rendered paste body Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 64.2.142.17:5060:
INVITE sip:1xxxxxxxxxx@outbound1.vitelity.net SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2fa31702;rport
From: "102" <sip:myusername@yyy.yyy.yyy.yyy>;tag=as55937ff3
To: <sip:1xxxxxxxxxx@outbound1.vitelity.net>
Contact: <sip:myusername@yyy.yyy.yyy.yyy>
Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "102" <sip:102@yyy.yyy.yyy.yyy>;privacy=off;screen=no
Date: Sat, 29 Dec 2007 23:31:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 473
v=0
o=root 2926 2926 IN IP4 yyy.yyy.yyy.yyy
s=session
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 11844 RTP/AVP 0 3 8 112 5 10 7 97 111 101 -- Re-regist
ration for myusername@inbound5.vitelity.net
[Dec 29 18:32:56] NOTICE[2933]: chan_sip.c:12113 handle_response_register: Outb
ound Registration: Expiry for inbound5.vitelity.net is 60 sec (Scheduling reregi
stration in 45 s)-G726-32/8000
Really destroying SIP dialog '567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy' M
ethod: INVITE16/8000
litclinux*CLI>
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
litclinux*CLI>
<--- SIP read from 64.2.142.17:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2fa31702;received=yyy.yyy.yyy.yyy;rport=5060
From: "102" <sip:myusername@yyy.yyy.yyy.yyy>;tag=as55937ff3
To: <sip:1xxxxxxxxxx@outbound1.vitelity.net>;tag=as4abecb20
Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a8efe03"
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 64.2.142.17:5060:
ACK sip:1xxxxxxxxxx@outbound1.vitelity.net SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2fa31702;rport
From: "102" <sip:myusername@yyy.yyy.yyy.yyy>;tag=as55937ff3
To: <sip:1xxxxxxxxxx@outbound1.vitelity.net>;tag=as4abecb20
Contact: <sip:myusername@yyy.yyy.yyy.yyy>
Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "102" <sip:102@yyy.yyy.yyy.yyy>;privacy=off;screen=no
Content-Length: 0
---
Audio is at yyy.yyy.yyy.yyy port 11844
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 64.2.142.17:5060:
INVITE sip:1xxxxxxxxxx@outbound1.vitelity.net SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5c65a47d;rport
From: "102" <sip:myusername@yyy.yyy.yyy.yyy>;tag=as55937ff3
To: <sip:1xxxxxxxxxx@outbound1.vitelity.net>
Contact: <sip:myusername@yyy.yyy.yyy.yyy>
Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "102" <sip:102@yyy.yyy.yyy.yyy>;privacy=off;screen=no
Proxy-Authorization: Digest username="myusername", realm="asterisk", algorithm=MD5,
uri="sip:1xxxxxxxxxx@outbound1.vitelity.net", nonce="0a8efe03", response="82d12
466c5c546b18f03534b55feb6ef", opaque=""
Date: Sat, 29 Dec 2007 23:31:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 473
v=0
o=root 2926 2927 IN IP4 yyy.yyy.yyy.yyy
s=session
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 11844 RTP/AVP 0 3 8 112 5 10 7 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
litclinux*CLI>
<--- SIP read from 64.2.142.17:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5c65a47d;received=yyy.yyy.yyy.yyy;rport=5060
From: "102" <sip:myusername@yyy.yyy.yyy.yyy>;tag=as55937ff3
To: <sip:1xxxxxxxxxx@outbound1.vitelity.net>
Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1xxxxxxxxxx@64.2.142.17>
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
[Dec 29 18:32:11] NOTICE[2933]: chan_sip.c:7145 sip_reregister: -- Re-regist
ration for myusername@inbound5.vitelity.net
[Dec 29 18:32:11] NOTICE[2933]: chan_sip.c:12113 handle_response_register: Outb
ound Registration: Expiry for inbound5.vitelity.net is 60 sec (Scheduling reregi
stration in 45 s)
Reliably Transmitting (NAT) to 64.2.142.17:5060:
OPTIONS sip:outbound1.vitelity.net SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK0aa2be62;rport
From: "asterisk" <sip:asterisk@yyy.yyy.yyy.yyy>;tag=as5032586a
To: <sip:outbound1.vitelity.net>
Contact: <sip:asterisk@yyy.yyy.yyy.yyy>
Call-ID: 4aa917382181b8ff6e1c90d7275db8bd@yyy.yyy.yyy.yyy
CSeq: 102 OPTIONS
[Dec 29 18:32:56] NOTICE[2933]: chan_sip.c:7145 sip_reregister:
Max-Forwards: 70
Date: Sat, 29 Dec 2007 23:32:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
litclinux*CLI>
<--- SIP read from 64.2.142.17:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK0aa2be62;received=yyy.yyy.yyy.yyy;rport=5060
From: "asterisk" <sip:asterisk@yyy.yyy.yyy.yyy>;tag=as5032586a
To: <sip:outbound1.vitelity.net>;tag=as09be9d34
Call-ID: 4aa917382181b8ff6e1c90d7275db8bd@yyy.yyy.yyy.yyy
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '4aa917382181b8ff6e1c90d7275db8bd@yyy.yyy.yyy.yyy'
Method: OPTIONS
litclinux*CLI>
<--- SIP read from 64.2.142.17:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5c65a47d;received=yyy.yyy.yyy.yyy;rport=5060
From: "102" <sip:myusername@yyy.yyy.yyy.yyy>;tag=as55937ff3
To: <sip:1xxxxxxxxxx@outbound1.vitelity.net>;tag=as2b448282
Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1xxxxxxxxxx@64.2.142.17>
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 5396 5396 IN IP4 64.2.142.17
s=session
c=IN IP4 64.2.142.17
t=0 0
m=audio 12534 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
<------------->
--- (11 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 64.2.142.17:12534
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format GSM for ID 3
Found description format telephone-event for ID 101
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|spee
x|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0xe (gsm|ulaw|
alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephon
e-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 64.2.142.17:12534
Scheduling destruction of SIP dialog '567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 64.2.142.17:5060:
CANCEL sip:1xxxxxxxxxx@outbound1.vitelity.net SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5c65a47d;rport
From: "102" <sip:myusername@yyy.yyy.yyy.yyy>;tag=as55937ff3
To: <sip:1xxxxxxxxxx@outbound1.vitelity.net>
Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
emote-Party-ID: "102" <sip:102@yyy.yyy.yyy.yyy>;privacy=off;screen=no
Content-Length: 0
---
Scheduling destruction of SIP dialog '567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy' in 6400 ms (Method: INVITE)
litclinux*CLI>
<--- SIP read from 64.2.142.17:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5c65a47d;received=yyy.yyy.yyy.yyy;rport=5060
From: "102" <sip:myusername@yyy.yyy.yyy.yyy>;tag=as55937ff3
To: <sip:1xxxxxxxxxx@outbound1.vitelity.net>;tag=as2b448282
Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
X-Asterisk-HangupCause: Network out of order
<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 64.2.142.17:5060:
ACK sip:1xxxxxxxxxx@outbound1.vitelity.net SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5c65a47d;rport
From: "102" <sip:myusername@yyy.yyy.yyy.yyy>;tag=as55937ff3
To: <sip:1xxxxxxxxxx@outbound1.vitelity.net>;tag=as2b448282
Contact: <sip:myusername@yyy.yyy.yyy.yyy>
Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "102" <sip:102@yyy.yyy.yyy.yyy>;privacy=off;screen=no
Content-Length: 0
---
litclinux*CLI>
<--- SIP read from 64.2.142.17:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5c65a47d;received=yyy.yyy.yyy.yyy;rport=5060
From: "102" <sip:myusername@yyy.yyy.yyy.yyy>;tag=as55937ff3
To: <sip:1xxxxxxxxxx@outbound1.vitelity.net>;tag=as2b448282
Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1xxxxxxxxxx@64.2.142.17>
Content-Length: 0
X-Asterisk-HangupCause: Network out of order
<------------->
--- (11 headers 0 lines) ---
litclinux*CLI>