Hi.
I downloaded pwlib_1.18.1 and openh323_1.15.1 to install Asterisk CVS
HEAD version.
I tried to install asterisk-oh323-0.7.1.
I patched openh323 as typing 'patch -p1
< /usr/src/asterisk-oh323-0.7.1/openh323_1.13.5-make.patch' in openh323
directory.
Then I compiled pwlib, openh323 and installed
We seem to be having the same problem. The cdr command is not found, so we
tried to do a make and install on the add-ons but it can't see to find the
files when we run 'make clean && make && make install'. We have downloaded
from CVS and the files look to be there but it still can't find the files.
Could somebody recommend a good software utility, preferably with a
web front, end for post paid billing in Asterisk?
I've seen a lot of
discussion on the various pre-paid and calling card based solutions,
but nothing that would allow me to configure different regex-based
locations/costs and gen
Hi,
We use Cacti (an MRTG based monitoring tool), and I would also like to
see how you set that up.
Any chance you are willing to share ?
Cheers,
Callum
Florian Overkamp wrote:
Hi,
-Original Message-
I use MRTG to graph Active/Configured SIP channels and
Active/Tot
On Tue, 2005-05-03 at 07:39 -0600, Rich Adamson wrote:
> > > To help identify the source of the delays, I built a new system this
> > > weekend from scratch. When that is complete, I'll use it to compare
> > > the differences in motherboards, OS distro's, and maybe kernel versions.
> >
> > Very go
i ,sorry but i've tried to made my extensions like you but nothing .now i've fxs card and i can recieve calls in my analog card from sip but i can not dial out anaother analog phone .
please help me
Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails !Cré
Hi
this is the macro used for that purpose ..
[macro-dialout-trunk]
exten => s,1,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4);check
for CID override for exten
exten => s,2,SetCallerID(${ECID${CALLERIDNUM}})
exten => s,3,Goto(6)
exten => s,4,GotoIf($[foo${OUTCID_${ARG1}} = foo]?6)
I also had problem faxing with spandsp with my old server (Athlon 700
on a VIA chipset). Now I've instaled asterisk on a P4 2.8Ghz (Asus
P5P800, btw great board, let's you assign the preferred interrupt for
each PCI slot), with 256Mb, and here's what I get (unpatched zttest):
(before I never got to
What's your end device? if it's a voip device (eg SIP phone or a soft
phone) then you shouldn't need a jitter buffer.
Also, you don't need bandwidth=low if you specify the codecs (the
disallow=all will override the bandwidth=low) and maxjitterbuffer is the
param you're after with this line "jit
Hello
all is right, the analog extension should ring, but maybe your dialplan is
not correct or you call a bad extension in you PBX.
can you post your dialplan?, to see it.
regards
- Original Message -
From: "Julio Saura" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, May 03, 2005 2:37 PM
Subjec
Hi all,
I'm using asterisk-1.0.7. I need to configure asterisk in such way
that it authenticates users from mysql DB. Is it possible to
authenticate SIP users from mysql database?
It seems to me that chan_sip2 code from Olle E. Johansson, Edvina.net,
[EMAIL PROTECTED] can authenticate users from m
On 03/05/05, Justin B Newman <[EMAIL PROTECTED]> wrote:
> Neil Cherry wrote:
>
> > What are your recommendations for a slightly fancy home phone?
> >
>
> The Sipura SPA-841 is a nice "compromise" between the Ciscos and the
> Grandstreams.
Check out the new Grandstream GXP-2000. I've been testing
Hi
i posted it this morning
i guess is a [EMAIL PROTECTED] problem... installing a new OS with * from
scratch
it does not even call outside connecting fxo to pots :?
El mié, 04-05-2005 a las 09:55 +0200, Mehdi Chouikh escribió:
> Hello
> all is right, the analog extension should ring, but
On Tue, 2005-05-03 at 13:41 -0500, Matthew Boehm wrote:
> > Personally, I presume you would need to bill your user for that 15
> > seconds, or else you will end up losing money.
>
> You're exactly right. Not only that, but if the call is "Answer'd()" by
> asterisk, the disposition becomes 'ANS
On Tue, 2005-05-03 at 20:05 -0500, Matthew Boehm wrote:
> Holy crap! You mean someone actually "read" my email?
>
> Thanks Andrew. Wish more people would read emails.
Just read it :)
I run from safe_asterisk and have the line
ASTARGS="-n"
in it.
Because I too hate the changing background. W
Hi,
can one summarize the main differences between Asterisk and [EMAIL PROTECTED] or
point me to a location where i can find such a list?
Much thanks,
Kib
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/list
In article <[EMAIL PROTECTED]>, xlab <[EMAIL PROTECTED]> wrote:
> When using phones that are using G.711 codec and the calls are recorded
> with "Monitor", when played back the files sound great.
>
> When we use gsm codec at one or both ends of the call, the recorded
> files sound very bad. Much
Hi
Try the step descibed at this link:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg86875.html
and make attention to edit correctly Makefile.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mail
On Tue, 2005-05-03 at 21:28 +, Tony Mountifield wrote:
> I've done a few more tests and think I may have uncovered a problem in
> the pseudo-driver. Whether it's relevant to Rich's problem I don't know,
> but it might have something to do with MeetMe drift on SIP channels.
>
> I modified Rich
Since you are referring to R2 signaling, it works like this:
The E1 R2 Call Blocking feature provides two ways to block incoming collect
calls-category-based and double answer. With category-based call blocking,
collect calls will be blocked based on a specific category. For example, in
Brazil, co
Hello,
I have 2 *, one is between a Siemens HiPath and the PSTN, having two PRIs
connected to each side.
When I call the Hipath to administer it (with Siemens HiPath Manager), I
usually call through the PSTN and all wents well.
However, I have a second Asterisk and when I call the first Asteris
Kib Eki wrote:
Hi,
can one summarize the main differences between Asterisk and
[EMAIL PROTECTED] or
point me to a location where i can find such a list?
See [EMAIL PROTECTED] site - http://asteriskathome.sf.net - and asterisk
site - www.asterisk.org.
Basically, asterisk is a program, and [EMAIL
for anyone is using RC8b-CVS: there are some major bugs in asterisk chan_sip
and utils. It's convenient to download new asterisk/utils.c and
asterisk/channels/chan_sip.c and reapply the kapejod patches to chan_sip.c
___
Asterisk-Users mailing list
Aster
Hi !
I'm a very happy user of Asterisk for my work since a few weeks now, and
I have almost everything working perfectly. I can get calls from our 3
T0 France Telecom lines, dial all SIP phones and queues internally or
externally, and also dial all national numbers.
My main problem is that I can
> Date: Tue, 03 May 2005 23:14:18 -0600
> From: Joseph <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] NVBackgroundDetect
>
> Is anybody using NVBackgroundDetect to detect fax signal on SIP protocol
> from ATA?
>
> --
> #Joseph
We're using NVBackgroundDetect with SIP and IAX. Several of our
produ
Hi friends !
Can anybody help me that how to use ser with asterisk server so that ser
can work like the front end of the asterisk and all other features of
the asterisk can be used.
I have tried the configuration given in asterisk-wiki/at+large but could
not succeed, still my asterisk in not listen
Hi all,
I'm trying to run a Quicknet LineJack as FXO under Asterisk, but I have
several problems like:
- what kind of drivers should I install? I currently have old ixj drivers.
Do I need to install ztdummy and zaptel to make it work?
- once solved first question, how can I configure this chann
On May 4, 2005 02:54 am, Peter Svensson wrote:
> Unfortunatly Asterisk as a cpe device neither lets the net end allocate
> the B channel, nor does it retry using a different B channel. The problem
> is that Asterisk does not see the whole PRI as a single link with several
> channels, it sees the in
Hi all.
At the end, i get atxfer with sip dowloading head cvs version of
asterisk and this is ok, but now i have errors with h323.
following the instructions i could compile h323 channel and load it, but
when i call from sip to h323 or viceversa, i obtain this.
debug
-
May 4 12:12:
> > Everyone has probably experienced this at some point in the past:
> > You pick up your analog phone. Rather than hearing dialtone, you are
> > connected with someone who has just called you. Neither you nor them
> > heard a ring.
> >
> > Maybe it's just me, but it seems these "freak incidents
> On Tue, 2005-05-03 at 09:48 -0600, Rich Adamson wrote:
> > TDM & X100P card users:
> >
> > Attached is a modified zaptel/zttest.c app called "attest-mod.c". It
> > has been modified to report the "delay" in receiving 8,192 bytes
> > from the TDM card (instead of reporting a percentage). It works
> >>
> >> I just got Mediatrix 1204 from ebay, but it is missing CD that
> conmtain the software and
>
> drivers, I am wondering if
>
> >> anybody knows where I could downloaded from.
> >>
>
>
> >The firmware is not openly available. Mediatrix approach is to "charge"
> >customer
Yes.
Quoting Henry Devito <[EMAIL PROTECTED]>:
> Are you using asterisk @ home?
> - Original Message -
> From: "Patrick M. Gray, Jr." <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Tuesday, May 03, 2005 9:22 PM
> Subject: Re: [Asterisk-Users
> > Everyone has probably experienced this at some point in the past:
> > You pick up your analog phone. Rather than hearing dialtone, you are
> > connected with someone who has just called you. Neither you nor them
> > heard a ring.
>
> I don't think this is a freak incident at all. It still ha
> > It would be very interesting to see everyone's results in running
> > this, and even more interesting to report the results with the OS
> > distro in use, mobo in use (if known), etc. If anyone actually
> > get's a result that is very close to 1.000 seconds, I'd really
> > like to know more abo
Yes I tried the rx and tx values, but no luck there. Then I removed
everything from this line, adsl, fax, etc. and left only asterisk and
still not working. Then I tried the following to get the dialtone and
dial digits myself
exten => _9,1,Dial(${TRUNK}/)
And that didn't work either. I also ad
Version 0.113 - 4. may 2005
* Can now transfer recorded conversations to your PC automatically
* You can configure a folder to hold recordings
* You can now specify that all conversations on an extension should be
recorded
* It's possible to attach a customised string to the recording file name
> > > It would be very interesting to see everyone's results in running
> > > this, and even more interesting to report the results with the OS
> > > distro in use, mobo in use (if known), etc. If anyone actually
> > > get's a result that is very close to 1.000 seconds, I'd really
> > > like to kno
Hi!
I've setup * with a Junghanns.net QuadBRI card agaisnt an Ericsson
MD-110 PBX. All four ISDN channels are setup to simulate EuroISDN
Point-to-Point ("Anlagenanschluss" in Germany) from the Ericsson's side.
Works well and I have had little problems at all.
Now what's happening during a call
On Wed, 4 May 2005, Alex Mack wrote:
> I've setup * with a Junghanns.net QuadBRI card agaisnt an Ericsson
> MD-110 PBX. All four ISDN channels are setup to simulate EuroISDN
> Point-to-Point ("Anlagenanschluss" in Germany) from the Ericsson's side.
> Works well and I have had little problems a
Does anyone know of a way to put a wait or a pause in a .call file?
When my * tries to make an outgoing call on a Zap channel, it does not
wait for a dialtone. It just starts dialing.
Thanks,
-Ronan
signature.asc
Description: This is a digitally signed message part
__
As no-one had actually put any technical details about how things work,
I wrote up a review of the GXP-2000 today.
http://www.gladstonewireless.net/tiki-index.php?page=GXP-2000
--Rob
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of P
Hi Peter!
Thanks for the quick response.
So I'm already doing ECT by using the bristuff'ed version of *?
Alex Mack
Peter Svensson schrieb:
On Wed, 4 May 2005, Alex Mack wrote:
I've setup * with a Junghanns.net QuadBRI card agaisnt an Ericsson
MD-110 PBX. All four ISDN channels are setup to simu
If "cdr mysql status" is 'command not found' then that means you haven't
loaded the module.
Check your module path to make sure it really is there.
(/usr/lib/asterisk/modules/)
If it is indeed there, do "load cdr_addon_mysql.so" from CLI*>
You might want to check modules.conf and make sure you h
Ezekiel Smith wrote:
> Could somebody recommend a good software utility, preferably with a
> web front, end for post paid billing in Asterisk?
>
> I've seen a lot of
> discussion on the various pre-paid and calling card based solutions,
> but nothing that would allow me to configure different regex
Ronan Eckelberry wrote:
Does anyone know of a way to put a wait or a pause in a .call file?
When my * tries to make an outgoing call on a Zap channel, it does not wait for
a dialtone. It just starts dialing.
Thanks,
-Ronan
This seems to be a serious shortcoming in Asterisk.
Can anyone explain
Interestingly enough, your subject says "Radius" yet you didn't say anything
about Radius in your email
MySQL auth on 1.0.7 was removed (I think). It might still be there but it
doesn't support NAT nor MWI.
Just download CVS and use RealTime. We are using yesterdays CVS in a
production enviro
Dave Cotton wrote:
> On Tue, 2005-05-03 at 20:05 -0500, Matthew Boehm wrote:
>> Holy crap! You mean someone actually "read" my email?
>>
>> Thanks Andrew. Wish more people would read emails.
>
> Just read it :)
>
> I run from safe_asterisk and have the line
>
> ASTARGS="-n"
>
> in it.
>
>
> Because
Hi everybody,
Firstly we have to connect our Asterisk system to a
Philips PBX throught QSIG protocol (interfaces S0), but we doesn't find any
documentation about the support of QSIG and S0 interfaces by
Asterisk.
[PSTN/ISDN] <---> Philips <-[QSIG over
S0]-> Asterisk <-[SIP]-> Final users
Put the call file into a folder and have cron copy it to the outgoing
spool after a pause
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronan
Eckelberry
Sent: Wednesday, May 04, 2005 9:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discuss
On Wed, 2005-05-04 at 08:46 -0500, Matthew Boehm wrote:
> Dave Cotton wrote:
> > On Tue, 2005-05-03 at 20:05 -0500, Matthew Boehm wrote:
> >> Holy crap! You mean someone actually "read" my email?
> >>
> >> Thanks Andrew. Wish more people would read emails.
> >
> > Just read it :)
> >
> > I run from
John,
Since you think its a serious shortcoming, either you fix it or you shut
up. To start bitching here and complain that its considered and not
implemented is bullshit. * is a great product, but all great product has
their flaws. Being OSS, you can always modify the code yourself. Otherwise
jus
Hi,
Background:
I'm running 2x * boxes.
Box A has a registered user which dials a number. The connection is sent
to Box B which acts as pstn gateway (sangoma 1xE1 card).
Problem:
On Box A before executing Dial() command I set
SetCallerPres(prohib_no_screened) but despite that Box B sends the
co
> Is there anyway of monitoring which extension is pressed on a IVR, I
> need to use it for voting application.
Look at AGI (or system() if you already have scripts)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/
Dave Cotton wrote:
> On Wed, 2005-05-04 at 08:46 -0500, Matthew Boehm wrote:
>> Dave Cotton wrote:
>>> On Tue, 2005-05-03 at 20:05 -0500, Matthew Boehm wrote:
Holy crap! You mean someone actually "read" my email?
Thanks Andrew. Wish more people would read emails.
>>>
>>> Just read it
Is there a way to have an agent choose whether they want
to press # to accept a call on an individual basis when they log in?
Also, the faq mentions that you can play an optional message
to the agent before they press '#', how is this performed? The
queue message seems to play AFTER they press
or create a file in another dir. Change the time on the file then put
it in the call spool. It should be covered on the WIKI as well. Or you
could write your own app to use the manager api to originate the calls
depending on the needs you have.
___
Asteri
DT wrote:
> Firstly we have to connect our Asterisk system to a Philips PBX
> throught QSIG protocol (interfaces S0), but we doesn't find any
> documentation about the support of QSIG and S0 interfaces by
> Asterisk.
>
> [PSTN/ISDN] <---> Philips <-[QSIG over S0]-> Asterisk <-[SIP]->
> Final u
Add some 'w' before the number, i.e., Zap/g0/ww1812121212
Julian J. M.
On 5/4/05, Ronan Eckelberry <[EMAIL PROTECTED]> wrote:
> Does anyone know of a way to put a wait or a pause in a .call file?
> When my * tries to make an outgoing call on a Zap channel, it does not
> wait for a dialtone. It j
Another social misfit appears.
Or is it a full moon tonight?
Didn't your mother teach you any manners?
John Novack
David Choo wrote:
John,
Since you think its a serious shortcoming, either you fix it or you shut
up. To start bitching here and complain that its considered and not
impleme
As anyone been able to make a TDM04B work in a Mac with Yellow Dog
3.01? (unless I have to use another version of Yellow Dog?)
I tried on a Power Mac 8500, a G3 Beige Desktop, G3 Blue & White and
G4 tower... I can compile zaptel and asterisk witthout any problem.
The card is seen but when I
Pat,
To my knowledge the only way to turn on and off the Call Waiting function
is on-screen with the phone itself. There are quite a few of these
'little' features I wish would be configurable via the config file but
don't seem to be...
Best wishes,
-Corey
> Great info! The only question I w
Jon Gabrielson wrote:
> Is there a way to have an agent choose whether they want
> to press # to accept a call on an individual basis when they log in?
> Also, the faq mentions that you can play an optional message
> to the agent before they press '#', how is this performed? The
> queue message se
Yes, we have to use QSIG directly with Asterisk or througt the Alcatel. So,
Asterisk will be able to handle H.323 to redirect to correct SIP users?
Regards.
- Original Message -
From: "Andreas Sikkema" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sen
Is this with the TDM400P card right?
-Original Message-
From: David Brodbeck [mailto:[EMAIL PROTECTED]
Sent: Monday, May 02, 2005 2:35 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Newer Dell Servers + TDM card
> -Original Message
We have this issue with every new Dell server out, we've tried different
distros even. A poweredge 800 was the last one we tried it on. It just
locks hard, don't get it. We're using the TDM400P (Not T1)..
-Original Message-
From: David John Walsh [mailto:[EMAIL PROTECTED]
Sent: Monday, Ma
Hello,
Thank you for replying !
These files are on the cisco ? or with Asterisk ? Because I don't have
Cisco phone. Is there a way independant of the phone ?
Best regards
Le mardi 03 mai 2005 Ã 19:22 -0500, Ing CIP Alejandro Celi MariÃtegui a
Ãcrit :
> El mar, 03-05-2005 a las 03:43, Deborah M
The best way to get PSTN into a Mac or Windows Asterisk setup is with a
Sipura SPA-3000. You can set it up as a trunk and it works great. I am
actually working on an article on how to configure it right now.
Kerry
http://geekgazette.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto
On Tue, May 03, 2005 at 05:27:33PM +, Tony Mountifield wrote:
> In article <[EMAIL PROTECTED]>,
> Rich Adamson <[EMAIL PROTECTED]> wrote:
> > - a modified zttest.c run on both systems to show the delays in reading
> >8192 bytes from the TDM card as 23,850 microseconds lateness on
> >th
Matthew,
Thank you very much for the help.
We know that the module is not loading because we can't do the make and make
install successfully for the add-ons. It's telling us that it can't find the
files necessary when we do a make(print out listed below). We have renamed
the add-ons dir and downl
Corey S. McFadden wrote:
Pat,
To my knowledge the only way to turn on and off the Call Waiting function
is on-screen with the phone itself. There are quite a few of these
'little' features I wish would be configurable via the config file but
don't seem to be...
# Call Waiting (0-disabled, 1-enable
I've just ordered up a new PC for my home office. I've decided upon a
VIA 500 MHz platform in a fanless/silent case with one PCI slot for my
TDM400 card. Instead of a HD I'm using an IDE <> CF convertor and
AstLinux.
To the user community I pose a question about throughput expectation on
such a p
I just got a cellsocket for my * box...I need help, will give you a channel
on my box in exchange for your time to help me out, if you have experience
to configure this things, please contact me out list...
Manny
Mawise(at)hotmaildotcom
Thanks
___
A
> We have this issue with every new Dell server out, we've
> tried different distros even. A poweredge 800 was the last
> one we tried it on. It just locks hard, don't get it. We're
> using the TDM400P (Not T1)..
For everyone's information, we are successfully using a TDM400P card with a
singl
Can I send an receive call on the same channel (line to the wall)
Thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
James,
jltaylor escreveu:
| Since you are referring to R2 signaling, it works like this:
I'm referring to ISDN PRI channels not R2.
|
| The E1 R2 Call Blocking feature provides two ways to block incoming collect
| calls-category-based and double answer.
On Tue, May 03, 2005 at 09:28:23PM +, Tony Mountifield wrote:
> I wrote:
> > In article <[EMAIL PROTECTED]>,
> > Rich Adamson <[EMAIL PROTECTED]> wrote:
> > >
> > > It would be very interesting to see everyone's results in running
> > > this, and even more interesting to report the results wit
>From your make output it looks like maybe you don't have the mySQL
development package installed on this box? The one with the associated
header files, etc.
On the RPM-based Linux systems I've used (like Redhat, Mandrake, CentOS,
Suse) the package is usually named "[packagname]-devel.version.rpm"
Hey guys,
Lots of nice people on the list using 7960s and using/discovering features
that I didn't think possible. (Re: Multi Line Appearance).
Wanted to know if anyone has gotten the 'CFwdAll' button to properly work.
The problem I am seeing is that if someone presses the button and types in
th
Doh. I didn't read close enough.
> app_addon_sql_mysql.c:31:19: mysql.h: No such file or directory
> cdr_addon_mysql.c:33:19: mysql.h: No such file or directory
> cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory
That means you don't have mysql installed. Or rather, you don't have mysq
I have a HFC-PCI based ISDN card.
How should an extension be constructed, when I want to set up a specific
outgoing number (I have 10 or so MSN numbers)?
For example, when I call 6546 from my SIP phone, I would like to call
"100" with an outgoing number of "555" - how should I do this?
exten =>
I've been trying to get PRI -> Email Fax to work for some time now. Several
months. Got newest everything and still some pages come out missing an inch
or two.
It was recommended to me to change my zaptel.conf so that span #1 used
itself as primary sync source. (It was set to 0).
I made the chang
On Wed, 2005-05-04 at 06:48 -0600, Rich Adamson wrote:
> > On Tue, 2005-05-03 at 09:48 -0600, Rich Adamson wrote:
> > > TDM & X100P card users:
> > I get average numbers very close to 1.024 (especially if I take some
> > rounding error into account).
>
> That's a very good point. Now I'm not sure
It works for me. Do you have reinvites enabled. I do not. That may
explain why * is sending a redirect.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Boehm
Sent: Wednesday, May 04, 2005 11:35 AM
To: asterisk-users@lists.digium.com
Subject: [Ast
The phone's context is "cytel-internal".
This allows us to hit "3XXX" to get someone on the inside.
If you hit "9" at the beginning, you Goto() the "cytel-outgoing" context.
So lets make a call..I'll dial 918005551212 (toll free directory).
The 9 sends it to cytel-outgoing. Call is made. Bridged
exten => 5646,1,SetCallerID(some name <555>)
exten => 5646,2,Dial(Zap/g0/98)
http://voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerID
On Wed, 2005-05-04 at 08:53, Tomasz Chmielewski wrote:
> I have a HFC-PCI based ISDN card.
>
> How should an extension be constructed, when I want
We ran into the same problem. Found out by reading the source that we
had to use joinempty=strict and leavewhenempty=strict to make it work.
Now if I could just get it to pause the agent when someone direct dials
the extension, and then unpause consistently when they hang up.
Anton Krall wrote
Title: Normal
So no one has any ideas about how to get MeetMe
to work with a codec other than ulaw?
Is anyone successfully doing it?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Morin
Sent: Tuesday, May 03, 2005 10:26
PM
To: Asterisk Users Mailing List
Is there an option for a caller to "quit" waiting in the queue and leave
a voicemail?
Thanks!
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
On May 4, 2005 12:05 pm, Matthew Boehm wrote:
> May 4 10:57:04 WARNING[25650]: chan_zap.c:4394 my_zt_write: Write
> returned -1 (Resource temporarily unavailable) on channel 2 - audio may
> have been lost
I think that something in asterisk (not zaptel) changed in the last week to
create this pro
On 5/4/05, Corey S. McFadden <[EMAIL PROTECTED]> wrote:
>
> Pat,
>
> To my knowledge the only way to turn on and off the Call Waiting function
> is on-screen with the phone itself. There are quite a few of these
> 'little' features I wish would be configurable via the config file but
> don't see
LiveVoip
has had numerous calls from some customers/brokers about this announcement. It
is a public announcement. We expect to close this subject to all
normal conditions, in very short order. LiveVoip will remain LiveVoip LLC
operating under RV Wireless as part of a public company, with a
Thanks guys, it's working now. I must have missed the mysql-devel on my last
build
Rick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brent DeShazer
Sent: Wednesday, May 04, 2005 9:04 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercia
Hi David
I was on site with this system and saw some other error something like this:
Avoided deadlock on zap 1-1 chan_lock. maximum retries 10
This came up between the errors:
May 3 17:43:12 pbxct kernel: qozap: dropped audio card 1 cardid 0 bytes 19 z1
71 z2 36
If you call into the system
The phone's context is "cytel-internal".
This allows us to hit "3XXX" to get someone on the inside.
If you hit "9" at the beginning, you Goto() the "cytel-outgoing"
context.
So lets make a call..I'll dial 918005551212 (toll free directory).
The 9 sends it to cytel-outgoing. Call is made. Bridged.
I turned off reinvite and I still get the same behavior.
What is your promiscredir set at?
-Matthew
Alexander Lopez wrote:
> It works for me. Do you have reinvites enabled. I do not. That may
> explain why * is sending a redirect.
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [m
Hi,
I have an Aastra 480i running with Asterisk. I can make local and long
distance calls on it no problem, but if I dial a number where another
phone system is involved and I need to punch in some numbers, this is no
go! I can hit all the numbers on the phone that I want and nothing
happens.
Folks,
This is a firmware bug in the TDMxxx and TExxx cards that Digium has
recently fixed.
I did an "advanced replacement" for mine which involved me buying
another one and them refunding me when they got my old one back.
Get onto their tech support.
Mark
Matt Schulte wrote:
Is this with the T
Andrew Kohlsmith wrote:
On May 4, 2005 12:05 pm, Matthew Boehm wrote:
May 4 10:57:04 WARNING[25650]: chan_zap.c:4394 my_zt_write: Write
returned -1 (Resource temporarily unavailable) on channel 2 - audio may
have been lost
I think that something in asterisk (not zaptel) changed in the last
Look at the app_conference description on the wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20app_conference
I believe it does what you want to do, but I really don't know if it works
with CVS_HEAD or stable releases. I'd be curious to hear how it affects
performance as well.
MATT--
1 - 100 of 231 matches
Mail list logo
