rendered paste bodypbx*CLI> dial 13806
-- Executing NoOp("OSS/dsp", "call_agi") in new stack
-- Executing Set("OSS/dsp", "__ORIG_CHAN=OSS/dsp") in new stack
-- Executing AGI("OSS/dsp", "agi://localhost:5555/") in new stack
-- AGI Script Executing Application: (DIAL) Options: (SIP/21.0.20.11/5004||ht)
-------------------------------------
Total wait time 36 seconds before anything else occurs
-------------------------------------
Mar 13 12:46:06 DEBUG[3671]: chan_sip.c:2085 sip_call: Outgoing Call for 5004
We're at 21.0.20.10 port 10030
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 21.0.20.11:5060:
INVITE sip:5004@21.0.20.11 SIP/2.0
Via: SIP/2.0/UDP 21.0.20.10:5060;branch=z9hG4bK78858200;rport
From: "asterisk" <sip:asterisk@21.0.20.10>;tag=as1e6f9d84
To: <sip:5004@21.0.20.11>
Contact: <sip:asterisk@21.0.20.10>
Call-ID: 1e1adfae607f8881463e5aa461b1d33e@21.0.20.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 13 Mar 2008 17:46:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 210
v=0
o=root 5435 5435 IN IP4 21.0.20.10
s=session
c=IN IP4 21.0.20.10
t=0 0
m=audio 10030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called 21.0.20.11/5004
pbx*CLI>
<-- SIP read from 21.0.20.11:1243:
SIP/2.0 100 Trying
To: <sip:5004@21.0.20.11>;tag=29655
From: "asterisk" <sip:asterisk@21.0.20.10>;tag=as1e6f9d84
Via: SIP/2.0/UDP 21.0.20.10:5060;branch=z9hG4bK78858200;rport
CSeq: 102 INVITE
Call-ID: 1e1adfae607f8881463e5aa461b1d33e@21.0.20.10
Server: ININ-CUMAICSRVR-36357178
Content-Length: 0
--- (8 headers 0 lines) ---
Mar 13 12:46:06 DEBUG[6064]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '1e1adfae607f8881463e5aa461b1d33e@21.0.20.10' Request 102: Found
pbx*CLI>
<-- SIP read from 21.0.20.11:1243:
SIP/2.0 180 Ringing
To: <sip:5004@21.0.20.11>;tag=29655
From: "asterisk" <sip:asterisk@21.0.20.10>;tag=as1e6f9d84
Via: SIP/2.0/UDP 21.0.20.10:5060;branch=z9hG4bK78858200;rport
CSeq: 102 INVITE
Call-ID: 1e1adfae607f8881463e5aa461b1d33e@21.0.20.10
Server: ININ-CUMAICSRVR-36357178
Content-Length: 0
--- (8 headers 0 lines) ---
Mar 13 12:46:06 DEBUG[6064]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '1e1adfae607f8881463e5aa461b1d33e@21.0.20.10' Request 102: Found
-- SIP/21.0.20.11-081d6678 is ringing
Mar 13 12:46:06 WARNING[6069]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
pbx*CLI>
<-- SIP read from 21.0.20.11:1243:
SIP/2.0 200 OK
To: <sip:5004@21.0.20.11>;tag=29655
From: "asterisk" <sip:asterisk@21.0.20.10>;tag=as1e6f9d84
Via: SIP/2.0/UDP 21.0.20.10:5060;branch=z9hG4bK78858200;rport
CSeq: 102 INVITE
Call-ID: 1e1adfae607f8881463e5aa461b1d33e@21.0.20.10
Contact: <sip:5004@21.0.20.11>
Content-Type: application/sdp
Server: ININ-CUMAICSRVR-36357178
Content-Length: 194
v=0
o=ININ 44362390 44362390 IN IP4 21.0.20.11
s=Interaction
c=IN IP4 21.0.20.12
t=0 0
m=audio 4770 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (10 headers 9 lines) ---
Mar 13 12:46:06 DEBUG[6064]: chan_sip.c:1392 __sip_ack: Acked pending invite 102
Mar 13 12:46:06 DEBUG[6064]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '1e1adfae607f8881463e5aa461b1d33e@21.0.20.10' of Request 102: Match Found
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 21.0.20.12:4770
Mar 13 12:46:06 DEBUG[6064]: chan_sip.c:3689 process_sdp: Peer audio RTP is at port 21.0.20.12:4770
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Mar 13 12:46:06 DEBUG[6064]: chan_sip.c:6282 build_route: build_route: Contact hop: <sip:5004@21.0.20.11>
list_route: hop: <sip:5004@21.0.20.11>
set_destination: Parsing <sip:5004@21.0.20.11> for address/port to send to
set_destination: set destination to 21.0.20.11, port 5060
Transmitting (no NAT) to 21.0.20.11:5060:
ACK sip:5004@21.0.20.11 SIP/2.0
Via: SIP/2.0/UDP 21.0.20.10:5060;branch=z9hG4bK2c09f15b;rport
From: "asterisk" <sip:asterisk@21.0.20.10>;tag=as1e6f9d84
To: <sip:5004@21.0.20.11>;tag=29655
Contact: <sip:asterisk@21.0.20.10>
Call-ID: 1e1adfae607f8881463e5aa461b1d33e@21.0.20.10
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/21.0.20.11-081d6678 answered OSS/dsp
<< Console call has been answered >>