All looks good
Chris
> On Jan 19 2016, at 9:24 pm, Nick Altmann nick.altm...@gmail.com
wrote:
>
> No repository – no problem with signing. ;-)
>
> Fixed now.
>
>
>
> \--
>
> Nick
>
>
>
> 2016-01-20 0:11 GMT+03
sips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml: [Errno
14] curl#7 - "Failed connect to yum.opensips.org:80; Connection refused"
Trying other mirror.
Chris
> On Jan 19 2016, at 11:05 am, Chris Stone axi...@gmail.com wrote:
I'll give it another trythanks
>
>
I'll give it another trythanks
Chris
> On Jan 19 2016, at 9:59 am, Nick Altmann nick.altm...@gmail.com
wrote:
>
> The repo is in testing mode.
>
> Have you tried today? Try again, all rpms are signed.
>
>
>
> \--
Nick
>
>
>
>
It is - but the packages (rpms) are all unsigned
Chris
> On Jan 19 2016, at 3:15 am, Bogdan-Andrei Iancu bog...@opensips.org
wrote:
Hey, it seems to be up again !!! :))
Regards,
>
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
>
, let me know.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 01/24/2013 07:09 PM, Chris Stone wrote:
Started getting some errors last night from Opensips:
Jan 23 19:19:01 mars /sbin/opensips[23215]: ORIGINAL-INVITE
[method=INVITE, from
Toyimo,
On Wed, Mar 9, 2011 at 11:19 AM, Toyima Dias toyim...@gmail.com wrote:
I can't find the opensips.pid on Red Hat, it is not in the path:
/var/run/opensips.pid
Has somebody installed OpenSIPS on Red Hat and be able to find the pid of
the process opensips? i'm trying to monitor the
Ovidiu,
On Wed, Feb 9, 2011 at 3:02 PM, Ovidiu Sas o...@voipembedded.com wrote:
It may be that something on the way out that is changing the message.
I've seen some routers messing with the INVITE that is sent out.
I want to see the message as it is when it leaves opensips and before
being
Well, looks like it WAS the ip_nat_sip and related kernel modules, but
not just on the Opensips server, also on the Asterisk server. I
unloaded all of the modules on the backend Asterisk server too and
tried a test call again and this time it worked just fine.
Appreciate all the help with this
Well, looks like it WAS the ip_nat_sip and related kernel modules, but
not just on the Opensips server, also on the Asterisk server. I
unloaded all of the modules on the backend Asterisk server too and
tried a test call again and this time it worked just fine.
Appreciate all the help with this
Henk,
On Tue, Feb 8, 2011 at 6:23 PM, Henk Hesselink opensips-us...@voipro.nl wrote:
What does an ngrep trace on the interface (interfaces?) of the machine
show? Are the packets changed when they leave the machine? In that
case it must be happening on that machine, i.e. no external firewall
Robin,
If you want to compile and install Opensips and all of the default
modules, then just use:
make all
make install
The default is to compile all modules except those listed here (from
the root Makefile):
exclude_modules?= b2b_logic jabber cpl-c xmpp rls mi_xmlrpc xcap_client \
Ovidiu,
On Wed, Feb 9, 2011 at 2:12 PM, Ovidiu Sas o...@voipembedded.com wrote:
Add the following xlog at the beginning of the script:
xlog(L_INFO, $mb\n);
Cool - forgot about the $mb to log the packet. Thanks - can be handy.
In the route() function, the packet was dumped to the log. Looked
Ovidiu,
On Wed, Feb 9, 2011 at 2:30 PM, Ovidiu Sas o...@voipembedded.com wrote:
well ... switch to tm and try again.
Did and no luck - still not firing local_route(). Changed my forward()
to t_relay(), thinking that might be it, and got a *ton* of error
messages about memory for xlog needing
Henk,
On Wed, Feb 9, 2011 at 3:48 PM, Henk Hesselink opensips-us...@voipro.nl wrote:
First off, the ngrep trace doesn't seem to match the debug log, f.i.
the From: tag is different. You might also want to check out -Wbyline
for ngrep, makes reading the output a lot easier.
They were
Dave,
On Tue, Feb 8, 2011 at 12:02 AM, Dave Singer dave.sin...@wideideas.com wrote:
Don't know what tools you are familiar with so here are some
suggestions for what they're worth.
Appreciate the input!
Am familiar with all - but included output below - always happy to
have another set of
Dave,
On Tue, Feb 8, 2011 at 1:09 PM, Dave Singer dave.sin...@wideideas.com wrote:
So weird.
Did you specify the -f path to custom config on the opensips cmd line?
That is the only thing left that I can think of as the problem, it is
using a default config that is somewhere else.
Yes - used:
Ovidiu,
On Tue, Feb 8, 2011 at 1:16 PM, Ovidiu Sas o...@voipembedded.com wrote:
Based on your description, it seems that you are dealing with a weird
firewall/NAT that is SIP aware.
Also, I don't know if this is a typo, but you are forwarding with
opensips to 67.212.153.179 and your INVITE is
Ovidiu,
On Tue, Feb 8, 2011 at 2:31 PM, Ovidiu Sas o...@voipembedded.com wrote:
You din't posted a proper SIP capture (no UDP src/dst IP for your
packets), but it seems that you are having a firewall issue. Just add
I agree that it almost sounds like a firewall issue butI
completely
On Mon, Feb 7, 2011 at 10:35 AM, Ovidiu Sas o...@voipembedded.com wrote:
You are re-posting the same question again without providing any
additional info:
http://lists.opensips.org/pipermail/users/2011-February/016626.html
Sorry, did not see my original message in my Gmail sent messages, or a
Ovidiu,
On Mon, Feb 7, 2011 at 11:22 AM, Ovidiu Sas o...@voipembedded.com wrote:
Just capture a call that is going through your SIP proxy and check the
SDP in the received and sent INVITE and the SDP in the received and
sent 200ok. The connection IP and port for each SDP should be the
same
Ovidiu,
On Mon, Feb 7, 2011 at 4:19 PM, Ovidiu Sas o...@voipembedded.com wrote:
By default, opensips does not modify the SDP.
Double check your config. If you don't need to touch SDP, make sure
that you are not loading nathelper or mediaproxy. Those are the two
modules that are changing
else is touching the message?
Regards,
Henk Hesselink
On 08-02-11 02:33, Chris Stone wrote:
Ovidiu,
On Mon, Feb 7, 2011 at 4:19 PM, Ovidiu Saso...@voipembedded.com wrote:
By default, opensips does not modify the SDP.
Double check your config. If you don't need to touch SDP, make sure
Sorry all for the last message - too quick on the Send button.
On Mon, Feb 7, 2011 at 6:48 PM, Henk Hesselink opensips-us...@voipro.nl wrote:
Hi Chris,
That config should't touch the Contact header, and yet that's also been
modified:
In: Contact:sip:+13038382386@208.94.157.10 ...
Out:
Hello - We have an Opensips 1.4 server that routes incoming calls to a
couple of different Asterisk servers and to upstream providers. All
working great and with the current config, the Opensips server only
handles the SIP traffic - all of the audio is between the UAs and
Asterisk servers.
Am
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