Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 02.10.2013 12:02, Iñaki Baz Castillo wrote:
Hi,
I have some users in OverSIP maillist complaining because the
documentation about OpenSIPS+OverSIP does not work:
http://www.opensips.org/Documentation/Tutorials-WebSocket
be about adding just 4 lines
:)
Thanks a lot.
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of the path column is
required.
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if it already works ok (but
just increasing the path column size to be able to store two URIs).
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OverSIP (since the non top Via header, that created by the
SIP WebSocket client, has WS as transport protocol).
And nothing else at all, but the above two points are important.
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support in OpenSIPS.
In a large deployment run N OverSIP proxies in front of your OpenSIPS
registrar. Just it.
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off of same DB.
The purpose of a SIP Edge Proxy (check RFC 5626) is very simple. No need
for DB access typically or for special configuration. Yes, having all
together in a single box may seem easier. Well, wait a bit then.
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2011/8/9 Bogdan-Andrei Iancu bog...@opensips.org:
Hi Nethra,
what RCS stands for ??
RCS is like the adoption of IMS for mobile devices, the industry and so...
http://en.wikipedia.org/wiki/Rich_Communication_Suite
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are partially shown?
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but a new
deployment should move to the standard: P-Asserted-Identity.
For more information about how all this stuf works, please read RFC
3325 in which *every* detail is fully explained.
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Users
.
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a request with *any* domain/IP)
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the session must be
ended, but for any other error it shouldn't be AFAIK.
You are right. RFC 5057 demostrates it (488 must only affect to its
transaction).
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be just higher than previous one, no matter how much higher).
But in your case, the problem is that when your UA receives the 488 it
will probably send a BYE (it's the usual policy when a re-INVITE is
rejected). But you can try it. You will not break SIP protocol at all.
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i
produce a loop at all, so it must be an error in your
conf.
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it to access just because
the From domain is the same as the IP of your Asterisk?
Use tables and functions in permissions module, just it.
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/SIP-router.
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response. If a worker fails to start when
running the chil init function, doesn't the parent (main process)
realize of it and exit?
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) to properly return
with failure code, but I must check it.
Best regards.
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procs) without getting any feedback on the init success
of its child processes.
Ok :)
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, so be careful.
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action. But it's
really a dirty hack. A daemonized process should not exit with 0 if
finally it fails to start.
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:
- Does the BYE arrive to Asterisk? (sure it does).
- Does Asterisk replies to the BYE? (maybe Asterisk doesn't implement
ending a call with BYE during early-dialog).
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*
pcapsipdump.cpp:}else if (htons(header_udp-source)==5060||
pcapsipdump.cpp:htons(header_udp-dest)==5060){
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if they do exist. not useful for me.
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2010/12/2 Erik Dekkers erik.dekk...@wvds.nl:
Would it be possible to run sipak from the opensips.cfg script? I'm also
looking for something like this.
Why do you need it?? Just run sipsak as an external process generating
a REGISTER in behalf of your proxy.
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i
inspect just the transaction identifier (Via's branch and CSeq).
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point me to some hints describing the proper way to
build transaction inspection?
It's well explained in RFC 3261.
An ACK for a [3456]XX response must have same branch and same CSeq
number (but ACK method) as the INVITE of the transaction.
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such PUT will be rejected, but the existing document (containing
external references) will remain *active*.
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where that is stated please?
I'm sorry but I can't remember it. But for sure I've read it in some
OMA spec. In fact you won't find 'entry-ref' in any XCAP document
defined by OMA.
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. Just external-ref is allowed.
entry-ref just exists in IETF's incomplete specs. What does OpenXCAP
implement exactly?
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should never relay a 503
response, but instead must transform it into a 500.
:)
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loading contacts based on registration
;q parameter? LCR provides this feature by using load_contacts() and
next_contacts().
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to
use it as a good starting point for customization.
Thanks a lot, it looks very well.
However I've already started creating my own application for
displaying cool SIP flows in HTML. Also note that in my case I don't
want it to display a real SIP capture.
Thanks a lot.
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receiving
port??
2) Why do you want something like that?
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408, 503 depending
on UA1 failover implementation) to UA1.
- UA1 receives such response (i.e 408) and reroutes the call.
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:
---
I think i was very clear in the begining, I addressed my questions to Bogdan as
a novice and not for profit, so please keep it quiet, i never asked and will
never ask for your help
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proxy then Contact uri
will be, obviously, modified in each proxy. Just don't do it.
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2010/8/30 S. Millard mga...@yahoo.fr:
The configuration file of OpenSIPS is not a XML file so such question
makes no sense. Neither asking for a FreeSwitch configuration file in
a OpenSIPS maillist.
Iñaki Baz Castillo,
I know I got mixed up because i stated in the begining that I am
in most of the countries).
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file so such question
makes no sense. Neither asking for a FreeSwitch configuration file in
a OpenSIPS maillist.
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perhaps in those cases the phone also make calls using a TEL
URI in the From, which would make the RFC 4474 unuseful.
Would there be other issues against the use of this RFC for this very purpose?
IMHO it's the best solution.
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for putting
a call on hold is no longer recommended, since it doesn't allow for
RTCP to be used with held streams, doesn't work with IPv6, and breaks
with connection oriented media.
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, it is placed on hold by marking it as sendonly. If the
stream to be placed on hold was previously a recvonly media stream,
it is placed on hold by marking it inactive.
So both a=sendonly and a=inactive should be considered.
Regards.
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to just use terminal to run SIPp.
Because I think webFronted SIPp is not just run,
but it needs to configure more.
I don't trust web interfaces for such kind of operations :)
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, 2)) {...
This had me confused for a while - it happens on trunk and on the 1.6 branch.
You are totally right, it's a bug in the doc. I've fixed it right now
in the trunk version.
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opensips using sipp.
Thanks before Inaki.
You are specially rude.
I just suggested you to ask for this kind of issue in the SIPp
maillist, as it makes MORE sense than asking it here.
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2010/6/30 Agung agoeng...@gmail.com:
Iñaki Baz Castillo i...@... writes:
So, do you have a segfault in OpenSips or in SIPp?
I got segfault on SIPp.
Then it's not a problem in OpenSIPS but in SIPp. I suggest you to
asking this question in SIPp maillist as it's not a OpenSIPS problem
to the
transaction.
t_relay() creates a transaction for the first time the INVITE arrives
so a retranmission would be absorbed by t_checktran() at the
beggining of the script.
I strongly recommend you to remove t_newtrans() and try.
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2010/6/29 Agung agoeng...@gmail.com:
Is that true??? I am really curious because it say that it have segmentation
fault, so I am suspicious that my config at server isn't right.
So, do you have a segfault in OpenSips or in SIPp?
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information about which responses are
being retransmitted.
Also, note that 200 OK is a end-to-end response rather than a
hop-by-hop response (like [3456]XX responses) so the proxy MUST NOT
retransmit it. It's a task of the UAS the retransmission of the 200
until it receives the ACK.
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of the SIP method, 100rel is the name of the whole
extension including the mechanism to negociate the usage (or not) of
PRACK method to confirm provisional responses.
http://tools.ietf.org/html/rfc3262
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such value?
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not the
first question about same subject.
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the subscriptions from
clients.
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code your own custom module in C for
OpenSIPS. Other choices would be using the Perl/Python module or the
SEAS module (but for this you need a server speaking SEAS protocol).
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phone and Eyebeam, right? (high ironic mode
on). When it works (this is: within the following 8 years) please tell
it to me, for sure SIMPLE/XCAP will replace XMPP in most of the
IM/presence deployments.
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1 5060 os1.phonetest.psu.edu.
_sip._udp.phonetest.psu.edu has SRV record 20 1 5060 os2.phonetest.psu.edu.
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2010/6/1 samoh dahmani.s...@gmail.com:
Hi,
I see that no one can help me ?!
Oh my god, nobody helps you! Have you paid something? If not, then you
should improve your question as after reading your mail I have no idea
of what you want to achieve.
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URI, so $ru $rU $rd $rp
b) changing the real destination URI, so $du $dU $dd $dp.
The To header is NOT modified as a proxy/gateway just takes into
account the RURI.
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, and this requirement is not
related to routing the request, the routing is working perfectly.
Then check out if the module modified the RURI or the real destination
($du) and use the resulted uri to replace the To header (textops).
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provided by Klaus is a good example of how SIMPLE
specs have failed. Does somebody doubt it? would the same annoying
issue occur with XMPP? (quick answer: no).
Sorry for being so realistic.
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2010/5/26 Klaus Darilion klaus.mailingli...@pernau.at:
Am 26.05.2010 10:33, schrieb Iñaki Baz Castillo:
Also remember: according to IETF's SIMPLE group brains, if you have a
working e-mail address (can receive mails) then your SIP presentity
should display you as available via mail (WTF
), this is, two
subnets, the the result in unexpected as the module just take the
first subnet (mask 32) for a given ip_addr. Confirmed by examining
the module code.
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2010/5/19 Iñaki Baz Castillo i...@aliax.net:
Commercial clients are available, Zoiper now implements full support for
XCAP and Publish method. CounterPath latest build is working finally against
OpenXCAP. Latest Snom software release 8 stores the contacts in XCAP server.
Yes, but I'm sure
that such XCAP doesn't work (interoperability issues at
the first try) = welcome to reality.
I strongly prefer to complain endlessly rather than expecting that all
this stuff will work some day.
But, of course, I will try to provide an alternative.
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2010/5/17 Bogdan-Andrei Iancu bog...@voice-system.ro:
I'm really interested to see the outcome of your specs - my question is
(assuming that from tech point of view, it is a good approach), how do
you foresee the wide spreading of the specs ?
Well, in the world I live SIMPLE just doesn't
as long as TLS is not mandatory.
Sure. However SIP over TCP/TLS is already specificied. Adding this
sane constrain to a new SIP based presence specification would be
acceptable IMHO.
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... and then you'll be able
to ask for specific questions and we'll be glad to help with them.
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already
tried to help you.
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2010/5/15 Aditya Kumar adityakumar...@yahoo.com:
Hi all,
I am having issue running the example script.
When creating a new topic please create a *new* mail instead of
pressing reply over any other thread.
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on groups...
nothing. And the reason is that it's not possible to make effective
usage of such painful specifications.
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hack over IETF *incomplete*
specifications.
Sorry for sounding so rude :)
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why this would happen??
Please, do a correct captures. Use -W byline option with ngrep command.
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2010/4/23 RAJNIKANT VANZA rajniva...@gmail.com:
Hi Chris,
You want to try for 2 opensips + 1 MySQL database.
As per my knowledge u can write listen = 0.0.0.0 ipaddress in both
opensip.crf file and try it.
I have configured for 2 kamailio + 1 MySQL database scenaris and its work
status.
Also, if I set offline status (other not to see me online) and I do
a call, will then my watchers receive a NOTIFY with status on the
phone?
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for it. This is what we have to deal with.
Hopefully SIMPLE presence specifications will dead soon and a new and
better specification will born ;)
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. Thanks.
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will see the
presence state that the buddy set in his client.
Hi, this feature sounds really reasonable for the real world in which we live :)
Could I see how such NOTIFY (mixing presence and dialog information)
looks? (just an example).
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please paste an example of such REGISTER?
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presentities are mixed/grouped with no real
criterion.
- A hypercomplex mechanism to publish a photo, mixing different protocols.
- And better if we don't speak about RLS and so...
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of the architecture, but the features
would remain so limited...)
So IMHO it's time to drop SIMPLE forever. IETF-SIMPLE has failed,
period, there is no need to continue adding more and more
specifications on top of a bad designed framework.
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.
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2010/3/10 Bogdan-Andrei Iancu bog...@voice-system.ro:
I just added flag 0x08 for Reason hdr propagation - both in trunk and
1.6 branch. It will part of the 1.6.2 release.
Already shown the commit, great ;)
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for subsequent notifications?
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the NATted client and the presence server. I suspect the topology is
as follows:
client --- Router-NAT (SIP TCP) Proxy (SIP UDP)
Presence-Server
Is it correct?
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2010/3/9 Patrice Dalle patda...@gmail.com:
According to RFC5262 it is legal to have a pidf-full XML element within an
application/pidf-diff+xml payload.
Is it a bug or it is simply not supported?
Not supported.
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and the
presence server you must add ;transport=tcp to $du (better than
rewriting $ru).
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explain the problem?
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mismatch will be observed and the Notify should be sent on
tcp as specified in the next hop.
Ok, thanks a lot.
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be better if you send such OPTIONS to the AoR so OpenSIPS would exec
lookup('location') and route the OPTIONS thought the appropriate socket.
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also append such parameter in the RURI when
processing the request in OpenSIPS, so it would be sent via TCP.
This is how SIP works.
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Iñaki Baz Castillo i...@aliax.net
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so you think it will work like this?
Un a regular expression to store your desired From username intoa a pseudo-
variable or AVP, and then use uac_replace_from with such pseudo-variable.
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Iñaki Baz Castillo i...@aliax.net
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El Jueves, 18 de Febrero de 2010, Кузьмицкий Александр escribió:
My understanting of script variables in opensips are middle.
I`m programmer with some skills in python, but opensips is very another
language.
Basic things i understand.
Interesting, and the question is???
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Iñaki Baz
Asterisk) and phones.
About missed call notifications, it's not so clear. Due to the lack of a
standard for this purpose, the most common approach is using a web interface
to show the missed calls (and commonly allow a click2dial system).
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Iñaki Baz Castillo i...@aliax.net
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