Hello to everybody,
I have Asterisk on Debian. To the Asterisk I have connected
Cisco7940(SIP firmware). I want use SIP to call to the Cisco7940-line
32 (only play some rtp stream). I try make my own scenario (shown
below). Line 31(SIPp) should call to line 32(Cisco7940). Line 31 is at
first registered on asterisk(accordingly to the scenario) - this is
ok. But when it try to send INVITE message it comes wrong. Asterisk
write this:
<------------>
Scheduling destruction of SIP dialog '1-3...@127.0.1.1' in 32000 ms
(Method: REGISTER)
gw-server*CLI>
<--- SIP read from 192.168.0.1:5061 --->
INVITE sip:3...@192.168.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5061;branch=z9hG4bK-3618-1-13
From: 31 <sip:3...@127.0.1.1:5061>;tag=1
To: <sip:3...@192.168.0.1:5060>
Call-ID: 1-3...@127.0.1.1
CSeq: 1 INVITE
Contact: <sip:3...@127.0.1.1:5061>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 142
v=0
o=user1 53655765 2353687637 IN IP4 127.0.1.1
s= SIPp Audio Call
c=IN IP4 127.0.1.1
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
<------------->
--- (11 headers 7 lines) ---
Ignoring this INVITE request
================================================================
I try change INVITE message in many ways but with no success.
Configuration of asterisk is basic. The SIPp scenario isn´t complete
but it never goes over the INVITE :((. Thanks for any help
Ondrej
================================================================
users.csv
SEQUENTIAL
#caller
31;[authentication username=31 password=uac]
32
================================================================
Scenario
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="UAC">
<send retrans="500" >
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
To: <sip:[fiel...@[remote_ip]:[remote_port]>
From: <sip:[fiel...@[remote_ip]:[remote_port]>
Contact: <sip:[fiel...@[local_ip]:[local_port]>;transport=[transport]
Expires: 300
Call-ID: [call_id]
CSeq: 1 REGISTER
Content-Length: 0
]]>
</send>
<recv response="100" optional="true" >
</recv>
<!-- 200OK -->
<recv response="200" optional="true" next="2">
</recv>
<!-- 404 - Not Found - User not found (sip:u...@domain...) -->
<recv response="404" optional="true" next="3">
</recv>
<recv response="401" auth="true" optional="true" next="1">
</recv>
<recv response="407" auth="true" optional="false" crlf="true">
</recv>
<label id="1" />
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
To: <sip:[fiel...@[remote_ip]:[remote_port]>
From: <sip:[fiel...@[remote_ip]:[remote_port]>
Contact: <sip:[fiel...@[local_ip]:[local_port]>;transport=[transport]
Expires: 300
Call-ID: [call_id]
CSeq: 2 REGISTER
Content-Length: 0
[field1]
]]>
</send>
<recv response="100" optional="true">
</recv>
<!-- 401 - Unauthorized. Unsuccessful Registration >> RFC 3665 -->
<recv response="401" optional="true" next="3">
</recv>
<!-- 403 - Forbidden (bad password) -->
<recv response="403" optional="true" next="3">
</recv>
<!-- 404 - Not Found - User not found -->
<recv response="404" optional="true" next="3">
</recv>
<!-- 200 OK -->
<recv response="200" optional="false" crlf="true" >
</recv>
<label id="2" />
<!-- Zacatek hovoru -->
<pause milliseconds="2000"/>
<send retrans="500">
<![CDATA[
INVITE sip:[field0 line...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0
line...@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[field0 line...@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: <sip:[field0 line...@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s= SIPp Audio Call
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="407" auth="true">
</recv>
<send>
<![CDATA[
ACK sip:[field0 line...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0
line...@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[field0 line...@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:[field0 line...@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
INVITE sip:[field0 line...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0
line...@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[field0 line...@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 2 INVITE
Contact: sip:[field0 line...@[local_ip]:[local_port]
[field1 line=0]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s= SIPp Audio Call
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="183" optional="true">
</recv>
<recv response="200" >
</recv>
<send>
<![CDATA[
ACK sip:[field0 line...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0
line...@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[field0 line...@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 ACK
Contact: sip:[field0 line...@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- Cancellation of Registration -->
<!-- Zruseni registrace -->
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
To: <sip:[fiel...@sip.com:[remote_port]>
From: <sip:[fiel...@[remote_ip]:[remote_port]>
Contact: <sip:[fiel...@[local_ip]:[local_port]>;transport=[transport]
[field1]
Expires: 0
Call-ID: [call_id]
CSeq: 3 REGISTER
Content-Length: 0
]]>
</send>
<recv response="100" optional="true" >
</recv>
<recv response="200" optional="false" next="3">
</recv>
<label id="3" />
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
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