Am Montag 14 November 2005 17:22 schrieb FaberK:
> Hi guys,
> this is the scenario:
> PRI <->Asterisk<->SER
> If I call from a Sip(SER) user everything is good, I can call
> anywhere, but if I try to call from outside(PRI) everything is
> wrong!!!
> This is the CLI for an incoming call:
> -
> Have I to use also prilocaldialplan ?
Can be left unknown.
Explains what you expect as the incoming number to look like
> Thanks again
>
>
>
> Giordano
>
> ; PRI Dialplan: Only RARELY used for PRI.
> ;
> ; unknown:Unknown
don't expect anything
> ; private:
Am Dienstag 05 April 2005 18:04 schrieb Henry Jensen:
> Hello,
>
> I've connected a TE110P from digium with a E1/T1 crossover cable,
> according to http://www.voip-info.org/wiki-crossover+T1+cable, to
> a s2m port on a Siemens Hipath 3750.
>
> The lights on the TE110p are flashing red and green, zt
As you have already started to debug this by yourself I would suggest a post
to the dev list as this might be a bug.
I'm not familiar with PRI signalling but I think you won't get any answer
here.
Jens
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Am Donnerstag 17 März 2005 17:49 schrieb John B Dunning:
> Daniel,
>
> Did you get much progress made on this? I'm new to asterisk - but we are
> a heavily invested LDAP shop and if I can demo an initial install that
> pulls telephony configs from LDAP it would really be nifty.
>
> I'd be happy to
Am Donnerstag 17 März 2005 12:26 schrieb J Thomas:
> I am trying to use TE110p card for Euro ISDN with Ericsson AMS switch. I
> consistently get one of the following errors:
>
> PRI got event: HDLC Abort (6) on Primary D-channel of span 1
> or
> PRI got event: HDLC Bad FCS (8) on Primary D-channel
Am Donnerstag 10 März 2005 00:55 schrieb Jens Kübler:
> If someone could explain me this (added some extra debugging code
>
> Mar 10 00:47:13 NOTICE[7561]: chan_sip.c:8448 handle_request: Registration
> from '' failed for '172.20.41.7'
> Mar 10 00:47:13
If someone could explain me this (added some extra debugging code
Mar 10 00:47:13 NOTICE[7561]: chan_sip.c:8448 handle_request: Registration
from '' failed for '172.20.41.7'
Mar 10 00:47:13 WARNING[7561]: res_config_odbc.c:97 realtime_odbc: SQL param
name is a7
Mar 10 00:47:13 WARNING[7561]: res
Am Samstag 05 März 2005 07:58 schrieb Remco Barende:
> Hi list!
>
> While looking for the Snom 190 I found another phone, the Elmeg IP 290
> (www.elmeg.de).
>
> Looking at the pictures & the specs they seem to be very similar beasts
> but the firmware is supposedly not interchangeable.
>
> Does any
The error messages are Postgres related.
You need to have a special postgres include file (postgres-dev files) to make
it compile or disable postpres support somehow.
I'm using debian and the the concering include file resided in a subdirectory
of what asterisk was told.
Jens
__
Am Freitag 04 März 2005 11:27 schrieb Giovanni Miano:
> Hi,
> I've a billion isdn0 card but i suppose than i cant patch kernel to
> use mISDN support because i've kernel 2.4 and patch on CVS i4l is for
>
> >= 2.6.8
use chan_capi
Jens
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Am Donnerstag 03 März 2005 14:59 schrieb R A:
> I need that my records cdr only get the calls that begin with 9 or any
> other rule is this possible??
>
yes
Jens
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Parts of your dialplan would be helpful to figure out
what you are doing
Jens
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Am Mittwoch 02 März 2005 12:54 schrieb Kanishka Somaratne:
> can i install this directly or do i have to install 1.0.0 and then upgrade
> ?
It's a complete package
no upgrade needed
Jens
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Am Montag 14 Februar 2005 12:57 schrieb Tor Setane:
> On Mon, 2005-02-14 at 12:22, Darren Ellis wrote:
> > Hi,
> >
> > I have * working with X-Lite and Sipura adapters, but I have one person
> > who is linux based, and is trying to use Linphone and Kphone. His end
> > works, but I get very bad ech
Am Montag 17 Januar 2005 11:26 schrieb Alberto Martínez:
> Hello.
>
> I have just arrived to Asterisk. I would like to know if Asterisk can
> perform some functionalities I am looking for.
>
> I want to allow voip over sip to some users. All of them must have
> their own user name and password to l
Am Sonntag, 19. Dezember 2004 21:40 schrieb Marcin Mazurek:
> HI,
> basic question. I've got a TE110P card and I'm trying to set it up with
> ztcfg with polish zone.
>
> ioctl(ZT_LOADZONE) failed: Invalid argument
> Notice: Configuration file is /etc/zaptel.conf
> line 206: Unable to register tone
Hi
I've bought the Wildcard TE110 some days ago but I'm unable to get it to work
with Siemens HiCom 300.
I've tried this so far:
1. I've used standard cat5 cable cut off on one edge and twisted wires 1 to 4
and 2 to 3 which is according to cisco a short circuit for E1 lines. Wildcard
takes a f
Hello
I found in the archives of this list, that the Digium Quad E1 Card is capable
of doing the "Master" for an E1 connection. I could not find any hint if the
E100 Card is also capable of doing this.
Is this feature the PRI / PRA property or is it the net and cpe property.
This is not explain
Am Samstag, 20. November 2004 18:48 schrieb Linux Dominicana:
> Hello Gregory
>
> Thanks for your tip, but this looks like a point to point encription,
> but how about between extensions registered in a Asterisk server.
>
> Let's say I got a building 200 users registered and a given set of
> extens
Am Montag, 18. Oktober 2004 14:21 schrieb Roy Sigurd Karlsbakk:
> hi
>
> how can I, using mysql sipfriends, allow one user to use g.729 while
> disallowing this codec on another user?
>
> thanks
>
> roy
>
There is currently no support for allow clauses unless you decide to add it.
Jens
__
Am Dienstag, 12. Oktober 2004 14:04 schrieb Goran Dj:
> I'm new to Asterisk.
> How big can be sip.conf (and other: iax.conf, extensions.conf...)
> Is there point when I must use DB (MySQL...) instead of pure .conf?
>
There is no real answer to this question.
Databases are always a good choice as th
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