The issue I have had with all other FC3 kernels apart from
the 2.6.9 one was that the zaptel build would throw lots of warnings up.
This would have the knock on of hanging the system, spinlock I think the problem
was, on a modprobe -r.
Lee
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi,
Sorry I can't help you in your questions but actually I have one.
I m using TDM22B card. I am in india.
I want to know are you able to get callerd ID?
What cidsignalling you have set for in zaptel.conf.?
On my system when a call comes it checks for caller ID and returns and error.
From: Damon Estep [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] How to use * and # as part of number
indialcommand
Good to hear you have found a temporary solution, although I think it is
the permanent solution.
Keypad protocol is a bandaid to fix the real problem, not a
Hi,
Emanuele Pucciarelli schrieb:
Tobias Wolf ha scritto:
I tried that successfully with my own SMS rig a couple of years ago. As
far as I could tell from experimenting and from the ETSI docs, the phone
knows it shouldn't ring, but it should answer and talk FSK to the SMSC,
by looking at
Hi,
I'm currently running asterisk 1.0.9-r1 and zaptel 1.0.9_p1-r1 (the
current gentoo ~x86 versions), with the UK CallerID patches from
http://www.lusyn.com/asterisk/patches.html applied.
The Zap interface itself seems to work fairly well - although it's a
little quiet, there is no echo.
Narcis GRATIANU a écrit :
Hello !
Anybody has any idea how can i configure my Asterisk box to send
outgoing SIP calls to Dualtalk provider ?
It depend if you have an dynamic IP or not. Here is a sip.conf with
dynamic IP
; Numbering
register = 6digits code:secret@sip.numbering.info
Hi Tim
Here are the my CID zapata.conf settings that are working with my TDM400P card
in the UK with a BT land line.
callerid=asreceived
usecallerid=yes
cidsignalling=v23
cidstart=polarity
ukcallerid=yes
Cheers
Graham
From: [EMAIL PROTECTED] on behalf
Kamran Ahmad ha scritto:
Hello
i am trying to use this exmple with SER-0.9.3
but still NATED Clients are not working any other
requirement
Look at the examples you find at www.onsip.org, they are really well
explained.
log every step taken with something like log(2,now I'm doing
Hello,
I have this already in sip.conf.
;sip.conf
[general]
context=default
port=5062
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no
autocreatepeer=yes
I have done sip reload and also restarted
asterisk with stop now and asterisk vvvgc.
Unfortunately Asterisk still does
From: Henry Junior [EMAIL PROTECTED]I want to issue a System cmd in my dialpan that is similar to the
unix echo command stated above. *EXCEPT* that I want to pipe theresults into a *new* variable (vs a text file.)Ideally, what would happen is in my diaplan I would issue the 'echo |date' command
AFAIK
you have to add port=5062 in the context general.
Stop
and restart asterisk, and everything should be fine..
[general]
port=5062
Regards
Guido
gwsNetTech
Guido Hecken
Quirrenbacher Str. 36
53639 Königswinter
Germany
fon +49(2244)
870663
fax +49(2244)
Richard,
Thanks for the pointer, and thanks to Nico at Siemens!
For future reference to procedure to recover from a Can't Boot!! situation
is:
1) Start the Siemens netbootserver
2) Configure the netbootserver with the FTP server where new software image
is located (must be in the FTP root
Mark Phillips wrote:
You want to speak with Matt Riddell. He's on the list
Indeed I am!
Call me 24 hours +64 3 4555770
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php
I agree you however i solved my problem with
app_voicemail.c
The table scheme provide in doc/README.odbcstorage
don't match to sql queries in app_voicemail.c
I think developpers who has written app_voicemail.c
for ARA provide a useable table !
Regards
Harry
--- Steve McMahon [EMAIL PROTECTED]
I agree you however i solved my problem with
app_voicemail.c
The table scheme provide in doc/README.odbcstorage
don't match to sql queries in app_voicemail.c
I think developpers who has written app_voicemail.c
for ARA provide a useable table !
Regards
Harry
--- Steve McMahon [EMAIL PROTECTED]
Hello,
I have SER and Asterisk running on the same box. I want SER
to listen on port 5060 (it is) and Asterisk to listen on port 5062. I have
configured my phones to register with x.x.x.x:5060 (SER) and Asterisk will
purely act as a voicemail server at the moment. However I cannot get
Hi there,
Sorry for the late reply. I had too many emails in my mailbox to clean up.
Anyway, I found out the problem is the sendmail in Linux did not work and
the voicemail.conf in our asterisk is ok. There is another issue for email
notification: some email servers rejects the email from
hi all, i m new to this list,
I have a big problem, how to configure a Queue to follow the behaivor of:
every incoming call should first ring the member listed first (in
queues.conf) - then the second and so on.
Is there a way to always start ringing with the first member of the queue?
Here
Hi Sr.
rrmemory i same like roundrobin, but this policy store which is the next
when a call get into your system.
For example with next queue:
SIP/1
SIP/2
SIP/3
and roundrobin, all calls stars with SIP/1 and with rrmemory first call
starts with SIP/1, second call with SIP/2 and so on.
Hi,
I create extensions started with *XXX and don't have any problem but to
create extensions started with # respond null tone. Asterisk support
extensions started with # or this is a problem in agi scripts?.
I see the scripts and probe diferents extensions formats (1*11, 1#11,
thanks for reply
but i tried it out: roundrobin - is doing like this
first Call rings sip/1 then sip/2 then sip/3
second Call rings sip/2 then sip/3 then sip/1
third Call rings sip/3 then sip/1 then sip/2
i want that: (this should be roundrobin)
first Call rings sip/1 then sip/2 then sip/3
[EMAIL PROTECTED] wrote:
Also - an outside chance - make sure Tip and Ring
are correct. You could be getting ground loops - depends on the noise.
I am having noise and slip errors between my TE110P and a legacy PBX T1
card. Could this be the same symptom? The connection is made using a 15 pin
Hi All,
I have configure my Asterisk as follow (using [EMAIL PROTECTED]):
[zaptel.conf]
span=1,1,0,ccs,hdb3,yellow
bchan=1-15,17-31
dchan=16
loadzone = uk
defaultzone=uk
[zapata.conf]
[channels]
switchtype=euroisdn
pridialplan=local
prilocaldialplan=local
internationalprefix=00
We use agents and queues, with CVS HEAD. I've read up on realtime queues
and queue members, (and actually understand it!) but there is no
reference to agents.
Is it possible to have realtime agents as well ?
Julian.
___
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Hi All
I have posted this problem many times on the list but
no reply, trying one more time may be someone will
response this time
When I call from 1 RTC Client to another without
Asterisk everything use to be fine but when asterisk
is there as a Registrar a problem use to occur in more
than
Hello,
I am writing a script (php script that runs via fastAGI) that takes
incoming calls and processes them in various ways depending on settings
from a database.
At some point, I need the script to receive an incoming fax. But the
problem is that if I run NVFaxDetect from the script, then
Have you resolved the problem, I find the same problem
Thanks
2005/8/23, Don Brearley [EMAIL PROTECTED]:
O come on now! Nothing? Not even a No idea! Good Luck! or anything?
Weak :)
Just kidding. Thanks just the same.
- Don
[EMAIL PROTECTED] 8/22/2005 11:09 AM
Hello,
nope, i havent :\
Keith Yoder wrote:
Casey Boone escreveu:
can someone who has a grandstream handytone 488 working with making
outgoing calls through the fxo port please post the parts of their
config that deal with this port? i cant quite seem to get it to make
outgoing calls despite having
What is CFU and CFNR?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Michel Koenen
Sent: Tuesday, August 30, 2005 1:46 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] How to use * and # as part of
[EMAIL PROTECTED] wrote:
What is CFU and CFNR?
Call forwarding unconditional
call forward not reachable
--
Andreas Sikkema bbned NV
Van Vollenhovenstraat 33016 BE Rotterdam
t: +31 (0)10 2245544 f: +31 (0)10 413 65 45
___
Hi,
Daniel Grad wrote:
I am writing a script (php script that runs via fastAGI) that takes
incoming calls and processes them in various ways depending on settings
from a database.
At some point, I need the script to receive an incoming fax. But the
problem is that if I run NVFaxDetect from
On Tue, 2005-08-30 at 07:19 -0600, Damon Estep wrote:
What is CFU and CFNR?
Call Forward Unconditional - Call Forward No Response
--
Dave Cotton [EMAIL PROTECTED]
___
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Asterisk-Users mailing
Hi John,
sorry to bother you but how can this be implemented?
can u give me a peice of the extensions.conf code that does that?
by the way the FXO ports are always green they never change even if theres no
line plugged in
-- Original Message --
From:
On Sun, 28 Aug 2005, Michel Koenen wrote:
Your example is still doing what I tried already, so eventually the
dial command ends like:
Dial(zap/4/*21*)
or
Dial(zap/4/*31*)
I prefer to use Dial(zap/4/*21*thenumber)
or Dial(zap/4/*31*thenumber)
But whatever I try, the error message as
There's not enough info in the posts below to help. The issue sounds
like a problem with disconnect supervision (or whatever you want to
call it).
In the case where a sip phone (or other asterisk phone) answers the
call and then hangs up, the zap channel is being hung up properly due
to the sip
Hi.
Try to set penalty for SIP/2 and SIP/3 so they won't ring unless SIP/1 is
busy.
Braz
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Christian
Gansberger
Envoyé : 30 août, 2005 08:09
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet
No, I don't have service with them.
I am thinking about getting service from them and I had some specific
questions about porting telephone numbers and clear up some things about their
packages.
Regards,
Chris
- Original Message -
From: Rick Baranowski [EMAIL
I´ve just downloaded the tarball from ftp.digium.com and it's still not
showing the version:
sertwo*CLI show version
Asterisk built by [EMAIL PROTECTED] on a i686 running Linux on 2005-08-30
13:46:04 UTC
On Sun, 28 Aug 2005, Kevin P. Fleming wrote:
Due to a packaging error, the tarball
bodra wrote:
Hi John,
sorry to bother you but how can this be implemented?
can u give me a peice of the extensions.conf code that does that?
by the way the FXO ports are always green they never change even if theres no line plugged in
I believe that on the TDM400 green only means
Asterisk Supporter wrote:
Asterisk has this error on compile:
flex ast_expr2.fl
ast_expr2.fl, line 50: unrecognized %option: reentrant
ast_expr2.fl, line 51: unrecognized %option: bison-bridge
ast_expr2.fl, line 52: unrecognized %option: bison-locations
make: *** [ast_expr2f.c] Error 1
Your
I use HT488, and I can make and receive FXO calls. It's actually quite
simple, you create a SIP acount in sip.conf. On the FXO section of HT488 web
admin page you enter these registration values. When you reboot the HT488
you should see it registering on Asterisk CLI.
What's left is a
On Tue, 2005-08-30 at 17:11 +0300, Soner Tari wrote:
I use HT488, and I can make and receive FXO calls. It's actually quite
simple, you create a SIP acount in sip.conf. On the FXO section of HT488 web
admin page you enter these registration values. When you reboot the HT488
you should see
Of course... Those are the basics to get HT488 working for the OP. In this
thread I am not trying to show how to create dialplans.
On Tue, 2005-08-30 at 17:11 +0300, Soner Tari wrote:
I use HT488, and I can make and receive FXO calls. It's actually quite
simple, you create a SIP acount in
Hi!
When I try to load the ztdummy driver via insmod ztdummy, I get the
following errors:
/lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_unregister
/lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_transmit
/lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_receive
Hiz;
I post you the debug, for seeing if anyone can help me.
---
Aug 30 12:53:36 VERBOSE[3562]: -- Accepting voice call from
'800245' to '800275' on channel 0/1, span 4
Aug 30 12:53:36 DEBUG[3562]: Enabled echo cancellation on channel 10
Aug 30
In article [EMAIL PROTECTED],
Christoph Eicke [EMAIL PROTECTED] wrote:
Hi!
When I try to load the ztdummy driver via insmod ztdummy, I get the
following errors:
/lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_unregister
/lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol
Your kernel has to be compile with CONFIG_CRC_CCITT=y or m.
Braz
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Christoph
Eicke
Envoyé : 30 août, 2005 10:47
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [Asterisk-Users] unresolved
i greatly appreciate the information and will be giving it a whirl later
today :)
Casey
Soner Tari wrote:
I use HT488, and I can make and receive FXO calls. It's actually quite
simple, you create a SIP acount in sip.conf. On the FXO section of HT488
web admin page you enter these
There is a lot of internal preset dial numbers starting with #xxx, that is
probably the reason for not to be able to use it on the dialplan
Is nothing relating to agi.
Carlos
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gabriel Perez
S.
Sent:
Date: Tue, 30 Aug 2005 08:56:33 -0500
Asterisk Supporter wrote:
Asterisk has this error on compile:
flex ast_expr2.fl
ast_expr2.fl, line 50: unrecognized %option: reentrant
ast_expr2.fl, line 51: unrecognized %option: bison-bridge
ast_expr2.fl, line 52: unrecognized %option: bison-locations
Speaking of GS..
I know polycom phones can eb rebooted with some script using sip_notify.
Can GS phones do this also?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Harald Holzer
|Sent: Lunes, 29 de Agosto de 2005 01:09 p.m.
|To: Asterisk Users
I'm interested for this thread, can you explain with an example please?
In my extensions.conf I have
...
[sip.proxy.com]
switch = Realtime/[EMAIL PROTECTED]
in extensions table on mysql I can insert on app field the command
include and in the appdata field my context ?
Luca
-Original
Am i alone with this problem ?
I just rewrote voicemessages table because of errors.
I read app_voicemail.c to fix my problem.
However app_voicemail.c support many schemes to query
the tables.
Harry
--- Jerris, Michael MI [EMAIL PROTECTED] a écrit :
harry gaillac
I agree you however i
Am i alone with this problem ?
I just rewrote voicemessages table because of errors.
I read app_voicemail.c to fix my problem.
However app_voicemail.c support many schemes to query
the tables.
Harry
--- Jerris, Michael MI [EMAIL PROTECTED] a écrit :
harry gaillac
I agree you however i
This is weird.
If I have 2 members call into meetme using zap PRI channels on the box,
they can here each other's keypresses.
If I have 2 members call into a separate box using the same PRI's and then
forward (dial(iax/...)) them to the previous box into the same meetme,
they only hear a
Luca Lafranchi wrote:
I'm interested for this thread, can you explain with an example please?
In my extensions.conf I have
...
[sip.proxy.com]
switch = Realtime/[EMAIL PROTECTED]
in extensions table on mysql I can insert on app field the command
include and in the appdata field my
Julian Lyndon-Smith wrote:
We use agents and queues, with CVS HEAD. I've read up on realtime queues
and queue members, (and actually understand it!) but there is no
reference to agents.
Is it possible to have realtime agents as well ?
Julian.
No there isn't. And there won't be until
having problems with installing [EMAIL PROTECTED] i downloaded the
asteriskathome-1.5.iso file from asteriskathome.sourceforge.net link burned it on a cd but it is not booting what seems to be the problem hoping for a quick reply
___
--Bandwidth
On Tue, 30 Aug 2005, Anton Krall wrote:
Speaking of GS..
I know polycom phones can eb rebooted with some script using sip_notify.
Can GS phones do this also?
You can reset the phones by requesting the right page from their built in
web server as long as you know the admin password.
Hi,
I have
just installed [EMAIL PROTECTED] from the link
below ... it works fine for me
http://switch.dl.sourceforge.net/sourceforge/asteriskathome/asteriskathome-1.5.iso
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of prashant
yadavSent: 30 August 2005 18:49To:
I am also having the same issue from the ftp tarball.
B. J.
-Original Message-
From: Martin Morey [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 30, 2005 8:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.2.0-beta1 tarball
Following up on a thread that I started about Agents/Queue and
acknowledging calls before bridging them...
Greg Boehnlein said that he was putting his efforts into ICD.
I downloaded and installed ICD, and I can get simple queue and agent
stuff working fine, and see that this new design is
does anyone had an experience with not hanging up the
call in astcc, thats my problem sometimes call does
not hang up automatically and even when pressing star
the call continue without stopping.
Regards;
jonny
Start
Title: Message
Did
you burn the iso properly or as a file on the cd?
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
ChandsSent: 30 August 2005 19:20To: 'prashant yadav';
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE:
Any chance anybody has asterisk at home with asterisk 1.2 beta? any problem if
I reinstall the beta on top of asterisk at home?
Thanks
CM
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Asterisk-Users mailing list
On 30/08/05, Graham Kiff [EMAIL PROTECTED] wrote:
Hi Tim
Here are the my CID zapata.conf settings that are working with my TDM400P
card in the UK with a BT land line.
callerid=asreceived
usecallerid=yes
cidsignalling=v23
cidstart=polarity
ukcallerid=yes
Cheers
Graham
Hi
Ok, if I have understood well... this mean that if I have configured a table
for extension.conf in real time mode, I can't use the static mode
(ast_config table) and consequently I can't use the include command ?
Luca
-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Title: Message
Did you check the FAQ on the [EMAIL PROTECTED] site? It talks about a few
reasons why this might be happening (IDE/SCSI, older/newer hardware,
etc). I just did it (older PIII) and it worked fine, too.
razza wrote:
Did you burn the iso properly or as a file
on the cd?
That's a bugger.
Forgive me for asking, but how is is possible to be able to have SIP
realtime (adding new sip phones in without having to reload) but we
can't have agent realtime ?
In my simple mind I substitute agent for SIP and can't compute :)
Julian.
Matthew Boehm wrote:
Julian
Is your computer set to boot from CD?
It's been noticed that 'cheap' CD's have had problems, try a different CD.
As this is a problem specific with [EMAIL PROTECTED], i suggest you post a message
on the dedicated forum for [EMAIL PROTECTED] on sourceforge - you'll get more help.
Woody
Julian Lyndon-Smith wrote:
That's a bugger.
Forgive me for asking, but how is is possible to be able to have SIP
realtime (adding new sip phones in without having to reload) but we
can't have agent realtime ?
In my simple mind I substitute agent for SIP and can't compute :)
because you
From voip-info.org:
Queue(queuename|options|optionalurl|announceoverride|timeout)
'optionalurl' allows you to send a URL to devices that support it.
Does anyone have details on the devices that support the
optionalurl method of the Queue application? I am wondering if there is
Does anyone have details on the “devices” that support the optionalurl
method of the Queue application? I am wondering if there is a softphone
that supports this. The only thing that seems to happen is the queue_log
is updated with whatever is placed in the “optionalurl” location of the
Hi,
I want to start managing my asterisk boxes with a centralized
graphical based interface so I can (due to customers request) give
control to customers to add/change extensions to their current PBX
intallations such as (not complete list)
Add/del/mod extensions
sound recordings (ivr or voice
Sounds to me like you copied the file to a disk rather than burn an ISO
image. A common mistake folks make especially if they've never done an
iso before.
What tools are you using? I prefer k3b. It rocks
Mark
prashant yadav wrote:
having problems with installing [EMAIL PROTECTED] i
From: Arik Funke
arik.funke at gmx.de Hi,my zaphfc is flooding my syslog with two messages (even without asterisk running). Is this normal?:--zaphfc: bchan rx fifo not enough bytes to receive! (z1=1360, z2=1353,
wanted 8 got 7), probably a buffer
Hi weicheng,
I found your e-mail at the list. I bought a F1000 and configured it to
connect on my [EMAIL PROTECTED]
But, some times the call is completed, some times no. Some times the F1000
call the other phone, but when I answer, I don't heard anything. Some times
I call, I answer and heard
Matthew Boehm wrote:
Julian Lyndon-Smith wrote:
That's a bugger.
Forgive me for asking, but how is is possible to be able to have SIP
realtime (adding new sip phones in without having to reload) but we
can't have agent realtime ?
In my simple mind I substitute agent for SIP and can't
Hello,
I'm currently researching a project that would enable us to pull the
actual signaling (SIP conversation) along with our CDRs
The best way I can tell to approach this is to set up a server on a
SPAN port which mirrors all my proxy servers' traffic.
I was curious if anyone else has ever
Is it possible to do nested dial() command on one line,
Dial number, wait new seconds, dial another number etc.
or dial number and jump to another line to continue dialing.
D(ww) doesn't work as it sends DTMF but before the call is bridged, and
I need to send numbers after the call is bridged.
In the next week to two weeks I'll be posting some information
concerning a system I've been designing. It currently does three layer
hosted VoIP pbx services as well as hosted ITSP services (the model is
System Owner - you, Affiliates - pbx owner/operators or ITSP operators,
and end users). The
For call accounting and billing, you can check out
www.aleph-com.net/astpp It is being prepared for another release which
will work closely with AMP.
Darren Wiebe
[EMAIL PROTECTED]
Erick Perez wrote:
Hi,
I want to start managing my asterisk boxes with a centralized
graphical based
is there any open source softpone for
windows?
Thanks
Matt
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
myphone fromopenh323 .
- Original Message -
From:
Matt
To: asterisk-users@lists.digium.com
Sent: Tuesday, August 30, 2005 4:54
PM
Subject: [Asterisk-Users] free open
source softphone for windows
is there any open source softpone for
windows?
iaxcomm: http://iaxclient.sourceforge.net/iaxcomm/
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of
MattSent: woensdag 31 augustus 2005 1:55To:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] free open
source softphone for
Hey - I know there's some other people out there that have the 9133i ...
has anyone gotten the DTMF tones to work after the far side picks up? I
didn't have any problems out of the box with my SPA-841 phones... the
aastra has been nicer so far, but I can't seem to get it to dial the
touch
On 8/30/2005, Geoff Manning [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
Also - an outside chance - make sure Tip and Ring
are correct. You could be getting ground loops - depends on the noise.
I am having noise and slip errors between my TE110P and a legacy PBX T1
card. Could this be
On 8/30/2005, Gulzar Hussain [EMAIL PROTECTED] wrote:
Hi All
I have posted this problem many times on the list but
no reply, trying one more time may be someone will
response this time
When I call from 1 RTC Client to another without
Asterisk everything use to be fine but when asterisk
is
Geoff Karl wrote:
Thanks Matt, that is a good strategy.
Any idea on how to pass the reason a call failed back through the
Asterisk Manager Interface? It would be great to send something back
like Busy, NoAnswer, etc...
You could use the dialstatus variable. Bear that if you were using a
It worked fine for me. I renamed all my /usr/src directories to
old_asterisk old_zaptel etc. so that when I downloaded the 1.2beta1 source
from CVS it would create and install the directories from scratch again.
So far AAH v1.5 is working perfectly with Asterisk v1.2beta1.
-Original
Hello,
Im having trouble figuring out how to setup Asterisk
so that its only a registrar not passing any RTP data during
phone calls.
So far I got this far:
Asterisk server holds registration information for phones
Phones register with Asterisk giving it their ip+port where
they
On 8/30/05, Gulzar Hussain [EMAIL PROTECTED] wrote:
When I call from 1 RTC Client to another withoutAsterisk everything use to be fine but when asteriskis there as a Registrar a problem use to occur in morethan 90% calls, Caller can hear the voice of thereceiving side
but the receiver cant be able
have you tried in the sip.conf for the
devices
canreinvite=yes
- Original Message -
From:
Tomas Florian
To: asterisk-users@lists.digium.com
Sent: Tuesday, August 30, 2005 8:48
PM
Subject: [Asterisk-Users] Registrar only
setup
Hello,
Im having
No I havent tried it but looks
like exactly what Im missing.
Thanks Ariel !
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ariel Batista
Sent: Tuesday, August 30, 2005
7:06 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re:
Hello group,
I'm running asterisk @ home 1.5 - I would like to change these messages(call
attend) to Spanish, how I can do that.
Thanks,
Nelson
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On Tue, 2005-08-30 at 20:25 -0500, Nelson Granados wrote:
Hello group,
I'm running asterisk @ home 1.5 - I would like to change these messages(call
attend) to Spanish, how I can do that.
You need to create o download any language pack.
Follow the instructions from:
On Tue, Aug 30, 2005 at 05:27:37PM -0400, Mark Phillips wrote:
Sounds to me like you copied the file to a disk rather than burn an ISO
image. A common mistake folks make especially if they've never done an
iso before.
But then also wrote:
What tools are you using? I prefer k3b. It
On Tue, Aug 30, 2005 at 10:08:51AM +0100, Graham Kiff wrote:
Hi Tim
Here are the my CID zapata.conf settings that are working with my TDM400P
card in the UK with a BT land line.
callerid=asreceived
usecallerid=yes
cidsignalling=v23
cidstart=polarity
This is for
Kevin Bockman a écrit :
Does anyone have details on the “devices” that support the optionalurl
method of the Queue application? I am wondering if there is a
softphone that supports this. The only thing that seems to happen is
the queue_log is updated with whatever is placed in the
On Tue, 30 Aug 2005, Hadar Pedhazur wrote:
Following up on a thread that I started about Agents/Queue and
acknowledging calls before bridging them...
Greg Boehnlein said that he was putting his efforts into ICD.
I downloaded and installed ICD, and I can get simple queue and agent
stuff
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