I use the permission before sending the calls to PSTN, however, I can't make it
to work; If I bypass route 3 on this config and go straight to route 4 I can
make call to PSTN through a few asterisk gateways.
modparam("permissions", "db_mode", 1)
modparam("permissions", "trusted_table", "truste
Nothing to worry about; if you are not using the permissions module but
loading it anyway, or are using it but keeping your list of authorised
peers in the database, you do not need to worry about this.
It only matters if you actually wish to list your authorised peers (the
things you use the
Thanks.
It is something I am missing or should I just don't worry about?
--- On Sun, 1/25/09, Iñaki Baz Castillo wrote:
From: Iñaki Baz Castillo
Subject: Re: [OpenSIPS-Users] Warning message at startup
To: users@lists.opensips.org
Date: Sunday, January 25, 2009, 10:46 PM
El Domingo, 25 de
El Domingo, 25 de Enero de 2009, Gonzalo Gonzalez escribió:
> Jan 25 17:40:44 sipproxy /usr/local/sbin/opensips[6640]:
> WARNING:permissions:parse_config_file: file not found:
> /usr/local/etc/opensips/permissions.allow Jan 25 17:40:44 sipproxy
> /usr/local/sbin/opensips[6640]: WARNING:permissions:
What this warning messages means?
Jan 25 17:40:44 sipproxy opensips: WARNING:core:fix_socket_list: could not rev.
resolve 192.168.1.102
Jan 25 17:40:44 sipproxy /usr/local/sbin/opensips[6640]: NOTICE:core:main:
version: opensips 1.4.4-notls (i386/linux)
Jan 25 17:40:44 sipproxy /usr/local/sbi
Dear Bogdan,
The issue has been solved
Thanks
On Sat, Jan 24, 2009 at 8:47 PM, Bogdan-Andrei Iancu wrote:
> Hi Michel,
>
> The error means you try to send a statefull reply from script (via t_reply)
> but there is no transaction already created (via t_newtran) - check in your
> request ruting
Dear Bogdan,
When trying to make a call I got now the following error:
Jan 25 23:17:14 [25302] ERROR:core:validate_msg: Message has no Call-ID
header
Jan 25 23:17:14 [25302] ERROR:carrierroute:rewrite_on_rule: could not hash
message with CRC32Jan 25 23:17:14 [25302] INFO:carrierroute:cr_do_route:
Dear Bogdan,
I changed my mind and need to use carrierroute instead of dispatcher because
I'm sure that carrierroute will accomplish what i need to do...
I added the following function to opensips.cfg:
if(!cr_route("default", "0", "$rU", "$rU", "call_id")){
sl_send_reply("403", "Not
Thanks; I have made some progress on my config, but now I have an issue sending
the calls to the gateways. I took a config file from "Building a telephony
system with openser" from Flavio E. Goncalves an adapted it to my needs, but
when I try to pass a call to PSTN I receive error 404, not found
On Jan 25, 2009, at 12:12 PM, Jimmy Svensson wrote:
Hi!
I'm looking for a pure call recording solution that generates pcap-
files of the data from one specific call. I've been looking at using
the a merged OpenSIPS+rtpproxy solution to achieve this. The
rtpproxy would not be proxying any R
Hi!
I'm looking for a pure call recording solution that generates pcap-files
of the data from one specific call. I've been looking at using the a
merged OpenSIPS+rtpproxy solution to achieve this. The rtpproxy would
not be proxying any RTP traffic in this case, just terminating and
recording. Doe
Hi Geoffrey,
in combination with t_relay(), you do not need to do anything special
for creating the transaction, as it will be automatically created
internally by t_relay(). But there is not the same behaviour for
t_reply() function - this function do expects to find the transaction
already c
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