Dear Rob
I am no longer receiving a 404 error since I am now using an xml file on linux
on the same device that my kamailio SIP is on. However, my calls are still
failing and receiving an "aborting call on UDP re-transmission timeout for
Call-ID. Please share any thoughts that you have on this predicament. My xml
file is attached below.
Thankfully,
David
<scenario name="UAC REGISTER + INVITE + call">
<!--
Use with CSV file struct like: 32;192.168.1.211;[authentication username=32
password=32];21; (user part of uri, server address, auth tag, call target)
-->
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0 Via: SIP/2.0/[transport]
[local_ip]:[local_port];branch=[branch] From:
<sip:[field0]@[field1]>;tag=[pid]SIPpTag00[call_number] To:
<sip:[field0]@[field1]> Call-ID: [call_id] CSeq: [cseq] REGISTER Contact:
sip:[field0]@[local_ip]:[local_port] Max-Forwards: 10 Expires: 120 User-Agent:
SIPp/Win32 Content-Length: 0
]]>
</send>
<!-- asterisk -->
<recv response="100" optional="true"></recv>
<recv response="401" auth="true"></recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0 Via: SIP/2.0/[transport]
[local_ip]:[local_port];branch=[branch] From:
<sip:[field0]@[field1]>;tag=[pid]SIPpTag00[call_number] To:
<sip:[field0]@[field1]> Call-ID: [call_id] CSeq: [cseq] REGISTER Contact:
sip:[field0]@[local_ip]:[local_port] [field2] Max-Forwards: 10 Expires: 120
User-Agent: SIPp/Win32 Content-Length: 0
]]>
</send>
<!-- asterisk -->
<recv response="100" optional="true"></recv>
<recv response="200"></recv>
<send retrans="500">
<![CDATA[
INVITE sip:[field3]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport]
[local_ip]:[local_port];branch=[branch] From: sipp
<sip:[field0]@[field1]>;tag=[pid]SIPpTag00[call_number] To:
<sip:[field3]@[field1]:[remote_port]> Call-ID: [call_id] CSeq: [cseq] INVITE
Contact: sip:[field0]@[local_ip]:[local_port] Max-Forwards: 10 Content-Type:
application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN
IP[local_ip_type] [local_ip] s=- 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:[field3]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport]
[local_ip]:[local_port];branch=[branch] From:
<sip:[field0]@[field1]>;tag=[pid]SIPpTag00[call_number] [last_To:] Call-ID:
[call_id] CSeq: [cseq] ACK Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 10 Content-Length: 0
]]>
</send>
<pause milliseconds="30000"/>
<send retrans="500">
<![CDATA[
BYE sip:[field3]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport]
[local_ip]:[local_port];branch=[branch] From:
<sip:[field0]@[field1]>;tag=[pid]SIPpTag00[call_number] [last_To:] Call-ID:
[call_id] CSeq: [cseq] BYE Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 10 Content-Length: 0
]]>
</send>
<!--
The 'crlf' option inserts a blank line in the statistics report.
-->
<recv response="200" crlf="true"></recv>
<!--
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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