search for: pb01

...rver mode and then restart Asterisk. In the extensions.conf I have exten => 2950,1,Answer exten => 2950,2,Festival(Hello Adam) exten => 2950,3,Hangup Festival.conf is all defaults, everything is commented out. When I call extension 2950 from my SIP phone, the Asterisk CLI shows: S-ATL-PB01*CLI> -- Executing Answer("SIP/2034-25df", "") in new stack -- Executing Festival("SIP/2034-25df", "Hello Adam") in new stack == Parsing '/etc/asterisk/festival.conf': Found Mar 11 11:24:07 WARNING[18073]: app_festival.c:445 festival_exec:...

Hi, All: For most but not all cases, qbinom is the inverse of pbinom. Consider the following example, which generates an exception: > (pb01 <- pbinom(0:1, 1, .5, log=T, lower.tail=FALSE)) [1] -0.6931472 -Inf Since "lower.tail=FALSE", Pr{X>1} = 0 in this context, and log(0) = -Inf, consistent with the documentation. However, the inverse of this does NOT recover 0:1: > qbinom(pb01,1, .5, log=T, lower.t...

Ok, I have tested with almost all versions both in 2.3.2--*-stable and 2.3.3-*-beta - I am getting same messages: May 9 10:55:26 WARNING[3961]: chan_zap.c:1925 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! and same Down state pb01*CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 Can anybody help me? br, dmitry Dmitry Zhukovski System developer ComX Networks A/S Naverland 31, 2 DK...

...: Sangoma A102 cards testing FIXED Ok, I have tested with almost all versions both in 2.3.2--*-stable and 2.3.3-*-beta - I am getting same messages: May 9 10:55:26 WARNING[3961]: chan_zap.c:1925 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! and same Down state pb01*CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 Can anybody help me? br, dmitry Dmitry Zhukovski System developer ComX Networks A/S Naverland 31, 2 DK-...

I've set up asterisk with my X100P as a home answering machine. Works great so far - answers the phone after 20 seconds, runs the phone tree, emails voicemail, etc. However, the one feature traditional answering machines have that I haven't been able to figure out is how to listen in on the call. Ideally I could just route through Console/dsp and hear it on my speakers. I've tried

...: Sangoma A102 cards testing FIXED Ok, I have tested with almost all versions both in 2.3.2--*-stable and 2.3.3-*-beta - I am getting same messages: May 9 10:55:26 WARNING[3961]: chan_zap.c:1925 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! and same Down state pb01*CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 Can anybody help me? br, dmitry Dmitry Zhukovski System developer ComX Networks A/S Naverland 31, 2 DK-...

I am attempting to update my Asterisk installation from 1.0 to the latest stable version. When I use CVS checkout, I am receiving the following messages on chan_sip.c: RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v retrieving revision 1.510.2.25 retrieving revision 1.510.2.27 Merging differences between 1.510.2.25 and 1.510.2.27 into chan_sip.c M asterisk/channels/chan_sip.c Then, when

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Greetings! Version 1.0.4 of Asterisk, Asterisk-addons, zaptel, and libpri has been released. The releases are available on the Digium ftp site. I have also started a web site called "The Asterisk Development Proxy" where I would like to start centralizing Asterisk development resources. ~ I am hoping that this will become valuable for