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...e and I have installed X-Lite on my Windows machine and I am able to connect it to the Asterisk server. I went ahead an created an account on Broadvoice today and followed the directions on http://voip-info.org/wiki-Asterisk+settings+Broadvoice and http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup but when ever I try and make a call from Xlite I get the all circuits are Busy now recording. Do I need to create a Trunk or get rid of the one that's there? Currently listed is the ZAP/g0 wich I think is for a hard line. Here is my current sip.conf and extensions.conf Thanks for any tips...

...ent in all of the online info is that nothing consistently works.... I have tried to combine the tutorial by William Pool at http://lists.digium.com/pipermail/asterisk-users/2005-January/086337.htm l with the latest instructions from Dan Weber at http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup . This has not worked. Right now, "sip show registry" shows this: Host Username Refresh State sip.broadvoice.com:5060 6033751414@s 120 Auth. Sent I added this to my sip.conf: externip=xxx.xxx.xxx.xxx localnet=10.1.0.0 localmask=255.255.0.0...

...ly I provided. If you were using the > instructions I provided originally, you would not be able to make > outbound calls. Here are the instructions that have been known to work; > Please read line by line and setup that way. > > http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup > > Dan > > On Sun, 6 Mar 2005, skamp wrote: > >> sorry still doesnt help with incoming calls, there is definatley >> something more wrong, my config was working fine until today and its >> worked fine for months. They have broken something. >> >> On Sun...

...PPPPPPP insecure=very context=sip authname=PPPPPPPPPP dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no Thank you On Wednesday 09 March 2005 08:23 am, Mike Matthews wrote: > Have you tried this: > > http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup > > Zanzamar Majere wrote: > >Thank you for the response. I still have the errors mentioned below, sip > >response and Failed to authenticate on INVITE > > > >[PPPPPPPPPP] > >type=peer > >username=PPPPPPPPPP > >fromuser=PPPPPPPPPP > >authuser=...

...me via BV my number is 8475100139 and it > >> is up. > >> > >> Chris > >> > >> On Mar 9, 2005, at 9:23 AM, Mike Matthews wrote: > >>> Have you tried this: > >>> > >>> http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup > >>> > >>> Zanzamar Majere wrote: > >>>> Thank you for the response. I still have the errors mentioned > >>>> below, sip response and Failed to authenticate on INVITE > >>>> > >>>> [PPPPPPPPPP] > >>>&...

Today, We have added INVITE Authentication. This seems to bring a large amount of problems to people in the way since they can't make outbound calls. Here's what needs to be done. You need to add three variables to your peers or friends, username, authuser, and secret. username=<phonenumber> authuser=<phonenumber> secret=<registration password> Dan

I have configured asterisk with the AMP php configuration utility. I am able to make outgoing calls through broadvoice but incoming calls are sent to BV's Voicemail and never actually enter the IVR. When I show sip debug info through the asterisk prompt it actually reads the incoming call from BV but then issues a busy signal sending the call to BV's voicemail. I also modified