Hi Olle,
Just a quick update..
I've gone through this in detail and the issue is actually that
asterisk sends an UPDATE with CSeq: 104 UPDATE
and when FS respond OK asterisk then sends its REINVITE with CSeq: 103 INVITE
As far as I can tell Freeswitch at this point is perfectly within its
righ
On 02/04/2014 01:44 AM, Alex Balashov wrote:
In fact, the only module that I am aware of that deals with
manipulation of encapsulated bodies at all (besides the far-end NAT
traversal and rtpproxy modules that modify RTP endpoints and other
SDP attributes) is the SIP-T module[1].
Well, I mean,
On 02/04/2014 01:35 AM, Ramya Ramamurthy wrote:
I am using kamailio for decoding of SIP Packets with Message
Body as a GSM PDU. But I am not sure of the modules which i need to use
for the same. Some help on this would be greatly appreciated.
Kamailio doesn't care much about the encaps
Dear SR-Users,
I am using kamailio for decoding of SIP Packets with Message Body as
a GSM PDU. But I am not sure of the modules which i need to use for the
same. Some help on this would be greatly appreciated.
Thanks,
~Ramya.
___
SIP Express Rout
Hi,
Can someone clarify here please?
Regards,
Shankar
From: Shankar [mailto:shankar...@plintron.com]
Sent: Monday, February 03, 2014 10:53 AM
To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List'
Subject: Regd. MAX_LDG_LOCKS
Hello,
Can anyone
Dear All,
Anybody has any clues about this issue?
I really want somebody to help me.
Regards,
Nandini
On Sat, Feb 1, 2014 at 3:42 PM, Nandini madhu wrote:
> Dear All,
>
> Anybody please help me in resolving this issue.
>
> Awaiting somebody's suggestions.
>
> Regards,
> Nandini
>
>
> On Wed,
Dear Kamailio'ns
I am working on Kamailio (V 4.0)+ Mediaproxy server (2.5.2) running on
Ubuntu 12.04. I am experiencing Latency, jitter, pixel led, choppy video
errors in my set-up. So i wanted increase buffer sizes in my kamailio
server , Mediaproxy server and Ubuntu also.
In that regards i have
Charles hi,
That's what I was trying to get at.
I have now set db_mode to 1 however I only get calls in state 4 (active) go
into the db not calls in state 2 (ringing).
Case is to collect calls in to a specific group of destinations (pick-up
group) into a profile and on receiving a call
I'm having trouble divining the proper way to authenticate calls to
remote destinations that require it. I'm hoping someone can clear it up.
Freeswitch client server A is authenticated to Kamailio.
Freeswitch test server B is authenticated to Kamailio.
Calls from A -> Kamailio -> B fail with rej
Hi John,
For the current call, you can access dialog attributes via the "$dlg(...)"
pseudo-variable. But I don't believe there is a way to access other dialogs
from within the config script without querying the database (and setting
db_mode to 1).
If you can explain your use case slightly, someon
I think you need to disable Authentication in Asterisk. Kamailio is doing
the authentication and then letting Asterisk know about the registration in
a new message. It doesn't know how to handle 401 challenge from Asterisk.
Dipak
On Mon, Feb 3, 2014 at 6:14 AM, *sm1Ly wrote:
> hello, I got the
Hi!
I was just testing the digest replay possibilities against Kamailio. (findings:
http://www.kamailio.org/wiki/tutorials/security/kamailio-security#digest_authentication)
It looks that by default (the typical default configs), a SIP replay attack can
be done during 300 seconds (?) .
Now I t
Am Montag, 3. Februar 2014, 12:42:50 schrieb Oliver Roth:
> Nobody an idea?
>
> To measure asr/ner ratio this is really important to us!
Hi Oliver,
do you need to get CDR information about a call or some special information
from inside the carrier route module? If you need the former, then you
Hi all
Nobody an idea?
To measure asr/ner ratio this is really important to us!
Regards,
Oli
Von: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Oliver Roth
Gesendet: Montag, 13. Januar 2014 09:26
An: Kamailio (SER) - Users Mailing List
Bet
hello, I got the same issue. my sip device getting ok from kamailio, but
asterisk reply 401.
but.
this is my config: http://pastebin.com/jGCak01E
and I changed regfwd route to
$du = "sip:50.0.0.10:5060;transport=udp";
forward();
and now I see this in logs:
Feb 3 15:13:38 kamaz /
Great! Thanks!
On Feb 2, 2014 8:56 AM, "Richard Fuchs" wrote:
> The master branch on github already fixes that.
>
> cheers
>
>
> On 02/01/14 19:40, Kelvin Chua wrote:
> > If you were to ask me, i would prefer that mp-ng would take the 2nd arg
> > to rtpproxy_manage and replace the sdp.
> >
> > Th
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