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.... Of course, no firmware on CAC AB1's I have the AAstra 480, Adtran 750 Channelbank (updated firmware), T100P card, and it worked fine on the first try with current CVS. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: Darren Nickerson [mailto:darren.nickerson@ifax.com] Sent: Monday, December 29, 2003 10:09 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Update. As I mentioned in my first post (below), I had managed to get the phone to accept a download from asterisk, bu...

...B8ZS Line Build Out: DSX-1 EQUALIZATION FOR 0-133 ft. (CSU 0dB) Loop Code Detection: ON Loopback: OFF FDL Type: None Can anyone familiar with the Adit 600 and/or TE405P see any obvious errors here? -Darren -- Darren Nickerson Senior Sales & Support Engineer iFAX Solutions, Inc. www.ifax.com darren.nickerson@ifax.com +1.215.438.4638 ext 8106 office +1.215.243.8335 fax

...ral] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls allow=ilbc|gsm|ulaw|g723.1|g711 ;allow=all dtmfmode=inband ;dtmfmode=inband|rfc2833 good luck... Girish >From: "Darren Nickerson" <darren.nickerson@ifax.com> >Reply-To: asterisk-users@lists.digium.com >To: <asterisk-users@lists.digium.com> >Subject: [Asterisk-Users] SJphone, Asterisk and DTMF tones ... >Date: Sun, 21 Dec 2003 01:29:16 -0500 > >Folks, > >I can't seem to get DTMF si...

...s stuff be working much better than this? We're willing to invest some time getting faxing to work, but I'd hate to ask people here to dedicate their time and energy to something that's never really going to work well due to limitations in Asterisk/zaptel technology. -Darren -- Darren Nickerson Senior Sales & Support Engineer iFax Solutions, Inc. www.ifax.com darren.nickerson@ifax.com +1.215.438.4638 +1.215.243.8335 (fax

...superseding the other? Would be good to put something together that details the different methods, and which situations maybe they better suit, if any. Ben -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Darren Nickerson Sent: 04 October 2004 16:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] echo cancellation: the never-ending quest fortruth "Bruce Komito" <brucek@bagel.com> spake thusly: > Asterisk apparently has five echo cancellation algorithms:...

I just installed the new version of SJPhone and it appears that it cannot work with * anymore? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040111/25c910bb/attachment.htm

T.38 is often put forward as the solution for reliable FAX over VOIP. Just wondering for anyone using T.38 (with any equipment), how well does it work as compared to a FAX PSTN call? Thanks. Jim James H. Thompson jht@lj.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041129/3ea08b2c/attachment.htm

...ffice, let alone busy. Is this the intended result when trying to dial a disconnected SIP extension, or have I misconfigured something? Does the dialplan above, which was built using analog handsets, need to be more intelligent to deal with SIP connectivity/registration status? -Darren -- Darren Nickerson Senior Sales & Support Engineer iFax Solutions, Inc. www.ifax.com darren.nickerson@ifax.com +1.215.438.4638 office +1.215.243.8335 fax

Hi, I would like to set up a high density FAX/PBX server, am looking at using Eicon Diva E1 card with Asterisk and Hylafax sharing channels, is this possible. I know Extensions can be reserved for voice OR fax with the combination of chan_capi used for * voice, capi4hylafax on fax but then channels must be dedicated rather than shared ie. would like to be able to answer voice/fax on same DID.

Asterisk apparently has five echo cancellation algorithms: STEVE, STEVE2, MARK, MARK2 and MARK3. The current default appears to be MARK2. My question is, has anyone had any experience with any of the others (other than MARK2), and is there some conventional wisdom as to when to use one over another? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

Hello, I have spandsp setup to accept incoming faxes and receiving tif files via Email. Using tiff2pdf, or tiff2ps -a2 or even tiffsplit, the last page of the fax is cut off and the quality of the text looks "squished". I "figure" it's a tiff parsing thing, as opposed to a problem with my spandsp installation (heh). Has anyone experienced the same thing, or can

I am wondering whether it is possible to have Puppet only stop and disable a Windows service if it exists. I have Puppet master 3.1.1 on CentOS, and Puppet clients 3.1.1 on many Windows machines. Right now, I have the bit below for certain nodes: service { "CVSNT": ensure => ''stopped'', enable => ''false'', } Some of the nodes have this

...mount of the folder that I'm exporting. NFS4 doesn't have a user_xattr option that I can use when mounting. Does anyone have any ideas for what I could do to get Samba to use ACLs over NFS4? If you need more configuration details, or if I could be more clear on any part, let me know. Paul Nickerson IT Systems Administrator & Support DeskNet Inc. Portland, Maine

What are your experiences with faxing through Asterisk to the PSTN? We are using g.711u as a codec, and are originating/terminating with Broadvox as well as through our own PSTN gateways. We have had some luck with incoming faxes coming into our network from Broadvox DIDs. They work 50% of the time. Not sure yet on PSTN incoming since nobody that is using FAX is in our local rate centers.

I'm trying to get a Carrier Access Corp. Channel Bank I working with a Digium T100P without success. What is stranger is that the status lights on the channel bank and T100P seem to change almost each time I power cycle the channel bank or reset the T100P. The channel bank has three status lights: T1, Framing, Status. T1 is green, Status is yellow, and Framing is usually red but sometime

All, I signed up for a broadvoice BYOD plan over the weekend (very excited about their offering) and after about an hour I had asterisk registered and was making in and out bound calls. However, the next day (without changing anything) I couldn't call in or out and haven't been able to get it going again. I can connect using a softphone (X-Lite) and make calls in and out

Hi all, We seem to be having an interesting issue maintaining our registrations with Broadvoice. It seems to be related to the fact that Asterisk does not normally include authentication with its first REGISTER attempt. Normally, Broadvoice doesn't care, responds with a 401 Proxy Auth Required, and asterisk answers accordingly. However, if the account is already registered, Broadvoice

Dear list members, I have a statistical question, that doesn't belong to this list, and I apologise for that in advance but I would appreciate your help very much. Is there some convention for selecting the a level for significance testing in scientific (e.g. chemical processes) studies? Most people use the 0.05 level but I could not find a reference to justify this. Why not 0.01 or 0.1?

How do you guys like the 330 and 320? I've been looking at this as my "standard" phone, since it's relatively cheaper than the 501 which is the phone I currently push with my PBX systems. Most of my customers do not use more than one line per phone, so having 3 lines on the 501 is not necessarily useful. Also, I read that the phone offers TLS security. What does that mean?

Hello all, We have an Asterisk server that has worked without issue for a while. Before, only Sipura and Polycom 500 series phones were used. Recently, we've added a few POE switches and 20 or so Polycom 330's. The 330's seem to lock up often. One easy way to do this is by hitting the * key. I looked thru the XML files for dialplan and what not, nothing stood out.