Someone once said "YOU CAN'T BE TO RICH OT HAVE TOO MUCH BANDWITH"
1 How much do you have? How many phone calls and how many other users on your connection? 2 Go to http://testmyvoip.com/ and test your bandwith Jeff Date: Sun, 13 Mar 2005 09:23:35 +0100 From: "Edward Banfa" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] RE: Asterisk-Users Digest, Vol 8, Issue 88 To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="US-ASCII" Hi, Thanks for the reply. I tried changing my allow and disallow entries to match yours below but still no luck. Could my problems be bandwidth related? If so what amount of bandwidth should I request? Cheers Edward -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff Glassman Sent: Sunday, March 13, 2005 12:17 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: Asterisk-Users Digest, Vol 8, Issue 88 These allow and disallow work with NuFone for me disallow=all allow=ulaw allow=alaw allow=gsm Jeff Message: 11 Date: Fri, 11 Mar 2005 11:15:51 +0100 From: "Edward Banfa" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] NuFone Configuration [problem] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" Hello, I am trying to configure the my asterisk box here with the following **iax.conf*** [NuFone] type=peer host=switch-1.nufone.net secret=xxxxxx ***extensions.conf:*** exten => _1NXXNXXXXXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} exten => _011N.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} I have a couple of Xlite softphones and 2 analogue phones connected to a mediatrix 1102 connected to our lan. The mediatrix talks sip to the asterisk box on the lan. We are running asterisk on FC3 . SOFTPHONES[XLITE] ---SIP--> ASTERISK----IAX--->NUFONE[ASTERISK] ANALOGPHONES---MEDIATRIX_1102---SIP--->ASTERISK---IAX--->NUFONE[ASTERISK ] Well the problem goes something like this. 1) I can dial a number form the softphones and when the call is answered I can hear the user on the other end but the user can't hear me 2) I can dial a number from the analog phones (via mediatrix tru to asterisk)(the mediatrix is properly registered with our asterisk box) and when the call is answered both ends can't hear a word, its just silent. I think I am having a codec problem here. What am I doing wrong. We would sincerely appreciate any help/pointers. Thank you all Edward Banfa _______________________________________________ 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
