[Asterisk-Users] Two internet connections cause unecessary bridging of calls

2005-11-17 Thread Keith Waters
Hi All! My asterisk box has 2 low speed ADSL connections (using pppoe straight from my [EMAIL PROTECTED] box). I've routed some IP ranges out over one of the ADSL lines and the other is the default route. The problem I have is that if one of the sip extentions that is routed over the one

Re: [Asterisk-Users] Adding caller name / ID to outbound meetme calls

2005-11-02 Thread Keith Waters
Just to follow up on my post of yesterday, the solution was simple (thanks to the asteriskTFOT book!) Simply add the following line (modified, of course!) to the call file: CallerID: Asterisk 800-555-1212 Regards, Keith - Original Message - I'm calling people on Zap interface using

[Asterisk-Users] Adding caller name / ID to outbound meetme calls

2005-11-01 Thread Keith Waters
Hi All - I'm calling people on Zap interface using /var/spool/asterisk/outgoing and then putting them into a MeetMe. This works 100%, but tends to give unknown name and number on the meetme list command... eg: User #: 01unknown no nameChannel: Zap/1-1 (unmonitored) I really

[Asterisk-Users] app_flash Flash command - flash lasting too long

2004-07-07 Thread Keith Waters
Hi all - I Need to use the Flash commend in app_flash to send a flash on a Zap channel. However, when I try it, it is holding the line too long, and essentially hanging up the line and picking it up again. Is there some way I can shorten the duration of the Flash event? exten = 555,1,Answer

[Asterisk-Users] Call files timeout on Flash command

2004-07-07 Thread Keith Waters
I managed to sort out my earlier query regarding flash times (changed delay in zapata.conf) Now, I am getting a timeout after the Flash command in an outgoing call-file based call: -- Attempting call on Zap/1/108 for [EMAIL PROTECTED]:4 (Retry 1) Channel Zap/1-1 was answered. --

Re: [Asterisk-Users] Busy message

2004-06-23 Thread Keith Waters
There are other users running the latest CVS-HEAD reporting that problem (asterisk segfaults when unable to create channel). Maybe you have to revert to a previous version till the bug is fixed. ( cvs -D ) OK, thanks, will try that (btw, cvs -D is an invalid command) Have you any idea why

Re: [Asterisk-Users] Busy message

2004-06-22 Thread Keith Waters
For example, a correct dialplan for a SIP extension would read: exten = _200Z,1,Dial(SIP/${EXTEN},20) exten = _200Z,2,Voicemail(u${EXTEN}) exten = _200Z,102,Voicemail(b${EXTEN}) exten = _200Z,103,Hangup Hi All... I'm a newbie, just busy getting to grips with asterisk. I've set up the

Re: [Asterisk-Users] Busy message

2004-06-22 Thread Keith Waters
Are you running Redhat or Fedora? If so, read this thread for a solution: http://lists.digium.com/pipermail/asterisk-users/2004-January/031953.html Nope, SUSE SLES 8 regards, Keith ___ Asterisk-Users mailing list [EMAIL PROTECTED]