I use ADAPTER "Eusso UTG7104"
VAD turn is off
dtmf_relay turn is on
this debug :
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
<-- SIP read from 85.115.115.125:5060:
INVITE sip:9@217.199.111.19 SIP/2.0
From: <sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501378-9c1e94-781
To: <sip:9@217.199.111.19>
Call-ID: 78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19
CSeq: 1 INVITE
Via: SIP/2.0/UDP 85.115.115.125:5060;branch=z9hG4bK-3a501378-9c1e94-3026
Max-Forwards: 70
Contact: <sip:75305101@85.115.115.125:5060>
Content-Type: application/SDP
Content-Length: 130
v=0
o=75305101 481519749 1606819611 IN IP4 85.115.115.125
s=-
c=IN IP4 85.115.115.125
t=0 0
m=audio 2076 RTP/AVP 18 4 0 101
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 0: INVITE
sip:9@217.199.111.19 SIP/2.0 (35)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 1: From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501378-9c1e94-781 (73)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 2: To:
<sip:9@217.199.111.19> (26)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 3: Call-ID:
78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19 (64)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 4: CSeq: 1
INVITE (14)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 5: Via:
SIP/2.0/UDP 85.115.115.125:5060;branch=z9hG4bK-3a501378-9c1e94-3026 (72)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 6:
Max-Forwards: 70 (16)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 7: Contact:
<sip:75305101@85.115.115.125:5060> (43)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 8:
Content-Type: application/SDP (29)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 9:
Content-Length: 130 (19)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 10: (0)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3349 parse_request: Line: v=0 (3)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3349 parse_request: Line: o=75305101
481519749 1606819611 IN IP4 85.115.115.125 (53)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3349 parse_request: Line: s=- (3)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3349 parse_request: Line: c=IN IP4
85.115.115.125 (23)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3349 parse_request: Line: t=0 0 (5)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3349 parse_request: Line: m=audio 2076
RTP/AVP 18 4 0 101 (31)
--- (10 headers 6 lines)---
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3149 find_call: = No match Their Call
ID: 5573737d-13c4-3a4feb87-1918-48a1 Their Tag
5573737d-13c4-3a50136d-9bf2cc-7811 Our tag: as73f26ac8
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3101 sip_alloc: Allocating new SIP
dialog for 78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19 - INVITE
(With RTP)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:10906 handle_request: **** Received
INVITE (5) - Command in SIP INVITE
Using INVITE request as basis request -
78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19
Sending to 85.115.115.125 : 5060 (non-NAT)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:7007 check_user_full: Setting NAT on
RTP to 0
Found user '75305101'
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 85.115.115.125:2076
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3501 process_sdp: Peer audio RTP is at
port 85.115.115.125:2076
Capabilities: us - 0x100 (g729), peer - audio=0x105 (g723|ulaw|g729)/video=0x0
(nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:10317 handle_request_invite: Checking
SIP call limits for device 75305101
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:2195 update_call_counter: Updating call
counter for incoming call
Looking for 9 in home (domain 217.199.111.19)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:6022 build_route: build_route: Contact
hop: <sip:75305101@85.115.115.125:5060>
list_route: hop: <sip:75305101@85.115.115.125:5060>
Transmitting (no NAT) to 85.115.115.125:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501378-9c1e94-3026;received=85.115.115.125
From: <sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501378-9c1e94-781
To: <sip:9@217.199.111.19>
Call-ID: 78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:9@217.199.111.19>
Content-Length: 0
---
Nov 16 02:52:03 DEBUG[22717]: chan_sip.c:11428 sip_devicestate: Checking device
state for peer 75305101
Nov 16 02:52:03 DEBUG[22717]: devicestate.c:187 do_state_change: Changing state
for SIP/75305101 - state 2 (In use)
Nov 16 02:52:03 DEBUG[22818]: pbx.c:1667 pbx_extension_helper: Launching
'Answer'
-- Executing Answer("SIP/75305101-5b84", "") in new
stack
Nov 16 02:52:03 DEBUG[22819]: app_queue.c:471 changethread: Device
'SIP/75305101' changed to state '2' (In use)
Nov 16 02:52:03 DEBUG[22818]: chan_sip.c:2505 sip_answer:
sip_answer(SIP/75305101-5b84)
We're at 217.199.111.19 port 15292
Adding codec 0x100 (g729) to SDP
Nov 16 02:52:03 DEBUG[22717]: chan_sip.c:11428 sip_devicestate: Checking device
state for peer 75305101
Adding non-codec 0x1 (telephone-event) to SDP
Nov 16 02:52:03 DEBUG[22717]: channel.c:770 channel_find_locked: Avoiding
initial deadlock for 'SIP/75305101-5b84'
Reliably Transmitting (no NAT) to 85.115.115.125:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501378-9c1e94-3026;received=85.115.115.125
From: <sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501378-9c1e94-781
To: <sip:9@217.199.111.19>;tag=as6aeea459
Call-ID: 78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:9@217.199.111.19>
Content-Type: application/sdp
Content-Length: 222
v=0
o=root 22818 22818 IN IP4 217.199.111.19
s=session
c=IN IP4 217.199.111.19
t=0 0
m=audio 15292 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Nov 16 02:52:03 DEBUG[22818]: chan_sip.c:1284 __sip_reliable_xmit: *** SIP
TIMER: Initalizing retransmit timer on packet: Id #54
Nov 16 02:52:03 DEBUG[22717]: devicestate.c:187 do_state_change: Changing state
for SIP/75305101 - state 2 (In use)
Nov 16 02:52:03 DEBUG[22820]: app_queue.c:471 changethread: Device
'SIP/75305101' changed to state '2' (In use)
<-- SIP read from 85.115.115.125:5060:
ACK sip:9@217.199.111.19 SIP/2.0
From: <sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501378-9c1e94-781
To: <sip:9@217.199.111.19>;tag=as6aeea459
Call-ID: 78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19
CSeq: 1 ACK
Via: SIP/2.0/UDP 85.115.115.125:5060;branch=z9hG4bK-3a501378-9c1f1c-4467
Max-Forwards: 70
Contact: <sip:75305101@85.115.115.125:5060>
Content-Length: 0
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 0: ACK
sip:9@217.199.111.19 SIP/2.0 (32)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 1: From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501378-9c1e94-781 (73)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 2: To:
<sip:9@217.199.111.19>;tag=as6aeea459 (41)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 3: Call-ID:
78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19 (64)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 4: CSeq: 1
ACK (11)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 5: Via:
SIP/2.0/UDP 85.115.115.125:5060;branch=z9hG4bK-3a501378-9c1f1c-4467 (72)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 6:
Max-Forwards: 70 (16)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 7: Contact:
<sip:75305101@85.115.115.125:5060> (43)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 8:
Content-Length: 0 (17)
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 9: (0)
--- (9 headers 0 lines)---
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3149 find_call: = Found Their Call ID:
78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19 Their Tag
5573737d-13c4-3a501378-9c1e94-781 Our tag: as6aeea459
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:10906 handle_request: **** Received ACK
(6) - Command in SIP ACK
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:1378 __sip_ack: ** SIP TIMER:
Cancelling retransmit of packet (reply received) Retransid #54
Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:1389 __sip_ack: Stopping retransmission
on '78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19' of Response
1: Match Found
Nov 16 02:52:07 DEBUG[22720]: chan_sip.c:1311 __sip_autodestruct: Auto
destroying call '5573737d-13c4-3a4feb87-1918-48a1'
Destroying call '5573737d-13c4-3a4feb87-1918-48a1'
*CLI>
*CLI>
*CLI>
*CLI> Nov 16 02:54:52 NOTICE[22874]: rtp.c:330 process_rfc3389: Comfort noise
support incomplete in Asterisk (RFC 3389). Please turn off on client if
possible. Client IP: 85.115.115.125
<-- SIP read from 85.115.115.125:5060:
INFO sip:9@217.199.111.19 SIP/2.0
From: <sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a50141c-9e9f56-29b8
To: <sip:9@217.199.111.19>;tag=as2af78cfd
Call-ID: 7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19
CSeq: 2 INFO
Via: SIP/2.0/UDP 85.115.115.125:5060;branch=z9hG4bK-3a501421-9eb4ae-2964
Max-Forwards: 70
Contact: <sip:75305101@85.115.115.125:5060>
Content-Type: application/dtmf-relay
Content-Length: 24
Signal=2
Duration=250
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 0: INFO
sip:9@217.199.111.19 SIP/2.0 (33)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 1: From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a50141c-9e9f56-29b8 (74)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 2: To:
<sip:9@217.199.111.19>;tag=as2af78cfd (41)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3: Call-ID:
7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19 (64)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4: CSeq: 2
INFO (12)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 5: Via:
SIP/2.0/UDP 85.115.115.125:5060;branch=z9hG4bK-3a501421-9eb4ae-2964 (72)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6:
Max-Forwards: 70 (16)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 7: Contact:
<sip:75305101@85.115.115.125:5060> (43)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8:
Content-Type: application/dtmf-relay (36)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 9:
Content-Length: 24 (18)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 10: (0)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: Signal=2 (8)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: Duration=250
(12)
--- (10 headers 2 lines)---
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3149 find_call: = Found Their Call ID:
7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19 Their Tag
5573737d-13c4-3a50141c-9e9f56-29b8 Our tag: as2af78cfd
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:10906 handle_request: **** Received
INFO (13) - Command in SIP INFO
Receiving INFO!
* DTMF-relay event received: '2'
Transmitting (no NAT) to 85.115.115.125:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501421-9eb4ae-2964;received=85.115.115.125
From: <sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a50141c-9e9f56-29b8
To: <sip:9@217.199.111.19>;tag=as2af78cfd
Call-ID: 7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19
CSeq: 2 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:9@217.199.111.19>
Content-Length: 0
---
== CDR updated on SIP/75305101-0ee9
Nov 16 02:54:52 DEBUG[22874]: pbx.c:1667 pbx_extension_helper: Launching
'Dial'
-- Executing Dial("SIP/75305101-0ee9",
"SIP/201@outbound") in new stack
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3101 sip_alloc: Allocating new SIP
dialog for (No Call-ID) - INVITE (With RTP)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:1860 create_addr_from_peer: Setting NAT
on RTP to 0
Nov 16 02:54:52 DEBUG[22874]: channel.c:2793 ast_channel_inherit_variables: Not
copying variable STACK-home-2-1.
Nov 16 02:54:52 DEBUG[22874]: channel.c:2793 ast_channel_inherit_variables: Not
copying variable STACK-home-9-1.
Nov 16 02:54:52 DEBUG[22874]: channel.c:2793 ast_channel_inherit_variables: Not
copying variable SIPCALLID.
Nov 16 02:54:52 DEBUG[22874]: channel.c:2793 ast_channel_inherit_variables: Not
copying variable SIPDOMAIN.
Nov 16 02:54:52 DEBUG[22874]: channel.c:2793 ast_channel_inherit_variables: Not
copying variable SIPURI.
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:2054 sip_call: Outgoing Call for 201
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:2195 update_call_counter: Updating call
counter for outgoing call
We're at 217.199.111.19 port 14292
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 0: INVITE
sip:201@85.115.115.125 SIP/2.0 (37)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 1: Via:
SIP/2.0/UDP 217.199.111.19:5060;branch=z9hG4bK2776613f;rport (65)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 2: From:
"75305101" <sip:75305101@217.199.111.19>;tag=as7fe1eb18 (61)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 3: To:
<sip:201@85.115.115.125> (28)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 4: Contact:
<sip:75305101@217.199.111.19> (38)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 5: Call-ID:
5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19 (56)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 6: CSeq: 102
INVITE (16)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 7:
User-Agent: Asterisk PBX (24)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 8:
Max-Forwards: 70 (16)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 9: Date:
Wed, 16 Nov 2005 00:54:52 GMT (35)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 10: Allow:
INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 11:
Content-Type: application/sdp (29)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 12:
Content-Length: 222 (19)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 13: (0)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: v=0 (3)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: o=root 22874
22874 IN IP4 217.199.111.19 (40)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: s=session (9)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: c=IN IP4
217.199.111.19 (23)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: t=0 0 (5)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: m=audio 14292
RTP/AVP 18 101 (28)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: a=rtpmap:18
G729/8000 (21)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: a=rtpmap:101
telephone-event/8000 (33)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: a=fmtp:101
0-16 (15)
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3349 parse_request: Line:
a=silenceSupp:off - - - - (25)
13 headers, 10 lines
Reliably Transmitting (no NAT) to 85.115.115.125:5060:
INVITE sip:201@85.115.115.125 SIP/2.0
Via: SIP/2.0/UDP 217.199.111.19:5060;branch=z9hG4bK2776613f;rport
From: "75305101" <sip:75305101@217.199.111.19>;tag=as7fe1eb18
To: <sip:201@85.115.115.125>
Contact: <sip:75305101@217.199.111.19>
Call-ID: 5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 16 Nov 2005 00:54:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 222
v=0
o=root 22874 22874 IN IP4 217.199.111.19
s=session
c=IN IP4 217.199.111.19
t=0 0
m=audio 14292 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:1284 __sip_reliable_xmit: *** SIP
TIMER: Initalizing retransmit timer on packet: Id #10
-- Called 201@outbound
Nov 16 02:54:52 DEBUG[22874]: channel.c:2331 set_format: Set channel
SIP/outbound-0ea4 to read format g729
Nov 16 02:54:52 DEBUG[22874]: channel.c:2331 set_format: Set channel
SIP/75305101-0ee9 to write format g729
Nov 16 02:54:52 DEBUG[22874]: channel.c:2331 set_format: Set channel
SIP/75305101-0ee9 to read format g729
Nov 16 02:54:52 DEBUG[22874]: channel.c:2331 set_format: Set channel
SIP/outbound-0ea4 to write format g729
<-- SIP read from 85.115.115.125:5060:
SIP/2.0 180 Ringing
From: "75305101"<sip:75305101@217.199.111.19>;tag=as7fe1eb18
To: <sip:201@85.115.115.125>;tag=5573737d-13c4-3a501422-9eb520-5b93
Call-ID: 5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19
CSeq: 102 INVITE
Via: SIP/2.0/UDP 217.199.111.19:5060;rport=5060;branch=z9hG4bK2776613f
Contact: <sip:201@85.115.115.125:5060>
Content-Length: 0
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 0: SIP/2.0
180 Ringing (19)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 1: From:
"75305101"<sip:75305101@217.199.111.19>;tag=as7fe1eb18 (60)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 2: To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501422-9eb520-5b93 (67)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3: Call-ID:
5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19 (56)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4: CSeq: 102
INVITE (16)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 5: Via:
SIP/2.0/UDP 217.199.111.19:5060;rport=5060;branch=z9hG4bK2776613f (70)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6: Contact:
<sip:201@85.115.115.125:5060> (38)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 7:
Content-Length: 0 (17)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8: (0)
--- (8 headers 0 lines)---
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3149 find_call: = Found Their Call ID:
5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19 Their Tag Our tag: as7fe1eb18
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:1433 __sip_semi_ack: *** SIP TIMER:
Cancelling retransmission #10 - INVITE (got response)
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:1442 __sip_semi_ack: (Provisional)
Stopping retransmission (but retaining packet) on
'5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19' Request 102: Found
Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:9423 handle_response_invite: SIP
response 180 to standard invite
Nov 16 02:54:52 DEBUG[22866]: chan_sip.c:11428 sip_devicestate: Checking device
state for peer outbound
Nov 16 02:54:52 DEBUG[22866]: devicestate.c:187 do_state_change: Changing state
for SIP/outbound - state 6 (Ringing)
Nov 16 02:54:52 DEBUG[22877]: app_queue.c:471 changethread: Device
'SIP/outbound' changed to state '6' (Ringing)
-- SIP/outbound-0ea4 is ringing
Nov 16 02:54:52 DEBUG[22874]: channel.c:2017 ast_indicate: Driver for channel
'SIP/75305101-0ee9' does not support indication 3, emulating it
Nov 16 02:54:52 DEBUG[22874]: channel.c:2331 set_format: Set channel
SIP/75305101-0ee9 to write format slin
Nov 16 02:54:52 DEBUG[22874]: rtp.c:1341 ast_rtp_write: Ooh, format changed from
unknown to g729
<-- SIP read from 85.115.115.125:5060:
SIP/2.0 200 OK
From: "75305101"<sip:75305101@217.199.111.19>;tag=as7fe1eb18
To: <sip:201@85.115.115.125>;tag=5573737d-13c4-3a501422-9eb520-5b93
Call-ID: 5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19
CSeq: 102 INVITE
Via: SIP/2.0/UDP 217.199.111.19:5060;rport=5060;branch=z9hG4bK2776613f
Contact: <sip:201@85.115.115.125:5060>
Content-Type: application/SDP
Content-Length: 116
v=0
o=201 812119400 215593963 IN IP4 85.115.115.125
s=-
c=IN IP4 85.115.115.125
t=0 0
m=audio 2074 RTP/AVP 18
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 0: SIP/2.0
200 OK (14)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 1: From:
"75305101"<sip:75305101@217.199.111.19>;tag=as7fe1eb18 (60)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 2: To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501422-9eb520-5b93 (67)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3: Call-ID:
5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19 (56)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4: CSeq: 102
INVITE (16)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 5: Via:
SIP/2.0/UDP 217.199.111.19:5060;rport=5060;branch=z9hG4bK2776613f (70)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6: Contact:
<sip:201@85.115.115.125:5060> (38)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 7:
Content-Type: application/SDP (29)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8:
Content-Length: 116 (19)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 9: (0)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: v=0 (3)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: o=201
812119400 215593963 IN IP4 85.115.115.125 (47)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: s=- (3)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: c=IN IP4
85.115.115.125 (23)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: t=0 0 (5)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: m=audio 2074
RTP/AVP 18 (23)
--- (9 headers 6 lines)---
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3149 find_call: = Found Their Call ID:
5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19 Their Tag
5573737d-13c4-3a501422-9eb520-5b93 Our tag: as7fe1eb18
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:1367 __sip_ack: Acked pending invite
102
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:1389 __sip_ack: Stopping retransmission
on '5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19' of Request 102:
Match Found
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:9423 handle_response_invite: SIP
response 200 to standard invite
Found RTP audio format 18
Peer audio RTP is at port 85.115.115.125:2074
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3501 process_sdp: Peer audio RTP is at
port 85.115.115.125:2074
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing),
combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:6022 build_route: build_route: Contact
hop: <sip:201@85.115.115.125:5060>
list_route: hop: <sip:201@85.115.115.125:5060>
set_destination: Parsing <sip:201@85.115.115.125:5060> for address/port to
send to
set_destination: set destination to 85.115.115.125, port 5060
Transmitting (no NAT) to 85.115.115.125:5060:
ACK sip:201@85.115.115.125:5060 SIP/2.0
Via: SIP/2.0/UDP 217.199.111.19:5060;branch=z9hG4bK7d1af83f;rport
From: "75305101" <sip:75305101@217.199.111.19>;tag=as7fe1eb18
To: <sip:201@85.115.115.125>;tag=5573737d-13c4-3a501422-9eb520-5b93
Contact: <sip:75305101@217.199.111.19>
Call-ID: 5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Nov 16 02:54:53 DEBUG[22874]: rtp.c:1341 ast_rtp_write: Ooh, format changed from
unknown to g729
-- SIP/outbound-0ea4 answered SIP/75305101-0ee9
Nov 16 02:54:53 DEBUG[22874]: channel.c:2331 set_format: Set channel
SIP/75305101-0ee9 to write format g729
Nov 16 02:54:53 DEBUG[22874]: channel.c:2331 set_format: Set channel
SIP/75305101-0ee9 to read format g729
Nov 16 02:54:53 DEBUG[22874]: channel.c:2331 set_format: Set channel
SIP/outbound-0ea4 to write format g729
Nov 16 02:54:53 DEBUG[22874]: channel.c:2331 set_format: Set channel
SIP/outbound-0ea4 to read format g729
Nov 16 02:54:53 DEBUG[22874]: channel.c:2331 set_format: Set channel
SIP/75305101-0ee9 to write format g729
-- Attempting native bridge of SIP/75305101-0ee9 and SIP/outbound-0ea4
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:12662 sip_set_rtp_peer: Sending
reinvite on SIP
'7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19' - It's
audio soon redirected to IP 85.115.115.125
set_destination: Parsing <sip:75305101@85.115.115.125:5060> for
address/port to send to
set_destination: set destination to 85.115.115.125, port 5060
We're at 217.199.111.19 port 14712
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 0: INVITE
sip:75305101@85.115.115.125:5060 SIP/2.0 (47)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 1: Via:
SIP/2.0/UDP 217.199.111.19:5060;branch=z9hG4bK38bafc54;rport (65)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 2: From:
<sip:9@217.199.111.19>;tag=as2af78cfd (43)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 3: To:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a50141c-9e9f56-29b8 (72)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 4: Contact:
<sip:9@217.199.111.19> (31)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 5: Call-ID:
7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19 (64)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 6: CSeq: 102
INVITE (16)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 7:
User-Agent: Asterisk PBX (24)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 8:
Max-Forwards: 70 (16)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 9: Allow:
INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 10:
X-asterisk-info: SIP re-invite (RTP bridge) (43)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 11:
Content-Type: application/sdp (29)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 12:
Content-Length: 221 (19)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 13: (0)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: v=0 (3)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: o=root 22874
22875 IN IP4 85.115.115.125 (40)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: s=session (9)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: c=IN IP4
85.115.115.125 (23)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: t=0 0 (5)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: m=audio 2074
RTP/AVP 18 101 (27)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: a=rtpmap:18
G729/8000 (21)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: a=rtpmap:101
telephone-event/8000 (33)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: a=fmtp:101
0-16 (15)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line:
a=silenceSupp:off - - - - (25)
13 headers, 10 lines
Reliably Transmitting (no NAT) to 85.115.115.125:5060:
INVITE sip:75305101@85.115.115.125:5060 SIP/2.0
Via: SIP/2.0/UDP 217.199.111.19:5060;branch=z9hG4bK38bafc54;rport
From: <sip:9@217.199.111.19>;tag=as2af78cfd
To: <sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a50141c-9e9f56-29b8
Contact: <sip:9@217.199.111.19>
Call-ID: 7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 221
v=0
o=root 22874 22875 IN IP4 85.115.115.125
s=session
c=IN IP4 85.115.115.125
t=0 0
m=audio 2074 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:1284 __sip_reliable_xmit: *** SIP
TIMER: Initalizing retransmit timer on packet: Id #11
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:12662 sip_set_rtp_peer: Sending
reinvite on SIP '5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19' -
It's audio soon redirected to IP 85.115.115.125
set_destination: Parsing <sip:201@85.115.115.125:5060> for address/port to
send to
set_destination: set destination to 85.115.115.125, port 5060
We're at 217.199.111.19 port 14292
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x4 (ulaw) to SDP
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 0: INVITE
sip:201@85.115.115.125:5060 SIP/2.0 (42)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 1: Via:
SIP/2.0/UDP 217.199.111.19:5060;branch=z9hG4bK33f70e49;rport (65)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 2: From:
"75305101" <sip:75305101@217.199.111.19>;tag=as7fe1eb18 (61)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 3: To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501422-9eb520-5b93 (67)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 4: Contact:
<sip:75305101@217.199.111.19> (38)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 5: Call-ID:
5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19 (56)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 6: CSeq: 103
INVITE (16)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 7:
User-Agent: Asterisk PBX (24)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 8:
Max-Forwards: 70 (16)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 9: Allow:
INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 10:
X-asterisk-info: SIP re-invite (RTP bridge) (43)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 11:
Content-Type: application/sdp (29)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 12:
Content-Length: 213 (19)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 13: (0)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: v=0 (3)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: o=root 22874
22875 IN IP4 85.115.115.125 (40)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: s=session (9)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: c=IN IP4
85.115.115.125 (23)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: t=0 0 (5)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: m=audio 2070
RTP/AVP 18 4 0 (27)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: a=rtpmap:18
G729/8000 (21)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: a=rtpmap:4
G723/8000 (20)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: a=rtpmap:0
PCMU/8000 (20)
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line:
a=silenceSupp:off - - - - (25)
13 headers, 10 lines
Reliably Transmitting (no NAT) to 85.115.115.125:5060:
INVITE sip:201@85.115.115.125:5060 SIP/2.0
Via: SIP/2.0/UDP 217.199.111.19:5060;branch=z9hG4bK33f70e49;rport
From: "75305101" <sip:75305101@217.199.111.19>;tag=as7fe1eb18
To: <sip:201@85.115.115.125>;tag=5573737d-13c4-3a501422-9eb520-5b93
Contact: <sip:75305101@217.199.111.19>
Call-ID: 5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 213
v=0
o=root 22874 22875 IN IP4 85.115.115.125
s=session
c=IN IP4 85.115.115.125
t=0 0
m=audio 2070 RTP/AVP 18 4 0
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
---
Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:1284 __sip_reliable_xmit: *** SIP
TIMER: Initalizing retransmit timer on packet: Id #12
Nov 16 02:54:53 DEBUG[22866]: chan_sip.c:11428 sip_devicestate: Checking device
state for peer outbound
Nov 16 02:54:53 DEBUG[22866]: devicestate.c:187 do_state_change: Changing state
for SIP/outbound - state 2 (In use)
Nov 16 02:54:53 DEBUG[22878]: app_queue.c:471 changethread: Device
'SIP/outbound' changed to state '2' (In use)
<-- SIP read from 85.115.115.125:5060:
SIP/2.0 200 OK
From: <sip:9@217.199.111.19>;tag=as2af78cfd
To: <sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a50141c-9e9f56-29b8
Call-ID: 7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19
CSeq: 102 INVITE
Via: SIP/2.0/UDP 217.199.111.19:5060;rport=5060;branch=z9hG4bK38bafc54
Contact: <sip:75305101@85.115.115.125:5060>
Content-Type: application/SDP
Content-Length: 121
v=0
o=75305101 177701337 159505494 IN IP4 85.115.115.125
s=-
c=IN IP4 85.115.115.125
t=0 0
m=audio 2070 RTP/AVP 18
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 0: SIP/2.0
200 OK (14)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 1: From:
<sip:9@217.199.111.19>;tag=as2af78cfd (43)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 2: To:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a50141c-9e9f56-29b8 (72)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3: Call-ID:
7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19 (64)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4: CSeq: 102
INVITE (16)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 5: Via:
SIP/2.0/UDP 217.199.111.19:5060;rport=5060;branch=z9hG4bK38bafc54 (70)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6: Contact:
<sip:75305101@85.115.115.125:5060> (43)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 7:
Content-Type: application/SDP (29)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8:
Content-Length: 121 (19)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 9: (0)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: v=0 (3)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: o=75305101
177701337 159505494 IN IP4 85.115.115.125 (52)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: s=- (3)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: c=IN IP4
85.115.115.125 (23)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: t=0 0 (5)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: m=audio 2070
RTP/AVP 18 (23)
--- (9 headers 6 lines)---
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3149 find_call: = No match Their Call
ID: 5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19 Their Tag
5573737d-13c4-3a501422-9eb520-5b93 Our tag: as7fe1eb18
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3149 find_call: = Found Their Call ID:
7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19 Their Tag
5573737d-13c4-3a50141c-9e9f56-29b8 Our tag: as2af78cfd
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:1367 __sip_ack: Acked pending invite
102
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:1378 __sip_ack: ** SIP TIMER:
Cancelling retransmit of packet (reply received) Retransid #11
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:1389 __sip_ack: Stopping retransmission
on '7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19' of Request
102: Match Found
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:9421 handle_response_invite: SIP
response 200 to RE-invite on outgoing call
7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19
Found RTP audio format 18
Peer audio RTP is at port 85.115.115.125:2070
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3501 process_sdp: Peer audio RTP is at
port 85.115.115.125:2070
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing),
combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:6022 build_route: build_route: Contact
hop: <sip:75305101@85.115.115.125:5060>
list_route: hop: <sip:75305101@85.115.115.125:5060>
set_destination: Parsing <sip:75305101@85.115.115.125:5060> for
address/port to send to
set_destination: set destination to 85.115.115.125, port 5060
Transmitting (no NAT) to 85.115.115.125:5060:
ACK sip:75305101@85.115.115.125:5060 SIP/2.0
Via: SIP/2.0/UDP 217.199.111.19:5060;branch=z9hG4bK4cfadd2d;rport
From: <sip:9@217.199.111.19>;tag=as2af78cfd
To: <sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a50141c-9e9f56-29b8
Contact: <sip:9@217.199.111.19>
Call-ID: 7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
<-- SIP read from 85.115.115.125:5060:
SIP/2.0 200 OK
From: "75305101"<sip:75305101@217.199.111.19>;tag=as7fe1eb18
To: <sip:201@85.115.115.125>;tag=5573737d-13c4-3a501422-9eb520-5b93
Call-ID: 5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19
CSeq: 103 INVITE
Via: SIP/2.0/UDP 217.199.111.19:5060;rport=5060;branch=z9hG4bK33f70e49
Contact: <sip:201@85.115.115.125:5060>
Content-Type: application/SDP
Content-Length: 116
v=0
o=201 83317681 1732424444 IN IP4 85.115.115.125
s=-
c=IN IP4 85.115.115.125
t=0 0
m=audio 2074 RTP/AVP 18
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 0: SIP/2.0
200 OK (14)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 1: From:
"75305101"<sip:75305101@217.199.111.19>;tag=as7fe1eb18 (60)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 2: To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501422-9eb520-5b93 (67)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3: Call-ID:
5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19 (56)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4: CSeq: 103
INVITE (16)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 5: Via:
SIP/2.0/UDP 217.199.111.19:5060;rport=5060;branch=z9hG4bK33f70e49 (70)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6: Contact:
<sip:201@85.115.115.125:5060> (38)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 7:
Content-Type: application/SDP (29)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8:
Content-Length: 116 (19)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 9: (0)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: v=0 (3)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: o=201
83317681 1732424444 IN IP4 85.115.115.125 (47)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: s=- (3)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: c=IN IP4
85.115.115.125 (23)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: t=0 0 (5)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: m=audio 2074
RTP/AVP 18 (23)
--- (9 headers 6 lines)---
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3149 find_call: = Found Their Call ID:
5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19 Their Tag
5573737d-13c4-3a501422-9eb520-5b93 Our tag: as7fe1eb18
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:1367 __sip_ack: Acked pending invite
103
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:1378 __sip_ack: ** SIP TIMER:
Cancelling retransmit of packet (reply received) Retransid #12
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:1389 __sip_ack: Stopping retransmission
on '5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19' of Request 103:
Match Found
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:9421 handle_response_invite: SIP
response 200 to RE-invite on outgoing call
5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19
Found RTP audio format 18
Peer audio RTP is at port 85.115.115.125:2074
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3501 process_sdp: Peer audio RTP is at
port 85.115.115.125:2074
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing),
combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:5965 build_route: build_route:
Retaining previous route: <sip:201@85.115.115.125:5060>
set_destination: Parsing <sip:201@85.115.115.125:5060> for address/port to
send to
set_destination: set destination to 85.115.115.125, port 5060
Transmitting (no NAT) to 85.115.115.125:5060:
ACK sip:201@85.115.115.125:5060 SIP/2.0
Via: SIP/2.0/UDP 217.199.111.19:5060;branch=z9hG4bK038520a8;rport
From: "75305101" <sip:75305101@217.199.111.19>;tag=as7fe1eb18
To: <sip:201@85.115.115.125>;tag=5573737d-13c4-3a501422-9eb520-5b93
Contact: <sip:75305101@217.199.111.19>
Call-ID: 5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
<-- SIP read from 85.115.115.125:5060:
INFO sip:9@217.199.111.19 SIP/2.0
From: <sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301
To: <sip:9@217.199.111.19>;tag=as53452dbd
Call-ID: 78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19
CSeq: 3 INFO
Via: SIP/2.0/UDP 85.115.115.125:5060;branch=z9hG4bK-3a501447-9f46e8-612c
Max-Forwards: 70
Contact: <sip:75305101@85.115.115.125:5060>
Content-Type: application/dtmf-relay
Content-Length: 24
Signal=6
Duration=250
Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 0: INFO
sip:9@217.199.111.19 SIP/2.0 (33)
Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 1: From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301 (74)
Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 2: To:
<sip:9@217.199.111.19>;tag=as53452dbd (41)
Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3: Call-ID:
78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19 (65)
Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4: CSeq: 3
INFO (12)
Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 5: Via:
SIP/2.0/UDP 85.115.115.125:5060;branch=z9hG4bK-3a501447-9f46e8-612c (72)
Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6:
Max-Forwards: 70 (16)
Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 7: Contact:
<sip:75305101@85.115.115.125:5060> (43)
Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8:
Content-Type: application/dtmf-relay (36)
Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 9:
Content-Length: 24 (18)
Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 10: (0)
Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: Signal=6 (8)
Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: Duration=250
(12)
--- (10 headers 2 lines)---
Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3149 find_call: = No match Their Call
ID: 05398b7e73a15e06401d749c76755a70@217.199.111.19 Their Tag
5573737d-13c4-3a501443-9f36d0-7500 Our tag: as5550c7ca
Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3149 find_call: = Found Their Call ID:
78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19 Their Tag
5573737d-13c4-3a501440-9f2e0a-7301 Our tag: as53452dbd
Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:10906 handle_request: **** Received
INFO (13) - Command in SIP INFO
Receiving INFO!
* DTMF-relay event received: '6'
Transmitting (no NAT) to 85.115.115.125:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501447-9f46e8-612c;received=85.115.115.125
From: <sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301
To: <sip:9@217.199.111.19>;tag=as53452dbd
Call-ID: 78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19
CSeq: 3 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:9@217.199.111.19>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
---
<-- SIP read from 85.115.115.125:5060:
INFO sip:9@217.199.111.19 SIP/2.0
From: <sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301
To: <sip:9@217.199.111.19>;tag=as53452dbd
Call-ID: 78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19
CSeq: 4 INFO
Via: SIP/2.0/UDP 85.115.115.125:5060;branch=z9hG4bK-3a501448-9f4c64-698a
Max-Forwards: 70
Contact: <sip:75305101@85.115.115.125:5060>
Content-Type: application/dtmf-relay
Content-Length: 24
Signal=5
Duration=250
Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 0: INFO
sip:9@217.199.111.19 SIP/2.0 (33)
Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 1: From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301 (74)
Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 2: To:
<sip:9@217.199.111.19>;tag=as53452dbd (41)
Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3: Call-ID:
78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19 (65)
Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4: CSeq: 4
INFO (12)
Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 5: Via:
SIP/2.0/UDP 85.115.115.125:5060;branch=z9hG4bK-3a501448-9f4c64-698a (72)
Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6:
Max-Forwards: 70 (16)
Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 7: Contact:
<sip:75305101@85.115.115.125:5060> (43)
Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8:
Content-Type: application/dtmf-relay (36)
Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 9:
Content-Length: 24 (18)
Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 10: (0)
Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: Signal=5 (8)
Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: Duration=250
(12)
--- (10 headers 2 lines)---
Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3149 find_call: = No match Their Call
ID: 05398b7e73a15e06401d749c76755a70@217.199.111.19 Their Tag
5573737d-13c4-3a501443-9f36d0-7500 Our tag: as5550c7ca
Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3149 find_call: = Found Their Call ID:
78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19 Their Tag
5573737d-13c4-3a501440-9f2e0a-7301 Our tag: as53452dbd
Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:10906 handle_request: **** Received
INFO (13) - Command in SIP INFO
Receiving INFO!
* DTMF-relay event received: '5'
Transmitting (no NAT) to 85.115.115.125:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501448-9f4c64-698a;received=85.115.115.125
From: <sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301
To: <sip:9@217.199.111.19>;tag=as53452dbd
Call-ID: 78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19
CSeq: 4 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:9@217.199.111.19>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
---
<-- SIP read from 85.115.115.125:5060:
INFO sip:9@217.199.111.19 SIP/2.0
From: <sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301
To: <sip:9@217.199.111.19>;tag=as53452dbd
Call-ID: 78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19
CSeq: 5 INFO
Via: SIP/2.0/UDP 85.115.115.125:5060;branch=z9hG4bK-3a50144a-9f52e2-6418
Max-Forwards: 70
Contact: <sip:75305101@85.115.115.125:5060>
Content-Type: application/dtmf-relay
Content-Length: 24
Signal=4
Duration=250
Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 0: INFO
sip:9@217.199.111.19 SIP/2.0 (33)
Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 1: From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301 (74)
Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 2: To:
<sip:9@217.199.111.19>;tag=as53452dbd (41)
Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3: Call-ID:
78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19 (65)
Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4: CSeq: 5
INFO (12)
Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 5: Via:
SIP/2.0/UDP 85.115.115.125:5060;branch=z9hG4bK-3a50144a-9f52e2-6418 (72)
Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6:
Max-Forwards: 70 (16)
Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 7: Contact:
<sip:75305101@85.115.115.125:5060> (43)
Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8:
Content-Type: application/dtmf-relay (36)
Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 9:
Content-Length: 24 (18)
Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 10: (0)
Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: Signal=4 (8)
Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: Duration=250
(12)
--- (10 headers 2 lines)---
Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3149 find_call: = No match Their Call
ID: 05398b7e73a15e06401d749c76755a70@217.199.111.19 Their Tag
5573737d-13c4-3a501443-9f36d0-7500 Our tag: as5550c7ca
Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3149 find_call: = Found Their Call ID:
78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19 Their Tag
5573737d-13c4-3a501440-9f2e0a-7301 Our tag: as53452dbd
Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:10906 handle_request: **** Received
INFO (13) - Command in SIP INFO
Receiving INFO!
* DTMF-relay event received: '4'
Transmitting (no NAT) to 85.115.115.125:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a50144a-9f52e2-6418;received=85.115.115.125
From: <sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301
To: <sip:9@217.199.111.19>;tag=as53452dbd
Call-ID: 78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19
CSeq: 5 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:9@217.199.111.19>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
---
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
<-- SIP read from 85.115.115.125:5060:
BYE sip:9@217.199.111.19 SIP/2.0
From: <sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301
To: <sip:9@217.199.111.19>;tag=as53452dbd
Call-ID: 78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19
CSeq: 6 BYE
Via: SIP/2.0/UDP 85.115.115.125:5060;branch=z9hG4bK-3a501454-9f7888-1456
Max-Forwards: 70
Content-Length: 0
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 0: BYE
sip:9@217.199.111.19 SIP/2.0 (32)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 1: From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301 (74)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 2: To:
<sip:9@217.199.111.19>;tag=as53452dbd (41)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3: Call-ID:
78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19 (65)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4: CSeq: 6
BYE (11)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 5: Via:
SIP/2.0/UDP 85.115.115.125:5060;branch=z9hG4bK-3a501454-9f7888-1456 (72)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6:
Max-Forwards: 70 (16)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 7:
Content-Length: 0 (17)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8: (0)
--- (8 headers 0 lines)---
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3149 find_call: = No match Their Call
ID: 05398b7e73a15e06401d749c76755a70@217.199.111.19 Their Tag
5573737d-13c4-3a501443-9f36d0-7500 Our tag: as5550c7ca
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3149 find_call: = Found Their Call ID:
78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19 Their Tag
5573737d-13c4-3a501440-9f2e0a-7301 Our tag: as53452dbd
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:10906 handle_request: **** Received BYE
(8) - Command in SIP BYE
Sending to 85.115.115.125 : 5060 (non-NAT)
Transmitting (no NAT) to 85.115.115.125:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501454-9f7888-1456;received=85.115.115.125
From: <sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301
To: <sip:9@217.199.111.19>;tag=as53452dbd
Call-ID: 78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19
CSeq: 6 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:9@217.199.111.19>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
---
Nov 16 02:55:42 DEBUG[22889]: rtp.c:1700 ast_rtp_bridge: Oooh, got a hangup
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:12662 sip_set_rtp_peer: Sending
reinvite on SIP '05398b7e73a15e06401d749c76755a70@217.199.111.19' -
It's audio soon redirected to IP 217.199.111.19
set_destination: Parsing <sip:201@85.115.115.125:5060> for address/port to
send to
set_destination: set destination to 85.115.115.125, port 5060
We're at 217.199.111.19 port 12976
Adding codec 0x100 (g729) to SDP
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317 parse_request: Header 0: INVITE
sip:201@85.115.115.125:5060 SIP/2.0 (42)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317 parse_request: Header 1: Via:
SIP/2.0/UDP 217.199.111.19:5060;branch=z9hG4bK58497b12;rport (65)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317 parse_request: Header 2: From:
"75305101" <sip:75305101@217.199.111.19>;tag=as5550c7ca (61)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317 parse_request: Header 3: To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501443-9f36d0-7500 (67)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317 parse_request: Header 4: Contact:
<sip:75305101@217.199.111.19> (38)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317 parse_request: Header 5: Call-ID:
05398b7e73a15e06401d749c76755a70@217.199.111.19 (56)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317 parse_request: Header 6: CSeq: 104
INVITE (16)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317 parse_request: Header 7:
User-Agent: Asterisk PBX (24)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317 parse_request: Header 8:
Max-Forwards: 70 (16)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317 parse_request: Header 9: Allow:
INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317 parse_request: Header 10:
X-asterisk-info: SIP re-invite (RTP bridge) (43)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317 parse_request: Header 11:
Content-Type: application/sdp (29)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317 parse_request: Header 12:
Content-Length: 166 (19)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317 parse_request: Header 13: (0)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3349 parse_request: Line: v=0 (3)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3349 parse_request: Line: o=root 22889
22891 IN IP4 217.199.111.19 (40)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3349 parse_request: Line: s=session (9)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3349 parse_request: Line: c=IN IP4
217.199.111.19 (23)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3349 parse_request: Line: t=0 0 (5)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3349 parse_request: Line: m=audio 12976
RTP/AVP 18 (24)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3349 parse_request: Line: a=rtpmap:18
G729/8000 (21)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3349 parse_request: Line:
a=silenceSupp:off - - - - (25)
13 headers, 8 lines
Reliably Transmitting (no NAT) to 85.115.115.125:5060:
INVITE sip:201@85.115.115.125:5060 SIP/2.0
Via: SIP/2.0/UDP 217.199.111.19:5060;branch=z9hG4bK58497b12;rport
From: "75305101" <sip:75305101@217.199.111.19>;tag=as5550c7ca
To: <sip:201@85.115.115.125>;tag=5573737d-13c4-3a501443-9f36d0-7500
Contact: <sip:75305101@217.199.111.19>
Call-ID: 05398b7e73a15e06401d749c76755a70@217.199.111.19
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 166
v=0
o=root 22889 22891 IN IP4 217.199.111.19
s=session
c=IN IP4 217.199.111.19
t=0 0
m=audio 12976 RTP/AVP 18
a=rtpmap:18 G729/8000
a=silenceSupp:off - - - -
---
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:1284 __sip_reliable_xmit: *** SIP
TIMER: Initalizing retransmit timer on packet: Id #19
Nov 16 02:55:42 DEBUG[22889]: channel.c:3449 ast_channel_bridge: Returning from
native bridge, channels: SIP/75305101-85cf, SIP/outbound-1d4a
Nov 16 02:55:42 DEBUG[22889]: channel.c:1308 ast_hangup: Hanging up channel
'SIP/outbound-1d4a'
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:2401 sip_hangup: Hangup call
SIP/outbound-1d4a, SIP callid 05398b7e73a15e06401d749c76755a70@217.199.111.19)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:2409 sip_hangup:
update_call_counter(201) - decrement call limit counter
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:2195 update_call_counter: Updating call
counter for outgoing call
Nov 16 02:55:42 DEBUG[22889]: app_dial.c:1587 dial_exec_full: Exiting with
DIALSTATUS=ANSWER.
Nov 16 02:55:42 DEBUG[22889]: pbx.c:2306 __ast_pbx_run: Spawn extension
(home,2,1) exited non-zero on 'SIP/75305101-85cf'
Nov 16 02:55:42 DEBUG[22889]: channel.c:1308 ast_hangup: Hanging up channel
'SIP/75305101-85cf'
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:2401 sip_hangup: Hangup call
SIP/75305101-85cf, SIP callid
78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19)
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:2409 sip_hangup:
update_call_counter(75305101) - decrement call limit counter
Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:2195 update_call_counter: Updating call
counter for outgoing call
Nov 16 02:55:42 DEBUG[22866]: chan_sip.c:11428 sip_devicestate: Checking device
state for peer outbound
Nov 16 02:55:42 DEBUG[22866]: devicestate.c:187 do_state_change: Changing state
for SIP/outbound - state 1 (Not in use)
Nov 16 02:55:42 DEBUG[22894]: app_queue.c:471 changethread: Device
'SIP/outbound' changed to state '1' (Not in use)
Nov 16 02:55:42 DEBUG[22866]: chan_sip.c:11428 sip_devicestate: Checking device
state for peer 75305101
Nov 16 02:55:42 DEBUG[22866]: devicestate.c:187 do_state_change: Changing state
for SIP/75305101 - state 1 (Not in use)
Nov 16 02:55:42 DEBUG[22895]: app_queue.c:471 changethread: Device
'SIP/75305101' changed to state '1' (Not in use)
<-- SIP read from 85.115.115.125:5060:
SIP/2.0 200 OK
From: "75305101"<sip:75305101@217.199.111.19>;tag=as5550c7ca
To: <sip:201@85.115.115.125>;tag=5573737d-13c4-3a501443-9f36d0-7500
Call-ID: 05398b7e73a15e06401d749c76755a70@217.199.111.19
CSeq: 104 INVITE
Via: SIP/2.0/UDP 217.199.111.19:5060;rport=5060;branch=z9hG4bK58497b12
Contact: <sip:201@85.115.115.125:5060>
Content-Type: application/SDP
Content-Length: 118
v=0
o=201 2262112151 2598113056 IN IP4 85.115.115.125
s=-
c=IN IP4 85.115.115.125
t=0 0
m=audio 2078 RTP/AVP 18
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 0: SIP/2.0
200 OK (14)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 1: From:
"75305101"<sip:75305101@217.199.111.19>;tag=as5550c7ca (60)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 2: To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501443-9f36d0-7500 (67)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3: Call-ID:
05398b7e73a15e06401d749c76755a70@217.199.111.19 (56)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4: CSeq: 104
INVITE (16)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 5: Via:
SIP/2.0/UDP 217.199.111.19:5060;rport=5060;branch=z9hG4bK58497b12 (70)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6: Contact:
<sip:201@85.115.115.125:5060> (38)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 7:
Content-Type: application/SDP (29)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8:
Content-Length: 118 (19)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 9: (0)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: v=0 (3)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: o=201
2262112151 2598113056 IN IP4 85.115.115.125 (49)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: s=- (3)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: c=IN IP4
85.115.115.125 (23)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: t=0 0 (5)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: m=audio 2078
RTP/AVP 18 (23)
--- (9 headers 6 lines)---
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3149 find_call: = Found Their Call ID:
05398b7e73a15e06401d749c76755a70@217.199.111.19 Their Tag
5573737d-13c4-3a501443-9f36d0-7500 Our tag: as5550c7ca
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:1367 __sip_ack: Acked pending invite
104
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:1378 __sip_ack: ** SIP TIMER:
Cancelling retransmit of packet (reply received) Retransid #19
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:1389 __sip_ack: Stopping retransmission
on '05398b7e73a15e06401d749c76755a70@217.199.111.19' of Request 104:
Match Found
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:9423 handle_response_invite: SIP
response 200 to standard invite
Found RTP audio format 18
Peer audio RTP is at port 85.115.115.125:2078
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3501 process_sdp: Peer audio RTP is at
port 85.115.115.125:2078
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing),
combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:5965 build_route: build_route:
Retaining previous route: <sip:201@85.115.115.125:5060>
set_destination: Parsing <sip:201@85.115.115.125:5060> for address/port to
send to
set_destination: set destination to 85.115.115.125, port 5060
Transmitting (no NAT) to 85.115.115.125:5060:
ACK sip:201@85.115.115.125:5060 SIP/2.0
Via: SIP/2.0/UDP 217.199.111.19:5060;branch=z9hG4bK688850f1;rport
From: "75305101" <sip:75305101@217.199.111.19>;tag=as5550c7ca
To: <sip:201@85.115.115.125>;tag=5573737d-13c4-3a501443-9f36d0-7500
Contact: <sip:75305101@217.199.111.19>
Call-ID: 05398b7e73a15e06401d749c76755a70@217.199.111.19
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
set_destination: Parsing <sip:201@85.115.115.125:5060> for address/port to
send to
set_destination: set destination to 85.115.115.125, port 5060
Reliably Transmitting (no NAT) to 85.115.115.125:5060:
BYE sip:201@85.115.115.125:5060 SIP/2.0
Via: SIP/2.0/UDP 217.199.111.19:5060;branch=z9hG4bK530d3ec0;rport
From: "75305101" <sip:75305101@217.199.111.19>;tag=as5550c7ca
To: <sip:201@85.115.115.125>;tag=5573737d-13c4-3a501443-9f36d0-7500
Contact: <sip:75305101@217.199.111.19>
Call-ID: 05398b7e73a15e06401d749c76755a70@217.199.111.19
CSeq: 105 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:1284 __sip_reliable_xmit: *** SIP
TIMER: Initalizing retransmit timer on packet: Id #20
Destroying call
'78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19'
<-- SIP read from 85.115.115.125:5060:
SIP/2.0 200 OK
From: "75305101"<sip:75305101@217.199.111.19>;tag=as5550c7ca
To: <sip:201@85.115.115.125>;tag=5573737d-13c4-3a501443-9f36d0-7500
Call-ID: 05398b7e73a15e06401d749c76755a70@217.199.111.19
CSeq: 105 BYE
Via: SIP/2.0/UDP 217.199.111.19:5060;rport=5060;branch=z9hG4bK530d3ec0
Content-Length: 0
Nov 16 02:55:43 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 0: SIP/2.0
200 OK (14)
Nov 16 02:55:43 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 1: From:
"75305101"<sip:75305101@217.199.111.19>;tag=as5550c7ca (60)
Nov 16 02:55:43 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 2: To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501443-9f36d0-7500 (67)
Nov 16 02:55:43 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3: Call-ID:
05398b7e73a15e06401d749c76755a70@217.199.111.19 (56)
Nov 16 02:55:43 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4: CSeq: 105
BYE (13)
Nov 16 02:55:43 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 5: Via:
SIP/2.0/UDP 217.199.111.19:5060;rport=5060;branch=z9hG4bK530d3ec0 (70)
Nov 16 02:55:43 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6:
Content-Length: 0 (17)
Nov 16 02:55:43 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 7: (0)
--- (7 headers 0 lines)---
Nov 16 02:55:43 DEBUG[22869]: chan_sip.c:3149 find_call: = Found Their Call ID:
05398b7e73a15e06401d749c76755a70@217.199.111.19 Their Tag
5573737d-13c4-3a501443-9f36d0-7500 Our tag: as5550c7ca
Nov 16 02:55:43 DEBUG[22869]: chan_sip.c:1378 __sip_ack: ** SIP TIMER:
Cancelling retransmit of packet (reply received) Retransid #20
Nov 16 02:55:43 DEBUG[22869]: chan_sip.c:1389 __sip_ack: Stopping retransmission
on '05398b7e73a15e06401d749c76755a70@217.199.111.19' of Request 105:
Match Found
Destroying call '05398b7e73a15e06401d749c76755a70@217.199.111.19'
----- Original Message -----
From: X-Files
To: asterisk-users@lists.digium.com
Sent: Wednesday, November 16, 2005 2:06 AM
Subject: [Asterisk-Users] not work DTMF
You do not wish to help me, I have " Internet Telephony Gateway "
from it I can to call on
asterisk and to type on local numbers from ports FXS and FXO (an example:
voicemail
the problem works perfectly on all inquiries), but that asterisk cannot
transfer dtmf in
ports FXS and FXO when I wish to type figures on phone all of them ignore.
In what a problem?
X-Files
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