Hi Armin,
On 18/02/2012 19:28, Arik Raffael Funke wrote:
in NT mode, the B-channel is not activated automatically. You have to
signal
the TE side that early-B3 data is available. Then the TE side can
activate
the B-channel. If the NT-side is chan_capi, use
exten => _X.,n,capicommand(progr
Hello Armin,
On 15/02/2012 22:53, Armin Schindler wrote:
I hear no progress indication. EVEN when using the r-option of the dial
command. It works however with
exten => _X.,1,Answer
exten => _X.,n,Dial(CAPI/contr1/12345)
in NT mode, the B-channel is not activated automatically. You
My apologies, I just realised I copied the wrong section of the debug
log. So once again, when pressing the "park call" button, I get the
following "capi debug" output:
CAPI: ApplId=0x0002 Command=0x80 SubCommand=0x82 MsgNum=0xe446
NCCI=0x1403
FACILITY_IND ID=002 #0xe446 LEN
002 #0xe3a5 LEN=0014
Controller/PLCI/NCCI= 0x1303
Reason = 0x0
DISCONNECT_RESPID=002 #0xe3a5 LEN=0012
Controller/PLCI/NCCI= 0x1303
On 14/02/2012 18:18, Arik Raffael Funke wrote:
Hi,
I am using ISDN phones which have a "Park call"
Hi,
I am using ISDN phones which have a "Park call" button. The idea is: you
are on a call, push the button and hang up. You can then go to another
phone and pickup the call without having to remember parking slots, etc.
Unfortunately I cannot figure out how to get it to work with asterisk. I
Hi,
I am trying to run asterisk on an AVM Fritz!Box Fon 7270 embedded DSL
router. This works quite well after getting rid of the preinstalled
phone server but I am encountering some unexpected behaviour.
Background: I am using two CAPI controllers provided by the hardware
- one in MSN mode fo
Hi,
I have installed Asterisk 1.4 with mISDN with the
install-asterisk.tar.gz script from beronet.com. On my system I have two
cards, one a AVM Frit!Card Pci 2.0 and one HFC-S chip. I know both to
work well with mISDN on my system from a previous installation.
Now however, the AVM card works w
Joshua Colp wrote:
Arik Raffael Funke wrote:
The auto setting also does not encompass the info
DTMF option for sending.
Thanks. I was not aware of this.
Ragards,
- Arik
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asterisk-users
Hi,
it seems that there is a bug in asterisk's dtmf mode autodetection.
Assume following sip.conf:
[sipprovider]
disallow=all
allow=g726
dtmfmode=auto
DTMF does not work. It seems rfc2833 mode is chosen despite it being
obvious that this cannot work!
The following configuration is necessar
Hello,
I am having something of an odd problem: about every 100 calls or so,
when a call comes in via an external mISDN interface and I route it to
an internal mISDN interface by dialing an internal msn that is
programmed for multiple phones on the internal bus, somtimes the other
phones cont
Hi,
I am using asterisk's SMS functionality for sending messages. Most of
the time it works without problems (as in situation 1) but sometimes
something seems to go wrong with the transmission (as in situation 2). I
am using "Deutsche Telekom", Germany's main TELCO, so I suppose the
problem m
Hi,
I wrote a few days ago about my problem with calls via voipstunt
stopping ringing after 5-6 rings, subj. "SIP Dial out timeout". Though
if the remote station picks up before that, everything works flawlessly.
I am not entirely sure when this phenomenon popped up, but I used the
this confi
Eric "ManxPower" Wieling wrote:
Arik Raffael Funke wrote:
I am having a problem that is a miracle to me: If I dial out via
voipstunt.com the call rings for a few seconds and then gives me a
busy sign.
Start out with not using the "r" option to the Dial line. That wi
Hi,
I am having a problem that is a miracle to me: If I dial out via
voipstunt.com the call rings for a few seconds and then gives me a busy
sign.
- I do not have a timeout set in my dial command
- the remote station does not cause the busy either
- dialing the number with the voipstunt clien
account by asterisk...
Kind regards,
Arik
Arik Raffael Funke wrote:
Hi,
I have switched a while back from chan_capi to chan_misdn. When the
number is dialed and the phone is then picked up everything works just
fine. Some users however FIRST pick up the phone and then start to
dial... I did n
Hi,
I have switched a while back from chan_capi to chan_misdn. When the
number is dialed and the phone is then picked up everything works just
fine. Some users however FIRST pick up the phone and then start to
dial... I did not get this to work with misdn.
When two digits have been dialed, a
Hello,
does anybody know if it is possible to save the IP address of an IAX
client logging into asterisk into the DB for future reference?
I.e. one could distinguish between cases, where the client was last seen
on the local net or on the road... even when it is not currently online.
Cheers
domU kernel. Does anyone know if
this has been fixed yet?
Arik Raffael Funke wrote:
Thanks. What kernels do you use for dom0 and the domU's? Custom-built
or out of the box?
- Arik
jason wrote:
I would vote RAM. I've been using a FXO card in xen for a good year
now with no iss
Thanks. What kernels do you use for dom0 and the domU's? Custom-built or
out of the box?
- Arik
jason wrote:
I would vote RAM. I've been using a FXO card in xen for a good year now
with no issues at all. In fact, my zttest timings are the same between
xen and native.
Arik Raf
Hello,
I have the following call file:
Channel: Local/[EMAIL PROTECTED]/n
Callerid: 27
MaxRetries: 2
RetryTime: 10
Context: test2
Extension: s
And the following dialplan:
[test1]
exten = s,1,NoOp(${CALLERIDNUM})
But my CALLERIDNUM and CALLERIDNAME variables are both empty. I tried
without s
Anselm Martin Hoffmeister wrote:
Am Montag, den 06.11.2006, 11:04 + schrieb Arik Raffael Funke:
I guess, in short my question reduces to: How do I find the IP adress of
a specific iax client?
With a set of statements like
exten => _5XX,1,Set(IPADDR=${CUT(${DB(SIP/sip${EXTEN})},
Hi,
want to have calls directed to internal fixed phones, when my employees
are home and automatically to their IAX connection when they are logged
in remotely. How do I do this?
The picture is as follows:
--- HOME ---
- user logged in via IAX with local IP adress
- fixed line should ring
-
Hi,
has anybody gotten the following message when trying to modprobe a isdn
adapter card driver?
mISDN_capi: no version for "capi_cmd2str" found: kernel tainted.
Anybody knows whether it's a problem or not and how to get rid of it?
Cheers,
Arik
__
Hi,
I am using Asterisk 1.2 with internal isdn phones connected via a hfcpci
card in nt-mode with misdn. Bridging calls from the internal hfcpci via
a avmfritz card (also chan_misdn) to the PSTN works flawlessly. However
when I use a sip channel to route the outgoing call via voipstunt, it
al
Hi,
does anybody currently use voipstunt from finarea? I place a call to
sip.voipstunt.com I get a 302 redirection. Unfortunately the second
server seems to support only a different set of codecs than the first:
-- Called [EMAIL PROTECTED]
-- Got SIP response 302 "Moved temporarely" b
Hello,
I moved a system from asterisk 1.0.10 based on bristuff and zaphfc to
1.2 based on mISDN.
Unfortunately I can no longer park my calls? I have hold_allowed=yes in
misdn.conf, include => parkedcalls in extensions.conf and even tries a
tT options in the Dial command.
I switched on debu
Hi,
has anybody experience running asterisk on a (i.e. fedora-based) Xen
system? What about mISDN support etc.?
For a low-load system I thought about using:
1. Sempron 2800+
2. some memory, in your opinion how much should I attribute to the
asterisk guest system?
3. A AVM Fritz!PCI card for
Hi,
How do the 302 redirects work in asterisk, and what is the
"promiscredir" directive doing? I am not getting the documentation on this.
I have following happening on my asterisk box:
-- Executing Dial("mISDN/1-1", "SIP/[EMAIL PROTECTED]||Tt") in
ne
Tzafrir Cohen wrote:
On Thu, Sep 07, 2006 at 08:56:54PM +0200, Arik Raffael Funke wrote:
Tzafrir Cohen wrote:
On Thu, Sep 07, 2006 at 01:27:36PM +0200, Arik Raffael Funke wrote:
Hi,
has anybody had success compiling bristuff with kernel 2.6.17.11? Error
messages are below...
This is not
Tzafrir Cohen wrote:
On Thu, Sep 07, 2006 at 01:27:36PM +0200, Arik Raffael Funke wrote:
Hi,
has anybody had success compiling bristuff with kernel 2.6.17.11? Error
messages are below...
This is not code that is touched by the bristuff patch.
Anyway, I'd try the latest 0.3.0 bristuff
Hi,
has anybody had success compiling bristuff with kernel 2.6.17.11? Error
messages are below...
Cheers,
Arik
---
/usr/src/asterisk_1.0.10/bristuff-0.2.0-RC8s/zaptel-1.0.10/zaptel.c:6520:
warning: passing argument 4 of 'class_device_create' from incompatible
pointer type
/usr/src/asterisk
Tim Sharp wrote:
I am running on 1.2.7.1 and have an intermittent problem when making outgoing calls. Sometimes the calling party does not hear the ring tone in their handset, but the call goes through. From my extension I have only had 3 calls like this in the last couple of weeks, other people
Hello,
I am getting lots of messages as the ones attached below. Is this a
problem anybody can explain. (My internet connection is NOT slow or
instable... thus I don't get it.) Maybe does this result from incorrect
registration?
Cheers,
Arik
- sip.conf --
[general]
qualify=no
srvlo
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