Hello
I have installed OpenSips 1.6.4 on CentOS 5.5 32bit with load balancing
mysql support ; I want to balance calls to 2 asterisk servers . I am sending
traffic to opensips from 1 x gnudialer 1 x vicidial ( so from predictive
dialers ). Situation is like this:
http://opensips-open-sip-server.1449251.n2.nabble.com/file/n6081772/Screenshot.png
1. How can I passed the voice from the UA to another by configuration the
opensips.cfg and using the rtpproxy command line ?
--
View this message in context:
2-3 times per day my opensips configuration with 300 channels and a load
balancer fails ; It's opensips 1.6.4 on centos 5.5. 32 bit
The erors I get are :
Any ideas?
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1718]: ERROR:tm:new_t:
out of mem
Mar 11 11:15:08 opensipsh
Hello
I had the same problem on 1.6.4, you should use 1.6.4-2 version
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Iulian Macare
Sent: Friday, March 11, 2011 12:24 PM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] opensips 1.6.4 out
That's the version I am using. 1.6.4-2 ; I tried to double the SHM parameter
and recompile but the same problem happened.
opensips-1.6.4-2-notls_src.tar.gz
On Fri, Mar 11, 2011 at 11:37 AM, Denis Putyato denis7...@mail.ru wrote:
Hello
I had the same problem on 1.6.4, you should use 1.6.4-2
Hi Iulian,
I suggest to update the dialog module from svn, branch 1.6. There was a
memory leak discovered after the 1.6.2 release and it was fixed in
January. It might also be showing up in your configuration.
Regards,
Anca Vamanu
OpenSIPS Developer
On 03/11/2011 11:39 AM, Iulian
Hello Iulian,
I took a look at the error output, and something seems very wrong. The
errors
ERROR:core:print_rr_body: too many RR
suggest the fact the your SIP messages have more than 64 Record-Route
headers, which is huge. Are you sure you are not having a traffic loop
problem ? Please
I put the latest dialog module from svn; I will see if it will still happen.
Thank you
On Fri, Mar 11, 2011 at 1:04 PM, Anca Vamanu a...@opensips.org wrote:
Hi Iulian,
I suggest to update the dialog module from svn, branch 1.6. There was a
memory leak discovered after the 1.6.2 release and
Hi,
Is there anyone with experience using rtpproxy_stream2uac command. I'm
trying to use with OpenSIPS 1.6.4 with rtpproxy-1.2.1. I'm getting the
following error:
ERROR:nathelper:rtpproxy_stream: required functionality is not supported by
the version of the RTPproxy running on the selected node.
Hello,
I have a question about connectivity to the public network, if a user wants
to make a call to the pstn, where does he define the originator of the call?
in the from header? or using a RPID? i'm quite confuse aboout this?
Many thanks!
___
Users
Depends what the terminating carrier gateway wants to see. At least here in
the US my experience is most prefer at least an RPID, some a
P-Asserted-Identity header with a Privacy header if you want to indicate
restricted caller ID.
- Jeff
From: Toyima Dias
Let me reword that…
Most prefer at least an RPID header.
Alternatively, some prefer a P-Asserted-Identity header.
If you use a PAI header, and want to indicate caller privacy, you'd use the
Privacy header in conjunction with it.
- Jeff
From: Jeff Pyle
Hello,
If opensips receives a request with a R-URI the same of the proxy it will
make any changes on the R-URI as i want, right? but what about if the
domain of the R-URI is not the one of the opensips proxy? (its behavior
should be as stated on the section 16.12 of the rfc..) could i make
so, the RPID will identify the originator of the call (in terms of the
PSNT), right? what if i do not set any RPID or PAI, what will be presented
as teh originator of the call on the pstn destiny?
2011/3/11 Jeff Pyle jp...@fidelityvoice.com
Let me reword that…
Most prefer at least an RPID
2011/3/11 Toyima Dias toyim...@gmail.com
so, the RPID will identify the originator of the call (in terms of the
PSNT), right? what if i do not set any RPID or PAI, what will be presented
as teh originator of the call on the pstn destiny?
Then it will be taken from the From
($fU)
Hi Flavio,
Can you please give me the output of the command rtpproxy -v?
The word session specifies RTPProxy to search for the codec list in
the initial Update command (when OpenSIPS calls rtpproxy_offer), so
there is nothing wrong with it.
This is how RTPProxy works when it receives a Play
Ok, so the RPID is not mandatory...
2011/3/11 Laszlo las...@voipfreak.net
2011/3/11 Toyima Dias toyim...@gmail.com
so, the RPID will identify the originator of the call (in terms of the
PSNT), right? what if i do not set any RPID or PAI, what will be presented
as teh originator of the
Hi Razvan,
rtpproxy -v
Basic version: 20040107
Extension 20050322: Support for multiple RTP streams and MOH
Extension 20060704: Support for extra parameter in the V command
Extension 20071116: Support for RTP re-packetization
Extension 20071218: Support for forking (copying) RTP stream
Extension
All depends on the carrier you're sending it to. I've got some where this is
the case. I've got others I have to use an RPID, and others still I have to
use a PAI. I build AVPs on the inbound portion my proxy called $avp(s:id_name)
and $avp(s:id_num) with a message flag to indicate privacy.
Hi Razvan,
I got to make it work using a version downloaded from GIT. There was a
mistake in the name of the file, it is working, I have removed the check
from rtpproxy_stream.c
Thanks for helping.
Flavio E. Goncalves
2011/3/11 Razvan Crainea razvancrai...@opensips.org
Hi Flavio,
Can
Hello Flavio,
I don't know why you have to take the check out. Probably this is a
little bug. I will dig into this and let you know as soon as I solve the
problem.
Regards,
Razvan
On 03/11/2011 04:35 PM, Flavio Goncalves wrote:
Hi Razvan,
I got to make it work using a version downloaded
Razvan,
I need to implement this feature with GSM and G729, but I see that this
files do not appear in the rtpproxy distribution. Any tip?
Best regards,
Flavio E. Goncalves
2011/3/11 Razvan Crainea razvancrai...@opensips.org
Hello Flavio,
I don't know why you have to take the check
Each AOR should be checked once during a timer interval.
When the timer interval is configured, an internal timer interval is
computed by dividing the configured timer interval over the number of
hash tables.
For large number of hash tables and a small timer interval, more then
one hash tables
Flavio:
http://en.wikipedia.org/wiki/G.729
http://en.wikipedia.org/wiki/G.729G.729 includes
patentshttp://en.wikipedia.org/wiki/Software_patent from
several companies and is licensed by Sipro Lab Telecom.
http://en.wikipedia.org/wiki/GSM#Voice_codecs
Hi Kamen,
Thanks for the indication, I found this in the source code of makeann.c
+then
+ echo
***
$ECHO_C 16
+ echo * Please contact sa...@sippysoft.com
if you need G.729 support in makeann * $ECHO_C 16
+ echo
In case someone finds this later on with similar issues, I'll tell the rest of
the story.
Setting the skipNormalize field to 1 or true in each data source did stop the
normalization. This stopped the problem of fields changing when I didn't want
them too. But, it also caused CDRTool to not
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