[OpenSIPS-Users] RTPProxy + OpenSIPS 1.7 Integration
Hello Everyone,
I am having trouble getting the RTPProxy 1.2.0 to work with OpenISP 1.7.
Starting the proxy using:
./rtpproxy -s udp:127.0.0.1:12221 -l public/private -p
/var/run/rtpproxy.pid -u root root -F -d INFO LOG_LOCAL0
opensips.cf relevant pieces
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:12221")
modparam("rtpproxy", "rtpproxy_autobridge", 1)
modparam("rtpproxy", "rtpproxy_timeout", "0.5")
modparam("rtpproxy", "rtpproxy_retr", 3)
if (has_totag() && is_method("INVITE")) {
engage_rtp_proxy("ie");
}
if (is_method("ACK") && has_body("application/sdp")) {
rtpproxy_answer();
}
route[1] {
xlog("Enter route 1");
if (has_body("application/sdp")) rtpproxy_answer();
if (is_method("INVITE")) {
xlog("New Call for route [ fu=$fu/ tu=$tu /ru=$ru/ ci=$ci]");
if (rtpproxy_offer()) {
t_on_reply("1");
}
else {
t_on_reply("2");
}
t_on_branch("2");
t_on_failure("1");
}
if (!t_relay()) {
sl_reply_error();
}
exit;
}
onreply_route[1] {
xlog("incoming reply\n");
if (has_body("application/sdp")) rtpproxy_answer();
exit;
}
onreply_route[2] {
xlog("incoming reply\n");
if (has_body("application/sdp")) rtpproxy_offer();
exit;
}
When starting OpenSIPS everything looks fine:
Dec 19 21:38:58 [3397] INFO:rtpproxy:rtpp_test: rtp proxy
found, support for it enabled
Dec 19 21:38:58 [3400] INFO:rtpproxy:rtpp_test: rtp proxy
found, support for it enabled
Dec 19 21:38:58 [3398] INFO:rtpproxy:rtpp_test: rtp proxy
found, support for it enabled
Dec 19 21:38:58 [3395] INFO:rtpproxy:rtpp_test: rtp proxy
found, support for it enabled
Dec 19 21:38:58 [3401] INFO:rtpproxy:rtpp_test: rtp proxy
found, support for it enabled
Dec 19 21:38:58 [3402] INFO:rtpproxy:rtpp_test: rtp proxy
found, support for it enabled
Making a call howver, yields the following error (watch it work now ;):
ERROR:rtpproxy:force_rtp_proxy_body: no available proxies
/var/log/syslog
Dec 19 21:28:18 opensips1 rtpproxy[3348]: INFO:main: rtpproxy started, pid 3348
Dec 19 21:28:31 opensips1 rtpproxy[3348]: INFO:handle_command: new
session f679c215-bae17257-a0a8a5b8@192.168.2.11, tag C0847B09-9D90A2;1
requested, type strong
Dec 19 21:28:31 opensips1 rtpproxy[3348]: ERR:create_twinlistener:
can't bind to the IPv4 port 50570: Cannot assign requested address
Dec 19 21:28:31 opensips1 rtpproxy[3348]: ERR:handle_command: can't
create listener
Dec 19 21:38:49 opensips1 rtpproxy[3379]: INFO:main: rtpproxy started, pid 3379
Dec 19 21:39:36 opensips1 rtpproxy[3379]: INFO:handle_command: new
session 5537fa8e-69de2248-78eef465@192.168.2.11, tag
A2311CB2-BB413627;1 requested, type strong
Dec 19 21:39:36 opensips1 rtpproxy[3379]: ERR:create_twinlistener:
can't bind to the IPv4 port 57636: Cannot assign requested address
Dec 19 21:39:36 opensips1 rtpproxy[3379]: ERR:handle_command: can't
create listener
Your help is greatly appreciated,
Nick.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] About msilo module config
Hi Friends I'm so sorry to trouble you. Who can tell me how to configure the msilo module to store all of the off line messages? Or give me a configure example? Thank you very much Regards,Kevin ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips-CP Dialplan Add Failure
Fixed specifying database parameters into /config/tools/system/dialplan/db.inc.php. I thought they were included in the global params….evidently not. Trevor Francis trevor.fran...@tgrahamcapital.com +1 (405) 757-7620 PO Box 54771 Oklahoma City, OK 73154 Personal Email: tfran...@fas.harvard.edu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips-CP Dialplan Add Failure
Whenever I add a dialplan entry in Opensips-CP the following occurs: First dial plan enters correctly. Second dialplan removes all of the expressions and leaves them blank. I also get an error: mysql_real_escape_string() expects parameter 2 to be resource, object given in /var/www/opensips-cp/web/tools/system/dialplan/dialplan.php on line 289 undefined index: dialplan_id in /var/www/opensips-cp/web/tools/system/dialplan/template/dialplan.main.php on line 28, All other modules work correctly. Trevor Francis trevor.fran...@tgrahamcapital.com +1 (405) 757-7620 PO Box 54771 Oklahoma City, OK 73154 Personal Email: tfran...@fas.harvard.edu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Blank domain field in subscriber table
I don't quite remember how we did it, but at a previous employer we migrated all our users from one domain to another by 'rewriting' all incoming messages to the new domain. All the accounts were converted to the be domain using a script just before going live We didn't quite rewrite, by substituted the domain when authenticating and perhaps when routing calls but it's been at least 7 years... -- Andreas ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Blank domain field in subscriber table
Sorry Schneur. Hopefully someone else can answer if this is possible. Otherwise I would start writing a script to add a domain to all the users in Subscriber table. On , Schneur Rosenberg wrote: Its not multi tenant, they are all going to the same exact place the only difference is the domain set in the clients hardware, On Mon, Dec 19, 2011 at 8:27 PM, Duane Larson duane.lar...@gmail.com> wrote: > If you are doing a multidomain/multitenant install why don't you want to > input the domain info in the subscriber table? > > On Dec 19, 2011 11:15 AM, "Schneur Rosenberg" rosenberg11...@gmail.com> > wrote: >> >> here is my problem, my server has multiple domain names, different >> customers register with a different domain name (we combined a few >> servers into one, therefore some register with different names) now as >> long as I dont have the correct domain set in the subscriber table, >> the system will keep on replying with a 401 Unauthorized. >> >> I assume it has to do with the way ha1 passwords are handled, I set >> the system to use plain text passwords, I also tried setting static >> challenge for www_authorize, I thought this will eliminate the need >> for the domain in the challenge, but still no registration. >> >> in my debug I get the following 2 lines >> >> Dec 19 09:11:06 opensipsmaster /sbin/opensips[4937]: >> DBG:db_mysql:db_mysql_convert_rows: no rows returned from the query >> Dec 19 09:11:06 opensipsmaster /sbin/opensips[4937]: >> DBG:auth_db:get_ha1: no result for user 'gra...@sipsvr1.mydomain.com' >> >> Schneur Rosenberg >> >> ___ >> 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Blank domain field in subscriber table
Its not multi tenant, they are all going to the same exact place the only difference is the domain set in the clients hardware, On Mon, Dec 19, 2011 at 8:27 PM, Duane Larson wrote: > If you are doing a multidomain/multitenant install why don't you want to > input the domain info in the subscriber table? > > On Dec 19, 2011 11:15 AM, "Schneur Rosenberg" > wrote: >> >> here is my problem, my server has multiple domain names, different >> customers register with a different domain name (we combined a few >> servers into one, therefore some register with different names) now as >> long as I dont have the correct domain set in the subscriber table, >> the system will keep on replying with a 401 Unauthorized. >> >> I assume it has to do with the way ha1 passwords are handled, I set >> the system to use plain text passwords, I also tried setting static >> challenge for www_authorize, I thought this will eliminate the need >> for the domain in the challenge, but still no registration. >> >> in my debug I get the following 2 lines >> >> Dec 19 09:11:06 opensipsmaster /sbin/opensips[4937]: >> DBG:db_mysql:db_mysql_convert_rows: no rows returned from the query >> Dec 19 09:11:06 opensipsmaster /sbin/opensips[4937]: >> DBG:auth_db:get_ha1: no result for user 'gra...@sipsvr1.mydomain.com' >> >> Schneur Rosenberg >> >> ___ >> 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
Re: [OpenSIPS-Users] Blank domain field in subscriber table
because I have customers coming in from a bunch different domains, I will have to go through my whole database and set the domain in the table for each user according to the domain they are using On Mon, Dec 19, 2011 at 8:27 PM, Duane Larson wrote: > If you are doing a multidomain/multitenant install why don't you want to > input the domain info in the subscriber table? > > On Dec 19, 2011 11:15 AM, "Schneur Rosenberg" > wrote: >> >> here is my problem, my server has multiple domain names, different >> customers register with a different domain name (we combined a few >> servers into one, therefore some register with different names) now as >> long as I dont have the correct domain set in the subscriber table, >> the system will keep on replying with a 401 Unauthorized. >> >> I assume it has to do with the way ha1 passwords are handled, I set >> the system to use plain text passwords, I also tried setting static >> challenge for www_authorize, I thought this will eliminate the need >> for the domain in the challenge, but still no registration. >> >> in my debug I get the following 2 lines >> >> Dec 19 09:11:06 opensipsmaster /sbin/opensips[4937]: >> DBG:db_mysql:db_mysql_convert_rows: no rows returned from the query >> Dec 19 09:11:06 opensipsmaster /sbin/opensips[4937]: >> DBG:auth_db:get_ha1: no result for user 'gra...@sipsvr1.mydomain.com' >> >> Schneur Rosenberg >> >> ___ >> 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
Re: [OpenSIPS-Users] Blank domain field in subscriber table
If you are doing a multidomain/multitenant install why don't you want to input the domain info in the subscriber table? On Dec 19, 2011 11:15 AM, "Schneur Rosenberg" wrote: > here is my problem, my server has multiple domain names, different > customers register with a different domain name (we combined a few > servers into one, therefore some register with different names) now as > long as I dont have the correct domain set in the subscriber table, > the system will keep on replying with a 401 Unauthorized. > > I assume it has to do with the way ha1 passwords are handled, I set > the system to use plain text passwords, I also tried setting static > challenge for www_authorize, I thought this will eliminate the need > for the domain in the challenge, but still no registration. > > in my debug I get the following 2 lines > > Dec 19 09:11:06 opensipsmaster /sbin/opensips[4937]: > DBG:db_mysql:db_mysql_convert_rows: no rows returned from the query > Dec 19 09:11:06 opensipsmaster /sbin/opensips[4937]: > DBG:auth_db:get_ha1: no result for user 'gra...@sipsvr1.mydomain.com' > > Schneur Rosenberg > > ___ > 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] Totally Stunned about this No Audio Going Out
I think the time when you were having full audio w/o any RTP proxy was due
to the fact that you've everything on the same subnet.
In my opinion you'll need RTPproxy eventually whenever you deploy it in
real environments.
I cant tell the exact reason for destination unreachable, maybe the virtual
IP has something to do with it.
To configure RTP proxy you need to do following.
1- Start RTPproxy in bridged mode i.e #*/usr/sbin/rtpproxy -l
/ blah blah switches*
2- Set module params in opensips.cfg file, and find out the point in main
route where call is forwarded to Asterisk Server VIP, just before that
write the function "*force_rtp_proxy("ei")*"
3- hmmm..on BYE or CANCEL you need to unforce rtp proxy as well.
Thats all I could think about it so far.
Regards,
Sammy
On Mon, Dec 19, 2011 at 10:36 PM, Nick Khamis wrote:
> What happened to my nice diagram? Argh Sorry guys!
>
> Router -> OpenSIPS -> Asterisk -> ITPS
>
> On Mon, Dec 19, 2011 at 12:20 PM, Nick Khamis wrote:
> > Hello Sammy,
> >
> > Thank you for your response. I now have outgoing audio again which is
> > half the battle.
> > The second half (incoming audio), has proven to be a challenge. Maybe
> > if I start with
> > a description of the setup:
> >
> > * This is a test environment done on virtual machines
> >
> >
> > Network:
> >
> > RouterL (192.168.2.1)
> > Polycom Phone (192.168.2.11)
> > OpenSIPS (192.168.2.102)
> > Asterisk Virtual IP for AST1 and AST2 (192.168.2.6)
> > Asterisk1 (192.168.2.110)
> > Asterisk2 (192.168.2.111)
> >
> >
> > - Port FWD (1)
> > ---
> > | Router |--> |OpenSIPS/RTPProxy|--> |
> > Asterisk GTWY | --- Internet/ITSP
> > -
> > -
> > ---
> >
> >
> > 1) The port forwarding range is:
> > SIP: 5060
> > RTP: 10,000-50,000
> > RTP Proxy: 7789
> >
> >
> > I just want to clear some things up. I had outgoing audio the whole
> > time without RTPProxy.
> > All the test UC (Polycom Phones) are within the same network. Do I
> > need to use RTPProxy
> > to get incomming audio working? As you can see in the diagram, I did
> > try using RTP Proxy
> > but never succeeded.
> >
> > Doing a raw UDP trace from ports (1-5) I found this:
> > http://pastebin.com/yzgBZQ9S
> > There is a "Destination unreachable" at first attempt being returned
> > by opensips server,
> > and then it dissapears, the it comes back again. Not sure if this is
> > related to the no
> > outgoing audio, but I will need to resolve it nevertheless.
> >
> > As for a SIP trace without RTP Proxy proxy running:
> > http://pastebin.com/PUXJ3wpK.
> > Wanted to turn your attention to:
> >
> > * The network architecture consists of OpenSIPS sending requests to
> > the Asterisk virtual IP (192.168.2.6),
> > which is connected to the Asterisk physical machines (192.168.2.110,
> > 192.168.2.111). The responding
> > asterisk box, in this particular eaxample, was 192.168.2.111. I hope
> > this would not be the problem?
> >
> > * A summary of the SDP trace is as follows:
> >
> > INVITE from UC: m=audio 10006 RTP/AVP 0 8 18 101
> > OK from Asterisk to OpenSIPS: m=audio 31576 RTP/AVP 8 0 101.
> > OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
> > OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101.
> > OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
> > OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101.
> > OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101.
> > OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101.
> > OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
> > OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101.
> > OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
> > OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
> >
> > Is taht my problem right there? My system is unable to connect
> > the initial request from the UC on port 10006, to the followup response
> > of the ITSP on port 34030? There is no OK from the UC to OpenSIPS.
> >
> > I've been struggling with this for a week now. Any help would be greatly
> > appreciated!
> >
> > Kind Regards,
> >
> > Nick.
>
> ___
> 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] Totally Stunned about this No Audio Going Out
What happened to my nice diagram? Argh Sorry guys! Router -> OpenSIPS -> Asterisk -> ITPS On Mon, Dec 19, 2011 at 12:20 PM, Nick Khamis wrote: > Hello Sammy, > > Thank you for your response. I now have outgoing audio again which is > half the battle. > The second half (incoming audio), has proven to be a challenge. Maybe > if I start with > a description of the setup: > > * This is a test environment done on virtual machines > > > Network: > > RouterL (192.168.2.1) > Polycom Phone (192.168.2.11) > OpenSIPS (192.168.2.102) > Asterisk Virtual IP for AST1 and AST2 (192.168.2.6) > Asterisk1 (192.168.2.110) > Asterisk2 (192.168.2.111) > > > - Port FWD (1) > --- > | Router |--> |OpenSIPS/RTPProxy|--> | > Asterisk GTWY | --- Internet/ITSP > - > - > --- > > > 1) The port forwarding range is: > SIP: 5060 > RTP: 10,000-50,000 > RTP Proxy: 7789 > > > I just want to clear some things up. I had outgoing audio the whole > time without RTPProxy. > All the test UC (Polycom Phones) are within the same network. Do I > need to use RTPProxy > to get incomming audio working? As you can see in the diagram, I did > try using RTP Proxy > but never succeeded. > > Doing a raw UDP trace from ports (1-5) I found this: > http://pastebin.com/yzgBZQ9S > There is a "Destination unreachable" at first attempt being returned > by opensips server, > and then it dissapears, the it comes back again. Not sure if this is > related to the no > outgoing audio, but I will need to resolve it nevertheless. > > As for a SIP trace without RTP Proxy proxy running: > http://pastebin.com/PUXJ3wpK. > Wanted to turn your attention to: > > * The network architecture consists of OpenSIPS sending requests to > the Asterisk virtual IP (192.168.2.6), > which is connected to the Asterisk physical machines (192.168.2.110, > 192.168.2.111). The responding > asterisk box, in this particular eaxample, was 192.168.2.111. I hope > this would not be the problem? > > * A summary of the SDP trace is as follows: > > INVITE from UC: m=audio 10006 RTP/AVP 0 8 18 101 > OK from Asterisk to OpenSIPS: m=audio 31576 RTP/AVP 8 0 101. > OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101. > OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101. > OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101. > OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101. > OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101. > OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101. > OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101. > OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101. > OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101. > OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101. > > Is taht my problem right there? My system is unable to connect > the initial request from the UC on port 10006, to the followup response > of the ITSP on port 34030? There is no OK from the UC to OpenSIPS. > > I've been struggling with this for a week now. Any help would be greatly > appreciated! > > Kind Regards, > > Nick. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Totally Stunned about this No Audio Going Out
Hello Sammy, Thank you for your response. I now have outgoing audio again which is half the battle. The second half (incoming audio), has proven to be a challenge. Maybe if I start with a description of the setup: * This is a test environment done on virtual machines Network: RouterL (192.168.2.1) Polycom Phone (192.168.2.11) OpenSIPS (192.168.2.102) Asterisk Virtual IP for AST1 and AST2 (192.168.2.6) Asterisk1 (192.168.2.110) Asterisk2 (192.168.2.111) - Port FWD (1) --- | Router |--> |OpenSIPS/RTPProxy|--> | Asterisk GTWY | --- Internet/ITSP - - --- 1) The port forwarding range is: SIP: 5060 RTP: 10,000-50,000 RTP Proxy: 7789 I just want to clear some things up. I had outgoing audio the whole time without RTPProxy. All the test UC (Polycom Phones) are within the same network. Do I need to use RTPProxy to get incomming audio working? As you can see in the diagram, I did try using RTP Proxy but never succeeded. Doing a raw UDP trace from ports (1-5) I found this: http://pastebin.com/yzgBZQ9S There is a "Destination unreachable" at first attempt being returned by opensips server, and then it dissapears, the it comes back again. Not sure if this is related to the no outgoing audio, but I will need to resolve it nevertheless. As for a SIP trace without RTP Proxy proxy running: http://pastebin.com/PUXJ3wpK. Wanted to turn your attention to: * The network architecture consists of OpenSIPS sending requests to the Asterisk virtual IP (192.168.2.6), which is connected to the Asterisk physical machines (192.168.2.110, 192.168.2.111). The responding asterisk box, in this particular eaxample, was 192.168.2.111. I hope this would not be the problem? * A summary of the SDP trace is as follows: INVITE from UC: m=audio 10006 RTP/AVP 0 8 18 101 OK from Asterisk to OpenSIPS: m=audio 31576 RTP/AVP 8 0 101. OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101. OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101. OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101. OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101. OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101. OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101. OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101. OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101. OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101. OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101. Is taht my problem right there? My system is unable to connect the initial request from the UC on port 10006, to the followup response of the ITSP on port 34030? There is no OK from the UC to OpenSIPS. I've been struggling with this for a week now. Any help would be greatly appreciated! Kind Regards, Nick. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Blank domain field in subscriber table
here is my problem, my server has multiple domain names, different customers register with a different domain name (we combined a few servers into one, therefore some register with different names) now as long as I dont have the correct domain set in the subscriber table, the system will keep on replying with a 401 Unauthorized. I assume it has to do with the way ha1 passwords are handled, I set the system to use plain text passwords, I also tried setting static challenge for www_authorize, I thought this will eliminate the need for the domain in the challenge, but still no registration. in my debug I get the following 2 lines Dec 19 09:11:06 opensipsmaster /sbin/opensips[4937]: DBG:db_mysql:db_mysql_convert_rows: no rows returned from the query Dec 19 09:11:06 opensipsmaster /sbin/opensips[4937]: DBG:auth_db:get_ha1: no result for user 'gra...@sipsvr1.mydomain.com' Schneur Rosenberg ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] cdr accounting on opensips restart
Hi Razwan, This is the pastebin of logs after shutdown: http://pastebin.com/tvmrSqwB This is the pastebin of logs after start which is huge: http://pastebin.com/C6K4Jt5y --- Jayesh On Wed, Dec 7, 2011 at 5:46 PM, Razvan Crainea wrote: > Hi, Jayesh! > > I need the logs after opensips restarts. The result might also be pretty > large. > > > Regards, > > -- > Răzvan Crainea > OpenSIPS Developer > > > On 12/07/2011 02:14 PM, Jayesh Nambiar wrote: > > Hi Razwan, > I have applied the patch and made it working. Do you still only need the > logs after opensips shutdown or also the logs after opensips restarts. > > --- Jayesh > > On Wed, Dec 7, 2011 at 4:35 PM, Razvan Crainea > wrote: > >> Hi, Jayesh! >> >> It seems like the problem appears while parsing the string got from the >> database, so after opensips is restarted. The patch attached should give us >> more information about the error. >> >> >> Regards, >> >> -- >> Răzvan Crainea >> OpenSIPS Developer >> >> >> On 12/07/2011 12:37 PM, Jayesh Nambiar wrote: >> >> This is the string what I see when I query the dialog table for vars: >> >> accX_flags# |accX_db#( 2 1 >> 11 1002 >> 12013386166 >> 919833171405 0 203.153.53.158 203.153.53.136 0 17 0 2 IP India - Mobile >> 919 9198 2 1 2 4 0.0150 0.0130 0 0. 0.0226 1. 6 1 6 1 1 2 2 2 >> Aal Izz Well >> Aal Izz Well 1 3 1.2.3.4|accX_leg#|accX_core# INVITE/ >> 100eb870-9e3599cb-13c4-50029-3b71-60de50d0-3b71\# >> 97.208.30.751120+1+31850054+b8c0a68/ >> 100f7670-9e3599cb-13c4-50029-3b71-3c625412-3b71 200 O á=ÃN >> |accX_created#Ã=ÃN| >> >> --- Jayesh >> >> On Wed, Dec 7, 2011 at 4:00 PM, Razvan Crainea < >> razvancrai...@opensips.org> wrote: >> >>> Hi, Jayesh! >>> >>> And what is the string in the database? You can see it in the database >>> after you kill opensips with the following command: >>> >>> select vars from dialog; >>> >>> >>> Regards, >>> >>> -- >>> Răzvan Crainea >>> OpenSIPS Developer >>> >>> >>> On 12/07/2011 12:24 PM, Jayesh Nambiar wrote: >>> >>> Hi Razvan, >>> Applied the patch and re-tested it. Here the logs that you are >>> interested in specifically: >>> >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: Dumping var name: value: <#026> >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: compare char 22 - 0 - 11 >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: compare char 0 - 1 - 12 >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: compare char 0 - 2 - 13 >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: compare char 0 - 3 - 14 >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: Serialized string (16) >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: 0) char: a hex: 61 >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: 1) char: c hex: 63 >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: 2) char: c hex: 63 >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: 3) char: X hex: 58 >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: 4) char: _ hex: 5F >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: 5) char: f hex: 66 >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: 6) char: l hex: 6C >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: 7) char: a hex: 61 >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: 8) char: g hex: 67 >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: 9) char: s hex: 73 >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: 10) char: # hex: 23 >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: 11) char: #026 hex: 16 >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: 12) char: #000 hex: 00 >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: 13) char: #000 hex: 00 >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: 14) char: #000 hex: 00 >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: 15) char: | hex: 7C >>> Dec 7 15:50:33 dev /usr/local/sbin/opensips[8011]: >>> DBG:dialog:write_pair: Dumping var name: value: <(> >>> >>> Apart from this, I have pasted everything in syslog after opensips >>> shutdown in the paste-bin here: >>> http://pastebin.com/gx0ZxFLb >>> >>> Let me know if there is anything more to test. >>> >>> --- Jayesh >>> >>> On Wed, Dec 7, 2011 at 3:36 PM, Razvan Crainea < >>> razvancrai...@opensips.org> wrote: >>> Sorr, I forgot to attach it. Here it is. Regards, -- Răzvan Crainea
Re: [OpenSIPS-Users] Count SIP requests based on their method
Hello, Check out the STATISTICS module [1]. It will allow you to create statistics from your script and increment/decrement them. Also, take into account that if you do not want to count retransmissions, you have to take care of this at TM level. [1] http://www.opensips.org/html/docs/modules/devel/statistics.html Regards, Vlad Paiu OpenSIPS Developer On 12/19/2011 09:51 AM, M.Abdulaziz wrote: Hello all, Is there anyway (MI command or any function) to count number of sip requests based on its method (INVITE - OPTIONS - MESSAGE)? Thank you -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Count-SIP-requests-based-on-their-method-tp7107398p7107398.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
