Hi Luis,
To have a LB scenario, you need to have at least 2 destinations where to
send the traffic - like 2 GW. The LB stuff is not related to users at all.
In the example from tutorial, the 6 means the maximum number of channels
(calls) accepted by that particular GW.
For User ID and PWD, yo
Hi Mehdi,
Unfortunately CPL does not implement the diversion hdr. Right now I see
no easy way to make it work (like manually add the hdr) as CPL
interpreter is completly encapsulated and you cannot interact with it -
only changing the code.
Regards,
Bogdan
mehdi boudou wrote:
> This is an
Hi Wesley,
Are you sure it gets there ? put some xlogs to see the execution flow
through your script (also print the RURI - $ru - to seeif changed).
Regards,
Bogdan
Wesley Volcov wrote:
> My opensips is forwarding the INVITE without to tag, but I don't know
> why. Would drouting must do it ?
Hi Alex,
You mentioned you generate these calls with SIPP - right ? are all of
them the same ?
looking at signalling, we should see why that calls have the
callee_route_set (some RR done after your proxy)..
also, have you checked the patch I sent you?
Regards,
Bogdan
Alex Massover wrote:
> H
Alex Massover wrote:
> One more observation. In my test UAS answers with 180 and 200OK to INVITE
> without delay, i.e. the delay between 180 and 200 is only about some ms.
>
> BAD scenario (rare):
> 180 is not processed by OpenSIPS - received from UAS but not forwarded to UAC
> route_set appears a
Hi,
We've got an issue with some SNOM phones that use a different port
every time they renew their registration, and this is causing us to
have lots of entries in our 'location' table for that phone.
Is there any way for us to know that it's the same phone - maybe using
the sip.instance informatio
Hi,
> You mentioned you generate these calls with SIPP - right ? are all of
> them the same ?
>
[Alex] Yes, they all are the same.
> looking at signalling, we should see why that calls have the
> callee_route_set (some RR done after your proxy)..
>
[Alex] UAS is also a SIPP script. But it's not
Hi all,
SIPv6 UA can't receive a 423 (Interval Too Brief) response from
openser-1.3.4-12.fc13.i686 SIPv6 Registrar Server when expiration of a REGISTER
request is less than a registrar-configured minimum.The details can be seen as
follows. Is it a bug or can be resolved by modifying configuratio
Hi all,
A Record-Route header field exists in 200 OK from openser register server after
sending a REGISTER request with a Record-Route header field. The details can be
as follows:
1 Test topology:
NUT(REG && PX) UA11 UA12
Hi all,
Can't receive a 500 (Server Internal Error) response from openser register
server when sending a REGISTER with the equal CSeq header field value to the
last value of CSeq or lower than the number. The details can be as follows:
1 Test topology:
NUT(REG && PX)
Hi all,
Can't receive a 200 0k response from openser register server when URI in a To
header field in a REGISTER request contains escaped characters. The details can
be as follows:
1 Test topology:
NUT(REG && PX) UA11 UA12
Hi all,
SIPv6 UA Can't receive 404 (Not Found) response from openser-1.3.4-12.fc13.i686
register server after sending REGISTER request with an invalid AoR. The
details can be as follows:
1 Test topology:
NUT(REG && PX) UA11 UA12
Hi all,
SIPv6 UA Can't receive a 500 (Server Error) response from openser register
server after sending REGISTER request with two Contact header field values. The
details can be as follows:
1 Test topology:
NUT(REG && PX) UA11 UA12
Hi all,
SIPv6 UA Can't receive a 420 (Bad Extension) response from
openser-1.3.4-12.fc13.i686 register server after sending Registrar with
Non-supported option-tag in Require filed. The details can be as follows:
1 Test topology:
NUT(REG && PX) UA11
Hi all,
SIPv6 UA Can't receive a 200 ok response from openser register server after
sending Forwarded REGISTER request. The details can be as follows:
1 Test topology:
NUT(REG && PX) UA11 UA12
DNS
Hi all,
Can't receive a 200 0k response from openser register server when URI in a To
header field in a REGISTER request contains escaped characters. The details can
be as follows:
1 Test topology:
NUT(REG && PX) UA11 UA12
Hello,
There is a new release for SIP SIMPLE client SDK available, version
0.15.0. This version incorporates fixes for the bugs discovered at
SIPIT26 and new features implemented since the previous release.
SIP SIMPLE client SDK is the main dependency of Blink QT due to be
released in July.
Hi all, I'm working for a small ITSP, we have been setup an openSIPs server,
now we want to develop a dialer for our
user, we have been tested the portsip sdk(www.portsip.com), it's working
fine with openSIPS so far, but we not doing long time test,
does anyone knows this product is working good wi
OpenSIPs is a SIP compliant SIP Proxy. It will work well with other entities
that are SIP compliant. :)
Maybe not the answer you are looking for, but in general SIP is SIP. Unless
it's not some weird proprietary variant of their own creation, it should
work well.
-Brett
On Tue, Jun 22, 2010 at 1
To get outgoing caller id to work I have to change my FROM header to a
trunk-specific format. While this is ugly, it works, unless I try to do it
more then once. Using gateway lists in drouting this scenario can happen as
the routing script is run for each gateway on failure.
Is there another way
Thank you Brett, then we will go on test.
On Tue, Jun 22, 2010 at 11:59 PM, Brett Nemeroff wrote:
> OpenSIPs is a SIP compliant SIP Proxy. It will work well with other
> entities that are SIP compliant. :)
>
> Maybe not the answer you are looking for, but in general SIP is SIP. Unless
> it's no
Let us know how it works for you. :)
Thanks,
Brett
On Tue, Jun 22, 2010 at 11:05 AM, list mail wrote:
> Thank you Brett, then we will go on test.
>
>
> On Tue, Jun 22, 2010 at 11:59 PM, Brett Nemeroff wrote:
>
>> OpenSIPs is a SIP compliant SIP Proxy. It will work well with other
>> entities th
Hi, it seems the modparam for module acc is overwriting the log_facility config
on opensips.cfg ?
/usr/local/etc/opensips/opensips.cfg
log_facility=LOG_LOCAL0
modparam("acc", "log_facility", "LOG_LOCAL1")
/etc/syslog.conf
local0.*-/var/log/opensip.log
local1.*
At least each time we are getting a different error, now we are seeing:
# opensipsdbctl create
[: 488: ==: unexpected operator
-e \E[37;33mINFO: test server charset
ERROR 1045 (28000): Access denied for user 'root'@'localhost' (using
password: NO)
ERROR 1045 (28000): Access denied for user 'root'@
Bogdan,
In my testing I found that this error still occurs in revision 6981, but only
if debug=4 or higher.
#0 0xb785c58b in build_rule (values=0x819acd4) at dp_db.c:382
382 LM_DBG("references:%d ,
max:%d\n",namecount,repl_comp->max_pmatch);
(gdb) bt
#0 0xb785c58b in build_rule (v
Hi Brett, thanks for your thoughts,
> You'll need to either:
> 1. Join the conferences together
I think thats impossible, because Asterisk is not able to handle something like
inter system communication (sharing channels on demand)
>
> 2. Have opensips track the creation of the conference room
You might get a better response if you kept your subject line shorter
and grouped all your messages into one.
AF.
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