must be
On Wed, 31 Aug 2005, Andreas Moroder wrote:
Hello,
is it possible to use a eicon diva server 2M as FXO ?
What do you mean? Do you want to connect this ISDN card to your analog line?
Armin
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On Wed, 2005-08-31 at 08:46 -0500, Kevin P. Fleming wrote:
B. J. Bomar wrote:
I am also having the same issue from the ftp tarball.
I've tested the tarball on a bunch of different systems and it worked
properly.
Just downloaded, compiled it on X86_64 and just to really throw a
spanner in
In case anyone is online at this very moment and interested
in voip in the call centre check this out
2pm EST
Cheers,
Dean
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Anyone out there running 1.2beta with a PRI and having CDR problems?
I just upgraded to most recent everything and now my CDR's look like this:
,,9035646130,copper_routing,,Zap/65-1,SIP/netl-a3ac,Dial,SIP/[EMAIL PROTECTED]|60,2005-08-31
13:03:09,2005-08-31 13:03:20, 2005-08-31
I copied my exact queues.conf, agents.conf and sections of the dialplan over
from my 1.0.7 * server to my 1.0.9 * server and the optionalurl is working!
I had to use the DIAX 910g app though (MozPhone worked without an issue on
1.0.9). The 915a would not accept the URL.
Are there any (dare I say)
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