rendered paste bodybender*CLI> sip set debug peer 880
SIP Debugging Enabled for IP: 192.168.8.2:5060
-- Accepting AUTHENTICATED call from 200.174.44.76:
> requested format = gsm,
> requested prefs = (),
> actual format = gsm,
> host prefs = (alaw|gsm),
> priority = mine
-- Executing [889@ramais:1] Dial("IAX2/central-7", "SIP/880|30|Ttr") in new stack
Audio is at 192.168.8.1 port 14142
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.8.2:5060:
INVITE sip:880@192.168.8.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK737a8895;rport
From: "Flavio do Carmo Junior" <sip:402@192.168.8.1>;tag=as5d5f06ee
To: <sip:880@192.168.8.2:5060>
Contact: <sip:402@192.168.8.1>
Call-ID: 2cb94a57528739a343de46b759ef9e7f@192.168.8.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 10 Jul 2007 18:00:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 260
v=0
o=root 3328 3328 IN IP4 192.168.8.1
s=session
c=IN IP4 192.168.8.1
t=0 0
m=audio 14142 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 880
bender*CLI>
<--- SIP read from 192.168.8.2:5060 --->
SIP/2.0 100 Trying
To: <sip:880@192.168.8.2:5060>
From: "Flavio do Carmo Junior" <sip:402@192.168.8.1>;tag=as5d5f06ee
Call-ID: 2cb94a57528739a343de46b759ef9e7f@192.168.8.1
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK737a8895
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
bender*CLI>
<--- SIP read from 192.168.8.2:5060 --->
SIP/2.0 180 Ringing
To: <sip:880@192.168.8.2:5060>;tag=3af6fd51637e6f1ei0
From: "Flavio do Carmo Junior" <sip:402@192.168.8.1>;tag=as5d5f06ee
Call-ID: 2cb94a57528739a343de46b759ef9e7f@192.168.8.1
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK737a8895
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
-- SIP/880-082080a8 is ringing
bender*CLI>
<--- SIP read from 192.168.8.2:5060 --->
SIP/2.0 200 OK
To: <sip:880@192.168.8.2:5060>;tag=3af6fd51637e6f1ei0
From: "Flavio do Carmo Junior" <sip:402@192.168.8.1>;tag=as5d5f06ee
Call-ID: 2cb94a57528739a343de46b759ef9e7f@192.168.8.1
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK737a8895
Contact: 880 <sip:880@192.168.8.2:5060>
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 229
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 65980 65980 IN IP4 192.168.8.2
s=-
c=IN IP4 192.168.8.2
t=0 0
m=audio 16450 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.8.2:16450
Found description format PCMU for ID 0
Found description format NSE for ID 100
Found description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.8.2:16450
list_route: hop: <sip:880@192.168.8.2:5060>
set_destination: Parsing <sip:880@192.168.8.2:5060> for address/port to send to
set_destination: set destination to 192.168.8.2, port 5060
Transmitting (no NAT) to 192.168.8.2:5060:
ACK sip:880@192.168.8.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK396d2ed0;rport
From: "Flavio do Carmo Junior" <sip:402@192.168.8.1>;tag=as5d5f06ee
To: <sip:880@192.168.8.2:5060>;tag=3af6fd51637e6f1ei0
Contact: <sip:402@192.168.8.1>
Call-ID: 2cb94a57528739a343de46b759ef9e7f@192.168.8.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/880-082080a8 answered IAX2/central-7
[Jul 10 15:00:24] WARNING[3338]: chan_iax2.c:7760 socket_process: Received mini frame before first full voice frame
<--- SIP read from 192.168.8.2:5060 --->
BYE sip:402@192.168.8.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.2:5060;branch=z9hG4bK-5a618fcb
From: <sip:880@192.168.8.2:5060>;tag=3af6fd51637e6f1ei0
To: "Flavio do Carmo Junior" <sip:402@192.168.8.1>;tag=as5d5f06ee
Call-ID: 2cb94a57528739a343de46b759ef9e7f@192.168.8.1
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Linksys/PAP2-3.1.9(LSc)
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.8.2 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.8.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.2:5060;branch=z9hG4bK-5a618fcb;received=192.168.8.2
From: <sip:880@192.168.8.2:5060>;tag=3af6fd51637e6f1ei0
To: "Flavio do Carmo Junior" <sip:402@192.168.8.1>;tag=as5d5f06ee
Call-ID: 2cb94a57528739a343de46b759ef9e7f@192.168.8.1
CSeq: 101 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:402@192.168.8.1>
Content-Length: 0
<------------>
== Spawn extension (ramais, 889, 1) exited non-zero on 'IAX2/central-7'
-- Hungup 'IAX2/central-7'
Really destroying SIP dialog '2cb94a57528739a343de46b759ef9e7f@192.168.8.1' Method: BYE
bender*CLI>