search for: woodoo

...with -c, i have no disconnected, but see the stuff restarting. i've tried to recompile, older version, virgin config, etc. same results. it's happened after a power loss, on a ext3 fs, sitting on a raid1. astdb was deleted, log is not showing any interesting things. any ideas please? -- WoodOO-[P]an[G]alaktikan[A]gent-People <][> http://shadow.pganet.com wpeople@shadow.pganet.com]iCQ#33118021[wpeople.on.iRCNet]wpeople@RedHat.users

Dear Forum Members, I just purchased two VoiSmart GSM cards. Tried to install one of them on my Fedora Core 5 system, The compilation was not smooth, as expected, but after a small fix, it went through. Then I put two SIM cards in the card's slots. Then I loaded the modules. Then I started the Asterisk. After all I configured the vgsm.conf file according to my settings, that is just changed

Hi all, I've been having problems with my A20002D lately - callers from the PSTN don't hear me when I answer, but I hear them. Disabling echo cancellation in zapata.conf brings the audio (and echo) back. This used to work fine, until two days ago. The only weird thing in the logs is this: May 12 07:42:53 steerpike wan_ecd: wp1ec: The H100 slave has lost its framing on the bus! May

...gt;> SPNEGO(gse_krb5) NEG_TOKEN_INIT failed: NT_STATUS_LOGON_FAILURE > > Looks like: https://bugzilla.samba.org/show_bug.cgi?id=12262 Thank you Mark but it doesn't feels the same to me... In subsequent tests I wasn't able any more even to join. The first time was a lucky one, woodoo. I discovered, that the generated smb.conf was not enough for an AD-DC. Despite having: server role = active directory domain controller ... the default settings for: domain logons = no (?) domain master = auto (aka equally NO) local master = yes (not specifically mentioned in the generated s...

----- V?stule no Rowland Penny <rowlandpenny at googlemail.com> --------- Datums: Fri, 20 Mar 2015 13:12:20 +0000 S?t?t?js: Rowland Penny <rowlandpenny at googlemail.com> Temats: Re: [Samba] Samba AD DC and browsing of shares Sa??m?js: samba at lists.samba.org >> the necessity for external heimdal even for Samba 4 in case of >> Slackware I found somehwere in

...led: NT_STATUS_LOGON_FAILURE > > > > Looks like: https://bugzilla.samba.org/show_bug.cgi?id=12262 > > Thank you Mark > > but it doesn't feels the same to me... > > In subsequent tests I wasn't able any more even to join. The first > time was a lucky one, woodoo. > > I discovered, that the generated smb.conf was not enough for an AD-DC. > > Despite having: > > server role = active directory domain controller > > ... the default settings for: > > domain logons = no (?) > domain master = auto (aka equally NO) > local...

...s unable to restore format, but not sure. The strange is, same freepbx (amp) works nice on 1.0.x I have installed 1.2.6 from source, but spandsp and appfax got from package. I've tried to install newest spandsp, and appfax, but appfax got failed on compilation. Anyone knows the problem? -- WoodOO-[P]an[G]alaktikan[A]gent-People <][> http://shadow.pganet.com wpeople@shadow.pganet.com]iCQ#33118021[wpeople.on.iRCNet]wpeople@RedHat.users

Hi all, I have a "Cologne Chip Designs GmbH ISDN network controller" and I want to terminate voip calls via this ISDN card. My question is: How I must to wire the ISDN equipment with my ISDN card? With normal cable or crossover? How I can to check if ISDN card is linked with ISDN equipment? In this moment I have 1:1 cable between ISDN's, the mISDN is installed and "misdn

...Called 11@80.98.x.x -- H.323 call 'ip$localhost/15803-305e28d0' cleared, reason 24 (Call ended with Q.931 cause [28 - Invalid number format]) -- Hungup 'OH323/11@80.98.x.x-305e28d0' if Panasonic 'd like to dial, i've got cause [111 on it anyone meet with this? -- WoodOO-[P]an[G]alaktikan[A]gent-People <][> http://shadow.pganet.com wpeople@shadow.pganet.com]iCQ#33118021[wpeople.on.iRCNet]wpeople@RedHat.users

hi does anyone have an idea how it could be possible to do email -> fax gatewaying with asterisk + app_txfax, but still keep track of who sent the fax? i've thought a little about smtp auth, but it doesn't look too easy to integrate smoothly with asterisk.... roy

Hello list, I'd like to share something u all , so that i could understand whats going on into my Asterisk box. i have a setup like this client(ip phone) -----ip network------- [Asterisk]----ip network -------[Service provider] i have configured A2biling in my Asterisk box. so when client call to my Asterisk A2billing's ivr respoce , my client authenticate there pin and call . all

...blem: to detect trunk status (ok/unreachable) in *, it's a must, to set qualify=yes as * connecting to SER, it's not replying to qualify messages, so even i can use it well without qualify, with qualify it's says unreachable immediately. What i have to set in SER to reply? Thanks! -- WoodOO-[P]an[G]alaktikan[A]gent-People <][> http://shadow.pganet.com wpeople@shadow.pganet.com]iCQ#33118021[wpeople.on.iRCNet]wpeople@RedHat.users

Hi all, I hava SER with many clients (sipura SPA2100). One of these is an Asterisk which have others clients (sipuraSPA2100). I also have a Cisco GW which give me access to the PSTN. I make calls to all IP phones in my network, but I can't call PSTN numbers. After I dial, I hear 2 ringbacks but at the same time Asterisk says: Called pstn_number@SER_ip_address SIP/SER_ip_address-ec75 is

Hi, I tried to find a reference in terms of size but got back a bunch of tech documents and couldn't get the idea of wav49 format. wav49 format is supposed to be half the size of a normal wav right? so, how much disk space takes to save one minute of audio in wav49? I trying to do some capacity planning for a voicemail server. -- ------------------------------------------------------------

Hi List, I would like to know if there is a way to detect gateways which time out (because of network problems or hardware failure for instance) when you send traffic to them. So when you do: Dial(SIP/number@gateway) If a call couldn't get through because the gateway has timed out, i want to do something about it. The idea would be to suspend gateway which time out for 60 minutes,

...omputers joining it? >> >> No, --with or --without-ad-dc, you can configure Samba4 as NT-4 style DC. > > yes, it was clear. The only thing that wasn't - possibility to join Win8 to > NT-styla domain. I learned, it is theoretically possible, but taking into > account the woodoo part of it, not the solution for particular case. Hmmm, as far as I examined , Windows 8 can join to NT-style domain, whose DC is Samba 4, compiled --with-ad-dc (default). >> From: J?nis <je at ktf.rtu.lv> >> Date: Fri, 20 Mar 2015 21:39:00 +0200 >> >>> How fast...

...which is no longer required to build Asterisk. > > You install Zaptel, but DAHDI is now the "latest version of Zaptel". > Zaptel is now frozen and has not been updated for quite some time. > > You set the /usr/src/linux-2.4 symlink , which is a pointless old piece > of woodoo. Can you point me to the place in the documentation that > pointed you to it? > > (Not to mention that the whole messing with kernels is so bad in Centos. > It is not better in SUSE, where you have to install the full kernel > source to build Zaptel / DAHDI). > > Next: no nee...

Hi, Did anyone try to install Asterisk on the Mini-Box M300 with a Versa mini-ITX board 1GHz VIA x86 CPU? The box looks promissing, but I am not sure if Digium cards are compatible with the mother board (Versa mini-ITX) Also I am not sure if the 1GHz VIA processor can handle a Digium 24 port analog board, or an E1 digital board. If anyone had tried the Mini-Box, the processor, of the mother

Hello everyone. I've got a HFC ISDN card that I'm using with chan_misdn and it basically behaves like crap. Echo is waaay worst then echo I get TDM400 card, sound is "choppy" (there other side is allays complaining about sound interruptions) and to top it all it detects fake DTMF's all the time. Is this a chan_misdn problem or is it a card problem? I really need to get

Dear Asterisk Comunity, I'm thinking about developing a callback application based on the following scenario: 1. Customer Calls the outgoing number which is a PSTN line connected to my Zap channel 2. Asterisk captures the Caller ID and calls back the customer. 3. As soon as the customer picks up the phone, asterisk plays a promt to enter the Destination number. 4. Asterisk Connects the