06-23-2013 04:08 PM - last edited on 03-25-2019 08:23 PM by ciscomoderator
HI,
I have a Cube Router configured
CUCM <-> h323 <-> Router <-> SIP <-> Telco <-> PSTN
Where when initiating an outbound call all I get is a busy tone.
However if I initiate an inbound call, which works fine.
I can then make outbound calls which work fine.
If there are no calls made for a period of time, I go back to the orginial problem.
Has any one seen a simialr probelm or have an idea wher the issue may lay !!
Many thanks
06-23-2013 05:41 PM
please add config, debug ccsip when making a call (we are not psychics)
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06-23-2013 07:06 PM
Recreate the problem
- run Debugs on CUBE (debug ccsip messages, debug voip ccapi inout)
- mention calling and called party numbers
- IP address of Phone; CUCM and running conf of CUBE if possible)
Tapan
06-23-2013 09:30 PM
Along with the ccsip and ccapi inout debugs, please collect the below debugs for H.323 call leg.
debug h225 q931
debug h225 asn1
debug h245 asn1
Also make sure the inbound and outbound dial-peers are correctly configured.if the configuration looks fine, then with the error code captured from the debugs would help us what is wrong and where.
06-24-2013 04:58 AM
06-24-2013 05:12 AM
06-24-2013 05:58 AM
Hi Ian,
It looks like the Telco is not responding to the SIP Invite we sent and hence the call is getting disconnected with
Cause Value=102 - CC_CAUSE_RECOVERY_ON_ TIMER_EXPIRY - Invite expires timer reached maximum number of retries allowed.
There are 3 Invite messages sent to telco but no response from them. Please have a word with them.
06-24-2013 06:24 AM
Hi,
The intresting point is though, it would appear the Router does not send the invite, itself.
As I have watched traffic out my FW and seen nothing, when calls are made
Until I have done the inbound call, when traffic is then sent both ways !
This is the point I have got to !
Regards.
06-24-2013 06:41 AM
Why is traffic Blocked in direction towards TELCO on the SIP call leg. If you can fix that Calls will work.
Router is processing the call properly.
Tapan
06-24-2013 06:46 AM
74158 105585: *Jun 24 12:05:51 UTC: //401/80B5408F7701/SIP/Msg/ccsipDisplayMsg:
74366 105586: Sent:
74586 105587: INVITE sip:1112918888402822@Y.Y.Y.Y:5060 SIP/2.0#015
74881 105588: Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK13E34A#015
75100 105589: Remote-Party-ID: <4165727915>;party=calling;screen=yes;privacy=off#0154165727915>
75352 105590: From: <4165727915>;tag=FAA2920-17B4#0154165727915>
75549 105591: To: <1112918888402822>#0151112918888402822>
75991 105592: Date: Mon, 24 Jun 2013 12:05:51 GMT#015
76237 105593: Call-ID: 3FCFF528-DBFD11E2-8309AB10-9523C09@X.X.X.X#015
76461 105594: Supported: 100rel,timer,resource-priority,replaces,sdp-anat#015
76683 105595: Min-SE: 1800#015
76899 105596: User-Agent: Cisco-SIPGateway/I
77114 105597: OS-15.3.1.T1#015
77344 105598: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER#015
77538 105599: CSeq: 101 INVITE#015
77779 105600: Max-Forwards: 70#015
78149 105601: Timestamp: 1372075551#015
78372 105602: Contact: <4165727915>#0154165727915>
78557 105603: Call-Info:
78787 105604: Expires: 180#015
79089 105605: Allow-Events: telephone-event#015
80845 105606: Content-Length: 0#015
81048 105607: #015
>> This is the invite sent out from CUBE to Telco. better to crosscheck the FW and make sure the SIP Invite is reaching the Telco.
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