Hi,
sipp version 3.1 has a bug in call.cpp in function get_remote_port_media().
The function improperly calculates the start position of the port.
Also gcc-4.3.2 doesn't compile call.cpp and scenario.cpp since these
files use INT_MAX and dont include limits.h.
I am providing two patch files
You could try including
[authentication username=your_username password=your_password]
in your second REGISTER.
Have a look at the enclosed file. This scenario does not register, but
sets up a call through a proxy that requires digest authorization.
Br, Dmitry
Nikolay Kondratyev wrote:
Hi
a standard repository…
Is there an error in my scenario below?
Could somebody give me a working example of registration scenario or
try my scenario?
Thanks and regards,
Nikolay.
*From:* Dmitry Goncharov [mailto:dgoncha
Hi,
Apparently, your scenario doesn't expect ACK when it comes. You should,
probably, follow the sip exchange and figure out if ACK is correct. If
it is correct then fix your scenario.
HTH, Dmitry
John Barry wrote:
Hello,
I want to measure the performance of a OpenSER (1.3.2)proxy doing
I can guess that your sip server gets overloaded and doesn't answer as
quickly as your scenario expects. However, this is just a guess. As is
said in my previous post, you should sniff the sip exchange and figure
out if ACK is correct according to SIP. If it is correct then, you
should read
歐德旺 wrote:
Dear all
i had dowload SIPp-3.1 and 3.0 version ,run make or make ossl have
follow error
scenario.cpp:900: error: 「INT_MAX」 (在此作用欄位中尚未宣告 )was not
declared in this scope
my os ubuntu 8.10
dickson
E.H.Eefting wrote:
After a lot of debugging with asterisk and sipp in pcap mode we found the
following problem:
If asterisk is configured to allow rtp ports greater than , it seems the
last digit is trimmed away some how.
An example of a tcpdump followed by an immediate netstat shows
catalina oancea wrote:
Hello Mike
Thanks for your answer. I tried with '-t u1' and I get the same error.
I also tried to run -mp 1024 and I added net.ipv4.ip_local_port_range
= 1024 65000 to /etc/sysctl.conf, and then sysctl -p /etc/sysctl.conf.
In my xml I use [auto_media_port], which now
catalina oancea wrote:
Hi,
Thanks for the idea.
What I see using strace is that normal sendto looks like this:
sendto(14, \...@$\0\264z\200\0\31\345\0'\273T\21S3\17vvswwvwsvy}\376...,
180, MSG_DONTWAIT, {sa_family=AF_INET, sin_port=htons(9764),
sin_addr=inet_addr(192.168.12.12)}, 16) = 180
Andrew Wood wrote:
Hi
Im trying to use SIP-P to test a SIP proxy, but the proxy keeps
rejecting the 180 Ive created in SIP-P saying:
Missing mandatory header fields (To, From, CSeq, Call-Id or Via)
The 180 is defined as follows:
send
![CDATA[
SIP/2.0 180 Ringing
Huve, Frederic wrote:
Folks,
I'd like to know whether it is possible to use sipp to generate a SIP call to a
SIP UAS, and set the media address (mi option) to an non-local IP address ?
The aim is to oblige the UAS to send the RTP stream(s) only to a non local
end-point (not deployed on the
Tincho ylm wrote:
Hi all!
My SIPp only allow 12 simultaneous calls. If a use -l 10 everything
work perfect!
If I put -l 25, I get this error at 13th call:
Aborting call on unexpected message for Call-Id '92-4...@ip-uac':
while expecting '100' (index 1), received 'SIP/2.0 500 Server internal
mwilliam prusty wrote:
Hi All
I am using SIPP unstable version.After the call is established
(Exchane of INVITE,180 ringing, 200 ok,ACK between the client server
) When SIPP client is sending Bye(cseq=2), It is getting 407 from the
CSCF. After that from SIPP client side i am sending the
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