we noticed that if sr 3.1 config does not contain
syn_branch=0
then acks to 200 oks that are t_relayed by sr, contain branch=0
param in topmost via.
in kamailio 1.5, via branch param has proper value even when tm param
syn_branch is not set.
a couple of questions:
1) according to core
Hello,
On 3/7/11 12:51 AM, Andy Lippitt wrote:
Hello all,
I've read as many of the asterisk balancing threads as I can find.
Either my situation is unusual or I simply haven't understood anything
I've read.
In short, I'm building an web/phone mashup which uses Asterisk's AGI
to get its
On 3/7/11 9:50 AM, Juha Heinanen wrote:
we noticed that if sr 3.1 config does not contain
syn_branch=0
then acks to 200 oks that are t_relayed by sr, contain branch=0
param in topmost via.
in kamailio 1.5, via branch param has proper value even when tm param
syn_branch is not set.
What is
On Sunday 06 March 2011, Daniel-Constantin Mierla wrote:
it took me quite a while due to traveling, but now the issue should be
fixed on git. Indeed there was an issue with the indexes when accessing
the xavp as PV.
Thanks for the fix, they work fine now.
--
Met vriendelijke groet,
Alex
Jiri Kuthan writes:
What is proper value?
jiri,
according to rfc3261 section 20.42:
For implementations compliant to this specification, the value of the branch
parameter MUST start with the magic cookie “z9hG4bK”, as discussed in
Section 8.1.1.
it is sad if sr is not BY DEFAULT rfc3261
On 3/7/11 10:40 AM, Juha Heinanen wrote:
see above. we have seen rfc3261 sip user agents that don't understand
branch=0 value
That's probably good enough argument, as some newer implementations may
give up on RFC2543 backwards compatibility and complaing about lack of
branch.
and matching
Hello Joe,
Thanks for your response. Setting $dlg_ctx(timeout_bye) = 1 did the trick.
One more question: after the session is ended, the following line is
printed:
WARNING: dialog [dlg_req_within.c:173]: inconsitent dlg timer data on dlg
0xb368c954 [1975:901494976] with clid
Hi everybody.
I am using Kamailio 3.1 and RTP proxy for internet calls. I'm using external
test accounts to check if the calls are established and the media flow is ok.
When I use a sip2sip.info or VoIP Talk accounts, then all is working fine
between my internal and these external
Greetings-
I'm attempting an installation of Siremis on a CentOS 5.5 64-bit system running
Kamailio 3.1.0. Following the instructions on the wiki [1], I find that I
cannot pass step 3 for the database configuration.
I've triple checked all of my details are correct and verified the permissions
Hi,
sorry for the delay... it was a busy week.
I have just commited a change to the current master branch so that you
may call ds_mark_dst(a) in a reply route. Setting the gateway to
active will now also reset the failure counter. I have adapted the
docs accordingly.
Please feel free to test...
Hi,
i have just changed the functionality in the master branch: If we get
a positive reply from the OPTIONS-Request and the destination has been
deactivated, it should now no longer become active again, only the
error-counter will be set to zero.
I will do some tests in the next few days...
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