search for: kueppers

hi, we are using a voip h323 switch. the switch sends all caals to our Gatekeeper (gnugk). gnugk musst send all calls to asterisk and asterisk must do his choice (sip endpoint or out to PSTN) Making calls to our h323 switch works fine over asterisk. what must i configure to get inboung h323 calls from our gnugk to asterisk? any hints for me? thx -- Thomas K?pper 01063 Telecom GmbH &

Hi List, we are using openh323 gatekeeper for voip telefony. We also have a voip over ss7 TELES Switch for voip into POSTN Network. Know we want to use Asterisk for converting SIP to h323. Now my question. Is Asterisk an full h323 gatekeeper like openh323? Do we need openh323 GK for astrerisk, too?. And how can i tell asterisk to sent all none SIP-ip calls to the gatekeeper over h323? thx

I thought it would be nice to have vboxes insead of user logins ... Now I am confuesed and in trouble .. My server serves 2 domains so user1 at domain1.com is the same as user1 at domain2.org I've created the mail_location string with %n and static userdb with %n dovecot creates 2 mail user directories user1 at domain1.com user1 at domain2.org What the heck am I missing .. ?? Your

...> <http://mail.yourdomain.com>?and randomly uses one of the two ips. You > can't control which one, but this gives you active/active loadbalancing. > In case one server is down the MUA just uses the other ip. Are you sure that this is working? Regards Patrick -- Westenberg + Kueppers GbR Spanische Schanzen 37 ---- Buero Koeln ---- 47495 Rheinberg pwestenberg at wk-serv.de Tel.: +49 (0)2843 90369-06 http://www.wk-serv.de Fax : +49 (0)2843 90369-07 Gesellschafter: Sebastian Kueppers & Patrick Westenberg

with a h323 client over my gatekepper a call comes over asrerisk to my sip endpoint: == Spawn extension (sip-phones, 01634255122, 1) exited non-zero on 'SIP/0699073201-528d' -- Executing Dial("H323/ip$10.0.0.124:49638/18690", "SIP/0699073201") in new stack -- Called 0699073201 -- SIP/0699073201-dc61 is ringing -- SIP/0699073201-dc61 answered

Hi, list, I'm going to deploy postfix + dovecot + CephFS( as Mail Storage). Basically I want to use two servers for them, which is kind of HA. My idea is that using keepalived or Pacemaker to host a VIP, which could fail over the other server once one is down. And I'll use Haproxy or Nginx to schedule connections to one of those server based on source IP( Session stickiness),

Hi ! I have a small problem. When switching a call (pstn -> sip user), I get the sip phone ringing - ie. everything is OK, but I do not get a ringtone in the handset on the pstn side. Can anyone help me out in how to make * play tones ? My setup: E1 IP pstn ------ Asterisk ------ sip phone Regards, Dave

Sorry if this is posted to the wrong forum but as it is related to a problem I have with Asterisk it may just scrape through!! I am running Fedora 1 and I can telnet in to my asterisk box as any user except root and am using the same credentials as logging in locally. I am new to Linux and any help would be gratefully appreciated. Thanks Neil -------------- next part -------------- An

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