[asterisk-users] Queue Timeout
Hello, We're running asterisk 16 with Realtime. We have queues configured in realtime. The "Timeout" setting appears to have an upper 2 minute limit. Even when setting the timeout in the queue to 600 seconds, the agent is no longer rung after exactly 120 seconds. The asterisk CLI claims "Exiting due to time-out cycle". We are calling the queue with options "tin". Removing "n" does keep the entire queue from exiting. But the agent is still stopped from ringing and then rung again after the announcements fire. I have also tried dynamically passing a timeout (and not) when calling the queue from the dial plan. IE. Queue(queuename,tin,,,600). This does make the queue completely exit after 600 seconds. But does not cause the agent to just ring for 600 seconds straight. We are answering the call in the dial plan first before entering the queue. So this is not an instance of the incoming call being canceled by some underlying carrier for being in the ring state > 60 seconds. Any thoughts? -- _ -- 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
Re: [asterisk-users] Odd issue CentOS 7 vs Ubuntu 20.04
On Fri, Feb 4, 2022 at 12:42 PM Jerry Geis wrote: > > > On Wed, Feb 2, 2022 at 1:06 PM Jerry Geis wrote: > >> >> >> On Wed, Feb 2, 2022 at 10:44 AM Jerry Geis wrote: >> >>> >>> >>> On Wed, Feb 2, 2022 at 9:26 AM Jerry Geis wrote: >>> So I have CentOS 7 server running asterisk 18.8.0 - all is good. I unplug that server - plug in a ubuntu 20.04 server at the same IP address. let my 3 devices reconnect to the ubuntu server When I pick up the polycom phone and dial it connects. I hear the other ends 'tone" - but when I press digits - nothing happens (to select a port) Seems everything is set for rfc2833. The devices are a TOA SIP Gateway, and a TOA N-8000 device connected to the GW. I have compared the settings of the polycom extension on both boxes - they match and also the SIP gateway. I tried to compare the sip debug from the Ubuntu to the centos and "looked" the same to me. Where might I look next or what might I look at ? Thanks, Jerry >>> >>> >>> ok - if I "rtp set debug on " on the CentOS 7 server I get a tone of >>> logging. >>> >>> if I do the same on the ubuntu 20.04 all i get is like 2 lines. >>> I have done "systemctl stop firewalld" on the ubuntu box - same result. >>> >>> Where do I look next ? >>> >>> Jerry >>> >> >> >> I dont get it - I certainly getting RTP traffic because I defined an >> extension to playback the demo-congrats messages. >> I call that extension - and ALL kinds of RTP traffic prints on teh >> console. >> >> But when I call the one extension - 103 - all it prints is 2 lines. >> >> I also removed the source tree - un tarred - ran the >> contrib/scripts/install_prereq install script, it did install a couple >> packages - I dont think they mattered. >> do the ./configure, make, make install and started up again - same issue >> though. >> >> Jerry >> > > > > So - still on this... > > I was just dialing the SIP Gateway with Dial(SIP/103) > > if I change my Dial command to this: > > Dial(SIP/103,20,D(15)) > So I send out the DTMF in the dial command - this works and connects me > and the DTMF is delivered and I get the right port. > > The problem still remains - Dialing just Dial(SIP/103) from the polycom > phone - and then doing 15 for DTMF does not work. Cant figure out why ? > > Any thoughts ? > > Jerry > This ended up being a simple canreinvite situation... I had yes - and needed to be set to NO. Jerry -- _ -- 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
Re: [asterisk-users] app TRANSFER PJSIP URI format (SOLVED)
mistake in __TRANSFER_CONTEXT Dne 07/02/2022 v 11:17 marek napsal(a): hi, i have two asterisk boxes need transfer call from second box to first one pstn -> asterisk1 -> dial(number 555) -> asterisk2 -> TRANSFER (number 444) -> asterisk1 dialplan on asterisk1 (using chan_sip) [some_context] exten => 555,1,Noop() same => n,dial(SIP/asterisk2/555) exten => 444,1,Noop() dialplan on asterisk2 (using chan_pjsip) exten => 444,1,Noop() same => n,Answer() same => n,Transfer(PJSIP/sip:4...@example.com) but its not working on asterisk1 i see chan_sip.c: Got OK on REFER Notify message but in NOTIFY message from asterisk1 to asterisk2 is SIP/2.0 404 Not found so what's the correct URI for TRANSFER? thanks Marek -- _ -- 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
[asterisk-users] app TRANSFER PJSIP URI format
hi, i have two asterisk boxes need transfer call from second box to first one pstn -> asterisk1 -> dial(number 555) -> asterisk2 -> TRANSFER (number 444) -> asterisk1 dialplan on asterisk1 (using chan_sip) [some_context] exten => 555,1,Noop() same => n,dial(SIP/asterisk2/555) exten => 444,1,Noop() dialplan on asterisk2 (using chan_pjsip) exten => 444,1,Noop() same => n,Answer() same => n,Transfer(PJSIP/sip:4...@example.com) but its not working on asterisk1 i see chan_sip.c: Got OK on REFER Notify message but in NOTIFY message from asterisk1 to asterisk2 is SIP/2.0 404 Not found so what's the correct URI for TRANSFER? thanks Marek -- _ -- 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
