rendered paste body
*CLI>
<--- SIP read from 10.200.201.2:5061 --->
INVITE sip:6000089@10.156.1.242;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.200.201.2:5061;branch=z9hG4bKc6655e8b3
Call-ID: 2297cac624f0358f0e5a61e64517a4ec@10.200.201.2
From: <sip:918613631@10.200.201.2;user=phone>;tag=1d567e44
To: <sip:6000089@10.156.1.242;user=phone>
CSeq: 1 INVITE
Contact: <sip:918613631@10.200.201.2:5061;user=phone>
Supported: 100rel
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER
Content-Length: 296
Content-Type: application/sdp
v=0
o=HuaweiSoftX3000 135538 135538 IN IP4 10.200.201.2
s=Sip Call
c=IN IP4 10.200.201.3
t=0 0
m=audio 10776 RTP/AVP 8 0 18 4 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes
<------------->
--- (12 headers 13 lines) ---
Sending to 10.200.201.2 : 5061 (no NAT)
Using INVITE request as basis request - 2297cac624f0358f0e5a61e64517a4ec@10.200.201.2
Found peer 'soft-x-ngn'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 97
Peer audio RTP is at port 10.200.201.3:10776
Found description format PCMA for ID 8
Found description format PCMU for ID 0
Found description format G729 for ID 18
Found description format G723 for ID 4
Found description format telephone-event for ID 97
Capabilities: us - 0x4 (ulaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/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 10.200.201.3:10776
Looking for 6000089 in ivr-1-in (domain 10.156.1.242)
list_route: hop: <sip:918613631@10.200.201.2:5061;user=phone>
<--- Transmitting (no NAT) to 10.200.201.2:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.200.201.2:5061;branch=z9hG4bKc6655e8b3;received=10.200.201.2
From: <sip:918613631@10.200.201.2;user=phone>;tag=1d567e44
To: <sip:6000089@10.156.1.242;user=phone>
Call-ID: 2297cac624f0358f0e5a61e64517a4ec@10.200.201.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6000089@10.156.1.242>
Content-Length: 0
<------------>
Audio is at 10.156.1.242 port 16558
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 10.200.201.2:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.201.2:5061;branch=z9hG4bKc6655e8b3;received=10.200.201.2
From: <sip:918613631@10.200.201.2;user=phone>;tag=1d567e44
To: <sip:6000089@10.156.1.242;user=phone>;tag=as4a040f32
Call-ID: 2297cac624f0358f0e5a61e64517a4ec@10.200.201.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6000089@10.156.1.242>
Content-Type: application/sdp
Content-Length: 237
v=0
o=root 13496 13496 IN IP4 10.156.1.242
s=session
c=IN IP4 10.156.1.242
t=0 0
m=audio 16558 RTP/AVP 0 97
a=rtpmap:0 PCMU/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
<--- SIP read from 10.200.201.2:5061 --->
ACK sip:6000089@10.156.1.242 SIP/2.0
Via: SIP/2.0/UDP 10.200.201.2:5061;branch=z9hG4bK1e7385b15
Via: SIP/2.0/UDP 10.200.201.2:5061;branch=z9hG4bK1e7385b15
Call-ID: 2297cac624f0358f0e5a61e64517a4ec@10.200.201.2
From: <sip:918613631@10.200.201.2;user=phone>;tag=1d567e44
To: <sip:6000089@10.156.1.242;user=phone>;tag=as4a040f32
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from 10.200.201.2:5061 --->
BYE sip:6000089@10.156.1.242 SIP/2.0
Via: SIP/2.0/UDP 10.200.201.2:5061;branch=z9hG4bK566bd1b8f
Call-ID: 2297cac624f0358f0e5a61e64517a4ec@10.200.201.2
From: <sip:918613631@10.200.201.2;user=phone>;tag=1d567e44
To: <sip:6000089@10.156.1.242;user=phone>;tag=as4a040f32
CSeq: 2 BYE
Reason: Q.850;cause=100;text="unknown"
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Sending to 10.200.201.2 : 5061 (no NAT)
<--- Transmitting (no NAT) to 10.200.201.2:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.201.2:5061;branch=z9hG4bK566bd1b8f;received=10.200.201.2
From: <sip:918613631@10.200.201.2;user=phone>;tag=1d567e44
To: <sip:6000089@10.156.1.242;user=phone>;tag=as4a040f32
Call-ID: 2297cac624f0358f0e5a61e64517a4ec@10.200.201.2
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6000089@10.156.1.242>
Content-Length: 0
<------------>
== Spawn extension (ivr-1-in, 6000089, 2) exited non-zero on 'SIP/10.200.201.2-007efe80'
Really destroying SIP dialog '2297cac624f0358f0e5a61e64517a4ec@10.200.201.2' Method: BYE