Thank you Marc.
On Thu, Feb 12, 2015 at 11:51 PM, Marc Soda wrote:
> Our config is based on the example config and the WebRTC bits are based on
> Carlos'.
> I've attached the relevant parts. It's pretty heavily customized to our
> specific environment. The main differences are the way that w
On 13/02/15 01:32 PM, Marc Soda wrote:
> How does Kamailio load balance traffic to rtpengine? Is it load based,
> round robin, etc? The module makes mention of this but I don't see how
> it works. Also, it talks about weighting the proxies. How is that
> accomplished?
This part of the module i
How does Kamailio load balance traffic to rtpengine? Is it load based,
round robin, etc? The module makes mention of this but I don't see how it
works. Also, it talks about weighting the proxies. How is that
accomplished?
Thanks,
Marc
___
SIP Express
Good day,
I’m experiencing some problems with our VoiP providers handling of REGISTER
requests. We are using a Gigaset PRO N720 as UAC behind a Juniper SSG 140 with
SIP-Alg enabled. This setup kind of works with UDP but our provider wants us to
use TCP. With TCP enforced incoming calls don’t wo
>Hi,
>
>If you are using rtpengine for the rtp you might have to upgrade to a
>newer version.
>
>On my system crypto negotiation in rtpengine started failing after a
>openssl update in January. Recompiling a newer version from git fixed that.
>
>regards
>
>M
Ok, now I found the problem: Updating w
>* Is the SIP signaling still going fine via websocket? Do you get callee
*>* ringing?
*>>* Cheers,
*>* Daniel*
Hello Daniel
Actually SIP signalling goes perfect, and i can see in RTP asterisk
log that RTP packets are being sent, so I suppose there is something
stuck somewhere in
On 2/13/15 11:36 AM, Juha Heinanen wrote:
i tried to use change_reply_status() to change 415 to 200, but got:
ERROR: textopsx [textopsx.c:283]: change_reply_status_f(): ERROR:
textops: the class of provisional or positive final replies cannot be
changed
textopsx/README does not mention anything
Hi,
If you are using rtpengine for the rtp you might have to upgrade to a
newer version.
On my system crypto negotiation in rtpengine started failing after a
openssl update in January. Recompiling a newer version from git fixed that.
regards
M
On 02/13/2015 09:56 AM, Rahul MathuR wrote:
i tried to use change_reply_status() to change 415 to 200, but got:
ERROR: textopsx [textopsx.c:283]: change_reply_status_f(): ERROR:
textops: the class of provisional or positive final replies cannot be
changed
textopsx/README does not mention anything about what can or cannot be
changed. why i
Hi Olle,
Thanks for getting back to me.
Kamailio seems to be receiving inbound sip udp correctly on the virtual IP's -
and is able to respond to that received traffic.
The issue seems more to be related to the mhomed mechanism where Kamailio tries
to identify the correct interface to use for f
On 13 Feb 2015, at 09:48, wrote:
> Hi,
>
> I am trying to install a Kamailio server in a HA configuration, with IP's for
> 3 attached networks. I am using Keepalived/VRRP to manage the VIP's. I have
> tested v 4.1.6 and latest 4.2.2
>
> I have 3 x listen entries, one for each network an
Hello Manuel,
To support the hypothesis of crypt libs screwing the logic, you can try a
'secure call' without using webrtc.
If them are to be blamed; your 'secure call' won't be successful.
Aside this, you can get a better idea of what has dwell-ed behind the
curtains by looking at syslogs.
Hop
Hi,
I am trying to install a Kamailio server in a HA configuration, with IP's for 3
attached networks. I am using Keepalived/VRRP to manage the VIP's. I have
tested v 4.1.6 and latest 4.2.2
I have 3 x listen entries, one for each network and mhomed=1. Also, for info,
we have net.ipv4.ip_non
On 12/02/15 17:04, Manuel Camargo Lominchar wrote:
> This might be a weird question
> I've been operating for some months with my kam + asterisk + webrtc
> sipml5 based system inside an Ubuntu Server
>
> Today I had the idea to apt-get upgrade my system and... now the whole
> system webrtc communi
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