Busy tone when dialing a specific external number

[shaharshad]

Hi,

We always get a busy dial tone when dail a specific number(external) otherwise there is no issue in making external calls. I have also dialed that number from mobile and landline that's working fine. Can someone help?

We are using Call manager 10.5

Thanks

[Ruslan Amrahov]

Hi Shaharshad

Can you please tell us what is your scenario, how do you reach PSTN (FXO, E1/T1 or SIP)?

Thanks,

[shaharshad]

Sorry to get back to you late.

Its a SIP trunk. phone-->call manager--->Gateway router 

[Ruslan Amrahov]

You're welcome anytime

Kindly send us debug ccsip messages from the gateway and Callmanager traces. Also Please confirm number you're dialing and time of the call establishment.

Thanks

[Devansh]

Hi Shaharshad,

1. Is this new set up and did external calls are working earlier?

2. if external calls are working earlier, Please let us know did you make any changes recently.

3.check whether the calls are hitting gateway.

4.Do DNA.

5. if the calls are hitting gateway check for disconnect messaged and who is sending the disconnect message. check for cause code.

6. if you get disconnect message from service provider then there is issue with Telco.

[shaharshad]

hi,

1. It's an existing set up and external calls are still fine except that particular number.

2. No changes are made recent

3. I did debug 

debug ccsip Call 

but couldn't get any debug message when I made call. Kindly correct if I m doing it wrong 

will do thorough debug tomorrow morning.

4. DNA?

thanx

[Devansh]

Use Dialing number analyzer to check whether we are sending the calls to gateway or not.

1. check whether you have any route pattern matching the specific number.

2. if you have the route pattern. check the partition of the route pattern included the CSS of the users.

3. is it happening for one user or for all users.

[shaharshad]

Hi,

1. Route pattern is fine for every other number except this one and its same for every user.

2. Kindly find attached DNA & small chunk of logging pasted.

Kindly let me know if you require further out or any particular debug command output. 

=========================================

pr 22 12:46:06 WST: //38867/1E8AC0800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.227.X.Y:5060;branch=z9hG4bK15674248D
From: "XYZ" <sip:0894266804@10.227.X.Y>;tag=BFCB9B74-2105
To: <sip:0894881300@sipconnectpipn.voip.net.au>
Call-ID: F5AE9BDB-77B11E6-B4D5FBA5-F25B68F2@bwas.voip.net.au
CSeq: 102 INVITE
Timestamp: 1461300366

Apr 22 12:46:07 WST: //38867/1E8AC0800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/UDP 10.227.93.18:5060;branch=z9hG4bK15674248D
From: "XYZ" <sip:0894266804@10.227.X.Y>;tag=BFCB9B74-2105
To: <sip:0894881300@sipconnectpipn.voip.net.au>;tag=1899920614-1461300366988
Call-ID: F5AE9BDB-77B11E6-B4D5FBA5-F25B68F2@bwas.voip.net.au
CSeq: 102 INVITE
Timestamp: 1461300366
Content-Length: 0

Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/cc_api_call_disconnected:
Cause Value=18, Interface=0x3CE1D224, Call Id=38867
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=18, Retry Count=0)
Apr 22 12:46:07 WST: //38866/1E8AC0800000/CCAPI/ccCallReleaseResources:
release reserved xcoding resource.
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/ccCallSetAAA_Accounting:
Accounting=0, Call Id=38867
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/ccCallDisconnect:
Cause Value=18, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=18)
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/ccCallDisconnect:
Start Calling Accounting;
Call Entry(Incoming=FALSE)
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/ccCallDisconnect:
Cause Value=18, Call Entry(Disconnect Cause=18)
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/ccCallDisconnect:

--More--   Call Entry(Disconnect Cause=18)
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/ccCallDisconnect:
Cause Value=18, Call Entry(Responsed=TRUE, Cause Value=18)
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/cc_api_update_interface_cac_resource:
Hwidb=GigabitEthernet0/1, Bandwidth=-80, Call Id=38867
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/cc_api_update_interface_cac_resource:
Total Call Count=8, Voip Call Count=8, MMoip Call Count=0x0, Bandwidth=560
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/cc_decr_if_call_volume:
Remote IP Address=10.222.2.91, Hwidb=GigabitEthernet0/1
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/cc_decr_if_call_volume:
Total Call Count=7, Voip Call Count=7, MMoip Call Count=0
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x3CE1D224, Tag=0x0, Call Id=38867,
Call Entry(Disconnect Cause=18, Voice Class Cause Code=0, Retry Count=0)
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/cc_delete_guid_pod_entry:
Incoming=FALSE
Apr 22 12:46:07 WST: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Apr 22 12:46:07 WST: :cc_free_feature_vsa freeing 22207F00
Apr 22 12:46:07 WST: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Apr 22 12:46:07 WST: vsacount in free is 15
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/cc_delete_call_entry:
Total Call Count=8, Call Entry(Call Count On=FALSE, Incoming Call=FALSE)
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/cc_delete_call_entry:
Deleting profileTable[0x2B60A78C]
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/ccCallGetVoipFlag:
Data Bitmask=0x2, Call Id=38867
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/ccCallGetVoipFlag:
Flag=FALSE
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/ccCallSetVoipFlag:
Flag=FALSE, Data Bitmask=0x2, Call Id=38867
Apr 22 12:46:07 WST: //38867/1E8AC0800000/CCAPI/ccCallSetVoipFlag:
Call Entry(Voip AAA Flags=0x0)
Apr 22 12:46:07 WST: //38867/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:
Call Entry Is Not Found
Apr 22 12:46:07 WST: //38867/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:
Call Entry Is Not Found
Apr 22 12:46:07 WST: //38867/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:
Call Entry Is Not Found
Apr 22 12:46:07 WST: //0/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:
Call Entry Is Not Found
Apr 22 12:46:07 WST: //38867/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:
Call Entry Is Not Found
Apr 22 12:46:07 WST: //38866/1E8AC0800000/CCAPI/ccCallDisconnect:
Cause Value=18, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
Apr 22 12:46:07 WST: //38866/1E8AC0800000/CCAPI/ccCallDisconnect:
Start Calling Accounting;
Call Entry(Incoming=TRUE)
Apr 22 12:46:07 WST: //38866/1E8AC0800000/CCAPI/ccCallDisconnect:
Cause Value=18, Call Entry(Disconnect Cause=0)
Apr 22 12:46:07 WST: //38866/1E8AC0800000/CCAPI/ccCallDisconnect:
Cause Value=18, Call Entry(Responsed=TRUE, Cause Value=18)
Apr 22 12:46:07 WST: //38866/1E8AC0800000/CCAPI/cc_api_update_interface_cac_resource:
Hwidb=GigabitEthernet0/0, Bandwidth=-80, Call Id=38866
--More--  Apr 22 12:46:07 WST: //38866/1E8AC0800000/CCAPI/cc_api_update_interface_cac_resource:
Total Call Count=8, Voip Call Count=8, MMoip Call Count=0x0, Bandwidth=320
Apr 22 12:46:07 WST: //38867/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:
Call Entry Is Not Found
Apr 22 12:46:07 WST: //38867/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:
Call Entry Is Not Found
Apr 22 12:46:07 WST: TCB3B69F82C setting property TCP_TOS (11) 3D192E24
Apr 22 12:46:07 WST: //38866/1E8AC0800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 480 Temporarily Not Available
Via: SIP/2.0/TCP 172.16.8.16:5060;branch=z9hG4bK4bbcddfca861
From: "XYZ" <sip:804@172.X.Y.Z>;tag=610747~562e307b-bd39-a55e-bb24-fb1bcd76d786-43782357
To: <sip:0894881300@10.227.93.18>;tag=BFCB9CC0-14E8
Date: Fri, 22 Apr 2016 04:46:06 GMT
Call-ID: 1e8ac080-7191ac8e-47062-100810ac@172.X.Y.Z
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M6
Reason: Q.850;cause=18
Content-Length: 0

[Manish Gogna · Accepted solution]

Since cucm is receiving SIP/2.0 480 Temporarily Not Available you should check with the Telco. Also, it would help if you can paste the initial invite message that goes out from cucm.

Manish

[Devansh]

Are you able to reach the specific number from you are personal mobile?.

We are getting disconnect message from service provider and issue could be with Telco..

as per cause code 18.

This cause is used when a called party does not respond to a call establishment message with either an alerting or connect indication within the prescribed period of time allocated.

What it means:
The equipment on the other end does not answer the call. Usually this is a misconfiguration on the equipment being called.

[shaharshad]

Hi All,

I have attached output starting from invite.(may be little bit more). We have escalated to our teleco earlier but they said they made call from their sip system and it worked. 

As i mentioned we made calls from other land line and mobile phone and that number works fine. 

service provider said may be it is due to codec mismatch. But you guys are more experience can guide me

[Ruslan Amrahov]

try to use delayed offer so your side will accept whichever codec SP wants

[shaharshad]

Hi,

I don't know Nadeem's answer is automatically considered "Best Answer"? I want to keep this open unless I get a proper solution. 

@Nadeem: we are using SIP trunk and "Show dial-peer voice summary" shows same sip server in "Sess-Target. 

[Ruslan Amrahov]

Manish is ablosutely correct, check with telko, something bad is on their side.

[Nadeem Ahmed · Accepted solution]

There is one thing i would like to understand hows your GW is connected to PSTN? is PRI/FXO/SIP?

I see 480 is being sent from GW to CM but would like to see what message is coming in from PSTN side?

Can you please provide the output of "show dial-peer voice summary"

if your GW is connected to PSTN over PRI then collect isdn debug this time.

Br,

nadeem