Hello * users
I've problems with sound quality on zaphfc
Asterisk works fine good sound quality.
If I do "make load" in the bristuf.xx zaphfc dir then sound quality
drops directly.
Even if I don't load the chan_zap in the modules.conf
I use this config on more (even old 400Mhz machines) and work
Anybody have any issues running tdmoe on kernel 2.6+?
I've got Suse 9.1 + 9.2 running 2.6.5 and 2.6.8 respectively, and when I
enable dynamic spans between them, both boxes dump something similar to:
Badness in local_bh_enable at kernel/softirq.c:141
[] local_bh_enable+0x48/0x60
[] dev_queue_x
You need to set the asterisk as a gateway
in the polycom.. then to dial out. Lets say you set the * as GW 0 on the
polycom you would dial 0*{exten} in order to dial through a gw on the ip3000
you have to use the prefix for the gateway. So 0* for GW 0 and 1* for GW 1
Hope this helps if
>Need help with how to configure for parked calls in the Flash Operator
>Panel's op_buttons.cfg file ...
>
>I've looked on the wiki, google and asternic's site and can't seem to find
>how to setup op_buttons.cfg to monitor parked calls.
>
>For example, if someone parks in 701, I'd like to see that
Trevor,
I used to sell Nuance when I worked for Fujitsu. When I first came
across Angel.com about 12 months ago I knew this was the right way to
approach NLVR.
The Nuance costs are unrealistically astronomical mainly due to the
'high touch' consulting fees that are imposed on this kind of rollout.
dean collins wrote on Saturday, 12 February 2005 9:16 PM:
> Check out www.angel.com
For that matter, check out Tellme. 1-800-555-TELL
Speaker-independent automatic speech recognition, when implemented properly,
is VERY good right now. However, good ASR is usually fairly expensive. Do
not c
I've seen the same behavior on my TDM400P. I solved it by simply scripting
a stop/start of the zaptel drivers and asterisk in the middle of the night
each night.
Of course, that might not be practical in a more seriously production
environment
Paul
- Original Message -
From: "Mi
>asterisk-users-bounces at lists.digium.com wrote:
>> Hi everyone,
>>
>> I was working yesterday and after I provide my IAXy box it
>> loose any network comunication, the link light (green) is on
>> and the activity light (orange) when the power is turned on
>> it does nothing, but when I pickup t
I am trying to figure out call parking. It is my understanding that it
is built into *. I have edited the features.conf like I want it but am
unsure where to add the include statement. Right now if I am on a call
from the FXO bridged to the FXS port and I hit the # key, nothing
happens.
I have tri
Disagree with you Matt.
Check out www.angel.com
If anyone wants some contacts over there email me. I'm sure they would
be happy to set up on API for utilizing their services in conjunction
with asterisk.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECT
Robert Rozman wrote:
Hi,
I noticed that agents logins (agentcallbacklogin) are reset if Asterisk is
restarted. Can this be avoided in some way ?
As Kevin insinuated, there is support for this in CVS Head. It's called
persistentagents and is set through agents.conf:
; Define whether callbacklo
[EMAIL PROTECTED] wrote:
> Hi everyone,
>
> I was working yesterday and after I provide my IAXy box it
> loose any network comunication, the link light (green) is on
> and the activity light (orange) when the power is turned on
> it does nothing, but when I pickup the phone connected to the
> box,
Matthew Boehm wrote:
I didn't write app_addon_sql_mysql.c so I have no clue as to its current
state.
-Matthew
Matthew,
I know that. You were just the last person to reply to that thread,
and I always include the person's name as though I am writing a message
to them personally. It's just habit
I honestly can't understand what all the confusion is about.
There are two versions of Asterisk, CVS-Head and Stable. Head has no
version numbers, it seems to be delineated by date.
If you download cvs v1-0 then you will always get the current release of
stable whether it be 1.0.5, 1.0.6. or wha
Steve Underwood wrote:
This seems to be a problem with the current wctdm driver. It seems to be
broken for audio going out. I used to be able to send faxes reliably
using spandsp and a TDM40P card, but I no longer can. I haven't had time
to look in detail at what is wrong.
And I think the CPU "s
Agreed, Steve. Iq, Maybe it is for your voice, but speech to text is a
long ways away from being as advanced as you think it is. Check out dragon
speek, and see what it takes to train a voice...
-m
On Sun, 13 Feb 2005, Steve Underwood wrote:
Iqbal wrote:
Hi
I dont know jack about speech recognit
Yeh this could be explained a little better.
If you log into the concole and type the command "help-aah"
It will bring up all of the commands available to change passwords for
the various user names (AMP, maintenance etc)
When you change the passwords it will give you the user name.
Because [EM
Thanks
I'll look into it, but from the little that I read on RealTime, I was
under the impression that it did not use MySQL or PostgreSQL which is a
database feature that I was hoping to use.
--Lonnie
> Why not just use the built-in database features to do what you want? Its
> called RealTi
I didn't write app_addon_sql_mysql.c so I have no clue as to its current
state.
-Matthew
- Original Message -
From: "Kristian Kielhofner" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, February 12, 2005 7:46 PM
Subject: Re: [Asterisk-Us
Why not just use the built-in database features to do what you want? Its
called RealTime. Lots of info on it on the wiki.
-Matthew
- Original Message -
From: <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Saturday, February 12, 2005 4:56 PM
Sub
Hi everyone,
I was working yesterday and after I provide my IAXy box it loose any
network comunication, the link light (green) is on and the activity
light (orange) when the power is turned on it does nothing, but when I
pickup the phone connected to the box, this light start blinking once
per sec
Hi all!
I'm newie to asterisk and I've been trying to make it work in order to
use it with Linux softphones (H.323, SIP or IAX, I don't mind) and none
hardware phone.
I'm using asterisk packages from Debian SID (my distribution), asterisk,
asterisk-config, asterisk-sounds, asterisk-h323. I've sti
Hi,
I changed the dialplan and the same error. By the way the * server has
public IP address and the firefly clients are behind firewall(iptables).
here is the error and config
chan_iax2.c:5718 socket_read: Rejected
connect attempt from 33.44.5.55, request '[EMAIL PROTECTED]' does not exist
cha
I use Fedora core 2, and openssl-0.9.7, expat-1.95.7 is installed by rpm packages.
I downloaded pwlib-1.5.2 and openh323-1.12.2 at my home directory(/root/root_src), asterisk 1.0.4 at directory /usr/src/ and have installed successfully.
Asterisk is executed normally, but module chan_h323.so c
Hello All,
Has anyone been able to patch the latest CVS Asterisk with the ast_data from:
http://svn.asteriskdocs.org/res_data/ast_data/
I am having troubles getting a patch version together.
Any help would be greatly appreciated.
Thanks,
Lonnie
___
On February 12, 2005 09:21 pm, David Coulson wrote:
> If he gets a green light with a loopback plug wired like that, his
> controller is definatly screwed up :-)
>
> 1->4
> 2->5
>
> That was how I always learned to wire a loop plug anyway.
You're absolutely right, I made a pretty big (and public)
I have problems getting into maintenance screen of AMP,
What is the user I should use? I must be missing something easy ...
Thanks
robert
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Guys.. Im using iaxphone softphone for testing my asterisk config and I
noticed that FWD offers a IAX to FWD gateway using asterisk or any IAX
softphone..
Has anybody configured iaxphone to use this iax to fwd gateway?
I want to try this out before messing with asterisk and fwd.
Thx!
If you want to set up call forwarding in a device independent way, see
http://lists.digium.com/pipermail/asterisk-users/2003-July/016872.html
Unfortunately, if you get Asterisk to do the forwarding, there is no way to
tell just by looking at the phone that your calls are forwarded. If you use
an I
Andrew Kohlsmith wrote:
Unplug it and plug in a loopback plug (pin 1->5, pin 2->6) -- if the T1 alarm
doesn't go away, the T1 controller itself is kaput. If it goes green (or
off), then your wire is suspect.
If he gets a green light with a loopback plug wired like that, his
controller is defin
I had the same problems,
I changed network card first, same problem then I changed the burner and
everything started to work.
Make sure that on the second pc you have different burner.
Oh and I use the nero 6 ... With no problems.
robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:
Hello,
I understand that only the destination of a call should do jitter
buffering. So, if IAX2/PhoneA calls IAX2/PhoneB through my server (no
transfers), PhoneA and PhoneB need to perform their own jitter buffering,
and Asterisk will just forward the frames, correct?
What happens
Matthew Boehm wrote:
This is most likely due to the HUGE CHANGES in recent code in regards to
linked-lists. Be patient. They are being fixed and optimized.
-Matthew
Matthew,
Has there been any progress on this? I am still getting the same error
(from CVS-HEAD as of a few minutes ago):
app_addon
Andrew Kohlsmith wrote:
Unplug it and plug in a loopback plug (pin 1->5, pin 2->6) -- if the T1 alarm
doesn't go away, the T1 controller itself is kaput. If it goes green (or
off), then your wire is suspect.
I really love it when a poster asks such a question: "Could my cross
over cable have so
Rich Adamson wrote:
Need to replace our older soho fax machine with something more current.
Would like to run the fax line through *, but haven't been able to
make spandsp work correctly with digium TDM04b card. Our fax volume
This seems to be a problem with the current wctdm driver. It seems to be
you cannot since asterisk checks for the existance of zaptel on
initial compile and you will have chan_zap missing.
On Sat, 12 Feb 2005 15:30:11 -0800, Nitesh Divecha
<[EMAIL PROTECTED]> wrote:
> Hello,
>
> Is it ok to install Zaptel afterwards and go ahead and install Asterisk?
>
> For some re
I tried downloading to a different PC - the download speed was much
faster (or was it the mirror I was using?) - 4MB/sec. But again, I
burned the image and got the same error.
I suppose it's Nero that's messing it up? I've never had any problems
on the many .iso's I've burned before with Nero
On February 12, 2005 07:31 pm, Richard Reina wrote:
> On thing that is odd is that although the t1 cross
> over cable is plugged in to both * and the Adit. Both
> t1 and t1 leds on the Adit are red. How can they both
> have the same status if one is hooke up and on is not?
> Could my cross over
> Did a cable come out of the Adit(like the T1
> cable???)?
On thing that is odd is that although the t1 cross
over cable is plugged in to both * and the Adit. Both
t1 and t1 leds on the Adit are red. How can they both
have the same status if one is hooke up and on is not?
Could my cross over
beonice wrote:
I'm still (doggedly) trying to get asterisk to read my
voicemail configuration from MySQL. I'm using the
stable release of Asterisk, from back in December,
before realtime was included.
If anyone has got it to work, please contact me ...
I've posted details, but everyone who's respo
See my comments inline:
; mgcp audit endpoint aaln/[EMAIL PROTECTED] (vg200)
[general]
port = 2427
bindaddr = 128.100.10.10 (asterisk server)
[vg200]
Whatever host name that you put here should be resolvable either by
/etc/hosts or DNS lookup. If not, set it to the IP address of the host,
i.e.
I see that typo I made for this suggestion, but the real problem is
that the system doesn't seem to register with Asterisk.
I can't dial out or even if I fix the error in my config will I be able
to dial the extension.
This phone just doesn't seem to want to work with Asterisk. I have
found
--- Lyle Giese <[EMAIL PROTECTED]> wrote:
> Did a cable come out of the Adit(like the T1
> cable???)?
t1 cable is connected on bothe ends.
>
> There should be a 'craft' port to hook up a serial
> port with a term program
> and you can poke around and see what alarms it's
> reporting.
Do you me
Hello,
Is it ok to install Zaptel afterwards and go ahead and install Asterisk?
For some reason I install Asterisk first and I wanted to use Conference
Bringing which requires Meetme.
Can I install Zaptel on top of Asterisk?
Any help will be appreciated.
Nitesh
Did a cable come out of the Adit(like the T1 cable???)?
There should be a 'craft' port to hook up a serial port with a term program
and you can poke around and see what alarms it's reporting.
What alarms is * reporting?
Lyle
- Original Message -
From: "Richard Reina" <[EMAIL PROTECTED]
Hello all!
It's my try to make b2bua from asterisk. It's patched asterisk and
some AGI script for it. What it support? Full vovida's b2bua radius
emulation, radius failover, LCR, Call failover, Codec based routing,
Session-Timeout and much other things that can be useful.
Any suggestions welcome!
Hello all,
I have just been trying to install the latest ast_data from:
http://svn.asteriskdocs.org/res_data/ast_data/
into my cvs version of Asterisk and have found that the install patching
fails.
-
patching file contrib/scripts/sip-friends.sql
patc
Hello.
LCR means least cost routing, and it's billing system problem where to
route a call, not b2bua's. But currently I dunno any free billing
system that support it, so i moved this logic to b2bua.
On Sat, 12 Feb 2005 07:05:39 +0330, mohammad <[EMAIL PROTECTED]> wrote:
> Hi Mike;
> Thanks for
Using digit networks x100p clone card.
On both incoming and outgoing calls, once the call is
connected a second dial tone is generated.
Any ideas?
I have tried both jacks on the x100p clone; both produce the
same result.
-Mark
__
Roger Hanson wrote:
I've downloaded 2x and burned 2 cds and get an error "invalid
compressed format (err=2) system halted" message both times.
It'd be nice to have a MD5 to verify my download is OK. It'd narrow
down the problem to either the download or the burn, wouldn't it?
The other day I was g
I've been working with RealTime configuration from MySQL Server, and have had
good results. You might check it out. You can do a search for 'realtime' on
the Wiki and get some good documentation on how to set it up. I think in the
extconfig.conf file, not only do you need to identify the engine (
Spinvox have a distinct advantage over most telephony applications in that
their speech recognition does not have to occur in realtime - it simply
records the speech and then processes it afterwards.
I strongly suspect since the company always acts very tight-lipped about
their technology that it
Hello All,
I followed all of the steps to install Asterisk on my Fedora2 and it
worked great.
Now I want to uninstall Asterisk because I want to make a fresh install
along with some additional modules.
I have found that there does not seem to be a "make uninstall" to go with
the "make install"
I've downloaded 2x and burned 2 cds and get an error "invalid compressed
format (err=2) system halted" message both times.
It'd be nice to have a MD5 to verify my download is OK. It'd narrow
down the problem to either the download or the burn, wouldn't it?
- Original Message -
From:
Need to replace our older soho fax machine with something more current.
Would like to run the fax line through *, but haven't been able to
make spandsp work correctly with digium TDM04b card. Our fax volume
is very low (maybe a few per week), but we have multiple offices in
three geographic locati
Hi,
I'd like to use one card to interface with existing ISDN pbx output. How
stable are those cards for this ?
Where can I find more info how to setup ?
Regards,
Rob.
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Hi,
I'm currently deciding on what card to pruchase for octo/quad BRI card to
use with Asterisk on EuroISDN lines.
I'm aware of at least two options (Junghanns or Beronet), but don't know how
stable and well supported they are. Which ones are better supported ? Any
experiences? Any advice ? How t
You need to tell us what type of device you are
using to make the phone calls. Are you using
a ZAP FXS, a softphone, a sip phone, or an iax phone.
Also, how are you terminating the call. Is it via a
ZAP FXO device like a t100p, is it another VOIP phone,
or is it via a service provider like iax.cc
- Original Message -
From: "Michiel van Baak" <[EMAIL PROTECTED]>
To:
Sent: Saturday, February 12, 2005 11:57 AM
Subject: Re: [Asterisk-Users] chan_capi or chan_mISDN vs bristuff
> On 14:10, Fri 11 Feb 05, Remco Barende wrote:
> > Hi list!
> >
> > I'm currently using a HFC-S card for m
I have been using Asterisk to make phone calls and
when i tried to use it today the volume was
unexpectedly very low. Changing the volume in the
volume control didn't effect it. I believe the problem
lies with Asterisk and not the volume control. I
appreciate any feedback of where and what to check
Hi!
I would like to have feedback on wireless (wifi / 802.11b) IP phone to use
with Asterisk PBX. Can you sugest model, The best and also the worst to
use.
Thanks,
eric.
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Daniel Eboa wrote:
I downloaded the iso file of the last release, but unable to burn it
on CD. Got error at 90%. Did anyone experience the same problem ?
Maybe the iso file is corrupted.
I just burned the CD and it installed just fine on my test box.
Regards.
Daniel.
-Original Message-
F
> Your original posting said the sidetone was coming from the distant
> phone and did not even come close to implying that sidetone is
> something always engineered into the local phone, regardless of
> whether its analog or digital. Sidetone is always local phone
> generated by design.
I went b
While trying to compile asterisk, I get the following errors -
--
bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c
ast_expr.y:118: unrecognized: %locations
ast_expr.y:118: Skipping to next %
ast_expr.y:149: invalid @-construct
ast_expr.y:149: $. i
What's the recommended way to show my exact build of Asterisk -- down
to the minor-minor version number?
I ask because I am setting up a small testbed and need to keep myself
straight and would prefer something more authoritative than a post-it
note and my addled memory.
If I do Asterisk -V, I
> Maybe http://www.grandstream.com.cn/BT100_Spec_cn.pdf is a bit
> confusing if you can't read Chinese, but I think G.168 should
> be easy to identify :-)
ok, I did miss that. Then again, the grandstream does have a
speaker phone. I guess the problem is that I don't know of a
SIP hardphone tha
Astcc is mysql driven w/ perl based web ui
Areski is same concept based on postgres w/ a php frontend also tied
in w/ Areski other scripts for reports and such
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correct your dialplan. something like this
[from-iax]
exten => 105,1,Dial(IAX2/QIax1,20)
exten => 106,1,Dial(IAX2/QIax2,20)
exten => 107,1,Dial(IAX2/QIax3,20)
hth
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Hello All,
Could someone please give me your impressions as to which Calling Card
application is better?
I'm trying to decide on the one that I will implement while I'm learning
about my new Asterisk server that I have just installed on Fedora 2.
Thanks in advance,
Lonnie
__
I have two phones which are callerid num and name capable connected to
asterisk 1.0.3. Both of these phones will display number and name of
caller when available and when connected to the French phone company
(France Télécom). However, one of these phones will not show it on
asterisk connected via
Daniel Eboa wrote:
I downloaded the iso file of the last release, but unable to burn it on CD. Got
error at 90%. Did anyone experience the same problem ?
Maybe the iso file is corrupted.
Not as of approx 5:)) PM yesterday. I downloaded, burned, and in last
stage of the install ( compiling * )
On Feb 12, 2005, at 17:58, Roy Sigurd Karlsbakk wrote:
Does anyone know of a speech recognition module (like say yes or no,
or numbers) I guess due to the complexity of speech recognition it
might just be found in commercial applications or am I wrong like
always?
What's wrong with the old and n
Does anyone know if
the 7912G (which the wiki says can do either sccp or sip) uses the 7940/60 sip
firmware? I ask this because the only firmware I can seem to find on TAC
for the 7912G is sccp, no sip...if it takes it's own firmware and doesn't use
7940/60 firmware, can someone point me t
Gary Reuter wrote:
I need to test if the app, MYSQL() in this case, returned -1 or 0.
You can't. If it returns -1, execution stops and if there is a channel
active, it is hung up.
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> > The sidetone is 'always' generated within analog and digital phones.
> > It never comes from any source outside the phone. In analog phones,
> > it derived from the hybrid within the phone. On digital phones, its
> > basically firmware.
>
> I never said that sidetone was generated outside the
After months of setting up Asterisk. I completed the
final testing last night We would go live on Monday.
Or so I thought.
As I moved the Adit 600 back out of the way, sliding
it six inches. I noticed the major light was red as
were both T1 and T2 lights ( eventhough only one t1
port is being u
Hi,
I'll probably kick myself when I read the replies to this...
How do I test the return code of an app in the dialplan?
I need to test if the app, MYSQL() in this case, returned -1 or 0.
It's easy to see after-the-fact in the log output, but I need the
result in the dialplan, I just can't fin
Does anyone know of a speech recognition module (like say yes or no,
or numbers) I guess due to the complexity of speech recognition it
might just be found in commercial applications or am I wrong like
always?
What's wrong with the old and non-fancy IVR?
Voice recognition menus only piss people
> > Those type changes to iax.conf require a full stop of
> > asterisk followed by a cold asterisk startup. A restart
> > from the CLI won't cut it.
>
> Ahh! That's a very important piece of information!
>
>
> > Were you previously doing the CLI restart?
>
> I did lots a CLI "reloads", and f
Asterisk wrote:
persistentmembers = yes
That is only in CVS HEAD, and it does not apply to logins generated via
AgentCallbackLogin, it applies to dynamic members added via AddQueueMember.
There is, however, some analogous persistence in chan_agent (again CVS
HEAD only) that may do what the OP wa
PROBLEM SOLVED (read the rest)
Weird. That's all I did to get my two Fritz! cards working.
Only did the modprobe.
Er... that's really not enough (in this case, at least).
Asterisk will only be happy with the initialization when you can see all
four channels listed in the "imon" utility (from t
Iqbal wrote:
Hi
I dont know jack about speech recognition, however since this topic came
up anyonw know how spinvox do speech ercognition, in fact its so good it
converst the speech to text and sends the voicemail as a SMS, I think a
awesome addone to the sms module in asterisk.
If it works real
Howdy,
Who knows? If I do an md5sum on the 0.5 iso, I get
9d5657b7c833830b8a1fd1f024215d46 asteriskathome-0.5.iso
They don't tell you the right md5 that I can tell,
so nobody must care if the downloads work.
Good day,
Ralph
On Sat, 2005-02-12 at 13:32 +0100, Daniel Eboa wrote:
> I downloaded t
Hi
I dont know jack about speech recognition, however since this topic came
up anyonw know how spinvox do speech ercognition, in fact its so good it
converst the speech to text and sends the voicemail as a SMS, I think a
awesome addone to the sms module in asterisk.
Iqbal
On 2/12/2005, "Race Va
*snipped
depending on which directly the call is traveling the option is 'T" or
"t", have you tried both?
i think i 'directly' need to go find some coffee! (meant direction, sorry)
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Need help with how to configure for parked calls in the Flash Operator
Panel's op_buttons.cfg file ...
I've looked on the wiki, google and asternic's site and can't seem to find
how to setup op_buttons.cfg to monitor parked calls.
For example, if someone parks in 701, I'd like to see that represe
Hi Rich -
Those type changes to iax.conf require a full stop of
asterisk followed by a cold asterisk startup. A restart
from the CLI won't cut it.
Ahh! That's a very important piece of information!
Were you previously doing the CLI restart?
I did lots a CLI "reloads", and few cold restarts to
Ryan Stark wrote:
Hi I'm having trouble # transfering queue calls.
in extensions.conf I have:
[macro-queue]
;
; Places caller in queue
; ${ARG1} - Queue name to place caller in.
; ${ARG2} - Voicemail Extention
; ${ARG3} - Caller ID to Set.
exten => s,1,DBget(temp=nm/on) ; Get Night key, if not exis
Is anybody familiar with the recent bristuff packages released ?
There is only a 3 hour difference in release time between them and the
CHANGES files are the same.
Also what's strange RC7 has 163K and RC7a has only 87K.
Ideas anyone ?
Regards,
Ivan
___
Nope, works fine. Several people have already downloaded and installed it
yesterday.
Try again.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Eboa
Sent: Saturday, February 12, 2005 7:32 AM
To: Asterisk Users Mailing List - Non-Co
Hello:
I want to receive calls from my SIP proxy and re-route them to one
of the analog lines on my Cisco VG200 ia MGCP and Asterisk.
Inbound SIP calls will arrive with the five digit called number preficed
with an "m" by the proxy. I'd like to them match these calls against
a rule like exten =>
I'm using Asterisk on a system described as below:
Asterisk version 1.0.5
on Linux Debian version 3.0 (unstable) with kernel version 2.6.10
(hardware: PC, i386 class).
My Asterisk works with a phone card Digium TDM400P, where
2 FXS and 2 FX0 modules are provided.
It works, but I notice an annoying
Hi,
I'm using the macro below in extensions.conf for most of my outbound
calls. One issue with my current configuration is that when I make an
outbound call it doesn't properly detect that my PSTN line (Zap/1) is
busy with another call and then overflow to my outbound IAX
connections. I think the
On 13:05, Sat 12 Feb 05, JunkMail wrote:
> Sure !
> But that's not the complete initialization of the isdn system.
> With "modprobe -v hisax type=21,21 protocol=2,2" ALONE, not even the first
> card answers calls...
> I just wonder what else does "isdntool" does to initialize the isdn
> system...
>
Sure !
But that's not the complete initialization of the isdn system.
With "modprobe -v hisax type=21,21 protocol=2,2" ALONE, not even the first
card answers calls...
I just wonder what else does "isdntool" does to initialize the isdn
system...
Thank you for your reply.
M.G.
- Original Messa
hello
try: exten => 8908,1,Dial(h323/8908,20,Ttr) !
harry
--- Scott Henderson <[EMAIL PROTECTED]> a écrit :
> I am trying to add a Polycom IP 3000 to our Asterisk
> system and am not
> getting anywhere.
>
> h323.conf
>
> [8908]
> type=friend
> host=192.168.104.25
> secret=polycom
> cont
On 12:20, Sat 12 Feb 05, JunkMail wrote:
>
> It all starts with "modprobe -v hisax type=21,21" (loading hisax and
> telling
> it that we'll use two teles pci cards)
> and then ? what else ???
>
try adding protocol=2,2 (for euroisdn, replace with your
type)
--
Michiel van Baak
http://lunteren.van
I downloaded the iso file of the last release, but unable to burn it on CD. Got
error at 90%. Did anyone experience the same problem ?
Maybe the iso file is corrupted.
Regards.
Daniel.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello!
After LOTS of research on this list and internet in general I managed
to get
an old Teles PCI card working with Asterisk throught ISDN4Linux.
No echos, no delays, simply perfect -- electronic poetry ! :)
eheheheheh
I just didn't get it to work
Hello all!
It's my try to make b2bua from asterisk. It's patched asterisk and
some AGI script for it. What it support? Full vovida's b2bua radius
emulation, radius failover, LCR, Call failover, Codec based routing,
Session-Timeout and much other things that can be useful.
Any suggestions and crit
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