Messages

1

Other Yate server issues / Re: Limit number of calls per trunk

« on: February 29, 2016, 12:01:42 PM »

The route.js script file content



################
Engine.debugName("route");
Message.trackName(Engine.debugName());

line = "asteradmin";
line.count = 0;
line.max = 2;
line.chans = [];

function onHangup(msg)
{
    var id = "" + msg.id;
    var idx = line.chans.indexOf(id);
    if (idx >= 0) {
        line.count--;
        delete line.chans[idx];
        Engine.debug(Engine.DebugAll,"Line " + line + " removed chan " + id +
            " count=" + line.count);
    }
    return false;
}

function onRoute(msg)
{
    var called = "" + msg.called;
    if (!called)
        return false;
    if (called.startsWith("456")) {
        // Route to 'someprovider' line
        if (line.count >= line.max) {
            msg.retValue("-");
            // Return some error (optional)
            msg.error = "busy";
            return true;
        }
        line.chans.push(msg.id);
        line.count++;
        Engine.debug(Engine.DebugAll,"Line " + line + " added chan " + msg.id +
            " count=" + line.count);
        msg.line = line;
        msg.forward_sdp = true;
        msg.callername = msg.caller;
        msg.formats = "alaw,mulaw,g729";
        msg.retValue("sip/99013307" + called.substr(2));
        return true;
    }
    return false;
}

Message.install(onHangup,"chan.hangup");
Message.install(onRoute,"call.route",10);

2

Other Yate server issues / Re: Yate changing RTP port

« on: February 11, 2016, 07:53:20 AM »

Dear Marian,

Thanks for answers.

but after deep investigation i found sniffed messages presented below.
As i understand chan.rtp is releasing the rtp legs while receiving reinvite and after changing localport param.
Please can you advise how can we fix this issue

Sniffed 'chan.rtp' time=1455181133.881092
  thread=0x1940a20 'YSIP EndPoint'
  data=(nil)
  retval='(null)'
  param['rtpid'] = 'yrtp/1362960942'
  param['media'] = 'audio'
  param['transport'] = 'RTP/AVP'
  param['terminate'] = 'true'
  param['call_direction'] = 'incoming'
  param['call_address'] = 'y.y.y.y:5060'
  param['call_status'] = 'answered'
  param['call_billid'] = '1454933764-119801'
Returned true 'chan.rtp' delay=0.000047
  thread=0x1940a20 'YSIP EndPoint'
  data=(nil)
  retval='(null)'
  param['rtpid'] = 'yrtp/1362960942'
  param['media'] = 'audio'
  param['transport'] = 'RTP/AVP'
  param['terminate'] = 'true'
  param['call_direction'] = 'incoming'
  param['call_address'] = 'y.y.y.y:5060'
  param['call_status'] = 'answered'
  param['call_billid'] = '1454933764-119801'
  param['handlers'] = 'yrtp:100'
  param['localip'] = 'x.x.x.x'
  param['localport'] = '25318'
  param['stats'] = 'PS=45538,OS=910760,PR=45000,OR=900000,PL=0'
  param['status'] = 'terminated'



20160211085853.881178 >>> DataTranslator::detachChain(0x7fbee40b14a0,0x7fbcfc0d26a0)
20160211085853.881189 <<< DataTranslator::detachChain
20160211085853.881198 >>> DataTranslator::detachChain(0x7fbcfc150080,0x7fbee409f0d0)
20160211085853.881204 <<< DataTranslator::detachChain
20160211085853.881216 <ALL> Cleaning up RTP 0x7fbee40ac480 [0x7fbee4044970]





Sniffed 'chan.rtp' time=1455181133.881291
  thread=0x1940a20 'YSIP EndPoint'
  data=0x7fbedc1155f0
  retval='(null)'
  param['id'] = 'sip/241959'
  param['module'] = 'sip'
  param['call_direction'] = 'incoming'
  param['call_address'] = 'y.y.y.y:5060'
  param['call_status'] = 'answered'
  param['call_billid'] = '1454933764-119801'
  param['media'] = 'audio'
  param['transport'] = 'RTP/AVP'
  param['direction'] = 'bidir'
  param['format'] = 'g729'
  param['ipv6_support'] = 'false'
  param['remoteip'] = 'y.y.y.y'
  param['remoteport'] = '9746'
  param['evpayload'] = '96'
  param['sdp_maxptime'] = '20'
  param['sdp_3gOoBTC'] = ''
20160211085853.881409 <INFO> DataTranslator::attachChain [0x7fbedc109700] '(null)' -> [0x7fbcfc0d26a0] 'g729' not possible
20160211085853.881428 <INFO> DataTranslator::attachChain [0x7fbcfc150080] 'g729' -> [0x7fbedc081fc0] '(null)' not possible
20160211085853.881456 >>> DataTranslator::detachChain(0x7fbedc109700,0x7fbcfc0d26a0)
20160211085853.881466 <<< DataTranslator::detachChain
20160211085853.881476 <ALL> DataTranslator::attachChain [0x7fbedc109700] 'g729' -> [0x7fbcfc0d26a0] 'g729' succeeded
20160211085853.881482 >>> DataTranslator::detachChain(0x7fbcfc150080,0x7fbedc081fc0)
20160211085853.881486 <<< DataTranslator::detachChain
20160211085853.881492 <ALL> DataTranslator::attachChain [0x7fbcfc150080] 'g729' -> [0x7fbedc081fc0] 'g729' succeeded
Returned true 'chan.rtp' delay=0.000257
  thread=0x1940a20 'YSIP EndPoint'
  data=0x7fbedc1155f0
  retval='(null)'
  param['id'] = 'sip/241959'
  param['module'] = 'sip'
  param['call_direction'] = 'incoming'
  param['call_address'] = 'y.y.y.y:5060'
  param['call_status'] = 'answered'
  param['call_billid'] = '1454933764-119801'
  param['media'] = 'audio'
  param['transport'] = 'RTP/AVP'
  param['direction'] = 'bidir'
  param['format'] = 'g729'
  param['ipv6_support'] = 'false'
  param['remoteip'] = 'y.y.y.y'
  param['remoteport'] = '9746'
  param['evpayload'] = '96'
  param['sdp_maxptime'] = '20'
  param['sdp_3gOoBTC'] = ''
  param['handlers'] = 'yrtp:100'
  param['localip'] = 'x.x.x.x'
  param['localport'] = '16950'
  param['rtpid'] = 'yrtp/70563133'
  param['status'] = 'created'

3

Other Yate server issues / Re: Yate changing RTP port

« on: February 09, 2016, 08:40:03 AM »

Dear Marian,

Also please could You help to create a datatranslator for alaw->g729
as we face with some
No DataTranslator created for 'alaw' -> 'g729'

4

Other Yate server issues / Re: Yate changing RTP port

« on: February 09, 2016, 06:28:22 AM »

Dear Marian,

As we detected Yate is changing RTP port on one leg only,
Like we have 2 interfaces Yate is binded.
While receiving Invite from B interface Yate answers with 200 OK without changing rtp port,
but during the same call when Yate is receiving Invite from A interface it answers with 200 OK and with changed rtp port.
Also as we can see from sniffer debug there is Operation = verify in logs while receiving Invite.
Can you please investigate how force Yate not to change rtp port while session refresh ?

Huge thanks

5

Other Yate server issues / Re: Yate changing RTP port

« on: January 29, 2016, 06:29:56 AM »

Dear Marian,

It will look like

;regexroute example:

[extra]
call.update=50

[call.update]
${sip_Invite}^$= ? ? ? (What must be written here for to return true from message handle to let media change) 

6

Other Yate server issues / Re: Yate changing RTP port

« on: January 29, 2016, 05:02:52 AM »

Hi Marian,

Please could you share how can i handle the message(the command list) and in which file it must be specified?

7

Other Yate server issues / Re: Yate changing RTP port

« on: January 29, 2016, 02:06:13 AM »

Dear Marian,

Thanks for prompt responce,

Yate version is YATE 5.2.0-1 r5801

Invite contains SDP with the same values as in first one.

I can not attach trace because i didnt saved it.

But anyway Invite contains SDP 

8

Other Yate server issues / Re: Routing call based on SIP message

« on: January 29, 2016, 01:27:57 AM »

Dear Marian,

Before i can collect logs because its a live server with huge amount of simultaneous calls could You PLEASE answer my new Topic "Yate changing RTP port ".
Its very important for us.
Thanks for understanding.

9

Other Yate server issues / Re: Routing call based on SIP message

« on: January 25, 2016, 01:28:07 PM »

Dear Marian

I have added with pipe character but in fact while receiving 480 SIP response the fork stops

^+1\(.*\)$=fork;fork.stop=noconn\|noanswer^;callto.1=sip/sip:1\1@11.22.33.44;callto.2=|;callto.3=sip/sip:1\1@55.66.77.88;

10

Other Yate server issues / Yate changing RTP port

« on: January 25, 2016, 08:07:50 AM »

Hi,

Dear Yate Team,

Please could you advise why Yate server acting as SBC while receiving Re-invite(for session refresh in dialog invite) from remote side is answering with 200 OK where in SDP is changing RTP source port.
Is there anyway to configure Yate not to change port ?

Thanks in advance

11

Other Yate server issues / Re: Routing call based on SIP message

« on: January 22, 2016, 04:57:44 AM »

Thanks a lot Marian,

I'll try and let You know results

12

Other Yate server issues / Re: Routing call based on SIP message

« on: January 22, 2016, 12:26:45 AM »

Hi,

Dear Marian,

Could we fix two end causes for to be checked in case of forking like

^123$=fork;fork.stop=noanswer,failure^;callto.1=sip/sip:123@127.0.0.1;callto.2=|;callto.3=wave/play//myfile.au

thank you

13

Other Yate server issues / Re: How to strip some characters from A-Number

« on: May 02, 2015, 03:41:01 PM »

Hi

Thanks a lot Marian

14

Other Yate server issues / How to strip some characters from A-Number

« on: March 27, 2015, 02:49:14 AM »

Hi,

Please could You help me to strip some characters from A -number then route it to specified destination.
I have tried this way '
${caller}^\(.*\)\(374\)=;caller=\1
In this case A-number is 37411444444 but in outgoing invite From header is displayed with B-number.

Thanks in advance

15

Other Yate server issues / Yate diconnects call with "nomediaUnsupported Media Type"

« on: November 10, 2014, 05:12:53 PM »

Hi,

Dears,

Yate disconnects call with "nomedia   Unsupported Media Type" after 7 seconds while Yate already has sent back 180 Ringing with SDP.
How can be the problem solved?