search for: kumbe

...s device registers with us username=202 ; Username to use when calling this device before registration limitonpeers = yes call-limit = 2 busy-level = 1 The directive busy-level is ignored.... I've also tried busy-limit but without any result... Thanks -- /*************/ nik600 http://www.kumbe.it

...y plan to add the Skills Based Routing strategy in queues.conf? I think that it will be enough to add an int skill to the struct member and then order the member by skill desc. Is it enough to add this type of strategy in calc_metric in app_queue.c ? thanks -- /*************/ nik600 http://www.kumbe.it -- /*************/ nik600 http://www.kumbe.it

...39; /></response> </ajax-response> But i can't heard nothing on the channel, i've tried to send the tone both on channel and link, but with no results. If i use normal dtmf from keyboards they works properly. What can i check? Thanks -- /*************/ nik600 http://www.kumbe.it

...after the hangup) the call is forwarded to xxx at 10.10.10.2 that is the wrong address. I've tried to force SIP_HEADER(CONTACT) in the dialplan with a Set but it seems that i can't due to security reason. Is it possible to avoid this problem? Thanks -- /*************/ nik600 http://www.kumbe.it

...[|announceoverride][|timeout][|AGI]) So, setting up an appropriate AGI script can i play an audio file (or create it with some tts) to the call? After the AGI script the call is linked with the operator even if there is an Answer into the AGI? Thanks to all -- /*************/ nik600 http://www.kumbe.it

...sk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- /*************/ nik600 http://www.kumbe.it -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160630/a55cb982/attachment-0001.html>

...-o imapc_password=bar -o pop3c_password=bar backup -R -u user at domain imapc: doveadm -o imapc_user=foo -o pop3c_user=foo -o imapc_password=bar -o pop3c_password=bar backup -R -u user at domain pop3c: Or imapc is enough? Do you have any suggestion? Thanks -- /*************/ nik600 http://www.kumbe.it

...;> >> Step 1,2,3 works properly but i'm not able to link the two channels, even >> using redirect,goto or pickupChan. >> >> Any idea or help will be appreciated! >> >> Thanks >> >> -- >> /*************/ >> nik600 >> http://www.kumbe.it >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.ast...

...to (.... transfer) And then manually match information between unique ID and queue_log to consider info on queue A,B,C,D, as a single queue. Or is there some magic sauce to specify an "IVR script" that is executed when a call is in a queue? Thanks -- /*************/ nik600 http://www.kumbe.it

...2. You only have to load DAHDI drivers if you want to take advantage of DAHDI services. One option is to unload DAHDI modules if you don't need them. 3. If you need DAHDI services, you must correctly configure DAHDI. Where am i wrong? Thanks -- /*************/ nik600 http://www.kumbe.it

...irect the current call to the channel2Link, connecting the call in queue with the remote number (?) Step 1,2,3 works properly but i'm not able to link the two channels, even using redirect,goto or pickupChan. Any idea or help will be appreciated! Thanks -- /*************/ nik600 http://www.kumbe.it -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160630/45cfa95c/attachment.html>

...i handle per server? each server will have: 4 core 3.0 Ghz 4 GB of RAM I need an aproximate sizing: 0-100 calls per server ? 100-200 calls per server ? 200-300 calls per server ? 300-400 calls per server? 400-500 calls per server? Thanks to all in advance -- /*************/ nik600 http://www.kumbe.it

...2.149.202.150 (format ilbc) -- Format for call is ilbc -- IAX2/my-iax-provider-361 is circuit-busy -- Hungup 'IAX2/my-iax-provider-361' == Everyone is busy/congested at this time (1:0/1/0) Have you got any idea? Thanks to all in advance -- /*************/ nik600 http://www.kumbe.it

...annels, >>>> even using redirect,goto or pickupChan. >>>> >>>> Any idea or help will be appreciated! >>>> >>>> Thanks >>>> >>>> -- >>>> /*************/ >>>> nik600 >>>> http://www.kumbe.it >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>> New to Asterisk? Join us for a live introductory webinar every Thurs...

...lastdata and lastapp with "ResetCDR" So, what can i do? Is it better to do some customization to generate a CDR event on each dialplan step or is better to parse the logfile and extract the information needed? I'm using Asterisk 1.4.23.1 TIA -- /*************/ nik600 http://www.kumbe.it

...M is the master and TO the slave. Finally, an additional feature could be to save in some metadata info of dst files (ie. file_A.gz,file_B.gz) the checksum of source files if you want to compare them by content checksum. What to you think about that? Thanks -- /*************/ nik600 http://www.kumbe.it -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.samba.org/pipermail/rsync/attachments/20141205/2ac33fd2/attachment.html>

Hi to all. Is possible with the Asterisk 1.4 cli view the current calls and their codec? Thanks to all -- /*************/ nik600 http://www.kumbe.it

...nterface that will be switched from one to the other in case of hardware problem. The question is: can one server with those settings manage up to 200 simultaneous call? The server will receive SIP calls and forward them through a CISCO router. Thanks to all -- /*************/ nik600 http://www.kumbe.it

...GI application problem as i get the "wrong" dtmf tone directly from Asterisk. It's not a phone problem as the same phone may retry and then it works. Is it possible to relate it with the load of the server? Can you suggest me something? Thansk -- /*************/ nik600 http://www.kumbe.it

...ck [Jun 30 21:56:56] WARNING[5617]: channel.c:2558 ast_waitfordigit_full: Unexpected control subclass '-1' -- User entered nothing. Any idea? if i call from number xxx to an extension that goes to testDTMF at cRETEUNICA it works properly. Thanks -- /*************/ nik600 http://www.kumbe.it -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160630/45708fd4/attachment.html>