Re: [OpenSIPS-Users] OpenSIPS or Kamailio
Toyima, This question is asked many time before :). Kamailio and OpenSIPS have the same roots. However, they differ more and more over time since each project is going it's own way. Bogdan has made a nice description of his (and the community's) vision on the future of Opensips: http://www.opensips.org/Development/NewDesign I would say: install both in e.g. a VMware environment and use what you like most. I've installed both kamailio and opensips and selected opensips in the end. My reason for choosing opensips: - Opensips is from my point of view more 'cleaner'. My experience with Kamailio was that it's looks pretty chaotic with the pre - past 3.0 modules/configs. For a beginner and noob like me that's not very useful - Opensips has a active IRC channel where you can ask questions and get (mostly) direct response. - Book (https://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book ) of opensips to study. - EbootCamp (http://www.opensips.org/Training/EBootcamp )to learn hands-on how opensips is working. All done via the Internet and good source for study either. Again, everything above is based on my own experience. But if you asked me, I definitely would recommend OpenSIPS. Erik Van: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] Namens Toyima Dias Verzonden: dinsdag 1 februari 2011 17:19 Aan: OpenSIPS users mailling list Onderwerp: [OpenSIPS-Users] OpenSIPS or Kamailio Hello Community, I've a question about OpenSIPS; Kamailio and OpenSIPS are the same application right? i mean, both are Proxy servers for high enterprise productions environmets...what i want to know is if there is any difference between them? why the Kamailio community is higher than OpenSIPS community, or at least seems to be like this...or not? Best Regards ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Memory leak
Hi, I am using opensips 1.6.2. I have modified the opensips.cfg script and it generates a memory leak. The memory leak seems to be on pkmem since when I use the following command 'opensipsctl fifo get_statistics pkmem: shmem: | grep free', pkmem dicreases I have followed the documentation from http://www.opensips.org/Resources/DocsTsMem, but unfortunately the memory status does not give any idea. What can I do to find the reason of the leak? Regards, Yannick LE COENT NEXCOM Systems ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] new install and INVITES not being forwarded
Hi Brett, Brett Nemeroff wrote: On Wed, Jan 12, 2011 at 9:04 AM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Gareth, Gareth Blades wrote: I am having a problem with running opensips in debug level 6. When opensips is set to this I am finding that it takes a long time to respond to register requests (over 5 seconds compared to a fraction of a second) which means that my phone times out when trying to register so I cannot then debug trying to send calls through. woow...that is really strange.never encountered something like thatdo you log to stderror(console) or to syslog ? Bogdan / Gareth, I've seen this on several installations. Exactly as you describe. If you search the archives, you'll see others have reported this as well. The problem seems to be in the syslog configuration. If you switch your configuration to use asynchronous writes, you'll see performance improve dramatically. In syslog, this is done by placing a minus sign before the filename. For example /var/log/opensips.log becomes -/var/log/opensips. Your distribution / syslogger may vary. And yes, this is apparent on a system with no load, and SINGLE SIP users. With full logging and no modifications to syslog, I've seen on *many* systems REGISTER timeouts. Took me a while to track it down; suspecting all sorts of other IO issues. Bogdan, I'm not sure if this should be expected or if it's an indication of a more serious problem in OpenSIPs. If it's not an indication of a coding problem, then I believe this information should be in the documentation somewhere (if it's not already)? The problem is not opensips related - the syslog is seen as an external I/O (as DB, DNS) which unfortunately blocks. And, yes, it may be a good idea to have it somewhere - for example you can add something to the Tips and FAQs section: http://www.opensips.org/Resources/DocsTipsFaqs Regards, Bogdan -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] memory trouble on 1.6.4
Hi Denis, Be sure you the latest version of 1.6.4 (SVN/tarball/debs) - there was a mem leak in the dialog module, issues that was fixed 1 or 2 weeks ago. Regards, bogdan Denis Putyato wrote: Hello! I have such error messages in syslog while using 1.6.4 Jan 24 09:40:04 opensips /usr/local/opensips1.6.4/sbin/opensips[10176]: WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation Jan 24 09:40:04 opensips /usr/local/opensips1.6.4/sbin/opensips[10176]: ERROR:core:build_res_buf_from_sip_req: out of pkg memory ; needs 477 Jan 24 09:40:04 opensips /usr/local/opensips1.6.4/sbin/opensips[10176]: WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation Jan 24 09:40:04 opensips /usr/local/opensips1.6.4/sbin/opensips[10176]: ERROR:core:db_allocate_columns: no private memory left After restart Opensips everything is OK but only during some time. I increase shm memory from 32M to 256M in config.h, but it didn`t help. I have such module loading: loadmodule db_mysql.so loadmodule sl.so loadmodule tm.so loadmodule signaling.so loadmodule auth.so loadmodule rr.so loadmodule maxfwd.so loadmodule textops.so loadmodule mi_fifo.so loadmodule uri.so loadmodule domain.so loadmodule drouting.so loadmodule siptrace.so loadmodule avpops.so loadmodule dialplan.so loadmodule dialog.so loadmodule permissions.so loadmodule usrloc.so loadmodule registrar.so loadmodule alias_db.so loadmodule auth_db.so loadmodule nathelper.so loadmodule acc.so loadmodule uac.so loadmodule aaa_radius.so For accounting I use cdr_flag feature. During call process Opensips makes about 4 SQL queries to db and receives from it some account parameters which stores to AVP. Thank you for any help ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Pointing auth_db at remote database, OpenSIPS is expecting a local-like schema?
Hi Ambert,
Just to be sure it is clear. When you configure a DB connection for a
certain module, OpenSIPS will use that particular connection to check in
a version table for the entries corresponding to that module.
So, for your case, opensips will check in the remote DB, in the version
table, only for the version of the subscriber table (used by auth_db)
module ; for the rest of the modules (pointing to local DB), the version
from the local Db will be queried.
Regards,
Bogdan
ambertch wrote:
I pointed auth_db at a remote DB and table via the following:
modparam(auth_db, db_url, mysql://user:pass@server/MY_REMOTE_DB)
...
if (!www_authorize(serve.name, users)) {
Doing this, it complains that the table MY_REMOTE_DB.version doesn't
exist.
When I create MY_REMOTE_DB.version and copy over opensips.version, it then
complains about users not being the correct version.
I addressed this by noting a db entry in opensips.version for the default
user storage table of subscriber = 7 and adding users = 7 that into
MY_DB_NAME.version
It works, but it looks to me that pointing to a remote db, some logic in
OpenSIPS is expecting the same things as if auth_db were pointed locally. Is
there any way to address this so I can remove the opensips version table
from MY_REMOTE_DB?
Thanks!
Ambert
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Re: [OpenSIPS-Users] My OpenSIPS apparently ignoring 100s
Hi Jock, Jock McKechnie wrote: Greetings; I apologise in advance for this one. I _know_ I screwed it up, but I just cannot see how. I'm sure it's something blazingly obvious, but I just cannot find it and it's driving me nuts. I've written an OpenSIPS config that uses an external perl 'helper' to do an LCR lookup (it incorporates a bunch more things that the built-in OpenSIPS LCR can't do, elsewise I'd use it), Have you looked at Dynamic Routing module (a more powerful LCR) - http://www.opensips.org/html/docs/modules/1.6.x/drouting.html I've rewritten the configuration several times over, and somewhere along the way I've borked it, I guess. When the system receives a call it'll do the LCR lookup, find a route, and sends the call out to that route. The gateway it sends the call to responds with a '100 Trying' and then a second later OpenSIPS sends the INVITE again, and gets another '100 Trying'. And then a second later, OpenSIPS sends the INVITE again, etc. Even when the call comes up, sometimes OpenSIPS isn't seeing the '200 OK' and continues sending INVITES until it times out the call. Set debug=6, make a call, and post the output somewhere - most probably the replies from GW are not matching the INVITE transactionbut let's see what the logs say. (attaching a SIP capture of the call will help) I've pasted the whole config here - I hate to gobsmack people with the inanity and insanity of the config, remember it is in development, but I think I really need to show 'everything' to have someone work out what the heck I'm doing wrong here. Config: http://pastebin.com/bqy9P9bt looks ok on a first look. Regards, Bogdan Thank you very much for your help! - Jock ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] call forwarding with replace from uri
Hi Jesse,
Lost me a bit between those pieces of script.
Anyhow, as far I understood from your problem, you want to change the
FROM URI in order to use the new value later in the script (in the auth
part). Well, this does not work - in opensips, the changes you do over
the message are not visible until the message is sent out. So you cannot
use your own changes later in script.
To avoid this issue, you can use a variable to store the correct value
of the FROM URI ($fu directly or the value from DB) - later in script,
use for auth (or other purposes) this variable in order to deal with the
FROM URI value (the right one).
Regards,
Bogdan
Jesse Cloutier wrote:
Hi list,
I having trouble with my script when trying to call forward by
reseting the $ru and doing a route(1)
My problem seems to be coming from the fact that I am changing my $fu
with uac_replace_from. When I xlog the $fu right before the route() It
shows the correct value (the original $fu before it was changed by
uac_replace_from). But on the request to the forwarded number it tries
to authenticate the user using the new value (the value that
uac_replace_from put in)
If I don't replace the $fu everything works fine.
Thanks A lot for any help!!
here is the relavant parts of my script:
Replacing the uri in the original request:
if (is_avp_set($avp(s:uri))) {
if (is_avp_set($avp(s:fromname))) {
xlog(L_INFO,Fromname set to
$avp(s:fromname) and URI set to $avp(s:uri));
uac_replace_from($avp(s:fromname),$avp(s:uri));
} else {
uac_replace_from(,$avp(s:uri));
xlog(L_INFO,Only Fromname Set);
}
}
The fowrwarding:
if(avp_db_load($ru,$avp(s:unavailcallfwd))) {
#xlog(call forward is set
to: $avp(s:unavailcallfwd));
avp_pushto($ru,$avp(s:unavailcallfwd));
xlog(call forward is set
to: $ru from $fu);
route(1);
exit;
}
And the proxy authorize
xlog(Checking if we should attempt authentication on $fu);
if (!(method==REGISTER))
{
#Do not authenticate calls from the gateways
xlog(Checking if its from a gateway);
if(!is_from_gw()) # This check is from the drouting module
{
xlog(Checking if it is an IP Authed IP);
if(!check_source_address(0, $avp(i:9)))
#This check looks in the address table
{
xlog(Checking if it is a subscriber);
xlog(from is $fu);
if (!proxy_authorize(, subscriber)) {
proxy_challenge(, 0);
xlog(Sent proxy challange to
$fu);
exit;
}
if (!db_check_from()) {
sl_send_reply(403,Forbidden
auth ID);
exit;
}
}
}
}
--
Jesse Cloutier
Network Administrator
Cronomagic Canada
5143411579 x210
je...@cronomagic.com
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Re: [OpenSIPS-Users] Permissions module
Hi Anton, It seams that variables are not supported for this function params. More or less as this functions are not so used in the late time (the idea of using files for data provisioning is deprecated). You have two options: - use dialplan module to set rules for checking the allowance (see http://www.opensips.org/html/docs/modules/1.6.x/dialplan.html#id249065 1.4.1.3). - open a feature request to see if variable support can be added. Regards, Bogdan Anton Zagorskiy wrote: Hi. Can I use pseudo variable as first parameter in the function allow_routing? I've made files domain1.allow, domain1.deny, domain2.allow, domain2.deny. Next, I'm calling allow_routing($dlg_val(cur_domain)). In the log file I see DBG:permissions:check_routing: no rules = allow any routing and nothing more from permissions module. In each file there is at least one rule. What I'm doing wrong? WBR, Anton Zagorskiy VoIP Developer, Oyster Telecom Phone.: +7 812 601-0666 Fax: +7 812 601-0593 a.zagors...@oyster-telecom.ru www.oyster-telecom.ru ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] BYE request for proper signalling
Hi Denis, From SIP point of view, the BYE must be sent to the contact URIs . I guess your contact is different than the layer3 IP because of some NAT presence, right ? if so, use fix_nated_contact() for INVITE and 200 OK, so that the received contact will be fixed with the layer3 IP, so the dialog module will use the contact with a useful info. Regards, Bogdan Denis Putyato wrote: Hello! I am using dialog module for control of call duration. When timeout of dialog expires I need Opensips send BYE not to caller and callee contact (which is stored during creation of dialog) but to IP address and port from which INVITE (caller) and 200 OK (callee) had been received. Thank you for any help ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to test if a message is from myself
Hi Dave,
do : if (src_ip==myself) {}
Regards,
Bogdan
Dave Singer wrote:
Is there any way to check if the source IP/port is one that opensips
is listening on or one ? something like if (sip:$si:$sp == myself) {
...bla; bla;}
Thanks
Dave
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Re: [OpenSIPS-Users] Installing OpenSIPS on Red Hat
Hi Guys, Just to clarify a bit here - from OpenSIPS perspective, all LINUX distro are fully supported (like compiling and runing) and most of UNIX-like OS (BSD, SOLARIS, etc). There is no difference is running OpenSIPS on Debian or RedHat - you just need to take care of the dependencies (packaging is different on the 2 distros) and the init script. Otherwise it is the same. Again, OpenSIPS is not Debian-only supported/focused. Regards, Bogdan Jeff Pyle wrote: Toyima, Adrian is right. We started on a CentOS infrastructure with Opensips. It works, but it's a pain. We're migrating to a complete Debian infrastructure. We started with Debian because of Opensips, Mediaproxy and CDRtool. But now that we understand it we find it to be much more lightweight, configurable and just easier than CentOS and the other RedHat derivatives. Is Debian better than CentOS? Not the question, and not the point. I can say in our experience it is far easier to manage Opensips-based systems in Debian than in CentOS. Xen is a lot more flexible, too, and we've made great use of that with Opensips. - Jeff From: Adrian Georgescu a...@ag-projects.com mailto:a...@ag-projects.com Reply-To: OpenSIPS users mailling list users@lists.opensips.org mailto:users@lists.opensips.org Date: Mon, 31 Jan 2011 11:35:13 -0500 To: OpenSIPS users mailling list users@lists.opensips.org mailto:users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Installing OpenSIPS on Red Hat You should use Debian in production as the software is developed on Debian. If you use Redhat you will always be behind new developments or any bug fixes as there might be nobody porting them to Redhat. Adrian On Jan 31, 2011, at 5:28 PM, Toyima Dias wrote: Hello, I've seen many information on how to install OpenSIPS in DEBIAN, but in my case i need to install it on Red Hat, i've found some information but not very well supported, i would like to have a very clean and stable installation for a very large and stable production server, is there any recommendations please? Regards ___ 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 OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Installing OpenSIPS on Red Hat
Hi John, Can I list you here: http://www.opensips.org/Resources/Downloads#osipmi Thanks and regards, Bogdan John Khvatov wrote: Hello. I maintain OpenSIPS package in fedora/epel. Of course, let me know if you have issues with OpenSIPS rpm package from official fedora/epel repos. On 31.01.2011, at 20:03, Adrian Georgescu wrote: Would the maintainers of those packages please step forward so that people who need support know exactly who to ask when in need? This would help everyone to better understand who is maintaing what so that there is no fear uncertainty and doubts but only sure things. Adrian -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] BYE request for proper signalling
Hello Bogdan because of some NAT presence, right ? No, I need use IP address when there is more than one SIP proxy in call path. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Wednesday, February 02, 2011 3:36 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] BYE request for proper signalling Hi Denis, From SIP point of view, the BYE must be sent to the contact URIs . I guess your contact is different than the layer3 IP because of some NAT presence, right ? if so, use fix_nated_contact() for INVITE and 200 OK, so that the received contact will be fixed with the layer3 IP, so the dialog module will use the contact with a useful info. Regards, Bogdan Denis Putyato wrote: Hello! I am using dialog module for control of call duration. When timeout of dialog expires I need Opensips send BYE not to caller and callee contact (which is stored during creation of dialog) but to IP address and port from which INVITE (caller) and 200 OK (callee) had been received. Thank you for any help ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA OpenSIPS solutions and know-how ___ 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] drouting / is_from_gw - matching for groups and not types
Hello, regarding opensips-cp and drouting i came across a small problem, maybe someone already tried something similar and wants to share his knowledge :) | opensips-cp -- Drouting / Settings, Gateway Types / Group ID?s is what i am talking about. | Is there any function to check for the Group ID?s instead of Gateway types inside the routing script? |is_from_gw and goes_to_gw only checks for types of Gateways but i can not find any equivalent to check for gateway group ids? The Group ids are assigned via permissions and i am selecting the group ids via avp_db_query. My goal is to decide by group ids which calls (permissions/group-based) are routed directly to load_balance function instead of going through the normal drouting process of rules/gateway(lists). I could eventually use a avp_db_query to get the group id for every call but this would probably use lots of Database Resources? Maybe there is another smarter way to do all of this? Best Regards Max M. | ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] $auth.resp script variable
Hello. There is problem with $auth.resp script variable. Line from opensips.cfg: xlog(L_INFO, $auth.resp\n); opensips -f opensips.cfg -c results: Feb 2 16:33:12 [24453] NOTICE:core:main: config file ok, exiting... Runtime error: Feb 2 16:34:27 aki /usr/sbin/opensips[24485]: ERROR:core:pv_parse_spec: pvar auth.resp not found Feb 2 16:34:27 aki /usr/sbin/opensips[24485]: ERROR:core:pv_parse_spec: wrong char [p/112] in [$auth.resp#012] at [9 (0)] Feb 2 16:34:27 aki /usr/sbin/opensips[24485]: ERROR:core:fix_actions: wrong fomat [$auth.resp#012] for value param Feb 2 16:34:27 aki /usr/sbin/opensips[24485]: ERROR:core:fix_actions: fixing failed (code=-5) at cfg line 216 Feb 2 16:34:27 aki /usr/sbin/opensips[24485]: ERROR:core:main: failed to fix configuration with err code -5 -- WBR, John Khvatov ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Installing OpenSIPS on Red Hat
Hi Bogdan. On 02.02.2011, at 15:45, Bogdan-Andrei Iancu wrote: Hi John, Can I list you here: http://www.opensips.org/Resources/Downloads#osipmi Sure. Also, you can update 'Fedora' (link: 'http://fedoraproject.org', latest version of OpenSIPS available in all currently supported Fedora branches) and add 'EPEL' with link: http://fedoraproject.org/wiki/EPEL. John Khvatov wrote: Hello. I maintain OpenSIPS package in fedora/epel. Of course, let me know if you have issues with OpenSIPS rpm package from official fedora/epel repos. -- WBR, John Khvatov ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Installing OpenSIPS on Red Hat
Anyone doing Fedora RPMs for MediaProxy? I have only a CentOS link here: http://mediaproxy-ng.org/wiki/InstallationGuide Adrian On Feb 2, 2011, at 8:52 AM, John Khvatov wrote: Hi Bogdan. On 02.02.2011, at 15:45, Bogdan-Andrei Iancu wrote: Hi John, Can I list you here: http://www.opensips.org/Resources/Downloads#osipmi Sure. Also, you can update 'Fedora' (link: 'http://fedoraproject.org', latest version of OpenSIPS available in all currently supported Fedora branches) and add 'EPEL' with link: http://fedoraproject.org/wiki/EPEL. John Khvatov wrote: Hello. I maintain OpenSIPS package in fedora/epel. Of course, let me know if you have issues with OpenSIPS rpm package from official fedora/epel repos. -- WBR, John Khvatov ___ 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] Lookup contact from user part of RURI
Hi, using opensips 1.6.2. I am trying to use the user part of incoming RURI
to look up a contact, reason being is full RURI is incorrect, this is due to
bogus proxy upstream so need a workaround.
lookup(location) seems to be only if you use AOR.
For exmaple i need to reroute incoming ACK to real address of UA
So i would like to lookup 1234 user part of RURI below and rewrite the
RURI with the correct ip. 12.34.56.78 is Opensips IP rather than end UA ip.
1...@domain.com is stored in db. How do i lookup contact just with user part
and rewrite the RURI.
ie ACK sip:1234@12.34.56.78;rinstance=A89B5393
Need something for below
if(method==ACK)
{
xlog(ACK received \n);
if( $rd == 12.34.56.78) // check if opensips ip
{
lookup(user); // ??? // need to lookup with user or rinstance
// rewrite RURI with correct address
}
}
Hope its clear, thanks
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Re: [OpenSIPS-Users] Lookup contact from user part of RURI
Hi,
you could set OpenSIPS to not use domain part of uri, so your issue is
solved.
stefano
Il 02/02/2011 15:30, Nauman Sulaiman ha scritto:
Hi, using opensips 1.6.2. I am trying to use the user part of incoming RURI
to look up a contact, reason being is full RURI is incorrect, this is due to
bogus proxy upstream so need a workaround.
lookup(location) seems to be only if you use AOR.
For exmaple i need to reroute incoming ACK to real address of UA
So i would like to lookup 1234 user part of RURI below and rewrite the
RURI with the correct ip. 12.34.56.78 is Opensips IP rather than end UA ip.
1...@domain.com is stored in db. How do i lookup contact just with user part
and rewrite the RURI.
ie ACK sip:1234@12.34.56.78;rinstance=A89B5393
Need something for below
if(method==ACK)
{
xlog(ACK received \n);
if( $rd == 12.34.56.78) // check if opensips ip
{
lookup(user); // ??? // need to lookup with user or rinstance
// rewrite RURI with correct address
}
}
Hope its clear, thanks
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Re: [OpenSIPS-Users] call forwarding with replace from uri
Hi Jesse,
in your script you are replacing from header twice.
Double check to your script and delete the second uac_replace_from.
This function can be used just once a phone call.
ciao
s
Il 02/02/2011 16:41, Jesse Cloutier ha scritto:
Thanks for the answer, I was not very clear in my first email though,
sorry for that. Basically when caller A initiates a call, his $fu may
be pulled from the db and replaced with a uac_replace_from. Then if
the call progresses and is redirected to a new destination the call
fails to authenticate because the new FROM URI is not in the DB. I
have tried to restore his original FROM URI using uac_restore_from,
but this doesnt seem to do it and if I store the original uri and call
uac_replace_from a second time I get something like
sip:111@1.1.1.1sip:222@1.1.1.1
Multumesc!
On Wed 02 Feb 2011 07:28:28 AM EST, Bogdan-Andrei Iancu wrote:
Hi Jesse,
Lost me a bit between those pieces of script.
Anyhow, as far I understood from your problem, you want to change the
FROM URI in order to use the new value later in the script (in the
auth part). Well, this does not work - in opensips, the changes you
do over the message are not visible until the message is sent out. So
you cannot use your own changes later in script.
To avoid this issue, you can use a variable to store the correct
value of the FROM URI ($fu directly or the value from DB) - later in
script, use for auth (or other purposes) this variable in order to
deal with the FROM URI value (the right one).
Regards,
Bogdan
Jesse Cloutier wrote:
Hi list,
I having trouble with my script when trying to call forward by
reseting the $ru and doing a route(1)
My problem seems to be coming from the fact that I am changing my
$fu with uac_replace_from. When I xlog the $fu right before the
route() It shows the correct value (the original $fu before it was
changed by uac_replace_from). But on the request to the forwarded
number it tries to authenticate the user using the new value (the
value that uac_replace_from put in)
If I don't replace the $fu everything works fine.
Thanks A lot for any help!!
here is the relavant parts of my script:
Replacing the uri in the original request:
if (is_avp_set($avp(s:uri))) {
if (is_avp_set($avp(s:fromname))) {
xlog(L_INFO,Fromname set to $avp(s:fromname) and URI set to
$avp(s:uri));
uac_replace_from($avp(s:fromname),$avp(s:uri));
} else {
uac_replace_from(,$avp(s:uri));
xlog(L_INFO,Only Fromname Set);
}
}
The fowrwarding:
if(avp_db_load($ru,$avp(s:unavailcallfwd))) {
#xlog(call forward is set to: $avp(s:unavailcallfwd));
avp_pushto($ru,$avp(s:unavailcallfwd));
xlog(call forward is set to: $ru from $fu);
route(1);
exit;
}
And the proxy authorize
xlog(Checking if we should attempt authentication on $fu);
if (!(method==REGISTER))
{
#Do not authenticate calls from the gateways
xlog(Checking if its from a gateway);
if(!is_from_gw()) # This check is from the drouting module
{
xlog(Checking if it is an IP Authed IP);
if(!check_source_address(0, $avp(i:9))) #This check looks in the
address table
{
xlog(Checking if it is a subscriber);
xlog(from is $fu);
if (!proxy_authorize(, subscriber)) {
proxy_challenge(, 0);
xlog(Sent proxy challange to $fu);
exit;
}
if (!db_check_from()) {
sl_send_reply(403,Forbidden auth ID);
exit;
}
}
}
}
--
Jesse Cloutier
Network Administrator
Cronomagic Canada
5143411579 x210
je...@cronomagic.com
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Re: [OpenSIPS-Users] Logging to many files
On Wed, Feb 2, 2011 at 10:10 AM, Anton Zagorskiy a.zagors...@oyster-telecom.ru wrote: Hi. Is it possible to log not to a 1 file via xlog? Not entirely sure what you are asking, but it's probably not an opensips question. It's probably a syslog question. To log to more than one file, you'll need to set syslog to log to more than one file and then set xlog to log to more than one facility (facility?) -Brett ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Nathelper / rtpproxy , problem with trusted IP
Hello all, Small problem with rtpproxy and nathelper as you can see in the log bellow , I use the force_rtp_proxy method with the 'r' flag to indicate to trust the IP of the SDP. In the example bellow, the trusted IP is 95.128.80.6 3966 , the problem is that the old rtp source (in this case a MoH server) continue to send 1 or 2 packet so the RTP proxy change them with 95.128.80.92:1086 so packet are not correctly forwarded. If the IP must be trusted why he take care of a received Packet ? I use the rtpproxy 1.2.1. Regards Laurent Feb 2 16:58:42 dns2 kamailio-1.5.4-notls/kamailio[31518]: route 5 | INVITE has SDP | we trust the IP in SDP Feb 2 16:58:42 dns2 kamailio-1.5.4-notls/kamailio[31518]: route 5 | force_rtp_proxy ocrf Feb 2 16:58:42 dns2 rtpproxy[8101]: DBUG:handle_command: received command 31518_84 Uc0,101 408970232 95.128.80.6 3966 16e48326;1 2107358281;1 Feb 2 16:58:42 dns2 rtpproxy[8101]: INFO:handle_command: adding strong flag to existing session, new=1/0/0 Feb 2 16:58:42 dns2 rtpproxy[8101]: INFO:handle_command: lookup on ports 3162/3820, session timer restarted Feb 2 16:58:42 dns2 rtpproxy[8101]: INFO:handle_command: update Feb 2 16:58:42 dns2 rtpproxy[8101]: INFO:handle_command: Unless the address provided by client historically Feb 2 16:58:42 dns2 rtpproxy[8101]: INFO:handle_command: pre-filling callee's address with 95.128.80.6:3966 Feb 2 16:58:42 dns2 rtpproxy[8101]: DBUG:doreply: sending reply 31518_84 3820 95.128.80.6 Feb 2 16:58:42 dns2 rtpproxy[8101]: INFO:rxmit_packets: callee's address filled in: 95.128.80.92:1086 (RTP) Feb 2 16:58:42 dns2 rtpproxy[8101]: INFO:rxmit_packets: guessing RTCP port for callee to be 1087 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] B2B + top hiding + origination media IP
Hi Guys,
I am testing the following call flow:
Soft Phone = opensips (configured for B2B) = third party termination SIP
proxy
Here is my config:
modparam(b2b_entities, script_req_route, b2b_request)
modparam(b2b_entities, script_reply_route, b2b_reply)
local_route {
xlog(LOCAL_ROUTE ($rm - $rr)\n);
setflag(22);
if (is_method(INVITE)) {
engage_rtp_proxy(e,OPENSIPS_IP);
exit;
}
else if (is_method(BYE) ) {
xlog(BYE\n);
}
}
route[b2b_request] {
$avp(s:source_ip_address) := $si;
perl_exec(messagedump_route, messages);
xlog(b2b_request ($ci) ($rm - $rr)\n);
}
route[b2b_reply] {
$avp(s:source_ip_address) := $si;
perl_exec(messagedump_reply, messages);
xlog(b2b_reply ($ci) - $rm - $rr\n);
}
route{
...
if (is_method(INVITE) perl_exec(check_for_forwarding_number))
{
engage_rtp_proxy(e,OPENSIPS_IP);
xlog(LOG: INVITE AUTHENTICATED TO: $avp(s:uid) ; FWD TO:
$avp(s:fwd_ip)\n);
setflag(1); # do accounting
xlog(L_ERR, LOG: to uri=[$tu] [$avp(s:sip_proxy_ip)]\n);
b2b_init_request(top hiding);
exit;
};
...
}
What happens is:
- INVITE from the soft phone to the opensips
- catched by the B2B and relayed to the third party SIP proxy + trying
returned to the soft phone
- Session Progress received from the third party SIP proxy - opensips -
my soft phone
At that stage, here is what I have on the soft phone log:
18:56:50 UDP Packet Received from OPENSIPS_IP:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.2:5070;rport=5070;branch=z9hG4bK673604
To: sip:359883409291@OPENSIPS_DOMAIN:5060;tag=B2B.113.667
From: 359883327749 sip:359883327749@OPENSIPS_DOMAIN:5060;tag=1040
Call-ID: 1296636915-3604-SALASWORK@192.168.1.2
CSeq: 361 INVITE
Content-Type: application/sdp
Contact: sip:OPENSIPS_IP:5060;transport=udp
Server: OpenSIPS (1.6.3-notls (x86_64/linux))
Content-Length: 184
v=0
o=SBCSIPUAS 900116523 1 IN IP4 THIRD_PARTY_SIP_PROXY_IP
s=SBCSIPUAS SIP STACK v1.0
c=IN IP4 THIRD_PARTY_SIP_PROXY_IP
t=0 0
m=audio 17900 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=maxptime:30
As can be seen, the media IP is not rewritten by the opensips and the IP
passed to my soft phone is the IP of the termination IP for the opensips
(i.e. the third party SIP proxy IP). Because of that, my soft phone starts
the RTP directly to my provider instead trough the RTP proxy that is
attached to the opensips.
Just to clarify, the media IP of my soft phone is not passed to my provider
- that case is handled good.
Any idea what is missing ?
Thanks in advance.
-- Kamen
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Re: [OpenSIPS-Users] My OpenSIPS apparently ignoring 100s
On Wed, Feb 2, 2011 at 6:20 AM, Bogdan-Andrei Iancu bog...@opensips.orgwrote: Hi Jock, Jock McKechnie wrote: Greetings; I apologise in advance for this one. I _know_ I screwed it up, but I just cannot see how. I'm sure it's something blazingly obvious, but I just cannot find it and it's driving me nuts. I've written an OpenSIPS config that uses an external perl 'helper' to do an LCR lookup (it incorporates a bunch more things that the built-in OpenSIPS LCR can't do, elsewise I'd use it), Have you looked at Dynamic Routing module (a more powerful LCR) - http://www.opensips.org/html/docs/modules/1.6.x/drouting.html I've rewritten the configuration several times over, and somewhere along the way I've borked it, I guess. When the system receives a call it'll do the LCR lookup, find a route, and sends the call out to that route. The gateway it sends the call to responds with a '100 Trying' and then a second later OpenSIPS sends the INVITE again, and gets another '100 Trying'. And then a second later, OpenSIPS sends the INVITE again, etc. Even when the call comes up, sometimes OpenSIPS isn't seeing the '200 OK' and continues sending INVITES until it times out the call. Set debug=6, make a call, and post the output somewhere - most probably the replies from GW are not matching the INVITE transactionbut let's see what the logs say. (attaching a SIP capture of the call will help) Thanks, Bogdan. I'm staring at this and I'm not seeing where it's getting the '100 Tryings' at all, but perhaps it's forest/trees for me. I've stripped off all the syslog date/time headers, but during this time space it sent out the initial INVITE, received a 100, send a second INVITE, a second 100 back, received a 183 Session Progress (presumably from the first INVITE)... after the time frame included it sent another three INVITEs and received two 183s back before everything BYE'd out. [Wed Feb 2 09:05:38 2011] Attempting to relay call to sip:+1641456@192.168.1.99 sip%3A%2B1641456@192.168.1.99 DBG:tm:t_newtran: transaction on entrance=0x DBG:core:parse_headers: flags= DBG:core:parse_headers: flags=78 DBG:tm:t_lookup_request: start searching: hash=22751, isACK=0 DBG:tm:matching_3261: RFC3261 transaction matching failed DBG:tm:t_lookup_request: no transaction found DBG:tm:run_reqin_callbacks: trans=0x7f5d8c2a14e8, callback type 1, id 1 entered DBG:core:parse_headers: flags=78 DBG:dialog:new_dlg_val: inserting accX_created= DBG:tm:run_reqin_callbacks: trans=0x7f5d8c2a14e8, callback type 1, id 0 entered DBG:dialog:get_dlg_timeout: invalid AVP value, use default timeout DBG:core:parse_headers: flags= DBG:core:check_ip_address: params 10.10.101.101, 10.10.101.101, 0 DBG:core:_shm_resize: resize(0) called DBG:tm:_reply_light: reply sent out. buf=0x7b21d8: SIP/2.0 1..., shmem=0x7f5d8c2942b8: SIP/2.0 1 DBG:tm:_reply_light: finished DBG:core:mk_proxy: doing DNS lookup... DBG:tm:set_timer: relative timeout is 50 DBG:tm:insert_timer_unsafe: [4]: 0x7f5d8c2a1708 (44600) DBG:tm:set_timer: relative timeout is 30 DBG:tm:insert_timer_unsafe: [0]: 0x7f5d8c2a1738 (475) DBG:tm:t_relay_to: new transaction fwd'ed DBG:tm:t_unref: UNREF_UNSAFE: [0x7f5d8c2a14e8] after is 0 DBG:dialog:unref_dlg: unref dlg 0x7f5d8c294d68 with 1 - 2 DBG:core:destroy_avp_list: destroying list (nil) DBG:core:receive_msg: cleaning up DBG:tm:utimer_routine: timer routine:4,tl=0x7f5d8c2a1708 next=(nil), timeout=44600 DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x7f5d8c2a14e8, INVITE si ... ) DBG:tm:set_timer: relative timeout is 100 DBG:tm:insert_timer_unsafe: [5]: 0x7f5d8c2a1708 (44700) DBG:tm:retransmission_handler: retransmission_handler : done DBG:tm:utimer_routine: timer routine:5,tl=0x7f5d8c2a1708 next=(nil), timeout=44700 DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x7f5d8c2a14e8, INVITE si ... ) DBG:tm:utimer_routine: timer routine:7,tl=0x7f5d8c292a20 next=(nil), timeout=44800 DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x7f5d8c292800, INVITE si ... ) DBG:tm:set_timer: relative timeout is 400 DBG:tm:insert_timer_unsafe: [7]: 0x7f5d8c292a20 (45200) DBG:tm:retransmission_handler: retransmission_handler : done DBG:tm:utimer_routine: timer routine:6,tl=0x7f5d8c2a1708 next=(nil), timeout=44900 DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x7f5d8c2a14e8, INVITE si ... ) DBG:tm:set_timer: relative timeout is 400 DBG:tm:insert_timer_unsafe: [7]: 0x7f5d8c2a1708 (45300) DBG:tm:retransmission_handler: retransmission_handler : done DBG:tm:utimer_routine: timer routine:7,tl=0x7f5d8c2a4408 next=(nil), timeout=44900 DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x7f5d8c2a41e8, INVITE si ... ) DBG:tm:set_timer: relative timeout is 400 DBG:tm:insert_timer_unsafe: [7]:
[OpenSIPS-Users] not storing registers and auth not working
Hello, Something is happening with my configuration, every time i restart opensips, opensips lost the registrations and i have to restart the phonesby the way, i configured the authentication and is not workink, for non-Register and Register OpenSIPS is not sending the 401 or 407...i don't now what is going on...a will paste here my configs, opensips.cfg and openctlrc *opensips.cfg* # # $Id: opensips.cfg 7027 2010-07-15 13:48:29Z razvancrainea $ # # OpenSIPS basic configuration script # by Anca Vamanu a...@voice-system.ro # # Please refer to the Core CookBook at: # http://www.opensips.org/index.php?n=Resources.DocsCookbooks # for a explanation of possible statements, functions and parameters. # ### Global Parameters # debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 #total number of UDP SIP worker processes per interface /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes server_header=Server: OpenSIPS SIP Proxy /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes #total number of TCP SIP worker processes in total /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = //etc/opensips/tls/user/user-cert.pem #tls_private_key = //etc/opensips/tls/user/user-privkey.pem #tls_ca_list = //etc/opensips/tls/user/user-calist.pem port=5060 /* uncomment and configure the following line if you want opensips to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:172.30.140.8:5060 ### Modules Section #set module path mpath=//lib/opensips/modules/ /* uncomment next line for MySQL DB support */ 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 acc.so /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule auth.so loadmodule auth_db.so /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule alias_db.so /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see multi-module params section ) */ #loadmodule domain.so /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule presence.so #loadmodule presence_xml.so # - setting module-specific parameters --- # - mi_fifo params - modparam(mi_fifo, fifo_name, /tmp/opensips_fifo) # - rr params - # add value to ;lr param to cope with most of the UAs modparam(rr, enable_full_lr, 1) # do not append from tag to the RR (no need for this script) modparam(rr, append_fromtag, 0) # - registrar params - /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam(registrar, max_contacts, 10) # - usrloc params - modparam(usrloc, db_mode, 2) #All changes are made to memory and database synchronization is done in the timer /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam(usrloc, db_url,mysql://opensips:opensipsro@localhost/opensips) # - uri params - modparam(uri, use_uri_table, 0) # - acc params - /* what sepcial events should be accounted ? */ modparam(acc, early_media, 1) modparam(acc, report_ack, 1) modparam(acc, report_cancels, 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable append_fromtag in rr module */ modparam(acc, detect_direction, 0) /* account triggers (flags) */ modparam(acc, failed_transaction_flag, 3) modparam(acc, log_flag, 1) modparam(acc, log_missed_flag, 2) /* uncomment the following lines to enable DB accounting also */ modparam(acc, db_flag, 1) modparam(acc, db_missed_flag, 2) # - auth_db params - /* uncomment the following lines if you want to enable the DB based authentication */ modparam(auth_db, calculate_ha1, yes) modparam(auth_db, password_column, password) modparam(auth_db, db_url,mysql://opensips:opensipsro@localhost/opensips) modparam(auth_db,
Re: [OpenSIPS-Users] B2B + top hiding + origination media IP
The B2B module is operating on the received INVITE. Any changes that
you make to the received INVITE are not visible by the B2B module.
Use a proxy to perform whatever you want to do (rtpproxy, accounting,
etc.) and a separate server only for b2b (top hiding).
Regards,
Ovidiu Sas
On Wed, Feb 2, 2011 at 12:11 PM, Kamen Petrov kamen.pet...@gmail.com wrote:
Hi Guys,
I am testing the following call flow:
Soft Phone = opensips (configured for B2B) = third party termination SIP
proxy
Here is my config:
modparam(b2b_entities, script_req_route, b2b_request)
modparam(b2b_entities, script_reply_route, b2b_reply)
local_route {
xlog(LOCAL_ROUTE ($rm - $rr)\n);
setflag(22);
if (is_method(INVITE)) {
engage_rtp_proxy(e,OPENSIPS_IP);
exit;
}
else if (is_method(BYE) ) {
xlog(BYE\n);
}
}
route[b2b_request] {
$avp(s:source_ip_address) := $si;
perl_exec(messagedump_route, messages);
xlog(b2b_request ($ci) ($rm - $rr)\n);
}
route[b2b_reply] {
$avp(s:source_ip_address) := $si;
perl_exec(messagedump_reply, messages);
xlog(b2b_reply ($ci) - $rm - $rr\n);
}
route{
...
if (is_method(INVITE) perl_exec(check_for_forwarding_number))
{
engage_rtp_proxy(e,OPENSIPS_IP);
xlog(LOG: INVITE AUTHENTICATED TO: $avp(s:uid) ; FWD TO:
$avp(s:fwd_ip)\n);
setflag(1); # do accounting
xlog(L_ERR, LOG: to uri=[$tu] [$avp(s:sip_proxy_ip)]\n);
b2b_init_request(top hiding);
exit;
};
...
}
What happens is:
- INVITE from the soft phone to the opensips
- catched by the B2B and relayed to the third party SIP proxy + trying
returned to the soft phone
- Session Progress received from the third party SIP proxy - opensips -
my soft phone
At that stage, here is what I have on the soft phone log:
18:56:50 UDP Packet Received from OPENSIPS_IP:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.2:5070;rport=5070;branch=z9hG4bK673604
To: sip:359883409291@OPENSIPS_DOMAIN:5060;tag=B2B.113.667
From: 359883327749 sip:359883327749@OPENSIPS_DOMAIN:5060;tag=1040
Call-ID: 1296636915-3604-SALASWORK@192.168.1.2
CSeq: 361 INVITE
Content-Type: application/sdp
Contact: sip:OPENSIPS_IP:5060;transport=udp
Server: OpenSIPS (1.6.3-notls (x86_64/linux))
Content-Length: 184
v=0
o=SBCSIPUAS 900116523 1 IN IP4 THIRD_PARTY_SIP_PROXY_IP
s=SBCSIPUAS SIP STACK v1.0
c=IN IP4 THIRD_PARTY_SIP_PROXY_IP
t=0 0
m=audio 17900 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=maxptime:30
As can be seen, the media IP is not rewritten by the opensips and the IP
passed to my soft phone is the IP of the termination IP for the opensips
(i.e. the third party SIP proxy IP). Because of that, my soft phone starts
the RTP directly to my provider instead trough the RTP proxy that is
attached to the opensips.
Just to clarify, the media IP of my soft phone is not passed to my provider
- that case is handled good.
Any idea what is missing ?
Thanks in advance.
-- Kamen
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Re: [OpenSIPS-Users] Nathelper / rtpproxy , problem with trusted IP
How did you started the session? Maybe you should start the session from the beginning in trusted mode. Regards, Ovidiu Sas On Wed, Feb 2, 2011 at 11:26 AM, Laurent Schweizer laurent.schwei...@peoplefone.com wrote: Hello all, Small problem with rtpproxy and nathelper as you can see in the log bellow , I use the force_rtp_proxy method with the 'r' flag to indicate to trust the IP of the SDP. In the example bellow, the trusted IP is 95.128.80.6 3966 , the problem is that the old rtp source (in this case a MoH server) continue to send 1 or 2 packet so the RTP proxy change them with 95.128.80.92:1086 so packet are not correctly forwarded. If the IP must be trusted why he take care of a received Packet ? I use the rtpproxy 1.2.1. Regards Laurent Feb 2 16:58:42 dns2 kamailio-1.5.4-notls/kamailio[31518]: route 5 | INVITE has SDP | we trust the IP in SDP Feb 2 16:58:42 dns2 kamailio-1.5.4-notls/kamailio[31518]: route 5 | force_rtp_proxy ocrf Feb 2 16:58:42 dns2 rtpproxy[8101]: DBUG:handle_command: received command 31518_84 Uc0,101 408970232 95.128.80.6 3966 16e48326;1 2107358281;1 Feb 2 16:58:42 dns2 rtpproxy[8101]: INFO:handle_command: adding strong flag to existing session, new=1/0/0 Feb 2 16:58:42 dns2 rtpproxy[8101]: INFO:handle_command: lookup on ports 3162/3820, session timer restarted Feb 2 16:58:42 dns2 rtpproxy[8101]: INFO:handle_command: update Feb 2 16:58:42 dns2 rtpproxy[8101]: INFO:handle_command: Unless the address provided by client historically Feb 2 16:58:42 dns2 rtpproxy[8101]: INFO:handle_command: pre-filling callee's address with 95.128.80.6:3966 Feb 2 16:58:42 dns2 rtpproxy[8101]: DBUG:doreply: sending reply 31518_84 3820 95.128.80.6 Feb 2 16:58:42 dns2 rtpproxy[8101]: INFO:rxmit_packets: callee's address filled in: 95.128.80.92:1086 (RTP) Feb 2 16:58:42 dns2 rtpproxy[8101]: INFO:rxmit_packets: guessing RTCP port for callee to be 1087 ___ 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] B2B + top hiding + origination media IP
Hi Ovidu,
I do not perform any changes on the received invite.
The top hiding does it and the problem is.. it does not change only the
media IP. Everything else goes OK.
Are you saying the top hiding does not work properly with the nathelper ?
Thanks
-- Kamen
On 2 February 2011 19:41, Ovidiu Sas o...@voipembedded.com wrote:
The B2B module is operating on the received INVITE. Any changes that
you make to the received INVITE are not visible by the B2B module.
Use a proxy to perform whatever you want to do (rtpproxy, accounting,
etc.) and a separate server only for b2b (top hiding).
Regards,
Ovidiu Sas
On Wed, Feb 2, 2011 at 12:11 PM, Kamen Petrov kamen.pet...@gmail.com
wrote:
Hi Guys,
I am testing the following call flow:
Soft Phone = opensips (configured for B2B) = third party termination
SIP
proxy
Here is my config:
modparam(b2b_entities, script_req_route, b2b_request)
modparam(b2b_entities, script_reply_route, b2b_reply)
local_route {
xlog(LOCAL_ROUTE ($rm - $rr)\n);
setflag(22);
if (is_method(INVITE)) {
engage_rtp_proxy(e,OPENSIPS_IP);
exit;
}
else if (is_method(BYE) ) {
xlog(BYE\n);
}
}
route[b2b_request] {
$avp(s:source_ip_address) := $si;
perl_exec(messagedump_route, messages);
xlog(b2b_request ($ci) ($rm - $rr)\n);
}
route[b2b_reply] {
$avp(s:source_ip_address) := $si;
perl_exec(messagedump_reply, messages);
xlog(b2b_reply ($ci) - $rm - $rr\n);
}
route{
...
if (is_method(INVITE)
perl_exec(check_for_forwarding_number))
{
engage_rtp_proxy(e,OPENSIPS_IP);
xlog(LOG: INVITE AUTHENTICATED TO: $avp(s:uid) ; FWD TO:
$avp(s:fwd_ip)\n);
setflag(1); # do accounting
xlog(L_ERR, LOG: to uri=[$tu]
[$avp(s:sip_proxy_ip)]\n);
b2b_init_request(top hiding);
exit;
};
...
}
What happens is:
- INVITE from the soft phone to the opensips
- catched by the B2B and relayed to the third party SIP proxy + trying
returned to the soft phone
- Session Progress received from the third party SIP proxy - opensips
-
my soft phone
At that stage, here is what I have on the soft phone log:
18:56:50 UDP Packet Received from OPENSIPS_IP:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.2:5070;rport=5070;branch=z9hG4bK673604
To: sip:359883409291@OPENSIPS_DOMAIN:5060;tag=B2B.113.667
From: 359883327749 sip:359883327749@OPENSIPS_DOMAIN:5060;tag=1040
Call-ID: 1296636915-3604-SALASWORK@192.168.1.2
CSeq: 361 INVITE
Content-Type: application/sdp
Contact: sip:OPENSIPS_IP:5060;transport=udp
Server: OpenSIPS (1.6.3-notls (x86_64/linux))
Content-Length: 184
v=0
o=SBCSIPUAS 900116523 1 IN IP4 THIRD_PARTY_SIP_PROXY_IP
s=SBCSIPUAS SIP STACK v1.0
c=IN IP4 THIRD_PARTY_SIP_PROXY_IP
t=0 0
m=audio 17900 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=maxptime:30
As can be seen, the media IP is not rewritten by the opensips and the IP
passed to my soft phone is the IP of the termination IP for the opensips
(i.e. the third party SIP proxy IP). Because of that, my soft phone
starts
the RTP directly to my provider instead trough the RTP proxy that is
attached to the opensips.
Just to clarify, the media IP of my soft phone is not passed to my
provider
- that case is handled good.
Any idea what is missing ?
Thanks in advance.
-- Kamen
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Re: [OpenSIPS-Users] B2B + top hiding + origination media IP
The nathelper module is performing changes on the received INVITE
(changing the SDP).
Those changes are not visible by the b2b module and therefor discarded.
As a result, the nathelper module (and any module that is changing the
initial INVITE) doesn't work with the b2b module.
The only change visible to the b2b module is the RURI.
Regards,
Ovidiu Sas
On Wed, Feb 2, 2011 at 12:52 PM, Kamen Petrov kamen.pet...@gmail.com wrote:
Hi Ovidu,
I do not perform any changes on the received invite.
The top hiding does it and the problem is.. it does not change only the
media IP. Everything else goes OK.
Are you saying the top hiding does not work properly with the nathelper ?
Thanks
-- Kamen
On 2 February 2011 19:41, Ovidiu Sas o...@voipembedded.com wrote:
The B2B module is operating on the received INVITE. Any changes that
you make to the received INVITE are not visible by the B2B module.
Use a proxy to perform whatever you want to do (rtpproxy, accounting,
etc.) and a separate server only for b2b (top hiding).
Regards,
Ovidiu Sas
On Wed, Feb 2, 2011 at 12:11 PM, Kamen Petrov kamen.pet...@gmail.com
wrote:
Hi Guys,
I am testing the following call flow:
Soft Phone = opensips (configured for B2B) = third party termination
SIP
proxy
Here is my config:
modparam(b2b_entities, script_req_route, b2b_request)
modparam(b2b_entities, script_reply_route, b2b_reply)
local_route {
xlog(LOCAL_ROUTE ($rm - $rr)\n);
setflag(22);
if (is_method(INVITE)) {
engage_rtp_proxy(e,OPENSIPS_IP);
exit;
}
else if (is_method(BYE) ) {
xlog(BYE\n);
}
}
route[b2b_request] {
$avp(s:source_ip_address) := $si;
perl_exec(messagedump_route, messages);
xlog(b2b_request ($ci) ($rm - $rr)\n);
}
route[b2b_reply] {
$avp(s:source_ip_address) := $si;
perl_exec(messagedump_reply, messages);
xlog(b2b_reply ($ci) - $rm - $rr\n);
}
route{
...
if (is_method(INVITE)
perl_exec(check_for_forwarding_number))
{
engage_rtp_proxy(e,OPENSIPS_IP);
xlog(LOG: INVITE AUTHENTICATED TO: $avp(s:uid) ; FWD
TO:
$avp(s:fwd_ip)\n);
setflag(1); # do accounting
xlog(L_ERR, LOG: to uri=[$tu]
[$avp(s:sip_proxy_ip)]\n);
b2b_init_request(top hiding);
exit;
};
...
}
What happens is:
- INVITE from the soft phone to the opensips
- catched by the B2B and relayed to the third party SIP proxy + trying
returned to the soft phone
- Session Progress received from the third party SIP proxy - opensips
-
my soft phone
At that stage, here is what I have on the soft phone log:
18:56:50 UDP Packet Received from OPENSIPS_IP:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.2:5070;rport=5070;branch=z9hG4bK673604
To: sip:359883409291@OPENSIPS_DOMAIN:5060;tag=B2B.113.667
From: 359883327749 sip:359883327749@OPENSIPS_DOMAIN:5060;tag=1040
Call-ID: 1296636915-3604-SALASWORK@192.168.1.2
CSeq: 361 INVITE
Content-Type: application/sdp
Contact: sip:OPENSIPS_IP:5060;transport=udp
Server: OpenSIPS (1.6.3-notls (x86_64/linux))
Content-Length: 184
v=0
o=SBCSIPUAS 900116523 1 IN IP4 THIRD_PARTY_SIP_PROXY_IP
s=SBCSIPUAS SIP STACK v1.0
c=IN IP4 THIRD_PARTY_SIP_PROXY_IP
t=0 0
m=audio 17900 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=maxptime:30
As can be seen, the media IP is not rewritten by the opensips and the IP
passed to my soft phone is the IP of the termination IP for the opensips
(i.e. the third party SIP proxy IP). Because of that, my soft phone
starts
the RTP directly to my provider instead trough the RTP proxy that is
attached to the opensips.
Just to clarify, the media IP of my soft phone is not passed to my
provider
- that case is handled good.
Any idea what is missing ?
Thanks in advance.
-- Kamen
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Re: [OpenSIPS-Users] Pointing auth_db at remote database, OpenSIPS is expecting a local-like schema?
O ok thanks for the clarification. Just have to make sure to include
that in my web server code to make sure that table is always there
then.
On Wed, Feb 2, 2011 at 4:11 AM, Bogdan-Andrei Iancu-2 [via OpenSIPS
(Open SIP Server)] ml-node+5984529-875472476-313...@n2.nabble.com
wrote:
Hi Ambert,
Just to be sure it is clear. When you configure a DB connection for a
certain module, OpenSIPS will use that particular connection to check in
a version table for the entries corresponding to that module.
So, for your case, opensips will check in the remote DB, in the version
table, only for the version of the subscriber table (used by auth_db)
module ; for the rest of the modules (pointing to local DB), the version
from the local Db will be queried.
Regards,
Bogdan
ambertch wrote:
I pointed auth_db at a remote DB and table via the following:
modparam(auth_db, db_url, mysql://user:pass@server/MY_REMOTE_DB)
...
if (!www_authorize(serve.name, users)) {
Doing this, it complains that the table MY_REMOTE_DB.version doesn't
exist.
When I create MY_REMOTE_DB.version and copy over opensips.version, it then
complains about users not being the correct version.
I addressed this by noting a db entry in opensips.version for the default
user storage table of subscriber = 7 and adding users = 7 that into
MY_DB_NAME.version
It works, but it looks to me that pointing to a remote db, some logic in
OpenSIPS is expecting the same things as if auth_db were pointed locally.
Is
there any way to address this so I can remove the opensips version table
from MY_REMOTE_DB?
Thanks!
Ambert
--
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OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami, USA
OpenSIPS solutions and know-how
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Re: [OpenSIPS-Users] B2B + top hiding + origination media IP
yes!
On Wed, Feb 2, 2011 at 1:12 PM, Kamen Petrov kamen.pet...@gmail.com wrote:
Ok, that brings some clarification. Thanks.
So the correct scenario is:
1) softphone --registered to-- opensips A (pure)
2) call is relayed from opensips A to opensips B (the B2B one)
3) the opensips B connects to the termination
4) the RTP goes between the softphone - opensips A - rtpproxy
Would that work ? :)
On 2 February 2011 20:04, Ovidiu Sas o...@voipembedded.com wrote:
The nathelper module is performing changes on the received INVITE
(changing the SDP).
Those changes are not visible by the b2b module and therefor discarded.
As a result, the nathelper module (and any module that is changing the
initial INVITE) doesn't work with the b2b module.
The only change visible to the b2b module is the RURI.
Regards,
Ovidiu Sas
On Wed, Feb 2, 2011 at 12:52 PM, Kamen Petrov kamen.pet...@gmail.com
wrote:
Hi Ovidu,
I do not perform any changes on the received invite.
The top hiding does it and the problem is.. it does not change only
the
media IP. Everything else goes OK.
Are you saying the top hiding does not work properly with the
nathelper ?
Thanks
-- Kamen
On 2 February 2011 19:41, Ovidiu Sas o...@voipembedded.com wrote:
The B2B module is operating on the received INVITE. Any changes that
you make to the received INVITE are not visible by the B2B module.
Use a proxy to perform whatever you want to do (rtpproxy, accounting,
etc.) and a separate server only for b2b (top hiding).
Regards,
Ovidiu Sas
On Wed, Feb 2, 2011 at 12:11 PM, Kamen Petrov kamen.pet...@gmail.com
wrote:
Hi Guys,
I am testing the following call flow:
Soft Phone = opensips (configured for B2B) = third party
termination
SIP
proxy
Here is my config:
modparam(b2b_entities, script_req_route, b2b_request)
modparam(b2b_entities, script_reply_route, b2b_reply)
local_route {
xlog(LOCAL_ROUTE ($rm -
$rr)\n);
setflag(22);
if (is_method(INVITE)) {
engage_rtp_proxy(e,OPENSIPS_IP);
exit;
}
else if (is_method(BYE) ) {
xlog(BYE\n);
}
}
route[b2b_request] {
$avp(s:source_ip_address) := $si;
perl_exec(messagedump_route, messages);
xlog(b2b_request ($ci) ($rm - $rr)\n);
}
route[b2b_reply] {
$avp(s:source_ip_address) := $si;
perl_exec(messagedump_reply, messages);
xlog(b2b_reply ($ci) - $rm - $rr\n);
}
route{
...
if (is_method(INVITE)
perl_exec(check_for_forwarding_number))
{
engage_rtp_proxy(e,OPENSIPS_IP);
xlog(LOG: INVITE AUTHENTICATED TO: $avp(s:uid) ; FWD
TO:
$avp(s:fwd_ip)\n);
setflag(1); # do accounting
xlog(L_ERR, LOG: to uri=[$tu]
[$avp(s:sip_proxy_ip)]\n);
b2b_init_request(top hiding);
exit;
};
...
}
What happens is:
- INVITE from the soft phone to the opensips
- catched by the B2B and relayed to the third party SIP proxy +
trying
returned to the soft phone
- Session Progress received from the third party SIP proxy -
opensips
-
my soft phone
At that stage, here is what I have on the soft phone log:
18:56:50 UDP Packet Received from OPENSIPS_IP:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.2:5070;rport=5070;branch=z9hG4bK673604
To: sip:359883409291@OPENSIPS_DOMAIN:5060;tag=B2B.113.667
From: 359883327749
sip:359883327749@OPENSIPS_DOMAIN:5060;tag=1040
Call-ID: 1296636915-3604-SALASWORK@192.168.1.2
CSeq: 361 INVITE
Content-Type: application/sdp
Contact: sip:OPENSIPS_IP:5060;transport=udp
Server: OpenSIPS (1.6.3-notls (x86_64/linux))
Content-Length: 184
v=0
o=SBCSIPUAS 900116523 1 IN IP4 THIRD_PARTY_SIP_PROXY_IP
s=SBCSIPUAS SIP STACK v1.0
c=IN IP4 THIRD_PARTY_SIP_PROXY_IP
t=0 0
m=audio 17900 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=maxptime:30
As can be seen, the media IP is not rewritten by the opensips and the
IP
passed to my soft phone is the IP of the termination IP for the
opensips
(i.e. the third party SIP proxy IP). Because of that, my soft phone
starts
the RTP directly to my provider instead trough the RTP proxy that is
attached to the opensips.
Just to clarify, the media IP of my soft phone is not passed to my
provider
- that case is handled good.
Any idea what is missing ?
Thanks in advance.
-- Kamen
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[OpenSIPS-Users] MySQL tables using the opensipsdbctl shell script
Step 3: Create MySQL tables using the opensipsdbctl shell script. The syntax for this utility follows: opensipsdbctl create db name or db_path, optional I'm getting the following error for the above syntax bash: syntax error near unexpected token `newline' what's wrong here ? might be silly question ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to test if a message is from myself
Wow I missed that one. Thanks.
Does that work for PVs so I can test other IPs like one from another
header, say X-src-ip:192.168.0.5. Last I tried I couldn't get it to
work. Not sure if that was 1.6.2 or 1.6.3. I'm using 1.6.4 now. :)
Thanks Again
Dave
On Wed, Feb 2, 2011 at 4:37 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote:
Hi Dave,
do : if (src_ip==myself) {}
Regards,
Bogdan
Dave Singer wrote:
Is there any way to check if the source IP/port is one that opensips
is listening on or one ? something like if (sip:$si:$sp == myself) {
...bla; bla;}
Thanks
Dave
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2 - 4 February 2011, ITExpo, Miami, USA
OpenSIPS solutions and know-how
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Re: [OpenSIPS-Users] OpenSIPS 1.6 on Ubuntu
Anyone can help please in installing Opensip on Ubuntu. On Wed, Feb 2, 2011 at 1:15 AM, Duane Larson duane.lar...@gmail.com wrote: The first thing you should do is http://www.packtpub.com/article/installation-of-opensips-1.6 You can watch the webinars here http://www.opensips.org/Resources/Webinars You should join the mailing list http://www.opensips.org/Resources/MailingLists To search old mailing list posts I use http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html Sounds like what you need to do is to actually create a user/subscriber so that opensips can register the x-lite client. For that you need to use the opensipsctl command or the osipsconsole. On Tue, Feb 1, 2011 at 12:21 PM, Robin Malhotra rocky...@gmail.comwrote: Guys I a newbie to OpenSIPS I have installed opensips and mysql on ubuntu following some instructions. I have also installed x-lite. Now how to register a user in opensips and to use it with the client ? I am stuck, please let me know Regards Ricky ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- thanking you, Pradeep Patil Cell No: 9676206432 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS 1.6 on Ubuntu
The best place to start is http://www.opensips.org/ In the left column of the web page there is a section titled Resources with links to many very helpful resources. Your using the mailing list so you probably already have seen them to get here. So. Where are you getting stuck? We need specifics in order to help out. Also when you have a question you should start your own thread and not use an existing thread unless it is completely relevant to what your asking/stating. FYI: The webinars are VERY important for getting an understanding of how the whole thing works. With SIP the big picture is very important! With out them you'll learn a lot of things the hard way like I did before they were available. Another good way to learn is to follow the mailing list discussions. Welcome to the club, ;-) Dave P.S. The software, documentation, mailing list, IRC, etc are all free resources. The people helping you out are not getting paid to do it. So an attitude of appreciation with patience will get you the best millage. If you need more support there are those willing to do contract support. See http://www.opensips.org/Resources/Business On Wed, Feb 2, 2011 at 7:16 PM, Pradeep Patil pradeep.pati...@gmail.com wrote: Anyone can help please in installing Opensip on Ubuntu. On Wed, Feb 2, 2011 at 1:15 AM, Duane Larson duane.lar...@gmail.com wrote: The first thing you should do is http://www.packtpub.com/article/installation-of-opensips-1.6 You can watch the webinars here http://www.opensips.org/Resources/Webinars You should join the mailing list http://www.opensips.org/Resources/MailingLists To search old mailing list posts I use http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html Sounds like what you need to do is to actually create a user/subscriber so that opensips can register the x-lite client. For that you need to use the opensipsctl command or the osipsconsole. On Tue, Feb 1, 2011 at 12:21 PM, Robin Malhotra rocky...@gmail.com wrote: Guys I a newbie to OpenSIPS I have installed opensips and mysql on ubuntu following some instructions. I have also installed x-lite. Now how to register a user in opensips and to use it with the client ? I am stuck, please let me know Regards Ricky ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- thanking you, Pradeep Patil Cell No: 9676206432 ___ 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
