search for: linuxautrement

Just done a clean install of rc5 on a totally new machine and found the following:- /etc/init.d/asterisk start errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. Dave Cotton

Hello all. Has anyone managed to get SIPCALL.co.uk's service working with the [*] box? I've managed to register with other SIP providers but not SIPcall. The debug just show's [*] attempting to register. But receiving a 401 error everytime. Cheers Matt

...rom 11:00 CET today * crashes at this point. == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg0000.txt': == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg0000.txt': Found Sheriff*CLI> Disconnected from Asterisk server -- Dave Cotton <dcotton@linuxautrement.com>

Hi, I used to have a Grandstream phone connected to Asterisk a few months ago. Worked just great! Then today I do a new install, rather than an upgrade, and all of a sudden I cannot check voicemail with it. No problem calling or receiving call. It simply speeds through the vm greetings but I cannot hear them. If I check the same VM with an analog phone it works fine. So I wanted to check

...lock); //#else static spinlock_t zaptimerlock = SPIN_LOCK_UNLOCKED; static spinlock_t bigzaplock = SPIN_LOCK_UNLOCKED; //#endif Things compile, but I don't know what it actually does. Any comments? > Message: 18 > Date: Wed, 15 Mar 2006 11:18:36 +0100 > From: Dave Cotton <dcotton@linuxautrement.com> > Subject: Re: [Asterisk-Users] Zaptel compile errors on x86_64 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <1142417917.22407.0.camel@RobinHood.LinuxAutrement.local> > Content-Type: text/plain &gt...

...RESERVED-50 NetHandle: NET-50-0-0-0-0 Parent: NetType: IANA Reserved Comment: RegDate: Updated: 2002-08-23 OrgTechHandle: IANA-ARIN OrgTechName: Internet Corporation for Assigned Names and Number OrgTechPhone: +1-310-823-9358 OrgTechEmail: res-ip@iana.org ? -- Dave Cotton <dcotton@linuxautrement.com>

How to change available options (behaving) during listening of voice mail? (They are unnecessarily complicated) For example, I don't want to press 3 (advanced options) and again 3 for envelope. I just want to play envelope. Also, when saving message, I do not want to choose folder, I want that message as default be saved in old messages. And, I don't want to press 6 for next message, I do

Dunno if anyone else has seen this yet: http://www.scmagazine.com/us/news/article/563800/vulnerabilities+put+asterisk+telephone+systems+risk/ -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Running slackware 9.1 with compiled kernel from source 2.6.5 running ok. I have 2 HFC-S chipbased Billion Bipac PCI ISDN BRI cards installed in PC. Would like to use one card as in TE and one in NT mode. System works fine running pbx4linux.But want to use SIP functionality, so I would like to try out the Asterisk. Trying to install the bri-stuff.0.0.2.tar.gz (May 10 2004)package, getting the

I did cvs update on asterisk, zaptel, libpri as of today (November 5, 2003). I also did 'make clean' on each of them. My previous version of asterisk was cvs of September 15, 2003. No other changes have been made to my system other that these updates. when running 'make asterisk' the following error appears term.c:55: conflicting types for `term_color'

I just setup an Asterisk system on a small Shuttle box; I am only using SIP channels and have no FXO/FXS cards. The system works fine in that I can call my inbound number (Broadvoice) and have the system answer and I can make outgoing calls. The problem is that every time I want to change something in the sip.conf file, I have to do a 'restart now' instead of a 'reload' or

Lets talk about SIP and IAX2 1. The good and bad of both 2. What is the better one and why 3. and any other information that maybe use full -- Best regards, Al Bochter Bochter Services (Voip PBX) Toll Free: 866-638-1254 EXT: 250 (Voip PBX) Free World DialUp: 780217 EXT: 250 (Voip) Cellular: 712-432-5401 http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold

...f the mobile proceeded by the 0 to dial out as the destination. Apparently this is a bug with the SNOM 200. If you move it 30cm., they didn't say in which direction, it automatically chooses a phone number and sets up the redirection. It's not 1st April is it? -- Dave Cotton <dcotton@linuxautrement.com>

For those who don't wake up at 5.00 am and start reading /. http://searchnetworking.techtarget.com/originalContent/0,289142,sid7_gci935769,00.html -- Dave Cotton <dcotton@linuxautrement.com>

I just got an email from SIPphone advising that there have been problems with the above firmware and advising to reload from their server. This does in fact reload 1.0.4.35 into the phone. And now voicemail has gone AWOL again. -- Dave Cotton <dcotton@linuxautrement.com>

...pi.c:2205 capi_handle_msg: Command.Subcommand = 0x5.0x81 but when I try and recompile I get this chan_capi.c:60: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) any help would be greatly appreciated. -- Dave Cotton <dcotton@linuxautrement.com>

...middle of the string any more. I have had to change exten => 123,1,SayUnixTime(||AdBY 'digits/at' IM) to exten => 123,1,SayUnixTime(||AdBY) exten => 123,2,Playback(digits/at) exten => 123,3,SayUnixTime(||IM) Peter -----Original Message----- From: Dave Cotton [mailto:dcotton@linuxautrement.com] Sent: 24 June 2004 06:45 To: Asterisk List Subject: Re: -- [Asterisk-Users] Serious issues with current CVS? For me chan_capi set up the call but no sound available. -- Dave Cotton <dcotton@linuxautrement.com> _______________________________________________ Asterisk-Users mailing li...

it appears the cvs for zaptel as of 12/13/04 am is missing at least 1 file -- wcfxs.c greg Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development & Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When

...on controller 2, i.e. perfect. The other calls come in on both controllers at the same time. I have checked the patches etc. Is this possibly a problem caused by a PtP / PtMP config error? What would happen if the telco thought it was PtP and we dealt with it as PtMP? -- Dave Cotton <dcotton@linuxautrement.com>

...ng response... 404 Not Found 09:22:14 ERROR 404: Not Found. make[1]: *** [/var/lib/asterisk/sounds/.asterisk-core-sounds-en-wav-1.4.0] Error 1 make[1]: Leaving directory `/usr/src/asterisk.svn/sounds' make: *** [datafiles] Error 2 but the file is definitely there. -- Dave Cotton <dcotton@linuxautrement.com>