similar to: RFD: With echo and other distortion, can ulaw/alaw line quality ever be good enough for faxing?

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

What kind of channelbank/FXS port are you connecting to? I've seen problems connecting to some of the older versions of the Adtran Total Access 750's. I wouldn't doubt there would be problems on other channelbanks with older firmwares. 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

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.

We're having some difficulty connecting a T1 fax board card to asterisk, with E&M Wink. We've successfully connected it to other E&M Wink T1s, but have been unable to get asterisk to hear it's dialing. Asterisk can dial down to it with no problem, and it hears all the digits, but whenever it sends digits down to asterisk, * always only "gets" the first one. This is

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.

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

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'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, I am using SJPhone here for testing ivr with Asterisk. I haven't seen any problem here yet. I have tried different things for that and finally got it working. I am not an expert to explain more about that, but here is the general section form my sip.conf. dont know whether it will help... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ;

What options do the others offer over one another? Is there a difference, or is it simply a case of one 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

Folks, We have several people using SIP softphones in the office. When they leave for the day, they power down their workstations, causing their registration with Asterix to quickly timeout. Here's the entry for one such extension in extensions.conf: exten => 8102,1,Dial(SIP/someone,20) exten => 8102,2,Voicemail(u8102) exten => 8102,3,Hangup exten => 8102,102,Voicemail(b8102)

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

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

On your web you have a link http://www.voip-info.org/wiki-Asterisk+settings+Diamondcard To Setup Calling with Diamondcard.us and I signed up and paid the money according to Stephen Karrington it was all automated... And it was automated to take money but when you look for service hookups or information you don't get it. I have tried now for last little while to contact them for support

I'm having issues with echo. My setup is Polycom IP501 phones connected to an Adit 600 via T100P. 13 Lines going to the Adit. All echo so far is on the local side (Employees hears own voice, but only on some calls). Watching the channel with ztmonitor, I notice that TXGAIN is pegged out most of the time. RXGAIN is anywhere between 45 and 60%. Is it safe to assume that playing around

Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. Thanks dave -- Dave Weis "I believe there are more instances of the abridgment djweis@sjdjweis.com of the freedom of the