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hemanshurpatel at gmail.c · Guest

hello there

I am doing performance testing for VICIDialers, so that we can
understand about our basic needs regarding hardware to fulfill our
requirements.

i have follow all performancetesting.txt file and i have also created
remoteagent with 999999999999 extention.
but when i make thsi agent active i got 404 from the other sip server
here is the complete log file:

INVITE sip:VICItest:test@192.168.1.32 ([email]test%40192.168.1.32[/email]):5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK1c21d189;rport
From: "T12101048458600051" <sip:asterisk@192.168.1.65 ([email]asterisk%40192.168.1.65[/email])>;tag=as7b57232a
To: <sip:VICItest:test@192.168.1.32 ([email]test%40192.168.1.32[/email]):5070>
Contact: <sip:asterisk@192.168.1.65 ([email]asterisk%40192.168.1.65[/email])>
Call-ID: 40bca60d23710fcc4ff075196599bec3@192.168.1.65 ([email]40bca60d23710fcc4ff075196599bec3%40192.168.1.65[/email])
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 10 Dec 2007 05:18:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 23511 23511 IN IP4 192.168.1.65
s=session
c=IN IP4 192.168.1.65
t=0 0
m=audio 16210 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (13 headers 12 lines) ---
Using INVITE request as basis request -
40bca60d23710fcc4ff075196599bec3@192.168.1.65 ([email]40bca60d23710fcc4ff075196599bec3%40192.168.1.65[/email])
Sending to 192.168.1.65 : 5060 (NAT)
Found peer 'VICItest'
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.65:16210
Peer video RTP is at port 192.168.1.65:65535
Found description format GSM
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0xe
(gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for VICItest:test in default (domain 192.168.1.32)
Reliably Transmitting (no NAT) to 192.168.1.65:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.1.65:5060;branch=z9hG4bK1c21d189;received=192.168.1.65;rport=5060
From: "T12101048458600051" <sip:asterisk@192.168.1.65 ([email]asterisk%40192.168.1.65[/email])>;tag=as7b57232a
To: <sip:VICItest:test@192.168.1.32 ([email]test%40192.168.1.32[/email]):5070>;tag=as71ebff8e
Call-ID: 40bca60d23710fcc4ff075196599bec3@192.168.1.65 ([email]40bca60d23710fcc4ff075196599bec3%40192.168.1.65[/email])
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---
localhost*CLI>
<-- SIP read from 192.168.1.65:5060:
ACK sip:VICItest:test@192.168.1.32 ([email]test%40192.168.1.32[/email]):5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK1c21d189;rport
From: "T12101048458600051" <sip:asterisk@192.168.1.65 ([email]asterisk%40192.168.1.65[/email])>;tag=as7b57232a
To: <sip:VICItest:test@192.168.1.32 ([email]test%40192.168.1.32[/email]):5070>;tag=as71ebff8e
Contact: <sip:asterisk@192.168.1.65 ([email]asterisk%40192.168.1.65[/email])>
Call-ID: 40bca60d23710fcc4ff075196599bec3@192.168.1.65 ([email]40bca60d23710fcc4ff075196599bec3%40192.168.1.65[/email])
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

--- (10 headers 0 lines) ---
Destroying call '40bca60d23710fcc4ff075196599bec3@192.168.1.65 ([email]%2740bca60d23710fcc4ff075196599bec3%40192.168.1.65[/email])'

i am not getting what can be the problem, and looking for your help!!!!

regards,
Hemanshu Patel

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