[Asterisk-Users] SIP H323 gateway
Hi pros, Newbie to asterisk, need some help. My existing senerio is we have 6 analog quintums and 1 digital H323, and our gatekeeper is gnugk openh323 located in US. Our business is Call Center and our method of dial is using prefix and gateway IP provided my Carrier. I also brought two AudioCodes MP108 8 FXS gateways, as our gateway runs on h323 my friend suggested to go for Asterisk. If I'm not mistaken Asterisk can entertain both H323 and SIP so I need to configure Asterisk as SIP and H323 gatekeeper to take calls and route to our International Carrier, I installed Asterisk on Fedora core 3, installation was successful but when I start asterisk with vvvc after 5-10 mins the box freezes, don`t know why, and it only happens whn I start asterisk, now i`m installing RH 9. The digital quintum have 4 E1 ports and we have brought iDCS with dialogic card installed, is thr anyway I could use Asterisk for this ? My question what files I need to modify on asterisk for SIP and H323 working and place calls. Can any 1 send a sample config for SIP and h323 please? Best regards, Mohd. Imran Kamal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP H323 gateway
I installed Asterisk on Fedora core 3, installation was successful but when I start asterisk with vvvc after 5-10 mins the box freezes, don`t know why, and it only happens whn I start asterisk, now i`m installing RH 9. fc3 is fine u can run it in runlevel 3cli mode Best regards, Mohd. Imran Kamal ___ -- Allen Niven GlobalFone 350 Fifth Avenue #6206 Empire State Building New York, NY 10118 +1-212-678-4381 office +1-646-246-7415 cell http://www.GlobalFone.biz Instant Messaging Accounts ICQ 137763656 Yahoo Messenger [EMAIL PROTECTED] MSN Messenger [EMAIL PROTECTED] PLEASE NOTE I NEVER NEVER NEVER RECEIVE EMAILS ON HOTMAIL OR YAHOO ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP - H323 GAteway
Quoting Anton Tinchev [EMAIL PROTECTED]: What gatekeeper do you use. It seems that is programed to make outgoing calls only to registered h.323 users. Just program it to forward unknown number to the asteris (or switch everything to SIP) I can't forward unknown number to the asterisk because the unknown numbers go to another gatekeeper. I am in a university where they have implemented VoIP with H.323. So they have a gatekeeper, and if a telephone is not in this gatekeeper it looks in another gatekeeper... I have to implement my SIP thing without touching this.. What does it mean to switch everything to SIP? Thanks a lot for your help Regards, Mireia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP - H323 GAteway
Mireia Munoz de jesus wrote: Hi! I am in a H.323 network with a gatekeeper and some terminals. Asterisk is a gateway between this network and the SIP network. Now I can do calls from de foreign network (SIP) to the locla (H.323) but I don't know how to do the inverse. The H.323 terminals use NetMeeting, and when I try to make a call, it says that the number dialed must be registered in the gatekeeper. How can I register a SIP number in a H.323 gatekeeper? I know that with NetMeeting I can make calls peer-to-peer dialing an IP... but if the softtelephone of the other terminal is an SIP UA that is not going to work, is it? Please any help will be welcome. Regards, Mireia What gatekeeper do you use. It seems that is programed to make outgoing calls only to registered h.323 users. Just program it to forward unknown number to the asteris (or switch everything to SIP) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
