Thanks to Klaus Darilion, QjSimple 0.6.3 works on most of desktop OS:
Linux(es), Windows and Mac OS X.
It is a very handy application, especially for developers, but not olny,
with simple UI and debugger window. A perfect replacement (for my very
old good Linux testing tool) KPhone since QjSimple has SIMPLE presence
support, tls ... more at:
http://www.ipcom.at/index.php?id=560/
Cheers,
Daniel
--
Daniel-Constantin Mierla
* SIP Router Bootcamp
* Kamailio (OpenSER) and Asterisk Training
* …
[View More]Berlin, Germany, Sep 1-4, 2009
* http://www.asipto.com/index.php/sip-router-bootcamp/
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Hello,
I installed two plugins for dokuwiki and flyspray bug tracker to
reference easier the git repository and bugs.
For FlySpray, when you describe or comment a bug you can reference to a
git commit by using: GIT#hash
- hash must be at list the first six characters of git commit hash value
Unfortunately FlySpray accepts only plain text in the "Close Task"
description, so the git reference does not work there, yet.
In dokuwiki same syntax can be used. In addition, you can add …
[View More]references
to FlySpray items via: FlySpray#itemid
- itemid is the integer id of FlySpray tracked item
I tried to collect all these plus other options for DW syntax in a wiki
page:
http://sip-router.org/wiki/tutorials/contribute-to-dokuwiki
Feel free to add more details there.
Cheers,
Daniel
--
Daniel-Constantin Mierla
* SIP Router Bootcamp
* Kamailio (OpenSER) and Asterisk Training
* Berlin, Germany, Sep 1-4, 2009
* http://www.asipto.com/index.php/sip-router-bootcamp/
[View Less]
A brief look at the IPTel pages show pretty much nothing but comment
spam for pages and pages. Does anyone even bother to maintain the pages
anymore?
I'm looking for more relevant documentation to SER 2.0 (or even the SIP
Router project). The docs in the doc subtree of the source for both of
these projects is outdated and no longer valid with respect to any
information stored in the database.
It still references subscriber tables and the like. All 0.9.X format DB
data.
Is there an update somewhere?
N.
Hi!
I have problem compiling the OSP-module in Kamailio 1.5.x
OS: Ubuntu Server 8.04LTS
Compiling and installing OSPToolkit is OK.
The problem occures when it comes to compiling the OSP-module in Kamailio:
make[1]: Entering directory `/usr/src/kamailio/modules/osp'
make[1]: Leaving directory `/usr/src/kamailio/modules/osp'
make[1]: Entering directory `/usr/src/kamailio/modules/osp'
Compiling destination.c
Compiling globals.c
Compiling orig_transaction.c
In file included from …
[View More]orig_transaction.c:40:
sipheader.h:48: error: expected declaration specifiers or '...' before 'OSPTCALLID'
orig_transaction.c: In function 'ospLoadRoutes':
orig_transaction.c:84: error: 'OSPE_DEST_PROT' undeclared (first use in this function)
orig_transaction.c:84: error: (Each undeclared identifier is reported only once
orig_transaction.c:84: error: for each function it appears in.)
orig_transaction.c:84: error: expected ';' before 'protocol'
orig_transaction.c:85: error: 'OSPE_DEST_OSP_ENABLED' undeclared (first use in this function)
orig_transaction.c:85: error: expected ';' before 'enabled'
orig_transaction.c:150: error: 'protocol' undeclared (first use in this function)
orig_transaction.c:154: error: 'OSPE_DEST_PROT_SIP' undeclared (first use in this function)
orig_transaction.c:157: error: 'OSPE_DEST_PROT_H323_LRQ' undeclared (first use in this function)
orig_transaction.c:158: error: 'OSPE_DEST_PROT_H323_SETUP' undeclared (first use in this function)
orig_transaction.c:159: error: 'OSPE_DEST_PROT_IAX' undeclared (first use in this function)
orig_transaction.c:163: error: 'OSPE_DEST_PROT_UNDEFINED' undeclared (first use in this function)
orig_transaction.c:164: error: 'OSPE_DEST_PROT_UNKNOWN' undeclared (first use in this function)
orig_transaction.c:170: error: 'enabled' undeclared (first use in this function)
orig_transaction.c:174: error: 'OSPE_OSP_FALSE' undeclared (first use in this function)
orig_transaction.c:189: error: 'OSPC_SOURCE' undeclared (first use in this function)
orig_transaction.c: In function 'ospRequestRouting':
orig_transaction.c:260: error: 'OSPTCALLID' undeclared (first use in this function)
orig_transaction.c:260: error: 'callids' undeclared (first use in this function)
orig_transaction.c:282: error: too many arguments to function 'ospGetCallId'
orig_transaction.c:325: error: 'OSPC_E164' undeclared (first use in this function)
make[1]: *** [orig_transaction.o] Error 1
make[1]: Leaving directory `/usr/src/kamailio/modules/osp'
make: *** [modules] Error 2
Has tried with both OSPToolkit 3.5.2 and 3.3.6.
And with both Kamailio 1.5.1 and 1.5.2
Same problem.
Does anyone know what is the problem?
Regards,
Martin
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Hi all
In kamailio 1.5.2, I used dlg_isflagset inside a ONREPLY_ROUTE.
////////////////scripts//////////////////
Route[1]{
...
Dlg_manage();
Dlg_setflag("1");
...
}
onreply_route[1] {
#dlg_get("$ci", "$ft", "$tt");
#xlog("L_INFO", " Dialog: $dlg(h_id) . $dlg(h_entry)\n");
if (dlg_isflagset("1")) {
xlog("L_INFO", " here is it \n");
}
}
/////////////////////////////////////////
The dlg_isflagset seems do not match anything until there is
dlg_get("…
[View More]$ci", "$ft", "$tt");
before it.
Questions are:
(1) Is it the correct usage or a pontential bug.
(2) How dialog is cheched/matched in the onreply_route internally?
Kind Regards
Min Wang
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Hi all,
could you please give me suggestions how to simultaneously send call to
PSTN destination for UAC that is logged in. So, when call comes for
sip:bob@example.com it must also go to its PSTN destination
1234(a)10.0.0.2 (avp in mysql) and ring both destinations simultaneously.
Example below doesn't work and I keep ending up with two INVITEs to
PSTN GW and proxy doesn't fork call to UAC. in this case, if I understand
correctly, this is proper behavior because append_branch() has its …
[View More]function only if UAC
isn't logged in or on 480,486... And how to handle this when user is logged in?
if (!lookup("location")) {
...
}
if(...) {
avp_pushto("$ru", "$avp(s:PSTN_fwd)");
append_branch();
}
Thanks a lot
/dubravko
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For those interested in building SIP and SIMPLE rich presence applications in
Ruby language:
http://xcapclient.rubyforge.org/http://dev.sipdoc.net/projects/ruby-xcapclient/wiki/
Ruby XCAPClient
--------------------------------------------------
A XCAP client library written in Ruby language.
XCAP (RFC 4825) is a protocol on top of HTTP which allows a client (usually a
VoIP/SIP device) to manipulate the contents of Presence Information Data
Format (PIDF) based presence documents. These …
[View More]documents are stored in a server
in XML format.
Ruby xcapclient library implements the XCAP protocol in client side, allowing
the applications to get, store, modify and delete XML documents in the server.
== Features
* Fetch, create/replace and delete a document.
* Fetch, create/replace and delete a document element (XML node).
* Fetch, create/replace and delete an element attribute.
* Full configurable parameters allowing customized fields for each XCAP
application, such auid, XML namespace, MIME-Type, scope (:user or :global) and
default document name ("index" if not set).
* Fetch the namespaces and prefixes of a document node as they are used in the
server.
* Manage of multiple documents per XCAP application.
* Fetch the XCAP server auids, extensions and namespaces ("xcap-caps"
application).
* SSL.
* Digest and Basic HTTP authentication.
* Raise custom Ruby exception for each HTTP error response.
--------------------------------------------------
Enjoy.
--
Iñaki Baz Castillo <ibc(a)aliax.net>
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Hello,
some time ago I created a new branch in GIT named 'daniel/xavp' that
includes code for a new pseudo-variable class 'xavp' - name being a
shortcut to 'extended attribute value pair'. I collected some design
details in wiki:
http://sip-router.org/wiki/devel/xavp
It is not a replacement for current AVP api implementation, although in
the future can replace it. In the past we had couple of discussions
about the limitations of avps which I tried to overcome. Here are some
highlights …
[View More]of xavp design:
- string names only - various tests showed that having the complex
addressing scheme with integer id and string name does not pay off
performance gains. XAVP computes and integer id for each name that is
used to speed up comparison
- possibility to store more data types - AVP allow string and integer
data types. More can be handled by XAVP, including a generic data type
where you can build your own structure and store it in the list. This is
good for example to store custom structures for transaction lifetine --
right now dialog module needs to store reference to dialog structure.
Optimizations can be done for any case of serial forking -- uri, dst
uri, q, etc can be stored in a structure without a need to parse and
build it from an encoded value stored in one or many avps
- possibility to group XAVPs inside another XAVP - practically is
building lists of XAVPs. This should overcome issues with naming
conflicts between modules - e.g., dispatacher module can group its avps
in a list named "dispatacher", lcr module in a list named 'lcrl, then
there can be XAVP with name 'uri' in both lists, without the risk of
misusage. Another benefit would be reducing the number of parameters,
right now each module exports a parameter to set the name for each AVP,
with xavp there is need for one parameter to set the name of the list,
then inside that list the module is free to use predefined names without
the risk of overlapping.
- xavp having value integer or string can be used directly in config,
whether they are in a sublist or not
Check the wiki (link above) for more. Looking forward for comments,
suggestions and improvements.
Cheers,
Daniel
--
Daniel-Constantin Mierla
* SIP Router Bootcamp
* Kamailio (OpenSER) and Asterisk Training
* Berlin, Germany, Sep 1-4, 2009
* http://www.asipto.com/index.php/sip-router-bootcamp/
[View Less]
Hello,
the new sip router core and librarization of kamailio management (MI)
and statistics interfaces affected the module interface. At this time,
although all the fields are still in the module interface, some are passive:
- mi commands exports
- statistics exports
- extra processes
The modules in the public GIT repository use the either mod_init or
mod_register to export them, so everything should work as expected.
fyi: if you didn't know so far - mod_register is a function
…
[View More]automatically called when a module is loaded, giving the possibility to
export components from module in very early stage, e.g., transfomations
must be exported via mod_register since they are lookep up while parsing
the routing blocks -- mod_register is available in kamailio since 1.5,
back ported from sip router core, you just need to implement it in the
module, it is not part of module interface.
Back to module interface, the three fields mislead private module
implementations as everything compiles ok, but at startup the fields are
not actually exported. I propose to remove them and document the
transition process to new interface, which is practically couple of code
lines. It this way people migrating private stuff will notice the change
and update.
Since we are here and we have to go through each K module and remove the
fields from the module interface, maybe we can create a single interface
for k and s modules. Doing a short analysis and considering the above
three fields removed from k interface, then:
- K interface has extra
- dlopen flags - needed to specify how the .so is open, required so
far by perl module, iirc
- PV exports - the list of pseudo-variables exported by module
- the cmd (function) exports structure has a field to specify a free
fixup function (not implemented for all exported commands). The goal is
to allow calling functions exported by modules for higher application
languages, like perl so far, where it is needed to free the memory
allocated by fixup function
- S interface has extra
- rpc exports - exported rpc commands
- onbreak function - is this used?!? couldn't find docs about or a
use case, so I consider it obsolete
Then common interface would be:
- name
- dlopen flags
- mod parameters
- mod commands with free fixup
- rpc commands
- PV exports
- mod init
- child init
- mod destroy
- onreply callback
Probably we can normalize it a bit more, comments and suggestions are
welcome!
Thanks,
Daniel
--
Daniel-Constantin Mierla
* SIP Router Bootcamp
* Kamailio (OpenSER) and Asterisk Training
* Berlin, Germany, Sep 1-4, 2009
* http://www.asipto.com/index.php/sip-router-bootcamp/
[View Less]
hi
I use pa module to support SUBSCRIBE request which Event header is "presence",
and the function I used in ser.cfg is as follow :
if (method=="SUBSCRIBE") {
if (!t_newtran()) {
log(1, "newtran error\n");
sl_reply_error();
};
handle_subscription("registrar");
break;
};
But the response of this request is 400 error while parsing header Server: …
[View More]SIP Express Route (0.9.6(i386/linux))
I find there is no place in source code to parse Server header
I am I miss something in ser.cfg or any other modules should be loaded ?
Could any one tell me why?
___________________________________________________________
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