Hello,
I have a question about CANCEL processing.
I read on the mailing list that CANCEL will be automatically matched by t_relay
to transaction it's cancelling, if needed transformation to RURI will be automatically
applied and then it will be automatically send to correct destination.
I'm experimenting with openser 0.10.x and it seems to be true, but I'd like to confirm
that the following is OK:
if (loose_route()) {
do something
t_relay();
break;
}
if (is_method("CANCEL")) {
t_relay();
break;
}
if (uri==myself) {
do lookups that rewrite RURI
t_relay();
}
Thank you,
--
See you later,
Michael
Hello,
I have installed ser 8.14 with serweb that comes with it.
Ser works smothly, I can apply aliases too, but the problem is when I
open the web interface the page opens fine, but there is a red line
saying:
serweb: "sorry -- cannot open write fifo"
I changed the rules of ser_fifo file in /tmp to 777, didn't help. And
both ser and serweb look for the same fifo file, I can see that in
config files.
How can I get rid of this problem, becouse this way serweb doesn't do
instant messages or many other things.
I also added fifo_mode=0666 to my ser.cfg file and restarted, nothing changed
Thanks in advance
Ertugrul
Hi all,
I'd like to call several users at one time i.e :
1001 1002 1003 are usernames
1010 is a numbers which refers to 1001 1002 and 1003.
I'd like that when I call 1010, then 1001, 1002 and 1003 receive the INVITE.
Does anybody knows how I can do that?
Thanks.
B.R.
Xavier.
Alberto.
I tried VOVIDA B2BUA with SER+MEDIAPROXY some time ago and worked
fine. I stoped using it because I noticed the VOVIDA B2BUA had a poor
interface, any custom should be done in the code, so I had to recompile
after any change. Due to this I changed it for the Asterisk B2BUA, really
nice interface and works great with SER+MEDIAPROXY, you should try it.
*B2BUA: http://developer.berlios.de/projects/b2bua/
*B2BUA mailing list: http://lists.berlios.de/mailman/listinfo/b2bua-users
*B2BUA (installation
manual):http://lists.berlios.de/pipermail/b2bua-users/2005-April/000047.html
EnjoY.
MaV.-
> -----Mensaje original-----
> De: Alberto Cruz [SMTP:acruz@tekbrain.com]
> Enviado el: Jueves, 21 de Julio de 2005 12:35 p.m.
> Para: 'serusers(a)lists.iptel.org'
> Asunto: [Serusers] Does VOVIDA B2BUA works with SER+MEDIAPROXY?
> Importancia: Alta
>
> Hi does anybody is running SER+MEDIAPROXY and VOVIDA B2BUA?
>
> Please I need help to handle the CANCEL message.
>
> I've already try searching the
> http://www.archivum.info/serusers@lists.iptel.org/ and the
> http://lists.iptel.org/pipermail/serusers/ for any tip or configuration
> example but I didn't find anything.
>
> Regards
>
> Alberto Cruz
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
If i want to disable a remote SIP device by changing the SIP login
password , how do i force it to counter register again with my SIP
server? I have tried changing the password , restarting the server , but
the device wouldn't register after a successful registration. The only
time when it register is when i reboot it manually.
Any advice ?
Regards,
Sam
Hello!
I have some confuse in asterisk +ser combine,My ser is working well and I can register to call .Than I install asterisk well in same linux host.asterisk is working , I can watched the process can receiving on the port 5060 and listening on the port 5038 . this is differ of ser's process ,it's listening and receiving on 5060. when I stop ser and runing asterisk alone ,I can not used asterisk for x-lite register ,My problem is my asterisk is working correct ?some doc tell me I need to edit the sip.conf port:5061 to combine with ser and forward fail call to in ser.cfg configure in port 5061. Just only do like that ? who have a correct ser.cfg with asterisk configure and asterisk conf sample give me some referrence.Wish anyones help!
Thanks advance
Zhaomin
Aarthy G - CTD, Chennai. wrote:
>>Hi,
>>
>>We are using asterisk for testing our home gateway setup.
>>We have implemented Call Hold feature in our application.
>
> In our Application we have written code in such a way that for an INVITE for
> putting a SIP phone on HOLD
> will contain connection address "0.0.0.0" in the SDP message.
> We expect the same connection address i.e "0.0.0.0" in the 200 OK response
> for the INVITE that is sent.
>
>>This feature works when we tested without involving Asterisk.
>>Now after configuring Asterisk as our Registrar and OutBound Proxy, we
>>find that Call hold is not getting through. But we are getting a 200 0K
>>with connection address as the host ip of Asterisk. We see that the this
>>ReInvite is not getting forwarded to the appropriate detsination from the
>>asterisk. We are not looking for music on hold feature.
>>Can somebody here please tell us about how to configure asterisk for this
>>to work?
>>
Without seeing a SIP debug output it's very hard to diagnose your
problem. Surely Asterisk supports on hold by putting 0.0.0.0 in the SDP,
but it seems like your system is doing something else.
/O
Hello everybody,
Am trying SER debian 0.9.3-0.2 and MySQL accounting.
It seems there are some problems of compatibility with the old ser.cfg 0.9.3 or the db and modparam and flags !?
Anyone have a working ser.cfg or a hint about changes ?
Thanks in advance,
George