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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: <sip:1649;phone-context=Exp_Net.ascom@ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d To: <sip:1715;phone-context=Exp_Net.ascom@ascom.be;user=phone> Call-ID: f8b5ad8-de671...

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...

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 support (autoreconf has to be run after patching the source). thanks, best regards, -- aspa -------------- nex...

...all modes, samplerates, > bitrates, stereo/mono, etc. However, an HDL implementation of a decoder > would be more impressive, and maybe more useful to others. > > I recommend starting with an encoder. > > > On Fri, Oct 4, 2013 at 2:15 AM, Fredrik Bonde <Fredrik.Bonde at ascom.se > <mailto:Fredrik.Bonde at ascom.se>> wrote: > > __ __ > > Hi,____ > > __ __ > > We would like to use the OPUS codec @ 16 kHz sampling rate and max > 32 kbps. ____ > > __ __ > > What about implem...

...sses 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 the network. The PBX itself will use that information to determine that for example, if a number begins with '4' then it is 'complete' after a total of 3 digits (eg. 411) or if it begins with 1, then it's complete a...

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

...onf. on SYSTEM SCIENCE and ENGINEERING (ICOSSE 2004) 4th WSEAS Int.Conf. on POWER ENGINEERING SYSTEMS (ICOPES 2004) **************************************************************** Cancun, Mexico, May 12-15, 2004 6th WSEAS Int.Conf. on ALGORITHMS, SCIENTIFIC COMPUTING, MODELLING AND SIMULATION (ASCOMS '04) ********************************************************** NOTE THAT IN WSEAS CONFERENCES YOU CAN HAVE PROCEEDINGS 1) HARD COPY 2) CD-ROM and 3) Web Publishing SELECTED PAPERS are also published (after further review) * as regular papers in WSEAS TRANSACTIONS (Journals) or...

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

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

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, 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

...o implement the entire standard, at all modes, samplerates, bitrates, stereo/mono, etc. However, an HDL implementation of a decoder would be more impressive, and maybe more useful to others. I recommend starting with an encoder. On Fri, Oct 4, 2013 at 2:15 AM, Fredrik Bonde <Fredrik.Bonde at ascom.se>wrote: > ** ** > > 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 c...

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...

Guys. I know there are wifi sip phones out there but I have a question, are any of these phones "anti explosive"? By that I mean, there are certain regulations about phones or cel phones that are not recommended to operate in environments like gas stations due to sparks and the chance of ingiting gas fumes. Are there any wifi sip phones out here that have complaince with regulations to

Hey folks, I have a T100P card with a PRI; when doing outbound dialing over the PRI, I can use SetCIDNum(2024561414|a) to force caller ID to display as "The White House" on a land line. This is apparently done as a reverse lookup by Verizon, as I do not hand the PRI the words "The White House" -- 202-456-1414 is an actual White House number. Using SetCIDName("Flying

...-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110128/234a2dbb/attachment-0001.htm> ------------------------------ Message: 4 Date: Fri, 28 Jan 2011 18:02:31 +0100 From: Urs Buob <Urs.Buob at ascom.CH> Subject: Re: [asterisk-users] Disabling Music On Hold To: asterisk-users at lists.digium.com Message-ID: <OFF4184A08.48C9E0D1-ONC1257826.005CF123-C1257826.005D9DA7 at ln.ascom.ch> Content-Type: text/plain; charset="us-ascii" > On 11-01-28 07:37 AM, Urs Buob wrote...

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. Ich h?tte es aber gern so das ich erstmal den H?rer abnehmen kann und dann ein internen W?hlton bekomme und jetzt erst die Nebenstelle anw?hle. Ich vermute mal ich muss etwas sp...

All, has anyone any experience in using Wifi smartphones as SIP clients? Does this work properly? What models/brands are optimal for this (in terms of ease of use, battery life etc)? Thx!! B.