Hi All,
when i make called line1 to line2 can connected and communicated, so on when
line3 called to line1, line1 will receive sound waiting "beep beep" and can
accepted line 3 (on hold line 2) after that i have unhold for switch on line2,
i have sound hangup.
i have logger in file :
Dec 24 14
Hi,
> [app_capiCD.so]Dec 23 19:21:45 WARNING[1076850816]: loader.c:242
> ast_load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined
> symbol: capidebug
> Dec 23 19:21:45 WARNING[1076850816]: loader.c:423 load_modules: Loading
> module app_capiCD.so failed!
in * modules.conf, be sure to
I means that asterisk can be used such as PBX,and streams across PBX.
and if asterisk used by sip proxy,I can use asterisk in internet network.
and could I let enterprise asterisk PBX behind nat network connect each other
by internet asterisk sip proxy ?
On Thu, 2004-12-23 at 21:29 -0800, Erik Espinoza wrote:
> Because it works well and supports a lot of codecs. All the other ones
> I have used were underdocumented and buggy. Firefly works perfectly,
> is very easy/customizable and free. Works great with Asterisk in both
> IAX2 or SIP.
>
> Erik
>
Any idea why this:
Record("IAX2/[EMAIL PROTECTED]/5", "/tmp/whatever.gsm|6|25")
Would result in this:
WARNING[3293201]: app_record.c:117 record_exec: No extension found
Thanks!
--
Start Your Own ISP!
http://www.YourOwnISP.com
___
Asterisk-Users mailing
in the phone's sip.conf entry add
subscribecontext=context_name
and in extensions.conf add a hint (info can be found near the bottom
of the following page)
http://www.voip-info.org/wiki-Asterisk+standard+extensions
On Fri, 24 Dec 2004 10:15:17 +1100, Paul Hales <[EMAIL PROTECTED]> wrote:
> If any
I received my new power supply and swapped it in last night and so far
the IAXy has gone "back to work". If it keeps working reliably, I
would come to assume that the other less frequent problems were due to
the power supply being flaky.
The new PS is a Radio Shack 110-240v to 9VDC @ 1500ma job sp
Can I ask why? This is clearly the easiest/best way to go about it.
On Thu, 23 Dec 2004 16:21:12 -0500, Tony Nichols <[EMAIL PROTECTED]> wrote:
> The wikki has an example that uses a db
>
> ;Login with *801, log out with *802
> exten => *801,1,DBPut(auto/attendant=1)
> exten => *802,1,DBPut
Nabeel Jafferali wrote:
I had the exact same problem, but was in the process of trying to figure it
out myself. I did remove the Asterisk source directory before downloading
the stable version. How do I "remove the modules that are in CVS-HEAD"?
As long as you are only using the stock modules that
Because it works well and supports a lot of codecs. All the other ones
I have used were underdocumented and buggy. Firefly works perfectly,
is very easy/customizable and free. Works great with Asterisk in both
IAX2 or SIP.
Erik
On Fri, 24 Dec 2004 01:32:09 +0100, Bruno Hertz <[EMAIL PROTECTED]>
in zapata.conf change fxsks to fxs_ks
On Thu, 23 Dec 2004 23:48:10 -0500, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> im doing modprobe wct4xxp and then modprobe wcfxo
>
> -jon
>
> - Original Message -
> From: "Lyle Giese" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-
Can you give me an example of how a call would end up in the timeout ext?
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: "Seth Remington" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, December 21, 2004 7:33 A
Anyone know what this error message means?
**
Dec 23 23:12:31 WARNING[3031057]: chan_sip.c:1874 sip_write: Asked to
transmit frame type 2, while native formats is 4 (read/write = 4/4)
**
I see this in my CLI when I call into Asterisk and press * which should hang
up the call sinc
Hi folks,
just a quick update we managed to solve our problem by getting
telstra to turn of the STD pips on outbound calls.
We don't the effect yet on inbound (but we haven't had problems in the
past) as most of our lines with x100p's are
generally only used for outboung.
Gary
.
___
im doing modprobe wct4xxp and then modprobe wcfxo
-jon
- Original Message -
From: "Lyle Giese" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, December 23, 2004 7:59 PM
Subject: Re: [Asterisk-Users] TE410P & X100P Troubles
What modules
I've been contracted by a company in NYC area to install and
congfigure an * system. However I live 70 miles from nyc, they want a
service contract with shomeon local. Are you interested? must show
that s/he know linux and asterisk good enough.
___
Asteri
On December 23, 2004 10:37 pm, James Sizemore wrote:
> Try commenting out
> ;echocancel=yes
> ;echotraining=yes
> I bet your faxs start working in both directions. But of course you will
> now have
> occasional echo problems.
echocancel=no
It's always disabled by * when it hears the fax tones any
I just have to make my view known about this.
1) I agree that one is needed but!
2) I feel that there should be a way to get a self
study course which will lead to a way to take a test for the
Certification.
3) Cost and who set this up is really something
that I think should be done firs
Try commenting out
;echocancel=yes
;echotraining=yes
I bet your faxs start working in both directions. But of course you will
now have
occasional echo problems.
Andrew Kohlsmith wrote:
On December 23, 2004 08:29 pm, Steve Underwood wrote:
This point is interesting. On most systems, if you can
Hi,
Is there a way to restart the DISA to the enter phone number? For
instance, Bell Calling Cards let you hit # at any point which lets you
enter another number to call. This is useful to reduce the number of
digits dialed and to utilize per-minute calls.
I was not able to find anything on the web
Jeff
remove the following directory /usr/lib/asterisk/modules/
On Thu, 2004-12-23 at 21:01, Nabeel Jafferali wrote:
> [EMAIL PROTECTED] wrote:
> > Tim Lewis wrote:
> > > Just upgraded to the current stable ver. when I start asterisk with
> > > -vcg I get the following error
> > >
> > >
> >
[EMAIL PROTECTED] wrote:
> Tim Lewis wrote:
> > Just upgraded to the current stable ver. when I start asterisk with
> > -vcg I get the following error
> >
> >
> > [pbx_loopback.so]Dec 23 19:25:33 WARNING[1633]: loader.c:258
> > ast_load_resource: /usr/lib/asterisk/modules/pbx_loopback.so:
> >
Upgraded to Asterisk CVS-v1-0-12/23/04-18:34:44 and txgian / rxgain
don't seem to work any more. Is this a know problem? I am using two
X100P cards.
-Thanks
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Look at canreinvite= in the sip.conf.
If you ‘remove’ Asterisk from
the stream them you are using Asterisk more like a Proxy and less like a PBX.
If this is the case and you want to support ‘tons’ of users look at
something like SER. Asterisk is not a Sip proxy but rather a PBX and Med
As per Caleer ID spec. Caller ID info is tranmitted between first and
second rings. This is to allow the phone to 'wake up' and receive the
Caller ID information. If you were to pick up the phone right before it
rang the first time or shortly after the first ring stopped you will not
get caller id.
On Fri, 24 Dec 2004 12:17:50 +1000, Gary wrote:
>Hi folks,
>
>this is specifically directed to Australia Asterisk users..
>
>We are having a roblem with x100p 's when dialing STD.
>Upon receipt of the approximately the 5th (out of the ten) PIP's
>asterisk will hang up
>
>Now I am wondering
Hi folks,
this is specifically directed to Australia Asterisk users..
We are having a roblem with x100p 's when dialing STD.
Upon receipt of the approximately the 5th (out of the ten) PIP's
asterisk will hang up
Now I am wondering if others are suffering the same problem ??
Any ideas ??
Hello ,
I learn handbook-draft.but I think I don't understand asterisk.
where I can find some learning book about asterisk?
thank u.
B.R.
John.
-原始邮件-
发件人: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED]
发送时间: 2004年12月24日 7:51
收件人: asterisk-users@lists.digium.com
On December 23, 2004 08:29 pm, Steve Underwood wrote:
> This point is interesting. On most systems, if you cannot here regular
> ticks its pretty certain there are no slips. With the Digium cards, for
> some reason, many people have slips (usually due to configuration issues
> rather than faulty ca
Tim Lewis wrote:
Just upgraded to the current stable ver. when I start asterisk with
-vcg I get the following error
[pbx_loopback.so]Dec 23 19:25:33 WARNING[1633]: loader.c:258
ast_load_resource: /usr/lib/asterisk/modules/pbx_loopback.so: undefined
symbol: pbx_substitute_variables_varshead
Dec
Michael Welter wrote:
I posted last week that taking timing from either the T100P or the
Adtran TA750 had no effect--that my fax transmissions crashed either
way. It turns out that the T100P card is bad and delivers random
slips during a fax transmission--sometimes after several pages
printed.
Just upgraded to the current stable ver. when I start asterisk with
-vcg I get the following error
[pbx_loopback.so]Dec 23 19:25:33 WARNING[1633]: loader.c:258
ast_load_resource: /usr/lib/asterisk/modules/pbx_loopback.so: undefined
symbol: pbx_substitute_variables_varshead
Dec 23 19:25:33 WAR
What modules are you loading and in what order?
Lyle
- Original Message -
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, December 23, 2004 3:25 PM
Subject: [Asterisk-Users] TE410P & X100P Troubles
> All,
>
> I've got an asterisk
On Thu, 2004-12-23 at 15:50 -0800, Erik Espinoza wrote:
> I'd recommend Firefly by Virbiage. It's free and works on third party
> networks with sip/iax2 support.
Any specifics as to the why? I browsed their site and it made a good
impression, but I didn't try it yet. The phone itself though seems
I'd recommend Firefly by Virbiage. It's free and works on third party
networks with sip/iax2 support.
On Fri, 24 Dec 2004 00:27:28 +0100, Bruno Hertz <[EMAIL PROTECTED]> wrote:
> On Thu, 2004-12-23 at 16:36 -0600, Michael Van Donselaar wrote:
>
> > iaxComm is Open Source, and currently runs on W
Did you do a "make config" in the zaptel source directory?
THat works for me.
Jerry
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At 03:43 PM 12/23/04, you wrote:
Oh, I see. This is the realtime connected problem.
Can't say too much constructive about that without info, I'm not a fan of it.
We need a debug trace of the registration process (SIP trace and *
messages) to debug why it failed,
not just a one-line message, and a
On Thu, 2004-12-23 at 16:36 -0600, Michael Van Donselaar wrote:
> iaxComm is Open Source, and currently runs on Win32 and i386Linux platforms.
> Earlier versions run on Mac OSX, but I don't have hardware to compile it, and
> have not had any recent reports.
Thanks Michael
I've tried it and it se
I can't get dial-out working. I'm trying to call 523936. Is there
something wrong with my setup here? Could someone please give me a few
pointers?
[fwd-out]
exten => _8.,1,SetCallerID(${FWDUSERID})
exten => _8.,2,SetCIDName(${FWDUSERNAME})
exten => _8.,3,Dial(SIP/[EMAIL PROTECTED],70)
I found ou
> > > Are there any common silent failure modes for email
> > > notification from the Voicemail module. I put the
> > > email and pager email addresses in my entry in
> > > voicemail.conf but no mail gets sent when I leave
> > > a voicemail. No obvious error messages either,
> > > unless I'm just
On Fri, 2004-12-24 at 04:52, Matthew Boehm wrote:
> Whoever was listed first in the list always got the call first. This isn't
> what I was expecting RR to do. I was expecting call #1 to goto agent 1. if
> call 2 comes in and 1 is still on phone it goes to 2. if 1 is not on phone
> it still goes to
If anyone has a good guide to the buddy function, I would also love to read
it!
Regards,
PaulH
-Original Message-
From: Nihal [mailto:[EMAIL PROTECTED]
Sent: Friday, 24 December 2004 6:11 AM
To: Asterisk Users Mailing List
Subject: [Asterisk-Users] Polycom Buddies
I've got two Polyco
At 01:21 PM 12/23/04, you wrote:
Greg - Cirelle Enterprises wrote:
Read it, makes no difference, it's broken :)
Also, it doesn't say why the table structure is the
way it is. just poor data modeling.
God, I'm sure everyone on the list must be thinking, "Oh, why oh why
didn't *Greg* write Asterisk
> In wiki pages it is stated that The audio channels (RTP) may go directly
> from phone to phone or may go through Asterisk's media bridge.
>
> Currently with my settings, I notice that all rtps are passing through
> my asterisk. How could I achieve that they go directly from phone to
> phone?
> > >I've got a configuration with PSTN line connected to FXO
> > >on TDM400P ringing through to a phone connected on a
> > >Sipura SPA-3000. The phone *does* ring before the
> > >caller-id is available. In fact, it shoes some
> > >alternate message like "waiting for caller id info"
> > >right
On Thu, Dec 23, 2004 at 04:51:34PM -0600, Rich Adamson wrote:
> > Are there any common silent failure modes for email
> > notification from the Voicemail module. I put the
> > email and pager email addresses in my entry in
> > voicemail.conf but no mail gets sent when I leave
> > a voicemail. No
Hi,
I can't get dial-out working. I'm trying to call 523936. Is there
something wrong with my setup here? Could someone please give me a few
pointers?
Regards,
Norman Zhang
[fwd-out]
exten => _8.,1,SetCallerID(${FWDUSERID})
exten => _8.,2,SetCIDName(${FWDUSERNAME})
exten => _8.,3,Dial(SIP/[EMAIL
> Are there any common silent failure modes for email
> notification from the Voicemail module. I put the
> email and pager email addresses in my entry in
> voicemail.conf but no mail gets sent when I leave
> a voicemail. No obvious error messages either,
> unless I'm just not looking in the righ
Hi !
Can somebody tell me how to turn the "*" Hangup option utrned off in
queues. I have not used any H option but still as an agent if I press "*"
key the user gets disconnected. Somehow it is turned on by
default. Can I turn this option off In my extensions.conf I have
written :
exte
canreinvite=yes
Aterisk stays in the signaling path so unless you're running tcpdump or the
like you'll never notice this.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of bijan
> Sent: Thursday, December 23, 2004 4:46 PM
> To: a
In wiki pages it is stated that The audio channels (RTP) may go directly from phone to phone or may
go through Asterisk's media bridge.
Currently with my settings, I notice that all rtp’s are passing through my asterisk. How could I
achieve that they go directly from phone to phone? I ass
Title: Asterisk queue_log
Dear All,
I'm running asterisk 0.7.1 on a production call center and second call center on Asterisk CVS-HEAD-05/18/04-01:57:42.
On the Asterisk CVS-HEAD-05/18/04-01:57:42 box I have a queue_log file that I can use to get statistics.
I get no such file in 0.7.1.
On Thu, 23 Dec 2004 22:43:43 +0100, Bruno Hertz <[EMAIL PROTECTED]> wrote:
>
>After having been toying around with asterisk and various VoIP stuff
>for a couple of weeks now, I want to recommend a preferred protocol
>and softphone to friends and family for calling me up.
>
>As SIP and H323 are suc
William Betts wrote:
found the answer in causes.h, but i'd really like to know what this means
Dec 23 17:11:43 WARNING[4845]: channel.c:1921 ast_request: No channel
type regis tered for 'IAX'
IAX2 is the default IAX protocol now.
___
Asterisk-Users maili
Are there any common silent failure modes for email
notification from the Voicemail module. I put the
email and pager email addresses in my entry in
voicemail.conf but no mail gets sent when I leave
a voicemail. No obvious error messages either,
unless I'm just not looking in the right place.
T
On Thu, Dec 23, 2004 at 04:25:55PM -0600, Christopher L. Wade wrote:
> Dorn Hetzel wrote:
> >I've got a configuration with PSTN line connected to FXO
> >on TDM400P ringing through to a phone connected on a
> >Sipura SPA-3000. The phone *does* ring before the
> >caller-id is available. In fact, it
Dorn Hetzel wrote:
I've got a configuration with PSTN line connected to FXO
on TDM400P ringing through to a phone connected on a
Sipura SPA-3000. The phone *does* ring before the
caller-id is available. In fact, it shoes some
alternate message like "waiting for caller id info"
right after the fi
Try IAX2 not IAX
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of William Betts
> Sent: Thursday, December 23, 2004 4:13 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Re: IAX cause codes
>
> found the answe
I've got a configuration with PSTN line connected to FXO
on TDM400P ringing through to a phone connected on a
Sipura SPA-3000. The phone *does* ring before the
caller-id is available. In fact, it shoes some
alternate message like "waiting for caller id info"
right after the first ring and then
found the answer in causes.h, but i'd really like to know what this means
Dec 23 17:11:43 WARNING[4845]: channel.c:1921 ast_request: No channel
type regis tered for 'IAX'
On Thu, 23 Dec 2004 14:06:00 -0600, William Betts
<[EMAIL PROTECTED]> wrote:
> Where can I find a list of IAX cause codes
Andrei (MPI) wrote:
Hi Jared,
Thank you for your reply. That server is for asterisk only, things like
X-windows and samba were not even installed there. I limited what I
could from system point of view. Digium support has qualified the box as
"clean" for TDM400P operations.
It is not clicks and
The WiKi can give you step by step instructions, but I have had only
failure with TDMoE.
--
Christopher Dobbs
FCG ZHAO Zigang wrote:
>who can tell me how to do TDM over enthernet ?
>
>pc a connect pc b only use TDM card?
>
>thank you
>
>John.
>
>
>
Thanks for the reply. However that did not do the trick.
Both hosts have fixed, static IP addresses, so I am not using register
statements, just IAX2 dial commands.
Any other ideas?
thanks
Mike
On Thu, 23 Dec 2004 14:29:33 -0500, Gene Willingham
<[EMAIL PROTECTED]> wrote:
>
> I received this
After having been toying around with asterisk and various VoIP stuff
for a couple of weeks now, I want to recommend a preferred protocol
and softphone to friends and family for calling me up.
As SIP and H323 are such a mess to set up in NATed environments, the
only reasonable protocol option righ
All,
I've got an asterisk box thats been running a TE410P without any problems.
I recently added an X100P for our back office line, and now asterisk wont
start. Any help is greatly appreciated.
Zaptel.conf
span=1,1,0,esf,b8zs
span=2,2,0,esf,b8zs
span=3,3,0,esf,b8zs
span=4,4,0,esf,b8zs
bchan=1-2
I just tried to order the CON-SNT-CP7960 part
from CDW… This is the ~$8 1yr support contract that’s supposed to
give access to the Cisco download site for firmware for the 7960 …
we, I got a call from CDW saying that Cisco wouldn’t authorize them
to sell that product to me. The sales r
The wikki has an example that uses a db
;Login with *801, log out with *802
exten => *801,1,DBPut(auto/attendant=1)
exten => *802,1,DBPut(auto/attendant=0)
;Incoming calls- check if autoattendant is logged in, otherwise goto "auto"
exten => s,1,DBGet(autoattendant=auto/attendant)
exten
Just out of interest, why is there no zaptel script included in the
tarballs of 1.0.3?
I used to use the RPMS but they haven't been updated for some time but now
I'm missing the zaptel init.d script.
Or should I have the modules loaded another way?
Cheers!
Remco
> > > Should I set the channel bank to provide timing or receive timing?
> >
> > Proper symantics: timing is "always" encoded into a T1/E1 transmit
> > signal. You can't disable it.
> >
> > "Clock Sync" on the other hand is a setable option. If the 750 is
> providing
> > fxs interfaces to phones, t
GoTo(6275,1) or (6275|1) are the same, and that would be
goto(extension,priority) in the same context as you've said.
Regards,
Jim
On Thu, 23 Dec 2004, Dorn Hetzel wrote:
>
> I understand some of the basic Goto() forms,
> such as Goto(context,extension,priority) and
> Goto(extension,priority
> I agree -- I must admit I added the IAXy to "pad out" my argument -- my
main
> beef is with the TDM4XXP card/driver.
Nonsense the IAXy not only has some driver / hardware issues but the feature
set make it unuseable in profession corporate enviroments
no echo can
no cpu for std codec like g723/g7
Andrei (MPI) wrote:
richard wrote:
Asterisk also ring (so that all three of them are ringing), and then
someone can then choose which phone they want to answer?
Hi Richard,
Absolutely, you can do that with Asterisk. Though VoiP telephone
(Asterisk) may start ringing a second later than analog ph
Oh, I see. This is the realtime connected problem.
Can't say too much constructive about that without info, I'm not a fan
of it.
We need a debug trace of the registration process (SIP trace and *
messages) to debug why it failed,
not just a one-line message, and anything after that is useless,
richard wrote:
Hi,
I have the following scenario:
We currently have 1 incoming line, that 2 POT phones plug into, and
when we have an incoming call, both phones ring. Is it possible to
have Asterisks in parallel, so that when the 2 POT phones ring, I can
have a Voip phone, which "is" plugged in
Andrew Kohlsmith wrote:
On December 23, 2004 02:31 pm, Steven Critchfield wrote:
While I agree, I also must point out
- TDM400P - this card seems to be the #1 source of problems. I believe
the FXO module issues are solved but the FXS issues are still around.
Hopefully the same fix works.
Hi,
I have the following scenario:
We currently have 1 incoming line, that 2 POT phones plug into, and when
we have an incoming call, both phones ring. Is it possible to have
Asterisks in parallel, so that when the 2 POT phones ring, I can have a
Voip phone, which "is" plugged in and configured
On December 23, 2004 02:31 pm, Steven Critchfield wrote:
> > While I agree, I also must point out
> > - TDM400P - this card seems to be the #1 source of problems. I believe
> > the FXO module issues are solved but the FXS issues are still around.
> > Hopefully the same fix works.
> I think this
On Thu, 2004-12-23 at 19:28 +0100, Aldo Bergamini wrote:
> [app_capiCD.so]Dec 23 19:21:45 WARNING[1076850816]: loader.c:242
> ast_load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined
> symbol: capidebug
> Dec 23 19:21:45 WARNING[1076850816]: loader.c:423 load_modules: Loading
> module
Where can I find a list of IAX cause codes and what they mean? I'm
looking for Cause code 66.
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Jon,
Yes, I have tried that. The problem does moves with the particular phone.
Andrei
Jon Radon wrote:
Try swapping the working phone and the non working phone. See if the
problem moves with the phone.
On Thu, 23 Dec 2004 10:52:27 -0500, Andrei (MPI)
<[EMAIL PROTECTED]> wrote:
Hi Jared,
Thank y
ok thanks
- Original Message -
From: "listas iPfone" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, December 23, 2004 4:03 PM
Subject: Re: [Asterisk-Users] Linksys PAP2-NA Config
> Hi!
>
> Use the spa2000 configuration info, the softwar
A lot of this has to do with the stupid linux kernel changing HDLC api time
after time after time.. I think at least 4 times thus far. I think if you
use 2.4.18 kernel you'll be able to get it to work.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PR
On Thu, Dec 23, 2004 at 01:26:38PM -0600, Steven Critchfield wrote:
>
> I seem to remember that mysql made a stink recently about the dual
> licensing of asterisk. That was the cause of the mysql code getting
> pulled and placed in the add-ons sections so the core didn't have any
> licensing issue
On Thu, 2004-12-23 at 13:53 -0500, Brent Franks wrote:
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of TC
> Sent: Thursday, December 23, 2004 1:29 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] RedAlar
> # Next come the dynamic span definitions, in the form:
> # dynamic=,,,
> #
> # Where is the name of the driver (e.g. eth), is the
> # driver specific address (like a MAC for eth), is the number
> # of channels, and is a timing priority, like for a normal span.
> # use "0" to not use this as a
I wrote:
> I wonder if anyone has come across this odd behavour with a T1 PRI using
> NI2 signalling from a Nortel switch.
>
> Sometimes, when bringing up a PRI trunk, a channel gets into a state
> where asterisk can't request a channel, and gets reason 0, but the
> channel is not busy. The only t
On Thu, 2004-12-23 at 13:26 -0500, Andrew Kohlsmith wrote:
> On December 23, 2004 12:58 pm, Steven Critchfield wrote:
> > Maybe some of the users just aren't up to the task of getting the job
> > done. My experience thus far has been very good. All the hardware with
> > the exception of one which w
I received this before, and it is because you are using the wrong context in
the iax.conf.
For example:
The context must match the username in the register statement.
Iax.conf...
register => username:[EMAIL PROTECTED]
[username]
type=friend
context=iax-in
user=username
secret=secret
auth=plai
On Thu, 2004-12-23 at 13:58 -0500, Dorn Hetzel wrote:
> On Thu, Dec 23, 2004 at 12:05:21PM -0600, Steven Critchfield wrote:
> >
> > Close on the complete reason. There is also a licensing conflict with
> > Dialogic drivers and GPL software. You have to get a commercial license
> > for asterisk to
After I leave a voicemail for an extension and hangup,
my asterisk console (with debug turned up quite high)
shows two error messages like:
WARNING[7664]: app_queue.c:341 changethread: Can't change device with no
technology!
WARNING[7668]: app_queue.c:341 changethread: Can't change device with n
Damon Estep wrote:
Has anyone had success with the TDM400 in production? I have multiple
boxes where these cards lock up and the only thing that will fix them is
to unload *, modprobe -r wctdm, modprobe wctdm, load asterisk. Does not
matter if it is a FXS/FXO module.
I know this topic has been disc
No, you just need a phone that allows you to accept more than one call.
Otherwise, you need to set up your dial plan so that the unanswered call
goes to voice mail
or elsewhere perhaps.
Norman Zhang wrote:
Hi,
Currently * is registered to 1 FWD #. If that line is busy people
can't call in? Do I
I've got two Polycom 500's that I'm playing with, and I want view the status of
either phone, (busy/on the phone/etc.) from the other.
I've got this cute little 'Buddies' button, and I can add contacts to that.
But the status doesnt actually update.
Do I need to setup realtime for asterisk? C
This is still a nasty design flaw (bug) in Asterisk.
IAX is similarly bugged.
I can only ask you to wait a little bit longer until I post the solution.
Ian Chilton wrote:
Hi,
I have a few accounts with sipgate.co.uk to get some different DiD
numbers. However, when an incoming call comes in, it see
I have a problem where an Asterisk server is sending a premature DRQ... Not
sure why..
Here's the setup - Asterisk using inAccess networks H323 replacement channel
driver
Connecting to a Lucent iMerge...
The call connects fine - I get the out of the box greeting - but after
exactly one
Minute - the
Hi!
Use the spa2000 configuration info, the software is the same.
Miklos
- Original Message -
From: "Listas" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, December 23, 2004 4:46 PM
Subject: Re: [Asterisk-Users] Linksys PAP2-NA Config
O
I posted last week that taking timing from either the T100P or the
Adtran TA750 had no effect--that my fax transmissions crashed either
way. It turns out that the T100P card is bad and delivers random slips
during a fax transmission--sometimes after several pages printed.
Replacing the card so
On Thu, Dec 23, 2004 at 12:05:21PM -0600, Steven Critchfield wrote:
>
> Close on the complete reason. There is also a licensing conflict with
> Dialogic drivers and GPL software. You have to get a commercial license
> for asterisk to clear the licensing issue. Beware that I think as soon
> as you
Try swapping the working phone and the non working phone. See if the
problem moves with the phone.
On Thu, 23 Dec 2004 10:52:27 -0500, Andrei (MPI)
<[EMAIL PROTECTED]> wrote:
> Hi Jared,
>
> Thank you for your reply. That server is for asterisk only, things like
> X-windows and samba were not e
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of TC
Sent: Thursday, December 23, 2004 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RedAlarm (t100p - Adtran Total Access 750)
> Why would you say channel ba
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