cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
4672
Views
14
Helpful
4
Replies

CUBE do not send Invite message error 503 service SIP/2.0 503 Service Unavailable

memo.perlop
Level 1
Level 1

Hi Guys,

I hope everything goes fine.


We have a issues with on cube, the thing is that this cube do not send the invite message to an SBC, but the strange thing is that this same cube can send invites to another CUCM and the call can complete without problem.

the call flow is CUCM->Cube->SBC (PSTN)

 

CUCM ip add 10.10.12.100

Cube ip add 10.10.10.162

SBC ip add 10.10.10.170

the confing of the cube is:

sh run
Building configuration...


Current configuration : 3786 bytes
!
! Last configuration change at 11:43:03 CDT Mon May 25 2015 by ingres
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname GW3925_SARA_PSTN_EMPRESARIAL
!
boot-start-marker
boot system flash0:c3900-universalk9-mz.SPA.151-4.M7.bin
boot-end-marker
!
!
! card type command needed for slot/vwic-slot 0/0
! card type command needed for slot/vwic-slot 0/1
enable secret 4 pBqOHsbcfZZOVKUsUt96/h1qcaVAWknA8Az8VuTXpp6
enable password cisco
!
no aaa new-model
clock timezone cst -6 0
clock summer-time CDT recurring 1 Sun Apr 3:00 last Sun Oct 2:00
!
no ipv6 cef
ip source-route
ip cef
!
!
!
!
!
multilink bundle-name authenticated
!
!
!
!
!
crypto pki token default removal timeout 0
!
!
voice-card 0
 dspfarm
 dsp services dspfarm
!
!
!
voice service voip
 ip address trusted list
  ipv4 10.10.10.170
 no ip address trusted authenticate
 mode border-element
 allow-connections sip to sip
 no supplementary-service sip handle-replaces
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
  bind control source-interface Loopback0
  bind media source-interface Loopback0
  early-offer forced
  midcall-signaling passthru
!
voice class codec 2
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
!
!
!
!
voice translation-rule 1
 rule 1 /^9\(........$\)/ /55\1/
 rule 2 /^9\(.*\)/ /\1/
!
voice translation-rule 2
 rule 1 /.*/ /5563640003/
!
!
voice translation-profile PSTN-SBC
 translate calling 2
 translate called 1
!
!
license udi pid C3900-SPE100/K9 sn FOC18402TY0
hw-module pvdm 0/0
!
hw-module pvdm 0/1
!
!
!
username ingres privilege 15 secret 4 pBqOHsbcfZZOVKUsUt96/h1qcaVAWknA8Az8VuTXpp6
!
redundancy
!
!
!
!
!
!
interface Loopback0
 ip address 10.10.10.162 255.255.255.0
!
interface Embedded-Service-Engine0/0
 no ip address
 shutdown
!
interface GigabitEthernet0/0
 ip address 10.11.11.10 255.255.255.240
 duplex auto
 speed auto
!
interface GigabitEthernet0/1
 ip address 10.11.11.24 255.255.255.240
 duplex auto
 speed auto
!
interface GigabitEthernet0/2
 no ip address
 shutdown
 duplex auto
 speed auto
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip route 0.0.0.0 0.0.0.0 172.20.34.1
!
!
nls resp-timeout 1
cpd cr-id 1
!
!
control-plane
!
!
!
!
mgcp profile default
!
sccp ccm 10.10.12.100 priority 1 version 7.0
sccp ccm 10.10.12.101 priority 2 version 7.0
!
dial-peer voice 1 voip
 session protocol sipv2
 incoming called-number .
 dtmf-relay rtp-nte
 codec transparent
 no vad
!
dial-peer voice 9000 voip
 translation-profile outgoing PSTN-SBC
 destination-pattern 90445532227987
 session protocol sipv2
 session target ipv4:10.10.10.170:5060
 session transport tcp
 voice-class codec 2  
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 20000 voip
 destination-pattern 4....$
 session protocol sipv2
 session target ipv4:10.10.12.22
 dtmf-relay rtp-nte
 codec transparent
 no vad
!
dial-peer voice 20001 voip
 preference 10
 destination-pattern 4....$
 session protocol sipv2
 session target ipv4:10.10.12.24
 dtmf-relay rtp-nte
 codec transparent
 no vad
!
dial-peer voice 9001 voip
 session protocol sipv2
 session transport tcp
 incoming called-number 90445532227987
 voice-class codec 2  
 dtmf-relay rtp-nte
 no vad
!
!
gateway
 timer receive-rtp 1200
!
sip-ua
 no remote-party-id
 retry invite 2
!
!
!
gatekeeper
 shutdown
!
!
!

And the Debug CCSip Mess

May 25 16:43:15.566: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:90445532227999@10.10.10.162:5060 SIP/2.0
Via: SIP/2.0/TCP 10.10.12.100:5060;branch=z9hG4bK42712fadf37b
From: "Memo Ip Com" <sip:2015@10.10.12.100>;tag=17009~309b2431-1c35-40b1-bfc5-b389a2ebafd2-20245215
To: <sip:90445532227999@10.10.10.162>
Date: Mon, 25 May 2015 16:48:56 GMT
Call-ID: ed84d180-56315278-4261-640c10ac@10.10.12.100
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  3600
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Cisco-Guid: 3984904576-0000065536-0000000108-1678512300
Session-Expires:  3600
P-Asserted-Identity: "Memo Ip Com" <sip:2015@10.10.12.100>
Remote-Party-ID: "Memo Ip Com" <sip:2015@10.10.12.100>;party=calling;screen=yes;privacy=off
Contact: <sip:2015@10.10.12.100:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 203

v=0
o=CiscoSystemsCCM-SIP 17009 1 IN IP4 10.10.12.100
s=SIP Call
c=IN IP4 10.10.12.100
t=0 0
m=audio 24644 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

May 25 16:43:15.570: //2357/ED84D1800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.12.100:5060;branch=z9hG4bK42712fadf37b
From: "Memo Ip Com" <sip:2015@10.10.12.100>;tag=17009~309b2431-1c35-40b1-bfc5-b389a2ebafd2-20245215
To: <sip:90445532227999@10.10.10.162>
Date: Mon, 25 May 2015 16:43:15 GMT
Call-ID: ed84d180-56315278-4261-640c10ac@10.10.12.100
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


May 25 16:43:20.570: //2357/ED84D1800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/TCP 10.10.12.100:5060;branch=z9hG4bK42712fadf37b
From: "Memo Ip Com" <sip:2015@10.10.12.100>;tag=17009~309b2431-1c35-40b1-bfc5-b389a2ebafd2-20245215
To: <sip:90445532227999@10.10.10.162>;tag=87C0320-1462
Date: Mon, 25 May 2015 16:43:15 GMT
Call-ID: ed84d180-56315278-4261-640c10ac@10.10.12.100
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=38
Content-Length: 0


May 25 16:43:20.570: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:90445532227999@10.10.10.162:5060 SIP/2.0
Via: SIP/2.0/TCP 10.10.12.100:5060;branch=z9hG4bK42712fadf37b
From: "Memo Ip Com" <sip:2015@10.10.12.100>;tag=17009~309b2431-1c35-40b1-bfc5-b389a2ebafd2-20245215
To: <sip:90445532227999@10.10.10.162>;tag=87C0320-1462
Date: Mon, 25 May 2015 16:48:56 GMT
Call-ID: ed84d180-56315278-4261-640c10ac@10.10.12.100
User-Agent: Cisco-CUCM10.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0

 

any idea?

 

Thanks in advance

 

Memo

 

 

 

 

 

1 Accepted Solution

Accepted Solutions

Hello Memo,

CUBE is unable to establish TCP session with address 10.10.10.170

Below is the error message capture in the ccsip errror.

May 25 20:44:29.579: //145/9F80C5800000/SIP/Error/sipTransportPostSendFailure: Posting send failure msg
May 25 20:44:29.579: //-1/xxxxxxxxxxxx/SIP/Error/act_idle_send_msg_failure: Send Error to 10.10.10.170:5060 for transport TCP

While checking the config CUBE is selecting Outgoing Dial-peer 9000

May 25 20:44:24.579: //144/9F80C5800000/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=9000

dial-peer voice 9000 voip
 translation-profile outgoing PSTN-SBC
 destination-pattern 90445532227987
 session protocol sipv2
 session target ipv4:10.10.10.170:5060
 session transport tcp=====================This dial-peer TCP enforced to make a connection with your service provider where it causing an issue.
 voice-class codec 2  
 dtmf-relay rtp-nte
 no vad
 
CUBE is disconnecting the call with 503 service unavailable as it cannot establish TCP session with Provider and thus unable route and construct the outgoing SIP invite to terminating side as TCP connection is not getting established

Can you please try removing "session transport tcp" from dial-peer 9000 and make one test call. if its still fails try taking debug ip tcp and udp packets. but i am pretty it will work.

 

 

Regards

Nadeem Ahmed

 

Br, Nadeem Please rate all useful post.

View solution in original post

4 Replies 4

Nadeem Ahmed
Cisco Employee
Cisco Employee

Hello Memo,

 

Can  you please collect another following debugs

 

1) debug ccsip message

2) debug ccsip events

3) debug ccsip error

4) debug voice ccapi inout

 

Looks to me CUBE is unable to construct the INVITE back to SBC due to fact of having connectivity issue with CUBE and SBC over or TCP /UDP.

 

Can you run a test call and we can conclude anything out of it.

 

Br,

Nadeem

Br, Nadeem Please rate all useful post.

Hi Nadeem,

 

Thanks for your time,

 

Here are the debugs you ask.

 

thanks in advance

 

Memo

Hello Memo,

CUBE is unable to establish TCP session with address 10.10.10.170

Below is the error message capture in the ccsip errror.

May 25 20:44:29.579: //145/9F80C5800000/SIP/Error/sipTransportPostSendFailure: Posting send failure msg
May 25 20:44:29.579: //-1/xxxxxxxxxxxx/SIP/Error/act_idle_send_msg_failure: Send Error to 10.10.10.170:5060 for transport TCP

While checking the config CUBE is selecting Outgoing Dial-peer 9000

May 25 20:44:24.579: //144/9F80C5800000/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=9000

dial-peer voice 9000 voip
 translation-profile outgoing PSTN-SBC
 destination-pattern 90445532227987
 session protocol sipv2
 session target ipv4:10.10.10.170:5060
 session transport tcp=====================This dial-peer TCP enforced to make a connection with your service provider where it causing an issue.
 voice-class codec 2  
 dtmf-relay rtp-nte
 no vad
 
CUBE is disconnecting the call with 503 service unavailable as it cannot establish TCP session with Provider and thus unable route and construct the outgoing SIP invite to terminating side as TCP connection is not getting established

Can you please try removing "session transport tcp" from dial-peer 9000 and make one test call. if its still fails try taking debug ip tcp and udp packets. but i am pretty it will work.

 

 

Regards

Nadeem Ahmed

 

Br, Nadeem Please rate all useful post.

Hi Nadeem,

 

I hope you are doing good,

 

Thanks for your time,

 

Yesterday I worked with the guys of the SBC and they confirm me that the communication need to be TCP but apparently the issue is with the port, is not 5060 they are using 6081, the weird thing is that I change the dial-peer to use the port 6081 but the issue is the same.

 

But I will continue working with these and as soon as  I have a useful answer from the  guys of the SBC I will post it.

 

thanks for your help