Configure Sip; Configure Basic Sip Parameters - Cisco 8831 Administration Manual

Configure SIP

<ExecuteItem Priority="0" URL="http://xmlserver.com/event.xml"/>
</CiscoIPPhoneExecute>
Authentication:
challenge
":" MD5(A2) )
where A1 = username ":" realm ":" passwd
and A2 = Method ":" digest-uri
Configure SIP
SIP settings for the Cisco Unified IP Conference Phone 8831 for Third-Party Call Control are configured
for the phone in general and for the extensions.

Configure Basic SIP Parameters

To configure general SIP parameters, navigate to Admin Login > advanced > Voice > SIP. Under SIP
Parameters, make these changes:
Parameter
Max Forward
Max Redirection
SIP User Agent Name User-Agent header used in outbound requests. The default is $VERSION.
SIP Server Name
SIP Reg User Agent
Name
SIP Accept Language The preferred languages for reason phrases, session descriptions, or status
RFC 2543 Call Hold
SIP TCP Port Min
Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide
3-4
= MD5( MD5(A1) ":" nonce ":" nc-value ":" cnonce ":" qop-value
Description
The number of proxies or gateways that can forward the request to the next
downstream server. The Max-Forwards value is an integer in the range of
0 to 255 indicating the remaining number of times the request message is
allowed to be forwarded. This count is decremented by each server that
forwards the request. The initial value is 70.
Number of times an invite can be redirected to avoid an infinite loop. The
default is 5.
If empty, the header is not included. Macro expansion of $A to $D
corresponding to GPP_A to GPP_D allowed.
Server header used in responses to inbound responses. The default is
$VERSION.
User-Agent name used in a REGISTER request. If not specified, the SIP
User Agent Name is used for the REGISTER request.
responses carried as message bodies in the response. If blank, the header
is not included and the server assumes that all languages are acceptable to
the client. Defaults to blank.
If set to Yes, the Cisco Unified IP Conference Phone 8831 for Third-Party
Call Control includes Session Description Protocol (SDP) syntax
c=0.0.0.0 when sending a SIP re-INVITE to a peer to hold the call. If set
to No, the phone does not include the c=0.0.0.0 syntax in the SDP. With
either setting, the phone includes a=sendonly syntax in the SDP. Defaults
to Yes.
Lowest TCP port number that can be used for SIP sessions. Defaults to
5060.
Chapter 3 Configure SIP and NAT