[asterisk-users] iaxmodem errors.

john atux at null.net
Wed Nov 16 03:26:57 CST 2016


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/Username    Host                 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 at 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 at PBX: /var/log/iaxmodem $



Any ideas?




On 15/11/2016 5:40 μμ, Larry Moore wrote:
> 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 <lmoore at 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 at from-itsp:1] NoOp("SIP/itsp-00000044",
>>         "Fax Detected 2016-11-13 12:33:40 +0800") in new stack
>>             -- Executing [fax at from-itsp:2]
>>         GotoIf("SIP/itsp-00000044", "0?3:8") in new stack
>>             -- Goto (from-itsp,fax,8)
>>             -- Executing [fax at from-itsp:8] NoOp("SIP/itsp-00000044",
>>         "Finish if_from-itsp_237") in new stack
>>             -- Executing [fax at from-itsp:9]
>>         GotoIf("SIP/itsp-00000044", "0?10:13") in new stack
>>             -- Goto (from-itsp,fax,13)
>>             -- Executing [fax at from-itsp:13] NoOp("SIP/itsp-00000044",
>>         "Finish if_from-itsp_238") in new stack
>>             -- Executing [fax at from-itsp:14] Set("SIP/itsp-00000044",
>>         "FAXOPT(gateway)=yes") in new stack
>>             -- Executing [fax at from-itsp:15]
>>         Queue("SIP/itsp-00000044", "hylafax-iax,dRt,,,15") in new stack
>>             -- Started music on hold, class 'silence', on
>>         SIP/itsp-00000044
>>             -- 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-00000044
>>             -- IAX2/iaxmodem2-3086 answered SIP/itsp-00000044
>>                > 0x89bac000 -- Probation passed - setting RTP source
>>         address to <ITSP IP Address>:18998
>>           == Using UDPTL TOS bits 184
>>             -- Executing [h at from-itsp:1] GotoIf("SIP/itsp-00000044",
>>         "0?2:3") in new stack
>>             -- Goto (from-itsp,h,3)
>>             -- Executing [h at from-itsp:3] NoOp("SIP/itsp-00000044",
>>         "Finish if_from-itsp_239") in new stack
>>             -- Executing [h at from-itsp:4] NoOp("SIP/itsp-00000044",
>>         "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-00000044'
>>
>>     I'm sure you've already checked and confirmed you have 'alaw' and
>>     'ulaw' codecs permitted in your IAX Modems, iax.conf and sip.conf
>>     configurations
>>
>>     To test your configuration you could set it up your environment
>>     so that you send an outgoing fax to yourself i.e. your dial your
>>     number at the VoIP provider, this assumes when you dial your VoIP
>>     number a connection is made back to you, you can then
>>     troubleshoot the communication.
>>
>>     This is how I performed the majority of my tests.
>>
>>     Not sure why you haven't explored the option of terminating a fax
>>     call in Asterisk, you will need some scripts to convert the
>>     received image to a PDF which is then e-mailed. An offer was made
>>     to you to provide scripts, if you set this up when your
>>     iaxmodem's aren't working a fallback will be for Asterisk to
>>     accept the call as it falls through, one thing you should know,
>>     if you use the T.38 Gateway in your dialplan you will need to
>>     disabled it prior to Asterisk terminating the call. I use
>>     extensions.ael so here is an example, I've included the macro I
>>     use to receive a fax in Asterisk:
>>
>>         context from-itsp {
>>
>>                 s => {
>>                         NoOp(Call Received ${STRFTIME(,,%F %T %z)});
>>                         Set(CHANNEL(language)=en_AU);
>>                         Set(DIALTIMEOUT=30);
>>                         Progress();
>>                         NoOp(Call Received from ${CALLERID(name)},
>>         Tel: ${CALLERID(num)});
>>                 .
>>                 . other conditions checked and extensions dialled
>>                 .
>>                 };
>>
>>                 fax => {
>>                         NoOp(Fax Detected ${STRFTIME(,,%F %T %z)});
>>                         Set(FAXOPT(gateway)=yes);
>>                         Queue(hylafax-iax,dRt,,,15);
>>
>>                         Set(FAXOPT(gateway)=no);
>>         &fax-receive(<TSID>,<Header>,FaxMaster,lmoore);
>>                         Hangup();
>>                 };
>>
>>                 h => {
>>                         if ( "X${FAXRXFILE}" != "X" )
>>                         {
>>                                 &email_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
>>             inhttp://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 at fax-in:1]
>>             Dial("SIP/mytrunk-00000001", "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-00000001'
>>             status is 'CHANUNAVAIL'
>>             RasPBX*CLI>
>>             the extensions.conf has
>>             [fax-in]
>>             exten => 00302112152130,1,Dial(IAX2/iaxmodem)
>>             any ideas, please?
>>
>>
>>
>>
>>
>>
>>     --
>>     _____________________________________________________________________
>>     -- Bandwidth and Colocation Provided
>>     by<http://www.api-digital.com>http://www.api-digital.com -- Check
>>     out the new Asterisk community forum
>>     at:https://community.asterisk.org/
>>     <https://community.asterisk.org/> New to Asterisk? Start
>>     here:https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>     <https://wiki.asterisk.org/wiki/display/AST/Getting+Started>
>>     asterisk-users mailing list To UNSUBSCRIBE or update options
>>     visit:http://lists.digium.com/mailman/listinfo/asterisk-users
>>     <http://lists.digium.com/mailman/listinfo/asterisk-users> 
>>
>>
>>
>
>
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20161116/546a52db/attachment.html>


More information about the asterisk-users mailing list