|
||||||
|
Asterisk can be integrated with Jabber xmpp in order to integrate gtalk, yahoo, etc. Here is the sample conf for asterisk: Please open jabber.conf (/etc/asterisk/jabber.conf) in your favourite editor and add following configuration: jabber.conf This is where you set your gmail/gtalk account info and will register you with the Google server. [general] debug=yes autoprune=no . . . → Read More: How to integrate Asterisk setup with gtalk, yahoo etc for calling? Registration string is used to secure the SIP Trunking. The trunks will be activated only if the authentication is completed. For configuring registration trunk we have to add this line (format) under trunk configuration as username:password@your.provider.com:5060 …….. Edit sip.conf (/etc/asterisk/sip.conf) in your favourite editor and add the following example configuration: [didforsale_1] type=peer host=209.216.2.211 nat=no canreinvite=no disallow=all allow=ulaw allow=g729 dtmfmode=rfc2833 insecure=very context=from-trunk [didforsale_2] type=peer host=209.216.15.70 nat=no canreinvite=no disallow=all allow=ulaw allow=g729 dtmfmode=rfc2833 insecure=very context=from-trunk After copying the above lines, save and reload Asterisk. Multi-tenancy is an architecture in which a single instance of a software application serves multiple customers. Each customer is called a tenant. For creating multitenant we need to create custom extensions in /etc/asterisk/extensions_custom.conf and give relevant context route calls: [company1] exten => 1234512345,1,Set(__FROM_DID=${EXTEN}) exten => 1234512345,n,Gosub(app-blacklist-check,s,1) exten => 1234512345,n,ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)})) . . . → Read More: How to create a multitenant users in asterisk-freepbx? This configures a ‘virtual’ extension that rings a group of phones simultaneously, stopping when any one of them is picked up. For this we need to create Ringgroup and add extensions. Then create context for incoming calls so that when someone calls the DID, it will go to the ring group. [ext-did-0002] include => ext-did-0002-custom . . . → Read More: How to use ring groups to follow the extension one by one in asterisk-freepbx? VoiceMail is used to leave a message if someone is not answering your call. The configuration in Asterisk is done in /etc/asterisk/voicemail.conf. In order to configure voicemail, add following line: 1000 => myemail@gmail.com,,attach=no|saycid=no|envelope=no|delete=no This will configure a voicemail system to email the received voice message (with attach file option) to myemail@gmail.com, when a call . . . → Read More: How to configure voicemail system in Asterisk? A SIP extension is configured in the SIP channel driver configuration file, called sip.conf. (This file resides in the Asterisk configuration directory, which is typically /etc/asterisk.) Open sip.conf with your favorite text editor, scroll to the bottom of the file, and add a section for your extension. You’ll need to choose your own unique . . . → Read More: How to create extensions in asterisk-freepbx? Did For Sale is happy to announce the launch of new website! The website has been redesigned to improve user friendliness and appeal. In addition to the changed design and layout of the pages, new functions have been implemented in this version. Design and Navigation The design of the web pages and the structure . . . → Read More: Website Launch Announcement: DidforSale launches new site DIDforSale is proud to announce that we have now expanded our DID coverage into Canada. In addition to US and UK, now you can purchase Candian DID’s as Metered, FlatRate or Unlimited per Channel basis. Our goal as always is to provide high quality DIDs at a very competitive price without compromising on . . . → Read More: DIDForSale Offers Canada DID Numbers
SIP is a Session Initiation Protocol at application layer which controls creating, modifying, and terminating sessions with multiple participants. A SIP Trunk can be referred to as a logical connection between an IP PBX and a Service Provider’s application servers that allows voice over IP traffic to be exchanged between the two. While . . . → Read More: What is SIP Trunking?  Comments are closed  All about VoIP, SIP TRUNKING, Voip Getting started  sip trunking, voip service provider |
||||||
|
|
||||||
Recent Comments