Hi,
I was wondering, when a SIP user publishes his presence to the openser
presence module, is this presence data cached in memory by the server or
is this immediately stored in the database?
Is this somehow configurable?
If the presence data is cached in memory, is there an interface
available for an external server to query the presence of one or more
resources?
In particular, we are looking for a performant way to query the presence
of multiple users (for example 20). We would like to avoid the overhead
of creating a SIP subscription for each resource:
- to avoid traffic overhead
- we only need to know the current presence of the resource(s) and are
not interested in future presence changes.
Do you have any suggestions on if/how this can be achieved with openser?
kind regards,
Sigrid Thijs
Would appreciate some help with CarrierRoute module
I get the following error
ERROR:carrierroute:carrier_rewrite_msg: error during rewrite_uri_recursor
This is my Route
route[22]
{
# route calls based on hash over callid
# choose route domain 0 of the default carrier
if(!cr_rewrite_uri("0", "call_id")){
sl_send_reply("403", "Not allowed");
} else {
# In case of failure, re-route the request
t_on_failure("3");
# Relay the request to the gateway
t_relay();
}
}
--
TC
I have a scenario in which an OpenSER machine distributes load among
several Asterisk machines for outgoing PSTN. I use an external program
which I call through exec_dset() to select which server a call goes to.
I use record_route().
All calls are completed OK, but when I dial from some SIP user-agents
and hang up from the caller side, OpenSER gets crazy and doesn't know
where to route the BYE to. Here's the reason:
Good scenario:
- I make a call to <5551234@mydomain>
- OpenSER calls exec_dset, which turns it into
<providercode#5551234@chosenserver>
- Call is answered
- I hang up
- My UA sends a BYE like this: "BYE
sip:providercode#5551234@chosenserver SIP/2.0"
- I get an OK back, all is well.
(Sometimes the # in the uri above is sent as %23, but it works either way)
Bad scenario:
- I make a call to <5551234@mydomain>
- OpenSER calls exec_dset, which turns it into
<providercode#5551234@chosenserver>
- Call is answered
- I hang up
- My UA sends a BYE like this: "BYE sip:5551234@mydomain SIP/2.0"
- My openser.cfg doesn't know how to handle that and I get a "Loop
Detected" back. The destination never gets a hangup signal.
Is the second UA from the "bad" scenario disrespecting the RFC? Do I
have any recourse to route a BYE that comes like that to the right
Asterisk server?
Thanks,
Juan
The OpenXCAP 0.9.7(version 0.9.7) allows the HTTP URI that is not percent-encoded while the
percent-encoded URI is rejected by the server. There is no problem of installation. Two types of
http URIs are shown below:
Accept: http://xcap-root/resource-lists/users/sip:alice@example.com/index/~~/resour…"friends"]
Reject: http://xcap-root/resource-lists/users/sip:alice@example.com/index/~~/resour…
Another problem is about fetching element that within multiple-folded elements, for example:
<list name="friend">
<entry uri="sip:alice@example.com">
<display-name>Alice</display-name>
</entry>
<entry uri="sip:bob@example.com">
<display-name>Bob</display-name>
</entry>
</list>
<list name="closefriends">
<entry uri="sip:tom@example.com">
<display-name>Tom</display-name>
</entry>
<entry uri="sip:mike@example.com">
<display-name>Mike</display-name>
</entry>
</list>
If fetching the element of "mike", the responsed to the following URI is "404 Not found".
http://xcap-root/resource-lists/users/sip:alice@example.com/index/~~/resour…
However, an alternative way of uing element sequence number works.
http://xcap-root/resource-lists/users/sip:alice@example.com/index/~~/resour…
Hope anyone can give me some clues, thanks a billion
Regards,
Kevin
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>
> > Hello all,
> >
> > I've got a working conf on which RTPs work perfectly with TCP, but
> > fails to work with UDP... can anyont tell me why??
>
> Could you detail more what the issue is? Do you mean that the SIP works
> fine
> over TCP and fails over UDP? RTP has nothing to do here.
>
>
> --
> I?aki Baz Castillo
>
Hello,
That's exactly what I mean, signalling being TCP or UPD got nothing to
do here.... but it DOES work on TCP and NOT over UDP!
Weird...
David
yes,
when i used same authenticating on both regiest and invite.the problem was resolved.many many thx.
======= 2008-01-08 15:16:35 您在来信中写道:=======
>
>
>Alex Balashov schrieb:
>> On Tue, 8 Jan 2008, liu york wrote:
>>
>>> I want to restrict invite when regiest unsuccessful even the invite
>>> request have right authentication. actually,i just want to only one
>>> person can call to pstn,others cant call even have sipid and
>>> password.now i has set by modparam("registrar", "max_contacts", 10).but
>>> cant restrict the invite by the sipphone no need regiestion.
>>
>> Are you authenticating INVITEs differently than you are authenticating
>> registrations? In other words, are the authentication realm and required
>> credentials different for your proxy_challenge() than they are for
>> your www_challenge() for handling REGISTER requests?
>>
>> If they are, it seems that all you should have to do is do a lookup()
>> and see if it fails. Presumably, you are doing a save() into your
>> registrar/usrloc only on valid REGISTER challenges?
>>
>>
>> i.e.
>>
>> if(is_method("INVITE")) {
>> if(!lookup("location")) {
>
> if(!registered("location")) {
>is sufficient
>
>btw: this does not give you any security
>
>regards
>klaus
>
>
>> sl_send_reply("403", "Forbidden -- REGISTER invalid.");
>> exit;
>> }
>> }
>>
>>
>> --
>> Alex Balashov
>> Evariste Systems
>> Web : http://www.evaristesys.com/
>> Tel : +1-678-954-0670
>> Direct : +1-678-954-0671
>>
>> _______________________________________________
>> Users mailing list
>> Users(a)lists.openser.org
>> http://lists.openser.org/cgi-bin/mailman/listinfo/users
= = = = = = = = = = = = = = = = = = = =
致
礼!
york
york(a)goldentek.biz
2008-01-09
Sorry, not all of the text I originally sent has appeared on the list archives, so here is the full email another time! :P
Hi,
can anyone provide any advice on getting CDRTool to work?
I have installed CDRTool 6.0 on a Debian Linux system 2.6 Kernel with Apache/2.2.6 (Unix) PHP/5.2.5
I am attempting to get it to work with OpenSER only (I have commented out the Asterisk and sip_trace sections
in global.inc) and I am recieving the error:
Error initializing CDRTool datasource
>From in my web browser.
It seems that in callsearch.phtml "$cdr_source" is not being set, but I dont understand why not and I am not
sure how to get more debugging info out of the whole thing (Im not used to PHP), I dont see any errors
in the syslog :(
With regards global.inc, I believe all database usernames and passwords are setup and have successfully tested
connecting to all databases with this data - and I believe all db_class sections have the corresponding sections
as required (the username, password, tablename etc).
thanks for any help! Andy.
mmm, again! :S well here is the bottom half of my email....
It seems that in callsearch.phtml "$cdr_source" is not being set, but I dont understand why not and I am not
sure how to get more debugging info out of the whole thing (Im not used to PHP), I dont see any errors
in the syslog :(
With regards global.inc, I believe all database usernames and passwords are setup and have successfully tested
connecting to all databases with this data - and I believe all db_class sections have the corresponding sections
as required (the username, password, tablename etc).
thanks for any help! Andy.
Hi,
can anyone provide any advice on getting CDRTool to work?
I have installed CDRTool 6.0 on a Debian Linux system 2.6 Kernel with Apache/2.2.6 (Unix) PHP/5.2.5
I am attempting to get it to work with OpenSER only (I have commented out the Asterisk and sip_trace sections
in global.inc) and I am recieving the error:
Error initializing CDRTool datasource
>From in my web browser.
It seems that in callsearch.phtml "$cdr_source" is not being set, but I dont understand why not and I am not
sure how to get more debugging info out of the whole thing (Im not used to PHP), I dont see any errors
in the syslog :(
With regards global.inc, I believe all database usernames and passwords are setup and have successfully tested
connecting to all databases with this data - and I believe all db_class sections have the corresponding sections
as required (the username, password, tablename etc).
thanks for any help! Andy.