Hi,
And if you don't use background mode, does this scenario work correctly?
It could be problem with your script used in new SIPp version, not in bg
mode...
Maybe you should try to use some -trace_* argument, it seems that sipp
was not able to start (or after starting, it must be terminated
It is not that sipp doesn't run in background mode. sipp would have exited
because of some failure like ipaddress:port wrong/already in use. to know
what would have happened, run the same senarion without background mode
(without -bg option), you would see what the error is
Thanks
Dhana
On
I usually see these type of errors if call-id's differs.
Dhana
On Mon, Nov 24, 2008 at 8:14 AM, Venkat Narasimhan
[EMAIL PROTECTED]wrote:
Hi,
I am having the following issue ... i know it looks wierd(bold text) but i
dont know why ...
any help appreciated. thanks in advance
2008-11-24
Is uas.xml located in the same directory where your ./sipp is located?
If it is in same location, then some xml editing error.
run this uas.xml through InternetExplorer, any syntax errors will be shown.
Dhana
On Fri, Nov 21, 2008 at 2:41 AM, ~肖士锋!~ [EMAIL PROTECTED] wrote:
hi,
i'm a
1. when you receive re-invite does it's call-id match to you curren't call's
call-id?
2. after receiving re-invite you are jumping to id 3 using next. after your
label id=3 do you have scenario to expect 200 for the INFO which you sent
earlier.
3. what if 200 for your INFO comes and then re-invite
I think -aa option will auto answer INFO and NOTIFY only, NOT for UPDATE
Dhana
On Fri, Nov 21, 2008 at 2:04 PM, Katwala, Kalpesh N (Kalpesh)** CTR **
[EMAIL PROTECTED] wrote:
All,
I tried using the -aa option on the command line to send out
an automatic 200OK for UPDATE/NOTIFY
what do you mean by inside server?
Dhana
On Wed, Nov 19, 2008 at 11:16 AM, Kelle [EMAIL PROTECTED] wrote:
Hi
I would like to run SIPp inside the server to check server performance
but I can't do that. The packets don't flow!
Anyone can help me?
Thanks
Raquel
I didn't understand you query. can you send me scenario file.
Dhana
On Mon, Nov 17, 2008 at 3:54 AM, Sumeet Bhardwaj
[EMAIL PROTECTED] wrote:
Hi all,
I am using SIPp3.1.
When I use any variable inside the send tag of invite to store the
[remote_ip] value then it shows error message -
I see you are sending Route: header (with the value of Service-Route you
received in 401) While you are sending the REGISTER after 401 challenge, DO
NOT any Route header in REGISTER. This one observation
Other observation is uri param in Authorization header has ipaddress
and port. It is supposed
You can create an out-of-call scenario that will create a new call for all
unexpected messages using the -oocsf option.
Charles
Dhananjaya Reddy Eadala [EMAIL PROTECTED]
11/26/2008 11:04 PM
To
[EMAIL PROTECTED]
cc
sipp-users@lists.sourceforge.net
Subject
Re: [Sipp-users] How to process
Or you can use SIP INFO.
On Wed, Nov 26, 2008 at 7:39 PM, Dhananjaya Reddy Eadala
[EMAIL PROTECTED]wrote:
you can't detect dtmf in signaling path. DTMF goes in media (RTP)
Dhana
On Fri, Nov 14, 2008 at 8:57 AM, vijay kant gupta
[EMAIL PROTECTED] wrote:
HI,
Anybody know how sip
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