Hi Everyone,
Which is the best tool to test OpenSIPS especially as Load Balancer as I want
100% results from that tool.
With Best Regards,
Faisal Rehman___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listin
Hi,
On Sep 22, 2011, at 2:10 AM, k1028 wrote:
> There got to be something I am missing for 1.7. on top of this error i have
> no audio both way.
>
> IP Phone -> Opensips -> PSTN have no problem
>
> PSTN -> OpenSIPS -> IP Phone with no audio. The funny part is Linksys IP
> Phone, Linksys PAP2, P
Hi Faisal,
You can try SIPP.
http://sipp.sourceforge.net/
Regards
Khue Nguyen
From: Faisal Rehman
To: "users@lists.opensips.org"
Sent: Thursday, September 22, 2011 2:26 PM
Subject: [OpenSIPS-Users] OpenSIPS Load Balancing Test Tool
Hi Everyone,
Which is t
Hi Khue,
Thanks a lot, you are always helpful.
With Best Regards,
Faisal Rehman
From: nguyen khue
To: Faisal Rehman ; "users@lists.opensips.org"
Sent: Thursday, September 22, 2011 12:35 PM
Subject: Re: [OpenSIPS-Users] OpenSIPS Load Balancing Test Tool
H
Hi Alex,
According to your scenario, there is a late negotiation (SDP is
advertised in 200OK and ACK). I see there is 'rtpproxy_offer' called for
a 200OK, but I can't see any 'rtpproxy_answer' for an ACK message.
Perhaps that's why the SDP in the ACK remains the same and I don't think
it is a
Hi Răzvan Crainea ,
yes, you're right. It is totally wrong logic with wrong config. I wonder why
it was working even in one direction :) Anyway I fix that with
unforce_rtp_proxy() in the INVITEs from REGISTRAR thus releasing all
previous bindings, and made rtpproxy_offer("f") for 200OK from the ph
Hello,
Are you getting other errors just before getting the mediaproxy error ?
If yes, please paste them here.
If not, could you please put your server in debug=6 and attach the log
for such a call where mediaproxy fails to create the dialog ?
Regards,
Vlad Paiu
OpenSIPS Developer
On 09/21/
Hi,
When using topology_hiding in OpenSIPS 1.7.0 (64bit), opensips segfaults.
Is this a known problem or am i doing somthing wrong?
You'll find my test opensips.conf, as well as the debug output including a "bt
full" from gdb here:
http://keenserver.nl/files/opensips.conf
http://keenserver.nl/fi
Hi everyone,
Is it normal for OpenSIPs has a lot of replies for just one call? My OpenSIPs
is giving me a bunch of replies messages for just one call.
Sep 22 11:16:36 oceano /sbin/opensips[24366]: incoming reply
Sep 22 11:16:36 oceano /sbin/opensips[24366]: Call from:
sip:1340101...@oceano.vip
Hi,
It's not normal. It's look like you are caught in a loop: the request is
forwarded back to Opensips till the CSeq hits 0.
Increase the log level and check if it is the same request with decreasing CSeq
each time.
Regards,
- vma
.
On Sep 22, 2011, at 4:22 PM, Rodrigo Ferreira wrote:
> Hi e
I don't see any error before. Will do the debug level 6 when i get to work in
few hour.
--
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Sent from the OpenSIPS - Users mailing
Looks like some kinda of loop, because I was checking the SIP transaction
with "ngrep" and I can see a lot of invites for the same call.
I will increase my debug level, and see what I can find it
--
From: "Vallimamod ABDULLAH"
Sent: Thursday, Sep
Hello,
Forgot to mention this in the previous email, but the SIP trace would
also help a lot, especially the initial INVITE for that particular dialog.
Regards,
Vlad Paiu
OpenSIPS Developer
On 09/22/2011 05:32 PM, k1028 wrote:
I don't see any error before. Will do the debug level 6 when i
I didn't find anything on my opensips.log.
The last change that I made, was update my opensips to 1.6.4-2, but I dont
know if theres a connection.
My opensips.cfg is attached, if anyone can take a look and help me out, I
would appreciate.
Thanks.
--
Do you have any suggestion on what is the easier way to do it or you want me
to attach everything from the debug log? I can't reproduce this problem on
my testing environment and it only happen on production. The debug log and
siptrace fill up very quickly.
--
View this message in context:
http:
Hi Andrew,
Thanks for the advice. I should have found that function!
Using set_advertised_address() has fixed it. All is working ok now.
I also used alias= to set both internal and external addresses.
It already added Record-Route header, but I added some logic to choose
whether rr should be intern
Sorry everyone. I been working on OpenSER and OpenSIPs for 5 years. This is
the first time i experienced so many problem upgrading.
1. Receiving ERROR:mediaproxy:__tm_request_in: could not create new dialog
on Production only not testing environment.
2. No Audio only on inbound to IP Phone
3. N
I uploaded lvl 6 debug and SIP trace. I am not sure what is going on and am
very confused too. There is no audio and call drop only on inbound call to
IP Phone from OpenSIPs. What driving me crazy is that it happen on ATA, IP
Phone, and other Dialer but it doesn't happen to Zoiper Dialer (I tried m
Hi Everyone,
I want to do load balancing of two asterisk servers which are dealing with PSTN
calls only, the script that I found on OpenSIPS website is a little complex and
not my requirement. Let's say I have two asterisk boxes to be load balanced &
whose IP addresses with allocated resources
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