Hi,
I am having difficulty running Avantfax on Debian. When I try to launch the
web UI, I get a whole page of PHP codes. It looks like my apache is not
recognizing the PHP file. However, I am able to run phpmyadmin no problem
which proves that apache2 is working with PHP.
Any idea?
Thanks,
My problem with Avantfax on Debian is resolved. It is just a simple dumb
permission problem. Sorry to bother everyone.
Pete
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
Is there any way to get Asterisk to keep redialing automatically if a number is
busy? How about the case where it is very difficult to get through, due to a
large number of people trying to simultaneously call the same number? In
particular, I am asking these questions for an outgoing SIP
On Tue, Mar 25, 2008 at 11:14:35PM -0700, Pete Kay wrote:
Hi,
I am having difficulty running Avantfax on Debian. When I try to launch the
web UI, I get a whole page of PHP codes. It looks like my apache is not
recognizing the PHP file. However, I am able to run phpmyadmin no problem
Hi all,
can anyone help me. I'm finding the softphone which can trigger web browser
and use callerid to go web page
thanks,
ti
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or
Hello,
use faxdetect=both in zapata.conf and then in extensions.conf you can
use
exten = fax,1,Goto(fax-hardware,s,1)
where fax-hardware is a context having instruction what to do with
fax calls.
if you still didn't get idea, write to me personally and i will send
you a
Hi all,
I want to be able to achieve the incoming fax inside mysql along with the
dailed number and CID. I understand one way to do it is to do a
fax-to-email to a centralized adress and then use procmail to do the db
storing. I am just wondering if there is a more straight forward why of
doing
Anyone who is willing to try out an image please send me a private email.
CS
Von: Christian Stredicke
Gesendet: Sonntag, 23. März 2008 11:56
An: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Betreff: AW: [asterisk-users] BLF and Snom phones
I
Hi,
Anyone know what the settings in SIPDefault.cnf should be for Cisco 7940
phones this year?
Came in today to find they'd all moved one hour ahead (NTP server is
correct and ok). Found the day was set to 26, but on trying to
change the settings to the below, my test phone isn't changing back:
Ah ok,
Those settings do seem to work (test phone was going to a different
tftpd server..)
Anyone know if the Ciscos re-download SIPDefault.cnf periodically, or
only on boot ?
Thanks,
Adrian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian
Marsh
Could you point a link to the DUAL MASTER Replication.
I swear i have been all over the docs and have NOT found this,
thx
Tilghman Lesher wrote:
On Tuesday 25 March 2008 20:43:49 Edgar Guadamuz wrote:
I am trying DB replication with MySQL. I have two nodes, and Linux-HA
running on both of
-Ursprüngliche Nachricht-
Von: Mojo with Horan Company, LLC [mailto:[EMAIL PROTECTED]
Gesendet: Dienstag, 25. März 2008 23:23
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Asterisk parking hold and
transferdigittimeout
It seems that
I am willing to try out an image.
/Benny
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Take it:
http://www.onlamp.com/pub/a/onlamp/2006/04/20/advanced-mysql-replication.html?page=1
Regards,
Ricardo Carvalho.
On Wed, Mar 26, 2008 at 10:45 AM, Al Baker [EMAIL PROTECTED] wrote:
Could you point a link to the DUAL MASTER Replication.
I swear i have been all over the docs and have
Benny Amorsen [EMAIL PROTECTED] writes:
I am willing to try out an image.
Yes this was intended to be private mail. Alas, the Asterisk lists
suffer from broken Reply-To, and I fell for it. Sorry.
/Benny
___
-- Bandwidth and Colocation Provided by
On Wed, 2008-03-26 at 12:30 +0100, Guido Hecken wrote:
Now, what happens:
Call for 9556230 reaches capi-in, is redirected through include statement to
capi-in-sub and executed.
So far so fine, expected behaviour.
Call for 95562315 reaches capi-in and is executed direct, the include
Just a test, please discard
Looks like something is eating my messages on their way :-(
Martin
--
http://mujblog.atlas.cz/___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Hi,
Could you explain for the benefit of the list what you have changed in
the snom image that will benefit this ticket? I am already receiving
your current beta images, through our distributor, up-to about
2008-13-19, and am not aware of any changes that affect BLF behaviour
or short-dials...
One of the major reasons we use DL320 / DL380's is the ease of swapping drives,
and the integrated ILO BIOS level access.We can support remote sites with
ease.
If a drive dies we get a notification, a new one is sent and a non-techie can
replace it with guidance.No onsite visit.
On Wed, Mar 26, 2008 at 10:43:13AM -, Adrian Marsh wrote:
Ah ok,
Those settings do seem to work (test phone was going to a different
tftpd server..)
Anyone know if the Ciscos re-download SIPDefault.cnf periodically, or
only on boot ?
As far as I can see, only on reboot.
You will
This just adds a new drop down called BLF in the function key area. This is a
mix of the already existing Extension (which displays dialog-state
information) and Speed Dial. The LED is controlled by the dialog state like
in the Extension mode, while the key is controlled by the Speed Dial
Hi...
I've been fighting this for a while now, trying clean builds of Asterisk
1.14.18, 1.14.19rc3, and then 1.6 Beta 6 today.
No workee. :-(
Here's the results for various calls made off-hook (push the blue
Speakerphone button on the Polycom 430):
988852700 - Phone waits for me to
On Wednesday 26 March 2008 05:45:19 Al Baker wrote:
Could you point a link to the DUAL MASTER Replication.
I swear i have been all over the docs and have NOT found this,
thx
http://dev.mysql.com/books/hpmysql-excerpts/ch07.html
See Figure 7-3.
--
Tilghman
Thanks for that!! I'd written a script to do remote telnet/reboot
commands, but maybe that's better!
...
Ok tried it... it didn't reboot the phone, but the phone does re-request
:
syncinfo.xml and dialplan.xml
But not SIPDefault.cnf or anything else, and no reboot..
I'm intrigued though.. I
What does your digitmap on your phone look like? This is what
controls sending the call to * when it recognizes a complete dial
pattern. The phone does not send digit by digit. If it is waiting for
you to press send, then it does not recognize your pattern.
On Mar 26, 2008, at 8:18 AM,
On Sun, Mar 23, 2008 at 06:34:01PM -0700, Steve Edwards wrote:
Does large mean voluminous or complex?. AGI's allow you to wrap
complex logic into a single dialplan step. I confess I've never used the
MySQL dialplan interface, but the idea of keeping track of several
(nested) result sets and
On Mon, Mar 24, 2008 at 01:57:53PM -0800, Mojo with Horan Company, LLC wrote:
P.S. This is not typical, right? If I do NOT have write access to a
directory, I can still write to files that already exist in that
directory, as long as I have write access to said files, I think...
Maybe
24 mar 2008 kl. 06.10 skrev [EMAIL PROTECTED]:
maybe the objections to this patch should have been raised *2 years
ago*?
I've objected to the architecture of the patch for a long time, but
not the functionality.
The functionality is a requirement in many PBX installations.
I still think
25 mar 2008 kl. 20.31 skrev Robert Norton - SophTelecom LLC:
Hey List,
I’m trying to work out a resolution for domain based SIP
authentication. We are working on a virtual PBX type product and
want to allow username overlaps on separate domains so that [EMAIL PROTECTED]
and [EMAIL
Hello!
Is it possible play background sounds while talking?
I would like to make an outgoing campaign with the possibility playing
sounds in background by command. But the extra is I would like to
choose which sound to be played. In short operator calls a number,
talking to callee and sometimes
From sip.cfg:
digitmap
dialplan.digitmap=[2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]
x|[2-9]xxxT
dialplan.digitmap.timeOut=3|3|3|3|3|3/
Don't think it's been modified from the original supplied.
...brig
-Original Message-
From: [EMAIL PROTECTED]
On Wed, Mar 26, 2008 at 09:59:04AM -0500, Jason Parker wrote:
Continuing the top-posting madness...
For future reference (and for the archives), you could have done `make
dist-clean` and re-run configure, rather than remove the directory.
Or just delete menuselect.makeopts
I have come to
-Ursprüngliche Nachricht-
Von: Jared Smith [mailto:[EMAIL PROTECTED]
Gesendet: Mittwoch, 26. März 2008 13:01
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Asterisk parking hold and
transferdigittimeout
On Wed, 2008-03-26 at 12:30
Jay R. Ashworth wrote:
On Mon, Mar 24, 2008 at 07:49:31AM +, Gordon Henderson wrote:
On Mon, 24 Mar 2008, mark morreny wrote:
What I need to do is to try to route called based on the dialed number as I
have multiple DIDs on my line. Is this something that can be done? Is this
On Mon, Mar 24, 2008 at 07:49:31AM +, Gordon Henderson wrote:
On Mon, 24 Mar 2008, mark morreny wrote:
What I need to do is to try to route called based on the dialed number as I
have multiple DIDs on my line. Is this something that can be done? Is this
something to do with the
Continuing the top-posting madness...
For future reference (and for the archives), you could have done `make
dist-clean` and re-run configure, rather than remove the directory.
Kyle Gibbons wrote:
All,
Thank you very much for your help, I have solved the problem. After
installing
On Tue, Mar 25, 2008 at 06:31:50PM +0800, mark morreny wrote:
Yes, it is that silly space!!! Thanks for all your help.
The origin of the she-bang, the
#!/usr/bin/whatever
command interpreted by the general loader on most *nixes, is that it
was a magic number, just like the ones on binary
Guido Hecken wrote:
-Ursprüngliche Nachricht-
Von: Mojo with Horan Company, LLC [mailto:[EMAIL PROTECTED]
Gesendet: Dienstag, 25. März 2008 23:23
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Asterisk parking hold and
Thank you very much Kevin for your help and attention.
best regards.
- Mensaje original
De: Kevin P. Fleming [EMAIL PROTECTED]
Para: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Enviado: martes 25 de marzo de 2008, 13:59:13
Asunto: Re:
According to this post:
http://lists.digium.com/pipermail/asterisk-dev/2007-April/027281.html
Includes are tacked on to the end of the dialplan they are mentioned
in, not where they stand.
So, since your exten = _955623XX,1,DoSomethingReallyImpressive()
matches, asterisk doesn't need to
Dear all,
I am working on customizing hylafax's faxrcvd script into PHP. Does anyone
has any sample or guideline that can share with me to give me a quick start?
Two questions I have are: 1. How to simulate the receival of fax without
actually sending one? 2. Where can I find the log that is
On Thu, Mar 27, 2008 at 02:21:45AM +0800, mark morreny wrote:
Dear all,
I am working on customizing hylafax's faxrcvd script into PHP. Does anyone
has any sample or guideline that can share with me to give me a quick start?
Yeah. Ask hylafaxs questions on hylafax mailing lists?
I figure
mark morreny wrote:
Two questions I have are: 1. How to simulate the receival of fax without
actually sending one? 2. Where can I find the log that is echo from
faxrcvd? 3. How to I config Hylafax so that it uses my PHP script
instead of the original .sh script?
Any help will be greatly
Hi!
I have some IVRs made in python.
If the caller hangup before the end of the script I can´t register in my
database the cdr.
Any idea to do this?
Thanks
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
You don't have to build Supermicro stuff yourself if you don't want to.
Most Supermicro dealers do it for you if you buy all the parts from them.
It's true that what your doing with Dell/HP is paying for emotional support.
When it comes to PBX's you not getting any value paying for Dell/HP
Supermicro with hotswap bays and KVM card does the same thing.
From: Darren Wright [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 26, 2008 5:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Had it with Dell Garbage - HP Question
One of the
On Wed, 26 Mar 2008, equis software wrote:
Hi!
I have some IVRs made in python.
If the caller hangup before the end of the script I can?t register in my
database the cdr.
From your description, I'm not sure exactly what you are asking, but 1 of
these should solve your problem.
1) Trap
Is anyone on the list reselling (or just using) EvolutionPBX
from Intuitive Voice Technologies?? If so, please contact me
off list. Thanks.
bill at mwdental.com
Bill
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Thomas Klettke wrote:
On Sat, 2008-03-22 at 12:48 -0400, John Novack wrote:
Assuming you have also checked the obvious possible defects regarding
cords from the XO device to the Digium card, what happens if you reverse
tip and ring?
John,
you were right on the money: I've found
Hi All,
Anyone have any idea what could cause incoming calls on a SIP channel
to no longer be able to use DTMF? DTMF on incoming calls on zaptel and
on local SIP softphones and ATAs all work fine. Nothing gets
registered in the CDR or on the console in verbose level 10, it just
times out. I
Inband only works with the ulaw and alaw codecs.
David Nedved wrote:
Hi All,
Anyone have any idea what could cause incoming calls on a SIP channel
to no longer be able to use DTMF? DTMF on incoming calls on zaptel and
on local SIP softphones and ATAs all work fine. Nothing gets
David Nedved wrote:
Hi All,
Anyone have any idea what could cause incoming calls on a SIP channel
to no longer be able to use DTMF? DTMF on incoming calls on zaptel and
on local SIP softphones and ATAs all work fine. Nothing gets
registered in the CDR or on the console in verbose level
I have a new installation where an Asterisk server is connected to an
Avaya PBX via a PRI E1. We are having a problem that I attribute to
their firewall but I just want to make sure.
When we make a call from the Avaya to a SIP extension there is only
sound on the receiving end.
I ended up solving this by using mailcmd
I changed mailcmd from /usr/bin/sendmail -t
to my own custom script.
Then I parsed out the data I need, do some database lookups, and
customize and send the email based on what I've got.
Now we have unlimited custom email possibilities without changing
I'm getting Got SIP response 406 Not Acceptable back from 10.0.1.2
occasionally when try to dial to SPA942 ,
anyone has any idea on this before i consider Firmware upgrade?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
I thought you weren't going to let anyone know which Asterisk gui you
were using?
PaulH
On Wed, 2008-03-26 at 15:25 -0500, Bill Andersen wrote:
Is anyone on the list reselling (or just using) EvolutionPBX
from Intuitive Voice Technologies?? If so, please contact me
off list. Thanks.
I have found 'make menuselect' useful to find out what is/isn't built
and sometimes a hint as to why.
PaulH
On Tue, 2008-03-25 at 14:37 +1100, Rob Hillis wrote:
Only if you're trying to compile Asterisk 1.2. Asterisk 1.4 also has
the menuselect configuration, though for most applications
Well, the cat is out of the bag now! I'm in the process of
rebuilding my box from scratch after I did their upgrade last
night and everything went to shit... The upgrade fixed 1 problem
and broke many others.
Called in this morning to ask why parking a call doesn't work
any more. Was told, Oh,
Seems a codec problem, check the sip.conf from that spa942
On Wed, Mar 26, 2008 at 11:59 PM, Al lists [EMAIL PROTECTED] wrote:
I'm getting Got SIP response 406 Not Acceptable back from 10.0.1.2
occasionally when try to dial to SPA942 ,
anyone has any idea on this before i consider Firmware
Steve Edwards wrote:
On Tue, 25 Mar 2008, Justin Newman wrote:
Does anyone have use for a broadcast/annouce app?
I wrote SystemAnnounce which will play a specified file to all active
channels (in an UP or bridged state). This was originally to tell users
to get off the system, but
Nope,
Coded is Ulaw on both sides and also this issue happens occasionally with no
change.
On Wed, Mar 26, 2008 at 6:17 PM, Adrià Vidal [EMAIL PROTECTED] wrote:
Seems a codec problem, check the sip.conf from that spa942
On Wed, Mar 26, 2008 at 11:59 PM, Al lists [EMAIL PROTECTED] wrote:
Howdy,
Whats the best way to change the callerid for internal
and external calls.
At the moment using callerid- Fred 04412345
sends callerid as Fred 04412345 for internal calls
when his internal extension is 200.
How can i change the callerid for internal calls but
also keep the specific
hi you,
I'm having problem with voice quality on my trixbox using TDM2400B.The
trixbox is connected via 20 FXO ports on a TDM2400 with the hardware echo
cancel module. Echo cancel almost works, but the users hear what they describe
as a 'background crackle/buzz' coming back when they
I have an old zapata tormenta 2 quad port pci card. I'd like to get it
working and play with it but was curious to see if that was possible.
Does anyone know if it will work for 2.6 kernels, or where I can find
decent drivers? I tried getting the tor driver from
64 matches
Mail list logo
