Yes it helped, thank you.-Original Message-From: wesleyvol...@gmail.comSent: Sun, 12 Dec 2010 19:32:15 -0200To: users@lists.opensips.orgSubject: Re: [OpenSIPS-Users] Opensips LogrotateHey Sergio,I'm using log/messages to log opensips messages, then I have in my /etc/logrotate.d/syslog
Hello list,
Has anyone used MRTG with SNMP and CACTI to monitor OpenSIPS?
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Hello list,
Has anyone used MRTG with SNMP and CACTI to monitor OpenSIPS?
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Sim ajudou, muito obrigado.-Original Message-From: wesleyvol...@gmail.comSent: Sun, 12 Dec 2010 19:32:15 -0200To: users@lists.opensips.orgSubject: Re: [OpenSIPS-Users] Opensips LogrotateHey Sergio,I'm using log/messages to log opensips messages, then I have in my /etc/logrotate.d/sysl
Some other weird stuff that happens if I remove the fix_nated_register call:
Sometimes everything will work fine, audio is fine, etc..
However, sometimes I'll call a phone, and the phone will immediately
place a call back out to the number I'm dialing from!
(very strange).
-- James
On Mon, Dec 13
Hi,
I'm having some issues getting a correct NAT configuration going.
The problem I'm having is
I get a "479 We don't forward to private IP addresses" back when
receiving a call to a phone from Asterisk, presumably because the
location table has private IPs in it for some reason.
This seems to be
Is there an easy way/example from a scripting perspective to check if a
dialog's status is 3 (not received an ACK yet), after a certain period of
time, time out ONLY dialogs in this state?
I've identified a situation where, if the UAC goes unresponsive, and never
sends an ACK, we will eventually s
On 13 Dec 2010, at 14:06, Bogdan-Andrei Iancu wrote:
> Hi Marty,
>
> That is really oddI know cases where opensips with rtpproxy are used on
> Solaris, but without this problem :(
>
> Short 2 questions:
> - what revision of opensips are you using (opensips -V)
> - do you do any nh_rel
Not actually from SVN but from the website. Here is the actual download
link:
http://opensips.org/pub/opensips/latest/src/opensips-1.6.3-tls_src.tar.gz
Btw, I have also tried recompiling the source code successfully. However,
the previously mentioned errors when running OpenSIPS itself remains the
Hi Marty,
That is really oddI know cases where opensips with rtpproxy are used
on Solaris, but without this problem :(
Short 2 questions:
- what revision of opensips are you using (opensips -V)
- do you do any nh_reload before getting that error?
Regards,
Bogdan
Marty Lee wrote:
Hello Anca, Stefano,
thanks for reply, i've already tried to loop the signalling through
localhost, i'm still debugging it though. However, in my opinion, it's not
the ideal solution... It would be nice if there could be a possibility to
manipulate with message parameters generated by b2bua module.
Awesome. Thanks.
On Mon, Dec 13, 2010 at 7:01 AM, Anca Vamanu wrote:
> Hi Duane,
>
> I have found the bug there - it was a wrong definition of xcap-diff event
> and I also noticed it in message-summary. I have fixed it now. I suggest
> you to update the code again.
>
> Thanks,
>
> --
> Anca Va
Hi Ronald,
Are you sure you have the latest 1.6.3 sources from SVN ?
Regards,
Bogdan
Ronald Cepres wrote:
Hi to all,
I have a problem about the drouting module. Here is a snippet of my
script configuration:
...
loadmodule "drouting.so"
modparam("drouting", "db_url", "mysql://user:p...@loca
Hi Bela,
On 12/13/2010 12:48 PM, beci345 wrote:
However, i have problem if i would like to extend the configuration with
topology hiding functionality - byusing the B2B modules.
By calling the scenario with b2b_init_request("top hiding"), Opensips fires
out the INVITE with wrong SDP Connection i
Hi David,
Are you sure you have configured ejabber to accept component connection
from OpenSIPS?
You have to edit the /etc/ejabberd/ejabberd.conf and add a block in the
listen list:
%% External OpenSIPS component
{5347, ejabberd_service, [
{host, "domain.com", [{passw
Hi Duane,
I have found the bug there - it was a wrong definition of xcap-diff
event and I also noticed it in message-summary. I have fixed it now. I
suggest you to update the code again.
Thanks,
--
Anca Vamanu
www.voice-system.ro
On 12/10/2010 11:02 PM, Duane Larson wrote:
Thanks for wo
Dec 13 12:46:27 [9576] DBG:xmpp:xmpp_component_child_process: server read
[xmlns='urn:ietf:params:xml:ns:xmpp-streams'/>]
Dec 13 12:46:27 [9576] DBG:xmpp:stream_node_callback: stream callback:
0: stream:error
Dec 13 12:46:27 [9576] DBG:xmpp:xode_send: xode_send
[72117fa8284621f1aa0745039a7a22584
Make logging to a file and set debug=9.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:users-
> boun...@lists.
I am trying to connect to my ejabber server and getting stream errors;
I am not sure how to connect this module; I set all the parameters as
follows;
modparam("xmpp", "backend", "component")
#modparam("xmpp", "xmpp_domain", "domain.com")
modparam("xmpp", "xmpp_host","domain.com")
modparam("xmp
Could you post your cfg?
regards,
s
Il 13/12/2010 11:48, beci345 ha scritto:
Hello to all,
i'm using the Opensips as proxy in multihomed mode (one public IPaddr and
one internal), with relaying RTP traffic through RTP proxy in bridged mode:
UA --->OpenSipsIP1-OpenSipsIP2 >UA (signalling)
UA
Hello to all,
i'm using the Opensips as proxy in multihomed mode (one public IPaddr and
one internal), with relaying RTP traffic through RTP proxy in bridged mode:
UA --->OpenSipsIP1-OpenSipsIP2 >UA (signalling)
UA>RTPproxyIP1-RTPproxyIP2--->UA (RTP), RTP proxy in bridged mode.
Every
Ok this is a really pointless discussion; Please use Asterisk or
FreeSWITCH forum for these things. This is not a debate forum.
Thanks to everyone for thei wonderful feedback;
On 12/10/10 10:31 AM, Laszlo wrote:
Hmm, it's like Ferrari owners talking about which one is better:
Volkswagen or T
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