asterisk users - Jan 2005 - Snom hint for ZAP channels?

no we have a tdm400 at this site does this still apply?

Hi

Il giorno ven, 21-01-2005 alle 08:54 -0600, Justin Carlson ha
scritto:
> is the hint
> 
> 99,hint,ZAP/1
that works only for sip channels.
if you want hint working also for zap, you should
check very latest bristuff at junghanns.net website.
Afaik he has added (among support for bri cards)
extension states also for zap channels.

Matteo.

Hi everybody,

I have setup a Mediatrix 1204, the calls worked fine, both incoming and
outgoing.

The problem here is the delay.

When I do a call to the PSTN or receive a call from the PSTN exists a delay of 4
seconds after answer or sending the call.

For OUTGOING

My Dialplan for the Mediatrix box is the following, here at Mexico we use 8
digits for local calls.

([1-9]xxxxxxx|01xxxxxxxxxx|1111|060|0xx)

I have verified that inmediatly after I dial from my IP phone, the in-use light
turns on in Mediatrix but the call is not pass until the 4 seconds timer
expires.

I have tried disabling the Dial plan but it didnt help

Form Mediatrix documentation....

The Timer is set to 4 seconds. It can be used to indicate that if users have not
dialed a digit for 4 seconds, it is likely that they have finished dialing and
the gateway can make the call. A Dial Map for this could be:
[2-9]xxxxxxT

FOR INCOMING 

The same 4 seconds delay after the call is sent to Asterisk.

The problem here, is that despite we answer or not the call, once the call is
sent to Mediatrix, the calling party hear 2 ring-back tones generated by
Mediatrix, then the ringback for Asterisk

Once the call is passed to Asterisk and starts ringing, if we call from a cell
phone,home or office the call is marked as answered and the call timer starts no
matter if is answered or not.

Any ideas?

I have tried sending the # at the end with no success.

Thanks!




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Hello

I'm trying for a while to compile and install OH323 channels on my two
distribs...

I have downloaded the src pwlib and h323 files versions given in the
documentation.
Make some RPMS with "googled" SPECs (and seems to give good results)
Tried to compile the channels failed each time... (I have also tried
with at-rpms oh323 and pwlib versions).

Did someone who have already done the job could help me ? -I'm looking
for working specs to compile pwlib and oh323-

Thanks

Nicolas.

Hi List!

I have an interesting problem. I am behind a NAT Firewall which works 
fine with SIP. I am connected to T-DSL in Germany and there the 
DSL-Connection is interrupted every 24hours and buck a few seconds later 
with a new dynamic IP.
My Asterisk is registered with several SIP-Providers and this works 
fine. In addition the *-Server has a HFC-S ISDN interface card installed 
in NT mode (using zaphfc) with an ISDN-Phone connected.
Everything works fine when I make a call from the ISDN-Phone through my 
SIP-Provider to other Phones or SIP-Users (external).

BUT: As soon as I get the new IP (either automatically due to the forced 
interrupt of my DSL line each 24hrs, or manually forced) every still 
seems to work fine but audio goes only in one direction from now on 
(alwasy I can't hear the other party but they can hear me and signalling 
also works fine). So I make a call, but while the phone I am calling 
rings I can't hear the ringtone in my phone. When the other side answers 
the phone I can't hear them while they can hear me loud and clear.
A "relaod" on the CLI solves the problem till next IP-Change.

I know there were already some things reported with dynamic IP's but in 
most cases nothing worked anymore after the IP changed. What can I do 
(maybe with the settings ind some conf-files). In addition I found that 
if I have srvlookup=yes in my sip.conf Asterisk can't register with my 
sip provider. But many example configs dsay you should use srvlookup=yes 
and I hoped that this might solve my problem but I can't use this 
setting set to yes at all.

Any ideas on what to try?

Thanks,

Jui

is the hint

99,hint,ZAP/1

supposed to work or how do I get the lights on the phones to display
channels in use in addition to extensions in use?