similar to: SV: Mailing list growth

Hi! Are the GIPS codecs now implemented with the Asterisk? If I need more analog lines, say around 30, what's the easiest way doing it? I checked the Mediatrix box with 24 connections, maybe that would be a good (and rel. cheap) way to go? Any other suggestions? The ports has to support fax machines. rgds, /staffan --------------------------------------

Hi I'm running exactly the same setup. Asterisk is running on my FW/NAT/Router with two interfaces. My local phones are situated behind the NAT and connects to the outer interface of the */FW/NAT/Router. * is then connected to my SIP providers (since I'm only using the SIP-part of *, PSTN connection through my SIP-provider). Works fine! rgds, /staffan kerker sweden -----Ursprungligt

Hi Is it possible (or recommended) to run both Asterisk and say SER on the same physical machine? How about port conflicts? Maybe the easiest way is to change the default SIP port on Asterisk? But how will that work if I register some SIP accounts directly from asterisk (like my SIP provider) but then wanna dial outbound pure SIP calls via my SER... Has anyone got a functional system like this

I have an interesting problem now. I use asterisk to connect to both FWD and a sip provider here in sweden. suddenly, (i know my provider upgraded from a unknown SIP proxy to SER) Asterisk crashes when I try to make a call using this provider. FWD still works fine, and I can call directly towards the GW to POTS without any problems. But, as I call using my providers SER, Asterisk crashes.

Hi! Does * do DNS-lookups when outgoing calls are placed to a different SIP domain? Can I call from <sip:1000@mydomain.com> to <sip:2000@remote_domain.com>? Can * work as a regular SIP proxy in that aspect? Can * handle SIP URI:s that are complete SIP URI:s (sip:user@domain) instead of numbers only? Or should I run a SIP proxy on a different machine to handle pure SIP requests and let

Hi I know this is a bit off topic, but still pretty interesting. I'm running Asterisk on my Linux router/NAT/FW connected via cable (1mbit/200kbit) to the internet. Now, I wanna do local QoS implementation. Just very simple to give RTP (UDP) highest priority on my outbound interface. So, whenever I got an ongoing call, the RTP traffic should be handled first and other data (file transfers

Hi Is there any work being done on implementing IM/SIMPLE support for SIP on Asterisk? Like a presence server? rdgs, /Staffan Kerker

Dear package mgcv users, I am using package mgcv to describe presence of a migratory bird species as a function of several variables, including year, day number (i.e. day-of-the-year), duration of survey, latitude and longitude. Thus, the "global model" is: global_model<-gam(present ~ as.factor(year) + s(dayno, k=5) + s(duration, k=5) + s(x, k=5) + s(y, k=5), family =

Hi!. Is there any way to know which extension answered a call , when dialing from an AGI Script?? Thanks! Luciano -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/22a585ea/attachment.htm

I would recommend that you look at the Pika Technologies Daytona MM board. It has onboard DSP and onboard analog bridging taking up much less horsepower. Please contact me off-list if you would like more information. Bill Hunt Stroudwater Contact Point 207 347 8080 x219 877 870 1234 Toll Free www.stroudwater.com "Realize the Value of Customer Contact!"TM This e-mail is intended

Hi list, I encounter a problem again, which I thought resolved : when joining a domain, sometimes the workstation says it cannot join, or the domain does not exist, or something like that, because there is a DNS resolution probleme. I read the Micro$oft documentation listed in the error message (http://www.microsoft.com/windows2000/dns/tshoot/dns_tshoot2A.asp) I solved the problem, the

In our experience, the total cost of softphones(money, reduced sound quality and lower reliability) in a large call center environment is actually greater over time than the cost of a channelbank and cheap analog headphones. We've tried 2 softphones, 2 kinds of SIP VOIP hardphones, 2 kinds of SIP analog adapters and we've tried channelbanks over the last 3 years. Right now we are half done

So far in January, we've had 726 messages on -users. December 2003: 2.978 messages November 2003: 3.410 messages October 2003: 3.177 messages December 2002: 741 messages December 2001: 67 messages ...the project is growing. /Olle

Fellow R users, I'm using the BCE {BCE} function to run a Bayesian sediment mixing model. The aim is to find the optimum % contribution from each of the 4 source areas that can yield the target geochemistry. I have geochemistry for 4 source areas called Rat: Rat<-read.table(text="CaO MgO Na2O Al2O3 Topsoils 2.511250 0.7445500 0.7085500 14.10375 ChannelBanks

Does anyone have any recommendations on how to connect 160 analogue phones to an asterisk PBX? Background information: A client wishes to replace their current PBX with a new VoIP system. Currently they have 2 PRIs. I intent to set up 2 asterisk PBXs with Debian GNU/Linux on raided drives. These drives will be mounted only read-only to recover gracefully from power-cycles. I am considering 2

What they are probably marketing is putting in their own equipment out there. I install a product that does exactly that. A paradyne jet fusion. It takes care of the part of which channels are data and which are voice. If it's anything like these, the lines will come out on pairs. You will then have to use channelbank and FXO/FXS cards to get it into your phone system. The jet Fusion

My suggestion would be the one-line eyeBeam phone under development. Check out support.xten.com. //Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r afoc@interconnessioni.it Skickat: den 18 oktober 2005 14:48 Till: Asterisk Users Mailing List - Non-Commercial Discussion ?mne: Re: SV: [Asterisk-Users] Queues

Hi to all! I have this config, PSTN <--> AS5300 <--> ASTERISK I am using the Cisco as5300 to receive incoming calls and routing them to Asterisk for IVR. When I ran asterisk this is what I get when calling the voicemail demo. *CLI> -- Executing Playback("SIP/-081058b8", "transfer|skip") in new stack -- Executing Macro("SIP/-081058b8",

hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. it will be a slow migration, the asterisk server will be inserted between the telco E1 and the hicom. new phones will be sip ones. the customer has several fax machines and analog phones (some of them have to be explosion-proof). around 50 analog ports in total are needed. as we are in

Sorry about that, its just outlook that does that by default. But manually deleted your adress now in reply. I don't know what you mean with "top posting"? What I mean is that if you have another security on the connection (be it physical security - the connection doesn't go over public means, or VPN - connection level encryption) then you don't need another encryption on