Can I do a cname or txt query in openser?
I'm trying to create a cname for a enum soa.
That is, I want to do:
soa: e164.test.com
cname: other.test.com -> e164.test.com
Is there any way to accomplish this sort of thing
through DNS?
-g
--
Greg Fausak
greg(a)thursday.com
Hello members,
I have SER and SEMS on the same machine and it works fine. For
performance reasons, I would like to them to be on separate machines.
My query is the fact that SEMS uses the email address from the
subscriber table in SER db. Do I need to have SER db on the machine
where SEMS is running (means 2 identical dbs) or how can I achieve this?
Regards,
Amos.
Dear all,
I think it will be useful for everybody.
I have discussion with SIP device manufacturer about expiration time of
call attempt. When call is made from UA it sends INVITE with "Expires:
65". It is controlled by device's timer setting "ring_time_limit". This
timer can be set to anything between 10 and 600 seconds, we set it to 65
seconds. Fine.
However at the same time it starts a special timer, say
"max_response_time_invite". It can be set to anything between 8 and 20
seconds. We set it to 20 sec. Fine so far.
CASE 1: Imagine we call from UA to PSTN number, using one of VoIP->PSTN
providers. We have more than one provider, so we setup SER with
"fr_timer" value of, let's say 25 seconds and "fr_inv_timer" of 60. We
prepare failure routes in case when those timers hits in PSTN call. When
"fr_timer" hits, we simply reroute to another gateway. Great.
SER forwards INVITE from UA to remote PSTN gateway and at the same time
sends back "100 Trying" to UA. Fine.
Imagine now that it is busy-hour and it takes PSTN provider 21 seconds
to send back "180 Ringing" (or "183 Session Progress") and after 8
seconds more remote callee picks up ("200 OK").
Unfortunately device wil send CANCEL to SER, because it hasn't received
"180 Ringing" or "183 Session Progress" within the
"max_response_time_invite" setting of 20 seconds. It only received "100
Trying" at the beginning.
This ruined any failure route attempts from SER, as call failed.
CASE 2: Imagine now a second call. This time PSTN provider send "183
Session Progress" after 14 seconds and remote callee picked up after
further 15 seconds. This time everything went good. UA did not hit
"max_response_time_invite" timer - it waited patiently for whole 29
seconds for remote callee to pick up.
I think that when UA receives provisional response (like "100 Trying")
from SER it should stop timer "max_response_time_invite" and use
'ring_time_limit'. What device does now, it uses:
* "max_response_time_invite" when "100 Trying" received
* "ring_time_limit" when "180 Ringing" or "183 Session Progress" received
I haven't really found explanation of this in RFC3261. When you look at
SIP state machine on page 128, it is not explicitly said which timer
controls behaviour of UA when in "Proceeding" state ("Proceeding" is
entered oin receipt of ANY provisional response).
QUESTION 1: what timer controls behaviour of UA when in "Proceeding"
state? Especially regarding timeout. Is this still the 'timer B' started
at the beginning of INVITE transaction? If it is 'timer B', why UA is
CANCELing transaction after 20 seconds, if it informed by "Expires: 65"
field in INVITE that it set 'timer B' to 65 seconds.
QUESTION 2: What should I propose to manufacturer for maximum value for
"max_response_time_invite"?
--
Thanks,
Arek Bekiersz
Hi Ramin!
Please always cc to the list
Ramin Dousti wrote:
> Thanks, Klaus.
>
> Here is the full picture:
>
> I receive a SIP URI "sip:user@dest" on a UDP socket. The only thing I need to do
> is to proxy this session on a TCP socket to "dest".
>
> I fugured I can do:
> forward_tcp(dest);
>
> But when I extract dest into an AVP, forwasrd_tcp does not work with an AVP as
> its argument.
>
> avp_printf("s:dst", "$td");
> forward_tcp("s:dst");
>
> it tries to resolve "s:dst" literally.
forward_tcp does not support pseudo variables.
try forward_tcp() without parameters.
regards
klaus
>>If the destination is stored in an AVP, use avp_pushto
>>(http://openser.org/docs/modules/1.0.x/avpops.html#AEN291) to push the
>>new URI into the request URI.
>>
>>Then use t_relay (make sure the URI contains a transport=tcp parameter).
>>
>>If the destination is static, use t_relay_to_tcp(ip, port)
>>http://openser.org/docs/modules/1.0.x/tm.html#AEN320
>>
>>regards
>>klaus
>>
>>
>>
>>Ramin Dousti wrote:
>>
>>>Hello,
>>>
>>>I'm trying to forward a SIP message that's been received by UDP, to
>>>$td through TCP. but the following configuration doesn't work:
>>>
>>> send_tcp("$td", 5060 );
>>>or
>>> send_tcp($td, 5060 );
>>>or
>>> avp_printf("s:dst", "$td");
>>> send_tcp("s:dst", 5060 );
>>>
>>>Could you please help?
>>>
>>>--
>>>Ramin
>>>
>>>_______________________________________________
>>>Users mailing list
>>>Users(a)openser.org
>>>http://openser.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>
>
> --
> Ramin
>
>
Hi,
i want to change the "From:" tag with uac_replace_from when the
displayname is "anonymous".
with from_uri i can only check the uri.
is there a way to check the displayname in the "From: " tag?
for example:
From: "Anonymous <sip:1234@blah.net>"
i want to change it to:
From: "Anonymous <sip:Anonymous@Anonymous>"
Ciao
hello,
i have followed serveb installation instructions, and know i am on the
last step : "Open the file
http://<your-host>/<your-install-dir>/admin/index.php"
But i don't understand what i have to put on "your-host" ?
Thanks,
Ismaël
Hello,
I want to install SER on my debian. Could you explain me step by step
how to install and configure ser server on my computer ??
How attach mysql to SER server ?
Thanks to advance,
Ismaël
Hello,
I installed Debian on my computer, and now i want to install ser proxy
ans serweb. To install SER, i did :
// installation of apache, mysql, and phpmyadmin
apt-get install apache mysql-server phpmyadmin
// installation of SER
dpkg -i ser_0.9.6-0.1_i386.deb
// restart of SER service
/etc/init.d/ser restart
To install serweb, i just did :
tar -xvzf serweb-0.9.4.tar.gz
What do i have to do now ??
Thanks for your help,
Ismaël
Hi,
This is a question that has been bugging me for some time now. If I use SER
and the clients connected to it use only TCP, will there be a problem with
the number of sockets that can be open at the same time, if these clients
are behind NAT? I mean that for NAT bindings to stay open, each client has
to keep open TCP connection to SER all the time after registration. What is
the maximum number of open sockets that Linux can handle at the same time?
I've changed the maximum number of open files to 65535 (ulimit -n 65535).
Doesn't this also mean the maximum number of sockets? Is this the absolute
maximum? Also, does this mean the maximum number of clients that can connect
to my proxy?
Regards,
Teemu
--
Teemu Harju
http://www.teemuharju.net
Hi,
I want to be able to extract QoS parameters such as jitter, delay and
packet loss from VoIP calls made using my SER(both peer to peer for
my public clients and also for clients where the voice will be routed
with mediaproxy). I would like to develop something as opposed to
purchasing a software package.
>From searching the Internet the last few days I have seen that the
following options seem to be available:
1) RTCP
RTCP stream will contain this information. The problem is that
sniffers cannot clearly distinguish RTCP packets from other UDP
packets unless something like an IP address or specific UDP port is
supplied. I had considered using TCPDUMP but then where do I position
this in the network when voice is going peer to peer between client
and I'm not sure how this can work to monitor all calls when I might
not be aware of the end user IP addresses.
2) SIP INFO extension
This also seem to be an option but most off the shelf phones don't
seem to support this and this would required modification of a SIP ua.
3) SNMP
I thought maybe SNMP might be an option but the SER snmp module no
longer exists...
Does anyone have any comments on the above? Are the statements that I
made correct or can anyone think of other ways to monitor the voice
QoS? I am trying to understand how commercial applications have
accomplished this.
Many thanks in advance for advice.
Aisling.
>---- Original Message ----
>From: clona(a)cyberhouse.no
>To: ashling.odriscoll(a)cit.ie
>Subject: Re: [Serusers] Status of SER SNMP module
>Date: Sun, 29 Jan 2006 20:30:02 +0100
>
>>
>>Hi Aisling,
>>
>>The SNMP module is dead as a killed turtle :(
>>
>>-Atle
>>* Aisling O'Driscoll <ashling.odriscoll(a)cit.ie> [060129 20:26]:
>>> Hello,
>>>
>>> I'm just wondering what the current status of the SER SNMP module
>is?
>>> Is it currently supported and is there somewhere I can find
>>> documentation on it?
>>>
>>> Many thanks,
>>> Aisling.
>>>
>>>
>>>
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>>>
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-------------------Legal Disclaimer---------------------------------------
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