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<DIV><FONT face=Arial size=2>I use ADAPTER "Eusso UTG7104"</FONT></DIV>
<DIV><FONT face=Arial size=2>VAD turn is off</FONT></DIV>
<DIV><FONT face=Arial size=2>dtmf_relay turn is on</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>this debug :</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>*CLI> <BR>*CLI> <BR>*CLI> <BR>*CLI>
<BR>*CLI> <BR>*CLI> <BR>*CLI> <BR>*CLI> <BR>*CLI> <BR>*CLI>
<BR>*CLI> <BR>*CLI> <BR>*CLI> <BR>*CLI> <BR><-- SIP read from
85.115.115.125:5060: <BR>INVITE sip:9@217.199.111.19 SIP/2.0<BR>From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501378-9c1e94-781<BR>To:
<sip:9@217.199.111.19><BR>Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19">78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19</A><BR>CSeq:
1 INVITE<BR>Via: SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501378-9c1e94-3026<BR>Max-Forwards:
70<BR>Contact: <sip:75305101@85.115.115.125:5060><BR>Content-Type:
application/SDP<BR>Content-Length: 130</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>v=0<BR>o=75305101 481519749 1606819611 IN IP4
85.115.115.125<BR>s=-<BR>c=IN IP4 85.115.115.125<BR>t=0 0<BR>m=audio 2076
RTP/AVP 18 4 0 101</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>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)<BR>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)<BR>Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 2:
To: <sip:9@217.199.111.19> (26)<BR>Nov 16 02:52:03 DEBUG[22720]:
chan_sip.c:3317 parse_request: Header 3: Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19">78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19</A>
(64)<BR>Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 4:
CSeq: 1 INVITE (14)<BR>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)<BR>Nov 16 02:52:03
DEBUG[22720]: chan_sip.c:3317 parse_request: Header 6: Max-Forwards: 70
(16)<BR>Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 7:
Contact: <sip:75305101@85.115.115.125:5060> (43)<BR>Nov 16 02:52:03
DEBUG[22720]: chan_sip.c:3317 parse_request: Header 8: Content-Type:
application/SDP (29)<BR>Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317
parse_request: Header 9: Content-Length: 130 (19)<BR>Nov 16 02:52:03
DEBUG[22720]: chan_sip.c:3317 parse_request: Header 10: (0)<BR>Nov 16
02:52:03 DEBUG[22720]: chan_sip.c:3349 parse_request: Line: v=0 (3)<BR>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)<BR>Nov 16 02:52:03 DEBUG[22720]:
chan_sip.c:3349 parse_request: Line: s=- (3)<BR>Nov 16 02:52:03 DEBUG[22720]:
chan_sip.c:3349 parse_request: Line: c=IN IP4 85.115.115.125 (23)<BR>Nov 16
02:52:03 DEBUG[22720]: chan_sip.c:3349 parse_request: Line: t=0 0 (5)<BR>Nov 16
02:52:03 DEBUG[22720]: chan_sip.c:3349 parse_request: Line: m=audio 2076 RTP/AVP
18 4 0 101 (31)<BR>--- (10 headers 6 lines)---<BR>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<BR>Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3101 sip_alloc:
Allocating new SIP dialog for <A
href="mailto:78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19">78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19</A>
- INVITE (With RTP)<BR>Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:10906
handle_request: **** Received INVITE (5) - Command in SIP INVITE<BR>Using INVITE
request as basis request - <A
href="mailto:78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19">78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19</A><BR>Sending
to 85.115.115.125 : 5060 (non-NAT)<BR>Nov 16 02:52:03 DEBUG[22720]:
chan_sip.c:7007 check_user_full: Setting NAT on RTP to 0<BR>Found user
'75305101'<BR>Found RTP audio format 18<BR>Found RTP audio format 4<BR>Found RTP
audio format 0<BR>Found RTP audio format 101<BR>Peer audio RTP is at port
85.115.115.125:2076<BR>Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3501
process_sdp: Peer audio RTP is at port 85.115.115.125:2076<BR>Capabilities: us -
0x100 (g729), peer - audio=0x105 (g723|ulaw|g729)/video=0x0 (nothing), combined
- 0x100 (g729)<BR>Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)<BR>Nov 16 02:52:03
DEBUG[22720]: chan_sip.c:10317 handle_request_invite: Checking SIP call limits
for device 75305101<BR>Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:2195
update_call_counter: Updating call counter for incoming call<BR>Looking for 9 in
home (domain 217.199.111.19)<BR>Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:6022
build_route: build_route: Contact hop:
<sip:75305101@85.115.115.125:5060><BR>list_route: hop:
<sip:75305101@85.115.115.125:5060><BR>Transmitting (no NAT) to
85.115.115.125:5060:<BR>SIP/2.0 100 Trying<BR>Via: SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501378-9c1e94-3026;received=85.115.115.125<BR>From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501378-9c1e94-781<BR>To:
<sip:9@217.199.111.19><BR>Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19">78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19</A><BR>CSeq:
1 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER, SUBSCRIBE, NOTIFY<BR>Max-Forwards: 70<BR>Contact:
<sip:9@217.199.111.19><BR>Content-Length: 0</FONT></DIV>
<DIV> </DIV><FONT face=Arial size=2>
<DIV><BR>---<BR>Nov 16 02:52:03 DEBUG[22717]: chan_sip.c:11428 sip_devicestate:
Checking device state for peer 75305101<BR>Nov 16 02:52:03 DEBUG[22717]:
devicestate.c:187 do_state_change: Changing state for SIP/75305101 - state 2 (In
use)<BR>Nov 16 02:52:03 DEBUG[22818]: pbx.c:1667 pbx_extension_helper: Launching
'Answer'<BR> -- Executing Answer("SIP/75305101-5b84", "") in
new stack<BR>Nov 16 02:52:03 DEBUG[22819]: app_queue.c:471 changethread: Device
'SIP/75305101' changed to state '2' (In use)<BR>Nov 16 02:52:03 DEBUG[22818]:
chan_sip.c:2505 sip_answer: sip_answer(SIP/75305101-5b84)<BR>We're at
217.199.111.19 port 15292<BR>Adding codec 0x100 (g729) to SDP<BR>Nov 16 02:52:03
DEBUG[22717]: chan_sip.c:11428 sip_devicestate: Checking device state for peer
75305101<BR>Adding non-codec 0x1 (telephone-event) to SDP<BR>Nov 16 02:52:03
DEBUG[22717]: channel.c:770 channel_find_locked: Avoiding initial deadlock for
'SIP/75305101-5b84'<BR>Reliably Transmitting (no NAT) to
85.115.115.125:5060:<BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501378-9c1e94-3026;received=85.115.115.125<BR>From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501378-9c1e94-781<BR>To:
<sip:9@217.199.111.19>;tag=as6aeea459<BR>Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19">78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19</A><BR>CSeq:
1 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER, SUBSCRIBE, NOTIFY<BR>Max-Forwards: 70<BR>Contact:
<sip:9@217.199.111.19><BR>Content-Type: application/sdp<BR>Content-Length:
222</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=root 22818 22818 IN IP4 217.199.111.19<BR>s=session<BR>c=IN IP4
217.199.111.19<BR>t=0 0<BR>m=audio 15292 RTP/AVP 18 101<BR>a=rtpmap:18
G729/8000<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101
0-16<BR>a=silenceSupp:off - - - -</DIV>
<DIV> </DIV>
<DIV>---<BR>Nov 16 02:52:03 DEBUG[22818]: chan_sip.c:1284 __sip_reliable_xmit:
*** SIP TIMER: Initalizing retransmit timer on packet: Id #54<BR>Nov 16
02:52:03 DEBUG[22717]: devicestate.c:187 do_state_change: Changing state for
SIP/75305101 - state 2 (In use)<BR>Nov 16 02:52:03 DEBUG[22820]: app_queue.c:471
changethread: Device 'SIP/75305101' changed to state '2' (In use)</DIV>
<DIV> </DIV>
<DIV><-- SIP read from 85.115.115.125:5060: <BR>ACK sip:9@217.199.111.19
SIP/2.0<BR>From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501378-9c1e94-781<BR>To:
<sip:9@217.199.111.19>;tag=as6aeea459<BR>Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19">78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19</A><BR>CSeq:
1 ACK<BR>Via: SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501378-9c1f1c-4467<BR>Max-Forwards:
70<BR>Contact: <sip:75305101@85.115.115.125:5060><BR>Content-Length:
0</DIV>
<DIV> </DIV>
<DIV><BR>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)<BR>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)<BR>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)<BR>Nov 16 02:52:03
DEBUG[22720]: chan_sip.c:3317 parse_request: Header 3: Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19">78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19</A>
(64)<BR>Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 4:
CSeq: 1 ACK (11)<BR>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)<BR>Nov 16 02:52:03
DEBUG[22720]: chan_sip.c:3317 parse_request: Header 6: Max-Forwards: 70
(16)<BR>Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header 7:
Contact: <sip:75305101@85.115.115.125:5060> (43)<BR>Nov 16 02:52:03
DEBUG[22720]: chan_sip.c:3317 parse_request: Header 8: Content-Length: 0
(17)<BR>Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:3317 parse_request: Header
9: (0)<BR>--- (9 headers 0 lines)---<BR>Nov 16 02:52:03 DEBUG[22720]:
chan_sip.c:3149 find_call: = Found Their Call ID: <A
href="mailto:78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19">78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19</A>
Their Tag 5573737d-13c4-3a501378-9c1e94-781 Our tag: as6aeea459<BR>Nov 16
02:52:03 DEBUG[22720]: chan_sip.c:10906 handle_request: **** Received ACK (6) -
Command in SIP ACK<BR>Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:1378 __sip_ack:
** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid
#54<BR>Nov 16 02:52:03 DEBUG[22720]: chan_sip.c:1389 __sip_ack: Stopping
retransmission on <A
href="mailto:'78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19'">'78923c-5573737d-13c4-3a501378-9c1e92-a8b@217.199.111.19'</A>
of Response 1: Match Found<BR>Nov 16 02:52:07 DEBUG[22720]: chan_sip.c:1311
__sip_autodestruct: Auto destroying call
'5573737d-13c4-3a4feb87-1918-48a1'<BR>Destroying call
'5573737d-13c4-3a4feb87-1918-48a1'</DIV>
<DIV>*CLI> <BR>*CLI> </DIV>
<DIV>*CLI> <BR>*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</DIV>
<DIV> </DIV>
<DIV><-- SIP read from 85.115.115.125:5060: <BR>INFO sip:9@217.199.111.19
SIP/2.0<BR>From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a50141c-9e9f56-29b8<BR>To:
<sip:9@217.199.111.19>;tag=as2af78cfd<BR>Call-ID: <A
href="mailto:7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19">7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19</A><BR>CSeq:
2 INFO<BR>Via: SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501421-9eb4ae-2964<BR>Max-Forwards:
70<BR>Contact: <sip:75305101@85.115.115.125:5060><BR>Content-Type:
application/dtmf-relay<BR>Content-Length: 24</DIV>
<DIV> </DIV>
<DIV>Signal=2<BR>Duration=250</DIV>
<DIV> </DIV>
<DIV>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)<BR>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)<BR>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)<BR>Nov 16 02:54:52
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3: Call-ID: <A
href="mailto:7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19">7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19</A>
(64)<BR>Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4:
CSeq: 2 INFO (12)<BR>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)<BR>Nov 16 02:54:52
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6: Max-Forwards: 70
(16)<BR>Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 7:
Contact: <sip:75305101@85.115.115.125:5060> (43)<BR>Nov 16 02:54:52
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8: Content-Type:
application/dtmf-relay (36)<BR>Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317
parse_request: Header 9: Content-Length: 24 (18)<BR>Nov 16 02:54:52
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 10: (0)<BR>Nov 16
02:54:52 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: Signal=2 (8)<BR>Nov
16 02:54:52 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: Duration=250
(12)<BR>--- (10 headers 2 lines)---<BR>Nov 16 02:54:52 DEBUG[22869]:
chan_sip.c:3149 find_call: = Found Their Call ID: <A
href="mailto:7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19">7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19</A>
Their Tag 5573737d-13c4-3a50141c-9e9f56-29b8 Our tag: as2af78cfd<BR>Nov 16
02:54:52 DEBUG[22869]: chan_sip.c:10906 handle_request: **** Received INFO (13)
- Command in SIP INFO<BR>Receiving INFO!<BR>* DTMF-relay event received:
'2'<BR>Transmitting (no NAT) to 85.115.115.125:5060:<BR>SIP/2.0 200 OK<BR>Via:
SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501421-9eb4ae-2964;received=85.115.115.125<BR>From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a50141c-9e9f56-29b8<BR>To:
<sip:9@217.199.111.19>;tag=as2af78cfd<BR>Call-ID: <A
href="mailto:7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19">7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19</A><BR>CSeq:
2 INFO<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY<BR>Max-Forwards: 70<BR>Contact:
<sip:9@217.199.111.19><BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR>---<BR> == CDR updated on SIP/75305101-0ee9<BR>Nov 16 02:54:52
DEBUG[22874]: pbx.c:1667 pbx_extension_helper: Launching
'Dial'<BR> -- Executing Dial("SIP/75305101-0ee9", "<A
href="mailto:SIP/201@outbound">SIP/201@outbound</A>") in new stack<BR>Nov 16
02:54:52 DEBUG[22874]: chan_sip.c:3101 sip_alloc: Allocating new SIP dialog for
(No Call-ID) - INVITE (With RTP)<BR>Nov 16 02:54:52 DEBUG[22874]:
chan_sip.c:1860 create_addr_from_peer: Setting NAT on RTP to 0<BR>Nov 16
02:54:52 DEBUG[22874]: channel.c:2793 ast_channel_inherit_variables: Not copying
variable STACK-home-2-1.<BR>Nov 16 02:54:52 DEBUG[22874]: channel.c:2793
ast_channel_inherit_variables: Not copying variable STACK-home-9-1.<BR>Nov 16
02:54:52 DEBUG[22874]: channel.c:2793 ast_channel_inherit_variables: Not copying
variable SIPCALLID.<BR>Nov 16 02:54:52 DEBUG[22874]: channel.c:2793
ast_channel_inherit_variables: Not copying variable SIPDOMAIN.<BR>Nov 16
02:54:52 DEBUG[22874]: channel.c:2793 ast_channel_inherit_variables: Not copying
variable SIPURI.<BR>Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:2054 sip_call:
Outgoing Call for 201<BR>Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:2195
update_call_counter: Updating call counter for outgoing call<BR>We're at
217.199.111.19 port 14292<BR>Adding codec 0x100 (g729) to SDP<BR>Adding
non-codec 0x1 (telephone-event) to SDP<BR>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)<BR>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)<BR>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)<BR>Nov 16 02:54:52
DEBUG[22874]: chan_sip.c:3317 parse_request: Header 3: To:
<sip:201@85.115.115.125> (28)<BR>Nov 16 02:54:52 DEBUG[22874]:
chan_sip.c:3317 parse_request: Header 4: Contact:
<sip:75305101@217.199.111.19> (38)<BR>Nov 16 02:54:52 DEBUG[22874]:
chan_sip.c:3317 parse_request: Header 5: Call-ID: <A
href="mailto:5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19">5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19</A>
(56)<BR>Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 6:
CSeq: 102 INVITE (16)<BR>Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317
parse_request: Header 7: User-Agent: Asterisk PBX (24)<BR>Nov 16 02:54:52
DEBUG[22874]: chan_sip.c:3317 parse_request: Header 8: Max-Forwards: 70
(16)<BR>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)<BR>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)<BR>Nov 16 02:54:52 DEBUG[22874]:
chan_sip.c:3317 parse_request: Header 11: Content-Type: application/sdp
(29)<BR>Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 12:
Content-Length: 222 (19)<BR>Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3317
parse_request: Header 13: (0)<BR>Nov 16 02:54:52 DEBUG[22874]:
chan_sip.c:3349 parse_request: Line: v=0 (3)<BR>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)<BR>Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3349 parse_request: Line:
s=session (9)<BR>Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3349 parse_request:
Line: c=IN IP4 217.199.111.19 (23)<BR>Nov 16 02:54:52 DEBUG[22874]:
chan_sip.c:3349 parse_request: Line: t=0 0 (5)<BR>Nov 16 02:54:52 DEBUG[22874]:
chan_sip.c:3349 parse_request: Line: m=audio 14292 RTP/AVP 18 101 (28)<BR>Nov 16
02:54:52 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: a=rtpmap:18
G729/8000 (21)<BR>Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:3349 parse_request:
Line: a=rtpmap:101 telephone-event/8000 (33)<BR>Nov 16 02:54:52 DEBUG[22874]:
chan_sip.c:3349 parse_request: Line: a=fmtp:101 0-16 (15)<BR>Nov 16 02:54:52
DEBUG[22874]: chan_sip.c:3349 parse_request: Line: a=silenceSupp:off - - - -
(25)<BR>13 headers, 10 lines<BR>Reliably Transmitting (no NAT) to
85.115.115.125:5060:<BR>INVITE sip:201@85.115.115.125 SIP/2.0<BR>Via:
SIP/2.0/UDP 217.199.111.19:5060;branch=z9hG4bK2776613f;rport<BR>From: "75305101"
<sip:75305101@217.199.111.19>;tag=as7fe1eb18<BR>To:
<sip:201@85.115.115.125><BR>Contact:
<sip:75305101@217.199.111.19><BR>Call-ID: <A
href="mailto:5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19">5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19</A><BR>CSeq:
102 INVITE<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Date: Wed, 16 Nov
2005 00:54:52 GMT<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY<BR>Content-Type: application/sdp<BR>Content-Length: 222</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=root 22874 22874 IN IP4 217.199.111.19<BR>s=session<BR>c=IN IP4
217.199.111.19<BR>t=0 0<BR>m=audio 14292 RTP/AVP 18 101<BR>a=rtpmap:18
G729/8000<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101
0-16<BR>a=silenceSupp:off - - - -</DIV>
<DIV> </DIV>
<DIV>---<BR>Nov 16 02:54:52 DEBUG[22874]: chan_sip.c:1284 __sip_reliable_xmit:
*** SIP TIMER: Initalizing retransmit timer on packet: Id
#10<BR> -- Called <A
href="mailto:201@outbound">201@outbound</A><BR>Nov 16 02:54:52 DEBUG[22874]:
channel.c:2331 set_format: Set channel SIP/outbound-0ea4 to read format
g729<BR>Nov 16 02:54:52 DEBUG[22874]: channel.c:2331 set_format: Set channel
SIP/75305101-0ee9 to write format g729<BR>Nov 16 02:54:52 DEBUG[22874]:
channel.c:2331 set_format: Set channel SIP/75305101-0ee9 to read format
g729<BR>Nov 16 02:54:52 DEBUG[22874]: channel.c:2331 set_format: Set channel
SIP/outbound-0ea4 to write format g729</DIV>
<DIV> </DIV>
<DIV><-- SIP read from 85.115.115.125:5060: <BR>SIP/2.0 180 Ringing<BR>From:
"75305101"<sip:75305101@217.199.111.19>;tag=as7fe1eb18<BR>To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501422-9eb520-5b93<BR>Call-ID:
<A
href="mailto:5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19">5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19</A><BR>CSeq:
102 INVITE<BR>Via: SIP/2.0/UDP
217.199.111.19:5060;rport=5060;branch=z9hG4bK2776613f<BR>Contact:
<sip:201@85.115.115.125:5060><BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR>Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 0:
SIP/2.0 180 Ringing (19)<BR>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)<BR>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)<BR>Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3:
Call-ID: <A
href="mailto:5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19">5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19</A>
(56)<BR>Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4:
CSeq: 102 INVITE (16)<BR>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)<BR>Nov 16 02:54:52
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6: Contact:
<sip:201@85.115.115.125:5060> (38)<BR>Nov 16 02:54:52 DEBUG[22869]:
chan_sip.c:3317 parse_request: Header 7: Content-Length: 0 (17)<BR>Nov 16
02:54:52 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8: (0)<BR>---
(8 headers 0 lines)---<BR>Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:3149
find_call: = Found Their Call ID: <A
href="mailto:5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19">5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19</A>
Their Tag Our tag: as7fe1eb18<BR>Nov 16 02:54:52 DEBUG[22869]:
chan_sip.c:1433 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #10 -
INVITE (got response)<BR>Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:1442
__sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on
<A
href="mailto:'5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19'">'5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19'</A>
Request 102: Found<BR>Nov 16 02:54:52 DEBUG[22869]: chan_sip.c:9423
handle_response_invite: SIP response 180 to standard invite<BR>Nov 16 02:54:52
DEBUG[22866]: chan_sip.c:11428 sip_devicestate: Checking device state for peer
outbound<BR>Nov 16 02:54:52 DEBUG[22866]: devicestate.c:187 do_state_change:
Changing state for SIP/outbound - state 6 (Ringing)<BR>Nov 16 02:54:52
DEBUG[22877]: app_queue.c:471 changethread: Device 'SIP/outbound' changed to
state '6' (Ringing)<BR> -- SIP/outbound-0ea4 is ringing<BR>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<BR>Nov 16
02:54:52 DEBUG[22874]: channel.c:2331 set_format: Set channel SIP/75305101-0ee9
to write format slin<BR>Nov 16 02:54:52 DEBUG[22874]: rtp.c:1341 ast_rtp_write:
Ooh, format changed from unknown to g729</DIV>
<DIV> </DIV>
<DIV><-- SIP read from 85.115.115.125:5060: <BR>SIP/2.0 200 OK<BR>From:
"75305101"<sip:75305101@217.199.111.19>;tag=as7fe1eb18<BR>To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501422-9eb520-5b93<BR>Call-ID:
<A
href="mailto:5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19">5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19</A><BR>CSeq:
102 INVITE<BR>Via: SIP/2.0/UDP
217.199.111.19:5060;rport=5060;branch=z9hG4bK2776613f<BR>Contact:
<sip:201@85.115.115.125:5060><BR>Content-Type:
application/SDP<BR>Content-Length: 116</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=201 812119400 215593963 IN IP4 85.115.115.125<BR>s=-<BR>c=IN IP4
85.115.115.125<BR>t=0 0<BR>m=audio 2074 RTP/AVP 18</DIV>
<DIV> </DIV>
<DIV>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 0:
SIP/2.0 200 OK (14)<BR>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)<BR>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)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3:
Call-ID: <A
href="mailto:5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19">5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19</A>
(56)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4:
CSeq: 102 INVITE (16)<BR>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)<BR>Nov 16 02:54:53
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6: Contact:
<sip:201@85.115.115.125:5060> (38)<BR>Nov 16 02:54:53 DEBUG[22869]:
chan_sip.c:3317 parse_request: Header 7: Content-Type: application/SDP
(29)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8:
Content-Length: 116 (19)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317
parse_request: Header 9: (0)<BR>Nov 16 02:54:53 DEBUG[22869]:
chan_sip.c:3349 parse_request: Line: v=0 (3)<BR>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)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349
parse_request: Line: s=- (3)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349
parse_request: Line: c=IN IP4 85.115.115.125 (23)<BR>Nov 16 02:54:53
DEBUG[22869]: chan_sip.c:3349 parse_request: Line: t=0 0 (5)<BR>Nov 16 02:54:53
DEBUG[22869]: chan_sip.c:3349 parse_request: Line: m=audio 2074 RTP/AVP 18
(23)<BR>--- (9 headers 6 lines)---<BR>Nov 16 02:54:53 DEBUG[22869]:
chan_sip.c:3149 find_call: = Found Their Call ID: <A
href="mailto:5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19">5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19</A>
Their Tag 5573737d-13c4-3a501422-9eb520-5b93 Our tag: as7fe1eb18<BR>Nov 16
02:54:53 DEBUG[22869]: chan_sip.c:1367 __sip_ack: Acked pending invite
102<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:1389 __sip_ack: Stopping
retransmission on <A
href="mailto:'5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19'">'5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19'</A>
of Request 102: Match Found<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:9423
handle_response_invite: SIP response 200 to standard invite<BR>Found RTP audio
format 18<BR>Peer audio RTP is at port 85.115.115.125:2074<BR>Nov 16 02:54:53
DEBUG[22869]: chan_sip.c:3501 process_sdp: Peer audio RTP is at port
85.115.115.125:2074<BR>Capabilities: us - 0x100 (g729), peer - audio=0x100
(g729)/video=0x0 (nothing), combined - 0x100 (g729)<BR>Non-codec capabilities:
us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0
(nothing)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:6022 build_route:
build_route: Contact hop: <sip:201@85.115.115.125:5060><BR>list_route:
hop: <sip:201@85.115.115.125:5060><BR>set_destination: Parsing
<sip:201@85.115.115.125:5060> for address/port to send
to<BR>set_destination: set destination to 85.115.115.125, port
5060<BR>Transmitting (no NAT) to 85.115.115.125:5060:<BR>ACK
sip:201@85.115.115.125:5060 SIP/2.0<BR>Via: SIP/2.0/UDP
217.199.111.19:5060;branch=z9hG4bK7d1af83f;rport<BR>From: "75305101"
<sip:75305101@217.199.111.19>;tag=as7fe1eb18<BR>To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501422-9eb520-5b93<BR>Contact:
<sip:75305101@217.199.111.19><BR>Call-ID: <A
href="mailto:5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19">5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19</A><BR>CSeq:
102 ACK<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Content-Length:
0</DIV>
<DIV> </DIV>
<DIV><BR>---<BR>Nov 16 02:54:53 DEBUG[22874]: rtp.c:1341 ast_rtp_write: Ooh,
format changed from unknown to g729<BR> -- SIP/outbound-0ea4
answered SIP/75305101-0ee9<BR>Nov 16 02:54:53 DEBUG[22874]: channel.c:2331
set_format: Set channel SIP/75305101-0ee9 to write format g729<BR>Nov 16
02:54:53 DEBUG[22874]: channel.c:2331 set_format: Set channel SIP/75305101-0ee9
to read format g729<BR>Nov 16 02:54:53 DEBUG[22874]: channel.c:2331 set_format:
Set channel SIP/outbound-0ea4 to write format g729<BR>Nov 16 02:54:53
DEBUG[22874]: channel.c:2331 set_format: Set channel SIP/outbound-0ea4 to read
format g729<BR>Nov 16 02:54:53 DEBUG[22874]: channel.c:2331 set_format: Set
channel SIP/75305101-0ee9 to write format g729<BR> --
Attempting native bridge of SIP/75305101-0ee9 and SIP/outbound-0ea4<BR>Nov 16
02:54:53 DEBUG[22874]: chan_sip.c:12662 sip_set_rtp_peer: Sending reinvite on
SIP <A
href="mailto:'7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19'">'7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19'</A>
- It's audio soon redirected to IP 85.115.115.125<BR>set_destination: Parsing
<sip:75305101@85.115.115.125:5060> for address/port to send
to<BR>set_destination: set destination to 85.115.115.125, port 5060<BR>We're at
217.199.111.19 port 14712<BR>Adding codec 0x100 (g729) to SDP<BR>Adding
non-codec 0x1 (telephone-event) to SDP<BR>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)<BR>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)<BR>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)<BR>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)<BR>Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 4:
Contact: <sip:9@217.199.111.19> (31)<BR>Nov 16 02:54:53 DEBUG[22874]:
chan_sip.c:3317 parse_request: Header 5: Call-ID: <A
href="mailto:7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19">7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19</A>
(64)<BR>Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 6:
CSeq: 102 INVITE (16)<BR>Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317
parse_request: Header 7: User-Agent: Asterisk PBX (24)<BR>Nov 16 02:54:53
DEBUG[22874]: chan_sip.c:3317 parse_request: Header 8: Max-Forwards: 70
(16)<BR>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)<BR>Nov
16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 10:
X-asterisk-info: SIP re-invite (RTP bridge) (43)<BR>Nov 16 02:54:53
DEBUG[22874]: chan_sip.c:3317 parse_request: Header 11: Content-Type:
application/sdp (29)<BR>Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317
parse_request: Header 12: Content-Length: 221 (19)<BR>Nov 16 02:54:53
DEBUG[22874]: chan_sip.c:3317 parse_request: Header 13: (0)<BR>Nov 16
02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: v=0 (3)<BR>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)<BR>Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349
parse_request: Line: s=session (9)<BR>Nov 16 02:54:53 DEBUG[22874]:
chan_sip.c:3349 parse_request: Line: c=IN IP4 85.115.115.125 (23)<BR>Nov 16
02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: t=0 0 (5)<BR>Nov 16
02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: m=audio 2074 RTP/AVP
18 101 (27)<BR>Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request:
Line: a=rtpmap:18 G729/8000 (21)<BR>Nov 16 02:54:53 DEBUG[22874]:
chan_sip.c:3349 parse_request: Line: a=rtpmap:101 telephone-event/8000
(33)<BR>Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line:
a=fmtp:101 0-16 (15)<BR>Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349
parse_request: Line: a=silenceSupp:off - - - - (25)<BR>13 headers, 10
lines<BR>Reliably Transmitting (no NAT) to 85.115.115.125:5060:<BR>INVITE
sip:75305101@85.115.115.125:5060 SIP/2.0<BR>Via: SIP/2.0/UDP
217.199.111.19:5060;branch=z9hG4bK38bafc54;rport<BR>From:
<sip:9@217.199.111.19>;tag=as2af78cfd<BR>To:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a50141c-9e9f56-29b8<BR>Contact:
<sip:9@217.199.111.19><BR>Call-ID: <A
href="mailto:7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19">7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19</A><BR>CSeq:
102 INVITE<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Allow: INVITE,
ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<BR>X-asterisk-info: SIP
re-invite (RTP bridge)<BR>Content-Type: application/sdp<BR>Content-Length:
221</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=root 22874 22875 IN IP4 85.115.115.125<BR>s=session<BR>c=IN IP4
85.115.115.125<BR>t=0 0<BR>m=audio 2074 RTP/AVP 18 101<BR>a=rtpmap:18
G729/8000<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101
0-16<BR>a=silenceSupp:off - - - -</DIV>
<DIV> </DIV>
<DIV>---<BR>Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:1284 __sip_reliable_xmit:
*** SIP TIMER: Initalizing retransmit timer on packet: Id #11<BR>Nov 16
02:54:53 DEBUG[22874]: chan_sip.c:12662 sip_set_rtp_peer: Sending reinvite on
SIP <A
href="mailto:'5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19'">'5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19'</A>
- It's audio soon redirected to IP 85.115.115.125<BR>set_destination: Parsing
<sip:201@85.115.115.125:5060> for address/port to send
to<BR>set_destination: set destination to 85.115.115.125, port 5060<BR>We're at
217.199.111.19 port 14292<BR>Adding codec 0x100 (g729) to SDP<BR>Adding codec
0x1 (g723) to SDP<BR>Adding codec 0x4 (ulaw) to SDP<BR>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)<BR>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)<BR>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)<BR>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)<BR>Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 4:
Contact: <sip:75305101@217.199.111.19> (38)<BR>Nov 16 02:54:53
DEBUG[22874]: chan_sip.c:3317 parse_request: Header 5: Call-ID: <A
href="mailto:5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19">5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19</A>
(56)<BR>Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 6:
CSeq: 103 INVITE (16)<BR>Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317
parse_request: Header 7: User-Agent: Asterisk PBX (24)<BR>Nov 16 02:54:53
DEBUG[22874]: chan_sip.c:3317 parse_request: Header 8: Max-Forwards: 70
(16)<BR>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)<BR>Nov
16 02:54:53 DEBUG[22874]: chan_sip.c:3317 parse_request: Header 10:
X-asterisk-info: SIP re-invite (RTP bridge) (43)<BR>Nov 16 02:54:53
DEBUG[22874]: chan_sip.c:3317 parse_request: Header 11: Content-Type:
application/sdp (29)<BR>Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3317
parse_request: Header 12: Content-Length: 213 (19)<BR>Nov 16 02:54:53
DEBUG[22874]: chan_sip.c:3317 parse_request: Header 13: (0)<BR>Nov 16
02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: v=0 (3)<BR>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)<BR>Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349
parse_request: Line: s=session (9)<BR>Nov 16 02:54:53 DEBUG[22874]:
chan_sip.c:3349 parse_request: Line: c=IN IP4 85.115.115.125 (23)<BR>Nov 16
02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: t=0 0 (5)<BR>Nov 16
02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: m=audio 2070 RTP/AVP
18 4 0 (27)<BR>Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request:
Line: a=rtpmap:18 G729/8000 (21)<BR>Nov 16 02:54:53 DEBUG[22874]:
chan_sip.c:3349 parse_request: Line: a=rtpmap:4 G723/8000 (20)<BR>Nov 16
02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line: a=rtpmap:0 PCMU/8000
(20)<BR>Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:3349 parse_request: Line:
a=silenceSupp:off - - - - (25)<BR>13 headers, 10 lines<BR>Reliably Transmitting
(no NAT) to 85.115.115.125:5060:<BR>INVITE sip:201@85.115.115.125:5060
SIP/2.0<BR>Via: SIP/2.0/UDP
217.199.111.19:5060;branch=z9hG4bK33f70e49;rport<BR>From: "75305101"
<sip:75305101@217.199.111.19>;tag=as7fe1eb18<BR>To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501422-9eb520-5b93<BR>Contact:
<sip:75305101@217.199.111.19><BR>Call-ID: <A
href="mailto:5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19">5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19</A><BR>CSeq:
103 INVITE<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Allow: INVITE,
ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<BR>X-asterisk-info: SIP
re-invite (RTP bridge)<BR>Content-Type: application/sdp<BR>Content-Length:
213</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=root 22874 22875 IN IP4 85.115.115.125<BR>s=session<BR>c=IN IP4
85.115.115.125<BR>t=0 0<BR>m=audio 2070 RTP/AVP 18 4 0<BR>a=rtpmap:18
G729/8000<BR>a=rtpmap:4 G723/8000<BR>a=rtpmap:0 PCMU/8000<BR>a=silenceSupp:off -
- - -</DIV>
<DIV> </DIV>
<DIV>---<BR>Nov 16 02:54:53 DEBUG[22874]: chan_sip.c:1284 __sip_reliable_xmit:
*** SIP TIMER: Initalizing retransmit timer on packet: Id #12<BR>Nov 16
02:54:53 DEBUG[22866]: chan_sip.c:11428 sip_devicestate: Checking device state
for peer outbound<BR>Nov 16 02:54:53 DEBUG[22866]: devicestate.c:187
do_state_change: Changing state for SIP/outbound - state 2 (In use)<BR>Nov 16
02:54:53 DEBUG[22878]: app_queue.c:471 changethread: Device 'SIP/outbound'
changed to state '2' (In use)</DIV>
<DIV> </DIV>
<DIV><-- SIP read from 85.115.115.125:5060: <BR>SIP/2.0 200 OK<BR>From:
<sip:9@217.199.111.19>;tag=as2af78cfd<BR>To:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a50141c-9e9f56-29b8<BR>Call-ID:
<A
href="mailto:7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19">7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19</A><BR>CSeq:
102 INVITE<BR>Via: SIP/2.0/UDP
217.199.111.19:5060;rport=5060;branch=z9hG4bK38bafc54<BR>Contact:
<sip:75305101@85.115.115.125:5060><BR>Content-Type:
application/SDP<BR>Content-Length: 121</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=75305101 177701337 159505494 IN IP4 85.115.115.125<BR>s=-<BR>c=IN
IP4 85.115.115.125<BR>t=0 0<BR>m=audio 2070 RTP/AVP 18</DIV>
<DIV> </DIV>
<DIV>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 0:
SIP/2.0 200 OK (14)<BR>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)<BR>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)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3:
Call-ID: <A
href="mailto:7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19">7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19</A>
(64)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4:
CSeq: 102 INVITE (16)<BR>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)<BR>Nov 16 02:54:53
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6: Contact:
<sip:75305101@85.115.115.125:5060> (43)<BR>Nov 16 02:54:53 DEBUG[22869]:
chan_sip.c:3317 parse_request: Header 7: Content-Type: application/SDP
(29)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8:
Content-Length: 121 (19)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317
parse_request: Header 9: (0)<BR>Nov 16 02:54:53 DEBUG[22869]:
chan_sip.c:3349 parse_request: Line: v=0 (3)<BR>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)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349
parse_request: Line: s=- (3)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349
parse_request: Line: c=IN IP4 85.115.115.125 (23)<BR>Nov 16 02:54:53
DEBUG[22869]: chan_sip.c:3349 parse_request: Line: t=0 0 (5)<BR>Nov 16 02:54:53
DEBUG[22869]: chan_sip.c:3349 parse_request: Line: m=audio 2070 RTP/AVP 18
(23)<BR>--- (9 headers 6 lines)---<BR>Nov 16 02:54:53 DEBUG[22869]:
chan_sip.c:3149 find_call: = No match Their Call ID: <A
href="mailto:5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19">5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19</A>
Their Tag 5573737d-13c4-3a501422-9eb520-5b93 Our tag: as7fe1eb18<BR>Nov 16
02:54:53 DEBUG[22869]: chan_sip.c:3149 find_call: = Found Their Call ID: <A
href="mailto:7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19">7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19</A>
Their Tag 5573737d-13c4-3a50141c-9e9f56-29b8 Our tag: as2af78cfd<BR>Nov 16
02:54:53 DEBUG[22869]: chan_sip.c:1367 __sip_ack: Acked pending invite
102<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:1378 __sip_ack: ** SIP TIMER:
Cancelling retransmit of packet (reply received) Retransid #11<BR>Nov 16
02:54:53 DEBUG[22869]: chan_sip.c:1389 __sip_ack: Stopping retransmission on <A
href="mailto:'7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19'">'7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19'</A>
of Request 102: Match Found<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:9421
handle_response_invite: SIP response 200 to RE-invite on outgoing call <A
href="mailto:7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19">7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19</A><BR>Found
RTP audio format 18<BR>Peer audio RTP is at port 85.115.115.125:2070<BR>Nov 16
02:54:53 DEBUG[22869]: chan_sip.c:3501 process_sdp: Peer audio RTP is at port
85.115.115.125:2070<BR>Capabilities: us - 0x100 (g729), peer - audio=0x100
(g729)/video=0x0 (nothing), combined - 0x100 (g729)<BR>Non-codec capabilities:
us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0
(nothing)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:6022 build_route:
build_route: Contact hop:
<sip:75305101@85.115.115.125:5060><BR>list_route: hop:
<sip:75305101@85.115.115.125:5060><BR>set_destination: Parsing
<sip:75305101@85.115.115.125:5060> for address/port to send
to<BR>set_destination: set destination to 85.115.115.125, port
5060<BR>Transmitting (no NAT) to 85.115.115.125:5060:<BR>ACK
sip:75305101@85.115.115.125:5060 SIP/2.0<BR>Via: SIP/2.0/UDP
217.199.111.19:5060;branch=z9hG4bK4cfadd2d;rport<BR>From:
<sip:9@217.199.111.19>;tag=as2af78cfd<BR>To:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a50141c-9e9f56-29b8<BR>Contact:
<sip:9@217.199.111.19><BR>Call-ID: <A
href="mailto:7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19">7894d4-5573737d-13c4-3a50141c-9e9f56-e2a@217.199.111.19</A><BR>CSeq:
102 ACK<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Content-Length:
0</DIV>
<DIV> </DIV>
<DIV><BR>---</DIV>
<DIV> </DIV>
<DIV><-- SIP read from 85.115.115.125:5060: <BR>SIP/2.0 200 OK<BR>From:
"75305101"<sip:75305101@217.199.111.19>;tag=as7fe1eb18<BR>To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501422-9eb520-5b93<BR>Call-ID:
<A
href="mailto:5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19">5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19</A><BR>CSeq:
103 INVITE<BR>Via: SIP/2.0/UDP
217.199.111.19:5060;rport=5060;branch=z9hG4bK33f70e49<BR>Contact:
<sip:201@85.115.115.125:5060><BR>Content-Type:
application/SDP<BR>Content-Length: 116</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=201 83317681 1732424444 IN IP4 85.115.115.125<BR>s=-<BR>c=IN IP4
85.115.115.125<BR>t=0 0<BR>m=audio 2074 RTP/AVP 18</DIV>
<DIV> </DIV>
<DIV>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 0:
SIP/2.0 200 OK (14)<BR>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)<BR>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)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3:
Call-ID: <A
href="mailto:5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19">5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19</A>
(56)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4:
CSeq: 103 INVITE (16)<BR>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)<BR>Nov 16 02:54:53
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6: Contact:
<sip:201@85.115.115.125:5060> (38)<BR>Nov 16 02:54:53 DEBUG[22869]:
chan_sip.c:3317 parse_request: Header 7: Content-Type: application/SDP
(29)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8:
Content-Length: 116 (19)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3317
parse_request: Header 9: (0)<BR>Nov 16 02:54:53 DEBUG[22869]:
chan_sip.c:3349 parse_request: Line: v=0 (3)<BR>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)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349
parse_request: Line: s=- (3)<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:3349
parse_request: Line: c=IN IP4 85.115.115.125 (23)<BR>Nov 16 02:54:53
DEBUG[22869]: chan_sip.c:3349 parse_request: Line: t=0 0 (5)<BR>Nov 16 02:54:53
DEBUG[22869]: chan_sip.c:3349 parse_request: Line: m=audio 2074 RTP/AVP 18
(23)<BR>--- (9 headers 6 lines)---<BR>Nov 16 02:54:53 DEBUG[22869]:
chan_sip.c:3149 find_call: = Found Their Call ID: <A
href="mailto:5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19">5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19</A>
Their Tag 5573737d-13c4-3a501422-9eb520-5b93 Our tag: as7fe1eb18<BR>Nov 16
02:54:53 DEBUG[22869]: chan_sip.c:1367 __sip_ack: Acked pending invite
103<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:1378 __sip_ack: ** SIP TIMER:
Cancelling retransmit of packet (reply received) Retransid #12<BR>Nov 16
02:54:53 DEBUG[22869]: chan_sip.c:1389 __sip_ack: Stopping retransmission on <A
href="mailto:'5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19'">'5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19'</A>
of Request 103: Match Found<BR>Nov 16 02:54:53 DEBUG[22869]: chan_sip.c:9421
handle_response_invite: SIP response 200 to RE-invite on outgoing call <A
href="mailto:5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19">5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19</A><BR>Found
RTP audio format 18<BR>Peer audio RTP is at port 85.115.115.125:2074<BR>Nov 16
02:54:53 DEBUG[22869]: chan_sip.c:3501 process_sdp: Peer audio RTP is at port
85.115.115.125:2074<BR>Capabilities: us - 0x100 (g729), peer - audio=0x100
(g729)/video=0x0 (nothing), combined - 0x100 (g729)<BR>Non-codec capabilities:
us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0
(nothing)<BR>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><BR>set_destination: Parsing
<sip:201@85.115.115.125:5060> for address/port to send
to<BR>set_destination: set destination to 85.115.115.125, port
5060<BR>Transmitting (no NAT) to 85.115.115.125:5060:<BR>ACK
sip:201@85.115.115.125:5060 SIP/2.0<BR>Via: SIP/2.0/UDP
217.199.111.19:5060;branch=z9hG4bK038520a8;rport<BR>From: "75305101"
<sip:75305101@217.199.111.19>;tag=as7fe1eb18<BR>To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501422-9eb520-5b93<BR>Contact:
<sip:75305101@217.199.111.19><BR>Call-ID: <A
href="mailto:5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19">5b86e0df1bb1f6b3713bcef20ebebf43@217.199.111.19</A><BR>CSeq:
103 ACK<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Content-Length:
0</DIV>
<DIV> </DIV>
<DIV><BR>---</DIV>
<DIV> </DIV>
<DIV>*CLI> <BR>*CLI> <BR>*CLI> <BR>*CLI> </DIV>
<DIV>*CLI> <BR><-- SIP read from 85.115.115.125:5060: <BR>INFO
sip:9@217.199.111.19 SIP/2.0<BR>From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301<BR>To:
<sip:9@217.199.111.19>;tag=as53452dbd<BR>Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19">78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19</A><BR>CSeq:
3 INFO<BR>Via: SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501447-9f46e8-612c<BR>Max-Forwards:
70<BR>Contact: <sip:75305101@85.115.115.125:5060><BR>Content-Type:
application/dtmf-relay<BR>Content-Length: 24</DIV>
<DIV> </DIV>
<DIV>Signal=6<BR>Duration=250</DIV>
<DIV> </DIV>
<DIV>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)<BR>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)<BR>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)<BR>Nov 16 02:55:30
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3: Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19">78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19</A>
(65)<BR>Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4:
CSeq: 3 INFO (12)<BR>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)<BR>Nov 16 02:55:30
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6: Max-Forwards: 70
(16)<BR>Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 7:
Contact: <sip:75305101@85.115.115.125:5060> (43)<BR>Nov 16 02:55:30
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8: Content-Type:
application/dtmf-relay (36)<BR>Nov 16 02:55:30 DEBUG[22869]: chan_sip.c:3317
parse_request: Header 9: Content-Length: 24 (18)<BR>Nov 16 02:55:30
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 10: (0)<BR>Nov 16
02:55:30 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: Signal=6 (8)<BR>Nov
16 02:55:30 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: Duration=250
(12)<BR>--- (10 headers 2 lines)---<BR>Nov 16 02:55:30 DEBUG[22869]:
chan_sip.c:3149 find_call: = No match Their Call ID: <A
href="mailto:05398b7e73a15e06401d749c76755a70@217.199.111.19">05398b7e73a15e06401d749c76755a70@217.199.111.19</A>
Their Tag 5573737d-13c4-3a501443-9f36d0-7500 Our tag: as5550c7ca<BR>Nov 16
02:55:30 DEBUG[22869]: chan_sip.c:3149 find_call: = Found Their Call ID: <A
href="mailto:78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19">78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19</A>
Their Tag 5573737d-13c4-3a501440-9f2e0a-7301 Our tag: as53452dbd<BR>Nov 16
02:55:30 DEBUG[22869]: chan_sip.c:10906 handle_request: **** Received INFO (13)
- Command in SIP INFO<BR>Receiving INFO!<BR>* DTMF-relay event received:
'6'<BR>Transmitting (no NAT) to 85.115.115.125:5060:<BR>SIP/2.0 200 OK<BR>Via:
SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501447-9f46e8-612c;received=85.115.115.125<BR>From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301<BR>To:
<sip:9@217.199.111.19>;tag=as53452dbd<BR>Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19">78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19</A><BR>CSeq:
3 INFO<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY<BR>Max-Forwards: 70<BR>Contact:
<sip:9@217.199.111.19><BR>Content-Length: 0<BR>X-Asterisk-HangupCause:
Normal Clearing</DIV>
<DIV> </DIV>
<DIV><BR>---</DIV>
<DIV> </DIV>
<DIV><-- SIP read from 85.115.115.125:5060: <BR>INFO sip:9@217.199.111.19
SIP/2.0<BR>From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301<BR>To:
<sip:9@217.199.111.19>;tag=as53452dbd<BR>Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19">78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19</A><BR>CSeq:
4 INFO<BR>Via: SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501448-9f4c64-698a<BR>Max-Forwards:
70<BR>Contact: <sip:75305101@85.115.115.125:5060><BR>Content-Type:
application/dtmf-relay<BR>Content-Length: 24</DIV>
<DIV> </DIV>
<DIV>Signal=5<BR>Duration=250</DIV>
<DIV> </DIV>
<DIV>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)<BR>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)<BR>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)<BR>Nov 16 02:55:31
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3: Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19">78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19</A>
(65)<BR>Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4:
CSeq: 4 INFO (12)<BR>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)<BR>Nov 16 02:55:31
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6: Max-Forwards: 70
(16)<BR>Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 7:
Contact: <sip:75305101@85.115.115.125:5060> (43)<BR>Nov 16 02:55:31
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8: Content-Type:
application/dtmf-relay (36)<BR>Nov 16 02:55:31 DEBUG[22869]: chan_sip.c:3317
parse_request: Header 9: Content-Length: 24 (18)<BR>Nov 16 02:55:31
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 10: (0)<BR>Nov 16
02:55:31 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: Signal=5 (8)<BR>Nov
16 02:55:31 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: Duration=250
(12)<BR>--- (10 headers 2 lines)---<BR>Nov 16 02:55:31 DEBUG[22869]:
chan_sip.c:3149 find_call: = No match Their Call ID: <A
href="mailto:05398b7e73a15e06401d749c76755a70@217.199.111.19">05398b7e73a15e06401d749c76755a70@217.199.111.19</A>
Their Tag 5573737d-13c4-3a501443-9f36d0-7500 Our tag: as5550c7ca<BR>Nov 16
02:55:31 DEBUG[22869]: chan_sip.c:3149 find_call: = Found Their Call ID: <A
href="mailto:78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19">78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19</A>
Their Tag 5573737d-13c4-3a501440-9f2e0a-7301 Our tag: as53452dbd<BR>Nov 16
02:55:31 DEBUG[22869]: chan_sip.c:10906 handle_request: **** Received INFO (13)
- Command in SIP INFO<BR>Receiving INFO!<BR>* DTMF-relay event received:
'5'<BR>Transmitting (no NAT) to 85.115.115.125:5060:<BR>SIP/2.0 200 OK<BR>Via:
SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501448-9f4c64-698a;received=85.115.115.125<BR>From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301<BR>To:
<sip:9@217.199.111.19>;tag=as53452dbd<BR>Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19">78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19</A><BR>CSeq:
4 INFO<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY<BR>Max-Forwards: 70<BR>Contact:
<sip:9@217.199.111.19><BR>Content-Length: 0<BR>X-Asterisk-HangupCause:
Normal Clearing</DIV>
<DIV> </DIV>
<DIV><BR>---</DIV>
<DIV> </DIV>
<DIV><-- SIP read from 85.115.115.125:5060: <BR>INFO sip:9@217.199.111.19
SIP/2.0<BR>From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301<BR>To:
<sip:9@217.199.111.19>;tag=as53452dbd<BR>Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19">78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19</A><BR>CSeq:
5 INFO<BR>Via: SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a50144a-9f52e2-6418<BR>Max-Forwards:
70<BR>Contact: <sip:75305101@85.115.115.125:5060><BR>Content-Type:
application/dtmf-relay<BR>Content-Length: 24</DIV>
<DIV> </DIV>
<DIV>Signal=4<BR>Duration=250</DIV>
<DIV> </DIV>
<DIV>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)<BR>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)<BR>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)<BR>Nov 16 02:55:33
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3: Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19">78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19</A>
(65)<BR>Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4:
CSeq: 5 INFO (12)<BR>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)<BR>Nov 16 02:55:33
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6: Max-Forwards: 70
(16)<BR>Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 7:
Contact: <sip:75305101@85.115.115.125:5060> (43)<BR>Nov 16 02:55:33
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8: Content-Type:
application/dtmf-relay (36)<BR>Nov 16 02:55:33 DEBUG[22869]: chan_sip.c:3317
parse_request: Header 9: Content-Length: 24 (18)<BR>Nov 16 02:55:33
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 10: (0)<BR>Nov 16
02:55:33 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: Signal=4 (8)<BR>Nov
16 02:55:33 DEBUG[22869]: chan_sip.c:3349 parse_request: Line: Duration=250
(12)<BR>--- (10 headers 2 lines)---<BR>Nov 16 02:55:33 DEBUG[22869]:
chan_sip.c:3149 find_call: = No match Their Call ID: <A
href="mailto:05398b7e73a15e06401d749c76755a70@217.199.111.19">05398b7e73a15e06401d749c76755a70@217.199.111.19</A>
Their Tag 5573737d-13c4-3a501443-9f36d0-7500 Our tag: as5550c7ca<BR>Nov 16
02:55:33 DEBUG[22869]: chan_sip.c:3149 find_call: = Found Their Call ID: <A
href="mailto:78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19">78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19</A>
Their Tag 5573737d-13c4-3a501440-9f2e0a-7301 Our tag: as53452dbd<BR>Nov 16
02:55:33 DEBUG[22869]: chan_sip.c:10906 handle_request: **** Received INFO (13)
- Command in SIP INFO<BR>Receiving INFO!<BR>* DTMF-relay event received:
'4'<BR>Transmitting (no NAT) to 85.115.115.125:5060:<BR>SIP/2.0 200 OK<BR>Via:
SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a50144a-9f52e2-6418;received=85.115.115.125<BR>From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301<BR>To:
<sip:9@217.199.111.19>;tag=as53452dbd<BR>Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19">78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19</A><BR>CSeq:
5 INFO<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY<BR>Max-Forwards: 70<BR>Contact:
<sip:9@217.199.111.19><BR>Content-Length: 0<BR>X-Asterisk-HangupCause:
Normal Clearing</DIV>
<DIV> </DIV>
<DIV><BR>---</DIV>
<DIV> </DIV>
<DIV>*CLI> <BR>*CLI> <BR>*CLI> <BR>*CLI> <BR>*CLI> <BR>*CLI>
<BR>*CLI> <BR>*CLI> <BR>*CLI> <BR>*CLI> <BR>*CLI> <BR>*CLI>
<BR>*CLI> <BR>*CLI> <BR>*CLI> <BR>*CLI> <BR><-- SIP read from
85.115.115.125:5060: <BR>BYE sip:9@217.199.111.19 SIP/2.0<BR>From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301<BR>To:
<sip:9@217.199.111.19>;tag=as53452dbd<BR>Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19">78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19</A><BR>CSeq:
6 BYE<BR>Via: SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501454-9f7888-1456<BR>Max-Forwards:
70<BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR>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)<BR>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)<BR>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)<BR>Nov 16 02:55:42
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3: Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19">78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19</A>
(65)<BR>Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4:
CSeq: 6 BYE (11)<BR>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)<BR>Nov 16 02:55:42
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6: Max-Forwards: 70
(16)<BR>Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 7:
Content-Length: 0 (17)<BR>Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317
parse_request: Header 8: (0)<BR>--- (8 headers 0 lines)---<BR>Nov 16
02:55:42 DEBUG[22869]: chan_sip.c:3149 find_call: = No match Their Call ID: <A
href="mailto:05398b7e73a15e06401d749c76755a70@217.199.111.19">05398b7e73a15e06401d749c76755a70@217.199.111.19</A>
Their Tag 5573737d-13c4-3a501443-9f36d0-7500 Our tag: as5550c7ca<BR>Nov 16
02:55:42 DEBUG[22869]: chan_sip.c:3149 find_call: = Found Their Call ID: <A
href="mailto:78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19">78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19</A>
Their Tag 5573737d-13c4-3a501440-9f2e0a-7301 Our tag: as53452dbd<BR>Nov 16
02:55:42 DEBUG[22869]: chan_sip.c:10906 handle_request: **** Received BYE (8) -
Command in SIP BYE<BR>Sending to 85.115.115.125 : 5060 (non-NAT)<BR>Transmitting
(no NAT) to 85.115.115.125:5060:<BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP
85.115.115.125:5060;branch=z9hG4bK-3a501454-9f7888-1456;received=85.115.115.125<BR>From:
<sip:75305101@217.199.111.19>;tag=5573737d-13c4-3a501440-9f2e0a-7301<BR>To:
<sip:9@217.199.111.19>;tag=as53452dbd<BR>Call-ID: <A
href="mailto:78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19">78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19</A><BR>CSeq:
6 BYE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY<BR>Max-Forwards: 70<BR>Contact:
<sip:9@217.199.111.19><BR>Content-Length: 0<BR>X-Asterisk-HangupCause:
Normal Clearing</DIV>
<DIV> </DIV>
<DIV><BR>---<BR>Nov 16 02:55:42 DEBUG[22889]: rtp.c:1700 ast_rtp_bridge: Oooh,
got a hangup<BR>Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:12662 sip_set_rtp_peer:
Sending reinvite on SIP <A
href="mailto:'05398b7e73a15e06401d749c76755a70@217.199.111.19'">'05398b7e73a15e06401d749c76755a70@217.199.111.19'</A>
- It's audio soon redirected to IP 217.199.111.19<BR>set_destination: Parsing
<sip:201@85.115.115.125:5060> for address/port to send
to<BR>set_destination: set destination to 85.115.115.125, port 5060<BR>We're at
217.199.111.19 port 12976<BR>Adding codec 0x100 (g729) to SDP<BR>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)<BR>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)<BR>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)<BR>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)<BR>Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317 parse_request: Header 4:
Contact: <sip:75305101@217.199.111.19> (38)<BR>Nov 16 02:55:42
DEBUG[22889]: chan_sip.c:3317 parse_request: Header 5: Call-ID: <A
href="mailto:05398b7e73a15e06401d749c76755a70@217.199.111.19">05398b7e73a15e06401d749c76755a70@217.199.111.19</A>
(56)<BR>Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317 parse_request: Header 6:
CSeq: 104 INVITE (16)<BR>Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317
parse_request: Header 7: User-Agent: Asterisk PBX (24)<BR>Nov 16 02:55:42
DEBUG[22889]: chan_sip.c:3317 parse_request: Header 8: Max-Forwards: 70
(16)<BR>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)<BR>Nov
16 02:55:42 DEBUG[22889]: chan_sip.c:3317 parse_request: Header 10:
X-asterisk-info: SIP re-invite (RTP bridge) (43)<BR>Nov 16 02:55:42
DEBUG[22889]: chan_sip.c:3317 parse_request: Header 11: Content-Type:
application/sdp (29)<BR>Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3317
parse_request: Header 12: Content-Length: 166 (19)<BR>Nov 16 02:55:42
DEBUG[22889]: chan_sip.c:3317 parse_request: Header 13: (0)<BR>Nov 16
02:55:42 DEBUG[22889]: chan_sip.c:3349 parse_request: Line: v=0 (3)<BR>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)<BR>Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3349
parse_request: Line: s=session (9)<BR>Nov 16 02:55:42 DEBUG[22889]:
chan_sip.c:3349 parse_request: Line: c=IN IP4 217.199.111.19 (23)<BR>Nov 16
02:55:42 DEBUG[22889]: chan_sip.c:3349 parse_request: Line: t=0 0 (5)<BR>Nov 16
02:55:42 DEBUG[22889]: chan_sip.c:3349 parse_request: Line: m=audio 12976
RTP/AVP 18 (24)<BR>Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:3349 parse_request:
Line: a=rtpmap:18 G729/8000 (21)<BR>Nov 16 02:55:42 DEBUG[22889]:
chan_sip.c:3349 parse_request: Line: a=silenceSupp:off - - - - (25)<BR>13
headers, 8 lines<BR>Reliably Transmitting (no NAT) to
85.115.115.125:5060:<BR>INVITE sip:201@85.115.115.125:5060 SIP/2.0<BR>Via:
SIP/2.0/UDP 217.199.111.19:5060;branch=z9hG4bK58497b12;rport<BR>From: "75305101"
<sip:75305101@217.199.111.19>;tag=as5550c7ca<BR>To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501443-9f36d0-7500<BR>Contact:
<sip:75305101@217.199.111.19><BR>Call-ID: <A
href="mailto:05398b7e73a15e06401d749c76755a70@217.199.111.19">05398b7e73a15e06401d749c76755a70@217.199.111.19</A><BR>CSeq:
104 INVITE<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Allow: INVITE,
ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<BR>X-asterisk-info: SIP
re-invite (RTP bridge)<BR>Content-Type: application/sdp<BR>Content-Length:
166</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=root 22889 22891 IN IP4 217.199.111.19<BR>s=session<BR>c=IN IP4
217.199.111.19<BR>t=0 0<BR>m=audio 12976 RTP/AVP 18<BR>a=rtpmap:18
G729/8000<BR>a=silenceSupp:off - - - -</DIV>
<DIV> </DIV>
<DIV>---<BR>Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:1284 __sip_reliable_xmit:
*** SIP TIMER: Initalizing retransmit timer on packet: Id #19<BR>Nov 16
02:55:42 DEBUG[22889]: channel.c:3449 ast_channel_bridge: Returning from native
bridge, channels: SIP/75305101-85cf, SIP/outbound-1d4a<BR>Nov 16 02:55:42
DEBUG[22889]: channel.c:1308 ast_hangup: Hanging up channel
'SIP/outbound-1d4a'<BR>Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:2401 sip_hangup:
Hangup call SIP/outbound-1d4a, SIP callid <A
href="mailto:05398b7e73a15e06401d749c76755a70@217.199.111.19">05398b7e73a15e06401d749c76755a70@217.199.111.19</A>)<BR>Nov
16 02:55:42 DEBUG[22889]: chan_sip.c:2409 sip_hangup: update_call_counter(201) -
decrement call limit counter<BR>Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:2195
update_call_counter: Updating call counter for outgoing call<BR>Nov 16 02:55:42
DEBUG[22889]: app_dial.c:1587 dial_exec_full: Exiting with
DIALSTATUS=ANSWER.<BR>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'<BR>Nov 16
02:55:42 DEBUG[22889]: channel.c:1308 ast_hangup: Hanging up channel
'SIP/75305101-85cf'<BR>Nov 16 02:55:42 DEBUG[22889]: chan_sip.c:2401 sip_hangup:
Hangup call SIP/75305101-85cf, SIP callid <A
href="mailto:78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19">78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19</A>)<BR>Nov
16 02:55:42 DEBUG[22889]: chan_sip.c:2409 sip_hangup:
update_call_counter(75305101) - decrement call limit counter<BR>Nov 16 02:55:42
DEBUG[22889]: chan_sip.c:2195 update_call_counter: Updating call counter for
outgoing call<BR>Nov 16 02:55:42 DEBUG[22866]: chan_sip.c:11428 sip_devicestate:
Checking device state for peer outbound<BR>Nov 16 02:55:42 DEBUG[22866]:
devicestate.c:187 do_state_change: Changing state for SIP/outbound - state 1
(Not in use)<BR>Nov 16 02:55:42 DEBUG[22894]: app_queue.c:471 changethread:
Device 'SIP/outbound' changed to state '1' (Not in use)<BR>Nov 16 02:55:42
DEBUG[22866]: chan_sip.c:11428 sip_devicestate: Checking device state for peer
75305101<BR>Nov 16 02:55:42 DEBUG[22866]: devicestate.c:187 do_state_change:
Changing state for SIP/75305101 - state 1 (Not in use)<BR>Nov 16 02:55:42
DEBUG[22895]: app_queue.c:471 changethread: Device 'SIP/75305101' changed to
state '1' (Not in use)</DIV>
<DIV> </DIV>
<DIV><-- SIP read from 85.115.115.125:5060: <BR>SIP/2.0 200 OK<BR>From:
"75305101"<sip:75305101@217.199.111.19>;tag=as5550c7ca<BR>To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501443-9f36d0-7500<BR>Call-ID:
<A
href="mailto:05398b7e73a15e06401d749c76755a70@217.199.111.19">05398b7e73a15e06401d749c76755a70@217.199.111.19</A><BR>CSeq:
104 INVITE<BR>Via: SIP/2.0/UDP
217.199.111.19:5060;rport=5060;branch=z9hG4bK58497b12<BR>Contact:
<sip:201@85.115.115.125:5060><BR>Content-Type:
application/SDP<BR>Content-Length: 118</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=201 2262112151 2598113056 IN IP4 85.115.115.125<BR>s=-<BR>c=IN IP4
85.115.115.125<BR>t=0 0<BR>m=audio 2078 RTP/AVP 18</DIV>
<DIV> </DIV>
<DIV>Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 0:
SIP/2.0 200 OK (14)<BR>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)<BR>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)<BR>Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3:
Call-ID: <A
href="mailto:05398b7e73a15e06401d749c76755a70@217.199.111.19">05398b7e73a15e06401d749c76755a70@217.199.111.19</A>
(56)<BR>Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4:
CSeq: 104 INVITE (16)<BR>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)<BR>Nov 16 02:55:42
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6: Contact:
<sip:201@85.115.115.125:5060> (38)<BR>Nov 16 02:55:42 DEBUG[22869]:
chan_sip.c:3317 parse_request: Header 7: Content-Type: application/SDP
(29)<BR>Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 8:
Content-Length: 118 (19)<BR>Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3317
parse_request: Header 9: (0)<BR>Nov 16 02:55:42 DEBUG[22869]:
chan_sip.c:3349 parse_request: Line: v=0 (3)<BR>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)<BR>Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3349
parse_request: Line: s=- (3)<BR>Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:3349
parse_request: Line: c=IN IP4 85.115.115.125 (23)<BR>Nov 16 02:55:42
DEBUG[22869]: chan_sip.c:3349 parse_request: Line: t=0 0 (5)<BR>Nov 16 02:55:42
DEBUG[22869]: chan_sip.c:3349 parse_request: Line: m=audio 2078 RTP/AVP 18
(23)<BR>--- (9 headers 6 lines)---<BR>Nov 16 02:55:42 DEBUG[22869]:
chan_sip.c:3149 find_call: = Found Their Call ID: <A
href="mailto:05398b7e73a15e06401d749c76755a70@217.199.111.19">05398b7e73a15e06401d749c76755a70@217.199.111.19</A>
Their Tag 5573737d-13c4-3a501443-9f36d0-7500 Our tag: as5550c7ca<BR>Nov 16
02:55:42 DEBUG[22869]: chan_sip.c:1367 __sip_ack: Acked pending invite
104<BR>Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:1378 __sip_ack: ** SIP TIMER:
Cancelling retransmit of packet (reply received) Retransid #19<BR>Nov 16
02:55:42 DEBUG[22869]: chan_sip.c:1389 __sip_ack: Stopping retransmission on <A
href="mailto:'05398b7e73a15e06401d749c76755a70@217.199.111.19'">'05398b7e73a15e06401d749c76755a70@217.199.111.19'</A>
of Request 104: Match Found<BR>Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:9423
handle_response_invite: SIP response 200 to standard invite<BR>Found RTP audio
format 18<BR>Peer audio RTP is at port 85.115.115.125:2078<BR>Nov 16 02:55:42
DEBUG[22869]: chan_sip.c:3501 process_sdp: Peer audio RTP is at port
85.115.115.125:2078<BR>Capabilities: us - 0x100 (g729), peer - audio=0x100
(g729)/video=0x0 (nothing), combined - 0x100 (g729)<BR>Non-codec capabilities:
us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0
(nothing)<BR>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><BR>set_destination: Parsing
<sip:201@85.115.115.125:5060> for address/port to send
to<BR>set_destination: set destination to 85.115.115.125, port
5060<BR>Transmitting (no NAT) to 85.115.115.125:5060:<BR>ACK
sip:201@85.115.115.125:5060 SIP/2.0<BR>Via: SIP/2.0/UDP
217.199.111.19:5060;branch=z9hG4bK688850f1;rport<BR>From: "75305101"
<sip:75305101@217.199.111.19>;tag=as5550c7ca<BR>To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501443-9f36d0-7500<BR>Contact:
<sip:75305101@217.199.111.19><BR>Call-ID: <A
href="mailto:05398b7e73a15e06401d749c76755a70@217.199.111.19">05398b7e73a15e06401d749c76755a70@217.199.111.19</A><BR>CSeq:
104 ACK<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Content-Length:
0</DIV>
<DIV> </DIV>
<DIV><BR>---<BR>set_destination: Parsing <sip:201@85.115.115.125:5060> for
address/port to send to<BR>set_destination: set destination to 85.115.115.125,
port 5060<BR>Reliably Transmitting (no NAT) to 85.115.115.125:5060:<BR>BYE
sip:201@85.115.115.125:5060 SIP/2.0<BR>Via: SIP/2.0/UDP
217.199.111.19:5060;branch=z9hG4bK530d3ec0;rport<BR>From: "75305101"
<sip:75305101@217.199.111.19>;tag=as5550c7ca<BR>To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501443-9f36d0-7500<BR>Contact:
<sip:75305101@217.199.111.19><BR>Call-ID: <A
href="mailto:05398b7e73a15e06401d749c76755a70@217.199.111.19">05398b7e73a15e06401d749c76755a70@217.199.111.19</A><BR>CSeq:
105 BYE<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Content-Length:
0</DIV>
<DIV> </DIV>
<DIV><BR>---<BR>Nov 16 02:55:42 DEBUG[22869]: chan_sip.c:1284
__sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet:
Id #20<BR>Destroying call <A
href="mailto:'78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19'">'78923c-5573737d-13c4-3a501440-9f2e08-2a0b@217.199.111.19'</A></DIV>
<DIV> </DIV>
<DIV><-- SIP read from 85.115.115.125:5060: <BR>SIP/2.0 200 OK<BR>From:
"75305101"<sip:75305101@217.199.111.19>;tag=as5550c7ca<BR>To:
<sip:201@85.115.115.125>;tag=5573737d-13c4-3a501443-9f36d0-7500<BR>Call-ID:
<A
href="mailto:05398b7e73a15e06401d749c76755a70@217.199.111.19">05398b7e73a15e06401d749c76755a70@217.199.111.19</A><BR>CSeq:
105 BYE<BR>Via: SIP/2.0/UDP
217.199.111.19:5060;rport=5060;branch=z9hG4bK530d3ec0<BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR>Nov 16 02:55:43 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 0:
SIP/2.0 200 OK (14)<BR>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)<BR>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)<BR>Nov 16 02:55:43 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 3:
Call-ID: <A
href="mailto:05398b7e73a15e06401d749c76755a70@217.199.111.19">05398b7e73a15e06401d749c76755a70@217.199.111.19</A>
(56)<BR>Nov 16 02:55:43 DEBUG[22869]: chan_sip.c:3317 parse_request: Header 4:
CSeq: 105 BYE (13)<BR>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)<BR>Nov 16 02:55:43
DEBUG[22869]: chan_sip.c:3317 parse_request: Header 6: Content-Length: 0
(17)<BR>Nov 16 02:55:43 DEBUG[22869]: chan_sip.c:3317 parse_request: Header
7: (0)<BR>--- (7 headers 0 lines)---<BR>Nov 16 02:55:43 DEBUG[22869]:
chan_sip.c:3149 find_call: = Found Their Call ID: <A
href="mailto:05398b7e73a15e06401d749c76755a70@217.199.111.19">05398b7e73a15e06401d749c76755a70@217.199.111.19</A>
Their Tag 5573737d-13c4-3a501443-9f36d0-7500 Our tag: as5550c7ca<BR>Nov 16
02:55:43 DEBUG[22869]: chan_sip.c:1378 __sip_ack: ** SIP TIMER: Cancelling
retransmit of packet (reply received) Retransid #20<BR>Nov 16 02:55:43
DEBUG[22869]: chan_sip.c:1389 __sip_ack: Stopping retransmission on <A
href="mailto:'05398b7e73a15e06401d749c76755a70@217.199.111.19'">'05398b7e73a15e06401d749c76755a70@217.199.111.19'</A>
of Request 105: Match Found<BR>Destroying call <A
href="mailto:'05398b7e73a15e06401d749c76755a70@217.199.111.19'">'05398b7e73a15e06401d749c76755a70@217.199.111.19'</A></DIV>
<DIV> </DIV>
<DIV></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=x-files@x-files.lv href="mailto:x-files@x-files.lv">X-Files</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Wednesday, November 16, 2005 2:06
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] not work
DTMF</DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial size=2>You do not wish to help me, I have " Internet
Telephony Gateway " from it I can to call on </FONT></DIV>
<DIV><FONT face=Arial size=2>asterisk and to type on local numbers from ports
FXS and FXO (an example: voicemail </FONT></DIV>
<DIV><FONT face=Arial size=2>the problem works perfectly on all inquiries),
but that asterisk cannot transfer dtmf in </FONT></DIV>
<DIV><FONT face=Arial size=2>ports FXS and FXO when I wish to type figures on
phone all of them ignore. </FONT></DIV>
<DIV><FONT face=Arial size=2>In what a problem?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>X-Files</FONT></DIV>
<P>
<HR>
<P></P>_______________________________________________<BR>--Bandwidth and
Colocation sponsored by Easynews.com --<BR><BR>Asterisk-Users mailing
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