That might be :) I'm now running into problems with the dialog module (which i use to limit concurrent calls). Calls seem to stick in the dialog module (thus denying additional calls) while the endpoint isn't listing the same amount of calls :/
Regards, Erik Op 6-5-2010 11:02, Bogdan-Andrei Iancu schreef: > So, after all, it was a network layer configuration issue... :) > > Regards, > Bogdan > > Erik Versaevel wrote: >> The destination (in this case) is the 1st server in the loadbalancer >> list (as there are no other calls). >> I've upgraded this machine to ubuntu 10 (from 8) and started getting >> Connection Tracking drop messages in my >> syslog. I've disabled connection tracking and the issue hasn't >> appeared since... >> >> >> Op 5-5-2010 12:24, Bogdan-Andrei Iancu schreef: >> >>> Hi Erik, >>> >>> have you tried to print the destination of the requests that fail? >>> >>> regards, >>> Bogdan >>> >>> Erik Versaevel wrote: >>> >>>> Hi All, >>>> >>>> I attempted an migration last night (from our current environment to >>>> this new setup) but i ran into this >>>> problem as soon as i tried to make some test calls, funny thing is i >>>> can't get it reproduced :/ Any clues >>>> on how to debug this any further? >>>> >>>> Kind regards, >>>> >>>> Erik >>>> >>>> Op 27-4-2010 15:17, Erik Versaevel schreef: >>>> >>>>> Hi all, >>>>> >>>>> I'm building a setup in which opensips is acting as registar for my >>>>> endpoints and loadbalancing >>>>> calls made by those endpoint over an cluster of asterisk machines. >>>>> (so that if we need more asterisk >>>>> power, we just have to add another destination to the loadbalancer >>>>> module) >>>>> Opensips is listening on multiple IP addresses and uses the >>>>> loadbalancer module to poll my asterisk >>>>> machines and select the destination. >>>>> My problem is that every now and then opensips fails to forward an >>>>> invite to my asterisk cluster and >>>>> generates >>>>> >>>>> "ERROR:core:udp_send: >>>>> sendto(sock,0x77b81280,1353,0,0x77b81b04,16): Operation not >>>>> permitted(1)" >>>>> >>>>> there is some iptables filtering on this machine, however it is not >>>>> showing drops in the logfile (and it keeps >>>>> occuring even without any iptable rules). >>>>> I tried stracing opensips but all i get is: >>>>> >>>>> opensipstrace.7423:sendto(6, "INVITE >>>>> sip:e164_dst_phone...@opensips_ip_address SIP/2.0 >>>>> Record-Route: >>>>> <sip:OPENSIPS_IP_ADDRESS;lr=on;ftag=AI05ED431A05432EB8;nat=yes;did=fd6.e1f16fe3;vsf=AAAAAAMIBgl3AggLFgF5HAAFGhwBHzE3NC44MQ--> >>>>> >>>>> Via: SIP/2.0/UDP OPENSIPS_IP_ADDRESS;branch=z9hG4bK3177.1e0e38b7.0 >>>>> Via: SIP/2.0/UDP >>>>> 192.168.178.44:5060;received=CPE_IP_ADDRESS;rport=61008;branch=z9hG4bK2010Apr222938466E164_DST_PHONE_NR >>>>> >>>>> To: <sip:e164_dst_phone...@opensips_ip_address> >>>>> From: \"3961\" >>>>> <sip:3...@opensips_ip_address>;tag=AI05ED431A05432EB8 >>>>> Call-ID: aif001c45e85f79...@192.168.178.44 >>>>> CSeq: 2 INVITE >>>>> Max-Forwards: 69 >>>>> Contact: >>>>> <sip:e164phone...@cpe_ip_address:61008;line=AIF8F01E8DF866D7CB> >>>>> Accept: application/sdp >>>>> Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER >>>>> Allow-Events: dialog,message-summary >>>>> P-Preferred-Identity: <sip:e164phone...@opensips_ip_address> >>>>> Privacy: none >>>>> User-Agent: SomeStrangeDude >>>>> Content-Type: application/sdp >>>>> Content-Length: 324 >>>>> I-FromDisp: <null> >>>>> I-FromUri: E164PHONE_NR >>>>> I-CustId: 3961 >>>>> >>>>> v=0 >>>>> o=intelligate 1133701155 1133701155 IN IP4 192.168.178.44 >>>>> s=call >>>>> c=IN IP4 CPE_IP_ADDRESS >>>>> t=0 0 >>>>> m=audio 5004 RTP/AVP 18 8 101 >>>>> a=rtpmap:18 G729/8000 >>>>> a=fmtp:18 annexb=no >>>>> a=rtpmap:8 PCMA/8000 >>>>> a=rtpmap:101 telephone-event/8000 >>>>> a=fmtp:101 0-15 >>>>> a=sendrecv >>>>> a=ptime:20 >>>>> a=direction:active >>>>> a=oldmediaip:192.168.178.44 >>>>> ", 1253, 0, {sa_family=AF_INET, sin_port=htons(5060), >>>>> sin_addr=inet_addr("ASTERISK_IP_ADDRESS")}, 16) = -1 EPERM >>>>> (Operation not permitted) >>>>> >>>>> I also use the uac_replace_from() to mangle the from header so >>>>> asterisk uses the correct user/peer/client to connect the call >>>>> (codec/dialplan etc). >>>>> I'm having trouble reproducing the error as it's not allways >>>>> occuring, the errors i straced where mainly the initial invite >>>>> towards my asterisk >>>>> cluster and a few 200 OK's which didn't get processed correctly. >>>>> >>>>> Any clues on how to debug this further? >>>>> >>>>> Kind regards, >>>>> >>>>> Erik Versaevel >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users@lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >> >> >> >> Erik Versaevel >> >> > > Erik Versaevel -- Core Network Engineer Infopact Network Solutions Hoogvlietsekerkweg 170 3194 AM Rotterdam Hoogvliet Telefoon +31 (0)88 - 4636777 Fax +31 (0)88 - 4636799 Mobile +31 (0)6 - 11116070 e.versae...@infopact.nl www.infopact.nl _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users