[Asterisk-Users] Variable Substitution

2003-07-29 Thread Justin Eckhouse
Hi, Can I do variable substitution in the [globals] section of extensions.conf? For example something like this: [globals] EXT_BOB=4206 PHONE_BOB=SIP/${EXT_BOB} Thanks, Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digiu

[Asterisk-Users] Multiple Phones for 1 Extension

2003-07-16 Thread Justin Eckhouse
Hi, I'd like to have a SIP phone at home and at the office and have them both ring when my extension is dialed. Right now I used the same config for the phones (Cisco 7960's). So they both register with the same login & pw. This doesn't seem to work quiet right, where only the last phone to regist

[Asterisk-Users] Cisco 7960 Transfer Call drop problem

2003-07-14 Thread Justin Eckhouse
Title: Message Hi,   I'm having problems with transfer from an analog line via a X100p and Cisco 7960's running SIP.   With an attended transfer the a call comes in, I transfer it to another 7960, they answer I announce the call, press transfer again, the two parties talk for 1-2 seconds the

[Asterisk-Users] No Sound via Sip Phone

2003-07-11 Thread Justin Eckhouse
Hi, I just setup a box with RH 9, and latest asterisk via CVS. The box as a T100P card in it that is currently hooked up to nothing. I did have the sample configs in place via make samples, and the only change I made was to add an entry to sip.conf for my Cisco 7960. When I dial 1000 to get to the

[Asterisk-Users] H.323 Gateway Connection

2003-07-01 Thread Justin Eckhouse
Hi, I'm trying to setup Asterisk to allow users to dial out to the PSTN using a remote box supporting h.323. I'm using chan_h323.so, and I'm able to make outbound calls to a client like netmeeting with a line like this: exten => 242,1,Dial(h323/xxx.xxx.xxx.xxx) And I'm able to receive incoming c