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<td id="header-avatar-image-container" valign="top" style="padding: 0px; border-collapse: collapse; vertical-align: top; width: 32px; padding-right: 9px"><a href="https://wiki.asterisk.org/wiki/display/~rnewton?src=email" style="color: #3b73af; text-decoration: none"><img id="header-avatar-image" class="image_fix" src="cid:avatar_925838d3d8b1f71935f30b314c925d64" height="32" width="32" border="0" style="border-radius: 3px; vertical-align: top" /></a></td>
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<table class="content-excerpt-pattern" cellspacing="0" cellpadding="0" border="0" width="100%" style="border-collapse: collapse; mso-table-lspace: 0pt; mso-table-rspace: 0pt; color: #333; font-family: Arial, sans-serif; font-size: 14px; line-height: 20px; mso-line-height-rule: exactly; mso-text-raise: 1px">
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<td class="content-excerpt-pattern-container mobile-resize-text " style="padding: 0px; border-collapse: collapse; padding: 0 0 0 24px"> <p class="diff-context-placeholder" style="margin: 10px 0 0 0; margin-top: 0">...</p> <h1 id="Configuringres_pjsip-Overview" class="diff-block-target diff-block-context" style="margin: 10px 0 0 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0"> <span class="diff-html-changed" id="changed-diff-0" style="background-color: #d6f0ff;">Overview</span> </h1> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-changed" style="background-color: #d6f0ff;">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.</span> </p> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-changed" style="background-color: #d6f0ff;">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 </span><a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none"><span class="diff-html-changed" style="background-color: #d6f0ff;">Creating SIP Accounts</span></a><span class="diff-html-changed" style="background-color: #d6f0ff;"> for some basic info or check out the sip.conf sample file included in the /configs directory of your Asterisk source files.</span> </p> <h1 id="Configuringres_pjsip-BeforeYouConfigure" class="diff-block-target diff-block-context" style="margin: 10px 0 0 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0"> <span class="diff-html-changed" style="background-color: #d6f0ff;">Before You Configure</span> </h1> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-changed" style="background-color: #d6f0ff;">This page assumes certain knowledge, or that you have completed a few prerequisites</span> </p>
<ul class="diff-block-target diff-block-context" style="margin: 10px 0 0 0">
<li> <span class="diff-html-changed" style="background-color: #d6f0ff;">You have </span><a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none"><span class="diff-html-changed" style="background-color: #d6f0ff;">installed pjproject</span></a><span class="diff-html-changed" style="background-color: #d6f0ff;">, a dependency for res_pjsip.</span> </li>
<li> <span class="diff-html-changed" style="background-color: #d6f0ff;">You have Installed Asterisk including the </span><a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none"><span class="diff-html-changed" style="background-color: #d6f0ff;">res_pjsip and chan_pjsip modules</span></a><span class="diff-html-changed" style="background-color: #d6f0ff;"> (implying you installed their dependencies as well)</span> </li>
<li> <span class="diff-html-changed" style="background-color: #d6f0ff;">You understand basic Asterisk concepts. Including the </span><a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none"><span class="diff-html-changed" style="background-color: #d6f0ff;">role of extensions.conf</span></a><span class="diff-html-changed" style="background-color: #d6f0ff;"> (dialplan) in your overall Asterisk configuration.</span> </li>
</ul> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-changed" style="background-color: #d6f0ff;">If you don't know anything about Asterisk yet, then you should probably start at the </span><a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none"><span class="diff-html-changed" style="background-color: #d6f0ff;">Getting Started section</span></a><span class="diff-html-changed" style="background-color: #d6f0ff;">.</span> </p> <h1 id="Configuringres_pjsip-QuickStart" class="diff-block-target diff-block-context" style="margin: 10px 0 0 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0"> <span class="diff-html-changed" style="background-color: #d6f0ff;">Quick Start</span> </h1> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-changed" style="background-color: #d6f0ff;">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.</span> </p>
<ul class="diff-block-target diff-block-context" style="margin: 10px 0 0 0">
<li> <span class="diff-html-changed" style="background-color: #d6f0ff;">Understand that res_pjsip is configured through pjsip.conf. This is where you'll be configuring everything related to your SIP trunks or phones.</span> </li>
<li> <span class="diff-html-changed" style="background-color: #d6f0ff;">Look at the </span><a href="https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip#Configuringres_pjsip-Fullres_pjsipconfigurationexamplesbyscenario" rel="nofollow" style="color: #3b73af; text-decoration: none"><span class="diff-html-changed" style="background-color: #d6f0ff;">Full res_pjsip configuration examples by scenario</span></a><span class="diff-html-changed" style="background-color: #d6f0ff;"> section. Grab the example most appropriate to your goal and use that to replace your pjsip.conf.</span> </li>
<li> <span class="diff-html-changed" style="background-color: #d6f0ff;">Reference documentation for all configuration parameters is available on the wiki:</span>
<ul style="margin: 10px 0 0 0">
<li> <a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none"><span class="diff-html-changed" style="background-color: #d6f0ff;">Core res_pjsip configuration options</span></a> </li>
<li> <a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none"><span class="diff-html-changed" style="background-color: #d6f0ff;">Configuration options for ACLs in res_pjsip_acl</span></a> </li>
<li> <a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none"><span class="diff-html-changed" style="background-color: #d6f0ff;">Configuration options for outbound registration, provided by res_pjsip_outbound_registration</span></a> </li>
<li> <a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none"><span class="diff-html-changed" style="background-color: #d6f0ff;">Configuration options for endpoint identification by IP address, provided by res_pjsip_endpoint_identifier_ip</span></a> </li>
</ul> </li>
<li> <span class="diff-html-changed" style="background-color: #d6f0ff;">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.</span> <br /> <span class="diff-html-changed" style="background-color: #d6f0ff;">Refer back to this page, the sample pjsip.conf, or the reference documentation if you get confused.</span> </li>
</ul>
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</table> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;"> </span> </p> <h1 id="Configuringres_pjsip-Dialingusingchan_pjsip" class="diff-block-target diff-block-context" style="margin: 10px 0 0 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">Dialing using chan_pjsip</span> </h1> <h2 id="Configuringres_pjsip-Dialingfromdialplaninextensions.conf" class="diff-block-target diff-block-context" style="margin: 10px 0 0 0; font-size: 20px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0; margin-top: 10px"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">Dialing from dialplan in extensions.conf</span> </h2> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">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 </span><a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none"><span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">Application_Dial</span></a> </p> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">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.</span> </p>
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<span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">exten => _6XXX,1,Dial(PJSIP/${EXTEN})</span>
</pre> </td>
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</table> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">Heres how you would dial with an explicit SIP URI, user and domain, via an endpoint (in this case dialing out a trunk), but not using its associated AOR/contact objects.</span> </p>
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<span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">exten => _9NXXNXXXXXX,1,Dial(PJSIP/mytrunk/sip:${EXTEN:1}@203.0.113.1:5060)</span>
</pre> </td>
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</table> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">This uses a contact(and its domain) set in the AOR associated with the </span><strong><span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">mytrunk</span></strong><span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;"> 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.</span> </p>
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<span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">exten => _9NXXNXXXXXX,1,Dial(PJSIP/${EXTEN:1}@mytrunk)</span>
</pre> </td>
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</table> <h1 id="Configuringres_pjsip-OldtoNew-sip.conftopjsip.confexamplecomparison" class="diff-block-target diff-block-context" style="margin: 10px 0 0 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">Old to New - sip.conf to pjsip.conf example comparison</span> </h1> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">We want to provide some examples of what similar configurations would look like between the old sip.conf and the new pjsip.conf</span> </p> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">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</span> </p> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">There is also a </span><a href="http://svnview.digium.com/svn/asterisk/trunk/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py?view=markup" class="external-link" rel="nofollow" style="color: #3b73af; text-decoration: none"><span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">script available</span></a><span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;"> to provide a basic conversion of a sip.conf config to a pjsip.conf config</span><strong><span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">.</span></strong> </p> <h2 id="Configuringres_pjsip-ExampleEndpointConfiguration" class="diff-block-target diff-block-context" style="margin: 10px 0 0 0; font-size: 20px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">Example Endpoint Configuration</span> </h2> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">This examples shows the configuration required for:</span> </p>
<ul class="diff-block-target diff-block-context" style="margin: 10px 0 0 0">
<li> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">two SIP phones need to make calls to or through Asterisk, we also want to be able to call them from Asterisk</span> </li>
<li> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">for them to be identified as users (in the old chan_sip) or endpoints (in the new res_sip/chan_pjsip)</span> </li>
<li> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">both devices need to use username and password authentication</span> </li>
<li> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">6001 is setup to allow registration to Asterisk, and 6002 is setup with a static host/contact</span> </li>
</ul>
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<span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">[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</span>
</pre> </td>
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</table> <p style="margin: 10px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;"> </span> </p> <p style="margin: 10px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;"> </span> </p> </td>
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<span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">[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</span>
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</div> <h2 id="Configuringres_pjsip-ExampleSIPTrunkConfiguration" class="diff-block-target diff-block-context" style="margin: 10px 0 0 0; font-size: 20px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">Example SIP Trunk Configuration</span> </h2> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">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.</span> </p>
<ul class="diff-block-target diff-block-context" style="margin: 10px 0 0 0">
<li> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">SIP provider requires registration to their server with a username of "myaccountname" and a password of "1234567890"</span> </li>
<li> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">SIP provider requires registration to their server at the address of 203.0.113.1:5060</span> </li>
<li> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">SIP provider requires outbound calls to their server at the same address of registration, plus using same authentication details.</span> </li>
<li> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">SIP provider will call your server with a user name of "mytrunk". Their traffic will only be coming from 203.0.113.1</span> </li>
</ul>
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<span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">[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</span>
</pre> </td>
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<span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">[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</span>
</pre> </td>
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</div> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;"> </span> </p> <h1 id="Configuringres_pjsip-Disablingres_pjsipandchan_pjsip" class="diff-block-target diff-block-context" style="margin: 10px 0 0 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">Disabling res_pjsip and chan_pjsip</span> </h1> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">There are several methods to disable or remove modules in Asterisk. Which method is best depends on your intent.</span> </p> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">If you have built Asterisk with the PJSIP modules, but don't intend to use them at this moment, you might consider the following:</span> </p>
<ol class="diff-block-target diff-block-context" style="margin: 10px 0 0 0">
<li> <p style="margin: 10px 0 0 0; margin-top: 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">Edit the file </span><strong><span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">modules.conf</span></strong><span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;"> in your Asterisk configuration directory. (typically /etc/asterisk/)</span> </p>
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<td class="diff-macro-body" style="background-color: #fff;border: 1px solid #dddddd;padding: 10px;; padding: 0px; border-collapse: collapse"> <pre style="margin: 10px 0 0 0; margin-top: 0">
<span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">noload => res_pjsip.so
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</span>
</pre> </td>
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</table> <p style="margin: 10px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">Having a noload for the above modules should (at the moment of writing this) prevent any PJSIP related modules from loading.</span> </p> </li>
<li> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">Restart Asterisk!</span> </li>
</ol> <p class="diff-block-target diff-block-context" style="margin: 10px 0 0 0"> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">Other possibilities would be:</span> </p>
<ul class="diff-block-target diff-block-context" style="margin: 10px 0 0 0">
<li> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">Remove all PJSIP modules from the modules directory (often, /usr/lib/asterisk/modules)</span> </li>
<li> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">Remove the configuration file (pjsip.conf)</span> </li>
<li> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">Un-install and re-install Asterisk with no PJSIP related modules.</span> </li>
<li> <span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">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</span> </li>
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