similar to: Sip dialback

Hi all, We've been experimenting with the app_queue application, and it works quite well. The only problem we encountered was that outgoing calls (to the operators) aren't logged in CDR. Example, * operators dial a specific number/extension, and AddQueueMember(..) runs (they get added without any problems), and they Hangup. * normal users dial the support/hotline number, get added

Hallo, ich schreibe einfach mal deutsch, und hoffe dass Du das auch verstehst (if not write me back in English). OS X 10.3.9 Ich habe R 2.1.0 installiert, und das Programm hat auch funktioniert. Will wenig sp??ter wieder mit R arbeiten, es l??sst sich aber nicht mehr starten. "Das Programm R wurde unerwartet beendet", w??hrend ich versuche es durch Doppelklicken auf das R.app-symbol

Is it possible to control the dialing format used when making a call from asterisk separately from the inbound format? In my case I have a single robbed-bit T1 configured for featd (to receive ANI and DNIS) on a TE410P. This configuration works fine for inbound calls, however when an outbound call is made on this same T1 asterisk dials outbound in the featd format. I need asterisk to only dial

How does one do translation for calls that come in from other pbx's where the incoming caller ID is an internal extension number on their pbx? Eg. when I get a call from Free-World-Dial the CID shows up as "429102" which is essentially their internal extension number sans any routing prefix. To dial the number back I need to dial the extension with FWD's routing prefix

I've got my ATA's setup with DHCP, host=3Ddynamic, and turned the registration on in the ATA All is well as long as my sip.conf is configured using the phone number like so... [1231231234] type=friend host=dynamic context=main canreinvite=no username=1231231234 secret=worknow disallow=all allow=ulaw allow=alaw It appears to me that the Cisco ATA (v3.1) will register the device using the

Alright, I am basically cheap, and I have a cellular plan which allows for free incoming calls (Nextel). I was wondering if there was any way to do sort of a dialback trick in the extensions.conf. I call into the system from my cell phone (maybe via DISA), I dial an internal extension, and dial a phone number. Then * sends to my cellphone the number dialed thus giving me a in call on the cell. Or

Pardon the newbie post. I installed Asterisk on a "test" system using the Asterisk@home cd image. When you boot from Asterisk@home is installs an O/S and Asterisk on your PC. How cool is that. But I was wondering if someone could point me in the right direction for updating the version that I have. I'm new to CVS, how do I determine what version to build? Is there a primer on how to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Here's a version I modified which grabs either a development or stable verision, and does a backup before updating from CVS. It also asks for addon's and cc. Leif Madsen did the original development and Mark released it. My changes does the minimum changes to previous version, to get what I need. It does the same version checking as

Hi, Just wondering: What are you guys using to deploy software on your windows (7) workstations within the samba4 AD? These are options I know of: * wpkg (but it seems it's development is not too active) * wpkg-gp using the group policies * www.opsi.org (Client Management System for Windows clients) * standard windows software deployment, but I think that's msi-only This page also has

Couldn't see the forrest for all the fascinating tree-like applications that are out there: For future reference, see: http://www.voip-info.org/wiki-Asterisk+call+parking :-) -----Original Message----- From: Kris Boutilier [mailto:Kris.Boutilier@scrd.bc.ca] Sent: August 11, 2004 1:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Inband announcement of parking slot from

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hmm, My music on hold has always worked fine. But I discovered that under incoming IAX2 calls they don't get any MOH! All I could find was a comment saying let me know if you find a solution... Nor does the debugger does say: Started music on hold So it's not starting the MOH, why? I do have it configured and it does play under other

I had the same problem with snom 190 phones. Using the transfer with # instead of "Transfer Button on the phone" worked for me. In my configuration "REFER" was not send, so the transfer with the button on the phone did not work. Guido Hecken -----Urspr?ngliche Nachricht----- Von: steve szmidt [mailto:steve@szmidt.org] Gesendet: Sonntag, 26. Dezember 2004 17:14 An:

I agree with that fact the same questions get posted, but that problem is compounded by the fact the archives are not really searchable. If the were as lease some users would search. The archives need to be fully indexed. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of steve szmidt Sent: Monday, June 27, 2005

On Mar 30, 2011, at 11:08 AM, David Sehr wrote: > All, > > As a part of the PNaCl project we're doing some investigations into how we do linking, and have started looking at llvm-ld. Several of us have heard that this tool is no longer supported. Could someone please clarify? Hi David, It is supported and in use by many projects. Most people who care about LTO have switched to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've forgotten the command to add a vm box, and searching google and wiki I'm surpriced I cannot find it. I'd love to know where this is written, so I can see how I managed to miss it! - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety."

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, A thought occurred to me; Background; In the early days of cable, the cable people seemed clueless to things like over selling bandwidth. But as time went along they got it better and better under control. Of course their natural competitor, DSL, created a bigger demand for them to get things under control. Today cable is often giving

We were just testing forwarding one of our numbers to a VoIP CLEC and ran into an issue that we've seen before but never figured out the cause of. It seems if you call us and immediately hang up - or if the call is forwarded by Verizon (which stills rings about ? of one time) the Zhone detects the ring and answers the call.. The Zhone however DOESN"T detect that it's no longer

I have not been able to find anything that relates to this problem. The agents are using Cisco phones. Calls goes into a queue. but once an agent picks it up it cannot be transferred. However if they call directly to the agents extension it's not a problem transferring calls. It sounds like a misconfiguration but I cannot see what's wrong. Any takers? -- Steve Szmidt "They

Hi Sanjiv, Sanjiv Gupta <sanjiv.gupta <at> microchip.com> writes: > > Hi Mikhail, > > I have one more problem related to driver/llvm-ld at my hand for which i > could not think of any solution. May be you can help. > [...] > > Is this possible in driver? or should I hack llvm-ld? On second thought, I think I'll add a workaround for this even if

I'm sure that this question gets asked frequently, but a quick perusal of the list archives shows that it hasn't been asked in a least a month or so, so pardon any repetition. What is the current state of asterisk on linux 2.6? I ask, because I spent yesterday giving it a whirl, and everything seems to go just fine till the very last minute. Zaptel, libpri and asterisk compile just