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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. Th...

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

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

...s deployment of telephone system for a small city (2000-5000 users for 2 years) shortly after New Year. I evaluating using of * for this deployment. Unfortunately, our team didn't include expirience C programmers, so i think that we will need some external support in this field. Thanks. Anton Tinchev Unix Solutions ltd Sofia, Bulgaria. P.S. If i can help to someone with my test resources or make some interoperability tests/repots for my devices, just drop me a line.

Can someone point me for good VRRPD (rfc2338) implementation on linux. Some stable and live project Thanks _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO: http://lartc.org/

Anybody sucessfully got stable 1000Hz clock without Digium harware and kernel 2.6? We need to consult some peoples how to clock asterisk stable with exactly 1000 Hz without much kernel/drives patching/tweaking. Some test results we made so far: 2.6 with digium card - stable 1000 Hz. 2.6 with ztdummy - uses RTC and the clock is 1024, not 1000. 2.6 with some Realtime kernel patch - provides stable

Has anyone succesfully integrated * with a cisco voice gateway ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031214/3b1ba7b3/attachment.htm

...on) > 2. Re: Has Nufone gone belly-up (John Baker) > 3. GSM phone to *? (Max Tulyev) > 4. Re: Has Nufone gone belly-up (David Liu) > 5. GSM modems (Steve Underwood) > 6. Re: rc.local dont works (listas iPfone) > 7. Need Europian vendor for Digium hardware. (Anton Tinchev) > 8. Wildcard X100P usable in Germany? (Roger Schreiter) > 9. RE: Need Europian vendor for Digium hardware. (Low, Adam) > 10. RE: Asterisk Indications (Philipp von Klitzing) > 11. He really doesn't care (Bill Michaelson) > 12. RE: GSM modems (Florian Overkamp) >...

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

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 :).

Must accepts wire transfers and ships to Sofia. Thanks

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

Hey all, I have an as5300 I use for dial in customers, we have 4 PRIs on it. We have a few free channels on it. I'm wondering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco

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

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

...m Hart" <adam@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 :)

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