Hi,
first of all, excuse me for posting these questions twice, but
the previous mail was bloated with too verbose sentences.
I'll be straightforward then.
1) Why do all the configuration files I was given to see on the
web call record_route() when the method is *not* a REGISTER ?
2) What is the harm of calling fix_nated_contact() in addition
to fix_nated_register() in a REGISTER message ? The configuration
file [1] carefully avoids doing this.
[1] http://voip-info.org/wiki/view/OpenSER+And+RTPProxy
Thank you very much.
Regards,
--
Jeremie Le Hen
< jeremie at le-hen dot org >< ttz at chchile dot org >
How should I configure ser.cfg to enable Ser to send calls to Asterisk
Voicemail when after ringing for some while, the phone isn't answered?
Thanks,
Ricardo.
>Message: 1
>Date: Tue, 01 Aug 2006 15:27:20 +0300
>From: OpenSer Users <openser-users(a)list.coretech.ro>
>Subject: [Users] heders in route & failure_route blocks
>To: users(a)openser.org
>Message-ID: <44CF48A8.4020503(a)list.coretech.ro>
>Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>hello,
>I want to add Remote-Party-ID header in route block and change it in
>failure_route block.
>the problem is that after the failure_route block the message sent out
>has two Remote-Party-ID headers.
>in failure_route I am using remove_hf("Remote-Party-ID")
>scenario:
>original invite has no Remote-Party-ID header
>in route block I use: remove_hf("Remote-Party-ID"); append_rpid_hf("<",
>">;party=calling;privacy=full")
>in failure_route block I use: remove_hf("Remote-Party-ID");
>append_rpid_hf("<", ">;party=calling;privacy=off")
>the invite sent out after failure_route block has two Remote-Party-ID
>headers, one with privacy=full and another with privacy=off.
>is there a way to correct the problem ?
>from what I understand the failure_route block works on a copy of the
>original message, but the original message has no Remote-Party-ID
>header, so why the header from the route block is still visible ?
>on the other hand if the header added from the route block is visible
>why is it no seen by the remove_hf function ?
>thanks,
>Razvan Radu
Hi I all so have a similar problem! Is this behavior normal? Is it
possible to have this type of manipulation of the message on the
failure-route?
Thanks in advance,
Jose Silva
Okay,
Since I've been unable to pass SIP MESSAGEs back and forth through our server,
but knowing full well that I once was able, I decided to take our
configuration and remove everything in it that we've added over time (slowly
regressing back to the beginning of our SVN code repository for the SER
config) and see if I could tell what had changed so that I can no longer pass
a SIP MESSAGE back and forth between clients.
It didn't work.
SO... I decided to take the hello-world config and try with it.
No luck.
What am I missing here? Is there some special handling that is needed to pass
a SIP MESSAGE in SER 0.9.6? Is there a chance that something basic would have
changed between 0.9.0 (our original server) and 0.9.6 that would have broken
our ability to pass the MESSAGE method without additional tinkering?
Sure, if I put in a block like:
if(method=="MESSAGE")
{
sl_send_reply("200", "OK");
};
... it's clear the message gets there and I receive the 200 OK, but of course
it doesn't go anywhere after that.
Without that block in there, it ends up getting a 513 Message too big response
(which seems to be somewhat of a default for messages that don't end up
getting handled in any other way that SER understands).
I'm now completely at a loss. Pointers. Tips. Snide remarks. Anything helpful
would be greatly appreciated.
N.
Hi
I have installed SER and testing
But in my both sipura config same Extension
both are registering, how can i restrict the extensions.
If the user already loging, reject other user to login using same extension.
any sample rules.
Ram
Hi,
first of all, I am new to the SIP world, so please forgive me if
I ask anything dumb.
I am trying to set up a NAT-traversal SIP proxy, with the help
of the nathelper module and rtpproxy. I only have access to the
proxy and one softphone behind it.
I am facing two problems: when someone in the outer world wants to call
my softphone (which is behind the proxy), the proxy forwards the
INVITE message to itself and recurses until the mf_process_maxfwd_header()
statement stops further processing. I am not sure how to handle this.
My guess is that I should use the registrar module and when the
proxy receives an INVITE message, it should use registered() method
in order to know where to forward the request ; but I don't know how
to do that. Help would be welcome.
My second problem is that when the peer hangs off the phone, I should
receive a BYE message but I can't see it. Given I have no access on
the other end, I guessed that the BYE message had been sent elsewhere.
That's why I fixed the Contact header field. However, I still can't
see actions coming from the outer phone. Any idea ?
(I didn't attached any packet dump for now because I think it can
be more misleading than informative because there are two proxies
on the same machine: one that is a tunnel broker to encapsulate
VoIP traffic in HTTP, and the other one is OpenSER. Basically,
every SIP session is being tunneled in HTTP, then the datagram
in unwrapped by the first proxy and then forwarded to the second
one, which is OpenSER. Nonetheless I would be glad to provide
any packet dump upon request.)
My configuration file is attached. If you find any other
mistakes or inconsistencies in this configuration file, please tell me.
Best regards,
--
Jeremie Le Hen
< jeremie at le-hen dot org >< ttz at chchile dot org >
Hi,
I'm deploying a SER + Asterisk architecture, where SER is used to manage
acc, users database and sip routing, and Asterisk is used for voicemail
and PSTN gateway.
How should I edit ser.cfg to make Ser redirect calls to Asterisk when
the user doesn't answer his phone or when the user exists but it hasn't
any phone registered?
If I'm using also Asterisk for Gateway purpose, I guess all calls to be
sent to PSTN, Ser redirects them to Asterisk, but it will try to find a
mailbox for every number dialled, when it should distinguish calls for
PSTN not to be sent to Voicemail. How can Asterisk distinguish between a
call sent for voicemail and a call sent by ser to be redirected to PSTN?
It depends on the configuration made in ser.cfg?
Thanks,
Ricardo.
Hi,
how can i execute an external script to lookup if a prefix exists in
a mysqltable?
I want to have prefix(""); filled from a database.
--
Mit freundlichen Grüßen / Kind regards
Dominik Bay
Cablesurf Technik
Hi everybody,
Short question: Are there any best practices regarding the routing of
ACK Messages?
Currently we have the following logic implemented:
route {
[...]
##################################################################################################################
# Check for Re-Transmissions (not ACK/CANCEL)
###############################################################################################################
if ((method != "CANCEL") && (method != "ACK")) {
if (t_check_trans()) {
log(1, "Re-Transmission detected, message dropped.\n");
# Drop the message silently.
return;
}
}
##################################################################################################################
# Loose-Routing (RFC3261)
###############################################################################################################
if (loose_route()) {
route(10);
return;
}; # if (loose_route()) {
##################################################################################################################
# ACK/CANCEL Messages may be relayed
###############################################################################################################
if ((method == "CANCEL") || (method == "ACK")) {
if (uri != myself) {
# Und das Paket entsprechend weiterleiten
if (!t_relay("udp:outbound_proxy:outbound_proxy_port")) {
log(1, "Not possible to relay\n");
# Fehler melden
sl_reply_error();
return;
}
return;
}
}
[...]
##################################################################################################################
# Rest of "normal" Routing logic with number manipulation / Routing
to different Gateways / User Authorization/Authentication
###############################################################################################################
}
Is there any other method, to improve the routing of ACK/CANCEL
Messages? I would like to avoid all the number manipulation, checking
and choosing of a proper gateway....
I thought, this must (somehow) be possible since from a technical
perspective the ACK and CANCEL Message can be associated with a
corresponding INVITE Message (and the corresponding Transaction).
I've looked in the manual of the TM-Module and found the following example:
if ( is_method("CANCEL") ) {
if ( t_check_trans() )
t_relay();
exit;
}
(http://openser.org/docs/modules/1.1.x/tm.html#AEN460)
I guess, this example does not mean, that the URI gets rewritten
properly and the CANCEL-/ACK-Message would get properly relayed to the
destination, where it is supposed to end.
Is there any way to do it best? I've seen, that some clients do proper
loose-routing for ACK-Messages, but not all clients, so this is not a
solution.
Are there any other suggestions?
Thanks in advance,
Carsten
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