[asterisk-users] Nortel CS1000 Asterisk with SIP
Koen Van Impe
koenvi at gmail.com
Tue Nov 21 07:12:06 MST 2006
Skipped content of type multipart/alternative-------------- next part --------------
Nov 21 14:17:47 VERBOSE[32580] logger.c:
<-- SIP read from 172.25.103.222:5060:
INVITE sip:1715;phone-context=exp_net.ascom at ascom.be:5060;maddr=172.25.96.48;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 1 INVITE
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b31-5f87fa39-63c2
Max-Forwards: 70
Supported: 100rel,sipvc,replaces,timer
User-Agent: Nortel CS1000 SIP GW release_4.5 version_sse-4.50.88
P-Asserted-Identity: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>
Privacy: none
History-Info: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;transport=udp;user=phone>;index=1
x-nt-corr-id: 000000b70e1423150b at 0001af0807d6-f087c208
x-nt-calling-id: <sip:123452001649 at ascom.be>
Contact: <sip:1649;phone-context=Exp_Net.ascom at ascom.be:5060;maddr=172.25.103.222;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Type: multipart/mixed ;boundary=unique-boundary-1
Content-Length: 770
--unique-boundary-1
Content-Type: application/SDP
v=0
o=- 4162 1 IN IP4 172.25.103.222
s=-
t=0 0
m=audio 5234 RTP/AVP 18 8 0
c=IN IP4 172.25.103.229
a=fmtp:18 annexb=no
a=ptime:30
a=sendrecv
--unique-boundary-1
Content-Type: application/x-nt-mcdn-frag-hex ;version=sse-4.50.88 ;base=x2611
Content-Disposition: signal ;handling=optional
0500b201
0107130081900000a2
09090f00e9a08300010032
131e070011fd1800a1160201010201a1300e8102010582010184020000850104
1315070011fa0f00a10d02010102020100cc04aa028503
1e0403008183
460e01000a0001000100010000000000
--unique-boundary-1
Content-Type: application/x-nt-epid-frag-hex ;version=sse-4.50.88 ;base=x2611
Content-Disposition: signal ;handling=optional
011201
00:02:b3:f6:5a:ec
--unique-boundary-1--
Nov 21 14:17:47 VERBOSE[32580] logger.c: --- (18 headers 31 lines) ---
Nov 21 14:17:47 DEBUG[32580] chan_sip.c: Allocating new SIP dialog for f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be - INVITE (With RTP)
Nov 21 14:17:47 DEBUG[32580] chan_sip.c: * SIP extension value: 7 for call f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
Nov 21 14:17:47 VERBOSE[32580] logger.c: Using INVITE request as basis request - f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
Nov 21 14:17:47 VERBOSE[32580] logger.c: Sending to 172.25.103.222 : 5060 (non-NAT)
Nov 21 14:17:47 VERBOSE[32580] logger.c: Found peer 'CS1000'
Nov 21 14:17:47 DEBUG[32580] chan_sip.c: Setting NAT on RTP to 0
Nov 21 14:17:47 VERBOSE[32580] logger.c: Found RTP audio format 18
Nov 21 14:17:47 VERBOSE[32580] logger.c: Found RTP audio format 8
Nov 21 14:17:47 VERBOSE[32580] logger.c: Found RTP audio format 0
Nov 21 14:17:47 VERBOSE[32580] logger.c: Peer audio RTP is at port 172.25.103.229:5234
Nov 21 14:17:47 DEBUG[32580] chan_sip.c: Peer audio RTP is at port 172.25.103.229:5234
Nov 21 14:17:47 VERBOSE[32580] logger.c: Capabilities: us - 0x100 (g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x100 (g729)
Nov 21 14:17:47 VERBOSE[32580] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Nov 21 14:17:47 DEBUG[32580] chan_sip.c: Checking SIP call limits for device
Nov 21 14:17:47 VERBOSE[32580] logger.c: Looking for 1715 in pra-incoming (domain ascom.be)
Nov 21 14:17:47 DEBUG[32580] chan_sip.c: build_route: Contact hop: <sip:1649;phone-context=Exp_Net.ascom at ascom.be:5060;maddr=172.25.103.222;transport=udp;user=phone>
Nov 21 14:17:47 VERBOSE[32580] logger.c: list_route: hop: <sip:1649;phone-context=Exp_Net.ascom at ascom.be:5060;maddr=172.25.103.222;transport=udp;user=phone>
Nov 21 14:17:47 VERBOSE[32580] logger.c: Transmitting (no NAT) to 172.25.103.222:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b31-5f87fa39-63c2;received=172.25.103.222
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1715 at 172.25.96.48>
Content-Length: 0
---
Nov 21 14:17:47 DEBUG[991] pbx.c: Launching 'Macro'
Nov 21 14:17:47 VERBOSE[991] logger.c: -- Executing Macro("SIP/1649-08f029f0", "eva-on-sip") in new stack
Nov 21 14:17:47 DEBUG[991] pbx.c: Launching 'Dial'
Nov 21 14:17:47 VERBOSE[991] logger.c: -- Executing Dial("SIP/1649-08f029f0", "SIP/evavox/1715") in new stack
Nov 21 14:17:47 DEBUG[991] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP)
Nov 21 14:17:47 DEBUG[991] chan_sip.c: Setting NAT on RTP to 0
Nov 21 14:17:47 DEBUG[991] channel.c: Not copying variable STACK-macro-eva-on-sip-s-1.
Nov 21 14:17:47 DEBUG[991] channel.c: Not copying variable MACRO_DEPTH.
Nov 21 14:17:47 DEBUG[991] channel.c: Not copying variable MACRO_PRIORITY.
Nov 21 14:17:47 DEBUG[991] channel.c: Not copying variable MACRO_CONTEXT.
Nov 21 14:17:47 DEBUG[991] channel.c: Not copying variable MACRO_EXTEN.
Nov 21 14:17:47 DEBUG[991] channel.c: Not copying variable STACK-pra-incoming-1715-1.
Nov 21 14:17:47 DEBUG[991] channel.c: Not copying variable SIPCALLID.
Nov 21 14:17:47 DEBUG[991] channel.c: Not copying variable SIPUSERAGENT.
Nov 21 14:17:47 DEBUG[991] channel.c: Not copying variable SIPDOMAIN.
Nov 21 14:17:47 DEBUG[991] channel.c: Not copying variable SIPURI.
Nov 21 14:17:47 DEBUG[991] chan_sip.c: Outgoing Call for 1715
Nov 21 14:17:47 VERBOSE[991] logger.c: We're at 172.25.96.48 port 11986
Nov 21 14:17:47 VERBOSE[991] logger.c: Adding codec 0x100 (g729) to SDP
Nov 21 14:17:47 VERBOSE[991] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Nov 21 14:17:47 VERBOSE[991] logger.c: 13 headers, 11 lines
Nov 21 14:17:47 VERBOSE[991] logger.c: Reliably Transmitting (no NAT) to 172.25.96.47:5060:
INVITE sip:1715 at 172.25.96.47 SIP/2.0
Via: SIP/2.0/UDP 172.25.96.48:5060;branch=z9hG4bK4df8041e;rport
From: "1649;phonecontext=Exp_Netascom" <sip:1649;phonecontext=Exp_Netascom at 172.25.96.48>;tag=as0a5a7b3c
To: <sip:1715 at 172.25.96.47>
Contact: <sip:1649;phonecontext=Exp_Netascom at 172.25.96.48>
Call-ID: 0ff188833a12041d6203999156019426 at 172.25.96.48
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 21 Nov 2006 13:17:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 239
v=0
o=root 32564 32564 IN IP4 172.25.96.48
s=session
c=IN IP4 172.25.96.48
t=0 0
m=audio 11986 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Nov 21 14:17:47 VERBOSE[991] logger.c: -- Called evavox/1715
Nov 21 14:17:47 VERBOSE[32580] logger.c:
<-- SIP read from 172.25.96.47:5060:
SIP/2.0 180 Ringing
From: "1649;phonecontext=Exp_Netascom"<sip:1649;phonecontext=Exp_Netascom at 172.25.96.48>;tag=as0a5a7b3c
To: <sip:1715 at 172.25.96.47>;tag=46d1550-0-13c4-120a2-5b839b6-120a2
Call-ID: 0ff188833a12041d6203999156019426 at 172.25.96.48
CSeq: 102 INVITE
Via: SIP/2.0/UDP 172.25.96.48:5060;rport=5060;branch=z9hG4bK4df8041e
Supported: replaces
Contact: <sip:1715 at 172.25.96.47>
Content-Type: application/SDP
Content-Length: 259
v=0
o=Intel_IPCCLib 74284544 74284545 IN IP4 172.25.96.47
s=Intel_SIP_CCLIB
c=IN IP4 172.25.96.47
t=0 0
m=audio 49152 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=ptime:30
a=fmtp:101 0-15
a=sendrecv
Nov 21 14:17:47 VERBOSE[32580] logger.c: --- (10 headers 12 lines) ---
Nov 21 14:17:47 DEBUG[32580] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0ff188833a12041d6203999156019426 at 172.25.96.48' Request 102: Found
Nov 21 14:17:47 VERBOSE[32580] logger.c: Found RTP audio format 18
Nov 21 14:17:47 VERBOSE[32580] logger.c: Found RTP audio format 101
Nov 21 14:17:47 VERBOSE[32580] logger.c: Peer audio RTP is at port 172.25.96.47:49152
Nov 21 14:17:47 DEBUG[32580] chan_sip.c: Peer audio RTP is at port 172.25.96.47:49152
Nov 21 14:17:47 VERBOSE[32580] logger.c: Found description format G729
Nov 21 14:17:47 VERBOSE[32580] logger.c: Found description format telephone-event
Nov 21 14:17:47 VERBOSE[32580] logger.c: Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Nov 21 14:17:47 VERBOSE[32580] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Nov 21 14:17:47 VERBOSE[991] logger.c: -- SIP/evavox-08f09cb0 is ringing
Nov 21 14:17:47 VERBOSE[991] logger.c: Transmitting (no NAT) to 172.25.103.222:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b31-5f87fa39-63c2;received=172.25.103.222
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=as3c2b6b62
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1715 at 172.25.96.48>
Content-Length: 0
---
Nov 21 14:17:47 VERBOSE[991] logger.c: -- SIP/evavox-08f09cb0 is making progress passing it to SIP/1649-08f029f0
Nov 21 14:17:47 VERBOSE[991] logger.c: We're at 172.25.96.48 port 12058
Nov 21 14:17:47 VERBOSE[991] logger.c: Adding codec 0x100 (g729) to SDP
Nov 21 14:17:47 VERBOSE[991] logger.c: Transmitting (no NAT) to 172.25.103.222:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b31-5f87fa39-63c2;received=172.25.103.222
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=as3c2b6b62
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1715 at 172.25.96.48>
Content-Type: application/sdp
Content-Length: 183
v=0
o=root 32564 32564 IN IP4 172.25.96.48
s=session
c=IN IP4 172.25.96.48
t=0 0
m=audio 12058 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=silenceSupp:off - - - -
---
Nov 21 14:17:47 VERBOSE[32580] logger.c:
<-- SIP read from 172.25.103.222:5060:
OPTIONS sip:1715 at 172.25.96.48 SIP/2.0
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=as3c2b6b62
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 2 OPTIONS
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b31-5f87fa7c-3510
Max-Forwards: 70
Supported: 100rel,sipvc,replaces,timer
User-Agent: Nortel CS1000 SIP GW release_4.5 version_sse-4.50.88
x-nt-corr-id: 000000b70e1423150b at 0001af0807d6-f087c208
Contact: <sip:1649;phone-context=Exp_Net.ascom at ascom.be:5060;maddr=172.25.103.222;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0
Nov 21 14:17:47 VERBOSE[32580] logger.c: --- (13 headers 0 lines) ---
Nov 21 14:17:47 VERBOSE[32580] logger.c: Looking for 1715 in pra-incoming (domain 172.25.96.48)
Nov 21 14:17:47 VERBOSE[32580] logger.c: Transmitting (no NAT) to 172.25.103.222:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b31-5f87fa7c-3510;received=172.25.103.222
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=as3c2b6b62
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 2 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1715 at 172.25.96.48>
Accept: application/sdp
Content-Length: 0
---
Nov 21 14:17:47 DEBUG[991] rtp.c: Ooh, format changed from unknown to g729
Nov 21 14:17:48 DEBUG[991] rtp.c: Ooh, format changed from unknown to g729
Nov 21 14:17:48 VERBOSE[32580] logger.c:
<-- SIP read from 172.25.96.47:5060:
SIP/2.0 200 OK
From: "1649;phonecontext=Exp_Netascom"<sip:1649;phonecontext=Exp_Netascom at 172.25.96.48>;tag=as0a5a7b3c
To: <sip:1715 at 172.25.96.47>;tag=46d1550-0-13c4-120a2-5b839b6-120a2
Call-ID: 0ff188833a12041d6203999156019426 at 172.25.96.48
CSeq: 102 INVITE
Via: SIP/2.0/UDP 172.25.96.48:5060;rport=5060;branch=z9hG4bK4df8041e
Supported: replaces
Contact: <sip:1715 at 172.25.96.47>
Content-Type: application/SDP
Content-Length: 259
v=0
o=Intel_IPCCLib 74284544 74284545 IN IP4 172.25.96.47
s=Intel_SIP_CCLIB
c=IN IP4 172.25.96.47
t=0 0
m=audio 49152 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=ptime:30
a=fmtp:101 0-15
a=sendrecv
Nov 21 14:17:48 VERBOSE[32580] logger.c: --- (10 headers 12 lines) ---
Nov 21 14:17:48 DEBUG[32580] chan_sip.c: Acked pending invite 102
Nov 21 14:17:48 DEBUG[32580] chan_sip.c: Stopping retransmission on '0ff188833a12041d6203999156019426 at 172.25.96.48' of Request 102: Match Found
Nov 21 14:17:48 VERBOSE[32580] logger.c: Found RTP audio format 18
Nov 21 14:17:48 VERBOSE[32580] logger.c: Found RTP audio format 101
Nov 21 14:17:48 VERBOSE[32580] logger.c: Peer audio RTP is at port 172.25.96.47:49152
Nov 21 14:17:48 DEBUG[32580] chan_sip.c: Peer audio RTP is at port 172.25.96.47:49152
Nov 21 14:17:48 VERBOSE[32580] logger.c: Found description format G729
Nov 21 14:17:48 VERBOSE[32580] logger.c: Found description format telephone-event
Nov 21 14:17:48 VERBOSE[32580] logger.c: Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Nov 21 14:17:48 VERBOSE[32580] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Nov 21 14:17:48 DEBUG[32580] chan_sip.c: build_route: Contact hop: <sip:1715 at 172.25.96.47>
Nov 21 14:17:48 VERBOSE[32580] logger.c: list_route: hop: <sip:1715 at 172.25.96.47>
Nov 21 14:17:48 VERBOSE[32580] logger.c: set_destination: Parsing <sip:1715 at 172.25.96.47> for address/port to send to
Nov 21 14:17:48 VERBOSE[32580] logger.c: set_destination: set destination to 172.25.96.47, port 5060
Nov 21 14:17:48 VERBOSE[32580] logger.c: Transmitting (no NAT) to 172.25.96.47:5060:
ACK sip:1715 at 172.25.96.47 SIP/2.0
Via: SIP/2.0/UDP 172.25.96.48:5060;branch=z9hG4bK57ad5ca3;rport
From: "1649;phonecontext=Exp_Netascom" <sip:1649;phonecontext=Exp_Netascom at 172.25.96.48>;tag=as0a5a7b3c
To: <sip:1715 at 172.25.96.47>;tag=46d1550-0-13c4-120a2-5b839b6-120a2
Contact: <sip:1649;phonecontext=Exp_Netascom at 172.25.96.48>
Call-ID: 0ff188833a12041d6203999156019426 at 172.25.96.48
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Nov 21 14:17:48 VERBOSE[991] logger.c: -- SIP/evavox-08f09cb0 answered SIP/1649-08f029f0
Nov 21 14:17:48 DEBUG[991] chan_sip.c: sip_answer(SIP/1649-08f029f0)
Nov 21 14:17:48 VERBOSE[991] logger.c: We're at 172.25.96.48 port 12058
Nov 21 14:17:48 VERBOSE[991] logger.c: Adding codec 0x100 (g729) to SDP
Nov 21 14:17:48 VERBOSE[991] logger.c: Reliably Transmitting (no NAT) to 172.25.103.222:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b31-5f87fa39-63c2;received=172.25.103.222
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=as3c2b6b62
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1715 at 172.25.96.48>
Content-Type: application/sdp
Content-Length: 183
v=0
o=root 32564 32565 IN IP4 172.25.96.48
s=session
c=IN IP4 172.25.96.48
t=0 0
m=audio 12058 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=silenceSupp:off - - - -
---
Nov 21 14:17:48 VERBOSE[991] logger.c: -- Attempting native bridge of SIP/1649-08f029f0 and SIP/evavox-08f09cb0
Nov 21 14:17:48 VERBOSE[32580] logger.c:
<-- SIP read from 172.25.103.222:5060:
ACK sip:1715 at 172.25.96.48 SIP/2.0
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=as3c2b6b62
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 1 ACK
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b32-5f87fe64-2809
Max-Forwards: 70
User-Agent: Nortel CS1000 SIP GW release_4.5 version_sse-4.50.88
x-nt-corr-id: 000000b70e1423150b at 0001af0807d6-f087c208
Contact: <sip:1649;phone-context=Exp_Net.ascom at ascom.be:5060;maddr=172.25.103.222;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0
Nov 21 14:17:48 VERBOSE[32580] logger.c: --- (12 headers 0 lines) ---
Nov 21 14:17:48 DEBUG[32580] chan_sip.c: Ignoring too old SIP packet packet 1 (expecting >= 2)
Nov 21 14:17:48 DEBUG[32580] chan_sip.c: SIP message could not be handled, bad request: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
Nov 21 14:17:49 VERBOSE[32580] logger.c: Retransmitting #1 (no NAT) to 172.25.103.222:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b31-5f87fa39-63c2;received=172.25.103.222
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=as3c2b6b62
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1715 at 172.25.96.48>
Content-Type: application/sdp
Content-Length: 183
v=0
o=root 32564 32565 IN IP4 172.25.96.48
s=session
c=IN IP4 172.25.96.48
t=0 0
m=audio 12058 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=silenceSupp:off - - - -
---
Nov 21 14:17:49 VERBOSE[32580] logger.c:
<-- SIP read from 172.25.103.222:5060:
ACK sip:1715 at 172.25.96.48 SIP/2.0
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=as3c2b6b62
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 1 ACK
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b32-5f87fe64-2809
Max-Forwards: 70
User-Agent: Nortel CS1000 SIP GW release_4.5 version_sse-4.50.88
x-nt-corr-id: 000000b70e1423150b at 0001af0807d6-f087c208
Contact: <sip:1649;phone-context=Exp_Net.ascom at ascom.be:5060;maddr=172.25.103.222;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0
Nov 21 14:17:49 VERBOSE[32580] logger.c: --- (12 headers 0 lines) ---
Nov 21 14:17:49 DEBUG[32580] chan_sip.c: Ignoring too old SIP packet packet 1 (expecting >= 2)
Nov 21 14:17:49 DEBUG[32580] chan_sip.c: SIP message could not be handled, bad request: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
Nov 21 14:17:50 DEBUG[991] rtp.c: Got RTCP report of 76 bytes
Nov 21 14:17:50 VERBOSE[32580] logger.c: Retransmitting #2 (no NAT) to 172.25.103.222:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b31-5f87fa39-63c2;received=172.25.103.222
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=as3c2b6b62
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1715 at 172.25.96.48>
Content-Type: application/sdp
Content-Length: 183
v=0
o=root 32564 32565 IN IP4 172.25.96.48
s=session
c=IN IP4 172.25.96.48
t=0 0
m=audio 12058 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=silenceSupp:off - - - -
---
Nov 21 14:17:50 VERBOSE[32580] logger.c:
<-- SIP read from 172.25.103.222:5060:
ACK sip:1715 at 172.25.96.48 SIP/2.0
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=as3c2b6b62
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 1 ACK
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b32-5f87fe64-2809
Max-Forwards: 70
User-Agent: Nortel CS1000 SIP GW release_4.5 version_sse-4.50.88
x-nt-corr-id: 000000b70e1423150b at 0001af0807d6-f087c208
Contact: <sip:1649;phone-context=Exp_Net.ascom at ascom.be:5060;maddr=172.25.103.222;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0
Nov 21 14:17:50 VERBOSE[32580] logger.c: --- (12 headers 0 lines) ---
Nov 21 14:17:50 DEBUG[32580] chan_sip.c: Ignoring too old SIP packet packet 1 (expecting >= 2)
Nov 21 14:17:50 DEBUG[32580] chan_sip.c: SIP message could not be handled, bad request: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
Nov 21 14:17:52 VERBOSE[32580] logger.c: Retransmitting #3 (no NAT) to 172.25.103.222:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b31-5f87fa39-63c2;received=172.25.103.222
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=as3c2b6b62
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1715 at 172.25.96.48>
Content-Type: application/sdp
Content-Length: 183
v=0
o=root 32564 32565 IN IP4 172.25.96.48
s=session
c=IN IP4 172.25.96.48
t=0 0
m=audio 12058 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=silenceSupp:off - - - -
---
Nov 21 14:17:52 VERBOSE[32580] logger.c:
<-- SIP read from 172.25.103.222:5060:
ACK sip:1715 at 172.25.96.48 SIP/2.0
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=as3c2b6b62
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 1 ACK
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b32-5f87fe64-2809
Max-Forwards: 70
User-Agent: Nortel CS1000 SIP GW release_4.5 version_sse-4.50.88
x-nt-corr-id: 000000b70e1423150b at 0001af0807d6-f087c208
Contact: <sip:1649;phone-context=Exp_Net.ascom at ascom.be:5060;maddr=172.25.103.222;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0
Nov 21 14:17:52 VERBOSE[32580] logger.c: --- (12 headers 0 lines) ---
Nov 21 14:17:52 DEBUG[32580] chan_sip.c: Ignoring too old SIP packet packet 1 (expecting >= 2)
Nov 21 14:17:52 DEBUG[32580] chan_sip.c: SIP message could not be handled, bad request: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
Nov 21 14:17:53 DEBUG[991] rtp.c: Got RTCP report of 76 bytes
Nov 21 14:17:56 VERBOSE[32580] logger.c: Retransmitting #4 (no NAT) to 172.25.103.222:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b31-5f87fa39-63c2;received=172.25.103.222
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=as3c2b6b62
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1715 at 172.25.96.48>
Content-Type: application/sdp
Content-Length: 183
v=0
o=root 32564 32565 IN IP4 172.25.96.48
s=session
c=IN IP4 172.25.96.48
t=0 0
m=audio 12058 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=silenceSupp:off - - - -
---
Nov 21 14:17:56 VERBOSE[32580] logger.c:
<-- SIP read from 172.25.103.222:5060:
ACK sip:1715 at 172.25.96.48 SIP/2.0
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=as3c2b6b62
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 1 ACK
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b32-5f87fe64-2809
Max-Forwards: 70
User-Agent: Nortel CS1000 SIP GW release_4.5 version_sse-4.50.88
x-nt-corr-id: 000000b70e1423150b at 0001af0807d6-f087c208
Contact: <sip:1649;phone-context=Exp_Net.ascom at ascom.be:5060;maddr=172.25.103.222;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0
Nov 21 14:17:56 VERBOSE[32580] logger.c: --- (12 headers 0 lines) ---
Nov 21 14:17:56 DEBUG[32580] chan_sip.c: Ignoring too old SIP packet packet 1 (expecting >= 2)
Nov 21 14:17:56 DEBUG[32580] chan_sip.c: SIP message could not be handled, bad request: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
Nov 21 14:17:59 DEBUG[991] rtp.c: Got RTCP report of 76 bytes
Nov 21 14:18:00 VERBOSE[32580] logger.c: Retransmitting #5 (no NAT) to 172.25.103.222:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b31-5f87fa39-63c2;received=172.25.103.222
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=as3c2b6b62
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1715 at 172.25.96.48>
Content-Type: application/sdp
Content-Length: 183
v=0
o=root 32564 32565 IN IP4 172.25.96.48
s=session
c=IN IP4 172.25.96.48
t=0 0
m=audio 12058 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=silenceSupp:off - - - -
---
Nov 21 14:18:00 VERBOSE[32580] logger.c:
<-- SIP read from 172.25.103.222:5060:
ACK sip:1715 at 172.25.96.48 SIP/2.0
From: <sip:1649;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d
To: <sip:1715;phone-context=Exp_Net.ascom at ascom.be;user=phone>;tag=as3c2b6b62
Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
CSeq: 1 ACK
Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b32-5f87fe64-2809
Max-Forwards: 70
User-Agent: Nortel CS1000 SIP GW release_4.5 version_sse-4.50.88
x-nt-corr-id: 000000b70e1423150b at 0001af0807d6-f087c208
Contact: <sip:1649;phone-context=Exp_Net.ascom at ascom.be:5060;maddr=172.25.103.222;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0
Nov 21 14:18:00 VERBOSE[32580] logger.c: --- (12 headers 0 lines) ---
Nov 21 14:18:00 DEBUG[32580] chan_sip.c: Ignoring too old SIP packet packet 1 (expecting >= 2)
Nov 21 14:18:00 DEBUG[32580] chan_sip.c: SIP message could not be handled, bad request: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37 at ascom.be
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