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
