How do I make a user dial a passcode if he wants to make an international call?
This electronic message contains information from BOSH Global Services which
may be company sensitive, proprietary, privileged or otherwise protected from
disclosure. The informatio
How do I make a user dial a passcode to make calls through asterisk?
We would like to place a phone at a client's location for our employee but are
afraid it may get abused by the other workers.
This electronic message contains information from BOSH Global Serv
I would like to auto provision the SPA504G phones we have in the office. What
is the best method for this task?
Felix
This electronic message contains information from BOSH Global Services which
may be company sensitive, proprietary, privileged or otherwise
: Call from
'' to extension '90111235551212' rejected because extension not found.
Felix
This electronic message contains information from BOSH Global Services which
may be company sensitive, proprietary, privileged or otherwise protected f
uch about it on the net. anyway, if it's well known:
what would be the downside of just (silently, implicitly) taking the
right adress? like when when resolving the peer-host, take a look into
the routing table...?
regards
felix
--
___
ses and routing.
this is about asterisk on sid, version 1:1.8.13.0~dfsg-1, but of course
i'm going to switch to whatever you might suggest.
regards and thanks
felix
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On Behalf Of *Felix Dong
> *Sent:* Tuesday, March 15, 2011 4:19 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] signal amplified by asterisk
>
>
>
> Hi there,
>
>
>
> i called one asterisk server from another asterisk ser
er the name is”.
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong
> *Sent:* Tuesday, March 15, 2011 4:29 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
how can I correct it to use the right Microsoft PCM wav? Where can I find
the File foo?
Thank you!
best regards
Felix
2011/3/15 Danny Nicholas
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com
fault rxgain and txgain (with another words: no
setting these values)?
Thanks a lot.
best regaud
Felix
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lto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong
> *Sent:* Monday, March 07, 2011 6:07 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Loudness of recorded wav-audio
>
>
>
> I tried to aj
-
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong
> *Sent:* Friday, March 04, 2011 8:55 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-u
uce the incoming volume by 4 decibels. You’ll have to do a “sip reload”
> for this to take effect.
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
> Sent: Friday, March 04, 2011 8:33 AM
> To: Asterisk U
Thank you! How can I reduce the RXgain?
Am 04.03.2011 um 15:21 schrieb "Danny Nicholas" :
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
> Sent: Friday, March 04, 2011 2:31 AM
> To: asteri
a lot.
best regards
Felix
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asterisk
ists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong
> *Sent:* Wednesday, February 16, 2011 6:22 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] function Echo() doesn't work
>
>
>
:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong
> *Sent:* Wednesday, February 16, 2011 5:33 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] function Echo() doesn't work
>
>
>
> * == Using SIP RTP CoS mar
; asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong
> *Sent:* Wednesday, February 16, 2011 5:33 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] function Echo() doesn't work
>
>
>
> * == Using SIP RTP CoS mark 5*
ported codec or send CLI log for
> further troubleshooting.
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong
> *Sent:* Wednesday, February 16, 2011 5:14 PM
> *To:* Asterisk Users Mailing List - Non-Co
.@lists.digium.com] *On Behalf Of *Felix Dong
> *Sent:* Wednesday, February 16, 2011 4:48 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] function Echo() doesn't work
>
>
>
> Hi guys,
>
>
>
> the function Echo() did work on CAPI,
Hi guys,
the function Echo() did work on CAPI, but doesn't work for SIP connection.
Can anybody help?
thanks a lot.
best regards,
Felix
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New to Ast
Hello,
can anyboby tell me, how can I disable the echo cancellation for sip?
thx a lot...
best regards,
Felix
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how did you increase it?
Am 16.02.2011 um 00:11 schrieb Hans Witvliet :
> On Tue, 2011-02-15 at 18:06 +0100, Felix Dong wrote:
>> Hello,
>>
>>
>> could I adjust the Rx and Tx gains for SIP and CAPI? If it is
>> possible, how should I do it?
>> Thanks a
Hello,
could I adjust the Rx and Tx gains for SIP and CAPI? If it is possible, how
should I do it?
Thanks a lot.
best regards,
Felix
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New to Asterisk
How can I use the application Monitor() in the Python AGI skripts?
Thanks a lot.
best regards,
Feilx
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Hallo everybody,
I got a question to asterisk 1.6. Is it possible to playback a Audiofile in
uplink and to record the downlink channel in another Audifile at the same
time?
If it is possible, how should I do it? Please explain it.
Thank you for your help to my thesis!
best regards,
Felix
en with the newer libpri) makes
things work again. So I'm almost sure that it's somewhere in the
DAHDI/vzaphfc code.
Felix
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New to Asteris
Hi,
thank you very much. this solved my problem.
greets
felix
Am 31.03.2010 um 22:52 schrieb Mark Michelson:
> Felix Tiefenthaler wrote:
>> Hi list,
>>
>> can anyone tell me how to reset/delete all modifications (personal
>> greeting message, personal name, ...) I
Hi list,
can anyone tell me how to reset/delete all modifications (personal
greeting message, personal name, ...) I made in my voicemail?
I just want to get the default automatic computer messages back.
thank you!
greets
felix
sing OpenVZ but when using
KVM, XEN, ESX, ...
Please tell me your opinion. I definitely want to run the Asterisk via
virtualization - so we have to find a solution for this ;-)
Thank you very much!
felix
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-- Bandwidt
si, al principio, de ahora en adelante en todas la instalaciones qe hago codificamos ese parametroOn 6/2/06, Anton Krall <
[EMAIL PROTECTED]> wrote:
Muchas gracias Felix, voy a probar a ver que tal
jala.
Tu tuviste ese miusmo problema?
From: [EMAIL PROTECTED]
[mailto:
Cambiando un timer que existe en el archivo mfcr2.c
La variable DEFAULT_T1 tiene el valor 5000, incrementalo a 2, compilas, instalas y listo…
mas o menos en la linea de codigo 102…
actual
#define DEFAULT_T1 5000
despues
#define DEFAULT_T1
Our friend record all the annoucements, so we dont have disparity problems...
She can also record the annoucements in Spanish...On 3/13/06, Alexander Lopez <[EMAIL PROTECTED]> wrote:
Placing plasic bag over ones head whist listnting to Allison Promptsmakes the 'breathy-ness' go away.
> -Origi
Hi Waldo,
We have a friend who does the recording for us. She is very good, if you are interested contact me.
FelixOn 3/9/06, Tom <[EMAIL PROTECTED]> wrote:
I have one that we work with. Digium also does this with Allison.Contact me off list for more info.TomAt 05:19 PM 3/8/2006, you wrote:>Can
f.
>
We are using 4 BRI channels in Germany connected to a AVM C4. We are using
PTP (Anlagenanschluss). You will need another config for PTMP
(Mehrgeräteanschluss).
I hope this will help you a bit.
Regards
Felix
/etc/asterisk/capi.conf
;
; CAPI config
;
;
[general]
nationalprefix=0
inter
no podria decirte, porqe tengo problemas con los scripts de email2fax y
Asterfax...espero resolverlos pronto y verificar el correcto envio de
faxes...On 12/21/05, Jorge Cisneros <[EMAIL PROTECTED]> wrote:
gracias Felix por el tip, ya lo hice y si funciono todo bien. tengo
otro problema no
Es un timeout...necesitas incrementarlo...en la libreria de unicall existe un archivo qe se llama mfcr2.c...
#define BLOCKING_RELEASE_TIME 450
#define
ANSWER_GUARD_TIME
100
#define
DEFAULT_T1
5000 <-Dale una valor mas alto...2 p
Hi, I´ve recently installed my first Asterisk and it´s working. I can only
make outbound calls trough internet. I was willing to record the phone calls
in files maybe with wav or gsm extension. Can someboy help me a little with
this?
Thanks
Felix
you need a stun server on asterisk side...I use the one that vovida.org provides...it is very easy to install and configure...On 11/23/05,
jeffery chen <[EMAIL PROTECTED]> wrote:
Asterisk server behind NAT,and SIP clinet behind another NAT.SIP.conf have set NAT=yes,SIP client can register with Ast
two things...verify the content of your /etc/ld.so.conf file must have
the path included(/usr/local/lib) and recompile and install...first
span and then asterisk...On 11/22/05, Dominik Simon <[EMAIL PROTECTED]> wrote:
Hi all,today I installed asterisk 1.2stable and than spandsp-0.0.2pre21 withrxfax
problably you need a stun server...vovida.org works fineOn 11/15/05, Matt Riddell <[EMAIL PROTECTED]
> wrote:John Biundo wrote:> I can't forward 1-2 with my router. So I used
rtp.conf to> narrow the band of ports down to something like 14000-14030 and> forwarded those ports That seems to
You need a Stun Server...vovida.org works for meOn 11/11/05, Enrique Leon <[EMAIL PROTECTED]
> wrote:Second postI have installed Asterisk on SuSE 10.0 with an active firewall/NAT filter.
The server has connection to my own Intranet (private IP) and to InternetEverything works well for clients behin
The Asterisk I biult only does outbound calls, and it do them by LAN, I
don´t have any special hardware. Please help with the Dial Plan.
Thanks a lot
Felix Amaral
I.T. - Information Technology
Grupo PyD S.A.
Reconquista 1011 4º (C1003ABU)
Cap. Fed.- Argentina
TeL: +54-11--4800 Ext. 555
Hi, I´ve just installed an Asterisk Server on a Fedora Core 4, and made it
work between diferrent extensions in the office and now I need to make it
work on calling outside the office and I think I need a Dial Plan, can
somebody help me a little with this?
Thanks a lot
_
I'll throw in a few requests as well-
A "pause" feature.
The ability to mark a recording "urgent".
The ability to change the prompt features around and edit voicemail prompts,
recording abilities, while retaining defaults and customizations for
different extensions.
I'm still studying AGI's
I have also been looking for a way to customize voicemail (I want to add a
"pause" feature and change the promps). I have come to the same conclusions
as to where to do it, but have not yet created a solution. I have found
this posting/forum which gives insight into modifying the "app_voicemai
The people who have been documenting Asterisk have been working on a book
for the last few months, it has been published by O'reilly (Asterisk-The
Future of Telephony)and is just now finding it's way into the major
bookstores, listed under Open-Source at Barns&Noble.
While it will not answer e
I want to have customers make payments by keying in their cc#'s.
I can see it's possible, I just want to know if anyone out there is doing
this and what financial institutions are supporting Asterisk PBX's.
So far I have found a few leads but would like to check here at the same
time.
Thank
a lot for the reply.
When exactly are the /dev/zap/ctl files supposed to be created?
I have been trying different things to see if they'll show up but known
when to expect them to show up would help a lot.
-jachin
On Sep 6, 2005, at 5:53 PM, FELIX E SKOWRONEK wrote:
I had this problem
I had this problem with White Box Enterprise Linux running the 2.6 kernel.
When I went back to the 2.4 kernel it created the /dev/zap/ctl files. Still
having other issues setting up AMP, but asterisk still recommends the 2.4
kernel.
From: Jachin Rupe <[EMAIL PROTECTED]>
Reply-To: Asterisk
ss
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Success
Once these messages start showing, I must stop my asterisk (stop now)
because the load goes sky high.
I'm using an Asterisk CVS-HEAD.
Looking forward for your
3 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Success
Once these messages start showing, I must stop my asterisk (stop now)
because the load goes sky high.
I'm using an Asterisk CVS-HEAD.
Looking forward for your
ss
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Success
Once these messages start showing, I must stop my asterisk (stop now)
because the load goes sky high.
I'm using an Asterisk CVS-HEAD.
Looking forward for your
ccess
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Success
Once these messages start showing, I must stop my asterisk (stop now)
because the load goes sky high.
I'm using an Asterisk CVS-HEAD.
Looking forward for your
streaming part.
I used Icecast, but without the conferencing part:
exten => 9779619,1,Ices(/home/feklee/asterisk/asterisk-ices.xml)
It works fine!
--
Felix E. Klee
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vailable in a format so that people can
listen to it using XMMS or similar software.
Comments? Would Asterisk fit the bill? Alternatives?
[1] It's Monday's EU Council of Ministers with Software Patents on the
agenda:
http://wiki.ffii.org/
nnel.c:472 ast_channel_walk_locked:
Avoided initial deadlock for 'CAPI[contr1/1429092]/279', 10 retries!
Feb 3 14:04:12 WARNING[23065]: channel.c:472 ast_channel_walk_locked:
Avoided initial deadlock for 'CAPI[contr1/1429092]/279', 10 retries!
Any idea?
Regards
Felix
> > Ja
d enable faxing?
Has anybody experiences with it? If there is a problem why is not kapejod
solving that?
I hope you could help me, I have some really angry customers.
Regards
Felix
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htt
Hi Steve,
I also want the commercial details, so if you can send them to me or put
me in touch with somebody who can it would be very helpfull.
Thank you in advance,
Felix Skwarczynski
Steve Underwood wrote:
Hi Bartosz,
We have a commercial SS7 for Asterisk that is running at a few test
sites
Hello,
take the capisuite
Regards
Feli x
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Robert Rozman
> Sent: Sunday, January 02, 2005 11:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Can
Hi. I am evaluating the installation of ~ 20 extensions and 4 telco lines. The customer asked me to compare costs and features of doing it all with voip phones or using analog phones.
I think that the analog route would involve a T1 card, a channelbank (probably adit 600) and 20 new phones (prob
t; channel 1" says "Offhook" but both incoming and outgoing calls work> fine. I believe this is related to a incomplete fix to bug #2359 but> my limited knowledge does not enable me to track down or fix the> problem.> > > - Original Message -> From: Felix
se there was two lines on that same plug (red/green and yellow/black) and plugging it into a fxo would short them out and make then off-hook.
Michel Belleau
De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Felix PizarroEnvoyé : 24 septembre 2004 11:53À : [EMAIL PROTECTED]Objet :
Hello, everyone
I am having problems with a TDM400 that has 3 fxs modules and 1 fxo. When plug a line from the telco to the fxo module it changes state from onhook to offhook, and of course I can not receive any calls. (When I tried to call from the outside to that line it shows as busy). Coul
I need a cheap platform for installing a tdm400. Could someone tell me if the cheap cpubuilders computer at sams $179 (cbs110l) is pci 2.2 compliance? I ve got a compaq deskpro en 700 that does not seems to be compliant and I need to change it to start developing. Thanks for the help. Computer
I think you should install the openssl and openssl-devel packagesDinesh <[EMAIL PROTECTED]> wrote:
cd ../asterisk# make clean; make installHello when I do a make clean and make install, I get this error message onmy asterisk box.bdb1.a stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv -lssl/u
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
immediate=no
usecallingpres=yes
overlapdial=yes
pridialplan = local
context = Amt595xxx-In
switchtype = euroisdn
signalling = pri_cpe
group = 1
channel => 1-15
channel => 17-31
Regards
Felix Dei
Hello,
it seems that opencall.org is down.
Could anybody send me the instructions and sources for fax? (pm:
[EMAIL PROTECTED])
Thanks
Felix Deierlein
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D: Implicit, PRI Spare: 0,
Exclusive Dchan: 0
<ChanSel: Reserved
< Ext: 1 Coding: 0 Number Specified
Channel Type: 3
< Ext: 1 Channel: 1 ]
-- Processing IE 24 (cs0
Hi,
could you post your capi.conf..
Regards
Felix
> >I would set the MSN's to 855285 and 859609 > >They do not
> usually include the area code.
> >
>
> [local]
> exten => _9XX.,1,Dial,CAPI/855285:bBYEXTENSION:1
> exten => _9XX.,2,Congestion
>
e Dchan:
0<
ChanSel:
Reserved<
Ext: 1 Coding: 0 Number Specified Channel Type:
3<
Ext: 1 Channel: 1 ]-- Processing IE 24 (cs0, Channel
Identification)
With kind regards
Felix
Deierlein
Hi all,
are you able to see incoming calls at the isdnlog? I have guessed I have a
problem
with the capi/isdn/card itsself and not really with asterisk.
Felix
> Thanks I will give that a try.
>
> Looks like this may need a bug report? We are all getting the
> same errors.
>
I hope, that you could help me...
Thanks
Felix Deierlein
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Hello,
do you have overlapdial=yes in your zapata.conf?
Felix
> When I call a PBX system and enter digits, Asterisk is eating
> away some digits. For example when I call AT&T and when the
> system prompts me to enter my phone number, Asterisk eats
> away some digits, so A
be mobile) with block transfer, I get
899312 and it works.
For me it seems that chan_capi does not supply inbound overlap-dial. Could
anybody clearify that, please?
Bye
Felix
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ement = **some_number**
>
>
> I read in the mailing list archives of commenting out line
> 2615 in chan_capi.c, but that did not change anything.
>
> Has anybody got an idea what the error:
>
> "Channel 'CAPI[contr1/**some_number**]/0'
ng, which is suggested in the output?
> 'make cloneconfig && make dep' in /usr/src/linux/
Felix
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also need capi...
Bye FElix
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Hi,
at SuSE 9.0 helped:
> > I am not able to compile zaptel...
> > Could you give me a hint?
> Have you tried the following, which is suggested in the output?
> 'make cloneconfig && make dep' in /usr/src/linux/
Felix
> -Original Message-
>
digits.
Do you have overlapdial=yes in your zapata.conf?
Cheers
Felix
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Hi,
you can integrate it via PRI or BRI.
Regards
Felix
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronaldo
Sent: Friday, June 11, 2004 7:04 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Integration with
Hi Dan,
could you support alaw/mlaw? Is that a big problem?
Regards
Felix
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Hi Patrick,
could you please give us a feedback if that have worked?
Because I have hacked the source to disable fax..
Thanks
Felix
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Nicolas Gudino
> Sent: Wednesday, June 09, 20
-step basis,
> as I'm not much into this telco world ;)
Sorry, that is not that easy because the receipt depends much on the
circumstances.
What connection do you have between pstn and hicom?
And you should read everything about the leagacy integration, so you will
get an idea,
Hello Martin,
how would you like to integrate? PRI (E1) or BRI (ISDN)?
We have a running integration with PRI and a Hicom 150..
If you have any questions...
Bye
Felix
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Martin M
Hello Holger,
I guess that you must configure your /etc/capi.conf
options = p2p..
Bye
Felix
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Holger Schurig
> Sent: Monday, June 07, 2004 5:04 PM
> To: [EMAIL PROTECTED]
> S
Hi,
I have really googled and read the wiki but I still no idea, how to supress
the fax recognizion.
Our users are not able to fax and that is bad... Could you give me an hint,
please?
Thanks
Felix
>
> Hello,
>
> we have a PRI (E1) to a carrier and a second one to a legacy PBX
;Zap/62-1'
-- Executing Dial("Zap/62-1", "Zap/g1/01081fax|30|TrH") in new stack
-- Called g1/01081fax
-- Channel 2, span 1 got hangup
-- Hungup 'Zap/2-1'
What have I to change? Could I supress that?
Thanks
Felix Deierlein
___
ing source make sure the hicom is not clocking off
> of this line
> Jason
I have allready tried it with 0 and with 1. Normally the Hicom should give
the timing, but it does not matter. It works for hours or only for minutes
and then it crashes.
I cannot close * and have
t +49 (333) is not a local number, so that
another 0 should be added (or a 9).
Regards
Felix
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart
> Sent: Monday, May 31, 2004 3:01 AM
> To: [EMAIL PROTECTED]
> Subject:
I hope that is okay? -> It seems not: so I have uploadet it to:
http://ePyron.de/log.zip
I would be very happy with any help you could provide.
Thanks
Felix Deierlein
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ace for asterisk box? What hardware would You recommend for
> > > this setup?
>
> Before ordering any equipment you should first of all test
> the H.323 setup/connectivity between HiPath and Asterisk.
You also could place * between PSTN and the HIPATH. Have a look at the
Wildcar
Hello Vit,
just try the indications from the UK. That worked fine in Germany.
Bye
Felix
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Dudlik
> Sent: Monday, May 17, 2004 9:20 AM
> To: [EMAIL PROTECTED]
> Subjec
still english.
The question where to place the subdirectories. In the wiki is not a real
answer..
Bye
Felix
> Hi there,
>
> today I made the German language prompts available for download:
> http://www.karl.aegee.org/asterisk.nsf/HT/sound-de
>
> Be aware: Asterisk doesn&#
com to DTAG,
the Hicom sends digit after digit.
My dial line is:
exten => _0.,1,Dial(Zap/g1/${EXTEN:1},60)
and that works fine with SIP and IAX. But with the Hicom I get only the
first two digits and then it trys to dial out: error.
Does I have to use schemes like exten => _0XXX
But I
wn (67) '' ]
> Called Number (len= 9) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
What pridialplan should I use with an
E1 with Euroisdn from the German Telekom (DTAG or T-Com).
Thanks
Felix
> -Original Messa
working fine (from both: Carrier and Siemens).
But I am not able to dial wether outbound nor to the Siemens PBX.
I allways get the message:
== Everyone is busy at this time
After hours of googling and reading and trying I seek help...
Thank you very much.
Felix Deierlein
My extension.conf
Hi Andreas,
I guess it is better to buy a B1 or C2 :-). They are not very expensive at
ebay. Or you buy digium hardware, it surely runs with *...
Or have a look at www.junghanns.net (author of chan_capi)
He sells a 4 Port BRI ...
Bye
Felix
> -Original Message-
> From:
affected?
The point is, I really would like to use IAX Phone, but is has no alaw
codec... (it seems that there is not any win iax client with alaw/mylaw)...
I hope you have some ideas and hits
Thanks
Bye
Felix Deierlein
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Hello Pertti,
we would be interessted to, if you could send further informations...
Thanks
Regards
Felix Deierlein
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-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Pertti
Pikkarainen
Gesendet: Samstag, 10. April 2004 11:26
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