Hello All,
Please ignore my previous message. It appears that by updating my
Wireshark to the latest stable version (1.8.5), I am able to merge the
files just fine.
Thanks,
Seth
On 2/28/2013 5:49 PM, Seth Schultz wrote:
Hello,
I have a PCAP generated from a Homer SIPCapture server and a P
Unless you can get over the idea of a centralized repository, GIT is
an over curve, some would say counter intuitive. Please look into
subtrees, branches, tags.
Did not have time to look at everyone's post, but I am pretty sure
everyone will be enthused about the idea. I would say, if it's not
bro
> "BI" == Bogdan-Andrei Iancu writes:
BI> So, after all everything reduces to GH versus SF
It does not need to be an either-or. You can keep using SF for the
mailing lists (you should do that no matter what) and anything else
you like better.
You can keep copies of the git repo on both. M
Hello Everyone,
When managing incoming calls, or "!do_routing" function is correctly
denying the route. Reason being it's not possible for us to add all
the users that could call into the "dr_groups" table.
To compensate for this we added the following script hard coding the
"group_id", for reques
Setting modparam ice_candidate to low-priority will enable ICE for all
calls in media proxy with media proxy's own address as TURN service with
low-priority. However if you want to control priority on per call basis
then then you can define ice_candidate_avp and set priority of media proxy
relay IC
Hello saul:
I've now running the OpenSIPS server with the Mediaproxy.
Some natted UA are working and others no. Those that can't, thay can call but
no audio/video.
I would like to enable the ICE (cause the UA also support it) but i'm really
stuck on the wiki page you've recomended. Could you hel
Hi Jorge,
trying to terminate the call on 200 ok is not really a good idea - you
have to wait for the call to establish properly (have the ACK) and do
the terminate.
Anyhow, sending a CANCEL to caller is bogus - CANCEL is all the time
sent to callee sides only.
So, I suggest to wait for th
Hi, thank you for your feedback.
User A (caller) sends an Invite to User B (callee). When the Invite arrives to
opensips, b2b_init_request is called with 'topology hiding' and the Invite is
sent to User B.
Callee reply with a 200OK with sdp content, when this messages arrives to b2b,
if somethin
Maybe, on the first pass through opensips you could save the domain part
into a custom hdr (before sending to Asterisk) and configure Asterisk to
propagate this hdr. On the second pass through, opensips will do the
uac_replace_from() based on that hdr...
Just an idea :)
Regards,
Bogdan-Andre
On Mar 1, 2013, at 9:40 AM, Yuming Zheng wrote:
> Thanks, I have added Bob to Alice pres-rules also, just like:
>
>
>
>
> id="sip:101@172.22.198.1"/>
>
>
>
> allow
>
>
>
>
> Can this help to resolve this problem t
On Mar 1, 2013, at 4:40 AM, zhengyumingnanjing wrote:
> So,you mean OMA mode must be used with openxcap and opensips to receive the
> presence.winfo message.
> Is it possible when I just use the IETF mode?
> Or may I modify some code to implememt this feature due to I have only IETF
> implement
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