Hi,
I'm curious if anyone knows of any possibility to use video VOIP client
(like Ekiga or Linphone or...) under Linux that could be operated by
touchscreen friendly GUI (bigger buttons, large keypad, etc...) ?
I like Ekiga, but GUI is small and cannot be operated via touchscreen... But
maybe
Hi,
X-Lite demo version has only one SIP account possible. I'd like to set it up
in such manner that I could register with Asterisk being at home (local LAN,
local ip) and at work (external ip).
Is this possible since X-Lite allows only one sip account settings ?
Thanks in advance,
regards,
- Original Message -
From: Robert Rozman [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, March 20, 2008 7:40 PM
Subject: [asterisk-users] 423 Interval Too Brief and expiry settings
insip.conf
Hi,
I'm getting this error when registering with SIP server using
Hi,
has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto
or more info about needed Asterisk SW and setup ?
Thanks in advance,
regards,
Rob.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
- Original Message -
From: Michiel van Baak [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, April 02, 2008 10:51 AM
Subject: Re: [asterisk-users] Howto connect to Cirpack softswitch
withAsterisk ?
On 10:11, Wed 02 Apr 08, Robert Rozman wrote:
Hi,
has anyone
Hi,
I'm getting this error when registering with SIP server using Asterisk
1.4.10 and Freepbx...
I'm getting this error no matter what I try to setup in sip.conf :
- I'm getting confused whether options are maxexpirey=36000 or
maxexpiry=36000 ?
- Can I solve this with some settings in
Hi,
I'm about to test VOIP connection (from my ISP provider) directly through
dedicated network card instead of going through ADSL gateway with analog
phone port - SPA 3000 - Asterisk.
I need to have eth2 set on dhcp (to retrieve IP automatically) and then work
with it under Asterisk as
Hi,
I have an older phone with touch screen from Philips. It have it connected
to Sipura 3000 FXS port and majority of features work ok.
But phone also has touchscreen and web browser that I'd love to use for
accessing my local web pages. But the phone only allows me to setup ISP
phone number
Hi,
I have an older phone with touch screen from Philips. It have ti connected
to Sipura 3000 FXS port and majority of features work ok.
But phone also has touchscreen and web browser that I'd love to use for
accessing my local web pages. But the phone only allows me to setup ISP
phone number
Hi,
some time ago we used bristuffed Asterisk for our hfc cards cause it offered
more features (echo cancellation most important) and was quite stable...
I'm seeing now (I'm putting together Asterisk after a long time with hfc
card) that there are now 3 choices for hfc chipsets :
vISDN,
Hi,
we're writting interface module for our speech recognition system. We would
like to export stream of audio samples to external app, but to preserve dtmf
recognition and dialplan progress.
I wonder if recording application would be a good start for that (recording
application obviously
Cohen wrote:
On Fri, Sep 15, 2006 at 02:58:02PM +0200, Robert Rozman wrote:
I'm banging my head on compiling bristuff modules for Suse 10.0 with
kernel
Linux laps1 2.6.13-15.11-smp #1 SMP Mon Jul 17 09:43:01 UTC 2006 x86_64
x86_64 x86_64 GNU/Linux
and Asterisk 1.2.10-BRIstuffed-0.3.0-PRE
I'm banging my head on compiling bristuff modules for Suse 10.0 with kernel
:
Linux laps1 2.6.13-15.11-smp #1 SMP Mon Jul 17 09:43:01 UTC 2006 x86_64
x86_64 x86_64 GNU/Linux
and Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s.
I get this :
laps1:~/Voipy/Bristuff/bristuff-0.3.0-PRE-1s/zaptel #
- Original Message -
From: Ronald Wiplinger [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, April 26, 2006 2:41 PM
Subject: [Asterisk-Users] I am looking for a webphone on MY SITE
I am looking for a way
:39, Robert Rozman wrote:
- Original Message - From: Time Bandit
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 20, 2006 4:18 PM
Subject: Re: [Asterisk-Users] Jingle support - can we test the feature
Hi,
we would like to build IM-Voice community for our students around Asterisk,
Jingle, Jabber.
Can we already test those features ? Anyone already running such setup? Any
more info ?
Thanks in advance,
regards,
Rob.
___
--Bandwidth and
Hi,
I want to start call between A and B. Currently call can be triggerred with
either first calling A or B number and then the other number after fist
picks up.
I'd like to call A and B at the same time and connect them in call when
possible...
One way would probably be with putting both
- Original Message -
From: Time Bandit [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 20, 2006 4:18 PM
Subject: Re: [Asterisk-Users] Jingle support - can we test the feature ?
we would like to build
Hi,
I'm speech recognition researcher and would like to do some research on
recognition robustness in echo distortion of speech signal. Since VOIP is
becoming wide spread, I'd like to simulate (one or more) common echo
distortions that mostly appear in voip communications ? Any example, FIR
Hi,
we've connected Sphinx4 through eagi script (modified eagi example) to
Asterisk. Users can now say their wishes - but for gradual evolution we
would like to provide older way of DTMF navigation too - can we recognize
DTMF while reading sound in eagi ?
Any advice or examples ?
Thanks in
Hi,
we've proof of conecpt system for speech recognition on Asterisk. We would
like to send results of recognition back to user in standard way.
Currently we're considering using sendtext command and it works with
Firefly. But I'm curious what soft or hard ip phones that can connect to
Hi,
I'm banging my head on compiling bristuff modules for Suse 10.0 with kernel
:
Linux laps1 2.6.13-15.7-smp #1 SMP Tue Nov 29 14:32:29 UTC 2005 x86_64
x86_64 x86_64 GNU/Linux
and Asterisk 1.2.1-BRIstuffed-0.3.0-PRE-1f .
I get this :
laps1:~/Voipy/1.2.1/bristuff-0.3.0-PRE-1f/zaphfc #
Hi,
we've connected Sphinx4 through eagi script (modified eagi example) to
Asterisk. Users can now say their wishes - but for gradual evolution we
would like also to provide older way of DTMF navigation too - can we
recognize
DTMF while reading sound in eagi ?
Any advice or examples ?
Thanks
Hi,
I have Asterisk connected to BRI interface in parallel to my ordinary ISDN
phone. Can I make internal calls between those two without going through
telco provider and taking both voice channels ?
Thanks in advance,
regards,
Rob.
___
Hi,
I had the same problem... I've solved it by recording desconnect tone line
is sending and then do frequency analysis and then you can specify
custom disconnect tone on sipura 3000 configuration Procedure is
described in more details on voxilla web page..
HTH,
regards,
Rob.
Hi,
I have following setup : PBX - Voxip from Parlay -PRI- Asterisk
-SIP- SIP IP GSM Gateway (2n)
on outgoing call from pbx through Voxip and to IP GSM gateway : latter only
responds with SIP session progress but no SIP Ringing message when
connection starts to ring, so Voxip is
Hi,
we have following setup : PBX - Parlay -ISDN PRI- Asterisk -SIP- GSM
Gateway
Call comes from PBX through Parlay to Asterisk and it routes it over SIP to
GSM gateway. GSM gateway gives back call progress (it takes some time to
ring or get through), but this info won't get back to Parlay
Hi,
we're having quite some problems with new hardware we're testing - Parlay
Voxip ISDN-SIP gateway...
So we're curious if anyone is using this in connection to Asterisk and what
are experiences on this HW ?
Thanks in advance,
regards,
Rob.
Hi,
I've found quite some docs on this, but many of them deprecated...
I'm curious what is the latest window messenger version that works as
registered client to Asterisk... I've tried 4.7, but it registers only if I
leave password empty.
Am I missing something or is there any better way to
Hi,
I'm trying to setup Sipura to work with Siemens Combiset 1009 on PSTN line
(GSM gateway - produced for germany)..
I have two problems:
- Sipura doesn't detect Caller ID
- Sipura doesn't detect hangup condition
I have 3.1.7(GWg) firmware on Sipura and Asterisk 1.0.9...
Anyone has
Hi,
I guess you know this project, but just in case:
http://jivesoftware.org/asterisk-im/
IMHO, Egroupware would be best groupware solution to start on, but they have
little interest in doing that (searching their mailing list for voip
returned 2 hits...).
We will gradually start working
Hi,
I have Grandstream 100 as only ever present extension for my Asterisk AMP
home setup. Incoming call comes to ring group and then proceeds to
voicemail.
But Grandstream 100 occasionally loses registration (have anyone found any
solution to this ?) and then AMP's dialparties.agi won't
Tole spada v DTMF zgodbo...
- Original Message -
From: Ryan [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, October 23, 2005 6:35 AM
Subject: Re: [Asterisk-Users] DTMF detection
On Sat, Oct 22, 2005 at 12:53:39PM -0600, Ryan exclaimed:
snip
I just setup
Sorry, went on wrong address
Regards,
Rob.
- Original Message -
From: Robert Rozman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, October 29, 2005 9:22 AM
Subject: Re: [Asterisk-Users] DTMF detection
Hi,
I have Asterisk between PBX and telco line. PBX is reporting number in
overlap dial manner.
I'd like to early connect to telco line as soon as I get for instance two
numbers, that distinguish telco calls. But the problem is if I receive 3
numbers at once, then two numbers dialplan rule
Hi,
we have strange problem on our new card. Sometimes it reports all channels
busy, so call cannot be made (it doesn't even appear in log).
We've contacted Digium support, but received no usable answer (they've told
us that this card should work on stable Asterisk version - AFAIK this is
Hi,
this java video softphone claims it can operate with Windows messenger. It's
also mentioned on this web page
http://www.voip-info.org/wiki/view/SIP+COMMUNICATOR
But I couldn't find any more info on how to set it up with Asterisk and how
compatible is with other video softphones...
Hi,
I found E400P quad PRI card quite cheap (749USD):
http://www.govarion.com/product_info.php?cPath=1products_id=2osCsid=68cdd6e3d08754
in comparison to te410p (approx 1500 USD )
http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TE410P
Now newer generation with HW
Hi,
I spot weird behaviour of latest Firefly 3rd party on my laptop. Sometimes
it comes to state that it won't start (hangs on Initializing ) and it
again works after system restart... Didn't yet figured out how to recreate
it.
Any similar experience ?
Also - how can I force Firefly to
Hi,
I'm aware that incoming and outgoing calls are going fine when isdn channels
are involved - caller id properly identifies calling party, so user can call
back
But how to properly handle this for iax, sip calls
I have few questions :
- BTW, what to type for instance in remote
Hi,
I'm using AMP and its dialparties.agi as most important script in system.
I'd like to port configuration to more embedded system, where I don't have
Perl available.
So I'd like to implement dialparties.agi functionality as closest as
possible with dialplan language.
Are there any
Hi,
I'm not sure if DTMF is convenient solution for user that has cellular on
his ear
Regards,
Rob.
- Original Message -
From: Dean Collins [EMAIL PROTECTED]
To: Ed Greenberg [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
- Original Message -
From: Richard Koch [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, July 08, 2005 4:38 PM
Subject: [Asterisk-Users] Speech Recognition
Ed,
Check this out:
http://turnkey-solution.com/asterisk-sphinx.html
That got me up in running in no
Robert Rozman wrote:
I wanted to do this (it's principle I always follow) , but we even
haven't received offer to pay for the stuff (we applied twice for offer
of two cards), so bought where we actually could buy something...
A customer of mine has had the same problem with the Italian dealer
Robert Rozman wrote:
I'm pulling my hair down and getting bold :-) . I have Asterisk
between
Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff
Asterisk)
(hint: spend the extra $$ and support who's written the software!)
Hi,
I wanted to do this (it's principle
Hi,
I'm pulling my hair down and getting bold :-) . I have Asterisk between
Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff
Asterisk)
I'm trying to do just plain transfer of call from pbx to ISDN through
Asterisk...
It seems like PBX hangsup, when call is
Hi,
I'm getting unreliable dtmf recognition (it works fine for 4-5 digits,
errors (duplicates) on more), when transferred inband from gsm gateway to NT
port of quadbri under bristuffed Asterisk.
Since Asterisk is claimed to have good dtmf recognizer, I suspect there are
some settings to
Hi,
I wonder how I could dimension Asterisk system that will be used solely as
callback server :
- when user calls it registers ring, hangup and calls back - it gives him a
dial signal and calls dialed number on another ISDN channel out
that means plain transfer between two ISDN
alle 19:54 +0200, Robert Rozman ha scritto:
Hi,
I'm pulling my hair out, cause cannot connect to EuroISDN BRI in Italy
with
octobri card from Beronet. I use bristuff and have following
zaptel.conf...
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
#
# First come the span definitions
Hi,
I'm pulling my hair out, cause cannot connect to EuroISDN BRI in Italy with
octobri card from Beronet. I use bristuff and have following zaptel.conf...
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
#
# First come the span definitions, in the format
# span=span num,timing,line
, 6 Jun 2005, Robert Rozman wrote:
I'd like to use DISA properly for my case - I'd like to handle it right,
if
user when in DISA doesn't dial any number - how does Asterisk return from
DISA cmd ?
The file app_disa.c is hardwired to hang up the call if too many incorrect
passwords are attempted
Hi,
I'd like to use DISA properly for my case - I'd like to handle it right, if
user when in DISA doesn't dial any number - how does Asterisk return from
DISA cmd ?
I'd like to dial some default number if user doesn't dial anything or give
him some message - but I don't know what gets
number)
exten = s,2,DigitTimeout(5) ; Vhodni
exten = s,3,ResponseTimeout(20) ;
exten = t,1,DBget(temp=DYNAMIC/${CALLERIDNUM})
exten = t,2,DBdel(DYNAMIC/${CALLERIDNUM})
exten = t,3,Dial(Local/[EMAIL PROTECTED]/n)
exten = t,102,Goto(from-pstn,s,1) ;
On 6/5/05, Robert Rozman [EMAIL PROTECTED
it was called from - but
I'd like to continue with other actions in dialplan - how to do that ?
Regards,
Rob.
On 6/5/05, Robert Rozman [EMAIL PROTECTED] wrote:
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Hi,
we've purchased new Beronet Octobri card and have problems loading modules
for stock Asterisk for Debian Sarge (it has bristuff patches in it).
when loading qozap it says that no multibri card was found although lspci
shows it... There were quite some rumours about bristuff not liking
?
Robert Rozman wrote:
when loading qozap it says that no multibri card was found although
lspci shows it... There were quite some rumours about bristuff not
liking other than junghanns cards, but don't know if something
happened
http://www.beronet.com/download
Hi,
I'm getting unusable DTMF detection with DISA on incoming ZAP channel
(bristuffed *) on quadbri from GSM gateway. DTMF detection works ok in
normal ISDN incoming line.
How can I check what's going on ? What settings to check ?
Anyone with more experience on such scenarios ?
Thanks in
) on Bristuffed * and quadbri
Robert Rozman wrote:
I wonder if I can hide caller id for just certain users. Can I override
caller id setting for show or hide on the fly from dialplan ?
Did you try setcallerid()?
--
I tried but this will work if calling internal line. I'm after dynamically
hiding caller
Hi,
I wonder if I can hide caller id for just certain users. Can I override
caller id setting for show or hide on the fly from dialplan ?
Thanks in advance,
regards,
Rob.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi,
I'm finding long timeout before DISA really calls extension user entered
annoying. I wonder what workarounds are you using for this ?
Playtones is one possibility , but it won't stop when user starts entering
numbers...
Regards,
Rob.
___
Hi,
I'd kindly ask if anyone can provide working configuration examples for
Asterisk-Fritz-mISDN combo.
Thanks in advance,
regards,
Rob.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi,
I'm trying to setup one of free low bandwidth codecs for Grandstream (ilbc,
g726, ...), but with ilbc I just hear engine running in handset.
Is anyone using ilbc sucessfully with Grandstream? Quality ? Any other
alternative ?
I use Bristuffed Asterisk
Thanks in advance,
regards,
Rob.
Hi,
I have problem with Quadbri and bristuffed Asterisk - I guess this is only
configuration trick. I'd like Asterisk to respond only to 1 number on BRI
interface and do nothing on other. Right now, even if I leave out that
number in incoming context, Asterisk takes out and rejects call as
Hi,
I'm trying to make a OH323 call with Grandstream, Asterisk and H323 voip
provider. Everything seems to work fine, except callerid is set to some
value for all calls (10100), despite setcallerid statements in Asterisk.
Are there any special considerations or tricks to get this working ? I've
Hi,
I always thought that if there is no called extension in context, then 's'
extension is started (I use latest bristuffed Asterisk)
I have context 'from-isdn' :
[from-isdn]
exten = s,1,Wait,2
exten = s,2,NoOp(ISDN call from outside ${CALLERID}: Name: ${CALLERIDNAME},
Number:
before doing a modprobe -r qozap
On Apr 6, 2005 4:45 PM, Robert Rozman [EMAIL PROTECTED] wrote:
uname -a
Hi,
I'm using latest Bristuffed Asterisk under Suse 9.2 and upgraded kernel.
Everything seems to be working fine, except crash when removing qozap
with
modprobe -r qozap.
Hi,
I'm still getting
- Original Message -
From: Eric Wieling aka ManxPower [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Monday, April 11, 2005 9:43 PM
Subject: Re: [Asterisk-Users] Why 's' doesn't take over unknown
Hi,
I'd like to setup delayed dial under Asterisk. That means that at the caller
side I set up number *YY and call Asterisk PBX (XXX... is number of
Asterisk PBX, * means pause (2 secs), YY is internal number).
Has anyone experience with receiving such calls ? How should I setup
- Original Message -
From: Mick Hastings [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, April 08, 2005 1:07 PM
Subject: [Asterisk-Users] Re: Delayed dial under Asterisk ?
Hi Robert,
I just set this up today for dialing international using a calling card
account.
uname -a
Hi,
I'm using latest Bristuffed Asterisk under Suse 9.2 and upgraded kernel.
Everything seems to be working fine, except crash when removing qozap with
modprobe -r qozap.
Any hint what's wrong ?
Thanks in advance,
regards,
Rob.
Linux voip 2.6.8-24.13-smp #1 SMP Fri Mar 18 10:19:42 UTC
Hi,
I'm trying to go route some of Asterisk users already proposed for Asterisk
minimal system. I've started from Suse Rescue system image - I've put it
into HD partition. But since rescue is spawned from working system it has
empty /boot directories and is not directly bootable if put on HD.
Hi,
I have strange behavior on outgoing calls (I can receive calls and I can
make outgoing calls to ISDN lines ok (035778421 and 5778421 for instance -
03 is area code).
I use latest bristuffed Ast. under Suse 9.2.
My zapata.conf and zaptel.conf are at the end of mail.
Any help, advice - I
Hi,
I'm confused whether I setup PC for Asterisk right or not. Module qozap is
alone (yet not sharing) in interrupt 209 (isn't this too high for native
interrupt).
Is this good state or not? If not, how to setup better ?
Thanks in advance,
regards,
Rob.
voip:~ # cat /proc/interrupts
Hi,
sorry for my h323 dumbness. VOIP provider terminates H323 calls - it can be
used as gatekeeper or gateway (they claim so). What option and what setup is
best to connect Asterisk to this provider ?
Any working examples ?
Thanks in advance,
regards,
Rob.
Hi,
I'm trying to connect quadbri between powered ISDN phone and ISDN line:
ISDN ---1--- TE - * - NT --2-- Phone
I use quadbri, suse 9.2 and latest 0.2.0-RC7k bristuff. I've used sample
settings provided with package, but do get strange error (I think that I
have wrong setting for P2P or P2MP
Hi,
I had problems described in another thread so went from a start, but now
have problems when loading qozap module. I get :
# insmod qozap.ko ports=9
insmod: error inserting 'qozap.ko': -1 Unknown symbol in module
and in /var/log/messages:
module qozap unsupported by SUSE/Novell, tainting
Hi,
I'd like to setup oneway connection - so asteriskB can place calls on
asteriskA and be safely authenticated with rsa keys. I just don't get any
response on asteriskA.
I've generated pair of keys: name.key, name.pub and put them on both servers
- is it right to only have name.key on
Hi,
I'd kindly ask if anyone can provide working example of RSA authentication
and IAX ?
Thanks in advance,
regards,
Rob.
- Original Message -
From: Robert Rozman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent
- Original Message -
From: Robert Rozman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, March 25, 2005 2:13 AM
Subject: Re: [Asterisk-Users] RSA interasterisk IAX problems ?
Hi,
I'd kindly ask if anyone can
Hi,
we've installed te110p with Suse 9.2 on Siemens primergy. We're connecting
to voxsteam i60 to test PRI interface.
We have problems, after reboot sometimes it goes green, otherwise stays
blinking red.
How could we debug this situation ?
Are there any common advices what to check ?
Are CVS
Hi,
I've compiled Asterisk cvs stable (few days ago) unde Suse 9.2 without any
problems. We're using te110p and wcte11xp module that is autoloaded by Suse
9.2.
Card goes green after reboot, but this meesages appear in logs:
Mar 22 11:28:51 linux kernel: Zapata Telephony Interface Registered on
- Original Message -
From: Robert Rozman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, March 19, 2005 12:36 AM
Subject: [Asterisk-Users] Te110P initial installation problems ?
Hi,
thank you for last info
primergy econel server under Suse 9.2 freezes
hard... Is this normal behaviour ? Do we have any debug options on loading
module ? How to track this problems ?
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
loadzone=fr
defaultzone=fr
Thanks in advance,
regards,
Robert Rozman
Hi,
call me stupid, but cable is not delivered with te110p. What cable can I use
to connect to telco NT PRI line ? Is it same as for BRI interface ?
Thanks,
regards,
Rob.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi,
I'd also like to see alternative op_style.cfg. Can we establish some place
to share them ? I've also one with smaller buttons (but will have to count
them :-) ...
Regards,
Rob.
- Original Message -
From: Nicolás Gudiño [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Hi,
I'd like to know which version of Asterisk performs best and most stable
with TE110P.
I don't need any other features (it'll just terminate interasterisk calls
without any other feature - so there is no need for CVS Head features or
? ).
Any info on setting up secure interasterisk IAX
Hi,
could anyone provide any working examples? I have same problem, I do
redirect to s-${Dialstatus} when making outgoing calls and I also get s-
into CDRs.
Thanks,
Rob.
- Original Message -
From: William M. Sandiford [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Hi,
I'm integrating Asterisk to legacy PBX via ISDN router. If I want to call
legacy PBX internal extension I need to specify MSN as caller id and local
number to call.
I wonder if I can cal out via Fritz CAPI on any msn I want, or are there
any limitations - I've read something about 5 MSNs
I don't know if this is still true, but Iax clients had problems when you
check them with qualify (set latter to no)...
HTH,
Rob.
- Original Message -
From: Anton Krall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent:
Hi,
I've updated my Asterisk 3 times with :
cvs checkout -r v1-0 zaptel asterisk asterisk-addons
and then do
cd asterisk
make clean make make install
make samples
make progdocs
and then when I run Asterisk I get :
Asterisk CVS-v1-0-02/11/05-01:46:25, Copyright (C) 1999-2004 Digium.
Is
- Original Message -
From: Adnan Ahmed [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, March 04, 2005 1:28 AM
Subject: Re: [Asterisk-Users] Wrong CVS version ?
you are compiling in wrong sequence first zaptel
Hi,
I'm trying to implement dynamic routing of incoming calls to local extension
if previous outgoing call was unanswered.
But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to
's-NOANSWER'. I guess this is normal, but I don't understand why ? How to
workaround on this one ?
Thanks in
:23 +0100, Robert Rozman [EMAIL PROTECTED]
wrote:
Hi,
I'm trying to implement dynamic routing of incoming calls to local
extension
if previous outgoing call was unanswered.
But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to
's-NOANSWER'. I guess this is normal, but I don't understand
04:11 pm, Robert Rozman wrote:
Hi,
I'm trying to implement dynamic routing of incoming calls to local
extension if previous outgoing call was unanswered.
But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to
's-NOANSWER'. I guess this is normal, but I don't understand why ? How
Hi,
I wonder if I can securely authenticate two Asterisk servers with IAX
connection. I know for RSA authentication for IAX2 channel, but that seems
to be meant for peer authentication...
Has anyone done RSA (or any other secure way) authentication between two
Asterisk servers ? Any example ?
- Original Message -
From: Michael Bielicki [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, February 26, 2005 5:01 PM
Subject: Re: [Asterisk-Users] Digium BRI or quad BRI
Hmm don't know about you but I
Hi,
I guess I'd need to run Beronet quad and octo bri cards under bristuff to
get zaptel features (echo canceling, timing source) Am I right or could
I achieve this also with chan_misdn - their native driver ?
Running bristuff on Beronet cards is unsupported. Has anyone succesfully run
Hi,
I had similar questions. I've emailed few questions and got no response in
10 days from junghanns. So I decided to try Beronet cards (they will arrive
shortly). I just cannot imagine to have support from someone that is not
able to answer few simple technical questions about their cards in
Hi,
I mistakenly posted this to Dev list
I wonder if I use quadbri or octobri cards to insert Asterisk between ISDN
PBX and ISDN line - if power of Asterisks fails - will those card connect
PBX directly to ISDN line ? If not are there any other simple switching
devices, that would do this
Hi,
I'd like to establish way to exchange data between two remote Asterisk
server. Something like call over IAX and send some structured data.
Any advice ?
Regards,
Rob.
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