Re: [asterisk-users] Can't block intrusion

[Larry Moore]

On 2/04/2020 6:35 AM, D'Arcy Cain wrote:

On 2020-04-01 16:28, Mark Boyce wrote:

On 1 Apr 2020, at 22:14, Greg Troxel mailto:g...@lexort.com>> wrote:

I think you need to use tcpdump and turn up firewall debugging.

sngrep is your friend …My bet is UDP vs TCP on firewall rules :-)

block drop in log quick on bge0 from  to any
block drop out log quick on bge0 from any to 

Am I misunderstanding pf?  I thought that that would block TCP, UDP,
ICMP and anything else trying to get through.

Since I started looking at this closer I did find that only some
connections have this problem.  Most get blocked as soon as the IP is
passed to the AUTOBLOCK table.


I suspect you have a good understanding of pf.

Have you included in your script running 'pfctl -k ' to kill 
any states that may exists after you update your  table?


In pf, like IP Filter, the last matching rule wins.

What can't be determined from the information provided is whether any 
connections that have been established from networks you have listed in 
the table , also appear in the  table.


Removing the 'quick' parameter from the rule for  will allow 
packets to fall through to the next rules. Alternatively, moving the 
'pass' rule to below your 'block' rules will allow any connections 
originating from networks listed in your  table and also exists 
in the  table, will be blocked.


Larry.

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Re: [asterisk-users] Can't block intrusion

[Larry Moore]

On 2/04/2020 5:39 AM, Larry Moore wrote:

On 2/04/2020 5:28 AM, Mark Boyce wrote:
On 1 Apr 2020, at 22:14, Greg Troxel <mailto:g...@lexort.com>> wrote:


I think you need to use tcpdump and turn up firewall debugging.


sngrep is your friend …My bet is UDP vs TCP on firewall rules :-)

Mark


Or the stateful entry still exists when the table entry is updated.

Does your script also issue a command to kill existing states from 
that host after it has updated the table, e.g.  pfctl -k 45.143.220.235


Larry.



Hmm, missed that in your original post. Could 'pfctl -K' be of help, I 
would suggest either removing 'quick' from your 'pass' rule or placing 
that line after the 'block' rules.


Larry.
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Re: [asterisk-users] Can't block intrusion

[Larry Moore]

On 2/04/2020 5:28 AM, Mark Boyce wrote:
On 1 Apr 2020, at 22:14, Greg Troxel > wrote:


I think you need to use tcpdump and turn up firewall debugging.


sngrep is your friend …My bet is UDP vs TCP on firewall rules :-)

Mark


Or the stateful entry still exists when the table entry is updated.

Does your script also issue a command to kill existing states from that 
host after it has updated the table, e.g.  pfctl -k 45.143.220.235


Larry.
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Re: [asterisk-users] Asterisk Fax Receive - how to get the remoteheader?

[Larry Moore]

Hi,

I haven't found anything definitive however I expect the TSI that is 
sent during initial fax call establishment is stored by the receiving 
terminal, see pages 28 & 29 of the English version of the document at 
https://www.itu.int/rec/T-REC-T.30-200509-I/en , I expect the header, 
which will include the TSI, is all part of the image (Tagline in 
HylaFAX) and not stored separately on the receiving terminal.


Cheers,

Larry.

On 18/12/2016 6:20 PM, Yves wrote:

Hi,

thanks for your answer. Unfortunately this is, what I already know. I 
was wondering, why it is possible to set ID and Header for an outgoing 
fax (which will then in turn
be inserted via asterisk on top of the transferred "image") , while it 
seems to not be possible to get the Header from a received fax (only 
the id), although it is present in the faxdocument.
The ID is also present in the faxdocument and there does a 
Faxopt(remotestationid) exist... so I thought, this info must be 
transferred not only binary within the "image", but
also within the "meta-data" / protocol-data of the fax (within the 
TSI) otherwise asterisk must do some kind of ocr to get the ID, 
what it definitely does not...


btw... when using sendfax, asterisk inserts the date, the id, the 
header and the pagenum on top of each faxpage... someone knows how to 
modify some settings like font, position, and so on?


thanks,
yves


Am 18.12.2016 um 00:02 schrieb Larry Moore:
The list of options available are listed here 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_FAXOPT


It doesn't appear that a received header is available unless it is 
written into the 'headerinfo' variable after it is received, I 
haven't checked for this.


From my days working with fax machines, the header could be inserted 
in the line the TSI is on or in the image being transmitted, if you 
receive a fax that has been sent to you with the latter set, then the 
'headerinfo' will not be of any use. Perhaps someone with more 
knowledge may be able to explain this better.


A quick Google search for 'fax header outside of tsi' will provide a 
list of manuals, here's one - 
http://manuals.konicaminolta.eu/bizhub-C554-C454-C364-C284-C224/EN/contents/sh3_378.html#qitem13 



Expand the line for


To specify the position of Header Position printed on a sent
fax ([Header Position])

Larry.

On 18/12/2016 4:30 AM, Yves wrote:

Hi,

I am using asterisk 11.8 in combination with spandsp to send and 
receive T38 Faxes. All works fine, but I do not know

how to get the remoteheader from the fax I receive.

When I send a fax, there are Faxopts to set the localstationid and 
the headerinfo, but for receiving, there seems to only exist

the Faxopts remotestationid

but for sure on any fax I receive there is a remoteheaderinfo 
besides the remotestationid... it is on the tiff-file, but I need this

info in a channel-variable...

Does anybody know how to get the remoteheaderinfo for a received fax?

thanks

yves












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Re: [asterisk-users] Asterisk Fax Receive - how to get the remoteheader?

[Larry Moore]

The list of options available are listed here 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_FAXOPT


It doesn't appear that a received header is available unless it is 
written into the 'headerinfo' variable after it is received, I haven't 
checked for this.


From my days working with fax machines, the header could be inserted in 
the line the TSI is on or in the image being transmitted, if you receive 
a fax that has been sent to you with the latter set, then the 
'headerinfo' will not be of any use. Perhaps someone with more knowledge 
may be able to explain this better.


A quick Google search for 'fax header outside of tsi' will provide a 
list of manuals, here's one - 
http://manuals.konicaminolta.eu/bizhub-C554-C454-C364-C284-C224/EN/contents/sh3_378.html#qitem13 



Expand the line for


   To specify the position of Header Position printed on a sent fax
   ([Header Position])

Larry.

On 18/12/2016 4:30 AM, Yves wrote:

Hi,

I am using asterisk 11.8 in combination with spandsp to send and 
receive T38 Faxes. All works fine, but I do not know

how to get the remoteheader from the fax I receive.

When I send a fax, there are Faxopts to set the localstationid and the 
headerinfo, but for receiving, there seems to only exist

the Faxopts remotestationid

but for sure on any fax I receive there is a remoteheaderinfo besides 
the remotestationid... it is on the tiff-file, but I need this

info in a channel-variable...

Does anybody know how to get the remoteheaderinfo for a received fax?

thanks

yves




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Re: [asterisk-users] iaxmodem errors.

[Larry Moore]

Hi,

If the 'fax show version' command doesn't work on your system it will 
also indicate you don't have the 'res_fax' module installed, thus you 
won't be able to take advantage of the T.38 gateway functionality, did 
you try the command?


You're indicating you are having problems with IAX registrations. It 
would seem that setting up at least 2 iaxmodem's should be done so you 
can test connectivity between them before attempting connections externally.


I'll assume you are only using iaxmodem on the IAX channels here is my 
'iax.conf' configuration with minor alterations:


   [general]
   iaxcompat=yes
   delayreject=yes
   language=en_AU
   disallow=all
   allow=alaw,ulaw
   jitterbuffer=yes
   forcejitterbuffer=yes
   ;maxjitterbuffer=1000
   maxjitterinterps=10
   autokill=yes
   shrinkcallerid=no

   [iaxmodem0]
   type=friend
   host=dynamic
   auth=md5
   secret=
   disallow=slin,slin16
   context=FAX-T30
   callerid="MODEM-1" <5001>
   tos=ef
   sendani=yes
   sendrpid=yes
   jitterbuffer=no
   requirecalltoken=no
   qualify=no
   qualifysmoothing=yes
   qualifyfreqok=6
   qualifyfreqnotok=1
   trunk=no
   deny=0.0.0.0/0.0.0.0
   permit=127.0.0.1/255.255.255.255

   [iaxmodem1]
   type=friend
   host=dynamic
   auth=md5
   secret=
   disallow=slin,slin16
   context=FAX-T30
   callerid="MODEM-2" <5002>
   tos=ef
   sendani=yes
   sendrpid=yes
   jitterbuffer=no
   requirecalltoken=no
   qualify=no
   qualifysmoothing=yes
   qualifyfreqok=6
   qualifyfreqnotok=1
   trunk=no
   deny=0.0.0.0/0.0.0.0
   permit=127.0.0.1/255.255.255.255

I'll include the ttyIAX configuration files in /etc/iaxmodem too, please 
note your 'owner' information may be different than what my system require.:


/etc/iaxmodem/ttyIAX0:

   device  /dev/ttyIAX0
   owner   uucp:dialer
   mode660
   port4570
   ;iax2debug
   refresh 60
   server  127.0.0.1
   peernameiaxmodem0
   secret  
   nojitterbuffer
   ;nodaemon
   codec   alaw,ulaw
   ;codec  slin,alaw,ulaw,g726aal2,gsm

/etc/iaxmodem/ttyIAX1:

   device  /dev/ttyIAX1
   owner   uucp:dialer
   mode660
   port4571
   ;iax2debug
   refresh 60
   server  127.0.0.1
   peernameiaxmodem1
   secret  
   nojitterbuffer
   ;nodaemon
   codec   alaw,ulaw
   ;codec  slin,alaw,ulaw,g726aal2,gsm

If you launch iaxmodem without specifying any configuration files it 
will run each one listed in /etc/iaxmodem or whatever the compiled 
location is.


You've not indicated what is utilising your IAX modems, if it is HylaFAX 
you will need to ensure the modem is configured appropriately and 
something is monitoring the device to accept a call.




On 16/11/2016 5:26 PM, john wrote:


Hi. the fax show version does not work since i am not using the digium 
modem.


the iax2 show peers is the command for me and the output is:

PBX*CLI> iax2 show peers
Name/UsernameHost Mask Port  Status  
Description
iaxmodem/iaxmod  127.0.0.1   (S)  255.255.255.255 4570  OK 
(1 ms)

1 iax2 peers [1 online, 0 offline, 0 unmonitored]
PBX*CLI>


the problem is that in logs i am getting errors and i do not know how 
to fix it.


root@PBX: /var/log/iaxmodem $ more ttyIAX0
[2016-11-16 09:08:12.483144] Registration failed.
[2016-11-16 09:13:05.118692] Terminating on signal 15...
[2016-11-16 09:21:49.181872] Registration failed.
[2016-11-16 09:22:30.731893] Terminating on signal 15...
[2016-11-16 09:22:30.759221] Registration failed.
[2016-11-16 09:25:11.014642] Registration failed.
root@PBX: /var/log/iaxmodem $



Any ideas?





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Re: [asterisk-users] iaxmodem errors.

[Larry Moore]

I suspect I followed a guide much like the one you have used including 
information found on voip-info - sorry, I can't seem to find any 
bookmarks of relevant information.


I spent an enormous amount of time getting it working and working very 
well, the real issue was getting T.38 working - I applied a patch to 
Asterisk version 1.8 to get the T.38 gateway functionality.


I would have started off my testing by confirming communications between 
two IAX modems, I presume you are using HylaFAX too.


Once the communications between the two IAX modems was working I 
progressed with testing sending and receiving faxes using G711A through 
my VoIP service and a modem attached to a PSTN service, suffice to say 
T.38 functionality was the key to getting reliable faxes working through 
VoIP at least when traversing the Internet, fortunately my VoIP provider 
facilitates T.38.


Using an SPA8800 on my network I tested sending and receiving faxes with 
a modem attached to the SPA8800, it worked in G711A and T.38.


I progressed to Asterisk 11 where the T.38 gateway functionality is 
better along with other improvements.


What is the output on your system for:

fax show version


Cheers,

Larry.

On 15/11/2016 8:09 PM, tux john wrote:
Hi. Since I am messing a lot with it without seeing the end of, may I 
ask if there is any solid guide for that please?

On 13/11/2016, 07:42 Larry Moore <lmo...@starwon.com.au> wrote:

Some additional information which may help you with your installation.

I have 4 IAX Modems named iaxmodem0 - iaxmodem3. I use iaxmodem3
for outbound fax transmissions.

I created a queue for the other 3 modems, here is my entry in
queues.conf:

[hylafax-iax]
strategy=linear
ringinuse=yes
autopause=no
retry=4
timeout=5
timeoutpriority=conf
reportholdtime=no
joinempty=strict
leavewhenempty=strict
musicclass=silence

member => IAX2/iaxmodem2
member => IAX2/iaxmodem1
member => IAX2/iaxmodem0

In case you are wondering about the 'musicclass' I have used, here
is the section from musiconhold.conf, the actual location of the
files may be elsewhere on your system:

[silence]
mode=files
directory=/usr/local/share/asterisk/silence
; ls /usr/local/share/asterisk/silence
; 10.gsm
;
; The file 10.gsm came from
/usr/local/share/asterisk/sounds/en/silence

I changed 'callbackextension' in my sip.conf for the trunk so that
it would go directly to the 'fax' extension in the dialplan i.e.
'callbackextension=fax'.

I've included the console output when an incoming fax is received:

  == Using SIP RTP TOS bits 184
-- Executing [fax@from-itsp:1] NoOp("SIP/itsp-0044",
"Fax Detected 2016-11-13 12:33:40 +0800") in new stack
-- Executing [fax@from-itsp:2] GotoIf("SIP/itsp-0044",
"0?3:8") in new stack
-- Goto (from-itsp,fax,8)
-- Executing [fax@from-itsp:8] NoOp("SIP/itsp-0044",
"Finish if_from-itsp_237") in new stack
-- Executing [fax@from-itsp:9] GotoIf("SIP/itsp-0044",
"0?10:13") in new stack
-- Goto (from-itsp,fax,13)
-- Executing [fax@from-itsp:13] NoOp("SIP/itsp-0044",
"Finish if_from-itsp_238") in new stack
-- Executing [fax@from-itsp:14] Set("SIP/itsp-0044",
"FAXOPT(gateway)=yes") in new stack
-- Executing [fax@from-itsp:15] Queue("SIP/itsp-0044",
"hylafax-iax,dRt,,,15") in new stack
-- Started music on hold, class 'silence', on
SIP/itsp-0044
-- Call accepted by 127.0.0.1 (format alaw)
-- Format for call is (alaw)
-- IAX2/iaxmodem2-3086 is ringing
-- Stopped music on hold on SIP/itsp-0044
-- IAX2/iaxmodem2-3086 answered SIP/itsp-0044
   > 0x89bac000 -- Probation passed - setting RTP source
address to :18998
  == Using UDPTL TOS bits 184
-- Executing [h@from-itsp:1] GotoIf("SIP/itsp-0044",
"0?2:3") in new stack
-- Goto (from-itsp,h,3)
-- Executing [h@from-itsp:3] NoOp("SIP/itsp-0044",
"Finish if_from-itsp_239") in new stack
-- Executing [h@from-itsp:4] NoOp("SIP/itsp-0044",
"Call/Fax Ended 2016-11-13 12:36:41 +0800") in new stack
-- Hungup 'IAX2/iaxmodem2-3086'
  == Spawn extension (from-itsp, fax, 15) exited non-zero on
'SIP/itsp-0044'

I'm sure you've already checked and confirmed you have 'alaw' and
'ulaw' codecs permitted in your IAX 

Re: [asterisk-users] iaxmodem errors.

[Larry Moore]

X${FAXRXFILE}" != "X" )
{
_rxfax();
}
NoOp(Call/Fax Ended ${STRFTIME(,,%F %T %z)});
};
   };

   macro fax-receive( fax-number, header-info, sender, recipient ) {
   /*
${ARG1} is Receiving Station Fax Number
${ARG2} is Fax Header Information
${ARG3} is Fax Sender E-mail Address
${ARG4} is Fax Recipient E-mail Address
   */
NoOp( FAX RECEIVE );
Set(FAXOPT(localstationid)=${LOCAL(fax-number)});
Set(FAXOPT(headerinfo)=${LOCAL(header-info)});
Set(FROMADDR=${LOCAL(sender)});
Set(TOADDR=${LOCAL(recipient)});
NoOp( SETTING FAXOPT );
NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)});
NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)});
NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)});
Set(RXSTART=${EPOCH});
Set(FAXRXPATH=/var/spool/asterisk/fax/received);
Set(FAXRXFILE=fax-${CALLERID(number)}-${UNIQUEID});
        NoOp( RECEIVING FAX : ${FAXRXFILE} );
ReceiveFAX(${FAXRXPATH}/${FAXRXFILE}.tif,f);
NoOp( Subroutine Return );
return;
};

Cheers,

Larry.


On 13/11/2016 8:07 AM, Larry Moore wrote:
Is your network/firewall configuration permitting the ports for UDPTL, 
runn the command:  udptl show config


UDPTL Global options

udptlstart:  4000
udptlend:4999
udptlfecentries: 3
udptlfecspan:3
use_even_ports:  No
udptlchecksums: Yes

In your sip configuration for your 'mytrunk' peer have you set 
applicable options e.g.:


t38pt_udptl=yes,redundancy,maxdatagram=400

In your extensions.conf you could and probably should set the 
following option prior to dialing the IAX channel, this is to enable 
the T.38 gateway feature of Asterisk 11:


Set(FAXOPT(gateway)=yes)

I have it working in my installation however I have incoming voice 
calls too hence I use 'faxdetect' to direct the call to the 'fax' 
extension.


Cheers,

Larry.

On 12/11/2016 5:24 AM, tux john wrote:
hi. i am using asterisk 11.24.1 in my raspberry. i do have a sip 
trunk with a provider with g711a. I am trying to setup a fax server 
by following the guide in 
http://the-asterisk-book.com/1.6/faxserver.html.

i do live in Greece and the number is 00302112152130
the problem is that i am getting the following error and i am stuck:
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Executing [00302112152130@fax-in:1] 
Dial("SIP/mytrunk-0001", "IAX2/iaxmodem") in new stack

-- Called IAX2/iaxmodem
-- Hungup 'IAX2/iaxmodem-3818'
  == Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/mytrunk-0001' status is 
'CHANUNAVAIL'

RasPBX*CLI>
the extensions.conf has
[fax-in]
exten => 00302112152130,1,Dial(IAX2/iaxmodem)
any ideas, please?








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Re: [asterisk-users] iaxmodem errors.

[Larry Moore]

Is your network/firewall configuration permitting the ports for UDPTL, 
runn the command:  udptl show config


   UDPTL Global options
   
   udptlstart:  4000
   udptlend:4999
   udptlfecentries: 3
   udptlfecspan:3
   use_even_ports:  No
   udptlchecksums: Yes

In your sip configuration for your 'mytrunk' peer have you set 
applicable options e.g.:


   t38pt_udptl=yes,redundancy,maxdatagram=400

In your extensions.conf you could and probably should set the following 
option prior to dialing the IAX channel, this is to enable the T.38 
gateway feature of Asterisk 11:


   Set(FAXOPT(gateway)=yes)

I have it working in my installation however I have incoming voice calls 
too hence I use 'faxdetect' to direct the call to the 'fax' extension.


Cheers,

Larry.

On 12/11/2016 5:24 AM, tux john wrote:
hi. i am using asterisk 11.24.1 in my raspberry. i do have a sip trunk 
with a provider with g711a. I am trying to setup a fax server by 
following the guide in http://the-asterisk-book.com/1.6/faxserver.html.

i do live in Greece and the number is 00302112152130
the problem is that i am getting the following error and i am stuck:
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Executing [00302112152130@fax-in:1] 
Dial("SIP/mytrunk-0001", "IAX2/iaxmodem") in new stack

-- Called IAX2/iaxmodem
-- Hungup 'IAX2/iaxmodem-3818'
  == Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/mytrunk-0001' status is 
'CHANUNAVAIL'

RasPBX*CLI>
the extensions.conf has
[fax-in]
exten => 00302112152130,1,Dial(IAX2/iaxmodem)
any ideas, please?




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Re: [asterisk-users] Change Asterisk MulticastRTP codec

[Larry Moore]

On my Asterisk 11 system I have the following in extensions.ael for 
chan_sip.


8001=> {
Set(SIP_CODEC=alaw);
//Dial(MulticastRTP/linksys/224.168.168.168:34567/224.168.168.168:6061);
Page(MulticastRTP/linksys/224.168.168.168:34567/224.168.168.168:6061,q,5);
Hangup();
};


I believe in your case you need to set PJSIP_MEDIA_OFFER(ulaw) in a 
pre-dial handler prior to making the call.


See 
https://wiki.asterisk.org/wiki/display/AST/Pre-dial+handlers+Specification.





On 1/10/2015 1:51 AM, Matthew Murphy wrote:

Greetings everyone,

I was wondering if there was a way to change the codec that Asterisk 
uses when streaming via MulticastRTP. Or perhaps a way to transcode 
the multicast stream.


In the CLI, when I have a multicast stream in progress, I am typing 
'core show channel MulticastRTP/0x7f7' to get lots of helpful 
information.


I have noticed that when I do a MULTICAST page and* send data from 
MP3Player*, I get no sound on my speakers and get the following from 
'core show channel PJSIP/xxx':


NativeFormats: (slin)
WriteFormat: slin
ReadFormat: slin
*WriteTranscode: No *
*ReadTranscode: No *

I have noticed that when I do a UNICAST page and* send data from 
MP3Player*, everything works flawlessly and I get the following from 
'core show channel MulticastRTP':


NativeFormats: (ulaw)
WriteFormat: slin
ReadFormat: slin
*WriteTranscode: Yes (slin@8000)->(ulaw@8000)*
*ReadTranscode: Yes (ulaw@8000)->(slin@8000)*


The *only* thing that is changing is the following line in my 
extensions.conf file:


; For Multicast Paging
same => n(next),Page(MulticastRTP/basic/239.1.1.2:6061,q)

; For Unicast Paging
same => 
n(next),Page(PJSIP/107/108,ib(pjsip-auto-answer-header^addheader^1)|p})



Is there any way to get the MP3Player stream to transcode (as it does 
on the UNICAST stream) when I try to MULTICAST?


Thanks for the help,

--Matt




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Re: [asterisk-users] Respond to an out of call SIP MESSAGE

[Larry Moore]

Could it be in the [general] section you should have;

accept_outofcall_message=yes

Your line appears to be missing the 'p' in accept and an extraneous 's' 
in message.


Larry.

On 21/09/2015 2:48 PM, Emil Ohlsson wrote:

Hi,

I'm having trouble configuring Asterisk to respond to an incoming out of call 
SIP MESSAGE. The transport protocol is TLS and the Asterisk version is 10 (it's 
old, but I'm kind of stuck with it at the moment). Currently I have roughly the 
following configuration and handling:

sip.conf:

[general]
accet_outofcall_messages=yes
outofcall_message_context=sip-im

and extensions.conf

[sip-im]
exten _X!, 1, NoOp(Got message)
exten _X!, n, Answer()
exten _X!, n, Agi(agi://localhost/messagehandler.agi?...)
exten _X!, n, SendText(Message received)

I can see in the log from Asterisk that the script in the sip-im context is 
running, but there is no message sent. I have followed the code in the call, 
and it seems like there is no channel registered with the SendText application. 
Is there some other approach that I could use to send a SIP MESSAGE back to the 
client? Does the client need to register for this to work?

BR,
Emil




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Re: [asterisk-users] PJSIP T.38 issues

[Larry Moore]

On 29/07/2015 12:13 PM, Jean-Denis Girard wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Thanks for your reply Larry.

Le 27/07/2015 01:22, Larry Moore a écrit :

I think the 488 Not acceptable here is occurring because the channel
connecting through is not T.38 capable, that will be the IAX channel
from iaxmomdem.


This is what T38gateway is supposed to do. And I'm very happy to report
that after one more day of efforts, I have everything working as I wante
d.




Pleased you have managed to get it working.

Was it enabling alaw/ulaw which helped or did you need to use another 
method to route the IAX channel through PJSIP or some other 
configuration setting such as 'faxdetect' which may have been disabled?


Cheers,

Larry.


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Re: [asterisk-users] PJSIP T.38 issues

[Larry Moore]

I think the 488 Not acceptable here is occurring because the channel 
connecting through is not T.38 capable, that will be the IAX channel 
from iaxmomdem.


I've not used PJSIP so cannot offer any advice regarding it however you 
may try to make iaxmodem connect through another context using either 
SIP or IAX (experiment with both bu most probably IAX) in an attempt to 
prevent the rejection of the T.38 establishment forcing the call to 
terminate. What I seem to recall when experimenting with SIP as the 
trunk, have UDPTL disabled i.e. t38pt_udptl=no, this would also induce 
488 Not acceptable here.


Looking at a legacy configuration where I tested iaxmodem 
(context=faxgateway-iax) going through Asterisk 1.2 which then forwarded 
the request to Asterisk 11 (context=FAX-T30) where it then went out 
through the trunk with Fax Gateway enabled.


In short;

Asterisk 1.2
IAX Modem in context faxgateway-iax, could change to faxgateway-sip.

[faxgateway-iax]
; Incoming calls from iaxmodem to Asterisk are directed to
; other Asterisk server.
exten = _XX.,1,Dial(IAX2/faxgw-iax@faxgw-iax/${EXTEN},55,t)
exten = _XX.,n,Wait(1)
exten = _XX.,n,Hangup
;

[faxgateway-sip]
; Incoming calls from iaxmodem to Asterisk are directed to
; other Asterisk server.
exten = _XX.,1,Dial(SIP/${EXTEN}@faxgw-sip,55,t)
exten = _XX.,n,Wait(1)
exten = _XX.,n,Hangup
;


Asterisk 11
IAX user faxgw-iax is in context FAX-T30

extensions.ael on Asterisk 11 contains

context FAX-T30 {
snip
_ = {
//  Set(FAXOPT(t38gateway)=yes);
Dial(SIP/${EXTEN}@itsp-fax,55);
Hangup();
};
snip
};


One other note, enable alaw  ulaw in iaxmomdem and your iax peer 
configuration in Asterisk, just to be sure!


I know this isn't specific to your case but maybe you can make something 
from this that helps.


Please note, I don't have the old set up to test so I can't be certain 
of the above configurations.


Cheers,

Larry.

On 27/07/2015 11:15 AM, Jean-Denis Girard wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi list,

2 weks ago I asked questions about PJSIP and T.38 but got no replies. I
upgraded Asterisk to git as of yesterday (309dd2a), and I'm still having
the same issues.

In the trace below, I'm sending a fax from Hylafax server through
iaxmodem on Asterisk-13 (tiare) to a second Asterisk (11.18.0 t0gw)
connected to the PSTN via ISDN; the call is to my test fax machine,
connected to the PSTN. chan_pjsip is used on Asterisk-13, and chan_sip
is used on Asterisk-11.

This how endpoint t0gw (Asterisk-11) is configured on tiare (Asterisk-13
):
tiare*CLI pjsip show endpoint t0gw
...
t38_udptl : true
t38_udptl_ec : fec
t38_udptl_ipv6 : false
t38_udptl_maxdatagram : 400
t38_udptl_nat : false
...

Could someone explain why I'm getting Not acceptable below?

-- Accepting AUTHENTICATED call from 127.0.0.1:4570:
 -- requested format = slin,
 -- requested prefs = (),
 -- actual format = slin,
 -- host prefs = (slin),
 -- priority = mine
 -- Executing [40ZZ@fax-sortant:1] NoOp(IAX2/iaxmodem0-7838, 
calls 40ZZ (local)) in new stack
 -- Executing [40ZZ@fax-sortant:2] Set(IAX2/iaxmodem0-7838,
FAXOPT(gateway)=yes) in new stack
 -- Executing [40ZZ@fax-sortant:3] Dial(IAX2/iaxmodem0-7838,
PJSIP/40ZZ@t0gw) in new stack
 -- Called PJSIP/40ZZ@t0gw
--- Transmitting SIP request (936 bytes) to UDP:192.168.0.10:5060 ---
INVITE sip:40zzz...@gw.sysnux.pf SIP/2.0
Via: SIP/2.0/UDP
192.168.0.200:5060;rport;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e3
8e5f1
From: SysNux
sip:+68940XX@192.168.0.200;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
To: sip:40zzz...@gw.sysnux.pf
Contact: sip:63035284-ad7d-484f-8e54-f5ea54f39104@192.168.0.200:5060
Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
CSeq: 31693 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk GPL PBX
Content-Type: application/sdp
Content-Length:   238

v=0
o=- 1710591484 1710591484 IN IP4 192.168.0.200
s=Asterisk
c=IN IP4 192.168.0.200
t=0 0
m=audio 8834 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

--- Received SIP response (585 bytes) from UDP:192.168.0.10:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.200:5060;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e38e5f1;
received=192.168.0.200;rport=5060
From: SysNux
sip:+68940XX@192.168.0.200;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
To: sip:40zzz...@gw.sysnux.pf
Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
CSeq: 31693 INVITE
Server: Asterisk PBX 11.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: 

Re: [asterisk-users] Dialplan for receiving faxes on Asterisk

[Larry Moore]

On 30/01/2015 1:25 PM, Simon Humbert wrote:

  Hi all,

It looks like people commonly use this kind of dialplan when receiving
faxes on Asterisk, with a jump to extension fax during the Wait() if a
fax tone is detected:

[start-here]
exten = _X.,1,Answer()
exten = _X.,n,Wait(n)
exten = _X.,n,...do stuff...
exten = _X.,n,Hangup()

exten = fax,1,Goto(fax-rx,receive,1)

[fax-rx]
exten = receive,1,...
exten = receive,n,...do stuff...
exten = receive,n,ReceiveFAX()

This is well suited in case Asterisk needs to receive both voice and fax
calls. But what if Asterisk is only used to receive fax calls, can we
start directly at the fax-rx context? I've heard that it's better to
wait a few seconds before calling ReceiveFAX(), is it still necessary in
case we don't actually need fax detection?



If you don't have the need to detect the fax tone then I don't see any 
need to wait.


You should disable the 'faxdetect' option in your peer otherwise it may 
attempt to redirect to the 'fax' extension upon detecting the fax 
signalling.


Assuming you are using a SIP trunk to accept the call you could use in 
your sip.conf peer something like;


context=fax-rx
disallow=all
allow=alaw,ulaw
jbenable=no
faxdetect=no
directmedia=no
callbackextension=receive
t38pt_usertpsource=yes
encryption=no

Note, in this example I am using 'callbackextension' instead of 
'register =', refer to the default sip.conf for further information.


Larry.

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Re: [asterisk-users] T.38 not working - help needed with log interpretation

[Larry Moore]

Just a thought regarding testing.

Create a suitable TIFF file with more than 30 seconds worth of data and 
send it from Asterisk using SendFAX() to convince yourself whether 
Asterisk will work with your ITSP, you may still need to enable session 
timers.


Have you considered setting up an extension on your Asterisk server 
which will receive the fax e.g. using ReceiveFAX() and see if your 
connection problem persists?


Larry.


On 15/12/2014 5:10 AM, Larry Moore wrote:



On 14/12/2014 7:24 PM, Recursive wrote:


On 11.12.2014 11:53, Larry Moore wrote:

You may very well find getting T.38 working in your environment in a
way you would like will consume a large amount of your time, you will
also find yourself doing a lot of research. What you should have
found out by now (or perhaps deduced) is that the T.38 is a standard
that is varied thus one cannot be assured a T.38 solution will always
work.


Agreed. But firstly, I really need a working fax, and secondly, I am
just trying to make it work with one specific provider (I have
identified two providers in Germany which could be a possibility and
have signed up a full account with both of them for testing purposes
(no problem since the fees are small and the cancellation period is
short in both cases), and as soon as one of these works like expected
I'll cancel the contract with the other one). So I don't need to have
a general solution which works with every provider around the world.


exten = _00., 1, NoOp()
same = n, Answer()
same = n, Progress()
same = n, Set(FAXOPT(gateway)=yes)
same = n, Dial(SIP/${EXTEN}@27XgY8YwfI2S9NAg)
same = n, Hangup()


One may assume this is your dialplan for the outgoing connection with
which you want T.38 to be supported.


You are right.


To obtain better assistance you will need to include information such
as what the local T.38 endpoint is and how it connects to your
system. If it is in fact a T.38 capable endpoint then you should
setting FAXOPT(gateway) to no. having Answer() Progress() for an
outgoing T.38 connection doesn't seem to make sense to me!


The endpoint is T.38 capable. I have configured FAXOPT(gateway)=yes
because I have read that the gateway automatically goes out of the way
if Asterisk determines that the two endpoints which should be
connected use the same protocols and parameters, and otherwise
translates between the endpoints (see
https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway, section
Using T.38 gateway mode).

Furthermore, T.38 passthrough never worked for me. My initial tries
were with Asterisk 1.8 which is included in Debian wheezy (and does
not have the gateway capability anyway), but whatever I did there, no
T.38 INVITE was accepted by Asterisk (this was some weeks ago, so I
don't remember the details). I then downloaded, compiled and installed
Asterisk 13.0.1 and tried T.38 passthrough, but to no avail as well.
When I added FAXOPT(gateway)=yes, I saw correct T.38 INVITEs for the
first time.



See comment regarding 'canreinvite'.


Regarding Progress(), I am not sure if I need this. I think one day I
could solve a certain problem by using it, but I really have done a
million of tries and don't remember every one.

Regarding Answer(), I think I need this. Without Answer(), no T.38
would be used in many cases; it seems that switching from initial
G.711 to T.38 is done in-band, and that couldn't be done without the
Answer() in the dialplan, could it?



Your ITSP should be sending the T.38 invite to Asterisk when they detect
the fax tones from the answered call (callee), it would appear you are
_forcing_ the T.38 session by using the Answer() function, you are then
relying on the activity detection period of the FAXOPT(gateway) function
for the call to be established.

Asterisk 1.8 with the T.38 Gateway patch (not the one by Niccolò Belli)
sends a T.38 invite to the ITSP when the fax tones are detected from the
callee, the T.38 Gateway implementation in Asterisk 10 and Niccolò's
back-port for Asterisk 1.8.11 does not behave this way.

The real issue here is having current versions of Asterisk  T.38
working as expected all the way through.


You should also include information relating to your SIP
configuration (with appropriate obfuscations) for the connection to
peer 27XgY8YwfI2S9NAg as well as what T.38 options you have set in
the general section of sip.conf.


You are right. I will now provide every detail and the logs. By the
way, I have switched back to chan_sip today for the purpose of testing
again and generating logs.

General remarks:

- Asterisk runs on the IP address 192.168.20.48.
- The hostname spock-asterisk.home.omeganet.de resolves to that IP
address.
- The fax software is Tobit David which is T.38 capable. I can't use
another fax software; an extensive explanation for that would be
off-topic here, but if somebody is interested, I will send respective
information via PM.
- The fax software runs on another machine than Asterisk.
- The fax software runs on the IP

Re: [asterisk-users] T.38 not working - help needed with log interpretation

[Larry Moore]

On 11/12/2014 4:52 PM, Recursive wrote:

Hello,

at first, thanks for helping!

In the meantime, I have done a lot of research and trial and error, and I could solve 
that specific problem. Obviously, the dialplan application Answer was playing 
a key role here. My original dialplan snippet (which produced that problem) was:



You may very well find getting T.38 working in your environment in a way 
you would like will consume a large amount of your time, you will also 
find yourself doing a lot of research. What you should have found out by 
now (or perhaps deduced) is that the T.38 is a standard that is varied 
thus one cannot be assured a T.38 solution will always work.



exten =  _00., 1, NoOp()
   same =  n, Set(FAXOPT(gateway)=yes)
   same =  n, Dial(SIP/${EXTEN}@27XgY8YwfI2S9NAg)
   same =  n, Hangup()

The problem vanished when I changed that to:

exten =  _00., 1, NoOp()
   same =  n, Answer()
   same =  n, Progress()
   same =  n, Set(FAXOPT(gateway)=yes)
   same =  n, Dial(SIP/${EXTEN}@27XgY8YwfI2S9NAg)
   same =  n, Hangup()



One may assume this is your dialplan for the outgoing connection with 
which you want T.38 to be supported.


To obtain better assistance you will need to include information such as 
what the local T.38 endpoint is and how it connects to your system. If 
it is in fact a T.38 capable endpoint then you should setting 
FAXOPT(gateway) to no. having Answer()  Progress() for an outgoing T.38 
connection doesn't seem to make sense to me!


You should also include information relating to your SIP configuration 
(with appropriate obfuscations) for the connection to peer 
27XgY8YwfI2S9NAg as well as what T.38 options you have set in the 
general section of sip.conf.


Larry.

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Re: [asterisk-users] SPA504G auto answer

[Larry Moore]

On 23/10/2014 4:57 PM, Larry Moore wrote:


On 23/10/2014 5:43 AM, Leandro Dardini wrote:

Hello,
I am struggling to have a SPA504G to auto answer (for intercom/paging).
I have tried the following SIP headers (not all together), but without
luck:

SIPAddHeader(Call-Info:\;answer-after=0);
SIPAddHeader(Call-Info: answer-after=0);
SIPAddHeader(Alert-Info: info=intercom);
SIPAddHeader(Alert-Info: Ring Answer);
SIPAddHeader(P-Auto-Answer: normal);



What does your dialplan look like that makes the paging call?

Listing from my Asterisk:

'8000' = 1. Set(SIP_CODEC=alaw)
2. Dial(MulticastRTP/linksys/224.168.168.168:45678/224.168.168.168:6061)
3. Hangup()


Using the Web Interface on you SPA501, log in as admin and select the
advanced view. Select the SIP tab, down the bottom of the page there is
a section headed 'Linksys Key System Parameters'.

You will want settings much like

Linksys Key System: yes
Multicast Address: 224.168.168.168:6061
Key System Auto Discovery: no
Key System IP Address: leave blank
Force LAN Codec: 711a may be set to none, G711a or G711u


For the benefit of others I encountered a situation where I was getting 
one-way audio in a call regardless of it not being a paging call, this 
was because the negotiated codecs for the call was one other than the 
one selected in the 'Force LAN Codec:' setting.


It would appear setting the 'Force LAN Codec:' to either G711u or G711a 
_always_ enforces the phone to use this codec for its Encoder regardless 
of what is negotiated in SIP.


My advice, leave the 'Force LAN Codec:' setting at its default value 
which is 'none'.


Larry.


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Re: [asterisk-users] OT - Is T.38 possible on SPA8800 FXO port ?

[Larry Moore]

On 17/11/2014 6:42 PM, Olivier wrote:

Hello,

If I'm not mistaken, it is not possible to get T.38 on a SPA3102 FXO
port (it is possible with the FXS port).
Do you know, by experience preferably, if this is possible with an
SPA8800 FXO port ?




Perhaps page 50 (Using a Fax Machine) of this document 
http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf 
will answer your question.


There is also another document you may want to reference.

https://supportforums.cisco.com/sites/default/files/legacy/8/1/0/42018-SPA8800_Asterisk_101909.pdf

In your sip.conf configuration you will need;

canreinvite=no

For newer version of Asterisk I expect you would set;

directmedia=no


You may want to change the SPA8800 option 'FAX T38 Redundancy' to the 
value you have set for redundancy in udptl.conf or 3 if using the 
default settings.


It's been a while since I played with my SPA8800, with the test I 
performed it did work, that was inbound and outbound calls.


Larry.

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Re: [asterisk-users] OT - Is T.38 possible on SPA8800 FXO port ?

[Larry Moore]

On 17/11/2014 7:34 PM, Larry Moore wrote:



On 17/11/2014 6:42 PM, Olivier wrote:

Hello,

If I'm not mistaken, it is not possible to get T.38 on a SPA3102 FXO
port (it is possible with the FXS port).
Do you know, by experience preferably, if this is possible with an
SPA8800 FXO port ?






Reading your question again, no the FXO port will not perform T.38, you 
are limited to G711 pass-through, from memory my SPA8800 was on the LAN 
the Asterisk PBX was attached and faxes were pretty good though I would 
have preferred to see T.38.


Grandstream ATA's support T.38 on the FXO port.

Larry.

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Re: [asterisk-users] OT - Is T.38 possible on SPA8800 FXO port ?

[Larry Moore]

On 17/11/2014 9:24 PM, Olivier wrote:

2014-11-17 12:55 GMT+01:00 Larry Moorelmo...@omninet.net.au:



On 17/11/2014 7:34 PM, Larry Moore wrote:




On 17/11/2014 6:42 PM, Olivier wrote:


Hello,

If I'm not mistaken, it is not possible to get T.38 on a SPA3102 FXO
port (it is possible with the FXS port).
Do you know, by experience preferably, if this is possible with an
SPA8800 FXO port ?






Reading your question again, no the FXO port will not perform T.38, you are
limited to G711 pass-through, from memory my SPA8800 was on the LAN the
Asterisk PBX was attached and faxes were pretty good though I would have
preferred to see T.38.


Thanks for sharing this here:
while it's not always easy to discover what a product can do, it's
harder to know what it CANNOT do.

Thanks


Grandstream ATA's support T.38 on the FXO port.


I had a lot of trouble to localize SPA3102 settings for Caller ID
presentation, for instance.



I was fortunate that there is plenty of information for configuring an 
SPA3000  SPA3102 for my country.


Perhaps page 53 of this document 
http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf 
may help, the other parameter I needed to adjust to receive Caller ID 
was the 'PSTN Answer Delay' setting.


I can't recall any great pain setting up an HT-503, I suspect I based 
settings upon an SPA3102/SPA8800, the T.38 faxing on the FXO was 
successful after a Firmware Upgrade, it was a couple of years ago when I 
was setting this up.


Larry.

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Re: [asterisk-users] SPA504G auto answer

[Larry Moore]

On 23/10/2014 4:57 PM, Larry Moore wrote:


On 23/10/2014 5:43 AM, Leandro Dardini wrote:

Hello,
I am struggling to have a SPA504G to auto answer (for intercom/paging).
I have tried the following SIP headers (not all together), but without
luck:

SIPAddHeader(Call-Info:\;answer-after=0);
SIPAddHeader(Call-Info: answer-after=0);
SIPAddHeader(Alert-Info: info=intercom);
SIPAddHeader(Alert-Info: Ring Answer);
SIPAddHeader(P-Auto-Answer: normal);



What does your dialplan look like that makes the paging call?

Listing from my Asterisk:

'8000' = 1. Set(SIP_CODEC=alaw)
2. Dial(MulticastRTP/linksys/224.168.168.168:45678/224.168.168.168:6061)
3. Hangup()




In the process of setting up another system, there is an additional 
requirement for the multicast paging to work.


Asterisk will need to know where to route the multicast traffic, on your 
Asterisk system, check your routing table and see if there is a route to 
the multicast address through the interface which connects to your 
phones. If not, create a routing entry, in this case to 224.168.168.168 
through the desired interface.



Larry.

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Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?

[Larry Moore]

On 24/10/2014 12:49 AM, Tim Nelson wrote:

- Original Message -



On 23/10/2014 10:07 PM, Larry Moore wrote:



On 22/10/2014 11:23 AM, Tim Nelson wrote:

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.

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Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?

[Larry Moore]

On 25/10/2014 11:43 PM, Larry Moore wrote:



On 24/10/2014 12:47 AM, Tim Nelson wrote:

- Original Message -



On 22/10/2014 11:23 AM, Tim Nelson wrote:

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:
  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-000b is making progress passing it to
IAX2/faxgw-iax-1210
-- SIP/itsp-fax-000b is making progress passing it to
IAX2/faxgw-iax-1210
== Using SIP RTP TOS bits 184
-- SIP/itsp-fax-000b 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-000b [1] Sending T.38 Params
Peer Is IAX2/faxgw-iax-1210 [0]
-- Request on IAX2/faxgw-iax-1210 [0] Storing I: SIP/itsp-fax-000b [1]
== Using UDPTL TOS bits 184
-- Negotiated on SIP/itsp-fax-000b [4] Ignoring I:
IAX2/faxgw-iax-1210 [0]
-- T.38 Gateway starting for chan SIP/itsp-fax-000b 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 01210/4 00010/5
0ms -0001ms ms 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-000 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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Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?

[Larry Moore]

On 24/10/2014 12:47 AM, Tim Nelson wrote:

- Original Message -



On 22/10/2014 11:23 AM, Tim Nelson wrote:

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:
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-000b is making progress passing it to 
IAX2/faxgw-iax-1210
-- SIP/itsp-fax-000b is making progress passing it to 
IAX2/faxgw-iax-1210

  == Using SIP RTP TOS bits 184
-- SIP/itsp-fax-000b 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-000b [1] Sending T.38 
Params Peer Is IAX2/faxgw-iax-1210 [0]
-- Request on IAX2/faxgw-iax-1210 [0] Storing I: 
SIP/itsp-fax-000b [1]

  == Using UDPTL TOS bits 184
-- Negotiated on SIP/itsp-fax-000b [4] Ignoring I: 
IAX2/faxgw-iax-1210 [0]
-- T.38 Gateway starting for chan SIP/itsp-fax-000b and peer 
IAX2/faxgw-iax-1210


pbx*CLI iax2 show channels
Channel   Peer UsernameID (Lo/Rem)  Seq 
(Tx/Rx)  Lag  Jitter  JitBuf  Format  FirstMsgLastMsg
IAX2/faxgw-iax-1210   192.168.54.18faxgw-iax   01210/4 
00010/5  0ms  -0001ms  ms  alawRx: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-000 Spandsp1  T.38  receiveActive 
(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'

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Re: [asterisk-users] SPA504G auto answer

[Larry Moore]

On 23/10/2014 5:43 AM, Leandro Dardini wrote:

Hello,
I am struggling to have a SPA504G to auto answer (for intercom/paging).
I have tried the following SIP headers (not all together), but without
luck:

SIPAddHeader(Call-Info:\;answer-after=0);
SIPAddHeader(Call-Info: answer-after=0);
SIPAddHeader(Alert-Info: info=intercom);
SIPAddHeader(Alert-Info: Ring Answer);
SIPAddHeader(P-Auto-Answer: normal);



What does your dialplan look like that makes the paging call?

Listing from my Asterisk:

  '8000' = 1. Set(SIP_CODEC=alaw)
2. 
Dial(MulticastRTP/linksys/224.168.168.168:45678/224.168.168.168:6061)

3. Hangup()


Using the Web Interface on you SPA501, log in as admin and select the 
advanced view. Select the SIP tab, down the bottom of the page there is 
a section headed 'Linksys Key System Parameters'.


You will want settings much like

Linksys Key System:  yes
Multicast Address:   224.168.168.168:6061
Key System Auto Discovery:  no
Key System IP Address:   leave blank
Force LAN Codec: 711a may be set to none, G711a or G711u
Auto Ans GrPage On Active Call:  no

Select the User tab and check

Auto Answer Page:  yes

If you have it all configured much like I have listed hear and it still 
doesn't work then you need to check the firewall configuration on your 
Asterisk  system and ensure it is allowing outbound Multicast traffic.


Larry.

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Re: [asterisk-users] SPA504G auto answer

[Larry Moore]

On 23/10/2014 4:57 PM, Larry Moore wrote:


On 23/10/2014 5:43 AM, Leandro Dardini wrote:

Hello,
I am struggling to have a SPA504G to auto answer (for intercom/paging).
I have tried the following SIP headers (not all together), but without
luck:

SIPAddHeader(Call-Info:\;answer-after=0);
SIPAddHeader(Call-Info: answer-after=0);
SIPAddHeader(Alert-Info: info=intercom);
SIPAddHeader(Alert-Info: Ring Answer);
SIPAddHeader(P-Auto-Answer: normal);



What does your dialplan look like that makes the paging call?

Listing from my Asterisk:

'8000' = 1. Set(SIP_CODEC=alaw)
2. Dial(MulticastRTP/linksys/224.168.168.168:45678/224.168.168.168:6061)
3. Hangup()


Using the Web Interface on you SPA501, log in as admin and select the
advanced view. Select the SIP tab, down the bottom of the page there is
a section headed 'Linksys Key System Parameters'.

You will want settings much like

Linksys Key System: yes
Multicast Address: 224.168.168.168:6061
Key System Auto Discovery: no
Key System IP Address: leave blank
Force LAN Codec: 711a may be set to none, G711a or G711u
Auto Ans GrPage On Active Call: no

Select the User tab and check

Auto Answer Page: yes

If you have it all configured much like I have listed hear and it still
doesn't work then you need to check the firewall configuration on your
Asterisk system and ensure it is allowing outbound Multicast traffic.

Larry.




Hmm, my SPA525G doesn't auto-answer a page however my SPA92X do.

The above is for paging, I use a macro to perform an intercom, here is 
what I have in my extensions.ael.


context {

_8XXX = {
sip_intercom(${EXTEN:1});
};

};


macro sip_intercom( extension ) {
ChanIsAvail(SIP/${LOCAL(extension)},s);
NoOp( Status : ${AVAILSTATUS} );
switch(${AVAILSTATUS}) {
case 1:
Set(TIMEOUT(absolute)=1920);
SIPAddHeader(Alert-Info: Ring Answer);
SIPAddHeader(Alert-Info: Info=Alert-Autoanswer);
SIPAddHeader(Alert-Info: info=ringAutoAnswer);
SIPAddHeader(Call-Info:\;Answer-After=0);
SIPAddHeader(P-Auto-Answer: normal);
SIPAddHeader(Answer-Mode: Auto);
Dial(SIP/${LOCAL(extension)});
Hangup();
break;
case 2:
Busy();
Hangup();
break;
case 5:
Congestion();
Hangup();
break;
default:
PlayBack(invalid);
Hangup();
break;
};
return;
};


Larry.

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Re: [asterisk-users] SPA504G auto answer

[Larry Moore]

On 23/10/2014 6:41 PM, Larry Moore wrote:




snip

Listing from my Asterisk:

'8000' = 1. Set(SIP_CODEC=alaw)
2. Dial(MulticastRTP/linksys/224.168.168.168:45678/224.168.168.168:6061)
3. Hangup()


snip


Hmm, my SPA525G doesn't auto-answer a page however my SPA92X do.


snip

Just upgraded the firmware on my SPA525G from 7.5.4 to 7.5.6 and the 
paging function is now working!


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Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?

[Larry Moore]

On 23/10/2014 3:55 AM, Tim Nelson wrote:

- Original Message -


Greetings-



Working with the T.38 gateway functionality that is sparsely
documented [1], I'm attempting to get the following functional:



Asterisk calling system -  Asterisk system in T.38 Gateway Mode (box
in question) -  SIP Provider



The problem is:



-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)



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].



Thank you,



--Tim



[1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway
[2]
http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html



*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.


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Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?

[Larry Moore]

On 22/10/2014 11:23 AM, Tim Nelson wrote:

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.

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Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?

[Larry Moore]

On 23/10/2014 10:07 PM, Larry Moore wrote:



On 22/10/2014 11:23 AM, Tim Nelson wrote:

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)'.

Larry.

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Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?

[Larry Moore]

On 24/10/2014 12:49 AM, Tim Nelson wrote:

- Original Message -



On 23/10/2014 10:07 PM, Larry Moore wrote:



On 22/10/2014 11:23 AM, Tim Nelson wrote:

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.

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Re: [asterisk-users] debugging T.38 issues

[Larry Moore]

It's been a while since I played with a Cisco SPA8800 with T.38 and with 
incoming faxes. There were settings in the SIP configurations for the 
SPA8800 to get it working with T.38 with asterisk, I don't know if the 
same will apply for this device.


If memory serves me you don't need, or perhaps, shouldn't use the T.38 
Gateway feature for the incoming call unless you are expecting your 
endpoint to be running T.30, in your case the SPA-112.


A quick glance over the data sheet for this device suggests it supports 
T.38 thus providing you have T.38 enabled on the SPA-211 I would advise 
against attempting to use the gateway feature for the incoming call.


In my experience Cisco use UDP redundancy and the default value is 1, 
you may want to experiment with this value and set it to 3. You may want 
to adjust the values in asterisk's udptl.conf by setting udptlfecentries 
 udptlfecspan to the same value, don't forget to restart asterisk after 
the change, I can't remember if you can force an unload and reload of 
the udptl module in asterisk.


You have not provided any information relating to the SPA-112's 
configuration in sip.conf so here are some settings I have;


[9003]
; Cisco SPA8800 FXS Port 3
; Analogue FAX Modem attached
type=friend
call-limit=2
qualify=yes
canreinvite=no
;directmedia=no
host=dynamic
context=FAX-T38
faxdetect=no
;t38pt_udptl=yes,redundancy,maxdatagram=400

I can't recall if I tested receiving a fax using G711 and used the 
faxgateway option when directing the call to the SPA8800


Larry.



On 14/10/2014 8:59 PM, Frederic Van Espen wrote:

Hello list,

We're currently facing some issues concerning T.38 gateway faxing.
This is a device used almost exclusively for receiving faxes. Calls
are incoming to asterisk on a SIP trunk (sangoma netborder) using
G711A. Gateway mode is activated in the asterisk dialplan towards a
Cisco SPA 112 running firmware 1.3.5. We are using asterisk 1.8.13.0
with the T.38 gateway patch applied (I know I know, we should move to
asterisk 11. I'm trying that tonight after business hours).

The issue we're seeing is that faxes incoming from some specific fax
machines consistenly fail. I have indentified 2 types of failures:
- Incoming call is sent to SPA112, reinvite to T.38 is initiated and
accepted. One T.38 packet is sent  from the SPA112 to asterisk
(t30ind: no-signal, see pcap output below) followed by 20 seconds of
nothingness and then the call is hung up.
- Incoming call is sent to SPA112, reinvite to T.38 is initiated and
accepted. A number of T.38 packets is sent from the SPA112 to asterisk
(it repeats NSF,CSI,DIS signals 3 times) followed ba call hangup.

In none of the above 2 cases do I see T.38 packets flowing from
asterisk to the SPA112.

In the logs I see these things that indicate T.38 gateway being started:
[Oct 14 14:16:17] DEBUG[11426] res_fax.c: SIP/SDSD0005-0007cb05 is
attempting to negotiate T.38 but SIP/SOV20001-0007cb04 does not
support it
[Oct 14 14:16:17] DEBUG[11426] res_fax.c: starting T.38 gateway for
T.38 channel SIP/SDSD0005-0007cb05 and G.711 channel
SIP/SOV20001-0007cb04
[Oct 14 14:16:17] DEBUG[11426] res_fax.c: Requesting a new FAX session
from 'Spandsp FAX Driver'.
[Oct 14 14:16:17] DEBUG[11426] res_fax.c: channel
'SIP/SOV20001-0007cb04' using FAX session '660'
[Oct 14 14:16:17] DEBUG[11426] chan_sip.c: T38 state changed to 3 on
channel SIP/SDSD0005-0007cb05


PCAP text output of 1st case:
216.025063 192.168.196.3 -  192.168.196.94 RTP PT=ITU-T G.711 PCMA,
SSRC=0x6118CC28, Seq=4877, Time=1179871936
216.042133 192.168.196.94 -  192.168.196.3 RTP PT=ITU-T G.711 PCMA,
SSRC=0x205734A, Seq=8641, Time=2227760
216.045031 192.168.196.3 -  192.168.196.94 RTP PT=ITU-T G.711 PCMA,
SSRC=0x6118CC28, Seq=4878, Time=1179872096
216.062169 192.168.196.94 -  192.168.196.3 RTP PT=ITU-T G.711 PCMA,
SSRC=0x205734A, Seq=8642, Time=2227920
216.065031 192.168.196.3 -  192.168.196.94 RTP PT=ITU-T G.711 PCMA,
SSRC=0x6118CC28, Seq=4879, Time=1179872256
216.082161 192.168.196.94 -  192.168.196.3 RTP PT=ITU-T G.711 PCMA,
SSRC=0x205734A, Seq=8643, Time=2228080
216.085022 192.168.196.3 -  192.168.196.94 RTP PT=ITU-T G.711 PCMA,
SSRC=0x6118CC28, Seq=4880, Time=1179872416
216.102145 192.168.196.94 -  192.168.196.3 RTP PT=ITU-T G.711 PCMA,
SSRC=0x205734A, Seq=8644, Time=2228240
216.105045 192.168.196.3 -  192.168.196.94 RTP PT=ITU-T G.711 PCMA,
SSRC=0x6118CC28, Seq=4881, Time=1179872576
216.122158 192.168.196.94 -  192.168.196.3 RTP PT=ITU-T G.711 PCMA,
SSRC=0x205734A, Seq=8645, Time=2228400
216.125042 192.168.196.3 -  192.168.196.94 RTP PT=ITU-T G.711 PCMA,
SSRC=0x6118CC28, Seq=4882, Time=1179872736
216.142145 192.168.196.94 -  192.168.196.3 RTP PT=ITU-T G.711 PCMA,
SSRC=0x205734A, Seq=8646, Time=2228560
216.145043 192.168.196.3 -  192.168.196.94 RTP PT=ITU-T G.711 PCMA,
SSRC=0x6118CC28, Seq=4883, Time=1179872896
216.146259 192.168.196.94 -  192.168.196.3 SIP/SDP Request: INVITE
sip:0043767079@192.168.196.3:5060, with session description
216.147280 

Re: [asterisk-users] debugging T.38 issues

[Larry Moore]

I don't know if this may help 
https://www.escaux.com/docs/DRD_T38Support_AdminGuide.html assuming you 
can enable T.38 on the Sangoma Netborder, then you can turn off the 
faxgateway option.


Larry.

On 15/10/2014 3:19 PM, Larry Moore wrote:

It's been a while since I played with a Cisco SPA8800 with T.38 and with
incoming faxes. There were settings in the SIP configurations for the
SPA8800 to get it working with T.38 with asterisk, I don't know if the
same will apply for this device.

If memory serves me you don't need, or perhaps, shouldn't use the T.38
Gateway feature for the incoming call unless you are expecting your
endpoint to be running T.30, in your case the SPA-112.

A quick glance over the data sheet for this device suggests it supports
T.38 thus providing you have T.38 enabled on the SPA-211 I would advise
against attempting to use the gateway feature for the incoming call.

In my experience Cisco use UDP redundancy and the default value is 1,
you may want to experiment with this value and set it to 3. You may want
to adjust the values in asterisk's udptl.conf by setting udptlfecentries
 udptlfecspan to the same value, don't forget to restart asterisk after
the change, I can't remember if you can force an unload and reload of
the udptl module in asterisk.

You have not provided any information relating to the SPA-112's
configuration in sip.conf so here are some settings I have;

[9003]
; Cisco SPA8800 FXS Port 3
; Analogue FAX Modem attached
type=friend
call-limit=2
qualify=yes
canreinvite=no
;directmedia=no
host=dynamic
context=FAX-T38
faxdetect=no
;t38pt_udptl=yes,redundancy,maxdatagram=400

I can't recall if I tested receiving a fax using G711 and used the
faxgateway option when directing the call to the SPA8800

Larry.



On 14/10/2014 8:59 PM, Frederic Van Espen wrote:

Hello list,

We're currently facing some issues concerning T.38 gateway faxing.
This is a device used almost exclusively for receiving faxes. Calls
are incoming to asterisk on a SIP trunk (sangoma netborder) using
G711A. Gateway mode is activated in the asterisk dialplan towards a
Cisco SPA 112 running firmware 1.3.5. We are using asterisk 1.8.13.0
with the T.38 gateway patch applied (I know I know, we should move to
asterisk 11. I'm trying that tonight after business hours).

The issue we're seeing is that faxes incoming from some specific fax
machines consistenly fail. I have indentified 2 types of failures:
- Incoming call is sent to SPA112, reinvite to T.38 is initiated and
accepted. One T.38 packet is sent from the SPA112 to asterisk
(t30ind: no-signal, see pcap output below) followed by 20 seconds of
nothingness and then the call is hung up.
- Incoming call is sent to SPA112, reinvite to T.38 is initiated and
accepted. A number of T.38 packets is sent from the SPA112 to asterisk
(it repeats NSF,CSI,DIS signals 3 times) followed ba call hangup.

In none of the above 2 cases do I see T.38 packets flowing from
asterisk to the SPA112.

In the logs I see these things that indicate T.38 gateway being started:
[Oct 14 14:16:17] DEBUG[11426] res_fax.c: SIP/SDSD0005-0007cb05 is
attempting to negotiate T.38 but SIP/SOV20001-0007cb04 does not
support it
[Oct 14 14:16:17] DEBUG[11426] res_fax.c: starting T.38 gateway for
T.38 channel SIP/SDSD0005-0007cb05 and G.711 channel
SIP/SOV20001-0007cb04
[Oct 14 14:16:17] DEBUG[11426] res_fax.c: Requesting a new FAX session
from 'Spandsp FAX Driver'.
[Oct 14 14:16:17] DEBUG[11426] res_fax.c: channel
'SIP/SOV20001-0007cb04' using FAX session '660'
[Oct 14 14:16:17] DEBUG[11426] chan_sip.c: T38 state changed to 3 on
channel SIP/SDSD0005-0007cb05


PCAP text output of 1st case:
216.025063 192.168.196.3 - 192.168.196.94 RTP PT=ITU-T G.711 PCMA,
SSRC=0x6118CC28, Seq=4877, Time=1179871936
216.042133 192.168.196.94 - 192.168.196.3 RTP PT=ITU-T G.711 PCMA,
SSRC=0x205734A, Seq=8641, Time=2227760
216.045031 192.168.196.3 - 192.168.196.94 RTP PT=ITU-T G.711 PCMA,
SSRC=0x6118CC28, Seq=4878, Time=1179872096
216.062169 192.168.196.94 - 192.168.196.3 RTP PT=ITU-T G.711 PCMA,
SSRC=0x205734A, Seq=8642, Time=2227920
216.065031 192.168.196.3 - 192.168.196.94 RTP PT=ITU-T G.711 PCMA,
SSRC=0x6118CC28, Seq=4879, Time=1179872256
216.082161 192.168.196.94 - 192.168.196.3 RTP PT=ITU-T G.711 PCMA,
SSRC=0x205734A, Seq=8643, Time=2228080
216.085022 192.168.196.3 - 192.168.196.94 RTP PT=ITU-T G.711 PCMA,
SSRC=0x6118CC28, Seq=4880, Time=1179872416
216.102145 192.168.196.94 - 192.168.196.3 RTP PT=ITU-T G.711 PCMA,
SSRC=0x205734A, Seq=8644, Time=2228240
216.105045 192.168.196.3 - 192.168.196.94 RTP PT=ITU-T G.711 PCMA,
SSRC=0x6118CC28, Seq=4881, Time=1179872576
216.122158 192.168.196.94 - 192.168.196.3 RTP PT=ITU-T G.711 PCMA,
SSRC=0x205734A, Seq=8645, Time=2228400
216.125042 192.168.196.3 - 192.168.196.94 RTP PT=ITU-T G.711 PCMA,
SSRC=0x6118CC28, Seq=4882, Time=1179872736
216.142145 192.168.196.94 - 192.168.196.3 RTP PT=ITU-T G.711 PCMA,
SSRC=0x205734A, Seq=8646, Time=2228560
216.145043 192.168.196.3 - 192.168.196.94

Re: [asterisk-users] SIP over 3G Mobile Network using NAT

[Larry Moore]

On 9/10/2014 9:28 PM, Mitul Limbani wrote:

Oops its qualify= n not notify=

Also check if your asterisk sip server I available with ports on the
public ip that your phone is trying to register from 3G nw.



In your devices sip configuration set;

qualify=no
nat=yes

in Bria;

Settings - Advanced - Default Network Traversal
Network Traversal Strategy  Custom Configuration
STUN Wi-Fi On
STUN Mobile On
STUN Server stun.counterpath.com (or another if appropriate)

Accounts - SIP Account - Account Advanced
Media Network Traversal
Suppress STUN Wi-Fi Off
Suppress STUN Mobile On

SIP NETWORK TRAVERSAL
Global IP Wi-Fi On
Global IP Mobile On

KEEP ALIVE
Wi-Fi Interval 0
Mobile Interval 0


Depending on your 3G provider, you may need to adjust Suppress STUN 
Mobile.



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Re: [asterisk-users] error receiving a fax ... but with a fax that was received without problems

[Larry Moore]

On 21/09/2014 9:11 PM, Bart Remmerie wrote:

Dear Joshua,

I don't think this is it:
first, this has been working in the past
second, why would I get a message like the one below exited non-zero
if everything is normal.



Joshua is correct, to reliably process the received fax you will need to 
process it in the 'h' extension.


You may want to refer to this thread 
http://lists.digium.com/pipermail/asterisk-users/2013-June/279625.html, 
note the afax2email script will e-mail the received tiff fax image if 
the PDF conversion fails.



Larry.

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Re: [asterisk-users] Fax buffer overflow detected

[Larry Moore]

On 7/02/2014 3:38 AM, Tech Support wrote:

All;

I’m running Asterisk 1.8.15-cert3 with the newest version of spandsp.
I’ve even tried unloading that and using Digium’s FFA module but I
receive the same error on an outbound transmission:

[2014-02-06 14:35:14] ERROR[19066]: udptl.c:294 encode_open_type: UDPTL
(SIP/XXX_outbound-): Buffer overflow detected (59 + 127
  175)

I only get this with one specific upstream provider. Has anyone seen
this before? Any help at all would be greatly appreciated.



Not sure if this relates to T38FaxMaxDatagram, here is an extract from a 
sample sip.conf file.


; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value 
(during T.38 setup) that
; is based on an incorrect interpretation of the T.38 recommendation, 
and results in failures
; because Asterisk does not believe it can send T.38 packets of a 
reasonable size to that
; endpoint (Cisco media gateways are one example of this situation). In 
these cases, during a
; T.38 call you will see warning messages on the console/in the logs 
from the Asterisk UDPTL
; stack complaining about lack of buffer space to send T.38 FAX packets. 
If this occurs, you
; can set an override (globally, or on a per-device basis) to make 
Asterisk ignore the
; T38FaxMaxDatagram value specified by the other endpoint, and use a 
configured value instead.
; This can be done by appending 'maxdatagram=value' to the t38pt_udptl 
configuration option,

; like this:
;
; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error 
correction and overrides
;   ; the other endpoint's provided 
value to assume we can
;   ; send 400 byte T.38 FAX packets 
to it.

;

Larry.

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Re: [asterisk-users] auto-answer call

[Larry Moore]

On 6/02/2014 2:21 AM, Salaheddine Elharit wrote:

thanks for your response ,

i test this solution but the issue still the same

any other solution
thanks and regards


2014-02-04 Steve Edwards asterisk@sedwards.com
mailto:asterisk@sedwards.com:

On Tue, 4 Feb 2014, Salaheddine Elharit wrote:

i have asterisk 1.4.43 installed and i want to configure the
auto-answer

exten = 506,1,SIPAddHeader(Call-Info:__\; answer-after=0)


I'm just a 1.2 Luddite...

I have this for a Sipura:

 exten = _!.,n,sipaddheader(Call-Info:\__;answer-after=0)

Maybe the quotes or the space after the semi-colon?

Maybe wireshark would yield a clue?

--
Thanks in advance,


Here is a list of headers used for various vendors, I can't remember 
which one is for Polycom.



SIPAddHeader(Alert-Info: Ring Answer);
SIPAddHeader(Alert-Info: Info=Alert-Autoanswer);
SIPAddHeader(Call-Info:\;Answer-After=0);
SIPAddHeader(P-Auto-Answer: normal);
SIPAddHeader(Answer-Mode: Auto);

Larry.

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Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

[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.

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Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

[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.

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Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

[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.

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Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

[Larry Moore]

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.

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Re: [asterisk-users] Asterisk Fax detection *11.7

[Larry Moore]

Hello,

Perhaps you need to have directmedia=no set for the channel, the call 
doesn't appear to have been answered hence asterisk won't be able to 
hear any tones to determine for itself if the call is an incoming fax.


Larry.

On 21/01/2014 6:51 PM, Jakob-Matthias Böttger wrote:

Hello everybody

I'm trying to enable the Digium res_fax app at my *11.7 Server.

a fax show stats comes up with
FAX Statistics:
---

Current Sessions : 0
Reserved Sessions : 0
Transmit Attempts : 0
Receive Attempts : 1
Completed FAXes : 1
Failed FAXes : 1

Digium G.711
Licensed Channels : 1
Max Concurrent : 0
Success : 0
Switched to T.38 : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure : 0
Protocol Error : 0
IO Partial : 0
IO Fail : 0

Digium T.38
Licensed Channels : 1
Max Concurrent : 1
Success : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure : 1
Protocol Error : 0
IO Partial : 0
IO Fail : 0

so that should be ok.

The corresponding dialplan section starts with


[from-sip]
include = inbound

[inbound]
exten = _X.,1,Answer()
exten = _X.,n,GotoIf(${BLACKLIST()}?black,1)
exten = _X.,n,Ringing
exten = _X.,n,Progress()
exten = _X.,n,Wait(5)
exten = _X.,n,Dial(SIP/123SIP/456,30,oxX)
...
exten = fax,1,NoOp( FAX DETECTED )
exten = fax,n,Goto(fax-rx,receive,1)

in the sip.conf i specified

[general]
sendrpid=rpid
trustrpid=yes
language=de
videosupport=yes
callevents=yes
caninvite=yes
qualify=yes
nat=force_rport,comedia
faxdetect=yes
t38pt_udptl=yes

...

[abcde]
type=peer
insecure=invite
defaultuser=12345678912
fromuser=12345678912
fromdomain=abcde.ab
secret=guess-what
host=abcde.ab
qualify=yes
context=from-sip
dtmfmode=rfc2833
callbackextension=12345678912


but all i can see if i try to send a testfax is

== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [12345678912@from-sip:1] Answer(SIP/abcde-0016, )
in new stack
  0x7fd11404cd00 -- Probation passed - setting RTP source address to
123.456.789.123:17108
-- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
0?black,1) in new stack
-- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016, )
in new stack
-- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016, )
in new stack
-- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 5) in
new stack
-- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
SIP/123SIP/456,30,oxX) in new stack
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
-- Called SIP/200
-- Called SIP/201
-- SIP/123-0018 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/456-0017 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/123-0018 is ringing
-- SIP/456-0017 is ringing


Any hints why thats not working?

Best Regards Jakob




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Re: [asterisk-users] Asterisk Fax detection *11.7

[Larry Moore]

Sorry, I missed the line showing the call had been answered.

On 22/01/2014 8:11 AM, Larry Moore wrote:

Hello,

Perhaps you need to have directmedia=no set for the channel, the call
doesn't appear to have been answered hence asterisk won't be able to
hear any tones to determine for itself if the call is an incoming fax.

Larry.

On 21/01/2014 6:51 PM, Jakob-Matthias Böttger wrote:

Hello everybody

I'm trying to enable the Digium res_fax app at my *11.7 Server.

a fax show stats comes up with
FAX Statistics:
---

Current Sessions : 0
Reserved Sessions : 0
Transmit Attempts : 0
Receive Attempts : 1
Completed FAXes : 1
Failed FAXes : 1

Digium G.711
Licensed Channels : 1
Max Concurrent : 0
Success : 0
Switched to T.38 : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure : 0
Protocol Error : 0
IO Partial : 0
IO Fail : 0

Digium T.38
Licensed Channels : 1
Max Concurrent : 1
Success : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure : 1
Protocol Error : 0
IO Partial : 0
IO Fail : 0

so that should be ok.

The corresponding dialplan section starts with


[from-sip]
include = inbound

[inbound]
exten = _X.,1,Answer()
exten = _X.,n,GotoIf(${BLACKLIST()}?black,1)
exten = _X.,n,Ringing
exten = _X.,n,Progress()
exten = _X.,n,Wait(5)
exten = _X.,n,Dial(SIP/123SIP/456,30,oxX)
...
exten = fax,1,NoOp( FAX DETECTED )
exten = fax,n,Goto(fax-rx,receive,1)

in the sip.conf i specified

[general]
sendrpid=rpid
trustrpid=yes
language=de
videosupport=yes
callevents=yes
caninvite=yes
qualify=yes
nat=force_rport,comedia
faxdetect=yes
t38pt_udptl=yes

...

[abcde]
type=peer
insecure=invite
defaultuser=12345678912
fromuser=12345678912
fromdomain=abcde.ab
secret=guess-what
host=abcde.ab
qualify=yes
context=from-sip
dtmfmode=rfc2833
callbackextension=12345678912


but all i can see if i try to send a testfax is

== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [12345678912@from-sip:1] Answer(SIP/abcde-0016, )
in new stack
 0x7fd11404cd00 -- Probation passed - setting RTP source address to
123.456.789.123:17108
-- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
0?black,1) in new stack
-- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016, )
in new stack
-- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016, )
in new stack
-- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 5) in
new stack
-- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
SIP/123SIP/456,30,oxX) in new stack
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
-- Called SIP/200
-- Called SIP/201
-- SIP/123-0018 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/456-0017 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/123-0018 is ringing
-- SIP/456-0017 is ringing


Any hints why thats not working?

Best Regards Jakob




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Re: [asterisk-users] Asterisk not receiving call from VPN address

[Larry Moore]

Have you checked your localnet=, deny=, permit=, contactdeny=  
contactpermit= settings?


My 2c worth.

On 20/01/2014 10:51 AM, David Cunningham wrote:

Hi,

We have a Kamailio and Asterisk cluster, both machines being on a real
103.x IP address and also on a 172.x OpenVPN address.

The problem is that when Kamailo receives a call from the VPN and
forwards it to the Asterisk server on it's 103.x address, Asterisk never
sees the call.

If Kamailio receives a call from the VPN and forwards the call to the
Asterisk server on it's 172.x address then it works. However, if the
call isn't from the VPN then forwarding it to the 172.x address doesn't
work. So basically the problem is going between the real network and the
VPN.

The question is, how can we make this work when calls are received on
either network on the Kamailio server and are forwarded to Asterisk?

Using ngrep on the Asterisk server we see that it does receive the
INVITE, but Asterisk's logging shows no sign it at all. We guess it's a
Linux networking issue rather than Asterisk's fault, but don't know
where to fix it. We do have net.ipv4.ip_forward = 1 on both the Kamailio
and Asterisk servers.

Thanks in advance for any help.

The ngrep on the Asterisk server:

U 2014/01/17 13:15:15.599557 172.x.x.x:5060 - 103.y.y.y:5060
INVITE sip:9067268@103.y.y.y:5060;transport=udp SIP/2.0.
Record-Route: sip:172.x.x.x;lr=on.
Via: SIP/2.0/UDP 172.x.x.x;branch=z9hG4bK50c7.f49ceb73.0.
Via: SIP/2.0/UDP 192.z.z.z:5062;rport=5062;branch=z9hG4bK806710997.
From: 9067271 sip:9067271@172.x.x.x;tag=198791249.
To: sip:9067268@172.x.x.x.
Call-ID: 1905625787@192.z.z.z.
...

172.x.x.x is the Kamailio server's VPN address
103.y.y.y is the Asterisk server's real address
192.z.z.z is the calling phone's LAN address

--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019




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Re: [asterisk-users] Asterisk not receiving call from VPN address

[Larry Moore]

Is Kamalio running on the same system as Asterisk?

On 21/01/2014 2:41 PM, David Cunningham wrote:

Hi Larry,

Thanks for the reply. We have all of those settings left out of our
sip.conf, so this should allow everything, right?



On 21 January 2014 17:38, Larry Moore lmo...@omninet.net.au
mailto:lmo...@omninet.net.au wrote:

Have you checked your localnet=, deny=, permit=, contactdeny= 
contactpermit= settings?

My 2c worth.


On 20/01/2014 10:51 AM, David Cunningham wrote:

Hi,

We have a Kamailio and Asterisk cluster, both machines being on
a real
103.x IP address and also on a 172.x OpenVPN address.

The problem is that when Kamailo receives a call from the VPN and
forwards it to the Asterisk server on it's 103.x address,
Asterisk never
sees the call.

If Kamailio receives a call from the VPN and forwards the call
to the
Asterisk server on it's 172.x address then it works. However, if the
call isn't from the VPN then forwarding it to the 172.x address
doesn't
work. So basically the problem is going between the real network
and the
VPN.

The question is, how can we make this work when calls are
received on
either network on the Kamailio server and are forwarded to Asterisk?

Using ngrep on the Asterisk server we see that it does receive the
INVITE, but Asterisk's logging shows no sign it at all. We guess
it's a
Linux networking issue rather than Asterisk's fault, but don't know
where to fix it. We do have net.ipv4.ip_forward = 1 on both the
Kamailio
and Asterisk servers.

Thanks in advance for any help.

The ngrep on the Asterisk server:

U 2014/01/17 13:15:15.599557 172
tel:15.599557%20172.x.x.x:5060 - 103.y.y.y:5060
INVITE sip:9067268@103.y.y.y:5060;__transport=udp SIP/2.0.
Record-Route: sip:172.x.x.x;lr=on.
Via: SIP/2.0/UDP 172.x.x.x;branch=z9hG4bK50c7.__f49ceb73.0.
Via: SIP/2.0/UDP
192.z.z.z:5062;rport=5062;__branch=z9hG4bK806710997.
From: 9067271 sip:9067271@172.x.x.x;tag=__198791249.
To: sip:9067268@172.x.x.x.
Call-ID: 1905625787@192.z.z.z.
...

172.x.x.x is the Kamailio server's VPN address
103.y.y.y is the Asterisk server's real address
192.z.z.z is the calling phone's LAN address

--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092 tel:%2B1%20213%20221%201092
UK: +44 (0) 20 3298 1642 tel:%2B44%20%280%29%2020%203298%201642
Australia: +61 (0) 2 8063 9019
tel:%2B61%20%280%29%202%208063%209019



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--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019




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Re: [asterisk-users] Movistar sip Mexico

[Larry Moore]

On 22/11/2013 6:49 AM, Alyed wrote:

Have you followed the instructions in:
https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway
and: http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway
??

If possible try with a different ATA since it seems not all of them work
fine with fax pass trough.

Alyed



My understanding of the original posting is that when a voice call 
arrives from the SIP provider it includes T38 information though the 
user only wants to accept the g729 component of the call and carry out a 
voice conversation.


If a fax is being received by the SIP provider it only has a the T38 
information for the call thus no audio (g729) information is in the SIP 
message.


I don't believe the original poster is attempting to receive a 
facsimile, instead use voice calls.


Larry.

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Re: [asterisk-users] Movistar sip Mexico

[Larry Moore]

On 21/11/2013 3:32 AM, Damian Gonzalez wrote:

Hello,

I have a problem with movistar in Mexico with a sip calls. Movistar send
to me T38 and G729 in the INVITE and they say that I have to ignore T38
and use G729 in the voice call.

When a fax call is made Movistar send only T38 in the INVITE.

Invite example:

v=0
o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2
s=sip call
c=IN IP4 192.168.1.2
t=0 0
m=audio 6370 RTP/AVP 18 101
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=image 6372 udptl t38
a=T38FaxVersion:0
a=T38FaxMaxBuffer:1100
a=T38FaxMaxDatagram:612
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

How can I  ignore T38 and use only G729 for this call?.

Thanks for your help.

Damian




Perhaps you could add the following to the peer configuration

faxdetect=no

Larry.

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

[Larry Moore]

On 8/11/2013 12:58 AM, Olivier wrote:
.
.
.

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.



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.



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!


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.

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

[Larry Moore]

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

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-0007]: res_fax.c:1698
receivefax_t38_init: channel 'SIP/myline-000e' 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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Re: [asterisk-users] IAX qualify timers

[Larry Moore]

On 21/08/2013 5:34 PM, Frederic Van Espen wrote:

Hi,

I think I encountered a bug in the qualify timers for IAX on asterisk
1.8 but I'd like to check if I'm not messing up in my config somewhere
before reporting a bug.

In my IAX peer configuration I have this:
[remote-host]
type=friend
host=172.16.6.45
username=remote-host
secret=test
notransfer=yes
qualify=16000


Perhaps you should use either yes or no for the qualify= line.
Here is an extract from the sample file

;qualify=yes; Make sure this peer is alive
;qualifysmoothing = yes ; use an average of the last two PONG
; results to reduce falsely detected 
LAGGED hosts

; Default: Off
;qualifyfreqok = 6  ; how frequently to ping the peer when
; everything seems to be ok, in 
milliseconds

;qualifyfreqnotok = 1   ; how frequently to ping the peer when it's
; either LAGGED or UNAVAILABLE, in 
milliseconds


Larry.

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Re: [asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

[Larry Moore]

On 01/08/2013, at 2:20 PM, Zoltán Fekete bl...@gyoz.info wrote:

 
 2013/8/1 Joshua Colp jc...@digium.com
 Larry Moore wrote:
 On 31/07/2013 8:08 PM, Joshua Colp wrote:
 Zoltán Fekete wrote:
 Thank You Larry!
 
 I have discussed with my provider. They are not able to insert the
 T38MaxBitRate value into the sip answer. :(
 https://gist.github.com/anonymous/6120148 (line 559)
 
 That means we are not able to passtrough T38 Faxes with any asterisk
 version at all?
 What do you mean? Am I able to modify and compile the source? Is it
 compicated? (I'm not a developer)
 
 Based on the SDP in your gist the remote implementation has given no
 attributes with the T.38 stream which makes it pretty broken
 (T38FaxRateManagement is mandatory) and fun. The two hard parts really
 would be 1. Modifying Asterisk in a sane fashion to cope and 2.
 Determining the exact settings to make the implementation happy.
 Defaults as defined in the spec are fine and good, but my experience has
 taught me to throw those out the window when it comes to actual
 implementations.
 
 
 It would seem that having a configurable option would be an idea for
 this scenario.
 
 That implies it would solve the problem, which my gut and experience tells 
 me... it wouldn't. I think the T.38 implementation is just cobbled together 
 and without knowing exactly how it behaves getting it to work would likely be 
 a nightmare (trust me, I've spent time in those deep dark reaches). Throwing 
 assumptions and defaults at it to try to make it work is of course an option.
 
 My testing with Asterisk 1.8 and T.38, I obserevd that setting
 FAXOPT(minrate) or FAXOPT(maxrate)had no effect, I concluded that when
 Astrerisk is receiving it uses hard coded values - is this a sane thing
 to do?!
 
 When Asterisk is receiving the stack implementation offers what it wants, 
 with the ability to override. So Asterisk doesn't hard code those values, the 
 stack provides them. What is hard coded is the default values if none are 
 received.
 
 I would even say it's a bug that the negotiation doesn't fail, since the 
 remote side isn't providing a mandatory attribute.
 
 -- 
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at:  www.digium.com   www.asterisk.org
 
 --
 _
 
 
 Yes you're right! As I know FAXOPT() value affect only when asterisk woks as 
 gateway. 
 We need passtrouh because my endpoints and also my provider supports T38.
 
 https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway
 Using T.38 Gateway mode
 T.38 Gateway mode should be used when one leg of a call is not capable of 
 T.38 mode. In the event that both legs are capable and Gateway mode is 
 configured, then the Gateway will step out of the way, allowing transparent 
 T.38 passthrough.
 
 The main problem is that I can't use G711 for the entire fax session because 
 the endpoints has 20-30ms response time.
 
 When I try to use my Asterisk as FAXOPT gateway (endpoint leg T38 and 
 provider leg G711) can I force somehow to not accept the T38 re-INVITEs from 
 the provider? 
 They have ~1ms response time, so G711 on that leg would be fine but they also 
 detect fax CED tones and sends the re-INVITEs.
 

Have you tried setting in your sip.conf for your provider t38pt_udptl=no whilst 
having the gateway option enabled?

Sorry I cant test this myself.

Cheers,

Larry.--
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Re: [asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

[Larry Moore]

On 31/07/2013 8:08 PM, Joshua Colp wrote:

Zoltán Fekete wrote:

Thank You Larry!

I have discussed with my provider. They are not able to insert the
T38MaxBitRate value into the sip answer. :(
https://gist.github.com/anonymous/6120148 (line 559)

That means we are not able to passtrough T38 Faxes with any asterisk
version at all?
What do you mean? Am I able to modify and compile the source? Is it
compicated? (I'm not a developer)


Based on the SDP in your gist the remote implementation has given no
attributes with the T.38 stream which makes it pretty broken
(T38FaxRateManagement is mandatory) and fun. The two hard parts really
would be 1. Modifying Asterisk in a sane fashion to cope and 2.
Determining the exact settings to make the implementation happy.
Defaults as defined in the spec are fine and good, but my experience has
taught me to throw those out the window when it comes to actual
implementations.



It would seem that having a configurable option would be an idea for 
this scenario.


My testing with Asterisk 1.8 and T.38, I obserevd that setting 
FAXOPT(minrate) or FAXOPT(maxrate)had no effect, I concluded that when 
Astrerisk is receiving it uses hard coded values - is this a sane thing 
to do?!


If Asterisk T.38 reception could be configured to use the values defined 
in FAXOPT(minrate) or FAXOPT(maxrate)this would be a good starting point 
for situations like this one.


Cheers,

Larry.


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Re: [asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

[Larry Moore]

On 22/07/2013 5:40 AM, Zoltán Fekete wrote:


Hi!

I have exactly the same problem on asterisk 1.8.22.0  and also on
separate 11.2.1 when sending fax to PSTN.
Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper softphone.
SpanDsp also works without any problem on my box.

As I remember it was a bug in 1.8.1.x that the a=T38MaxBitRate paramater
was sent as maxBitRate. Without capital M.

Are you closer to the solution?
I have tryed almost anything and I don't understand why sends the
T38MaxBitRate:2400 parameter.



Could it be because the remote endpoint does not supply the 
T38MaxBitRate attribute in its reply which then leads to Asterisk 
applying the Minimum Rate to your ATA!?


I am referring to the information around lines 403  404 of 
https://gist.github.com/anonymous/5701207.


Do you know what the Fax Rate was for the connection in 
https://gist.github.com/anonymous/5701150.


What happens if you insert in your dialplan something like

Set(FAXOPT(minrate)=4800)

Cheers,

Larry.

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Re: [asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

[Larry Moore]

On 23/07/2013 6:18 AM, Kevin Larsen wrote:

The a=T38MaxBitRate issue you refer to was one that was actually
discovered at my company and submitted by a colleague. It was fixed in
11.3.0 and 1.8.21.0. However, I think that it wouldn't help based on the
description below being that the parameter was missing altogether. I
think if that parameter is missing, then the code would in fact default
to 2400 as a safe value.



The Case Insensitive checking of the T.38 attributes was introduced in 
these versions.


Looking at the ITU-T T.38 Implementors' Guide (11 May 2012)in Table H.2, 
if the T38MaxBitRate attribute is omitted they suggest using the default 
value, they indicate a default value of 14400.


Larry.

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Re: [asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

[Larry Moore]

On 22/07/2013 10:19 PM, Larry Moore wrote:

On 22/07/2013 5:40 AM, Zoltán Fekete wrote:


Hi!

I have exactly the same problem on asterisk 1.8.22.0  and also on
separate 11.2.1 when sending fax to PSTN.
Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper
softphone.
SpanDsp also works without any problem on my box.

As I remember it was a bug in 1.8.1.x that the a=T38MaxBitRate paramater
was sent as maxBitRate. Without capital M.

Are you closer to the solution?
I have tryed almost anything and I don't understand why sends the
T38MaxBitRate:2400 parameter.



Could it be because the remote endpoint does not supply the
T38MaxBitRate attribute in its reply which then leads to Asterisk
applying the Minimum Rate to your ATA!?

I am referring to the information around lines 403  404 of
https://gist.github.com/anonymous/5701207.

Do you know what the Fax Rate was for the connection in
https://gist.github.com/anonymous/5701150.

What happens if you insert in your dialplan something like

Set(FAXOPT(minrate)=4800)



Another suggestion might be to set the variable in the peer 
configuration like;


setvar=FAXOPT(minrate)=4800


Larry.

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Re: [asterisk-users] Mailing a fax with mutt does not succeed

[Larry Moore]

On 22/06/2013 2:17 PM, Steve Edwards wrote:

On Sat, 22 Jun 2013, Larry Moore wrote:


echo  $MSGFILE
printf %18s Sender:   $MSGFILE; printf %-20s\n
${REMOTESTATIONID}  $MSGFILE
printf %18s Pages:   $MSGFILE; printf %-20s\n ${FAXPAGES}
 $MSGFILE
printf %18s Signal Rate:   $MSGFILE; printf %-20s\n
${FAXBITRATE} bit/s  $MSGFILE
printf %18s CallerID Number:   $MSGFILE; printf %-20s\n
${CIDNUMBER}  $MSGFILE
printf %18s CallerID Name:   $MSGFILE; printf %-20s\n
${CIDNAME}  $MSGFILE
printf %18s Call Duration:   $MSGFILE; printf %-20s\n
${DURATION}  $MSGFILE
printf %18s Status:   $MSGFILE; printf %-20s\n ${FAXERROR}
 $MSGFILE


How about:

#!/bin/bash
 FORMAT='%18s %-20s\n'
 (
 printf ${FORMAT} 'Sender:'${REMOTESTATIONID}
 printf ${FORMAT} 'Pages:'${FAXPAGES}
 printf ${FORMAT} 'Signal Rate:'${FAXBITRATE} bits/s
 printf ${FORMAT} 'CallerID Number:'${CIDNUMBER}
 printf ${FORMAT} 'CallerID Name:'${CIDNAME}
 printf ${FORMAT} 'Call Duration:'${DURATION}
 printf ${FORMAT} 'Status:'${FAXERROR}
 ) ${MSGFILE}



Thank Steve,

That makes the section more readable and it works with /bin/ksh too.

Cheers,

Larry.

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Re: [asterisk-users] Mailing a fax with mutt does not succeed

[Larry Moore]

On 20/06/2013 2:03 AM, Daniel - Asterisk wrote:

Hello everyone,
I'm trying to send a received fax with mutt, when I try it from the
Linux shel it works, but when trying with Asterisk's System command it
doesn't.
Successful Linux command:
echo | mutt -s New fax earohua...@gmail.com
mailto:earohua...@gmail.com -a /tmp/faxes/20130619.tif
Unsuccessful Asterisk Command:
same = n,System(mutt -s New fax elder.arohua...@gmail.com
mailto:elder.arohua...@gmail.com -a ${FAXDEST}/${tempfax}.tif)
I'm using Asterisk 1.8.19.0 on Debian 6.0.6, Asterisk was installed by root.
Any hint will be appreciated.
Elder D. Arohuanca
Lima - Peru


--


Reading the responses to this thread I suspect the issue is that the 
Muttrc file cannot be found, I am assuming the user:group Asterisk is 
running as does not have a shell associated with it nor a home directory.


I am using OpenBSD and the above is true for my environment.

I have in my Asterisk Spool directory the folder structure 
/var/spool/asterisk/fax/received. This folder is read-write for the 
account Asterisk runs as.


On my system I set it up to use macros, one to receive the fax and 
another to send the e-mail if one was received. The calling of the macro 
to send the e-mail is performed in the Hangup extension.


I use extensions.ael on my system so here are the macros I set up;

macro fax-receive( fax-number, header-info, sender, recipient ) {
/*
${ARG1} is Receiving Station Fax Number
${ARG2} is Fax Header Information
${ARG3} is Fax Sender E-mail Address
${ARG4} is Fax Recipient E-mail Address
*/
NoOp( FAX RECEIVE );
Set(FAXOPT(localstationid)=${LOCAL(fax-number)});
Set(FAXOPT(headerinfo)=${LOCAL(header-info)});
Set(FROMADDR=${LOCAL(sender)});
Set(TOADDR=${LOCAL(recipient)});
NoOp( SETTING FAXOPT );
NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)});
NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)});
NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)});
Set(RXSTART=${EPOCH});
Set(FAXRXPATH=/var/spool/asterisk/fax/received);
Set(FAXRXFILE=fax-${CALLERID(number)}-${UNIQUEID});
NoOp( RECEIVING FAX : ${FAXRXFILE} );
ReceiveFAX(${FAXRXPATH}/${FAXRXFILE}.tif,fd);
NoOp( Subroutine Return );
return;
};

macro email_rxfax() {
Set(CALLDURATION=$[(${EPOCH}-${RXSTART})]);
System(FAXRXFILE=${FAXRXFILE} FAXRXPATH=${FAXRXPATH} 
FAXRXSTATUS=${FAXOPT(status)} FAXERROR=${FAXOPT(statusstr)}
FROMADDR=${FROMADDR} TOADDR=${TOADDR} 
LOCALSTATIONID=${FAXOPT(localstationid)} 
REMOTESTATIONID=${REMOTESTATIONID} FAX
PAGES=${FAXOPT(pages)} FAXBITRATE=${FAXOPT(rate)} 
FAXRESOLUTION=${FAXOPT(resolution)} CIDNUMBER=${CALLERID(number)} CIDNAME
=${CALLERID(name)} CALLDURATION=${CALLDURATION} 
/usr/local/sbin/afax2email);

NoOp( STATUS : ${SYSTEMSTATUS} );
NoOp( Subroutine Return );
return;
};

Note, the System() line in email_rxfax() is one long line before the NoOp().

Here is an example entry in extensions.ael;


context fax-pstn1 {

fax = {
NoOp(Fax Received ${STRFTIME(,,%F %T %z)});
fax-receive(+61 8 9XXX,Your Business 
Name,FaxMaster@your.domain.name,lmoore@your.domain.name);

};

h = {
if ( X${FAXRXFILE} != X )
{
email_rxfax();
}
NoOp(Call/Fax Ended ${STRFTIME(,,%F %T %z)});
};

};

Here is a sample extensions.ael entry for an incioming fax.

context fax-pstn1 {

fax = {
NoOp(Fax Received ${STRFTIME(,,%F %T %z)});
fax-receive(+61 8 9XXX,Your Business 
Name,FaxMaster@your.domain.name,lmoore@your.domain.name);

};

h = {
if ( X${FAXRXFILE} != X )
{
email_rxfax();
}
NoOp(Call/Fax Ended ${STRFTIME(,,%F %T %z)});
};

};


I created a Muttrc file in /etc/asterisk, I use msmtp for the mail 
progam, sendmail works but I didn't like the Envelope-From information 
sendmail would add hence using msmtp, because the user asterisk runs as 
does not have a home directory nor a shell I _had_ to use an RC file for 
mutt. Here is my Muttrc file.


set sendmail=/usr/local/bin/msmtp
set use_from=yes
set realname=Asterisk Fax Agent
set from=FaxMaster
set record=
#set envelope_from=yes


This is the afax2email script I use, note that mutt is called with -F 
/etc/asterisk/Muttrc.



#! /bin/ksh
#
# Script to e-mail a received fax from Asterisk
#

#
# Environment variables imported from Asterisk
#

EMAIL=Asterisk Fax Agent $FROMADDR

export EMAIL SUBJECT TOADDR FROMADDR

TIFF2PDF=/usr/local/bin/tiff2pdf
TIFFINFO=/usr/local/bin/tiffinfo
MAILER=/usr/local/bin/mutt
MAILERARGS=-F /etc/asterisk/Muttrc
FILEPATH=$FAXRXPATH/$FAXRXFILE
FAXIMAGE=$FILEPATH.tif

Re: [asterisk-users] Asterisk T.38 Pass-Through doesn't work

[Larry Moore]

On 4/06/2013 4:53 AM, Andrey Polovov wrote:

On 06/03/2013 05:03 PM, Larry Moore wrote:

Have you checked the installed version of firmware against the latest
available from Cisco?

Oh! I didn't guess to check. The firmware was not fresh, but upgrading
doesn't help.

Looking at your SIP information when your ITSP initiated a T.38
session it did not indicate a maxmimum bitrate, it would appear your
spa112 attempted to negotiate a connection at 2400bps.

Whether there is a way to force my provider to indicate maximum bitrate?

Do you have a sip debug session when you sent a fax from your Asterisk
box to the PSTN, it would be interesting to see if it sends it as a
t.38 or reverts to G711 audio.

I have collect a set of debugs (with fresh SPA112 firmware) and actual
config files:



I would suggest you test the SPA112 directly against your SIP provider 
however ensure you have the latest firmware, there appear to have been 
some FAX related fixes in recent versions.


To do this change the following (based upon the screen shot you made 
available);


Nat Mapping Enable: yes

Call Waiting Serv: no
MWI Serv: no

Proxy: 80.75.130.136

Register: no
Make Call Without Reg: yes
Ans Call Without Reg: yes

User ID: 7495777
Password: remotesecret

FAX T38 Redundancy: 3
FAX Tone Detect Mode: callee
FAX T38 Return to Voice: no

When you get this working you can then look at making it work through 
Asterisk - this is how I got my SPA8800 working.


I am assuming your network configuration is set up correctly on the spa112.

You may want to look at enabling the following options, on my SPA8800 
they are located under the SIP tab in the section headed NAT Parameters:


Handle via rport: yes
Insert via rport: yes
Send Resp to Src Port: yes

In addition, once it is working and providing your SIP provider permits 
ECM through a T.38 service I would encourage one to enable options such 
as FAX T38 ECM Enable.



If you are still experiencing problems you may want to perform a packet 
capture (set the snap size to 1500) of the communications between the 
spa112 and the other end point and run it through Wireshark and examine 
the VoIP calls in the captured packets.


Good luck.

Larry.

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Re: [asterisk-users] Asterisk T.38 Pass-Through doesn't work

[Larry Moore]

On 3/06/2013 8:04 PM, Andrey Polovov wrote:

Thank you for reply, Larry!

On 06/03/2013 05:14 AM, Larry Moore wrote:

1) On SPA112 set FAX T38 Redundancy = 3

I have tried to change this value with no effect.

2) Add t38pt_usertpsource=yes in [mtt] section

This option take no positive effect for me, asterisk continues to use
ports from udptl.conf range. But in the oser side, I think, that there
are no problems with network, because:
* on router, that provide internet for asterisk configured full DNAT
from white IP to asterisk server,
* fax between asterisk cmd SendFax and PSTN fax works great.

3) Change maxdatagram=200 to maxdatagram=1400
4) In udptl.conf change T38FaxMaxDatagram to a value of 1400
5) In udptl.conf change use_even_ports to yes

This settings doesnt help me.

You don't appear to list the sip.conf entry for the SPA112.

I`m sorry, spa112 peer settings are in mysql extconfig, config file
analogue:
[297]
context=office
secret=xxx
host-dynamic
nat=yes
type=friend
callerid=297
disallow=all
allow=alaw

Where did t38pt_rtp  t38_tcp come from?

This settings I have googled on some forum and then add to my config
with despair.

You may also want to experiment with the SPA112 setting FAX T38 ECM
Enable

This option doesnt help me too :(.

Additional info: fax between two SPA112 works perfect! Now I returned
settings to original state as described in first letter.




Have you checked the installed version of firmware against the latest 
available from Cisco?


Looking at your SIP information when your ITSP initiated a T.38 session 
it did not indicate a maxmimum bitrate, it would appear your spa112 
attempted to negotiate a connection at 2400bps.


Do you have a sip debug session when you sent a fax from your Asterisk 
box to the PSTN, it would be interesting to see if it sends it as a t.38 
or reverts to G711 audio.


You will only want to experiment with the ECM option once you have t.38 
working.


Larry.

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Re: [asterisk-users] Asterisk T.38 Pass-Through doesn't work

[Larry Moore]

On 2/06/2013 9:34 PM, Андрей Половов wrote:

What I have is:
* Asterisk 1.8.10.1~dfsg-1ubuntu1,
* SPA112 ATA with analog fax in 1-st FXS port connected,
* SIP trunk with provider supporting T.38.

My network looks like this:
* spa112 (192.168.33.200/24) and Asterisk (192.168.5.253/24) in
neighbouring LANs,
* Asterisk connects to the provider (80.75.130.136) via router
(82.200.7.184). Router has full DNAT to Asterisk server.

What happens?
* The fax from SPA112 to Asterisk cmd ReceiveFax works well,
* The fax from Asterisk cmd SendFax to PSTN fax works well,
* However, the fax from SPA112 to PSTN fax doesn't work. Using udptl
debug, I can see packets between Asterisk and both sides (SPA112 and
PSTN fax) but it seems that faxes can't agree how to send image.

== sip.conf:
[general]
tcpenable=yes
videosupport=yes
transport=udp,tcp
dtmfmode=rfc2833
qualify=yes
directmedia=no
allowguest=no
alwaysauthreject=yes
rtcachefriends=yes
rtupdate=no
callcounter=yes
t38pt_udptl=yes,redundancy,maxdatagram=200
t38pt_rtp=no
t38pt_tcp=no
ignoresdpversion=yes
disallow=all
allow=alaw
allow=ulaw
externip=82.200.7.184
localnet=192.168.0.0/255.255.0.0

[mtt]
type=peer
host=80.75.130.136
fromuser=7495777
disallow=all
allow=alaw,ulaw
directmedia=no
canreinvite=no
nat=yes

== udptl.conf:
[general]
udptlstart=4000
udptlend=4999
T38FaxUdpEC=t38UDPRedundancy
T38FaxMaxDatagram=200
udptlfecentries = 3
udptlfecspan = 3
use_even_ports = no




I don't have access to one of these devices though some suggestions you 
could try;


1) On SPA112 set FAX T38 Redundancy = 3
2) Add t38pt_usertpsource=yes in [mtt] section
3) Change maxdatagram=200 to maxdatagram=1400
4) In udptl.conf change T38FaxMaxDatagram to a value of 1400
5) In udptl.conf change use_even_ports to yes


You don't appear to list the sip.conf entry for the SPA112.

Where did t38pt_rtp  t38_tcp come from?

You may also want to experiment with the SPA112 setting FAX T38 ECM Enable

Cheers,

Larry.

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Re: [asterisk-users] Faxdetect + T38gateway

[Larry Moore]

On 18/02/2013 6:25 AM, Jean-Denis Girard wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi list,

I'm using faxdetect so that users can receive faxes on their phone
numbers. It works fine.

Fax is actually received by Hylafax through iaxmodem.

I'm also using T.38 between asterisk and ATA (HT502/503) for better
reliability. Sending from / to Hylafax works fine too; I checked with
udptl set debug on that T.38 is actually used.

The problem I see is when other end tries to use T.38: re-invite is to
the SIP phone, which obviously rejects it (488 not applicable here), so
T.38 is not used when hitting fax extension. Is there a solution to
combine fax detect and t38getaway on the same call?

This is with asterisk-11.2.1.




I have Asterisk 1.8 and use a HT503.

In sip.conf for the HT503 I use the following;

[ht503]
.
.
faxdetect=t38
directmedia=update  ; may need to set directmedia=no
.
.

and for the handset;

[sip-phone]
.
faxdetect=no
.


You have not provided any information relating to your configurations.

I believe you could have the following in the HT503 sip.conf entry in 
Asterisk 11;


setvar=FAXOPT(gateway)=yes

Presumably you have also configured the fax extension according to your 
environment.


Larry.

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Re: [asterisk-users] Asterisk 11 with t38modem 2.0: 488 Not acceptable here

[Larry Moore]

My 2 cents worth.

Turn off faxdetect in the peer configuration for Asterisk.

Failing that, try the Fax Gateway feature in Asterisk 11 to Hylafax 
listening on an IAX2 channel.


Larry.

On 24/01/2013 6:32 AM, Carsten Maass wrote:

Hello all,

we do have a problem here with Asterisk 11 talking T.38 to a t38modem
2.0. The callflow is:

ISDN PRI -- Berofix (10.1.1.150) -- Asterisk (10.1.1.148) -- t38modem
(10.1.1.148) -- Hylafax [1]

Although the call gets connected, both parties are unable to negotiate
the audio codecs:

[2013-01-23 21:59:57] VERBOSE[8805][C-] chan_sip.c: Got T.38
offer in SDP in dialog 237c0a65027630cd0c8bf2e70b5b3dc7@10.1.1.148:5060
[2013-01-23 21:59:57] VERBOSE[8805][C-] chan_sip.c:
Capabilities: us - (alaw), peer -
audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
[2013-01-23 21:59:57] VERBOSE[8805][C-] chan_sip.c: Non-codec
capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing),
combined - 0x0 (nothing)
[2013-01-23 21:59:57] VERBOSE[8805][C-] chan_sip.c: Got T.38
Re-invite without audio. Keeping RTP active during T.38 session.

which results in a SIP/2.0 488 Not acceptable here from the Asterisk
and the call gets disconnected.

Asterisk full log with SIP trace is at http://pastebin.com/NNr6BTdp

This looks a lot like
https://issues.asterisk.org/jira/browse/ASTERISK-15596 which doesn't
seems to be solved by now.

Is this still a known issue in Asterisk 11? What can I do to make
Asterisk 11 play nice with t38modem v2.0?


Environment:
--
Red Hat Enterprise Linux Server release 6.3 (Santiago)

Linux myhost.mydomain.local 2.6.32-279.19.1.el6.x86_64 #1 SMP Sat Nov 24
14:35:28 EST 2012 x86_64 x86_64 x86_64 GNU/Linux

T38Modem Version 2.0.0
  (OPAL-3.9.0/3.9beta0, PTLIB-2.9.0/2.9beta0 (svn:24165)) by Vyacheslav
Frolov on Unix Linux (2.6.32-279.19.1.el6.x86_64-x86_64)

Asterisk 11.1.0
Hylafax 6.0.6


Thanx in advance and greetings,
Carsten.


[1] Yes, I know: the Berofix appliance can talk directly to the
t38modems, which works perfectly well here. But there is a limitation of
140 SIP Accounts in the Berofix and we have to serve ~500 fax numbers.
So we had to set Asterisk between the Berofix and the t38modems, bearing
the SIP accounts.




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Re: [asterisk-users] MaxCallBR Peer Setting

[Larry Moore]

On 4/01/2013 9:39 PM, XBrian wrote:

Hi

sip show peer 21342

gives me peer 21342's parameters. I am interested in the MaxCallBR line i.e.

   MaxCallBR: 384 kbps


What exactly does this mean?




Extracted from sample sip.conf file;

;maxcallbitrate=384 ; Maximum bitrate for video calls 
(default 384 kb/s)
; Videosupport and maxcallbitrate is 
settable

; for peers and users as well

Larry.


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Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

[Larry Moore]

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.

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Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

[Larry Moore]

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.

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Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

[Larry Moore]

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.



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Re: [asterisk-users] ReceiveFax

[Larry Moore]

On 19/12/2012 5:46 AM, Andrew Nowrot wrote:

Hi

I am trying to get faxes to work with Asterisk and ReceiveFax. The
problem is that I am getting only about 60% of success (for example two
of six fax transmissions always fails). The most common errors are
Error : Unexpected DCN after requested retransmission and Error :
Disconnected after permitted retries.

The scenario is like this:

Fax machine - ATA -- (SIP) -- Asterisk 11.1.0 - ReceiveFax

I am using Cisco SPA122 and Sipura SPA2102 (both with the newest
firmware). Both are configured to use T38. I can see that Asterisk and
ATAs negotiate fax T38.
Is this a common issue with these ATAs. I am obviously doing something
wrong. Can anyone point me in to right direction. I can provide the logs
form both good and failing transmission.
BTW
Fax machine to fax machine transmission with these ATAs almost always works.

Cheers



Just a suggestion, check the current T.38 redundancy value in the SPA's, 
if it isn't set to 3 then set it to 3.


If Asterisk 11 is still using udptl.conf then check the following;

udptlfecentries = 3
udptlfecspan = 3

You may also want to set;

use_even_ports = yes

Good luck.

Larry.

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Re: [asterisk-users] Queues and Distinctive Ring with Alert-Info

[Larry Moore]

On 26/11/2012 10:14 AM, Klaverstyn, David C wrote:

Hi All,

I’m new to Queues and I have created one as follows which seems to work ok.

[david-test]

strategy = rrmemory

timeout = 10

retry = 0

maxlen = 0

announce-frequency = 0

announce-holdtime = no

member = SIP/121

member = SIP/122

member = SIP/123

I’m wondering how do you change the SipAddHeader/Alert-Info when a call
comes from a queue so users know it is a queue that is calling?

Is something like the following supposed to work?

exten = 0453451564,1,SipAddHeader(Alert-Info: n=Classic-4;w=3;c=4)

exten = 0453451564,2,Queue(david-test)




Seems to work with Asterisk 1.8.18.0.

I'm using extensions.ael and have tested the following;

400 = {
SIPAddHeader(Alert-Info: n=Classic-4;w=3;c=4);
Queue(400,inrt,,,30);
Hangup();
};


Larry.


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Re: [asterisk-users] Incompatible voice frame ulaw/alaw

[Larry Moore]

On 26/08/2012 3:45 AM, Markus wrote:

Hi list, I'm resending this.. no one answered. No one got an idea? Thank
you!


When I receive an incoming call from a SIP peer where I've configured

disallow=all
allow=alaw
(and no other codec)

I can see the following NOTICE on the console:

Dropping incompatible voice frame SIP/peer07-007c of format ulaw
since our native format has changed to (alaw)

My question is: where can I change the native format from ulaw to alaw
(or something else)? Is ulaw, as the native format, adjustable in some
config file or is it hard-coded into Asterisk?

It doesn't seem to have any effect on the voice quality but the messages
on the console are quite annoying.



I suspect you will find the frequency of these messages is the value you 
have set for rtpkeepalive.


I would suggest you include the following in your peer's configuration;

rtpkeepalive=0

Larry.

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Re: [asterisk-users] How to set SIP to auto answer in the dial plan .

[Larry Moore]

I have the following in my intercom macro in extensions.ael;

SIPAddHeader(Alert-Info: Ring Answer);
SIPAddHeader(Call-Info:\;Answer-After=0);
SIPAddHeader(P-Auto-Answer: normal);


If memory serves me, respectively they are for the following vendors;

Grandstream
Linksys/Cisco SPA
Yealink

Larry.

On 14/07/2012 1:50 PM, upendra wrote:

Hi,

its not working for me ! let me know anyone having sample dialplan
so that i can use for test 1 sip call answer.



regards
Upendra

On Fri, Jul 13, 2012 at 9:57 PM, Jared Baxley jared.bax...@gmail.com
mailto:jared.bax...@gmail.com wrote:

You also have to send the alert info you particular phone needs to
make it autoanswer.

On Jul 13, 2012 4:53 AM, upendra uppi...@gmail.com
mailto:uppi...@gmail.com wrote:

Hi,

thanks , i need to put this in the sip context...

regards
Upendra.

On Fri, Jul 13, 2012 at 3:15 PM, Zohair Raza
engineerzuhairr...@gmail.com
mailto:engineerzuhairr...@gmail.com wrote:

try with SipAddHeader(uri=answer-after=0)

check syntax for Addheader

Regards,
Zohair Raza




On Fri, Jul 13, 2012 at 1:42 PM, upendra uppi...@gmail.com
mailto:uppi...@gmail.com wrote:
  Hi,
 
 
  I am trying to write dial plan for sip to auto answer
(auto attend) the
  incoming call to the sip phone.
 
  - If i call from sip1 to sip2 then sip2 should
automatically answer the call
  and play some sound file.
  I am trying to do this but as new to the asterisk dial
plan configuration ,
  so not able Todo this.
  help me if anyone already done this setup.
 
 
 
  Regards
  Upendra.
 
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Re: [asterisk-users] IAX2 passing back and forth variables

[Larry Moore]

On 23/05/2012 10:46 AM, Ruddy Gbaguidi wrote:


I cannot find it

*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny 
Nicholas

*Sent:* 2012-05-21 10:25
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* Re: [asterisk-users] IAX2 passing back and forth variables

There was a nice thread on this back in April.




Perhaps it is the thread which started on the 15th of April with the 
subject line


Set variables from one asterisk ta a second.





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Re: [asterisk-users] how to set iaxmodem receiving speed

[Larry Moore]

On 17/05/2012 2:47 PM, gincantalupo wrote:

Hi Steve,

you are telling me there is no way to set a particular speed on my 
iaxmodem in order to force the sender speed?
I have some problems with a customer who gets malformed faxes even if 
no error occurs. Since I cannot tell the sender to lower its fax 
speed, my idea is to force my iaxmodem to a lower fixed speed so the 
sender is oblidged to negotiate at that speed (or lower, of course) 
without the customer could realize it, at least at first. :)
There is no ATA in the middle (I'm using it for my tests but my 
customer does not have any), all faxes are received thru a primary 
channel to a bunch of iaxmodems. Sometimes some faxes are corrupted, 
that's why I thought to lower the speed. I could try to disable ECM 
but that's even harder to do (found nothing on internet).




Hi Giorgio,

You may want to try these settings to set the most basic form of 
transmission on your receiving modems, I would however have thought, ECM 
being on would be better for you as it could then deal with lost frames.


Class1MRSupport:no
Class1MMRSupport:   no
Class1ECMSupport:   no

To set the ECM frame size to a lower value than the default of 64, you 
would set the following


Class1PersistentECM:yes
Class1ECMFrameSize: 64

Perhaps the corruption is occurring at the senders end before the data 
is pushed through the modem.


Cheers,

Larry.

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Re: [asterisk-users] how to set iaxmodem receiving speed

[Larry Moore]

I have iaxmodem version 1.2.0 installed on my system.

I have set the following in the IAX configuration file, SIGHUP'd 
FaxGetty and submitted a single page outbound fax via Asterisk;


Class1RMQueryCmd:   !24,48,72 # enable this to disable V.17 
receiving
Class1TMQueryCmd:   !24,48,72 # enable this to disable V.17 
sending


The resulting output from my T.38 Gateway reports the following;

-- Connection Statistics
Bit Rate :7200
ECM : No
Pages : 1
-- Hungup 'IAX2/iaxmodem0-11055'

I also tested with the maximum speed set to 4800, the image was received 
however the responses to EOP timed out, I don't know if the is to do 
with my Asterisk T.38 gateway or my VoIP providers T.38 gateway. The 
result was the fax was retried for the defined number of attempts.


Cheers,

Larry.

On 16/05/2012 6:28 PM, gincantalupo wrote:

Hi all,

I'm trying to lower my iaxmodem speed but still I haven't found any 
solution...I tried to add
Class1RMQueryCmd:   !24,48,72
to config.IAXtty but does not work...Hylafax says it it running at 9600 
(sometimes at 14400) baud..

Any ideas?

Thank you.

Giorgio


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Re: [asterisk-users] how to set iaxmodem receiving speed

[Larry Moore]

Read the subject line more closely.

Tested receiving too,

I set the Send  Receive speed of the receiving analogue modem to that 
below, the log file on the sending modem (iaxmodem) reported it capable 
of 9600.


May 16 21:32:04.28: [ 2335]: REMOTE best rate 9600 bit/s
May 16 21:32:04.28: [ 2335]: REMOTE max A3 page width (303 mm)
May 16 21:32:04.28: [ 2335]: REMOTE max unlimited page length
May 16 21:32:04.28: [ 2335]: REMOTE best vres R16 x 15.4 line/mm
May 16 21:32:04.28: [ 2335]: REMOTE format support: MH, MR, MMR
May 16 21:32:04.28: [ 2335]: REMOTE supports T.30 Annex A, 256-byte ECM
May 16 21:32:04.28: [ 2335]: REMOTE best 0 ms/scanline
May 16 21:32:04.28: [ 2335]: USE 9600 bit/s

Perhaps the issue is with Hylafax.

Setting the Transmit  Receive strings to !24,48,72,96 seems to yield 
the most reliability in transmission


Cheers,

Larry.

On 16/05/2012 7:23 PM, Larry Moore wrote:

I have iaxmodem version 1.2.0 installed on my system.

I have set the following in the IAX configuration file, SIGHUP'd 
FaxGetty and submitted a single page outbound fax via Asterisk;


Class1RMQueryCmd:   !24,48,72 # enable this to disable V.17 
receiving
Class1TMQueryCmd:   !24,48,72 # enable this to disable V.17 
sending


The resulting output from my T.38 Gateway reports the following;

-- Connection Statistics
Bit Rate :7200
ECM : No
Pages : 1
-- Hungup 'IAX2/iaxmodem0-11055'

I also tested with the maximum speed set to 4800, the image was 
received however the responses to EOP timed out, I don't know if the 
is to do with my Asterisk T.38 gateway or my VoIP providers T.38 
gateway. The result was the fax was retried for the defined number of 
attempts.


Cheers,

Larry.

On 16/05/2012 6:28 PM, gincantalupo wrote:

Hi all,

I'm trying to lower my iaxmodem speed but still I haven't found any 
solution...I tried to add
Class1RMQueryCmd:   !24,48,72
to config.IAXtty but does not work...Hylafax says it it running at 9600 
(sometimes at 14400) baud..

Any ideas?

Thank you.

Giorgio


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Re: [asterisk-users] how to set iaxmodem receiving speed

[Larry Moore]

On 17/05/2012 1:24 AM, Steve Underwood wrote:

Hi,

On 05/16/2012 09:59 PM, Larry Moore wrote:

Read the subject line more closely.

Tested receiving too,

I set the Send  Receive speed of the receiving analogue modem to 
that below, the log file on the sending modem (iaxmodem) reported it 
capable of 9600.


May 16 21:32:04.28: [ 2335]: REMOTE best rate 9600 bit/s
May 16 21:32:04.28: [ 2335]: REMOTE max A3 page width (303 mm)
May 16 21:32:04.28: [ 2335]: REMOTE max unlimited page length
May 16 21:32:04.28: [ 2335]: REMOTE best vres R16 x 15.4 line/mm
May 16 21:32:04.28: [ 2335]: REMOTE format support: MH, MR, MMR
May 16 21:32:04.28: [ 2335]: REMOTE supports T.30 Annex A, 256-byte ECM
May 16 21:32:04.28: [ 2335]: REMOTE best 0 ms/scanline
May 16 21:32:04.28: [ 2335]: USE 9600 bit/s

Perhaps the issue is with Hylafax.

Setting the Transmit  Receive strings to !24,48,72,96 seems to 
yield the most reliability in transmission
If you have an ATA in the path that is often the case. Many of them 
badly mess up a FAX signal. Without such a distortion machine V.17 
should be fine.


The receiving analogue modem is directly connected to the PSTN network 
and was used to to determine if the reported issue could be reproduced 
on a non-iaxmodem.


My outgoing connections were through an iaxmodem with T.38 gateway 
enabled and disabled, with most successful transmissions being when T.38 
gateway was used.


Larry.

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Re: [asterisk-users] OT - Incoming fax cuts ADSL line

[Larry Moore]

On 17/05/2012 12:18 AM, James Sharp wrote:

On 5/16/2012 12:07 PM, Tim Nelson wrote:

- Original Message -

Hi,

I'm facing a strange situation.
Though it's not directly related to Asterisk, I do think it is
interesting to this mailing list.


The setup is a single line which is split between an ADSL
modem/routeur and a fax machine (Asterisk was removed from the
equation).

Any time the fax machine rings (incoming fax), the ADSL service is
troubled to the VPN users are disconnected.
It can be reproduced at will.

I've changed the ADSL filter twice (a different unit, then a
different
model) without any visible change.
What could explain this ?



I've experienced this quite a few times, and after working with a 
local telco, it has become policy to not place ADSL on lines where 
fax is going to be used. I'm unsure of the exact technical reasons 
behind this other than 'the fax signals/frequencies interfere with 
the ADSL signalling/frequencies used on the circuit'. It sounds like 
you might want to separate your fax/ADSL lines.


--Tim


You might also be able to limit the Fax machines maximum transmission 
rate so the modem's transmission spectrum doesn't inch up into where 
the ADSL service is.




I have clients with their ADSL2+ service attached to their fax lines 
with no problems observed.


Perhaps the issue is the fax machines attenuators are not set correctly 
are are to _loud_ on the PSTN.


In Australia Telstra advised the signal level received at the exchange 
should be between -15dB and -17dB. They have a Fax On Line Diagnostic 
System (FOLDS) which you can send a transmission to, a report is 
returned advising of the quality of your transmission including measured 
signal and noise levels.


In old days the fax machine might have a wired jumper block to set the 
attenuation, more modern devices would be configured from the front 
panel, typically in a maintenance mode. Your good old dial-up modems 
with fax capabilities would have an S-Register or two to set the 
attenuation.


Larry.

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Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)

[Larry Moore]

On 18/04/2012 6:39 AM, Kevin P. Fleming wrote:

On 04/17/2012 06:17 AM, Larry Moore wrote:

The send log you have posted does not show any outgoing T.38 packets
from your system.

I set up a test build of 1.8.11.0 using the patch recently released, I
have difficulties sending T.38 with this patch, in fact I cannot send
successfully however I can receive. I did however observe some outgoing
T.38 packets. The analogue fax modem I was dialling into is under my
control hence the log files showed there was no signalling coming from
my ITSP,

The T.38 session on my Asterisk server show the CRP's which were sent
from the analogue fax device during the negotiation.

The patch I have used for a while seems to give me outgoing
functionality as well as incoming.

I can't reproduce your scenario whereby your T.38 session is
communicating with a different gateway to the SIP server you use hence
can only speculate that Asterisk has difficulty wanting to send T.38 SDP
traffic when it is a different device than the SIP server it negotiates
with.


We know for a fact that Asterisk has no trouble with the signaling and 
media going to different addresses/ports. Honestly, I just don't 
understand why all of this effort is being put into trying to use an 
old (and clearly broken) patch for adding T.38 gateway support to 
Asterisk 1.8.


You guys know that it works in Asterisk 10, but you say you can't use 
Asterisk 10 for some reason that I don't understand.




I have downloaded asterisk 10.3.0 and compiled on a Centos 6 system I 
setup to compare behaviour on OpenBSD with a Linux version of asterisk 
based upon the OpenBSD port.


Unfortunately the T.38 Gateway functionality in my build of 10.3.0 
doesn't appear to work. Looking at the upgrade documentation from 1.8 
there doesn't appear to be any considerations applicable to my setup.


As an excercise in futility I downloaded the Asterisk 1.8.11.0 source 
and compiled using the version of T.38 patch I have maintained and 
tested by sending a fax via an IAX channel out through my SIP provider, 
the fax was sent successfully.


I then removed and recreated the asterisk 1.8.11.0 directory and applied 
the back-port patch and observed the same problem when attempting to 
send through the T.38 gateway as was observed in Asterisk 10.



Console output of Asterisk 1.8.11.0 with Asterisk 10 backport patch:

 asterisk-dev*CLI
-- Accepting AUTHENTICATED call from 192.168.54.12:
 requested format = slin,
 requested prefs = (),
 actual format = slin,
 host prefs = (slin|alaw|ulaw),
 priority = mine
-- Executing [@FAX-T30:1] Set(IAX2/iaxmodem1-4445, 
FAXOPT(t38gateway)=yes) in new stack
-- Executing [@FAX-T30:2] Dial(IAX2/iaxmodem1-4445, 
SIP/@itsp-fax,55) in new stack

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called SIP/@itsp-fax
-- SIP/itsp-fax-0004 is making progress passing it to 
IAX2/iaxmodem1-4445

-- SIP/itsp-fax-0004 answered IAX2/iaxmodem1-4445
[Apr 23 21:14:11] NOTICE[14165]: channel.c:4152 __ast_read: Dropping 
incompatible voice frame on SIP/itsp-fax-0004 of format slin since 
our native format has changed to 0x8 (alaw)

  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
[Apr 23 21:14:12] ERROR[14165]: astobj2.c:110 INTERNAL_OBJ: user_data is 
NULL
[Apr 23 21:14:23] WARNING[14165]: res_rtp_asterisk.c:2135 ast_rtp_read: 
RTP Read too short
[Apr 23 21:14:24] WARNING[14165]: res_rtp_asterisk.c:2135 ast_rtp_read: 
RTP Read too short
[Apr 23 21:14:24] WARNING[14165]: res_rtp_asterisk.c:2135 ast_rtp_read: 
RTP Read too short
[Apr 23 21:14:24] WARNING[14165]: res_rtp_asterisk.c:2135 ast_rtp_read: 
RTP Read too short
[Apr 23 21:14:24] WARNING[14165]: res_rtp_asterisk.c:2135 ast_rtp_read: 
RTP Read too short
[Apr 23 21:14:24] WARNING[14165]: res_rtp_asterisk.c:2135 ast_rtp_read: 
RTP Read too short
[Apr 23 21:14:24] WARNING[14165]: res_rtp_asterisk.c:2135 ast_rtp_read: 
RTP Read too short
[Apr 23 21:14:24] WARNING[14165]: res_rtp_asterisk.c:2135 ast_rtp_read: 
RTP Read too short




Console output of Asterisk 10.3.0:


Connected to Asterisk 10.3.0 currently running on asterisk-dev (pid = 14798)
Verbosity is at least 3
Core debug is at least 3
-- Accepting AUTHENTICATED call from 192.168.54.12:
 requested format = slin,
 requested prefs = (),
 actual format = slin,
 host prefs = (slin|alaw|ulaw),
 priority = mine
-- Executing [@FAX-T30:1] Set(IAX2/iaxmodem1-863, 
FAXOPT(t38gateway)=yes) in new stack
-- Executing [@FAX-T30:2] Dial(IAX2/iaxmodem1-863, 
SIP/@itsp-fax,55) in new stack

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called SIP/@itsp-fax
-- SIP/itsp-fax- is making progress passing it to 
IAX2/iaxmodem1-863

-- SIP/itsp-fax- answered IAX2/iaxmodem1-863
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
[Apr 23 21:25:15] ERROR[14983]: astobj2.c:110 INTERNAL_OBJ: user_data

Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)

[Larry Moore]

The send log you have posted does not show any outgoing T.38 packets 
from your system.


I set up a test build of 1.8.11.0 using the patch recently released, I 
have difficulties sending T.38 with this patch, in fact I cannot send 
successfully however I can receive. I did however observe some outgoing 
T.38 packets. The analogue fax modem I was dialling into is under my 
control hence the log files showed there was no signalling coming from 
my ITSP,


The T.38 session on my Asterisk server show the CRP's which were sent 
from the analogue fax device during the negotiation.


The patch I have used for a while seems to give me outgoing 
functionality as well as incoming.


I can't reproduce your scenario whereby your T.38 session is 
communicating with a different gateway to the SIP server you use hence 
can only speculate that Asterisk has difficulty wanting to send T.38 SDP 
traffic when it is a different device than the SIP server it negotiates 
with.


Cheers,

Larry.

On 17/04/2012 6:47 PM, Niccolò Belli wrote:

Il 17/04/2012 01:10, Niccolò Belli ha scritto:

Tomorrow I will try without directmedia=yes.


Unfortunately it didn't help.

Niccolò

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Re: [asterisk-users] 10.3 : sip loses registration ?

[Larry Moore]

I have experienced this issue with a provider with Asterisk 1.2, 1.6  1.8.

I never got to the root cause of the problem however it used to occur 
quite frequently, now it appear to occur once every month or two - 
haven't seen it occur for a while now but then I have been incrementally 
updating my version of asterisk, currently 1.8.11.0


The preceding events I observed was that there would be a timeout 
communicating with the peer followed by retry attempts and finally a 
message reporting Wrong password, this is the point at which 
registration attempts stopped despite the value in sip.conf being set to 0.


As per your observations a 'sip reload' gets things going again.

When the problem was occurring within a 24-hour period I set up an 
SPA-942 phone to register to the service and captured packets between 
them, I don't recall seeing any issues over a period of a few days with 
the SPA phone hence was baffled by this phenomenon and have been since.


I was considering writing a script to check for the No Authentications 
status and to then issue a 'sip reload' but as the problem is rarely 
seen now I haven't had to do this.


My suspicion to the cause of the problem is that the authentication 
database at the VSP may have been offline momentarily hence why the 
response of a wrong password, I wasn't convinced of this as the packet 
capture of the SPA-942 did not reveal any authentication errors.


Cheers,

Larry.



On 16/04/2012 10:26 PM, sean darcy wrote:
We found this morning we had no SIP connection to another site. sip 
show registry on the main site gave no authentication. sip show 
peers on the other site showed the peer unspecified.


The odd part about this:  doing sip reload on the main site made it 
all work. Nothing else was changed.


Main site:

 sip show registry

SFO:5060  N  sip_outgoin   105 No 
AuthenticationSat, 14 Apr 2012 14:48:15

4 SIP registrations.
..

PBX*CLI sip reload
 Reloading SIP
  == Parsing '/etc/asterisk/sip.conf':   == Found
  == Parsing '/etc/asterisk/users.conf':   == Found
  == Using SIP TOS bits 96
  == Using SIP CoS mark 3
  == Parsing '/etc/asterisk/sip_notify.conf':   == Found
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5

On other site:

sip show peers
.
sip_outgoing/s   (Unspecified)D   
N 0Unmonitored

..
13 sip peers [Monitored: 3 online, 7 offline Unmonitored: 2 online, 1 
offline]


result of sip reload on main site:
-- Registered SIP 'sip_outgoing' at xxx.yyy.zzz.aaa:9345


How do I/can I set the main site to retry a registration? I've now 
changed sip.conf to add:


registertimeout=20
registerattempts=0;Default is 0 tries, continue forever

But these are the defaults anyhow!

Thanks,

sean


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Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)

[Larry Moore]

I applied the patch to my 1.8.11.0 build and observed the same error as 
shown in you t38_send.log.


I have maintained a private patch file for this functionality and 
reverted to it when I too observed the INTERNAL_OBJ: user_data is NULL 
message.


Do you have directmedia=no in your SIP configuration?

Cheers,

Larry.


On 14/04/2012 8:33 PM, Niccolò Belli wrote:

Il 04/04/2012 07:45, Anton Kvashenkin ha scritto:

Check it out, thank you.


You're welcome.

New packages against dahdi-linux-2.6.0, dahdi-tools-2.6.0, libpri 
1.4.12+svn20120409 and spandsp-0.0.6~pre20:
http://www.linuxsystems.it/2012/04/asterisk-1-8-11-0-debian-squeeze-packages-with-t-38-gateway-queue-hints-and-fixed-rfc4235/ 



If someone can help me there is a bug with T38 gw and eutelia: 
http://lists.digium.com/pipermail/asterisk-dev/2012-April/054681.html


Asterisk 10.4-rc1 does work, so it should be a matter of identifying 
the problem and backporting the fix.


Thanks,
Niccolò

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Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)

[Larry Moore]

Perhaps your problem may be that Asterisk doesn't like to send T.38 to a 
peer other than the one it negotiates the SIP connection with.


If I recall correctly you mentioned a while back that eutelia made a 
change which broke your outgoing T.38 functionality, did you ever find 
out what the change was?


Larry.

On 17/04/2012 4:58 AM, Niccolò Belli wrote:

Hi,

Il 16/04/2012 22:50, Larry Moore ha scritto:

Do you have directmedia=no in your SIP configuration?


Yes I have.

Niccolò

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Re: [asterisk-users] Problem with ReceiveFax

[Larry Moore]

On 13/03/2012 8:10 PM, Ishfaq Malik wrote:

On Tue, 2012-03-13 at 00:10 +0800, Larry Moore wrote:

On 12/03/2012 10:53 PM, Ishfaq Malik wrote:

Thanks for the input so far. I'm going to keep plugging away and if
anyone has any insights, they will be gladly appreciated. Ish

In SIP Account Configuration on Draytek;

Set Voice Active Detect to Off

In Phone Settings on the Draytek;

Enable Symmetric RTP
Check Start  End RTP Ports match values set in /etc/asterisk/udptl.conf
for udptlstart  udptlend

In /etc/asterisk/udptl.conf set;

use_even_ports=yes


Thanks for the above, I was hoping to have replied earlier with a
success message buy alas, no joy to be had.

Could I be having some sort of DTMF issue? I noticed this in amongst the
console output once I set the console logging level to include dtmf

[2012-03-13 12:06:39] DTMF[24784]: channel.c:3976 __ast_read: DTMF end 'f' 
received on SIP/588-000c, duration 0 ms
[2012-03-13 12:06:39] DTMF[24784]: channel.c:4002 __ast_read: DTMF begin 
emulation of 'f' with duration 100 queued on SIP/588-000c
[2012-03-13 12:06:39] DTMF[24784]: channel.c:4138 __ast_read: DTMF end 
emulation of 'f' queued on SIP/588-000c

does the above look correct for an inbound fax?

Thanks in advance (again!)

Ish


It's now time to do some debugging.

I would suggest you capture packets between asterisk and peer 588 using 
tcpdump, make sure you enable a large enough snaplen (-s) to ensure you 
capture all packets in the frame.


Submit your fax and upon completion of the session whether or not it is 
received successfully, transfer the file where you can open the captured 
file in Wireshark and select VoIP Calls located in the Telephony menu. 
You can then select the relevant line or lines in the session and click 
on the Flow button and review what is happening.


I have a Grandstream HT-503 at the other end of an IPSEC vpn which has 
the FXO port connected to a PSTN line.


I have configured the HT-503 to call the fax extension in the dialplan 
when it answers a call hence I have disabled faxdetect in the peer 
configuration.


Looking at the Draytek manual I think this would be setup in VoIP  
Phone Settings by enabling Call Forwarding and setting it to Always 
and defining the SIP URL as fax@astersk_server_ip, assuming you have a 
fax extension enabled in the context of the peer. I am assuming you 
currently have this set to 200@astersk_server_ip.


Did you disable VAD on the Draytek.

I would also suggest you disable Call Waiting  Call Transfer.

You may also want to look at Volume Gain in case that affects the 
level of the signal being converted to T.38 on the Draytek. Testing by 
progressively decreasing the level and if that doesn't help then 
increasing it.


Here is the peer configuration I just tested with my HT-503.

T.38 is enabled in the [general] section of sip.conf

[0123456789]
type=peer
defaultuser=0123456789
secret=you_guessed_it
call-limit=2
host=dynamic
disallow=g722
g726nonstandard=yes ;(this is required for Sipura and 
Grandstream ATAs, among others).

transport=udp,tcp
encryption=no
directmedia=no
faxdetect=no
context=Fax-Test
qualify=yes


Good luck.

Larry.

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Re: [asterisk-users] Transfer to fax

[Larry Moore]

On 14/03/2012 5:18 AM, Mike Diehl wrote:

So I'm still trying to get this to work... (I'm top posting, but the details
are below, if you want/need background info)

I'd like Asterisk to detect incoming faxes and redirect them elsewhere.  The
details aren't important, as long as I get the detection working.

I've added this to my sip.conf file.  Probably overkill, but I'll tune it once
it works:

[general]
faxdetect=both

My sip registrations are all in a Mysql RT database, so I added this column to
my table:

faxdetect char(3) default 'no'

I've set faxdetect to 'yes' for the devices that I expect to be receiving fax
calls.

I did a sip reload from the console after adding and updating this column.

I've added a fax extension to the appropriate context in extensions.conf:
exten =  fax,1,noop(I hear a fax!)

Since most of my dialplan is in an AGI script, I've added this to the code
that handles my test number:

$main::agi-answer();
$main::agi-exec(ringing);
$main::agi-exec(wait,5);


So, now that all of this is in place, I call the extension from my fax
machine... and I don't get any indication on the console that Asterisk heard a
fax.  My extension simply rings and I answer it.

What am  missing?



In your peer config set directmedia=no and faxdetect=cng, Asterisk needs 
to be in the path to hear the CNG tones.


Larry.

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Re: [asterisk-users] Problem with ReceiveFax

[Larry Moore]

On 12/03/2012 5:27 PM, Ishfaq Malik wrote:

On Fri, 2012-03-02 at 15:32 +, Ishfaq Malik wrote:

I've tried this with the f option on receiveFax but it still isn't
working. Any insight would be helpful as this is driving me a bit potty

   == Using UDPTL CoS mark 5
   == Using SIP RTP CoS mark 5
 -- Executing [200@local:1] Goto(SIP/588-, fax-in,s,1)
 -- Goto (fax-in,s,1)
 -- Executing [s@fax-in:1] Answer(SIP/588-, )
 -- Executing [s@fax-in:2] Wait(SIP/588-, 3)
 -- Executing [s@fax-in:3] Set(SIP/588-, 
FAXFILE=/tmp/fax-588-20120312-092231.tiff)
 -- Executing [s@fax-in:4] ReceiveFAX(SIP/588-, 
/tmp/fax-588-20120312-092231.tiff,f)
 -- Channel 'SIP/588-' receiving FAX 
'/tmp/fax-588-20120312-092231.tiff'
[2012-03-12 09:22:34] WARNING[4529]: res_fax.c:1508 receivefax_t38_init: 
channel 'SIP/588-' refused to negotiate T.38
 -- Channel 'SIP/588-' FAX session '0' started
[2012-03-12 09:22:34] WARNING[3763]: chan_sip.c:8740 process_sdp: Unsupported 
SDP media type in offer: audio 0 RTP/AVP 8
[2012-03-12 09:22:34] WARNING[3763]: chan_sip.c:8827 process_sdp: Failing due 
to no acceptable offer found
[2012-03-12 09:22:44] ERROR[4529]: res_fax.c:1344 generic_fax_exec: channel 
'SIP/588-' FAX session '0' failure, reason: 'fax session timed-out' 
(TIMEOUT)
 -- Executing [s@fax-in:5] Hangup(SIP/588-, )
   == Spawn extension (fax-in, s, 5) exited non-zero on 'SIP/588-'
 -- Executing [h@fax-in:1] NoOP(SIP/588-, ###   FAXSTATUS: 
FAILED)
 -- Executing [h@fax-in:2] NoOP(SIP/588-, ###FAXERROR: 
TIMEOUT)
 -- Executing [h@fax-in:3] NoOP(SIP/588-, ### FAXMODE: )
 -- Executing [h@fax-in:4] NoOP(SIP/588-, ###FAXPAGES: 
0)
 -- Executing [h@fax-in:5] NoOP(SIP/588-, ###  FAXBITRATE: )
 -- Executing [h@fax-in:6] NoOP(SIP/588-, ###   FAXRESOLUTION: )
 -- Executing [h@fax-in:7] NoOP(SIP/588-, ### REMOTESTATIONID: )
 -- Executing [h@fax-in:8] System(SIP/588-, mail -s FaxToEmail 
i...@-net.co.uk  /tmp/fax-588-20120312-092231.tiff)
 -- FAX handle 0: [ 040.001588 ], entering CLOSING state
 -- Channel 'SIP/588-' FAX session '0' is complete, result: 
'FAILED' (FAX_NO_FAX), error: 'T1_TIMEOUT', pages: 0, resolution: 'unknown', 
transfer rate: '2400', remoteSID: ''

Thanks in Advance



Looking at the information you have sent in this posting in certainly 
appears that the 'f' option has indeed helped however you have another 
matter to overcome.


You may wish to set the following parameters in your peer configuration 
for 588.


ignoresdpversion=yes
directmedia=no

I use Spandsp FAX successfully.

I have also attached an analogue Fax Modem to the FXS port on an SPA8800 
and an HT-502 and have been able to receive faxes on them when I last 
tested, the SPA8800 like the HT-502 are now in storage.


Looking at the User Guide for the Vigor 2701 there is an option in the 
configuration to enable T.38 mode, did you enable it?



In my sip.conf I have the following;

[general]
.
.
.
faxdetect=cng
t38pt_udptl=yes,redundancy,maxdatagram=400
;t38pt_usertpsource=yes
.
.
.
[903]
; Cisco SPA8800 FXS Port 3
; Grandstream HT502 FXS Port 1
; Analogue FAX Modem attached
type=friend
defaultuser=903
secret=you_guessed_it
call-limit=2
disallow=g722
transport=udp
qualify=yes
canreinvite=no
directmedia=no
host=dynamic
context=FAX-T38
faxdetect=no



Larry.

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Re: [asterisk-users] Problem with ReceiveFax

[Larry Moore]

On 12/03/2012 10:53 PM, Ishfaq Malik wrote:
Thanks for the input so far. I'm going to keep plugging away and if 
anyone has any insights, they will be gladly appreciated. Ish 


In SIP Account Configuration on Draytek;

Set Voice Active Detect to Off

In Phone Settings on the Draytek;

Enable Symmetric RTP
Check Start  End RTP Ports match values set in /etc/asterisk/udptl.conf 
for udptlstart  udptlend


In /etc/asterisk/udptl.conf set;

use_even_ports=yes



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Re: [asterisk-users] Asterisk Version 1.8.9.2 Question About SIP/SRTP/TLS

[Larry Moore]

On 28/02/2012 7:58 PM, DHAVAL INDRODIYA wrote:

Hi All,

I have one question that if my device is registered over TLS on 
asterisk .


is it required that it can only use SRTP for making an outbound calls 
or incoming calls too.




No.


how we can disable srtp and only enable TLS.



tlsenable=yes

Make sure you have your certificates setup prior to enabling this.

AFAIK Encryption of RTP  isn't on by default, to disable it in your peer 
configurations use


encryption=no

is there any dial-plan functions that can help to disable/enable this 
SRTP.




Not that I am aware of.


I want following settings.

 TLS  UDP
USERAGENT === ASTERISK = VoIpProvider




TLS is used for SIP signalling, what you do with RTP is up to you.

I have phones configured to use TLS and I have enforced RTP encryption 
i.e. SRTP by using encryption=yes in the peers configuration an setting 
the corresponding setting on the phone.


When an outgoing call is made to my ITSP the communications between the 
phone and Asterisk is all encrypted, the communications with my ITSP are 
all un-encrypted.


I have also used encryption=allow, this permits the administrator of the 
UA to decide if it should use SRTP or otherwise traditional RTP is used.



So is it possible with asterisk.



Yes!

Was that one question!?

Larry.

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Re: [asterisk-users] Asterisk 1.8.7 and ReceiveFAX

[Larry Moore]

On 9/10/2011 1:29 AM, Administrator TOOTAI wrote:

Le 07/10/2011 16:32, Kristijan Vrban a écrit :

remove the c argument


Done but now I have

[Oct  8 19:20:20] WARNING[8771]: res_fax.c:1508 receivefax_t38_init: 
channel 'SIP/tootaiAUDIO-00ea' refused to negotiate T.38
[Oct  8 19:20:20] WARNING[8771]: res_fax.c:1529 receivefax_t38_init: 
Audio FAX not allowed on channel 'SIP/tootaiAUDIO-00ea' and T.38 
negotiation failed; aborting.
[Oct  8 19:20:20] ERROR[8771]: res_fax.c:1734 receivefax_exec: error 
initializing channel 'SIP/tootaiAUDIO-00ea' in T.38 mode


How can I allow Audio FAX?

I saw a discussion on asterisk-devel from january 2010 about new 
spandsp where Kevin P. Fleming told you to do an core show 
application ReceiveFAX to find out how to enable this feature. I'm 
perhaps a little bit stupid but can't find any usable information 
while using this command :-(




The Fallback option to T.30 is 'f'.

ReceiveFAX(filename,f)

See 
https://wiki.asterisk.org/wiki/display/AST/Application_ReceiveFAX+%28res_fax%29


Larry.



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Re: [asterisk-users] T.38 client for Linux?

[Larry Moore]

On 22/09/2011 4:12 AM, Ian Pilcher wrote:

I am looking for a simple way to send occasional faxes via the FXO
port on my SPA3102 -- without having to connect a fax modem to an
ATA.  In an ideal world, this would be some sort of softfax that
runs on my Linux desktop and talks (via Asterisk) to the SPA3102 with
T.38.




The SPA3102 doesn't use T.38 on the FXO port.

You can do it by using HylaFAX with an iaxmodem and a HylaFAX client on 
your desktop.


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Re: [asterisk-users] Cisco SPA 941 and auto-answer with SIPheader Call-Info

[Larry Moore]

On 5/09/2011 4:27 PM, Jonas Kellens wrote:

Hello,

I'm trying to page the Cisco SPA 941 by adding the SIP-header 
Call-Info: answer-after=0


dialplan :

exten = _*XX*,n,SIPAddHeader(Call-Info: answer-after=0)



Try

exten = _*XX*,n,SIPAddHeader(Call-Info:\;Answer-After=0)

Larry.
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Re: [asterisk-users] problems with hylafax + iaxmodem +asterisk1.8.5

[Larry Moore]

On 5/09/2011 10:05 PM, Alessio wrote:

someone can help me to solve this problem?

thanks

--
From: Alessio ales...@asistar.it
Sent: Friday, September 02, 2011 5:10 PM
To: Lee Howard fax...@howardsilvan.com
Cc: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] problems with hylafax + iaxmodem 
+asterisk1.8.5



1: from the phone i called  the fax-server
2: from external fax i tried to send a fax to fax-server

the results:
_


G'Day Alessio,

I replied to your original post suggesting you set up two IAX modems and 
get successful transmission working between them.


I suspect you want to use T.38 with IAX modem, I don't believe the IAX2 
channel supports T.38 hence I would suggest you remove the t38pt_udptl 
lines from your iax.conf files to avoid confusion.


I am assuming you are receiving your incoming facsimile using SIP, if so 
I would suggest you have only one reference to t38pt_udptl in that peers 
configuration and set it to no.


Depending on whether the peer is dedicated to receiving facsimiles I 
would suggest you also include in your peer's configuration faxdetect=no 
otherwise if this is an Audio/FAX line I would suggest you set it to 
faxdetect=cng.


Once you have this working but really want to use T.38 then you will 
need to apply the T.38 Gateway patch to your 1.8.5.0 build, see 
https://issues.asterisk.org/view.php?id=13405 .


Changes you will need to make to your SIP peer is to set t38pt_udptl=yes 
and in your dial plan before the Dial() enable the gateway with 
Set(FAXOPT(t38gateway)=yes).


Larry.

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Re: [asterisk-users] problems with hylafax + iaxmodem + asterisk1.8.5

[Larry Moore]

On 2/09/2011 12:13 AM, Alessio wrote:

Hi!
from 2 days I'm trying to run hylafax server and iaxmodem with 
Asterisk 1.8.5.
I have 2 computers in the lan, one is the Asterisk PBX and the other 
is the server with hylafax and iaxmodem installed.

In Asterisk I set up an IAX trunk in this way:


I would suggest you set up at least two IAX modems and confirm you can 
send a fax out through one and receive it on the other.


Once this is working you can look at the next stage of accepting a call 
through another channel such as SIP we can resolve the configuration you 
will need for it to work.


Larry.
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Re: [asterisk-users] Asterisk 1.8.3.3 T.38 Gateway

[Larry Moore]

On 1/09/2011 7:04 PM, Tim King wrote:
I have found numerous claims that 1.8 can do T.38 gateway with a 
patch, however I am yet to find the patch or any instructions on 
implementing it. Anyone have a link?


https://issues.asterisk.org/view.php?id=13405

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Re: [asterisk-users] Wanted a modified SIP message body

[Larry Moore]

On 31/08/2011 11:23 PM, Olle E. Johansson wrote:

31 aug 2011 kl. 14:42 skrev Kevin P. Fleming:


On 08/31/2011 02:46 AM, Jaime Lozano wrote:

Hello,
I agree with you, I'm not explaining the problem in a proper manner,
because of my lack of Asterisk knowings. I send the Wireshark captures.

3com telephones take the timezone TZ:7200 from the 3Com PBX to show the
time right. But what if I want a 3Com telephone to work with Asterisk
PBX? Then the telephone time is wrong, 2 hours lower. It seems 3Com
telephones need the TZ:7200. 3Com telephones work with Asterisk and it
is great, but we would like to log the calls.

OK, so the first clarification is that you are talking about responses to 
REGISTER requests specifically, not all responses to all requests. That's good 
:-)

On to the meat of the issue... indeed, the '200 OK' response to a REGISTER 
request does not normally have a message body; nothing in the SIP RFCs even 
suggests that there would be one (although it's certainly allowed should the 
registrar want to include it) or what would be present in it.

As has been previously replied here, there is no facility in Asterisk to 
include a message body in a REGISTER request response, so providing one will 
definitely require source code modifications. They wouldn't be terribly 
difficult, but they would only be applicable to these particular phones, which 
reduces the benefit of making the changes to the community at large.

With that said... it's certainly possible to do this, but it's going to take 
some non-trivial code changes. The REGISTER handling code does not use any of 
the methods that exist in chan_sip to add message body content to its 
responses, it uses simpler methods that assume there won't be a message body.

In addition, this mechanism is really pretty broken anyway; the server would 
have to know where each phone is physically located in order to be able to 
provide the correct TZ value to it, and would have to be updated if the phone 
is moved. Not an ideal situation.

The RFC states that a phone could use the Date: header in the response to set 
the local time in the device. It's always in GMT which makes it stupid to add a 
time zone any where.

-1 for this implementation.


Perhaps researching the specs./capabilities of the phone for other 
capabilities setting its time zone. A DHCP server can offer a 
time-offset value, whether the phone can be provisioned with a defined  
time zone offset or accept the offset in DHCP is a matter of further 
research.


Larry.

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Re: [asterisk-users] [OT] Yealink T26/28/38 and Open-VPN

[Larry Moore]

On 24/08/2011 10:17 PM, --[ UxBoD ]-- wrote:

Hi,

Sorry for an OT post but striking out a bit at the moment trying to 
get a response from Yealink RD.  Has anybody successfully managed to 
get a Yealink phone to work across Open-VPN when using tlsauth ?  We 
really do hope that it is possible due to the benefits tlsauth offers 
against DoS.




I have used a Yealink T22 accross an IPSEC VPN  using TLS Auth however I 
have since configured it to connect directly via the Internet.


I have been keeping the devices firmware updated as they are released.

My two-bobs worth!

Larry.
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Re: [asterisk-users] T38 Fax

[Larry Moore]

On 2/08/2011 1:02 AM, Robert Huddleston wrote:


Anyone have any testing experience with T38 and HT-502 Grandstream?

I just want to confirm that t.38 is working on this device.

Thanks




Yes, it works.

I currently have latest firmware installed and it still works in T.38. I 
am using UDP transport for this device as I seem to encounter problems 
with TCP or TLS.


I am currently running Asterisk 1.8.5.0.

*Product Model: * HT-502 V1.1C
*Software Version: * 	  Program-- 1.0.5.5Bootloader-- 1.0.0.9 
   Core-- 1.0.5.2Base-- 1.0.5.2



Some settings I have set and you may wish to check for the FXS port are;

/Force INVITE: /No Yes (Always refresh with INVITE instead of UPDATE)
/Send Re-INVITE After Fax: /No Yes


/VAD: / No Yes
/Symmetric RTP: /   No Yes
/Fax mode: /T.38 (Auto Detect) Pass-Through
/Fax tone detection mode: / Caller Callee Caller or Callee
/Jitter buffer type: /  Fixed Adaptive
/Jitter buffer length: /Low Medium High



You will need to ensure you are using redundancy mode instead of FEC.

I am able to send a fax via my voice provider seemingly without errors 
even though ECM is not enabled, this is because redundancy mode is 
working as expected on the outbound communication.


Unfortunately my voice provider only sends one data item in the incoming 
UDPTL hence the occasional missed line.


Here is an extract from my sip.conf

[general]
.
.
t38pt_udptl=yes,redundancy,maxdatagram=400
.
.
[906]
; Grandstream HT502 FXS Port
; Analogue FAX Modem attached
type=friend
defaultuser=906
md5secret=c5bca943c9b0cc303c496fbf9d48a48e
call-limit=1
disallow=g722
transport=udp
qualify=yes
directmedia=no
host=dynamic
context=FAX-T38
faxdetect=no
deny=0.0.0.0/0.0.0.0
permit=10.0.0.0/255.0.0.0
permit=172.16.0.0/255.240.0.0
permit=192.168.0.0/255.255.0.0

Larry.
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Re: [asterisk-users] T38 Fax

[Larry Moore]

On 2/08/2011 4:13 AM, Robert Huddleston wrote:


Thanks -- and did you find a provider with T.38 DIDs? I don't see many 
pay as you go providers with T.38





I am not looking for VoIP providers for such functionality and is 
subjective to geographic location, however I am of the opinion that one 
should have a PSTN line connected directly to a fax device at CPE for 
receiving said communications and one could use T.38 for Outbound faxing 
providing the transmissions are of high enough quality


Larry..


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Re: [asterisk-users] FAX with SIP

[Larry Moore]

On 22/07/2011 5:43 AM, Israel Gottlieb wrote:



On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming 
kpflem...@digium.com mailto:kpflem...@digium.com wrote:


On 07/21/2011 04:34 PM, Joaquin Sosa wrote:

On Mon, Jul 18, 2011 at 07:58, Steve
Daviesdavies...@gmail.com mailto:davies...@gmail.com  wrote:

The magic sauce that you need is T.38 - Asterisk 1.6
supports this
to a limited degree, and your ITSP will need to support it.

The sip.conf.sample file and the voip-info wiki has all the
information you need to try it out.


Correct. However it would be helpful to note T.38 support in
Asterisk
is broken. Take a T.38 ITSP, T.38 enabled Asterisk and T.38
ATA and
try to send a fax. It won't work!


We do this in our testing all the time, and it works fine. Since
you didn't specify any particular version of Asterisk, there's no
way to associate your It won't work statement with anything in
particular. Given the variations of T.38 implementations that
exist in ATAs, carrier networks and other places, *any* T.38
connection that involves implementations from more than one vendor
is (unfortunately) likely to have problems, whether any version of
Asterisk is involved or not



well I tried  a linksys spa 8000 and 2102 thru
asterisk 1.8.3
1.8.4
1.6.2.16-19
sonus switch at itsp (012 israel)

and no luck



Cisco released updated firmware earlier this year for the SPA8000  
SPA2102 which addresses T.38 problems.


I have an SPA8800 which I was able to use T.38 mode to send faxes 
successfully, I recently updated Asterisk box and also updated to 1.8.5, 
haven't tested the SPA8800 with this config but I am expecting it will 
still work. The key to my success was to ensure the SPA8800 did not do a 
re-invite to the ISP for the RTP stream.


Larry.

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Re: [asterisk-users] No audio after a reinvite changing codec ---- with SIP DEBUG!!

[Larry Moore]

On 28/06/2011 6:59 PM, Matteo Campana wrote:



Hi Larry,
I have the SIP debug taken from asterisk.
In this debug: 1.2.3.4 --- IP SIP PROXY
 5.6.7.8 --- IP UAC (Linksys SPA 962)
 9.10.11.12 --- IP ASTERISK to connect to the 
provider

 13.14.15.16 -- IP PROVIDER
 17.18.19.20 -- IP ASTERISK


The SIP debug is available at this link: http://pastebin.com/9DrFDmeC




You mention you have an SPA962, I expect the configuration will be the 
same if not similar to an SPA942. It would be worth checking what your 
Symmetric RTP setting is, you can find it listed in the RTP Parameters 
section under the SIP section of your phone 
http://ip_address_of_spa962/admin/advanced.


If it is set to no set it to yes.

Larry.
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Re: [asterisk-users] dialplan execution stops after ReceiveFax

[Larry Moore]

On 29/06/2011 5:13 PM, Ruben Rögels wrote:

Hello,

I have a noticed strange behavior in Asterisk 1.6.18.2 with ReceiveFax
Digium FAX Driver: 1.6.2.0_1.3.0 (optimized for i686_32).

I use a context [capi-in] for icoming ISDN calls:


==
[capi-in]
; Faxe fuer Ruben
exten =  12345,1,Macro(faxin,ruben.roeg...@jumping-frog.org,${EXTEN})
==

My macro for the fax receiving looks like that:

==
[macro-faxin]
; Faxe
; ARG1 = eMail-Adresse
exten =  s,1,Verbose(${BOUNDARY} Eingehender Ruf von ${CALLERID(num)})
exten =  s,n,Verbose(${BOUNDARY} BCHANNELINFO ${BCHANNELINFO})
; nur verarbeiten, wenn B-Kanal frei ist
exten =  s,n,GotoIf($[${BCHANNELINFO} = 2]?hangup:free)
exten =  s,n(free),NoOp()
exten =  s,n,Set(TO=${ARG1})
exten =  s,n,Verbose(1,${BOUNDARY} Eingehendes Fax ${CDR(uniqueid)})
exten =  s,n,Set(FAXFILE=/var/spool/fax/fax-${TO}-${CDR(uniqueid)}.tif)
exten =  s,n,Set(LOCALSTATIONID=jumping frog)
exten =  s,n,Answer()
exten =  s,n,Wait(3)
exten =  s,n,ReceiveFAX(${FAXFILE},d)
exten =  s,n,Verbose(1,${BOUNDARY} Nach dem Fax!)
exten =  s,n,System(/usr/local/bin/fax2mail.sh ${FAXFILE} ${TO})
;exten =  s,n,capicommand(receivefax,${FAXFILE},+00497613821,Headline,k)
exten =  s,n(hangup),HangUp()
exten =  h,1,System(/usr/local/bin/fax2mail.sh ${FAXFILE} ${TO})
==

As you can see, the received fax file should be processed by a
bash-script, but after the call hangs up, the script is never executed.

The console log shows:

==
  -- Channel 'CAPI/ISDN1#02/3821-5' FAX session '4' is complete,
result: 'SUCCESS' (FAX_SUCCESS), error: 'NO_ERROR', pages: 1,
resolution: '204x98', transfer rate: '9600', remoteSID: '4932123847885'
   == Spawn extension (macro-faxin, s, 11) exited non-zero on
'CAPI/ISDN1#02/12345-5' in macro 'faxin'
   == Spawn extension (capi-in, 12345, 1) exited non-zero on
'CAPI/ISDN1#02/12345-5'
   == ISDN1#02: CAPI Hangingup for PLCI=0x101 in state 2
  ISDN1#02: CAPI INFO 0x3490: Normal call clearing
==

Anyone seeing what I'm missing?


Hi Ruben,

You should be looking at this thread 
http://lists.digium.com/pipermail/asterisk-users/2011-June/263995.html


Presently I don't have the time to generate and send logs however soon 
after my last post I did perform additional testing.


I am using ReceiveFAX using SPANDSP technology.

The occasions the System() call would not be executed, whether it was in 
'h' of the dialplan  or the main part of the macro after ReceiveFAX(), 
was when a T.38 fax was being received, when it was a G.711 fax no 
matter what I did to the call it would always execute the System() call 
whether it was in the macro or the 'h'.


Cheers,

Larry.


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Re: [asterisk-users] Problem with ReceiveFAX app from FFA

[Larry Moore]

On 20/06/2011 8:18 AM, Steve Underwood wrote:

On 06/20/2011 03:38 AM, khalid touati wrote:

Hi Guys,
I solved temporarely my issue by kind of tricking Asterisk, I used 
the following line instead of the old:

exten = h,n,System('/usr/local/
bin/fax2mail -p -f ${FAXFILENOEXT} --cid-number ${CALLERID(num)} 
--cid-name ${CALLERID(name)} --dest-name Sir/Madam')
now when it hang up I receive my fax through email, and let me tell 
you (first time using Free Fax from Asterisk) ReceiveFAX catch well 
faxes, just a couple tries but got them all, let's see with more 
faxes what will happen.


Why do you consider this a temporary fix? The far end machine will 
normally hang up at the end of the FAX, so the hangup option in the 
dialplan is exactly where you should expect to be.


I don't know the specifics of how an Asterisk application should exit 
however WRT ReceiveFAX() using SPANDSP Technology I would expect the 
call to descend to any functions below ReceiveFAX() whether or not the 
facsimile was received successfully, the status codes from ReceiveFAX() 
can be used by whatever is called next, e.g. a script to e-mail the 
received facsimile or a report advising errors were encountered.


I am using a macro to receive faxes, I have placed my System() call back 
to were the macro returns after execution due to the function not being 
called after ReceiveFAX() under certain conditions.


This however does not guarantee getting an e-mail of what has been 
received if the sender decides to abort the transmission.


I can reproduce this using HylaFAX to send a fax to an extension which 
Asterisk ReceiveFAX(filename,f) will accept, granted it will fall back 
to G.711 mode when receiving, when I abort the transmission using WHFC 
client, it is as though ReceiveFAX() goes of somewhere else or simply 
decides to forget where it came from as it does not appear to return 
hence the System() call is never made.


I should point out I am using extensions.ael for my dialplan.

I personally have considered this behaviour to possibly be a bug.

Cheers,

Larry.

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