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...1'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, but it still said PLEASE PROGRAM...

...300 ms - WINK ends 1000 ms - digit 1 1150 ms - digit 2 1300 ms - digit 3 1450 ms - digit 4 (this was configured to only accept 4 incoming digits) What settings are we missing to get asterisk to see more than 1 digit? Thanks, -- Aidan Van Dyk aidan@ifax.com Senior Software Developer +1 215 438-4638 x8103 iFAX Solutions, Inc. http://www.ifax.com/

...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

...39;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

...; 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 signaling working properly using S...

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

...nd those who do have echo problems for any reason (like us) dig a little deeper). He said that MARK2 is the recommended one, and that some people have found the aggressive option help. FWIW, we run MARK2 in aggressive mode here. -Darren -- Darren Nickerson Senior Sales & Support Engineer iFax Solutions, Inc. www.ifax.com darren.nickerson@ifax.com +1.215.438.4638 x8106 +1.215.243.8335 (fax) _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options...

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

...esult 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

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

Hi, does anyone have any recommended (read tried and tested) way of making asterisk be able to handle incoming faxes. I've a PC running asterisk with a digium E1 card in and simply want to be able to route a call to some application which will take a fax call and save the fax as an image. I guess the bits I'm unsure of are how to terminate a fax call without a modem (if this is

...ON END ____________________ HylaFAX(tm) Users Mailing List _______________________ To subscribe/unsubscribe, click http://lists.hylafax.org/cgi-bin/lsg2.cgi On UNIX: mail -s unsubscribe hylafax-users-request@hylafax.org < /dev/null *To learn about commercial HylaFAX(tm) support, mail sales@ifax.com.*

Okay fellow Asterisk users... Listen up (or perhaps more appropriately, "read down"). The Asterisk community is growing at a remarkable pace. I know there are thousands of you out there -- in fact there are over eight *thousand* subscribers to asterisk-users alone, and almost one *thousand* registered users on the bug tracker. The #asterisk IRC channel is consistently in the top 10

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

Folks, I recently swapped a TDM400 FXS card that was working perfectly into a new server (running recent CVS), and it's either misbehaving (unlikely), or I've missed something obvious (much more probable). Everything seems to be working, but I can't get any dialtone from it when I plug a phone into any of the 4 ports!! All of the jacks are lit (green lights) but they all seem dead