Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Larry Moore omninet.net.au> writes: > > sip.conf > > [general] > faxdetect=t38 > > [sipcall.ch] > directmedia=no > > In extensions.conf change Wait(2) to Wait(5), if your VSP sends you a > T.38 re-invite this will trigger the switch to the Fax extension. > > If this proves successful you can work on removing the Wait() from your > dialplan as Asterisk will remain in the audio path and should > successfully switch to the fax extension if extension 200 or 201 answer > a call that happens to be a fax. > > Larry. > Hi to all Sorry to bump this old thread. Well, I found a while ago finally the reason why T.38 doesn't work in conjunction with Swiss VoIP provider sipcall. Despite T.38 is stated as "supported", that provider does NOT support T.38. Their T.38 gateway has some fundamental negotiation problems, - it "exceeds the T4 timer of the T.30 protocol". Therefore, T.38 faxing does not work. http://wiki.innovaphone.com/index.php?title=Howto:Sipcall_business_-_SIP_Provider_Compatibility_Test Sipcall has confirmed me that they work now on a solution. Will see... Kind regards, Clemens -- _ -- 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] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
On 3/02/2014 3:34 PM, Jakob-Matthias Böttger wrote:
.
.
.
[sipcall.ch]
type=peer
insecure=invite
defaultuser=123456789
fromuser=123456789
fromdomain=voipdomain.com
secret=gueswhat
host=voipdomain.com
qualify=yes
context=from-sip
dtmfmode=rfc2833
callbackextension=123456789
add
directmedia=no
setvar=FAXOPT(gateway)=no
change
insecure=port,invite
[fax-rx]
exten = receive,1,NoOp( FAX RECEIVE )
exten = receive,n,Set(GLOBAL(FAXCOUNT)=$[${GLOBAL(FAXCOUNT)} + 1])
exten = receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
Do you want to keep your received faxes or is it OK to overwrite them
the next time asterisk is re-started!?
udptl.conf
[general]
udptlstart=4000
udptlend=4999
udptlfecentries = 3
udptlfecspan = 3
use_even_ports = no
You may want to change
use_even_ports=yes
You will need to restart Asterisk for this change.
Some other suggestion if the above doesn't help are;
faxdetect=cng
t38pt_udptl=no
Larry.
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Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Hi, changing
faxdetect=cng
and
t38pt_udptl=no
helped making it work.
Thanks
Am 03.02.2014 11:57, schrieb Larry Moore:
On 3/02/2014 3:34 PM, Jakob-Matthias Böttger wrote:
.
.
.
[sipcall.ch]
type=peer
insecure=invite
defaultuser=123456789
fromuser=123456789
fromdomain=voipdomain.com
secret=gueswhat
host=voipdomain.com
qualify=yes
context=from-sip
dtmfmode=rfc2833
callbackextension=123456789
add
directmedia=no
setvar=FAXOPT(gateway)=no
change
insecure=port,invite
[fax-rx]
exten = receive,1,NoOp( FAX RECEIVE )
exten = receive,n,Set(GLOBAL(FAXCOUNT)=$[${GLOBAL(FAXCOUNT)} + 1])
exten = receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
Do you want to keep your received faxes or is it OK to overwrite them
the next time asterisk is re-started!?
udptl.conf
[general]
udptlstart=4000
udptlend=4999
udptlfecentries = 3
udptlfecspan = 3
use_even_ports = no
You may want to change
use_even_ports=yes
You will need to restart Asterisk for this change.
Some other suggestion if the above doesn't help are;
faxdetect=cng
t38pt_udptl=no
Larry.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
On 3/02/2014 7:15 PM, Jakob-Matthias Böttger wrote: Hi, changing faxdetect=cng and t38pt_udptl=no helped making it work. Hmm, the fax will be received as an audio call rather than T.38, setting t38pt_udptl=no has turned off T.38. Do you know if your upstream provider supports T.38? 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] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data Am 03.02.2014 12:34, schrieb Larry Moore: On 3/02/2014 7:15 PM, Jakob-Matthias Böttger wrote: Hi, changing faxdetect=cng and t38pt_udptl=no helped making it work. Hmm, the fax will be received as an audio call rather than T.38, setting t38pt_udptl=no has turned off T.38. Do you know if your upstream provider supports T.38? 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] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data In udptl.conf set use_even_ports=yes and then issue a reload. You can confirm the settings have been applied by performing udptl show config. Change the the t38 line to read as; t38pt_udptl=yes,redundancy,maxdatagram=400 Reload sip and test. -- _ -- 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] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data In udptl.conf set use_even_ports=yes and then issue a reload. You can confirm the settings have been applied by performing udptl show config. Change the the t38 line to read as; t38pt_udptl=yes,redundancy,maxdatagram=400 Reload sip and test. after that i started udptl debug as well and now i'm getting lots of UDPTL (SIP/sipcall.ch-0007): packet to 212.117.203.76:24492 (seq 152, len 11) and in between [Feb 3 13:18:30] WARNING[26313][C-0007]: res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short and in the end [Feb 3 13:18:37] WARNING[9942]: chan_sip.c:4409 __sip_autodestruct: Autodestruct on dialog '24d15e0d-28df847a-9fae13c-7...@sip.iforb.com~1o' with owner SIP/sipcall.ch-0007 in place (Method: BYE). Rescheduling destruction for 1 ms [Feb 3 13:18:41] ERROR[26313][C-0007]: res_fax.c:1535 generic_fax_exec: channel 'SIP/sipcall.ch-0007' FAX session '7' failure, reason: 'fax session timed-out' (TIMEOUT) == Spawn extension (fax-rx, receive, 11) exited non-zero on 'SIP/sipcall.ch-0007' Thx, Jakob -- _ -- 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] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger: Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data In udptl.conf set use_even_ports=yes and then issue a reload. You can confirm the settings have been applied by performing udptl show config. Change the the t38 line to read as; t38pt_udptl=yes,redundancy,maxdatagram=400 Reload sip and test. after that i started udptl debug as well and now i'm getting lots of UDPTL (SIP/sipcall.ch-0007): packet to 212.117.203.76:24492 (seq 152, len 11) and in between [Feb 3 13:18:30] WARNING[26313][C-0007]: res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short and in the end [Feb 3 13:18:37] WARNING[9942]: chan_sip.c:4409 __sip_autodestruct: Autodestruct on dialog '24d15e0d-28df847a-9fae13c-7...@sip.iforb.com~1o' with owner SIP/sipcall.ch-0007 in place (Method: BYE). Rescheduling destruction for 1 ms [Feb 3 13:18:41] ERROR[26313][C-0007]: res_fax.c:1535 generic_fax_exec: channel 'SIP/sipcall.ch-0007' FAX session '7' failure, reason: 'fax session timed-out' (TIMEOUT) == Spawn extension (fax-rx, receive, 11) exited non-zero on 'SIP/sipcall.ch-0007' Thx, Jakob may do i have to open more ports then udp 1:2 (RTP), udp 4000:4999 (UDPTL) and tcp 5060,5061(SIP/TLS) -- _ -- 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] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
On 3/02/2014 8:42 PM, Jakob-Matthias Böttger wrote: Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger: Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data In udptl.conf set use_even_ports=yes and then issue a reload. You can confirm the settings have been applied by performing udptl show config. Change the the t38 line to read as; t38pt_udptl=yes,redundancy,maxdatagram=400 Reload sip and test. after that i started udptl debug as well and now i'm getting lots of UDPTL (SIP/sipcall.ch-0007): packet to 212.117.203.76:24492 (seq 152, len 11) and in between [Feb 3 13:18:30] WARNING[26313][C-0007]: res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short and in the end [Feb 3 13:18:37] WARNING[9942]: chan_sip.c:4409 __sip_autodestruct: Autodestruct on dialog '24d15e0d-28df847a-9fae13c-7...@sip.iforb.com~1o' with owner SIP/sipcall.ch-0007 in place (Method: BYE). Rescheduling destruction for 1 ms [Feb 3 13:18:41] ERROR[26313][C-0007]: res_fax.c:1535 generic_fax_exec: channel 'SIP/sipcall.ch-0007' FAX session '7' failure, reason: 'fax session timed-out' (TIMEOUT) == Spawn extension (fax-rx, receive, 11) exited non-zero on 'SIP/sipcall.ch-0007' Thx, Jakob may do i have to open more ports then udp 1:2 (RTP), udp 4000:4999 (UDPTL) and tcp 5060,5061(SIP/TLS) The T.38 connection will be attempted when ReceiveFax is called. The port number to use should be in the SDP information, yes, allow udp ports 4000-4999 in and out. If your firewall can be so configured you could set it to allow traffic in and out based upon the user ID Asterisk is running as, assuming it is using a unique unprivileged id. You may like to try the following to see if your SIP provider will initiate a T.38 re-invite. sip.conf [general] faxdetect=t38 [sipcall.ch] directmedia=no In extensions.conf change Wait(2) to Wait(5), if your VSP sends you a T.38 re-invite this will trigger the switch to the Fax extension. If this proves successful you can work on removing the Wait() from your dialplan as Asterisk will remain in the audio path and should successfully switch to the fax extension if extension 200 or 201 answer a call that happens to be a fax. 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
