Re: [asterisk-users] Busy problem
Erik Wartusch wrote: > - Got SIP response 486 "Busy Here" back from 172.10.3.31 > I see that response when someone presses the DND button on our Polycom phones. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Busy problem
Hi, I've a huge problem with the following: Setup: Asterisk 1.4.11 I've got two Thomson ST2030s in an queue. After a while Asterisk logs the following if somebody calls the queues number: - Got SIP response 486 "Busy Here" back from 172.10.3.31 -- SIP/office1-0823d190 is busy -- Nobody picked up in 0 ms The phones are NOT busy (show channels show nothing). Also "show queues" says not in use. Then obviously the phones gets stucked (the Thomson phone's display gets freezing) . I have the same problem also now with an Linksys SPA942. Is the Queue implementation buggy? Any idea? Kind Regards, Erik ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Busy problem
Hello, I have a busy problem with Asterisk when I try to transfer a call from PRI directly to IVR. This problem appear sometime after 2 hours or 2 minutes. The log file contain : Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) When this problem appear I must restart Asterisk to solve it. Another thing, I don't know why the alarm is set to NOP on SPAN 2 ?Maybe it comes from here ? Any ideas ? The configuration : PRI (France Telecom) 15 channels <> (SPAN1) Asterisk (SPAN2) <=> IVR Server : - Dell power edge 1800SC - 2 Ethernet cards (LAN + VoIP LAN) - Digium card : TE 405P - Linux Mandriva LE 2005 (10.2) : Linux ASTERISK 2.6.11-12mdksmp #1 SMP i686 Intel(R) Xeon(TM) CPU 3.00GHz unknown GNU/Linux - Asterisk 1.2.4 - Zaptel 1.2.3 - Libpri 1.2.2 ASTERISK*CLI> zap show status Description Alarms IRQbpviol CRC4 T4XXP (PCI) Card 0 Span 1OK 0 0 0 T4XXP (PCI) Card 0 Span 2NOP0 0 0 T4XXP (PCI) Card 0 Span 3NOP0 0 0 T4XXP (PCI) Card 0 Span 4RED/NOP0 0 0 zaptel.conf span=1,1,0,ccs,hdb3 span=2,1,0,ccs,hdb3,crc4,yellow span=3,1,0,ccs,hdb3,crc4,yellow span=4,1,0,ccs,hdb3,crc4,yellow bchan = 1-15, 17-31 dchan = 16 bchan = 32-46,48-62 dchan = 47 bchan = 63-77,79-93 dchan = 78 bchan = 94-108,110-124 dchan = 109 loadzone = fr defaultzone = fr zapata.conf [channels] switchtype=euroisdn pridialplan=national signalling=pri_cpe usecallerid=yes hidecallerid=yes usecallingpres=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=-6.0 group=1 callgroup=1 pickupgroup=1 immediate=no callprogress=no busydetect=no callerid=asreceived group=1 context=from-pstn signalling=pri_cpe channel => 1-15;,17-31 => only 15 first channels on PRI group=2 context=from-ivr signalling=pri_net channel => 32-46,48-62 group=3 context=from-ivr-bis signalling=pri_net channel => 63-77,79-93 group=4 signalling=pri_net channel => 94-108,110-124 Regards Jerome ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Busy - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
Frank Sautter schrieb: * i can't signal Busy to the calling party. asterisk receives busy from the ericsson PBX but does not forward this to the external caller. i tried with exten => _.,102,Busy() with no effect. this is the part of the extensions.conf i'm using: peter svensson gave me the hint to set priindication=outofband now i'm able to signal busy to the calling party and with setting PRI_CAUSE there are even more possibilities see http://www.voip-info.org/wiki-Asterisk+cmd+Hangup regards frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Busy - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
hi, well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * i can't signal Busy to the calling party. asterisk receives busy from the ericsson PBX but does not forward this to the external caller. i tried with exten => _.,102,Busy() with no effect. this is the part of the extensions.conf i'm using: [pri-external] exten => _5678.,1,SetCIDNum(0${CALLERIDNUM}) ; Add a leading zero exten => _5678.,2,Goto(${EXTEN:4}|1) ; Strip trunk digits from the DDI exten => h,HangUp() include => durchwahl include => pri-external-route [pri-external-route] exten => _.,1,Dial(Zap/g3/${EXTEN}) exten => _.,2,Hangup() exten => _.,102,Busy() this is a excerpt from /var/log/asterisk/full a call from a mobile phone (017212345678) to extension 134 which is busy: -- Starting simple switch on 'Zap/35-1' -- Executing SetCIDNum("Zap/35-1", "017212345678") in new stack -- Executing Goto("Zap/35-1", "134|1") in new stack -- Goto (pri-external,134,1) -- Executing Dial("Zap/35-1", "Zap/g3/134") in new stack -- Called g3/134 -- Zap/38-1 is making progress passing it to Zap/35-1 Requested indication 14 on channel Zap/35-1 Received AST_CONTROL_PROGRESS on Zap/35-1 Dunno what to do with control type 15 -- Zap/38-1 is busy Set option AUDIO MODE, value: ON(1) on Zap/38-1 Hangup: channel: 38 index = 0, normal = 63, callwait = -1, thirdcall = -1 Not yet hungup... Calling hangup once with icause, and clearing call disabled echo cancellation on channel 38 Set option TDD MODE, value: OFF(0) on Zap/38-1 Updated conferencing on 38, with 0 conference users Set option AUDIO MODE, value: OFF(0) on Zap/38-1 disabled echo cancellation on channel 38 -- Hungup 'Zap/38-1' == Everyone is busy/congested at this time (1:1/0/0) Exiting with DIALSTATUS=BUSY. -- Executing Busy("Zap/35-1", "") in new stack Requested indication 5 on channel Zap/35-1 == Spawn extension (pri-external, 134, 102) exited non-zero on 'Zap/35-1' -- Executing Dial("Zap/35-1", "Zap/g3/h") in new stack -- Called g3/h Set option AUDIO MODE, value: ON(1) on Zap/38-1 Hangup: channel: 38 index = 0, normal = 63, callwait = -1, thirdcall = -1 Not yet hungup... Calling hangup once with icause, and clearing call disabled echo cancellation on channel 38 Set option TDD MODE, value: OFF(0) on Zap/38-1 Updated conferencing on 38, with 0 conference users Set option AUDIO MODE, value: OFF(0) on Zap/38-1 disabled echo cancellation on channel 38 -- Hungup 'Zap/38-1' Exiting with DIALSTATUS=CANCEL. == Spawn extension (pri-external, h, 1) exited non-zero on 'Zap/35-1' Set option AUDIO MODE, value: ON(1) on Zap/35-1 Hangup: channel: 35 index = 0, normal = 60, callwait = -1, thirdcall = -1 Not yet hungup... Calling hangup once with icause, and clearing call disabled echo cancellation on channel 35 Set option TDD MODE, value: OFF(0) on Zap/35-1 Updated conferencing on 35, with 0 conference users Set option AUDIO MODE, value: OFF(0) on Zap/35-1 disabled echo cancellation on channel 35 -- Hungup 'Zap/35-1' regards frank sautter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users