Re: [Asterisk-Users] Squished faxes with txfax
Josué, My setup is: txfax -[Asterisk] - E1 Pri -[Cisco 5350]- t.38 with Sip - [Terminating Carrier] So the t.38 is all done by the Cisco 5350 and not by asterisk at all. I have done some tests with a Cisco ATA186, but with limited success using g.711 pass through. I can receive faxes using rxfax with a similar setup but a different carrier. Finally, if I change the Cisco - Terminating Carrier connection to use g.711 the fax quality is much improved, which indicates that either the Cisco's t.38 or the terminating carriers t.38 is broken or incompatible. - Scott On Tuesday 28 March 2006 7:33 am, Josué Conti wrote: Scott, as you made to install the T38 in asterisk? To receive fax it is normal? You use an ATA? Greetings 2006/3/28, Scott Eisert [EMAIL PROTECTED]: Hello, I have been getting squished faxes very reliably when sending through Asterisk using txfax. It looks as if all horizontal white space has been removed. Interestingly it is perfectly repeatable, which seems to rule out timing related issues. My configuration is: Asterisk SVN-branch-1.2-r7337M spandsp-0.0.2pre21 ( though I have tried a 0.0.3 snapshot as well as 0.0.2pre25 ) libtiff 3.8.0-3 I am sending the fax out from Asterisk over a E1 to a Cisco 5350, which then is sending t.38 out to the terminating carrier. The fax completes every attempt, so it doesn't appear to be IP related, and as I mentioned before, the fax on the receiving end always looks almost exactly the same. Also, if I send a fax with a vertical bar down one side the image is not squished nearly as much as something with just text (like an invoice) which will be about 1/3 the size of the original. I have also tried sending faxes using iaxmodem and HylaFax to Asterisk, which gave perfect results about 1 in 30 tries, with most failures resulting in incomplete page. Any suggestions will be greatly appreciated. - Scott ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Squished faxes with txfax
Hello, I have been getting squished faxes very reliably when sending through Asterisk using txfax. It looks as if all horizontal white space has been removed. Interestingly it is perfectly repeatable, which seems to rule out timing related issues. My configuration is: Asterisk SVN-branch-1.2-r7337M spandsp-0.0.2pre21 ( though I have tried a 0.0.3 snapshot as well as 0.0.2pre25 ) libtiff 3.8.0-3 I am sending the fax out from Asterisk over a E1 to a Cisco 5350, which then is sending t.38 out to the terminating carrier. The fax completes every attempt, so it doesn't appear to be IP related, and as I mentioned before, the fax on the receiving end always looks almost exactly the same. Also, if I send a fax with a vertical bar down one side the image is not squished nearly as much as something with just text (like an invoice) which will be about 1/3 the size of the original. I have also tried sending faxes using iaxmodem and HylaFax to Asterisk, which gave perfect results about 1 in 30 tries, with most failures resulting in incomplete page. Any suggestions will be greatly appreciated. - Scott ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto Answer Fax
I have been just working on the same thing today. You can start by taking a look at this application for inbound IP faxes: http://www.voip-info.org/tiki-index.php?page=NVFaxDetect I can currently detect the fax but can't seem to capture it. - Scott On Thursday 29 September 2005 4:52 pm, Rene Nelson wrote: Can anyone point me to a good howto or example on how to get * to recognize inbound faxes and adjust accordingly? Ideally I would like it to grab the fax and email it to me, but I dont know if that is really possible yet or not. Thanks Neri ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Tandem Inbound only.
Hello, I have a carrier that is supplying me with DID inbound over SIP to my asterisk server. Because the CID is different with every call that is coming in the only way I have to authenticate this carrier is IP based. In my sip.conf I want to define this user as type=user, however this can't work because Asterisk only authenticates users by username, not IP. I can take calls in if I set type=friend or type=peer which will allow authentication by IP. The problem with this is that asterisk sends sip OPTIONS messages to the carrier, because asterisk thinks that the carrier will be receiving calls as well as sending calls. The options messages make the carrier very unhappy, and just throw errors on their end. I believe that if I were to put Ser in front of asterisk it would resolve this issue, but that seems a bit drastic, and it is not justified I think. Does anyone have an idea of how to stop asterisk from sending options messages to peers or friends, or how to authenticate based on IP address for a user? Thanks, - Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Tandem Inbound only.
On Tuesday 27 September 2005 3:12 pm, Peter Bowyer wrote: On 27/09/05, Scott Eisert [EMAIL PROTECTED] wrote: Hello, I have a carrier that is supplying me with DID inbound over SIP to my asterisk server. Because the CID is different with every call that is coming in the only way I have to authenticate this carrier is IP based. In my sip.conf I want to define this user as type=user, however this can't work because Asterisk only authenticates users by username, not IP. Check out 'insecure=very' for sip.conf. Peter It doesn't look like insecure can solve my problem. If I have type=user, I send back a 404 regardless of the insecure setting. If I have type=peer or type=friend I can receive calls but asterisk sends out Options messages regardless of the insecure setting (yes or very). Any other suggestions? - Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Tandem Inbound only.
That did it! I had a qualify=yes in the carrier definition, which was causing Asterisk to issue the OPTIONS. For the record, this is the entry in sip.conf for an inbound tandem carrier who you will not be terminating to. [carriername] type=peer host=gatewayIP context=sipInbound qualify=no dtmfmode=RFC2833 insecure=very allow=g729 Thanks guys. - Scott On Tuesday 27 September 2005 3:53 pm, Joshua Colp - Asterlink wrote: Hi Scott, To do what you want to do you do indeed need to use a peer entry, with the IP address where INVITEs will come from specified as the host, and insecure=very. Your OPTIONS though is being caused by qualify being turned on somewhere. Joshua Colp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Eisert Sent: Tuesday, September 27, 2005 4:48 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Tandem Inbound only. On Tuesday 27 September 2005 3:12 pm, Peter Bowyer wrote: On 27/09/05, Scott Eisert [EMAIL PROTECTED] wrote: Hello, I have a carrier that is supplying me with DID inbound over SIP to my asterisk server. Because the CID is different with every call that is coming in the only way I have to authenticate this carrier is IP based. In my sip.conf I want to define this user as type=user, however this can't work because Asterisk only authenticates users by username, not IP. Check out 'insecure=very' for sip.conf. Peter It doesn't look like insecure can solve my problem. If I have type=user, I send back a 404 regardless of the insecure setting. If I have type=peer or type=friend I can receive calls but asterisk sends out Options messages regardless of the insecure setting (yes or very). Any other suggestions? - Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
