Cisco Gateway generates 488 "Media Type(s) Unavailable"

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Jun 27th, 2014
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I have a trouble with incoming calls from a ISR3945 to a ISR 2951.. When I send the call I receive a 488 Not acceptable here, I added the dial-peer config:

 

voice class codec 100
 codec preference 1 transparent
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g729r8

!

!

dial-peer voice 300 voip
 destination-pattern 61[0-9][0-9]
 session protocol sipv2
 session target ipv4:10.2.201.2

voice-class codec 100

 voice-class sip bind control source-interface GigabitEthernet0/0
 voice-class sip bind media source-interface GigabitEthernet0/0
 dtmf-relay rtp-nte sip-kpml sip-notify

no vad
 !

!

Here you have a trace matching this dial-peer, the call is refused inmediately after receiving the invite:

INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.227.19.233:5060;branch=z9hG4bK1642BD
Remote-Party-ID: <sip:[email protected]>;party=calling;screen=no;privacy=off
From: <sip:[email protected]>;tag=242C8A8C-14D
To: <sip:[email protected]>
Date: Mon, 23 Jun 2014 17:16:42 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 4006857602-4193063395-2465580347-3968770196
User-Agent: Cisco-SIPGateway/IOS-15.4.1.T
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1403543802
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 65
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 2397 6528 IN IP4 10.227.19.233
s=SIP Call
c=IN IP4 10.227.19.233
t=0 0
m=audio 17028 RTP/AVP 8 101
c=IN IP4 10.227.19.233
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

--------------------------------------------------

SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 10.227.19.233:5060;branch=z9hG4bK1642BD
From: <sip:[email protected]>;tag=242C8A8C-14D
To: <sip:[email protected]>;tag=80CBF7B0-128A
Date: Mon, 23 Jun 2014 17:19:21 GMT
Call-ID: [email protected]
Timestamp: 1403543802
CSeq: 101 INVITE
Allow-Events: telephone-event
Warning: 304 172.26.1.2 "Media Type(s) Unavailable"
Reason: Q.850;cause=65
Server: Cisco-SIPGateway/IOS-15.2.4.M4
Content-Length: 0

In the other hand, when I send outbound calls, they are succesfully completed as follows:

 

dial-peer voice 200 voip

 description CELULAR Local - Telecomm Atlas
 translation-profile outgoing remove7
 destination-pattern 044..........
 session protocol sipv2
 session target ipv4:10.227.19.233

 voice-class sip bind control source-interface GigabitEthernet0/1
 voice-class sip bind media source-interface GigabitEthernet0/1
 voice-class codec 100  
 dtmf-relay rtp-nte sip-kpml sip-notify
 no vad

!

!

Here you have a tracecall of a succesfull outbound call:

INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.26.1.2:5060;branch=z9hG4bK8425C3
Remote-Party-ID: "SA-Remigio Salvador Sanchez" <sip:[email protected]>;party=calling;screen=yes;privacy=off
From: "SA-Remigio Salvador Sanchez" <sip:[email protected]>;tag=808F9518-25F0
To: <sip:[email protected]>
Date: Mon, 23 Jun 2014 16:13:23 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1264244992-0000065536-0000000791-0046727690
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M4
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1403540003
Contact: <sip:[email protected]:5060>
Call-Info: <sip:172.26.1.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 69
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 241

v=0
o=CiscoSystemsSIP-GW-UserAgent 8112 1672 IN IP4 172.26.1.2
s=SIP Call
c=IN IP4 172.26.1.2
t=0 0
m=audio 20720 RTP/AVP 8 101
c=IN IP4 172.26.1.2
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

 

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.26.1.2:5060;branch=z9hG4bK8425C3
From: "SA-Remigio Salvador Sanchez" <sip:[email protected]>;tag=808F9518-25F0
To: <sip:[email protected]>;tag=23F03944-21E6
Date: Mon, 23 Jun 2014 16:10:45 GMT
Call-ID: [email protected]
Timestamp: 1403540003
CSeq: 101 INVITE
Require: 100rel
RSeq: 7688
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:[email protected]>;party=called;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.227.19.233:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.4.1.T
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 194

v=0
o=CiscoSystemsSIP-GW-UserAgent 6163 8882 IN IP4 10.227.19.233
s=SIP Call
c=IN IP4 10.227.19.233
t=0 0
m=audio 17010 RTP/AVP 8
c=IN IP4 10.227.19.233
a=rtpmap:8 PCMA/8000
a=ptime:20

 

 

PRACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.26.1.2:5060;branch=z9hG4bK85183B
From: "SA-Remigio Salvador Sanchez" <sip:[email protected]>;tag=808F9518-25F0
To: <sip:[email protected]>;tag=23F03944-21E6
Date: Mon, 23 Jun 2014 16:13:23 GMT
Call-ID: [email protected]
CSeq: 102 PRACK
RAck: 7688 101 INVITE
Allow-Events: kpml, telephone-event
Max-Forwards: 70
Content-Length: 0

 

SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.26.1.2:5060;branch=z9hG4bK85183B
From: "SA-Remigio Salvador Sanchez" <sip:[email protected]>;tag=808F9518-25F0
To: <sip:[email protected]>;tag=23F03944-21E6
Date: Mon, 23 Jun 2014 16:10:49 GMT
Call-ID: [email protected]
Server: Cisco-SIPGateway/IOS-15.4.1.T
CSeq: 102 PRACK
Content-Length: 0

 

SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.26.1.2:5060;branch=z9hG4bK8425C3
From: "SA-Remigio Salvador Sanchez" <sip:[email protected]>;tag=808F9518-25F0
To: <sip:[email protected]>;tag=23F03944-21E6
Date: Mon, 23 Jun 2014 16:10:49 GMT
Call-ID: [email protected]
Timestamp: 1403540003
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
Contact: <sip:[email protected]:5060>
Supported: replaces
Call-Info: <sip:10.227.19.233:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.4.1.T
Session-Expires:  1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 194

 

I checked both SDP info in both calls, but I dont have any idea to try to figure out this issue. I alrerady tried to forced the codec without succesfull.

 

Thanks

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George Thomas Fri, 06/27/2014 - 14:37
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Did you try what Ayodeji suggested earlier in the other post? Please provide full configurations along with debug voip ccapi inout and debug ccsip messages. Please dont truncate the logs as it becomes difficult to understand.. You also had H323 dial-peers on the remote gateway hence the reason for full logs/configs. If you cannot share the configs/debugs here, I would suggest you open a TAC case.

ilana_ilana Mon, 11/30/2015 - 12:07
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Hello carmenozcisco,

How did you ended up with your issue ?

Right now I am facing exactly same problem. One way call between CME's are successful and in reverse call immediately fail with error SIP/2.0 488 Not Acceptable Media error.

Would appreciate if you can share your finding.

Regards,

Tagir Temirgaliyev Mon, 11/30/2015 - 12:14
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  • Silver, 250 points or more

Please provide full configurations along with  debug ccsip messages.

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