Hi,I am still struggling with the E1 cardDoes anyone has some experience with sangoma E1 card? I have this card in soekris net 4801. First I was runnig it with deactivated DMA and I was receiving overruns (even with no channels in use). Then I enabled the DMA. Now I have the overruns only
Hello,
I've testing soekris with isdn card for few months,
with hight speed hard disk...
the box have not enought power to run asterisk
properly.
their is a problem with irqand ide
controler.
definitively not a good box without à faster cpu
clock...
you can translate 2-3 call only...
the
Hi,
i have change my sip.conf and my extensions.conf but
unfortunately nothing change. Should i not see the
hint priority in the CLI?
richard
--- Steve Davies [EMAIL PROTECTED] wrote:
On 5/12/06, Jerry Jones [EMAIL PROTECTED] wrote:
I believe the hint priority must be in the same
context
Hello Cosmin,
Friday, May 12, 2006, 10:45:05 AM, you wrote:
CP Hello everyone.
CP I've got a HFC ISDN card that I'm using with chan_misdn and it basically
CP behaves like crap. Echo is waaay worst then echo I get TDM400 card,
CP sound is choppy (there other side is allays complaining about
Hi to all,
I've ever post many times some questions about snmp to
monitor asterisk .
I need to be adviced to extend res_snmp in order to
monitor both hardware and softs of asterisk .
I wish to monitor digium cards to get call and line
statistics as well as status and errors (traps).
Which
Hi all,
I want to be able to see the status of my Agents on a web interface. I
have no idea how to do so.
I have found a few sample script to communicate with queues manager to
view queues.But I couldn't find any on viewing the agent status. Could
anybody give me a clue?
Regards,
Pim
I've been running into an issue where chan_agent gets stuck and all queues
stop working. Here's a show channels from when it's stuck:
Channel Location State Application(Data)
SIP/56-be24 [EMAIL PROTECTED]:10 RingDial(Agent/19|50|tw)
Local/[EMAIL PROTECTED]
Pimjai Wesnarat wrote:
I want to be able to see the status of my Agents on a web interface. I
have no idea how to do so.
I have found a few sample script to communicate with queues manager to
view queues.But I couldn't find any on viewing the agent status. Could
anybody give me a clue?
You
I am seeking for the SIP Adapter which is providing the dual FXs ports. I
can get some in the market, did some one experience that using Zyxel P-2002
ATA compatible with Asterisk?
Further more, does Auto-Provisioning ATA useful to work with Asterisk?
Please advice, Good experience ATA is
Hi all
What the lists thinks about to have in Dial application, in S
option some functionality present in L option, that is, play a sound
file to the parties to announce the time is nearly finished, and all
those stuff present in L option??
It could be good, and polite for pre-paid
Hi, in the menu of voicemailmain, appear a
lot of options, there is a way to leave only some of them?
Also I want to know if there is a option that erase
all message in a user box.
Best REgards
Ever Zalazar
___
--Bandwidth and Colocation
Hi!
in the menu of voicemailmain, appear a lot of options, there is a way to
leave only some of them?
A simple solution is to just edit/remove some of the voice prompts that
announce the unwanted options, so the user will not be informed about
their existence.
Also I want to know if there
Harry,
I've ever post many times some questions about snmp to
monitor asterisk .
I need to be adviced to extend res_snmp in order to
monitor both hardware and softs of asterisk .
I wish to monitor digium cards to get call and line
statistics as well as status and errors (traps).
You've been
Rich Adamson schrieb:
Harry,
I've ever post many times some questions about snmp to
monitor asterisk .
I need to be adviced to extend res_snmp in order to
monitor both hardware and softs of asterisk .
I wish to monitor digium cards to get call and line
statistics as well as status and
Klaus Darilion wrote:
Even if I answer n, the Setup script still compiles wanpipe modules.
Thus I guess answering yes is only needed if I want to have kernel
sources synchronized with the installed modules (binaries). If I do not
care about a kernel source tree which includes latest wanpipe
Hi Adibar,
It took me some time to answer because
I was waiting for a positive confirmation from that client of mine.
I have no confirmation, but hey, I concider that as good news.
It appears that Sam and Cyber-telecom did a good job
providing the right support.
That model of GSM-gateway works
On 15 May 2006, at 09:44, [EMAIL PROTECTED] hgaillac-
[EMAIL PROTECTED] wrote:
Hi to all,
I've ever post many times some questions about snmp to
monitor asterisk .
I need to be adviced to extend res_snmp in order to
monitor both hardware and softs of asterisk .
I wish to monitor digium
I am using Mitel 52xx dual mode phones in SIP mode. They
work excellent, I am however having a problem with Voicemail retrieval. The
Mitel Phones have a voicemail button on them. The light lites and clears
correctly but I am not able to retrieve the voicemails using this button. In
the
hidecallerid=yes lets me make the calls from asterisk to the panasonic, but now
I do not have the CID number either.
What is the proper way to configure asterisk to send a callerID number, but NOT
send any name info???
zapata.conf:
context=panasonic
swichtype=national
pridialplan=unknown
Great,
What is the trick to call it announce the Park position during the transfer
instead of a call back??
--
--
Steven
http://www.glimasoutheast.org
Andrew Kohlsmith [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
On Friday 12 May 2006 17:38, Steven wrote:
Does anyone have a
I'm seeing a similar thing...
We have Asterisk 1.2.7.1, Chan CAPI and a home-brewed IVR/Auto Attendant
that
places in-bound calls to role-based riinging groups like sales, support,
admin etc.
which works well, but from a 7960G phone (SIP 7.5) if the person that
answers a
call then transfers
Are you interesting in monitoring asterisk with snmp
before i translate the text in english ?
Harry
--- Michael Labuschke [EMAIL PROTECTED] a écrit :
Rich Adamson schrieb:
Harry,
I've ever post many times some questions about
snmp to
monitor asterisk .
I need to be adviced to
Hi all
I have setup sips accounts to an asterisk server from a
provider, I know that there are using asterisk real time for sip users
definitions.
Sometimes in a ramdom basis I receive:
chan_sip.c:9596 handle_response_register:
Forbidden - wrong password
Hi,
I wish to use asterisk with asterfax to send and receive fax's over PSTN.
Is this possible using a standard 56k modem? I know voice calls are
impracticle because a modem cannot send and receive data at the same
time, but has anybody done this using with fax?
Thanks
Mark
Hi,
Is it possible to echo cancel a voip (sip) channel/trunk in asterisk
somehow? If not, this function would be neat since some providers really
suck at echocancelling when you call out on pstn.
Regards,
Jan
___
--Bandwidth and Colocation provided by
Hello,
Please join us Wednesday May 17th for an informal kickoff to
openly discuss and formulate a plan to resurrect the Ottawa Asterisk User Group
as well as discuss additional topics of interest that impact the VOIP
marketplace and may also be included in our collective strategy.
Hi folks,
It seems that BV has messed it up yet again.
I noted this weekend that any call going in or out had no incoming
audio. All my other SIP providers seem to be OK. Is anyone else having
this problem?
Perhaps it's time to move on. What providers do you recommend that
provide unlimited
On 5/15/06, Mark Phillips [EMAIL PROTECTED] wrote:
Hi folks,
It seems that BV has messed it up yet again.
I noted this weekend that any call going in or out had no incoming
audio. All my other SIP providers seem to be OK. Is anyone else having
this problem?
Perhaps it's time to move on. What
Hi,
I have been written an small script for test the use these commands. I had done
massive test with commands, but I didn´t get success
it. Any of the cases, I don´t listen nothing on channel that call 2100
extension. I dial 2100 extension through an cisco phone 7912 with SIP, also I
dialed
have you tried EXPLICITLY disabling busydetect? It could cause confusion on digital (BRI PRI) lines...If you have busydetect=yes in previuos channel definitions, it will be inherited by your BRI channels also.
Just a thing to try...2006/5/12, stoffell [EMAIL PROTECTED]:
On 5/11/06, Tim Robinson
Hi,
I have a question on
VoIP adapters. As far as I understand, those adapters are usually used to
connect DSL/Cable access to a normal phone (Internet to Adapter, then to PSTN
phones).
I want to know if
you can use those adapters to do the opposite: connect a few lines (1-4 let`s
say)
Mike,Yes, this is absolutely possible. You're just looking for FXO adapters. Off the top of my head, look for Linksys/Sipura, Grandstream Handy Tone, and Aastra devices for this purpose.Alex
On 5/15/06, Mike [EMAIL PROTECTED] wrote:
Hi,
I have a question on
VoIP adapters. As far as I
I have no hands-on experience with this product, but it seems that the Mediatrix 1204 would do the trick for you:http://www.twacomm.com/catalog/model_MEDIATRIX-1204.htm
There are probably lots of other ones as well out there. As the previous poster indicated, you're looking for 4-port FXO adapters
Sorry, didn't mean to direct you to a vendor site - here's the mfg's website:http://www.mediatrix.com/products_devices.php?prodid=13
On 5/15/06, Lachek Butalek [EMAIL PROTECTED] wrote:
I have no hands-on experience with this product, but it seems that the Mediatrix 1204 would do the trick for you:
It didn't work. In fact, I changed:
exten = _XX,2,VoiceMail([EMAIL PROTECTED],u|j)
to
exten = _XX,2,VoiceMail([EMAIL PROTECTED],j)
and now I'm getting
-- Executing VoiceMail(SIP/02-c7df, [EMAIL PROTECTED]|j) in new stack
May 15 10:29:06 WARNING[2859]: app_voicemail.c:2411 leave_voicemail: No
Lacheck,Cheers, I had forgotten about the Mediatrix line. They're pretty popular too.AlexOn 5/15/06, Lachek Butalek
[EMAIL PROTECTED] wrote:I have no hands-on experience with this product, but it seems that the Mediatrix 1204 would do the trick for you:
in the dialplan, before dialing to your legacy pbx, do a:Set(CALLERID(name)=)to blank the CID name.2006/5/15, Steven
[EMAIL PROTECTED]:hidecallerid=yes lets me make the calls from asterisk to the panasonic, but now I do not have the CID number either.
What is the proper way to configure asterisk
On 12 May 2006, at 19:21, Me wrote:
Can someone tell me if passwords are sent in plain text when using
IAX?
I have been told already that SIP automatically encrypts the password?
Anyone know of some good Asterisk security links, docs, articles?
Thanks!
There are 4 options (as configured
Hi Klaus,
Yes, you will need to patch the kernel to run the Sangoma cards in TDM Voice
mode.
David Yat Sin
Sangoma Technologies
(905) 474 1990 x119
(800) 388 2475 x199
MSN: [EMAIL PROTECTED]
Email: [EMAIL PROTECTED]
Wiki: http://sangoma.editme.com
Message: 2
Date: Thu, 11 May 2006 21:22:39
Hi Armin!
I have problems on debian sarge with standard 2.6.8-2-386 kernel.
I've installed the packages:
kernel-headers-2.6.8-2
kernel-headers-2.6.8-2-386
kernel-image-2.6.8-2-386
kernel-kbuild-2.6-3
kernel-source-2.6.8
Then I unpacked the kernel sources into /usr/src/kernel-source-2.6.8
Hello list,
we have prepared a short tutorial that will teach you to turn your
[EMAIL PROTECTED] box into a full-fledged call center within minutes, with
both always-on and callback agents available and the very extensive
reporting facilities that QueueMetrics provides.
You can download
Rodney G. McDuff wrote:
Just out of curiosity does anyone have a guestimate of how many SER
and Asterisk servers FWD uses to provision their +500K customers?
It would be interesting the know the rough spec of the machines as well,
not just the quantity.
Sangoma Techdesk wrote:
Hi Klaus,
Yes, you will need to patch the kernel to run the Sangoma cards in TDM Voice
mode.
That's not 100% correct. You do not have to patch the kernel SOURCES -
this step is optional. If you do not patch the kernel SOURCES, the
wanpipe installer still compiles the
Hi!
I've now tried on another server with a custom 2.6 kernel. It fails with
other errors.
regards
klaus
[EMAIL PROTECTED]:~/asterisk/divas4linux-melware-3.0.e-106.622-1$ make
Searching for configured kernel in /usr/src/linux
Kernel version is 2.6.14
Building divas4linux kernel modules...
On Mon, 15 May 2006, Klaus Darilion wrote:
Hi Armin!
I have problems on debian sarge with standard 2.6.8-2-386 kernel.
I've installed the packages:
kernel-headers-2.6.8-2
kernel-headers-2.6.8-2-386
kernel-image-2.6.8-2-386
kernel-kbuild-2.6-3
kernel-source-2.6.8
Then I unpacked the
Is this a plain 2.5.14 kernel and is module support activated in the kernel
config?
If yes, can you please send me the kernel config (.config) you use?
Armin
PS: the warnings about the capi symbols can be ignored.
On Mon, 15 May 2006, Klaus Darilion wrote:
Hi!
I've now tried on another
Howdy,
How can you tell if RTP traffic has been reinvited/is bypassing an * server?
Sincerely,
Brent A. Torrenga
[EMAIL PROTECTED]
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
+1 219 836 8918 x325 Voice
+1 219 836 1138 Facsimile
www.torrenga.com
Thanks, I will give it a shot tonight.
-- -- Steven
http://www.glimasoutheast.org
"picciuX" [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]...in
the dialplan, before dialing to your legacy pbx, do
a:Set(CALLERID(name)=)to "blank" the CID name.
2006/5/15, Steven [EMAIL
For starters, let's remove some clutter
On Mon, May 15, 2006 at 06:47:54PM +0200, Klaus Darilion wrote:
Hi Armin!
I have problems on debian sarge with standard 2.6.8-2-386 kernel.
I've installed the packages:
kernel-headers-2.6.8-2
kernel-headers-2.6.8-2-386
kernel-image-2.6.8-2-386
Nope, that didn't work.
The idea made sense though.
It must be a PRI thing and any CIDName info, even
null, makes the Legacy PBX stop responding on that channel.
It doesn't hang-up, by it never reports ringing
over the PRI either.
-- -- Steven
http://www.glimasoutheast.org
"Steven"
Thanks Alex and Lachek, I'll look into
those!
Mike
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Sorry if I post in this forum, this may be not the right one, but I hope to find in here some experts which could help me out.I have in one location 8 extensions from a Panasonic PBX KX-TD1232 connected to FXO Ports on an MultiVoIP Gateway from Multitech.On the other location I have 8
Ok, I finally fix it. It seems to be a bug in the app_voicemail.c file:
http://bugs.digium.com/view.php?id=7164
--
Atly.
Alvaro Palma
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, May 15, 2006 8:19 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] GET DATA and STREAM FILE commands,
don´t work
...
Now, below is my script
Hi,
Which fax-modem would you pick if you had to test
fax capabilities ?
For instance, before releasing a new PBX system
offering fax connectivity, you would like to make sure you "comply"
withmost fax machines and protocols.
As you can't afford you buy and maintain tens of
such fax
Greetings,
I am trying to gauge the level of interest in an Asterisk users'
group in Vancouver, BC (or in BC in general). If you would be
interested, please reply off-list.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed:
On Mon, May 15, 2006 20:04, Tzafrir Cohen said:
For starters, let's remove some clutter
On Mon, May 15, 2006 at 06:47:54PM +0200, Klaus Darilion wrote:
Hi Armin!
I have problems on debian sarge with standard 2.6.8-2-386 kernel.
I've installed the packages:
kernel-headers-2.6.8-2
Has anyone had any experience connecting Asterisk to Cbeyond's
SIPconnect service (http://www.sipconnect.info)? Any opinions?
Thanks,
-Brian
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or
On Mon, May 15, 2006 20:04, Tzafrir Cohen said:
For starters, let's remove some clutter
On Mon, May 15, 2006 at 06:47:54PM +0200, Klaus Darilion wrote:
Hi Armin!
I have problems on debian sarge with standard 2.6.8-2-386 kernel.
I've installed the packages:
*PLONK*
[EMAIL PROTECTED] wrote:
Please change the email address
of [EMAIL PROTECTED] to [EMAIL PROTECTED]
Thanks
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
I don't see anything obviously wrong with your configs.
You don't want relaxdtmf. That can cause the problem, not fix it.
POST 2 -- got no response
Hi Eric,
at the begining - Thanks for your help.
relaxdtmf is not written in my config, so it should be at the default, i
guess i remember
This doesn't appear to be asterisk related -- you may want to contact
your other vendors for product support (Panasonic, Multitech or Micronet).
Looks like this is a kludgey setup you could have had cheaper using
asterisk, however.
housi mueller wrote:
Sorry if I post in this forum, this may
We are a Cbeyond partner and have implemented their SIPConnect product. My
main complaint is that they don't let me spoof the outbound caller id yet.
They lock it down to one specific number. So if users with their own DID's
want their number to go out for caller id, you cannot do that at this
On Mon, May 15, 2006 at 09:27:41PM +0200, Armin Schindler wrote:
On Mon, May 15, 2006 20:04, Tzafrir Cohen said:
For starters, let's remove some clutter
On Mon, May 15, 2006 at 06:47:54PM +0200, Klaus Darilion wrote:
Hi Armin!
I have problems on debian sarge with standard
Which fax-modem would you pick if you had to test fax capabilities ?
For instance, before releasing a new PBX system offering fax
connectivity, you would like to make sure you comply with most fax
machines and protocols.
As you can't afford you buy and maintain tens of such fax machines nor
Is any way to install the res_snmp.c module without downloading all the svn
for the trunk?
I already have a production box on 1.2.7, and I realize that res_snmp wasn't
inside the core.
Due to I had recompiled a couple of times due to the inclusion of the
SpanDSP on it, I tried to minimize all
I bet Signate will love this.
Lenz wrote:
Hello list,
we have prepared a short tutorial that will teach you to turn your
[EMAIL PROTECTED] box into a full-fledged call center within minutes, with
both always-on and callback agents available and the very extensive
reporting facilities that
On Mon, 15 May 2006, Tzafrir Cohen wrote:
On Mon, May 15, 2006 at 09:27:41PM +0200, Armin Schindler wrote:
On Mon, May 15, 2006 20:04, Tzafrir Cohen said:
For starters, let's remove some clutter
On Mon, May 15, 2006 at 06:47:54PM +0200, Klaus Darilion wrote:
Hi Armin!
I
I have run into a small snag with SNOM phones, asterisk, and autoanswer.
I direct an extension to autoanswer a SNOM 320 phone. Call is
autoanswered, and call progresses correctly.
I then execute the agi command to transfer to another SNOM 320.
Unfortunately, Asterisk does not clear the
Chris,
I have it working now using Playback but I am looking for a way to reduce the
wait time.
The system can hold up to four numbers per each DID number. When a call comes
in the first message says
Please wait while I attempt to connect you. At any time during this call you
may press 1 to
On 5/15/06, picciuX [EMAIL PROTECTED] wrote:
have you tried EXPLICITLY disabling busydetect? It could cause confusion
on digital (BRI PRI) lines...
If you have busydetect=yes in previuos channel definitions, it will be
inherited by your BRI channels also.
hi, thanks for the tip, but
Why not use background?
On 5/15/06, Tim Sharp [EMAIL PROTECTED] wrote:
Chris,
I have it working now using Playback but I am looking for a way to reduce the
wait time.
The system can hold up to four numbers per each DID number. When a call comes
in the first message says
Please wait while I
Hello,
I´m have a CCM 3.3 and Asterisk in my
LAN.
I need connect my Asterisk in my CCM
3.3.
You can a help me?
Thank´s
Gustavo Souza Queiroz.___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or
On Mon, May 15, 2006 21:37, Tzafrir Cohen said:
The sangoma drivers build without patching the source? That's news to
me. Good to know. If anybody needs my help in making a proper deb out
of them, I'd be glad to. I just don't have the hardware to test with.
Yes, they do. Just answer No when
On Mon, May 15, 2006 21:53, Armin Schindler said:
On Mon, 15 May 2006, Tzafrir Cohen wrote:
...
The sangoma drivers build without patching the source? That's news to
me. Good to know. If anybody needs my help in making a proper deb out
of them, I'd be glad to. I just don't have the hardware to
I do a sip debug on the appropriate channel or IP address and look at the SIP messages. Would be great if there were an easier way though?pFrom: "Brent Torrenga" [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Mon, 15 May 2006 12:52:19 -0500Subject: [Asterisk-Users] How to tell
On Mon, 2006-05-15 at 17:40 -0300, Gustavo Souza Queiroz wrote:
Hello,
I´m have a CCM 3.3 and Asterisk in my LAN.
I need connect my Asterisk in my CCM 3.3.
You can a help me?
I hate to say it, but your best bet is to upgrade to CCm 4.0 and use
SIP.. It is a free cisco upgrade assuming
I am running unixODBC to connect to postgres for your realtime data for
things like call forwarding, dnd and have noticed a significant delay
when running the realtime application. Has anyone else encountered this?
Even from the CLI if I do realtime load cf_data exten 4501 it lags for
Hello
I Installed the
Ceptral voicesand Iam trying tu use the swift module with
asterisk.
But when I start
it show:
[app_swift.so]May 15 17:53:09 WARNING[18876]:
loader.c:325 __load_resource: libswift.so.4: cannot open shared object file:
No such file or
hello,
Is there somebody wit a h323 terminal ?
___
Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos
services préférés : vérifiez vos nouveaux mails,
Does anybody know anyone who offers encrypted IAX termination at
reasonable rates? I googled, searched the WIKI, but didn't find a whole
lot of information.
Thanks,
David
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users
On Mon, 15 May 2006, Klaus Darilion wrote:
On Mon, May 15, 2006 21:53, Armin Schindler said:
On Mon, 15 May 2006, Tzafrir Cohen wrote:
...
The sangoma drivers build without patching the source? That's news to
me. Good to know. If anybody needs my help in making a proper deb out
of them,
I had a similar problem back in the day, except it was backwards.
Asterisk was fast as hell when accessing postgres but everything sucked.
Check your indexes to make sure you aren't indexing stuff you don't need
to index, and that you are indexing what needs to be.
Also, have you tried doing
hello,
I have to test asterisk/gnugk is their somebody, sur
cette putain de liste, with a h323 terminal ?
harry
___
Faites de Yahoo! votre page d'accueil sur le web pour retrouver
On Monday 15 May 2006 07:01, Ed Greenberg wrote:
I've got my res_mysql.conf stating:
[general]
dbhost = 127.0.0.1
dbname = switchref
dbuser = asteriskuser
dbpass = xxx
dbport = 3306
and my extconfig.conf stating:
sipusers = mysql,switchref,sip_buddies
sippeers =
Hello
I Installed the
Ceptral voicesand Iam trying tu use the swift module with
asterisk.
But when I
start it show:
[app_swift.so]May 15 17:53:09 WARNING[18876]:
loader.c:325 __load_resource: libswift.so.4: cannot open shared object file:
Kerry,
We are also a Cbeyond partner focusing mainly on SIPconnect, I thought I
would chime in b/c we don't have that problem setting outbound callerid.
It's true we can't set it to a number not on the customer's account, but
we can set it to any number on the account including DIDs. We do this
[EMAIL PROTECTED] wrote:
hello,
I have to test asterisk/gnugk is their somebody, sur
cette putain de liste, with a h323 terminal ?
No need to be aggressive like that, I don't think it will help your
request. And if you think what you wrote, feel free to unsubscribe.
--
Daniel
If you're trying to check on a particular call, you can do
a 'sip show channel xxx' to display a bunch of info...look for Audio IP which
will tell you where the audio is coming from for a particular
call...
Marty
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philippe
I had the same problem, I emailed Sven, the author and wokred through the following solution:you need to put the path/opt/swift/libinto /etc/ld.so.conf or /etc/ld.so.conf.d/local.conf (depending onyour linux distribution)
and then runldconfigI also reinstalled Cepstral after doing this and it
is their nobody here with a h323 terminal, netmmeting
...
I just need a h323 terminal register with
asterisk/oh323/gnugk just five minutes
just aggressive because of I'm feeling tired
--- Administrator TOOTAI [EMAIL PROTECTED] a écrit :
[EMAIL PROTECTED] wrote:
hello,
I have to
Maybe agent and caller connect in meetme room, and agent uses web
interface to add/remove music-playing extensions to the room?
Moj
nzrh wrote:
Hi,
3-way calling is implemented by the endpoint that
starts the call. I need something that I can make on
server side.
Like all of the agents in a
If you're looking for real-time help, maybe the irc channel would be a
better place?
Peter
On 15/05/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
is their nobody here with a h323 terminal, netmmeting
...
I just need a h323 terminal register with
asterisk/oh323/gnugk just five minutes
just
hi all, i am having problems with my queue. I just found out that my queue is unable to hold simultaneous calls. What i mean is that when a call comes into the queue and it hasent been answered, and a second call come into the queue, the second call is dropped while the first is left ringing. if
So, We want to be able to put a fax machine on the line port of the
X100P in our asterisk server. We however also want to use this card for
911 calling. We need some sort of mechanisim to disable the line out
port on the x100p by software to interrupt a call on the line.
Anyone done
Check your queues.conf for maxlen=1
in the queue config you are having problems with. (Maxlen sets the maximum
length of a queue.)
Wes Baehr
Ability Business Computing, Ltd.
Office/Cell/Fax: 330.882.0455 x25
[EMAIL PROTECTED]
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hello,
is it possible to delete global variables during runtime?
Regards
Bastian
Virus checked by G DATA AntiVirusKit
Version: AVK 16.7382 from 15.05.2006
Virus news: www.antiviruslab.com
___
--Bandwidth and Colocation provided
On 5/15/06, Corey Frang [EMAIL PROTECTED] wrote:
So, We want to be able to put a fax machine on the line port of the
X100P in our asterisk server. We however also want to use this card for
911 calling. We need some sort of mechanisim to disable the line out
port on the x100p by software to
On Tue, May 16, 2006 at 01:09:41AM +0200, Bastian Schern wrote:
Hello,
is it possible to delete global variables during runtime?
Is setting the variable to an empty value good enough? How do you use
it?
-- Tzafrir
___
--Bandwidth and Colocation
There's a been a long standing issue with voicemail volume levels for files
saved in WAV49 format as compared to WAV format. WAV49 is much smaller in
emails and that's great, but it's also less than half the volume level than
the exact same voicemail saved in WAV format. I've seen this
1 - 100 of 133 matches
Mail list logo
