I'm using the latest opensips release 2.1.2 on Debian and have a problem with
the add_diversion header.
For every callforward I have to add a diversion header at top level with the
counter increased by one. I can put the counter number in manual, but not
use a variable:
Do you see the incoming traffic on the opensips machine (check with
ngrep/tcpdump). Have you checked if OpenSIPS is listening on the right
port (where the traffic is sent to) via "netstat -ulnp| greop opensips"
? Do you see any pending data on the socket ?
Regards,
Bogdan-Andrei Iancu
Yes as I said ngrep shows traffic but no action by opensips. Very annoying
as I have had it working on Amazon in the past.
Netstat shows opensips listening on my local host address on port 5060 UDP.
Opensips logs not showing any traffic.
I have exact same config on digital ocean and it works.
And do you see any data pending on the opensips listening socket (via
netstat in the Recv-Q column) ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 26.01.2016 11:08, Jason Bedward wrote:
Yes as I said ngrep shows traffic but no action by
Hi Colin,
Which of the in which module do you use the uac authentication feature :
uac_auth
b2b_entities
uac_registrant
?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 26.01.2016 11:22, Colin Martin wrote:
Just one extra bit of
Hello all,
The upcoming public meeting will be held on IRC (#opensips on FreeNode),
Wednesday, 27th of January 2016, at 15:00 [1].
The discussions will be based on finding solutions / making improvements
to the way different data types are both stored in and retrieved from
scripting
Maybe some blocking iptables rules on the INPUT chain ??
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 26.01.2016 11:18, Jason Bedward wrote:
No shows 0
On 26 Jan 2016 09:16, "Bogdan-Andrei Iancu" >
Just one extra bit of information, this doesn't happen on the first call
post-restart, but does occur for each subsequent call.
I'm happy to ignore it if it won't affect stability, but I don't know how to
verify that!
Colin
> On 25 Jan 2016, at 18:25, Colin Martin wrote:
>
So after all, the problem was so slow/blocking communication with the
Radius server. For the future, to debug such issue you can use the
exec_msg_threshold to see what are the slow parts of your script:
http://www.opensips.org/Documentation/Script-CoreParameters-2-1#toc57
Regards,
Bogdan,
To answer the first question, I created all of the tables including the
extras. I did another git pull and recreated the tables. All of the
permissions were in place except for the table route_tree_id_seq.
Thanks
Nathaniel Keeling
On 1/25/16 6:07 AM, Bogdan-Andrei Iancu wrote:
Hi
Hi Aqs,
It looks like you have some sql connectivity problems. Are you sure your
opensips can reach the mysql server ? all your errors (about long query
time and about the failure to reconnect) do point to a connectivity
problem to mysql server.
Regards,
Bogdan-Andrei Iancu
OpenSIPS
can uac_replace_from read real phone number from databases?
On 01/25/2016 01:03 PM, MichaelLeung wrote:
thanks for reply
no , it is just a asking , i don't have real phone number database, or
should i have one ?
can you tell me what is the name of this technology ?
On 01/24/2016 07:33 PM,
Hi, Tito!
Can you send me a trace?
Thanks,
Răzvan
On 01/25/2016 10:41 PM, Tito Cumpen wrote:
Hey Razvan,
This is still an issue with the latest dev build. The event is
entirely empty when it is transmitted to the queue. I've tried
modparam("event_rabbitmq", "sync_mode", 1) but it does not
Travis,
The documentation on formating the variables is available under:
http://www.opensips.org/Documentation/Script-CoreVar-2-1
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 26.01.2016 00:16, Travis Manson-Drake wrote:
That did it!
Thanks Bogdan,
Why this problem exists only in Opensips 2.1?
Best regards,
Dragomir
2016-01-26 9:59 GMT+02:00 Bogdan-Andrei Iancu :
> So after all, the problem was so slow/blocking communication with the
> Radius server. For the future, to debug such issue you can use the
Hi Nathaniel,
Thanks for confirming the fix. Indeed, I found out the answer to my own
question after sending you my first reply...after digging more into the
code of opensipsdbctl.
Anyhow, thanks for report !
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
Hi Edwin,
Yes, the add_diversion() does not support variables in parameters.
Still, you can simply add the header via append_hf() (or insert_hf()),
by building the whole header by hand - these functions do support
variables. And the syntax of the Diversion hdr is trivial.
You can also open
Hey Razvan,
This is still an issue with the latest dev build. The event is entirely
empty when it is transmitted to the queue. I've tried
modparam("event_rabbitmq", "sync_mode", 1) but it does not make a
difference.
THanks,
Tito
On Fri, Oct 30, 2015 at 1:20 PM, Răzvan Crainea
Hi Bogdan,
It looks like this document. I have searched in Polycom documentations and
community forum and I can not find any mention of the document.
So I posted the question on Polycom community forum hoping someone can give
an answer.
In Asterisk source code for chan_sip, they have a comment
Hey Razvan,
This is still an issue with the latest dev build. The event is entirely
empty when it is transmitted to the queue. I've tried
modparam("event_rabbitmq", "sync_mode", 1) but it does not make a
difference.
THanks,
Tito
On Mon, Jan 25, 2016 at 3:39 PM, Tito Cumpen
Hi Dragomir,
What RADIUS problem do you still have in 2.1 ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 26.01.2016 10:17, Dragomir Haralambiev wrote:
Thanks Bogdan,
Why this problem exists only in Opensips 2.1?
Best regards,
Dragomir
Hi,
Fixed it (at least tried to) with this commit
https://github.com/OpenSIPS/opensips/commit/be43fcdb7696c6ee53673b351380fe701c209a44.
Can you please test it an tell me if everything works
correctly? If no path provided '/RPC2' shall be used as before, but if
you do provide a path
(like
Hi Julian,
Could you test please this fix made by John:
https://github.com/OpenSIPS/opensips/commit/be43fcdb7696c6ee53673b351380fe701c209a44
If it works ok, we will do the backport to the stable releases too.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
I know I have used presence extensively with Polycom phones using pidf+xml. I
know they support it. Maybe there is some setting in your model specifying the
remote server type? If that is set to Microsoft Lync the Polycom may be sending
xpidf for compatibility.
But Polycom phones absolutely
I had a similar issue using the older Polycom OS and BLF using the directory
and buddy watch (xpidf+xml) that Freeswitch supports really well.
However, after upgrading the Polycom’s and using the newer
attendant.resourceList method for BLF, I found that they use the standard
dialoginfo
Hi Colin,
The issue should be fixed on GIT repo, on all maintained versions.
Thanks and Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 26.01.2016 11:41, Colin Martin wrote:
Hi Bogdan,
It's the uac_auth module, coupled with the uac
Greetings all;
We have several hundred 1.8.8 OpenSIPS proxies across our corporate
networks which are running Debian Wheezy. I apparently haven't built a
new one in a while as sometime between then and now the 1.8 LTS has
become deprecated (although I can't find any eMails or notices on the
Hello, everybody!
This year we'll be again present at the biggest open-source conference,
FOSDEM'16[1].
This year, me and Liviu we'll be showing you how to enhance your VoIP
capabilities by using it as an Edge Proxy/SBC[2]. Looking forward to
seeing you there!
[1] https://fosdem.org/2016/
Hi there!
I use opensip 1.7.2 and noted retransmissions of INVITEs in 450ms despite
the fact I got 100 Trying from my external carrier.
FYI:
1) it's is from opensips exactly (not from UA).
2) Opensips sends absolutely same INVITEs (same cseg, tags, etc).
I can be wrong but RCF says:
"After
Hello Jarrod,
Thanks for pointing that. That's true what you are saying for
attendant.resourceList parameter. We also use it for BLF and call pickup.
XPIDF is used by Polycom directory presence.
It looks like XPIDF is the content type Polycom uses for P2P presence.
So, if I configure OpenSIPS to
Bogdan,
That’s brilliant. Thanks very much.
Colin
> On 26 Jan 2016, at 16:45, Bogdan-Andrei Iancu wrote:
>
> Hi Colin,
>
> The issue should be fixed on GIT repo, on all maintained versions.
>
> Thanks and Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and
Ben,
I had a similar issue using the older Polycom OS and BLF using the directory
and buddy watch (xpidf+xml).
However, after upgrading the Polycom’s and using the newer
attendant.resourceList method for BLF, I found that they use the standard
dialoginfo supported by presence_dialoginfo
Hello Ben,
Thank you for the hint with Lync. I did check our settings and we do not
have any Lync related configuration. It looks like default settings do not
have any special lync stuff.
The Polycom phones do support PIDF with BLF/call pick. The event is dialog.
Meanwhile for directory, when
:O.the classical problem strikes again !!!
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 26.01.2016 11:28, Jason Bedward wrote:
Thanks. Somehow iptables had turned itself back on! All working now.
On 26 Jan 2016 09:23, "Bogdan-Andrei Iancu"
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