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
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)
UARTPproxyIP1-RTPproxyIP2---UA (RTP), RTP proxy in bridged mode.
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)
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(xmpp, sip_domain,
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-
Dec 13 12:46:27 [9576] DBG:xmpp:xmpp_component_child_process: server read
[stream:errorinvalid-namespace
xmlns='urn:ietf:params:xml:ns:xmpp-streams'//stream:error]
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:
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
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,
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
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,
Awesome. Thanks.
On Mon, Dec 13, 2010 at 7:01 AM, Anca Vamanu a...@opensips.org 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,
--
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
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:
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
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_reload
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
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
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
Hello list,
Has anyone used MRTG with SNMP and CACTI to monitor OpenSIPS?
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Hello list,
Has anyone used MRTG with SNMP and CACTI to monitor OpenSIPS?
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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
21 matches
Mail list logo
