The first part of this is kind of off topic as it doesn't answer OP's original question, but instead is a reply to one of the replies.
Cisco is certainly not the only option for doing T38 gatewaying with Asterisk. I believe Asterisk 1.6 with app_fax supports T.38 origination and termination, that is not gatewaying, however if origination and termination are already there, gatewaying should be fairly trivial to implement. I haven't actually tested 1.6 using T.38, however I have read: http://www.asterisk.org/node/48457 "11873, Added core API changes to handle T.38 origination and termination (The version of app_fax in Asterisk-addons now supports this.)" Additionally, there are some 3rd party modules available for Asterisk 1.4 that will add T.38 termination, origination, and gatewaying. The ones I am thinking of specifically are the ones made by Attractel in there Attrafax package (previously known as Faxterisk): http://www.attrafax.com/attrafax.php I have used Attrafax before and it works great for us. We use it in combination with Linksys SPA2102 ATAs. We had problems with it at first but upgrading the firmware on the Linksys ATAs made the problem go away. In our case we have a PRI however and are not using SIP connections over the internet. Another option as you have already stated is using a SIP provider that supports T.38 such as gafachi. However in this particular case I understand the OP has already provisioned DIDs from a SIP provider, assuming one of these DIDs is your fax number you may find yourself with a bit of a problem if your provider does not support T.38. You may have some luck with faxing w/o T.38 using G.711a/u over the internet, but it will be patchy at best, you will probably find you will have many failed faxes doing this. Using G.711a/u internally over a LAN is one thing (still wouldn't recommend it, but you would get a high success rate), but doing it over the internet is a completely different story. If you have no real PSTN connections and are SIP only, your provider *must* support T.38 to achieve an acceptable success rate. If your DIDs are already on print materials and your provider doesn't support T.38, the only options I would see for you are: 1) Have your Fax DID ported to another SIP provider that does support T.38, you can leave your voice ones with your current provider 2) Get a new DID from a another SIP provider and re-print all of your materials (probably incredibly expensive) -- Matt -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem Helge Sent: Monday, May 19, 2008 11:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax Machine Options Cisco gateway with T.38 support. That's the only real way to do faxing through asterisk. I think a VG200 with newer firmware will support SIP + T.38 but don't buy on my suggestion because I've never used that device outside call manager configuration. Or see if your VoIP provider supports T.38 fax but you must use SIP in that case. It will work very well once you get it working.... hint: check udptl.conf On Mon, May 19, 2008 at 11:27 PM, Joseph L. Casale <[EMAIL PROTECTED]> wrote: > Is my only solution to add a fax machine to our VOIP only setup by using an > IAXy? > I should specify the office people want a traditional fax machine in the > sense that > > fax's be sent and received from a physical unit, they don't want an email to > fax setup. > They have a dedicated sip did provisioned just for the fax. > > > > What are others using? > > > > Thanks! > jlc > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
