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.
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
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?
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
Wilmer, I'd suggest opening a ticket with support. If the two devices are truly on the same subnet with no NAT involved, then there shouldn't be any real chance for an audio issue relating to firewall. Support can help you with the debugs and potentially getting a DSP capture. Thanks
Hello Jay, thank you for you reply. The issue was resolved, I stop iptables within the server and now all is working properly. Configuration posted for the Adtran and Asterisk are OK. Regards