Call and hang up before respond - phone rings forever

Asterisk Asterisk Support
#1

If someone call and hang up before anyone respond then all the phones rings forever (until someone open and simply close the phone - he will only hear busy tone). It is very annoying.

I’m using Asterisk 1.6.2.16.1.

If someone have any ideas please help.

Extensions.conf is very simple:

[size=85][local]
exten => _X.,1,Dial(SIP/${EXTEN}@gwfxo)

[default]
exten => _X.,1,Hangup

[gwfxo]
exten => _[1-4],1,Answer()
exten => _[1-4],2,Goto(coada,1,1)

[coada]
exten => 1,1,Queue(callcenter,3600000,Tt)
exten => 1,2,Wait(1)
exten => 1,3,4,Hangup

[gwfxs]
exten => _[12]XX,1,Dial(SIP/${EXTEN},3600000,Tt)
exten => _3X.,1,Answer()
exten => _3X.,2,Playback(beep)
exten => _3X.,3,ConfBridge(${EXTEN})

exten => _0.,1,Dial(SIP/${EXTEN}@gwfxo,3600000,Tt)

exten => _*,1,Pickup(1@coada)[/size]

In Sip.conf all extensions are like:

[size=85][211]
type = friend
host = dynamic
nat = no
disallow = all
allow = alaw
allow = ulaw
username = 211
fromuser = 211
secret = Iui211
context = gwfxs
qualify = 1000
canreinvite=yes
callwaiting = yes
insecure=port,invite
[/size]

#2

I seem to remember that there were problems if a call was cancelled before the 100 Trying response, but I thought that had been fixed. A SIP trace is needed.

#3

This line isn’t right…

The second parameter is the options, not a timeout.

why timeouts of 41.666666 days? Might as well leave it off.

Should provide some CLI output for a call.

#4

== Using SIP RTP CoS mark 5
– Executing [3@gwfxo:1] Answer(“SIP/gwfxo-00000117”, “”) in new stack
– Executing [3@gwfxo:2] GotoIf(“SIP/gwfxo-00000117”, “0?10:3”) in new stack
– Goto (gwfxo,3,3)
– Executing [3@gwfxo:3] Goto(“SIP/gwfxo-00000117”, “coada,1,1”) in new stack
– Goto (coada,1,1)
– Executing [1@coada:1] Queue(“SIP/gwfxo-00000117”, “callcenter,3600000,Tt”) in new stack
– Started music on hold, class ‘default’, on SIP/gwfxo-00000117
== Using SIP RTP CoS mark 5
– SIP/112-00000118 is ringing
– Nobody picked up in 15000 ms
== Using SIP RTP CoS mark 5
– SIP/113-00000119 is ringing
– SIP/113-00000119 answered SIP/gwfxo-00000117
– Stopped music on hold on SIP/gwfxo-00000117
– Native bridging SIP/gwfxo-00000117 and SIP/113-00000119
== Spawn extension (coada, 1, 1) exited non-zero on ‘SIP/gwfxo-00000117’

And I use another application that listening the asterisk on port 5038, maybe the log is useful:

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/114
PeerStatus: Registered
Address: 10.77.4.12
Port: 5060

Event: QueueMemberStatus
Privilege: agent,all
Queue: callcenter
Location: SIP/114
MemberName: SIP/114
Membership: static
Penalty: 0
CallsTaken: 1
LastCall: 1298477459
Status: 1
Paused: 0

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/113
PeerStatus: Registered
Address: 10.77.4.12
Port: 5060

Event: QueueMemberStatus
Privilege: agent,all
Queue: callcenter
Location: SIP/113
MemberName: SIP/113
Membership: static
Penalty: 0
CallsTaken: 3
LastCall: 1298563263
Status: 1
Paused: 0

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/112
PeerStatus: Registered
Address: 10.77.4.12
Port: 5060

Event: QueueMemberStatus
Privilege: agent,all
Queue: callcenter
Location: SIP/112
MemberName: SIP/112
Membership: static
Penalty: 0
CallsTaken: 2
LastCall: 1298477682
Status: 1
Paused: 0

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/111
PeerStatus: Registered
Address: 10.77.4.12
Port: 5060

Event: QueueMemberStatus
Privilege: agent,all
Queue: callcenter
Location: SIP/111
MemberName: SIP/111
Membership: static
Penalty: 0
CallsTaken: 4
LastCall: 1298645127
Status: 1
Paused: 0

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/gwfxo
PeerStatus: Registered
Address: 10.77.4.11
Port: 5060

Event: Newchannel
Privilege: call,all
Channel: SIP/gwfxo-00000117
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum: 0213108826
CallerIDName:
AccountCode:
Exten: 3
Context: gwfxo
Uniqueid: 1298645771.279

Event: Newstate
Privilege: call,all
Channel: SIP/gwfxo-00000117
ChannelState: 4
ChannelStateDesc: Ring
CallerIDNum: 0213108826
CallerIDName:
Uniqueid: 1298645771.279

Event: Newstate
Privilege: call,all
Channel: SIP/gwfxo-00000117
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 0213108826
CallerIDName:
Uniqueid: 1298645771.279

Event: Join
Privilege: call,all
Channel: SIP/gwfxo-00000117
CallerIDNum: 0213108826
CallerIDName: unknown
Queue: callcenter
Position: 1
Count: 1
Uniqueid: 1298645771.279

Event: MusicOnHold
Privilege: call,all
State: Start
Channel: SIP/gwfxo-00000117
UniqueID: 1298645771.279

Event: Newchannel
Privilege: call,all
Channel: SIP/112-00000118
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
AccountCode:
Exten:
Context: gwfxs
Uniqueid: 1298645771.280

Event: QueueMemberStatus
Privilege: agent,all
Queue: callcenter
Location: SIP/112
MemberName: SIP/112
Membership: static
Penalty: 0
CallsTaken: 2
LastCall: 1298477682
Status: 1
Paused: 0

Event: Newstate
Privilege: call,all
Channel: SIP/112-00000118
ChannelState: 5
ChannelStateDesc: Ringing
CallerIDNum: 0213108826
CallerIDName:
Uniqueid: 1298645771.280

Event: QueueMemberStatus
Privilege: agent,all
Queue: callcenter
Location: SIP/112
MemberName: SIP/112
Membership: static
Penalty: 0
CallsTaken: 2
LastCall: 1298477682
Status: 1
Paused: 0

Event: Hangup
Privilege: call,all
Channel: SIP/112-00000118
Uniqueid: 1298645771.280
CallerIDNum: 0213108826
CallerIDName:
Cause: 16
Cause-txt: Normal Clearing

Event: QueueMemberStatus
Privilege: agent,all
Queue: callcenter
Location: SIP/112
MemberName: SIP/112
Membership: static
Penalty: 0
CallsTaken: 2
LastCall: 1298477682
Status: 1
Paused: 0

Event: Newchannel
Privilege: call,all
Channel: SIP/113-00000119
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
AccountCode:
Exten:
Context: gwfxs
Uniqueid: 1298645792.281

Event: QueueMemberStatus
Privilege: agent,all
Queue: callcenter
Location: SIP/113
MemberName: SIP/113
Membership: static
Penalty: 0
CallsTaken: 3
LastCall: 1298563263
Status: 1
Paused: 0

Event: Newstate
Privilege: call,all
Channel: SIP/113-00000119
ChannelState: 5
ChannelStateDesc: Ringing
CallerIDNum: 0213108826
CallerIDName:
Uniqueid: 1298645792.281

Event: QueueMemberStatus
Privilege: agent,all
Queue: callcenter
Location: SIP/113
MemberName: SIP/113
Membership: static
Penalty: 0
CallsTaken: 3
LastCall: 1298563263
Status: 1
Paused: 0

Event: Newstate
Privilege: call,all
Channel: SIP/113-00000119
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 0213108826
CallerIDName:
Uniqueid: 1298645792.281

Event: MusicOnHold
Privilege: call,all
State: Stop
Channel: SIP/gwfxo-00000117
UniqueID: 1298645771.279

Event: Leave
Privilege: call,all
Channel: SIP/gwfxo-00000117
Queue: callcenter
Count: 0
Uniqueid: 1298645771.279

Event: NewAccountCode
Privilege: call,all
Channel: SIP/113-00000119
Uniqueid: 1298645792.281
AccountCode:
OldAccountCode:

Event: Bridge
Privilege: call,all
Bridgestate: Link
Bridgetype: core
Channel1: SIP/gwfxo-00000117
Channel2: SIP/113-00000119
Uniqueid1: 1298645771.279
Uniqueid2: 1298645792.281
CallerID1: 0213108826
CallerID2: 0213108826

Event: QueueMemberStatus
Privilege: agent,all
Queue: callcenter
Location: SIP/113
MemberName: SIP/113
Membership: static
Penalty: 0
CallsTaken: 3
LastCall: 1298563263
Status: 1
Paused: 0

Event: Unlink
Privilege: call,all
Channel1: SIP/gwfxo-00000117
Channel2: SIP/113-00000119
Uniqueid1: 1298645771.279
Uniqueid2: 1298645792.281
CallerID1: 0213108826
CallerID2: 0213108826

Event: Hangup
Privilege: call,all
Channel: SIP/113-00000119
Uniqueid: 1298645792.281
CallerIDNum: 0213108826
CallerIDName:
Cause: 16
Cause-txt: Normal Clearing

Event: Hangup
Privilege: call,all
Channel: SIP/gwfxo-00000117
Uniqueid: 1298645771.279
CallerIDNum: 0213108826
CallerIDName:
Cause: 16
Cause-txt: Normal Clearing

Event: QueueMemberStatus
Privilege: agent,all
Queue: callcenter
Location: SIP/113
MemberName: SIP/113
Membership: static
Penalty: 0
CallsTaken: 4
LastCall: 1298645796
Status: 1
Paused: 0

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/114
PeerStatus: Registered
Address: 10.77.4.12
Port: 5060

Event: QueueMemberStatus
Privilege: agent,all
Queue: callcenter
Location: SIP/114
MemberName: SIP/114
Membership: static
Penalty: 0
CallsTaken: 1
LastCall: 1298477459
Status: 1
Paused: 0

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/113
PeerStatus: Registered
Address: 10.77.4.12
Port: 5060

Event: QueueMemberStatus
Privilege: agent,all
Queue: callcenter
Location: SIP/113
MemberName: SIP/113
Membership: static
Penalty: 0
CallsTaken: 4
LastCall: 1298645796
Status: 1
Paused: 0

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/112
PeerStatus: Registered
Address: 10.77.4.12
Port: 5060

Event: QueueMemberStatus
Privilege: agent,all
Queue: callcenter
Location: SIP/112
MemberName: SIP/112
Membership: static
Penalty: 0
CallsTaken: 2
LastCall: 1298477682
Status: 1
Paused: 0

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/111
PeerStatus: Registered
Address: 10.77.4.12
Port: 5060

Event: QueueMemberStatus
Privilege: agent,all
Queue: callcenter
Location: SIP/111
MemberName: SIP/111
Membership: static
Penalty: 0
CallsTaken: 4
LastCall: 1298645127
Status: 1
Paused: 0

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/218
PeerStatus: Registered
Address: 10.77.4.13
Port: 5060

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/217
PeerStatus: Registered
Address: 10.77.4.13
Port: 5060

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/216
PeerStatus: Registered
Address: 10.77.4.13
Port: 5060

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/215
PeerStatus: Registered
Address: 10.77.4.13
Port: 5060

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/214
PeerStatus: Registered
Address: 10.77.4.13
Port: 5060

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/213
PeerStatus: Registered
Address: 10.77.4.13
Port: 5060

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/212
PeerStatus: Registered
Address: 10.77.4.13
Port: 5060

Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/211
PeerStatus: Registered
Address: 10.77.4.13
Port: 5060

#5

[quote=“mazzic”]This line isn’t right…

The second parameter is the options, not a timeout.

why timeouts of 41.666666 days? Might as well leave it off.

Should provide some CLI output for a call.[/quote]

Setting up 5 min for timeout doesn’t solve the problem.

exten => 1,1,Queue(callcenter,Tt,300)

#6

<------------->
[Feb 28 13:36:58] VERBOSE[2588] chan_sip.c: — (14 headers 12 lines) —
[Feb 28 13:36:58] VERBOSE[2588] chan_sip.c: Really destroying SIP dialog ‘44d3212201ae6e332b7f65ed5cda2860@10.77.4.10’ Method: OPTIONS
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.11:5060 —>
INVITE sip:3@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.77.4.11;branch=z9hG4bKac621179059
Max-Forwards: 70
From: sip:0722304336@gwfxo;tag=1c621174192
To: sip:3@asterisk;user=phone
Call-ID: 621173736261200013452@10.77.4.11
CSeq: 1 INVITE
Contact: sip:gwfxo@10.77.4.11:5060
Supported: em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 251

v=0
o=AudiocodesGW 621163958 621163839 IN IP4 10.77.4.11
s=Phone-Call
c=IN IP4 10.77.4.11
t=0 0
m=audio 6140 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: — (14 headers 12 lines) —
[Feb 28 13:37:02] VERBOSE[2588] netsock.c: == Using SIP RTP CoS mark 5
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: Sending to 10.77.4.11 : 5060 (no NAT)
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: Using INVITE request as basis request - 621173736261200013452@10.77.4.11
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: Found peer ‘gwfxo’ for ‘0722304336’ from 10.77.4.11:5060
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c:
<— Reliably Transmitting (no NAT) to 10.77.4.11:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.77.4.11;branch=z9hG4bKac621179059;received=10.77.4.11
From: sip:0722304336@gwfxo;tag=1c621174192
To: sip:3@asterisk;user=phone;tag=as2f627fd4
Call-ID: 621173736261200013452@10.77.4.11
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="47161817"
Content-Length: 0

<------------>
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: Scheduling destruction of SIP dialog ‘621173736261200013452@10.77.4.11’ in 6400 ms (Method: INVITE)
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.11:5060 —>
ACK sip:3@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.77.4.11;branch=z9hG4bKac621179059
Max-Forwards: 70
From: sip:0722304336@gwfxo;tag=1c621174192
To: sip:3@asterisk;user=phone;tag=as2f627fd4
Call-ID: 621173736261200013452@10.77.4.11
CSeq: 1 ACK
Contact: sip:gwfxo@10.77.4.11:5060
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Length: 0

<------------->
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: — (12 headers 0 lines) —
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.11:5060 —>
INVITE sip:3@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.77.4.11;branch=z9hG4bKac621234000
Max-Forwards: 70
From: sip:0722304336@gwfxo;tag=1c621174192
To: sip:3@asterisk;user=phone
Call-ID: 621173736261200013452@10.77.4.11
CSeq: 2 INVITE
Authorization: Digest username=“gwfxo”,realm=“asterisk”,nonce=“47161817”,uri=“sip:3@asterisk”,algorithm=MD5,response="f1931398a982ebf80938229423164640"
Contact: sip:gwfxo@10.77.4.11:5060
Supported: em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 251

v=0
o=AudiocodesGW 621163958 621163839 IN IP4 10.77.4.11
s=Phone-Call
c=IN IP4 10.77.4.11
t=0 0
m=audio 6140 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: — (15 headers 12 lines) —
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: Sending to 10.77.4.11 : 5060 (no NAT)
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: Using INVITE request as basis request - 621173736261200013452@10.77.4.11
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: Found peer ‘gwfxo’ for ‘0722304336’ from 10.77.4.11:5060
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: Found RTP audio format 8
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: Found RTP audio format 0
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: Found RTP audio format 101
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: Found audio description format PCMA for ID 8
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: Found audio description format PCMU for ID 0
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: Found audio description format telephone-event for ID 101
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: Peer audio RTP is at port 10.77.4.11:6140
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: Looking for 3 in gwfxo (domain asterisk)
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c: list_route: hop: sip:gwfxo@10.77.4.11:5060
[Feb 28 13:37:02] VERBOSE[2588] chan_sip.c:
<— Transmitting (no NAT) to 10.77.4.11:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.77.4.11;branch=z9hG4bKac621234000;received=10.77.4.11
From: sip:0722304336@gwfxo;tag=1c621174192
To: sip:3@asterisk;user=phone
Call-ID: 621173736261200013452@10.77.4.11
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:3@10.77.4.10
Content-Length: 0

<------------>
[Feb 28 13:37:02] VERBOSE[1638] pbx.c: – Executing [3@gwfxo:1] Answer(“SIP/gwfxo-00000130”, “”) in new stack
[Feb 28 13:37:02] VERBOSE[1638] chan_sip.c: Audio is at 10.77.4.10 port 15448
[Feb 28 13:37:02] VERBOSE[1638] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Feb 28 13:37:02] VERBOSE[1638] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Feb 28 13:37:02] VERBOSE[1638] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 28 13:37:02] VERBOSE[1638] chan_sip.c:
<— Reliably Transmitting (no NAT) to 10.77.4.11:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.77.4.11;branch=z9hG4bKac621234000;received=10.77.4.11
From: sip:0722304336@gwfxo;tag=1c621174192
To: sip:3@asterisk;user=phone;tag=as0bf359f4
Call-ID: 621173736261200013452@10.77.4.11
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:3@10.77.4.10
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1300453006 1300453006 IN IP4 10.77.4.10
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 10.77.4.10
t=0 0
m=audio 15448 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[Feb 28 13:37:03] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.11:5060 —>
ACK sip:3@10.77.4.10 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.11;branch=z9hG4bKac621369356
Max-Forwards: 70
From: sip:0722304336@gwfxo;tag=1c621174192
To: sip:3@asterisk;user=phone;tag=as0bf359f4
Call-ID: 621173736261200013452@10.77.4.11
CSeq: 2 ACK
Contact: sip:gwfxo@10.77.4.11:5060
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Length: 0

<------------->
[Feb 28 13:37:03] VERBOSE[2588] chan_sip.c: — (12 headers 0 lines) —
[Feb 28 13:37:03] VERBOSE[1638] pbx.c: – Executing [3@gwfxo:2] GotoIfTime(“SIP/gwfxo-00000130”, “19:00-10:00|mon-fri||?closed,c,1”) in new stack
[Feb 28 13:37:03] VERBOSE[1638] pbx.c: – Executing [3@gwfxo:3] GotoIfTime(“SIP/gwfxo-00000130”, “|sat-sun||*?closed,c,1”) in new stack
[Feb 28 13:37:03] VERBOSE[1638] pbx.c: – Executing [3@gwfxo:4] GotoIf(“SIP/gwfxo-00000130”, “1?1000:999”) in new stack
[Feb 28 13:37:03] VERBOSE[1638] pbx.c: – Goto (gwfxo,3,1000)
[Feb 28 13:37:03] VERBOSE[1638] pbx.c: – Executing [3@gwfxo:1000] Dial(“SIP/gwfxo-00000130”, “SIP/213”) in new stack
[Feb 28 13:37:03] VERBOSE[1638] netsock.c: == Using SIP RTP CoS mark 5
[Feb 28 13:37:03] VERBOSE[1638] chan_sip.c: Audio is at 10.77.4.10 port 16468
[Feb 28 13:37:03] VERBOSE[1638] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Feb 28 13:37:03] VERBOSE[1638] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Feb 28 13:37:03] VERBOSE[1638] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 28 13:37:03] VERBOSE[1638] chan_sip.c: Reliably Transmitting (no NAT) to 10.77.4.13:5060:
INVITE sip:213@10.77.4.13:5060 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK1c26df3a;rport
Max-Forwards: 70
From: “0722304336” sip:213@10.77.4.10;tag=as4f511f35
To: sip:213@10.77.4.13:5060
Contact: sip:213@10.77.4.10
Call-ID: 1439d8171d0d828430c7ef4e0d0c4580@10.77.4.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.16.1
Date: Mon, 28 Feb 2011 11:37:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1230819595 1230819595 IN IP4 10.77.4.10
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 10.77.4.10
t=0 0
m=audio 16468 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[Feb 28 13:37:03] VERBOSE[1638] app_dial.c: – Called 213
[Feb 28 13:37:03] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK1c26df3a;rport
From: “0722304336” sip:213@10.77.4.10;tag=as4f511f35
To: sip:213@10.77.4.13:5060;tag=1c636829756
Call-ID: 1439d8171d0d828430c7ef4e0d0c4580@10.77.4.10
CSeq: 102 INVITE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Length: 0

<------------->
[Feb 28 13:37:03] VERBOSE[2588] chan_sip.c: — (10 headers 0 lines) —
[Feb 28 13:37:03] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK1c26df3a;rport
From: “0722304336” sip:213@10.77.4.10;tag=as4f511f35
To: sip:213@10.77.4.13:5060;tag=1c636829756
Call-ID: 1439d8171d0d828430c7ef4e0d0c4580@10.77.4.10
CSeq: 102 INVITE
Contact: sip:213@10.77.4.13:5060
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Length: 0

<------------->
[Feb 28 13:37:03] VERBOSE[2588] chan_sip.c: — (11 headers 0 lines) —
[Feb 28 13:37:03] VERBOSE[1638] app_dial.c: – SIP/213-00000131 is ringing
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK1c26df3a;rport
From: “0722304336” sip:213@10.77.4.10;tag=as4f511f35
To: sip:213@10.77.4.13:5060;tag=1c636829756
Call-ID: 1439d8171d0d828430c7ef4e0d0c4580@10.77.4.10
CSeq: 102 INVITE
Contact: sip:213@10.77.4.13:5060
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Type: application/sdp
Content-Length: 248

v=0
o=AudiocodesGW 636845120 636844999 IN IP4 10.77.4.13
s=Phone-Call
c=IN IP4 10.77.4.13
t=0 0
m=audio 6080 RTP/AVP 8 101
c=IN IP4 10.77.4.13
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: — (12 headers 12 lines) —
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Found RTP audio format 8
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Found RTP audio format 101
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Found audio description format PCMA for ID 8
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Found audio description format telephone-event for ID 101
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Peer audio RTP is at port 10.77.4.13:6080
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: list_route: hop: sip:213@10.77.4.13:5060
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: set_destination: Parsing sip:213@10.77.4.13:5060 for address/port to send to
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: set_destination: set destination to 10.77.4.13, port 5060
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Transmitting (no NAT) to 10.77.4.13:5060:
ACK sip:213@10.77.4.13:5060 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK565b48d3;rport
Max-Forwards: 70
From: “0722304336” sip:213@10.77.4.10;tag=as4f511f35
To: sip:213@10.77.4.13:5060;tag=1c636829756
Contact: sip:213@10.77.4.10
Call-ID: 1439d8171d0d828430c7ef4e0d0c4580@10.77.4.10
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.16.1
Content-Length: 0

[Feb 28 13:37:09] VERBOSE[1638] app_dial.c: – SIP/213-00000131 answered SIP/gwfxo-00000130
[Feb 28 13:37:09] VERBOSE[1638] rtp.c: – Native bridging SIP/gwfxo-00000130 and SIP/213-00000131
[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: set_destination: Parsing sip:gwfxo@10.77.4.11:5060 for address/port to send to
[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: set_destination: set destination to 10.77.4.11, port 5060
[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: Audio is at 10.77.4.10 port 15448
[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: Reliably Transmitting (no NAT) to 10.77.4.11:5060:
INVITE sip:gwfxo@10.77.4.11:5060 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK2b80a322;rport
Max-Forwards: 70
From: sip:3@asterisk;user=phone;tag=as0bf359f4
To: sip:0722304336@gwfxo;tag=1c621174192
Contact: sip:3@10.77.4.10
Call-ID: 621173736261200013452@10.77.4.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 1300453006 1300453007 IN IP4 10.77.4.13
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 10.77.4.13
t=0 0
m=audio 6080 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: set_destination: Parsing sip:213@10.77.4.13:5060 for address/port to send to
[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: set_destination: set destination to 10.77.4.13, port 5060
[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: Audio is at 10.77.4.10 port 16468
[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: Reliably Transmitting (no NAT) to 10.77.4.13:5060:
INVITE sip:213@10.77.4.13:5060 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK078c01e8;rport
Max-Forwards: 70
From: “0722304336” sip:213@10.77.4.10;tag=as4f511f35
To: sip:213@10.77.4.13:5060;tag=1c636829756
Contact: sip:213@10.77.4.10
Call-ID: 1439d8171d0d828430c7ef4e0d0c4580@10.77.4.10
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 1230819595 1230819596 IN IP4 10.77.4.11
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 10.77.4.11
t=0 0
m=audio 6140 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK078c01e8;rport
From: “0722304336” sip:213@10.77.4.10;tag=as4f511f35
To: sip:213@10.77.4.13:5060;tag=1c636829756
Call-ID: 1439d8171d0d828430c7ef4e0d0c4580@10.77.4.10
CSeq: 103 INVITE
Contact: sip:213@10.77.4.13:5060
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Type: application/sdp
Content-Length: 248

v=0
o=AudiocodesGW 636845120 636845000 IN IP4 10.77.4.13
s=Phone-Call
c=IN IP4 10.77.4.13
t=0 0
m=audio 6080 RTP/AVP 8 101
c=IN IP4 10.77.4.13
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: — (12 headers 12 lines) —
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Found RTP audio format 8
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Found RTP audio format 101
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Found audio description format PCMA for ID 8
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Found audio description format telephone-event for ID 101
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Peer audio RTP is at port 10.77.4.13:6080
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: set_destination: Parsing sip:213@10.77.4.13:5060 for address/port to send to
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: set_destination: set destination to 10.77.4.13, port 5060
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Transmitting (no NAT) to 10.77.4.13:5060:
ACK sip:213@10.77.4.13:5060 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK2c12cf3e;rport
Max-Forwards: 70
From: “0722304336” sip:213@10.77.4.10;tag=as4f511f35
To: sip:213@10.77.4.13:5060;tag=1c636829756
Contact: sip:213@10.77.4.10
Call-ID: 1439d8171d0d828430c7ef4e0d0c4580@10.77.4.10
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.16.1
Content-Length: 0

[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.11:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK2b80a322;rport
From: sip:3@asterisk;user=phone;tag=as0bf359f4
To: sip:0722304336@gwfxo;tag=1c621174192
Call-ID: 621173736261200013452@10.77.4.11
CSeq: 102 INVITE
Contact: sip:gwfxo@10.77.4.11:5060
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 248

v=0
o=AudiocodesGW 621163958 621163840 IN IP4 10.77.4.11
s=Phone-Call
c=IN IP4 10.77.4.11
t=0 0
m=audio 6140 RTP/AVP 8 101
c=IN IP4 10.77.4.11
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: — (13 headers 12 lines) —
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Found RTP audio format 8
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Found RTP audio format 101
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Found audio description format PCMA for ID 8
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Found audio description format telephone-event for ID 101
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Peer audio RTP is at port 10.77.4.11:6140
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: set_destination: Parsing sip:gwfxo@10.77.4.11:5060 for address/port to send to
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: set_destination: set destination to 10.77.4.11, port 5060
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Transmitting (no NAT) to 10.77.4.11:5060:
ACK sip:gwfxo@10.77.4.11:5060 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK6190d538;rport
Max-Forwards: 70
From: sip:3@asterisk;user=phone;tag=as0bf359f4
To: sip:0722304336@gwfxo;tag=1c621174192
Contact: sip:3@10.77.4.10
Call-ID: 621173736261200013452@10.77.4.11
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.16.1
Content-Length: 0

[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: set_destination: Parsing sip:213@10.77.4.13:5060 for address/port to send to
[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: set_destination: set destination to 10.77.4.13, port 5060
[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: Audio is at 10.77.4.10 port 16468
[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 28 13:37:09] VERBOSE[1638] chan_sip.c: Reliably Transmitting (no NAT) to 10.77.4.13:5060:
INVITE sip:213@10.77.4.13:5060 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK76a04578;rport
Max-Forwards: 70
From: “0722304336” sip:213@10.77.4.10;tag=as4f511f35
To: sip:213@10.77.4.13:5060;tag=1c636829756
Contact: sip:213@10.77.4.10
Call-ID: 1439d8171d0d828430c7ef4e0d0c4580@10.77.4.10
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 1230819595 1230819597 IN IP4 10.77.4.11
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 10.77.4.11
t=0 0
m=audio 6140 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK76a04578;rport
From: “0722304336” sip:213@10.77.4.10;tag=as4f511f35
To: sip:213@10.77.4.13:5060;tag=1c636829756
Call-ID: 1439d8171d0d828430c7ef4e0d0c4580@10.77.4.10
CSeq: 104 INVITE
Contact: sip:213@10.77.4.13:5060
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Type: application/sdp
Content-Length: 248

v=0
o=AudiocodesGW 636845120 636845001 IN IP4 10.77.4.13
s=Phone-Call
c=IN IP4 10.77.4.13
t=0 0
m=audio 6080 RTP/AVP 8 101
c=IN IP4 10.77.4.13
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: — (12 headers 12 lines) —
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Found RTP audio format 8
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Found RTP audio format 101
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Found audio description format PCMA for ID 8
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Found audio description format telephone-event for ID 101
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Peer audio RTP is at port 10.77.4.13:6080
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: set_destination: Parsing sip:213@10.77.4.13:5060 for address/port to send to
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: set_destination: set destination to 10.77.4.13, port 5060
[Feb 28 13:37:09] VERBOSE[2588] chan_sip.c: Transmitting (no NAT) to 10.77.4.13:5060:
ACK sip:213@10.77.4.13:5060 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK5a4accf1;rport
Max-Forwards: 70
From: “0722304336” sip:213@10.77.4.10;tag=as4f511f35
To: sip:213@10.77.4.13:5060;tag=1c636829756
Contact: sip:213@10.77.4.10
Call-ID: 1439d8171d0d828430c7ef4e0d0c4580@10.77.4.10
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.6.2.16.1
Content-Length: 0

[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
BYE sip:213@10.77.4.10 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657452689
Max-Forwards: 70
From: sip:213@10.77.4.13:5060;tag=1c636829756
To: “0722304336” sip:213@10.77.4.10;tag=as4f511f35
Call-ID: 1439d8171d0d828430c7ef4e0d0c4580@10.77.4.10
CSeq: 1 BYE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Reason: Q.850 ;cause=16 ;text="local"
Content-Length: 0

<------------->
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: — (12 headers 0 lines) —
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Sending to 10.77.4.13 : 5060 (no NAT)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— Transmitting (no NAT) to 10.77.4.13:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657452689;received=10.77.4.13
From: sip:213@10.77.4.13:5060;tag=1c636829756
To: “0722304336” sip:213@10.77.4.10;tag=as4f511f35
Call-ID: 1439d8171d0d828430c7ef4e0d0c4580@10.77.4.10
CSeq: 1 BYE
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
[Feb 28 13:37:11] VERBOSE[1638] chan_sip.c: set_destination: Parsing sip:gwfxo@10.77.4.11:5060 for address/port to send to
[Feb 28 13:37:11] VERBOSE[1638] chan_sip.c: set_destination: set destination to 10.77.4.11, port 5060
[Feb 28 13:37:11] VERBOSE[1638] chan_sip.c: Audio is at 10.77.4.10 port 15448
[Feb 28 13:37:11] VERBOSE[1638] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Feb 28 13:37:11] VERBOSE[1638] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 28 13:37:11] VERBOSE[1638] chan_sip.c: Reliably Transmitting (no NAT) to 10.77.4.11:5060:
INVITE sip:gwfxo@10.77.4.11:5060 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK4956fa2d;rport
Max-Forwards: 70
From: sip:3@asterisk;user=phone;tag=as0bf359f4
To: sip:0722304336@gwfxo;tag=1c621174192
Contact: sip:3@10.77.4.10
Call-ID: 621173736261200013452@10.77.4.11
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 1300453006 1300453008 IN IP4 10.77.4.10
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 10.77.4.10
t=0 0
m=audio 15448 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[Feb 28 13:37:11] VERBOSE[1638] pbx.c: == Spawn extension (gwfxo, 3, 1000) exited non-zero on ‘SIP/gwfxo-00000130’
[Feb 28 13:37:11] VERBOSE[1638] chan_sip.c: Scheduling destruction of SIP dialog ‘621173736261200013452@10.77.4.11’ in 6400 ms (Method: ACK)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
REGISTER sip:gwfxs8 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657474464
Max-Forwards: 70
From: sip:218@10.77.4.10;tag=1c657471472
To: sip:218@10.77.4.10
Call-ID: 76918875511200054738@10.77.4.13
CSeq: 49681 REGISTER
Contact: sip:218@10.77.4.13:5060;expires=180
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 180
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Length: 0

<------------->
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: — (12 headers 0 lines) —
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Sending to 10.77.4.13 : 5060 (no NAT)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— Transmitting (no NAT) to 10.77.4.13:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657474464;received=10.77.4.13
From: sip:218@10.77.4.10;tag=1c657471472
To: sip:218@10.77.4.10;tag=as2e18bd66
Call-ID: 76918875511200054738@10.77.4.13
CSeq: 49681 REGISTER
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="16a7fee2"
Content-Length: 0

<------------>
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Scheduling destruction of SIP dialog ‘76918875511200054738@10.77.4.13’ in 32000 ms (Method: REGISTER)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Really destroying SIP dialog ‘1439d8171d0d828430c7ef4e0d0c4580@10.77.4.10’ Method: BYE
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
REGISTER sip:gwfxs8 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657495134
Max-Forwards: 70
From: sip:217@10.77.4.10;tag=1c657492079
To: sip:217@10.77.4.10
Call-ID: 76918955811200054738@10.77.4.13
CSeq: 49679 REGISTER
Contact: sip:217@10.77.4.13:5060;expires=180
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 180
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Length: 0

<------------->
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: — (12 headers 0 lines) —
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Sending to 10.77.4.13 : 5060 (no NAT)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— Transmitting (no NAT) to 10.77.4.13:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657495134;received=10.77.4.13
From: sip:217@10.77.4.10;tag=1c657492079
To: sip:217@10.77.4.10;tag=as231ee8a4
Call-ID: 76918955811200054738@10.77.4.13
CSeq: 49679 REGISTER
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="5149c055"
Content-Length: 0

<------------>
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Scheduling destruction of SIP dialog ‘76918955811200054738@10.77.4.13’ in 32000 ms (Method: REGISTER)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
REGISTER sip:gwfxs8 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657517349
Max-Forwards: 70
From: sip:218@10.77.4.10;tag=1c657471472
To: sip:218@10.77.4.10
Call-ID: 76918875511200054738@10.77.4.13
CSeq: 49682 REGISTER
Authorization: Digest username=“218”,realm=“asterisk”,nonce=“16a7fee2”,uri=“sip:gwfxs8”,algorithm=MD5,response="6f846cf33cbe114803619c49b505d480"
Contact: sip:218@10.77.4.13:5060;expires=180
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 180
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Length: 0

<------------->
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: — (13 headers 0 lines) —
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Sending to 10.77.4.13 : 5060 (no NAT)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Reliably Transmitting (no NAT) to 10.77.4.13:5060:
OPTIONS sip:218@10.77.4.13:5060 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK11113e92;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.77.4.10;tag=as1e4bfd2e
To: sip:218@10.77.4.13:5060
Contact: sip:asterisk@10.77.4.10
Call-ID: 1da9d38564cd979b37b8d83445797e5d@10.77.4.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.16.1
Date: Mon, 28 Feb 2011 11:37:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— Transmitting (no NAT) to 10.77.4.13:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657517349;received=10.77.4.13
From: sip:218@10.77.4.10;tag=1c657471472
To: sip:218@10.77.4.10;tag=as2e18bd66
Call-ID: 76918875511200054738@10.77.4.13
CSeq: 49682 REGISTER
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 180
Contact: sip:218@10.77.4.13:5060;expires=180
Date: Mon, 28 Feb 2011 11:37:11 GMT
Content-Length: 0

<------------>
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Scheduling destruction of SIP dialog ‘76918875511200054738@10.77.4.13’ in 32000 ms (Method: REGISTER)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
REGISTER sip:gwfxs8 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657528086
Max-Forwards: 70
From: sip:216@10.77.4.10;tag=1c657525119
To: sip:216@10.77.4.10
Call-ID: 76919031811200054738@10.77.4.13
CSeq: 49679 REGISTER
Contact: sip:216@10.77.4.13:5060;expires=180
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 180
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Length: 0

<------------->
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: — (12 headers 0 lines) —
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Sending to 10.77.4.13 : 5060 (no NAT)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— Transmitting (no NAT) to 10.77.4.13:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657528086;received=10.77.4.13
From: sip:216@10.77.4.10;tag=1c657525119
To: sip:216@10.77.4.10;tag=as02359ac5
Call-ID: 76919031811200054738@10.77.4.13
CSeq: 49679 REGISTER
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="5639e265"
Content-Length: 0

<------------>
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Scheduling destruction of SIP dialog ‘76919031811200054738@10.77.4.13’ in 32000 ms (Method: REGISTER)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
REGISTER sip:gwfxs8 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657551301
Max-Forwards: 70
From: sip:217@10.77.4.10;tag=1c657492079
To: sip:217@10.77.4.10
Call-ID: 76918955811200054738@10.77.4.13
CSeq: 49680 REGISTER
Authorization: Digest username=“217”,realm=“asterisk”,nonce=“5149c055”,uri=“sip:gwfxs8”,algorithm=MD5,response="e234ac60ff71c2c65ead854a3992c8e2"
Contact: sip:217@10.77.4.13:5060;expires=180
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 180
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Length: 0

<------------->
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: — (13 headers 0 lines) —
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Sending to 10.77.4.13 : 5060 (no NAT)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Reliably Transmitting (no NAT) to 10.77.4.13:5060:
OPTIONS sip:217@10.77.4.13:5060 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK6764a459;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.77.4.10;tag=as6f22c381
To: sip:217@10.77.4.13:5060
Contact: sip:asterisk@10.77.4.10
Call-ID: 34ca34e33459cced52ea851a3e0d0bee@10.77.4.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.16.1
Date: Mon, 28 Feb 2011 11:37:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— Transmitting (no NAT) to 10.77.4.13:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657551301;received=10.77.4.13
From: sip:217@10.77.4.10;tag=1c657492079
To: sip:217@10.77.4.10;tag=as231ee8a4
Call-ID: 76918955811200054738@10.77.4.13
CSeq: 49680 REGISTER
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 180
Contact: sip:217@10.77.4.13:5060;expires=180
Date: Mon, 28 Feb 2011 11:37:11 GMT
Content-Length: 0

<------------>
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Scheduling destruction of SIP dialog ‘76918955811200054738@10.77.4.13’ in 32000 ms (Method: REGISTER)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
REGISTER sip:gwfxs8 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657565761
Max-Forwards: 70
From: sip:215@10.77.4.10;tag=1c657562770
To: sip:215@10.77.4.10
Call-ID: 76919107411200054738@10.77.4.13
CSeq: 49680 REGISTER
Contact: sip:215@10.77.4.13:5060;expires=180
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 180
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Length: 0

<------------->
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: — (12 headers 0 lines) —
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Sending to 10.77.4.13 : 5060 (no NAT)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— Transmitting (no NAT) to 10.77.4.13:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657565761;received=10.77.4.13
From: sip:215@10.77.4.10;tag=1c657562770
To: sip:215@10.77.4.10;tag=as6469d1a0
Call-ID: 76919107411200054738@10.77.4.13
CSeq: 49680 REGISTER
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3fefd26a"
Content-Length: 0

<------------>
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Scheduling destruction of SIP dialog ‘76919107411200054738@10.77.4.13’ in 32000 ms (Method: REGISTER)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK11113e92;rport
From: “asterisk” sip:asterisk@10.77.4.10;tag=as1e4bfd2e
To: sip:218@10.77.4.13:5060;tag=1c657580461
Call-ID: 1da9d38564cd979b37b8d83445797e5d@10.77.4.10
CSeq: 102 OPTIONS
Contact: sip:10.77.4.13:5060
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.023.006
X-Resources: telchs=8/0;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 248

v=0
o=AudiocodesGW 657588617 657588484 IN IP4 10.77.4.13
s=Phone-Call
c=IN IP4 10.77.4.13
t=0 0
m=audio 6000 RTP/AVP 8 0 96
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv

<------------->
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: — (14 headers 12 lines) —
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Really destroying SIP dialog ‘1da9d38564cd979b37b8d83445797e5d@10.77.4.10’ Method: OPTIONS
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.11:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK4956fa2d;rport
From: sip:3@asterisk;user=phone;tag=as0bf359f4
To: sip:0722304336@gwfxo;tag=1c621174192
Call-ID: 621173736261200013452@10.77.4.11
CSeq: 103 INVITE
Contact: sip:gwfxo@10.77.4.11:5060
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 248

v=0
o=AudiocodesGW 621163958 621163841 IN IP4 10.77.4.11
s=Phone-Call
c=IN IP4 10.77.4.11
t=0 0
m=audio 6140 RTP/AVP 8 101
c=IN IP4 10.77.4.11
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: — (13 headers 12 lines) —
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Found RTP audio format 8
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Found RTP audio format 101
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Found audio description format PCMA for ID 8
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Found audio description format telephone-event for ID 101
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Peer audio RTP is at port 10.77.4.11:6140
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: set_destination: Parsing sip:gwfxo@10.77.4.11:5060 for address/port to send to
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: set_destination: set destination to 10.77.4.11, port 5060
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Transmitting (no NAT) to 10.77.4.11:5060:
ACK sip:gwfxo@10.77.4.11:5060 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK75d5d5a8;rport
Max-Forwards: 70
From: sip:3@asterisk;user=phone;tag=as0bf359f4
To: sip:0722304336@gwfxo;tag=1c621174192
Contact: sip:3@10.77.4.10
Call-ID: 621173736261200013452@10.77.4.11
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.16.1
Content-Length: 0

[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: set_destination: Parsing sip:gwfxo@10.77.4.11:5060 for address/port to send to
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: set_destination: set destination to 10.77.4.11, port 5060
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Reliably Transmitting (no NAT) to 10.77.4.11:5060:
BYE sip:gwfxo@10.77.4.11:5060 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK11586d84;rport
Max-Forwards: 70
From: sip:3@asterisk;user=phone;tag=as0bf359f4
To: sip:0722304336@gwfxo;tag=1c621174192
Call-ID: 621173736261200013452@10.77.4.11
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.16.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Scheduling destruction of SIP dialog ‘621173736261200013452@10.77.4.11’ in 6400 ms (Method: ACK)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
REGISTER sip:gwfxs8 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657603438
Max-Forwards: 70
From: sip:214@10.77.4.10;tag=1c657600451
To: sip:214@10.77.4.10
Call-ID: 76919184411200054738@10.77.4.13
CSeq: 49680 REGISTER
Contact: sip:214@10.77.4.13:5060;expires=180
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 180
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Length: 0

<------------->
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: — (12 headers 0 lines) —
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Sending to 10.77.4.13 : 5060 (no NAT)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— Transmitting (no NAT) to 10.77.4.13:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657603438;received=10.77.4.13
From: sip:214@10.77.4.10;tag=1c657600451
To: sip:214@10.77.4.10;tag=as57430691
Call-ID: 76919184411200054738@10.77.4.13
CSeq: 49680 REGISTER
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="313d9822"
Content-Length: 0

<------------>
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Scheduling destruction of SIP dialog ‘76919184411200054738@10.77.4.13’ in 32000 ms (Method: REGISTER)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
REGISTER sip:gwfxs8 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657626596
Max-Forwards: 70
From: sip:213@10.77.4.10;tag=1c657623639
To: sip:213@10.77.4.10
Call-ID: 76919261711200054738@10.77.4.13
CSeq: 49677 REGISTER
Contact: sip:213@10.77.4.13:5060;expires=180
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 180
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Length: 0

<------------->
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: — (12 headers 0 lines) —
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Sending to 10.77.4.13 : 5060 (no NAT)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— Transmitting (no NAT) to 10.77.4.13:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657626596;received=10.77.4.13
From: sip:213@10.77.4.10;tag=1c657623639
To: sip:213@10.77.4.10;tag=as55596569
Call-ID: 76919261711200054738@10.77.4.13
CSeq: 49677 REGISTER
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="2e9ff004"
Content-Length: 0

<------------>
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Scheduling destruction of SIP dialog ‘76919261711200054738@10.77.4.13’ in 32000 ms (Method: REGISTER)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
REGISTER sip:gwfxs8 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657649575
Max-Forwards: 70
From: sip:216@10.77.4.10;tag=1c657525119
To: sip:216@10.77.4.10
Call-ID: 76919031811200054738@10.77.4.13
CSeq: 49680 REGISTER
Authorization: Digest username=“216”,realm=“asterisk”,nonce=“5639e265”,uri=“sip:gwfxs8”,algorithm=MD5,response="36123d3dadfc53bfb2d9a4aec442822b"
Contact: sip:216@10.77.4.13:5060;expires=180
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 180
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Length: 0

<------------->
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: — (13 headers 0 lines) —
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Sending to 10.77.4.13 : 5060 (no NAT)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Reliably Transmitting (no NAT) to 10.77.4.13:5060:
OPTIONS sip:216@10.77.4.13:5060 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK180ce6f2;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.77.4.10;tag=as6a0af76d
To: sip:216@10.77.4.13:5060
Contact: sip:asterisk@10.77.4.10
Call-ID: 5d72a6a80e91f57016d389095da9e1db@10.77.4.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.16.1
Date: Mon, 28 Feb 2011 11:37:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— Transmitting (no NAT) to 10.77.4.13:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657649575;received=10.77.4.13
From: sip:216@10.77.4.10;tag=1c657525119
To: sip:216@10.77.4.10;tag=as02359ac5
Call-ID: 76919031811200054738@10.77.4.13
CSeq: 49680 REGISTER
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 180
Contact: sip:216@10.77.4.13:5060;expires=180
Date: Mon, 28 Feb 2011 11:37:11 GMT
Content-Length: 0

<------------>
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Scheduling destruction of SIP dialog ‘76919031811200054738@10.77.4.13’ in 32000 ms (Method: REGISTER)
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK6764a459;rport
From: “asterisk” sip:asterisk@10.77.4.10;tag=as6f22c381
To: sip:217@10.77.4.13:5060;tag=1c657666157
Call-ID: 34ca34e33459cced52ea851a3e0d0bee@10.77.4.10
CSeq: 102 OPTIONS
Contact: sip:10.77.4.13:5060
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.023.006
X-Resources: telchs=8/0;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 248

v=0
o=AudiocodesGW 657670234 657670110 IN IP4 10.77.4.13
s=Phone-Call
c=IN IP4 10.77.4.13
t=0 0
m=audio 6000 RTP/AVP 8 0 96
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv

<------------->
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: — (14 headers 12 lines) —
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Really destroying SIP dialog ‘34ca34e33459cced52ea851a3e0d0bee@10.77.4.10’ Method: OPTIONS
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.11:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.77.4.10:5060;branch=z9hG4bK11586d84;rport
From: sip:3@asterisk;user=phone;tag=as0bf359f4
To: sip:0722304336@gwfxo;tag=1c621174192
Call-ID: 621173736261200013452@10.77.4.11
CSeq: 104 BYE
Contact: sip:gwfxo@10.77.4.11:5060
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Length: 0

<------------->
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: — (11 headers 0 lines) —
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c: Really destroying SIP dialog ‘621173736261200013452@10.77.4.11’ Method: ACK
[Feb 28 13:37:11] VERBOSE[2588] chan_sip.c:
<— SIP read from UDP:10.77.4.13:5060 —>
REGISTER sip:gwfxs8 SIP/2.0
Via: SIP/2.0/UDP 10.77.4.13;branch=z9hG4bKac657698111
Max-Forwards: 70
From: sip:212@10.77.4.10;tag=1c657695165
To: sip:212@10.77.4.10
Call-ID: 76919340411200054738@10.77.4.13
CSeq: 49677 REGISTER
Contact: sip:212@10.77.4.13:5060;expires=180
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 180
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006
Content-Length: 0

#7

Please close this post, i solve the problem.
I set Polarity Reversal to Enable on my gateway, my mistake.
Asterisk is running ok.

Thx