search for: xfering

...al Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rich > Adamson > Sent: Wednesday, January 12, 2005 3:46 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Xfering a call > >> I'm having an issue when I transfer a call to another SIP extension it > sees >> that the sip phone is not there and goes to voicemail but in my case >> it >> transfers to the main voicemail instead of the users voicemail. >> >> Here is wha...

All, During development of a backup solution with rsync I experienced some failed backups because of RC 23, partial transfer. These were because the application using the data I was backing up, was still active, and had deleted a file inbetween rsync compiling the file list and then transfering the file. After some feedback from Wayne Davidson and JW Schultz suggesting the solution to this,

Hi I have setup forwarding - xfering - where you press # and then the extension. I add t to the dial cmd. My problem is that when you call something like internet banking they want #, but when # is pressed asterisk gets it instead. is there a way around this ? I haven't been able to get asterisk to listen to flash either Alex...

hi! I've got cisco 7960G working with * box. Calls could be Blind Xfered through the phone but not the supervised transfer( Message on the phone: Transfer failed). Even when I put the caller on hold and resume it later, I can't hear the other side but the otherside can hear me. (It shows as the line is connected though. Yet the respective caller entry blinks.) Any suggestions most

I just copied a file system using xfsdump|xfsrestore At least 1 new directory had been created on the source during the xfer (took 9+hours -- 7TB), so I wanted to verify I hadn't missed anything. Using rsync: > rsync --version rsync version 3.1.0 protocol version 31 Capabilities: 64-bit files, 64-bit inums, 64-bit timestamps, 64-bit long ints, socketpairs, hardlinks,

Hi; As you probably know, SER style of handling an incoming call is : 1) try to look-up it from registrar DB 2) if not found there, try to do some thing else Is there any possibility of doing the above at "Asterisk Dial-plan"? Regards Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL:

...com Subject: [Asterisk-Users] Transfering from a device to a queue crashesAsterisk Asterisk crashes with no errors when I transfer from a device (my phone) to a queue Asterisk crashes with no errors. Also if I xfer from a sip device to another and dont wait for the other user to pickup before xfering the call gets dropped. Any ideas? Im using the latest cvs of asterisk, amp 1.10.009 and our phones are polycom 501. _____ Yahoo! for Good Click here to donate <http://store.yahoo.com/redcross-donate3/> to the Hurricane Katrina relief effort. -------------- next part -------------...

Hi , Thought I would keep the same topic as a previous post in June - still is a problem with pre49 Setting dhcp options 209 and 210 is breaking gpxe/pxelinux - everything else being the same, it works fine in 3.86 and not 4.00. ..Inserting following into pxelinux.gpxe ... set 209:string pxelinux.cfg/default set 210:string http://xxxxxxxx.xxx.xx.uk/pxe/ It is hanging with 'Trying to