Messages

1

Yate based projects / Re: simple install - from SIP message

« on: September 04, 2018, 09:21:47 PM »

ok.

local interface 10.154.xx.xx but where to add the external ip in the config?

will the regex ${role}local=sip/sip:${called}@ADDR;oconnection_id=OUTBOUND sends all the SIP to the ADDR ? i.e REGISTER, INVITE , NOTIFY PUBLISH etc ?

2

Yate based projects / simple install - from SIP message

« on: September 02, 2018, 09:01:46 PM »

Hi team,

How are things? It's a long time i tested my first yate installation but never used it in our production.
I am already using opensips but i want to test yate in parallel.
Just simple requirements to start with.

Listen on 5060 UDP/TCP port and forward it to another IP(opensips). so anything comes to 5060 port , forward it to another SIP server.  My machine is running 10.154.0.2 IP but the Public (static) IP is xx.xx.xx.xx

Can you please guide me what file do i need to change for the above ?

Thank you team.

3

Other Yate server issues / Re: SS7 - SIGTRAN - SIP

« on: May 23, 2016, 04:44:36 PM »

Hello guys
i have Cisco 5400 routers, i am reading yate does the ss7 over IP. can i use yate to route calls directly from 5400 and vice versa? so i dont need to use pgw anymore?

4

Yate users hangout place / Re: incoming Registration send in roundrobin

« on: February 25, 2015, 08:08:43 PM »

i am using another hardware based failover to send the requests on different yate servers in roundrobin fashion.  i did not do in the yate. yate does not have that feature

5

Yate users hangout place / Re: simple sip phone dial

« on: December 09, 2014, 05:50:36 PM »

so i changed it to below. and my 100phone rang for 20 seconds then it just hangup saying Temporarily unavailable.  it did not try on my second server after the timeout. you might be able to figure out from the attached logs. by the way i am dialing from 100 to 100.

 $ev->retval = "fork";
  $ev->params["callto.1"] = "sip/sip:100@xx.xx.xx.xx:6542";
  $ev->params["callto.2"] = "|";
  $ev->params["callto.3"] = "sip/sip:100@mysecondserver.com";
  $ev->params["callto.1.maxcall"] = 20000;
  $ev->params["stoperror"] = "noanswer|busy^";
  $ev->handled = true;
  $ev->Acknowledge();
  break;

6

Yate users hangout place / Re: simple sip phone dial

« on: December 09, 2014, 02:24:01 AM »

sure let's say i have location saved and now i want to route it.

the location is sip/sip:100@xx.xx.xx.xx:6055

how to route in route.php to this location?

         $location = my above location value from db
         if ($ev->getValue("called")=="100") {
           $ev->retval = $location;
           $ev->params["line"] = "100";
           //$ev->params["targetid"] = $ev->getValue("callid");
           $ev->handled = true;
           $ev->Acknowledge();
           break;
         }

will this work? and how to timeout on busy or no answer? if the call timeouts or busy i will put different server IP in the location and route to anywhere else. that;s what i want. Can you please guide this ? ihave hard time to get this done

7

Yate users hangout place / Re: simple sip phone dial

« on: December 08, 2014, 06:41:12 AM »

Actually i want to handle the call by route.php so its fine. but its not working i tried all the possibilities.
The below should work, but its not. any clue please

           $ev->retval = "sip/".$ev->getValue("called");
           $ev->params["line"] = "100";
           //$ev->params["targetid"] = $ev->getValue("callid");
           $ev->handled = true;
           $ev->Acknowledge();
           break;
         }

8

Yate users hangout place / Re: simple sip phone dial

« on: December 07, 2014, 01:21:06 AM »

also

  param['rtp_forward'] = 'possible'
  param['handlers'] = 'register:50,regexroute:100,cdrbuild:50,route.php:80,gvoice:20,queues:45,yrtp:50,cdrbuild:50,lateroute:75,dbwave:90,analyzer:90,iax:90,tone:90,conf:90,dumb:90,filetransfer:90,sip:90,wave:90,callfork:100,extmodule:100,pbx:100,callgen:100,moh:100,park:100'
  param['line'] = '100'
  param['targetid'] = 'sip/6213646a5a29720d/2e57b003/'
  param['callto'] = 'sip/100'
  param['error'] = 'offline'

here the logs says error is offline. unable to figure out where the issue

9

Yate users hangout place / Re: simple sip phone dial

« on: December 07, 2014, 12:44:56 AM »

agreed. thats what i am wondered about. when i restart the yate it says register:120 you can see below. but when i make a call i saw register 50. 
what it could be any clue?

param['handlers'] ='engine:100,callgen:100,extmodule:100,cdrcombine:100,cdrbuild:100,ilbcwebrtc:100,isaccodec:100,accfile:100,regfile:100,regexroute:110,analogdetect:120,subscription:120,mysqldb:120,mux:120,socks:120,openssl:120,zlibcompress:120,analyzer:120,enumroute:120,tonedetect:120,callfork:120,stun:120,iax:120,tone:120,filetransfer:120,conf:120,fileinfo:120,pbx:120,dumb:120,queues:120,users:120,wave:120,sip:120,yrtp:120,gvoice:120,park:120,monitoring:120,presence:120,queuesnotify:120,cache:120,register:120,snmpagent:120,mrcp:120,sipfeatures:120'

10

Yate users hangout place / Re: simple sip phone dial

« on: December 04, 2014, 05:53:32 PM »

yah that's weired. please see my attached register.conf
i dont want to use native call.route actually. i want to route it using route.php beacuse after 20 seconds i have to send the call to another IP.
this is for testing where i have checked if the dialed number is 100 then first route it to extension 100 if nobody picks up or busy then forward it to another ip for example myotherip.com  thats what i want

         if ($ev->getValue("called")=="100") {
           $ev->retval = "sip/".$ev->getValue("called");
           $ev->params["line"] = "100";
           //$ev->params["targetid"] = $ev->getValue("callid");
           $ev->handled = true;
           $ev->Acknowledge();
           break;
         }

11

Yate users hangout place / Re: simple sip phone dial

« on: December 04, 2014, 02:06:19 AM »

weired. i have call.route=yes in register.conf. but all the calls are going to route.php
and if i do a check in route.php phone still does not ring. its ok to route the call via route.php because i have to send the call to voicemail server if it timeouts on the user.
here is my route.php

        Yate::Output("Route Message: " . $ev->name . " id: " . $ev->id . " called: " . $ev->params["called"] . " caller: " . $ev->params["caller"]);
         if ($ev->getValue("called")=="100") {
           $ev->retval = "sip/".$ev->getValue("called");
           $ev->params["line"] = "100";
           $ev->params["targetid"] = $ev->getValue("callid");
           $ev->handled = true;
           $ev->Acknowledge();
           break;
         }


have a look at the attached logs, i can see call goes to route.php but it is again saying offline in the error. i also have now added location in the sip register/unregister query. can you please have a look. thanks. also if i have to timeout the call to my voicemail server how it will be acheived? say after 20 seconds if nobody picks up, route to voicemail.com server.
thanks

12

Yate users hangout place / Re: hangup event?

« on: December 04, 2014, 01:31:22 AM »

great! much appreciated.

do you have any example for php? since all my other scripts are running in scripts. just if you have any or any idea

13

Yate users hangout place / Re: simple sip phone dial

« on: December 02, 2014, 01:54:37 AM »

should i put ${data} in the location field? currently i am not using location.

14

Yate users hangout place / Re: hangup event?

« on: November 26, 2014, 10:59:43 PM »

any example will be helpful please. right now i have

cdr_finalize=UPDATE yate_cdr SET billtime = '${billtime}', ringtime  = '${ringtime}', `status` = '${status}', reason = '${reason}' WHERE chan = '${chan}' AND billid='${billid}'

so if i call a script how the value will be returned back ?
any example will be helpful.
thanks

15

Yate users hangout place / simple sip phone dial

« on: November 26, 2014, 10:55:10 PM »

Hi team,

i have used db to register my sip users. this works fine. but when i try to dial this user it does not route and says NOT FOUND.
[user.register]
; Query for the user.register message
query=UPDATE sip_users SET fullcontact='${data}',expires=DATE_ADD(NOW(), INTERVAL ${expires} SECOND) WHERE username='${username}'


i have user 100 is regsiter and calling 100. here is my route.php. the script is run and i see it set the line but dont route and say 404 Not found.

         if ($ev->getValue("called")=="100") {
           $ev->retval = "sip/".$ev->getValue("called");
           $ev->params["line"] = "100";
           //$ev->params["targetid"] = $ev->getValue("callid");
           $ev->handled = true;
           $ev->Acknowledge();
           break;
         }


here r logs
  param['rtp_forward'] = 'possible'
  param['handlers'] = 'register:50,regexroute:100,cdrbuild:50,register:50,route.php:80,gvoice:20,queues:45,yrtp:50,cdrbuild:50,lateroute:75,dbwave:90,analyzer:90,iax:90,tone:90,conf:90,dumb:90,filetransfer:90,sip:90,wave:90,callfork:100,extmodule:100,pbx:100,callgen:100,moh:100,park:100'
  param['line'] = '100'
  param['targetid'] = 'sip/4c578f4d7a34f816/e9155934/'
  param['callto'] = 'sip/100'
  param['error'] = 'offline'