Re: [Asterisk-Users] Asterisk + GrandStream SIP phones

2004-03-30 Thread pesb
Hi, Thanks for the help. You were correct. There was some data missing in the extension.conf file I was able to call one SIP phone from the other. I was even able to call an H323 IP phone registered to the gnugk GK (It has Asterisk registered to him as a GW). But, I have another problem rigth

RE: [Asterisk-Users] Asterisk + GrandStream SIP phones

2004-03-29 Thread David J Carter
What does your extensions.conf look like? Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of pesb Sent: 29 March 2004 18:48 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk + GrandStream SIP phones -This is my 'sip.conf' file:

Re: [Asterisk-Users] Asterisk + GrandStream SIP phones

2004-03-29 Thread Tilghman Lesher
On Monday 29 March 2004 12:25, pesb wrote: I have 2 SIP GrandStream phones, both phones are correctly registered to the Asterisk server. But, when I try to make a call from registered phone '1005' to registered phone '1004', dialing 1004, Asterisk responds with the 'Status: 404 Not Found'

RE: [Asterisk-Users] Asterisk + GrandStream SIP phones

2004-03-29 Thread David J Carter
Try this small extensions.conf Don't think I have missed owt. My config files are here, you just need to add your own extension numbers. http://www.codepipe.com/id25.htm Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of pesb Sent: 29 March 2004 19:26

RE: [Asterisk-Users] Asterisk + GrandStream SIP phones

2004-03-29 Thread Robinson, Eliot S.
] Asterisk + GrandStream SIP phones winmail.dat

Re: [Asterisk-Users] Asterisk + GrandStream SIP phones

2004-03-29 Thread Tilghman Lesher
On Monday 29 March 2004 13:58, Robinson, Eliot S. wrote: how do you get the phone message button to light when there is a message? mailbox=1234 OR [EMAIL PROTECTED] ; if you're not using the [default] context ; in voicemail.conf -Tilghman

RE: [Asterisk-Users] Asterisk + GrandStream SIP phones

2004-03-29 Thread Sergio Serrano
Try to add a qualify= to sip.conf, and try to exec a sip show peers. In spite of phones appears like register, if you use NAT, your firewall can cut communication. Try the next: Just after phone register call to it, and then wait for a minutes and try to call again. Could you call first time