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You wrote: >On Fri, Aug 15, 2003 at 03:37:21PM -0700, Rob McMillin wrote: >> This is on rsync v2.4.5 on RedHat 7.3. >> >> If I do something like >> >> rsync ... 'somehost:/path/to/files.*' /local/path >> >> it works fine, but if I do >> >> rsync ... '/local/path/to/files.*' somehost:/pat...

...question(s) (Tony Mountifield) 8. Re: TDM400P Issues (Jeremy Bogan) 9. Re: TDM400P Issues (Jeremy Bogan) 10. Re: X100P and NTL (ex Cable + Wireless) (Stephen Davies) 11. Re: TDM400P Issues (Vic Cross) 12. Re: TDM400P Issues (Jeremy Bogan) 13. Re: Dial Outside SIP address from AGI (Ron McMillin) 14. Re: X100P and NTL (ex Cable + Wireless) (Vic Cross) --__--__-- Message: 1 From: "James Gardiner" <asterisk@crafted.com.au> To: <asterisk-users@lists.digium.com> Date: Tue, 13 Apr 2004 16:12:15 +1000 Subject: [Asterisk-Users] VoiceMailBox wav file format in EMAIL. Repl...

Hi all, I have encountered this problem: if the caller is connected to the callee using Dial() command called from extensions in extensions.conf, there is no problem. But if the same caller and callee are connected using an AGI->exec('Dial'...), the line is disconnected when asnwer. There's a problem bridging. If the codecs are the same on both ends then there is no problem.

Hi all, Is it possible to dial an OUTSIDE SIP address while inside AGI application? For example, within extension context, I could use [from-sip] exten => 7723,1,Dial(SIP/897224@fwd) and this works whereas when I'm inside agi app, $AGI->exec('Dial',"SIP/897224@fwd") and this DOESN'T work. There some errors about invalid argument. If I were to do

Hi all, Do you know if there's any free public SIP proxy server that is more reliable that FWD and Iptel? Thanks Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040415/85192715/attachment.htm

This is on rsync v2.4.5 on RedHat 7.3. If I do something like rsync ... 'somehost:/path/to/files.*' /local/path it works fine, but if I do rsync ... '/local/path/to/files.*' somehost:/path/to globbing fails, yielding an error about being unable to find a file named 'files.*': link_stat /local/path/to/files.* : No such file or directory rsync error: partial

I am trying implement two-stage dialing. Scenario is following: 1. * Dials SIP agent 2. SIP agent answer the phone and provide dial tone 3. * Sends DTMF string 4. "Bridge" channel with calling party I thought that something like: exten => _2XX,2,Dial_but_not_connect_(SIP/BYEXTENSION,10) exten => _2XX,3,Wait,1 exten => _2XX,4,SendDTMF($DTMF_DIGITS) Should do it. Thank

...n@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. Re: Asterisk Security Audit? (Steven Critchfield) > 2. DTMF Detection Problem (Ron McMillin) > 3. Re: Caller entered digits ignored during wait.... (Tilghman Lesher) > 4. Re: Sipcall.co.uk & [*] (Dave Cotton) > 5. Re: IAX2 trunk mode over satellite (clive18@webmail.co.za) > 6. Register vith SIP provider from behind NAT (Simon Brown) > 7. Can't talk on...

...n@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. Re: Asterisk Security Audit? (Steven Critchfield) > 2. DTMF Detection Problem (Ron McMillin) > 3. Re: Caller entered digits ignored during wait.... (Tilghman Lesher) > 4. Re: Sipcall.co.uk & [*] (Dave Cotton) > 5. Re: IAX2 trunk mode over satellite (clive18@webmail.co.za) > 6. Register vith SIP provider from behind NAT (Simon Brown) > 7. Can't talk on...

This might have been talked about before, but I'm posting anyhow. I've got down to testing Asterisk yesterday, and I couldn't help but compare it with Skype (a Windoze only product, yet, but extremely efficient for some reason). Skype has almost unperceptible delay (LAN), while there is almost half a second of delay regardless of the codec on Asterisk. An even if we were to

I''m a complete newbie when it comes to Ruby on Rails. I have lots of experience in PHP, but yesterday was the first time I sat down and began developing using Ruby on Rails (I''m impressed!). Anyway, I''m having a problem with edit/update of records when using form validation. Here are my files for reference. ticket.rb ============ class Ticket <

Sorry to bust in, folks, but I'm having some trouble integrating the OpenSSH RPMs in my Red Hat 6.1 system and was hoping somebody could point me at an appropriate place to ask my newbie questions. (I've been over the docs and they don't seem to apply to the problems I'm having.) -- http://www.pricegrabber.com | The best deals, all the time.

I have several machines that must have trusted root accounts, that is, I need to be able to run "ssh targethost command" on each by each, for the root user. I have had no success thus far doing so. Normally for the non-root users, all I have to do is set the /etc/ssh/known_hosts, build up the users' ~/.ssh/known_hosts, and the users will work without requiring passwords. (I'm

Hi all, If I'm just using Asterisk as PBX and calls going through between ouside lines and inside extensions, (not using any softphone running on the asterisk pc), does what soundcard I use affect voice quality at all? Do I have to get a full duplex soundcard? Thanks Ron