similar to: Zaptel causes kernel crash - zt_init_tone_state

I''ve upgraded to linux 3.10 and enabled extended inode refs and skinny metadata extent refs with these commands: btrfstune -r /dev/sdc1 btrfstune -x /dev/sdc1 Since then, I have "WARNING: at fs/btrfs/backref.c:903 find_parent_nodes+0x616/0x815 [btrfs]()" showing up like crazy: # grep -c "WARNING: at fs/btrfs/backref.c:903" syslog 181819 That''s after just

Not sure if it''s anything interesting - I had the following entry in dmesg a few days ago, on a server with 32 GB RAM. The system is still working fine. [1878432.675210] btrfs-qgroup-re: page allocation failure: order:5, mode:0x104050 [1878432.675319] CPU: 5 PID: 22251 Comm: btrfs-qgroup-re Not tainted 3.11.0-rc7 #2 [1878432.675417] Hardware name: System manufacturer System Product

hi, is there a possibility to provide german dialtones on an IAXy S100IPWRD? 'language=de' sets only the messages to german (voicemail, etc.) is there something like 'loadzone' as in /etc/zaptel.conf regards frank

In AU we have a number of different dialtones defined for various purposes. >From indications.conf: au <ringcadance> 400,200,400,2000 au dial 413+438 au busy 425/375,0/375 au ring 413+438/400,0/200,413+438/400,0/2000 au congestion 425/375,0/375,420/375,0/375 au callwaiting 425/200,0/200,425/200,0/4400 au

I think I have something misconfigured regarding voicemails. They work great, I have this setup: Sip.conf [ext1] Context=phones Mailbox=201 Voicemail.conf [home] 201,password,name,email@mail Voicemail delivery and all works great but when I check sip extension ext1 (analog phone using a Granstream ATA 286), the stutter tone signaling message waiting does not work. Anything wrong with

Hi, I'm planning to buy a new ISDN-PBX (I hope this is the right term for an ISDN phone system). I would also like to connect it to asterisk. As far as I know there is no ISDN card where I can connect an ISDN-Phone to directly working together with asterisk (please correct me if I'm wrong). So what I was thinking of doing was to get a regular ISDN PBX and add a second internal S0 bus

Hi all. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten => 123,2,Dial(SIP/3000,,wW) ; wW allow one-touch recording During the call, I press *1 but it records nothing. David Morrow

Hi, Just wondering if someone could better explain how the indications.conf file actually affects Asterisk? I am using a Cisco 7940 from my Asterisk system, and have set in indications.conf "country=au" thinking that this would make the dialtones/call progress sound like the familiar Australian tones? However when I call another extension on my system, it still sounds like

Hey all..I'm looking to start with a single FXS card but with 3 extensions for VM purposes only. I'd like to know if there is a way that you can have different stuttering dialtones depending on which extension has a VM. For example If x103 and x104 have VM can there be a distinctly different stutter for each mail box and have them play back at once or back to back so that when you

Hi, I am thinking of using * with IP phones instead of a hardware PBX. The situation is like this. I have 3 different companies having an analogue line for each of them. I want to make sure that when a call comes in, we have an indication on the IP phone which line the call comes from. Is this possible? Do I need IP phones with 3 lines? Are there any IP Phones with at least 3 lines? Thanks,

I have a TDM400P with one FXO module and two FXS modules in it. I also have a Wildcard X101P. After trying hard to get things working on various Intel computers, but having echo problems that made it not really usable, I decided to try it on some older PowerPC (Macintosh) hardware running Yellow Dog Linux. Things started off smoothly. Both zaptel and asterisk seemed to compile okay, and both

I have asterisk running on my personal computer and am using Kphone to connect to it. My provider is broadvoice which is Ulaw and I had kphone connected as GSM. The lag was terrible coming from Pots-->--Broadvoice-->Kphone. About 2.5 seconds! Going the other direction seemed fine. I did a: show translation recalc 200 and see that the translation time should be about 2 ms. When I do the

Hi, I'm trying to configure Asterisk and DISA. Asterisk is working, but I cannot have DISA dialing out. This is a snippet of my extensions.conf: [internal] exten => 1003,1,DISA(no-password|outgoing2) [outgoing2] exten => 1003,1,Playback(beep.gsm) exten => 1005,1,Playback(beep.gsm) My understanding is that if I dial the extension 1003, I should then be redirected to the context

Hello list, Im writing my dial plan, in witch every SIP phone begins with 74 and has more 3 numbers (like 74XXX). So, I want to route all 74XXX calls to my sip channel. For this I wrote this line: exten => s,1,Dial(SIP/74118@193.136.252.5,30,r) but this way all calls go to 74118@193.136.252.5 ..... Then I tried: exten => s,1,Dial(SIP/${EXTEN}@193.136.252.5,30,r) but this way, the

I cant seem to get outgoing calls to be placed properly .. No matter what I try I get an error from the PSTN company saying that the "call can not be completed as dialed" or "you need to dial a one..." The asterisk debugging seems to show the correct number being dialed out of the zap interface... the "9" is being stripped and there is a "1" where it is

I am trying to write a define type which will use an array but in the meantime have an argument pass to it that sets a case. See for example : define link_files ($linkcase) { case $linkcase { "var" : { file { "${name}_exelink" : path => "/var/log/puppet/${name}_log", ensure => link,

Hi, Chris (cc) and I try to get the SGE master monitor work with Apple Leopard dtrace. Unfortunately we are stuck with the error msg below. Anyone having an idea what could be the cause? What I can rule out as cause is function inlining for the reasons explained below. Background information on SGE master monitor implementation is under http://wiki.gridengine.info/wiki/index.php/Dtrace