Hi,
I am using Ubuntu 8.04 and I have just installed Openser 1.3.1 via debian
package (with mysql and presence modules) to do presence with my computer
but I am not able to find "openser_mysql.sh" script to configure my
database. So I downloaded the src package on openser ftp and unpack it but
the script is still missing.
Is someone could tell me where I could download it ? Or if I did something
wrong during my installation ?
Regards
--
Yann Derichard
Hi,
We are running openeser version 1.2.2. Currently we have defined 3 groups of subscribers. We have a requirement of limiting max number of concurrent calls per group. For example, 10 maximum number of concurrent ( incoming /outgoing ) calls for subscribers belonging to group 1. How do we go about it. Any sample config will be appreciated.
thanks,
dhekial
_________________________________________________________________
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Hi Lee
You are not running the 'fix_nated_contact' function for INVITE messages.
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Hi Experts,
I have one SIP UA on private IP behind NAT and another 1 on public IP.
They both registered to OpenSER and they could call each other. The
problem is on call release. The BYE message sent from the UA on public
IP could never get to the other UA on private IP. OpenSER kept on
sending BYE to the private IP instead of sending it to the external
NAT IP. I tried to use fix_nat_register() and fix_nat_contact() when
REGISTER was received from the UA behind NAT but it didn't work. I
also tried putting fix_nat_contact() on BYE and still no help. Am I
missing anything?
modparam("nathelper", "received_avp", "$avp(s:rcv)")
. . . . .
if (method == REGISTER) {
if (nat_uac_test("2") || nat_uac_test("16")) {
fix_nated_register(); # or fix_nated_contact();
}
}
. . . . .
if (method == BYE) {
fix_nated_contact();
}
Thanks,
whglee
Hi Experts,
In my openser.cfg file, I tried to use the fix_nated_register()
function to save the IP:port information when a REGISTER message is
received. After I restart OpenSER, none of my SIP UAs could register
to it anymore. Network traces showed "port unreachable". Can someone
help?
Thanks,
whglee
Hi Klaus:
Thanks for your answer.
After I add dns_try_ipv6=yes in openser.cfg,
my problem has been solved.
But I wonder why the default value of
dns_try_ipv6 is "yes" in version 1.2.x,
but "no" in version 1.3.x?
For my personal opinion, setting default
value to "yes" is better than "no".
Best Regards,
Fang
> Hi!
>
> Openser does support AAAA, when enabled:
> http://www.openser.org/dokuwiki/doku.php/core-cookbook:1.3.x#dns_try_ipv6
>
> regards
> klaus
>
> Fangyu Ling wrote:
>> In my DNS server, there are only AAAA records for any nodes.
>> My test topology is as following:
>>
>> -----+---------+--------+------
>> | | |
>> siphone1 openser siphone2
>> (UA1) (UA2)
>>
>> For the testing procedure, I only use domain name of the URI.
>> When UA1 invited UA2 via openser, openser only ask A record
>> of UA2. There are no A records in my DNS server! So, openser
>> then replied 476 Unresolvable Destination message to UA1.
>>
>> In normal conditon, if there were no such name of A record,
>> it should ask AAAA record instead.
>> But why OPENSER only ask A record?
>>
>> Best Regards,
>> Fang
Hi
Is it possible to replicate incoming SIP requests (eg. REGISTER) to another
remote user SIP URI instead of host URI:
For instance I want to do (having previously registered to OpenSER the user
pascal.maugeri(a)company.com):
t_replicate("sip:pascal@company.com <sip%3Apascal(a)company.com>");
instead of
t_replicate("sip:10.2.3.4:5060");
I tried it with OpenSER 1.3 but I get the error message when it tries to
replicate:
3(8113) Main routing block. Method [REGISTER]
3(8113) ERROR: mk_proxy: could not resolve hostname: "company.com"
3(8113) ERROR: uri2proxy: bad host name in URI
<sip:pascal@company.net<sip%3Apascal(a)company.net>
>
3(8113) ERROR:tm:t_forward_nonack: failure to add branches
I'm just wondering if what I want to do is possible ...
Regards,
Pascal
Hello OpenSER Users,
O'Reilly Media has been working on an Asterisk Cookbook. Using the
same printed format as other O'Reilly Cookbooks (Perl, Python, etc)
O'Reilly Cookbooks offer practical solutions to problems in a fun
"recipe" format.
I myself have been working quite a bit with OpenSER in the last
couple of years and I've asked the other authors if we should open up
submissions for other recipes (including OpenSER). Everyone thinks
it's a great idea.
We are looking for a potentially new name and URL but the current
"Asterisk Cookbook" Wiki can be found here:
http://www.asteriskcookbook.com/
We'll be covering "Asterisk and friends". That means Asterisk,
OpenSER, Cisco gateways, etc. Pretty much anything VoIP with at least
an Open Source slant.
Request an account and add your recipes to the recipe heap (you'll
see my OpenSER recipes towards the bottom). See you there!
--
Kristian Kielhofner
http://blog.krisk.org
I have problem with CDR manipulation in case of call forwarding from client.
I'm testing with Linksys pap2 and I've set unconditional call forwarding.
When linksys receives the call, it'll send 302, temporarily moved. I have
done proper ACL checking using Diversion header in the new INVITE but I'm
having difficulty associating the forwarded leg CDR to original call as this
is totally new INVITE and none of the fields match in CDR. Can anyone guide
me in right direction?
In my DNS server, there are only AAAA records for any nodes.
My test topology is as following:
-----+---------+--------+------
| | |
siphone1 openser siphone2
(UA1) (UA2)
For the testing procedure, I only use domain name of the URI.
When UA1 invited UA2 via openser, openser only ask A record
of UA2. There are no A records in my DNS server! So, openser
then replied 476 Unresolvable Destination message to UA1.
In normal conditon, if there were no such name of A record,
it should ask AAAA record instead.
But why OPENSER only ask A record?
Best Regards,
Fang