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tnelson at rockbochs.com
Guest
|
 Posted: Thu Oct 23, 2014 11:43 am Post subject: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force R |
 |
|
----- Original Message -----
| Quote: | On 10/22/2014 03:55 PM, Tim Nelson wrote:
| Quote: | ----- Original Message -----
| Quote: | Working with the T.38 gateway functionality that is sparsely
documented [1], I'm attempting to get the following functional:
|
| Quote: | Asterisk calling system -> Asterisk system in T.38 Gateway Mode
(box
in question) -> SIP Provider
|
| Quote: | -The provider is not initiating a reinvite to T.38, even though it
is
100% supported
-Asterisk is not detecting the CNG tones from the far side of the
call and initiating a T.38 session on that call leg (with the SIP
provider), but *DOES* attempt to initiate a T.38 session with the
calling Asterisk system (which rejects with SIP/488 as expected)
|
| Quote: | So, how does one force a reinvite to T.38 on the outbound call leg
in
this scenario? I did find the same problem from another user on
the
list in the archives, but didn't find a solution contained within
the responses [2].
|
*bump*
Any thoughts? I'm quite familiar with the T.38 functionality within
Callweaver, and a function is provided there to do exactly what I
need ( SipT38SwitchOver() ). However, given Callweaver is ancient
at this point, and better T.38 features such as "gateway" do not
function, I am pressed to use Asterisk (11.13.1) with SpanDSP
(0.0.5, latest from Github since spandsp.org is down) for this
job. 
Thanks!
--Tim
|
I can't help with your root problem (maybe check "core show function
FAXOPT"?), but the spandsp site is up. Try using www.spandsp.org.
Downloads are available here:
http://www.spandsp.org/downloads/spandsp/
|
It is up now, thanks!
--Tim
--
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tnelson at rockbochs.com
Guest
|
| Posted: Thu Oct 23, 2014 11:44 am Post subject: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force R |
|
|
----- Original Message -----
| Quote: |
On 23/10/2014 3:55 AM, Tim Nelson wrote:
| Quote: | ----- Original Message -----
| Quote: | Working with the T.38 gateway functionality that is sparsely
documented [1], I'm attempting to get the following functional:
|
| Quote: | Asterisk calling system -> Asterisk system in T.38 Gateway Mode
(box
in question) -> SIP Provider
|
| Quote: | -The provider is not initiating a reinvite to T.38, even though it
is
100% supported
-Asterisk is not detecting the CNG tones from the far side of the
call and initiating a T.38 session on that call leg (with the SIP
provider), but *DOES* attempt to initiate a T.38 session with the
calling Asterisk system (which rejects with SIP/488 as expected)
|
| Quote: | So, how does one force a reinvite to T.38 on the outbound call leg
in
this scenario? I did find the same problem from another user on
the
list in the archives, but didn't find a solution contained within
the responses [2].
|
*bump*
Any thoughts? I'm quite familiar with the T.38 functionality within
Callweaver, and a function is provided there to do exactly what I
need ( SipT38SwitchOver() ). However, given Callweaver is ancient
at this point, and better T.38 features such as "gateway" do not
function, I am pressed to use Asterisk (11.13.1) with SpanDSP
(0.0.5, latest from Github since spandsp.org is down) for this
job.
|
No thoughts on your problem, I do think you will need a newer version
of
spandsp through - the site seems to be up now.
|
The version of SpanDSP is not in question at this point. The problem lies in I need a way to use the T38 Gateway function, but *also* initiate the reinvite to T.38 on the call as the provider will not do this, saying it is the *caller*'s responsibility. This is contrary to past experience however...
--Tim
--
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tnelson at rockbochs.com
Guest
|
| Posted: Thu Oct 23, 2014 11:47 am Post subject: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force R |
|
|
----- Original Message -----
| Quote: |
On 22/10/2014 11:23 AM, Tim Nelson wrote:
| Quote: | Greetings-
Working with the T.38 gateway functionality that is sparsely
documented
[1], I'm attempting to get the following functional:
|
What type of endpoint are you using which is originating the call and
is
it T.38 capable?
|
The originating endpoint is an IAXmodem controlled by Hylafax. Actual call flow is IAXmodem --G.711u via localhost--> Asterisk (old version with no T.38 support) --G.711u--> Asterisk 11.x --G.711u/T.38--> ITSP
The problem lies on the Asterisk 11.x system not being able to reinvite to T.38 on the call leg with the ITSP, and given the ITSP does not do this either, the call is stuck in G.711u with varying performance. :/
--Tim
--
_____________________________________________________________________
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tnelson at rockbochs.com
Guest
|
| Posted: Thu Oct 23, 2014 11:49 am Post subject: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force R |
|
|
----- Original Message -----
| Quote: |
On 23/10/2014 10:07 PM, Larry Moore wrote:
| Quote: |
On 22/10/2014 11:23 AM, Tim Nelson wrote:
| Quote: | Greetings-
Working with the T.38 gateway functionality that is sparsely
documented
[1], I'm attempting to get the following functional:
|
What type of endpoint are you using which is originating the call
and is
it T.38 capable?
Larry.
|
Have you had a look at
https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
As an exercise you could disable T.38 on 'Asterisk calling system',
if
you have an ATA which is originating the call to 'Asterisk calling
system' disable T.38 on that device too and disable in your sip.conf
using t38pt_udptl=no.
If you are using SendFax() on 'Asterisk calling system' ensure T.38
is
not able to be used.
If using an ATA connecting to 'Asterisk calling system' ensure you
have
set in your peer's configuration canreinvite=no or directmedia=no,
depending on the version of Asterisk you are running on this system.
On Asterisk system in '(box in question)' set directmedia=no for the
peer which is connecting to 'SIP Provider' and also to 'Asterisk
calling
system', you may want to set setvar=FAXOPT(gateway)=yes in your peer
config to 'SIP Provider' otherwise it will need to be set in your
dialplan.
Set your verbose & debug to at least 3 on '(box in question)',
possibly
a little higher and send a fax - you may now see the Fax Gateway
detect
CED. Not sure if this is suppressed in
You may want enable udptl debugging on '(box in question)'.
|
I do *not* want to disable reinvites or udptl media as it is required for T.38 operation. All testing shows (via packet capture) no reinvite for T.38 is happening on the call leg with the ITSP.
Thank you for the idea however on setting the FAXOPT for gateway in the provider SIP peer definition, I will test that shortly.
--Tim
--
_____________________________________________________________________
-- 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
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lmoore at omninet.net.au
Guest
|
| Posted: Thu Oct 23, 2014 5:48 pm Post subject: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force R |
|
|
On 24/10/2014 12:49 AM, Tim Nelson wrote:
| Quote: | ----- Original Message -----
| Quote: |
On 23/10/2014 10:07 PM, Larry Moore wrote:
| Quote: |
On 22/10/2014 11:23 AM, Tim Nelson wrote:
| Quote: | Greetings-
Working with the T.38 gateway functionality that is sparsely
documented
[1], I'm attempting to get the following functional:
|
What type of endpoint are you using which is originating the call
and is
it T.38 capable?
Larry.
|
Have you had a look at
https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
As an exercise you could disable T.38 on 'Asterisk calling system',
if
you have an ATA which is originating the call to 'Asterisk calling
system' disable T.38 on that device too and disable in your sip.conf
using t38pt_udptl=no.
If you are using SendFax() on 'Asterisk calling system' ensure T.38
is
not able to be used.
If using an ATA connecting to 'Asterisk calling system' ensure you
have
set in your peer's configuration canreinvite=no or directmedia=no,
depending on the version of Asterisk you are running on this system.
On Asterisk system in '(box in question)' set directmedia=no for the
peer which is connecting to 'SIP Provider' and also to 'Asterisk
calling
system', you may want to set setvar=FAXOPT(gateway)=yes in your peer
config to 'SIP Provider' otherwise it will need to be set in your
dialplan.
Set your verbose& debug to at least 3 on '(box in question)',
possibly
a little higher and send a fax - you may now see the Fax Gateway
detect
CED. Not sure if this is suppressed in
You may want enable udptl debugging on '(box in question)'.
|
I do *not* want to disable reinvites or udptl media as it is required for T.38 operation. All testing shows (via packet capture) no reinvite for T.38 is happening on the call leg with the ITSP.
Thank you for the idea however on setting the FAXOPT for gateway in the provider SIP peer definition, I will test that shortly.
|
The canreinvite= option is an old setting, this is replaced by the
directmedia= option in newer versions of Asterisk, it doesn't prevent a
re-invite, it keeps the audio going through asterisk rather than
negotiating an audio channel directly with the other endpoint.
The reason I suggested disabling udptl at that end is because my
understanding of how the implementation of T.38 Gateway works on
Asterisk is;
1) it does not utilise any of the T.38 gateway features in spandsp
2) the gateway will not step in if the originator negotiates T.38
Considering the other post you sent, are you suing IAX between the two
Asterisk boxes?
To test the T.38 Gateway can work on your box in question set up an IAX
modem and configure HylaFAX modem to use the iaxmodem on the box in
question, test the gateway functionality.
When I tested Asterisk 11 a little while back I configured HylaFAX on my
current system to communicate with an IAX modem on my Asterisk 11 test
box and was able to observe the T.38 gateway function.
I can't tell from the information you've provided if the old Asterisk
box is on the same network or having to traverse a WAN link to make the
connection out through to your SIP provider.
Perhaps you could provide more information about your set up such as
entries from your sip.conf, iax.conf, udptl.conf etc.
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:
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lmoore at omninet.net.au
Guest
|
| Posted: Sat Oct 25, 2014 10:44 am Post subject: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force R |
|
|
On 24/10/2014 12:47 AM, Tim Nelson wrote:
| Quote: | ----- Original Message -----
| Quote: |
On 22/10/2014 11:23 AM, Tim Nelson wrote:
| Quote: | Greetings-
Working with the T.38 gateway functionality that is sparsely
documented
[1], I'm attempting to get the following functional:
|
What type of endpoint are you using which is originating the call and
is
it T.38 capable?
|
The originating endpoint is an IAXmodem controlled by Hylafax. Actual call flow is IAXmodem --G.711u via localhost--> Asterisk (old version with no T.38 support) --G.711u--> Asterisk 11.x --G.711u/T.38--> ITSP
The problem lies on the Asterisk 11.x system not being able to reinvite to T.38 on the call leg with the ITSP, and given the ITSP does not do this either, the call is stuck in G.711u with varying performance. :/
--Tim
|
IAXmodem (other host on network) -> Asterisk 1.2 (IAX) -> Asterisk 1.8
with Fax Gateway Patch -> SIP provider -> PSTN Fax destination
I have successfully sent a fax using a full page image via an Asterisk
1.2 system which forwards the request to my Asterisk 1.8 over an IAX
channel, Asterisk 1.8 has the T.38 Fax Gateway patch installed. The
outbound call triggered the T.38 gateway and the fax was received
without error. I have ECM disabled in my IAX modem configuration in Hylafax.
I don't have Asterisk 11 running to test with at this time however I
confirmed the T.38 Gateway functions in Asterisk 11 when testing it.
-- Accepting AUTHENTICATED call from 192.168.54.18:
| Quote: | requested format = ulaw,
requested prefs = (ulaw|alaw|slin),
actual format = alaw,
host prefs = (alaw|ulaw),
priority = mine
|
-- Executing [<PSTN Number>@FAX-T30:1] Dial("IAX2/faxgw-iax-1210",
"SIP/<PSTN Number>@itsp-fax,55") in new stack
== Using SIP RTP TOS bits 184
-- Called SIP/<PSTN Number>@itsp-fax
-- SIP/itsp-fax-0000000b is making progress passing it to
IAX2/faxgw-iax-1210
-- SIP/itsp-fax-0000000b is making progress passing it to
IAX2/faxgw-iax-1210
== Using SIP RTP TOS bits 184
-- SIP/itsp-fax-0000000b answered IAX2/faxgw-iax-1210
[Oct 25 23:24:11] NOTICE[27896]: channel.c:4220 __ast_read: Dropping
incompatible voice frame on IAX2/faxgw-iax-1210 of format slin since our
native format has changed to 0x8 (alaw)
-- Got Fax Tone CED Chan SIP/itsp-fax-0000000b [1] Sending T.38
Params Peer Is IAX2/faxgw-iax-1210 [0]
-- Request on IAX2/faxgw-iax-1210 [0] Storing I:
SIP/itsp-fax-0000000b [1]
== Using UDPTL TOS bits 184
-- Negotiated on SIP/itsp-fax-0000000b [4] Ignoring I:
IAX2/faxgw-iax-1210 [0]
-- T.38 Gateway starting for chan SIP/itsp-fax-0000000b and peer
IAX2/faxgw-iax-1210
pbx*CLI> iax2 show channels
Channel Peer Username ID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format FirstMsg LastMsg
IAX2/faxgw-iax-1210 192.168.54.18 faxgw-iax 01210/00004
00010/00005 00000ms -0001ms 0000ms alaw Rx:NEW Tx:ACK
1 active IAX channel
pbx*CLI> fax show sessions
Current FAX Sessions:
Channel Tech FAXID Type Operation State
File(s)
SIP/itsp-fax-0000000 Spandsp 1 T.38 receive Active
(null)
1 FAX sessions
-- Executing [h@FAX-T30:1] GotoIf("IAX2/faxgw-iax-1210", "0?2:3")
in new stack
-- Goto (FAX-T30,h,3)
-- Executing [h@FAX-T30:3] NoOp("IAX2/faxgw-iax-1210", "Finish
if_FAX-T30_37") in new stack
-- Executing [h@FAX-T30:4] NoOp("IAX2/faxgw-iax-1210", "Call/Fax
Ended 2014-10-25 23:27:38 +0800") in new stack
-- Connection Statistics
Bit Rate :14400
ECM : No
Pages : 1
== Spawn extension (FAX-T30, <PSTN Number>, 1) exited non-zero on
'IAX2/faxgw-iax-1210'
-- Hungup 'IAX2/faxgw-iax-1210'
--
_____________________________________________________________________
-- 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 |
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lmoore at omninet.net.au
Guest
|
| Posted: Tue Oct 28, 2014 8:56 am Post subject: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force R |
|
|
On 24/10/2014 12:49 AM, Tim Nelson wrote:
| Quote: | ----- Original Message -----
| Quote: |
On 23/10/2014 10:07 PM, Larry Moore wrote:
| Quote: |
On 22/10/2014 11:23 AM, Tim Nelson wrote:
| Quote: | Greetings-
Working with the T.38 gateway functionality that is sparsely
documented
[1], I'm attempting to get the following functional:
|
What type of endpoint are you using which is originating the call
and is
it T.38 capable?
Larry.
|
Have you had a look at
https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
As an exercise you could disable T.38 on 'Asterisk calling system',
if
you have an ATA which is originating the call to 'Asterisk calling
system' disable T.38 on that device too and disable in your sip.conf
using t38pt_udptl=no.
If you are using SendFax() on 'Asterisk calling system' ensure T.38
is
not able to be used.
If using an ATA connecting to 'Asterisk calling system' ensure you
have
set in your peer's configuration canreinvite=no or directmedia=no,
depending on the version of Asterisk you are running on this system.
On Asterisk system in '(box in question)' set directmedia=no for the
peer which is connecting to 'SIP Provider' and also to 'Asterisk
calling
system', you may want to set setvar=FAXOPT(gateway)=yes in your peer
config to 'SIP Provider' otherwise it will need to be set in your
dialplan.
Set your verbose& debug to at least 3 on '(box in question)',
possibly
a little higher and send a fax - you may now see the Fax Gateway
detect
CED. Not sure if this is suppressed in
You may want enable udptl debugging on '(box in question)'.
|
I do *not* want to disable reinvites or udptl media as it is required for T.38 operation. All testing shows (via packet capture) no reinvite for T.38 is happening on the call leg with the ITSP.
Thank you for the idea however on setting the FAXOPT for gateway in the provider SIP peer definition, I will test that shortly.
--Tim
|
It would seem for Asterisk 11 and T.38 Gateway work for an IAX channel
you require the following;
IAX2 -> SIP -> T.38 Gateway -> ITSP (SIP)
Where as it would be nicer if it would accept acting as a gateway for an
IAX channel i.e.;
IAX2 - T.38 Gateway -> ITSP (SIP)
If an IAX2 channel is connected directly to a context with
FAXOPT(t38gateway) enabled I see 'ast_rtp_read: RTP Read too short'
messages and a failed transmission, the same is observed if using SIP
with udptl=no instead of IAX2 channel;
SIP (udptl=no) -> T.38 Gateway -> ITSP (SIP).
Not sure if this is by design!
Maybe time for another friendly chat with your ITSP in the hope they can
resolve the issue.
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 |
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lmoore at omninet.net.au
Guest
|
| Posted: Tue Oct 28, 2014 8:56 am Post subject: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force R |
|
|
On 25/10/2014 11:43 PM, Larry Moore wrote:
| Quote: |
On 24/10/2014 12:47 AM, Tim Nelson wrote:
| Quote: | ----- Original Message -----
| Quote: |
On 22/10/2014 11:23 AM, Tim Nelson wrote:
| Quote: | Greetings-
Working with the T.38 gateway functionality that is sparsely
documented
[1], I'm attempting to get the following functional:
|
What type of endpoint are you using which is originating the call and
is
it T.38 capable?
|
The originating endpoint is an IAXmodem controlled by Hylafax. Actual
call flow is IAXmodem --G.711u via localhost--> Asterisk (old version
with no T.38 support) --G.711u--> Asterisk 11.x --G.711u/T.38--> ITSP
The problem lies on the Asterisk 11.x system not being able to
reinvite to T.38 on the call leg with the ITSP, and given the ITSP
does not do this either, the call is stuck in G.711u with varying
performance. :/
--Tim
|
IAXmodem (other host on network) -> Asterisk 1.2 (IAX) -> Asterisk 1.8
with Fax Gateway Patch -> SIP provider -> PSTN Fax destination
I have successfully sent a fax using a full page image via an Asterisk
1.2 system which forwards the request to my Asterisk 1.8 over an IAX
channel, Asterisk 1.8 has the T.38 Fax Gateway patch installed. The
outbound call triggered the T.38 gateway and the fax was received
without error. I have ECM disabled in my IAX modem configuration in
Hylafax.
I don't have Asterisk 11 running to test with at this time however I
confirmed the T.38 Gateway functions in Asterisk 11 when testing it.
-- Accepting AUTHENTICATED call from 192.168.54.18:
| Quote: | requested format = ulaw,
requested prefs = (ulaw|alaw|slin),
actual format = alaw,
host prefs = (alaw|ulaw),
priority = mine
|
-- Executing [<PSTN Number>@FAX-T30:1] Dial("IAX2/faxgw-iax-1210",
"SIP/<PSTN Number>@itsp-fax,55") in new stack
== Using SIP RTP TOS bits 184
-- Called SIP/<PSTN Number>@itsp-fax
-- SIP/itsp-fax-0000000b is making progress passing it to
IAX2/faxgw-iax-1210
-- SIP/itsp-fax-0000000b is making progress passing it to
IAX2/faxgw-iax-1210
== Using SIP RTP TOS bits 184
-- SIP/itsp-fax-0000000b answered IAX2/faxgw-iax-1210
[Oct 25 23:24:11] NOTICE[27896]: channel.c:4220 __ast_read: Dropping
incompatible voice frame on IAX2/faxgw-iax-1210 of format slin since our
native format has changed to 0x8 (alaw)
-- Got Fax Tone CED Chan SIP/itsp-fax-0000000b [1] Sending T.38 Params
Peer Is IAX2/faxgw-iax-1210 [0]
-- Request on IAX2/faxgw-iax-1210 [0] Storing I: SIP/itsp-fax-0000000b [1]
== Using UDPTL TOS bits 184
-- Negotiated on SIP/itsp-fax-0000000b [4] Ignoring I:
IAX2/faxgw-iax-1210 [0]
-- T.38 Gateway starting for chan SIP/itsp-fax-0000000b and peer
IAX2/faxgw-iax-1210
pbx*CLI> iax2 show channels
Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format
FirstMsg LastMsg
IAX2/faxgw-iax-1210 192.168.54.18 faxgw-iax 01210/00004 00010/00005
00000ms -0001ms 0000ms alaw Rx:NEW Tx:ACK
1 active IAX channel
pbx*CLI> fax show sessions
Current FAX Sessions:
Channel Tech FAXID Type Operation State File(s)
SIP/itsp-fax-0000000 Spandsp 1 T.38 receive Active (null)
1 FAX sessions
-- Executing [h@FAX-T30:1] GotoIf("IAX2/faxgw-iax-1210", "0?2:3") in new
stack
-- Goto (FAX-T30,h,3)
-- Executing [h@FAX-T30:3] NoOp("IAX2/faxgw-iax-1210", "Finish
if_FAX-T30_37") in new stack
-- Executing [h@FAX-T30:4] NoOp("IAX2/faxgw-iax-1210", "Call/Fax Ended
2014-10-25 23:27:38 +0800") in new stack
-- Connection Statistics
Bit Rate :14400
ECM : No
Pages : 1
== Spawn extension (FAX-T30, <PSTN Number>, 1) exited non-zero on
'IAX2/faxgw-iax-1210'
-- Hungup 'IAX2/faxgw-iax-1210'
|
Well, forgive me as I should have had an Asterisk 11 system up and
running and performing tests before posting.
It would appear there is a behavioural difference with the patch created
for Asterisk 1.8 and the implementation applied to Asterisk 11.
The observations as listed above relating to the fax gateway stepping
in, occurs when an outbound fax call is made using either of the g711
codecs, Asterisk detects the fax tones in the calling leg about 3
seconds after the call has been answered and sends a T.38 re-invite to
the ITSP.
Using Asterisk 11, when an outbound call is made, the fax gateway
detection feature does not do anything on the leg of the call (as you
have observed) to the ITSP until it receives a T.38 re-invite from the
ITSP, my observations show this occurs about 4 seconds after the call is
answered. I suspect once the T.38 re-invite is received from the ITSP,
the T.38 Gateway sends a T.38 re-invite on the leg of the caller to
check if it is capable of T.38. I have not confirmed this definitively.
I'm obviously fortunate my ITSP is correctly handling T.38.
Larry.
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