similar to: Function transfer RFC 5589

Hello, Throwing calls from Asterisk to PSTN (via a VoIP gateway) some operators sends an explaining audio, in situations as: The phone number does is not assigned The phone is powered off etc. The audio is sent before the call to be answered. So, in an automatic dialling application I'd like to recognize that audio to know what to do with those calls (queue them to a service, mark as wrong

I have a problem transferring calls multiple times using DTMF sequences (#, *2). The scenario is: Transferee calls Transferor 1 Transferor 1 transfers to Transferor 2 Transferor 2 transfers to Transfer Target When Transferor 2 enters '#' or '*2', Asterisk no longer reacts and the call remains with Transferor 2. I have tested this with Asterisk 11.2 and 11.5 and

hi, I'm trying to configure a Cisco IP Phone 7911G in order to work with Asterisk. I have loaded the 8.3.3 SIP Firmware of Cisco through a DHCP and a TFTP server. All seems ok but a file that is downloaded : term06.default.loads (I understand that is for 7906 model) instead of term11.default.loads (I understand that is for 7911 model). In any case the phone reboots well. At this moment I

hello, I'm trying to use a remote SMB/CIFS share to backup a local directory tree. I use rsync "version 3.0.4 protocol version 30" and mount.cifs "1.10" with a kernel 2.6.18. The directory I'm trying to backup is quite big and has a lot of files inside. [root@linux dir]# du -hs . 50G . [root@linux dir]# find . -type f | wc -l 41929 I get the following error when

Hi, I read on http://nouveau.freedesktop.org/wiki/TestersWanted that you're looking for information for the integrated tvout functionality. I'm the author of nvtv, and I know some bits and pieces about that subject, so maybe we can pool efforts. The TV encoder used in the NVidia GPUs may be a Zoran CVE IP core, http://www.zoran.com/products/literature/index.html For the GF4 MX,

Bug report follows. Please CC me if you want me to read the reply, as I'm not subscribed to ext3-users. [1.] One line summary of the problem: Ext3 has just causing filesystem corruption which required fsck to fix. [2.] Full description of the problem/report: I just had my machine lock up with an ext3 oops, preceeded by a bunch of error messages in the logs. On rebooting, after replaying the

While on conversation with another party, I dial the atxfer key sequence. Asterisk says "Transfer" then gives you a dial tone, while put the other party on hold music. I dial the transferee number and talk with the transferee, then I hang up and the other party must be connected with the transferee. But this doesn't work the transferee hears a beep. -- Playing 'beep'

Hello, I'm trying to find more information about this Remote Attended Transfers, as is explained in https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Remote+Attended+Transfers for Asterisk 12 using pjsip stack Was Remote Attended Transfer implemented in previous versions of Asterisk (versions without PJSIP, Asterisk 11 and previous)? Where can I find configuration examples to do it work

I am trying to get alpha centrauri to work under wine. I know it exists on linux version also, but I already have bought the windows version some time ago (stupid me) so I would like to at least TRY to get it working.. so, my problem is, that I can start it, but I am missing some colors, I think, and graphics perhaps.. the first menu that comes, where you select the game type etc, looks OK

Hey all, Two issues with HL-CS. First, my system specs: XFree 4.2.0 stock Redhat one - January 23. 4.2.0-8 Redhat 7.3 - WineX(and Wine 20020228) 512mb RAM Matrox G400 card. All my OpenGL stuff works(ie. HL in single player mode, with the nice voice over narration et al.) So first issue: On startup, halflife complains that Could not open MCI file for playback: 266: There is an undetectable

This is a maintenance release of Matrox DDX. Just for disclosure, the code is currently broken for Millennium, Millennium II, and G550. For those models, you may have to disable acceleration to workaround the issue. (the workaround is untested) The code is untested on Mystique and G100 due to lack of access to the hardware at this time. The code works generally okay on G200, G400, and G450. That

Hi, In 1.6.1, it seems Attended Transfer do not behave exactly behave like Blind Transfer when transferer hangs up before callee answers : - in Blind Transfer, caller (ie transferee) is hearing Ringing tone when callee's phone is ringing - in Attended Transfer, caller (ie transferee) is hearing Music On Hold when callee's phone is ringing - in Attended Transfer, if callee don't answer

Hi All I have just finished a re-install of a box that was previously working fine under 4.7. The problem is that the matrox drm kernel module is failinf to initialise correctly. :from the dmesg: FreeBSD 4.8-STABLE #0: Thu May 8 11:41:26 SAST 2003 root@shagrat.prv.moria.org:/usr/src/sys/compile/SHAGRAT48 Timecounter "i8254" frequency 1193182 Hz CPU: Intel Pentium III

Upgraded from 16.2.14 to 1.6.2.15 on Fedora 13, with aastra 9133i and 57i. On 9133i and 57i: #<extension># works for a blind transfer. Xfer<extension>Xfer doesn't! All this worked on 1.6.2.14. Nothing useful on cli, verbose 3, DEBUG. Here extension 169 answers an outside call, and tries to transfer it to 145 using the Xfer button: -- SIP/169-0000009c answered

I'm trying to use Asterisk app_parkandannouce to build a global parking pool from within a couple of Norstar PBXes. Right now I can blind transfer calls into the parking lot, but the slot announcement relies on calling back the 'transferee' after the call is parked and I can't pass enough callerid data out from within the PBX to be able to route the call back in (ie. no PRI

Hi! Maybe someone could help me out? When a call is routed via a2billing AGI and user does a transfer, the call is dropped. If the trunk is called directly everyhing works. Here's a direct scenario (working fine): [pbx000001] exten => 101,1,Set(__TRANSFER_CONTEXT=pbx000001) exten => 101,n,Dial(SIP/pozitel/37129238254,45,t) exten => 102,1,Dial(SIP/12345,60) so, when user calls ext

Hi all, Consider the following scenario. Extension A is in a call with B. At some point, B transfers A to C by sending REFER: A Asterisk B C | | | | |<-- INVITE/200 -->|<-- INVITE/200 -->| | : : : : |

does * support call return? i want when the operator transfers a call if the transferee is busy or doesn't answer the call the call return back to operator again... this feature may be called: call return on busy call return on no answer Paradise Dove

Hi all Any of you guys could point me in the right direction? I need to make that a blind transfer to return to the transferrer when the transferee does not answer. Scenario: . Miss Jane Doe, our front desk attendant, picks up an external call to Mr. Smith; . Miss Doe flashes, dial Mr. Smith's extension and then hangup; . Mr Smith's phone rings until timeout; . At this point, how

I've read the Wiki on using asterisk's built-in transfer options (#8 and #6). They work fine but how does one cancle an attended transfer? Example: I have person on phone, I hit #6 to being att-transfer. I enter Sally's extension. I let it ring for a few seconds. Sally never picks up but her voicemail does. How do I hangup her voicemail and resume the previous call? The example on the