rendered paste body<-- SIP read from 10.0.2.81:5060:
ACK sip:%231@10.0.1.185 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.81;branch=z9hG4bKe20afe1514C31388
From: "Christopher Wall" <sip:1718@10.0.1.185>;tag=CDC3D18F-7A30EC04
To: <sip:%231@10.0.1.185>;tag=as2ab01f39
CSeq: 1 ACK
Call-ID: 8556aab3-c4ab2731-cd77ee56@10.0.2.81
Contact: <sip:1718@10.0.2.81>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.6.0036
Max-Forwards: 70
Content-Length: 0
--- (11 headers 0 lines)---
dev-asterisk*CLI>
<-- SIP read from 10.0.2.81:5060:
INVITE sip:%231@10.0.1.185 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.81;branch=z9hG4bKfc959a92F7D7E7CD
From: "Christopher Wall" <sip:1718@10.0.1.185>;tag=CDC3D18F-7A30EC04
To: <sip:%231@10.0.1.185>
CSeq: 2 INVITE
Call-ID: 8556aab3-c4ab2731-cd77ee56@10.0.2.81
Contact: <sip:1718@10.0.2.81>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.6.0036
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="1718", realm="asterisk", nonce="572cec2f", uri="sip:%231@10.0.1.185", response="0ad570a8d1af5372e45a9876a3367292", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 243
v=0
o=- 1154451132 1154451132 IN IP4 10.0.2.81
s=Polycom IP Phone
c=IN IP4 10.0.2.81
t=0 0
a=sendrecv
m=audio 2228 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
--- (15 headers 11 lines)---
Using INVITE request as basis request - 8556aab3-c4ab2731-cd77ee56@10.0.2.81
Sending to 10.0.2.81 : 5060 (non-NAT)
Found user '1718'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.0.2.81:2228
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for %231 in progrexion (domain 10.0.1.185)
Reliably Transmitting (no NAT) to 10.0.2.81:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.2.81;branch=z9hG4bKfc959a92F7D7E7CD;received=10.0.2.81
From: "Christopher Wall" <sip:1718@10.0.1.185>;tag=CDC3D18F-7A30EC04
To: <sip:%231@10.0.1.185>;tag=as2ab01f39
Call-ID: 8556aab3-c4ab2731-cd77ee56@10.0.2.81
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:%231@10.0.1.185>
Content-Length: 0
---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'evo_ivr' logged on from 10.0.1.186
== Manager 'evo_ivr' logged off from 10.0.1.186
dev-asterisk*CLI>
<-- SIP read from 10.0.2.81:5060:
ACK sip:%231@10.0.1.185 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.81;branch=z9hG4bKfc959a92F7D7E7CD
From: "Christopher Wall" <sip:1718@10.0.1.185>;tag=CDC3D18F-7A30EC04
To: <sip:%231@10.0.1.185>;tag=as2ab01f39
CSeq: 2 ACK
Call-ID: 8556aab3-c4ab2731-cd77ee56@10.0.2.81
Contact: <sip:1718@10.0.2.81>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.6.0036
Proxy-Authorization: Digest username="1718", realm="asterisk", nonce="572cec2f", uri="sip:%231@10.0.1.185", response="0ad570a8d1af5372e45a9876a3367292", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
--- (12 headers 0 lines)---
Destroying call '8556aab3-c4ab2731-cd77ee56@10.0.2.81'
Destroying call 'd2418ed5-2e900773-93ab3248@10.0.2.81'
-- Executing AgentLogin("SIP/ev_120-03c0", "") in new stack
-- Playing 'agent-user' (language 'en')
dev-asterisk*CLI>