cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
26669
Views
9
Helpful
3
Replies

RE-INVITE responding with 488 Not Acceptable Media with IP-IP Gateway

John Platts
Level 4
Level 4

I have been troubleshooting a problem with a IP to IP gateway, which is running IOS 12.4(20)T4.

Here is the relevant configuration (not with the real phone numbers or the real ITSP settings):

voice service voip
media flow-around
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
supplementary-service media-renegotiate
sip
  asserted-id pai
  rel1xx disable
  header-passing error-passthru
  registrar server
  midcall-signaling passthru
  sip-profiles 101
!
!
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
!
!
!
!
!
voice class sip-profiles 100
request INVITE sip-header P-Asserted-Identity modify "<sip:011(.*)" "<sip:+\1"
request INVITE sip-header Diversion modify "<sip:011(.*)" "<sip:+\1"
request INVITE sip-header P-Asserted-Identity modify "<sip:\+?(.*)" "<sip:+\1"
request INVITE sip-header Diversion modify "<sip:\+?(.*)" "<sip:+\1"
request REINVITE sip-header P-Asserted-Identity modify "<sip:011(.*)" "<sip:+\1"
request REINVITE sip-header Diversion modify "<sip:011(.*)" "<sip:+\1"
request REINVITE sip-header P-Asserted-Identity modify "<sip:\+?(.*)" "<sip:+\1"
request REINVITE sip-header Diversion modify "<sip:\+?(.*)" "<sip:+\1"
request INVITE sip-header Allow-Header modify " UPDATE, " " "
request REINVITE sip-header Allow-Header modify " UPDATE, " " "
response 180 sip-header Allow-Header modify " UPDATE, " " "
response 200 sip-header Allow-Header modify " UPDATE, " " "
!
voice class sip-profiles 101
request INVITE sip-header Allow-Header modify " UPDATE, " " "
request REINVITE sip-header Allow-Header modify " UPDATE, " " "
response 180 sip-header Allow-Header modify " UPDATE, " " "
response 200 sip-header Allow-Header modify " UPDATE, " " "
!
!
!
voice register global
mode cme
source-address 200.70.220.243 port 5060
max-dn 500
max-pool 185
authenticate register
authenticate realm ipipgw.example.com
!
voice register dn 1
number 14695552000
no-reg
!
voice register dn 2
number 14695552001
no-reg
!
voice register dn 3
number 14695552002
no-reg
!
voice register dn 4
number 14695554000
no-reg
!
voice register dn 5
number 14695554001
no-reg
!
voice register dn 6
number 14695554002
no-reg
!
voice register pool  1
id mac 001C.5555.3201
number 1 dn 1
number 2 dn 2
number 3 dn 3
dtmf-relay rtp-nte
voice-class codec 1
username 14695552000 password site1
no vad
!
voice register pool  2
id mac 001C.5555.9023
number 1 dn 4
number 2 dn 5
number 3 dn 6
dtmf-relay rtp-nte
voice-class codec 1
username 14695554000 password site2
no vad
!
!
voice translation-rule 1
rule 1 /^\+1/ /1/
rule 2 /^\+\([2-9]\)/ /011\1/
rule 3 /^\([2-9][0-9][0-9][2-9][0-9][0-9][0-9][0-9][0-9][0-9]\)$/ /1\1/
!
voice translation-rule 2
rule 1 /^011\([2-9]\)/ /+\1/
!
voice translation-profile Incoming_PSTN_Call
translate calling 1
translate called 1
translate redirect-target 1
translate redirect-called 1
!
voice translation-profile Outgoing_PSTN_Call
translate calling 2
!
dial-peer voice 1000 voip
translation-profile incoming Incoming_PSTN_Call
voice-class codec 1
session protocol sipv2
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 65 voip
translation-profile outgoing Outgoing_PSTN_Call
destination-pattern .%
voice-class codec 1
voice-class sip rel1xx disable
voice-class sip profiles 100
session protocol sipv2
session target dns:sipgateway.exampleitsp.com
session transport udp
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
!
!
!
sip-ua
no remote-party-id
!
!
!
gatekeeper
shutdown
!

I am able to make phone calls with the above configuration. I can also successfully forward calls with the above configuration. Not all calls fail to transfer properly. But on some phone calls, a 488 Not Acceptable Error response is returned instead of sending the re-INVITE to the ITSP gateway.

Here is what I really want to accomplish:

  • Media flow-around for calls
  • Re-INVITE messages sent to ITSP gateway

I had made a few changes, and I will try to see if rebooting it will solve the problem. Are there other settings that I should apply, or should I upgrade to a newer version of IOS?

3 Replies 3

John Platts
Level 4
Level 4

I actually executed the debug ccsip messages command on the IP-IP gateway. Here is the output of the re-INVITE, and the failed response:

Mar  8 23:53:20.691 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:19725559000@168.90.100.243:5060 SIP/2.0
Via: SIP/2.0/UDP 76.250.30.1:5060;branch=z9hG4bK79C25ED
From: <14695554002>;tag=5B031CC-1164
To: <19725559000>;tag=2ACB58-1E1C
Date: Tue, 09 Mar 2010 05:43:33 GMT
Call-ID: CF8C6AF1-2A7611DF-80A3FA72-EEE44B4E@168.90.100.243
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3481966201-0712380895-2157836914-4007938894
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1268113413
Contact: <14695554002>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 5465 8927 IN IP4 76.250.30.1
s=SIP Call
c=IN IP4 76.250.30.1
t=0 0
m=audio 18542 RTP/AVP 0 101
c=IN IP4 76.250.30.1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Mar  8 23:53:20.691 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 76.250.30.1:5060;branch=z9hG4bK79C25ED
From: <14695554002>;tag=5B031CC-1164
To: <19725559000>;tag=2ACB58-1E1C
Date: Tue, 09 Mar 2010 05:53:20 GMT
Call-ID: CF8C6AF1-2A7611DF-80A3FA72-EEE44B4E@168.90.100.243
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


Mar  8 23:53:20.695 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 76.250.30.1:5060;branch=z9hG4bK79C25ED
From: <14695554002>;tag=5B031CC-1164
To: <19725559000>;tag=2ACB58-1E1C
Date: Tue, 09 Mar 2010 05:53:20 GMT
Call-ID: CF8C6AF1-2A7611DF-80A3FA72-EEE44B4E@168.90.100.243
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

How do I get the IP-IP gateway to actually send the re-INVITE out to the ITSP SIP Gateway instead of sending back the 488 Not Acceptable Media? All three endpoints actually support the G.711 codec, and I still want media flow-around enabled. I do not want to transcode this call. How do I get this problem fixed? This re-INVITE is being handled incorrectly as we are getting a 488 response instead of the response being sent out to the ITSP SIP Gateway.

Did you resolve this issue and how?

For issues like this, run 'debug voip ccapi inout' along with 'debug ccsip mess' and see what inbound dial-peer is being matched.  Usually the inbound dial-peer match doesn't match the codec of which is being offered in the INVITE.

Getting Started

Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: