Hi,
El 02/02/10 16:50, Leonid Nasedkin escribió:
Hi there.
Is it posible to use dns-name at media_relay_avp?
Now I just get error:
media-dispatcher[356]: warning: user requested media_relay
relay-01.test.local is not available
No, you need to specify the IP address, as the dispatcher keeps
Hi,
El 03/02/10 5:28, CheeWii escribió:
Havn't resloved the former problem,I got into the new trouble. The
errors shows as follows,could you give me some suggestions? Is it the
fault of python？？
Unfortunately the traceback doesn't show anything useful :-/ I'll try to
reproduce your scenario
Hello to everybody!
I'm newbie with opensips. I have a voip network with about 30 sip clients
and I use opensips as sip server.
one of this clients should have the highest priority and should be able to
speak to everybody, so all active calls should be hanged up! how can I set
the highest
Hi.
I'd like to set that in my voip network (using opensips as sip server) no
more than one call should be allowed. in case of multiple calls I'd like to
put them in a queue and serve one call at a time. is there the possibility
to set this feature in opensips?
thanks a lot!
--
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Hi,
El 03/02/10 6:38, Daniel Worrad escribió:
Hi All,
I have MediaProxy working in a multi-homed setup where it is acting as a
relay between an interface on a public IP and one on a private IP (We
connect to our SIP provider over a private network via the private
interface using static
El Miércoles, 3 de Febrero de 2010, Mike O'Connor escribió:
Authenticate by IP Address, yep but how ?
What is the recommend way of handling this and how do I route DID's to
them.
You can use address table (permissions module). Use the grp column to
identify the client based on the origin
El Miércoles, 3 de Febrero de 2010, Lei Tang escribió:
Hi Bogdan and Iñaki, Thank you every much.
Bogdan, Could you send me the patch?
Did you try my suggestion before??
http://www.opensips.org/Resources/DocsCoreFcn15#toc25
--
Iñaki Baz Castillo i...@aliax.net
El Miércoles, 3 de Febrero de 2010, wüber escribió:
Hello to everybody!
I'm newbie with opensips. I have a voip network with about 30 sip clients
and I use opensips as sip server.
one of this clients should have the highest priority and should be able to
speak to everybody, so all active
El Miércoles, 3 de Febrero de 2010, wüber escribió:
Hi.
I'd like to set that in my voip network (using opensips as sip server) no
more than one call should be allowed. in case of multiple calls I'd like to
put them in a queue and serve one call at a time. is there the possibility
to set this
yes, exactly!
thanks a lot.
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El Miércoles, 3 de Febrero de 2010, wüber escribió:
yes, exactly!
thanks a lot.
AFAIK this is not possible.
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Ok. So, I cann't put some calls in a waiting state if the callee is already
busy. is it right?
Can you suggest a way to do that?
Thanks a lot for your support!
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El Miércoles, 3 de Febrero de 2010, wüber escribió:
Ok. So, I cann't put some calls in a waiting state if the callee is
already busy. is it right?
Can you suggest a way to do that?
Again OpenSIPS is mainly a SIP proxy while you are asking for PBX features.
I recommend you to use a PBX
Hi Brian,
The test I mentioned was done with UDP.
The messages you are seeing means that the TCP MANAGER process sees that
all the TCP WORKER processes are already processing other messages, so
no one is free (idle), so , it will queue the current active connection
to one of the TCP
Hi Julien,
OpenSIPS does not support the 0.0.0.0 address - it is not able (due
internal stuff) to learn new IPs on the fly (at runtime)
Regards,
Bogdan
Julien Chavanton wrote:
This was well documented
in opensips.cfg
listen=udp:0.0.0.0:5060
or for both TCP/UDP
Hi Brian,
Regarding the crash you mentioned - do you have any backtraces ?
About some of your doubts:
1) t_relay() is not forcing any proto by itself: it preserves the
inbound proto if the RURI (or socket) is not saying otherwise.
2) turning off the double RR may brake some things as opensips
Hi Iñaki,
Well, that is for forcing a totally different port, not for forcing the
addition of the default port. But theoretically it may work...not sure
about the practical part :).
Regards,
Bogdan
Iñaki Baz Castillo wrote:
El Martes, 2 de Febrero de 2010, Bogdan-Andrei Iancu escribió:
Hi,
When accessing the dialog context for a sequential request (like ACK,
BYE), be sure you do it after loose_route() function - this function is
the one matching the request to the internal dialog and exposing the
dialog context.
Regards,
Bogdan
liuf wrote:
I have the same problem as
El Miércoles, 3 de Febrero de 2010, Bogdan-Andrei Iancu escribió:
Hi Iñaki,
Well, that is for forcing a totally different port, not for forcing the
addition of the default port. But theoretically it may work...not sure
about the practical part :).
Hi, by adding such option
Nice little utility, saves alot of time on typing. :-)
Here's a pastbin with the correct format (ngrep-sip b) of the same call:
http://pastebin.ca/1776903
Thanks,
Magnus
2010/2/2 Iñaki Baz Castillo i...@aliax.net
El Martes, 2 de Febrero de 2010, Magnus Burman escribió:
I'll start capturing
El Miércoles, 3 de Febrero de 2010, Magnus Burman escribió:
Nice little utility, saves alot of time on typing. :-)
Here's a pastbin with the correct format (ngrep-sip b) of the same call:
http://pastebin.ca/1776903
As you can see, the SDP in not modified by mediaproxy module for the
Hi Magnus,
El 03/02/10 12:38, Magnus Burman escribió:
Nice little utility, saves alot of time on typing. :-)
Here's a pastbin with the correct format (ngrep-sip b) of the same call:
http://pastebin.ca/1776903
Thanks,
Magnus
This looks as a configuration issue to me. Have a look at the
None of my users are behind NAT, they're all on public IPs (I control their
connection).
Sorry if it's a stupid question, but what do you mean with the SDP is not
modified by mediaproxy?
On line 276 in the re-invite (Opensips -- UA) the port used is different:
m=audio 40518 RTP/AVP 18 8 0 101'
El Miércoles, 3 de Febrero de 2010, Magnus Burman escribió:
None of my users are behind NAT, they're all on public IPs (I control their
connection).
It could occur that the gateway just allows RTP from certains IP's.
Sorry if it's a stupid question, but what do you mean with the SDP is not
Thanks for your help.
I've another question. I can't access any of the dialog vals on reply
route. loose_route() only can be used for REQUEST_ROUTE. Am I missing
something?
opensips.cfg
=
onreply_route {
if (status==100)
{
Now I see what you're saying. I thought mediaproxy used the wrong port in
the re-invite, while it is in fact not engaged at all and thus the original
IP and port is sent on. That makes a lot of sense, thank you.
According to the docs the engage_media_proxy should only be called once on
the
El Miércoles, 3 de Febrero de 2010, Magnus Burman escribió:
Now I see what you're saying. I thought mediaproxy used the wrong port in
the re-invite, while it is in fact not engaged at all and thus the original
IP and port is sent on. That makes a lot of sense, thank you.
Yes, that's the
Hi Franz,
Please update to svn branch, it contains a much stable version of b2b.
Regards,
--
Anca Vamanu
www.voice-system.ro
Franz Edler wrote:
Hi,
I observed the following behaviour with B2BUA(top hiding scenario):
Whenever the ringing phase lasts e few seconds longer a segmentation
Thanks for your help.
I've got it. Now I can access the dialog vals on request route and reply
route. Thanks.
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Thanks a lot !
Looking forward to your reply kindly~ : )
2010/2/3 Saúl Ibarra Corretgé s...@ag-projects.com
Hi,
El 03/02/10 5:28, CheeWii escribió:
Havn't resloved the former problem,I got into the new trouble. The
errors shows as follows,could you give me some suggestions? Is it the
Thank you for your help Iñaki, it's much appreciated.
By asking my last question I was hoping someone else might chime in. :-)
Best Regards,
Magnus
2010/2/3 Iñaki Baz Castillo i...@aliax.net
El Miércoles, 3 de Febrero de 2010, Magnus Burman escribió:
Now I see what you're saying. I thought
Hi,
El 03/02/10 14:43, Magnus Burman escribió:
Now I see what you're saying. I thought mediaproxy used the wrong port
in the re-invite, while it is in fact not engaged at all and thus the
original IP and port is sent on. That makes a lot of sense, thank you.
According to the docs the
Hello Bogdan,
An mer., févr 03, 2010, Bogdan-Andrei Iancu schrieb:
opensipsl...@encambio.com wrote:
I have eight TCP listeners configured and about sixteen UACs are
connected. I get a ton of these warnings whenever REGISTER or INVITE
messages come in:
Feb 02 18:17:22 name.host.tld warning
Hello Bogdan,
An mer., févr 03, 2010, Bogdan-Andrei Iancu schrieb:
Regarding the crash you mentioned - do you have any backtraces ?
There's quite a lot of situations in which OpenSIPS crashes, so
I'm not sure this one is related to TLS traffic arriving on a non
TLS port. In any case, here's the
Could any one please let me know if opensips support CID (content
indirection) in PIDF PUBLISH and NOTIFY ?. Is there any configuration to be
done to enable if it is available ?
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Hi Saúl,
Thanks for your help, as you suggested, it doesn't look like the rtp timeouts
are working. I am running CentOS kernel 2.6.18-92.el5, with the following
component versions:
gnutls-2.4.1
python-application-1.2.1
libnetfilter_conntrack-0.0.101
mediaproxy-2.3.10
mediaproxy-2.3.10
Hi Bogdan,
Sorry for not getting back sooner. I've updated my config a bit. I'm
including what our reinvite handling looks like and the two reinvites
that pass through opensips. The second one as you can see has no
payload (ngrep shows ...) I have verified this as well under wireshark.
For now,
I add quota on opensips.subscriber table;
I just change in opensips.cfg this case 1 part:
switch ($retcode) {
case 2:
# Call with no limit
xlog(L_INFO, Call control: no limit\n);
case 1:
# Call with a limit under
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