Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
Have you configured the canreinvite=yes in sip peer? I am currently off work for two days, but a 100% fail means a configuration problem for sure. Leandro 2012/12/27 Eric Wieling ewiel...@nyigc.com We are offering $100 (paid via paypal or check) to the first person who assists us in successfully sending and receiving faxes in the setup described below. Offer expires Dec 31. We are a direct customer of Level 3, there is no other carrier involved. What we want to work: Level 3 T.38 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
We have set directmedia=yes as well as directmedia=no. There is no NAT involved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, December 27, 2012 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through Have you configured the canreinvite=yes in sip peer? I am currently off work for two days, but a 100% fail means a configuration problem for sure. Leandro 2012/12/27 Eric Wieling ewiel...@nyigc.com We are offering $100 (paid via paypal or check) to the first person who assists us in successfully sending and receiving faxes in the setup described below. Offer expires Dec 31. We are a direct customer of Level 3, there is no other carrier involved. What we want to work: Level 3 T.38 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
directrtpsetup=yes in sip.conf? On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling ewiel...@nyigc.com wrote: We have set directmedia=yes as well as directmedia=no. There is no NAT involved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, December 27, 2012 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through Have you configured the canreinvite=yes in sip peer? I am currently off work for two days, but a 100% fail means a configuration problem for sure. Leandro 2012/12/27 Eric Wieling ewiel...@nyigc.com We are offering $100 (paid via paypal or check) to the first person who assists us in successfully sending and receiving faxes in the setup described below. Offer expires Dec 31. We are a direct customer of Level 3, there is no other carrier involved. What we want to work: Level 3 T.38 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
We have directrtpsetup=no because the comments in the sample config indicates it does not work in all situations. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher Harrington Sent: Thursday, December 27, 2012 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through directrtpsetup=yes in sip.conf? On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling ewiel...@nyigc.com wrote: We have set directmedia=yes as well as directmedia=no. There is no NAT involved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, December 27, 2012 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through Have you configured the canreinvite=yes in sip peer? I am currently off work for two days, but a 100% fail means a configuration problem for sure. Leandro 2012/12/27 Eric Wieling ewiel...@nyigc.com We are offering $100 (paid via paypal or check) to the first person who assists us in successfully sending and receiving faxes in the setup described below. Offer expires Dec 31. We are a direct customer of Level 3, there is no other carrier involved. What we want to work: Level 3 T.38 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
True, but it should bypass Asterisk when possible for SIP streams and may solve your problem. On Thu, Dec 27, 2012 at 12:16 PM, Eric Wieling ewiel...@nyigc.com wrote: We have directrtpsetup=no because the comments in the sample config indicates it does not work in all situations. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher Harrington Sent: Thursday, December 27, 2012 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through directrtpsetup=yes in sip.conf? On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling ewiel...@nyigc.com wrote: We have set directmedia=yes as well as directmedia=no. There is no NAT involved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, December 27, 2012 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through Have you configured the canreinvite=yes in sip peer? I am currently off work for two days, but a 100% fail means a configuration problem for sure. Leandro 2012/12/27 Eric Wieling ewiel...@nyigc.com We are offering $100 (paid via paypal or check) to the first person who assists us in successfully sending and receiving faxes in the setup described below. Offer expires Dec 31. We are a direct customer of Level 3, there is no other carrier involved. What we want to work: Level 3 T.38 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
It does not appear to make any difference. Calls are still failing. -Original Message- From: Christopher Harrington [mailto:ch...@acsdi.com] Sent: Thursday, December 27, 2012 1:20 PM To: Eric Wieling Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through True, but it should bypass Asterisk when possible for SIP streams and may solve your problem. On Thu, Dec 27, 2012 at 12:16 PM, Eric Wieling ewiel...@nyigc.com wrote: We have directrtpsetup=no because the comments in the sample config indicates it does not work in all situations. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher Harrington Sent: Thursday, December 27, 2012 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through directrtpsetup=yes in sip.conf? On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling ewiel...@nyigc.com wrote: We have set directmedia=yes as well as directmedia=no. There is no NAT involved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, December 27, 2012 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through Have you configured the canreinvite=yes in sip peer? I am currently off work for two days, but a 100% fail means a configuration problem for sure. Leandro 2012/12/27 Eric Wieling ewiel...@nyigc.com We are offering $100 (paid via paypal or check) to the first person who assists us in successfully sending and receiving faxes in the setup described below. Offer expires Dec 31. We are a direct customer of Level 3, there is no other carrier involved. What we want to work: Level 3 T.38 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
Last thing to check, just for sanity's sake: t38pt_udptl=yes in sip.conf? It defaults to off. On Thu, Dec 27, 2012 at 12:32 PM, Eric Wieling ewiel...@nyigc.com wrote: It does not appear to make any difference. Calls are still failing. -Original Message- From: Christopher Harrington [mailto:ch...@acsdi.com] Sent: Thursday, December 27, 2012 1:20 PM To: Eric Wieling Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through True, but it should bypass Asterisk when possible for SIP streams and may solve your problem. On Thu, Dec 27, 2012 at 12:16 PM, Eric Wieling ewiel...@nyigc.com wrote: We have directrtpsetup=no because the comments in the sample config indicates it does not work in all situations. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher Harrington Sent: Thursday, December 27, 2012 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through directrtpsetup=yes in sip.conf? On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling ewiel...@nyigc.com wrote: We have set directmedia=yes as well as directmedia=no. There is no NAT involved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, December 27, 2012 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through Have you configured the canreinvite=yes in sip peer? I am currently off work for two days, but a 100% fail means a configuration problem for sure. Leandro 2012/12/27 Eric Wieling ewiel...@nyigc.com We are offering $100 (paid via paypal or check) to the first person who assists us in successfully sending and receiving faxes in the setup described below. Offer expires Dec 31. We are a direct customer of Level 3, there is no other carrier involved. What we want to work: Level 3 T.38 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
We are using t38pt_udptl=yes,redundancy,maxdatagram=400 Without the maxdatagram we get errors in the CLI. We also tried using FEC instead of redundancy, but no difference. -Original Message- From: Christopher Harrington [mailto:ch...@acsdi.com] Sent: Thursday, December 27, 2012 2:23 PM To: Eric Wieling Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through Last thing to check, just for sanity's sake: t38pt_udptl=yes in sip.conf? It defaults to off. On Thu, Dec 27, 2012 at 12:32 PM, Eric Wieling ewiel...@nyigc.com wrote: It does not appear to make any difference. Calls are still failing. -Original Message- From: Christopher Harrington [mailto:ch...@acsdi.com] Sent: Thursday, December 27, 2012 1:20 PM To: Eric Wieling Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through True, but it should bypass Asterisk when possible for SIP streams and may solve your problem. On Thu, Dec 27, 2012 at 12:16 PM, Eric Wieling ewiel...@nyigc.com wrote: We have directrtpsetup=no because the comments in the sample config indicates it does not work in all situations. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher Harrington Sent: Thursday, December 27, 2012 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through directrtpsetup=yes in sip.conf? On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling ewiel...@nyigc.com wrote: We have set directmedia=yes as well as directmedia=no. There is no NAT involved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, December 27, 2012 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through Have you configured the canreinvite=yes in sip peer? I am currently off work for two days, but a 100% fail means a configuration problem for sure. Leandro 2012/12/27 Eric Wieling ewiel...@nyigc.com We are offering $100 (paid via paypal or check) to the first person who assists us in successfully sending and receiving faxes in the setup described below. Offer expires Dec 31. We are a direct customer of Level 3, there is no other carrier involved. What we want to work: Level 3 T.38 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs
Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
On 28/12/2012 1:55 AM, Eric Wieling wrote: We are offering $100 (paid via paypal or check) to the first person who assists us in successfully sending and receiving faxes in the setup described below. Offer expires Dec 31. We are a direct customer of Level 3, there is no other carrier involved. What we want to work: Level 3 T.38 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. In udptl.conf try the following option ; ; Some VoIP providers will only accept an offer with an even-numbered ; UDPTL port. Set this option so that Asterisk will only attempt to use ; even-numbered ports when negotiating T.38. Default is no. use_even_ports = yes ; Looking at some old notes other options I set for some devices to be able to pass through T.38 in sip.conf were, directmedia=no t38pt_udptl=no May be worth checking the following; directrtpsetup=no Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
On 28/12/2012 4:59 AM, Larry Moore wrote: On 28/12/2012 1:55 AM, Eric Wieling wrote: . snip . directmedia=no t38pt_udptl=no snip Hmm, the t38pt_udptl will need to be set to yes, this was set to no for non T.38 capable devices I had set faxdetect=no in the peer's configuration for the T.38 capable device, perhaps this was to prevent an attempt by Asterisk to redirect the call to the fax extension in the dialplan. Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
udptl.conf settings: [general] udptlstart=4000 udptlend=4998 udptlchecksums=no udptlfecentries = 3 udptlfecspan = 3 use_even_ports = yes T38FaxUdpEC = t38UDPRedundancy T38FaxMaxDatagram = 400 sip.conf settings: directmedia=yes faxdetect = no t38pt_udptl=yes,redundancy,maxdatagram=400 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Larry Moore Sent: Thursday, December 27, 2012 4:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through On 28/12/2012 1:55 AM, Eric Wieling wrote: We are offering $100 (paid via paypal or check) to the first person who assists us in successfully sending and receiving faxes in the setup described below. Offer expires Dec 31. We are a direct customer of Level 3, there is no other carrier involved. What we want to work: Level 3 T.38 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. In udptl.conf try the following option ; ; Some VoIP providers will only accept an offer with an even-numbered ; UDPTL port. Set this option so that Asterisk will only attempt to use ; even-numbered ports when negotiating T.38. Default is no. use_even_ports = yes ; Looking at some old notes other options I set for some devices to be able to pass through T.38 in sip.conf were, directmedia=no t38pt_udptl=no May be worth checking the following; directrtpsetup=no Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
On Thu, Dec 27, 2012 at 3:45 PM, Eric Wieling ewiel...@nyigc.com wrote: sip.conf settings: directmedia=yes I know you've said you tried it both ways, but consensus seems to be that directmedia needs to be =no when using UDPTL. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
I usually set directmedia=yes with good results... Leandro 2012/12/27 Christopher Harrington ch...@acsdi.com On Thu, Dec 27, 2012 at 3:45 PM, Eric Wieling ewiel...@nyigc.com wrote: sip.conf settings: directmedia=yes I know you've said you tried it both ways, but consensus seems to be that directmedia needs to be =no when using UDPTL. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
On 28/12/2012 5:45 AM, Eric Wieling wrote: udptl.conf settings: [general] udptlstart=4000 udptlend=4998 udptlchecksums=no udptlfecentries = 3 udptlfecspan = 3 use_even_ports = yes T38FaxUdpEC = t38UDPRedundancy T38FaxMaxDatagram = 400 sip.conf settings: directmedia=yes faxdetect = no t38pt_udptl=yes,redundancy,maxdatagram=400 From memory when I was doing this in March 2011 Asterisk would not allow a T.38 connection to successfully establish when canreinvite was set to yes, I did have NAT involved in my testing hence T.38 would be successful when canreinvite=no, the options to use now seeing as canreinvite is deprecated are; directmedia=no direcrtpsetup=no The T38Fax... options you have in udptl.conf are no longer supported. I have the T.38 Fax Gateway patch applied to my installation of 1.8.18.1 though I don't believe this will make any difference as I had got my T.38 relaying working prior to the patch. I have in my sip.conf; [general] t38pt_udptl=yes,redundancy,maxdatagram=1400 You may also want to enable; t38pt_usertpsource=yes Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
