Displaying 5 results from an estimated 5 matches for "webswitch".
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bbswitch
2003 May 27
1
chan_h323 + Ericsson Webswitch 100
I'm haveing trouble connecting an Ericsson Webswitch 100 to asterisk.
Has anyone gotten a Webswitch running? When I try to connect asterisk
thinks everything works fine, while the webswitch just rings. I belive
chan_h323 is picking the wrong port to talk at the webswitch on, however
I'm not sure, nor am I sure how to fix it. Any clues/hints?...
2004 Apr 06
0
Ericsson Webswitch 100 Model G4 with * ?
Does anyone have info on using the Ericsson Webswitch 100 Model G4 with *?
The documentation and brochures make it sound like it's some kind of
specialized FXO device purely to provide legacy PBX with access to VOIP,
and not a general purpose FXO-VOIP gateway.
Is it possible to use it to route incoming PSTN calls into * with full
callerid, an...
2003 Nov 13
1
how to interconnect gnugk and asterisk?
Hello folks.
We are trying to interconnect an asterisk installation with a gnugk 2.0.5
installation to become able to use some H323 hardware that needs a gatekeeper
(particulary an Ericsson WebSwitch 100).
We have managed asterisk to dial H323 endpoints successfully (although calls
are interrupted immediately after connection with "spawn extension exited
non-zero"), but we could not manage gnugk to "dial" to asterisk.
What I am confused most about is, what asterisk is f...
2003 Sep 12
3
h323 v oh323
...es asterisk,
> zaptel and libpri
> (as shown on asterisk.org) ?
>
> 2.
> If it is, is it h323 or oh323?
>
> 3.
> If it is not, does one just need to follow instructions in
> /asterisk/channels/h323/
> in order to get it enabled.
>
> 4.
> I have an Ericsson webswitch 100 G4 (4 FXO ports). Once h323 is
> enabled/configured
> in *, WHY there is no any authentication needed on Ericsson
> box for placing
> the calls?
> I presume, that when Ericsson, "assigns" LAN IPs, that only
> those boxes can
> communicate with it?
> Is that c...
2003 Sep 03
8
Asterisk Jitters
Hi,
Every time I dial into my asterisk box i hear nothing but asterisk
jittering.
The following is an example of what I get on the asterisk CLI
Thanks
*CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT
on RTP
to 0
DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res
DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user