search for: unixsol

I'm first to buy 5 pack. Even for > $30.

In my tftp usage I'm using the server to collect device configurations but I don't want once the files are uploaded to be accessible for download. I've solved my problem by adding --forbid-get/-F option which disables file downloads. -- Georgi Chorbadzhiyski | http://georgi.unixsol.org/ | http://github.com/gfto/ -------------- next part --------------

Must accepts wire transfers and ships to Sofia. Thanks

I just need to buy 5-6 ADSI Phones. I wondering which of these models to choose - aastra 390 - I don't know is this an ADSI phone at all. Is there versions with and without ADSI - aastra 350 - I'm sure that it have ADSI. If there is some other good working model, it will be great, if someone points me. Thanks

...ists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > --__--__-- > > Message: 7 > Date: Mon, 26 Jan 2004 11:52:03 +0100 > From: Anton Tinchev <atl@unixsol.org> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Need Europian vendor for Digium hardware. > Reply-To: asterisk-users@lists.digium.com > > Must accepts wire transfers and ships to Sofia. > Thanks > > --__--__-- > > Message: 8 > Date: Mon, 26...

Anyone out there have the AudioCodes MP-108 working with Asterisk? I am able to get the channels to registers with Asterisk, but anytime I try and send a call I receive these error messages: Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:771 __sip_ack: Stopping retransmission on '117801284512845hUxv-9991110061--17708185305@63.201.117.76' of Response 20587: Found Jul 6 15:12:10

...teragen.com.au> >To: <asterisk-users@lists.digium.com> >Subject: Re: [Asterisk-Users] Digium should develop and sell just Dummy card. For timing... >Date: Wed, 15 Oct 2003 14:57:55 +1000 >Reply-To: asterisk-users@lists.digium.com > >From: "Anton Tinchev" <atl@unixsol.org> >To: <asterisk-users@lists.digium.com> >Sent: Wednesday, October 15, 2003 1:06 PM >Subject: [Asterisk-Users] Digium should develop and sell just Dummy card. >For timing... > > > > >>I'm first to buy 5 pack. Even for > $30. >> >> &...

Hi, I wonder about the performance of a Linux box used as router (I guest I''m not the first :). Althought I know it mainly depends on the hardware, I''m trying to find some references on the topic or comparations with other routing solutions (FreeBSD box used as router, Cisco, etc). For example, http://facweb.cti.depaul.edu/jyu/Publications/Yu-Linux-TSM2004.pdf (althought is

I has a E1 trunk to PC and 4-5 SIP phones. Can * plays some music in all calls if all the phones are busy, and when one got free, to forward the call to the agent. Excuse me, if it is newbie question, but i'm googling and reading this list 4 hours and didn't found clear answer :)

Need to buy 2-3 channel banks for some asterisk deployments...

Hi all. I'm going to implement some large Asterisk based solution. Maybe 4-5 PCs with 1-2 E1/T1 trunks on each. Because some of the traffic will be sended to external VoIP provider, i has some questions 1. Which is the lowest bandwidth consuming codec in Asterisk, which is compatible with Cisco Gateways. Stability is needed too. 2. Have someone allready bulded such a systems and what

Just happy. hardware information: -------------- Some small factor IBM Celetron (coppermine) at 1100 (11*100FSB) 256 RAM 15GB Hard. 1 x Digium E100P - E1 Line from telco with 300 Dids 1 x TDM400P for local phones --------------- Few small machines (mainly brand PII at 233Mhz with TDM400P Cards. --------------- There is a lot of SIP equipment attached: 2 x Micronet SIP Gateways 1 x ata186 1 x

Yes, i know that there are many ISDN card on the market. But when i spend money for ISDN card, i prefer to be Digiums, to get all support and help Asterisk :).

I just got my Access Bank 2 (of course i'm happy now:)). Just need the wiring scheme Thanks

Whicch version of zaptel and Zapata should I use with 1.0?

I'm using T100P with CAC AB II, only FXS ports. After upgrading, asterisk stoped sending caller IDs to the phones. Even inside - port to port. I got 2 errors in the debug: __zt_exception: Exception 23, channle 2 (i'm ringing to channel 2) zt_handle_event: Didn't finish Caller-ID spill. Canceling.

As i see, bradcom released driver under GPL. Date is 09/12/2002. Has someone tried it? _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO: http://lartc.org/

I've reading this lists few months. We are small company, that makes some system intregration, development and deployments in VoIP scene. Completely under linux. Today i have 6 machines with asterisk, huge test base - including devices like AS5350, Audiocodes gateways, ATAs, IP phones ... Now is time to make a decition for including * in our future projects. Main goal for us is the Stability.

Gabe Bourque wrote: > Hello Anton Tinchev, > > I'm writing to you in hopes you can answer a few questions regarding > Asterisk/Digium and it's reliability. I saw your posting in the > Asterisk mailing list (Re: [Asterisk-Users] Is Asterisk ready for "real" > use?) and decided to write directly to you. The reason being that you > are one of only a few

Just spended ~ hour googling - all boards are based on GC-XX or I750X Chipsets - all for Xeons. There also some boards for Pentium 3. Can someone point me to a board with 64Bit 3.3v PCI for ordinary P4 with 800Mhz FSB. Thanks