I think I solved the problem. It's related to the fact that I use opensips as registrar also
For exampel I have users 300 with password 300 on opensips and also an extension 300 with pass 300 on the asterisk servers that I balance the calls. The solution for me was to remove the secret for extension 300 in asterisk. On Wed, Mar 7, 2012 at 11:40 AM, Iulian Macare <iulian.mac...@gmail.com> wrote: > I have installed Opensips 1.7.2 on debian and I have the same problem. > > > > Mar 7 10:23:59 opensipsnew /usr/sbin/opensips[18883]: sending call > with callid 3817a6411cec76292a8d1c2811ad7def@82.77.252.47 and ruri > sip:0103629834@82.77.252.40 to sip:82.77.252.48 > Mar 7 10:23:59 opensipsnew /usr/sbin/opensips[18882]: sending call > with callid 3817a6411cec76292a8d1c2811ad7def@82.77.252.47 and ruri > sip:0103629834@82.77.252.40 to sip:82.77.252.39 > Mar 7 10:23:59 opensipsnew /usr/sbin/opensips[18880]: sending call > with callid 3817a6411cec76292a8d1c2811ad7def@82.77.252.47 and ruri > sip:0103629834@82.77.252.40 to sip:82.77.252.48 > Mar 7 10:23:59 opensipsnew /usr/sbin/opensips[18880]: sending call > with callid 3817a6411cec76292a8d1c2811ad7def@82.77.252.47 and ruri > sip:0103629834@82.77.252.40 to sip:82.77.252.48 > > The called number is 0103629834 sent both to 82.77.252.39 and > 82.77.252.48 many times > > In the configuration file I have > > > if (uri=~"^sip:0[1-9][0-9]*@") { > load_balance("1","pstn"); > xlog("sending call with callid $ci and ruri $ru to $du\n"); > t_on_failure("1"); > > if (!t_relay()) > { sl_reply_error(); > }; > exit; > } > > > Destination:: sip:82.77.252.39 id=3 group=1 enabled=yes auto-re=on > Resource:: pstn max=200 load=36 > Destination:: sip:82.77.252.35 id=1 group=1 enabled=yes auto-re=on > Resource:: pstn max=200 load=36 > Destination:: sip:82.77.252.48 id=6 group=1 enabled=yes auto-re=on > Resource:: pstn max=200 load=36 > > > > > The number 0103629834 I don't see it in opensips in accounting ( I use > CDRTool ) and I also don't see it in any logs on 82.77.252.39 or > 82.77.252.48 ( wich are Asterisks ) > > > opensips 1.6.4 load balancing performance > > On Tue, Mar 15, 2011 at 6:46 PM, Iulian Macare <iulian.mac...@gmail.com> > wrote: >> I see also something like this: >> >> Mar 15 18:47:36 opensipsvl /usr/local/sbin/opensips[4547]: sending call with >> callid 3b77e3325a8c75f8571c4fcf37a8ba3c@192.168.52.10 and ruri >> sip:0758553307@192.168.52.20 to sip:192.168.254.241 >> Mar 15 18:47:36 opensipsvl /usr/local/sbin/opensips[4539]: sending call with >> callid 3b77e3325a8c75f8571c4fcf37a8ba3c@192.168.52.10 and ruri >> sip:0758553307@192.168.52.20 to sip:192.168.254.241 >> Mar 15 18:47:36 opensipsvl /usr/local/sbin/opensips[4556]: sending call with >> callid 3b77e3325a8c75f8571c4fcf37a8ba3c@192.168.52.10 and ruri >> sip:0758553307@192.168.52.20 to sip:192.168.254.242 >> >> Same call 3 times.. 2 to one server and another time to the other >> >> On Tue, Mar 15, 2011 at 12:10 PM, Bogdan-Andrei Iancu <bog...@opensips.org> >> wrote: >>> >>> Hi Iulian, >>> >>> >>> >>> Related to the second issue (with 2 logs for a single call), I suspect it >>> may be a retransmission issue. To be sure, print in your xlog the callid >>> also (to be sure it is the same request) . Var for callid is $ci. >>> >>> Regards, >>> Bogdan >>> >>> >>> Iulian Macare wrote: >>>> >>>> Hello >>>> >>>> >>>> I have installed OpenSips 1.6.4 on CentOS 5.5 32bit with load balancing & >>>> mysql support ; I want to balance calls to 2 asterisk servers . I am >>>> sending >>>> traffic to opensips from 1 x gnudialer & 1 x vicidial ( so from predictive >>>> dialers ). Situation is like this: >>>> >>>> >>>> >>>> +----+----------+--------------------------+-----------+------------+-------------+ >>>> | id | group_id | dst_uri | resources | probe_mode | >>>> description | >>>> >>>> +----+----------+--------------------------+-----------+------------+-------------+ >>>> | 1 | 1 | sip:192.168.254.241:5060 <http://192.168.254.241:5060> >>>> | pstn=300 | 0 | | >>>> | 2 | 1 | sip:192.168.254.242:5060 <http://192.168.254.242:5060> >>>> | pstn=300 | 0 | | >>>> >>>> >>>> +----+----------+--------------------------+-----------+------------+-------------+ >>>> >>>> 600 channels in total , and I send around 500 channels ; OpenSips drops a >>>> lot of calls; By drop I mean a call that is not sent to one of those 2 >>>> asterisk servers that I have. >>>> >>>> The code for balancing in this situation is: >>>> >>>> if (uri=~"^sip:0[1-9][0-9]*@") { >>>> load_balance("1","pstn"); >>>> xlog("sending call $ru to $du\n"); >>>> t_relay(); >>>> exit; >>>> } >>>> >>>> ! An important thing to say is that in /var/log/messages I see the >>>> specific number that is sent to 192.168.254.241 for example ; So the >>>> parameter xlog("sending call $ru to $du\n"); works ; The problem is that in >>>> logs on 192.168.254.241 that number never arrives in asterisk logs ; In >>>> logs >>>> of vicidial & gnudialer I see it like congestion. >>>> >>>> If I do something like this: >>>> >>>> >>>> +----+----------+---------------------+-----------+------------+-------------+ >>>> | id | group_id | dst_uri | resources | probe_mode | >>>> description | >>>> >>>> +----+----------+---------------------+-----------+------------+-------------+ >>>> | 1 | 1 | sip:192.168.254.241 | pstn=150 | 0 | >>>> | >>>> | 2 | 2 | sip:192.168.254.241 | pstn=150 | 0 | >>>> | >>>> | 3 | 1 | sip:192.168.254.242 | pstn=150 | 0 | >>>> | >>>> | 4 | 2 | sip:192.168.254.242 | pstn=150 | 0 | >>>> | >>>> >>>> And I split opensips balancing in 2 >>>> >>>> >>>> if(src_ip==192.168.3.10 ) >>>> { >>>> load_balance("1","pstn"); >>>> xlog("sending call to $du\n"); >>>> t_relay(); >>>> exit; >>>> }; >>>> >>>> if(src_ip==192.168.3.11 ) >>>> { >>>> load_balance("2","pstn"); >>>> xlog("sending call to $du\n"); >>>> t_relay(); >>>> exit; >>>> }; >>>> >>>> >>>> and by doing this I get the same numbers of channels on opensips ( around >>>> 500 channels ) but I am splitting in 2 groups of load balancing; It can >>>> process all the calls. >>>> >>>> Another question that I saw is that when I make a single call to opensips >>>> and I involve load balancing in /var/log/messages I get 2 times the same >>>> message .. just like it send 2 time to asterisk server the call .. but on >>>> asterisk I receive only one time. >>>> >>>> Mar 10 14:58:47 opensips /usr/local/sbin/opensips[27611]: sending call to >>>> sip:192.168.254.241 >>>> Mar 10 14:58:47 opensips /usr/local/sbin/opensips[27611]: sending call to >>>> sip:192.168.254.241 >>>> >>>> >>>> Isn't load balancing fast enough the process the calls made by predictive >>>> dialers, when over 300 channels is sent .. ? Or I have some mistakes made >>>> . >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users@lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> >>> >>> -- >>> Bogdan-Andrei Iancu >>> OpenSIPS eBootcamp - 28th February 2011 >>> 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