5 Replies Latest reply on Jan 23, 2017 2:19 PM by vasquezwilmer

    route voice traffic from an FXS port to SIP trunks

    wernerg New Member

      I need to route voice traffic from an FXS port which connected to a fax to SIP trunks on Eth0 which communicate with a VoIP PBX.  924e.

        • Re: route voice traffic from an FXS port to SIP trunks
          jay Employee

          Werner, assuming the SIP trunk is already set up, you would configure your voice user like this:

           

          voice user 5551000

            connect fxs 0/1

            sip-identity 5551000 T01 register auth-name 5551000 password 1234

           

          Please let me know if you have further questions. Thanks

          • Re: route voice traffic from an FXS port to SIP trunks
            jayh Hall_of_Fame

            You're placing a call from the FXS port to the PBX, correct?

             

            Is there a voice grouped-trunk attached to the PBX SIP trunk which matches the dialed digits from the FAX? Does the PBX expect to see that dialed pattern exactly as dialed on that trunk?

             

            What does "debug voice switchboard" show when you attempt a call? What do you hear in the speaker of the fax machine (or from an analog phone plugged in to that port) when you dial the call?

              • Re: route voice traffic from an FXS port to SIP trunks
                vasquezwilmer New Member

                hello, I have same issue, no audio between adtran (one phone connected to a FXS port) and one extension registered on Asterisk PBX. Debuging shows CODECS. RTP negotion OK but still no audio. Please let me know what is wrong. (NO NAT NO FIREWALL) Adtran and Asterisk are in the same network.

                 

                 

                ADTRAN-LAB#sh run

                Building configuration...

                !

                !

                ! ADTRAN, Inc. OS version R10.9.0

                ! Boot ROM version R10.9.0

                ! Platform: Total Access 908e (3rd Gen), part number 4243908F2

                ! Serial number CFG1236087

                !

                !

                hostname "ADTRAN-LAB"

                enable password

                !

                !

                !

                ip subnet-zero

                ip classless

                ip default-gateway 192.168.1.1

                ip routing

                ipv6 unicast-routing

                !

                !

                domain-name "google.com"

                domain-proxy

                name-server 8.8.8.8 8.8.4.4

                !

                !

                auto-config

                !

                event-history on

                no logging forwarding

                no logging email

                !

                no service password-encryption

                !

                 

                !

                banner motd X

                ===== =====

                 

                 

                X

                !

                !

                no ip firewall alg msn

                no ip firewall alg mszone

                no ip firewall alg h323

                !

                !

                !

                !

                !

                !

                !

                !

                no dot11ap access-point-control

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !      

                !

                !

                interface eth 0/1

                  description LAN Port

                  ip address  192.168.100.14  255.255.255.0

                  shutdown

                !

                !

                interface eth 0/2

                  description Connection to SW1

                  ip address  10.10.1.1  255.255.255.0

                  media-gateway ip primary

                  no shutdown

                !

                !

                !

                interface gigabit-eth 0/1

                  no ip address

                  no shutdown

                !

                !

                !

                !

                interface t1 0/1

                  description Connected to Asterisk PRI

                  tdm-group 1 timeslots 1-24 speed 64

                  no shutdown

                !

                interface t1 0/2

                  description Connected to BR1

                  tdm-group 2 timeslots 1-24 speed 64

                  no shutdown

                !

                interface t1 0/3

                  shutdown

                !

                interface t1 0/4

                  shutdown

                !

                !

                interface pri 1

                  isdn switch-type dms

                  role network b-channel-restarts enable

                  connect t1 0/1 tdm-group 1

                  no shutdown

                !

                !

                interface fxs 0/1

                  no shutdown

                !

                interface fxs 0/2

                  no shutdown

                !

                interface fxs 0/3

                  no shutdown

                !

                interface fxs 0/4

                  no shutdown

                !

                interface fxs 0/5

                  no shutdown

                !

                interface fxs 0/6

                  no shutdown

                !

                interface fxs 0/7

                  no shutdown

                !

                interface fxs 0/8

                  no shutdown

                !      

                !

                interface fxo 0/0

                  no shutdown

                !

                interface ppp 1

                  description PPP

                  ip address  192.168.1.2  255.255.255.252

                  media-gateway ip primary

                  no shutdown

                  cross-connect 1 t1 0/2 2 ppp 1

                !

                !

                isdn-group 1

                  connect pri 1

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                no tftp server

                no tftp server overwrite

                http server

                http session-timeout 5400

                http secure-server

                snmp agent

                no ip ftp server

                no ip scp server

                no ip sntp server

                !

                !

                !

                !

                !

                !

                !

                !

                sip

                sip udp 5060

                no sip tcp

                !

                !

                !

                voice feature-mode network

                voice forward-mode network

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                voice codec-list CODECS

                  codec g711ulaw

                  codec g729

                !

                !

                !

                voice trunk T01 type isdn

                  description "PRI PTP Link"

                  resource-selection circular descending

                  connect isdn-group 1

                  no early-cut-through

                  rtp delay-mode adaptive

                !

                voice trunk T02 type sip

                  description "to SONUS"

                  sip-server primary 192.168.100.200

                  trust-domain

                  transfer-mode network

                !

                voice trunk T03 type sip

                  description "to ASTERISK"

                  sip-server primary 10.10.1.200

                  codec-list CODECS both

                !

                !

                voice grouped-trunk SONUS

                  trunk T02

                  accept 2000 cost 0

                !

                !

                voice grouped-trunk ASTERISK

                  trunk T03

                  accept 1000 cost 0

                !

                !

                voice user 3000

                  connect fxs 0/1

                  password "1234"

                  did "9549054211"

                  sip-authentication password "1234"

                  codec-list CODECS

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                !

                ip rtp symmetric-filter

                !

                !

                !

                line con 0

                  no login

                !

                line telnet 0 4

                  login

                  password 12350WCS_noc

                  no shutdown

                line ssh 0 4

                  login local-userlist

                  no shutdown

                !

                sntp server time.nist.gov

                !

                !

                !

                !

                end

                ADTRAN-LAB#   

                 

                ASTERISK Extension.conf:

                 

                [general]

                static=yes

                writeprotect=no

                clearglobalvars=no

                 

                [globals]

                CONSOLE=Console/dsp

                 

                autofallthrough=yes

                VERYSHORTTIMEOUT=10

                SHORTTIMEOUT=20

                MEDTIMEOUT=45

                LONGTIMEOUT=60

                 

                [default]

                exten => _2XXX,1,Dial(SIP/${EXTEN}@ADTRAN-LAB)

                exten => 9549054211,1,Dial(SIP/${EXTEN}@ADTRAN-LAB)

                 

                exten => _1XXX,1,Dial(SIP/1000)

                 

                ASTERISK SIP.CONF:

                 

                [general]

                context=default

                allowguest=no

                allowoverlap=no

                allowtransfer=no

                bindport=5060

                bindaddr=0.0.0.0

                srvlookup=yes

                vmexten=vm

                disallow=all

                allow=alaw

                allow=ulaw

                allow=g723

                allow=all

                useragent=ClIeNt-PbX

                rtptimeout=60

                rtpholdtimeout=120

                canreinvite=yes

                alwaysauthreject = yes

                directmedia=no

                 

                [1000]

                type=peer

                host=dynamic

                secret=12350WCS_noc

                context=default

                qualify=yes

                port=5060

                nat=yes

                disallow=all

                allow=all

                dial=SIP/1000

                 

                =============================DEBUG ADTRAN VOICE VERBOSE==========================

                 

                ADTRAN-LAB#

                22:26:18.252 TM.T03 01 SipTM_Idle           rcvd SIP call-leg request: INVITE

                22:26:18.252 TM.T03 01 SipTM_Idle           call-leg -> Offering

                22:26:18.253 TM.T03 01 SipTM_Idle           State change      >> SipTM_Idle->SipTM_Trying

                22:26:18.254 TM.T03 01 SipTM_Trying         SDP offer is not loopback request

                22:26:18.254 TM.T03 01 SipTM_Trying         Processing From for Caller-ID.

                22:26:18.254 TM.T03 01 SipTM_Trying         Caller ID Name   = "1000"

                22:26:18.254 TM.T03 01 SipTM_Trying         Caller ID Number = "1000"

                22:26:18.255 TM.T03 01 SipTM_Trying         info: unable to set redirect number(s) from INVITE

                22:26:18.255 TM.T03 01 SipTM_Trying         sent: TA->InboundCall

                22:26:18.255 TM.T03 01 Looking up source address for destination 10.10.1.200

                22:26:18.255 TM.T03 01 call-leg (0x0x628bea60) -> src: 10.10.1.1 : 5060  dst: 10.10.1.200 : 5060

                22:26:18.257 TM.T03 01 SipTM_Trying         sent: 100 Trying

                22:26:18.258 TA.T03 01 TAIdle               rcvd: inboundCall from TM

                22:26:18.258 TA.T03 01 State change      >> TAIdle->TAInboundCall (TAS_Calling)

                22:26:18.258 TA.T03 01 Success - DID resolved 9549054211 to 3000

                22:26:18.259 TA.T03 01 TAIdle               sent: call to SB

                22:26:18.259 TM.T03 01 SipTM_Trying         tachg -> TAInboundCall

                22:26:18.259 TM.T03 01 SipTM_Trying         State change      >> SipTM_Trying->SipTM_Pending

                22:26:18.259 SB.CALL 213 Idle                 Called the call routine with 3000

                22:26:18.260 SB.CCM isMappable:

                22:26:18.260 SB.CCM  :  Call Struct 0x0x75150a10 :   Call-ID = 213

                22:26:18.260 SB.CCM  :  Org Acct = T03    Dst Acct = 3000

                22:26:18.260 SB.CCM  :  Org Port ID = SipTrunk 0/0.200   Dst Port ID = unknown 0/0

                22:26:18.260 SB.CCM  :  SDP Transaction = CallID: 213

                22:26:18.260 SB.CCM  :  SDP Offer = 0x75157610, (10.10.1.200:18000)

                22:26:18.261 SB.CCM isMappable: Call Connection Type is RTP_TO_TDM

                22:26:18.261 SB.CCM isMappable: Reserving RTP Channel 0/1.1

                22:26:18.264 SB.CCM translateOffer: offer codec list: PCMU GSM PCMA       G722  

                22:26:18.264 SB.CCM translateOffer: revised offer codec list: PCMU

                22:26:18.264 SB.CCM translateOffer: codec list after answerer: PCMU

                22:26:18.265 SB.CCM translateOffer: DTMF signaling: answerer has no restrictions configured, passing offer(NTE 101) through

                22:26:18.265 SB.CCM translateOffer: success

                22:26:18.266 MEDIA.MANAGER Allocating media port.

                22:26:18.266 MEDIA.MANAGER getSubstitutePort: No matching callIdMap entry found for call 213

                22:26:18.266 MEDIA.MANAGER Call ID map : Added new entry : call ID 213 : session root886965307INIP410.10.1.200 : version 886965307 : index 524

                22:26:18.266 MEDIA.MANAGER New media entry : type(0), callID(213), sessionID(root886965307INIP410.10.1.200), original IP(10.10.1.200) ports(18000-18001), substitute IP(::) ports(10524-10525), RtpChannel(0/1.1), connection(0x0x7516a510), sdpOverride(0), me(0x0x7516d910). RtpChannel 0/1.1

                22:26:18.266 SB.CALL 213 Idle                 Call sent from T03 to 3000 (3000)

                22:26:18.267 SB.CALL 213 State change      >> Idle->Delivering

                22:26:18.267 RTP.MANAGER fxs 0/1 - empty - RTP: Request resource

                22:26:18.267 RTP.MANAGER fxs 0/1 - Dsp 0/1.1 - RTP: DSP channel allocated for the resource

                22:26:18.267 RTP.PROVIDER fxs 0/1 - Dsp 0/1.1 - RTP: providing already allocated RTP channel

                22:26:18.267 TA.T03 01 TAInboundCall        CallResp event accepted

                22:26:18.268 TA.T03 01 State change      >> TAInboundCall->TAConnectWaitIn (TAS_Calling)

                22:26:18.268 SA.3000 rcvd: deliver from SB

                22:26:18.268 SA.3000 Ca:0 Idle                 sent: deliverResponse(accept) to SB

                22:26:18.268 SA.3000 Ca:0 Idle                 Set my destination sessionCookie to my call Appearance

                22:26:18.268 SA.3000 Ca:0 Idle                 State change      >> Idle->Ringing (CAS_Ringing)

                22:26:18.269 SA.3000 Ca:0 Ringing              sent: AcctPhoneMgr_cachg(CAS_Ringing) to PM

                22:26:18.269 PM.0:1 Idle                 Processed CACHG:Ring

                22:26:18.269 PM.0:1 Idle                 sent: Alert to SA

                22:26:18.269 PM.0:1 State change      >> Idle->Ringing

                22:26:18.270 SB.CALL 213 Delivering           Called the deliverResponse routine from Delivering

                22:26:18.270 SB.CALL 213 Delivering           DeliverResponse(accept) sent from 3000 to T03

                22:26:18.270 SA.3000 Ca:0 Ringing              rcvd: AcctPhoneMgr_alert from PM

                22:26:18.270 SA.3000 Ca:0 Ringing              sent: deliverResponse(alert) to SB

                22:26:18.270 TONESERVICES.EVENTS fxs 0/1 - empty - Caller-ID Generation: Request resource

                22:26:18.270 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Caller-ID Generation: DSP channel allocated for the resource

                22:26:18.271 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Caller-ID Generation: constructed

                22:26:18.271 TA.T03 01 TAConnectWaitIn      deliverResponse event accepted

                22:26:18.271 TA.T03 01 TAConnectWaitIn      ERROR! deliverResponse ignored

                22:26:18.271 SB.CALL 213 Delivering           Called the deliverResponse routine from Delivering

                22:26:18.271 SB.CALL 213 Delivering           Alert sent from 3000 to T03

                22:26:18.272 SB.CALL 213 State change      >> Delivering->Alerting

                22:26:18.272 TA.T03 01 TAConnectWaitIn      alert event accepted

                22:26:18.272 TM.T03 01 SipTM_Pending        tachg -> TAConnectWaitIn

                22:26:18.272 TM.T03 01 SipTM_Pending        State change      >> SipTM_Pending->SipTM_Alerting

                22:26:18.273 TM.T03 01 SipTM_Alerting       Sent 180 Ringing

                22:26:18 SB.CallStructObserver 213 Created

                22:26:18 SB.CallStructObserver 213 <-> 40b5d3c22fe8b6e370bca8717cf15ffd@10.10.1.200:5060

                22:26:20.797 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Caller-ID Generation: starting Caller-ID alert and sending Caller-ID information:

                22:26:20.797 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Caller-ID Generation:   chars = "....01202226..1000..1000?"

                22:26:20.797 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Caller-ID Generation:   bytes = "80 16 01 08 30 31 32 30 32 32 32 36 02 04 31 30 30 30 07 04 31 30 30 30 3F"

                22:26:20.797 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Caller-ID Generation: TDM map

                22:26:21.615 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Caller-ID Generation: received Caller-ID Done event

                22:26:21.615 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Caller-ID Generation: stopping

                22:26:21.615 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Caller-ID Generation: TDM unmap

                22:26:21.616 RTP.CHANNEL fxs 0/1 - Dsp 0/1.1 - Caller-ID Generation: releasing RTP resource

                22:26:21.616 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Caller-ID Generation: release

                22:26:23.917 PM.0:1 Ringing              Processed OFFHOOK

                22:26:23.917 PM.0:1 State change      >> Ringing->Connected

                22:26:23.917 SA.3000 Ca:0 Ringing              rcvd: AcctPhoneMgr_connect from PM

                22:26:23.917 SA.3000 Ca:0 Ringing              sent: connect to SB

                22:26:23.918 SA.3000 Ca:0 Ringing              State change      >> Ringing->Connecting (CAS_Active)

                22:26:23.918 SB.CALL 213 Alerting             Called the connect routine

                22:26:23.918 SB.CCM isResponseMappable:

                22:26:23.918 SB.CCM  :  Call Struct 0x0x75150a10 :   Call-ID = 213

                22:26:23.918 SB.CCM  :  Org Acct = T03    Dst Acct = 3000

                22:26:23.918 SB.CCM  :  Org Port ID = SipTrunk 0/0.200   Dst Port ID = FxsPhone 0/1

                22:26:23.919 SB.CCM  :  SDP Transaction = CallID: 213

                22:26:23.919 SB.CCM  :  SDP Offer = 0x75157610, (10.10.1.200:18000)

                22:26:23.919 SB.CCM  :  RTP Channel = 0/1.1

                22:26:23.919 SB.CCM isResponseMappable: reversing call connection type to compensate for event originator direction

                22:26:23.919 SB.CCM isResponseMappable: Call Connection Type is TDM_TO_RTP

                22:26:23.919 SB.CCM isResponseMappable: Creating SDP Answer based on SDP Offer

                22:26:23.920 SB.CCM createAnswer: creating SDP answer using RTP channel 0/1.1

                22:26:23.920 SB.CCM createAnswer : offer  codec list: PCMU

                22:26:23.920 SB.CCM              : answer codec list: PCMU

                22:26:23.921 SB.CCM createAnswer : result codec list: PCMU

                22:26:23.922 SB.CCM createAnswer : final DTMF signaling(NTE 101)

                22:26:23.922 MEDIA.MANAGER getSubstitutePort: Matching callIdMap entry found for call 213 sessionId root886965307INIP410.10.1.200 remote port 18000

                22:26:23.922 MEDIA.MANAGER getSubstitutePort: Matching sessionPortMap entry found for session

                22:26:23.922 MEDIA.MANAGER getSubstitutePort: Session port count (1)  Returning port (10524)

                22:26:23.922 SB.CCM updateMediaEntryForReinviteWithSameSdp : no associated port found for port (10524)

                22:26:23.923 SB.CCM translateAnswer: offer  codec list: PCMU

                22:26:23.923 SB.CCM                : answer codec list: PCMU

                22:26:23.923 SB.CCM translateAnswer: CODEC transcoding is not required

                22:26:23.924 SB.CCM translateAnswer: offer / answer DTMF signaling identical: DTMF transcoding not required

                22:26:23.924 SB.CCM translateAnswer: success

                22:26:23.924 MEDIA.MANAGER Allocating media port.

                22:26:23.925 MEDIA.MANAGER getSubstitutePort: Matching callIdMap entry found for call 213 sessionId -1484951183INIP4127.0.0.3 remote port 0

                22:26:23.925 MEDIA.MANAGER Call ID map : Added new session ID : call ID 213 : session -1484951183INIP4127.0.0.3 : version 1 : index 526

                22:26:23.925 MEDIA.MANAGER New media entry : type(0), callID(213), sessionID(-1484951183INIP4127.0.0.3), original IP(127.0.0.3) ports(10526-10527), substitute IP(::) ports(10526-10527), RtpChannel(0/1.1), connection(0x0x7516ca10), sdpOverride(0), me(0x0x7515d710). RtpChannel 0/1.1

                22:26:23.926 SB.CALL 213 Alerting             Connect sent from 3000 to T03

                22:26:23.926 SB.CALL 213 State change      >> Alerting->Connecting

                22:26:23.926 TA.T03 01 TAConnectWaitIn      connect event accepted

                22:26:23.926 TA.T03 01 State change      >> TAConnectWaitIn->TAConnectPending (TAS_Connected)

                22:26:23.926 TM.T03 01 SipTM_Alerting       tachg -> TAConnectPending

                22:26:23.927 TM.T03 01 SipTM_Alerting       State change      >> SipTM_Alerting->SipTM_Accept

                22:26:23.927 TM.T03 01 SDP DPI call ID 213 : No media bin.

                22:26:23.927 TM.T03 01 Processing new SDP entries.

                22:26:23.927 TM.T03 01 Checking for internal Media Gateway IP Address

                22:26:23.928 TM.T03 01 Using RTP Channel 0/1.1

                22:26:23.928 TM.T03 01 Inserting 10.10.1.1 into SDP for Media Gateway

                22:26:23.928 MEDIA.MANAGER getSubstitutePort: Matching callIdMap entry found for call 213 sessionId -1484951183INIP4127.0.0.3 remote port 10526

                22:26:23.928 MEDIA.MANAGER getSubstitutePort: Matching sessionPortMap entry found for session

                22:26:23.928 MEDIA.MANAGER getSubstitutePort: Session port count (1)  Returning port (10526)

                22:26:23.928 MEDIA.MANAGER Existing entry found for port reuse of SDP port 10526 and sub port 10526.

                22:26:23.929 MEDIA.MANAGER Reuse anchor entry with same SDP : call 213 : session -1484951183INIP4127.0.0.3 : IP 10.10.1.1 ports 10526 - 10527 : remote IP 127.0.0.3 ports 10526 - 10527.

                22:26:23.929 TM.T03 01 Adding RTP Media Gateway Entry: 127.0.0.3:10526 -> 10.10.1.1:10526

                22:26:23.929 TM.T03 01 Allocating anchor ports 10526 and 10527 for interface 10.10.1.1

                22:26:23.931 TM.T03 01 SipTM_Accept         call-leg -> Accepted

                22:26:23.931 TM.T03 01 SipTM_Accept         sent: 200 with SDP

                22:26:23.933 TM.T03 01 SipTM_Accept         rcvd SIP call-leg request: ACK

                22:26:23.933 TM.T03 01 SipTM_Accept         call-leg -> Connected

                22:26:23.933 TM.T03 01 SipTM_Accept         No body in message when trying to get SDP

                22:26:23.934 TM.T03 01 SipTM_Accept         info: unable to save SDP

                22:26:23.934 TM.T03 01 SipTM_Accept         sent: TA->Connect

                22:26:23.934 TM.T03 01 SipTM_Accept         State change      >> SipTM_Accept->SipTM_Connected

                22:26:23.934 TM.T03 01 SipTM_Connected      call-leg-mod -> Modify Idle

                22:26:23.934 TA.T03 01 TAConnectPending     rcvd: connect from TM

                22:26:23.935 TA.T03 01 State change      >> TAConnectPending->TAConnected (TAS_Connected)

                22:26:23.935 SB.CALL 213 Connecting           Called the connectResponse routine

                22:26:23.935 SB.CCM connect:

                22:26:23.935 SB.CCM  :  Call Struct 0x0x75150a10 :   Call-ID = 213

                22:26:23.935 SB.CCM  :  Org Acct = T03    Dst Acct = 3000

                22:26:23.935 SB.CCM  :  Org Port ID = SipTrunk 0/0.200   Dst Port ID = FxsPhone 0/1

                22:26:23.936 SB.CCM  :  SDP Transaction = CallID: 213

                22:26:23.936 SB.CCM  :  SDP Offer = 0x75157610, (10.10.1.200:18000)

                22:26:23.936 SB.CCM  :  SDP Answer = 0x75169510, (127.0.0.3:10526)

                22:26:23.936 SB.CCM  :  RTP Channel = 0/1.1

                22:26:23.937 SB.CCM connect: Call Connection Type is RTP_TO_TDM

                22:26:23.937 SB.CCM SDP offer is 10.10.1.200:18000, SDP answer is 127.0.0.3:10526

                22:26:23.937 MEDIA.MANAGER Trying to connect call ID 213 : SDP sessions root886965307INIP410.10.1.200 and -1484951183INIP4127.0.0.3

                22:26:23.937 MEDIA.MANAGER Found 1 ports for session root886965307INIP410.10.1.200

                22:26:23.937 MEDIA.MANAGER Found 1 ports for session -1484951183INIP4127.0.0.3

                22:26:23.938 MEDIA.MANAGER Connecting Disconnected Local [::]:10524 : Remote 10.10.1.200:18000

                22:26:23.938 MEDIA.MANAGER    and     Disconnected Local 10.10.1.1:10526 : Remote 127.0.0.3:10526

                22:26:23.938 MEDIA.MANAGER Setting up DSP Media Connection 213 for entry(type(0), callID(213), sessionID(root886965307INIP410.10.1.200), original IP(10.10.1.200) ports(18000-18001), substitute IP(::) ports(10524-10525), RtpChannel(0/1.1), connection(0x0x7516a510), sdpOverride(0), me(0x0x7516d910))

                22:26:23.938 MEDIA.MANAGER Setting up DSP Media Connection 213 for entry(type(0), callID(213), sessionID(-1484951183INIP4127.0.0.3), original IP(127.0.0.3) ports(10526-10527), substitute IP(10.10.1.1) ports(10526-10527), RtpChannel(0/1.1), connection(0x0x7516ca10), sdpOverride(0), me(0x0x7515d710))

                22:26:23.938 MEDIA.MANAGER Connection Fixup 1 DSP Port 10524

                22:26:23.939 MEDIA.MANAGER   Local [::]:10524 : Remote 10.10.1.200:18000

                22:26:23.939 MEDIA.MANAGER Connection Fixup 2 DSP Port 10526

                22:26:23.939 MEDIA.MANAGER   Local 10.10.1.1:10526 : Remote 127.0.0.3:10526

                22:26:23.939 MEDIA.MANAGER connectionFixup : Letting other side fixup connection : entry 10524 sub [::]:10524 remote 10.10.1.200:18000

                22:26:23.939 MEDIA.MANAGER                 : Other side : entry 10526 sub 10.10.1.1:10526 remote 127.0.0.3:10526

                22:26:23.939 MEDIA.MANAGER Connection Fixup 1 DSP Port 10526

                22:26:23.939 MEDIA.MANAGER   Local 10.10.1.1:10526 : Remote 127.0.0.3:10526

                22:26:23.940 MEDIA.MANAGER Connection Fixup 2 DSP Port 10524

                22:26:23.940 MEDIA.MANAGER   Local [::]:10524 : Remote 10.10.1.200:18000

                22:26:23.940 MEDIA.MANAGER connectionFixup : DSP media : Change entry 10526 remote from 127.0.0.3:10526 to 10.10.1.200:18000

                22:26:23.940 MEDIA.MANAGER Setup RTP Channel false for 0/1.1

                22:26:23.940 MEDIA.MANAGER Setup RTP Channel true for 0/1.1

                22:26:23.940 MEDIA.MANAGER Connection Result 1 DSP Port 10526

                22:26:23.940 MEDIA.MANAGER   Local 10.10.1.1:10526 : Remote 10.10.1.200:18000

                22:26:23.941 MEDIA.MANAGER Connection Result 2 Entry not activated

                22:26:23.941 MEDIA.MANAGER connectionFixup success for port 10526 and 10524

                22:26:23.941 MEDIA.MANAGER Marking setup complete for port 10526

                22:26:23.941 MEDIA.MANAGER Marking setup complete for port 10524

                22:26:23.941 MEDIA.MANAGER Connection Fixup 1 DSP Port 10525

                22:26:23.941 MEDIA.MANAGER   Local [::]:10525 : Remote 10.10.1.200:18001

                22:26:23.942 MEDIA.MANAGER Connection Fixup 2 DSP Port 10527

                22:26:23.942 MEDIA.MANAGER   Local 10.10.1.1:10527 : Remote 127.0.0.3:10527

                22:26:23.942 MEDIA.MANAGER connectionFixup : Letting other side fixup connection : entry 10525 sub [::]:10525 remote 10.10.1.200:18001

                22:26:23.942 MEDIA.MANAGER                 : Other side : entry 10527 sub 10.10.1.1:10527 remote 127.0.0.3:10527

                22:26:23.942 MEDIA.MANAGER Connection Fixup 1 DSP Port 10527

                22:26:23.942 MEDIA.MANAGER   Local 10.10.1.1:10527 : Remote 127.0.0.3:10527

                22:26:23.942 MEDIA.MANAGER Connection Fixup 2 DSP Port 10525

                22:26:23.943 MEDIA.MANAGER   Local [::]:10525 : Remote 10.10.1.200:18001

                22:26:23.943 MEDIA.MANAGER connectionFixup : DSP media : Change entry 10527 remote from 127.0.0.3:10527 to 10.10.1.200:18001

                22:26:23.943 MEDIA.MANAGER Connection Result 1 DSP Port 10527

                22:26:23.943 MEDIA.MANAGER   Local 10.10.1.1:10527 : Remote 10.10.1.200:18001

                22:26:23.943 MEDIA.MANAGER Connection Result 2 Entry not activated

                22:26:23.943 MEDIA.MANAGER connectionFixup success for port 10527 and 10525

                22:26:23.944 MEDIA.MANAGER Marking setup complete for port 10527

                22:26:23.944 MEDIA.MANAGER Marking setup complete for port 10525

                22:26:23.944 MEDIA.MANAGER Connected DSP Port 10526

                22:26:23.944 MEDIA.MANAGER   Local 10.10.1.1:10526 : Remote 10.10.1.200:18000

                22:26:23.944 MEDIA.MANAGER Connected associations Entry not activated

                22:26:23.944 SB.CCM connect: Connected RTP/TDM via MCM

                22:26:23.944 MEDIA.MANAGER Setup RTP Channel true for 0/1.1

                22:26:23.945 SB.CCM setupRtpChannel, source 2, silence 0

                22:26:23.945 SB.CCM setupRtpChannel: setup using media connection

                22:26:23.945 SB.CCM Looking up source address for destination 10.10.1.200

                22:26:23.945 SB.CCM setupRtpChannel: Source IP addr = 10.10.1.1, port = 10526

                22:26:23.945 SB.CCM setupRtpChannel: Target IP addr = 10.10.1.200, port = 18000

                22:26:23.946 SB.CCM setupRtpChannel: Undo of previous operation not required

                22:26:23.946 SB.CCM getFinalCodec: PCMU

                22:26:23.946 SB.CCM getFinalCodec: PCMU

                22:26:23.947 SB.CCM setupRtpChannel: Configuring RTP Channel 0/1.1 to Src 10.10.1.1:10526 Trg 10.10.1.200:18000 via PCMU Rx PCMU

                22:26:23.947 SB.CCM setupRtpChannel: fpp=2 echo=on dtmf=101/101 dscp=46 vad=off isOffer no

                22:26:23.947 SB.CCM setupRtpChannel: Starting RTP Channel

                22:26:23.948 RTP.CHANNEL Channel 0/1.1 session statistics cleared.

                22:26:23.948 RTP.CHANNEL Channel 0/1.1 started successfully.

                22:26:23.948 SB.CCM firewallConnectCall: Set up firewall from media connections

                22:26:23.948 SB.CCM sdpFirewall: invoked with offer - 10.10.1.1:10526, answer - 10.10.1.200:18000

                22:26:23.948 SB.CCM sdpFirewall: IPv4 firewall is not enabled, no action taken

                22:26:23.949 SB.CCM connect: TDM streams: port(SipTrunk 0/1.1) to port(FxsPhone 0/1)

                22:26:23.949 SB.CALL 213 Connecting           ConnectResponse sent from T03 to 3000

                22:26:23.949 RTP.CHANNEL fxs 0/1 - Dsp 0/1.1 - RTP: starting

                22:26:23.950 SA.3000 Ca:0 Connecting           rcvd: connectResponse from SB

                22:26:23.950 SA.3000 Ca:0 Connecting           State change      >> Connecting->Connected (CAS_Connected)

                22:26:23.950 SA.3000 Ca:0 Connected            sent: AcctPhoneMgr_cachg(CAS_Connected) to PM

                22:26:23.950 PM.0:1 Connected            Processed CACHG:Connected

                22:26:23.950 PM.0:1 State change      >> Connected->Connected

                22:26:23.950 PM.0:1 Connected            sent: finalizeConnect to SA

                22:26:23.951 SA.3000 Ca:0 Connected            sent: AcctPhoneMgr_info to PM

                22:26:23.951 PM.0:1 ERROR! APM_Info ignored

                22:26:23.951 SA.3000 Ca:0 Connected            rcvd: AcctPhoneMgr_finalizeConnect from PM

                22:26:23.951 SA.3000 Ca:0 Connected            sent: finalizeConnect to SB

                22:26:23.951 SB.CALL 213 Connecting           Called the finalizeConnect routine

                22:26:23.951 SB.CCM finalizeConnect: connection already finalized(2)

                22:26:23.951 SB.CALL 213 State change      >> Connecting->Connected

                22:26:27.769 TM.T03 01 SipTM_Connected      rcvd SIP call-leg request: BYE

                22:26:27.769 TM.T03 01 SipTM_Connected      call-leg -> Disconnected

                22:26:27.769 TM.T03 01 SipTM_Connected      CallLegStateChanged to Disconnected - TM change to closing state.

                22:26:27.769 TM.T03 01 SipTM_Connected      State change      >> SipTM_Connected->SipTM_Closing

                22:26:27.769 TM.T03 01 SipTM_Closing        sent: TA->Clear

                22:26:27.771 TM.T03 01 SipTM_Closing        call-leg -> Terminated

                22:26:27.771 TA.T03 01 TAConnected          rcvd: clear from TM

                22:26:27.771 TA.T03 01 State change      >> TAConnected->TATrunkClearing (TAS_Clearing)

                22:26:27.772 TM.T03 01 SipTM_Closing        tachg -> TATrunkClearing

                22:26:27.772 TM.T03 01 SipTM_Closing        State change      >> SipTM_Closing->SipTM_Terminated

                22:26:27.772 TM.T03 01 SipTM_Terminated     sent: TA->AppearanceOff

                22:26:27.772 TM.T03 01 SipTM_Terminated     State change      >> SipTM_Terminated->SipTM_Idle

                22:26:27.772 SB.CALL 213 Connected            Called the clearCall routine

                22:26:27.773 SB.CALL 213 Connected            ClearCall sent from T03 to 3000

                22:26:27.773 SB.CALL 213 State change      >> Connected->Clearing

                22:26:27.773 TA.T03 01 TATrunkClearing      rcvd: appearance off from TM

                22:26:27.773 TA.T03 01 State change      >> TATrunkClearing->TAClearingComplete (TAS_Clearing)

                22:26:27.773 TA.T03 01 TATrunkClearing      Processing an appearance OFF

                22:26:27.773 SA.3000 Ca:0 Connected            rcvd: clearCall from SB

                22:26:27.774 SA.3000 Ca:0 Connected            sent: clearResponse(pass) to SB

                22:26:27.774 SA.3000 Ca:0 Connected            State change      >> Connected->Idle (CAS_Idle)

                22:26:27.774 SA.3000 Ca:0 Idle                 sent: AcctPhoneMgr_cachg(CAS_Idle) to PM

                22:26:27.774 PM.0:1 Connected            Processed CACHG:IDLE on Primary CA

                22:26:27.774 PM.0:1 State change      >> Connected->Clearing Quiet

                22:26:27.775 SB.CALL 213 Clearing             Called the clearResponse routine

                22:26:27.775 SB.CALL 213 State change      >> Clearing->CallIdlePending

                22:26:27.775 SB.CCM disconnect:

                22:26:27.775 SB.CCM  :  Call Struct 0x0x75150a10 :   Call-ID = 213

                22:26:27.775 SB.CCM  :  Org Acct = T03    Dst Acct = 3000

                22:26:27.776 SB.CCM  :  Org Port ID = SipTrunk 0/1.1   Dst Port ID = FxsPhone 0/1

                22:26:27.776 SB.CCM  :  RTP Channel = 0/1.1

                22:26:27.776 SB.CCM disconnect: Call Connection Type is RTP_TO_TDM

                22:26:27.776 SB.CCM disconnect: Stopping RTP Channel 0/1.1

                22:26:27.776 RTP.CHANNEL Channel 0/1.1 stopped successfully.

                22:26:27.776 SB.CCM disconnect: Disconnecting TDM streams

                22:26:27.777 SB.CCM release:

                22:26:27.777 SB.CCM  :  Call Struct 0x0x75150a10 :   Call-ID = 213

                22:26:27.777 SB.CCM  :  Org Acct = T03    Dst Acct = 3000

                22:26:27.777 SB.CCM  :  Org Port ID = SipTrunk 0/1.1   Dst Port ID = FxsPhone 0/1

                22:26:27.778 SB.CCM  :  RTP Channel = 0/1.1

                22:26:27.778 SB.CCM release: Call Connection Type is RTP_TO_TDM

                22:26:27.778 SB.CCM release: Releasing RTP Channel 0/1.1

                22:26:27.778 RTP.CHANNEL Channel 0/1.1 released successfully.

                22:26:27.780 SB.CALL 213 CallIdlePending      ClearResponse sent from 3000 to T03

                22:26:27.780 SA.3000 Ca:0 Idle                 rcvd: AcctPhoneMgr_appearance(OFF) from PM

                22:26:27.780 SA.3000 Ca:0 Idle                 sent: AcctPhoneMgr_cachg(CAS_Idle) to PM

                22:26:27.780 PM.0:1 Clearing Quiet       Dropped CACHG w/Call State not RINGING

                22:26:27.781 TA.T03 01 TAClearingComplete   clearResponse event accepted

                22:26:27.781 TA.T03 01 TAClearingComplete   Clear Local Variables

                22:26:27.781 TA.T03 01 State change      >> TAClearingComplete->TAIdle (TAS_Idle)

                22:26:27.781 TM.T03 01 SipTM_Idle           tachg -> TAIdle

                22:26:27.781 RTP.CHANNEL fxs 0/1 - Dsp 0/1.1 - RTP: stopping

                22:26:27.782 RTP.CHANNEL fxs 0/1 - Dsp 0/1.1 - RTP: releasing RTP resource

                22:26:27.782 RTP.CHANNEL fxs 0/1 - Dsp 0/1.1 - RTP: releasing

                22:26:27 SB.CallStructObserver 213 Finalized

                22:26:28.678 PM.0:1 Clearing Quiet       FXS Port OffHook

                2017.01.20 22:26:28 SMDR 213        01/20/2017 22:26:18      0.1 0    E  00/01 1000            1000            00/01                 3000            0 N 

                22:26:29.776 PM.0:1 Clearing Quiet       Processed Clearing Timeout

                22:26:29.776 PM.0:1 State change      >> Clearing Quiet->Requesting Dialtone

                22:26:29.776 SA.3000 Ca:0 Idle                 rcvd: AcctPhoneMgr_appearance(ON) from PM

                22:26:29.776 SA.3000 Ca:0 Idle                 State change      >> Idle->DigitGathering (CAS_ReqDigits)

                22:26:29.777 SA.3000 Ca:0 DigitGathering       sent: AcctPhoneMgr_cachg(CAS_ReqDigits) to PM

                22:26:29.777 PM.0:1 Requesting Dialtone  CACHG:ReqDigits on primary CA

                22:26:29.777 PM.0:1 State change      >> Requesting Dialtone->SendingDigits

                22:26:29.778 TONESERVICES.EVENTS fxs 0/1 - empty - Tone Detection: Request resource

                22:26:29.778 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Tone Detection: DSP channel allocated for the resource

                22:26:29.778 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Tone Detection: constructed

                22:26:29.778 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Tone Detection: starting

                22:26:29.779 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Tone Detection: TDM map

                22:26:30.279 TONESERVICES.EVENTS fxs 0/1 - empty - DialTone Generation: Request resource

                22:26:30.280 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - DialTone Generation: DSP channel allocated for the resource

                22:26:30.280 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - DialTone Generation: constructed

                22:26:30.280 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - DialTone Generation: starting

                22:26:30.280 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - DialTone Generation: TDM map

                22:26:30.371 PM.0:1 SendingDigits        Processed ONHOOK

                22:26:30.371 PM.0:1 State change      >> SendingDigits->Idle

                22:26:30.372 SA.3000 Ca:0 DigitGathering       rcvd: AcctPhoneMgr_appearance(OFF) from PM

                22:26:30.372 SA.3000 Ca:0 DigitGathering       State change      >> DigitGathering->Idle (CAS_Idle)

                22:26:30.372 SA.3000 Ca:0 Idle                 sent: AcctPhoneMgr_cachg(CAS_Idle) to PM

                22:26:30.372 PM.0:1 Idle                 Dropped CACHG w/Call State not RINGING

                22:26:30.373 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - DialTone Generation: stopping

                22:26:30.373 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - DialTone Generation: TDM unmap

                22:26:30.373 RTP.CHANNEL fxs 0/1 - Dsp 0/1.1 - DialTone Generation: releasing RTP resource

                22:26:30.373 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - DialTone Generation: release

                22:26:30.374 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Tone Detection: stopping

                22:26:30.374 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Tone Detection: TDM unmap

                22:26:30.374 RTP.CHANNEL fxs 0/1 - Dsp 0/1.1 - Tone Detection: releasing RTP resource

                22:26:30.374 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Tone Detection: release

                22:26:30.374 SA.3000 rcvd: AcctPhoneMgr_COSOverride from PM

                22:26:34.364 PM.0:1 Idle                 Processed OFFHOOK

                22:26:34.364 PM.0:1 State change      >> Idle->Requesting Dialtone

                22:26:34.364 SA.3000 Ca:0 Idle                 rcvd: AcctPhoneMgr_appearance(ON) from PM

                22:26:34.365 SA.3000 Ca:0 Idle                 State change      >> Idle->DigitGathering (CAS_ReqDigits)

                22:26:34.365 SA.3000 Ca:0 DigitGathering       sent: AcctPhoneMgr_cachg(CAS_ReqDigits) to PM

                22:26:34.365 PM.0:1 Requesting Dialtone  CACHG:ReqDigits on primary CA

                22:26:34.365 PM.0:1 State change      >> Requesting Dialtone->SendingDigits

                22:26:34.366 TONESERVICES.EVENTS fxs 0/1 - empty - Tone Detection: Request resource

                22:26:34.366 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Tone Detection: DSP channel allocated for the resource

                22:26:34.366 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Tone Detection: constructed

                22:26:34.367 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Tone Detection: starting

                22:26:34.367 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Tone Detection: TDM map

                22:26:34.867 TONESERVICES.EVENTS fxs 0/1 - empty - DialTone Generation: Request resource

                22:26:34.868 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - DialTone Generation: DSP channel allocated for the resource

                22:26:34.868 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - DialTone Generation: constructed

                22:26:34.868 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - DialTone Generation: starting

                22:26:34.868 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - DialTone Generation: TDM map

                22:26:37.299 PM.0:1 SendingDigits        Processed ONHOOK

                22:26:37.299 PM.0:1 State change      >> SendingDigits->Idle

                22:26:37.300 SA.3000 Ca:0 DigitGathering       rcvd: AcctPhoneMgr_appearance(OFF) from PM

                22:26:37.301 SA.3000 Ca:0 DigitGathering       State change      >> DigitGathering->Idle (CAS_Idle)

                22:26:37.301 SA.3000 Ca:0 Idle                 sent: AcctPhoneMgr_cachg(CAS_Idle) to PM

                22:26:37.301 PM.0:1 Idle                 Dropped CACHG w/Call State not RINGING

                22:26:37.301 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - DialTone Generation: stopping

                22:26:37.302 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - DialTone Generation: TDM unmap

                22:26:37.302 RTP.CHANNEL fxs 0/1 - Dsp 0/1.1 - DialTone Generation: releasing RTP resource

                22:26:37.302 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - DialTone Generation: release

                22:26:37.302 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Tone Detection: stopping

                22:26:37.302 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Tone Detection: TDM unmap

                22:26:37.303 RTP.CHANNEL fxs 0/1 - Dsp 0/1.1 - Tone Detection: releasing RTP resource

                22:26:37.303 TONESERVICES.EVENTS fxs 0/1 - Dsp 0/1.1 - Tone Detection: release

                22:26:37.303 SA.3000 rcvd: AcctPhoneMgr_COSOverride from PM

                ADTRAN-LAB#

                 

                 

                 

                 

                 

                =============================SIP SET DEBUG ASTERISK ==========================

                 

                 

                 

                 

                 

                 

                <--- SIP read from UDP:10.10.1.11:5060 --->

                INVITE sip:9549054211@10.10.1.200:5060 SIP/2.0

                Via: SIP/2.0/UDP 10.10.1.11:5060;branch=z9hG4bK2302469184

                From: "1000" <sip:1000@10.10.1.200:5060>;tag=3049795389

                To: <sip:9549054211@10.10.1.200:5060>

                Call-ID: 0_2611340150@10.10.1.11

                CSeq: 1 INVITE

                Contact: <sip:1000@10.10.1.11:5060>

                Content-Type: application/sdp

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

                Max-Forwards: 70

                User-Agent: Yealink SIP-T46G 28.80.0.130

                Allow-Events: talk,hold,conference,refer,check-sync

                Supported: replaces

                Content-Length: 278

                 

                 

                v=0

                o=- 20020 20020 IN IP4 10.10.1.11

                s=SDP data

                c=IN IP4 10.10.1.11

                t=0 0

                m=audio 11832 RTP/AVP 18 0 8 101

                a=rtpmap:18 G729/8000

                a=fmtp:18 annexb=no

                a=rtpmap:0 PCMU/8000

                a=rtpmap:8 PCMA/8000

                a=ptime:20

                a=sendrecv

                a=rtpmap:101 telephone-event/8000

                a=fmtp:101 0-15

                <------------->

                --- (14 headers 14 lines) ---

                Sending to 10.10.1.11:5060 (NAT)

                Using INVITE request as basis request - 0_2611340150@10.10.1.11

                Found peer '1000' for '1000' from 10.10.1.11:5060

                 

                 

                <--- Reliably Transmitting (NAT) to 10.10.1.11:5060 --->

                SIP/2.0 401 Unauthorized

                Via: SIP/2.0/UDP 10.10.1.11:5060;branch=z9hG4bK2302469184;received=10.10.1.11;rport=5060

                From: "1000" <sip:1000@10.10.1.200:5060>;tag=3049795389

                To: <sip:9549054211@10.10.1.200:5060>;tag=as05661815

                Call-ID: 0_2611340150@10.10.1.11

                CSeq: 1 INVITE

                Server: ClIeNt-PbX

                Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

                Supported: replaces, timer

                WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6861faf6"

                Content-Length: 0

                 

                 

                 

                 

                <------------>

                Scheduling destruction of SIP dialog '0_2611340150@10.10.1.11' in 6400 ms (Method: INVITE)

                 

                 

                <--- SIP read from UDP:10.10.1.11:5060 --->

                ACK sip:9549054211@10.10.1.200:5060 SIP/2.0

                Via: SIP/2.0/UDP 10.10.1.11:5060;branch=z9hG4bK2302469184

                From: "1000" <sip:1000@10.10.1.200:5060>;tag=3049795389

                To: <sip:9549054211@10.10.1.200:5060>;tag=as05661815

                Call-ID: 0_2611340150@10.10.1.11

                CSeq: 1 ACK

                Content-Length: 0

                 

                 

                <------------->

                --- (7 headers 0 lines) ---

                 

                 

                <--- SIP read from UDP:10.10.1.11:5060 --->

                INVITE sip:9549054211@10.10.1.200:5060 SIP/2.0

                Via: SIP/2.0/UDP 10.10.1.11:5060;branch=z9hG4bK1627442362

                From: "1000" <sip:1000@10.10.1.200:5060>;tag=3049795389

                To: <sip:9549054211@10.10.1.200:5060>

                Call-ID: 0_2611340150@10.10.1.11

                CSeq: 2 INVITE

                Contact: <sip:1000@10.10.1.11:5060>

                Authorization: Digest username="1000", realm="asterisk", nonce="6861faf6", uri="sip:9549054211@10.10.1.200:5060", response="5aee0aa23f61483e8554b7f831014698", algorithm=MD5

                Content-Type: application/sdp

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

                Max-Forwards: 70

                User-Agent: Yealink SIP-T46G 28.80.0.130

                Allow-Events: talk,hold,conference,refer,check-sync

                Supported: replaces

                Content-Length: 278

                 

                 

                v=0

                o=- 20020 20020 IN IP4 10.10.1.11

                s=SDP data

                c=IN IP4 10.10.1.11

                t=0 0

                m=audio 11832 RTP/AVP 18 0 8 101

                a=rtpmap:18 G729/8000

                a=fmtp:18 annexb=no

                a=rtpmap:0 PCMU/8000

                a=rtpmap:8 PCMA/8000

                a=ptime:20

                a=sendrecv

                a=rtpmap:101 telephone-event/8000

                a=fmtp:101 0-15

                <------------->

                --- (15 headers 14 lines) ---

                Sending to 10.10.1.11:5060 (NAT)

                Using INVITE request as basis request - 0_2611340150@10.10.1.11

                Found peer '1000' for '1000' from 10.10.1.11:5060

                  == Using SIP RTP CoS mark 5

                Found RTP audio format 18

                Found RTP audio format 0

                Found RTP audio format 8

                Found RTP audio format 101

                Found audio description format G729 for ID 18

                Found audio description format PCMU for ID 0

                Found audio description format PCMA for ID 8

                Found audio description format telephone-event for ID 101

                Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)

                Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

                Peer audio RTP is at port 10.10.1.11:11832

                Looking for 9549054211 in default (domain 10.10.1.200)

                list_route: hop: <sip:1000@10.10.1.11:5060>

                 

                 

                <--- Transmitting (NAT) to 10.10.1.11:5060 --->

                SIP/2.0 100 Trying

                Via: SIP/2.0/UDP 10.10.1.11:5060;branch=z9hG4bK1627442362;received=10.10.1.11;rport=5060

                From: "1000" <sip:1000@10.10.1.200:5060>;tag=3049795389

                To: <sip:9549054211@10.10.1.200:5060>

                Call-ID: 0_2611340150@10.10.1.11

                CSeq: 2 INVITE

                Server: ClIeNt-PbX

                Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

                Supported: replaces, timer

                Contact: <sip:9549054211@10.10.1.200:5060>

                Content-Length: 0

                 

                 

                 

                 

                <------------>

                    -- Executing [9549054211@default:1] Dial("SIP/1000-00000018", "SIP/9549054211@ADTRAN-LAB") in new stack

                  == Using SIP RTP CoS mark 5

                We think we can do text

                Audio is at 18674

                Adding codec 0x4 (ulaw) to SDP

                Adding codec 0x2 (gsm) to SDP

                Adding codec 0x8 (alaw) to SDP

                Adding codec 0x10 (g726aal2) to SDP

                Adding codec 0x20 (adpcm) to SDP

                Adding codec 0x40 (slin) to SDP

                Adding codec 0x80 (lpc10) to SDP

                Adding codec 0x200 (speex) to SDP

                Adding codec 0x800 (g726) to SDP

                Adding codec 0x1000 (g722) to SDP

                Adding codec 0x8000 (slin16) to SDP

                Adding codec 0x200000000 (speex16) to SDP

                Adding codec 0x800000000000 (testlaw) to SDP

                Adding non-codec 0x1 (telephone-event) to SDP

                Reliably Transmitting (NAT) to 10.10.1.1:5060:

                INVITE sip:9549054211@10.10.1.1 SIP/2.0

                Via: SIP/2.0/UDP 10.10.1.200:5060;branch=z9hG4bK5892066f;rport

                Max-Forwards: 70

                From: "1000" <sip:1000@10.10.1.200>;tag=as3e38e9ea

                To: <sip:9549054211@10.10.1.1>

                Contact: <sip:1000@10.10.1.200:5060>

                Call-ID: 3df669b9702b004e5dffa5c912d0254b@10.10.1.200:5060

                CSeq: 102 INVITE

                User-Agent: ClIeNt-PbX

                Date: Fri, 20 Jan 2017 22:43:20 GMT

                Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

                Supported: replaces, timer

                Content-Type: application/sdp

                Content-Length: 528

                 

                 

                v=0

                o=root 53741299 53741299 IN IP4 10.10.1.200

                s=Asterisk PBX 1.8.32.3

                c=IN IP4 10.10.1.200

                t=0 0

                m=audio 18674 RTP/AVP 0 3 8 112 5 10 7 110 111 9 118 117 101

                a=rtpmap:0 PCMU/8000

                a=rtpmap:3 GSM/8000

                a=rtpmap:8 PCMA/8000

                a=rtpmap:112 AAL2-G726-32/8000

                a=rtpmap:5 DVI4/8000

                a=rtpmap:10 L16/8000

                a=rtpmap:7 LPC/8000

                a=rtpmap:110 speex/8000

                a=rtpmap:111 G726-32/8000

                a=rtpmap:9 G722/8000

                a=rtpmap:118 L16/16000

                a=rtpmap:117 speex/16000

                a=rtpmap:101 telephone-event/8000

                a=fmtp:101 0-16

                a=ptime:20

                a=sendrecv

                 

                 

                ---

                    -- Called SIP/9549054211@ADTRAN-LAB

                 

                 

                <--- SIP read from UDP:10.10.1.1:5060 --->

                SIP/2.0 100 Trying

                From: "1000"<sip:1000@10.10.1.200>;tag=as3e38e9ea

                To: <sip:9549054211@10.10.1.1>

                Call-ID: 3df669b9702b004e5dffa5c912d0254b@10.10.1.200:5060

                CSeq: 102 INVITE

                Via: SIP/2.0/UDP 10.10.1.200:5060;rport=5060;branch=z9hG4bK5892066f

                Contact: <sip:9549054211@10.10.1.1:5060;transport=UDP>

                Supported: 100rel,replaces

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

                User-Agent: ADTRAN_Total_Access_908e_3rd_Gen/R10.9.0

                Content-Length: 0

                 

                 

                <------------->

                --- (11 headers 0 lines) ---

                 

                 

                <--- SIP read from UDP:10.10.1.1:5060 --->

                SIP/2.0 180 Ringing

                From: "1000"<sip:1000@10.10.1.200>;tag=as3e38e9ea

                To: <sip:9549054211@10.10.1.1>;tag=62840680-7f000001-13c4-d8c9c-9b775e5-d8c9c

                Call-ID: 3df669b9702b004e5dffa5c912d0254b@10.10.1.200:5060

                CSeq: 102 INVITE

                Via: SIP/2.0/UDP 10.10.1.200:5060;rport=5060;branch=z9hG4bK5892066f

                Contact: <sip:9549054211@10.10.1.1:5060;transport=UDP>

                Supported: 100rel,replaces

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

                User-Agent: ADTRAN_Total_Access_908e_3rd_Gen/R10.9.0

                Content-Length: 0

                 

                 

                <------------->

                --- (11 headers 0 lines) ---

                list_route: hop: <sip:9549054211@10.10.1.1:5060;transport=UDP>

                    -- SIP/ADTRAN-LAB-00000019 is ringing

                 

                 

                <--- Transmitting (NAT) to 10.10.1.11:5060 --->

                SIP/2.0 180 Ringing

                Via: SIP/2.0/UDP 10.10.1.11:5060;branch=z9hG4bK1627442362;received=10.10.1.11;rport=5060

                From: "1000" <sip:1000@10.10.1.200:5060>;tag=3049795389

                To: <sip:9549054211@10.10.1.200:5060>;tag=as521f660f

                Call-ID: 0_2611340150@10.10.1.11

                CSeq: 2 INVITE

                Server: ClIeNt-PbX

                Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

                Supported: replaces, timer

                Contact: <sip:9549054211@10.10.1.200:5060>

                Content-Length: 0

                 

                 

                 

                 

                <------------>

                 

                 

                <--- SIP read from UDP:10.10.1.1:5060 --->

                SIP/2.0 200 OK

                From: "1000"<sip:1000@10.10.1.200>;tag=as3e38e9ea

                To: <sip:9549054211@10.10.1.1>;tag=62840680-7f000001-13c4-d8c9c-9b775e5-d8c9c

                Call-ID: 3df669b9702b004e5dffa5c912d0254b@10.10.1.200:5060

                CSeq: 102 INVITE

                Via: SIP/2.0/UDP 10.10.1.200:5060;rport=5060;branch=z9hG4bK5892066f

                Contact: <sip:9549054211@10.10.1.1:5060;transport=UDP>

                Supported: 100rel,replaces

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

                User-Agent: ADTRAN_Total_Access_908e_3rd_Gen/R10.9.0

                Content-Type: application/sdp

                Content-Length: 202

                 

                 

                v=0

                o=- 1484951427 1 IN IP4 10.10.1.1

                s=-

                c=IN IP4 10.10.1.1

                t=0 0

                m=audio 10538 RTP/AVP 0 101

                a=silenceSupp:off - - - -

                a=rtpmap:0 PCMU/8000

                a=rtpmap:101 telephone-event/8000

                a=fmtp:101 0-15

                <------------->

                --- (12 headers 10 lines) ---

                Found RTP audio format 0

                Found RTP audio format 101

                Found audio description format PCMU for ID 0

                Found audio description format telephone-event for ID 101

                Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)

                Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

                Peer audio RTP is at port 10.10.1.1:10538

                list_route: hop: <sip:9549054211@10.10.1.1:5060;transport=UDP>

                set_destination: Parsing <sip:9549054211@10.10.1.1:5060;transport=UDP> for address/port to send to

                set_destination: set destination to 10.10.1.1:5060

                Transmitting (NAT) to 10.10.1.1:5060:

                ACK sip:9549054211@10.10.1.1:5060;transport=UDP SIP/2.0

                Via: SIP/2.0/UDP 10.10.1.200:5060;branch=z9hG4bK671e2b08;rport

                Max-Forwards: 70

                From: "1000" <sip:1000@10.10.1.200>;tag=as3e38e9ea

                To: <sip:9549054211@10.10.1.1>;tag=62840680-7f000001-13c4-d8c9c-9b775e5-d8c9c

                Contact: <sip:1000@10.10.1.200:5060>

                Call-ID: 3df669b9702b004e5dffa5c912d0254b@10.10.1.200:5060

                CSeq: 102 ACK

                User-Agent: ClIeNt-PbX

                Content-Length: 0

                 

                 

                 

                 

                ---

                    -- SIP/ADTRAN-LAB-00000019 answered SIP/1000-00000018

                Audio is at 13180

                Adding codec 0x4 (ulaw) to SDP

                Adding codec 0x8 (alaw) to SDP

                Adding codec 0x100 (g729) to SDP

                Adding non-codec 0x1 (telephone-event) to SDP

                 

                 

                <--- Reliably Transmitting (NAT) to 10.10.1.11:5060 --->

                SIP/2.0 200 OK

                Via: SIP/2.0/UDP 10.10.1.11:5060;branch=z9hG4bK1627442362;received=10.10.1.11;rport=5060

                From: "1000" <sip:1000@10.10.1.200:5060>;tag=3049795389

                To: <sip:9549054211@10.10.1.200:5060>;tag=as521f660f

                Call-ID: 0_2611340150@10.10.1.11

                CSeq: 2 INVITE

                Server: ClIeNt-PbX

                Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

                Supported: replaces, timer

                Contact: <sip:9549054211@10.10.1.200:5060>

                Content-Type: application/sdp

                Content-Length: 306

                 

                 

                v=0

                o=root 1813601651 1813601651 IN IP4 10.10.1.200

                s=Asterisk PBX 1.8.32.3

                c=IN IP4 10.10.1.200

                t=0 0

                m=audio 13180 RTP/AVP 0 8 18 101

                a=rtpmap:0 PCMU/8000

                a=rtpmap:8 PCMA/8000

                a=rtpmap:18 G729/8000

                a=fmtp:18 annexb=no

                a=rtpmap:101 telephone-event/8000

                a=fmtp:101 0-16

                a=ptime:20

                a=sendrecv

                 

                 

                <------------>

                    -- Locally bridging SIP/1000-00000018 and SIP/ADTRAN-LAB-00000019

                 

                 

                <--- SIP read from UDP:10.10.1.11:5060 --->

                ACK sip:9549054211@10.10.1.200:5060 SIP/2.0

                Via: SIP/2.0/UDP 10.10.1.11:5060;branch=z9hG4bK2758645936

                From: "1000" <sip:1000@10.10.1.200:5060>;tag=3049795389

                To: <sip:9549054211@10.10.1.200:5060>;tag=as521f660f

                Call-ID: 0_2611340150@10.10.1.11

                CSeq: 2 ACK

                Contact: <sip:1000@10.10.1.11:5060>

                Max-Forwards: 70

                User-Agent: Yealink SIP-T46G 28.80.0.130

                Content-Length: 0

                 

                 

                <------------->

                --- (10 headers 0 lines) ---

                 

                 

                <--- SIP read from UDP:10.10.1.1:5060 --->

                BYE sip:1000@10.10.1.200:5060;transport=UDP SIP/2.0

                From: <sip:9549054211@10.10.1.1>;tag=62840680-7f000001-13c4-d8c9c-9b775e5-d8c9c

                To: "1000"<sip:1000@10.10.1.200>;tag=as3e38e9ea

                Call-ID: 3df669b9702b004e5dffa5c912d0254b@10.10.1.200:5060

                CSeq: 1 BYE

                Via: SIP/2.0/UDP 10.10.1.1:5060;branch=z9hG4bK-d8ca4-34ed6346-6fcb2c3

                Max-Forwards: 70

                Supported: 100rel,replaces

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

                User-Agent: ADTRAN_Total_Access_908e_3rd_Gen/R10.9.0

                Content-Length: 0

                 

                 

                <------------->

                --- (11 headers 0 lines) ---

                Sending to 10.10.1.1:5060 (NAT)

                Scheduling destruction of SIP dialog '3df669b9702b004e5dffa5c912d0254b@10.10.1.200:5060' in 32000 ms (Method: BYE)

                 

                 

                <--- Transmitting (NAT) to 10.10.1.1:5060 --->

                SIP/2.0 200 OK

                Via: SIP/2.0/UDP 10.10.1.1:5060;branch=z9hG4bK-d8ca4-34ed6346-6fcb2c3;received=10.10.1.1;rport=5060

                From: <sip:9549054211@10.10.1.1>;tag=62840680-7f000001-13c4-d8c9c-9b775e5-d8c9c

                To: "1000"<sip:1000@10.10.1.200>;tag=as3e38e9ea

                Call-ID: 3df669b9702b004e5dffa5c912d0254b@10.10.1.200:5060

                CSeq: 1 BYE

                Server: ClIeNt-PbX

                Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

                Supported: replaces, timer

                Content-Length: 0

                 

                 

                 

                 

                <------------>

                  == Spawn extension (default, 9549054211, 1) exited non-zero on 'SIP/1000-00000018'

                Scheduling destruction of SIP dialog '0_2611340150@10.10.1.11' in 6400 ms (Method: ACK)

                set_destination: Parsing <sip:1000@10.10.1.11:5060> for address/port to send to

                set_destination: set destination to 10.10.1.11:5060

                Reliably Transmitting (NAT) to 10.10.1.11:5060:

                BYE sip:1000@10.10.1.11:5060 SIP/2.0

                Via: SIP/2.0/UDP 10.10.1.200:5060;branch=z9hG4bK7ba53aaf;rport

                Max-Forwards: 70

                From: <sip:9549054211@10.10.1.200:5060>;tag=as521f660f

                To: "1000" <sip:1000@10.10.1.200:5060>;tag=3049795389

                Call-ID: 0_2611340150@10.10.1.11

                CSeq: 102 BYE

                User-Agent: ClIeNt-PbX

                Proxy-Authorization: Digest username="1000", realm="asterisk", algorithm=MD5, uri="sip:10.10.1.200", nonce="", response="fdd1f90bed78dbee5476d1a72b12fc84"

                X-Asterisk-HangupCause: Normal Clearing

                X-Asterisk-HangupCauseCode: 16

                Content-Length: 0