VoIP: Cara Mengkonfigurasi Trunk di Asterisk - Revision history

Onnowpurbo: /* Lebih Dalam */

2014-02-25T12:55:42Z

Lebih Dalam

← Older revision		Revision as of 12:55, 25 February 2014
Line 113:		Line 113:
	==Lebih Dalam==		==Lebih Dalam==

			* [[VoIP: Trunk]]
	* [[VoIP: Asterisk menerima Anonymous Call]]		* [[VoIP: Asterisk menerima Anonymous Call]]
	* [[VoIP: Asterisk route ke arah IP PBX lain dengan kode area]]		* [[VoIP: Asterisk route ke arah IP PBX lain dengan kode area]]
	* [[VoIP: Cara Mengkonfigurasi Trunk di Asterisk]]		* [[VoIP: Cara Mengkonfigurasi Trunk di Asterisk]]
	* [[VoIP: Cara Mengkonfigurasi Dial Out Melalui Trunk di Asterisk]]		* [[VoIP: Cara Mengkonfigurasi Dial Out Melalui Trunk di Asterisk]]
			* [[VoIP: Asterisk forward call ke IP softswitch lain]]

			===Asterisk Round Robin===

			* [[VoIP: Astersk Dial Round Robin]]
			* [[VoIP: Asterisk pakai GotoIf]]

	===OpenSIPS===		===OpenSIPS===
Line 122:		Line 129:
	* [[VoIP: OpenSIPS route ke arah Asterisk]]		* [[VoIP: OpenSIPS route ke arah Asterisk]]
	* [[OpenSIPS: Rewrite URI]]		* [[OpenSIPS: Rewrite URI]]
			* [[OpenSIPS: Rewritehostport]]

			===OpenSIPS round robin===

			* [[OpenSIPS: dispatcher]]


	==Pranala Menarik==		==Pranala Menarik==

Onnowpurbo at 05:42, 20 January 2014

2014-01-20T05:42:50Z

← Older revision		Revision as of 05:42, 20 January 2014
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	In-bound call membutuhkan untuk di terima oleh SIP dan di arahkan ke context call plan.		In-bound call membutuhkan untuk di terima oleh SIP dan di arahkan ke context call plan.
	Out-bound call aplikasi dial membutuhkan profile spesifik di konfigurasi SIP atau registrasi ke tujuan.		Out-bound call aplikasi dial membutuhkan profile spesifik di konfigurasi SIP atau registrasi ke tujuan.


	==Context==		==Context==
Line 46:		Line 45:
	exten => _91.,2,Dial(SIP/trunk-paris-bangkok/${EXTEN:2},20,rt)		exten => _91.,2,Dial(SIP/trunk-paris-bangkok/${EXTEN:2},20,rt)

	Agar memungkinkan struktur multi site dialin plan, kita memodifikasi caller ID dengan menambahkan Prefix Paris (98).		Agar memungkinkan struktur multi site dialin plan, kita memodifikasi caller ID dengan menambahkan Prefix Paris (98).


	~~The~~ Dial ~~application is directeing the~~ call ~~to the~~ SIP trunk ~~named~~ trunk-paris-bangkok, defined ~~in the~~ sip.conf ~~file between [ ]~~. ~~The~~ trunk ~~type is~~ set to peer ~~since we want to direct calls towards a specific~~ IP address~~, defined in the host~~ parameter~~, here set to asterisk-bangkok (see DNS or~~ host ~~table for resolution)~~.		Aplikasi Dial akan me-redirect call ke SIP trunk yang bernama trunk-paris-bangkok, yang di definisikan di defined sip.conf. Tipe trunk di set sebagai peer karena kita menginginkan call untuk langsung ke IP address yang di definisikan di parameter host.

	When a call needs to be established, a SIP INVITE message arrives on the Asterisk based in Bangkok. Asterisk looks in the SIP database for a profile which can accept this call, the IP address is used as the discriminator. The asterisk-paris profile match this requirement. The next action is to direct the call to the specified context and look for an extension match.		When a call needs to be established, a SIP INVITE message arrives on the Asterisk based in Bangkok. Asterisk looks in the SIP database for a profile which can accept this call, the IP address is used as the discriminator. The asterisk-paris profile match this requirement. The next action is to direct the call to the specified context and look for an extension match.

	==Call ~~on the reverse path~~==		==Call di Jalur Sebaliknya==

	Now that a call can be placed from Paris to Bangkok, we need the reverse configuration in order for Bangkok to be able to call Paris. Required modification to our configuration are in bold below:		Now that a call can be placed from Paris to Bangkok, we need the reverse configuration in order for Bangkok to be able to call Paris. Required modification to our configuration are in bold below:

Onnowpurbo at 04:54, 20 January 2014

2014-01-20T04:54:05Z

← Older revision		Revision as of 04:54, 20 January 2014
Line 2:		Line 2:


			Disain asterisk memisahkan antara in-bound dan out-bound call.
			In-bound call membutuhkan untuk di terima oleh SIP dan di arahkan ke context call plan.
			Out-bound call aplikasi dial membutuhkan profile spesifik di konfigurasi SIP atau registrasi ke tujuan.

	~~how to configure SIP trunk with Asterisk~~

			==Context==

			Kesulitan utama dalam konfigurasi SIP trunking di asterisk adalah berbagai parameter di sip.conf dan sip_trunk.conf.

	~~In order to enable voice call exchange between two PBX the straightforward protocol to use is SIP. It can allows device interconnection~~, regardless of their respective role, phone, proxy or gateway. This article explains how simply interconnecting two Asterisk PBX and could be used as a template in order to configure a SIP trunk between an Asterisk PBX and any other one available on the market.		Pada contoh ini, kita akan membuat dua asterisk server

	~~There is a separation by design in Asterisk between in~~-~~bound and out~~-bound calls. An in-bound call requires to be accepted by the SIP stack and oriented towards an call plan context. On the other hand, an out-bound call, the Dial application requires either a specific profile in the SIP configuration or a registration to the counterpart. We use registration when our IP address is dynamically allowed and can change, thus mainly for IP phones.		* asterisk-bangkok
			* asterisk-paris

	~~Context~~		kedua-nya menggunakan IP address statik dalam jaringan yang sama.

	The main complexity for SIP trunking configuration in Asterisk is the role of each parameter in the sip.conf file and sip_trunk.conf. In order to illustrate this article, we will use two Asterisk servers called respectivelly asterisk-bangkok and asterisk-paris. They are both using a static IP address and sharing the same IP network (no NAT in the middle to interfere with the SIP protocol). In order to exchange calls between Bangkok and Paris, we have build prefixes in our dial plan~~, 91 is used to call Bangkok and 98 to call, the routing is really simple.~~		Dalam dial plan

	~~Simple configuration~~		* 91 untuk menelepon Bangkok
			* 98 untuk menelepon Paris.

			==Konfigurasi Sederhana==

			Untuk mengijinkan Paris untuk menelpon Bangkok. Konfigurasi pada masing-masing sip.conf

	~~To start with, we need a configuration to allow calls to flow from Paris towards Bangkok. The configurations on both servers are the following, in sip.conf :~~

	Bangkok		Bangkok


	~~Paris~~

	[trunk-bangkok-paris]		[trunk-bangkok-paris]
	type=peer		type=peer
	host=asterisk-paris		host=asterisk-paris
	context=from-asterisk-paris		context=from-asterisk-paris

			Paris

	[trunk-paris-bangkok]		[trunk-paris-bangkok]
Line 36:		Line 41:
	host=asterisk-bangkok		host=asterisk-bangkok

	~~In the~~ dial plan (extensions.conf) ~~the~~ call ~~is leaving~~ Paris ~~with the following~~:		Pada dial plan (extensions.conf) call akan meninggalkan Paris dengan konfigurasi:

	exten => _91.,1,Set(CALLERID(num)=98${CALLERID(num)})		exten => _91.,1,Set(CALLERID(num)=98${CALLERID(num)})
	exten => _91.,2,Dial(SIP/trunk-paris-bangkok/${EXTEN:2},20,rt)		exten => _91.,2,Dial(SIP/trunk-paris-bangkok/${EXTEN:2},20,rt)

	~~In order to respect our~~ multi site ~~dialing~~ plan ~~structure~~, ~~we modify as the first action the~~ caller ~~identifier by perfixing it with the~~ Paris ~~prefix~~ (98).		Agar memungkinkan struktur multi site dialin plan, kita memodifikasi caller ID dengan menambahkan Prefix Paris (98).


	The Dial application is directeing the call to the SIP trunk named trunk-paris-bangkok, defined in the sip.conf file between [ ]. The trunk type is set to peer since we want to direct calls towards a specific IP address, defined in the host parameter, here set to asterisk-bangkok (see DNS or host table for resolution).		The Dial application is directeing the call to the SIP trunk named trunk-paris-bangkok, defined in the sip.conf file between [ ]. The trunk type is set to peer since we want to direct calls towards a specific IP address, defined in the host parameter, here set to asterisk-bangkok (see DNS or host table for resolution).
Line 47:		Line 53:
	When a call needs to be established, a SIP INVITE message arrives on the Asterisk based in Bangkok. Asterisk looks in the SIP database for a profile which can accept this call, the IP address is used as the discriminator. The asterisk-paris profile match this requirement. The next action is to direct the call to the specified context and look for an extension match.		When a call needs to be established, a SIP INVITE message arrives on the Asterisk based in Bangkok. Asterisk looks in the SIP database for a profile which can accept this call, the IP address is used as the discriminator. The asterisk-paris profile match this requirement. The next action is to direct the call to the specified context and look for an extension match.

	Call on the reverse path		==Call on the reverse path==

	Now that a call can be placed from Paris to Bangkok, we need the reverse configuration in order for Bangkok to be able to call Paris. Required modification to our configuration are in bold below:		Now that a call can be placed from Paris to Bangkok, we need the reverse configuration in order for Bangkok to be able to call Paris. Required modification to our configuration are in bold below:

	Bangkok		Bangkok


	~~Paris~~

	[trunk-bangkok-paris]		[trunk-bangkok-paris]
	type=peer		type=peer
	host=asterisk-paris		host=asterisk-paris
	context=from-asterisk-paris		context=from-asterisk-paris

			Paris

	[trunk-paris-bangkok]		[trunk-paris-bangkok]
Line 71:		Line 76:
	exten => _98.,2,Dial(SIP/trunk-bangkok-paris/${EXTEN:2},20,rt)		exten => _98.,2,Dial(SIP/trunk-bangkok-paris/${EXTEN:2},20,rt)

	Authenticated SIP trunk		==Authenticated SIP trunk==

	It is sometimes required to authenticate calls routed from one PBX to another. Most of the time the main point is billing or tracking calls. Authenticating each call acts as an approval for the associated fees or constraints. Keep in mind that configuration is at the PBX level and not at the phone level and is not as strong as what we can do with X.509 certificates. Authentication for SIP is using a digest exchange and MD5 as the signature, therefore the secret exchanged between both entities is never exchanged in clear over the network.		It is sometimes required to authenticate calls routed from one PBX to another. Most of the time the main point is billing or tracking calls. Authenticating each call acts as an approval for the associated fees or constraints. Keep in mind that configuration is at the PBX level and not at the phone level and is not as strong as what we can do with X.509 certificates. Authentication for SIP is using a digest exchange and MD5 as the signature, therefore the secret exchanged between both entities is never exchanged in clear over the network.
Line 79:		Line 84:
	Bangkok		Bangkok


	~~Paris~~

	[trunk-bangkok-paris]		[trunk-bangkok-paris]
	type=peer		type=peer
Line 88:		Line 90:
	username=trunk-paris-bangkok		username=trunk-paris-bangkok
	secret=strong_password		secret=strong_password

			Paris

	[trunk-paris-bangkok]		[trunk-paris-bangkok]
Line 99:		Line 102:
	The coloured parts should match in both configurations.		The coloured parts should match in both configurations.

	Conclusion		==Conclusion==

	A SIP trunk between two Asterisk PBX is as simple and allows to easily expand the IP telephony network. Interfacing an Asterisk with any other SIP PBX will require something similar, this is the case when connecting to an IP telephony provider. We will see in another article how to take care of the NAT issues and the impact on Asterisk configuration and the network infrastructure.		A SIP trunk between two Asterisk PBX is as simple and allows to easily expand the IP telephony network. Interfacing an Asterisk with any other SIP PBX will require something similar, this is the case when connecting to an IP telephony provider. We will see in another article how to take care of the NAT issues and the impact on Asterisk configuration and the network infrastructure.

Onnowpurbo: /* Referensi */

2014-01-20T04:33:10Z

Referensi

← Older revision		Revision as of 04:33, 20 January 2014
Line 109:		Line 109:

	* http://www.alkia.net/index.php/faqs/137-how-to-configure-sip-trunk-with-asterisk		* http://www.alkia.net/index.php/faqs/137-how-to-configure-sip-trunk-with-asterisk

			==Lebih Dalam==

			* [[VoIP: Asterisk menerima Anonymous Call]]
			* [[VoIP: Asterisk route ke arah IP PBX lain dengan kode area]]
			* [[VoIP: Cara Mengkonfigurasi Trunk di Asterisk]]
			* [[VoIP: Cara Mengkonfigurasi Dial Out Melalui Trunk di Asterisk]]

			===OpenSIPS===

			* [[VoIP: OpenSIPS route ke arah Asterisk]]
			* [[OpenSIPS: Rewrite URI]]





			==Pranala Menarik==

			* [[VoIP]]
			* [[OpenBTS]]

			===Latar Belakang===

			* [[Menjadikan VoIP dan 4G Legal]]
			* [[Sekitar VoIP Rakyat]]
			* [[VoIP: Dasar Hukum Internet Telepon]]
			* [[VoIP: Beberapa Skenario Topologi]]
			* [[VoIP: Pilihan Teknologi Internet Telepon]]
			* [[VoIP: Pengkodean Suara di Jaringan Komputer]]
			* [[VoIP: Konsep Video Conference]]

			===Untuk Pemula===
			* [[VoIP: Kebutuhan Peralatan dan Software]]
			* [[VoIP: Internet Telepon PC ke PC]]

			===Untuk Peneliti / Pencoba===
			* [[VoIP: Bandwidth Internet Telepon]]
			* [[VoIP: Softswitch / Server Internet Telepon]]
			* [[VoIP: Repository Software Internet Telepon]]
			* [[VoIP: Menghubungkan PSTN dan Selular]]

			===Untuk Operator===
			* [[VoIP: Server Video Conference]]
			* [[VoIP: Software dan peralatan client Internet Telepon]]
			* [[VoIP: Penggunaan DAHDI]]
			* [[VoIP: Hardware Client VoIP]]
			* [[VoIP: Hardware Server VoIP]]
			* [[VoIP: Interkoneksi dan Alokasi Nomor Telepon]]
			* [[VoIP: Peering Antar Operator VoIP]]
			* [[VoIP: Menghubungkan PSTN dan Selular]]
			* [[VoIP: Trunk]]

			===Topik Lanjut===
			* [[VoIP: ENUM untuk pengenalan nomor telepon di Internet Telepon]]
			* [[VoIP: Teknik Evaluasi Internet Telepon]]
			* [[VoIP: Troubleshooting]]
			* [[VoIP: Video Conference Server]]

			===Buku Teknologi VoIP===
			* [[Onno W. Purbo]], "VoIP Cikal Bakal Telkom Rakyat", Infokomputer, 2007.
			* http://125.160.17.21/speedyorari/index.php?dir=ebook-voip
			* http://opensource.telkomspeedy.com/speedyorari/index.php?dir=ebook-voip


			[[Category: VoIP]]
			[[Category: Internet Telepon]]

Onnowpurbo: New page: Sumber: http://www.alkia.net/index.php/faqs/137-how-to-configure-sip-trunk-with-asterisk how to configure SIP trunk with Asterisk In order to enable voice call exchange between ...

2014-01-08T02:49:20Z

New page: Sumber: http://www.alkia.net/index.php/faqs/137-how-to-configure-sip-trunk-with-asterisk how to configure SIP trunk with Asterisk In order to enable voice call exchange between ...

New page

Sumber: http://www.alkia.net/index.php/faqs/137-how-to-configure-sip-trunk-with-asterisk

how to configure SIP trunk with Asterisk

In order to enable voice call exchange between two PBX the straightforward protocol to use is SIP. It can allows device interconnection, regardless of their respective role, phone, proxy or gateway. This article explains how simply interconnecting two Asterisk PBX and could be used as a template in order to configure a SIP trunk between an Asterisk PBX and any other one available on the market.

There is a separation by design in Asterisk between in-bound and out-bound calls. An in-bound call requires to be accepted by the SIP stack and oriented towards an call plan context. On the other hand, an out-bound call, the Dial application requires either a specific profile in the SIP configuration or a registration to the counterpart. We use registration when our IP address is dynamically allowed and can change, thus mainly for IP phones.

Context

The main complexity for SIP trunking configuration in Asterisk is the role of each parameter in the sip.conf file and sip_trunk.conf. In order to illustrate this article, we will use two Asterisk servers called respectivelly asterisk-bangkok and asterisk-paris. They are both using a static IP address and sharing the same IP network (no NAT in the middle to interfere with the SIP protocol). In order to exchange calls between Bangkok and Paris, we have build prefixes in our dial plan, 91 is used to call Bangkok and 98 to call, the routing is really simple.

Simple configuration

To start with, we need a configuration to allow calls to flow from Paris towards Bangkok. The configurations on both servers are the following, in sip.conf :

Bangkok

Paris

[trunk-bangkok-paris]
type=peer
host=asterisk-paris
context=from-asterisk-paris

[trunk-paris-bangkok]
type=peer
host=asterisk-bangkok

In the dial plan (extensions.conf) the call is leaving Paris with the following:

exten => _91.,1,Set(CALLERID(num)=98${CALLERID(num)})
exten => _91.,2,Dial(SIP/trunk-paris-bangkok/${EXTEN:2},20,rt)

In order to respect our multi site dialing plan structure, we modify as the first action the caller identifier by perfixing it with the Paris prefix (98).

The Dial application is directeing the call to the SIP trunk named trunk-paris-bangkok, defined in the sip.conf file between [ ]. The trunk type is set to peer since we want to direct calls towards a specific IP address, defined in the host parameter, here set to asterisk-bangkok (see DNS or host table for resolution).

When a call needs to be established, a SIP INVITE message arrives on the Asterisk based in Bangkok. Asterisk looks in the SIP database for a profile which can accept this call, the IP address is used as the discriminator. The asterisk-paris profile match this requirement. The next action is to direct the call to the specified context and look for an extension match.

Call on the reverse path

Now that a call can be placed from Paris to Bangkok, we need the reverse configuration in order for Bangkok to be able to call Paris. Required modification to our configuration are in bold below:

Bangkok

Paris

[trunk-bangkok-paris]
type=peer
host=asterisk-paris
context=from-asterisk-paris

[trunk-paris-bangkok]
type=peer
host=asterisk-bangkok
context=from-asterisk-bangkok

In Bangkok we also need a specific set of action to allow a call, prefixed by 98 to be routed towards Paris:

exten => _98.,1,Set(CALLERID(num)=91${CALLERID(num)})
exten => _98.,2,Dial(SIP/trunk-bangkok-paris/${EXTEN:2},20,rt)

Authenticated SIP trunk

It is sometimes required to authenticate calls routed from one PBX to another. Most of the time the main point is billing or tracking calls. Authenticating each call acts as an approval for the associated fees or constraints. Keep in mind that configuration is at the PBX level and not at the phone level and is not as strong as what we can do with X.509 certificates. Authentication for SIP is using a digest exchange and MD5 as the signature, therefore the secret exchanged between both entities is never exchanged in clear over the network.

Authentication is based on login and password validation. The password is called secret in the configuration and login is called username. The username is not mandatory and can be derived from the configuration, but specifying it is clearer and easier to troubleshoot in case of. Configuration changes are availabe below:

Bangkok

Paris

[trunk-bangkok-paris]
type=peer
host=asterisk-paris
context=from-asterisk-paris
username=trunk-paris-bangkok
secret=strong_password

[trunk-paris-bangkok]
type=peer
context=from-asterisk-bangkok
host=asterisk-bangkok
username=trunk-bangkok-paris
secret=strong_password

The coloured parts should match in both configurations.

Conclusion

A SIP trunk between two Asterisk PBX is as simple and allows to easily expand the IP telephony network. Interfacing an Asterisk with any other SIP PBX will require something similar, this is the case when connecting to an IP telephony provider. We will see in another article how to take care of the NAT issues and the impact on Asterisk configuration and the network infrastructure.

==Referensi==

* http://www.alkia.net/index.php/faqs/137-how-to-configure-sip-trunk-with-asterisk