similar to: Nortel CS1000 Asterisk with SIP

Hello, We have been trying to add asterisk to our Ascom Ascotel 2050 PBX. We have a AVM Fritz!PCI Card connected to an S0 bus extension from the PBX. The fritz card is configured to use chan_capi, and we can make calls SIP->SIP SIP->PBX extension PBX extension->SIP all successfully, we have assigned more than one PBX extension number to the S0 port in the Ascom PBX (it has 8 positions)

hello! one thing i'd like to see in OpenSSH is (optional) IDEA algorith support. this would be useful especially in an environment which has a mix of old ssh v1.2.x and OpenSSH installations. according to Ascom non commercial use of IDEA is free (http://www.ascom.com/infosec/idea.html). also, there are countries (e.g. Finland) where IDEA is not patented. here's a patch suggestion for IDEA

Just to make sure, what's the goal here? Is the goal 1) to have a fast Opus implementation or are you 2) looking for an interesting FPGA implementation project? If 1), then an FPGA is most likely not necessary since Opus is not computationally expensive. If 2), then it depends on the desired size of the project and the desired quality. The simplest encoder possible is indeed simpler than the

The originating PRI system passes the entire dialed number in the d-channel setup frame, thus the concept of a wait time for additional digits is meaningless. Progressive digit gathering implies that the signalling is occuring 'in-band' as would be the case with DTMF signalling on analog lines. You need to look in the Ascom and find the configuration table that lays out the dialplan for

I'm writing a function and keep getting the following error message. myfunc <- function(lst) { lst <- list(roots = c("car insurance", "auto insurance"), roots2 = c("insurance"), prefix = c("cheap", "budget"), prefix2 = c("low cost"), suffix = c("quote", "quotes"), suffix2 = c("rate",

I'm not aware of an FPGA implementations yet. You could be the first! An encoder implementation would be much easier, because there are almost no rules about encoders. An encoder is free to behave any way it wants, so you could implement a very small subset of Opus and still have a compliant (and useful) encoder. A decoder implementation would be much harder, because decoders are required

On Thu, Nov 14, 2013 at 12:51 AM, Fredrik Bonde <Fredrik.Bonde at ascom.se> wrote: > 1. Is it possible to via SDP force the remote party to send a > specific mode of the 32 different possible modes or to force the remote side > to use CELT only? No, and if you cannot decode opus completely (all modes) your device is not conformant with the Opus specification. I'm

hi all this potential customer has some 70 ascom office 30 phones, and wonders if it's usable with *. Does anyone know if it is? I currently don't have one myself, so I can't test it... roy

When I run a macro that uses RExcel, I get a dialog box that says "Microsoft Excel is waiting for another application to complete an OLE action." There is no error in the RExcel commands in the macro, of that I am sure. The box appears to be related to the inclusion of RunRFile commands. The macro will run through the second RunRFile command, but will not execute the

I have a really long functions, and at the end of the function, I am using a if statement to tag certain keywords based on whether they have certain values contained in them. However, the if statement doesn't seem to work. When I had split up the commands into various functions, it worked fine, but I'm not sure what going on now that it's combined into a single function. myfunc

It may have gone to sleep. Chris Cooper Systems/Network Administrator EFC International 1940 Craigshire Blvd St. Louis, MO 63146 US Phone - 314-439-4325 Fax - 314-439-4443 Mobile - 314-402-8912 - -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent:

I have a repetative task in R and i'm trying to find a more efficient way to perform the following task. lst <- list(roots = c("car insurance", "auto insurance"), roots2 = c("insurance"), prefix = c("cheap", "budget"), prefix2 = c("low cost"), suffix = c("quote", "quotes"),

I just installed Compiz on Ubuntu Feisty (Gutsy does not work on my hardware as per the release notes) via the http://ppa.launchpad.net/amaranth/ubuntu repo, which is the repo suggested in the compiz-fusion website. The machine is a Dell Inspiron E1505 / 6400 with the horrible ATI X1400 graphics card and 1680x1050 monitor. I installed XGL and the proprietary ATI driver, a combination that worked

Sorry that I wrote in german : Ich benutze asterisk mit dem zaphfc Treiber. Jetzt hab ich folgendes Problem, habe 2 ISDN-Telefone angeschlossen. zaphfc im nt-mode. Anrufe von ausserhalb per sip (sipgate.de) kommen an. Wenn ich aber intern zwischen den zwei Telefonen (Ascom Eurit 30) sprechen m?chte geht das nur wie folgt : Erst die Nebenstelle w?hlen und dann den H?rer am Telefon abnehmen.

hey * folk, am trying to configure a WellGate 3504A FXS SIP ATA (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set up two SIP clients in sip.conf as follows: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls [1235] host = dynamic secret = somepass context = default type

I am getting the following information and asterisk 1.0.7 is not answering the call. Any ideas? jerry ------------------ Sip read: INVITE sip:2828;phone-context=cdp.udp@qg.com:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From: <sip:3173241052;phone-context=+1@qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed To:

All, I am connecting to a CS 1000 nortel PBX. I can call out, I have limited success with call in. I get debug traffic that a call is coming in but I get the message "Unable to create/find channel". I was expecting that incoming calls over the trunk would be handled from my sip definition and goto the nortel context. It is not. Below is the actual incoming call debug information. I am

Greetings to All! Help to solve a problem: There is an asterisk and two VoIP a sluice (NSGate 800 2FXS 2FXO, NSGate 800 2FXS). In sip.conf they are registered so: [3301] type=friend host=172.222.135.11 username=3301 secret=0000 defaultip=172.222.135.11 dtmfmode=rfc2833 context=it callerid="VoIPGate2Line1" <3301> allow=g723.1 [3302] type=friend host=172.222.135.11 username=3302

Hi eveybody again! I don't want to be annoying, but if nobody can help me with this, I'll have to desist of working with SIP.I have some questions about SIP, as I wrote in another mail. I have a SIP Gateway and I have two phones (an analog one and a DECT one) conected to it.Also, I have two Dlink dg102s with four phones conected to them. The main problems are two. Calls between the

Hello, I am working with 1.8.32.2 which I have patched with t38-gateway and PRACK. t38 is tested and working fine with Zoiper client but I can't get the t.38 software from Biscom (FAXCOM) to talk. In my first attempts I found FAXCOM announces that it supports 100rel so I added the PRACK patch hoping that would do the trick. Now it gets a little further but * complains about rejecting a