SIP Provider outbound calls fail until an inbound call comes in? Possible Timer issue?

[Cameron McKay]

We have a SIP trunk setup to MTS Allstream in Canada. Outbound calls will fail /w a fast busy unless an inbound call is made to a DID. After the first inbound call is made outbound calling will work for a while and then go back to fast busy. It seems that if the lines are idle then we run into this problem.

Configs attached... any suggestions? I have never encountered this type of issue before. CUCM 10.5. Router: C2921 ISR

[Zekeria Sammantar]

Hi Cameron,

Upon checking the debug_ccsip.txt, it can be seen that the CUBE is send a INVITE with SDP but no response, it tries again but fails so thats why you are receiving an fast busy.

*Feb  8 04:15:35.334: //3362/B6A2CF800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:1************@216.13.193.132:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.13:5060;branch=z9hG4bK84412D0
Remote-Party-ID: "Eagle" <sip:************@10.10.100.13>;party=calling;screen=yes;privacy=off
From: "Eagle" <sip:************@10.10.100.13>;tag=89DBF54-1C7E
To: <sip:1************@216.13.193.132>

After 2 tries it sends 408 Request to CUCM hence the fast busy on the phone.

*Feb  8 04:15:36.834: //3361/B6A2CF800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 10.10.100.16:5060;branch=z9hG4bK112d135bfa3a8
From: "Eagle" <sip:************@10.10.100.16>;tag=529172~7e7c7fcf-512e-9fd5-6119-4a0b9c30ffc5-20682594
To: <sip:1************@10.10.100.13>;tag=89DC530-16E0
Date: Sun, 08 Feb 2015 04:15:35 GMT
Call-ID: b6a2cf80-4d61e3e4-b9b1-10640a0a@10.10.100.16
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.3.3.M4
Reason: Q.850;cause=102
Content-Length: 0

As for the outbound_call_ok.txt, the same INVITE with SDP is sent but a meet with a response.

*Feb  8 04:17:06.462: //3367/ECE04F000000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:**************@216.13.193.132:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.13:5060;branch=z9hG4bK846233E
Remote-Party-ID: "Eagle" <sip:**************@10.10.100.13>;party=calling;screen=yes;privacy=off
From: "Eagle" <sip:**************@10.10.100.13>;tag=89F2348-16A3
To: <sip:**************@216.13.193.132>
Date: Sun, 08 Feb 2015 04:17:06 GMT
Call-ID: 2D27AF99-AE8011E4-954E96F3-A1A5DA2C@10.10.100.13
Cisco-Guid: 3974123264-0000065536-0000001142-0274991626

Response from ITSP.

*Feb  8 04:17:06.478: //3367/ECE04F000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.100.13:5060;branch=z9hG4bK846233E
From: "Eagle" <sip:**************@10.10.100.13>;tag=89F2348-16A3
To: <sip:**************@216.13.193.132>
Call-ID: 2D27AF99-AE8011E4-954E96F3-A1A5DA2C@10.10.100.13
CSeq: 101 INVITE
Timestamp: 1423369026
Content-Length: 0

I suggest you contact your ITSP for further information, since its unusual for and ITSP not responded to any message 

Also check this out:

MTS Allstream IP Trunking Service: Connecting Cisco Unified Communications Manager 8.5 via the Cisco Unified Border Element Release 8.6 using SIP.

Please Rate.

[Zekeria Sammantar · Accepted solution]

On your outgoing dial-peer to ITSP, I can see that you are removing the Min-SE header and Session-Expires header, why?

voice class sip-profiles 15
 request INVITE sip-header Supported remove
 request INVITE sip-header Min-SE remove
 request INVITE sip-header Session-Expires remove
 request INVITE sip-header Unsupported modify "Unsupported:" "timer"




MIN-SE
  When the header field is not present, its default value for is 90
   seconds.from RFC 4028

Remove the sip-profile from the dial-peer.

Voice-gateway and re-adjust your MIN-SE and max to value equal or less than 600.

min-se 600 session-expires 600

MIN-SE
  When the header field is not present, its default value for is 90
   seconds.

I can see that your ITSP is sending INVITES with Minimum Session Expiration Timer of (600).

Received:
INVITE sip:************@10.10.100.13;user=phone SIP/2.0
Max-Forwards: 69
Session-Expires: 86400;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:************@10.10.100.13;user=phone>
From: "CM" <sip:*************@216.13.193.132>;tag=-45026-157bed3-63edf97b-157bed3

Its removed by the incoming dial-peer from ITSP.

INVITE sip:************@10.10.100.16:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.13:5060;branch=z9hG4bK8451EEC
Remote-Party-ID: "CM" <sip:*************@10.10.100.13>;party=calling;screen=no;privacy=off
From: "CM" <sip:*************@10.10.100.13>;tag=89E8170-1D7A

[Cameron McKay]

The provider requests that we strip the min-SE and session-timer headers before it hits their network.

I've made the changes, but still get fast busy on outbound until the first inbound call. It's pretty quick.. about a minute passes and then I can't place the call out.

I will reach out to the provider.