[OpenSIPS-Users] How to enable rest_client module ??
Hi everyone, I am trying to use opensips rest_client module. Now i have *opensips 1.11.2-notls (x86_64/linux)* installed from Debian package on ubuntu. My question is how can i enable rest_client??? i mean if we want to use mysql we are going to install* opensips-mysql-module* like this which module is having rest_client?? I can't see rest_client.so in module folder also.. I know that if we installed opensips from source we can easily enable it by make menuselect then add-ons.. But i want to do the same in Debian package. any help is appreciated. Thank you, Venkatesh Macha, @ sillycodes4u http://sillycodes4u.blogspot.in -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/How-to-enable-rest-client-module-tp7592668.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to enable rest_client module ??
Hi Venkatesh, normally #apt-cache search opensips should list down all available packages for opensips, If its not listed it either not available in repo you'll have to contact repo maintainer to get the pkg build, I think its not available since its still in beta stage. On Tuesday, 29 July 2014, 11:42, Venkatesh Macha linuxven...@gmail.com wrote: Hi everyone, I am trying to use opensips rest_client module. Now i have *opensips 1.11.2-notls (x86_64/linux)* installed from Debian package on ubuntu. My question is how can i enable rest_client??? i mean if we want to use mysql we are going to install* opensips-mysql-module* like this which module is having rest_client?? I can't see rest_client.so in module folder also.. I know that if we installed opensips from source we can easily enable it by make menuselect then add-ons.. But i want to do the same in Debian package. any help is appreciated. Thank you, Venkatesh Macha, @ sillycodes4u http://sillycodes4u.blogspot.in -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/How-to-enable-rest-client-module-tp7592668.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Compress RTP packets in opensips or rtpproxy
Hello, I want to compress RTP packets in opensips or rtpproxy. Can anyone guide me or give me suggestion to achieve compression of media packets in opensips or rtpproxy? I have referred LZ77 , LZW , Huffman , ROHC algorithm for compression. Is there any specific file that can compress RTP packets in opensips or rtpproxy? Is there any library file that can be used in linphone to achieve media compression? I read that ROHC compression algorithm can do better job for RTP compression. It can remove 40 Bytes (IP Header 12 bytes + UDP header 8 Bytes + RTP Header 12 Bytes) of overhead and compress 40 bytes to 2-3 bytes. So rtp will send only 2-3 bytes of header plus 20 bytes of payload. Please Suggest best method to compress media packets. -- Kind regards, Kaushik Parmar ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] uac_auth to uac_registrant module
Hi,
If you try it from a failure route, you need to do :
$(replyhdr(Proxy-Authenticate))
(see http://www.opensips.org/Documentation/Script-CoreVar-1-11)
In failure route, the context is of the request message, so if you want
to access the reply, you need to switch to its context.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 29.07.2014 12:41, Igor Olhovskiy wrote:
Hi again.
Seems to be, $hdr(Proxy-Authenticate) is NULL at 401 response.
failure_route[1] {
...
if ( t_check_status(40[17]) ) {
...
xlog(L_INFO, Asterisk flavour $hdr(WWW-Authenticate), Proxy
flavour $hdr(Proxy-Authenticate));
}
}
becomes
/usr/sbin/opensips[18983]: Asterisk flavour null, Proxy flavour null
It's logic, cause in failure_route we work with initial INVITE, but
not 401 reply. Cause, if we working with reply directly, we can't
apply uac_auth function to it.
28.07.14 21:10, Игорь Ольховский написав(ла):
Hi,
Many thanks on your answer, will wait for a new feature and look at $hdr var
more close.
Anyway, I have a little trouble with CSeq change (means it is need to do
accurate), but for now it’s a solution.
Many thanks again.
28 июля 2014, в 20:46, Bogdan-Andrei Iancubog...@opensips.org написал(а):
Hi,
1) on changing cseq as a simple text - this is not wise as you break the
sequence of cseq number in the dialog; we are working on a feature to allow you
do that in sip-wise way.
2) about realm, the proxy/www -Authenticate header (in the 401/407 reply) has
the realm parameter; you can grab it by transformations; on
$hdr(Proxy-Authenticate) apply a regexp transformation
(seehttp://www.opensips.org/Documentation/Script-Tran-1-11#toc72) to get the
realm param from there.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 21.07.2014 20:15, Igor Olkhovskii wrote:
Made it work via modification of CSeq (remove_hf - append_hf) and now is a
question, how to get correct realm from response. OpenSIPs is very limitated to
text processing
21.07.2014 18:39, Igor Olhovskiy пишет:
Found this tread, but seems to be no luck in to work with INVITE on
Asterisk.
Is there any luck to get Asterisk auth (without touching Asterisk)
https://www.mail-archive.com/users@lists.opensips.org/msg25236.html
On 21.07.2014 16:14, Igor Olhovskiy wrote:
Hi!
I'm trying to get OpenSIPS 1.11 act as registrar proxy. Means it's not
only register on external servers, but take care of INVITE's and so.
I've configured modules as:
loadmodule uac_auth.so
loadmodule uac.so
loadmodule uac_registrant.so
modparam(uac,restore_mode,auto)
modparam(uac_auth,auth_realm_avp,$avp(uac_realm))
modparam(uac_auth,auth_username_avp,$avp(uac_username))
modparam(uac_auth,auth_password_avp,$avp(uac_password))
modparam(uac_registrant, timer_interval, 120)
modparam(uac_registrant, hash_size, 2)
modparam(uac_registrant, db_url,
mysql://opensips:opensips@localhost/opensips)
failure_route[1] {
..
# have we already tried to authenticate?
if (isflagset(8)) {
xlog(L_INFO, FAILUREROUTE_STATUS40X_SETFLAG8:
[F=$fu R=$ru D=$du M=$rm IP=($si:$sp $Ri:$Rp) ID=$ci]);
t_reply(503,Authentication failed);
exit;
}
if (is_method(INVITE)) {
# mark that auth was performed
setflag(8);
# trigger again the failure route
t_on_failure(1);
# repeat the request with auth response this time
$avp(uac_realm) = $td;
$avp(uac_username) = $fU;
avp_db_query(SELECT password FROM registrant
WHERE (registrar = 'sip:$avp(uac_realm)') AND ( username =
'$avp(uac_username)'),$avp(uac_password));
xlog(L_INFO,
FAILUREROUTE_STATUS40X_UACAUTHINVITE_DEBUG_VARIABLES: AVP_UAC_REALM:
$avp(uac_realm) AVP_UAC_USERNAME: $avp(uac_username) AVP_UAC_PASSWORD
:$avp(uac_password));
uac_auth();
t_relay();
}
}
.
}
I see correct vars in debug message, but uac_auth() not to append branch
to reply INVITE.
For example, I have such string
AVP_UAC_REALM: some-dns.example.net.ua AVP_UAC_USERNAME: 2225678
AVP_UAC_PASSWORD :SuperStrongPassword
What is wrong in this config/AVP's?
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Re: [OpenSIPS-Users] Handling Timeouts Between INVITE Provisional Responses
Hi, I will look into that, thanks for the report. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 28.07.2014 22:11, Jamuel Starkey wrote: Thanks, Bogdan--great feature in 1.11.x I was originally looking in the 1.8.x man pages. I just got 1.11.2 working here in dev and the feature works great. One question--any reason that if I xlog the $T_fr_timeout or $T_fr_inv_timeout variables they return 0 whenever they were not explicitly set in the script instead of returning the default values that were set in the mod_param section for tm module? In any case keep up the good work! On Jul 28, 2014, at 10:49 AM, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: Hi, fr_timeout is the interval between the request and the first provisional reply - shortly, how long to to wait for the first provisional reply. fr_inv_timeout is the interval between the request and the final reply (2xx or negative) - shortly, how long to wait for completing the transaction. So you can use different values for these 2 timers for a transaction to get the desired behavior. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 19.07.2014 07:53, Jamuel Starkey wrote: Hi, From the TM module man page it would see fr_timeout should fire whenever a final reply is not received for a request but it is only limited to ACK's for negative INVITE replies? Is this correct? Can someone be so kind as to clarify when fr_timeout is triggered. Is there a method to set a timeout between specific provisional replies in an INVITE? In other words let's say we receive a 100 TRYING but then see a very long delay until the next provisional reply (specifically 183 Ringing). Is there a way to timeout the request so that we can route advance to another gateway? We have a carrier who is randomly prone to lengthy PDD intervals in completing calls we send to them. On such a call our initial INVITE almost immediately sees the 100 Trying but then we might see another 25-30 seconds before we seeing another provisional response such as 183 Ringing or a final response. Is there a way to implement an INVITE specific provisional response timer so that if we get the 100 Trying and then don't see any other provisional response in a much shorter period of time (for example no more than 6 seconds) we could then route advance and attempt the call on another gateway. Is this at all possible? Serial forking to the next gateway would be fine but parallel forking would be preferred after the provisional timeout occurred. We'd just not want to route advance if we received the 183 Ringing provisional response. Any help or insight would be greatly appreciated. Cheers, JPS ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to enable rest_client module ??
Thank you for your reply Pasan, your right it is not available now. but it is available if we install from source.Anyway i installed opensips from source it is working fine. once again thank you for your Input. Venkatesh Macha, -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/How-to-enable-rest-client-module-tp7592668p7592678.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mediaproxy-dispatcher extra fields in database
On 28 Jul 2014, at 19:50, Edwin eahaselh...@gmail.com wrote: I am using mediaproxy-dispatcher 2.6.1 with the (mysql) database backend. The id, call_id, from_tag and to_tag are created on first start. The rest of the data is in the info field. I would like to add extra fields like caller_codec, callee_codec, caller_remote etc. Is this possible and if so how can I add this fields (like extra settings in config.ini?). IIRC you’d have to modify the source code for that. Or use RADIUS, which uses separated fields. -- Saúl Ibarra Corretgé AG Projects signature.asc Description: Message signed with OpenPGP using GPGMail ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Running two OpeniS installs on one MySQL host
Hi, Is it possible to have two installs of OpenSiPS use the same MySQL data base? Below are the modules that I will be using: loadmodule db_mysql.so loadmodule signaling.so loadmodule sl.so loadmodule tm.so loadmodule rr.so loadmodule maxfwd.so loadmodule usrloc.so loadmodule registrar.so loadmodule textops.so loadmodule mi_fifo.so loadmodule uri.so loadmodule dialog.so loadmodule load_balancer.so I know it has been done in the past by others, just not sure if with my set up it can be done. Regards, Dovid ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] DTMF Detection in RTP-Proxy
Hello, I am working on setting up freeswitch and opensips integrations. Actually followed this link. http://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegration Everything worked right out of the box with minor tweeks here and there. Thank you *Giovanni Maruzzelli * I am stuck at passing DTMF digits to freeswitch when two UAs are in the middle of conversations. Say user A called user B. Now user B want to park the call by pressing *8500, so that user C can pick up the call by pressing *8501. During this time user A should be on hold listening to music etc. Any idea what would be best approach to achieve this. All UAs are registered on Opensips. Thank you, -Jai ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
