Hi,
Out of curiosity, I tried this (-trace_calldebug) flag and sipp 
complaint that no such flag available.

# sipp -v
  SIPp v3.1-TLS-PCAP, version svn9910, built May  2 2008, 15:31:41.

Am I using the latest version?
Thanks and Regards,
Manish



ZiLi0n wrote:
> Mr. Charles, I have seen "hash 0" in calldebug when  SIPp sends the message 
> "BYE" :
> 
> 2008-10-10    08:12:14:785    1223619134.785339 Set Retransmission Hash: 0 
> (recv index 3, send index 9)
> 2008-10-10    08:12:14:785    1223619134.785348 Processing message 10 of type 
> 3 for call [EMAIL PROTECTED] at 183076.
> 2008-10-10    08:12:15:295    1223619135.295335 Processing message 10 of type 
> 3 for call [EMAIL PROTECTED] at 183586.
> 2008-10-10    08:12:15:295    1223619135.295344 Retransmisison required (1 
> retransmissions, max 9)
> 2008-10-10    08:12:15:295    1223619135.295362 Sending UDP message for call 
> [EMAIL PROTECTED] (index 9, hash 502009980):
> 
> Thanks a lot!
> 
>> To: [EMAIL PROTECTED]
>> From: [EMAIL PROTECTED]
>> Date: Thu, 9 Oct 2008 11:42:54 -0400
>> CC: [email protected]
>> Subject: Re: [Sipp-users] Why doesn´t SIPp re-send the ACK when it receive a 
>> lot of messages 200 OK?
>>
>> Ths is a -trace_msg trace not a -trace_calldebug trace.
>>
>> Charles
>>
>>
>>
>>
>> ZiLi0n <[EMAIL PROTECTED]> 
>> 10/09/2008 11:27 AM
>>
>> To
>> <[email protected]>
>> cc
>>
>> Subject
>> [Sipp-users]  Re:  Why doesn´t SIPp re-send the ACK when it receive a lot 
>> of messages 200 OK?
>>
>>
>>
>>
>>
>>
>> Mr Charles. Thank. Here is the traces
>>
>> In this traces, SIPp don´t resends the ACK even when Asterisk resends the 
>> message '200 OK':
>>
>> ----------------------------------------------- 2008-10-08 
>> 13:11:08:995.378
>> UDP message received [758] bytes :
>>
>> SIP/2.0 200 OK^M
>> Via: SIP/2.0/UDP 
>> 192.168.1.179:5060;branch=z9hG4bK-2882-230-0;received=192.168.1.179^M
>> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=2882SIPpTag00230^M
>> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as7fc476b0^M
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 1 INVITE^M
>> User-Agent: Asterisk PBX 1.6.0^M
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M
>> Supported: replaces, timer^M
>> Contact: <sip:[EMAIL PROTECTED]>^M
>> Content-Type: application/sdp^M
>> Content-Length: 261^M
>> ^M
>> v=0^M
>> o=root 377264768 377264768 IN IP4 192.168.1.246^M
>> s=Asterisk PBX 1.6.0^M
>> c=IN IP4 192.168.1.246^M
>> t=0 0^M
>> m=audio 18590 RTP/AVP 8 101^M
>> a=rtpmap:8 PCMA/8000^M
>> a=rtpmap:101 telephone-event/8000^M
>> a=fmtp:101 0-16^M
>> a=silenceSupp:off - - - -^M
>> a=ptime:20^M
>> a=sendrecv^M
>>
>> ----------------------------------------------- 2008-10-08 
>> 13:11:08:995.480
>> UDP message sent (383 bytes):
>>
>> ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0^M
>> Via: SIP/2.0/UDP 192.168.1.179:5060;branch=z9hG4bK-2882-230-4^M
>> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=2882SIPpTag00230^M
>> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as7fc476b0^M
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 1 ACK^M
>> Contact: sip:[EMAIL PROTECTED]:5060^M
>> Max-Forwards: 70^M
>> Subject: Performance Test^M
>> Content-Length: 0^M
>> ^M
>> ----------------------------------------------- 2008-10-08 
>> 13:11:09:993.684
>> UDP message received [758] bytes :
>>
>> SIP/2.0 200 OK^M
>> Via: SIP/2.0/UDP 
>> 192.168.1.179:5060;branch=z9hG4bK-2882-230-0;received=192.168.1.179^M
>> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=2882SIPpTag00230^M
>> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as7fc476b0^M
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 1 INVITE^M
>> User-Agent: Asterisk PBX 1.6.0^M
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M
>> Supported: replaces, timer^M
>> Contact: <sip:[EMAIL PROTECTED]>^M
>> Content-Type: application/sdp^M
>> Content-Length: 261^M
>> ^M
>> v=0^M
>> o=root 377264768 377264768 IN IP4 192.168.1.246^M
>> s=Asterisk PBX 1.6.0^M
>> c=IN IP4 192.168.1.246^M
>> t=0 0^M
>> m=audio 18590 RTP/AVP 8 101^M
>> a=rtpmap:8 PCMA/8000^M
>> a=rtpmap:101 telephone-event/8000^M
>> a=fmtp:101 0-16^M
>> a=silenceSupp:off - - - -^M
>> a=ptime:20^M
>> a=sendrecv^M
>>
>> ----------------------------------------------- 2008-10-08 
>> 13:11:10:994.064
>> UDP message received [758] bytes :
>>
>> SIP/2.0 200 OK^M
>> Via: SIP/2.0/UDP 
>> 192.168.1.179:5060;branch=z9hG4bK-2882-230-0;received=192.168.1.179^M
>> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=2882SIPpTag00230^M
>> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as7fc476b0^M
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 1 INVITE^M
>> User-Agent: Asterisk PBX 1.6.0^M
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M
>> Supported: replaces, timer^M
>> Contact: <sip:[EMAIL PROTECTED]>^M
>> Content-Type: application/sdp^M
>> Content-Length: 261^M
>> ^M
>> v=0^M
>> o=root 377264768 377264768 IN IP4 192.168.1.246^M
>> s=Asterisk PBX 1.6.0^M
>> c=IN IP4 192.168.1.246^M
>> t=0 0^M
>> m=audio 18590 RTP/AVP 8 101^M
>> a=rtpmap:8 PCMA/8000^M
>> a=rtpmap:101 telephone-event/8000^M
>>
>>
>> Other problem is SIPp retrans the message BYE even when Asterisk has sent 
>> the '200 OK':
>>
>> U +0.026130 192.168.1.136:5060 -> 192.168.1.246:5060
>> BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK-12878-227-9
>> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=12878SIPpTag00227
>> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as7e62485f
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 2 BYE
>> Contact: sip:[EMAIL PROTECTED]:5060
>> Max-Forwards: 70
>> Subject: Performance Test
>> Content-Length: 0
>>
>>
>> #
>> U +0.000146 192.168.1.246:5060 -> 192.168.1.136:5060
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP 
>> 192.168.1.136:5060;branch=z9hG4bK-12878-227-9;received=192.168.1.136
>> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=12878SIPpTag00227
>> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as7e62485f
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 2 BYE
>> User-Agent: Asterisk PBX 1.6.0
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Supported: replaces, timer
>> Contact: <sip:[EMAIL PROTECTED]>
>> Content-Length: 0
>>
>> U +0.152849 192.168.1.136:5060 -> 192.168.1.246:5060
>> BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK-12878-227-9
>> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=12878SIPpTag00227
>> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as7e62485f
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 2 BYE
>> Contact: sip:[EMAIL PROTECTED]:5060
>> Max-Forwards: 70
>> Subject: Performance Test
>> Content-Length: 0
>>
>>
>> #
>> U +0.000125 192.168.1.246:5060 -> 192.168.1.136:5060
>> SIP/2.0 481 Call leg/transaction does not exist
>> Via: SIP/2.0/UDP 
>> 192.168.1.136:5060;branch=z9hG4bK-12878-227-9;received=192.168.1.136
>> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=12878SIPpTag00227
>> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as7e62485f
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 2 BYE
>> User-Agent: Asterisk PBX 1.6.0
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Supported: replaces, timer
>> Content-Length: 0
>>
>> I think that SIPp CLIENT hasn´t got time for receive the response from 
>> Asterisk. this errors only occurrs when the load is heavy.
>>
>> Some solution?
>>
>> PD: For this test I use sipp.2008-09-25.tar
>>
>> Thanks.
>>
>>> To: [EMAIL PROTECTED]
>>> From: [EMAIL PROTECTED]
>>> Date: Tue, 7 Oct 2008 19:36:24 -0400
>>> CC: [email protected]
>>> Subject: Re: [Sipp-users] Why doesn´t SIPp re-send the ACK when it 
>> receive a lot of messages 200 OK?
>>> You should include the output of -trace_calldebug in your message.
>>>
>>> Charles
>>>
>>> ZiLi0n <[EMAIL PROTECTED]> wrote on 10/07/2008 04:33:10 PM:
>>>
>>>>> SIPp should retransmit the ACK if the retransmitted 200 is 
>> identical. 
>>> You 
>>>>> should enable -trace_msg and -trace_calldebug and see what the logs 
>>> tell 
>>>>> you.
>>>> The retrasnmitted meesages 200 OK are identicals.
>>>>
>>>> I´m using SIPp with Asterisk.Asterisk show this errors:
>>>> Sep 12 18:52:36 WARNING [4620]: chan_sip.c:1835 retrans_pkt: Maximum
>>>> retries exceeded on transmission 778f89593967725f0abe40eb1752504c (at)
>>>> 10.10.206.53 for seqno 1620 (Critical Response)
>>>>
>>>> Sep 12 18:52:36 WARNING [4620]: chan_sip.c:1835 retrans_pkt: Hanging
>>>> up call 778f89593967725f0abe40eb1752504c (at) 10.10.206.53 no reply to
>>>> our critical packet.
>>>>
>>>> A solution? Thanks
>>>>
>>>>> Charles
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ZiLi0n <[EMAIL PROTECTED]> 
>>>>> 10/05/2008 02:26 PM
>>>>>
>>>>> To
>>>>> <[email protected]>
>>>>> cc
>>>>>
>>>>> Subject
>>>>> [Sipp-users] Why doesn´t SIPp re-send the ACK when it receive a lot 
>> of 
>>>>> messages 200 OK?
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> I am testing my Asterisk Server. This is my configuration:
>>>>>
>>>>> SIPpCLIENT --- Asterisk Server --- SIPpSERVER
>>>>>
>>>>> If the simultaneous calls is less than 300 the test is OK, but when 
>>> the 
>>>>> simultaneous calls is approximately 350 calls, the calls hang up:
>>>>>
>>>>> Asterisk sends to SIPpCLIENT the message '200 OK'. SIPpCLIENT sends 
>> to 
>>>>> Asterisk the message 'ACK'. 
>>>>> Asterisk re-sends to SIPpCLIENT the message '200 OK'.
>>>>> SIPpCLIENT doesn´t re-send the message 'ACK' to Asterisk.
>>>>>
>>>>> I think that message 'ACK' is lost... but the cpu load Asterisk 
>> Server 
>>> is 
>>>>> approximately 50% and the net status is perfect.
>>>>>
>>>>> Why does not SIPpCLIENT re-send the ACK to Asterisk? Asterisk needs 
>>> the 
>>>>> ACK for complete the call
>>>>>
>>>>> Thank´s
>>>>>
>>>>> La cartera, las gafas. ¿te falta algo? Ahora llévate Messenger en tu 
>>> móvil
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>>>>> _______________________________________________
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>>>>>
>>>>>
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