similar to: Dial over IAX ahngs up after 3 rings

Anyone have any thoughts on why versions of asterisk I try (4 so far) after CVS-07/18/03 always end up locking up on me... which means no sip clients can register/re-register and if I type "reload" or "stop now" at the cli it just returns and does nothing. I have experienced this same issue on three separate boxes. Two running RedHat 9 and one running Redhat 8. I don't

Hi all, i have read in the * whitepaper the following: "s: The "start" extension. A call which does not have digits associated with it (for example, a loopstart analog line) begins at the "s" extension." I think this means the s extension will be execute when the phone is picked up. But when i pick up the phone the s extension will be never executed. Whats wrong

Hi all, have anybody an idea where to get adsi phones in europe ? Thanks, Thomas. ******************************************* beroNet technologies GmbH Dipl.- Ing. Thomas H?ger Potsdamer Str. 18 A 14513 Teltow FON: +49 (0) 3328 3077731 FAX: +49 (0) 3328 334779 Email: thomas.haeger@beronet.com *******************************************

Hi all, i made a test scenario with two windoze machines: On the first one callgen323 is running in listening mode On the second one, callgen323 strarting 25 calls to the asterisk pbx, and the asterisk calls the first windoze machine. But after the second one make a few calls (mostly after 11 - 14) asterisk crashes with the only message : Segmentation fault. Are this to many calls for oh323

Hi all, i use the PGSQL App, and i have found out, if you use a "(" or a ")" in your query the query crashes... My sample query was : SELECT count(*) from tbltest where fldtest='xxx' can somebody fix this ...?? Regards, Thomas. ******************************************* beroNet technologies GmbH Dipl.- Ing. Thomas H?ger Potsdamer Str. 18 A 14513 Teltow FON:

Hi all, i 've written a little Callgen script for generating calls through the outgoing spool directory. The calls goes over 8 ttyI devices to another pbx and come in through other 8 ttyI devices. But when i generate the calls, sometimes * register the calls but never initiate them. Especially when the files come to fast into the outgoing dir. What can be wrong ? Is it possible that the

Hi all, i have tried to connect to a clarent gatekeeper. I have used both of h323 drivers chan_h323.so and chan_oh323.so. But no one can register to this gatekeeper. Our ip is activated on this gatekeeper. Maybe, i do wrong anything.... I have only set the "gatekeeper" option in the h323.conf or oh323.conf to the ip address from the gatekeeper. gatekeeper=x.x.x.x But no one of the

Hi all, i don't know how often someone ask for this, but i ask agian: Is it possible to use G723.1 with * or not ? I tried to use G723.1 from * over OH323 to a gatekeeper from my provider. The situation is following: Zap/analog ---> IAX -----INTERNET-----IAX--->OH323---->GATEKEEPER/PROVIDER The provider supports G723.1. Can someone help me ? Regards, Thomas.

Hi all (Michael), how it is possible to get the ip address of the calling party ? (i know by using h323... but there're a few unknown segfaults...) and so i want to use oh323, but i have to get the ip from the caller to permit or deny the call with AGI. Is it possible at all ? Thanks, Thomas. ******************************************* beroNet technologies GmbH Dipl.- Ing. Thomas H?ger

On Sat, Feb 18, 2017 at 12:21:31AM +0800, Yunchih Chen wrote: > >`virsh list` hangs on my server that hosts a bunch of VMs. >This might be due to the Debian upgrade I did on Feb 15, which upgrades >`libvirt` from 2.4.0-1 to 3.0.0-2. >I have tried restarting libvirtd for a few times, without luck. > >Attached below are some relevant logs; let me know if you need some more

Hi all, i have two *. One makes call to another over chan_H323. At the called side i have defined a user (friend) like --> [thaeger] type=friend host=172.20.23.100 context=incomingh323 At the calling side i have following entry in extensions.conf exten => _X.,1,Dial,H323/thaeger@172.20.23.10 Now i have two questions: 1. how will be transfered the callerid (normaly

Hi all, i have big problems to make a h323 call over the gatekeeper from my provider. The provider demanded following account data: H323 ID: XXX-XXX-XX-X DetinationNumer: XXXXXXXXXXX I have configured the oh323.conf following: gatekeeper=XX.XX.XXX.XXX alias=XXX-XXX-XX-X Isx the alias equal to the h323id ? And how i have to make a call with the dial app ? I have following config: exten

Hi all, i want lock Zap channels via global var FREE1 if FREE1 = 1 then call should go on with nothing and waiting for digits to go in _X. Otherwise hangup the channel But if the GotoIf goes to s|4 (NoOp) then comes a ringing indication.... !? The "immediate" property in the zapat.conf is "yes" [tel1] exten => s,1,GotoIf($[${FREE1} = 1]?s|4:s|2) exten =>

Hi all, i have a few questions about PRI/ISDN: 1. Are "supplementary services" like conferencing, call brokering or call forwarding supported by * ? 2. Is there a way to switch calls "transparent" through * from one port to another port ? 3. Is it possible to configure the * so that * detecting dtmf during a call ? Thanks for answering questions, regards, Thomas. :-)

Hi all, i've installed the a new pwlib (1.5.0) / oh323lib (1.12.0) on my *. Then i've installed the new chan_oh323 (0.5.6). when i try to make a call with "netmeeting" through * ( * dial out with "Dial,OH323/${EXTEN}@xx.xxx.xxx.xx" ) the call will be blocked. Before, there was chan_oh323 0.5.5 and pwlib(1.4.11) and openh323(1.11.7) installed, and it worked. Is here

Hi all, inside my * is a E400P. The machine is a PII 400Mhz with 256MB Ram. OS is Debian woody. * is the newest cvs co. I have written a little callgen script which make outgoing calls through my *: #! /bin/sh set -e n=$1 # Nummer anz=$2 # Anzhal der Versuche anz2=$3 # Kan?le sle=$4 # Timeout bis zum n?chsten Versuch if [ -z $4 ]; then sle=0 fi s=1

Hi all, i tried to write a call generator script. It generate i file like sample.call in ast src tree. But when i set the callerid and make a call over Modem[i4l] then the caller id is not set for the outgoing call. Is it impossible to set the callerid at this time for ttI devices? Thanks for help, Thomas. ******************************************* beroNet technologies GmbH Dipl.- Ing.

hi all, i have one (teles) pbx with a BRI telephone and an outgoing E1 port. The outgoing E1 is connected to an pri_net port from my *. The incoming call will dail out to a h323 soft phone like openphone or sjphone or just netmeeting. The call will be conneted, but i don't hear any sound, from no one of the both sides. Can somebody help me? Thanks, Thomas.

Hi all, if i enter a "show codecs" at cli * response with: 1 (1 << 0) G.723.1 2 (1 << 1) GSM 4 (1 << 2) G.711 u-law 8 (1 << 3) G.711 A-law 16 (1 << 4) MPEG-2 layer 3 32 (1 << 5) ADPCM 64 (1 << 6) 16 bit Signed Linear PCM 128 (1 << 7) LPC10