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Unable to dial between SIP using CME,

zhir
Beginner
Beginner

hey there, i have two 7841 sip phones and successfully registered on the CME as SIP phones but i can't, Below is info regarding the current configuration,

Router# show run | section dial-peer
dial-peer voice 1 voip
destination-pattern 1000
session target ipv4:10.10.10.2
codec g711ulaw
dial-peer voice 2 pots
destination-pattern 1000
dial-peer voice 100 voip
destination-pattern 1000
session target ipv4:10.10.10.2
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 101 pots
destination-pattern 1000
forward-digits all
dial-peer voice 1001 voip
destination-pattern 1001
session target ipv4:10.10.10.2
no vad
dial-peer voice 1003 voip
destination-pattern 1003
session target ipv4:10.10.10.3
no vad
dial-peer voice 1002 voip
destination-pattern 1002
session target ipv4:10.10.10.5
codec g711ulaw
no vad
Router#
----------------------------------------


Router#show sip-ua status registrar
Line destination expires(sec) contact
transport call-id
peer
============================================================
1003 10.10.10.3 1412 10.10.10.3
UDP 2241368959@10.10.10.3
40002

1003 10.10.10.3 2078 10.10.10.3
UDP 3905824348@10.10.10.3
40002

1001 10.10.10.2 2998 10.10.10.2
UDP a4b43931-b8810016-52e542ec-69dfc7e8@10.10.10.2
40001

1002 10.10.10.5 1713 10.10.10.5
UDP a4b43931-b9df0002-1f1a9584-7517aa07@10.10.10.5
40003

Router#

------------------------------


Router#sh voice register pool all
Pool Tag 1
Config:
Mac address is A4B4.3931.B881
Type is 7841
Number list 1 : DN 1
Proxy Ip address is 0.0.0.0
Current Phone load version is Cisco-CP7841/12.0.1
DTMF Relay is disabled
Call Waiting is enabled
DnD is disabled
Video is disabled
Camera is disabled
Night Service Bell is disabled
Busy trigger per button value is 2
keep-conference is enabled
registration expires timer max is 3600 and min is 60
username user1 password 1234
kpml signal is enabled
Lpcor Type is none

Transport type is udp
service-control mechanism is supported
registration Call ID is a4b43931-b8810016-52e542ec-69dfc7e8@10.10.10.2
Registration method: per line
Privacy feature is not configured.
Privacy button is disabled
active primary line is: 1001

contact IP address: 10.10.10.2 port 5060

Phone SIS Version: 7.0.0
GW SIS Version: 1.0.0
conference admin: no
conference add mode: all
conference drop mode: never
paging-dn: config 0 [multicast] effective 0 [multicast]

Dialpeers created:

Dial-peers for Pool 1:

dial-peer voice 40001 voip
destination-pattern 1001$
session target ipv4:10.10.10.2:5060
session protocol sipv2
digit collect kpml
codec g711ulaw bytes 160
after-hours-exempt FALSE

Statistics:
Active registrations : 1

Total SIP phones registered: 3
Total Registration Statistics
Registration requests : 2
Registration success : 2
Registration failed : 0
unRegister requests : 1
unRegister success : 1
unRegister failed : 0
Auto-Register requests : 0
Attempts to register
after last unregister : 0
Last register request time : *07:12:58.227 UTC Sat Dec 2 2023
Last unregister request time : *07:12:44.171 UTC Sat Dec 2 2023
Register success time : *07:12:58.227 UTC Sat Dec 2 2023
Unregister success time : *07:12:44.177 UTC Sat Dec 2 2023


Pool Tag 2
Config:
Mac address is A4B4.3931.B9DF
Type is 7841
Number list 1 : DN 2
Proxy Ip address is 0.0.0.0
Current Phone load version is Cisco-CP7841/12.8.1
DTMF Relay is disabled
Call Waiting is enabled
DnD is disabled
Video is disabled
Camera is disabled
Night Service Bell is disabled
Busy trigger per button value is 2
keep-conference is enabled
registration expires timer max is 3600 and min is 60
username user2 password 1234
kpml signal is enabled
Lpcor Type is none

Transport type is udp
service-control mechanism is supported
registration Call ID is a4b43931-b9df0002-1f1a9584-7517aa07@10.10.10.5
Registration method: per line
Privacy feature is not configured.
Privacy button is disabled
active primary line is: 1002

contact IP address: 10.10.10.5 port 5060

Phone SIS Version: 7.0.0
GW SIS Version: 1.0.0
conference admin: no
conference add mode: all
conference drop mode: never
paging-dn: config 0 [multicast] effective 0 [multicast]

Dialpeers created:

Dial-peers for Pool 2:

dial-peer voice 40003 voip
destination-pattern 1002$
session target ipv4:10.10.10.5:5060
session protocol sipv2
digit collect kpml
codec g711ulaw bytes 160
after-hours-exempt FALSE

Statistics:
Active registrations : 1

Total SIP phones registered: 3
Total Registration Statistics
Registration requests : 1
Registration success : 1
Registration failed : 0
unRegister requests : 0
unRegister success : 0
unRegister failed : 0
Auto-Register requests : 0
Attempts to register
after last unregister : 0
Last register request time : *07:51:33.123 UTC Sat Dec 2 2023
Last unregister request time :
Register success time : *07:51:33.123 UTC Sat Dec 2 2023
Unregister success time :


Pool Tag 3
Config:
Mac address is 001B.D401.34CC
Type is 7912
Number list 1 : DN 3
Proxy Ip address is 0.0.0.0
Current Phone load version is Cisco-CP7912/8.0.1-060412A
DTMF Relay is disabled
Call Waiting is enabled
DnD is disabled
Video is disabled
Camera is disabled
Night Service Bell is disabled
Busy trigger per button value is 0
keep-conference is enabled
registration expires timer max is 3600 and min is 60
username user3 password 1234
Lpcor Type is none

Transport type is udp
service-control mechanism is supported
registration Call ID is 3905824348@10.10.10.3
Registration method: per line
Privacy feature is not configured.
Privacy button is disabled
active primary line is: 1003

contact IP address: 10.10.10.3 port 5060

conference admin: no
conference add mode: all
conference drop mode: never
paging-dn: config 0 [multicast] effective 0 [multicast]

Dialpeers created:

Dial-peers for Pool 3:

dial-peer voice 40002 voip
destination-pattern 1003$
session target ipv4:10.10.10.3:5060
session protocol sipv2
codec g711ulaw bytes 160
after-hours-exempt FALSE

Statistics:
Active registrations : 2

Total SIP phones registered: 3
Total Registration Statistics
Registration requests : 3
Registration success : 3
Registration failed : 0
unRegister requests : 1
unRegister success : 1
unRegister failed : 0
Auto-Register requests : 0
Attempts to register
after last unregister : 0
Last register request time : *07:57:37.587 UTC Sat Dec 2 2023
Last unregister request time : *08:15:08.943 UTC Sat Dec 2 2023
Register success time : *07:57:37.587 UTC Sat Dec 2 2023
Unregister success time : *08:15:08.943 UTC Sat Dec 2 2023

Debugging:

this what i got from chat gpt4 when tried to analyze my logs when i tried to dial between the phones :

INVITE Request Received: The router receives an INVITE request for a SIP call from Phone 1001 (Cisco-CP7841/12.0.1) to Phone 1002. The request includes details like the calling party (Phone 1001), called party (Phone 1002), and Session Description Protocol (SDP) data, which specifies the codecs and media capabilities.

Call Setup Indication: The router processes this INVITE and sets up the call parameters. It recognizes the calling number (1001) and the called number (1002). The call is handled by the incoming dial-peer 40001.

Sending 100 Trying: The router sends a "100 Trying" response, indicating it has received the INVITE and is attempting to process it. This is a standard part of the SIP call setup process.

Call Disconnect with Cause Value=3: Shortly after the "100 Trying" response, the router sends a "404 Not Found" response and disconnects the call. The "Cause Value=3" in the disconnect message is significant. In the SIP protocol, this typically indicates that the called party (Phone 1002 in this case) could not be found or is not available.

ACK Received: The router then receives an ACK for the "404 Not Found" response. This is part of the SIP protocol's way of acknowledging that the calling party (Phone 1001) has received the notification of the failed call setup.

The critical point here is the "404 Not Found" response with "Cause Value=3". This suggests that the call is failing because the router cannot route the call to the destination (Phone 1002). There are several potential reasons for this:

appreciate your help,

1 Accepted Solution

Accepted Solutions

Have you added allow-connections sip to sip under voice service voip?



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View solution in original post

2 Replies 2

Have you added allow-connections sip to sip under voice service voip?



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Thanks a lot it worked just fine, Appreciate it.

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