10.02.2014 13:24, Bogdan-Andrei Iancu пишет:
Hello,
In this scenario:
Astеrisk -- 3 -- OpenSIPS -- 4 -- UAC
What SIP requests the UAC is sending ? REGISTER ? INVITES ?
Hello!
UAC registered to Opensips, Opensips registered as UAC on Asterisk. When
the INVITE request comes from the
Hello Alec,
I'm happy you managed to make it work.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 11.02.2014 06:18, Alec Doran-Twyford wrote:
Hi Bogdan,
Thanks for the help it working a charm now the code look like this for
anyone else who
Hello,
Use the dialog based topology hiding
http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id296001
when forwarding the INVITE to Asterisk (in OpenSIPS).
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 11.02.2014 10:12, Александр
Hi
Now I have one DNS SRV domain (sip.domain.com) which points to two A
record inputs with different weight (sip1.sss.domain.com and
sip2.sss.domain.com).
If I add for sip.domain.com A record which will pont to the same IP as
sip1.sss.domain.com, could this be causing any problems for
On 11 Feb 2014, at 14:47, Miha m...@softnet.si wrote:
Hi
Now I have one DNS SRV domain (sip.domain.com) which points to two A record
inputs with different weight (sip1.sss.domain.com and sip2.sss.domain.com).
If I add for sip.domain.com A record which will pont to the same IP as
Hello,
OK, thanks for testing - I will push the fix I did in the public code. I
will also give a bit more thinking to see what should be the best
solution to expose the internal error.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On
Hello,
Two things:
1) be sure you have the latest 1.9.1 code (GIT or tar balls)
2) post the actual backtrace (from the core file), please
Thanks and regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 11.02.2014 06:25, Ahsan Hasan wrote:
So it
Hello,
OpenSIPS will automatically try NAPTR/SRV (of course, depending on the
URI form - like port must be unknown in order to go for SRV lookup).
If I got it right, your hackish DNS configuration may work - using the
same name for the SRV and for the A record - it should not be any
Hello,
Please keep the list CC'ed all the time !
You did no inserted the topo hiding triggering in the write place - you
need to do it only for initial INVITEs ; for sequential requests you
need the be sure to invoke the match_dialog() function. See:
On Thu, Feb 6, 2014 at 1:54 PM, Nick Altmann nick.altm...@gmail.com wrote:
Carrier is just a gateways list. You may also set weight for each gw
and drouting will choose gw's due its weight.
Route_to_carrier() gets carrier parameters (gateways list) into avps
and set $ru to first gw address.
Hi,
I have attached a copy from the pbx and the opensips/ I been told by one of
my colleagues that the contact header maybe the one which is effecting the
call working. I can't seem to find away to modify the contact header.
However I found this am I on the right track if I need to modify the
As I write before... Look into these avps after do_routing:
modparam(drouting, ruri_avp, '$avp(dr_ruri)')
modparam(drouting, gw_attrs_avp, '$avp(dr_gw_attrs)')
modparam(drouting, gw_prefixes_avp, '$avp(dr_gw_pref)')
Try to change order there (in all avps the same order).
use_next_gw() just
The Opensips version I am using is 1.9.1 compiled from src tar. The output
of opensips -V is below:
===
version: opensips 1.9.1-tls (x86_64/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST,
SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC,
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