similar to: CallerID when transferring calls.

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 8351 bytes Desc: image001.gif Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050601/65fd0455/attachment.gif

Today is my first day with sieve, so be gentle :) I'm trying to set up a pretty webmail interface to our Dovecot 1.2.4 server using roundcube. The managesieve config + roundcube 'managesieve' plugin work fine, and I'm able to use roundcube's UI to generate .dovecot.sieve files. We use winbind + LDAP lookups to do some exotic mail rewriting... ultimately user.name at domain.com

Hi Can anyone provide me with a Manchester (0161) UK DID number, preferably IAX2 but SIP is ok too, that I can use for my incoming calls? Call volume will be low. The critical thing is that DTMF must be correctly passed 100% of the time, unlike Sipgate, my current (free) provider, whose DTMF detection/passing is not at all reliable, making it useless for a virtual receptionist scenario. I

I known. I'm using the 1.44 firmware version relesed on 26 may. I worked for italian IVR an HTTP pgaes. So i can only update asterisk with CVS and try atxfer. Thanks for all -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill Inviato: luned? 30 maggio 2005 18.40 A:

Good...it almost works fine! I just have 't' in command Dial, but i also have 'T'. Is it a problem ? This is my Dial() exten => 605,1,Dial(${GIORDANO NAT},60,Ttr) I have only a problem: A and B are speaking, B calls C and ask it if wants speak with A, C accept but if B hang up A is waiting and C get busy tone. To make it works B don't have to hangup but habe to press

Hi :) I'm having enormous probs with the Budgetdedicated APT repo on Ubuntu dapper... 0.9.17 works a treat, but 0.9.18 and 0.9.19 immediately call the debugger on /any/ app, even winecfg + /usr/bin/notepad.. I've tried deleting my .wine dir, and always ensure there is no wineserver hanging around, but always to the same end effect: gdh@plip:~$ rm -r .wine gdh@plip:~$ notepad wine:

Hi, I'm putting together a system to manage agents with Realtime, and without chan_agent. In 1.2.13, there's a handy (although marked as deprecated in apps/Makefile) PGSQL application to let me do this: macro queue-addremove(queuename,penalty) { switch(${MACRO_EXTEN:0:1}) { case I: // Login PGSQL(Connect connid host=XXX user=XXX password=XXX

hi we need a predictive dialer which can be used with asterisk software. Is it possible? Bye Owais Bin Zuber --------------------------------- Do you Yahoo!? Yahoo! Search - Find what you’re looking for faster. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040310/a49a4b5c/attachment.htm

I have been impressed with the quality and meagre bandwidth of the G729 codec from Digium. I am in a testing phase of our roll out, we are using 5 Asterisk PBXs in various countries to provide connectivity for our employees, owners and family. As we are testing, and our setup is somewhat complex due to the peculiarities of our connectivity, there has had to be a lot of changes to servers, cards to

Hullo. Having found http://bugs.digium.com/bug_view_page.php?bug_id=0000875 I grabbed the original 0.0.1 and Dan's patch, and whilst it didn't apply, I was able to patch the zaprtc.c manually - the Makefile has changed a lot, and I wasn't able to understand the changes. (this is all on a machine that's never had any * on it before, and has a 2.6.5 kernel with a matching

Mm, I have a big Quad-Opteron.. thing.. that I''m trying to get Xen onto. I''ve used the 3.0.2 binary-install mode, updated menu.lst as per the README, but I need an initrd which contains the HP cciss RAID driver, and no Xen initrd image was installed into /boot. Now I notice xen-3.0.2-2-install/install/lib/modules/2.6.16-xen/kernel/drivers /block/cciss.ko But I

Hello, I have a little trouble when entering into a domU console (xm console mydomU) : i can''t use the CTRL-C sequence to stop a script/command (like a continuous ping for example) Is there any parameter / tip for that ? Arnaud _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Hullo :) Please be gentle with me, I don't have a working * install, and am just looking for background information. I'm always impressed by companies who implement a queue like "You are now number N in the queue. There are currently M agents answering calls, and your call should be answered in approx. O minutes" I've seen on

Nothing much to be said.. I backported ringinuse, autofill and the QueueLog application from 1.4.0 to 1.2.14. Any or all may be applied - order doesn't matter. They have received minimal testing but appear to function correctly. As always with these things, don't blame me if they connect your callers to a phonesex line, etc. http://bum.net/patches/ Cheers, Gavin.

Hi, I'm trying to use ChanIsAvail to build a resilient 'dialout' macro. The logic is simple; try Zap/g1 (a group of two E1s), and if that fails, try locating a channel via DUNDi. Here's a massively cut down version to illustrate the problem I'm having. macro dialout ( dest ) { ChanIsAvail(Zap/g1); noop(Value of AVAILCHAN is ${AVAILCHAN});

Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension

Hello. I'm currently evaluating the Sipura SPA-841, and snom 190 phones for use in an Asterisk PBX/call center environment. One feature the SPA-841 has, which I can't figure out how to implement on the snom 190, is the "make/accept calls without registration" feature. Or more specifically, "produce a dial tone even if I'm not registered." I would like to set our

Hi, Quick question. With an old phone system a receptionist receiving a call has 1 button to push to transfer calls to a specific extension, with Asterisk, a receptionist would actually put the caller on hold, pick up another line, call the extension, ask if the person is available, hang up pick up the caller again and transfer. To me it's seems a long way to simply do a receptionist

Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's

Hi, We are looking towards a 8+ CO line setup (20 extensions) in our office but we do not want an IVR(auto-attendant) feature. All incoming will be answered by a receptionist. I have read the multi-line configuration for cisco 7960 thread in this list but that way I believe we could only display 6 incoming lines. What will happen to the rest? Does the expansion module for the cisco 7960 work