Cisco 7940 Not Accepting Calls

Hello all,

I am struggling with setting up my Cisco 7940 phones. The phones are not accepting calls but outgoing calls are successful. I tel netted into the phone and this is what I get with an incoming call:

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.04.21 19:28:49 =~=~=~=~=~=~=~=~=~=~=~=

Password :*****

Cisco Systems, Inc. Copyright 2000-2005
Cisco IP phone MAC: 000f:23ac:3948
Loadid: SW: P0S3-8-12-00 ARM: PAS3ARM1 Boot: PC030301 DSP: 4.0(5.0)[A0]
Cisco7960> debug sip-task
Enabling bug logging on this terminal - use ‘tty mon 0’ to disable
debugs: timestamp sip-task
Cisco7960> debug sip-message
debugs: timestamp sip-task sip-messages
Cisco7960> [19:29:11:12335] SIPTaskProcessListEvent: cmd = 0x160200
[19:29:11:12335] SIPProcessUDPMessage: recv UDP message from <68.64.160.19>:<50195>, length=<1355>, message=
[19:29:11:12335] INVITE sip:[email protected]:5060;user=phone;transport=udp SIP/2.0

Via: SIP/2.0/UDP 68.64.160.19:5060;branch=z9hG4bK6e46e1ba

Max-Forwards: 70

From: “Adam M. Skalicky” sip:[email protected];tag=as6f61ecec

To: sip:[email protected]:5060;user=phone;transport=udp

Contact: sip:[email protected]:5060

Call-ID: [email protected]:5060

CSeq: 102 INVITE

User-Agent: FPBX-2.11.0(11.4.0)

Date: Tue, 22 Apr 2014 02:29:12 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 715

v=0

o=root 540977059 540977059 IN IP4 68.64.160.19

s=Asterisk PBX 11.4.0

c=IN IP4 68.64.160.19

t=0 0

m=audio 14562 RTP/AVP 0 8 3 115 7 110 9 5 102 4 10 111 18 97 112 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:115 G7221/32000

a=fmtp:115 bitrate=48000

a=rtpmap:7 LPC/8000

a=rtpmap:110 speex/8000

a=rtpmap:9 G722/8000

a=rtpmap:5 DVI4/8000

a=rtpmap:102 G7221/16000

a=fmtp:102 bitrate=32000

a=rtpmap:4 G723/8000

a=fmtp:4 annexa=no

a=rtpmap:10 L16/8000

a=rtpmap:111 G726-32/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:97 iLBC/8000

a=fmtp:97 mode=30

a=rtpmap:112 G726-32/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

[19:29:11:12339] SIPTaskProcessSIPMessage: Line filter: Determining destination line…
[19:29:11:12340] SIPTaskProcessSIPMessage: Line filter: Call ID match not found: INVITE: free ccb index = 0.
[19:29:11:12341] Unknown address in Request URI
[19:29:11:12342] sipSPICheckRequest: Request URI Not Found
[19:29:11:12342] SIPTaskProcessSIPMessage: Error: sipSPICheckRequest() returned error.
[19:29:11:12345] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=INVITE, to_tag=<>)
[19:29:11:12346] sipTransportSendMessage: Opened a one-time UDP send channel to server <68.64.160.19>:<5060>, handle = 8 local port= 5060
[19:29:11:12346] sipTransportSendMessage:Sent SIP message to <68.64.160.19>:<5060>, handle=<8>, length=<352>, message=
[19:29:11:12347] SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 68.64.160.19:5060;branch=z9hG4bK6e46e1ba

From: “Adam M. Skalicky” sip:[email protected];tag=as6f61ecec

To: sip:[email protected]:5060;user=phone;transport=udp

Call-ID: [email protected]:5060

Date: Tue, 22 Apr 2014 02:29:11 GMT

CSeq: 102 INVITE

Content-Length: 0

[19:29:11:12348] sipTransportSendMessage: Closed a one-time UDP send channel handle = 8
[19:29:11:12367] SIPTaskProcessListEvent: cmd = 0x160200
[19:29:11:12368] SIPProcessUDPMessage: recv UDP message from <68.64.160.19>:<50195>, length=<453>, message=
[19:29:11:12368] ACK sip:[email protected]:5060;user=phone;transport=udp SIP/2.0

Via: SIP/2.0/UDP 68.64.160.19:5060;branch=z9hG4bK6e46e1ba

Max-Forwards: 70

From: “Adam M. Skalicky” sip:[email protected];tag=as6f61ecec

To: sip:[email protected]:5060;user=phone;transport=udp

Contact: sip:[email protected]:5060

Call-ID: [email protected]:5060

CSeq: 102 ACK

User-Agent: FPBX-2.11.0(11.4.0)

Content-Length: 0

[19:29:11:12370] SIPTaskProcessSIPMessage: Line filter: Determining destination line…
[19:29:11:12372] SIPTaskProcessSIPMessage: Line filter: Previous Call ID. Message not in reTx list.
[19:29:11:12372] SIPTaskProcessSIPMessage: Line filter: Call ID match not found: Rejecting.

The config I am using has worked before, I am not sure what is going on. I would really appreciate any help.

As always, have you researched the 7940/7960 threads? There has to be 100 of them.

You must always post FreePBX and Asterisk version along with how system was installed (distro or by hand). If by hand what OS?

Common issues with these phones. NAT must be disabled in extension and the FreePBX generated password can sometimes be too complex. Make it shorter and take out special characters.

You also didn’t indicate if phones are on same network as server.

Thank you so much for the response. I am running Asterisk 11.4.0 and if I am correct, FreePBX version 3.211.63-10. The install is from the distro.

Regarding the phones, I have them on a seperate network from the PBX. Because of this I have NAT enabled but have not specific the NAT address.

The phones can make outbound calls and show the incoming call getting to them, but decline it.

Regarding the password. I did figure that out after many long hours, but I did shorten and the phones registered.

What is odd is my softphone can accept calls just not the Cisco phones.

Would it be beneficial to post the config files I am using?

I also just search for problems of a similar nature on the forum, I was unable to find anything that pertained to this exact situation.

You can post the if you are configuring by hand. If you used the end point manager don’t bother.

I have never had great luck using those over NAT. NAT can cause rejects because it answers back wrong.

Frankly I don’t know if it’s worth all of this effort for a $20 10 year old phone. You can get a Polycom 501 on eBay for under $50 that does NAT perfect.

Certainly you should have NAT configured on the Cisco config and you may have to specify your external IP (which is a pain on dynamics)

Your other option is to use a VPN to the server. Most any router today supports basic LAN 2 LAN VPN’s.

I did do allot of the config by hand and moved some things around so I posted it below. I had the phones working before in a near identical configuration so this is really frustrating. I know they can work it’s weird that they are not. If we could not resolve this issue, would it be possible to have a FreePBX vm at each location the phones are at and then use my cloud based one for the actual registration with the carrier?

SIP Phone Specific:
phone_label: “{$displayname.line.1}”
{line_loop}
line{$line}_name: "{$authname}"
line{$line}_shortname: "{$shortname}"
line{$line}_displayname: "{$shortname}"
line{$line}_authname: "{$authname}"
line{$line}_password: “{$secret}”
{/line_loop}

services_url: "{$services_url}"
directory_url: "{$directory_url}"
logo_url: “{$logo_url}”

SIP Default
image_version: “{$image_name}”

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”

http_proxy_addr: ""
http_proxy_port: 80
remote_party_id: 0
nat_enable: "1"
nat_address: “”

messages_uri: “{$vmail|*97}”

proxy1_address: "{$server.ip.1}"
proxy1_port:“5060”

proxy_emergency: "{$server.ip.1}"
proxy_emergency_port: "5060"
proxy_backup: "{$server.ip.1}"
proxy_backup_port: "5060"
outbound_proxy: "{$server.ip.1}"
outbound_proxy_port: “5060”

proxy_register: "1"
timer_register_expires: "120"
preferred_codec: “none”

enable_vad: "0"
dial_template: "dialplan"
network_media_type: "auto"
autocomplete: "1"
telnet_level: “2”

voip_control_port: "5060"
start_media_port: "16348"
end_media_port: "20134"
nat_received_processing: "0"
dyn_tftp_addr: "{$server.ip.1}"
tftp_cfg_dir: “./”

time_zone: {$sip_7940_7960_time_zone}
sntp_mode: "directedbroadcast"
sntp_server: "{$network_time_server}"
time_format_24hr: "1"
dst_offset: "1"
dst_start_month: "March"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "2"
dst_start_time: "2"
dst_stop_month: "Nov"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "1"
dst_stop_time: "2"
dst_auto_adjust: “1”

telnet_level: "2"
phone_prompt: "Cisco7960"
phone_password: "{$telnet_password}"
enable_vad: "0"
network_media_type: "auto"
user_info: phone
date_format: “YY-M-D”