similar to: no ring tone analog Zap --> I4L

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 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

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, can somebody explain this ? 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 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 use the DISA app for giving the user a trunk after a authentication through PGSQL as follows .... auth via PGSQL exten => s,1,DISA,no-password|test I think the user is now in context "test" and he could dial any number if the extension-conf in "test" is for example exten s,1,Dial,OH323/<myip> But if the user dial one digit the call build up

Hi all, i have configured incoming voip traffic as follows: [voipin] exten => _X.,1,SetCallerID(033283077734) exten => _X.,2,Dial,Zap/g4/${EXTEN} exten => _X.,3,Hangup If the call going out the pri dials with an additional '0' before the dialed number. This is for caller number AND called number. But i can't see anything that says set a '0' more in front of the

Hi all, this message occurs if i was connected or not: WARNING[213006]: File chan_sip.c, Line 432 (retrans_pkt): Maximum retries exceeded on call 0b03e0c6189a769b54e49eb471f32454@172.20.23.150 for seqno 102 (Response) If i was connected, the call will be disconnected after a few seconds. What does it means ? I don't see anything to configure like Max retries.... Thanks for help, Thomas.

Hi all together, i have following configuration: ISDN Phone ---> ASTERISK1/PRI ---> ASTERISK1/IAX ---> INTERNET --->INTERNET ROUTER (Port 5036 nat) ---> ASTERISK2/FXO/ANALOG DEV The call flows fine, but no sound will be transfered. On ASTERISK1 a message like "stopped sounds" occurs..... What' s wrong? Is there another port wich i have to nat ? Regards, thanks

Hi all, can somebody explain me why i can't hear a ringing tone (alerting) if i'am going to connect to my destination end point? Is it basically so that i have to configure like: exten => xxx,1,Dial,ChanTec/number|timout|r Is it really nessesary to use the "r" option everytime if i want to indicate a ringing tone? This suggest a wrong call flow for the user ... Thanks for

Hi all, i have following scenario: ____________* 1____ ______________* 2______ | | | | analog/Zap --> IAX2 ----> DSL ---> INTERNET ---> Backbone/100Mbit ----> IAX2 ---> Zap/pri(E400P) ----> PSTN And, if i make a call from *1 over *2 to PSTN, i can hear an echo in my analog phone, even though

Hi all, does anybody know if it is possible to make a modem connection (voice) through * over a handy which is connected to the RS232 port ? I get following messages when * starting: WARNING[16384]: File chan_modem.c, Line 356 (modem_setup): Modem reset failed: (No Response) WARNING[16384]: File chan_modem.c, Line 735 (mkif): Unable to configure modem '/dev/ttyS0' ERROR[16384]: File