Beginners in Yate - Revision history

Dana: /* Test the setup */

2015-03-10T13:05:22Z

‎Test the setup

Dana: /* Test the setup */

2015-03-10T13:01:44Z

‎Test the setup

Dana: /* Test the setup */

2015-03-10T12:59:40Z

‎Test the setup

Dana: /* SIP Configuration */

2015-03-10T12:49:09Z

‎SIP Configuration

Dana: /* Test the setup */

2015-03-10T12:44:19Z

‎Test the setup

Monica: /* Test the setup */

2013-10-31T12:59:28Z

‎Test the setup

Monica: /* Test the setup */

2013-10-31T12:58:36Z

‎Test the setup

Dana at 07:55, 31 May 2013

2013-05-31T07:55:47Z

Dana at 07:55, 31 May 2013

2013-05-31T07:55:37Z

Dana: /* Test the setup */

2013-04-08T10:09:16Z

‎Test the setup

@@ Line 114: / Line 114: @@
   99991003 - you should hear ring tone
-These are numbers defined for testing purposes and you can find all of them in the "regexroute.conf" file. When you call one of those numbers, you'll get a standard telephony tone.
+These numbers are defined for testing purposes and you can find all of them in the "regexroute.conf" file. When you call one of these numbers, you get a standard telephony tone.
-Then make a call from one phone to the other and check the audio, if you can't hear the other end then you may have a NAT problem.<br>
+Then make a call from one phone to the other and check the audio: if you cannot hear the other end then you may have a NAT problem.
 Also to see the flow of the messages, use a telnet client. Connect like this:

@@ Line 106: / Line 106: @@
 == Test the setup==
-If Yate was running when you wrote in configuration file, '[[Stopping and restarting Yate|reload Yate]]' so that the changes will take place.<br>
+If Yate is running when you change the configuration file, you need to [[Stopping and restarting Yate|reload Yate]] for the changes to become effective.
 From the phone, dial one of the following:

@@ Line 116: / Line 116: @@
 These are numbers defined for testing purposes and you can find all of them in the "regexroute.conf" file. When you call one of those numbers, you'll get a standard telephony tone.
-Then make a call from one phone to the other and check the audio, if you can hear the other end then you may have a NAT problem.<br>
+Then make a call from one phone to the other and check the audio, if you can't hear the other end then you may have a NAT problem.<br>
 Also to see the flow of the messages, use a telnet client. Connect like this:

@@ Line 85: / Line 85: @@
 ===SIP Configuration===
-The file that is used is [[SIP Configuration File|ysipchan.conf]]. No configuration is needed in this file because by default Yate will bind with all the network interfaces on your server on port 5060. If other programs use this port then you have to use another free port and put it in [general] section.
+The file used is [[SIP Configuration File|ysipchan.conf]]. No configuration is needed in this file because by default Yate will bind with all the network interfaces on your server on port 5060. If other programs use this port then you have to use another free port and put it in [general] section.
 ===Routing===

@@ Line 106: / Line 106: @@
 == Test the setup==
-If Yate was running when you written in configuration file [[Stopping and restarting Yate|reload Yate]] so that the changes will take place.<br>
+If Yate was running when you wrote in configuration file, '[[Stopping and restarting Yate|reload Yate]]' so that the changes will take place.<br>
-From the telephony, dial one of the following:
+From the phone, dial one of the following:
-  99991001 - you should here dial tone
+  99991001 - you should hear dial tone
-  99991002 - you should here busy tone
+  99991002 - you should hear busy tone
-  99991003 - you should here ring tone
+  99991003 - you should hear ring tone
-These are numbers defined for testing purposes and you can find all of them in the "regexroute.conf" file. When you call one of these, you'll get a standard telephony tone.
+These are numbers defined for testing purposes and you can find all of them in the "regexroute.conf" file. When you call one of those numbers, you'll get a standard telephony tone.
-Then make a call from one phone to the other and check the audio, if you can here the other end then you may have a NAT problem.<br>
+Then make a call from one phone to the other and check the audio, if you can hear the other end then you may have a NAT problem.<br>
-Also if you want to see the flow of the messages we can use a telnet client. Connect like this:
+Also to see the flow of the messages, use a telnet client. Connect like this:
   telnet localhost 5038
-And write next commands for enable the debug and color the output:
+And write next commands to enable the debug and color the output:
   debug on

← Older revision		Revision as of 12:59, 31 October 2013
Line 135:		Line 135:
	* [[Stopping and restarting Yate]]		* [[Stopping and restarting Yate]]

−	[[Category:~~Beginers~~]]	+	[[Category:Beginners]]

← Older revision		Revision as of 12:58, 31 October 2013
Line 134:		Line 134:
	* [[Starting Yate]]		* [[Starting Yate]]
	* [[Stopping and restarting Yate]]		* [[Stopping and restarting Yate]]
		+
		+	[[Category:Beginers]]

@@ Line 1: / Line 1: @@
-ssssssssssss
 [[What is Yate|Yate]] (acronym for Yet Another Telephony Engine) is a next-generation telephony engine, is a free and open source communications software with support for video, voice and instant messaging. <br>
 Based on Voice over Internet Protocol (VoIP) and PSTN, it can easily be extended. It supports SIP, H.323, IAX, MGCP, Jingle, Jabber, E1, T1, analogic, ISDN PRI, BRI, and SS7.<br>