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? Asterisk seems to either rather crash than to create a queue, or to do nothing.
Le 07/12/2018 à 15:56, hw a écrit :> 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? Asterisk seems to either > rather crash than to create a queue, or to do nothing.Why you don't just add members dynamically in a queu using AddQueueMember/RemoveQueueMember or even with pause/unpause members ? BTW the above dial string has nothing to do with queue, it just a cmd that rings all phones at once. -- Daniel
On 12/07/2018 04:14 PM, Administrator TOOTAI wrote:> Le 07/12/2018 à 15:56, hw a écrit : >> 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? Asterisk >> seems to either rather crash than to create a queue, or to do nothing. > > Why you don't just add members dynamically in a queu using > AddQueueMember/RemoveQueueMember or even with pause/unpause members ?So far, there's only one queue, and it's members are always the same. With dynamic queue members, how do you solve the problem of automatically recreating queues when restarting asterisk?> BTW the above dial string has nothing to do with queue, it just a cmd > that rings all phones at once.Yes --- I was looking for a way to do that, and the only way I found was using a queue. I have two cases in one of which a queue is just right while ringing several phones at once and not having a queue would be better in the other.
Alembic currently doesn't cover queue_logs. As of now it only covers configuration, voicemail and cdr. With best regards Florian Floimair Innovation - Software-Development COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 http://www.commend.com <http://www.commend.com/> Security and Communication by Commend FN 178618z | LG Salzburg Am 07.12.18, 15:56 schrieb "asterisk-users im Auftrag von hw" <asterisk-users-bounces at lists.digium.com im Auftrag von hw at gc-24.de>: 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? Asterisk seems to either rather crash than to create a queue, or to do nothing. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Oh, I didn't know that. Regards, Marcelo H. Terres <mhterres at gmail.com> IM: mhterres at jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On Mon, 10 Dec 2018 at 14:50, Floimair Florian <f.floimair at commend.com> wrote:> > Alembic currently doesn't cover queue_logs. > As of now it only covers configuration, voicemail and cdr. > > > > > With best regards > > Florian Floimair > Innovation - Software-Development > > COMMEND INTERNATIONAL GMBH > A-5020 Salzburg, Saalachstraße 51 > http://www.commend.com <http://www.commend.com/> > > Security and Communication by Commend > > FN 178618z | LG Salzburg > > Am 07.12.18, 15:56 schrieb "asterisk-users im Auftrag von hw" <asterisk-users-bounces at lists.digium.com im Auftrag von hw at gc-24.de>: > > 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? Asterisk seems to either > rather crash than to create a queue, or to do nothing. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users