Several suppliers SIP ITSP on SPA2102

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• PEI - SIP - CME - SIP - error CUCM Media is not Acceptable

  Hello world

  I have a problem with a TRUNK of SIP ITSP, the question is apparently "SIP/2.0 488 not acceptable media.

  I tried several things, I Don t know how to solve this problem.

  Outgoing calls is already OK, the problem is with incoming calls: of the ITSP to the CUCM.

  I have this topology of the ITSP SIP TRUNK:

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  Dial-peer voice voip 67
  Description * SIP trunk ITSP *.
  destination-model 591 [67]...
  session protocol sipv2
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  no interaction of dtmf
  Codec g711ulaw

  !

  Dial-peer voice voip 68
  Description * SIP trunk CUCM *.
  reply-to address. T
  session protocol sipv2
  session target ipv4:172.16.6.3
  voice-class sip forced early offer
  Codec g711ulaw

  !

  SIP - ua
  Disable-early-media 180
  connection-reuse

  !

  voip phone service
  list of approved IP addresses
  IPv4 0.0.0.0
  IPv4 0.0.0.0 0.0.0.0
  h323 connections allow h323
  allow connections h323 to SIP
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  allow sip to sip connections
  no service additional h450.7
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  no service additional sip refer
  Fax protocol t38 ls-redundancy version 0 0 hs-redundancy 0 help none
  SIP
  binding control source-interface GigabitEthernet0/1.5
  bind media source-interface GigabitEthernet0/1.5
  Registrar Server
  offer-early forced

  !

  On the side of CUCM:

  End point of media (checked)
  Disable the media beginning on 180 (unchecked)
  Requires the idle exchange of SDP for call Media Change (checked)
  Early support for voice and video calls (checked)
  Send send-receive SDP appealed INVITES (checked)

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  001062: 03:31:03: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Received:
  GUEST sip:[email protected]/ * /: 5060; user = phone SIP/2.0
  Via: SIP/2.0/UDP 172.17.0.11:5060; branch = z9hG4bKi2mhl410do70mrc4s141.1
  Call ID: [email protected]/ * /.
  From: "70965999"<>[email protected]/ * /; transport = udp; user = phone >; tag = 2u2uavrx-CC-1013
  To: '69200020'<>[email protected]/ * /; transport = udp; user = phone >
  CSeq: 1 INVITE
  Max-Forwards: 69
  Contact:
  Allow: INVITE, ACK, OPTIONS, CANCEL, INFO, BYE PRACK, NOTIFY, MESSAGE, UPDATE
  P - asserted-Identity has:
  Supported: 100rel, histinfo, prerequisite
  P-early-Media: support
  Content-Length: 362
  Content-Type: application/sdp

  v = 0
  o = HuaweiSoftx3000 1102026905 1102026906 IN IP4 172.17.0.11
  s = SipCall
  c = IN IP4 172.17.0.11
  t = 0 0
  m = audio RTP/AVP 8 18 116 10386
  a = rtpmap:8 PCMA/8000
  a G729/8000 rtpmap:18 =
  a rtpmap:116 telephone-event/8000 =
  a = ptime:5
  a = sendrecv local curr:qos
  a = distance zero curr:qos
  a = sendrecv local compulsory are: qos
  a = sendrecv distance optional with: qos
  a = 3gOoBTC

  001063: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
  Number = 69200020, called number = 69200020, Peer Info Type = DIALPEER_INFO_SPEECH
  001064: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
  Associate the rule of = DP_MATCH_DEST; Called number = 69200020
  001065: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
  No outbound dial-peer does not; Result = NO_MATCH(-1)
  001066: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
  dialstring = 69200020, saf_enabled = 1 saf_dndb_lookup = 1, dp_result =-1
  001067: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
  Result = NO_MATCH(-1)
  001068: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
  Number = 70965999, called number =, Voice-Interface = 0 x 0.
  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,
  Peer Type Info = DIALPEER_INFO_SPEECH
  001069: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
  Result = Success (0) after DP_MATCH_ANSWER; Incoming dial-peer = 68
  001070: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0
  001071: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
  Number = 70965999, called number =, Voice-Interface = 0 x 0.
  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,
  Peer Type Info = DIALPEER_INFO_SPEECH
  001072: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
  Result = Success (0) after DP_MATCH_ANSWER; Incoming dial-peer = 68
  001073: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0
  001074: 03:31:03: //-1/CD12136099D5/DPM/dpAssociateIncomingPeerCore:
  Number = 70965999, called number = 69200020, Voice-Interface = 0 x 0.
  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,
  Peer Type Info = DIALPEER_INFO_SPEECH
  001075: 03:31:03: //-1/CD12136099D5/DPM/dpAssociateIncomingPeerCore:
  Result = Success (0) after DP_MATCH_ANSWER; Incoming dial-peer = 68
  001076: 03:31:03: //-1/CD12136099D5/DPM/dpMatchSafModulePlugin:
  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0
  SIP: Attempt to analyze the attribute not supported at the level of the media
  001077: 03:31:03: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Envoy:
  SIP/2.0 488 Media is not Acceptable
  Via: SIP/2.0/UDP 172.17.0.11:5060; branch = z9hG4bKi2mhl410do70mrc4s141.1
  From: "70965999"<>[email protected]/ * /; transport = udp; user = phone >; tag = 2u2uavrx-CC-1013
  To: '69200020'<>[email protected]/ * /; transport = udp; user = phone >; tag = C139A0-1755
  Date: Thu, November 6, 2014 13:01:04 GMT
  Call ID: [email protected]/ * /.
  CSeq: 1 INVITE
  Allow-events: telephone-event
  Reason: Q.850; cause = 65
  Server: Cisco-SIPGateway/IOS-15.3.3.M2
  Content-Length: 0

  001078: 03:31:03: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Received:

  SIP GatewayTelf_JD #ACK:[email protected]/ * /: 5060; user = phone SIP/2.0
  Via: SIP/2.0/UDP 172.17.0.11:5060; branch = z9hG4bKi2mhl410do70mrc4s141.1
  CSeq: 1 ACK
  Call ID: [email protected]/ * /.
  From: "70965999"<>[email protected]/ * /; transport = udp; user = phone >; tag = 2u2uavrx-CC-1013
  To: '69200020'<>[email protected]/ * /; transport = udp; user = phone >; tag = C139A0-1755
  Max-Forwards: 69
  Content-Length: 0

  Little light at the end of the tunnel?

  Thanks in advance!

  Hello

  can you collect debug voice ccapi inout & debugging ccsip GCE message and attach the logs here please?

  your provider sends A Law G711 codec in the PROMPT message, but you have configured G711 U right in the dial-peers.

  can you try to fix G711 has the right dial-peers and check out them? Also make sure you have TPMS in the MRGL applied to CUCM SIP Trunk.

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  Hello

  Is it possible to use a SIP trunk to a provider SIP ITSP having the CUBE / router gateway behind a firewall using a NAT?

  Does anyone do this?

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  Tom

  Hello

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  https://Tools.ietf.org/html/rfc6314

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