- Overview
- Before You Configure
- Quick Start
- pjsip.conf Explained
- Full res_pjsip configuration examples by scenario
- Dialing using chan_pjsip
- Old to New - sip.conf to pjsip.conf example comparison
- Disabling res_pjsip and chan_pjsip
Overview
This page is intended to help an Asterisk administrator understand configuration fundamentals for the new SIP resources and channel driver included with Asterisk 12. It covers an explanation of configuration for pjsip.conf which configures the SIP resource modules utilized by the chan_pjsip driver.
If you are looking for info on how to configure sip.conf (the config used by the older SIP channel driver for Asterisk), then you'll want to go here Creating SIP Accounts for some basic info or check out the sip.conf sample file included in the /configs directory of your Asterisk source files.
Before You Configure
This page assumes certain knowledge, or that you have completed a few prerequisites
- You have installed pjproject, a dependency for res_pjsip.
- You have Installed Asterisk including the res_pjsip and chan_pjsip modules (implying you installed their dependencies as well)
- You understand basic Asterisk concepts. Including the role of extensions.conf (dialplan) in your overall Asterisk configuration.
If you don't know anything about Asterisk yet, then you should probably start at the Getting Started section.
Quick Start
If you like to play before you read or figure out things as you go; here's a few quick steps to get you started.
- Understand that res_pjsip is configured through pjsip.conf. This is where you'll be configuring everything related to your SIP trunks or phones.
- Look at the Full res_pjsip configuration examples by scenario section. Grab the example most appropriate to your goal and use that to replace your pjsip.conf.
- Reference documentation for all configuration parameters is available on the wiki:
- You'll need to tweak details in pjsip.conf and on your SIP device (for example IP addresses and authentication credentials) to get it working with Asterisk.
Refer back to this page, the sample pjsip.conf, or the reference documentation if you get confused.
pjsip.conf Explained
Configuration Section Format
pjsip.conf is a flat text file composed of sections like most configuration files used with Asterisk. Each section defines configuration for a configuration object within res_pjsip or an associated module.
Sections are identified by names in square brackets. (see SectionName below)
Each section has one or more configuration options that can be assigned a value by using an equal sign followed by a value. (see ConfigOption and Value below)These options and values are the configuration for a particular component of functionality provided by the configuration object's respective Asterisk modules.
Every section will have a type option that defines what kind of section is being configured. You'll see that in every example config section below.
[ SectionName ]
ConfigOption = Value
ConfigOption = Value
Section Types
Below is a brief description of each section type and an example showing configuration of that section only. The module providing the configuration object related to the section is listed in parentheses next to each section name.
There are dozens of config options for some of the sections, but the examples below are very minimal for the sake of simplicity.
ENDPOINT (res_pjsip)
Endpoint configuration provides numerous options relating to core SIP functionality and ties to other sections such as auth, aor and transport. You can't contact an endpoint without associating one or more AoR sections. An endpoint is essentially a profile for the configuration of a SIP endpoint such as a phone or remote server.
TRANSPORT (res_pjsip)
Configure how res_pjsip will operate at the transport layer. For example, it supports configuration options for protocols such as TCP, UDP or WebSockets and encryption methods like TLS/SSL. You can setup multiple transport sections and other sections (such as endpoints) could each use the same transport, or a unique one.
AUTH (res_pjsip)
Authentication sections hold the options and credentials related to inbound or outbound authentication. You'll associate other sections such as endpoints or registrations to this one. Multiple endpoints or registrations can use a single auth config if needed.
AOR (res_pjsip)
A primary feature of AOR objects (Address of Record) is to tell Asterisk where an endpoint can be contacted. Without an associated AOR section, an endpoint cannot be contacted. AOR objects also store associations to mailboxes for MWI requests and other data that might relate to the whole group of contacts such as expiration and qualify settings.
When Asterisk receives an inbound registration, it'll look to match against available AORs.
Registrations: The name of the AOR section must match the user portion of the SIP URI in the "To:" header of the inbound SIP registration. That will usually be the "user name" set in your hard or soft phones configuration.
REGISTRATION (res_pjsip_outbound_registration)
The registration section contains information about an outbound registration. You'll use this when setting up a registration to another system whether it's local or a trunk from your ITSP.
DOMAIN_ALIAS (res_pjsip)
Allows you to specify an alias for a domain. If the domain on a session is not found to match an AoR then this object is used to see if we have an alias for the AoR to which the endpoint is binding. This sections name as defined in configuration should be the domain alias and a config option (domain=) is provided to specify the domain to be aliased.
ACL (res_pjsip_acl)
The ACL module used by 'res_pjsip'. This module is independent of 'endpoints' and operates on all inbound SIP communication using res_pjsip. Features such as an Access Control List, as defined in the configuration section itself, or as defined in acl.conf. ACL's can be defined specifically for source IP addresses, or IP addresses within the contact header of SIP traffic.
IDENTIFY (res_pjsip_endpoint_identifier_ip)
Controls how the res_pjsip_endpoint_identifier_ip module determines what endpoint an incoming packet is from. If you don't have an identify section defined, or else you have res_pjsip_endpoint_identifier_ip loading after res_pjsip_endpoint_identifier_user, then res_pjsip_endpoint_identifier_user will identify inbound traffic by pulling the user from the "From:" SIP header in the packet. Basically the module load order, and your configuration will both determine whether you identify by IP or by user.
CONTACT (res_pjsip)
The contact config object effectively acts as an alias for a SIP URIs and holds information about an inbound registrations. Contact objects can be associated with an individual SIP User Agent and contain a few config options related to the connection. Contacts are created automatically upon registration to an AOR, or can be created manually by using the "contact=" config option in an AOR section. Manually configuring a CONTACT config object itself is outside the scope of this "getting started" style document.
Config Section Help and Defaults
Once we have the XML configuration help pulled onto the wiki we'll put a link here to that wiki section.
In the meantime use the built-in configuration help to your advantage. You can use "config show help res_pjsip <configobject> <configoption>" to get help on a particular option. That help will typically describe the default value for an option as well.
Relationships of Configuration Objects in pjsip.conf
Now that you understand the various configuration sections related to each config object, lets look at how they interrelate.
You'll see that the new SIP implementation within Asterisk is extremely flexible due to its modular design. A diagram will help you to visualize the relationships between the various configuration objects. The following entity relationship diagram covers only the configuration relationships between the objects. For example if an endpoint object requires authorization for registration of a SIP device, then you may associate a single auth object with the endpoint object. Though many endpoints could use the same or different auth objects.
Configuration Flow: This lets you know which direction the objects are associated to other objects. e.g. The identify config section has an option "endpoint=" which allows you to associate it with an endpoint object.
Entity Relationships | Relationship Descriptions | ||
---|---|---|---|
| ENDPOINT
REGISTRATION
AOR
CONTACT
IDENTIFY
ACL, DOMAIN_ALIAS
|
Full res_pjsip configuration examples by scenario
Below are some sample configurations to demonstrate various scenarios with complete pjsip.conf files.
An endpoint with a single SIP phone with inbound registration to Asterisk
A SIP trunk to your service provider, including outbound registration
Multiple endpoints with phones registering to Asterisk, using templates
Dialing using chan_pjsip
Dialing from dialplan in extensions.conf
We are assuming you already know a little bit about the Dial application here. To see the full help for it, see "core show help application dial" on the Asterisk CLI, or see Application_Dial
Below we'll simply dial an endpoint using the chan_pjsip channel driver. This is really going to look at the AOR of the same name as the endpoint and start dialing the contacts associated.
exten => _6XXX,1,Dial(PJSIP/${EXTEN})
Heres how you would dial with an explicit SIP URI, user and domain, via an endpoint (in this case dialing out a trunk), but not using its associated AOR/contact objects.
exten => _9NXXNXXXXXX,1,Dial(PJSIP/mytrunk/sip:${EXTEN:1}@203.0.113.1:5060)
This uses a contact(and its domain) set in the AOR associated with the mytrunk endpoint, but still explicitly sets the user portion of the URI in the dial string. For the AOR's contact, you would define it in the AOR config without the user name.
exten => _9NXXNXXXXXX,1,Dial(PJSIP/${EXTEN:1}@mytrunk)
Old to New - sip.conf to pjsip.conf example comparison
We want to provide some examples of what similar configurations would look like between the old sip.conf and the new pjsip.conf
These examples contain only the configuration required for sip.conf/pjsip.conf as the configuration for other files should be the same excepting the Dial statements in your extensions.conf
There is also a script available to provide a basic conversion of a sip.conf config to a pjsip.conf config. ADD A LINK TO SIP.CONF to PJSIP.CONF SCRIPT WHEN READY
Example Endpoint Configuration
This examples shows the configuration required for:
- two SIP phones need to make calls to or through Asterisk, we also want to be able to call them from Asterisk
- for them to be identified as users (in the old chan_sip) or endpoints (in the new res_sip/chan_pjsip)
- both devices need to use username and password authentication
- 6001 is setup to allow registration to Asterisk, and 6002 is setup with a static host/contact
sip.conf | pjsip.conf |
---|---|
[general] udpbindaddr=0.0.0.0 [6001] type=friend host=dynamic disallow=all allow=ulaw context=internal secret=1234 [6002] type=friend host=192.0.2.1 disallow=all allow=ulaw context=internal secret=1234
| [simpletrans] type=transport protocol=udp bind=0.0.0.0 [6001] type = endpoint transport = simpletrans context = internal disallow = all allow = ulaw aors = 6001 auth = auth6001 [6001] type = aor max_contacts = 1 [auth6001] type=auth auth_type=userpass password=1234 username=6001 [6002] type = endpoint transport = simpletrans context = internal disallow = all allow = ulaw aors = 6002 auth = auth6002 [6002] type = aor contact = sip:6002@192.0.2.1:5060 [auth6002] type=auth auth_type=userpass password=1234 username=6001 |
Example SIP Trunk Configuration
This shows configuration for a SIP trunk as would typically be provided by an ITSP. That is registration to a remote server, authentication to it and a peer/endpoint setup to allow inbound calls from the provider.
- SIP provider requires registration to their server with a username of "myaccountname" and a password of "1234567890"
- SIP provider requires registration to their server at the address of 203.0.113.1:5060
- SIP provider requires outbound calls to their server at the same address of registration, plus using same authentication details.
- SIP provider will call your server with a user name of "mytrunk". Their traffic will only be coming from 203.0.113.1
sip.conf | pjsip.conf |
---|---|
[general] udpbindaddr=0.0.0.0 register => myaccountname:1234567890@203.0.113.1:5060 [mytrunk] type=friend secret=1234567890 username=myaccountname host=203.0.113.1 disallow=all allow=ulaw context=from-external | [simpletrans] type=transport protocol=udp bind=0.0.0.0 [mytrunk] type=registration transport=simpletrans outbound_auth=mytrunk server_uri=sip:myaccountname@203.0.113.1:5060 client_uri=sip:myaccountname@192.0.2.1:5060 [mytrunk] type=auth auth_type=userpass password=1234567890 username=myaccountname [mytrunk] type=aor contact=sip:203.0.113.1:5060 [mytrunk] type=endpoint transport=simpletrans context=from-external disallow=all allow=ulaw outbound_auth=mytrunk aors=mytrunk [mytrunk] type=identify endpoint=mytrunk match=203.0.113.1 |
Disabling res_pjsip and chan_pjsip
There are several methods to disable or remove modules in Asterisk. Which method is best depends on your intent.
If you have built Asterisk with the PJSIP modules, but don't intend to use them at this moment, you might consider the following:
Edit the file modules.conf in your Asterisk configuration directory. (typically /etc/asterisk/)
noload => res_pjsip_pubsub.so noload => res_pjsip_session.so noload => chan_pjsip.so noload => res_pjsip_exten_state.so noload => res_pjsip_log_forwarder.so
Having a noload for the above modules should (at the moment of writing this) prevent any PJSIP related modules from loading.
- Restart Asterisk!
Other possibilities would be:
- Remove all PJSIP modules from the modules directory (often, /usr/lib/asterisk/modules)
- Remove the configuration file (pjsip.conf)
- Un-install and re-install Asterisk with no PJSIP related modules.
- If you are wanting to use chan_pjsip alongside chan_sip, you could change the port or bind interface of your chan_pjsip transport in pjsip.conf
Attachments:











