similar to: Asterisk segfault in dmesg

Dear listers, I'm facing a puzzling situation with Digium TDM2400 card (12 FXO / 12 FXS). Once a day or so we detect 1 or 2 zombie FXO channels. These can be either outbound or inbound calls. I thought this could be related to obsolete DAHDI or Asterisk versions, so I upgraded to 2.4.0 and 1.6.2.15 respectively (OS: Ubuntu 10.04 64 bits). To no avail; the zombie channels keep showing up.

Hello everyone, I have noticed thar our dahdi-channels.conf has some repeating directives, for instance for channel 2 (FXO) we have these settings: ;;; line="2 WCTDM/0/1 FXSKS" signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel => 2 callerid= group= context=default As you can see, a few directives are repeated (callerid, group, context). This was generated by

POTS line already did. It's like a zombie channel. When this happens we issue "hangup request DAHDI/x-x". Anyone faced similar issue? Current configuration: Ubuntu 10.04 64 bits, Asterisk 1.6.2.7, DAHDI 2.3.0. Thank you, Alex Saavedra --0016364d1dc91459020496ed0bbe Content-Type: text/html; charset=UTF-8 Content-Transfer-Encoding: quoted-printable <div

Dear R users,, I'm having difficulties when i use the gstat extensions in R especially with the graphical interface, as the variograms plots are depicted with gray background and cyan points for the number of pairs, Consequently the fitted variogram is drawn in cyan, How can i change that so i could have, white background and black lines for the fitted variogram, for the complete plot.

Hola, estoy tratando de hacer un boxplot para compara diferentes comportamientos. Tengo un set de datos con 17 columnas, quiero crear un boxplot considerando 3 de ellas. He buscado en foros y siempre responden esta pregunta con subconjuntos de datos dentro de una misma columna, y no he logrado realizarlo. Este es un ejemplo de mis datos: ID: id animal EP: episodio AA: acostado y alerta AD:

Hello list, what is the difference between these 2 codecs ? What codec to choose if bandwith is an issue ? (like in most cases I guess) Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101208/d611534a/attachment.htm

Buenas noches, Soy nueva en R y a veces me cuesta pensar los cálculos de manera más práctica, por los que les agradecería la ayuda. Tengo una matriz de datos con una dim = 35745 19, correspondientes a ubicaciones de 39 perros, cada perro tiene poco más o poco menos de 1000 datos. Necesito saber el % de uso de hábitat natural. Es un sencillo cálculo de porcentaje, pero me gustaría hacerlo mas

Hello Everyone We are upgrading to Samba4 from Samba 3.x, we got a fully functional domain over the last 4 years and we are trying to do our migration as smooth as possible. I have migrated, users, machines and everything to the new samba4 domain (not in production yet), however when i log and old machine from the samba 3.x domain to the samba 4 domains, the local policy still applies, i suppose

The Asterisk Development Team has announced the release of Asterisk 1.6.2.7. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.7 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.6.2.7. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.7 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

Hello, I just upgraded DAHDI from 2.3.0 to 2.4.0, both dahdi-linux and dahdi-tools (Ubuntu 10.04 64 bits). It took a few minutes, and it was straightforward. Everything is working. Now I'm preparing to upgrade Asterisk from 1.6.2.7 to 1.6.2.15. I made a backup of configuration files, codec licenses and CDR. Is there something else I should be aware of before upgrading? Thank you, Alex

Hi, I'm trying to connect an HDL F10 device for a friend living in Brazil to the TDM2400 on his Asterisk server. That device should behave like a normal doorbell and it is if connected to an analog PBX. I connected to the TDM2400 and everything works fine except for one thing: when the called party hangs up his phone, the F10 HDL device does not hang up. I'm not brazilian and not

Hi List... I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's, and it also has the echo canceller... I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel 2.6.9-34.0.2.EL I'm using Polycom's 501 with the SIP 1.6.2.0041 The problem is when someone dials to or from the PSTN through the TDM2400, the voice quality is crappy...Instead of hearing:

Dear Mr. Baddeley   I just graduated from a Masters in Applied Mathematics on Jun19th. My thesis was about spatial distribution /a nalysis of some trees in a part of  the Congo Basic Forest. In my thesis I used your spatial package in R, and today I'm doing a more deeper study about how we choise the r distance in Function F(). I would like to be able to understand about value of rmaxdefault

Hola Lorena: Lo prometido es deuda. Dije que miraría si iba este asunto: ggplot(data = dLSaa, aes(x = factor(AA), y = AD, colour = factor(AA))) + geom_boxplot() + xlab( "AA" ) + ylab( "AD" ) + labs(colour = "Leyenda") + ggtitle(label = "Comparación de AD frente a AA") Y este es el resultado: [image:

Hello In our company we have 5 remote office, i'm trying to setup a replicated domain across this sites, using vpn, is multiple DCs soported? can i use the RODC option or is not yet implemented? I try top setup the replication between this servers but i'm starting to get several WERR_BADFILE error and now i get a WERR_DS_DRA_INTERNAL_ERROR Appreciate any help

Hello, I have been trying to completely disable music on hold on my asterisk system. When a call is put on hold I do not want any music on hold, but I would like the remote user to get informed of this event (depending on the technology e.g. with a SIP reinvite and an SDP indicating the call is on hold). I have searched and tried out various approaches, but when putting the call on hold

Hello, I'm having issues connecting throu PRI with the following error "Requested transfer capability: 0x00 - SPEECH" Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660 at voipphones:1] Set("SIP/4856-00000003", "CALLERID(num)=xxxxxxxxx") in new stack -- Executing [97052660 at voipphones:2] Dial("SIP/4856-00000003",

Hi, I have installed Asterisk 1.8 and am using realtime dialplan contexts from a mysql table. Asterisk keeps segfaulting. When I trace the thread ids associated with the segfaults in the full log, all they have in common is netsock2.c: == Using SIP RTP CoS mark 5 which is probably a red herring since it appears so often in threads that do not segfault. When compiling, I ticked all the

Hey Guys! I have two asterisk 1.8.3.2 same version on both machine but why one asterisk has "reload" command but other doesn't ? satish-desktop*CLI> core show version Asterisk 1.8.3.2 built by root @ satish-desktop on a x86_64 running Linux on 2011-03-25 16:10:39 UTC satish-desktop*CLI> re <tab><tab> realtime reload shirley*CLI> core show version Asterisk