I'm trying to figure out how to handle our fax line when we switch to our asterisk for voice. After a lot of reading and poking about I have concluded, as have many others it would seem, that the best thing to do is either to have a separate pstn fax line or use some sort of internet faxing service rather than try and make faxing work in a way it's not meant to over voip lines. The question I can't seem to find a good answer to is if there is a service/software that would allow a DID to be transferred to them and then they perform the t.38 gateway/conversion functions to which I can connect with asterisk as a t.38 endpoint and originator, or if there is a way that I could host that on my own server? So essentially I am a bit confused that asterisk supports t.38 as an endpoint or originator, but there doesn't seem to be a way to convert to/from analog for interoperating with "normal" fax machines. I'm sure something exists or the code wouldn't have been written into asterisk... Can someone point me in the right direction? Brendan Martens
What version of *? Are you going all VOIP for your voice or are you using a T1/E1? *? 1.4 has t38 pass-through and 1.6 has pass-through and termination, but 1.6 was just release and I would not suggest using it in a production environment unless you can tolerate problem or even outages. If you are planning on using a T1/E1 then send incoming calls to iaxmodem/hylafax or to an ATA/FXS card. Either works very well. Jonn -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Brendan Martens Sent: Wednesday, October 22, 2008 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] fax / t38 gateway I'm trying to figure out how to handle our fax line when we switch to our asterisk for voice. After a lot of reading and poking about I have concluded, as have many others it would seem, that the best thing to do is either to have a separate pstn fax line or use some sort of internet faxing service rather than try and make faxing work in a way it's not meant to over voip lines. The question I can't seem to find a good answer to is if there is a service/software that would allow a DID to be transferred to them and then they perform the t.38 gateway/conversion functions to which I can connect with asterisk as a t.38 endpoint and originator, or if there is a way that I could host that on my own server? So essentially I am a bit confused that asterisk supports t.38 as an endpoint or originator, but there doesn't seem to be a way to convert to/from analog for interoperating with "normal" fax machines. I'm sure something exists or the code wouldn't have been written into asterisk... Can someone point me in the right direction? Brendan Martens _______________________________________________ -- 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
Olivier wrote:> > > 2008/10/24 Brendan Martens <brendan.martens at crosscomm.net > <mailto:brendan.martens at crosscomm.net>> > > Do you have any recommendations for good ones, or, non-buggy ones?Some of or resellers are using 2102 apparently with no issues :) Senad www.bicomsystems.com
> The folks that devloped the fax V.protocols took into acount typical > copper problems like noise or echo. But what they never conceived of as > even being possible is that a call might shift around in the time > domain. Thanks to jitter/latency, the delay time of a call can change in > the middle of the call. That isn't possible with copper technologies. > This makes faxing over even G.711 a dice roll. > > IMO, with a sufficiently large buffer and a rock-solid quartz clocking > system that goes way beyond what is typically seen, it might be > theoretically possible to send a fax over VOIP.Hi All, This is a good discussion. I can support most of the findings here as I have recently spent a lot of time in the lab with T38 equipment from several vendors. Interoperability is a toss up, some ATA's only work with the parent vendor gateways, some gateways are more forgiving and work with Asterisk, some ATA's work straight out of the box, others require a PHD to configure, etc..... What I ended up with for a rock solid "ON-NET" T38 gateway to T38 fax ATA is the Audiocodes Mediant 1000 with Audiocodes ATA's (MPXXX). I emphasis on-net because I control my environment end-to-end from PSTN-Data Center-T1's-Customer-QOS LAN. I was able to reliably push 100's of faxes, multi page, single page, high density, high resolution, natted ATA's, various scenarios successful. I was very excited about T38 and was convinced this was the solution I would build on. So I took one of the ATA's home to test over the Internet (great connection, low hops to my datacenter, low latency, low jitter, lots of bandwidth) and was very disappointed when I could not get more than a 3 page fax through without errors. I started getting protocol and various page errors. I tweaked every T38 parameter that Audiocodes had with zero improvement. So I have to say, my confidence in T38 is very low, at least where open Internet connections are being used. I'm now going to look at some other technology, fax over HTTPS. I will be testing the FaxBack products to see how they stack up. JR --------------------- JR Richardson Engineering for the Masses
Here is the QOS script that I use on my bridge. http://www.taylortelephone.com/asterisk/astshape I have also had a very high success rate with Fax-->ATA-->SIP-->Asterisk-->SIP-->PSTN and the other way. The fax is a Brother MFC-440CN. I have posted most of my hylafax iaxmodem configs and other asterisk setup scripts. All are welcome. Steve, you have done a wonder full job on your spandsp library! THANK YOU I have the ability to do a lot of pure VOIP testing if you need it. Jonn -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Underwood Sent: Thursday, October 30, 2008 6:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] fax / t38 gateway Jonn R Taylor wrote:> I have been able to repeat the results at other locations. The location that has 26 pages is a linksys PAP2T our accounting person uses remotely to fax stuff to the office. The ATA is behind a DIL-625 router with QOS on a DSL line. > > I can send faxes from my test sever at home that is using an IAX trunk to our office, same ATA. FAX--->ATA--->SIP--->Asterisk/gateway/QOS_shaper--->cablemodem--->internet--->cablemodem_office--->QOS_bridge--->asterisk/iaxmodem-hylafax So, if you look at the protocols that are used it is SIP--->IAX--->SIP--->PSTN or SIP--->IAX--->IAXmodem > > This same connection is currently handling 4000 emails a day, webmail, POP, IMAP, MAPI, VPN traffic, web traffic, and normal web surfing with downloading. One test that I did with the download is started a 650MB iso download at about 900kB. Now at the same time started to send and receive faxes at the same time and worked. > > Jonn >That signal chain is probably part of the reason you get away with this so often. iaxmodem does not have the real time constraints of a real FAX machine. Its still real time, but not as tightly constrained to a smooth flow of data. If packets are delayed, and jitter is high, iaxmodem can be very tolerant, as long as the packet loss is really low. Your QoS should groom things in the other direction, and ensure a reasonably smooth outward flow of packets. Steve _______________________________________________ -- 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
On 10/31/08, Jonn R Taylor <jonnt at taylortelephone.com> wrote:> Here is the QOS script that I use on my bridge. > > http://www.taylortelephone.com/asterisk/astshapeYou should upgrade to the newer astshape script. It classifies traffic using iptables, which is much more flexible. It also has beta support for the HFSC qdisc: http://astlinux.svn.sourceforge.net/viewvc/astlinux/trunk/package/iproute2/astshape -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com
Thanks Kristian I will checkout the new script and see how it goes! Jonn -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kristian Kielhofner Sent: Friday, October 31, 2008 1:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] fax / t38 gateway On 10/31/08, Jonn R Taylor <jonnt at taylortelephone.com> wrote:> Here is the QOS script that I use on my bridge. > > http://www.taylortelephone.com/asterisk/astshapeYou should upgrade to the newer astshape script. It classifies traffic using iptables, which is much more flexible. It also has beta support for the HFSC qdisc: http://astlinux.svn.sourceforge.net/viewvc/astlinux/trunk/package/iproute2/astshape -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com _______________________________________________ -- 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