Displaying 20 results from an estimated 400 matches similar to: "Fun with CAPI"
2004 Nov 29
1
Outbound E&M?
I've got a new setup (different building) where Asterisk is sitting
between the PBX and phone company on a E&M T1 line.
Mitel PBX <-> Asterisk <-> Phone company
Inbound works. Asterisk gets the in-band digits from the phone company
and hands the call off to the Mitel just fine.
Outbound is weird. Asterisk seems to expect that the mitel will send
routing information
2004 May 24
1
Chan_capi 0.3.1 , Asterisk , 3 x C4 active ISDN card Segmentation fault
Hi,
i use chan_capi 0.3.1 with asterisk (stable branch cvs) and 3 x c4
active ISDN card.
From Controller 1 - 7 there are no problems making calls between
asterisk and the pstn.
But when i make calls from controller 8 - 12 i get on every controller
(8 - 12) a segmentation fault in asterisk :(
I tried different linux distributions (gentoo 2004.1, redhat 9.0 , suse
9.1) but same error.
2005 Jun 15
1
Old but Gold
Everyone,
Im sure you've seen this error a million times, but Ive looked everywhere I
can think of & still haven't found a solution that works.
I'm trying to make an outside call, I can call the physical phone from a
xlite on another pc (and vice versa) but whenever I try to make a call to
the outside world, this happens:
on the CLI:
Jun 15 08:45:20 NOTICE[10390]:
2005 Jan 24
1
Cisco7905 keeps forwarding to voicemail
Hello All!
I have a strange problem with Cisco 7905. It is forwarding unanswered
calls to VoiceMail even thought I have setup it not to.
My ring timer on cisco 7905 is 60s, and my ForwardToVMDelay is 3000s.
This means that call should never be forwarded to VM!
This is true if I call from internal number then this happens on asterisk:
-- SIP/104-6073 is ringing
-- Nobody picked up in
2004 Dec 29
0
AstTAPI - Incoming Calls
Good day,
does anyone have AstTAPI running for incoming calls, and would like to
show some examples.
My setting right now looks like this:
sip.conf
--------
[22]
type=friend
dtmfmode=info
username=22
mailbox=22
secret=privat
host=dynamic
context=privat
canreinvite=yes
callgroup=1
incominglimit=2
extension.conf
--------------
exten => 123,1,noop
;Hint(SIP/22)
exten =>
2005 Mar 02
0
chan_capi - fax patch - crash
WARNING[<pid>]: CAPI[contr3/123456]/178 already has PBX structure??
WARNING[<pid>]: CAPI[contr3/123456]/178 already has a call record??
WARNING[<pid>]: CDR on channel 'CAPI[contr3/12345]/177' already started
WARNING[<pid>]: Thread 1109916592 Blocking
'CAPI[contr3/123456]/178', already blocked by thread 1116277680 in
procedure ast_waitfor_nandfds
2006 Mar 31
3
Echo cancellation problem
Hi!
I'm here again with echo canceller problem... :-(
I think I've done everything to enable echo canceller feature, but it
still doesn't work...
Can anybody tell me if there is some error or something missing in this
configuration please?
I'm using Eicon Diva Server 4Bri.
http://www.eicon.com/worldwide/products/MediaGateways/disv4bri.htm?techspec=1®ID=4999
Card
2003 Oct 14
1
Outgoing CallerID
Hello,
Does anyone know how to set the outgoing CallerID properly when using Snom200/SIP/CAPI/BRI?
Following doesn?t work:
exten => _0.,1,SetCallerID,526910
exten => _0.,2,Dial,CAPI/526980:${EXTEN:1}
Asterisk writes:
*CLI> -- Executing SetCallerID("SIP/226-ada0", "526910") in new stack
-- Executing Dial("SIP/226-ada0",
2004 Apr 29
1
CAPI ptp does not work
Hallo all,
I am trying to get * with chan_capi and a ptp-ISDN with 4 lines on a AVM C4
card to work.
But weather inbound nor outbound is working :(
My capi.conf:
;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
mode=immediate
isdnmode=ptp
msn=8993
incomingmsn=*
mode=immediate
controller=1,2,3,4
softdtmf=1
;accountcode=
context=demo
2007 Mar 18
1
Choppy sound with chan_capi + Fritz Card USB
Hi everybody,
I have a problem which I cannot eliminate on my own. Has anybody any idea
for the following:
I am using the asterisk-version from Debian-Testing (1.2.13) with the
latest chan_capi (also tried an older version).
When using the Capi-Channel, everything works fine except from the sound
it sounds extremely choppy and is unusable :-(
When e.g. capisuite is used for fax, everything
2005 Jul 15
2
Strange problem with SIP and CAPI
Hi,
I?ve strange problem when I?m making a call from SIP (Cisco 7960) to capi
(Fritz PCI). When I call a national number, I?m hearing the ringtone when
the called party is ringing but when I call an international number, I don?t
hear the ringtone and I?ve a silence until the called party answers. Both
call are going through the same extension.
Here?s 2 log files, one with a national number and
2006 Jan 24
2
Re: Asterisk-Users Digest, Vol 18, Issue 134
O.K. thanks a lot, Felix and Peer Oliver. But somehow asterisk keeps
telling me while startup:
[chan_capi.so] => (Common ISDN API for Asterisk)
Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused
contr1
Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused
contr2
Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused
contr3
Jan 24 14:30:47
2005 Feb 27
1
DISA and a long delay; ideas?
Hi,
I have just setup a DISA setup whereby people can dial in, authenticate, are
given a dialtone and can then call out.
Everything works however there is a 10 second delay after the user enters
the number and presses #, until the system does anything.
Here is the relevant section from my extensions.conf:
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2007 Sep 13
1
fax machine detection for outgoing call on DIVA card
Hello,
I need to detect both fax and answer machine, and it should be valuable that
the detection will be run by the Diva card itself. So :
- I read Diva Documentation, and I found that the Diva could send some
specific DTMF, if I had "[..] enabled [this functionnality] by the
application for a designated controller through a manufacturer request
command 9 [...]", but I didn't
2004 Jan 20
0
chan_capi capiECT
Hi all,
does anyone have capiECT working? I can't get it to work - please see
my logfiles below.
While using an * CVS version of late September and chan_capi-0.2.5 (I
guess), it worked!!! (I know, never change a running system ...
should've backuped ... etc.)
Here's the setup:
NT----AgfeoISDNPBX----AgfeoISDNPBX----AnalogPhones
in between the two PBX I attached my * server with a
2006 Jan 13
2
PrimuX Cards with chan_capi-cm
Hello List,
I'm trying to get a PrimuX Card (www.primuxisdn.de) working. The
Manufacturer says that chan_capi (the older one) used to work.
Now I'm trying with chan_capi-cm and have got the following problems:
Outgoing calls "ausgehend_ueber_ntba_primux.txt" the other phone rings
but when I answer the call asterisk crashes.
Incoming: first tried with immediate=no
2007 Jul 24
0
Diva Server BRI hangs up after about 25 seconds
I have a Diva Server BRI installed in a Debian system
running Asterisk 1.2.22. When a call comes in the
Background application plays a couple of times.
Halfway through the second time, the call is
disconnected (total time connected about 25 seconds).
I've posted various configs below. Any advice on how
to debug would be appreciated. We are located in New
Zealand.
CLI showing capi debug (just
2005 Sep 19
0
Call dropped 100% of time when incoming IAX routed to outgoing CAPI
Good day,
The unusual thing about this problem is that it doesn't occur just during a
CAPI call, or just during an IAX/SIP call. Only during IAX/CAPI
I'm having some trouble with the CAPI interface and it only occurs when a call
comes in on an IAX channel and goes out the CAPI interface.
The capi debug in the asterisk console is below as well as the relevent parts
of .conf files from
2005 Aug 24
2
Error when answering CAPI
Hi,
I've a Fritz card which was working fine, recently I changed
hardware and my nightmare started. Now when I call someone through the
chan_capi (0.3.5 or 0.4.0) it works fine but when I receive calls I
always get hungup. Can someone please give some help? Here are the logs:
*CLI>
-- CONNECT_IND ID=001 #0x0000 LEN=0049
Controller/PLCI/NCCI = 0x101
CIPValue
2003 Oct 22
0
capi incoming call
Hello,
i have asterisk with chan_capi ( AVM Fritz card ) working, isdn line
BRI, setup to point-to-multipoint.
working fine, but when both channel on interfaces are used and somebody
wants to call in - the response is "ha-la-li" - sounds like msn or
called number does not exists. i think that right response should be -
line busy tone... right ?
can anybody tell me what is wrong ?