Hello,
we have a TE405P running at DTAG.
Zapata.conf:
stern01:/etc/asterisk # cat zapata.conf
[channels]
faxdetect=no
language=de
usecallerid=yes
hidecallerid=no
restrictid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callretu
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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Sorry for the HTML-Messages, I have simply forgotten to change it before
sending.
Hello,
we had a running configruation where asterisk passed the phone
number and the ddi to the pstn (i.e. 595-431)
Now only the rootnumber arrives: 5950
I do not k
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
> exten => _9XX.,3,Hangup
>
> ;
> ; CAPI config
Hello,
we had a running configruation where asterisk passed
the phone number and the ddi to the pstn (i.e. 595-431)
Now only the rootnumber arrives:
5950
I do not know, what to do. I tried to use callingpres
(now i am just hiding every number, because 595-0 is no valid extension..) but
t
00 CET"
>
> If i revert back to cvs update -D "6/21/04 18:00:00 CET" the
> problem is gone.
>
> -- original message --
>
> I am also having the same problem. Latest CVS & Latest Capi
>
> When it does work and you pick up the phone, CAPI disconnec
Hi all,
* was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a
Today chan_capi stopped working, without any changings at the system.
It seems, that not * is the reason, because isdn-log also shows no calls.
If I try to call * from outside via capi, I only get a busy.
That is th
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 AT&T does not get the
Tobi,
> > Do you have DIDs (PTP-ISDN)?
> yes
then I guess that I have the same problem.
If I get a overlaped dial from PSTN, i get only the first did-digit as
extension
, p.e: my number 8993-12 then it goes to 89931 and that extension does not
exist
If I get a call from ISDN (or maybe mobile) wi
Hi Tobi,
> I installed Asterisk with CAPI support. Everything works fine
> while starting Asterisk, but when a call comes in Asterisk
> hangsup the call after two times of ringing.
>
> The output is like:
>
> Jun 24 22:19:49 NOTICE[1082178480]: chan_capi.c:1931 capi_handle_msg:
> CONNECT_IND
> Mike,
>
> I've been trying to install under SuSE 9.1, but cannot compile zaptel
>
> What's the secret incantation ??
>
> TIA
I was helped with:
> > 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 clon
Hi,
> >From recent experience:
> If you want to use digium hardware dont use suse 9.0. It
> seems to think the E1 card is a tigerjet bri card and the
> kernel hangs on ztcfg.
I have a WT405P running under SuSE 9.0 and it works great.
But I had only choosen SuSE because I also need capi...
Bye
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-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PR
Hi Thomas,
> I have got the following problem (E100P, pri_cpe):
> My number range is <6digits>xyz. (e.g. 123456-999)
> >From ISDN phones, everything's fine, but calling in from analogue
> >phones causes the
> following problem: Asterisk only receives the first 6 digits.
Do you have overlapdial=ye
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, 2004 8:48 PM
> To:
Hello Martin,
> >how would you like to integrate? PRI (E1) or BRI (ISDN)?
> Besides of making calls with VoIP from PC to PC, we'd like
> that our people abroad could dial company internal extensions
> through Asterisk using a SIP client. On a second approach,
> the same people abroad could dia
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 Mielke
> Sent: Tue
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]
> Subject: Re: [Asterisk-
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:
>
> DTAG --
Hello,
we have a PRI (E1) to a carrier and a second one to a legacy PBX:
DTAG ---pri * -- Hicmo
(PSTN) |
|
Sip
and
more
Many normal inbound calls are direcly routed to the hicom.
Outbound calls from the Hicom go th
Hello Jason,
> >Everything works really fine, but the connection breaks sometimes
> >(there is not really a time scheme), so that you could not dial from
> >the hicom to * or from * to hicom.
> I see from your config file you are using the hicom as the
> second timing source make sure the hicom
Hello Adam,
Hi Adam,
two features I would really like to have:
- the textbox from "Dial a URL" in the normal client (maybe optionally) so
that you could easily copy and paste numbers in
- a function that replaces +49 or wathever to 00. maybe it would be also
possible, to recognize that +49 (333)
Hello,
we have a connection to a leagacy pbx (Siemens Hicom 150E) via PRI (E1).
Everything works really fine, but the connection breaks sometimes (there is
not really a time scheme), so that you could not dial from the hicom to * or
from * to hicom.
The only one to get the connection back is to
Hi,
> > > enough get redirected to human consultant. There should be
> > > possibility for supervisors to connect to ongoing conversation.
> > > Expected traffic will not exceed 30 concurrent calls.
>
> Look at "ZapBarge" for the listening-in. As usual the Wiki is
> your friend. Also I assume
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]
> Subject: [Asterisk-Users] indications.co
Hello,
first: thanks. The normal prompts are working.
But I am still not sure, where I sould place the german digits, letters and
phonems.
First I placed everything under sounds/de/.. but then digits did not work,
then I linked it to /sounds/digits/de/ now I have german digits but
saynumber is st
e verbose level of output and post the call
> setup trace results here. Try the following command from the
> Asterisk CLI before making your next call:
>
> pri debug span x
>
> Where x = single integer digit for the PRI span that will be
> used to make the outgoing call.
ge-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> ePyron Felix Deierlein
> Sent: Sunday, May 09, 2004 6:48 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] No outbound calls at a PRI possible
>
> Hello all,
>
> the
Hello all,
the scenario:
Carrier S2M-- * -S2M--Siemens
|
|
SIP Clients
and many other features
With much help from the list, the PRI links are without alarms and inbound
calls are
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: [EMAIL PROTECTED
Hello all,
I have googled a bit, but was not able to a definite answer (maybe there is
not one..)
The question is, how different would be the voice qualitiy, if you let
translate * from alaw (PRI) to gsm instead of using alaw as codec for sip.
And also how would echo and the processor load be affe
Hello Pertti,
we would be interessted to, if you could send further informations...
Thanks
Regards
Felix Deierlein
[EMAIL PROTECTED]
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Pertti
Pikkarainen
Gesendet: Samstag, 10. April 2004 11:26
An
Hi,
thanks.
>> how can I checkout ztdummy?
>> Thank for you help.
>Checkout of cvs the zaptel source then follow these instructions:
>http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy
I have tried to follow, but I did not know, wich modul I had to check out..
Bye
Felix
Hi,
how can I checkout ztdummy?
Thank for you help.
Felix Deierlein
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