Comcast SIP trunks 480 error 1

[teletechman]

I am getting a 480 temporarily unavailable when trying to use a DID pointing to VM:AAName. It works if I point the DID to an ext or HG. This is very straight forward and I have done this 100 times but for whatever reason it will not work. I have tried to point it to the voicemail option as well and that does not work. The VM is working as I have a PRI DID pointed to the AA and it answers fine. The IPO is a 500V2 R11.1 FP2 SP2 with an App server on Hyper-V. Please help!
Mike

 

[derfloh]

How are your SIP URIs configured

IP Office remote service
IP Office certificate check
CLI based call blocking
SCN fallback over PSTN

 

[teletechman]

They have 2 URIs 1 is the test number from Comcast and one is set to auto.

 

[teletechman]

The setup is pretty basic, LAN2 is plugged into the Comcast SBC onsite so no firewall or network switch in between. I program the IPO for open internet and point the SIP trunks at the IP address of the Comcast SBC (Adtran).

 

[teletechman]

Here is a trace of the failure.
14:21:59 617043662mS SIP Rx: UDP 192.168.43.254:5060 -> 192.168.43.1:5060
INVITE sip:617xxxxxxx@192.168.43.1:5060 SIP/2.0
From: "USTELECENTERS" <sip:978xxxxxxx@192.168.43.1:5060;transport=UDP>;tag=7c2edb8-7f000001-13c4-136dd-148bb1e8-136dd
To: <sip:617xxxxxxx@192.168.43.1:5060>
Call-ID: 7ca29f8-7f000001-13c4-136dd-3d024c9f-136dd@192.168.43.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.43.254:5060;branch=z9hG4bK-136dd-4be4f87-5679f27b
Max-Forwards: 70
Supported: replaces
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, PRACK, REFER, REGISTER, OPTIONS
User-Agent: ADTRAN_NetVanta_3430/R13.9.0.HB.V (S0024454f56f1)
P-Asserted-Identity: "USTELECENTERS" <sip:978xxxxxxx@192.168.43.1>
Contact: <sip:978xxxxxxx@192.168.43.254:5060;transport=UDP>
Content-Type: application/sdp
Content-Length: 261

v=0
o=Sonus_UAC 870522 7268 IN IP4 192.168.43.254
s=SIP Media Capabilities
c=IN IP4 192.168.43.254
t=0 0
m=audio 10044 RTP/AVP 9 0 101
a=sendrecv
a=ptime:20
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
14:21:59 617043665mS Sip: Association found trunk: SIP Line (17)
14:21:59 617043666mS CMCallEvt: 0000000000000000 0.536874187.0 -1 BaseEP: NEW CMEndpoint f1a81834 TOTAL NOW=3 CALL_LIST=1
14:21:59 617043666mS Sip: SIP Line (17): sip_trunk_config_items 50020101, sip_trunk_config_items_2 00000000, voip.flags 00000949
14:21:59 617043666mS Sip: SIPDialog f1a7fb08 created, dialogs 1 txn_keys 1 video 1 presentation 1 camera 1 unsupp audio 0
14:21:59 617043666mS Sip: 0000000000000000 0.536874187.0 -1 SIPTrunk Endpoint(f1a7fb08) SetUnIntTransactionCondition to UnInt_None
14:21:59 617043667mS Sip: SIP Line (17): License, Valid 1, Available 25, Consumed 0
14:21:59 617043668mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForRequest from 192.168.43.254:5060 to 192.168.43.254:5060
14:21:59 617043668mS Sip: FindContactParameters mFarDisplayString <USTELECENTERS> mFarDisplayNumber <978xxxxxxx> restricted 0
14:21:59 617043669mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) Cloned
14:21:59 617043669mS Sip: SIPDialog::ExtractResponseParamsFromViaHeader remote sent_by: 192.168.43.254:5060 trunk
14:21:59 617043669mS Sip: SIPDialog::ExtractResponseParamsFromViaHeader remote sent by transport: SIP/2.0/UDP trunk
14:21:59 617043669mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 192.168.43.254:5060 to 192.168.43.254:5060
14:21:59 617043670mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) SendSIPResponse: INVITE code 100 SENT TO 192.168.43.254 5060
14:21:59 617043670mS SIP Tx: UDP 192.168.43.1:5060 -> 192.168.43.254:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.43.254:5060;branch=z9hG4bK-136dd-4be4f87-5679f27b
From: "USTELECENTERS" <sip:978xxxxxxx@192.168.43.1:5060;transport=UDP>;tag=7c2edb8-7f000001-13c4-136dd-148bb1e8-136dd
Call-ID: 7ca29f8-7f000001-13c4-136dd-3d024c9f-136dd@192.168.43.1
CSeq: 1 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 11.1.2.2.0 build 20
To: <sip:617xxxxxxx@192.168.43.1:5060>;tag=07668fefe1b2097d
Content-Length: 0

14:21:59 617043671mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) INVITE Received ep 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a81834), dialog f1a7fb08
14:21:59 617043671mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) UpdateSIPCallState SIPDialog::INITIAL(0) -> SIPDialog::INVITE_RCVD(9)
14:21:59 617043672mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) UpdateSDPState SIPDialog::IDLE(0) -> SIPDialog::OFFER_RCVD(2)
14:21:59 617043672mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) SdpClone
14:21:59 617043672mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) SIPDialog::BuildFastStartFromSDPOffer sdpmsg f1a7d690
14:21:59 617043672mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) SIPDialog::FindConnectionParameters: found media proto <RTP/AVP>
14:21:59 617043673mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) BuildFastStartFromSDPOffer: known_stream 1 media_type 0 stream_type 6 allow_stream 1
14:21:59 617043673mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) SIPDialog::BuildFastStartFromAudioMediaSDPOffer reinvite 0 mRTP_PType 255 prefer_existing_codec 0
14:21:59 617043673mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) HandleSDPContext: found_rfc2833 with payload 101
14:21:59 617043673mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) SetRfc2833TxPayload: use RFC2833 for dtmf
14:21:59 617043673mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) SetRfc2833TxPayload: payload 101 set_rx 1
14:21:59 617043674mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) Process SIP request dialog f1a7fb08, method INVITE in state SIPDialog::INVITE_RCVD(9)
14:21:59 617043675mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) UpdateClone mMesg f1a7bb8c smsg f58bd8b8
14:21:59 617043675mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) CMSetup forwarded to call model owner_ep 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a81834), dialog f1a7fb08 has sdp 1
14:21:59 617043675mS CMCallEvt: CREATE CALL:562 (f1ab0304)
14:21:59 617043676mS CMCallEvt: 0000000000000000 0.536874188.0 -1 BaseEP: NEW CMEndpoint f1abced0 TOTAL NOW=4 CALL_LIST=1
14:21:59 617043679mS CMLineRx: v=0
CMSetup
Line: type=SIPLine 17 Call: lid=17 id=536874187 in=1
Called[617xxxxxxx] Type=Default (100) Reason=CMDRdirect SndComp Calling[978xxxxxxx@192.168.43.1] Type=Unknown Plan=Default
BC: CMTC=Speech CMTM=CallIndSigCon CMTR=64 CMST=Default CMU1=ULaw
IE CMIEFastStartInfoData (6) 2 item(s)
IE CMIECallingPartyName (110)(Type=CMNameDefault) name=USTELECENTERS
IE CMIERespondingPartyName (228)(Type=CMNameDefault) name=USTELECENTERS
IE CMIERespondingPartyNumber (230)(P:100 S:100 T:0 N:100 R:4) number=978xxxxxxx@192.168.43.1
IE CMIEDeviceDetail (231) 0a00001320000ccb LOCALE=enu HW=15 VER=11 class=CMDeviceSIPTrunk type=0 number=17 channel=1 features=0x0 rx_gain=32 tx_gain=32
ep_callid=536874187 ipaddr=10.0.0.19 apps=0 loc=999 em_a_loc=999 em_d_loc=0 features2=0x0 is_spcall=1 ignores_dtmf=0
avgsid=<>
14:21:59 617043679mS CMCallEvt: 0a00001320000ccb 17.536874187.1 562 SIPTrunk Endpoint: StateChange: END=A CMCSIdle->CMCSDialInitiated
14:21:59 617043679mS CMTARGET: 0a00001320000ccb 17.536874187.1 562 SIPTrunk Endpoint: LOOKUP CALL ROUTE: GID=17 type=100 called_party=617xxxxxxx sub= calling=978xxxxxxx@192.168.43.1 calling_sub= dir=in complete=1 ses=0
14:21:59 617043680mS CMTARGET: 0a00001320000ccb 17.536874187.1 562 SIPTrunk Endpoint: SET BESTMATCH: length 0 vs -1 match= dest=VM:SomersetAA climatch -1 vs -1
14:21:59 617043680mS CMTARGET: 0a00001320000ccb 17.536874187.1 562 SIPTrunk Endpoint: SET BESTMATCH: length 10 vs 0 match=617xxxxxxx dest=VM:SomersetAA climatch -1 vs -1
14:21:59 617043680mS CMCallEvt: Priority hike: call 562 priority 0->1
14:21:59 617043680mS CMTARGET: 0a00001320000ccb 17.536874187.1 562 SIPTrunk Endpoint: LOOKUP ICR: DDI=617xxxxxxx CGPN=978xxxxxxx@192.168.43.1 (Destination VM:SomersetAA CLI ) => CDPN=VM:SomersetAA
14:21:59 617043680mS CMTARGET: 0a00001320000ccb 17.536874187.1 562 SIPTrunk Endpoint: ADD TARGET (N): number=VM:SomersetAA type=100 depth=1 nobar=1 setorig=1 ses=0
14:21:59 617043680mS CMTARGET: 0a00001320000ccb 17.536874187.1 562 SIPTrunk Endpoint: ADD VM TARGET
14:21:59 617043681mS CMTARGET: 0a00001320000ccb 17.536874187.1 562 SIPTrunk Endpoint: MakeVoicemailTarget pbx=<null> local=1 type=1
14:21:59 617043681mS CMMap: Mapper::AllocateCodec unable to allocate resource
14:21:59 617043681mS CMCallEvt: Cannot reserve a codec for Aend, aend is 0a00001320000ccb 17.536874187.1 562 SIPTrunk Endpoint
14:21:59 617043681mS CMTARGET: NULL: ADD PRIMARY
14:21:59 617043681mS CMTARGET: 0a00001320000ccb 17.536874187.1 562 SIPTrunk Endpoint: ADD VM TARGET: FAILED availability=2
14:21:59 617043681mS CMTARGET: 0a00001320000ccb 17.536874187.1 562 SIPTrunk Endpoint: NO INITIAL TARGETS: ??
14:21:59 617043681mS CMTARGET: 0a00001320000ccb 17.536874187.1 562 SIPTrunk Endpoint: Fallback() targeting failed
14:21:59 617043683mS CMLOGGING: CALL:2022/09/0714:21,00:00:00,000,978xxxxxxx@192.168.43.1,I,SomersetAA,617xxxxxxx,USTELECENTERS,,,0,,"",0,n/a
14:21:59 617043684mS CMLineTx: v=0
CMReleaseComp
Line: type=SIPLine 17 Call: lid=17 id=536874187 in=1
Cause=34, No cct/chan available
14:21:59 617043684mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a81834) received CMReleaseComp
14:21:59 617043684mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) Terminating dialog f1a7fb08, state SIPDialog::INVITE_RCVD(9) for cause CMCauseNoChannel
14:21:59 617043684mS Sip: SIPDialog::ExtractResponseParamsFromViaHeader remote sent_by: 192.168.43.254:5060 trunk
14:21:59 617043684mS Sip: SIPDialog::ExtractResponseParamsFromViaHeader remote sent by transport: SIP/2.0/UDP trunk
14:21:59 617043684mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 192.168.43.254:5060 to 192.168.43.254:5060
14:21:59 617043685mS Sip: 0a00001320000ccb 17.536874187.1 -1 SIPTrunk Endpoint(f1a7fb08) SendSIPResponse: INVITE code 480 SENT TO 192.168.43.254 5060
14:21:59 617043686mS SIP Tx: UDP 192.168.43.1:5060 -> 192.168.43.254:5060
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 192.168.43.254:5060;branch=z9hG4bK-136dd-4be4f87-5679f27b
From: "USTELECENTERS" <sip:978xxxxxxx@192.168.43.1:5060;transport=UDP>;tag=7c2edb8-7f000001-13c4-136dd-148bb1e8-136dd
Call-ID: 7ca29f8-7f000001-13c4-136dd-3d024c9f-136dd@192.168.43.1
CSeq: 1 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 11.1.2.2.0 build 20
Reason: Q.850;cause=34;text="No circuit/channel available"
To: <sip:617xxxxxxx@192.168.43.1:5060>;tag=07668fefe1b2097d
Content-Length: 0

 

[kwing112000]

this probably wont fix it but whats the network topology set to for the SIP trunk? Because this is a local IP you are routing to i would have it set to none. that way it doesnt use any of the LAN2 settings for the SIP trunk. there is no need for any of that because you are not going off the LAN.

Kevin Wing
ACSS Small and Medium Enterprise (SME) Communications
ACS- Implement IP Office
ACA- Implement IP Office
Vive Communications

 

[TouchToneTommy]

14:21:59 617043681mS CMMap: Mapper::AllocateCodec unable to allocate resource
14:21:59 617043681mS CMCallEvt: Cannot reserve a codec for Aend, aend is 0a00001320000ccb 17.536874187.1 562 SIPTrunk Endpoint

Do you have VCM resources, do the codec selections on the SIP trunk match the provider, is Re-invite Supported checked?

 

[teletechman]

TTT I feel like a fool, you are correct the IPO does not have a VCM in it. I will work with Sales to fix this. Thanks for catching this.