gmane.comp.telephony.yate

T.38 Transcoding

Philipp Hoffmann — 2013-05-17T20:12:45

Hi,

I am trying to realize T.38 transcoding with Yate:

A  (SIP)--> Yate (SIP) --> Carrier (SIP) --> B (ISDN)

A supports T.38 (but no g711), the Carrier does *not* support T.38. That means, that Yate needs to transcode T.38 to g711.

How can I configure this case in Yate?

We use YATE 2.2.0-1 (default debian package).

Regards,
Philipp.

German ISUP support

Philipp Hoffmann — 2013-05-15T18:32:42

Hi,

is there currently any SS7 stack with German ISUP dialect available for Yate?

Thanks,
Philipp.

Perform radius authentication lookups

Philipp Hoffmann — 2013-05-11T21:06:44

Hi,

I am trying to perform radius authentication requests using yradius.conf:

The radius server expect two standard attributes, which have to been set up by Yate: The "User-Name" (always the called party number, but without the first char, e. g. +49310 -> 49310) and the "User-Password" (always clear-text: "radius").

For reference:

--
[nas]
add:User-Name=${caller} ;;;;; question: how can we cut the first char (+)? ;;;;;
add:User-Password=radius
--

The query goes through the radius server, but Yate do not provide the "User-Password" attribute.

Trace:

--
<yradius:NOTE> Using sections [nas] and [radius common] for authentication
<yradius:GOON> Ignoring unknown attribute of type 1
--

In the source code, the "User-Password" attribute is definied as binary, while "User-Name" is defined as string. Is it possible, to set a binary (string) attribute in the yradius.conf file?

Thanks,
Philipp.

SS7 Localy blocked ciruit

Herman Bigos — 2013-05-09T17:16:12

Hello.
My config is YATE 4.3.1+ AS5400.

I've got problem with localy blocked circuits:
Circuits are blocked by Yate itselfs from unknown reson.


status sig isup1/97
%%+status:sig isup1/97
module=sig,trunk=isup1,type=ss7-isup;circuit=97,span=mg70+1,status=Disabled,lockedlocal=true,lockedremote=false,changing=false,flags=0x1
%%-status

control isup1/ISUP unblock circuit=97
Could not control isup1/ISUP unblock circuit=97

control isup1/ISUP unblock circuit=97 force=yes
Control 'isup1/ISUP' OK

--------------------------------------------- but circuit is still
blocked----------------------------------

status sig isup1/97
%%+status:sig isup1/97
module=sig,trunk=isup1,type=ss7-isup;circuit=97,span=mg70+1,status=Disabled,lockedlocal=true,lockedremote=false,changing=false,flags=0x1
%%-status



---------------------------------------------------------------------------------------------------------------------

So, anybody knows whats up ?

Regards
Hermozol

dtmf debug

2013-05-06T11:45:00

Hello

I have Cisco 5400 with SLT as PSTN to SIP GW. I have problems with  
DTMF that is received from PSTN site. When someone call my NGN number  
and press and hold dtmf tone for longer time, I receive it double.  
There is no problem for GN.
Only difference I see in IAM is that when someone call NGN ther is:

PropagationDelayCounter='126'
HopCounter='30'

when someone call GN number i see:

PropagationDelayCounter='0'
ParameterCompatInformation.PropagationDelayCounter='transit,nopass-param'
ParameterCompatInformation='31 c0'

Is there a way to check what goes as DTMF from PSTN site ? Or can  
PropagationDelayCounter cause this behaviour ?

Greetings
Andrzej

IVR with javascript

Mehdi Shirazi — 2013-05-03T12:12:42

Hi
Is it possible that someone kindly provides 
 sample IVR in javascript?

Thank you

RTP Packet Size in yate?

Bipin Patel — 2013-05-02T14:49:36

Problem with Callfork and RTP/DTMF in progressing state

Christian Beyerlein — 2013-04-30T10:31:58

Hi There!

We have a problem with relaying RFC2833 DTMF to forked calls in 
progressing state.

DTMF is received on the caller side and YRTPWrapper::gotDTMF queues a 
chan.dtmf message inside chan.masquerade.

The chan.dtmf message is not properly handled then by 
DTMFHandler::received, it cannot find the YRTPWrapper of progressing 
fork target, as it is searching for the targetid "fork/1234", but the 
YRTPWrapper of progressing target has an id of "sip/2345".

Any chance to have support for DTMF in forked progressing calls?

BR
   Christian

CDR Time issue

Egor — 2013-04-29T11:46:05

Hi, i am collecting cdrs to mysql accordint to this scheme:
http://yate.null.ro/pmwiki/index.php?n=Main.CollectingBillingInformation

All works fine, but time format...
I get this written in mysql

+------------+
| time       |
+------------+
| 1367245948 |
| 1367245948 |
+------------+

I try to edit cdrbuild.conf but with no positive result.
My cdrbuild.conf:

resolution=msec
updates=true
status=true
status_answer=true
call_start_time=YY/MM/DD HH:mm:SS.uuu UTC
call_answer_time=YY/MM/DD HH:mm:SS.uuu UTC
call_hangup_time=YY/MM/DD HH:mm:SS.uuu UTC
duration_call=HH:mm:SS.uuu

Can you tell me where is problem?
Yate runs at centos 5.8, yate from svn.
Thanks.


+----+------------+--------+--------+----------+----------+----------+
| id | time       | caller | called | billtime | duration | status   |
+----+------------+--------+--------+----------+----------+----------+
|  1 | 1367245948 | 200    | 100    |       10 |       12 | answered |
|  2 | 1367245948 | 200    | 100    |       10 |       12 | answered |
+----+------------+--------+--------+----------+----------+----------+

YATE with others PBX

Duany — 2013-04-26T15:02:39

Hi . I'm doing  interconnection YATE with others PBX IP for examples Asterisk. 
I would like know if exist any way to specific in YATE incoming calls only make 
a specific trunk.

Thanks



Ing. Duany Baró Menéndez
Jefe de Línea de Telecomunicaciones de Centro UCID
Teléfono: 837- 3607 
Correo: dbaro-NigQoAqfJP0< at >public.gmane.org

http://www.uci.cu

Yate as MGCP Media Gateway

Eugene Prokopiev — 2013-04-26T08:10:01

Hi,

Is it possible to configure Yate as MGCP Media Gateway? I see
/usr/lib/yate/server/mgcpgw.yate and /etc/yate/mgcpgw.conf files, but
I can't find key parameter in configutation file example. Is it
possible to configure it?

--
WBR,
Eugene Prokopiev

Fax problem

Egor — 2013-04-26T08:07:01

Hi, i am tryind to receive a fax with yate.
I use a script from 
https://github.com/shimaore/yate/blob/master/share/scripts/detector.php
I am iniciating a fax with pbxassist.
For example, in my regexroute are route
^964$=external/nodata/fax.php;called=${caller};
I use fax.php to iniciate fax receiving and detector.php
to write it into a file. The problem is, that in detector.php
filename consist of billid, called and called, but when it writes
a file it writes in form billid caller caller

For example, i make a call from extension 022768482 to 022244340
I make one change in detector.php i v added a print of $num varueble.

My pbxassist transfer trigger:
[transfer]
; blind transfer: make call on behalf of peer, hangup this
; key: *1nnnnn*
trigger=\*1\([0-9]\+\)\#$
target=\1
;trigger=*3\([0-9]\+\)\*$
;target=\1
First time script detects correctly, but after it detect both as caller.
I tryed regexroute with ^964$=external/nodata/fax.php
but it returns, that there is no called after making a transfer

<ExtModReceiver:WARN> Error: 'NUM 1366916013-1_022768482-022244340'
<pbxassist:CALL> Created assistant for 'sip/2'
<NOTE> Choosing started 'audio' format 'mulaw' [0x1959c1a0]
<sip:MILD> No formats for 'video', excluding from SDP [0x1958cff0]
<NOTE> Choosing started 'audio' format 'mulaw' [0x19591490]
<yrtp:WARN> Wrapper neither format nor payload specified [0x195ab360]
<pbxassist:NOTE> Chan 'sip/2' triggered operation 'transfer' in state 
'new' holding '(null)'
<ExtModReceiver:WARN> Error: 'NUM 1366916013-1_022768482-964'

And log with ^964$=external/nodata/fax.php;called=${caller};

<pbxassist:CALL> Created assistant for 'sip/9'
Starting detection on sip/9
<ExtModReceiver:WARN> Error: 'NUM 1366913760-5_022768482-022244340'
<pbxassist:NOTE> Channel 'sip/9' already assisted!
<pbxassist:CALL> Created assistant for 'sip/10'
<NOTE> Choosing started 'audio' format 'mulaw' [0x116aac50]
<sip:MILD> No formats for 'video', excluding from SDP [0x116ae040]
<NOTE> Choosing started 'audio' format 'mulaw' [0x116baa70]
<yrtp:WARN> Wrapper neither format nor payload specified [0x116b17b0]
<sip:NOTE> Registered user '022768482' expires in 120 s
<pbxassist:NOTE> Chan 'sip/10' triggered operation 'transfer' in state 
'new' holding '(null)'
Starting detection on sip/9
<ExtModReceiver:WARN> Error: 'NUM 1366913760-5_022768482-022768482'
PHP Installed: call.execute
PHP Answered: call.answered id: 1115985716517973d1de1b56.84525410
PHP: bye! Fax PHP

And here is fax.php

#!/usr/bin/php -q
<?php
/* Simple Fax2Mail Skript for the Yate PHP interface
    To test add in regexroute.conf

    ^NNN$=external/nodata/fax.php

    where NNN is the number you want to assign
*/
require_once("libyate.php");

/* Always the first action to do */
Yate::Init();
Yate::Debug(true);

date_default_timezone_set('Europe/Chisinau');

//$faxfile     = '/var/spool/fax/' . 'rx-file-' . date("Ymd_His") . '.tiff';
$faxfile     = '/var/spool/fax/rx-file-123.tiff';
$ourcallid   = 'fax/' . uniqid(rand(),1);
$partycallid = '';
$state       = 'call';

Yate::Install("call.execute",100);

/* The main loop. We pick events and handle them */
while ($state != "") {
     $ev=Yate::GetEvent();
     /* If Yate disconnected us then exit cleanly */
     if ($ev === false)
         break;
     /* Empty events are normal in non-blocking operation.
        This is an opportunity to do idle tasks and check timers */
     if ($ev === true)
         continue;
     /* If we reached here we should have a valid object */
     switch ($ev->type) {
         case "incoming":
             switch ($ev->name) {
                 case "call.execute":
                     $partycallid = $ev->params["id"];
                     $ev->params["targetid"] = $ourcallid;
                     $ev->handled = true;
                     // we must ACK this message before dispatching a 
call.answered
                     $ev->Acknowledge();
                     // we already ACKed this message
                     $ev = false;

                     $m = new Yate("call.answered");
                     $m->params["id"] = $ourcallid;
                     $m->params["targetid"] = $partycallid;
                     $m->Dispatch();

                     $m = new Yate("chan.attach");
                     $m->params["id"] = $ourcallid;
//                  $m->params["source"] = 
"wave/play//usr/share/yate/sounds/fax2mail/dialtone.sln";
                     //$m->params["source"] = 
"wave/play//usr/local/share/yate/sounds/ring.au";
                                         $m->params["source"] = 
"wave/play//home/rec.slin";

                     $m->id = ""; // don't notify about message result
                     $m->params["sniffer"] = "tone/*";
                     $m->params["fax_divert"] = 'fax/receive//' . $faxfile;
                                         //$m->params["fax_divert"] = 
'fax/receive/var/spool/fax/rx-file-123.tiff';
                     $m->Dispatch();
                     //Yate::Output("PHP Incoming: call.execute id: " . 
$ev->id);

                     break;
                 default:
                     Yate::Output("PHP Incoming: " . $ev->name . " id: " 
. $ev->id);
             }
             /* This is extremely important.
                We MUST let messages return, handled or not */
             if ($ev)
                 $ev->Acknowledge();
             break;
         case "answer":
             Yate::Output("PHP Answered: " . $ev->name . " id: " . $ev->id);
             break;
         case "installed":
             Yate::Output("PHP Installed: " . $ev->name);
             break;
         case "uninstalled":
             Yate::Output("PHP Uninstalled: " . $ev->name);
             break;
         default:
             Yate::Output("PHP Event: " . $ev->type . $ev->name);
     }
}

Yate::Output("PHP: bye! Fax PHP");

/* vi: set ts=8 sw=4 sts=4 noet: */
?>

Maximum number of extensions in Yate

gregory — 2013-04-25T10:31:43

Hi,

 

What is the (estimated) maximum number of extensions with which Yate will
work properly?

If there are so many extensions that Yate doesn't work properly, is it
possible to somehow use an additional Yate on another computer in order to
overcome this limitation?

 

Thanks

Yata on Windows with Sangoma card-problem

H Yavari — 2013-04-25T04:57:35

Hi to all Yaters,

I download yate 4.3 .exe and installed it on windows 7. I have a A102 card on my computer
and installed driver version 7.0.1.0 for that.
I configure wpcard.conf and ysigchan.conf to have a PRI trunk.When I run yate, I noticed that wanpipe module not loaded and in signalling channel module happen this :
<MILD> Factory could not create 'SignallingCircuitSpan' named 'trunk1/B/wanpipe1'
<sig:NOTE> Trunk('trunk1'). Create failure: Failed to build voice span 'wanpipe1'
<sig:WARN> Failed to initialize trunk 'trunk1' of type 'isdn-pri-net' 
I downloaded the source file for windows and compile in VS2008. Again problem in
wpcardw compilation,void* to SignallingComponent. I fixed this. But I have same problem
in initializing PRI trunk. 

I tested it with older version of sangoma driver (6.0.9.12) on Windows XP. but same problem
and more problem in wpcardw compilation.

Anybody did this job successfully ? Yate work on windows with Sangoma? 

Yate team not working on wpcardw ? 

If any body solved the problem of this module, please share with us.

Thank so for feedback's.

Forwarding audio

gregory — 2013-04-23T12:16:24

Hi, 

 

I want to record all calls done via Yate using an external program.

This program receives RTP and records it.

 

The solution of which I thought is to insert both sides of every call to a
conference, to join the conference using my own code (not related to Yate)
and to forward the RTP that I receive in my code.

 

Is there something simpler than the solution above?

For example, is there a way to get the SDP of every call in order to know
the RTP ports that should be recorded?

Or is it possible to forward the RTP (of both sides) to some known address?

 

Thanks?

Calling outside via some lines

Eugene Prokopiev — 2013-04-23T09:49:54

Hi,

I have this accfile.conf:

[line-1]
enabled=yes
protocol=sip
username=line-1
password=pwd-1
domain=10.10.10.100
registrar=10.10.10.100

[line-2]
enabled=yes
protocol=sip
username=line-2
password=pwd-2
domain=10.10.10.100
registrar=10.10.10.100

How to call via this lines using any free line? Need I use something
like round-robin routing or script to select free line and use it?

Please, give me example string for regexroute.conf.

--
WBR,
Eugene Prokopiev

Variables for call routing

Eugene Prokopiev — 2013-04-23T06:23:05

Hi,

How do I know all or at least most used variable names which can be
used in regexroute.conf both on the right and left of the '=' ?

--
WBR,
Eugene Prokopiev

Failed to send ISUP MSU

Rubén Gracia — 2013-04-22T13:00:18

Hello,
Im configuring a ss7 link with yate+cisco. The link is active, but i can 
not send isup message thought it.

module=sig,format=Type|Status|Uptime;count=1;CARRIER_link=cisco-slt|Normal 
Alignment|867

name=mgcpca,type=misc;spans=22,chans=0;GW24_6/0_CARRIER=s6/ds1-0/2< at >X.X.X.X:2427........

Routing table of 'ss7router': [0x7fc8600bbde8]
ITU     7-69-4 (allow) CANTV_linkset,100,unknown
ITU     7-68-3 (allow) CANTV_linkset,0,allow
Statistics for 'ss7router': Rx=4, Tx=2, Fwd=0, Fail=5, Cong=0

Logs:

<CARRIER_isup/ISUP:INFO> Remote user part is available
<signalling:ALL> Engine [0x7f38e80f3e50] sending notify from 
'CARRIER_isup/ISUP' [0x7f38e81c9e28]
<mgcpca:INFO> MGCPCircuit::status(Idle,false) 1002 [0x7f38e80fe370]
<mgcpca:INFO> MGCPCircuit::status(Idle,false) 1003 [0x7f38e80fe770]
.............
............


<CARRIER_isup/ISUP:INFO> Sending message (0x7f38e8143a10)
-----
GRA [cic=1002 label=1-1-1:3-3-3:10]
   RangeAndStatus='14'
   RangeAndStatus.map='00000000000000'
-----
<ss7router:MILD> Could not send ISUP MSU size 13 on any linkset
<ss7router:ALL> Failed to send ISUP MSU size 13 on allow route 14892
<CARRIER_isup/ISUP:INFO> Received message (0x7f38e80f6730)
-----
GRS [cic=1017 label=3-3-3:1-1-1:9]
   protocol-type='itu-t'
   message-type='GRS'
   RangeAndStatus='14'
-----
..................
...............

<CARRIER_isup/ISUP:INFO> Sending message (0x7f38e80f87c0)
-----
GRA [cic=1017 label=1-1-1:3-3-3:9]
   RangeAndStatus='14'
   RangeAndStatus.map='00000000000000'
-----
<ss7router:MILD> Could not send ISUP MSU size 13 on any linkset
<ss7router:ALL> Failed to send ISUP MSU size 13 on allow route 14892
<CARRIER_isup/ISUP:INFO> Received message (0x7f38e80f6730)
-----
GRS [cic=1002 label=3-3-3:1-1-1:10]
   protocol-type='itu-t'
   message-type='GRS'
   RangeAndStatus='14'
-----
............
..........
<CARRIER_isup/ISUP:INFO> Sending message (0x7f38e80f4220)
-----
GRA [cic=1002 label=1-1-1:3-3-3:10]
   RangeAndStatus='14'
   RangeAndStatus.map='00000000000000'
-----
<ss7router:MILD> Could not send ISUP MSU size 13 on any linkset
<ss7router:ALL> Failed to send ISUP MSU size 13 on allow route 14892
<CARRIER_linkset:ALL> Received SLTM ITU,2-2-2:1-1-1:0 (14883:6366:0) 
with 10 bytes
<CARRIER_linkset:ALL> Sending SLTA ITU,1-1-1:2-2-2:0 (6366:14883:0) with 
10 bytes
<CARRIER_linkset:ALL> Sending SLTM ITU,1-1-1:2-2-2:0 (6366:14883:0) with 
4 bytes
<CARRIER_linkset:ALL> Received SLTA ITU,2-2-2:1-1-1:0 (14883:6366:0) 
with 4 bytes
<CARRIER_isup/ISUP:INFO> Received message (0x7f38e80f4220)
-----
GRS [cic=1017 label=3-3-3:1-1-1:9]
   protocol-type='itu-t'
   message-type='GRS'
   RangeAndStatus='14'
-----

Have you got any idea about what can be the problem?

This is my ysigchan configuration:

[general]
debuglevel=10,10,10,10,10
enable=yes

[CARRIER_isup]
type=ss7-isup
enable=yes
pointcodetype=ITU
pointcode=1-1-1
defaultpointcode=1-1-1
remotepointcode=3-3-3
netindicator=international
voice=GW24_6/0_CARRIER,GW24_6/1_CARRIER,GW24_6/2_CARRIER,GW24_6/3_CARRIER,GW24_6/3_CARRIER,GW24_6/4_CARRIER,GW24_6/5_CARRIER,GW24_6/6_CARRIER,GW24_6/7_CARRIER,GW24_7/0_CARRIER,GW24_7/1_CARRIER,GW24_7/2_CARRIER,GW24_7/3_CARRIER,GW24_7/4_CARRIER,GW24_7/5_CARRIER,GW24_7/6_CARRIER,GW23_3/0_CARRIER,GW23_3/1_CARRIER,GW23_3/2_CARRIER,GW23_3/3_CARRIER,GW23_3/4_CARRIER,GW23_3/5_CARRIER
numtype=international
presentation=allowed
screening=user-provided
print-messages=yes
sls=auto

[CARRIER_linkset]
type=ss7-mtp3
netind2pctype=ITU
netindicator=international
route=ITU,3-3-3
adjacent=ITU,2-2-2
local=ITU,1-1-1
link=CARRIER_link

[CARRIER_link]
type=ss7-mtp2
sig=CARRIER_link
autostart=yes

Thanks in advance for your help and best regards,
Ruben Gracia

regfile users count wrong in 'status overview'

Bill Simon — 2013-04-21T12:08:07

It looks like recent updates to the regfile module cause a miscount when you use 'status overview'. When I use 'status overview' I see 0, but get a correct count when I request the detailed status information.

This line in regfile.cpp uses the usrCount variable which is incremented only when processing the detailed user listing - so overview is always 0.

    msg.retValue() << ",users=" << usrCount;

Prevent yate from changing local RTP port on re-INVITE

Thilo Bangert — 2013-04-19T10:04:53

Hi,

it appears that YATE changes its own RTP port upon a re-INVITE. Is there any 
way this can be prevented. i tried using ignore_sdp_port and ignore_sdp_addr, 
but they are meant for incoming RTP ports, if i can read the documentation 
correctly.

thanks
kind regards
Thilo

Using tonegen/tonedetect in C++ (for continuity testing)

2013-04-16T17:41:32

Hello,

We are implementing Continuity Testing for Sangoma cards (wpcard.cpp)
and I was wondering whether there is a nicer way to do the equivalent
of:

  bool WpCircuit::setupContinuityTest() {

    ToneSource* tone_source = ToneSource::getTone("cotv");
    if(!tone_source.attach(m_consumer,true))
      return false;

    ToneConsumer* tone_consumer = new ToneConsumer("cotv");
    if(!m_source.attach(tone_consumer,true))
      return false;

    return   true;
  }

This would require to somehow link tonegen and tonedetect with wpcard,
which sounds awkward.

Any pointers would be appreciated.
S.



PS: I feel source & consumer are reversed in tonegen/tonedetect wrt
    circuits, so the bit of code above might need more work as well.