Hi,
senario:
[UA]-[Opensips]-[Freeswitch]
UA sending correct ACK to freeswitch but Opensips loose_route() sending it
to itself and it break dialog, If use fix_dialog_route() then it works, I
don't have any IP address added in domain table also.
How do i check whether
thx for the reply, I think I fixed the issue, but do you know why am I seeing
so many of these error messages? Is it just means some of the SIP message
opensips gets can not be parsed or have some weird stuff in it? I checked
these IP, they are from VoIP devices manufactured by Grandstream (e.g.
On Thu, Mar 26, 2015 at 7:41 PM, Aqs Younas aqsyou...@gmail.com wrote:
Hi, users.
I am new to opensips, just started playing with it. Please pardon me for
my naive question, but I want cps in opensips.
I have tried the following snippet, but still calls went on.
if (is_method(INVITE)) {
Hello,
Is it possible to have a rule with a range of numbers in Dynamic routing?
For instance I want 8881231231 to 5 to be routed to a specific gw. Can I do
this with one rule only?
I don't want 8881231236 to 9 to be routed to that gateway.
Thanks,
Ali Pey
Hi,
Dialog state is hanging at state 5 when there is no response from the
destination.
What config we need to add to get the dialog moved to failure_route if,
opensips is not receiving any response from destination for any request
sent out.?
Or
How can we drop the dialog instead of moving to
Hi Ben,
If you run the dr_carrier_status MI command, do you see the same #012
in the ID of the carrier ? #012 is '\n' - are you sure you do not have
it by mistake in DB ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 25.03.2015 21:53,
Hi all,
again on BLF...i noticed that after correct SUBSCRIPTION on the server
and on the phones of the monitored extension, when one of that change
its status, the NOTIFY was send to the server that reply with 404 and
*not* to the monitoring phone.
Here's a siptrace of what's happens:
U
Hi,
First check if opensips actually started : run ps auxw | grep opensips
If it didn't , check the logs (messages or syslog) for errors.
If you see the process, check with netstat -lnp | grep opensips to see
the listening interfaces of OpenSIPS.
Regards,
Bogdan-Andrei Iancu
OpenSIPS
Hi, users.
I am new to opensips, just started playing with it. Please pardon me for my
naive question, but I want cps in opensips.
I have tried the following snippet, but still calls went on.
if (is_method(INVITE)) {
if (!rl_check($si, 3, RED)) {
xlog(L_ERR,Cps
Hi,
It may possible be that the received ACK is broken, leading to that kind
of looping. By broken I mean heaving wrong RURI or Route headers. Can
you post on a pastebin a sip capture of that call ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
ok, no worries :)
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 26.03.2015 17:32, Newlin, Ben wrote:
Bogdan,
After further investigation I have found that the characters are an
artifact of the syslog daemon on the system. When I run with
Bogdan,
It did occur to me that the code seemed to be the octal for newline, but I
don't know what the significance of that could be.
The MI command output does not show the extra characters:
~]$ opensipsctl dr carrier_status
ID:: Level3-VT Enabled=yes
ID:: IndyEdges Enabled=yes
I have also
Bogdan,
After further investigation I have found that the characters are an artifact of
the syslog daemon on the system. When I run with fork=no and log_stderr=yes
I do not see the characters in the screen output.
I have also now noticed both newline characters - #015 and #012 - at other
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