Difference between revisions of "VoIP Phones - Asterisk"

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<indicator name="VoIPConfiguring">[[File:menu-voip.svg|link=:Category:VoIP Phones|30px|Back up to the VoIP Configuring page]]</indicator>
*[[IPv6]] Works!
Asterisk is extremely flexible and covering different uses for it is outside the scope of this example as the setup used here was very basic. You should read through the included documentation, especially the security documentation, before configuring Asterisk for the first time.
= Configuration =
==Define proxies for inbound calls==
Asterisk has two methods to configure SIP connections. The legacy "sip.conf" (SIP) and the more modern "pjsip.conf" (PJSIP).
Some background first:
Newer installations of Asterisk should be configured to use PJSIP as it will be more supported as Asterisk development continues, '''however''' it's been reported that PJSIP doesn't support in-band DTMF detection properly. You may need to switch back to legacy sip.conf if this affects you. The official recommendation on the [https://trac.pjsip.org/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!
Asterisk performs a '''forward''' DNS lookup on incoming calls. This means that even where the Asterisk box has registered, if the call comes in from the IP it registered to on a dual stack box but the lookup returns the other IP, Asterisk gets confused about where to route the call. It does not seem to cache the IP address it finds for the duration of the TTL either, so you can register one minute, then have a call from the same IP fail the next.
It isn't a good idea to have an installation that mixes sip.conf with pjsip.conf.
Unfortunately the only way around this seems to be to define a peer for each potential IP address that a call can come from. We only have two live "voiceless" servers at present (and a third test box) but it means that as we expand the service in future Asterisk users would have to keep updating their SIP configs. We therefore created a list of 10 servers (C and onwards are just A records pointing to the A server for now). Sadly this means that we needed to created 20 peers as they will all be dual stack!
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.
If your Asterisk box is not dual stack you only need to include the IPv4 '''or''' [[IPv6]] hostnames; whichever you're using.
=sip.conf (SIP)=
If you are not registering you will need to set up the username, password and host name for the trunk in the "SIP to your server" settings on the control pages. We do not recommend registering unless you're on a dynamic IP or behind NAT.
Here is what the config looks like for a dual stack Asterisk box on static public IPs:
== Incoming Calls ==
=== User Section ===
*Accept authenticated calls and route them to a context.
<syntaxhighlight lang="ini">
; Config for inbound calls
; The following 2 values are set in the "SIP to your server" trunk settings on the control pages.
; They should be commented out, and trunk settings removed from the control pages, if using Asterisk to register to the far end.
fromuser=voiceless-in ; Their user for authenticating with us.
secret=incomingpass ; Their password for authenticating with us
*We send Remote-Party-Id with the privacy and screen settings, setting '''trustrpid=yes''' in the incoming SIP config will allow Asterisk to pass withheld/unknown on.
=== Authentication ===
; IPv4 hostnames
*Voiceless must authenticate so that calls are recognised as the above peer section.
*You need to use the '''match_auth_username=yes''' setting otherwise Asterisk will not recognise Voiceless' initial requests.
; [[IPv6]] hostnames
<syntaxhighlight lang="ini">
*We initially send an Authorization header with only a username, allowing Asterisk to identify Voiceless by username instead of by IP. By default Asterisk ignores the username when identifying peers.
==Withheld/unknown caller ID - trustrpid==
We send Remote-Party-Id with the privacy and screen settings, setting trustrpid=yes in the incoming SIP config will allow asterisk to pass withheld/unknown on.
Maybe the wiki page http://wiki.aa.org.uk/VoIP_Phones_-_Asterisk could
be updated to add this setting.
== Outgoing Calls ==
==Define a proxy for outbound calls==
*Either use a separate '''type=peer''' section or combine incoming and outgoing in one '''type=friend''' section
Defining us as a SIP proxy for outbound calls:
=== Separate Section ===
<syntaxhighlight lang="ini">
; Config for outbound calls.
remotesecret=outgoingpass ; Our password to their service *some older asterisk versions require secret, not remotesecret, here*
defaultuser=+441234567890 ; Authentication user for outbound *some older asterisk versions require username, not defaultuser, here*
=== Combined Section ===
<syntaxhighlight lang="ini">
; incoming
; outgoing
==Note: Order of sip.conf is important==
==Note: Asterisk and IPv6 SLAAC addresses==
Asterisk will bind to all [[IPv6]] addresses if it is set to use [[IPv6]]. This means that if you have a static IP and a SLAAC IP, Asterisk sometimes replies to invites sent to the static IP from the SLAAC IP instead which breaks things. We recommend using static IP addresses and disabling SLAAC (and privacy extensions) on the box running Asterisk until its [[IPv6]] support is more mature.
If you're behind NAT it is helpful to make Asterisk register. It re-registers every 120 seconds by default anyway so should keep NAT sessions open.
You can register (and tell Asterisk that it's behind NAT) with these settings under the [general] section:
<syntaxhighlight lang="ini">
register => +441234567980:outgoingpass@voiceless.aa.net.uk/extn
In this example, extn is the extension that Asterisk will pass the call to. Asterisk matches the hostname against the peers later on in the config and so the only change you have to make is to remove the fromuser and secret from the [common] template in the inbound proxy example above.
Localnet should of course be set to whatever RFC1918 range you are using on your LAN.
See Asterisk's included example sip.conf for examples of how to send the call to different contexts etc.
For outbound calls:
<syntaxhighlight lang="ini">
exten => _[+X]_X.,1,Dial(SIP/voiceless-out/${EXTEN})
For inbound calls (assuming you're routing call to a registered Snom):
<syntaxhighlight lang="ini">
exten => _[+X]_X.,1,Dial(SIP/snom)
=pjsip.conf (PJSIP)=
==PJSIP: Trunk registration==
==Test server==
Here is an example of a working pjsip.conf setup where Asterisk will register with A&A to receive calls.
We have a test server for debugging problems and testing new features.
If you wish to allow our test server to make calls in to you as well, you can include its IPs by defining these peers too:
; Test server IPv4 hostname
In pjsip.conf:
; Test server [[IPv6]] hostname
type = registration
retry_interval = 20
fatal_retry_interval = 20
forbidden_retry_interval = 20
max_retries = 9999
auth_rejection_permanent = no
contact_user = maininbound
expiration = 120
outbound_auth = auth_reg_442082881111
client_uri = sip:+442082881111@voiceless.aa.net.uk
server_uri = sip:voiceless.aa.net.uk
type = auth
password = BusinessPaidGrewCome
username = +442082881111
type = aor
contact = sip:+442082881111@voiceless.aa.net.uk
type = identify
endpoint = aaisptrunk
match = voiceless.aa.net.uk
type = endpoint
context = maininbound
dtmf_mode = rfc4733
disallow = all
allow = alaw
allow = ulaw
direct_media = no
aors = aaisptrunk
Calls come into the context "maininbound" in extensions.conf - in this example calls get sent onto extension 222 and 205 for 20 seconds and then go to voicemail.
exten = maininbound,1,Dial(PJSIP/222&PJSIP/205,20)
exten = maininbound,n,Voicemail(222@default,us)
In extensions.conf you can dial out via the trunk with:
=Further Help=
exten => _X.,1,Dial(PJSIP/${EXTEN}@aaisptrunk,,)
Customers using Asterisk and AAISP have created a website and IRC channel especially for this!
==PJSIP: Keep-Alive / Anti-Idle==
If you are using a firewall or NAT router with short timeouts on UDP sessions you can force packets to be sent over the connection to keep it alive.
Set qualify_frequency in the aor section; This triggers an OPTIONS message every X (as set) seconds.
An example of the aor section follows:
type = aor
contact = sip:+442082881111@voiceless.aa.net.uk
==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 ]
=Firewall & Security=
*You will also want to set up firewall rules, as per the [[VoIP Firewall]] page.
*Also see the [[VoIP Security]] page for information about securing your VoIP service.
[[Category:VoIP Phones|Asterisk]]
[[Category:VoIP IPv6]]

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