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Problems with Cisco Voice Gateway 2811

I have a cisco 2811 router with 8 ports FXS which functions as SIP voice gateway, it is registered to a SIP SERVER,

but I can only make calls to telephone and not cell phone, and receive calls from the PSTN either.


Please could help me.

Router#sh run
Building configuration...

Current configuration : 2578 bytes
!
! Last configuration change at 18:31:40 UTC Thu Jan 26 2012
! NVRAM config last updated at 18:33:22 UTC Thu Jan 26 2012
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
!
no aaa new-model
!
resource policy
!
!
!
ip cef   
!
!
ip name-server 172.24.106.6
!
!
voice-card 0
no dspfarm
!
!
!
voice service voip
srtp
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
sip
  rel1xx disable
!
!
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
!
!
!
!
interface FastEthernet0/0
no ip address
load-interval 30
duplex full
speed 100
!
interface FastEthernet0/0.2709
encapsulation dot1Q 2709
ip address 10.32.160.178 255.255.255.252
no snmp trap link-status
!
interface FastEthernet0/1
ip address 10.32.248.253 255.255.255.252
duplex auto
speed auto
!
ip route 0.0.0.0 0.0.0.0 10.32.160.177
!
!
no ip http server
no ip http secure-server
!
!
!
control-plane
!

!
voice-port 0/1/0
!
voice-port 0/1/1
!
voice-port 0/2/0
!
voice-port 0/2/1
!
voice-port 0/2/2
!
voice-port 0/2/3
!
voice-port 0/3/0
station-id number 5116604421
!        
voice-port 0/3/1
station-id number 5116604423
!
!
!
!
!
dial-peer voice 4421 voip
preference 1
destination-pattern .T
voice-class codec 1
session protocol sipv2
session target ipv4:172.24.106.6
dtmf-relay sip-notify
no vad
!
dial-peer voice 604421 pots
preference 1
destination-pattern 5116604421
incoming called-number 5116604421
port 0/3/0
prefix 5116604421
authentication username 5116604421 password 0253550A5D505F75181C58 realm 172.24.106.6
!
dial-peer voice 604423 pots
preference 1
destination-pattern 5116604423
incoming called-number 5116604423
port 0/3/1
prefix 5116604423
authentication username 5116604423 password 014657550D5D565B751E1D realm 172.24.106.6
!
dial-peer voice 4423 voip
preference 1
destination-pattern .T
voice-class codec 1
session protocol sipv2
session target ipv4:172.24.106.6
dtmf-relay sip-notify
no vad
!
gateway
!
sip-ua   
no redirection
retry register 10
timers connect 100
timers refer 300
timers register 100
registrar ipv4:172.24.106.6 expires 3600
sip-server ipv4:172.24.106.6
!
!
!
!
line con 0
line aux 0
line vty 0 4
login
!
scheduler allocate 20000 1000
!
webvpn context Default_context
ssl authenticate verify all
!
no inservice
!        
!
end

Router#
Router#show sip-ua register status
Line          peer           expires(sec)  registered
============  =============  ============  ===========
5116604421    604421           376           yes        
5116604423    604423           382           yes        
Router#

Jan 26 17:57:04.278: //544/FD39FA5282E6/SIP/Error/sipSPIDoQoSNegotiation: Error:Unsupported QoS Type Jan 26 17:57:04.278: //544/FD39FA5282E6/SIP/Error/sipSPIContinueNewMsgInvite: Unacceptable INVITE media/precondition Jan 26 17:57:04.342: //545/FD43BE9B82E8/SIP/Error/sipSPIDoQoSNegotiation: Error:Unsupported QoS Type Jan 26

4 REPLIES

Problems with Cisco Voice Gateway 2811

This is one of the errors.

Reason: Q.850;cause=49

cause: 49(0x31)[Quality of service unavailable] Reason: Q.850;cause=49
cause: 49(0x31)[Quality of service unavailable]

Regards

Ricardo

Problems with Cisco Voice Gateway 2811

The answer is in your log:

Jan 26 17:57:04.278:  //544/FD39FA5282E6/SIP/Error/sipSPIDoQoSNegotiation: Error:Unsupported  QoS Type Jan 26 17:57:04.278:  //544/FD39FA5282E6/SIP/Error/sipSPIContinueNewMsgInvite: Unacceptable  INVITE media/precondition Jan 26 17:57:04.342:  //545/FD43BE9B82E8/SIP/Error/sipSPIDoQoSNegotiation: Error:Unsupported  QoS Type Jan 26

The incoming SIP INVITE from Huawei Softswitch contains SIP Preconditions fields in SIP and SDP part:

INVITE sip:5116604421@10.32.160.178:5060;user=phone SIP/2.0

From: "990189507"<990189507>;tag=ec1c908d-CC-24

To: "16604421"<5116604421>

Call-ID: 6216ce3a9920d82557287dff6dbfbd8b@10.18.5.64

CSeq: 1 INVITE

Record-Route: <172.24.106.6:5060>

Via: SIP/2.0/UDP 172.24.106.6:5060;branch=z9hG4bK-1461e72-f9e6ed50-28813d5c

Via: SIP/2.0/UDP 10.0.26.7:5061;branch=z9hG4bKa8e49766c965c8a02c5884ec0;X-DptMsg=135

Max-Forwards: 70

Supported: 100rel,precondition

Contact: <10.0.26.7:5060>

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

P-Early-Media: supported

Content-Type: application/sdp

Content-Length: 517

v=0

o=HuaweiSoftx3000 1078487705 1078487706 IN IP4 10.0.26.7

s=SipCall

c=IN IP4 10.0.2.40

t=0 0

m=audio 13304 RTP/AVP 8 18 4 108 102 0 116

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:108 AMR/8000

a=fmtp:108 mode-change-neighbor=1;mode-change-period=2

a=rtpmap:102 AMR/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:116 telephone-event/8000

a=ptime:10

a=curr:qos local sendrecv

a=curr:qos remote none

a=des:qos mandatory local sendrecv

a=des:qos optional remote sendrecv

a=3gOoBTC

You can ask to your ITSP to disable Preconditions or try to handle it.

Try this command: no ip rsvp policy default-reject

or enable SIP RSVP following this doc:

http://docwiki.cisco.com/wiki/End-to-End_RSVP_Over_SIP_Trunk_System_Test_Configuration

or

http://www.cisco.com/en/US/docs/ios/voice/sip/configuration/guide/sip_cg-rsvp.html

Regards.

Problems with Cisco Voice Gateway 2811

I can not make calls to cell phones.

Please help to review these logs.

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

Router#
*Jan 27 23:07:24.674: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x48E57ED0) with key=[207] to table
*Jan 27 23:07:24.674: //-1/000000000000/SIP/Info/sipSPI_ipip_vcc_Initialization:  Entry...
*Jan 27 23:07:24.674: //-1/000000000000/SIP/Info/ccsip_offer_ans_init:
*Jan 27 23:07:24.674: //344/000000000000/SIP/State/sipSPIChangeState: 0x48E57ED0 : State change from (STATE_NONE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
*Jan 27 23:07:24.674: //344/000000000000/SIP/Info/ccsip_call_setup_request: Set Protocol information
*Jan 27 23:07:24.674: //344/000000000000/SIP/Info/ccsip_call_setup_request: Before processing SETUP REQccb->pld.flags_ipip = 200
*Jan 27 23:07:24.674: //344/000000000000/SIP/Info/ccsip_call_setup_request: This is a TDM-IP call: callID= 344, peer_callID = 342
*Jan 27 23:07:24.674: //344/000000000000/SIP/Info/ccsip_call_setup_request: This is a TDM-IP call: callID= 344, peer_callID = 342
*Jan 27 23:07:24.674: //344/000000000000/SIP/Info/ccsip_call_setup_request: After processing SETUP REQccb->pld.flags_ipip = 200
*Jan 27 23:07:24.674: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetOutboundHostAndDestHostPrivate: CCSIP: target_host : 172.24.106.6 target_port : 5060

*Jan 27 23:07:24.674: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP
*Jan 27 23:07:24.674: //344/79E80ECC8232/SIP/Info/ccsip_call_setup_request: Incrementing call counter to [1] in dial-peer [4423]
*Jan 27 23:07:24.674: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 2
*Jan 27 23:07:24.674: //344/79E80ECC8232/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 158 to table
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Info/sipSPIGetCallConfig: preferred_codec set[0] type :No Codec    bytes: 0
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Info/sipSPIGetCallConfig: Media forking disabled
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Info/sipSPIGetCallConfig: Media Antitrombone disabled
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Info/sipSPICanSetFallbackFlag: Local Fallback is not active
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Info/sipSPISetMediaFlowMode: Storing the configured mode as FLOW-THROUGH
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Info/sipSPISetMediaFlowMode: xcoder high-density disabled
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Info/sipSPISetMediaFlowMode: Flow Mode set to FLOW_THROUGH
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Info/sipSPIGetCallConfig: Checking Video Type Rate=-1 video_codec_allowed=1F
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Info/sipSPIGetModemInfoPerCall: peer_callID=342
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Info/sipSPIGetCallConfig: Media forking disabled
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Info/sipSPISetOverlapConfiguration: Overlap signaling: FALSE: Endpt: SIP Trunk
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Info/sipSPI_ipip_GetHdrPassthruCfg: Hdr passthrough config:1 tag:0
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Info/sipSPI_ipip_GetCopyListCfg: Copy-list config:2 tag:0
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Info/sipSPI_ipip_build_consolidated_header_list: Both passthru and copylist are disabled
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Info/sipSPIValidateAndCopyOutboundHost: CCSIP: copy target_host to outbound_host
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Info/preprocessSetup:
This is a not a SIGO Call -, could be DM call
*Jan 27 23:07:24.678: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: calling reg_invoke_ip_first_hop()
*Jan 27 23:07:24.678: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: calling ip_best_local_address()
*Jan 27 23:07:24.678: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: return addr 10.32.160.178
*Jan 27 23:07:24.678: //344/79E80ECC8232/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.32.160.178
*Jan 27 23:07:24.678: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
*Jan 27 23:07:24.678: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling ip_best_local_address()
*Jan 27 23:07:24.678: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 10.32.160.178
*Jan 27 23:07:24.682: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 17318 for stream 1
*Jan 27 23:07:24.682: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g729r8 codecbytes :20, ptime: 20
*Jan 27 23:07:24.682: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1
*Jan 27 23:07:24.682: //344/79E80ECC8232/SIP/Info/sip_generate_sdp_xcaps_list: Modem Relay and T38 disabled. X-cap not needed
*Jan 27 23:07:24.682: //344/79E80ECC8232/SIP/Info/sipSPIOutgoingCallSDP: Creating recv-only stream for outbound call
*Jan 27 23:07:24.682: //344/79E80ECC8232/SIP/Media/sipSPIProcessRtpSessions: No active streams.
*Jan 27 23:07:24.682: //344/79E80ECC8232/SIP/Info/sip_gw_pre_setup_add_sdp_container: SDP container added
*Jan 27 23:07:24.682: //344/79E80ECC8232/SIP/Info/sipSPIValidateGtd: Signal Forward disabled
*Jan 27 23:07:24.682: //344/79E80ECC8232/SIP/Info/sipSPIValidateTunnelData: RawMsg/QSIG Tunneling Not Enabled
*Jan 27 23:07:24.682: //344/79E80ECC8232/SIP/Info/sipSPIAddMLPPServicesInfo: No MLP Info available on incoming leg
*Jan 27 23:07:24.682: //344/79E80ECC8232/SIP/Info/sipSPIPreprocessUriFormat: Url cfg for 1: 2,phone-ctxt=FALSE
*Jan 27 23:07:24.686: //344/79E80ECC8232/SIP/Info/sipSPIAddCiscoGcid: Gcid value not set - not adding header.
*Jan 27 23:07:24.686: //344/79E80ECC8232/SIP/Info/sipSPIAddPrivacyandIdentityInfo: Removing "id" value from Privacy
*Jan 27 23:07:24.686: //344/79E80ECC8232/SIP/Info/sipSPICompareHistoryInfoWithMatchedDialpeer: call-route history-info CLI not enabled
*Jan 27 23:07:24.686: //344/79E80ECC8232/SIP/Info/sipSPI_ipip_set_history_info_header: No HI header recvd from container
*Jan 27 23:07:24.686: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
*Jan 27 23:07:24.686: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling ip_best_local_address()
*Jan 27 23:07:24.686: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 10.32.160.178
*Jan 27 23:07:24.686: //344/79E80ECC8232/SIP/Info/sipSPIRscmsmAvail: Value returned by check is = 0
*Jan 27 23:07:24.686: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
*Jan 27 23:07:24.686: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling ip_best_local_address()
*Jan 27 23:07:24.686: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 10.32.160.178
*Jan 27 23:07:24.686: //344/79E80ECC8232/SIP/Info/sipSPIUaddCcbToUACTable: ****Adding to UAC table.
*Jan 27 23:07:24.686: //344/79E80ECC8232/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x48E57ED0 key=82452BA4-487211E1-82388A0F-CF80AD9E@10.32.160.178
*Jan 27 23:07:24.686: //344/79E80ECC8232/SIP/Info/sipSPIUsetBillingProfile: sipCallId for billing records = 82452BA4-487211E1-82388A0F-CF80AD9E@10.32.160.178
*Jan 27 23:07:24.686: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 172.24.106.6,Port 5060, Transport 1, SentBy Port 5060
*Jan 27 23:07:24.686: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Clock Time Zone is UTC, same as GMT: Using GMT
*Jan 27 23:07:24.686: //344/79E80ECC8232/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
*Jan 27 23:07:24.690: //344/79E80ECC8232/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:0, container:4928E778
*Jan 27 23:07:24.690: //344/79E80ECC8232/SIP/Info/Session-Timer/sipSTSLSRReqSend: Session timer is not required
*Jan 27 23:07:24.690: //344/79E80ECC8232/SIP/Info/Session-Timer/sipSTSLMain:
SE: 0;refresher:none peer refresher:none, flags:2000, posted event:E_STSL_INVALID_PEER_EVENT, reason:4
Configured SE:1800, Configured Min-SE:1800
*Jan 27 23:07:24.690: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIgetRegistrarHost: registrar host retrieved : 172.24.106.6
*Jan 27 23:07:24.690: //344/79E80ECC8232/SIP/Event/sipSPICreateRpid: Received Octet3A=0x00 -> Setting ;screen=no ;privacy=off
SIP: (344) Group (a= group line) attribute, level 65535 instance 1 not found.
*Jan 27 23:07:24.690: //344/79E80ECC8232/SIP/Info/sipSPIGetCallExtensionSupported: anat enabled, src_sdp dont have anat
*Jan 27 23:07:24.690: //344/79E80ECC8232/SIP/Info/sipSPISendInvite: Associated container=0x4928E778 to Invite
*Jan 27 23:07:24.690: //344/79E80ECC8232/SIP/Transport/sipSPISendInvite: Sending Invite to the transport layer
*Jan 27 23:07:24.690: //344/79E80ECC8232/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Global configuration, Switch Transport is FALSE
*Jan 27 23:07:24.690: //344/79E80ECC8232/SIP/Transport/sipSPITransportSendMessage: msg=0x4737CA48, addr=172.24.106.6, port=5060, sentBy_port=0, local_addr=, is_req=1, transport=1, switch=0, callBack=0x41A3199C
*Jan 27 23:07:24.690: //344/79E80ECC8232/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
*Jan 27 23:07:24.694: //344/79E80ECC8232/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
*Jan 27 23:07:24.694: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:172.24.106.6, rport:5060 with laddr:

*Jan 27 23:07:24.694: //344/79E80ECC8232/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x4737CA48
*Jan 27 23:07:24.694: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x4737CA48, addr=172.24.106.6, port=5060, local_addr=, connId=2 for UDP
*Jan 27 23:07:24.694: //344/79E80ECC8232/SIP/Info/sentInviteRequest: Sent Invite in state STATE_IDLE
*Jan 27 23:07:24.694: //-1/xxxxxxxxxxxx/SIP/Info/sentInviteRequest: Transaction active. Facilities will be queued.
*Jan 27 23:07:24.694: //344/79E80ECC8232/SIP/State/sipSPIChangeState: 0x48E57ED0 : State change from (STATE_IDLE, SUBSTATE_NONE)  to (STATE_SENT_INVITE, SUBSTATE_NONE)
*Jan 27 23:07:24.694: //344/79E80ECC8232/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 344) to the VOIP RTP library
*Jan 27 23:07:24.694: //344/79E80ECC8232/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr
*Jan 27 23:07:24.694: //344/79E80ECC8232/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.32.160.178
*Jan 27 23:07:24.694: //344/79E80ECC8232/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
*Jan 27 23:07:24.694: //344/79E80ECC8232/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 10.32.160.178, lport = 17318, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE
src_callid = 344, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY
media_ip_addr =  - , vrf tableid = 0 media_addr_type = 1
*Jan 27 23:07:24.694: //344/79E80ECC8232/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
*Jan 27 23:07:24.694: //344/79E80ECC8232/SIP/Info/sipSPICreateRtpSession: sess: 4941B8A0 do_rtcp:0
*Jan 27 23:07:24.694: //344/79E80ECC8232/SIP/Media/sipSPICreateRtpSession: stun is disabled
*Jan 27 23:07:24.694: //344/79E80ECC8232/SIP/State/sipSPIChangeStreamState: Stream (callid =  344)  State changed from (STREAM_ADDING) to (STREAM_ACTIVE)
*Jan 27 23:07:24.698: //344/79E80ECC8232/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:990189507@172.24.106.6:5060 SIP/2.0

Via: SIP/2.0/UDP 10.32.160.178:5060;branch=z9hG4bK12214B8

Remote-Party-ID: <5116604421>;party=calling;screen=no;privacy=off

From: <5116604421>;tag=F70F44-25E6

To: <990189507>

Date: Fri, 27 Jan 2012 23:07:24 GMT

Call-ID: 82452BA4-487211E1-82388A0F-CF80AD9E@10.32.160.178

Supported: timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 2045251276-1215435233-2184350223-3481316766

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: 1327705644

Contact: <5116604421>

Call-Info: <10.32.160.178:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 217

v=0

o=CiscoSystemsSIP-GW-UserAgent 7164 4979 IN IP4 10.32.160.178

s=SIP Call

c=IN IP4 10.32.160.178

t=0 0

m=audio 17318 RTP/AVP 18

c=IN IP4 10.32.160.178

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=ptime:20


*Jan 27 23:07:24.766: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [172.24.106.6]:5060, local_address:[ - ]
*Jan 27 23:07:24.766: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
*Jan 27 23:07:24.766: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
*Jan 27 23:07:24.766: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
*Jan 27 23:07:24.766: //344/79E80ECC8232/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying

From: <5116604421>;tag=F70F44-25E6

To: <990189507>

Call-ID: 82452BA4-487211E1-82388A0F-CF80AD9E@10.32.160.178

CSeq: 101 INVITE

Via: SIP/2.0/UDP 10.32.160.178:5060;branch=z9hG4bK12214B8

Timestamp: 1327705644

Record-Route: <172.24.106.6:5060>

Content-Length: 0


*Jan 27 23:07:24.770: //344/79E80ECC8232/SIP/Info/sipSPICheckResponseExt: INVITE response with no RSEQ - disable IS_REL1XX
*Jan 27 23:07:24.770: //344/79E80ECC8232/SIP/State/sipSPIChangeState: 0x48E57ED0 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE)  to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)

Problems with Cisco Voice Gateway 2811

Can you add more log?

There is only tour sent INVITE but not responses.

Regards.

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