Llorenç Suau [EMAIL PROTECTED] writes:
Any suggestions, to how I can make that the PBX receives correctly the call,
PREFIX+number, to make the external call.
Does this link have the right to make calls to the outside world on
the PBX? Normally this feature is turned off on typical PBX.
cu,
Rich Adamson [EMAIL PROTECTED] writes:
Sean Cook wrote:
I have a sangoma 104d that is our main pbx now( legacy system died ). I
have replaced every phone in the building and things are going very well.
We have fax working well and calls are routing properly... All is well...
Except for
James Arscott [EMAIL PROTECTED] writes:
Hi
Small progress, though combining the suggest below, enabling overlapdial and
a few other things I have got the following :
When you hit 9 on the simenes, you hear a dial tone. As soon as you hit
another number to start dialling it complains with
Ron Wellsted [EMAIL PROTECTED] writes:
So while there HFC cards out there, it seems that they are going to get
harder to find.
We got a few of these from Conrad. They are in Germany and I am not
sure if this one is the same as ours. But at EUR 24.95 per card you
cannot loose to much.
James Arscott [EMAIL PROTECTED] writes:
Asterisk is not matching the extension from the siemens because the siemens
has not even sent one yet, it is still waiting for a dial tone. When I hit 9
on the siemens it does not get a dial tone from asterisk, I assume this is
because I have not told
James Arscott [EMAIL PROTECTED] writes:
My concern is still over if L2 and L3 are Œup¹ on my ISDN between the
asterisk and siemens, I do my settings look right ? I thought my timing on
span 2 maybe incorrect ?
If you do a pri show span 2 you get a short info about the state of
the span.
Eric \ManxPower\ Wieling [EMAIL PROTECTED] writes:
Wolfgang Zweimueller wrote:
Hi all,
when I receive incoming SIP calls on my Asterisk (1.2.9.1) where the
caller has a username in it's From-Address which also exists in my
sip.conf then my system answers with 407 Proxy Authentication
Hi all,
when I receive incoming SIP calls on my Asterisk (1.2.9.1) where the
caller has a username in it's From-Address which also exists in my
sip.conf then my system answers with 407 Proxy Authentication
Required. If it's nonexistent username then callin works fine!
It seems that this is a
Mimmus [EMAIL PROTECTED] writes:
Hi,
after a few of upgrades, I noticed these messages in full debug log:
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring
Tristan [EMAIL PROTECTED] writes:
This server ( an IBM X-Series 346 dual 3Ghz Xeon with 2gb ram ) has to
serve about 60-70 incoming/outgoing PSTN simultaneous calls ( IVR and
max 30-40 conferences... ) and about 10-20 SIP calls to begin...
I tried the Quad-port Digium cards in this special
Hi Alex,
Asterisk [EMAIL PROTECTED] writes:
Wolfgang: What kind of CPU-load issue on A104? Could you give me a
link or something? I also use A104 and it works very good, but
recently I noticed a behavior which is maybe connected with this
issue, so more info would be very helpful for me :)
Rich Adamson [EMAIL PROTECTED] writes:
Cosmin Prund wrote:
I wanted to see where those periodical spikes are coming from so I
started shutting things down. The first thing to go was
Asterisk.
[...]
Is there something funny happening with my zaptel?
Wolfgang Zweimueller, can you give
Hi All,
I have read many posts about problems with Asterisk on some systems. I
also set up Asterisk on many different boxes. But I have never seen
the following...
There is an IBM X346 (3.4GHz Xeon) with one Sangoma A104. This system
is currently idle, that means there is nothing running except
Hi Johann,
Johann Hanne [EMAIL PROTECTED] writes:
Hi,
we are still trying to properly connect a Tenovis PBX to an Asterisk server
(asterisk 1.2.6, libpri 1.2.2, zaptel 1.2.5, Digium Wildcard TE110P), this
time with QSIG.
I tried with asterisk 1.2.2 against Alcactel 4400 on Monday. We had
Christian Reelfs [EMAIL PROTECTED] writes:
Hi, sorry for this noop question,
but does anybody know how to cut the first digit of a variable?
example:
044612345
should be after cut operation:
44612345
Look at README.variables! It says:
,
| The format for removing characters from a
Dinesh Nair [EMAIL PROTECTED] writes:
On 03/31/06 19:49 Wolfgang Zweimueller said the following:
My conclusion with Q.SIG: do not use it at this implementation
level. YMMV.
i'll beg to differ. we've used Q.SIG successfully with an Ericsson
MD110 for a customer in thailand.
Well, that's
Marcus Hofbauer [EMAIL PROTECTED] writes:
Hi!
I'm looking for someone who has successfuly setup an asterisk in
austria with isdn in p2p mode and chan_capi.
There is is a special problem in austria with DID. If someone is
dialing the phone number of the asterisk pbx like 12345-0, zero is
Marcus Hofbauer [EMAIL PROTECTED] writes:
BUT ... If someone is dialing the PBX head number without any
extension, asterisk can't handle this ... the DID in this case is
empty
Any ideas how to handle this?
Try the WaitExten application.
cu,
Wolfgang
Igor Neves [EMAIL PROTECTED] writes:
Hi,
Does anyone have any experience connecting asterisk to alcatel 4200
series pbx with bri cards?
Does it should work with asterisk bri in NT mode, and alcatel bri with
TE mode?
Hi Igor,
we are doing that. Bristuffed Asterisk with two HFC-cards is
Johann Steinwendtner [EMAIL PROTECTED] writes:
I can only guess, but I think I can remember that the creflen needs
to be 2 octets for qsig. Check what the Alcatel switch sends in the
setup message to *.
Thanks, I will have a look at that.
Anyway, why do use QSIG ? Does name display work on
Hi all,
I tried to connect the bristuffed(0.3.0-PRE-1i) * to an Alcatel PBX
via BRI (zaphfc) and Q.SIG. The Alcatel PBX is connected to the
outside world and should forward our calls to the telco. This setup
works correctly as far as I use euroisdn as the switchtype.
The first problem was that
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