As I am trying to compile a fresh copy of the current svn release of
Asterisk 1.2 for a UK system with a combination of X100 and TDM
cards, can a kind soul email me the CLID patches for 1.2?
Many thanks
Vassilis
___
--Bandwidth and Colocation
On Fri, Dec 02, 2005 at 10:06:09PM -0800, Geo wrote:
Message not sent
Right, not the comma separator
I still have 2 small questions:
1) I have the following warnings:
WARNING[2943]: chan_zap.c:10916 setup_zap: Ignoring :callreturn
WARNING[2943]: res_musiconhold.c:124 spawn_mp3:
On Sun, Dec 04, 2005 at 08:14:07AM +, Vassilis Konstantinou wrote:
As I am trying to compile a fresh copy of the current svn release of
Asterisk 1.2 for a UK system with a combination of X100 and TDM
cards, can a kind soul email me the CLID patches for 1.2?
As announced before in this
On Sat, Dec 03, 2005 at 09:28:57AM +0100, Karsten Wemheuer wrote:
Hi,
On Tue, November 29, 2005 13:50 Francesco Peeters wrote:
BTW: BRIstuff is not included by default as it breaks PRI support.
Asterisk is already set up to use zap, so that is easy...
As far as I know, BRIstuff is not
Hi all,
I was testing the FXO system from sipura 3000 with asterisk PERL AGI.
But when we hangup the FXO phone the channel is not disconnecting and
the destination is continue ringing. even if we try to press the
disconnect button for destionations after some seconds again it start
to ringing.
Thanks Tzafrir,
But those patches do not work with the current 1.2. The asterisk_uk
patch fails on 3 accounts with chan_zap.c and callerid.h
The zaptel patch seems to be ok.
Vassilis
At 09:00 04/12/2005, you wrote:
On Sun, Dec 04, 2005 at 08:14:07AM +, Vassilis Konstantinou wrote:
On 3 Dec 2005, at 21:41, chawki hammoud wrote:Hi:Thanks for your answer, i tried all possible codecsand the same result the call failed,my asteriskverison is 1.0 ,I asked callshopcompany "the voipprovider" about whats the reason of the failure of thecalls and he said he didnt know whats the
there should be a way in agents.conf to autologoff agents after a while
the do not answer the phone.
l.
In data Sat, 03 Dec 2005 23:48:05 +0100, Vladimir S. Blazhkun
[EMAIL PROTECTED] ha scritto:
-- Called 1101
-- Agent/1101 is ringing
-- Got SIP response 480 Temporarily Unavailable
HI:
i tried to write asterisk -rv on console but no
such command message
appears,but when i make show version it gives me
this:
Asterisk CVS-v1-0-08/22/05-18:56:48 built by
[EMAIL PROTECTED] on a i686 running Linux.
--- tim panton [EMAIL PROTECTED] wrote:
On 3 Dec 2005, at 21:41, chawki
How can I configure asterisk to switch from my hardphone which is always
up and online, as soon as I register with my notebook's softphone on the
asterisk server?
The target is to receive all calls destinated to my hardphone on my
softphone when it's online.
Any ideas?
dima wrote:
How can I configure asterisk to switch from my hardphone which is
always up and online, as soon as I register with my notebook's
softphone on the asterisk server?
The target is to receive all calls destinated to my hardphone on my
softphone when it's online.
Any ideas?
I'm
Vassilis Konstantinou wrote:
Thanks Tzafrir,
But those patches do not work with the current 1.2. The asterisk_uk
patch fails on 3 accounts with chan_zap.c and callerid.h
The zaptel patch seems to be ok.
Worked fine for me...
Are you sure you used the 1.2.0 patch?
Tony
hi all, some days back i mailed to group abt my error on asterisk. Now also i m getting same error : cannot find extension context 'from-sip' . I tried DEFAULT context also. but at that time errot remains same: cannot fined extension context 'default'. I think problem is that it is not
On 4 Dec 2005, at 13:33, chawki hammoud wrote:HI:i tried to write "asterisk -rv" on console but "nosuch command" messageappears,but when i make "show version" it gives methis:Asterisk CVS-v1-0-08/22/05-18:56:48 built by[EMAIL PROTECTED] on a i686 running Linux.Weird. IAX should be fine with that
For starters, please provide a more descriptive subject for your
message. Something like cannot find extension context 'from-sip' would
be better than nothing.
Also see my reply inline.
On Sun, Dec 04, 2005 at 07:48:11AM -0800, Tejas Shah wrote:
hi all,
some days
I have tried Billion HFC-PCI and Eicon Diva cards for ISDN 2e in the UK
but both seem to have drawbacks/advantages.
I need to build a new Asterisk box for my tiny business (1 x ISDN2e from
BT and 1 x IAX link from Gradwell)
Is anyone prepared to go out on a limb and say which card they
Simon Faulkner wrote:
I have tried Billion HFC-PCI and Eicon Diva cards for ISDN 2e in the UK
but both seem to have drawbacks/advantages.
I need to build a new Asterisk box for my tiny business (1 x ISDN2e from
BT and 1 x IAX link from Gradwell)
Is anyone prepared to go out on a limb and
Hi,
Why do changes to musiconhold.conf require a reboot. Also if I put mp3's
into the /var/lib/asterisk/mohmp3 directory will the be played if I use
the -r option? Using Asterisk 1.2 and have run the make config in the
/usr/src/asterisk-addons directory.
Thanks
I'm thinking of Installing [EMAIL PROTECTED] on my PC, to contol and route calls
to each of my children's computer via SoftPhone X-lite. I downloaded the ISO
image, and familiar with the process to install Asterisk now. However I
don't have a digium modem.
Can I use any regular phone modem
hi all,Can anyone tell me how i can remove (uninstall) asterisk 1.2 from Red Hat 9. Also pls tell me which version of asterisk is most suitable for making VoIP gateway on Red Hat 9.Thankstejas
Yahoo! Personals
Single? There's someone we'd like you to meet.
Lots of someones,
Faris Raouf wrote:
Simon Faulkner wrote:
I have tried Billion HFC-PCI and Eicon Diva cards for ISDN 2e in the
UK but both seem to have drawbacks/advantages.
I need to build a new Asterisk box for my tiny business (1 x ISDN2e
from BT and 1 x IAX link from Gradwell)
Is anyone prepared to go
John Daragon wrote:
I'd second that. For a single ISDN2e connection the AVM Fritz card is
really hard to beat/
Yeah, single is the key word there. I have 2x ISDN2 (OnRamp2 in
Australia) and the AVM Fritz cards are a nightmare. Replaced the two
cards with an Eicon Diva V-4BRI (so I have two
You can simply put then in order in your dial plan:
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]
exten = _1NXXNXX,2,Dial(SIP/[EMAIL PROTECTED]
and so on. if carrier1 returns an error, * will dial out using carrier2.
SciptHead
On 12/2/05, Max Clark [EMAIL PROTECTED] wrote:
Hi all,I
Hi:
Sorry,but i dont know what ethereal is,and for my
asterisk version the iax is good on it because i made
a lot of succesful iax connections with many voip
providers like sixtel,voipjet...
--- tim panton [EMAIL PROTECTED] wrote:
On 4 Dec 2005, at 13:33, chawki hammoud wrote:
HI:
i
With this setup Asterisk will also dial out using carrier2 even if the
call via carrier1 does NOT fail. It will dial out via carrier2 if the
number is busy, disconnected, answered then ended, etc. This is BAD BAD
BAD.
Try looking at std-exten for an example of how to handle stuff when Dial
I have a Aastra 9133i phone and would like to do a simple test to make sure
everything works. I already assigned an IP address to the phone (I'm able to
ping it.) I have Asterisk running (installed Asterisk and Zaptel only) but not
configured. I don't have a FXS/FXO card yet but I would like
On Mon, 5 Dec 2005, Avi Miller wrote:
John Daragon wrote:
I'd second that. For a single ISDN2e connection the AVM Fritz card is
really hard to beat/
Yeah, single is the key word there. I have 2x ISDN2 (OnRamp2 in Australia)
and the AVM Fritz cards are a nightmare. Replaced the two cards
Is it possible to send data over the D Channel using ZAPHFC?
I'd like to send data between three servers (only one is live yet, but I
am thinking ahead and trying to plan...) to verify that each of their ISDN
connections is live.
Ie:
1 sends to 2
1 sends to 3
2 sends to 1
2 sends to 3
3 sends
Armin Schindler wrote:
With 'echo across the board', do you mean a connection bridged between ports
of the 4BRI only? If yes, there shouldn't be any echo when using line
interconnect (CAPI native bridging). If this doesn't work for you, please
let me know.
Heh, no -- I meant there's no (or
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.
Script Head wrote:
You can simply put then in order in your dial plan:
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]
exten = _1NXXNXX,2,Dial(SIP/[EMAIL PROTECTED]
and so on. if carrier1 returns an error, * will dial out using carrier2.
SciptHead
On 12/2/05, *Max Clark* [EMAIL
Hi all,
I was wondering whether the DISA function on the
latest asterisk 1.2 stable release
actually works better than the other prior
releases. Basically the [EMAIL PROTECTED]
version 2.0 BETA 4
I'm using does not recognise the DTMF tones all the time and sometime when
it does, it
UK, London Based DID £1 per month
All number begin with 0208 0xx
If you are interested please email ukdid AT cyber-telecom.net
SIP based and support standard ulaw or alaw.
Unlimited incoming minutes.
For multi channels please email for pricing.
Sam
Single port GSM Gateway support 900 / 1800 GSM mode with external antenna.
Brand new unit and all of them will be tested before dispatch.
Extremely easy to setup and can be used out of the box without any
configuration. So should be good alternatively of phonecell or nokia pbx
etc..
Units are
You have to have a Digium TDM400, a clone X100P (eBay, about $10), or Sipura
SPA-3000. A regular phone modem will not work. Another way is to get a cheap
VOIP account like Broadvoice for $10 a month, then you don't need a regular
phone line for the kids and you can have two calls going at the same
Avi Miller wrote:
John Daragon wrote:
I'd second that. For a single ISDN2e connection the AVM Fritz card is
really hard to beat/
Yeah, single is the key word there. I have 2x ISDN2 (OnRamp2 in
Australia) and the AVM Fritz cards are a nightmare. Replaced the two
cards with an Eicon Diva
You have a stray immediate=yes.
Try this: grep -v ; /etc/asterisk/zapata.conf
Remco Barende wrote:
I just upgraded my config from * 1.0.10 to 1.2
I removed caller ID from my configs because when I try to use CallerID
(new style) on my IAX provider (magrathea) but whenever I try to make a
You don't want immediate=yes
Remco Barende wrote:
On Sat, 3 Dec 2005, Begumisa Gerald M wrote:
On Sat, 3 Dec 2005, Remco Barende wrote:
Whenever I pick up that phone I get on the console:
Dec 3 16:37:36 WARNING[19551]: pbx.c:2347 __ast_pbx_run: Channel
'Zap/1-1' sent into
Better yet get a Telasip account at telasip.com
On 12/4/05, Kerry Garrison [EMAIL PROTECTED] wrote:
You have to have a Digium TDM400, a clone X100P (eBay, about $10), or Sipura
SPA-3000. A regular phone modem will not work. Another way is to get a cheap
VOIP account like Broadvoice for $10 a
[EMAIL PROTECTED] wrote:
We are currently an ISP offering broadband and wireless internet
connections. We are planning to start offering VoIP services to our
current customers. We have decided to use Asterisk as our PBX software.
If I may, I have a multi-part question.
1) If we do use
Hi!
I have been using Asterisk-1.0.3 for quite some time now.My main aim nowadays is to make iax-iax calls for which i am usin DIAX soft phone.The problem is that sometimes the phone doesn`t register and at others it gets out of the registration(after being registere for some time).Can anyonetell
I have 2 Asterisk servers running 1.2.0. One of them is a PSTN
gateway. Currently they are connected using IAX2. I wanted to play
with SIP.
I setup a sip entry (type=friend) in the PSTN gateway box and a sip
entry (type=user) in the second box in order to send calls using SIP
to the
Already HAVE Florz patch installed! :-(
What version of * and BRIstuff are you using?
Strange, sounds like the florz patch has not been effectively applied or
it's broken. I'm using an old version of bristuff :
Asterisk 1.0.9-BRIstuffed-0.2.0-RC8n built by [EMAIL PROTECTED]
on a x86_64
I have configured my SPA-3000 to direct all calls received over PSTN
interface to Asterisk. That is set up via a dial plan with S0 and works
fine. CID information is passed along as well. But I find it impossible
to setup call waiting on the PSTN line. I do subscribe to this service
from my PSTN
Has anybody been able to get call waiting on the PSTN line?
As far as I recall, you will only hear a tone in the audio stream when
a second call comes in. The Sipura does not detect or handle it, but
if you flash the line on the FXS interface after hearing the tone, the
Sipura will forward the
Any ideas on how to correctly set this up?
Try adding authuser= and/or username= to the configuration. Do a SIP
DEBUG and see what peer asterisk looks for when trying to authenticate
the INVITE. It probably can't find the right peer; authuser on the
initiating end should help in this case.
Hi all
I am a newbie to the asterisk. I just installed asterisk server and two
X-Lite softphones. I allready configured sip.conf and extension.conf.
Now when i call from one softphone to other , sip signaling is
going perfect. Both phone are in ringing mode. But i can't able to hear
ring. When i
Hi all,
I posted about this briefly, but haven't gotten a response as yet.
I've checked further, and it seems that the 'Wrapup Time' option just
isn't having any effect. We have agents logging into the queue using
[queueno]* and [queueno]** to log out. The calls are all processed fine,
but it
Any suggestions/comments on companies that provide hosted voicexml
solutions that work with SIP? Seems like a new enough market that the
pricing is pretty high and the number of vendors that will work with
smaller volumes is low. So far Voxeo is at the top of my list.
Chris
Hi, I am getting this error when compiling asterisk `ls *.c`: unrecognized optionh -DBUSYDETECT_MARTIN `ls *.c`Usage: /bin/sh [GNU long option] [option] ... /bin/sh [GNU long option] [option] script-file ...GNU long options: --debug --dump-po-strings --dump-strings --help --login
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