Hi,
I wouldn't say it a bug then, since its ok from specs poiint of view. The
real issue was that we had Asterisk realtime previously which had the same
extension set enabled in DB.
So the users with same extensions registering on Kamailio making calls via
asterisk had trouble because asterisk st
Hello,
On 12/2/11 5:24 AM, Sammy Govind wrote:
Hello again,
You were right, as soon as I made changes in asterisk SIP profile for
the Kamailio proxy server and stopped the 401 Auth from Asterisk to
Kamailio the CANCELS started to work fine.
well, the 401 from asterisk is ok from specs point o
Hello again,
You were right, as soon as I made changes in asterisk SIP profile for the
Kamailio proxy server and stopped the 401 Auth from Asterisk to Kamailio
the CANCELS started to work fine.
So the SIP flow now is:
- invite from phone to kamailio
- kamailio asks for authentication - 407
- ack
Hey Daniel,
I've exactly followed your point, I'll try some stuff on asterisk server to
stop asking for 401 Auth to Kamailio., maybe this will eliminate the need
for another INVITE with authentication params.
But one thing which just makes me curious is that a soft phone directly
coming from a Pu
Hello,
is the SIP trace complete?
What I could find inside is:
- invite from phone to kamailio
- kamailio asks for authentication - 407
- ack
- invite with credentials, kamailio forwards to asterisk
- asterisk asks for authentication - 401
- ack
- there is no new INVITE with credentials for kama
Hello,
I will look over it soon - since you sent pcap I couldn't look at it
directly from the email. ngrep outputs plain text which is easy to read
from email, the reason I am asking mainly for ngrep traces since many
times I am not around a computer where is convenient to open pcap file.
On
Hello again,
Please see the attached wireshark trace, I tried for a sipgrep trace but
couldn't somehow. I hope this will get me some clue on what I'm doing wrong.
This is a setup with Kamailio in front of Asterisk Servers. Kamailio is
multihomed and MS are on private IPs, all the calls are routed
Thanks for your reply I will attach the wireshark traces as soon as I get
to my workstation.
BR,
Sammy.
On Mon, Nov 28, 2011 at 3:33 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> send the ngrep trace of such call, from the initial INVITE, you can use:
>
> ngrep -d any -qt -W byline port 506
Hello,
send the ngrep trace of such call, from the initial INVITE, you can use:
ngrep -d any -qt -W byline port 5060
The sip trace will help to see what is wrong with that CANCEL.
Cheers,
Daniel
On 11/28/11 7:19 AM, Sammy Govind wrote:
Anyone please help.
On Sat, Nov 26, 2011 at 10:39 PM, S
Anyone please help.
On Sat, Nov 26, 2011 at 10:39 PM, Sammy Govind wrote:
> Hello list,
>
> I'm using Kamailio 3.1.5 in front of asterisk servers. Kamailio handles
> all the SIP registrations. Calls from SIP phones are forwarded to asterisks
> and then dialled out to Kamailio.
>
> root@SBCserver
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