search for: phonex

On 12/07/2018 03:36 PM, Administrator TOOTAI wrote: > Le 07/12/2018 à 14:32, hw a écrit : > > [...] >> >> Queues seem to be the only way to have several phones ring at once, or >> are there other ways? > > Dial(SIP/Phone1&SIP/Phone2&...&SIP/Phonex,,) > Good to know, thanks! What are the entries needed in the queue_members table when using odbc? Alembic made the primary key so that each queue can only have one entry (What is an interface here?), and there's probably a reason for that. How do you enter several members for a queue...

On 12/06/2018 08:43 PM, Antony Stone wrote: > On Thursday 06 December 2018 at 17:49:25, hw wrote: >>>> How dynamic are changes made in the database? >>> >>> If by "dynamic" you mean "quickly used" then the answer is "immediately". >> >> There's a note in some configuration file saying that dynamic extensions >>

I've just got a batch of 301s and 501s in and am trying to get them to work. I'd like to manually configure everything via FTP rather than the web or phone interfaces, but I can't seem to find a good source of definitions for all the options in the sip.cfg or phoneX.cfg files. Anyone know of any? Also, I'm having quite the problem getting the Polycom SP 501 to send *any* DTMF. Running tethereal, I'm just seeing G.711 packets; no other RTP packets being sent (using RFC2833, supposedly). Relevant info: Asterisk 1.2.0 beta1 PolyCom SP 501, sip 1.5.3...

...to add 2 extensions to a queue name [teknisk] Extension 301 and 302. I have a test setup now which I thought should look like this: When a external call come to my external number (67209611) this will ring for 5 seconds, and then transferred to queue "teknisk" And I thought that internal phonex/extensions 301 and 302 would ring. But, when I ring the external number, it just rings...and rings...until it hang-ups. CLI output shows that the commands are running, but maybe the wrong way, are the queue command routed to my sip provider? Info: 67209611 is my public phone number. 90015103 is...

...text=rwest host=dynamic mailbox=200@rwest callerid=Rob West <200> username=rwest200 qualify=no port=5060 nat=no dtmfmode=rfc2833 canreinvite=no This is the voicemail.conf [rwest] 200 => 1234,Rob West 201 => 1234,Julia Zeiter 202 => 1234,Larry Sallberg This is the phonex.cfg <?xml version="1.0" encoding="UTF-8" standalone="yes"?> <!-- Per-phone configuration in this file --> <reginfo> <reg reg.1.displayName="200 Rob West" reg.1.address="rwest200" reg.1.label="200" reg.1.auth.user...

Greetings, I'm having a miserable time getting Asterisk working with FWD. All the samples show something like... exten => _7., .... How do I get Asterisk to wait until the user is finished dialing instead of trying as soon as it gets the second digit? I can use _7XXX, and dial the FWD 3-digit test numbers fine, but I'd like to be able to dial others... Same problem for outside

I am getting read to roll out close to 100 polycom phones and wondered if any one knows of a program to take a list of MAC addresses, extensions, and names and generate the configuration files? -- Bruce Nortex Networks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060504/d3bb612a/attachment.htm