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[asterisk-users] How to enable T.38 between SPA3102 PSTN Line port and ReceiveFAX app ?


 
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oza_4h07 at yahoo.fr
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PostPosted: Tue Nov 05, 2013 5:09 am    Post subject: [asterisk-users] How to enable T.38 between SPA3102 PSTN Lin Reply with quote

Hello,


I've got an analog phone which is currently receiving unsollicited FAX calls from PSTN.



For learning purpose, I'm preparing an Asterisk/SPA3102 setup that would let voice calls come in and out and translate incoming FAX calls to TIF files (forwarded through email)).


My target setup is :



PSTN <-- analog--> SPA3102 Line Port <-- SIP --> Asterisk <-- SIP --> SPA3102 Phone Port <-- analog --> Analog phone



When a call comes in, analog phone rings.

If callee answers and a fax tone is detected, then the incoming call is sent by Asterisk to ReceiveFAX application which translates incoming audio to TIF file.


My setup is working ok when I'm using ReceiveFAX in fallback mode (with f option).


Then I would like to improve my setup letting ReceiveFAX negociate T.38 with SPA3102.

The trouble is SPA3102, as I configured it, seems to refuse T.38 negociation as I'm reading lines like this in Asterisk logs:

  == Using UDPTL CoS mark 5
[2013-11-05 10:36:50] WARNING[3061][C-00000007]: res_fax.c:1698 receivefax_t38_init: channel 'SIP/myline-0000000e' refused to negotiate T.38


My question is:

Any hint on how to configure SPA3102 PSTN Line port so that it  would accept to upgrade to T.38 ?



When Asterisk re-invites SPA3102, here is the dialog between both boxes:




INVITE sip:myline@172.16.2.12:5061 SIP/2.0
Via: SIP/2.0/UDP 172.16.2.1:5060;branch=z9hG4bK260a479e
Max-Forwards: 70
From: <sip:4250@172.16.2.1 ([email]sip%3A4250@172.16.2.1[/email])>;tag=as1a0daffe
To: myline <sip:myline@172.16.2.1 ([email]sip%3Amyline@172.16.2.1[/email])>;tag=2d2e7e5ad74dec0co1
Contact: <sip:4250@172.16.2.1:5060>
Call-ID: c8278ef2-b80a5454@172.16.2.12 (c8278ef2-b80a5454@172.16.2.12)
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.3.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 473469291 473469292 IN IP4 172.16.2.1
s=Asterisk PBX 11.5.0
c=IN IP4 172.16.2.1
t=0 0
m=image 4506 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC

---

<--- SIP read from UDP:172.16.2.12:5061 --->
SIP/2.0 488 Not Acceptable Here
To: myline <sip:myline@172.16.2.1 ([email]sip%3Amyline@172.16.2.1[/email])>;tag=2d2e7e5ad74dec0co1
From: <sip:4250@172.16.2.1 ([email]sip%3A4250@172.16.2.1[/email])>;tag=as1a0daffe
Call-ID: c8278ef2-b80a5454@172.16.2.12 (c8278ef2-b80a5454@172.16.2.12)
CSeq: 102 INVITE
Via: SIP/2.0/UDP 172.16.2.1:5060;branch=z9hG4bK260a479e
Contact: myline <sip:myline@172.16.2.12:5061>
Warning: 304 spa "Media type not available"
Server: Linksys/SPA3102-5.1.10(GW)
Content-Length: 0





Any hint ?


Regards
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lmoore at omninet.net.au
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PostPosted: Tue Nov 05, 2013 5:22 pm    Post subject: [asterisk-users] How to enable T.38 between SPA3102 PSTN Lin Reply with quote

On 5/11/2013 6:09 PM, Olivier wrote:
Quote:
Hello,

I've got an analog phone which is currently receiving unsollicited FAX
calls from PSTN.

For learning purpose, I'm preparing an Asterisk/SPA3102 setup that would
let voice calls come in and out and translate incoming FAX calls to TIF
files (forwarded through email)).

My target setup is :

PSTN <-- analog--> SPA3102 Line Port <-- SIP --> Asterisk <-- SIP -->
SPA3102 Phone Port <-- analog --> Analog phone


When a call comes in, analog phone rings.
If callee answers and a fax tone is detected, then the incoming call is
sent by Asterisk to ReceiveFAX application which translates incoming
audio to TIF file.

My setup is working ok when I'm using ReceiveFAX in fallback mode (with
f option).

Then I would like to improve my setup letting ReceiveFAX negociate T.38
with SPA3102.
The trouble is SPA3102, as I configured it, seems to refuse T.38
negociation as I'm reading lines like this in Asterisk logs:

== Using UDPTL CoS mark 5
[2013-11-05 10:36:50] WARNING[3061][C-00000007]: res_fax.c:1698
receivefax_t38_init: channel 'SIP/myline-0000000e' refused to negotiate T.38

My question is:
Any hint on how to configure SPA3102 PSTN Line port so that it would
accept to upgrade to T.38 ?



The SPA3102 only supports T.38 on the FXS port, the FXO port uses G711
for a fax session.

The Grandstream HT503 supports T.38 on both the FXO and FXS ports.

What problem do you have receiving a fax over G711?

Larry.

--
_____________________________________________________________________
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oza_4h07 at yahoo.fr
Guest





PostPosted: Thu Nov 07, 2013 11:58 am    Post subject: [asterisk-users] How to enable T.38 between SPA3102 PSTN Lin Reply with quote

2013/11/5 Larry Moore <lmoore@omninet.net.au (lmoore@omninet.net.au)>
Quote:
On 5/11/2013 6:09 PM, Olivier wrote:
Quote:
Hello,

I've got an analog phone which is currently receiving unsollicited FAX
calls from PSTN.

For learning purpose, I'm preparing an Asterisk/SPA3102 setup that would
let voice calls come in and out and translate incoming FAX calls to TIF
files (forwarded through email)).

My target setup is :

PSTN <-- analog--> SPA3102 Line Port <-- SIP --> Asterisk <-- SIP -->
SPA3102 Phone Port <-- analog --> Analog phone


When a call comes in, analog phone rings.
If callee answers and a fax tone is detected, then the incoming call is
sent by Asterisk to ReceiveFAX application which translates incoming
audio to TIF file.

My setup is working ok when I'm using ReceiveFAX in fallback mode (with
f option).

Then I would like to improve my setup letting ReceiveFAX negociate T.38
with SPA3102.
The trouble is SPA3102, as I configured it, seems to refuse T.38
negociation as I'm reading lines like this in Asterisk logs:

   == Using UDPTL CoS mark 5
[2013-11-05 10:36:50] WARNING[3061][C-00000007]: res_fax.c:1698
receivefax_t38_init: channel 'SIP/myline-0000000e' refused to negotiate T.38

My question is:
Any hint on how to configure SPA3102 PSTN Line port so that it  would
accept to upgrade to T.38 ?





The SPA3102 only supports T.38 on the FXS port, the FXO port uses G711 for a fax session.


That explains why I couldn't find the option  Wink


Quote:

The Grandstream HT503 supports T.38 on both the FXO and FXS ports. 


I never tried this one.
How would you rate this product ?


Is it easy to auto-provision an HT503 over TFTP or HTTP ?

Is it easy to localize FXO/FXS setttings for non-US countries (those having played with a SPA3102 sure know what I'm thinking about) ?

Do it T.38 implementation works ok with Asterisk ?



Quote:

What problem do you have receiving a fax over G711?

I tried few calls in G711 fax mode and it worked OK but I rated T.38 as more reliable solution in the long run and in general.

Quote:

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





PostPosted: Wed Nov 20, 2013 4:17 am    Post subject: [asterisk-users] How to enable T.38 between SPA3102 PSTN Lin Reply with quote

On 8/11/2013 12:58 AM, Olivier wrote:
.
.
.
Quote:
I never tried this one.
How would you rate this product ?


I don't use it for faxing only for the purposes of testing its
capabilities receiving. I have progressively upgraded the firmware as it
has been released.

Quote:
Is it easy to auto-provision an HT503 over TFTP or HTTP ?

It has these options though I have not used auto-provisioning with it.

Grandstream have configuration tools and templates, see
http://www.grandstream.com/support/tools.

Quote:
Is it easy to localize FXO/FXS setttings for non-US countries (those
having played with a SPA3102 sure know what I'm thinking about) ?

In short, Yes!

Quote:
Do it T.38 implementation works ok with Asterisk ?

Seems to though I really have only performed basic testing receiving a
fax through it from the PSTN (FXO port).

Regards,

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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