similar to: Installing the none commercialintelg729codecsinto Asterisk@Home 2.2?

I downloaded and installed the none commercial g729 codec very often now I only disable HT on my systems I think * doesn't like this One of the guys @ digium advised me to turn it of, since they haven't written * to be multi treading any way The codec I download is the http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium4.so It should work fine. Wouldn't know what it

Thanks but this is for a test, I didn't buy the first one as it's a non commercial installation. I'm trying to test bandwidth etc so I need to try out how 4 of them handle the link simultaneously, I just don't know how to add a second test license. Dean ________________________________ From: asterisk-users-bounces@lists.digium.com

Hi, I'm helping out with a political campaign and would like to use asterisk to blast out about 200,000 calls with a short message from the candidate. Provider: I'm thinking voipjet may be a good solution? Hardware setup: I will have access to several T-1 lines so I would just want to set up the dialers to limit the number of concurrent calls and so forth. I found teleyapper on

Greetings list, It seems that more and more phones these days are coming with XML mini-browsers. I'd like to have a go at developing something useful to use on them, but in all honesty, most of our customers use their phones to make and take calls and very little else. So I'm open to suggestions. What useful applications are you developing for these mini-browsers? What sort of things do

Yep I did the same. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Francesco Peeters (Asterisk) Sent: Saturday, 21 January 2006 5:34 PM To: fbraeuer@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

Greetings list, Does anyone have experience with SIP conference phones? I need to source a couple for a client, but I'm not really familiar with the market - i.e. what's available, what's decent quality, etc.. A cursory googling has led me to the Polycom Soundpoint IP4000 at around the ?450 mark - any thoughts on this? If anyone knows a good Polycom wholesaler in the UK, I'd be

Hi samba list!!! Douglas, /usr/sbin/samba_kcc is made in python. Does it have a link to source4/dsdb/kcc/kcc_periodic.c which is made in C? The errors that appear in my DCs have their output in the C code. Correct me if I'm wrong, but I read in some old posts on the list that samba would have a new code for kcc (python code?), which would be closer to what M$ uses. Could this have anything

We have antiquated caller ID schemes here in Australia. We barely support numbers from other local carriers, let alone OS ones. Certainly no names either.

Mine is very similar: i don't have echocancelwhenbridged=yes because it seems work only on TDM, is it ? And in Italy, I often have set pridialplan = unknown About echo I have some problems, but only at the beginning of the call. After 3-4 seconds the echo became almost null, specially with snom 190; with pa168s and ywh10 I have again some problem, the echo come up also after 1 minute of

Greetings list, I've noticed over the last couple of weeks that, unsurprisingly, nearly every new PC seems to be coming with Vista these days. I expect it'll only be a matter of time for all of us before clients start needing Vista-compatible softphones (if it's not already happened). So, what's the story with Vista compatibility amongst the softphones currently out there?

Hi all, I am new to asterisk and am looking for a voip provider that supports asterisk. I am aware that their are several vendors to choose from. Any opinions on the best one? thanks Burak Balasaygun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060128/0a97a302/attachment.htm

Well I'm stunned no one has suggested a webcast option. I mean we aren't talking a bunch of people unable to grasp the concepts of chat/voice/vision sessions with a log in/remote display capability. If you think this is an option let me know I have someone who has some software they wouldn't mind stress testing as a trial. Cheers, Dean > -----Original Message----- > From:

Hang on.... there's a non commercial G729 codec that will work with Asterisk? Can someone point me to where I can find it? Thanks, Doug. -----Original Message----- From: Francesco Peeters (Asterisk) [mailto:francesco@fampeeters.com] Sent: Sun 1/22/2006 8:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion

Greetings list, I remember a discussion many months ago which ISTR concluded that asterisk didn't play nicely at all in multi-homed setups (e.g. SIP packets not being sent out through the same interface they were received on, etc.). Is this still the case, or are there versions which have resolved the issue? Even if it's still the case, is this only a problem for SIP, or does it affect

Greetings list, Does anyone have any experiences they'd like to share deploying these phones in medium-size asterisk setups, e.g. 40+ users? I have a project coming up to deploy 100 phones over 2 offices and the client rather likes these phones. Are there any obvious pitfalls/configuration difficulties/quality issues etc. using these phones? If so, what alternatives would people suggest with

Greetings. I did some digging with Google, the wiki, and on the archives, but didn't find a recent conclusive answer. If this is answered in the wiki or archives somewhere, please point me to it. I'm in the process of setting up an Asterisk box for home use. I've got a X100P card on the way. I've not decided what analog adapter(s) to get yet. The only phone service to hook up

Greetings list, I need to set up a point to point SIP connection between two devices without either of them registering with a registrar/proxy/etc. at all. The devices I've tested so far all seem to insist on having a registration before they'll make or take calls. One of the devices needs to be an ATA with an FXO port (e.g. Sipura/Linksys SPA-3000/3102), the other device can be either

Great idea Dean, "I would also suggest that maybe instead of looking to set up within the disaster zones that you consider setting up in the relocation zone (eg where people are being sent to) yes they have payphones there but having a bank of 20 grandstream phones connected a T1 is a far smarter and more effective solution and probably more meaningful." > -----Original Message-----

Greetings all, Has anyone managed to get dialplan status hints working across multiple servers? I've separated a load of SIP users out across 2 servers today, but it'd be useful if they could still see each others' status. I've replaced the various hint lines for the sip devices now on another box with: exten => 210,hint,IAX2/otherserver/210 Where 210 is defined on the other

Greetings all, Not specifically an asterisk query, but a couple of transfer queries that I'm sure are obvious to folks who use these phones all the time: 1) how does one do a blind transfer? When a call is answered and one hits the transfer button, followed by an extension, one has to wait for the other party to answer, then hit transfer again, before the call is released. I'm sure there