Are you tracing using ethereal / wireshark? This will help with the
capturing the unexpected messages. If you are tracing, what are the
unexpected messages? Also, silly question, have you first registered the
users and are they successfully registered?
Simon
On 8/15/07, Christian Kraus <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> I want to test my Asterisk-Server. Therefore I want to use the included
> uac_pcap.xml.
> I´m posting my configs and only changing the IPs. Hopefully anyone could
> help, as I can´t find something googling for hours/days.
>
>
> I set up:
>
> SIPP-Client: 89.111.111.111
> SIPP-Server: 89.222.222.222
> Asterisk: 89.555.555.555
>
>
> on SIPP-Server I run:
> ./sipp -sn uas -i 89.222.222.222 89.222.222.222
>
> and an SIPP-Client:
> ./sipp -sf uac_pcap.xml -d 100000 -s 2000 89.555.555.555 -l 1000 -r 10
> -rp 1000
>
>
> To the asterisk´s sip.conf I added:
>
> [sipp]
> context=test
> language=en
> type=friend
> host=dynamic
> nat=no
>
> To the extension.conf I added:
>
> [test]
> exten => 2000,1,Dial(SIP/89.222.222.222)
>
> So every caller being 2000, will be directed to the SIPP-Server...
>
>
>
> As you can see the SIPP-Client is sending, but it gets "unexpected
> messages":
>
> ------------------------------ Scenario Screen -------- [1-9]: Change
> Screen --
> Call-rate(length) Port Total-time Total-calls Remote-host
> 10.0(100000 ms)/1.000s 5060 28.01 s 280 89.555.555.555:5060(UDP)
>
> 10 new calls during 1.000 s period 2 ms scheduler resolution
> 127 calls (limit 1000) Peak was 141 calls, after 14 s
> 0 Running, 128 Paused, 10 Woken up
> 0 out-of-call msg (discarded)
> 1 open sockets
> 0 Total RTP pckts sent 0.000 last period RTP rate (kB/s)
>
> Messages Retrans Timeout Unexpected-Msg
> INVITE ----------> 280 657 0
> 100 <---------- 0 0 0 153
> 180 <---------- 0 0 0 0
> 200 <---------- E-RTD1 0 0 0 0
>
> ACK ----------> 0 0
> [ NOP ]
> Pause [ 8000ms] 0 0
> [ NOP ]
> Pause [ 1000ms] 0 0
> BYE ----------> 0 0 0
> 200 <---------- 0 0 0 0
>
> ------ [+|-|*|/]: Adjust rate ---- [q]: Soft exit ---- [p]: Pause
> traffic -----
>
>
>
>
> The SIPP-Server is receiving messages and sends answers back:
>
> ------------------------------ Scenario Screen -------- [1-9]: Change
> Screen --
> Port Total-time Total-calls Transport
> 5060 35.02 s 209 UDP
>
> 10 new calls during 1.001 s period 2 ms scheduler resolution
> 164 calls Peak was 165 calls, after 30 s
> 0 Running, 164 Paused, 0 Woken up
> 1 open sockets
>
> Messages Retrans Timeout Unexpected-Msg
> ----------> INVITE 209 0 0
>
> <---------- 180 209 0
> <---------- 200 209 0 0
> ----------> ACK E-RTD1 209 0 0
>
> ----------> BYE 85 0 0
> <---------- 200 85 0
> [ 4000ms] Pause 85 0
> ------------------------------ Sipp Server Mode
> -------------------------------
>
>
>
>
> But as you can see on the SIPP-Client´s output, there are no succesfull
> calls. I logged everything and get the following failure types:
>
> sipp: The following events occured:
> 2007-08-15 16:38:16:570 1187188696.570840: Aborting call on unexpected
> message for Call-Id '[EMAIL PROTECTED]': while$
> Via: SIP/2.0/UDP
> 127.0.0.1:5060;branch=z9hG4bK-5633-96-0;received=89.110.157.78
> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=96
> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as52a75494
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> 2007-08-15 16:38:17:568 1187188697.568788: Aborting call on unexpected
> message for Call-Id '[EMAIL PROTECTED]': whil$
> Via: SIP/2.0/UDP
> 127.0.0.1:5060;branch=z9hG4bK-5633-106-0;received=89.110.157.78
> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=106
> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as6eb78372
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 02007-08-15 16:38:17:568 1187188697.568788: Aborting
> call on unexpected message for Call-Id '106-5633$
> Via: SIP/2.0/UDP
> 127.0.0.1:5060;branch=z9hG4bK-5633-106-0;received=89.110.157.78
> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=106
> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as6eb78372
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
>
>
>
>
> What is going wrong? What can I do to get correct sip calls running?
> It would be great if someone could help me.
>
> thanks
>
> Christian
>
>
>
>
>
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems? Stop.
> Now Search log events and configuration files using AJAX and a browser.
> Download your FREE copy of Splunk now >> http://get.splunk.com/
> _______________________________________________
> Sipp-users mailing list
> [email protected]
> https://lists.sourceforge.net/lists/listinfo/sipp-users
>
-------------------------------------------------------------------------
This SF.net email is sponsored by: Splunk Inc.
Still grepping through log files to find problems? Stop.
Now Search log events and configuration files using AJAX and a browser.
Download your FREE copy of Splunk now >> http://get.splunk.com/
_______________________________________________
Sipp-users mailing list
[email protected]
https://lists.sourceforge.net/lists/listinfo/sipp-users
