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This is the support site for Andrews & Arnold Ltd, a UK Internet provider. Information on these pages is generally for our customers but may be useful to others, enjoy!

VoIP Phones - Asterisk: Difference between revisions

DTMF PJSIP
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(DTMF PJSIP)
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Asterisk has two methods to configure SIP connections. The legacy "sip.conf" (SIP) and the more modern "pjsip.conf" (PJSIP).
 
Newer installations of Asterisk should be configured to use PJSIP as it will be more supported as Asterisk development continues., It'''however''' isnit's been reported that PJSIP doesn't asupport goodin-band ideaDTMF todetection haveproperly. anYou installationmay thatneed mixesto switch back to legacy sip.conf withif this affects you. The official recommendation on the [https://trac.pjsip.conforg/repos/wiki/FAQ#dtmf PJSIP FAQ] seems to be to write your own plugin if you need it. In-band DTMF support seems like an important thing to have, so we suggest raising a bug to report a missing feature in PJSIP if this affects you!
 
It isn't a good idea to have an installation that mixes sip.conf with pjsip.conf.
 
When reading the instructions below be aware which are for sip.conf and which are for pjsip.conf. PJSIP examples are below the SIP examples on this page.
In this example, extn is the extension that Asterisk will pass the call to.
Localnet should of course be set to whatever RFC1918 range you are using on your LAN.
 
=pjsip.conf (PJSIP)=
 
==Dialplan==
exten => _X.,1,Dial(SIP/snom)
</syntaxhighlight>
 
=pjsip.conf (PJSIP)=
 
==PJSIP: Trunk registration==
 
In extensions.conf you can dial out via the trunk with:
exten => _X.,1,Dial(PJSIP/0800500005${EXTEN}@aaisptrunk,,)
 
==PJSIP: Keep-Alive / Anti-Idle==
contact = sip:+442082881111@voiceless.aa.net.uk
qualify_frequency=20
 
==Status and Commands==
A good command within the asterisk software is the show registration command:
asterisk*CLI> pjsip show registrations
<Registration/ServerURI..............................> <Auth..........> <Status.......>
==========================================================================================
reg_442082881111/sip:voiceless.aa.net.uk auth_reg_442082881111 Registered
Objects found: 1
In this example it shows that the Asterisk server is successfully registered with the Andrews & Arnold SIP server.
 
=Further Help=
Customers using Asterisk and AAISP have created a website and IRC channel especially for this!
*http://www.aa-asterisk.org.uk/ [ Dead link @ Dec 2020 ]
*irc://zirc.jeaachat.net/a&a-asterisk
 
 
 
=Firewall & Security=
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