My cisco 7940 can call but not receive calls.
Here's my debug on my phone when I try to call it.
[02:02:05:490875] SIPTaskProcessListEvent: cmd = 0x160200
[02:02:05:490876] SIPProcessUDPMessage: recv UDP message from <213.179.55.150>:<50195>, length=<1113>, message=
[02:02:05:490876] INVITE sip:USERNAME@IP:16751 SIP/2.0
Via: SIP/2.0/UDP 213.179.55.150:5060;rport;branch=z9hG4bKPj40897d67-0ce9-44e7-bdb7-4dadab5471aa
From: "+PHONENUMBER" <sip:+PHONENUMBER@IP>;tag=a9cc87a5-9a6e-42be-b4b3-3c6399c79082
To: <sip:USERNAME@IP>
Contact: <sip:asterisk@213.179.55.150:5060>
Call-ID: 1a0119ac-8712-47e2-91a2-3f54dd8b2b67
CSeq: 16804 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Oyatel/1.0
Content-Type: application/sdp
Content-Length: 430
v=0
o=- 305020534 305020534 IN IP4 213.179.55.150
s=Oyatel
c=IN IP4 213.179.55.150
t=0 0
a=group:BUNDLE audio-0
m=audio 25530 RTP/AVP 8 0 107 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:494713794 cname:6bacf9f8-00e0-4fbf-acbb-fcfb40ceb7c8
a=mid:audio-0
[02:02:05:490880] SIPTaskProcessSIPMessage: Line filter: Determining destination line...
[02:02:05:490881] SIPTaskProcessSIPMessage: Line filter: Call ID match not found: INVITE: free ccb index = 0.
[02:02:05:490882] Port Mismatch(UDP), URL Port: 16751, Port Used: 5060
[02:02:05:490882] sipSPICheckRequest: Request URI Not Found
[02:02:05:490883] SIPTaskProcessSIPMessage: Error: sipSPICheckRequest() returned error.
[02:02:05:490883] sipSPISendErrorResponse: Sending response 404...
[02:02:05:490885] sipRelCoupledMessageStore: Storing for reTx (cseq=16804, method=INVITE, to_tag=<>)
[02:02:05:490886] sipTransportSendMessage: Opened a one-time UDP send channel to server <213.179.55.150>:<5060>, handle = 3 local port= 5060
[02:02:05:490887] sipTransportSendMessage:Sent SIP message to <213.179.55.150>:<5060>, handle=<3>, length=<376>, message=
[02:02:05:490888] SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 213.179.55.150:5060;rport;branch=z9hG4bKPj40897d67-0ce9-44e7-bdb7-4dadab5471aa
From: "+4746785889" <sip:+PHONENUMBER@213.179.55.150>;tag=a9cc87a5-9a6e-42be-b4b3-3c6399c79082
To: <sip:USERNAMRE@IP>
Call-ID: 1a0119ac-8712-47e2-91a2-3f54dd8b2b67
Date: Mon, 04 Jun 2018 09:02:05 GMT
CSeq: 16804 INVITE
Content-Length: 0
[02:02:05:490889] sipTransportSendMessage: Closed a one-time UDP send channel handle = 3
[02:02:05:490890] SIPTaskProcessListEvent: cmd = 0x160200
[02:02:05:490890] SIPProcessUDPMessage: recv UDP message from <213.179.55.150>:<50195>, length=<404>, message=
[02:02:05:490891] ACK sip:USERNAME@IP:16751 SIP/2.0
Via: SIP/2.0/UDP 213.179.55.150:5060;rport;branch=z9hG4bKPj40897d67-0ce9-44e7-bdb7-4dadab5471aa
From: "+PHONENUMBER" <sip:+PHONENUMBER@213.179.55.150>;tag=a9cc87a5-9a6e-42be-b4b3-3c6399c79082
To: <sip:USERNAME@IP>
Call-ID: 1a0119ac-8712-47e2-91a2-3f54dd8b2b67
CSeq: 16804 ACK
Max-Forwards: 70
User-Agent: Oyatel/1.0
Content-Length: 0
[02:02:05:490893] SIPTaskProcessSIPMessage: Line filter: Determining destination line...
[02:02:05:490894] SIPTaskProcessSIPMessage: Line filter: Previous Call ID. Message not in reTx list.
[02:02:05:490895] SIPTaskProcessSIPMessage: Line filter: Call ID match not found: Rejecting.
Here is my phone config
SIP Phone> show config
------ Current *FLASH* Configuration ------
Platform : Cisco Systems, Inc. IP Phone CP-7940G
Elapsed Time: 01:38:30
dhcp_server : 10.13.37.1
my_ip_addr : 10.13.37.62
subnet_mask : 255.255.255.0
defaultgw : 10.13.37.1
dyn_dns_addr_1 : 0.0.0.0
dyn_dns_addr_2 : 0.0.0.0
dns_addr : HIDDEN
dns_backup_1: HIDDEN
primary_tftp_addr : 10.13.37.55
dyn_tftp_addr : 0.0.0.0
my_mac_addr : 0013:c39b:2094
domain_name : HIDDEM
my_name : SIP0013C39B2094
Status Flags : 12310000
image_version : "P0S3-8-12-00"
FirmLoadID : "PC030301"
DSPLoadID : ""
network_media_type : Auto
network_port2_type : Hub/Switch
dscpForAudio : 184
phone_label : "UNPROVISIONED"
tftp_cfg_dir : "./"
phone_password : **********
phone_prompt : "SIP Phone"
language : english
sntp_mode : DirectedBroadcast
sntp_server : time.google.com
time_zone : PST
dst_offset : 1
dst_start_month : March
dst_start_day : 10
dst_start_day_of_week : Sun
dst_start_week_of_month : 1
dst_start_time : 2
dst_stop_month : Nov
dst_stop_day : 3
dst_stop_day_of_week : Sunday
dst_stop_week_of_month : 1
dst_stop_time : 2
dst_auto_adjust : 1
time_format_24hr : 1
date_format : M/D/Y
nat_enable : 1
nat_address : 10.13.37.62
voip_control_port : 5060
start_media_port : 16384
end_media_port : 20134
sync : "1"
xml_card_dir : ""
xml_card_file : "CARD.XML"
telnet_level : 2
services_url : ""
directory_url : ""
logo_url : "LOGO"
http_proxy_addr :
http_proxy_port : 80
garp_enable : 0
enable_vad : 0
dial_template : "dialplan"
callerid_blocking : 0
anonymous_call_block : 0
autocomplete : 1
messages_uri : "*9909412"
dnd_control : 0
preferred_codec : none
dtmf_outofband : avt
dtmf_avt_payload : 101
dtmf_db_level : 3
dtmf_inband : 1
call_manager1_addr : "UNPROVISIONED"
call_manager2_addr : "UNPROVISIONED"
call_manager3_addr : "UNPROVISIONED"
call_manager1_sip_port : 5060
call_manager2_sip_port : 5060
call_manager3_sip_port : 5060
call_manager5_addr : "UNPROVISIONED"
call_manager5_sip_port : 5060
call_manager4_addr : "UNPROVISIONED"
call_manager4_sip_port : 0
line1_name : "SIP USERNAME"
line2_name : ""
line1_authname : "SIP USERNAME"
line2_authname : "UNPROVISIONED"
line1_password : **********
line2_password : **********
line1_shortname : "SIP USERNAME"
line2_shortname : ""
line1_displayname : "SIP USERNAME"
line2_displayname : ""
line1_contact : "UNPROVISIONED"
line2_contact : "UNPROVISIONED"
proxy1_address : "SIP SERVER"
proxy2_address : "UNPROVISIONED"
proxy1_port : 5060
proxy2_port : 5060
sip_retx : 10
sip_invite_retx : 6
timer_t1 : 500
timer_t2 : 4000
timer_invite_expires : 180
timer_register_expires : 120
proxy_register : 1
proxy_backup : ""
proxy_emergency : ""
proxy_backup_port : 5060
proxy_emergency_port : 5060
outbound_proxy :
outbound_proxy_port : 5060
nat_received_processing : 1
mwi_status : 0
call_waiting : 1
user_info : none
cnf_join_enable : 0
remote_party_id : 0
semi_attended_transfer : 1
transfer_onhook_enabled : 0
call_hold_ringback : 0
stutter_msg_waiting : 0
cfwd_url : ""
call_stats : 0
auto_answer : 0
local_cfwd_enable : 1
timer_register_delta : 5
sip_max_forwards : 70
rfc_2543_hold : 0
version_stamp : ""
timer_keepalive_expires : 120
connection_monitor_duration : 120
encrypt_key : **********
SIP Phone>
Have searched the internet and this forum hoping to find out what's wrong. but nothing helps. Have tested the account in softphone apps and it works smoothly.
SIPMAC.cnf
line1_name: "usernamne"
line1_shortname: "username"
line1_displayname: "username"
line1_authname: "username"
line1_password: "password"
line2_name: ""
line2_shortname: ""
line2_displayname: ""
line2_authname: "UNPROVISIONED"
line2_password: "UNPROVISIONED"
messages_uri: "*9909412"
SIPDefault.cnf
image_version: "P0S3-8-12-00"
proxy1_address: "SIP SERVER IP"
# proxy2_address: "xxx.xxx.xxx.xxx"
# proxy3_address: "xxx.xxx.xxx.xxx"
# proxy4_address: "xxx.xxx.xxx.xxx"
# Proxy Server Port
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: ""
outbound_proxy_port: "5060"
nat_enable: "1"
nat_address: "10.13.37.62"
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: ""
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: "2"
cnf_join_enable: "0"
semi_attended_transfer: "1"
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"
sntp_mode: "directedbroadcast"
sntp_server: "time.google.com"
time_zone: "CEST"
time_format_24hr: "1"
dst_offset: "1"
dst_start_month: "March"
dst_start_day: "10"
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "1"
dst_start_time: "2"
dst_stop_month: "Nov"
dst_stop_day: "3"
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "1"
dst_stop_time: "2"
dst_auto_adjust: "1"
# services_url: "http://example.domain.tld/services/menu.xml"
# directory_url: "http://example.domain.tld/services/directory.php"
# logo_url: "http://example.domain.tld/imagename.bmp"
http_proxy_addr: ""
http_proxy_port: 80
remote_party_id: 0
