CISCO LDAP XML Directory

image_version: P0S3-06-2-00

proxy1_address: "serv2.simstream.com"		; Can be dotted IP or FQDN

proxy1_port: 5060 

proxy_register: 1 

# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 3600 

# Codec for media stream (g711ulaw (default), g711alaw, g729a)
preferred_codec: g711ulaw

# TOS bits in media stream [0-5] (Default - 5)
tos_media: 5

# Inband DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: 1

# Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt )
dtmf_outofband: avt

# DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up)
dtmf_db_level: 3

# SIP Timers
timer_t1: 500 			; Default 500 msec
timer_t2: 4000 			; Default 4 sec
sip_retx: 10			; Default 10
sip_invite_retx: 6 		; Default 6
timer_invite_expires: 180 	; Default 180 sec

####### New Parameters added in Release 2.0 #######

# Dialplan template (.xml format file relative to the TFTP root directory)
dial_template: dialplan

# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./sip-phones/"
 
# Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)
sntp_server: "134.59.1.5"	; SNTP Server IP Address
sntp_mode: directedbroadcast	; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: CET			; Time Zone Phone is in
dst_offset: 1			; Offset from Phone's time when DST is in effect 
dst_start_month: April		; Month in which DST starts
dst_start_day: ""		; Day of month in which DST starts
dst_start_day_of_week: Sun	; Day of week in which DST starts
dst_start_week_of_month: 1	; Week of month in which DST starts
dst_start_time: 02		; Time of day in which DST starts
dst_stop_month: Oct		; Month in which DST stops
dst_stop_day: ""		; Day of month in which DST stops
dst_stop_day_of_week: Sunday	; Day of week in which DST stops
dst_stop_week_of_month: 8	; Week of month in which DST stops 8=last week of month
dst_stop_time: 2		; Time of day in which DST stops
dst_auto_adjust: 1		; Enable(1-Default)/Disable(0) DST automatic adjustment
time_format_24hr: 1		; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)

# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control)
dnd_control: 0			; Default 0 (Do Not Disturb feature is off)

# Caller ID Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
callerid_blocking: 0		; Default 0 (Disable sending all calls as anonymous) 

# Anonymous Call Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: 0		; Default 0 (Disable blocking of anonymous calls)

# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: 101		; Default 101

# Sync value of the phone used for remote reset 
sync: 1				; Default 1

####### New Parameters added in Release 2.1 #######

# Backup Proxy Support
proxy_backup: ""		; Dotted IP of Backup Proxy
proxy_backup_port: 5060		; Backup Proxy port (default is 5060)

# Emergency Proxy Support
proxy_emergency: "" 		; Dotted IP of Emergency Proxy
proxy_emergency_port: 5060	; Emergency Proxy port (default is 5060)

# Configurable VAD option
enable_vad: 0			; VAD setting 0-disable (Default), 1-enable

####### New Parameters added in Release 2.2 ######

# NAT/Firewall Traversal
nat_enable: 0                   ; 0-Disabled (default), 1-Enabled
nat_address: ""		        ; WAN IP address of NAT box (dotted IP or DNS A record only)
voip_control_port: 5060      	; UDP port used for SIP messages (default - 5060)
start_media_port: 16384 	; Start RTP range for media (default - 16384)
end_media_port: 32766   	; End RTP range for media (default - 32766)
nat_received_processing: 0	; 0-Disabled (default), 1-Enabled

# Outbound Proxy Support
outbound_proxy: ""	 	; restricted to dotted IP or DNS A record only
outbound_proxy_port: 5060       ; default is 5060

####### New Parameter added in Release 3.0 #######

# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable : 1		; 0-Disabled, 1-Enabled (default)

####### New Parameters added in Release 3.1 #######

# Allow Transfer to be completed while target phone is still ringing
semi_attended_transfer: 1	; 0-Disabled, 1-Enabled (default)

# Telnet Level (enable or disable the ability to telnet into the phone) 
telnet_level: 2			; 0-Disabled (default), 1-Enabled, 2-Privileged

####### New Parameters added in Release 4.0 #######

# XML URLs
services_url: "http://serv1.simstream.com/phone/directory.php"		; URL for external Phone Services
directory_url: "http://serv1.simstream.com/phone/directory.php" ; URL for external Directory location
logo_url: "http://serv1.simstream.com/phone/logo.bmp"			; URL for branding logo to be used on phone display
messages_uri: "123"

# HTTP Proxy Support
http_proxy_addr: ""		; Address of HTTP Proxy server
http_proxy_port: 80		; Port of HTTP Proxy Server (80-default)

# Dynamic DNS/TFTP Support
dyn_dns_addr_1: ""              ; restricted to dotted IP
dyn_dns_addr_2: ""              ; restricted to dotted IP
dyn_tftp_addr: ""               ; restricted to dotted IP

# Remote Party ID
remote_party_id: 0		; 0-Disabled (default), 1-Enabled

####### New Parameters added in Release 4.4 #######

# Call Hold Ringback (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
call_hold_ringback: 0		; Default 0 (Disable ringback of held call)

<CiscoIPPhoneInput> 
<Title>Annuaire LDAP Simstream</Title> 
<Prompt>Prompt text.</Prompt> 
<URL>http://serv1.simstream.com/phone/result.php</URL> 
<InputItem> 
<DisplayName>Filtre</DisplayName> 
<QueryStringParam>filtre</QueryStringParam> 
<InputFlags></InputFlags> 
<DefaultValue></DefaultValue> 
</InputItem>  
</CiscoIPPhoneInput>

<?php
// This XML-PHP page searches contacts in LDAP and display it on CISCO directory service
// Script Provided by Gael Marronnier From Simstream Company www.simstream.com [email protected]
// 
// First I get the parameter
$filtre = $_REQUEST["filtre"];

// Now I contuct the LDAP query
$ds=ldap_connect("serv3.simstream.com"); 
$myldap="dc=simstream,dc=com";

if ($ds) {
//   $r=ldap_bind($ds);
   $filtreldap="(&(objectclass=person)(|(givenname=*".$filtre."*)(sn=*".$filtre."*)(cn=*".$filtre."*)(mail=*".$filtre."*)))";
   $sr=ldap_search($ds, $myldap, $filtreldap);
// Lets start displaying XML infos
echo "<CiscoIPPhoneDirectory>\n";
			echo "<Title>IP Telephony Directory</Title>\n";
   $info = ldap_get_entries($ds, $sr);
   echo "<Prompt>";
   echo $info["count"];
   echo " Reachable people</Prompt>\n";
	   for ($i=0; $i<$info["count"]; $i++) {
      if ($info[$i]["mobile"][0]) {
      echo "<DirectoryEntry>\n";
	  echo "<Name>M:" .  iconv("UTF-8","ISO-8859-1",$info[$i]["cn"][0]) . "</Name>\n";
	  // I replace the french international extension (+33) by 0. It s not really clean, fell free to improve
	  echo "<Telephone>" .str_replace(" ","",str_replace("+33","0", $info[$i]["mobile"][0])) ."</Telephone>\n";
      echo "</DirectoryEntry>\n";
	  }
      if ($info[$i]["telephonenumber"][0]) {
      echo "<DirectoryEntry>\n";
	  echo "<Name>T:" .  iconv("UTF-8","ISO-8859-1",$info[$i]["cn"][0]) . "</Name>\n";
	  echo "<Telephone>" .str_replace(" ","",str_replace("+33","0", $info[$i]["telephonenumber"][0])) ."</Telephone>\n";
      echo "</DirectoryEntry>\n";
	  }
      if ($info[$i]["telephonenumber"][1]) {
      echo "<DirectoryEntry>\n";
	  echo "<Name>T:" .  iconv("UTF-8","ISO-8859-1",$info[$i]["cn"][0]) . "</Name>\n";
	  echo "<Telephone>" .str_replace(" ","",str_replace("+33","0", $info[$i]["telephonenumber"][1])) ."</Telephone>\n";
      echo "</DirectoryEntry>\n";
	  }
	  if ($info[$i]["telephonenumber"][2]) {
      echo "<DirectoryEntry>\n";
      echo "<Name>T:" .  iconv("UTF-8","ISO-8859-1",$info[$i]["cn"][0]) . "</Name>\n";
      echo "<Telephone>" .str_replace(" ","",str_replace("+33","0", $info[$i]["telephonenumber"][2])) ."</Telephone>\n";
      echo "</DirectoryEntry>\n";
      }

  		 }
	  echo "</CiscoIPPhoneDirectory>";
      ldap_close($ds);
} else {
   echo "error";
}

?>