SIP<BYE sip

[colec12]

I currently have a sip trunk that ties to a vdn to a Cisco conference bridge. All has worked well in the past but after about 25-30 seconds calls disconnect. I will provide a trace from my extension 7237.

16:21:52 TRACE STARTED 02/24/2016 CM Release String cold-00.1.510.1-19917
16:21:56 active station 7237 cid 0x1e06
16:21:58 dial 4800
16:21:58 term station 4800 cid 0x1e06
16:21:58 0 0 ENTERING TRACE cid 7686
16:21:58 19 1 vdn e4225 bsr appl 0 strategy 1st-found override y
16:21:58 19 1 wait 0 secs hearing silent
16:21:58 19 2 collect 1 digits after annc 20672 for none
16:21:58 19 2 announcement: board 01B03 ann ext: 20672
16:21:58 call-forwarding 4225
16:21:58 term announcement 20672 cid 0x1e06
16:21:58 active announcement 20672 cid 0x1e06
16:21:58 xoip options: faxff modemff ttyff (ann-igc)
xoip ip: [10.2.2.29]:2060
16:21:58 xoip options: faxff modemff ttyff (ann-igc)
xoip ip: [10.2.2.4]:27356
16:21:58 hear annc board 01B03 ext 20672 cid 0x1e06
16:22:06 19 3 goto step 7 if digits = 1
16:22:06 19 3 digits = [1]
16:22:06 19 7 route-to number 3352 cov y if unconditionally
16:22:06 19 8 LEAVING VECTOR PROCESSING cid 7686
16:22:06 call-forwarding 3352
16:22:06 term trunk-group 10 cid 0x1e06
16:22:06 call-forwarding 3352
16:22:06 route-pattern 250 preference 1 location 1/ALL cid 0x1e06
16:22:06 seize trunk-group 10 member 7 cid 0x1e06
16:22:06 Calling Number & Name NO-CPNumber NO-CPName
16:22:06 SIP>INVITE sip:3352@hvcc.edu SIP/2.0
16:22:06 Call-ID: 09e189349dae51ce365128c7bf00
16:22:06 Setup digits 3352
16:22:06 Calling Number & Name *5186297237
16:22:06 SIP<SIP/2.0 100 Trying
16:22:06 Call-ID: 09e189349dae51ce365128c7bf00
16:22:06 Proceed trunk-group 10 member 7 cid 0x1e06
16:22:06 SIP<SIP/2.0 180 Ringing
16:22:06 Call-ID: 09e189349dae51ce365128c7bf00
16:22:06 Alert trunk-group 10 member 7 cid 0x1e06
16:22:06 SIP<SIP/2.0 200 OK
16:22:06 Call-ID: 09e189349dae51ce365128c7bf00
16:22:06 SIP>ACK sip:3352@hvcc.edu;gr=urn:uuid:5fc32f9f-838e-5464-90
16:22:06 SIP>f9-731cfab5b95b SIP/2.0
16:22:06 Call-ID: 09e189349dae51ce365128c7bf00
16:22:06 active trunk-group 10 member 7 cid 0x1e06
16:22:06 G711MU ssff ps:20
rgn:247 [151.103.19.78]:64696
rgn:1 [10.2.2.29]:2050
16:22:06 xoip options: fax:T38 modemT tty:US uid:0x50063
time data
xoip ip: [10.2.2.29]:2050
VOIP data from: [10.2.2.29]:2050
16:22:17 Jitter:1 0 0 0 0 0 0 0 0 0: Buff:16 WC:8 Avg:1
16:22:17 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
VOIP data from: [10.2.2.29]:2050
16:22:26 Jitter:1 1 1 1 1 1 1 1 0 0: Buff:30 WC:8 Avg:0
16:22:26 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
VOIP data from: [10.2.2.29]:2050
16:22:36 Jitter:1 1 1 1 1 1 1 1 1 1: Buff:14 WC:9 Avg:0
16:22:36 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
VOIP data from: [10.2.2.29]:2050
16:22:46 Jitter:1 1 1 1 1 1 1 1 1 1: Buff:14 WC:9 Avg:0
16:22:46 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
VOIP data from: [10.2.2.29]:2050
16:22:56 Jitter:1 1 1 1 1 1 1 1 1 1: Buff:14 WC:9 Avg:0
16:22:56 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
16:22:58 SIP<BYE sip:+5186297237@10.2.2.10:5061;transport=tls SIP/2.
16:22:58 SIP<0
16:22:58 Call-ID: 09e189349dae51ce365128c7bf00
16:22:58 SIP>SIP/2.0 200 OK
16:22:58 Call-ID: 09e189349dae51ce365128c7bf00
16:22:58 idle trunk-group 10 member 7 cid 0x1e06
16:27:43 TRACE COMPLETE station 7237 cid 0x0

 

[ZeroZeroOne]

I have found that Cisco does not play will with Avaya using TLS.

Create another trunk and signaling group that only uses TCP - or change the existing trunk group 10.

 

[colec12]

I changed the signaling group to TCP and port to 5061 and received a busy signal. I am assuming changes need to also be made in Session Manager?

Thanks

 

[kyle555]

Check your session refresh timers in your CM trunk and in the Cisco trunk

 

[ZeroZeroOne]

Yes - change from TLS to TCP all the way through, including the SIP Entity Links in Session Manager.

My Cisco phones would time out at 32 seconds for every single call until I switched to TCP. Maybe it's the same in your case.

 

[colec12]

I really did it know. I made a change on the signaling group station form (far-end Node Name)and then put it back to what it was and know when I do a status on my singnaling groups (10 and 11) I get far-end bypass. Also when I do a status on my trunks (10 and 11) I get OOS/FE-idle. I have tried busyout and release with no luck.

 

[ZeroZeroOne]

In CM:

Check Signaling Groups that go to Session Manager. If the Transport Type is TLS then make sure the Ports are 5061. If TCP, then port 5060.

display signaling-group 97                                      Page   1 of   2
                                SIGNALING GROUP

 Group Number: 97             Group Type: sip
  IMS Enabled? n        Transport Method: [highlight #FCE94F]tls[/highlight]
        Q-SIP? n
     IP Video? n                                   Enforce SIPS URI for SRTP? y
  Peer Detection Enabled? y  Peer Server: SM
 Prepend '+' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers? y
Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers? n
Alert Incoming SIP Crisis Calls? n
   Near-end Node Name: procr                 Far-end Node Name: ASM
 Near-end Listen Port: [highlight #FCE94F]5061[/highlight]                Far-end Listen Port: [highlight #FCE94F]5061[/highlight]
                                        Far-end Network Region: 2

Far-end Domain: company.com
                                             Bypass If IP Threshold Exceeded? n
Incoming Dialog Loopbacks: eliminate                  RFC 3389 Comfort Noise? n
         DTMF over IP: rtp-payload            Direct IP-IP Audio Connections? n
Session Establishment Timer(min): 3                     IP Audio Hairpinning? n
         Enable Layer 3 Test? y
                                                  Alternate Route Timer(sec): 6

In System Manager:

Go to Home / Elements / Routing / Entity Links.

Confirm the Entity Links for CM match what is in the Signaling Groups, TLS / 5061 or TCP / 5060

 

[colec12]

I checked all of that and it seems fine. The signaling and trunks groups will come into service for a couple of seconds after I busy/ release them.
Definitely out of sync, not sure how to fix at this point. Everything is back to original at this point.