Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RFC] OpenSIPS 2.0 - changes on the SIP messages
IMO lumps should go. They're a hackish solution that caused enough problems already. As I see it, we must have the original message and the current message (original plus all changes applied in realtime) so one can get at any moment the original message and the modified message without any complication (modified message should be per branch of course). Re- parsing can be kept to a minimum or even avoided completely in certain circumstances if the modified message is kept as a dictionary of SIP header name/value pairs, plus some extra info about the first line in the message and the content. In other words while the original message is kept both parsed and unparsed, the current modified messages are only kept parsed and are reconstructed in a text buffer only when sent out. On 28 Apr 2010, at 13:40, Bogdan-Andrei Iancu wrote: Hi, sorry for crossposting, but I thing this topic is interesting both for users (as experience) and for developers (as solution). So, right now the code for 2.0 got to a stage were we need to apply changes on the received messages (like adding VIA headers, changing headers, etc). And before starting the work on this area, I would like to get some opinions on the available options: 1) keep the current mechanism for changing the SIP messages by using lumps that are applied only when the message is sent out : Advantages: code is there and works; the mechanism is very fast ; no need to re-parse the message after the parsing Disadvantages : from script or code, you are not able to see the previous changes you did on the message, like: if you remove a hdr, it will be marked to be removed, but you will still see it there after the removed if you add a new hdr, you will not be able to see it (it will be actually added only when the message will be sent out) if you replace the contact hdr (like after fix_nated_contact() ) will not see the new contact, but the old one trying to apply 2 changes in the same area (like changing twice the contact) will result in bogus result. 2) find a new approach that will allow to push in realtime the changes on messages Advantages: the change will take affect instantly, so all the disadvantages from 1) (as user experience in operating opensips) will disappear . Disadvantages : the code will probably be slower (the changes part), but will speed up other parts of the code (where you need to manually identify the prev changes); changed parts of the SIP message will require re-parsing (so the code will have to check the state of a header (if parsed or not) before each usage) you will not be able to undo changes on the SIP messages in an easy way (for example during the parallel and serial forking on per-branch changes) So this is the dilemma: if the current mechanism (with applying changes at the end) such a bad user experience (scripting) in order to try for 2.0 a different approach ? I would like to hear some opinions from both people writing scripts and people writing code. Thanks and regards, Bogdan -- Bogdan-Andrei Iancu www.voice-system.ro ___ Devel mailing list de...@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel -- Dan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.6.2 w/ Control Panel 4.0 - Problem w/ CDRViews and Dialog
Alex this is my current setup - /var/www/opensips-cp/config/db.inc.php //database driver mysql or pgsql $config-db_driver = mysql; //database host $config-db_host = localhost; //database port - leave empty for default $config-db_port = ; //database connection user $config-db_user = root; //database connection password $config-db_pass = passwd root; //database name $config-db_name = opensips; if (!empty($config-db_port) ) $config-db_host = $config-db_host . : . $config-db_port; ? - /var/www/opensips-cp/config/boxes.global.inc.php /* DEFINITION OF BOXES (servers) */ // each server is a box $box_id=0; // mi host:port pair || fifo_file $boxes[$box_id]['mi']['conn']=/tmp/opensips_fifo; // monit host:port $boxes[$box_id]['monit']['conn']=127.0.0.1:2812; $boxes[$box_id]['monit']['user']=admin; $boxes[$box_id]['monit']['pass']=monit; $boxes[$box_id]['monit']['has_ssl']=0; // description (appears in mi , monit ) $boxes[$box_id]['desc']=Primary SIP server; $boxes[$box_id]['assoc_id']=1; // enable local smonitor charts on this box : 0=disabled 1=enabled // (cron) $boxes[$box_id]['smonitor']['charts']=1; - Opensips.cfg - # - mi_fifo params - - modparam(mi_fifo, fifo_name, /tmp/opensips_fifo) - modparam(mi_fifo, fifo_mode, 0666) Thanks Erick Ch. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Alex Ionescu Sent: Wednesday, April 28, 2010 3:07 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips 1.6.2 w/ Control Panel 4.0 - Problem w/ CDRViews and Dialog Hi Erick, The global db.inc.php (the one you have in /var/www/opensips-cp/config/) is used by all the modules as long as they don't find something more specific defined into their config files (like, for example, if you want a different db setup for module domains you can edit /var/www/opensips-cp/config/tools/system/domains/db.inc.php - and this will override the global db setup ). Dialog takes the call information using a MI command. So you must have OpenSIPS Control Panel and OpenSIPS configured properly (you must choose between xmlrpc and fifo) - see the OpenSIPS config file (to enable mi_xmlrpc module or the mi_fifo) and also check the config/boxes.global.inc.php to have the correct MI connection parameters set up. Regards, Alex On 4/27/2010 18:24, Erick Chinchilla Berrocal wrote: FYI According the manual http://opensips-cp.sourceforge.net/ Dialog Module (dialog) one option is change the setup in the file /opensips-cp/config/tools/system/dialog/db.inc.php in the configuration I don't have this file only those following. Please confirm if need the file db.inc.php /etc/opensips# ls -al /var/www/opensips-cp/config/tools/system/dialog/ total 16 drwxr-xr-x 2 www-data www-data 4096 2010-04-26 13:46 . drwxr-xr-x 16 www-data www-data 4096 2010-04-26 11:49 .. -rw-r--r-- 1 www-data www-data 1243 2010-03-08 12:54 local.inc.php -rw-r--r-- 1 www-data www-data 1094 2010-03-08 12:54 menu.inc.php Look this file in the other modules and get. In the modules./tools/. are in default setup /etc/opensips# find / -name db.inc.php /var/www/opensips-cp/config/tools/admin/add_admin/db.inc.php /var/www/opensips-cp/config/tools/admin/list_admins/db.inc.php /var/www/opensips-cp/config/tools/users/user_management/db.inc.php /var/www/opensips-cp/config/tools/users/alias_management/db.inc.php /var/www/opensips-cp/config/tools/system/smonitor/db.inc.php /var/www/opensips-cp/config/tools/system/cdrviewer/db.inc.php /var/www/opensips-cp/config/tools/system/permissions/db.inc.php /var/www/opensips-cp/config/tools/system/domains/db.inc.php /var/www/opensips-cp/config/tools/system/loadbalancer/db.inc.php /var/www/opensips-cp/config/tools/system/drouting/db.inc.php /var/www/opensips-cp/config/tools/system/siptrace/db.inc.php /var/www/opensips-cp/config/tools/system/pdt/db.inc.php /var/www/opensips-cp/config/tools/system/nathelper/db.inc.php /var/www/opensips-cp/config/tools/system/dialplan/db.inc.php /var/www/opensips-cp/config/tools/system/dispatcher/db.inc.php /var/www/opensips-cp/config/db.inc.php Now, I understand that if only changed this file is ok for all modules, can you confirm if is correct /var/www/opensips-cp/config/db.inc.php //database driver mysql or pgsql $config-db_driver = mysql; //database host $config-db_host = localhost; //database port - leave empty for default $config-db_port = ; //database connection user $config-db_user = root; //database connection password $config-db_pass = password for root; //database name $config-db_name = opensips; if (!empty($config-db_port) ) $config-db_host = $config-db_host . : . $config-db_port; I want to test the
[OpenSIPS-Users] MediaProxy 1.x - Plz Help
Hi, I'm looking for MediaProxy version 1.8.x. I can't use 2.x becouse of existing old version of openser at the installation site which i can't change. Where can i find archive versions of media proxy ? Best regards, -- Wojciech Wrona ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] uac_replace_from
Thanks help much apreciated modparam(rr, append_fromtag, 1) modparam(uac,restore_mode,auto) xlog($fu \n); uac_replace_from(2101573...@10.10.1.100 mailto:2101573...@10.10.1.100 ); xlog($fu \n); Output sip:02086847...@10.10.1.50 sip:02086847...@10.10.1.50 I cant seem to get it to change the value. am i missing something From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Antonio Anderson Souza Sent: 29 April 2010 00:26 To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] uac_replace_from The best way is to use the uac module. Calling the function uac_replace_from before the relay, follow a script snippet bellow: uac_replace_from(sip:2...@10.10.1.100 mailto:sip%3a2...@10.10.1.100 ) Best regards, Antonio Anderson M. Souza Voice Technology http://www.antonioams.com Em 28/04/2010 20:19, dcomms i...@dcomms.netescreveu: HI, Whats the best way to do this? i have asterisk configured with fromuser=22 xlog($fu \n) shows sip:222...@10.10.1.100 mailto:sip%3a222...@10.10.1.100 I want to change to sip:111...@10.10.1.100 mailto:sip%3a111...@10.10.1.100 before relay Even better can someone show me how to change $fu for any ip address 10.10.1.100 thanks -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/uac-replace-from-t p4977451p4977451.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] uac_replace_from
2010/4/29 David Adade david.ad...@dcomms.co.uk Thanks help much apreciated modparam(rr, append_fromtag, 1) modparam(uac,restore_mode,auto) xlog($fu \n); uac_replace_from(2101573...@10.10.1.100); uac_replace_from(sip:2101573...@10.10.1.100 sip%3a2101573...@10.10.1.100 ); xlog($fu \n); Output sip:02086847...@10.10.1.50 sip:02086847...@10.10.1.50 I cant seem to get it to change the value. am i missing something -- *From:* users-boun...@lists.opensips.org [mailto: users-boun...@lists.opensips.org] *On Behalf Of *Antonio Anderson Souza *Sent:* 29 April 2010 00:26 *To:* OpenSIPS users mailling list *Subject:* Re: [OpenSIPS-Users] uac_replace_from The best way is to use the uac module. Calling the function uac_replace_from before the relay, follow a script snippet bellow: uac_replace_from(sip:2...@10.10.1.100 sip%3a2...@10.10.1.100) Best regards, Antonio Anderson M. Souza Voice Technology http://www.antonioams.com Em 28/04/2010 20:19, dcomms i...@dcomms.netescreveu: HI, Whats the best way to do this? i have asterisk configured with fromuser=22 xlog($fu \n) shows sip:222...@10.10.1.100sip%3a222...@10.10.1.100 I want to change to sip:111...@10.10.1.100sip%3a111...@10.10.1.100before relay Even better can someone show me how to change $fu for any ip address 10.10.1.100 thanks -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/uac-replace-from-tp4977451p4977451.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] install cdrtools
Dear who can give me file of cdrtools installation I install opensips with mysql callcontrol install callcontrol app But I cann't install CDRTools apt-get install cdrtool tell me need geoip-database,OK I installed ,and do it again tell me need php5-geoip,OK installingbut this step remove the geoip-database My god? How to do? Kenny ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Modify Script variable
Hi, most of the variables (the once allowing access to the SIP message) are read-only, so you cannot change them - better see : http://www.opensips.org/Resources/DocsCoreVar16 Regards, Bogdan dcomms wrote: HI, Whats the best way to do this? i have asterisk configured with fromuser=22 xlog($fu \n) shows sip:222...@10.10.1.100 I want to change to sip:111...@10.10.1.100 before relay Even better can someone show me how to change $fu for any ip address 10.10.1.100 thanks -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] install cdrtools
A quick google of CDRTool found the CDRTool website which has a good installation guide: http://cdrtool.ag-projects.com/wiki/Install In the installation guide there is a link to where you can download the tar of it: http://download.ag-projects.com/CDRTool/ Hope this helps :) -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/install-cdrtools-tp4978787p4978878.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Issue with rls module?
Thanks Anca
I have another issue with RLS Subscription expiration. I send a Subscribe to
RLS and I expect a final (Subscription Ste=terminated) Notify when the
subscriptione expires. But I receive no such message.
On the other hand, after a few hours it seems indeed the subscription is not
active anymore.
Sorry to bother you on this, maybe it is a configuration issue.
Best regards
Andrés
2010/4/28 Anca Vamanu a...@opensips.org
Andres2 Cores2 wrote:
Thank you Anca
In fact I am using the user-agent (picked it up from one config file),
either my client either opensips.
if($hdr(User-Agent) =~ me)
{
xlog(L_DBG,RLS subscribe);
$var(ret_code)= rls_handle_subscribe();
if($var(ret_code)== 10){
handle_subscribe();
}
}
else
{
xlog( route 2 subscribe normal --- \n);
handle_subscribe();
}
Now, after some fixes of the configuration, it seems the flow is
correctly managed.
There is still one point I am not sure. I had some issues with the
NOTIFY sent by PS to RLS
that was not correctly managed and I fixed it by adding this condition
for NOTIFY
if
((is_method(SUBSCRIBE)||is_method(NOTIFY)) $rd == 10.62.0.155) {
# in-dialog subscribe requests
route(2);
exit;
}
Where route(2) invokes rls_handle_notify() for NOTIFY requests.
Do you think it is fine? It is not perfectly clear what is called the
in-dialog subscribe requests.
Yes, the check is all right. The most common check is if uri ==
myself, because then it is sure that the Notify is addressed to
OpenSIPS. The stricter check is uri== what you set in server_address
prameter for module rls.
That comment line with in-dialog requests is put there because that if
you look in the default configuration file - initial and in-dialog
rquests are processed separately. The type is determined at the
beginning and there are two different block for the two types of
messages. So, for Subscribes route(2) is called in two places in the
script - once for the initial ones and also for the in dialog ones.
Anyhow that comment is not something you should worry about.
Regards,
--
Anca Vamanu
www.voice-system.ro
Thanks for all.
Andrew
2010/4/28 Anca Vamanu a...@opensips.org mailto:a...@opensips.org
Hi Andrew,
Thanks for your appreciation.
The thing that you need to take care with RLS are the Subscribe
messages
sent by the RLS server ( because this is the mechanism - when
receiving
a Subscribe for RLS the rls server will send more Subscribe
messages to
the presence server, and I see that you have the presence server
collocated with rls). You need to make sure that these are handled by
the presence server and that they do not go into rls again. For
this you
can check the source of the message and if it is the same machine
just
call handle_subscribe.
if(is_method(SUBSCRIBE) si == _local_ip_)
handle_subscribe();
If you watch the network trace and log where each subscribe goes for
processing and still notice this errors, post again an email on
the list.
Regards,
--
Anca Vamanu
www.voice-system.ro http://www.voice-system.ro
Andres2 Cores2 wrote:
Hi,
First of all we would like to thank the whole opensips team. For
testing our client, we are mainly interested in the presence and
rls
topics and found that a pretty good set of features is available
right
now, so thanks for your job.
During our tests we encountered an issue running opensips 1.6 in
rls-server mode (simple presence works fine). It seems to us it is
just a configuration issue or a bug in the routing rules, but we
can't
find out it.
From the client point of view I cannot succeed in receiving
presence
Notify messages body, only rlmi parts are provided.
In fact it seems rls_handle_subscribe has issues when trying to
send
Subscribes to Presence on behalf of rls. I have systematically
errors
presence:get_stored_info: record not found in hash_table and
sometimes Duplicate entry
'sip:alice-l...@promethee.test.com-cbb99ea81711c30ed9c0edf6' for
key 2
Typically:
Apr 26 13:45:57 [20047] ERROR:db_mysql:db_mysql_do_prepared_query:
driver error: Duplicate entry
'sip:alice-l...@promethee.test.com-cbb99ea81711c30ed9c0edf6' for
key 2
Apr 26 13:45:57 [20047] ERROR:presence:update_db_subs: unsuccessful
sql insert
Apr 26 13:46:01 [20045] ERROR:presence:get_stored_info: record not
Re: [OpenSIPS-Users] uac_replace_from
Hi David, David Adade wrote: Thanks help much apreciated modparam(rr, append_fromtag, 1) modparam(uac,restore_mode,auto) xlog($fu \n); uac_replace_from(2101573...@10.10.1.100 mailto:2101573...@10.10.1.100); xlog($fu \n); Output sip:02086847...@10.10.1.50 sip:02086847...@10.10.1.50 I cant seem to get it to change the value. am i missing something It is ok, the $fu prints all the time the original FROM URI (it does not reflect the changes) - but check the outgoing SIP message to see if the FROM URI was indeed changed. Regards, Bogdan *From:* users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Antonio Anderson Souza *Sent:* 29 April 2010 00:26 *To:* OpenSIPS users mailling list *Subject:* Re: [OpenSIPS-Users] uac_replace_from The best way is to use the uac module. Calling the function uac_replace_from before the relay, follow a script snippet bellow: uac_replace_from(sip:2...@10.10.1.100 mailto:sip%3a2...@10.10.1.100) Best regards, Antonio Anderson M. Souza Voice Technology http://www.antonioams.com Em 28/04/2010 20:19, dcomms i...@dcomms.net mailto:i...@dcomms.netescreveu: HI, Whats the best way to do this? i have asterisk configured with fromuser=22 xlog($fu \n) shows sip:222...@10.10.1.100 mailto:sip%3a222...@10.10.1.100 I want to change to sip:111...@10.10.1.100 mailto:sip%3a111...@10.10.1.100 before relay Even better can someone show me how to change $fu for any ip address 10.10.1.100 thanks -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/uac-replace-from-tp4977451p4977451.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org mailto: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 -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Autehntification from DB with AVPOPS module ?
Hi everybody, Can I do the autehntification from DB with AVPOPS and UAC modules ? I want to do the authentification from DB (no opensips DB) but I don't know how to proceed. I don't know if my start's code is right ? (befor, I'd like to test on opesips DB): modparam(avpops,db_url,mysql://opensips:opensip...@localhost/opensips) modparam(avpops,avp_table,subscriber) modparam(uac, auth_username_avp, $avp(i:1)) modparam(uac, auth_password_avp, $avp(i:2)) modparam(uac, auth_realm_avp, $avp(i:3)) avp_db_query(SELECT username, password, realm FROM subscriber WHERE where username='$tu', $avp(i:1);$avp(i:2);$avp(i:3)); can anyone help me please ?. Thanks. Sam. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Autehntification-from-DB-with-AVPOPS-module-tp4979020p4979020.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
[OpenSIPS-Users] Question about B2BUA
Hi Guys, Do the B2Bua modules in Opensips permit the box to register to a remote sip proxy? If yes, can you please point me to an example or documentation how to implement that? Thanks, -- Best regards, Vadim Grinco ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Storage of registered users.
On 29/4/10 3:45 PM, brianpocock wrote: Just out of interest, where are registered users stored when a backend database is not used, is it a text file somewhere? Thanks If no persistent storage backend is used then they are stored in memory. Regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Storage of registered users.
Thanks for that, so if you were to stop opensips any user you had created would be lost? Thanks On 29 April 2010 15:19, Saúl Ibarra Corretgé [via OpenSIPS (Open SIP Server)] ml-node+4980220-183174806-480...@n2.nabble.comml-node%2b4980220-183174806-480...@n2.nabble.com wrote: On 29/4/10 3:45 PM, brianpocock wrote: Just out of interest, where are registered users stored when a backend database is not used, is it a text file somewhere? Thanks If no persistent storage backend is used then they are stored in memory. Regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list [hidden email] http://user/SendEmail.jtp?type=nodenode=4980220i=0 http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- View message @ http://opensips-open-sip-server.1449251.n2.nabble.com/Storage-of-registered-users-tp4980049p4980220.html To unsubscribe from OpenSIPS (Open SIP Server), click herehttp://opensips-open-sip-server.1449251.n2.nabble.com/subscriptions/Unsubscribe.jtp?code=YnJpYW5rZWl0aGphbWVzQGdvb2dsZW1haWwuY29tfDE0NDkyNTF8NTkyNDg5NzI0. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Storage-of-registered-users-tp4980049p4980232.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] Storage of registered users.
On 29/4/10 4:20 PM, brianpocock wrote: Thanks for that, so if you were to stop opensips any user you had created would be lost? Thanks Yes. A database is not really needed, db_text uses a text file as the persistent storage backend. Regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] NAT Problem using Nat helper
Hi,
I've configured OpenSIPs with FreeRadius, the version for OpenSIPs I'm using
is 1.6.1 and FreeRadius verison is update 2 date. When I register 2
sofphone, they got authenticated and authorized by radius and got
registered sucessfully. Even I made calls between two softphone
sucessfully(Can hear one another). The UAS configured on different network
means hosted with public IP and my softphones are registered other and NATed
network. I mapped a DID on UAS and mapped it on my one of my softphone. The
problem I'm facing is when call coming from DID and ring my phone the caller
can hear me but I can't hear the caller(one way calling issue). But not
facing the problem on when calling between to sip clients and also calling
from OpenSIPs to Asterisk. The configuration and cases for OpenSIPs is
listed down below;
UAC-- UAS(OpenSIPs) -
UACtwo way voice is establised
UAC-- UAS(OpenSIPs) - Asterisk
UACtwo way voice is establised
PSTN-- UAS(OpenSIPs)
- UAC one way
voice is establised
(hears the dest voice)(can't hear caller
voice)
#loadmodule auth_diameter.so
loadmodule aaa_radius.so
loadmodule auth_aaa.so
loadmodule permissions.so
loadmodule nathelper.so
#Settings For
Radius-
#modparam(auth_diameter, diameter_client_host, localhost)
modparam(aaa_radius,
radius_config,/usr/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, aaa_url,
radius:/usr/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, aaa_flag, 2)
modparam(acc, aaa_missed_flag, 3)
modparam(acc, aaa_extra, User-Name=$Au; \
Calling-Station-Id=$from; \
Called-Station-Id=$to; \
Sip-Translated-Request-URI=$ruri; \
Sip-RPid=$avp(s:rpid); \
Source-IP=$si; \
Source-Port=$sp; \
Canonical-URI=$avp(s:can_uri); \
Billing-Party=$avp(s:billing_party); \
Divert-Reason=$avp(s:divert_reason); \
X-RTP-Stat=$hdr(X-RTP-Stat); \
Contact=$hdr(contact); \
Event=$hdr(event); \
SIP-Proxy-IP=$avp(s:sip_proxy_ip); \
ENUM-TLD=$avp(s:enum_tld))
modparam(auth_aaa,aaa_url,radius:/usr/etc/radiusclient-ng/radiusclient.conf)
modparam(auth, rpid_prefix, sip:)
modparam(auth, rpid_suffix, @11.22.33.44;screen=yes;privacy=off)
modparam(auth, rpid_avp, $avp(s:rpid))
#modparam(uri,service_type,10)
# - setting module-specific parameters ---
modparam(dispatcher, db_url, mysql://opensips:opensip...@localhost
/opensips)
modparam(permissions, db_url, mysql://opensips:opensip...@localhost
/opensips)
#- setting NAT module parameters -
modparam(nathelper,ping_nated_only,1)
modparam(nathelper, natping_interval, 30)
modparam(nathelper,natping_processes,1)
#modparam(nathelper,rtpproxy_sock,udp:127.0.0.1:7890)
modparam(nathelper,rtpproxy_sock, )
modparam(nathelper,received_avp,$avp(i:42))
#modparam(nathelper, sipping_bflag, 7)
modparam(usrloc, nat_bflag, 6)
### Routing Logic
# main request routing logic
route{
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}
#NAT detection
log(# Go to Route 3 for NAT
Detection #);
route(3);
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method(BYE)) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
record_route();
}
# route it out to whatever destination was set by loose_route()
# in $du (destination URI).
route(1);
} else {
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK; must be an ACK after
# a 487 or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching transaction -
# ignore and discard
exit;
}
}
sl_send_reply(404,Not here);
}
exit;
}
#initial requests
# CANCEL processing
if (is_method(CANCEL))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
if (loose_route()) {
xlog(L_ERR,
Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]);
if (!is_method(ACK))
sl_send_reply(403,Preload Route denied);
exit;
}
# record routing
if (!is_method(REGISTER|MESSAGE))
record_route();
#$avp(i:27)=check_source_address(0);
#xlog(Check Source Address from Address TABLE : $(avp(i:27))\n);
$avp(s:checksrc) = check_source_address(0);
log(###\n);
xlog(Check
[OpenSIPS-Users] Opensips logging issue?
Hello list, We were used to disable logs (debug=0) due to problems regarding to old Openser versions, and this became standard when we were going to migrate from Test to Production environments. But in the last one we forget this setting, and the log was enabled in Production Env, but no one was concerned about this, we though that this problem didn' t happen in Opensips, but it did. So, the problem itself is: The Opensips receive a INVITE packet, and took too long to forward. Looking in the log, we could see the lines about the processing of INVITE (instantly forwarded). But the time between the packet arrival (capturing by ngrep) and your lines in the log its about 2, 3, 4 seconds of difference. This problem, as I said we already faced it on openser versions, and we only have disable logs (debug=0) in opensips.cfg, restart the service and it got back to work normally. Did someone faced this problem? -- Adelson ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Command opensipsctl dialplan reload randomly hangs
Hey Bogdan, Hereby some more tests (I should mention that I cannot block on demand the fifo, it simply randomly blocks, so I must be lucky when I am trying to reproduce it). Based on ps the cpu and memory consuption do not increase during the hang (CPU:0.0 Memory:1.0). Ta, DanB Log of the actions: --- Before reload: sip1:~# opensipsctl ps|grep FIFO Process:: ID=13 PID=26368 Type=MI FIFO sip1:~# ps uww -p 26368 USER PID %CPU %MEMVSZ RSS TTY STAT START TIME COMMAND opensips 26368 0.0 1.0 1147236 10788 ? S07:30 0:00 /usr/sbin/opensips -P /var/run/opensips/opensips.pid -m 1024 -u opensips -g opensips sip1:~# During hanging reload: sip1:~# ps uww -p 26368 USER PID %CPU %MEMVSZ RSS TTY STAT START TIME COMMAND opensips 26368 0.0 1.0 1147644 11040 ? R07:30 0:00 /usr/sbin/opensips -P /var/run/opensips/opensips.pid -m 1024 -u opensips -g opensips After reload interrupted with crtl+c and /etc/init.d/opensips restart sip1:~# ps uww -p 26368 USER PID %CPU %MEMVSZ RSS TTY STAT START TIME COMMAND sip1:~# opensipsctl ps|grep FIFO Process:: ID=13 PID=26716 Type=MI FIFO sip1:~# ps uww -p 26716 USER PID %CPU %MEMVSZ RSS TTY STAT START TIME COMMAND opensips 26716 0.0 0.1 1147164 1988 ?S14:59 0:00 /usr/sbin/opensips -P /var/run/opensips/opensips.pid -m 1024 -u opensips -g opensips On Mon, Apr 26, 2010 at 12:25 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Dan, Your descriptions point to a blocked fifo process. Blocking maybe because of some internal locking (you see 99% cpu usage) or some I/O (normal cpu usage). So, do the followings: 1) do opensipsctl fifo ps to see the PID of the fifo process 2) make fifo to block 3) check if the fifo process (by pid) is there - if yes, see how much cpu it uses and try to attache with gdb to it to get a backtrace. Regards, Bogdan DanB wrote: Hey Bogdan, Thanks for coming back so fast. There was no error reported neither on console nor in the syslog (debug 7). I will need to check for dead process since all I could spot was no reply back and console hanging, and be able to stop it only with Ctrl+C. After Ctrl+C could not get the any other opensipsctl commands to work. Will need to wait few days more to know about dead process. DanB On Thu, Apr 22, 2010 at 6:55 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Dan, Did you notice any error from the fifo process during the reload ? it may be something related to locking (during reload) of the table - this may affect all the other processes. So, any errors? any dead processes (like fifo one) ? Regards, Bogdan DanB wrote: Hey Guys, I have noticed the opensipsctl dialplan reload command randomly hanging, sometimes even the server itself becoming non responsible, other times reloading the dialplan into memory but not reporting anything on console, the last one becoming unusable until server restart. This happened in the past as well but with the traffic increase, it becomes more and more annoying. I suspect the same bug which was present in the past with fifo hanging. I should mention that I got about 2000 records in the dialplan table, so I would say not that much loaded. The version I am running: sip1:/home/employee/dan# opensips -V version: opensips 1.6.1-notls (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:6509M @(#) $Id: main.c 6169 2009-09-22 12:48:37Z bogdan_iancu $ main.c compiled on 14:43:30 Jan 11 2010 with gcc 4.3.2 All I could find in the logs was database reconnection in the case of hang but no error or something else reported (running debug 7). Bellow some of the log: Apr 22 14:27:19 sip1 /usr/sbin/opensips[16299]: DBG:mi_fifo:mi_fifo_server: entered consume Apr 22 14:27:19 sip1 /usr/sbin/opensips[16299]: DBG:mi_fifo:mi_fifo_server: done consume Apr 22 14:27:19 sip1 /usr/sbin/opensips[16299]: DBG:mi_fifo:mi_fifo_server: done parsing the mi tree Apr 22 14:27:19 sip1 /usr/sbin/opensips[16299]: DBG:dialplan:dp_load_db: init Apr 22 14:27:19 sip1 /usr/sbin/opensips[16299]: INFO:db_mysql:db_mysql_submit_query: disconect event for 0x77f060 Apr 22 14:27:19 sip1 /usr/sbin/opensips[16299]: INFO:db_mysql:reset_all_statements: reseting all statements on connection: (0x77fd18) 0x77f060 Apr 22 14:27:19 sip1 /usr/sbin/opensips[16299]: DBG:db_mysql:db_mysql_connect: opening connection: mysql://:x...@192.168.11.253/sipeandb Apr 22 14:27:19 sip1 /usr/sbin/opensips[16299]: DBG:db_mysql:db_mysql_connect: connection type is 192.168.11.253 via TCP/IP
Re: [OpenSIPS-Users] Question about B2BUA
Hi Vadim, B2BUA only handles calls. What do you mean by registering to another box? Regards, -- Anca Vamanu www.voice-system.ro Vadim Grinco wrote: Hi Guys, Do the B2Bua modules in Opensips permit the box to register to a remote sip proxy? If yes, can you please point me to an example or documentation how to implement that? Thanks, -- Best regards, Vadim Grinco ___ 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] Question about B2BUA
Salut Anca, Thank you for your answer. That's exactly what I expected it to be, but someone told me that it is possible to use opensips as UA to register to a ITSP proxy server, to be able to receive incoming calls from it. I was always using asterisk between opensips and my ITSP, and am very curious if I can get rid of the asterisk server. The ITSP requires auth, so I can't use rewritehostport or other ways of routing the calls to it. Cheers, On Thu, Apr 29, 2010 at 5:15 PM, Anca Vamanu a...@opensips.org wrote: Hi Vadim, B2BUA only handles calls. What do you mean by registering to another box? Regards, -- Anca Vamanu www.voice-system.ro Vadim Grinco wrote: Hi Guys, Do the B2Bua modules in Opensips permit the box to register to a remote sip proxy? If yes, can you please point me to an example or documentation how to implement that? Thanks, -- Best regards, Vadim Grinco ___ 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 -- Best regards, Vadim Grinco ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Send a request to multiple gateways
Hi All, Is there a way I can send a 'call pickup' request to multiple Asterisk servers at the same time, then the first one to answer gets the call? I think this is some sort of fork or branch? If anyone can offer any advice or examples I would be very greatful. Kind regards, Ross _ http://clk.atdmt.com/UKM/go/19780/direct/01/ We want to hear all your funny, exciting and crazy Hotmail stories. Tell us now___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS defunct processes
Hello, Certainly, here they are from opensips.cfg and I've included the modparam in case they help: loadmodule db_mysql.so loadmodule sl.so loadmodule tm.so loadmodule usrloc.so loadmodule auth.so loadmodule auth_db.so loadmodule maxfwd.so loadmodule mi_fifo.so loadmodule nathelper.so loadmodule perl.so loadmodule registrar.so loadmodule rr.so loadmodule textops.so loadmodule uri.so modparam( auth, nonce_expire, 30 ) modparam( auth_db|domain|uri_db|usrloc, db_url, mysql://foo ) modparam( auth_db, calculate_ha1, yes ) modparam( auth_db, password_column, secret ) modparam( auth_db, use_domain, 0 ) modparam( auth_db, user_column, name ) modparam( mi_fifo, fifo_name, /tmp/opensips_fifo ) modparam( nathelper, natping_interval, 240 ) modparam( nathelper, ping_nated_only, 1 ) modparam( nathelper, sipping_bflag, 1 ) modparam( nathelper, sipping_from, sip:keepal...@foo ) modparam( nathelper|registrar, received_avp, $avp(i:42) ) modparam( perl, filename, /path/to/OpenSIPS.pm ) modparam( perl, modpath, /path/to/perllib ) modparam( registrar, append_branches, 1 ) modparam( rr, enable_full_lr, 1 ) modparam( usrloc, db_mode, 2 ) modparam( usrloc, desc_time_order, 1 ) modparam( usrloc, nat_bflag, 1 ) modparam( usrloc, timer_interval, 5 ) Thank you! On Wed, Apr 28, 2010 at 4:53 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi David, by chance, using the exec module ? Or, can you list the modules you are using ? Regards, Bogdan David Cunningham wrote: Hello, Thank you for the reply. I checked the parent of the zombie processes, and they seem to be SIP receiver processes as per the following ps -ef extract and opensipsctl fifo ps information. We're not running the respawn patch. Any more advice very welcome, thanks again! user 5830 1 0 06:38 ? 00:00:00 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5832 5830 0 06:38 ? 00:00:16 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5833 5830 0 06:38 ? 00:00:16 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5834 5830 0 06:38 ? 00:00:16 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5835 5830 0 06:38 ? 00:00:16 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5836 5830 0 06:38 ? 00:00:16 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5837 5830 0 06:38 ? 00:00:16 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5838 5830 0 06:38 ? 00:00:16 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5839 5830 0 06:38 ? 00:00:16 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5840 5830 0 06:38 ? 00:00:01 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5841 5830 0 06:38 ? 00:00:00 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5842 5830 0 06:38 ? 00:00:00 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5843 5830 0 06:38 ? 00:00:00 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5844 5830 0 06:38 ? 00:00:00 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5845 5830 0 06:38 ? 00:00:00 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5846 5830 0 06:38 ? 00:00:00 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5847 5830 0 06:38 ? 00:00:00 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5848 5830 0 06:38 ? 00:00:00 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5849 5830 0 06:38 ? 00:00:00 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5850 5830 0 06:38 ? 00:00:00 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 5851 5830 0 06:38 ? 00:00:00 /sbin/opensips -m 256 -P /var/run/user/opensips.pid user 7260 5833 0 08:30 ? 00:00:00 [opensips] defunct user 7261 5833 0 08:30 ? 00:00:00 [opensips] defunct user 7262 5833 0 08:30 ? 00:00:00 [opensips] defunct user 7263 5833 0 08:30 ? 00:00:00 [opensips] defunct user 7264 5833 0 08:30 ? 00:00:00 [opensips] defunct user 7265 5833 0 08:30 ? 00:00:00 [opensips] defunct user 9770 5835 0 08:37 ? 00:00:00 [opensips] defunct user 9771 5835 0 08:37 ? 00:00:00 [opensips] defunct user 9772 5835 0 08:37 ? 00:00:00 [opensips] defunct user 9838 5834 0 08:38 ? 00:00:00 [opensips] defunct user 9839 5834 0 08:38 ? 00:00:00 [opensips] defunct user 15519 5833 0 08:57 ? 00:00:00 [opensips] defunct user 15520 5833 0 08:57 ? 00:00:00 [opensips] defunct user 15521 5833 0 08:57 ? 00:00:00 [opensips] defunct user 15522 5833 0 08:57 ? 00:00:00 [opensips] defunct [r...@hostname ~]# opensipsctl fifo ps Process:: ID=0 PID=5830 Type=attendant Process:: ID=1 PID=5832 Type=SIP receiver udp:xxx.xxx.xxx.xxx:5060 Process:: ID=2 PID=5833 Type=SIP receiver
Re: [OpenSIPS-Users] Autehntification from DB with AVPOPS module ?
Hi Sam, Use the auth module (pv_www_authorize and pv_proxy_authorize ) functions to do auth with a password you have form other place (and not from subscriber table). Regards, Bogdan samoh wrote: Hi everybody, Can I do the autehntification from DB with AVPOPS and UAC modules ? I want to do the authentification from DB (no opensips DB) but I don't know how to proceed. I don't know if my start's code is right ? (befor, I'd like to test on opesips DB): modparam(avpops,db_url,mysql://opensips:opensip...@localhost/opensips) modparam(avpops,avp_table,subscriber) modparam(uac, auth_username_avp, $avp(i:1)) modparam(uac, auth_password_avp, $avp(i:2)) modparam(uac, auth_realm_avp, $avp(i:3)) avp_db_query(SELECT username, password, realm FROM subscriber WHERE where username='$tu', $avp(i:1);$avp(i:2);$avp(i:3)); can anyone help me please ?. Thanks. Sam. -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Send a request to multiple gateways
Hi Ross, that is parallel forking, and you can do it simple like : RURI will hold the first destination, use append_branch() to add the rest of them and simply relay it out: $ru = A1; append_branch(A2); t_relay(); Regards, Bogdan Ross Beer wrote: Hi All, Is there a way I can send a 'call pickup' request to multiple Asterisk servers at the same time, then the first one to answer gets the call? I think this is some sort of fork or branch? If anyone can offer any advice or examples I would be very greatful. Kind regards, Ross Get a free e-mail account with Hotmail. Sign-up now. http://clk.atdmt.com/UKM/go/19780/direct/01/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] tm - module out of memory
tm module went out of memory, we did face other problems with a non compliant SIP device, so this could well be a side effect of it. Any recommendation on tm memory usage ? opensips log --- Apr 29 19:17:33 osip /usr/local/sbin/opensips[20101]: ERROR:tm:new_t: out of mem Apr 29 19:17:33 osip /usr/local/sbin/opensips[20101]: ERROR:tm:t_newtran: new_t failed ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] tm - module out of memory
I have to diagnostic this system further, as before facing a memory shortage, there is a huge bunch of ERROR:tm:t_check: reply cannot be parsed Apr 29 19:16:58 osip /usr/local/sbin/opensips[20101]: ***BRANCH ROUTE** Apr 29 19:16:58 osip /usr/local/sbin/opensips[20104]: ***REPLY ROUTE - td[10.1.16.50]** Apr 29 19:16:58 osip /usr/local/sbin/opensips[20103]: ERROR:tm:t_check: reply cannot be parsed Apr 29 19:16:58 osip /usr/local/sbin/opensips[20101]: ***HOST ROUTED *** From: users-boun...@lists.opensips.org on behalf of Julien Chavanton Sent: Thu 29/04/2010 8:27 PM To: Users@lists.opensips.org Subject: [OpenSIPS-Users] tm - module out of memory tm module went out of memory, we did face other problems with a non compliant SIP device, so this could well be a side effect of it. Any recommendation on tm memory usage ? opensips log --- Apr 29 19:17:33 osip /usr/local/sbin/opensips[20101]: ERROR:tm:new_t: out of mem Apr 29 19:17:33 osip /usr/local/sbin/opensips[20101]: ERROR:tm:t_newtran: new_t failed ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] NAT Problem using Nat helper
Ahmed,
Could you send an wireshark trace to the list? It will be easier to check
what's going wrong.
Besta regards,
Antonio Anderson M. Souza
Voice Technology
http://www.antonioams.com
Em 29/04/2010 11:47, Ahmed Munir ahmedmunir...@gmail.comescreveu:
Hi,
I've configured OpenSIPs with FreeRadius, the version for OpenSIPs I'm using
is 1.6.1 and FreeRadius verison is update 2 date. When I register 2
sofphone, they got authenticated and authorized by radius and got
registered sucessfully. Even I made calls between two softphone
sucessfully(Can hear one another). The UAS configured on different network
means hosted with public IP and my softphones are registered other and NATed
network. I mapped a DID on UAS and mapped it on my one of my softphone. The
problem I'm facing is when call coming from DID and ring my phone the caller
can hear me but I can't hear the caller(one way calling issue). But not
facing the problem on when calling between to sip clients and also calling
from OpenSIPs to Asterisk. The configuration and cases for OpenSIPs is
listed down below;
UAC-- UAS(OpenSIPs) -
UACtwo way voice is establised
UAC-- UAS(OpenSIPs) - Asterisk
UACtwo way voice is establised
PSTN-- UAS(OpenSIPs)
- UAC one way
voice is establised
(hears the dest voice)(can't hear caller
voice)
#loadmodule auth_diameter.so
loadmodule aaa_radius.so
loadmodule auth_aaa.so
loadmodule permissions.so
loadmodule nathelper.so
#Settings For
Radius-
#modparam(auth_diameter, diameter_client_host, localhost)
modparam(aaa_radius,
radius_config,/usr/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, aaa_url,
radius:/usr/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, aaa_flag, 2)
modparam(acc, aaa_missed_flag, 3)
modparam(acc, aaa_extra, User-Name=$Au; \
Calling-Station-Id=$from; \
Called-Station-Id=$to; \
Sip-Translated-Request-URI=$ruri; \
Sip-RPid=$avp(s:rpid); \
Source-IP=$si; \
Source-Port=$sp; \
Canonical-URI=$avp(s:can_uri); \
Billing-Party=$avp(s:billing_party); \
Divert-Reason=$avp(s:divert_reason); \
X-RTP-Stat=$hdr(X-RTP-Stat); \
Contact=$hdr(contact); \
Event=$hdr(event); \
SIP-Proxy-IP=$avp(s:sip_proxy_ip); \
ENUM-TLD=$avp(s:enum_tld))
modparam(auth_aaa,aaa_url,radius:/usr/etc/radiusclient-ng/radiusclient.conf)
modparam(auth, rpid_prefix, sip:)
modparam(auth, rpid_suffix, @11.22.33.44;screen=yes;privacy=off)
modparam(auth, rpid_avp, $avp(s:rpid))
#modparam(uri,service_type,10)
# - setting module-specific parameters ---
modparam(dispatcher, db_url, mysql://opensips:opensip...@localhost
/opensips)
modparam(permissions, db_url, mysql://opensips:opensip...@localhost
/opensips)
#- setting NAT module parameters -
modparam(nathelper,ping_nated_only,1)
modparam(nathelper, natping_interval, 30)
modparam(nathelper,natping_processes,1)
#modparam(nathelper,rtpproxy_sock,udp:127.0.0.1:7890)
modparam(nathelper,rtpproxy_sock, )
modparam(nathelper,received_avp,$avp(i:42))
#modparam(nathelper, sipping_bflag, 7)
modparam(usrloc, nat_bflag, 6)
### Routing Logic
# main request routing logic
route{
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}
#NAT detection
log(# Go to Route 3 for NAT
Detection #);
route(3);
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method(BYE)) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
record_route();
}
# route it out to whatever destination was set by loose_route()
# in $du (destination URI).
route(1);
} else {
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK; must be an ACK after
# a 487 or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching transaction -
# ignore and discard
exit;
}
}
sl_send_reply(404,Not here);
}
exit;
}
#initial requests
# CANCEL processing
if (is_method(CANCEL))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
if (loose_route()) {
xlog(L_ERR,
Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]);
if (!is_method(ACK))
sl_send_reply(403,Preload Route denied);
exit;
}
# record routing
if (!is_method(REGISTER|MESSAGE))
record_route();
[OpenSIPS-Users] RADIUS AVP related Error
I have configured Opensips, Radius and CDRTool. But while making a call i am getting following error and no Data is inserted into 'radacct' table. ERROR = rc_avpair_new: unknown attribute 0 Apr 29 06:19:09 opensips /sbin/opensips[32327]: ERROR:aaa_radius:rad_avp_add: failure Apr 29 06:19:09 opensips /sbin/opensips[32327]: ERROR:acc:acc_aaa_request: failed to add Bcontact, 10 Please help me, where should i look for ? Thanks, Vishu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] tm - module out of memory
I found is a routing loop with another equipement, I was able to find some trace, this is not an Opensips related problem anymore. From: users-boun...@lists.opensips.org on behalf of Julien Chavanton Sent: Thu 29/04/2010 11:22 PM To: OpenSIPS users mailling list; Users@lists.opensips.org Subject: Re: [OpenSIPS-Users] tm - module out of memory I have to diagnostic this system further, as before facing a memory shortage, there is a huge bunch of ERROR:tm:t_check: reply cannot be parsed Apr 29 19:16:58 osip /usr/local/sbin/opensips[20101]: ***BRANCH ROUTE** Apr 29 19:16:58 osip /usr/local/sbin/opensips[20104]: ***REPLY ROUTE - td[10.1.16.50]** Apr 29 19:16:58 osip /usr/local/sbin/opensips[20103]: ERROR:tm:t_check: reply cannot be parsed Apr 29 19:16:58 osip /usr/local/sbin/opensips[20101]: ***HOST ROUTED *** From: users-boun...@lists.opensips.org on behalf of Julien Chavanton Sent: Thu 29/04/2010 8:27 PM To: Users@lists.opensips.org Subject: [OpenSIPS-Users] tm - module out of memory tm module went out of memory, we did face other problems with a non compliant SIP device, so this could well be a side effect of it. Any recommendation on tm memory usage ? opensips log --- Apr 29 19:17:33 osip /usr/local/sbin/opensips[20101]: ERROR:tm:new_t: out of mem Apr 29 19:17:33 osip /usr/local/sbin/opensips[20101]: ERROR:tm:t_newtran: new_t failed ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] tm - module out of memory
For what it's worth, I have also experienced out of memory error occurring from large quantities of routing loops that I hadn't detected. The memory loss was so bad that opensips stopped processing calls (but never seemed to have crashed). Memory errors went away when I resolved the route loops. :) -Brett On Thu, Apr 29, 2010 at 7:22 PM, Julien Chavanton j...@atlastelecom.comwrote: I found is a routing loop with another equipement, I was able to find some trace, this is not an Opensips related problem anymore. -- *From:* users-boun...@lists.opensips.org on behalf of Julien Chavanton *Sent:* Thu 29/04/2010 11:22 PM *To:* OpenSIPS users mailling list; Users@lists.opensips.org *Subject:* Re: [OpenSIPS-Users] tm - module out of memory I have to diagnostic this system further, as before facing a memory shortage, there is a huge bunch of ERROR:tm:t_check: reply cannot be parsed Apr 29 19:16:58 osip /usr/local/sbin/opensips[20101]: ***BRANCH ROUTE** Apr 29 19:16:58 osip /usr/local/sbin/opensips[20104]: ***REPLY ROUTE - td[10.1.16.50]** Apr 29 19:16:58 osip /usr/local/sbin/opensips[20103]: ERROR:tm:t_check: reply cannot be parsed Apr 29 19:16:58 osip /usr/local/sbin/opensips[20101]: ***HOST ROUTED *** -- *From:* users-boun...@lists.opensips.org on behalf of Julien Chavanton *Sent:* Thu 29/04/2010 8:27 PM *To:* Users@lists.opensips.org *Subject:* [OpenSIPS-Users] tm - module out of memory tm module went out of memory, we did face other problems with a non compliant SIP device, so this could well be a side effect of it. Any recommendation on tm memory usage ? opensips log --- Apr 29 19:17:33 osip /usr/local/sbin/opensips[20101]: ERROR:tm:new_t: out of mem Apr 29 19:17:33 osip /usr/local/sbin/opensips[20101]: ERROR:tm:t_newtran: new_t failed ___ 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] spam on callcontrol website
Hi all, I found some more spam, this time on the callcontrol website. The front page (one essay link), Installation page (two essay links) and the tickets (three german spams). -- bye, pabs http://bonedaddy.net/pabs3/ signature.asc Description: This is a digitally signed message part ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] cdrtool rating computation problem
I'm having this problem:
http://opensips.wordpress.com/2009/06/26/cdrtool-rating-computation-problem/
Looking at the code, it seems to be caused by rating.php line 6443:
if (!preg_match(/^0/,$CDR-CanonicalURINormalized)) {
Should I be working around this line of code by prefixing a 0 to the
can_uri avp before calling call_control() in opensips.cfg?
--
bye,
pabs
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