e list suggest rewriting the contact using subst() from the
textopts module but after some fiddling around with regexes I'm not able to
produce the desired result.
Could someone point me in the right direction?
Thanks in advance,
Remco.
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Hi Adrian,
Thanks for your reply. Unfortunately, changing the Asterisk side is not an
option as some PBX'es are not under my control and others are embedded
devices and such. So I'm looking for a solution on the OpenSIPS side...
Thanks,
Remco.
On Tue, Dec 6, 2011 at 2:04 AM, Adria
lacing this in the onreply_route but t_reply() is not allowed
there. What would be the best way to achieve this?
Thanks in advance.
Remco.
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Is there a way to log incoming calls using CDRTool?
(e.g. PSTN -> OpenSIPS -> User).
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Thanks Tijmen. Is there any documentation on the Radius configuration
needed?
On Mon, Apr 2, 2012 at 2:39 PM, Tijmen de Mes wrote:
> Hi,
>
>
> On 04/02/2012 02:05 PM, Remco . wrote:
>
>> Is there a way to log incoming calls using CDRTool?
>> (e.g. PSTN -> OpenSIPS
Hi,
An upstream carrier wants user=phone in the request uri, from header and to
header (and, if used in P-asserted-ID as well). To route traffic to the
carriers, drouting is being used. I'm using a routeid in dr_rules to loop
trough the custom route before the invite is sent out to the carrier.
I
dify the CDRtool code, however extending some class might
be a neat solution?
Thanks,
Remco.
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know if that's possible with Asterisk as a media server?).
The announcement works, however returns 200-OK. If I uncomment the
't_reply', the call is ended to soon without allowing the announcement to
be played.
Does anyone how to solve this? I tried branching but I cannot get it to
work.
:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *Im Auftrag von *Remco .
> *Gesendet:* Samstag, 13. Oktober 2012 22:52
> *An:* OpenSIPS users mailling list
> *Betreff:* [OpenSIPS-Users] Route to media-server, but reply negative
>
> ** **
>
>
described in the scenario above.
Any ideas?
Thanks,
Remco.
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Hi Saúl,
Thanks for your reply. I did send you the details off-list for the sake of
privacy.
Thanks,
Remco.
On Tue, Jan 8, 2013 at 11:00 AM, Saúl Ibarra Corretgé
wrote:
> Hi,
>
> On Jan 8, 2013, at 10:49 AM, Remco . wrote:
>
> > Hi all,
> >
> > I seem to experi
Hi,
No Sonicwall involved. There is a Cisco PiX between UA and OpenSIPS. SIP
inspection and helpers have been disabled.
UA is Asterisk PBX.
On Tue, Jan 8, 2013 at 3:58 PM, dotnetdub wrote:
> On 8 January 2013 09:49, Remco . wrote:
> > Hi all,
> >
> > I seem to exper
tant to us");
}
And then use t_relay("0x01") to avoid duplicate 100-replies. This avoids
race-conditions, it might help in your case as well.
Regards,
Remco.
On Tue, Jan 8, 2013 at 9:49 PM, Mariana Arduini wrote:
> Hello all,
>
> I hope somebody can give me a kind hint o
Some additional information on this one as it turned out that I wasn't
running the latest version.
I updated to the latest version 2.5.2 of mediaproxy, but the issue still
exists.
Running kernel 2.6.32-5-amd64 (debian package).
On Tue, Jan 8, 2013 at 4:19 PM, Remco . wrote:
> Hi
_* functions from the textops module?
According to the relevant RFC's, it shouldn't but somehow I can't get
it to work..
Regards,
Remco.
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nat=no is specified for the peer, however media is proxy'ied on a
different IP address than SIP is received from (could that explain
something?).
I hope this rings a bell to someone, as apart from this issue
mediaproxy is functioning perfect and I don't feel like replacing it.
Tha
4/udp (on both ends). An
update entry is visible in the syslog (level: debug).
Thanks,
Remco.
On Tue, Jan 22, 2013 at 12:45 PM, Saúl Ibarra Corretgé
wrote:
> Hi,
>
> Sorry it took me some time. I inspected the traces you sent me privately, see
> inline.
>
> On Jan 22, 2013, at
.
Regards,
Remco.
On Tue, Jan 22, 2013 at 4:04 PM, Remco . wrote:
> Hi Saúl,
>
> I have sent you the information you asked for by e-mail. For the
> interest of others, who might encounter the same problem in the future
> I'll post my findings here as well.
> The strea
which you often cannot avoid the
Virtual IP scenario).
Thanks,
Remco.
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interface -
no wonder the kernel picks the primary. I will see if I can modify the
resource agent a bit so it will add the VIP as the primary IP (swap the IP
addresses round). That way, the mhomed=yes option will work. I will report
back my findings.
Thanks,
Remco.
On Fri, Nov 1, 2013 at 12:27 PM
expected in
this scenario.
Regards,
Remco.
On Mon, Nov 4, 2013 at 12:28 PM, Bogdan-Andrei Iancu wrote:
> Hi Remco,
>
> OK, if your approach does not work, keep in mind you can still use
> local_route to change the outbound socket for the probing OPTIONs.
>
> Regards,
>
might be causing this
behavior?
Thanks,
Remco.
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Hi,
There are resource agents available for corosync which can automatically
start opensips and check sip response using sipsak. See www.linux-ha.org
Regards,
Remco
Op 1 dec. 2013 09:44 schreef "Miha" :
> Hi,
>
> is someone using pacemaker for cluster and have added
> ope
George,
Try enabling the force_dialog parameter in the uac module.
Regards,
Remco
Op 4 dec. 2013 20:43 schreef "George Lee" :
> Hi,
>
> Opensips UAC replace the From URI in the BYE to some garbage. Here is the
> BYE message:
>
> BYE sip:8661234...@domain.com:5060
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