[Asterisk-Users] Is this a bug or by design? Workaround?

2005-02-18 Thread Stig Andersson
Hi, I need to use the trailing 5 digits of a callerid. callerid may be anything from a length of 4 to 10 digits in this case. Using this: --- SubString,cid=${CALLERIDNUM}|-5|5 Works great, BUT shows this message: The use of Substring application is deprecated. Please use

[Asterisk-Users] bug? Unterminated comment detected beginning on line 0

2005-02-21 Thread Stig Andersson
Hi, Using latest cvs. A comment-line begins with semicolon ; However - if the line contains ;-- or like this ; -- blabla bla -- You get this error and * stops reading that file: Feb 21 13:47:12 WARNING[17393]: config.c:664 config_text_file_load: Unterminated comment detected beginning

[Asterisk-Users] Any luck with attended transfer and ATA186?

2005-02-21 Thread Stig Andersson
Hi, Using latest cvs. I (as many otheres it seems) can't get Attended transfer to work with Cisco ATA186 (using SIP) Has anyone else had any luck? Same with 3-part calling, if one drops off, all are disconnected... /Stig ___ Asterisk-Users mailing

Re: [Asterisk-Users] New Firefly Beta - with SIP and G.729

2004-03-16 Thread Stig Andersson
Hi again, Installed your new release today (after the sip bugfix). Now SIP registers OK with asterisk, but calling fails... Firefly says: Couldn't start call. Asterisk in SIP debug mode shows the registration, but shows no response when firefly tries to call. Using NO stun, asterisk and

[Asterisk-Users] Session numbers?

2004-03-18 Thread Stig Andersson
Hi, The messages produced by asterisk console, in vvv mode, what are the numbers after the brackets? in this example, /4 and /5 = Releasing [EMAIL PROTECTED]/4 and IAX2[ulf]/5 Are these session numbers or? Are they reused? When the first call comes after asterisk is restarted, they begin

RE: [Asterisk-Users] firefly softphone

2004-03-19 Thread Stig Andersson
Yup, also experienced the crashes, however... the version available from virbigage site (1.4) does not seem to support SIP even though it is available as choice when setting it up from the installation. Select IAX during setup. When finished setup, select options and choose codecs (they are

Re: [Asterisk-Users] If you know your party's extension # please dial it now ...

2004-03-21 Thread Stig Andersson
Asterisk doesn't accept keystrokes during playback, use BackGround to play while waiting for keystrokes. /Stig At 08:37 2004-03-21 -0500, you wrote: Hi all, I've built the usual press one for sales, 2 for support IVR which works fine but I'm having difficulty in allowing callers to type in

[Asterisk-Users] Is Wildcard TDM400P capable of sending DTMF callerid?

2004-03-30 Thread Stig Andersson
Hi, Is Wildcard TDM400P capable of sending DTMF callerid? Does asterisk support it? I know X100P does not, but I have found no info as to TDM400P... /Stig ___ Asterisk-Users mailing list [EMAIL PROTECTED]

Re: [Asterisk-Users] Caller entered digits ignored during wait....

2004-03-30 Thread Stig Andersson
Asterisk doesn't accept keys during wait, use Background and play 1 sec silence instead. /Stig At 23:46 2004-03-30 -0600, you wrote: On 2004 Mar 30, at 20:56, Gene Kochanowsky wrote: How would you use the t extension to accomplish this? exten = s,1,Wait(1) exten = s,2,Answer exten =

Re: [Asterisk-Users] PSTN calls do NOT hang up

2004-04-07 Thread Stig Andersson
Hi, Asterisk either need to know when the remote caller ends his call, or it must detect the silence. Simplest solution is to activate silence detection, see voicemail.conf. You may need to do some testing to get the proper silencethreshold setting. Also search the archive, this is a often

Re: [Asterisk-Users] TDM400P Issues

2004-04-13 Thread Stig Andersson
Just a note regarding this issue. I'm using RH9, two X100p and one TDM400 Loading them in this order: zaptel wcfxo wcfxs zapata.conf like this: fxsks=1