search for: brellsystem

Tengo en una maquia ubuntu 8.10 el kernel es Linux 2.6.27-14-generic Quiero configurar una tarjeta x100p i usarla con asterisk, asi que descague compile e instale lo siguiente: asterisk-1.4.24 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 Sin embargo no logro configurar la tarjeta con exito, ya probe casi? todo. Esto aparece si ejecuto lspci: 04:06.0 Communication controller: Motorola

I've got a question about codec_dahdi witrh a system running Asterisk 1.6.0.6 and DAHDI 2.1.0.4 with a TE410P card. The system is used primary to route calls between different PRI connections, so no transcoding between codecs is happening as far as I am aware. 1) How can I use codec_dahdi? Would it be useful when passing a call from one dahdi channel to another dahdi channel? 2) I'm

Hello, I have an ISDN-30 connection and a Digium TE-121 with VPMADT032 echo cancellation. Every 30-60 minutes I experience PRI dropping. @@@ /etc/zaptel.conf: loadzone=dk defaultzone=dk span=2,1,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 @@@ @@@ /etc/asterisk/zapata.conf [channels] switchtype=euroisdn usecallerid=yes group=2 signalling=pri_cpe context=incoming channel => 32-46

At the top of my /etc/dahdi/system.conf file is this line: # Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10 2009 -- do not hand edit OK, so how do I adjust the timing source and LBO numbers, and echo cancellers if I'm not supposed to edit this file? Brandon. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm working on an Asterisk system with all of it's PRI ports configured with the LBO setting at 5, like this: span=3,0,5,esf,b8zs As of yet, I am unwilling to change the LBO to 0 to where it probably should be because the system is working and I'm not sure exactly what the LBO does. I'm aware some changes were made to deal with low audio levels. Does anybody know what audio

Asterisk 1.6.0.6 with dahdi 2.1.0.4 is showing a strange "Unrecognized prilocaldialplan" error with the SIP username when a SIP call is dialed to a PRI trunk. The error shows up like this: Unrecognized prilocaldialplan TON modifier: a Unrecognized prilocaldialplan TON modifier: b Unrecognized prilocaldialplan TON modifier: c Where abc is the SIP username. Is this a bug

I've just noticed that mobile phones are starting to support HD Voice. http://thenextweb.com/apple/2012/09/21/iphone-5s-hd-voice-impressive-theres-still-work-done/Is there any hardware capable of connecting HD voice quality calls into Asterisk now? I see that Asterisk compatible GSM hardware is available, but does anything support HD voice presently or should in the future? -------------- next