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Asterisk DTMF issues

Its a common issue with asterisk as it sometimes wont pass dtmf properly. To solve the issue first we need to check the network have sufficient bandwidth, if bandwidth is sufficient then we need to add below parameters on all didforsale trunks Note that if you are using G729 codec then Inband wont work. . . . → Read More: Asterisk DTMF issues

Choppy lines or call breaks on asterisk

Choppy lines are common problem when we use sip signaling. The main reasons for the issues are Bandwidth, Codec, Lots of SIP trunks registered, Jitter.

Basic checklist for Choppy Lines

Initial step we have to take is to check the bandwidth usage of network Check the Codec ULAW, ALAW or G729 is allowed on . . . → Read More: Choppy lines or call breaks on asterisk

Asterisk one way audio issue

How to resolve one way or no audio issues

Its a common issue with PBX to have audio issues like one way audio or no audio. Sometime only caller can hear remote party or remote party only can hear the caller. This is mainly because of NAT issues. We recommend to use NAT with . . . → Read More: Asterisk one way audio issue

Local Numbers outside your Territory

Do you Advertise your business online, in print media or any other marketing channels? Do you give your business number on the Advertisements?

How good are the chances that your potential leads in Florida will call you if your number is from California as compared to your competitor whose has a local access number . . . → Read More: Local Numbers outside your Territory

Why DIDForSale is different

We want to take an opportunity to tell you why we believe we are better than our competitors.

We provide phone number in 11,000+ Rate Centers We have the largest coverage in US, Canada and UK. Even more than Verizon, Level 3 and most of other cLEC Companies) For 95% of the rate centers . . . → Read More: Why DIDForSale is different

How to Cut Company Costs by Using a SIP Trunk

A SIP trunk to the rescue? Yes, SIP trunks save companies thousands of dollars each year through Internet-based telephony technology, and are especially useful for companies that have many extensions but a low volume of simultaneous calls.

A SIP trunk is the external component of an in-house IP PBX. The PBX connects calls within . . . → Read More: How to Cut Company Costs by Using a SIP Trunk

How to setup asterisk and integrate with gtalk, yahoo etc for calling.

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?

What is the registration string and how can it secure the trunk?

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

……..

How to setup SIP trunks in Asterisk?

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.

How to create a multitenant users in asterisk-freepbx?

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?