similar to: Programming an unlocked ADSI phone?

Folks, I'm experimenting with bringing multiple (8) analog lines from our local telco into a Carrier Access ADIT 600 channel bank with an FXO module, then having this talk to Asterisk via the T1 TDM controller on the ADIT and a TE405P card. I don't know if this will work well (ie: give me decent echo cancellation, call disconnect supervision, caller ID etc) but I haven't had much

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

Folks, I've been following recent discussions regarding echo and echo training with much interest, since it's a problem we've never been able to eliminate here. We're facing two challenges presently, and they may be related (or not): a) Cisco 7960s in the office here echo back to our staff, but the customer hears decent sound. b) We've yet to be able to pass faxes through

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

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

Greetings, I have just purchased several Astra 390 phones ready for asterisk. I have placed a line with adsi=yes in the Zapata.conf file just before channel => 13 I have also added an extension exten => 6199,1,ADSIProg(asterisk.adsi) exten => 6199,2,Hangup in the extensions.conf file. When I try to program the phone I get the following: Asterisk CVS-03/28/04-12:02:10,

I've received my Aastra 390 phone. I got the unlock procedure from the vendor, and the services button now shows all four entrys as "<available>". When I give the phone "ADSIProg()" the phone displays: Asterisk PBX download refused Conflict with: <available> The asterisk.adsi hasn't been changed: DESCRIPTION "Asterisk PBX" VERSION 0x02 SECURITY

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

There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ;

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

Wondering, since they appear to be plentiful on eBay, whether I could get a Nortel 350 to use to learn my way around ADSI. The vendor claims that these are "generic," and looking through the archives I wonder if that means that they might be unlocked in the sense that the word is meaningful to asterisk. Of course I am green as could be on this topic, so this question may even be a

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

Ok, so I figured out my problem with my pt480s. But, now I have a few more. 1. When I dial into the voicemailmain or voicemailmain2 application, the phone and * start talking adsi, but then the phone tells me "programming download canceled, services is full.", but my services list isn't full, only "Asterisk PBX" occupies slot 2, slots 1, 3 and 4 are available. Any ideas?

Folks, I am in the process of fighting my way through my first foray into ADSI phones. I have a very nice Aastra 480 from Sayson, purchased through Cylogistics. Very nice phone. However, my experience in configuring ADSI has been pure pain so far, and it's not all working yet... 1) The locking mechanism for these ADSI phones is... baroque? Arcane? Perhaps there is a reason why you would

I have searched through the archives to no avail. Can someone give a quick overview or starting point for ADSI. I just received a 390 and was told that it is unlocked. But am not sure of the next step... I have tried a limited test using the adsi.conf and received: -- ADSI Available on CPE. Attempting Upload. WARNING[106510]: File app_adsiprog.c, Line 1211 (adsi_process): Invalid or

Greetings, I'm trying to setup an option in my greetingmenu that would allow the caller to select this particular option for emergency calls. That option would dial out on an available PSTN line to a cell phone number. Currently it is setup as such exten => 9,1,Dial(Zap/g1/<CELLPHONENUMBER> where <CELLPHONENUMBER> is the number it is calling out to. When option 9 is

I have an Aastra 390 ADSI phone. It's not locked. I can call ADSIProg without a problem and it programs my phone. Calling Voicemail2 also programs my phone. However, in order for the VMail option to appear on the screen I have to go into the Services menu, pick Asterisk PBX and pick Select. Then the VMail softbutton appears on the screen, but any time I make a call it goes back to the