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<!DOCTYPE html>
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<title>SIP Peers</title>
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<article>
<h1>SIP Peers</h1>
Example Cisco SIP peer configuration in <code class="literal">sip.conf</code>. New settings added by the patch are listed below.<br>
<br>
<h2 id="cisco_usecallmanager">cisco_usecallmanager <a href="#cisco_usecallmanager" title="Link">link</a></h2>
Enable support for Cisco SIP phone features, required for <code class="literal">USECALLAMANGER</code> phones. Do not enable on peers using phones from other vendors.<br>
<br>
<table>
<tbody>
<tr>
<td><b>no</b></td>
<td>Disabled</td>
<td class="vertical-rule"><b>yes</b></td>
<td>Enabled</td>
</tr>
</tbody>
</table>
<br>
<h2 id="dndbusy">dndbusy <a href="#dndbusy" title="Link">link</a></h2>
Have Asterisk automatically generate a busy response when calling a phone that has a presence state of DND, otherwise the call will be sent to the phone.<br>
<br>
<table>
<tbody>
<tr>
<td><b>no</b></td>
<td>Disabled</td>
<td class="vertical-rule"><b>yes</b></td>
<td>Enabled</td>
</tr>
</tbody>
</table>
<br>
<h2 id="subscribe">subscribe <a href="#subscribe" title="Link">link</a></h2>
Add a subscription for a BLF Speed Dial or BLF Directed Call Park line key, required for Cisco SIP phones as they do not send SUBSCRIBE requests. This option only applies to the phone's primary line. Multiple extensions can be specified separated by a comma.<br>
<br>
<table>
<tbody>
<tr>
<td><b>exten</b></td>
<td>Watch <code class="literal">exten</code> in <code class="literal">subscribecontext</code> if defined, otherwise <code class="literal">context</code></td>
</tr>
<tr>
<td><b>exten@context</b></td>
<td>Watch <code class="literal">exten</code> in specified <code class="literal">context</code></td>
</tr>
</tbody>
</table>
<br>
<h2 id="register">register <a href="#register" title="Link">link</a></h2>
Bulk-register specified peer automatically when this peer registers, required for Cisco SIP phones as they only send a REGISTER request for their primary line. The order in which the <code class="literal">register</code> entries are defined must match the <code class="literal">lineIndex</code> attribute defined in <a href="sepmac-cnf-xml.html#line">SEPMAC.cnf.xml</a>. Peers registered in this way do not support subscriptions, separate <code class="literal">qualify</code> values and should have the same <code class="literal">dndbusy</code> setting as the primary line. Multiple peers can be specified separated by a comma.<br>
<br>
<table>
<tbody>
<tr>
<td><b>name</b></td>
<td>Name of the peer.</td>
</tr>
</tbody>
</table>
<br>
<h2 id="huntgroup_default">huntgroup_default <a href="#huntgroup_default" title="Link">link</a></h2>
Default hunt group login state of a peer. This option only applies to the phone's primary line.<br>
<br>
<table>
<tbody>
<tr>
<td><b>no</b></td>
<td>Logged Out</td>
<td class="vertical-rule"><b>yes</b></td>
<td>Logged In</td>
</tr>
</tbody>
</table>
<br>
<h2 id="cisco_pickup_alert">cisco_pickupnotify_alert <a href="#cisco_pickup_alert" title="Link">link</a></h2>
When another phone in your pickup group is ringing alert the user with one or more of the following methods. Multiple alert options can be combined using a comma, eg: <code class="literal">from,to,beep</code>. A notification will only be sent if the phone is idle and has not activated DND.<br>
<br>
<table>
<tbody>
<tr>
<td><b>from</b></td>
<td>Display the caller ID number of the calling phone in the status line.</td>
</tr>
<tr>
<td><b>to</b></td>
<td>Display the extension of the phone being called in the status line.</td>
</tr>
<tr>
<td><b>beep</b></td>
<td>Play a beep tone through the speaker.</td>
</tr>
<tr>
<td><b>none</b></td>
<td>Do not send any notification.</td>
</tr>
</tbody>
</table>
<br>
<h2 id="cisco_pickup_timer">cisco_pickupnotify_timer <a href="#cisco_pickup_timer" title="Link">link</a></h2>
Display timeout in seconds for the pickup notify alert when either <code class="literal">from</code> or <code class="literal">to</code> are used.<br>
<br>
<h2 id="cisco_multiadmin_conference">cisco_multiadmin_conference <a href="#cisco_multiadmin_conference" title="Link">link</a></h2>
When joining another participant to an ad-hoc conference who also has <code class="literal">cisco_multiadmin_conference</code> set to <code class="literal">yes</code> then make that participant a conference administrator as well. That participant can now use the Conference List feature and mute or kick participants as well as adding other participants to the conference.<br>
<br>
<table>
<tbody>
<tr>
<td><b>no</b></td>
<td>Disabled</td>
<td class="vertical-rule"><b>yes</b></td>
<td>Enabled</td>
</tr>
</tbody>
</table>
<br>
<h2 id="cisco_keep_conference">cisco_keep_conference <a href="#cisco_keep_conference" title="Link">link</a></h2>
When there are no more administrators in an ad-hoc conference (see above for how to have more than one administrator), hang up the all remaining participants.<br>
<br>
<table>
<tbody>
<tr>
<td><b>no</b></td>
<td>Disabled</td>
<td class="vertical-rule"><b>yes</b></td>
<td>Enabled</td>
</tr>
</tbody>
</table>
<br>
<h2 id="cisco_qrt_url">cisco_qrt_url <a href="#cisco_qrt_url" title="Link">link</a></h2>
Tell the phone to access this URL when QRT is enabled to gather further information from the user regarding the call.<br>
<br>
<table>
<tbody>
<tr>
<td><b>url</b></td>
<td>URL to access. The phone's device-name is automatically included in the <code class="literal">name</code> parameter.</td>
</tr>
</tbody>
</table>
<br>
<div class="horizontal-rule"></div>
<h2 id="Extension-Template">Extension Template <a href="#Extension-Template" title="Link">link</a></h2>
Settings common to all SIP peers that are local extensions.<br>
<br>
<code class="prettify lang-astsip">[extension](!)
type=friend
context=extensions
host=dynamic
trustrpid=no
parkinglot=default
allowsubscribe=yes
notifyhold=no
callcounter=yes
videosupport=no
disallow=all
allow=g722,ulaw,alaw,g729
...</code>
<br>
<h2 id="Cisco-SIP-Phone-Template">Cisco SIP Phone Template <a href="#Cisco-SIP-Phone-Template" title="Link">link</a></h2>
Default settings that apply to all Cisco SIP phones, inherits the settings from the <code class="literal">extension</code> template.<br>
<br>
<code class="prettify lang-astsip">[cisco-usecallmanager](!,extension)
transport=tcp
nat=no
directmedia=no
sendrpid=rpid
rpid_update=yes
rpid_immediate=yes
send_diversion=yes
dndbusy=yes
cisco_usecallmanager=yes
cisco_pickupnotify_alert=from,to
cisco_pickupnotify_timer=5
cisco_keep_conference=no
cisco_multiadmin_conference=yes
huntgroup_default=no
...</code>
<br>
<h2 id="Direct-Media">Direct Media <a href="#Direct-Media" title="Link">link</a></h2>
To have RTP (media) flow directly between phones replace the <code class="literal">directmedia</code> setting in the above template with <code class="literal">update</code>.<br>
<br>
<code class="prettify lang-astsip">directmedia=update</code>
<br>
<h2 id="Transport-Layer-Security">Transport Layer Security <a href="#Transport-Layer-Security" title="Link">link</a></h2>
Cisco SIP Phones support three different transport security modes set using a combination of <<code class="tag">transportLayerProtocol</code>> and <<code class="tag">deviceSecurityMode</code>> in <a href="sepmac-cnf-xml.html#transportLayerProtocol">SEPMAC.cnf.xml</a>. The SSL certificate used by Asterisk must be included in <code class="literal">ITLFile.tlv</code> wih the <code class="literal">CCM</code> function or verifiable via TVS. See <a href="itl-file-tlv.html#tlvfile">Device Security</a> and <a href="trust-verification.html">Trust Verification</a> for more information.<br>
<br>
<table>
<tbody>
<tr>
<td><b>Non-Secure Mode</b></td>
<td>Connect using TCP, RTP is unencrypted</td>
</tr>
<tr>
<td><b>Authenticated Mode</b></td>
<td>Connect using TLS with the NULL cipher, RTP is unencrypted</td>
</tr>
<tr>
<td><b>Encrypted Mode</b></td>
<td>Connect using TLS with AES cipher, RTP is encrypted</td>
</tr>
</tbody>
</table>
<br>
<code class="prettify lang-astsip">[general]
...
; Only the following ciphers are supported, phone may fail to connect if others are specified
tlscipher=ECDHE-RSA-AES256-GCM-SHA384:ECDHE-RSA-AES128-GCM-SHA256:DHE-RSA-AES256-SHA:DHE-RSA-AES128-SHA
; NULL cipher is only needed if you are using Authenticated mode, otherwise is should not be enabled
;tlscipher+=:NULL
...
[non-secure-mode](!)
transport=tcp
[authenticated-mode](!)
transport=tls
[encrypted-mode](!)
transport=tls
; The res_srtp module must be loaded.
encryption=yes
encryption_taglen=80</code>
<br>
<b>Note</b>: To enable RTP encryption <code class="literal">libsrtp</code> must be installed. Additionally to use the newer <code class="literal">AES-128-GCM</code> and <code class="literal">AES-256-GCM</code> ciphers both Asterisk and <code class="literal">libsrtp</code> must have been compiled with support for them enabled. See <a href="itl-file-tlv.html#libsrtp">Device Security</a> for more information.<br>
<br>
<h2 id="Device-Model-Template">Device Model Template <a href="#Device-Model-Template" title="Link">link</a></h2>
Example individual device templates for each Cisco SIP phone model to handle the different features, inherits the settings from the <code class="literal">cisco-usecallmanager</code> template.<br>
<br>
<code class="prettify lang-astsip">[cisco-7941](!,cisco-usecallmanager)
; These should match <busyTrigger> and <maxNumCalls> in SEPMAC.cnf.xml
busylevel=3
call-limit=4
; Force huntgroup login so that the prompt does not show the logged out message
huntgroup_default=yes
[cisco-8841](!,cisco-usecallmanager)
busylevel=4
call-limit=5
[cisco-8865](!,cisco-usecallmanager)
busylevel=4
call-limit=5
; <videoCapability> also needs to be enabled in SEPMAC.cnf.xml
videosupport=yes
; Allow the video codec
allow=h264
[cisco-9951](!,cisco-usecallmanager)
busylevel=5
call-limit=6
; <videoCapability> also needs to be enabled in SEPMAC.cnf.xml
videosupport=yes
; Allow the video codec
allow=h264</code>
<br>
<h2 id="SIP-Phone-Peers">SIP Phone Peers <a href="#SIP-Phone-Peers" title="Link">link</a></h2>
Individual peer definitions for Cisco SIP phones, inherits the settings from the <code class="literal">cisco-<i>MODEL</i></code> template.<br>
<br>
<code class="prettify lang-astsip">[301](cisco-7941,non-secure-mode)
secret=Gsgf90tYZ26FNQgA
callerid="Alice" <301>
description=Alice
callgroup=1
pickupgroup=1
mailbox=301@default
; See extensions.conf for example on OPickup
setvar=OTHERPICKUPGROUP=2
; Encryption is enabled on this peer
[302](cisco-8841,encrypted-mode)
secret=eV4i5qrCxf0ohMyE
callerid="Bob" <302>
description=Bob
callgroup=1
pickupgroup=1
mailbox=302@default
; Extensions that the phone is watching, they need to be configured in SEPMAC.cnf.xml as well
subscribe=301
subscribe=303
subscribe=381
[303](cisco-8865,non-secure-mode)
secret=NH3d6r1WW1Kcvo9I
callerid="Cookie Monster" <303>
description=Cookie Monster
callgroup=1
pickupgroup=1
mailbox=303@default
; Abbreviated/speed dials for extensions
setvar=SPEEDDIAL_1=301
setvar=SPEEDDIAL_2=302</code>
<br>
<h2 id="Multiple-Lines">Multiple Lines <a href="#Multiple-Lines" title="Link">link</a></h2>
Cisco SIP phones that have more than one line must have each of those peers specified in their peer definition using <code class="literal">register</code>. Each additional bulk-registered peer will automatically have the same qualify, Do Not Disturb and Hunt Group state as the primary peer.<br>
<br>
<code class="prettify lang-astsip">; First line (lineIndex=1 in SEPMAC.cnf.xml)
[301](cisco-9951,non-secure-mode)
secret=LJoO6dgRJYzrCE5Y
callerid="Alice" <301>
description=Alice, Line 1
callgroup=1
pickupgroup=1
mailbox=301@default
; Second line (lineIndex=2 in SEPMAC.cnf.xml)
register=302
; Third line (lineIndex=3 in SEPMAC.cnf.xml)
register=303
[302](cisco-9951,non-secure-mode)
; Password is not used, but it prevents unauthenticated registration
secret=4FnipFu1YN6NoJhz
callerid="Alice" <302>
description=Alice, Line 2
[303](cisco-9951,non-secure-mode)
; Password is not used, but it prevents unauthenticated registration
secret=RzuR08uYYX0Pea5B
callerid="Alice" <303>
description=Alice, Line 3</code>
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<span class="icon">copyright</span> Gareth Palmer and individual contributors. Documentation distributed under <a href="LICENSE">CC BY 4.0</a>.
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