With TLS it still is not working
Mar 27 11:39:16 [4421] INFO:core:probe_max_receive_buffer: using a UDP receive
buffer of 255 kb
Mar 27 11:39:16 [4425] WARNING:usrloc:dbrow2info: non-local socket
...ignoring
Mar 27 11:39:18 [4431] INFO:core:tls_accept: client did not present a
certificate
Mar
Yups,
it works fine with TCP.
All packets are forwarded to other party perfectly and everything work smooth.
only with TLS, on softphone it doesnt work.
any more suggestions
Regards,
Hemanshu Patel
Sr. Software Engg
SIS,Ahmedabad
Mo:09601295238
On Sat 27/03/10 9:50 AM , "Hemanshu Patel"
I havent tested over TCP, let me check it
but hardphone, i mean hardware based phones from grandstream gvx3140 works
fine with same implementation on TLS.
--
Regards,
Hemanshu Patel
M: 09601295238
> Does eyebeam with SIP over TCP is working?
>
> Am 26.03.2010 13:43, schrieb Hemanshu Patel:
>>
Hello,
I propose to merge the users mailing lists, most of the traffic these
days is about 3.0 and even there are 2 stables branches now, they are
sync'ed, so same code more or less. For 3.1 will be one stable branch,
falvor selection will be only a matter of make command.
Lately common usef
Hello,
On 3/26/10 4:13 PM, Zappasodi Daniele wrote:
Hello,
this is what I get with gdb:
(gdb) bt full
#0 0x40137e54 in sched_yield () from /usr/local/lib2/libc.so.6
No symbol table info available.
I don't think that it can help, but I am not able to load the symbol
table for openser on th
Hector,
Seems like rtpproxy is allocating an rtp session properly. However
rtpproxy's stats show that there are no RTP packets from the caller. I
would recommend using wireshark or tcpdump, and
(a) Verifying the connection address in the SDP for messages leaving the
proxy
(b) Make sure RTP pac
Done,
I had a lot of errors so I'll just show the final version that works OK.
=~ "192\.168\.([0-9]{1,3})\.([0-9]{1,3})
The only drawback is that I could pass as valid 192.168.999.999 but as these
IPs come from a DNS query, I assume they'll be fine.
Cheers,
Uriel
On Fri, Mar 26, 2010 at 11:29
Hello,
this is what I get with gdb:
(gdb) bt full
#0 0x40137e54 in sched_yield () from /usr/local/lib2/libc.so.6
No symbol table info available.
I don't think that it can help, but I am not able to load the symbol table for
openser on the server.
thanks,
Daniele
-Messaggio origi
Vikram
Thanks, that solved that problem but now I can only hear audio in one direction
and here is what I see in the log
Mar 26 16:11:43 openser rtpproxy[30827]: DBUG:handle_command: received command
"21119_2 U 1de249c9-f1fb...@192.168.0.3 195.176.213.123 16476
a489077dc86f9d6o0;1"
Mar 26 16:1
Hi all! Kamailio newbie here.
Well my setup is like this
Kamailio as load balancer and proxy -> Multiple clustered asterisk as
registrars
What I wanted to do was have all clients registered to asterisk so that most
of the pbx stuff will be done thru it and I'm using Asterisk Realtime as
well and
I'm still pretty stucked.
This is the goal.
Cisco trunk <-> Kamailio/RTPProxy <-> Asterisk.
The Cisco trunk is located on a private 10.x.x.x/29 range, the SIP-GW
and RTP stream are on public IPs, but only reachable via the private
10.x.x.x network. SER should only be the man in the middle an
Hector,
Just out of curiosity, could you please share your config file?
If I don't do the NAT detection and try using the function force_rtp_proxy and
cannot hear sound in any direction, the called gets disconnected and see the
following in the log
Mar 26 14:15:11 openser /usr/sbin/openser[
Hi Alex,
Actually what I'm trying to do is check the IPs on a request on a
Kamailio+RTPProxy acting as border of our network.
So I have the ingress IP and egress IP and need to check if I have to bridge
ii, ei, ie or ee.
I managed to obtain all IPs in AVPs, but now I have to check if they are
pub
172.16.0.0/12 does not line up on octet boundaries. You will need to
do something other than a regular expression. Fortunately, 'src_ip'
is a composite that supports comparisons against subnets in shorthand
CIDR notation.
It might also be that whatever you are trying to accomplish can be
Hi guys,
Does anyone have a REGEX syntax to match a private IP on the 192.168.x.x
range?
I'm trying with:
if($avp(s:ip_origen)=~"192.168(\.([1]?\d{1,2}|2[0-4]{1}\d{1}|25[0-5]{1})){2}"
)
But all IPs pass as private, even public ones.
Thanks!
Uriel
___
Just out of curiosity, could you please share your config file?
If I don't do the NAT detection and try using the function force_rtp_proxy and
cannot hear sound in any direction, the called gets disconnected and see the
following in the log
Mar 26 14:15:11 openser /usr/sbin/openser[19389]: ONR
Am 26.03.2010 12:15, schrieb hector.or...@swisscom.com:
Hi, I solved my issue. There was nothing wrong with my configuration.
RTPProxy wasn't being enforced because the SIP Phone wasn't being detected as
being behind NAT.
This is why I always remove NAT-detection as I want the rtpproxy in a
Does eyebeam with SIP over TCP is working?
Am 26.03.2010 13:43, schrieb Hemanshu Patel:
i am still having this problem.
when i use two grandstream phone everything works fine,
i can make calls on TLS and users can talk to each other.
But when i use two eyebream phone, it doesnt work, gives fol
i am still having this problem.
when i use two grandstream phone everything works fine,
i can make calls on TLS and users can talk to each other.
But when i use two eyebream phone, it doesnt work, gives following error
:33 [2875] WARNING:core:init_ssl_ctx_behavior: server verification NOT
activa
Ah, yes, that would do it.
On 03/26/2010 08:11 AM, dotnetdub wrote:
Hi Alex,
It was an issue with the domain table being empty. Populated the domain
table and all works as expected. I have an extra check in the script now
to make sure this doesn't happen again.
Apologies.
Stephen
On 26 Marc
Hi Alex,
It was an issue with the domain table being empty. Populated the domain
table and all works as expected. I have an extra check in the script now to
make sure this doesn't happen again.
Apologies.
Stephen
On 26 March 2010 00:23, Alex Balashov wrote:
> On 03/25/2010 08:16 PM, dotnetdub
Hi, I solved my issue. There was nothing wrong with my configuration.
RTPProxy wasn't being enforced because the SIP Phone wasn't being detected as
being behind NAT.
I disabled SIP ALG on my ADSL Modem and now RTPProxy is being enforced and I'm
able to do the recording. Now I have to find out
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