VoIP Cookbook: Test your SIP Softphone - Revision history

Anton at 17:55, 20 August 2010

2010-08-20T17:55:46Z

← Older revision		Revision as of 17:55, 20 August 2010
Line 7:		Line 7:
	Go to [[VoIP Rakyat]]'s Service Number page, http://voiprakyat.or.id/services/. This page provides you with some numbers that can be used to test your [[VoIP]] connection and their functions. Some of them are:		Go to [[VoIP Rakyat]]'s Service Number page, http://voiprakyat.or.id/services/. This page provides you with some numbers that can be used to test your [[VoIP]] connection and their functions. Some of them are:

	901 – which indicates the time Jakarta's time and nearby countries.		901 - which indicates the time Jakarta's time and nearby countries.
	902 - noise		902 - noise
	903 - echo test		903 - echo test
			904 - record caller voice and playback to listen to the VoIP call quality

	[[Image:Voiprakyat-service2.jpeg\|left\|200px\|thumb\|Through VoIP Rakyat's Phonebook, you can see who's online]]		[[Image:Voiprakyat-service2.jpeg\|left\|200px\|thumb\|Through VoIP Rakyat's Phonebook, you can see who's online]]

Onnowpurbo at 02:54, 13 May 2010

2010-05-13T02:54:58Z

← Older revision		Revision as of 02:54, 13 May 2010
Line 1:		Line 1:
	Now that you have adjusted both ~~softphones~~ (or just one of them), the next important thing you have to do is to test whether it could run properly. Note that the quality of the voice produced by the softphone during the test may have been just fine, but when your softphone is connected to a VoIP provider, the voice quality could be poor, depending on many other things such as bandwidth availability and the type of codec run by the softphone. For this test purpose, VoIP providers usually provide the telephone number to which you can dial.		Now that you have adjusted both [[softphone]]s (or just one of them), the next important thing you have to do is to test whether it could run properly. Note that the quality of the voice produced by the [[softphone]] during the test may have been just fine, but when your softphone is connected to a VoIP provider, the voice quality could be poor, depending on many other things such as [[bandwidth]] availability and the type of codec run by the [[softphone]]. For this test purpose, [[VoIP]] providers usually provide the telephone number to which you can dial.

	If your computer is connected to an internet behind a firewall, the firewall might block your connectivity. In order to make your VoIP connectivity working behind the firewall, you have to open Port 5060-6060 to enable Session Initiation Protocol (SIP) and Port 8000-20000 for voice data delivery using Real Time Protocol (RTP). But if you're not sure what to do, you can simply ask your network administrator to do what is told here.		If your [[computer]] is connected to an [[internet]] behind a [[firewall]], the [[firewall]] might block your connectivity. In order to make your [[VoIP]] connectivity working behind the [[firewall]], you have to open Port 5060-6060 to enable [[Session Initiation Protocol]] ([[SIP]]) and Port 8000-20000 for voice data delivery using [[Real Time Protocol]] ([[RTP]]). But if you're not sure what to do, you can simply ask your network administrator to do what is told here.

	[[Image:Voiprakyat-service1.jpeg\|right\|200px\|thumb\|Just like other VoIP Providers, VoIP Rakyat provides its users with some numbers with which the users can use for testing their VoIP quality]]		[[Image:Voiprakyat-service1.jpeg\|right\|200px\|thumb\|Just like other VoIP Providers, VoIP Rakyat provides its users with some numbers with which the users can use for testing their VoIP quality]]

	Go to VoIP Rakyat's Service Number page, http://voiprakyat.or.id/services/. This page provides you with some numbers that can be used to test your VoIP connection and their functions. Some of them are:		Go to [[VoIP Rakyat]]'s Service Number page, http://voiprakyat.or.id/services/. This page provides you with some numbers that can be used to test your [[VoIP]] connection and their functions. Some of them are:

	901 – which indicates the time Jakarta's time and nearby countries.		901 – which indicates the time Jakarta's time and nearby countries.
Line 15:		Line 15:
	In testing this connectivity, what users will often do is to call anyone found online in http://www.voiprakyat.or.id/?inc=online_phones. So don't be surprised if someone dials your number. Depending on where the users are, the call comes from a variety of countries, including the U.S.		In testing this connectivity, what users will often do is to call anyone found online in http://www.voiprakyat.or.id/?inc=online_phones. So don't be surprised if someone dials your number. Depending on where the users are, the call comes from a variety of countries, including the U.S.

	There are of course other VoIP phone numbers which you can use to test your VoIP connection. These are provided in a long list available in http://www.voip-info.org/wiki/view/Phone+Numbers. If you want to call using SIP address format (sip@domain.com), the following is a table of some numbers you may use:		There are of course other [[VoIP]] phone numbers which you can use to test your [[VoIP]] connection. These are provided in a long list available in http://www.voip-info.org/wiki/view/Phone+Numbers. If you want to call using [[SIP]] address format (sip@domain.com), the following is a table of some numbers you may use:

Onnowpurbo: /* See Also */

2010-05-13T02:53:07Z

Onnowpurbo at 04:34, 8 March 2010

2010-03-08T04:34:21Z

← Older revision		Revision as of 04:34, 8 March 2010
Line 24:		Line 24:
	! Enum		! Enum
	\|-		\|-
	Autoattendant		\| Autoattendant
	BC Wireless (http://www.bcwireless.net/moin.cgi/NetworkServices/VoiceServices/PublicConferenceRoom).		\| BC Wireless (http://www.bcwireless.net/moin.cgi/NetworkServices/VoiceServices/PublicConferenceRoom).
	1000@mutual.bcwireless.net		\| 1000@mutual.bcwireless.net
	1 604 484 5289 x8600 through E164.org		\| 1 604 484 5289 x8600 through E164.org
	\|-		\|-
	\|		\|

Onnowpurbo: New page: Now that you have adjusted both softphones (or just one of them), the next important thing you have to do is to test whether it could run properly. Note that the quality of the voice produ...

2010-03-08T04:33:40Z

New page: Now that you have adjusted both softphones (or just one of them), the next important thing you have to do is to test whether it could run properly. Note that the quality of the voice produ...

New page

Now that you have adjusted both softphones (or just one of them), the next important thing you have to do is to test whether it could run properly. Note that the quality of the voice produced by the softphone during the test may have been just fine, but when your softphone is connected to a VoIP provider, the voice quality could be poor, depending on many other things such as bandwidth availability and the type of codec run by the softphone. For this test purpose, VoIP providers usually provide the telephone number to which you can dial.

If your computer is connected to an internet behind a firewall, the firewall might block your connectivity. In order to make your VoIP connectivity working behind the firewall, you have to open Port 5060-6060 to enable Session Initiation Protocol (SIP) and Port 8000-20000 for voice data delivery using Real Time Protocol (RTP). But if you're not sure what to do, you can simply ask your network administrator to do what is told here.

[[Image:Voiprakyat-service1.jpeg|right|200px|thumb|Just like other VoIP Providers, VoIP Rakyat provides its users with some numbers with which the users can use for testing their VoIP quality]]

Go to VoIP Rakyat's Service Number page, http://voiprakyat.or.id/services/. This page provides you with some numbers that can be used to test your VoIP connection and their functions. Some of them are:

901 – which indicates the time Jakarta's time and nearby countries.
902 - noise
903 - echo test

[[Image:Voiprakyat-service2.jpeg|left|200px|thumb|Through VoIP Rakyat's Phonebook, you can see who's online]]

In testing this connectivity, what users will often do is to call anyone found online in http://www.voiprakyat.or.id/?inc=online_phones. So don't be surprised if someone dials your number. Depending on where the users are, the call comes from a variety of countries, including the U.S.

There are of course other VoIP phone numbers which you can use to test your VoIP connection. These are provided in a long list available in http://www.voip-info.org/wiki/view/Phone+Numbers. If you want to call using SIP address format (sip@domain.com), the following is a table of some numbers you may use:

{| border="1" cellpadding=2 style="border-collapse: collapse"
! Function
! SIP Provider
! SIP
! Enum
|-
Autoattendant
BC Wireless (http://www.bcwireless.net/moin.cgi/NetworkServices/VoiceServices/PublicConferenceRoom).
1000@mutual.bcwireless.net
1 604 484 5289 x8600 through E164.org
|-
|
| Enum2go (http://enum2go.com/)
| 878107472000010@sip2go.com
|
|-
| Echo Test
| N3 Network Lab. (http://www.n3network.ch/)
| Echo test sip: echo@n3network.ch and sip: 905100@n3network.ch (no G.729)
|
|-
|
| Mouselike.org (UK) (http://www.mouselike.org/)
| 904@mouselike.org
| +441483604781
|-
|
| VoipTalk UK (http://www.voiptalk.org/)
| 904@voiptalk.org
|
|-
| Reread Called ID
|
| 95861111@mutual.bcwireless.net
|
|-
| Welcome Line
| FWD
| 55555@fwd.pulver.com
|
|-
|
| Ewing IT
| 611300766674@sip.like2fone.com
|
|-
|
| Xmission (http://xmission.com/transmission)
| xmission@pbx.xmission.com (tidak ada G.729)
|
|-
|
| UCLA (http://internet2.edu/sip.edu)
| 13108254321@ucla.edu (tidak ada G.729)
|
|-
|
| TELL
| 18005558355@proxy01.sipphone.com
|
|-
|
| U. Philippines
| 0116329818500@proxy01.sipphone.com
|
|-
|
| Personal Telco (http://wiki.personaltelco.net/moin.cgi/SipPhoneDirectory)
| 274185@fwd.pulver.com
|
|-
|
| Patton Electronics (http://www.patton.com/support)
| support@patton.com (tidak ada G.729)
|
|-
|
| Party Line
| 17475552663@proxy01.sipphone.com (VoIP conference setiap sabtu jam 20:00 GMT)
|
|-
|
| Ingate (http://www.ingate.com/trysip.php)
| music@trysip.ingate.com
|
|-
|
| MIT (http://sipphone.com/numbers)
| 16172531000@proxy01.sipphone.com
|
|}

==See Also==

* [[VoIP Cookbook: Building your own Telecommunication Infrastructure]]
* [[VoIP Cookbook: Becoming a user]]

← Older revision		Revision as of 02:53, 13 May 2010
Line 116:		Line 116:
	* [[VoIP Cookbook: Building your own Telecommunication Infrastructure]]		* [[VoIP Cookbook: Building your own Telecommunication Infrastructure]]
	* [[VoIP Cookbook: Becoming a user]]		* [[VoIP Cookbook: Becoming a user]]

			[[Category: VoIP]]