Hi,
We have Opensips 1.4.4. with openxcap 1.0.7.
We are using sip_subscribe_rls script of SIP Simple client to send
RLS-subscriptions.
Is it possible to user opensips with openxcap to build an XDM service for
RCS specification, where there is one shared resource list that is linked to
pres-rul
Hi All
Opensips 1.5 is installed in CentOS. Presence is loaded with simple
configuration i.e. any one can see the presence status of any one.
Presece, presence_xml, presence_mwi are loaded in order.
When server is started, it is crashing when presence is loaded.
pr 27 23:59:21 [6058] INFO:auth_db:
Hi JayaPrakash,
Can you print here the output of the gdb backtrace?
regards,
Anca
JayaPrakash wrote:
> Hi All
> Opensips 1.5 is installed in CentOS. Presence is loaded with simple
> configuration i.e. any one can see the presence status of any one.
> Presece, presence_xml, presence_mwi are loa
Hello,
i have configure the opensips.cfg (config.cfg) to get the RLS server support.
modparam("rls", "db_url", "mysql://opensips:passw...@db/opensips")
modparam("rls", "server_address", "sip:r...@10.0.0.1")
modparam("rls", "to_presence_code", 5)
modparam("rls", "integrated_xcap_server", 1)
Hi Toni,
TLahna wrote:
> Hi,
>
> We have Opensips 1.4.4. with openxcap 1.0.7.
>
> We are using sip_subscribe_rls script of SIP Simple client to send
> RLS-subscriptions.
>
> Is it possible to user opensips with openxcap to build an XDM service for
> RCS specification, where there is one shared re
Hi Khan,
All your error are RADIUS related.
for Freeradius: have you checked if the .sock file exists and what are
the permissions?
for Mediaproxy: check if you point to a existing valid radius cfg file
Regards,
Bogdan
Khan wrote:
> Dear Users,
>
> I have made bunch of changes in my opensips
Hello,
someone has a functioning main routing logic from Opensips where RLS is
configure. opensips work well and the modparam from RLS too, but i dont know
what code i need i the main routing logic to make RLS 100% functional.
it would be great if someone could help me
Regards
Michael
___
Hello,
someone has a functioning main routing logic from Opensips where RLS is
configure. opensips work well and the modparam from RLS too, but i dont know
what code i need i the main routing logic to make RLS 100% functional.
it would be great if someone could help me
Regards
Michael
___
Hi Bogdan,
here my 2 Euro Cents, following your brainstorm request:
> 1. Init Route (new)
Personally I do not have immmediate need for this feature, but as soon
as I start digging into memcache support it will be really useful. As
it also shouldn't be that hard to implement: please do it!
> 2.
OK, perfect.
Regards,
Bogdan
ASHWINI NAIDU wrote:
> Hi Bogdan,
>
> The Issue is solved. The crash was due to rtpproxy.
>
>
>
> On Tue, Apr 28, 2009 at 3:26 PM, Bogdan-Andrei Iancu
> mailto:bog...@voice-system.ro>> wrote:
>
> Hi Ashwini,
>
> Can you extract and post the backtrace fr
Hi all,
yes, that is correct and it is included in the 1.4 to 1.5 migration doc.
See : http://www.opensips.org/Resources/DocsMigration14to15
Regards
Bogdan
Alex Balashov wrote:
> It was removed in 1.5.0. Transactions are now automatically released
> where appropriate.
>
> Franz Edler wrote:
>
Have you read the documentation : http://www.opensips.org/Resources/Rls?
Michael Ciupka wrote:
> Hello,
>
> someone has a functioning main routing logic from Opensips where RLS is
> configure. opensips work well and the modparam from RLS too, but i dont know
> what code i need i the main routin
Hello,
oh no. thank you. i read the text now. i have another question.
when i enter this command in the webbrowser i get always the same output:
http://192.168.2.6:8000/xcap-root/pres-rules/users/alice/pres-rules.xml
http://192.168.2.6:8000/xcap-root/pres-rules/users/alice/pres-rules.xml
http://1
Check the source of the page in the browser, it might not display the tags in
the visible output.
Sebastian
> -Original Message-
> From: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Michael Ciupka
> Sent: Tuesday, 28. April 2009 12:56
> To: A
Hi Bogdan,
The Issue is solved. The crash was due to rtpproxy.
On Tue, Apr 28, 2009 at 3:26 PM, Bogdan-Andrei Iancu wrote:
> Hi Ashwini,
>
> Can you extract and post the backtrace from the core file (I see the
> corefile is generated). Use "gdb" and do "bt" to get the backtrace.
>
> Thank
HI Iñaki,
Actually I came across GW sending 180 + SDP (instead of 183 + SDP), so
the most sane thing you can do is to force RTP replay for all replies
advertising "applications/SDP" content type.
Regards,
bogdan
Iñaki Baz Castillo wrote:
> 2009/4/27 Sergio Gutierrez :
>
>> Hi Iñaki.
>>
>>
Hi,
There is no magic needed to make T38 working with opensips - T38 is a
codec, so it is involving only the end parties and not the mid proxies.
You just have to take care of correctly routing the sequential requests
in the dialogs as FAX usage typically requires a re-INVITE for
re-negotiatin
Hi Bogdahn,
> I guess the problem is with outbound calls, right, as only there you do
> use dispatcher ? or?
No, its with the inbound calls. The outbound calls are routed correctly.
BR
Uwe
>
> Regards,
> Bogdan
>
> Uwe Kastens wrote:
>> Hello,
>>
>> I am handling in and outbound calls with a
Hi Jeff,
Could you post the INVITE that gets the 422 ? Does it contain a min-se
header? If not, according to the RFC, the default min-se is 90 (from the
request) and the server has min-se 1800 -> this is why you get the 422.
Regards,
Bogdan
Jeff Pyle wrote:
> Hello,
>
> I noticed today that
Hi Sergio,
that means RTPProxy received an update command (U) (for request) , than
you have the callid , SDP IP and SDP port +from trag (1 is to tag which
is not available at request time)
Regards,
Bogdan
Sergio Gutierrez wrote:
>
> Hello all.
>
> Is there any guide about how to interpret rtp
Hi,
Get a ngrep capture of the SIP traffic between * and OSIPS . Typically a
retransmission is triggered by a lack of response from the other party,
but to see what response is lacking, you need to see the ngrep capture
of the SIP traffic.
Regards,
Bogdan
troxlinux wrote:
> Hi list , I have s
HI Uwe,
I guess the problem is with outbound calls, right, as only there you do
use dispatcher ? or?
Regards,
Bogdan
Uwe Kastens wrote:
> Hello,
>
> I am handling in and outbound calls with a different setup for
> loadbalancing and failover.
>
> 1) Inbound calls to UA:
> softswitch >LB> asteris
Ok, and how the inbound calls are affected by dispatcher:
1) Inbound calls to UA:
softswitch >LB> asterisk1 to asteriskn => opensips => opensips => UA
?
Regards,
Bogdan
Uwe Kastens wrote:
> Hi Bogdahn,
>
>> I guess the problem is with outbound calls, right, as only there you do
>> use dispa
Hi Ashwini,
Can you extract and post the backtrace from the core file (I see the
corefile is generated). Use "gdb" and do "bt" to get the backtrace.
Thanks and regards,
Bogdan
ASHWINI NAIDU wrote:
> Hi everyone,
>
> I have installed *opensips-1.5,CDRTool-6.7.7,Call-controller-2.0.3
> on a
> Ok, and how the inbound calls are affected by dispatcher:
>
> 1) Inbound calls to UA:
> softswitch >LB> asterisk1 to asteriskn => opensips => opensips => UA
This is what I am doing befor relaying to the asterisk. If I comment 1
and 2 out, I will have the typical problem, that I am missing some
Hi Thomas,
Thomas Gelf wrote:
> Hi Bogdan,
>
> here my 2 Euro Cents, following your brainstorm request:
>
>
>
>> 1. Init Route (new)
>>
>
> Personally I do not have immmediate need for this feature, but as soon
> as I start digging into memcache support it will be really useful. As
> it al
Hi Bogdan,
Here's the call flow:
Adtran CPE gateway (no timer support)
|
V
Opensips 1.4.4 with SST (P1)
|
V
Opensips 1.5.x with SST (P2)
|
V
PSTN carrier
The INVITE arri
Hi,
I made again some tests.
It looks like, that there is a bug in my configuration. I have to debug
further. It looks like that for any type of sip pkg there in no active
transaction.
So the ds_select_dst was the only option to know where to send the pkg to.
BR
Uwe
Uwe Kastens schrieb:
>> Ok
>
> 1. Init Route (new)
>
> A route to be executed at startup only - this will allow the script
> writer to load stuff from DB , to populate memcache-ul, so, to prepare
> the routing process. More or less you can do this first time load even
> now, but you need to check all the time if you already
Hi Sara,
check all /* to be sure you close them via */
Regard,
Bogdan
PS: please use the list all the time
Sara EL KABIRI wrote:
>
> Hello,
>
> I'm kind of completely stuck.
> It's about an issue i'm encountering while starting Opensips.
>
> Opensips is listening on port 5060 whatever i put i
Hi Brett,
Brett Nemeroff wrote:
>
> 1. Init Route (new)
>
> A route to be executed at startup only - this will allow the script
> writer to load stuff from DB , to populate memcache-ul, so, to prepare
> the routing process. More or less you can do this first time load even
> no
>
> The basic idea is that every module that "returns" a value should also
>> return some sort of primary key that was used for that value. Then it can be
>> logged into ACC. Displayed on logging "Sending call to Gateway #100"
>>
> I see your point and I agree with that - personally I had this need
Thanks Dan,
already solved upgrading ctypes.
Regards.
--
Antonio
2009/4/27 Dan Pascu :
>
> The trace suggests a broken ctypes module.
>
> On Saturday 25 April 2009, Antonio Reale wrote:
>> Hi all,
>> I installed mediaproxy 2.3.4, all python packages with easy_install and
>> gnutls 2.4.3 from so
Hi Jeff,
I reviewed the code, and it looks like if no SE hdr exists, the module
does add the SE hdr, but no MINSEstrangeI made a fix (see the
attached patch) -> apply it on P1
Regards,
Bogdan
Jeff Pyle wrote:
Hi Bogdan,
Here's the call flow:
Adtran CPE gateway (no timer support
Brett,
Brett Nemeroff wrote:
>
> 2. Native combined CDR format. One row, start, answer, end,
> duration. Yes, I know, not a B2BUA.. It'll have limitations,
> and a bit fat disclaimer and I'm sure we'll have "that talk"
> again. :)
>
> you mean to get directly C
Ok,
I tested it several times. The behaviour has been changed. I force a
record_route_preset("$si:5100") for incoming calls from pstn. That works
fine.
The other direction is my problem. If a UA at my external opensips will
make a call, some answers are missing. They are only relayed correctly
if
excuseme , I didn't remember that there was a list
2009/4/27 Alex Balashov :
> You may wish to consider posting this to the SER-Asterisk-Interwork list.
>
regardss
--
rickygm
http://gnuforever.homelinux.com
___
Users mailing list
Users@lists.opensip
>
>
>> have you tried the mysql procedure used by opensips-cp for creating CDR?
> the CP is consuming the ACC data to create CDRs in a different table via a
> mysql procedure that is triggered by cron. So far even with large CDR
> volume, I got good results with this.
>
> But making opensips to di
it is strange, I thought that it was asterisk the problem, but I
upgrade to a version that I consider stable 1.4.24
2009/4/28 Bogdan-Andrei Iancu :
> Hi,
>
> Get a ngrep capture of the SIP traffic between * and OSIPS . Typically a
> retransmission is triggered by a lack of response from the othe
Hi all,
I have been using OpenSER for a couple years now for routing our DIDs
from our carriers to the proper customer trunk on our SBC. We have been
using a static configuration with a routing entry for each number DID
number we have. As you can imagine, even with scripts building the
openser.c
Brett Nemeroff wrote:
>
>
> have you tried the mysql procedure used by opensips-cp for
> creating CDR? the CP is consuming the ACC data to create CDRs in a
> different table via a mysql procedure that is triggered by cron.
> So far even with large CDR volume, I got good results with
Hi Tim,Actually, the documentation is quite good IMO. LCR and CarrierRoute
are all but replaced by Drouting.. so don't bother reading up on LCR or
CarrierRoute.
For drouting, setup your termination points in dr_gateways and your routing
rules in dr_rules. It's really pretty simple.
Are you having
It seams you have an ACK routing problem. The caller (.30:5064)
correctly sends ACK with:
ACK sip:*...@192.168.10.3:5070 SIP/2.0
Route:
but opensips (.3:5060),sends it out as:
ACK sip:192.168.10.3;lr=on;ftag=d7f613072c3769a3 SIP/2.0
this means that OSIPS tinks that 192.16
At Bogdan's request, I checked out the stored proc for CDR Correlation in
the opensips-cp project.
I see the Invite cursor declared as:
DECLARE inv_cursor CURSOR FOR SELECT time, callid, from_tag, to_tag FROM
opensips.acc where method='INVITE' and cdr_id='0';
This seems like a problem to me.. Im
Well, not really since it wont work ;)
I guess i need more tuning..
Also the reply of reinvite from upstream is rewriting the to and from which
i need to mangle i guess
Thanks
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf
Are your media gateways (or end points) configured for T38? I had to
configure the Cisco media gateway to support T38.
Have you looked at packet capture to see where it is failing?
I was able to see a negative reply from the gateway when the fax server
tried to re-invite to negotiate T38.
-
2009/4/28 Bogdan-Andrei Iancu :
> It seams you have an ACK routing problem. The caller (.30:5064) correctly
> sends ACK with:
> ACK sip:*...@192.168.10.3:5070 SIP/2.0
> Route:
>
> but opensips (.3:5060),sends it out as:
> ACK sip:192.168.10.3;lr=on;ftag=d7f613072c3769a3 SIP/2.0
>
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