Sip trunk теле2


SIP Trunk Nedir ve Nasıl Çalışır?

Bir işletme sahibi veya çalışanı olarak harcamalarınızı azaltmanın yollarını arıyorsanız tam size göre bir yazımız var.

Bu yazımızda SIP Trunk’a yüksek seviyeli bir bakış ile birlikte SIP Trunk’ın nasıl çalıştığını anlatacağız. 

İşletmelerde genelde harcamalar önem sırasına göre dizilir. Yönetim ile birlikte bölüm sorumlularının, kullanılan tüm hizmetlerin ince detay ve bilgilerine sahip olamaması anlayış gösterilecek bir durumdur. Belki de SIP Trunk’ı duydunuz veya arkadaşlarınızdan biri işletmesinde bunu kullanarak maliyetlerini düşürüyor ama siz servis sağlayıcılar ile iletişime geçmeden önce belki de daha fazla bilgi sahibi olmak istiyorsunuz. 

SIP Trunk Nedir?SIP (Türkçe’ye çevirirsek Oturum Başlatma Protokolü), IP üzerinden ses iletimine erişiminde kullanılan en yaygın protokoldür. İki uç nokta arasında gerçek zamanlı ses ve/veya video görüşme sağlamakta kullanılan bir arabirimdir. Diğer bir deyişle SIP, ikili veya çok partili bir aramada, bir IP ağı içinde bir veya birden çok tarafın seanslarını yaratan, değiştiren ve sonlandıran bir teknolojidir.  

SIP Trunk, analog bir telefon hattının sanal versiyonudur. SIP Trunk’ları kullanarak bir SIP sağlayıcısı sizin telefon satralınıza bir, iki veya N tane kanalı bağlayabilir ve sizin internet aracılığıyla yerel, uzun mesafe ve uluslararası aramalar yapmanızı sağlar. Eğer ofisinizde kurulu bir telefon santralı varsa, bir SIP Trunk sağlayıcısı sizi bağlayabilir ve sizin var olan sisteminiz üzerinden eş zamanlı arama sayısı kısıtlaması olmadan dışarıya aramalar yapmanızı sağlar.

SIP Trunking’in Maliyeti Nedir? SIP Trunking’in maliyeti sizin işletme ihtiyaçlarınıza bağlı olarak değişir. Fakat bu teknolojinin iletişim maliyetlerinizi düşürmek için ortaya çıktığını düşünürsek, mevcut telefon hattı kullanımına göre çok daha düşük sabit ücret ve görüşme ücretleri sunacağını belirtebiliriz.

Bazı SIP Trunking sağlayıcıları kanalize olan planlara ek olarak bazı sayaçlı planlar sunarlar.

Sayaçlı SIP Trunking’in teslim edilmesinin ardından kullanıma bağlı olarak faturalandırılır, yani her dakika faturaya yansır. Sayaçlı SIP Trunking’de her aramanın her dakikası faturalandırıldığından dolayı eş zamanlı arama sayısında sınır yoktur ve oldukça esnektir. Sayaçlı hizmetler işletmelerin dinamik olarak arama yapmalarına ve her ek kullanım için ödeme yapmalarına olanak sağlar. 

Kanalize edilen SIP Trunking, her kanal ve arama üzerinden sınırsız olarak içeri ve dışarı, yerel ve uzun mesafe aramaları yapmanızı sağlayan ön ödemeli bir seçenektir. Her kanal tek bir arama yapma veya alma kapasitesini sağlar. Kanallarınızın hepsini doldurduğunuzda ek olarak arama yapamaz veya alamazsınız. Kanalları servis sağlayıcınız ile iletişim kurarak arttırabilirsiniz. Bu tip bir SIP Trunk hizmeti, işletmelerin kolay bir şekilde telekom harcamalarını kontrol etmelerine izin verir.

Sayaçlı SIP Trunk’ta Ne Kadar Eş Zamanlı Arama Yapılabilir? Peki ya Bant Genişliği?Sınırlama, internet bant genişliği kapasitesine bağlıdır. Her bir sıkıştırılmamış arama, yaklaşık olarak 85 Kbps(kullanılan kodek türüne göre değişebilir) bant genişliği kullanır. Birçok internet bağlantısında yükleme hızınız normal olarak indirme hızınızdan düşüktür, bu yüzden bağlantınızın destekleyeceği arama sayısına karar verirken yükleme hızınızı da göz önünde bulundurmalısınız. Örneğin, yükleme hızınız 4Mbps ise maksimum arama sayısı 47 olacaktır (4,000,000/85,000). 

En Meşgul Anlardaki Eş Zamanlı Arama SayısıXSaniye Başına 85 Kilobayt=SIP Trunking için Gerekli Olan Bant Genişliği (Saniye Başına Megabayt Olarak)

Not: Sıkıştırılmış Ses kullanarak bant genişliği kullanımını yarı yarıya düşürmek mümkündür. Bu yazıyı okuyanlar bunları da okudu;10 Soruda Çağrı MerkeziNAT Nedir ve VoIP ile Nasıl Çalışabilir?IP Santralı Hackerlardan Korumak

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Telekom Defteri: SIP Trunking Nedir?

SIP Trunking hizmetini ve bunu saglayan irili ufakli bircok firmanin haberlerini voip kanallarinda siklikla goruyoruz. Analizlerde SIP Trunking'in sirketlere getirdiği yararlardan bahsediliyor. Peki nedir bu SIP Trunking? Voip dunyasinda genel olarak 3 temel tanimina rastlayabilriz - Firmalara servis saglayicilar tarafindan sunulan ve cevrim anahtarlama (Circuit switching) yerini alan PSTN baglantisidir. - Sirketin SIP serverinda voicemain server, call center ve application server gibi diger voip serverlara baglanti amaciyla tanimlanmis bir port - IP-PBX ler arasinda TDM kanallarin yerini almasi icin kurulan baglanti.

Bu uc tanimida kapsayan genel bir tanimi da su sekilde yapabiliriz;Bir SIP server kendisine gelen SIP requestleri istedigi gibi isleyebilir. Ornegin alinan bir SIP requestin nasil route edilecegine, nasil authenticate edilecegine, PSTN'e yonlendirilecegine, requestin sonlandirilacagina veya hangi headerlarin eklenip hangilerinin cikarilacagina  karar verebilir. Bir SIP request'e (SIP request headerindaki "route" alani ile tanimlanmis) onceden tanimlanmis islemlerin uygulanmasina SIP trunking, islemler toplami ve SIP request tanimina ise SIP trunk denir. Ornegin;

- PSTN baglanti Trunk'i: onceden tanimlanmis bir SIP request server'a ulastiginda ilk olarak TLS mutual authentication uygulanir ve sadece onceden provize edilmis kullanicilarin devam etmesine izin verilir. Daha sonra server'a ait rouing tablolarinda tanimlanmis PSTN gateway'e yonlendirilir.- Filtering Trunk: Cogunlukla SBC (Session Border Controller) ya da benzer serverlar tarafindan kullanilir. Gelen SIP request headerlarini onceden belirlenmis header tablosu ile karsilastirir ve fazla olan headerlari requestten cikartarak requesti bir sonraki noda gonderir.- Voicemail Trunk: Voicemail server tarafindan kullanilir ve kullanicilari gecerli bir sertifika ile authenticate eder.

Gercek dunyada ise SIP trunk'in genel kullanim alani sirketlere tek bir IP baglantisi uzerinden coklu PSTN baglanti hizmeti vermektir. Bilindigi gibi gunumuzde voip hizmeti PSTN'e gore daha ucuz ve daha kaliteli ses baglantisi sunmaktadir. Bu yuzden cogu buyuk firma sirket icinde IP-PBX araciligi ile voip iletisimine gecmistir. PSTN ile baglanti kumak icin ise ITSP (Internet telephony service provider - Internet telefonu servis saglayicisi) sirketlerinin verdigi PSTN hizmetinden yararlanmaktadir. Boylece sirket ses hismeti icin ayrica TDM hat kiralamak ya da PSTN gateway kurmak zorunda kalmaz. Sekil 1 basit bir SIP PBX trunking ornegi gosterilmistir.

Sekil 1 - IP PBX Trunking
 Sekilde de goruldugu gibi sirket hem dis dunyadaki PSTN agina erisebilmekte hem de internet uzerinden ofis disindaki IP telefonlar ile baglanti kurabilmektedir. ITSP firmasi gereken QoS saglamakla yukumludur. Dikkat edilmesi gereken bir nokta internet uzerinden ulasilan IP telefonlar'a ofis icerisinde IP PBX araciligi ile sunulan tum servislerin sunulamamasidir.

Bir diger alternatifi ise hosted PBX trunking olarak adlandirilmaktdir. Bu yontem oncekine gore daha ucuzdur cunku musteri IP PBX sistemini ofise kurmak yerine baska bir firmadan kiralar. Sekil 2 de hosted PBX trunking gosterilmistir.

Sekil 2 - Hosted IP PBX Trunking
Sekilde de goruldugu gibi ofis icindeki ve disindaki telefonlar kiralanan PBX sistemine baglandigi icin hepsi aynı servislerden yararlanmaktadir. Sistemin dezajantaji ise ofisten direk internete baglanti kuruldugu icin bir onceki sistemde ITSP tarafindan sunulan QoS hizmetinden yaralanilamamasidir.

Ozet olarak musterilerin SIP trunking kullanimi ile elde ettigi ajavtajlar;

  • BRI ve PRI aboneliklerinden ya da lokal PBX PSTN gateway kurulumundan kurtulur.
  • Ofis agi basitlesir.
  • Yeni ofis eklemek ya da ofisi tasimak kolaylasir.
  • Bircok BRI/PRI aboneligi yerine tek bir IP aboneligi yeterlidir.
  • Ses ve veri iletisimi tek bir IP kanali paylasir.
Servis saglayicilara getirdigi avantajlar ise;
  • T1/E1 altyapi harcamalari azalir.
  • Haftalar suren provizyon islemlerini birkac saatte yapabilir.
  • UC, FMC gibi yeni nesil SIP uygulamalari musterilerine sunabilir.
What is a Session Initiation Protocol (SIP) Trunk Anyway?SIP Trunking — QoS at Its Simplest

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SIP Trunk Config

The following describes the IP Office configuration required to route calls to a CS1K (with NRS) via SIP. This example was built between a CS1K 5.5 and IP Office 8.1, and functions well. The configuration is fairly standard, and should work for other releases.

The following key settings are used in this example:

  • LAN1 (LAN): 10.10.10.5
  • LAN2 (WAN): 172.16.1.1
  • Admin PC: 10.10.10.100 (via LAN1 interface)
  • CS1K Node IP: 11.11.11.17 (TLAN, via LAN2 interface)
  • Primary Signalling Server IP: 11.11.11.15
  • ISP Gateway: 172.16.1.2 (path to the CS1K TLAN)
  • CS1K SIP Domain Name: cs1kSIP.com

In the CS1K NRS, the IP Office is added as a "Static SIP Endpoint", with "Name: Mikes_PBX", and "IP address: 172.16.1.1". CS1K NRS Routing entries must be added to route CS1K calls to the IP Office, and this is not covered here.

You can also download this configuration , and edit the IP addresses and domain names (within Manager) for your application.

The IP Office network must be reachable from the CS1K TLAN, and visa-versa. Confirm this with some Ping tests.

Click the links with to display the relevant screen shot.

System Settings:

1License

  • Enter license codes for Essential Edition, and SIP Trunk Channels (SIP Trunk instances = number of concurrent calls licensed, or allowed) Note: License keys are based on the licensed feature AND the system's Dongle Serial Number.

2System > System

  • Name: must match the SIP Endpoint name on the NRS, eg, Mikes_PBX
  • TFTP, Manager PC, File Writer, and Time Server all set to the Admin PC address
  • Local Number length: set to extension digit length, eg, 4
  • Time Setting Config Source: set to VoiceMail Pro/Manager

3System > LAN1 > LAN Settings

  • Enter the LAN1 IP address and subnet mask (the LAN address - hooked to local network, for IPO administration)

4System > LAN1 > VoIP

5System > LAN2 > LAN Settings

  • Enter the LAN2 IP address and subnet mask (the WAN address - hooked to Gateway/ISP, for SIP trunk path)

6System > LAN2 > VoIP

  • Check "SIP Trunks Enable" only.

7System > Telephony > Telephony

  • Check A-Law (in Europe) for Switch and Line. (Optional: check "Restrict Network Interconnect" to enable Network Type: Public/Private in SIP Line)

8System > CODECS

  • Select only the CODEC's needed: ie, G.729, and G.711, and position in order of preference. Note: to use G.711, the CS1K zone setting must be set to "Best Bandwidth".

SIP Line Settings:

1LINE > SIP Line

Create a new SIP line... Right-click LINE, select NEW, select SIP (eg, 17).

Then, click the new line, eg, 17, and....

  • Line number: auto assigned
  • ITSP Domain Name: the CS1K SIP domain name, eg, cs1kSIP.com
  • Call Routing Method: change To HEADER
  • Send Caller ID: change to DIVERSION HEADER

Click on the new SIP line, and continue to configure it:

2SIP LINE > Transport

  • ITSP Proxy Address: This depends on the CS1K release! - when the CS1K is at Rls 5.5 and below: enter the CS1K Node IP address. - when the CS1K is at Rls 7.0 and above: enter the Primary NRS IP address.
  • Layer 4 Protocol: change to UDP
Note: in the CS1K NRS, the IP Office is a "Static SIP Endpoint".

3SIP LINE > SIP URI

Click ADD, and enter the following:

  • Local URI: *
  • Contact: *
  • Display Name: *
  • Incoming Group: enter the local SIP line number, eg, 17.
  • Outgoing Group: enter the local SIP line number, eg, 17.
  • Max Calls per channel: this must match your SIP license "instances".

4SIP LINE > VoIP

  • Codec's should mirror CODEC's selected in System > Codec's
  • Change "Fax Transport Support" to G.711. - omitting this will inhibit faxing!

User Settings:

Optional steps, not conducive to SIP trunk programming!

1USER

What ever you enter under "No User" is applied to all users! eg, Source Numbers, add Enable_OTT

Button Programming: (Optional) for example....

  • Button 1, Appearance, a=
  • Button 2, Appearance, b=
  • Button 3, Follow Me To
  • Button 4, Call Pickup Members, mypug

2HUNT Groups

  • Create Pickup Groups in HuntGroup, Ring Mode: Collective, and ADD the required extensions.

Routing Settings:

1Short Codes

Add short codes for dialing patterns (access codes) as necessary in the format:

  • Add short codes - as required - to dial CS1K four digit extensions: - Code: 4XXX - Feature: Dial - Telephone Number: 4N"@11.11.11.17"   - this is the CS1K NODE IP address - Line Group: 17 - this is the SIP line group created earlier.
Note: Wildcard X (match one digit) goes in the Code, wildcard N (match any digits) goes in the Telephone Number.
  • Add short codes - as required - to dial CS1K access codes: 9, etc. eg: - Code 9N; - Feature: Dial - Telephone Number: 9N"@11.11.11.17"   - note the quotes! - Line Group: 17
Note: A semi-colon denotes "end-of-dialing" in the Code string.

2Incoming Call Route > Standard

  • Only one! Edit the "Any Voice" entry (zero) to the SIP Line Group ID, eg, 17.

3Incoming Call Route > Destinations

  • Enter a period (full-stop) in the Destination column, leave the Fallback Extension blank. A destination of period causes calls to ring the dialed extension.

4IP Routes

  • Add a default Route: 0.0.0.0 (this is wild, and means 'everything')
  • Enter the Gateway IP address (ISP gateway)
  • Destination is LAN2: the WAN network port.

And finally, when its all working, you can enjoy a cold one in the land of contentment!

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SIP Trunking Configuration Guides - Twilio

The following Configuration Guides are intended to help you connect your SIP Infrastructure (IP-PBX, SBC, etc) to a Twilio Elastic SIP Trunk.

Be aware, due to the large number of versions, variations, add-ons, and options for many of these systems, the settings you see may differ from those shown in our Configuration Guides. As such, these documents are intended as general guidelines, rather than configuration templates. There is an assumption of familiarity with your network and SIP infrastructure, and how they work.

Twilio cannot provide direct support for third-party products; you should contact the manufacturer for your PBX/SBC for assistance in configuring such products.

If you wish to share your PBX or SBC configuration guide to help us improve this section for other users, kindly submit them or any corrections to the existing guides to [email protected]

Asterisk IP-PBX

Assuming you have Asterisk already set up as your IP-PBX, with one or more telephones configured and running calls between them, the following guide provides detailed step-by-step instructions of how to configure your Trunk and your Asterisk IP-PBX. Optionally, Twilio Elastic SIP trunking also provides Secure Trunking (SIP TLS and SRTP), see guide for configuration details.

Click here to download the Asterisk Interconnection Guide

FreeSwitch IP-PBX

Assuming you have FreeSwitch already set up as your IP-PBX, with one or more telephones configured and running calls between them, the following Interconnection Guide provides you with step-by-step instructions to use FreeSwitch PBX with your Twilio Elastic SIP Trunk.

Click here to download the FreeSwitch PBX Interconnection Guide

FreeSwitch using Secure Trunking

This is supported. At this time there is no guide published but reach out to support if you have any questions.

Twilio Elastic SIP trunking also provides Secure Trunking (SIP TLS and SRTP).

This guide provides the configuration steps required to implement FreeSwitch PBX using a Twilio Elastic SIP trunk using Secure Trunks.

Click here to download the FreeSwitch PBX with Secure Trunking Interconnection Guide

3CX

Click here to see 3CX guide to configuring Twilio Elastic SIP Trunks

Assuming you have your 3CX already set up with one or more telephones configured and running calls between them, the following highlights specific configuration for use with your Twilio SIP Trunk.

  • Add a new VoIP Provider account in the 3CX phone system: "Twilio"
    • Set the SIP server hostname to: example.pstn.twilio.com
    • Set your Authentication ID/username and password (as you configured in your user credentials on your Twilio Trunk)
  • DID’s and Inbound Call Identification: Enter your Twilio numbers under the "DID" tab.
  • "Advanced" under "Codec priorities" only include G711 U-law
  • Create Outbound Call Rules: setting calls to numbers with a length of 10, and also prepend a "+1". This will ensure E164 formatting.

Click here to download the 3CX Interconnection Guide

Elastix

If you want to use Elastix IP-PBX with your Twilio Trunk, the following guide provides detailed step-by-step instructions of how to configure your Trunk and your IP-PBX.

Click here to download the Elastix Interconnection Guide

Free PBX

Assuming you have Free PBX already set up as your IP-PBX, with one or more telephones configured and running calls between them, the following highlights specific configuration for use with your Twilio Trunk.

Click here to download the Free PBX Interconnection Guide

GrandStream UCM

The following Interconnection Guide provides you with step-by-step instructions to use GrandStream UCM with your Twilio Elastic SIP Trunk.

Click here to download the Grandstream Interconnection Guide

Mitel MiVoice Business 7.2

The following guide is not maintained by Twilio. Please see Mitel Knowledge base for latest guide.

Click here to download the Mitel MiVoice configuration Guide

Acme Packet SBC

Assuming you have your SBC already set up with your IP-PBX, with one or more telephones configured and running calls between them, the following highlights specific configuration for use with your Twilio trunk.

Make sure you have your Network & Physical Interfaces appropriately configured.

Configure your Trunk SIP Interface towards Twilio:

sip-interface state enabled realm-id OUTSIDE description sip-port address X.X.X.X (add this to your Twilio IP ACL) port 5060 transport-protocol UDP tls-profile allow-anonymous agents-only ims-aka-profile carriers trans-expire 0 ...

Configure your Session Agent towards Twilio:

session-agent hostname example.pstn.twilio.com ip-address port 5060 state enabled app-protocol SIP app-type transport-method UDP realm-id OUTSIDE egress-realm-id description Twilio carriers allow-next-hop-lp enabled constraints disabled ...

The second example presented here illustrates adding +1 to called numbers (To and Request-URI headers) for all SIP trunk endpoints in a particular realm.

Firstly, define the session-translation with a called rule:

session-translation id addCalledPlusOne rules-calling rules-called addPlusOne

Then define the rule to append +1:

translation-rules id addPlusOne type add add-string +1 add-index 0 delete-string delete-index 0

Lastly, apply the translation as outgoing to the SIP trunk realm:

realm-config identifier OUTSIDE ... in-translationid out-translationid addCalledPlusOne ...

Set the preferred codec to G711 mu-law. In the example below, the Net-Net SD manipulates the codec list for all PBXs in the PBXs realm such that PCMU appear first in the media descriptor offered to the SIP trunk:

realm-config identifier PBXs ... options preferred-codec=PCMU ...

Cisco ISR (Cisco 28xx, 29xx, 38xx, 39xx, 43xx etc.)

Assuming you have your ISR already set up with one or more telephones configured and running calls between them, the following highlights specific configuration for use with your Twilio Trunk.

If you use credentials for outbound calls, you must use the B2BUA built into Cisco IOS:

sip-ua authentication username anniebp password 7 15431A0D1E0A1C171060302610 realm sip.twilio.com registrar dns:example.pstn.twilio.com expires 3600 sip-server dns:example.pstn.twilio.com !

Update your Trust List:

voice service voip ip address trusted list ipv4 54.172.60.0/23 ipv4 54.171.127.192/26 ipv4 54.65.63.192/26 ipv4 54.169.127.128/26 ipv4 54.252.254.64/26 ipv4 177.71.206.192/26 allow-connections sip to sip !
  • TWILIO accepts 'Early offer' only, so Cisco users/partners would have to force call as Early offer.
  • Use SIP normalization profile to change 'From' header to include IP address of CUBE router instead of DNS name

Ensure all numbers use full E.164 format, so transform all outbound calls to E.164 before sending to Twilio. The rules below are doing 2 things: changing this outbound call from 919803331212 to +19803331212 and changing the ANI from 4002 to 9802180999.

voice translation-rule 1 rule 1 /^91/ /+1/ ! voice translation-rule 2 rule 1 /4004/ /9802180971/ rule 2 /4002/ /9802180999/ rule 3 /4005/ /9802180980/ ! ! voice translation-profile twilio translate calling 2 translate called 1 !

Lastly, you may have a dial-peer with 91[2-9]..[2-9]...... in order to catch the calls. You can see the translation profile that is applied to translated the number to E.164. Also ensure G.711 codec is used. The ‘session target sip-server’ is what target the sip B2BUA configured above with the ‘sip-ua’ command.

dial-peer voice 200 voip translation-profile outgoing twilio destination-pattern 91[2-9]..[2-9]...... session protocol sipv2 session target sip-server dtmf-relay rtp-nte sip-kpml sip-notify codec g711ulaw no vad !

Sonus E-SBC 5000 using Microsoft Lync

Assuming you have your E-SBC already set up, the following highlights specific configuration for your Sonus E-SBC for interworking with Microsoft's Lync Server 2013 environment using your Twilio Trunk.

Click here to download the Sonus Microsoft Lync Interconnection Guide

Audiocodes SBC

Assuming you have your SBC already set up with your IP-PBX, with one or more telephones configured and running calls between them, the following highlights specific configuration for use with your Twilio Trunk.

Make sure you have an IP Group defined with:

IPGroup_Description: Twilio IPGroup_SIPGroupName: domain.pstn.twilio.com ...

Define your Proxy IP:

[ ProxyIp ] FORMAT ProxyIp_Index = ProxyIp_IpAddress, ProxyIp_TransportType, ProxyIp_ProxySetId; ProxyIp 1 = "54.172.60.0/23:5060", 0, 2; ProxyIp 2 = "54.171.127.192/26:5060", 0, 2; ProxyIp 3 = "54.65.63.192/26:5060", 0, 2; ProxyIp 4 = "54.169.127.128/26:5060", 0, 2; ProxyIp 5 = "54.252.254.64/26:5060", 0, 2; ProxyIp 6 = "177.71.206.192/26:5060", 0, 2; [ \ProxyIp ]

Have a Coders Group with:

CodersGroup0_Name: g711ulaw64k CodersGroup0_pTime: 20 CodersGroup0_PayloadType: 0

You will also need to define your IP Profiles & Routing rules.

AudioCodes using Microsoft Lync

Assuming you have your E-SBC already set up, the following highlights specific configuration for your AudioCodes E-SBC for interworking with Microsoft's Lync Server 2013 environment using your Twilio Trunk.

Click here to download the Audio Codes using Microsoft Lync Interconnection Guide

EdgeMarc

Assuming you have your SBC already set up with one or more telephones configured and running calls between them, the following highlights specific configuration for use with your Twilio Trunk.

Navigate to "VoIP">"SIP" to configure the SIP server info for Twilio. Enter in the SIP Server FQDN assigned for these services under the SIP Server Address field. Fill in the SIP Server Domain field with the proper Twilio domain.

Note: Make sure to check the "Limit Inbound to listed Proxies" and "Limit Outbound to listed Proxies" boxes to help prevent fraudulent activity sourced from a LAN side PBX or a WAN side DoS attack.

Navigate to "VoIP ALG" and then "B2BUA" to configure the SIP Trunk registration with the soft-switch (between the EdgeMarc and the WAN side soft-switch), the PBX for SIP registration mode (between the PBX and LAN side of the EdgeMarc), inbound rule (for sending SIP messages from the WAN side of the EdgeMarc to the PBX) and outbound rule (for sending the SIP messages from the EdgeMarc to the WAN soft-switch). RFC-4904 support will be handled by applying header manipulation action rules to the matched outbound rules.

Configuring the PBX for SIP registration mode (between PBX and the EdgeMarc). From the "Trunking Devices" section:

  • Click the "New Row" button to get to a new entry for a Trunking Device.
  • Enter a PBX name in the "Name" field.
  • Select the correlating PBX from the drop-down list of the "Model" field.

  • Select IP Registration mode by selecting the radio button for using the IP field and Port field.
  • Enter the PBX IP in the "IP" field.
  • Enter 5060 in the "Port" field. Click "Update" to create a Trunking Device for PBX. Click "Submit" at the bottom of the page to send the config to the EdgeMarc.

Configure the EdgeMarc default inbound rule (for sending the SIP messages from the EdgeMarc to the PBX). This is required in order for non-pilot DIDs to reach the PBX.

From the Actions section:

  • Click the "New Row" button to get a new entry for creating an inbound action.
  • Enter the action name in the "Name" field.
  • Select the radio button of "Trunking Device".
  • Select the PBX from the drop-down list next to "Trunking Device".

  • Click the "Update" button.

From the Match section:

  • Click the "New Row" button to get a new entry for an inbound rule.
  • Select "Inbound" in the "Direction" field.
  • Select the radio button of "Default".
  • Select "InboundAction" from the drop-down list of the "Action" field.
  • Click the "Update" button.

From the Match section:

  • Click the "New Row" button to start a new entry for an outbound rule.
  • Select "Outbound" in the "Direction" field.
  • Select the radio button of "Pattern match", select "Calling" from the drop-down list and enter a "." or match the partial DID map (for example, if there is 6785551111-1115, then use 678555111X) in the "Pattern match" field to match any calling numbers.
  • Select "Any" from the "Source" field.
  • Select OutboundAction1 from the drop-down list of the "Action" field.
  • Click the "Update" button.

  • Click "Submit" at the bottom of the page to send the config to EdgeMarc.

inGate SIParator

The following Interconnection Guide provides you with step-by-step instructions to use inGate SIParator E-SBC with Twilio Elastic SIP Trunk. Optional steps to configure SIP over TLS and SRTP (Secure Trunking) are also included in this guide.

Click here to download the inGate Interconnection Guide

Sansay

Assuming you have your SBC already set up with one or more telephones configured and running calls between them, the following highlights specific configuration for use with your Twilio Trunk.

xCally Call Center

The following Interconnection Guide provides you with step-by-step instructions to use XCally Call Center your with Twilio Elastic SIP Trunk.

Click here to download the xCally Interconnection Guide

www.twilio.com

Asterisk PBX SIP Trunk Configuration | VoiceHost

How to configure SIP Trunking for Asterisk IP PBX based systems

Our service is 100% compatible with Asterisk using either standard SIP registration, or IP authentication where SIP trunks are configured as such. Asterisk is the base software behind many open-source PBX distributions, including FreePBX, Trixbox and Elastix, and is also the enabler behind many other ITSPs and commercial PABX manufacturers.

This guide is aimed at Asterisk's SIP stack via the sip.conf file, it does not deal with realtime configuration via a back-end database however the principles are the same and the appropriate options should be transposed as such. Configuration of SIP trunks for the PJSIP stack within Asterisk versions 12, 13 and greater is forthcoming and will be posted in a separate FAQ entry.

Once you have setup and configured Asterisk, you can use the following details to start making calls.

STxxxxxTxxx = Your SIP trunk usernameppppp = Your SIP trunk password

These details are visible on your customer control panel if you have been allocated a SIP trunk.

Asterisk sip.conf SIP configuration using SIP registration

You should have the following in your sip.conf file:

[general]

register => STxxxxxTxxx:[email protected]

[voicehost]

host=st.sipconvergence.co.uk

defaultuser=[SIP Trunk username]

secret=[SIP Trunk password]

type=peer

disallow=all

allow=alaw

allow=ulaw

allow=g722

allow=gsm

context=inbound

insecure=port,invite

qualify=15000

These options are explained as follows:

register => STxxxxxTxxx:[email protected] ; Specifies Asterisk registers with given username, password and domain hostname or IP address, note this is not needed for SIP trunks configured for IP authentication

host=st.sipconvergence.co.uk ; VoiceHost domain hostname or public IP address, see http://help.voicehost.co.uk/content/6/9/en/sip-trunk-general-settings-and-pbx-compatibility.html for information on VoiceHost SIP trunk domains & IP addresses

defaultuser=REPLACEME ; Username of the VoiceHost SIP trunk is entered here (less the <> characters), note this is not needed for SIP trunks configured for IP authentication

secret=REPLACEME ; Password of the VoiceHost SIP trunk is entered here (less the <> characters), note this is not needed for SIP trunks configured for IP authentication

type=peer ; This specifies the SIP endpoint to be contacted for call handling, type=peer is usually used for SIP trunk connectivity, type=friend for IP phones authenticating per call to Asterisk

disallow=all ; Codec settings specifying no codecs, usually this is first in the list with allow= fields below it which then specify codecs permitted

allow=alaw ; Specify G.711a on INVITE and responses for calls

allow=ulaw ; Specify G.711u on INVITE and responses for calls

allow=g722 ; Specify G.722 on INVITE and responses fourth in priority list

allow=gsm ; Specify GSM 6.10 Full Rate on INVITE and responses for calls

context=inbound ; Context within the dialplan to which calls into this trunk will be matched

insecure=port,invite ; Specifies Asterisk to accept the call without requesting authorisation

qualify=15000 ; Sends OPTIONS SIP request to the host value every 15 seconds (value is in milliseconds, e.g. 15000 is 15 seconds)

 

For Asterisk major versions 1.6 and older, replace the "defaultuser=" option with "username=", otherwise authentication on outbound calls will fail.

You should have something similar to the following in your extensions.conf file to receive inbound calls:

[inbound]exten => _X.,1,Dial(SIP/100) ; Directs incoming calls via SIP to extension 100 within the dialplan

Please note: Incorrectly setting the 'context' can lead to vulnerabilities open to attack from hackers. Please check your configuration thoroughly as VoiceHost does not accept any liability for fraudulent use of your Asterisk system.

Inbound telephone numbers need to be mapped in the Asterisk system in E.164 format to receive calls correctly, e.g. +441603904090

If you require any further assistance, please feel free to contact VoiceHost Support who will be happy to answer any questions you may have.

 

Troubleshooting Asterisk configurations

Calls to VoiceHost error with "all circuits busy" or "congestion" - this is the default configuration of Asterisk regardless of the actual error generated, unless Asterisk is updated to send back the real error rather than the changed error. This error most commonly occurs when the call is not authenticating properly, at which point check the above in the SIP trunk configuration within Asterisk, and swap username= for defaultuser= to see if this solves the issue. Just because a trunk is showing as registered does not mean it will authenticate correctly. For inbound calls, ensure the mapping is correct and a viable destination set in the PBX, e.g. voicemail or auto-attendant.

Outbound calls to VoiceHost fail with SIP error 488 (Not Accepted Here) or I-SUP errors 58 (bearer capability not available) or 88 (incompatible destination) - check the codecs allowed in the SIP trunk configuration above, VoiceHost only supports:

  • alaw (G.711 alaw)
  • ulaw (G.711 ulaw)
  • g722 (G.722 wideband codec)
  • gsm (GSM 6.10 Full Rate)

If a codec is defined in Asterisk that is not one of the above, or is offering a differing sample rate or interval rate (e.g. 8000/20i - 8000Hz at 20ms cannot interwork with 16000/30i - 16000Hz at 30ms) and the call attempted the call will fail. The codecs in the SIP trunk configuration within Asterisk need to be aligned to use one of the above codecs. As a minimum, offer both alaw and ulaw codecs when sending and receiving calls for widest compatibility with fixed line and mobile operators. alaw should be first in the list for calls to and from the UK PSTN.

Inbound calls to Asterisk fail with SIP error 408 (Request Timeout) - check the inbound number is mapped in the system correctly, if necessary the SIP trunk on the portal can be configured to strip the plus when sending calls to Asterisk, e.g. if Asterisk is configured to use plus logic somewhere else. Check the trunk is registered. Ascertain how long the 408 error took to come back, if it was immediate the trunk is usually unregistered, if it took a few seconds the number is usually not mapped in correctly.

Calls fail to VoiceHost with SIP error 503 (Service unavailable), I-SUP errors 34 (No circuit available) or 38 (Network out of order) - If the VoiceHost platform replies back with SIP error 503 it usually means one of two things: the SIP trunk has exceeded the Calls-Per-Second (CPS) limit, or the gateway has rejected the call for some other reason.

Calls fail into Asterisk with SIP error 403 Forbidden (Bad Auth) - authentication is being requested from a VoiceHost gateway that is not recognised or not authorised in the SIP configuration, this usually happens if the SIP trunk configuration is set with insecure=very and host=st.sipconvergence.co.uk or st.voicehost.co.uk. Change type=peer, insecure=port,invite within the SIP configuration, and re-test inbound calls upon successfully re-registering. If calls continue to fail change the host= value from the domain to a public IP address under the domain

www.voicehost.co.uk

FAQs for SIP Trunk - SIP Trunk

About SIP Trunk

What is SIP Trunk?

SIP trunk is a service of delivering telephone services over the Internet to customers that have SIP enabled IP-PBX or VoIP devices. SIP utilizes both Voice over Internet Protocol and Session Initiation Protocol (SIP) and it replaces traditional telephone lines.

 

VoIPVoIP SIP Trunk Service

What services does VoIPVoIP provide?

VoIPVoIP offers SIP Trunk service consisting of Origination (inbound) and Termination (outbound) calling. We also offer DID telephone numbers in the US, Canada and from 50+ countries.

How quickly can I be up and running?

All of our services are provisioned instantly. In some cases verification of payment method may be required. If that is the case, it could take up to 24 hours to validate your payment information, but payment verification is typically done within 1 hour during normal business hours.

How do I sign up for service?

Its easy to get started, simply visit click on Sign Up button on this page and you will be forwarded to VoIPVoIP.

How do I get support?

The easiest way to get support is by opening a support ticket logging to your account You can also email [email protected] voipvoip.com which will automatically open a support ticket assuming you have an account established with us. If you need to call us, you can reach us at 925-395-5300, although we recommend logging a support ticket or live chat for the fastest response times.

 

PBX Support

How to Verify That Your PBX is SIP-Enabled

There are several ways to check and see if your PBX is SIP-enabled. First, if your PBX has a data jack or Ethernet jack on the back, there is a good chance that it is SIP-capable. Older PBX or key systems just have analog lines to connect to the PSTN, so if your system does not have a data jack or Ethernet jack it is probably not SIP-capable.  If your PBX has a data jack and you are still unsure if it’s SIP-capable, you can check the user manual. You’ll want to look for a section on ‘configuring a SIP Trunk’ or you might find it in the specifications section, typically located at the end of the manual. Look for words like SIP or SIP-enabled IP calling.  

Can you connect to a legacy analog PBX or key system?

Yes. We recommend the Grandstream multi-port ATA’s to connect analog PBX system. They come in 4,8 and 24 port increments. These multi-port ATA’s are very easy to configure and have proven to be very reliable with our customer deployments. They can be purchased from Amazon.

Which PBX systems and devices do you support?

We support any SIP-enabled PBX . This includes open source VoIP IP-PBX’s like Asterisk, FreePBX®, Trixbox, Switchvox, PBX in a Flash, Elastix, Bluebox, FusionPBX, 3CXand more. Many traditional PBX manufacturers such as Toshiba, Panasonic, NEC, Avaya, Cisco, Nortel, Intertel  also support SIP trunking with their latest software releases.  In addition, we fully support analog telephone adapters (ATAs) which can interface to legacy analog PBX’s and key systems. We recommend the Cisco SPA-2102 for a single port ATA interface and the Grandstream GXW400X series multi-port ATAs for interfacing with analog systems that require more than one line.

Does your SIP trunking service work with popular Graphical User Interfaces to configure and control Asterisk?

Our SIP trunking service works perfectly with Asterisk, FreeSwitch and other open source telephony applications including popular Graphical User Interface applications used to configure and control Asterisk. We provide detailed configuration instructions for these systems.

What VoIP protocols and voice codecs do you support?

We currently only support the SIP protocol. We support both G.711 and G.729 voice codecs for calling.

 

Supported Features

Do you provide e911 service on your DIDs?

Yes. We provide ‘Nomadic’ e911 service on all of our US DIDs. Our ‘Nomadic’ e911 service allows you to set ANY physical address in the United States as your address to be transmitted on 911 calls. This means your calls to 911 will route to the closest PSAP (Public Safety Answering Point) to your registered e911 address on our system. That address will also appear on the emergency services operator’s screen when you call. The address can be updated at anytime online via our control panel.

Do you pro 411 directory information service?

Calls to 411 will route on our system, we send those calls to 1-800-FREE-411. It is a free, ad-sponsored directory assistance application.

Do you provide CNAM (CallerID Name) service on inbound numbers?

Yes. Inbound CNAM (Caller ID with Name) delivery is available on all incoming calls (assuming it was provided to us by our upstream carrier).

Do you provide CNAM (CallerId Name) service on outbound calls?

Yes. We offer this service on most all of our DIDs. If you need to show your PSTN phone number as your Caller ID please login to your account and open a ticket from Help Desk.

Can I use SIP Trunk service for faxing?

While it is possible to use the SIP Trunk service for faxing by placing an ATA (analog telephone adapter) behind a standard fax machine, keep in mind that calls over the public Internet are not designed to deliver faxes reliably 100% of the time. Traditional fax machines have fax modems which rely on data being delivered to them predictably without jitter or latency. With VoIP calling (and faxes riding on that connection), we can’t guarantee that fax transmissions will appease the finicky fax modems contained in most fax machines.

 

Reporting

How do I check my Call History?

You can check your Call History on our Control Panel in Incoming Calls section. Our CDRs are written in real-time to our database. You have the ability to pull 90 days worth of CDRs on the Control Panel. If you require CDRs that are over 90 days old, please open a support ticket.

 

Telephone Numbers

What is DID?

Direct inward dialing (DID) is a feature offered by telephone service providers for use with their customers’ PBX systems. Individual numbers are provided to specific subscribers. This makes it possible for a 10 digit phone number to reach a specific telephone within a company, rather than reaching a main line.

What calling area do the unlimited SIP trunks cover?

Our unlimited SIP trunks cover outbound SIP calls to the anywhere in the world. Inbound local DIDs from 50+ countries are also included in the unlimited SIP Trunks.

What is your US DID footprint?

We have one of the largest DID footprints of any SIP trunking provider with access to over 6,000 rate centers in the United States. Whereas most other SIP trunk providers will only have a handful of numbers in each rate center, you will often find hundreds of numbers in each area code across the country in our inventory

Do you offer Toll-Free DIDs?

Yes. We offer toll-free DIDs from US and 40+ countries. Toll Free pricing can be found on our main pricing page.

Do you offer vanity telephone numbers?

We do not currently offer vanity telephone numbers. Toll-Free vanity numbers, on the other hand, are easier to come by. We partner with www.tollfreenumbers.com the leading provider of toll-free vanity numbers in the US. Simply browse and purchase a toll-free vanity number through their website, and port it over to us. They will provide you with the necessary documentation for porting, making the porting process smooth.

Can I port my telephone number to VoIPVoIP?

Porting your US (PSTN, Wireless, Toll Free) or International number to our service is an effortless and easy task with VoIPVoIP SIP Trunk service, provided that your number is portable. Please check your number’s portability with our support team.

 

Configuration, Deployment and Use

How to place a call using SIP Trunk

When placing a SIP call with SIP Trunk you will want to make sure that your PBX or device is configured properly using Username / Password authentication or IP address authentication. An easy way to test a SIP Call is to use a softphone, such as Xlite or Zoiper, and configure a SIP trunk directly in the softphone. When making your SIP call from the softphone, you’ll want to be sure to dial the country code followed by the area code and then the number.

What kind of Internet connection do I need for your service?

Any broadband Internet connection will work with our service. Cable, DSL, T1-data as well as 3G and 4G mobile services are all supported. Unlike some other SIP providers, do not require you to get connectivity from us.

How much bandwidth do telephone calls consume on your service?

Telephone calls on our system are by default configured to operate on the G.711 voice codec, which consumes 85kbps of Internet bandwidth up and down. For example, a small DSL connection of 512kbps up and 3M down will have a limiting factor of the upstream 512kbps limit. Take 512 and divide it by 85 to arrive at a total of 6 maximum simultaneous calls on that particular type of Internet connection. That’s the very low end. Most broadband Internet connections these days are much faster, to the point where dozens and dozens of calls can traverse the data connection.

Where can I find configuration instructions for your SIP trunking service?

Our website contains examples of SIP trunk configurations for a variety of systems and devices along with other useful information to help you get started.

Do you assist in configuring SIP trunks on my PBX or endpoints?

Yes, if provided with remote access to your system, we will make best efforts to configure your equipment to use our service. Simply open a new support ticket and we can help get you going quickly.

What ports do I need to forward to my system if it is behind NAT?

We recommend forwarding ports UDP/5060 and UDP/10000-20000 for standard FreePBX/Asterisk based installs. It may be possible to get your service working without port forwarding, but optimal service will be obtained with these ports.  

What methods of SIP trunk authentication do you offer?

We offer the ability to do username/password or IP address authentication for our SIP trunks. .

Do I need a dedicated public IP address to use your service ?

No. You are not required to have a dedicated public IP address to use our service, although if you are able to obtain one it is certainly recommended.

What is QoS (Quality of Service) and how do I enable it?

QoS or Quality of Service allows for the prioritization of traffic on your network and is usually configured in your router settings. It is imperative that QoS be properly enabled to insure the successful transmission of VoIP phone calls. Without QoS you run the risk of audio-breakup and voice choppiness whenever your broadband Internet connection becomes saturated. For example, if you are downloading a large video file from a website and you do not have QoS enabled and a VoIP telephone call was initiated at the same time the voice packets would get lumped in along with the packets of the download. The audio would break up and in some cases become unintellligible until the download was complete. (To test QoS is properly enabled, you can initiate a large download on your computer while making a telephone call.) Fortunately there are inexpensive routers with robust easy-to-configure QoS. These include the Linksys WRT54GL with the Tomato firmware as well as the Cisco SRP521W.

 

Payment

What methods of payment do you accept?

We accept all major forms of credit card including MasterCard, Visa, Discover and American Express as well as PayPal.

How can I change my method of payment on file?

To add or change a method of payment, please visit the billing section of the Control Panel.

How do I cancel my service and can I cancel at anytime?

You can cancel your service online via the Control Panel at any time. There are no partial-month refunds for unlimited SIP trunks or DID numbers. If you have a remaining balance in your pre-paid account, it will be refunded to you after you cancel (may take 3-5 business days to post to your card statement).

International Calling

Do you offer International DIDs?

Yes. We offer International DIDs from 50+ countries around the world. By default they are 2-channel max unlimited inbound numbers for a fixed flat rate per month. If additional channels are required beyond the 2-channel maximum, please open a support ticket to get a quote on boosting the number of channels per International phone number.

Do you offer international service and what are the rates?

Yes. We offer international outbound calling service with competitive rates.

sip-trunk.org

Panasonic NCP PBX - SIP Trunk Configuration Guide | VoiceHost

How to configure SIP Trunking on a Panasonic NCP IP PBX

Tentative Version 0.1(PSN) 18th, July, 2013

SIP Trunk – Port Property:Important Note: Programming the details of the SIP trunk is done in this field.In this example the system has been programmed to use the changed FAX setting and NAT Keep Alive ability.

- Reject T.38 Request change to “Enable”. (Default: Disable)  *Note SIP server does not support T.38. (Need to set reject T.38 request by PBX.)

Recommended setting- NAT - Keep Alive Packet Sending Ability change to “Enable”. (Default: Disable)Go to 1.Configuration - 1.Slot and select “IPCMPR Virtual Slot”. and click “Ous”.Move mouse over “V-SIPGW16” and click “Port Property”.

Main Tab:1. Channel Attribute:                                 Basic Channel2. Provider Name:                                     Enter a logical name3. SIP Server Location – Name:                  st.sipconvergence.co.uk – Enter your assigned server.4. SIP Server Location – IP Address:           Not required5. SIP Server port Number:                        Leave at 50606. SIP Service Domain:                             Not required7. Subscriber Number:                              Not required

Account Tab:1. User name:Enter the SIP Account (User name) as supplied by VoiceHost. Please note that this is just the SIP Account (user name) and DOES NOT include @st.voicehost.co.uk For example: SIP Account (User name) = ST00000T000 Enter: ST00000T000

2. Authentication ID:                      Enter the Authentication ID as supplied by VoiceHost. Please note that this is just the Authentication ID and DOES NOT include @st.voicehost.co.ukFor example: Authentication ID = ST00000T000 Enter: ST00000T000

3. Authentication Password:Enter the Password as supplied by VoiceHost

Register Tab:1. Register Ability:                                           Leave at Enable2. Register Interval:                                        Leave at 36003. Un-Register Ability:                                     Leave enabled4. Registrar Server – Name:                             Not required  * If SIP Server and Registrar Server are different, enter the Registrar Server.5. Registrar Server – IP Address:                      Not required6. Registrar Server port number:                      Leave at 5060

Go Back to “Slot”.Move mouse over “V-SIPGW16” again, and click “Shelf Property”.NAT - Keep Alive Packet Sending Ability:                   Change to EnableNAT - Keep Alive Packet Type:                                 Confirm Blank UDPNAT – Keep Alive Packet Sending Interval:                    Confirm 20

Then, click“OK”. Move mouse over“V-SIPGW16” again, and click “Ins”.

Incoming Call Routing:Go to “10. CO & Incoming call” and select “3.DDI /DID Table”1. DDI/DID Number:                       Enter the DDI number in the format 44+PSTN Number (as below)

  • Example: PSTN number=0843-9999999
  • Enter: 448439999999 (Remove “0” of 0843-)

2. DDI/DID Name:                Determined by the installer (optional setting)3. DDI/DID Destination:     Determined by the installer (extension number, group etc)All other settings can be left at default

Outbound Call CLI:Each extension that wish to present individual CLI need to be programmed with a usable CLI. The usable CLI is a PSTN number assigned with the SIP trunk.

 i.e. if the PSTN number is 0843-9999999, the CLI to be programmed is 08439999999

Go to “Calling Party” tab.1. From Header – User Part:          Change to PBX-CLIP

All other tabs may be left at default:- Header Type- From Header – SIP-URI (100 characters)- P-Preferred-Identity Header – User Part- P-Preferred-Identity Header – SIP-URI (100 characters)- Number Format- Remove Digit- Additional Dial- Anonymous format in “From” header- P-Asserted-Identity header

Go to “4.Extension, 1.Wired Extension, 1.Extension Settings” & select “ISDN CLIP”

1.   Enter a valid CLI for each extension that requires it in the CLIP ID field. This setting, callee side shown ‘08439999999’.2.   Enter the name for each extension that requires it in the Extension Name field

This setting, what characters shown callee side is now testing.

[T.38 Tab]

1. Reject T.38 Request from Network:      Change to Enable

All other tabs may be left at default- T38 FAX Max Datagram- T38 FAX UDPTL Error Correction - Redundancy- T38 FAX UDPTL Redundancy count for T.30 messages- T38 FAX UDPTL Redundancy count for data

www.voicehost.co.uk


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