Guys,Sure posing spam is a problem. That's a wiki's life. Do we really need
a voting system for posting information about developers?
I mean, it sounds neat and all, but that almost seems like it requires a
forum of it's own.. Which I think is a great idea.
-Brett
On Thu, Feb 12, 2009 at 12:46
Would it be acceptable to post a link to a company that uses OpenSIPS at
the core of its architecture?
The concern is that our services focus on the delivery of VoIP rather
than OpenSER related software or support and i'm not sure if your
intentions include companies in this category
Adrian Ge
Hi Ibrahim,
For the second leg (from DID to FWD number) the problem seams to be
setting the destination part, because of the multi destination. The
error is you push all the destinations into dst_leg AVP - note that AVPs
are per transaction and not per branch!
What you can do is to set the dst
Zahid,
Thank you for your reply.
I tried if (!lookup("location")) before the uri matching, but no change.
Unfortunately don't know how to setup xlog.
#from an internal domain -> inbound
#Native SIP destinations are handled using the location table
#Gateway destinations are handl
Hi Gonzalo,
Is lookup("location") being called before the uri matching or after?
Looking at the code snippet, If local callers are dialing a pattern that
matches one of the "if (uri=" Then it is sent to route(4) and
lookup part of the code is not touched.
You may want to add "x
What I have noticed is that open wiki pages, polling and voting
mechanisms tend to be easily polluted or misused. What can be faked
with much more effort are:
1. Links to customer PR sites
2. Software that advertises that it uses or complements OpenSIPS
3. Third party blogs
Practically anyth
It can, in some cases, create some interesting catch-22 issues, though.
On Thu, 12 Feb 2009 20:33:58 +0200, Bogdan-Andrei Iancu
wrote:
> Hi Adrian,
>
> I agree that something like that will be good, but if you have an idea
> about how to implement it. I'm ready for suggestions :).
>
> R
Each user has a DID number and the user name match the DID
umber, so on incoming calls to a DID if the user is online it ring that ext.
and if is offline it go to voice mail. Mt issue is when a user call another
user that is online it doesn’t go to that user directly even though shows in
the lo
Hi Adrian,
I agree that something like that will be good, but if you have an idea
about how to implement it. I'm ready for suggestions :).
Regards,
Bogdan
Adrian Georgescu wrote:
> Bogdan
>
> I know that is a very hairy subject and we have precedents behind us
> that we can learn from.
>
>
On Feb 12, 2009, at 5:53 PM, Brian Chamberlain wrote:
Hi Adrian,
I got my issue resolved with the rating not appearing. 2 Questions
for you:
Two? :-)
Say a Provider operating and the normal rules are 0 for local 00 for
international and the country code is 44
How should I write phon
Hello to all members.
A new feature has been introduced in nathelper module. Based on an idea
published by one of the developers, and with a complete rework made by Anca
Vamanu, dynamic reloading of RTP proxies list for nathelper module has been
implemented.
This feature allows to provision proxi
Can you register two sip accounts on http://sip2sip.info
Then make a T.38 fax call and see if it works. If it does not, let me
know the Call Id, there I have access to traces and we can find out
the cause of your problem.
Adrian
On Feb 12, 2009, at 3:44 PM, Carlo Dimaggio wrote:
Il gior
Hi Adrian,
I got my issue resolved with the rating not appearing. 2 Questions for
you:
Say a Provider operating and the normal rules are 0 for local 00 for
international and the country code is 44
How should I write phone number dialled for a special rate number for
example a fixed rate
Hm, you must have provisioned something wrongly. Min duration as far
as I remember has to do with the amount of seconds a call should have
to be billed. Please reduce that value to 0 see if this solves it.
Adrian
On Feb 12, 2009, at 3:23 PM, Brian Chamberlain wrote:
Hi,
My apologies I in
Bogdan
I know that is a very hairy subject and we have precedents behind us
that we can learn from.
An open Wiki page will likely yield, given enough time, on
opensips.org the N-th directory listing of 'I am also here doing some
xSER stuff'
I would rather make a page where relevant cons
> From: users-boun...@lists.opensips.org [mailto:users-
>
> 2009/2/12 John Rose :
> > I'm on SER, Kamalilo and OpenSIPS mailing lists. This is by far the
> most
> > lively list. Kamalilo list is dead as far as I can see...
>
> Kamailio list is really far from being dead. It has more or less the
>
Geoffrey,You shouldn't be put off by everyone's responses. Keep in mind that
all of us out here took time to respond to you. Which should be an indicator
of the willingness of everyone to help. Not everyone will have a
gentle demeanor, but the basic point is that what you were asking for
suggested
2009/2/12 John Rose :
> I'm on SER, Kamalilo and OpenSIPS mailing lists. This is by far the most
> lively list. Kamalilo list is dead as far as I can see...
Kamailio list is really far from being dead. It has more or less the
same traffic as OpenSIPS list. Both are alive.
--
Iñaki Baz Castillo
I'm on SER, Kamalilo and OpenSIPS mailing lists. This is by far the most
lively list. Kamalilo list is dead as far as I can see... Also, if you ask a
question on the SER list you often get dead air though that list is good for
historical quantity and quality.
John
> From: users-boun...@lists.ope
Hi Bogdan,
I have to use multileg accounting because my application records all
request leg to acc. I use mentioned issue for this scenario;
1. DID call come to my opensips and it add to multileg.
$avp(s:src_leg)=$fU; $avp(s:dst_leg)=$rU;
2. The user owner of the DID was set callforwarding to a f
Hi Geoffrey,
I'm sorry for the outcome, but I think you are missing the most
important aspect here:
nobody was flaming or reluctant to offer you support - but reviewing a
production script from 0 in 2 hours it is fantasy job. That was what all
the people tried to explain you.
I, as a consulta
Fair enough.
Thanks for putting it together!
-Brett
On Thu, Feb 12, 2009 at 9:05 AM, Bogdan-Andrei Iancu wrote:
> Brett,
>
> The idea is to prevent that page is becoming a place for marketing (where
> people do present and describe their products and services).
>
> To help a bit, I created the
Well, just figured I would provide an update. As I was very put off
by the general attitude of the community as it related to my request
for commercial support and assistance during my deployment phase I
started looking elsewhere.
Someone on this list suggested I contact the author of the dispatc
Brett,
The idea is to prevent that page is becoming a place for marketing
(where people do present and describe their products and services).
To help a bit, I created the "section" to be easier to see what a
company/person has to offer - at least as area. If you want more info,
than you have
Hi ibrahim,
For paralel forking you do not need multi-leg accounting. TM will
select the wining branch (the one that picked up) and acc will use for
accounting the wining branch. So you do not have to do anything about this.
Regards,
Bogdan
ibrahim tunali wrote:
> Hello,
>
> I am dealing with
Il giorno 11/feb/09, alle ore 12:00, Ruud Klaver ha scritto:
Hi Carlo,
The problem definitely lies with OpenSIPS in your first example, the
one that didn't work, since the media-dispatcher does not receive
anything from the mediaproxy module in OpenSIPs when the re-INVITE
is received. Co
Hello,
I am dealing with branch routing and multiple leg accounting. I would like
to forward call to several PSTN destinations. I want to account the
certainly which one get the call
I set destination leg avp with below;
branch_route[1]{
$avp(s:dst_leg)=$rU;
}
The Avp has a few values with
Hi,
My apologies I investigate further I see this is for debugging reasons
I assume:
Duration: 120 s
App: audio
Destination: 3531
Customer: subscriber=1...@sw1.dub.asterisk.ie
Increment: 60 s
Min duration: 300 s
My price is showing as 0 after renormalization. I have
Hi Mauro,
yes, it is true - only the faulty server had the info about the dialog
you want to end.
Regards,
Bogdan
Mauro Davi' wrote:
>
> Hi All,
>
> I’m setting up an opensips architecture with several opensips server.
> Some of this acts as Proxy same as UAS same as registrar.
>
> Every serve
Bogdan,Hey, great idea. However, I'm concerned that we can't list any
information. I understand why you say this, it certainly it's supposed to be
a free billboard. But how would a user know what to click on?
I think maybe we should allow a few bullet points to name some specialities.
I know, once
Hi ,
I installed the software dependencies regarding the openxcap but when I am
trying to run the openxcap server it is throwing the following error.
/opt/python2.5/bin/openxcap start
Traceback (most recent call last):
File "/opt/python2.5/bin/openxcap", line 41, in
from xcap.logutil impor
Iñaki Baz Castillo wrote:
> 2009/2/12 Bogdan-Andrei Iancu :
>
>> Hi Iñaki
>>
>> Iñaki Baz Castillo wrote:
>>
>>> 2009/2/12 Bogdan-Andrei Iancu :
>>>
>>>
2) a web page for listing business entities that may help you in
whatever OpenSIPS related issue. Keep in mind, this page
2009/2/12 Bogdan-Andrei Iancu :
> Hi Iñaki
>
> Iñaki Baz Castillo wrote:
>>
>> 2009/2/12 Bogdan-Andrei Iancu :
>>
>>>
>>> 2) a web page for listing business entities that may help you in
>>> whatever OpenSIPS related issue. Keep in mind, this page is just a
>>> listing (per sections) and not a plac
Hi All,
I'm setting up an opensips architecture with several opensips server.
Some of this acts as Proxy same as UAS same as registrar.
Every server works on a set of dialogs, if I want end a dialog simply
sent to it an XML/RPC command with the appropriate identifier and two
BYE are sent (one
Hi All,
I have copied one of my radacct tables into radacct200807 to test my
billing and rates. It is a table from January. When I renormalise for
a day I don't get any errors for the customer that I have added, I
have them pointing at a correct profile which in turn points at
correct rat
Hi Iñaki
Iñaki Baz Castillo wrote:
> 2009/2/12 Bogdan-Andrei Iancu :
>
>> 2) a web page for listing business entities that may help you in
>> whatever OpenSIPS related issue. Keep in mind, this page is just a
>> listing (per sections) and not a place for publicity or service/product
>> descript
2009/2/12 Bogdan-Andrei Iancu :
> 2) a web page for listing business entities that may help you in
> whatever OpenSIPS related issue. Keep in mind, this page is just a
> listing (per sections) and not a place for publicity or service/product
> descriptions. The idea is simple: if you do business ar
Hi,
There is a new functionality available for dispatcher module on 1.5 -
you can define (as module parameter) your custom set of SIP reply codes
to be considered as OK for probing:
http://www.opensips.org/html/docs/modules/devel/dispatcher.html#id271069
This is a new enhancement Anca did c
Hi everybody,
Norman Brandinger wrote:
> The old OpenSER/Kamailio site has a "business" mailing list that is
> dedicated to this type of topic. Seems like a good idea to create a
> similar mailing list for the OpenSIPS folks.
>
Just as a completion - Norm, thanks for your suggestion - I did
Hi,
I guess the yesterday discussion touched a hot topic - how to get
business work around OpenSIPS. By just looking at the number of messages
in the thread
(http://lists.opensips.org/pipermail/users/2009-February/003086.html)
and a lot of passion in the debate.
So, I consider this topic must
Hello Olle,
OpenSER/OpenSIPS never targeted the windows based platform. But this
does not mean that it is again. I heard about people using cygwin with
partial results.
REgards,
Bogdan
Olle Frimanson wrote:
> Hi,
>
> we need to run both a SIP client and a SIP server on a embedded
> systems
Hi Ashwini,
currently there are no packages for wesip so you have to compile and
install everything by yourself. You can follow the instructions at www.wesip.eu
(http://www.wesip.com/mediawiki/index.php/Installation and
http://www.wesip.com/mediawiki/index.php/Configuration)
Regards
i am trying to configure cdrtool , gui runs ok,
but when i start ratingEngine i am getting warning
PHP Warning: pcntl_signal(): Error assigning signal in
/usr/local/www/CDRTool/library/rating_server.php on line 100
Warning: pcntl_signal(): Error assigning signal in
/usr/local/www/CDRTool/library/
Hi all,
Sorry for budging in again.
And as we all know, we
all have lots of spare time. So I suggest even a voting system where the
mostly liked comment sticks on the top... J
And to dip my toes in to really shark infested waters;
mentioning dollars...
Is there any way contributing to th
Ok, this may be a silly question... but.. do you have any OpenSIPs in there?
:)
On Thu, Feb 12, 2009 at 2:07 AM, Amit Bansal wrote:
> I am using Salfin server as a SIP container, NetBeans as IDE, Java language
> and X-Lite as softphone.
> http://wiki.glassfish.java.net/Wiki.jsp?page=SipClickToD
I am using Salfin server as a SIP container, NetBeans as IDE, Java language
and X-Lite as softphone.
http://wiki.glassfish.java.net/Wiki.jsp?page=SipClickToDialExample2
I am also tried to deploy this sample application (above link) but facing
the same problem.
Plz help me to overcome this probl
I am using Salfin server as a SIP container, NetBeans as IDE, Java language
and X-Lite as softphone.
http://wiki.glassfish.java.net/Wiki.jsp?page=SipClickToDialExample2
I am also tried to deploy this sample application (above link) but facing
the same problem.
Plz help me to overcome this proble
Ultimately, the comments would be tacked onto the docs. I imagine it just
how it is today,with a comments field at the bottom, you post and it just
rolls the comment downwards..
On Thu, Feb 12, 2009 at 1:58 AM, Bogdan-Andrei Iancu wrote:
> Brett,
>
> I have to agree that from web perspective th
According to RFC 3261, the defining elements for a dialog are Callid,
To-tag and From-tag. TO-uri and From-uri are obsolete .
Regards,
Bogdan
Brett Nemeroff wrote:
> I was a little confused. I was trying to explain to the operator that
> the reply doesn't have a RURI and that I send it with the
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