Hi, Im newbie to cisco VoIP tech. Ive tried to set up some testing network with one phone stand, somehow managed to make it work, but calls still dont go through. I´ll attach all the config files and can someone please help me? It´s cisco 7940 phone, I know its pretty outdated, but for testing seems to be enough.
sipdefault.cnf :
image_version: "P0S3-8-12-00"
proxy1_address: "sip.viptel.sk"
# proxy2_address: "xxx.xxx.xxx.xxx"
# proxy3_address: "xxx.xxx.xxx.xxx"
# proxy4_address: "xxx.xxx.xxx.xxx"
proxy1_port:"5060"
# proxy2_port:"5060"
# proxy3_port:"5060"
# proxy4_port:"5060"
proxy_emergency: ""
proxy_emergency_port: "5060"
proxy_backup: ""
proxy_backup_port: "5060"
outbound_proxy: "sip.viptel.sk"
outbound_proxy_port: "5060"
nat_enable: "0"
nat_address: ""
voip_control_port: "5060"
start_media_port: "16348"
end_media_port: "20134"
nat_received_processing: "1"
dyn_dns_addr_1: ""
dyn_dns_addr_2: ""
dyn_tftp_addr: "192.168.88.2"
tftp_cfg_dir: "./"
proxy_register: "1"
timer_register_expires: "120"
preferred_codec: "none"
tos_media: "5"
enable_vad: "0"
dial_template: "dialplan"
network_media_type: "auto"
autocomplete: "1"
telnet_level: "0"
cnf_join_enable: "1"
semi_attended_transfer: "0"
call_waiting: "1"
anonymous_call_block: "0"
callerid_blocking: "0"
dnd_control: "0"
dtmf_inband: "1"
dtmf_outofband: "avt"
dtmf_db_level: "3"
dtmf_avt_payload: "101"
timer_t1: "500"
timer_t2: "4000"
sip_retx: "10"
sip_invite_retx: "6"
timer_invite_expires: "180"
messages_uri: "*97"
#services_url: "http://example.domain.ext/services/menu.xml"
#directory_url: "http://example.domain.ext/services/directory.php"
#logo_url: "http://example.domain.ext/imagename.bmp"
http_proxy_addr: ""
http_proxy_port: 80
remote_party_id: 0
XMLDefault.cnf.xml :
<?xml version="1.0"?>
<Default>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName>sip.viptel.sk</processNodeName>
</callManager>
</member>
<member priority="1">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName>sip.viptel.sk</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<loadInformation307 model="SIP: Cisco IP Phone 7911">SIP11.8-5-4S</loadInformation307>
<loadInformation30007 model="SIP: Cisco 7912">CP7912080000SIP060111A</loadInformation30007>
<loadInformation495 model="SIP: Cisco 6921">SIP69xx.9-4-1-3SR2</loadInformation495>
<loadInformation8 model="SIP: Cisco 7940">P0S3-8-12-00</loadInformation8>
<loadInformation7 model="SIP: Cisco 7960">P0S3-8-12-00</loadInformation7>
<loadInformation115 model="SIP: Cisco 7941">SIP41.8-5-4S</loadInformation115>
<loadInformation309 model="SIP: Cisco 7941G-GE">SIP41.8-5-4S</loadInformation309>
<loadInformation30018 model="SIP: Cisco 7961">SIP41.8-5-4S</loadInformation30018>
<loadInformation308 model="SIP: Cisco 7961G-GE">SIP41.8-5-4S</loadInformation308>
<loadInformation434 model="SIP: Cisco 7942">SIP42.8-5-4S</loadInformation434>
<loadInformation404 model="SIP: Cisco 7962">SIP42.8-5-4S</loadInformation404>
<loadInformation435 model="SIP: Cisco 7945">SIP45.8-5-4S</loadInformation435>
<loadInformation436 model="SIP: Cisco 7965">SIP45.8-5-4S</loadInformation436>
<loadInformation621 model="SIP: Cisco 7821">sip78xx.11-0-1-11</loadInformation621>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<servicesURL></servicesURL>
</Default>
SIP(macaddress).cnf :
proxy1_address: "sip.viptel.sk"
proxy1_port=5060
line1_name: "name"
line1_shortname: "name"
line1_displayname: "name"
line1_authname: "username"
line1_password: "password"
proxy_emergency: ""
proxy_emergency_port: "5060"
proxy_backup: ""
proxy_backup_port: "5060"
outbound_proxy: ""
outbound_proxy_port: "5060"
nat_enable: "0"
nat_address: ""
voip_control_port: "5060"
start_media_port: "16348"
end_media_port: "20134"
nat_received_processing: "0"
phone_label: "name"
time_zone: UTC
dialplan.xml :
<DIALTEMPLATE>
<TEMPLATE MATCH="." TIMEOUT="15" User="Phone"/>
<TEMPLATE MATCH="...." TIMEOUT="2" User="Phone"/>
<TEMPLATE MATCH="9......." TIMEOUT="2" User="Phone"/>
<TEMPLATE MATCH="13...." TIMEOUT="2" User="Phone"/>
<TEMPLATE MATCH="02........" TIMEOUT="2" User="Phone"/>
</DIALTEMPLATE>
plus i have some ringtones and firmware stuff in there, think that shouldnt really matter, Ive got it from a github template, so hopefully its okay. Thanks for any replies.
