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inbound and outbound calls failed between CUBE and Telco

Remon Adel
Level 1
Level 1

Dears 
we have this  scenario 
CUCM Cluster (Pub & Sub ) ----SIP TRUNK----CUBE----SIP Provider .

We face an issue with incoming calls with this error messages 
Cause Value=127
Also outbound calls failed with reason code
 Reason: Q.850;cause=102

Please check attached 
Show run of CUBE

logs of below debugs in case of inbound and outbound calls 

debug voip ccapi inout

debug ccsip messages

debug voip rtp session named-event


also find output of 
show sip-ua timers
show sip-ua retry
show sip-ua min-se

Thanks

2 Accepted Solutions

Accepted Solutions

The incoming call is now failing because of the same reason as the outgoing calls.
The call reaches CUCM and we get the far end is ringing.


We receive 180 Ringing from CUCM
Jul 26 08:57:47.056: //3658/CD08C7C08F00/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.112.4.50:5060;branch=z9hG4bK341C09

We forward this to Telco
Jul 26 08:57:47.057: //3657/CD08C7C08F00/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 202.163.63.213:5060;branch=z9hG4bK04B676bae13a8e8da89

But the Telco sends us an INVITE message again.

Jul 26 08:57:47.485: //3657/CD08C7C08F00/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:69326900@198.18.211.90:5060 SIP/2.0
Via: SIP/2.0/UDP 202.163.63.213:5060;branch=z9hG4bK04B676bae13a8e8da89

The Telco's equipment is behaving as though none of our messages are reaching them. Please check the network topology between the CUBE and the Telco and ensure the packets are not being dropped somewhere in between.
Please also involve the Telco and ask them if they are receiving the messages from the CUBE.

View solution in original post

Hello,
Issue has been Solved 
Telco informed us that we use source port different than 5060 ,so we configure sip source port to be 5060 be below commands
voice service voip
sip
isten-port non-secure 5060
listen port Change will be effective once the SIP service is re-started
call service stop
no call service stop

Also we add this commands 

sip-ua
connection-reuse

Thanks

View solution in original post

13 Replies 13

Sreekanth Narayanan
Cisco Employee
Cisco Employee
Outgoing calls are failing because there is no response to the INVITE message sent to the telco. Please check if you are able to ping that IP or establish a telnet connection to port 5060 on that IP. If you are able to do this, then try a traceroute as well. Perhaps there is a firewall blocking these messages being sent to telco. If that is not the case, then please contact the provider.

For the incoming call, the min-se parameter needs to be changed to 1800. The remote end of that call, 10.112.4.51, is asking for a minimum value of 1800.
Jul 25 07:52:23.683: //651/8024AC1B82FD/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 422 Session Timer too small
Min-SE: 1800


voice service voip
sip
min-se 1800 session-expires 1800

Hello  Sreekanth Narayanan

T
hanks for your replay 
Please note that I can ping SIP Proxy Server IP address successfully .

and this SIP circuit was works fine on another IPPBX 

Thanks


Do you have logs from an outbound call on the other IPPBX which is working? Are all IP addresses involved the same?

Hello Sreekanth Narayanan

Kindly check below logs of incoming call , i noticed that , called number has been repeated .It must be 
69326900 but in logs become  6932690069326900 .

Please advice

   Set Up Event Sent;

   Call Info(Calling Number=91239850(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),

   Called Number=6932690069326900(TON=Unknown, NPI=Unknown))

Jul 25 07:52:23.677: //650/8024AC1B82FD/CCAPI/cc_process_call_setup_ind:

   Event=0x7FF44E2BA4E8

Jul 25 07:52:23.677: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:

   Try with the demoted called number 6932690069326900

This is happening because of the inbound sip profile configured on Dial peer 1.

voice class sip-profiles 1
request ANY sip-header To copy "sip:(.*)@" u10
request ANY sip-header SIP-Req-URI modify "@(.*)" "\u10@\1"
!
The number in the To is being appended to the number in the Req URI. Do you need the number to be replaced? Is this a required config?

Hello It's not required config to change called number so i remove this profile , also i set Min-SE: 1800 But we face same issue as attached logs

The reason why it is failing now is because the Telco is sending an INVITE with Min-se: 120. Since we have set it to 1800 on our CUBE router, it responds to the Telco's INVITE with 422 session timer too small.

Can you please change the config to:
voice service voip
sip
min-se 120 session-expires 1080

Same issue ,check logs

It's the CUCM which is rejecting the call now. Please change the Min-SE parameter on the CUCM.

To configure CUCM to have a Minimum Session Expires time to 1080:

1. Log in to CUCM.

2. Go to

System > Service Parameters.

3. Select

Server = current server e.g. " (Active)".

4. Select

Service = Cisco CallManager (Active).

5. Search for

SIP Min-SE Value and set it to 1080.

After this changes incoming calls success ,but no rtp and disconnect after 20s 
attached logs

The incoming call is now failing because of the same reason as the outgoing calls.
The call reaches CUCM and we get the far end is ringing.


We receive 180 Ringing from CUCM
Jul 26 08:57:47.056: //3658/CD08C7C08F00/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.112.4.50:5060;branch=z9hG4bK341C09

We forward this to Telco
Jul 26 08:57:47.057: //3657/CD08C7C08F00/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 202.163.63.213:5060;branch=z9hG4bK04B676bae13a8e8da89

But the Telco sends us an INVITE message again.

Jul 26 08:57:47.485: //3657/CD08C7C08F00/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:69326900@198.18.211.90:5060 SIP/2.0
Via: SIP/2.0/UDP 202.163.63.213:5060;branch=z9hG4bK04B676bae13a8e8da89

The Telco's equipment is behaving as though none of our messages are reaching them. Please check the network topology between the CUBE and the Telco and ensure the packets are not being dropped somewhere in between.
Please also involve the Telco and ask them if they are receiving the messages from the CUBE.

Hello,
Issue has been Solved 
Telco informed us that we use source port different than 5060 ,so we configure sip source port to be 5060 be below commands
voice service voip
sip
isten-port non-secure 5060
listen port Change will be effective once the SIP service is re-started
call service stop
no call service stop

Also we add this commands 

sip-ua
connection-reuse

Thanks

Awesome! Good to know.