Re: [asterisk-users] help: H323 and SIP
Hi Alex, You should create a dial plan to route sip calls to H.323 calls. Take a look at : http://www.voip-info.org/wiki/ On 8/6/07, Alessandro Russo [EMAIL PROTECTED] wrote: Hi to all, I've installed Asterisk 1.4.9 with h323, and gnugk as soft gatekeeper. I've tested h323 using ohphone and I can talk between them, then I've tested SIP with Twinkle softphones and function very well. Now I have to perform call from h323 to sip and viceversa. How can I do it I receive h323 call from a Cisco Voice GW to my Asterisk and this call have to go to a SIP phone: - PSTN == CiscoVoiceGW(h323) == Asterisk == SIP - SIP == Asterisk == CiscoVoiceGW(h323) == PSNT I've now idea how to configure asterisk (conf file) and softphones... Thanks for all! -- AxR. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help: H323 and SIP
Hi, thanks for reply I'm reading more about Dialplan, but until now, I've not found anything...(like example or tutorial) With the word route you are intending the Goto command?? Please spent some minutes for helping me ^_^ If you are agree, I send you some information about configuration files. Thx On 8/6/07, map [EMAIL PROTECTED] wrote: Hi Alex, You should create a dial plan to route sip calls to H.323 calls. Take a look at : http://www.voip-info.org/wiki/ On 8/6/07, Alessandro Russo [EMAIL PROTECTED] wrote: Hi to all, I've installed Asterisk 1.4.9 with h323, and gnugk as soft gatekeeper. I've tested h323 using ohphone and I can talk between them, then I've tested SIP with Twinkle softphones and function very well. Now I have to perform call from h323 to sip and viceversa. How can I do it I receive h323 call from a Cisco Voice GW to my Asterisk and this call have to go to a SIP phone: - PSTN == CiscoVoiceGW(h323) == Asterisk == SIP - SIP == Asterisk == CiscoVoiceGW(h323) == PSNT I've now idea how to configure asterisk (conf file) and softphones... Thanks for all! -- AxR. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alessandro R. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help: H323 and SIP
Hi Alex, you should have a route for each extensions you would like to reach in your extension.conf file. Dial Plan is the main concept to understand in Asterisk. Feel free to send you conf and I'll take a look. On 8/6/07, Alessandro Russo [EMAIL PROTECTED] wrote: Hi, thanks for reply I'm reading more about Dialplan, but until now, I've not found anything...(like example or tutorial) With the word route you are intending the Goto command?? Please spent some minutes for helping me ^_^ If you are agree, I send you some information about configuration files. Thx On 8/6/07, map [EMAIL PROTECTED] wrote: Hi Alex, You should create a dial plan to route sip calls to H.323 calls. Take a look at : http://www.voip-info.org/wiki/ On 8/6/07, Alessandro Russo [EMAIL PROTECTED] wrote: Hi to all, I've installed Asterisk 1.4.9 with h323, and gnugk as soft gatekeeper. I've tested h323 using ohphone and I can talk between them, then I've tested SIP with Twinkle softphones and function very well. Now I have to perform call from h323 to sip and viceversa. How can I do it I receive h323 call from a Cisco Voice GW to my Asterisk and this call have to go to a SIP phone: - PSTN == CiscoVoiceGW(h323) == Asterisk == SIP - SIP == Asterisk == CiscoVoiceGW(h323) == PSNT I've now idea how to configure asterisk (conf file) and softphones... Thanks for all! -- AxR. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alessandro R. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help: H323 and SIP
Hi, you must choose the h323 channel, install and configure it. I've used ooh323 for a similar project and I have edited the ooh323.conffile with the Gnugk's IP address (your h323 gatekeeper) and a new context for your test. I've also configured the file .ini in Gnugk (I've used the Win version). In the extensions.conf file I've created a dialplan for the new context. I've registered the H.323 endpoint (in your case the Cisco GW) with Gnugk. This is what happens: H.323 Endpoint-GnuGK-Asterisk-SIP client. But the ooh323 channel in Asterisk must be active and properly configured. Regards Dino ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
