similar to: PRI/Zap premature dialing problem

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

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Jerry Glomph Black > Sent: Monday, January 30, 2006 11:59 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Polycom IP301: Pass-through ethernet port > unusable? > > Have just done a

Skipped content of type multipart/alternative-------------- next part -------------- Nov 21 14:17:47 VERBOSE[32580] logger.c: <-- SIP read from 172.25.103.222:5060: INVITE sip:1715;phone-context=exp_net.ascom@ascom.be:5060;maddr=172.25.96.48;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From:

Is it just me or are there problems? The list has just shutdown over the last 24 hours :( David

I can receive incoming calls. I can see via sip debug that I am communicating with BV just fine. When I call a BV number it goes through just fine. When I call any other # I get: "We're sorry your call can not be completed at this time. Please hang up and try your call again later" Broadvoice tech support does not see any errors and they see that I am registered just fine. I have

For work environments where you only get HTTP or HTTPS access, what is the feasibility of doing IAX over HTTP? I have heard of projects such as stunnel. Has anyone tried something like this? I did a quick search but didn't come up with much.

Hi, We would like to use the OPUS codec @ 16 kHz sampling rate and max 32 kbps. What about implementing an OPUS coder and decoder in an FPGA? Has this been done? Would either coder or decoder more suitable for FPGA implementation? Best regards Fredrik Bonde -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Since our device can only handle the Celt part of Opus (due to a MIPS limitation), we have two questions: 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? 2. In the reference implementation of Opus it looks like the only way to force the encoder to use CELT only

I had a working 2-way SIP connection running until about 2 days ago, now my outbound calls are promptly blocked with a "403 Forbidden" error. Inbound still functions OK. Perhaps they are fingerprinting and blocking Asterisk access (I hope not). They do not answer their support mail, or questions on their own forum. I'm sure there are other Asteriskers out there who have

I have a lot of experience, all of it pretty good, with various Sipura products, Grandstreams, Zultys, IAXy, and numerous SIP/IAX soft phones connecting into Asterisk as clients. Good sound quality, great reliability. I've tried two of the units named in the subject line, and frankly I'm frustrated. Calls usually start out OK, but within a brief period the sound goes totally to

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

Hi, I am wondering whether I can use Asterisk as SIP/H.323 Signalling Gateway. The setup I envisage looks as follows: H.323 end-point ---------(ETH)--------- Asterisk ---------(ETH)--------- SIP Proxy/Registrar ---------(ETH)--------- SIP end-point (ETH: Ethernet) In principle, Asterisk would just be used to integrate H.323 end-points into a fully SIP-based core-network. Hence, there

Hi, Does anybody have an idea on how to realise ad-hoc conferencing with Asterisk ? Although Asterisk MeetMe and maybe a procedure with Call Holding could somehow come close to ad-hoc conferencing, it doesn't seem to be the right way to do it. Any experience with ad-hoc conferencing using SIP in general as well as with Asterisk? Thanks, Joerg

hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. the customer is currently using a paging system (small receivers which display a callback number and a base station (transmitter) with several antennas at the site) the problem is, that the currently operative base station uses 4 ISDN BRI interfaces. But the protocol is old germany 1TR6

Hi. We install ruby, rail, rubygems, rails, mod_fastcgi and fastCGI nad we get from this URL http://bobcares.com/article38.html ans everything install with no errors, but when we try to run "test" we get this: Error from apache error_log: Premature end of script headers: /home/user/public_html/rails/dispatch.cgi /usr/local/lib/ruby/site_ruby/1.8/rubygems.rb:46:in

Hi all, I have a working ruby app that I developed on Window and I am trying to deploy it on Linox with Apache. The hosting company generated for me an empty rails app and I just replaced the files that I modified in my app (only rb/rhtml/css/js files). In addition I updatd the database.yml but now I am getting either page not found or Application error Rails application failed to start