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SIP Trunk Reason: Q.850;cause=102;text="recovery of timer expiry"

rasheedccvp
Level 1
Level 1

‎10-08-2012 08:59 AM - edited ‎03-16-2019 01:34 PM

                   Hello,  Need a help

i am getting below error

 

SIP Trunk Reason: Q.850;cause=102;text="recovery of timer expiry"

 

 

I configured the SIP trunk and the below is the Call Flow

 

IP Phone ----->>>>.CUCM------(SIP Trunk)----CUBE(2921)--------SIP server(ISP)

 

i can receive incoming call ringing on the IP phone andonce answer there is no audio or ring backtone.

Outgoing calls not allways provides busytone

 

 

 

interface GigabitEthernet0/0

description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$

ip address 10.1.1.x 255.255.255.0

duplex auto

speed auto

h323-gateway voip interface

h323-gateway voip bind srcaddr 10.1.1.x!

interface GigabitEthernet0/1

no ip address

shutdown

duplex auto

speed auto

!

interface GigabitEthernet0/2

description  SIP Interface

no ip address

duplex auto

speed auto

!

interface GigabitEthernet0/2.245

encapsulation dot1Q 245

ip address 10.x.x.254 255.255.255.252

 

 

 

dial-peer voice 51 voip

description Outgoing From STC Server to CUCM SIP TRUNK

translation-profile outgoing PSTN-01

preference 1

destination-pattern 2508...

rtp payload-type cisco-codec-fax-ack 111

rtp payload-type nte 97

session protocol sipv2

session target ipv4:10.x.x.19 (Call Manager)

session transport udp

incoming called-number .

dtmf-relay rtp-nte digit-drop

codec g711alaw

no vad

dial-peer voice 301 voip

description Outgoing for Local Calls

translation-profile outgoing Strip9

destination-pattern 9T

rtp payload-type cisco-codec-fax-ack 111

rtp payload-type nte 97

session protocol sipv2

session target ipv4:10.x.x.157

session transport udp

voice-class sip profiles 2

codec g711alaw

no vad

 

 

 

Message was edited by: Rasheed C.V. Ping test  fails from STC interface to the CallManager  ping 10.x.x.19(CM IP address) source GigabitEthernet0/2.245 (SIP interface) Is it normal behaviour, even i can not ping phone  ip from stc interface

5 people had this problem
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0 Helpful
12 Replies 12

Suresh Subramanian
Level 7
Level 7

‎10-09-2012 12:32 AM

Hi Rasheed,

The CUCM should be pingable if ICMP is not blocked by firewall. I checked the traces and saw that the CUBE is forwarding the INVITE message to CUCM but CUCM is not at all responding with '100 Tring' message. Please crosscheck the SIP trunk configuration in CUCM also.

Thanks

Suresh

//Suresh Please rate all the useful posts.

Ayodeji Okanlawon
VIP Alumni
VIP Alumni

‎10-09-2012 04:40 AM

Rahseed,

From your trace, I can see that your provider is not responding to your invite.  CUBE sent multiple invite but doesnt get any response back from your ITSP. You need to check with them or check that you have the right config to your ITSP

095487: Oct  1 15:57:57.435: //40199/99568259ACC9/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:2508001@10.x.x.19:5060 SIP/2.0
Via: SIP/2.0/UDP 10.x.x.254:5060;branch=z9hG4bK2E72493
Remote-Party-ID: <90509142839>;party=calling;screen=no;privacy=off
From: <90509142839>;tag=14EDAC7C-2679
To: <2508001>
Date: Mon, 01 Oct 2012 15:57:57 GMT
Call-ID: 9957BAA9-B1711E2-ACCFDC16-306C038F@10.x.x.254
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2572583513-0186061282-2898910230-0812385167
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1349107077
Contact: <90509142839>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 301

v=0
o=CiscoSystemsSIP-GW-UserAgent 1389 8563 IN IP4 10.x.x.254
s=SIP Call
         
c=IN IP4 10.x.x.254
t=0 0
m=audio 23240 RTP/AVP 8 0 18 97
c=IN IP4 10.x.x.254
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15

095488: Oct  1 15:57:59.435: //40199/99568259ACC9/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:2508001@10.x.x.19:5060 SIP/2.0
Via: SIP/2.0/UDP 10.x.x.254:5060;branch=z9hG4bK2E72493
Remote-Party-ID: <90509142839>;party=calling;screen=no;privacy=off
From: <90509142839>;tag=14EDAC7C-2679
To: <2508001>
Date: Mon, 01 Oct 2012 15:57:59 GMT
Call-ID: 9957BAA9-B1711E2-ACCFDC16-306C038F@10.x.x.254
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2572583513-0186061282-2898910230-0812385167
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1349107079
Contact: <90509142839>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 301

v=0
o=CiscoSystemsSIP-GW-UserAgent 1389 8563 IN IP4 10.x.x.254
s=SIP Call
c=IN IP4 10.x.x.254
t=0 0
m=audio 23240 RTP/AVP 8 0 18 97
c=IN IP4 10.x.x.254
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15

095489: Oct  1 15:58:03.435: //40199/99568259ACC9/SIP/Msg/ccsipDisplayMsg:
Sent:    
INVITE sip:2508001@10.x.x.19:5060 SIP/2.0
Via: SIP/2.0/UDP 10.x.x.254:5060;branch=z9hG4bK2E72493
Remote-Party-ID: <90509142839>;party=calling;screen=no;privacy=off
From: <90509142839>;tag=14EDAC7C-2679
To: <2508001>
Date: Mon, 01 Oct 2012 15:58:03 GMT
Call-ID: 9957BAA9-B1711E2-ACCFDC16-306C038F@10.x.x.254
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2572583513-0186061282-2898910230-0812385167
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1349107083
Contact: <90509142839>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 301

v=0
o=CiscoSystemsSIP-GW-UserAgent 1389 8563 IN IP4 10.x.x.254
s=SIP Call
c=IN IP4 10.x.x.254
t=0 0
m=audio 23240 RTP/AVP 8 0 18 97
c=IN IP4 10.x.x.254
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15

Please rate all useful posts

"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

Please rate all useful posts

benoit.petry
Level 1
Level 1

‎10-09-2012 08:30 AM

Hi all,

Isn't the problem with early offer in SIP invite ? I know it's an identified problem

Best regards

Benoît PETRY

Sent from Cisco Technical Support iPhone App

0 Helpful

Ayodeji Okanlawon
VIP Alumni
VIP Alumni
In response to benoit.petry

‎10-09-2012 10:30 AM

Early offer in SIP invite is perfectly normal. There is no problem with it. What problem do you know with it?

Please rate all useful posts

"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

Please rate all useful posts

asafayan
Level 4
Level 4

‎06-12-2013 06:56 PM

What was the resolution ?

Amir

Jorge Armijo
Level 4
Level 4
In response to asafayan

‎06-12-2013 07:34 PM

Amir,

This post is kind of old, if you have the same issue, you may open a new discussion. However, based on the trace, most likely was a ITSP problem. They were not replying to the SIP INVITE sent by the CUBE.

HTH

--
Jorge Armijo

Please remember to rate helpful responses and identify helpful or correct answers.

-- Jorge Armijo Please remember to rate helpful responses and identify helpful or correct answers.

asafayan
Level 4
Level 4
In response to Jorge Armijo

‎06-13-2013 10:15 AM

Hi Jorge,

Thanks for the reply.  I have a very strange situation where I have the following symptom:

1.  7925 <-> CUCM <-> CUBE <-> TSP <-> PSTN <-> Called Party  ----  FAILS

2.  7937 <-> CUCM <-> CUBE <-> TSP <-> PSTN <-> Called Party  ----  FAILS

3.  7945 <-> CUCM <-> CUBE <-> TSP <-> PSTN <-> Called Party  ----  SUCCEEDS

The above results are consistent and there is absolutely no difference in the configuration of the various calling phones. All are SCCP, all have the same DN and DEVICE level CSSs, DPs, MRGLs etc.

Very strange.

TIA,

Amir

Ayodeji Okanlawon
VIP Alumni
VIP Alumni
In response to asafayan

‎06-13-2013 11:24 AM

I suggest you open a new thread...Please send us debug ccsip messages from a failed call

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts
0 Helpful

asafayan
Level 4
Level 4
In response to Ayodeji Okanlawon

‎06-13-2013 03:53 PM

Hi Aokanlawon,

Will do.

TIA,

Amir

asafayan
Level 4
Level 4
In response to Ayodeji Okanlawon

‎06-13-2013 04:32 PM

Hi Aokanlawon,

Here is the new thread.  I know it will be easy for you to resolve! : )

https://supportforums.cisco.com/thread/2222892

TIA,

Amir

AlejandroMongeNieves64269
Level 1
Level 1
In response to asafayan

‎02-25-2021 02:11 PM

hello, What was the resolution ? i have the same issue some calls are not complete 

 

regards!

0 Helpful

TONY SMITH
Spotlight
Spotlight
In response to AlejandroMongeNieves64269

‎03-08-2021 08:27 AM

Post up your debugs from the CUBE "debug ccsip mess" to start with.  It's highly unlikely that your problem is the same as the OPs.

0 Helpful
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