similar to: multiple port

Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

Hi I have the following configuration | UA1 --|------ asterisk1 -----------------------+ UA2 --|------ asterisk2 -----------------------+ DB UA3 --|------ asterisk3 -----------------------+ UA4 --|------ asterisk4 -----------------------+ | All UA is located in the same area. A seperated PC is used as a centralized DB for storing a common dial plan, user account and register

I recollect that there used to be a fixed, finite limit to the number of call groups that could exist. Does anyone know if that limitation still exists in 1.2.1, or maybe where I could look in the code to find out if it's a fixed length array or not? Thanks. Doug.

Using 1.0.9: If I have: exten => s,1,Dial(SIP/5555&SIP/12345@192.168.1.1) How can I return the DIALSTATUS variable for the second SIP channel ONLY if the second SIP channel is busy, regardless of the dialstatus of the first SIP channel? What I want is, if the second SIP channel is busy go to n+1 or n+101 regardless of the status of the first SIP channel. tia

I'm pretty sure I already know the answer to this, but... Is there a way to copy/transfer/replicate sip subscriptions from one asterisk system to another, for the purposes of HA? You coudln't even write a script to do it I don't think. You can do an 'asterisk -rx sip show subscriptions' but there'd be no way to repopulate it on a second system. Yes/No? Doug.

Is there any technical difference between a T1 cable and a cat5e patch cable as far as using them with Digium T1/E1 cards? Can PRI circuits terminating at a smart jack connect successfully to Digium cards using straight through CAT5e cables? If so, are they using all of the pins in the cable? Thanks in advance

Hi, I have several Polycom 501 connected to asterisk. The phone has an ACD-login function that I'd like to use. But I can't find find much information about this. I've read a post on bugs@digium (http://bugs.digium.com/view.php?id=6119) about this function but I'm not really clear on if this is actually working or not? Has anyone actually used the Polycom ACD-login function

Thanks! Do you have any suggestions on what I might do next. I have the phones, I have asterisk, and I have everything setup. But i can't get the login to work with the Polycom function. Nothing happens...and I can't find any readmes' or manuals. Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com

Hi all, Does anyone know how to get the channel ID on the other side of the call? For example: When SIP/50 calls SIP/21, and the call is answered by SIP/21 I get: SIP/21-6735 answered SIP/50-b456 ${CHANNEL} will show me SIP/50-b456. Is there a parameter or a workaround to get the SIP/21-6735 part? Thanks. Assaf Benharoosh -------------- next part -------------- An HTML attachment was

Is there an app or softphone for meetings that displays the hosts screen like webex or intercall. Jordan Novak -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060321/df90d527/attachment.htm

Does anyone know where I can get replacement handsets for the Polycom SoundPoint IP phones? Or does anyone have any they want to sell? From the looks of it you have to buy a whole new phone to get a new handset. My vendor, TriaTechCOA, told me I had to buy a whole new phone to get a handset, which is pretty ridiculous. Maybe there is a more sane vendor I should be buying from? Thanks, -Ryan

Hi, I am designing a asterisk load balancing model as follow. There are 3 asterisks connected to a single DB and a single server storing all the configuration file and voicemail. Round Robin DNS will distribute the request to asterisks. DNS round robin ---+ asterisk1--------------------------+ DB and file server +---asterisk2-----------------------+

Amit Shah <amit.shah at redhat.com> writes: > On (Mon) 22 Jul 2013 [15:26:22], Rusty Russell wrote: >> Amit Shah <amit.shah at redhat.com> writes: >> > The removal functions act on the vqs, and the vq operations need to be >> > locked. >> > >> > Signed-off-by: Amit Shah <amit.shah at redhat.com> >> >> How can userspace

Amit Shah <amit.shah at redhat.com> writes: > On (Mon) 22 Jul 2013 [15:26:22], Rusty Russell wrote: >> Amit Shah <amit.shah at redhat.com> writes: >> > The removal functions act on the vqs, and the vq operations need to be >> > locked. >> > >> > Signed-off-by: Amit Shah <amit.shah at redhat.com> >> >> How can userspace

Hello, This series fixes a few bugs and races with port unplug and the various file operations: read(), write(), close() and poll(). There still might be more races lurking, but testing this series looks good to at least solve the easily-triggerable ones. I've run the virtio-serial testsuite and a few open/close/unplug tests, and haven't seen any badness. I've marked these patches

Hello, This series fixes a few bugs and races with port unplug and the various file operations: read(), write(), close() and poll(). There still might be more races lurking, but testing this series looks good to at least solve the easily-triggerable ones. I've run the virtio-serial testsuite and a few open/close/unplug tests, and haven't seen any badness. I've marked these patches

On Thu, Aug 08, 2019 at 05:06:05PM +0530, Pankaj Gupta wrote: > The commit a7a69ec0d8e4 ("virtio_console: free buffers after reset") > deferred detaching of unused buffer to virtio device unplug time. > > This causes unplug/replug of single port in virtio device with an > error "Error allocating inbufs\n". As we don't free the unused buffers >

Hello Michael, Thanks for your reply. > > On Mon, Mar 04, 2019 at 06:35:11PM +0530, Pankaj Gupta wrote: > > The commit a7a69ec0d8e4 ("virtio_console: free buffers after reset") > > deffered detaching of unused buffer to virtio device unplug time. > > > > This causes unplug/replug of single port in virtio device with an > > error "Error

Hello Michael, Thanks for your reply. > > On Mon, Mar 04, 2019 at 06:35:11PM +0530, Pankaj Gupta wrote: > > The commit a7a69ec0d8e4 ("virtio_console: free buffers after reset") > > deffered detaching of unused buffer to virtio device unplug time. > > > > This causes unplug/replug of single port in virtio device with an > > error "Error

Has anyone had any success using the MixMonitor() plus "command" as nothing I have tried works. I am using 1.2.1 I did google the archive but couldn't see any mention of anyone using this. What I am hoping to do is run a macro on hangup, current method I am using seems to miss some calls 5% of calls fail to mix / convert to mp3 etc. Was hoping that MixMonitor would fix this.