RE: [Asterisk-Users] Codec problem from SIP to H323

2006-04-21 Thread Alejandro Mejía Evertsz
asterisk to version 1.2.4 Are you using OH323 or H323 ? I had same problem with 1.2.1 using H323(addon) , Installed 1.2.4 and OH323 and everything worked fine. Cheers, Oliver -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Mejía Evertsz Sent

RE: [Asterisk-Users] extension match sip address

2006-04-21 Thread Alejandro Mejía Evertsz
exten = _sipuserX.,1,blah to match sipuser01, sipuser99... ? or exten = sipuser01,1,blah to match sipuser01 only ? Not to usefull when you want to match domain also :S Eg [EMAIL PROTECTED] -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jon-o Addleman

RE: [Asterisk-Users] extension match sip address

2006-04-21 Thread Alejandro Mejía Evertsz
Nop. . matches one or more from the previous carácter More info... http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Eric ManxPower Wieling Enviado el: Friday, April 21, 2006 2:36 PM

[Asterisk-Users] Error installing oh323

2006-04-21 Thread Alejandro Mejía Evertsz
I'm running: OS: FreeBSD 6.0 Asterisk: 1.2.4 Installing OH323: 0.7.3 I have this error when compiling chan_oh323.c: In function `reload_config': chan_oh323.c:4677: warning: implicit declaration of function `sscanf' chan_oh323.c: At top level: chan_oh323.c:3244: warning:

RE: [Asterisk-Users] CDRs and billing

2006-04-20 Thread Alejandro Mejía Evertsz
One billing solution that Works with your CDRs is AsterBill (www.cybexdev.com) for postpaid. (not opensource, you have to buy) It runs a cronjob to get the latest CDRs and bill each accountcode. For prepaid of course it does it with AGI. -Mensaje original- De: [EMAIL PROTECTED]

[Asterisk-Users] Codec problem from SIP to H323

2006-04-19 Thread Alejandro Mejía Evertsz
Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my

RE: [Asterisk-Users] Codec problem from SIP to H323

2006-04-19 Thread Alejandro Mejía Evertsz
1.2.4 Are you using OH323 or H323 ? I had same problem with 1.2.1 using H323(addon) , Installed 1.2.4 and OH323 and everything worked fine. Cheers, Oliver -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Mejía Evertsz Sent: Thursday, April 20

[Asterisk-Users] SIP incoming calls

2006-01-27 Thread Alejandro Mejía Evertsz
To which context of the dial-plan does asterisk tries to match incoming calls when acting as a sip client? To be more specific: In extensions.conf Under which context should I place  exten = 648064,1,Dial(TECH/peer) for an entry like this register = 648064:[EMAIL PROTECTED]/648064 ?

[Asterisk-Users] Context for SIP incoming (newbie question?)

2006-01-27 Thread Alejandro Mejía Evertsz
Please help me out with this To which context of the dial-plan does asterisk tries to match incoming calls when acting as a sip client? To be more specific: In extensions.conf Under which context should I place exten = 648064,1,Dial(TECH/peer) for an entry like this register =

[Asterisk-Users] To which context a registered peer is sent?

2006-01-26 Thread Alejandro Mejía Evertsz
Hi list. Im having problems handling the incoming calls when using asterisk as a sip client. For example, on sip.conf I have lines like this: register = user1:[EMAIL PROTECTED]/777 register = user2:[EMAIL PROTECTED]/888 [host1] context=hostnumber1 type=friend insecure=very