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wernerg · ‎12-30-2016

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.

Anonymous · ‎01-13-2017

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

jayh · ‎01-19-2017

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?

vasquezwilmer · ‎01-20-2017

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

22:26:23.947 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

Anonymous · ‎01-23-2017

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

vasquezwilmer · ‎01-23-2017

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

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

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

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route voice traffic from an FXS port to SIP trunks

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

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

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