Re: [asterisk-users] Queue Ring inuse is shared ?
I have 1.8.7.0, Realtime queue table with ringinuse set to 0, callcounter set to yes in sip .conf for my SIP members. Above allows me Queue not sending a call to a member when (s)he is on call(Be it from same Queue or any other call). Member can also transfer(through features.conf) a call without any issue. call-limit I think is deprecated in 1.8. --Satish Barot Ahmedabad, India On Sat, Jun 22, 2013 at 2:41 PM, Shanavaz E A shanava...@yahoo.com wrote: Hi, I use asterisk 1.8. My issue is : I have the same SIP members added to two queues. I use realtime configuration and has set the field ringinuse=0 for both the queues. But if an extension is answering the call in one queue, and some new call comes in the second queue, still that extension is ringed. In the queue_log table I am getting RINGNOANSWER events each second for the extension until the call gets answered. Is this a normal behaviour ? Can we prevent it? Can we set not to ring any queue member if he is answering a call either in the same queue or a different queue? Pls guide me. Regards Shanavaz. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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
Re: [asterisk-users] SIP Trunking Mantra (Origination)
Any other experts out there? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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
Re: [asterisk-users] Asterisk-11 loop behaviour
On Mon, Jun 24, 2013 at 2:37 PM, James B. Byrne byrn...@harte-lyne.cawrote: Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 Snom870 Handsets We are in the process of moving to an Asterisk based PBX. At the moment most things work as we wish. However, I have just notices that when I force a reload using 'amportal a reload' I see this loop start in 'asterisk -rvv': Channel Local/s@tc-maint-02a4;1 was answered. Launching NoCDR() on Local/s@tc-maint-02a4;1 [2013-06-24 15:22:32] NOTICE[32678]: pbx_spool.c:402 attempt_thread: Call completed to Local/s@tc-maint [2013-06-24 15:22:32] NOTICE[32678]: pbx_spool.c:402 attempt_thread: Call completed to Local/s@tc-maint == Spawn extension (tc-maint, s, 5) exited non-zero on 'Local/s@tc-maint-02a4;2' -- Attempting call on Local/s@tc-maint for application NoCDR() (Retry 1) -- Executing [s@tc-maint:1] NoCDR(Local/s@tc-maint-02a6;2, ) in new stack -- Executing [s@tc-maint:2] Set(Local/s@tc-maint-02a6;2, TCMAINT=RETURN) in new stack -- Executing [s@tc-maint:3] Gosub(Local/s@tc-maint-02a6;2, timeconditions,1,1()) in new stack -- Executing [1@timeconditions:1] GotoIfTime(Local/s@tc-maint-02a6;2, 08:00-17:00,mon-fri,*,*?truestate) in new stack -- Goto (timeconditions,1,9) -- Executing [1@timeconditions:9] GotoIf(Local/s@tc-maint-02a6;2, 0?falsegoto) in new stack -- Executing [1@timeconditions:10] ExecIf(Local/s@tc-maint-02a6;2, 0?Set(DB(TC/1)=)) in new stack -- Executing [1@timeconditions:11] Set(Local/s@tc-maint-02a6;2, DEVICE_STATE(Custom:TC1)=NOT_INUSE) in new stack -- Executing [1@timeconditions:12] ExecIf(Local/s@tc-maint-02a6;2, 0?Set(DEVICE_STATE(Custom:TCSTICKY)=INUSE)) in new stack -- Executing [1@timeconditions:13] GotoIf(Local/s@tc-maint-02a6;2, 0?ext-group,417,1) in new stack -- Executing [1@timeconditions:14] Set(Local/s@tc-maint-02a6;2, TCSTATE=true) in new stack -- Executing [1@timeconditions:15] Return(Local/s@tc-maint-02a6;2, ) in new stack -- Executing [s@tc-maint:4] System(Local/s@tc-maint-02a6;2, /var/lib/asterisk/bin/schedtc.php 60 /var/spool/asterisk/outgoing 1) in new stack -- Executing [s@tc-maint:5] Answer(Local/s@tc-maint-02a5;2, ) in new stack Channel Local/s@tc-maint-02a5;1 was answered. Launching NoCDR() on Local/s@tc-maint-02a5;1 [2013-06-24 15:22:32] NOTICE[32681]: pbx_spool.c:402 attempt_thread: Call completed to Local/s@tc-maint [2013-06-24 15:22:32] NOTICE[32681]: pbx_spool.c:402 attempt_thread: Call completed to Local/s@tc-maint It is not an infinite loop but it does go on for an inordinately long time. Does anyone here recognize what is happening and can provide me with an explanation? Since it is pbx_spool doing the processing, you probably have something creating a callfile in /var/spool/asterisk/outgoing on startup (or periodically). I did a quick Google search and found out that this particular context is used by FreePBX 2.9's Time Conditions feature - see http://www.freepbx.org/forum/freepbx-distro/distro-discussion-help/odd-failed-calls-in-logs for more information. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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
Re: [asterisk-users] Asterisk-11 loop behaviour
On Tue, June 25, 2013 09:57, Matthew Jordan wrote: On Mon, Jun 24, 2013 at 2:37 PM, James B. Byrne byrn...@harte-lyne.cawrote: It is not an infinite loop but it does go on for an inordinately long time. Does anyone here recognize what is happening and can provide me with an explanation? Since it is pbx_spool doing the processing, you probably have something creating a callfile in /var/spool/asterisk/outgoing on startup (or periodically). I did a quick Google search and found out that this particular context is used by FreePBX 2.9's Time Conditions feature - see http://www.freepbx.org/forum/freepbx-distro/distro-discussion-help/odd-failed-calls-in-logs for more information. Thank you. Could I ask what search term you used for google? -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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
Re: [asterisk-users] Asterisk-11 loop behaviour
On Tue, Jun 25, 2013 at 12:18 PM, James B. Byrne byrn...@harte-lyne.cawrote: On Tue, June 25, 2013 09:57, Matthew Jordan wrote: On Mon, Jun 24, 2013 at 2:37 PM, James B. Byrne byrn...@harte-lyne.cawrote: It is not an infinite loop but it does go on for an inordinately long time. Does anyone here recognize what is happening and can provide me with an explanation? Since it is pbx_spool doing the processing, you probably have something creating a callfile in /var/spool/asterisk/outgoing on startup (or periodically). I did a quick Google search and found out that this particular context is used by FreePBX 2.9's Time Conditions feature - see http://www.freepbx.org/forum/freepbx-distro/distro-discussion-help/odd-failed-calls-in-logs for more information. Thank you. Could I ask what search term you used for google? Sure - freepbx tc-maint But I did that after I looked at the FreePBX dialplan that came with AsteriskNOW and saw that the context tc-maint wasn't in there. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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
Re: [asterisk-users] SIP Trunking Mantra (Origination)
On 6/25/13, Jai Rangi jpra...@didforsale.com wrote: Not a problem, I wanted to tell you the diff between PRI and sip trunking. I am sure there are lots of option we are just fine what ever works best for you. Back to subject we strongly believe that sip trunking is far better option than PRI and that's the way to go in future. Jai Hello Jai the benefits of SIP trunking is well noted. However, there will always be an underline interconnect that makes SIP trunking possible. What I am trying to say is that there will always be ISUP/ISDN trunk groups that we throw a TCP/IP stack on top, and offer clients with SIP trunking. We are looking for information on how to accomplish interconnect using ISUP trunks (i.e., SS7 interconnect). Kind Regards, Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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
