[Asterisk-Users] SIP routing over IAX2
Hi All, I have two Asterisks, one on the LAN that handles the internal calls with a PSTN interface and one on the DMZ with a public interface. I would like to route SIP calls from the internal users to the Internet over IAX2 to the DMZ and onwards. All users have soft phones so they would enter sip:[EMAIL PROTECTED] to get a connection. I would like to avoid having number prefixes to dial external SIP phones. Can Asterisk do this? I am relatively new to Asterisk. I guess I'm after something like an email smarthost feature for SIP. I have googled and checked some of the getting started pages but all dial plans deal with number prefixes to route calls. I want to route calls starting with 'sip:' as a prefix. Thanks, Bart... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP routing over IAX2
2006/3/16, Bart J. Smit [EMAIL PROTECTED]: Can Asterisk do this? I am relatively new to Asterisk. I guess I'm after something like an email smarthost feature for SIP. Yes, Asterisk can do protocol conversion as well as codec conversion. Just configure phones and asterisk to connect correctly (i.e. echo test working) and make sure the audio codecs you are using are compatible or are enableded in asterisk. I.E. One case that will not work: phone or trunk A: protocols supported speex,iBLC. Asterisk: supports speex, iBLC, G711. Phone B: supports G729, G723. In this case, Asterisk should converted one of the codecs supported by B to one of supported by A, but Asterisk can't decode them because you don't installed any codec for G729 nor G723. Cases it will work: if A supports also G729 or G723: in this case, Asterisk don't need to do transcoding, then it does not matter if it has tihs codecs. If you install G729 and/or G723 in Asterisk. In this case, Asterisk can decode the audio and re-encode with speex or iBLC. -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP routing over IAX2
Thanks Alejandro, I'm sure the codecs are fine, as I can make calls inbound to the LAN Asterisk. Can you tell me which configuration changes I need to make on each Asterisk to route these calls? Bart... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Vargas Sent: 16 March 2006 08:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP routing over IAX2 2006/3/16, Bart J. Smit [EMAIL PROTECTED]: Can Asterisk do this? I am relatively new to Asterisk. I guess I'm after something like an email smarthost feature for SIP. Yes, Asterisk can do protocol conversion as well as codec conversion. Just configure phones and asterisk to connect correctly (i.e. echo test working) and make sure the audio codecs you are using are compatible or are enableded in asterisk. I.E. One case that will not work: phone or trunk A: protocols supported speex,iBLC. Asterisk: supports speex, iBLC, G711. Phone B: supports G729, G723. In this case, Asterisk should converted one of the codecs supported by B to one of supported by A, but Asterisk can't decode them because you don't installed any codec for G729 nor G723. Cases it will work: if A supports also G729 or G723: in this case, Asterisk don't need to do transcoding, then it does not matter if it has tihs codecs. If you install G729 and/or G723 in Asterisk. In this case, Asterisk can decode the audio and re-encode with speex or iBLC. -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip routing
Hello, Can we configure asterisk in order to send sip requests to a outbound proxy when asterisk get AOR of users agents with an private ip ? Asterisk AOR:[EMAIL PROTECTED] ip | | sip proxy/nat box---user agent 192.168.0.0/24 Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
