12 Replies Latest reply on Jan 13, 2016 8:27 AM by cbbteceri

    Problems with Inbound SIP trunk from a MetaSwitch

    quazar66 New Member

      Setting up a SIP trunk from a MetaSwitch that includes 2 additional DID numbers.  The main number on the SIP trunk works fine, but the two additional DID will not connect to additional voice accounts.  From a "debug voice verbose" and a "debug sip stack messages invite" it appears that the invite is not being accepted:

       

      SIP Trunk Config:

      voice trunk T07 type sip

        description "Velocity"

        no reject-external

        sip-server primary adtran.vnetvoice.com

        registrar primary adtran.vnetvoice.com

        max-number-calls 3

        register 8146510925

        codec-list g711_first both

        authentication username "8146510925" password "........."

       

       

       

      Invite:


      15:12:21.828 SIP. MSG INVITE REQ RX anonymous 8146510926

      INVITE sip:8146510925@64.8.57.21:5060;transport=UDP SIP/2.0

      From: "Anonymous"<sip:anonymous@adtran.vnetvoice.com>;tag=66.211.254.180+1+5b7515c4+57993a83

      To: <sip:8146510926@adtran.vnetvoice.com>

      Call-ID: 0gQAAC8WAAACBAAALxYAAOX74rLtTeE4DuQWYnlXURZJ6H4S/S1VyMy+i8osgpUz@66.211.254.180

      CSeq: 684079421 INVITE

      Via: SIP/2.0/UDP 66.211.254.180:5060;branch=z9hG4bK+6ac65ac59a93c456a2d9932d944224911+sip+1+ab5c368c

      Expires: 180

      Call-Info: <sip:66.211.254.180:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"

      Supported: resource-priority

      Supported: siprec

      Supported: 100rel

      Organization: Metaswitch Networks

      Max-Forwards: 67

      Alert-Info: <http://www.notused.com>#info=alert-internal

      Accept: application/sdp, application/dtmf-relay

      Contact: <sip:f2d1491729f1bf23fb7d2caa5df7a57f@66.211.254.180:5060>

      Allow-Events: message-summary,refer,dialog,line-seize,presence,call-info,as-feature-event,calling-name

      Content-Type: application/sdp

      Content-Length: 199

       

      v=0

      o=- 48035111654102 48035111654102 IN IP4 66.211.254.180

      s=-

      c=IN IP4 66.211.254.180

      t=0 0

      m=audio 49986 RTP/AVP 0 18 101

      a=rtpmap:101 telephone-event/8000

      a=fmtp:18 annexb=no

      a=ptime:20

       

      15:12:21.835 SIP. MSG INVITE RSP TX anonymous 8146510926

      SIP/2.0 100 Trying

      From: "Anonymous"<sip:anonymous@adtran.vnetvoice.com>;tag=66.211.254.180+1+5b7515c4+57993a83

      To: <sip:8146510926@adtran.vnetvoice.com>

      Call-ID: 0gQAAC8WAAACBAAALxYAAOX74rLtTeE4DuQWYnlXURZJ6H4S/S1VyMy+i8osgpUz@66.211.254.180

      CSeq: 684079421 INVITE

      Via: SIP/2.0/UDP 66.211.254.180:5060;branch=z9hG4bK+6ac65ac59a93c456a2d9932d944224911+sip+1+ab5c368c

      Contact: <sip:8146510925@64.8.57.21:5060;transport=UDP>

      Supported: 100rel,replaces

      Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

      User-Agent: ADTRAN_NetVanta_7100/R11.4.3.E

      Content-Length: 0

       

      Voice Verbose:


      15:09:07.075 PM.208 Ca:0 Sending Keep-alive: INFO

      15:09:07.078 PM.208 Ca:0 call-leg transaction -> Request Sent

      15:09:07.197 PM.208 Ca:0 SipPM_Connected      rcvd SIP call-leg response: 200 OK

      15:09:07.197 PM.208 Ca:0 call-leg transaction -> Final Response Rcvd

      15:09:07.197 PM.208 Ca:0 SipPM_Connected      Received keep-alive response: 200

      15:09:07.198 PM.208 Ca:0 call-leg transaction -> Terminated

      15:09:12.252 TM.T07 01 SipTM_Idle           rcvd SIP call-leg request: INVITE

      15:09:12.253 TM.T07 01 SipTM_Idle           call-leg -> Offering

      15:09:12.253 TM.T07 01 SipTM_Idle           State change      >> SipTM_Idle->SipTM_Trying

      15:09:12.253 TM.T07 01 SipTM_Trying         SDP offer is not loopback request

      15:09:12.254 TM.T07 01 SipTM_Trying         Processing From for Caller-ID.

      15:09:12.254 TM.T07 01 SipTM_Trying         Caller ID is private due to 'anonymous' detected in header

      15:09:12.254 TM.T07 01 SipTM_Trying         Caller ID Name   = "Anonymous"

      15:09:12.254 TM.T07 01 SipTM_Trying         Caller ID Number = "anonymous"

      15:09:12.255 TM.T07 01 SipTM_Trying         Caller ID is private

      15:09:12.255 TM.T07 01 SipTM_Trying         info: unable to set redirect number(s) from INVITE

      15:09:12.255 TM.T07 01 SipTM_Trying         sent: TA->InboundCall

      15:09:12.256 TM.T07 01 Looking up source address for destination 66.211.254.180

      15:09:12.256 TM.T07 01 call-leg (0x50ccd70) -> src: 64.8.57.21 : 5060  dst: 66.211.254.180 : 5060

      15:09:12.258 TM.T07 01 SipTM_Trying         sent: 100 Trying

      15:09:12.259 TA.T07 01 TAIdle               rcvd: inboundCall from TM

      15:09:12.259 TA.T07 01 State change      >> TAIdle->TAInboundCall (TAS_Calling)

      15:09:12.259 TA.T07 01 Failed - DID translation: no match for 8146510925, using 8146510925

      15:09:12.259 TA.T07 01 TAIdle               sent: call to SB

      15:09:12.260 TM.T07 01 SipTM_Trying         tachg -> TAInboundCall

      15:09:12.260 TM.T07 01 SipTM_Trying         State change      >> SipTM_Trying->SipTM_Pending

      15:09:12.260 SB.CALL 44974 Idle                 Called the call routine with 8146510925

      15:09:12.261 SB.CALL 44974 Idle                 No LOCAL station matched dialed number (8146510925)

      15:09:12.261 SB.CALL 44974 Idle                 No TRUNK accepted dialed number (8146510925)

      15:09:12.261 SB.CALL 44974 Idle                 Translating alias: 8146510925 to 208

      15:09:12.262 SB.CCM isMappable:

      15:09:12.262 SB.CCM  :  Call Struct 0x496d810 :   Call-ID = 44974

      15:09:12.262 SB.CCM  :  Org Acct = T07    Dst Acct = 208

      15:09:12.262 SB.CCM  :  Org Port ID = SipTrunk 0/0.384   Dst Port ID = unknown 0/0

      15:09:12.263 SB.CCM  :  SDP Transaction = CallID: 44974

      15:09:12.263 SB.CCM  :  SDP Offer = 0x06152910, (66.211.254.180:49792)

      15:09:12.263 SB.CCM isMappable: Call Connection Type is RTP_TO_RTP

      15:09:12.263 SB.CCM handleRtpToRtp: Modifying SDP Offer

      15:09:12.264 SB.CCM translateOffer: offer codec list: PCMU G729

      15:09:12.265 SB.CCM translateOffer: revised offer codec list: PCMU G729

      15:09:12.265 SB.CCM translateOffer: codec list after answerer: PCMU G729

      15:09:12.266 SB.CCM translateOffer: DTMF signaling: answerer has no restrictions configured, passing offer(NTE 101) through

      15:09:12.267 SB.CCM translateOffer: success

      15:09:12.267 MEDIA.MANAGER Allocating media port.

      15:09:12.268 MEDIA.MANAGER getSubstitutePort: No matching callIdMap entry found for call 44974

      15:09:12.268 MEDIA.MANAGER Call ID map : Added new entry : call ID 44974 : session -44637792333386INIP466.211.254.180 : version 197

      226058 : index 2064

      15:09:12.268 MEDIA.MANAGER New media entry : type(0), callID(44974), sessionID(-44637792333386INIP466.211.254.180), original IP(66.

      211.254.180) ports(49792-49793), substitute IP(::) ports(12064-12065), RtpChannel(NULL), connection(0x6158010), sdpOverride(0), me(

      0x611a110). No RtpChannel

       

       

      So basically what happens is that any of the inbound SIP calls on this trunk go to the voice account that the main number is tied to.

       

      What am I missing?