Re: [OpenSIPS-Users] mmgeoip
Hi Bodgan, Thank you very much. We got it to work. The problem is we have the country edition loaded instead of the city edition. See you at cluecon. Thank You -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/mmgeoip-tp7580620p7580793.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] mmgeoip
I am getting unknown when using mmgeoip. I tried both the citylite version
and city version from maxmind. Thank You
version: opensips 1.7.2-notls (x86_64/linux)
if (mmg_lookup("lon:lat:cc:reg","$si","$avp(lat_lon)")) {
xlog("GEOIP: $(avp(lat_lon)[0]) $(avp(lat_lon)[1])
$(avp(lat_lon)[2]) $(avp(lat_lon)[3]) $(avp(lat_lon)[4]) $avp(lat_lon)");
}
DBG:mmgeoip:mmg_lookup_cmd: 'x.x.x.x'--> 'Unknown'.
GEOIP:
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Re: [OpenSIPS-Users] libmysqlclient-dev for CentOS?
You need mysql-libs.x86_64. This is what i have in my Centos to install OpenSIPS with MySQL mysql.x86_64 mysql-devel.x86_64 mysql-libs.x86_64 -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/libmysqlclient-dev-for-CentOS-tp7580351p7580592.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] FW: Opensips memory problem
I have the same problem with location and address tables before and i increased the memory pool fixed it. Hope this will help you as well. Take a look at http://www.opensips.org/Resources/DocsTsMem -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/FW-Opensips-memory-problem-tp7580585p7580591.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:mediaproxy:__tm_request_in: could not create new dialog
I uploaded lvl 6 debug and SIP trace. I am not sure what is going on and am very confused too. There is no audio and call drop only on inbound call to IP Phone from OpenSIPs. What driving me crazy is that it happen on ATA, IP Phone, and other Dialer but it doesn't happen to Zoiper Dialer (I tried many many time to confirm this too). -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/ERROR-mediaproxy-tm-request-in-could-not-create-new-dialog-tp6817274p6821578.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:mediaproxy:__tm_request_in: could not create new dialog
Sorry everyone. I been working on OpenSER and OpenSIPs for 5 years. This is the first time i experienced so many problem upgrading. 1. Receiving ERROR:mediaproxy:__tm_request_in: could not create new dialog on Production only not testing environment. 2. No Audio only on inbound to IP Phone 3. Now it kept crashing when IP Phone trying to register. /usr/local/sbin/opensips[16389]: CRITICAL:core:del_lump: offset exceeds message size (485328 > 682) aborting... Will upload the debug lvl 6 and sip trace asap. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/ERROR-mediaproxy-tm-request-in-could-not-create-new-dialog-tp6817274p6821413.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:mediaproxy:__tm_request_in: could not create new dialog
Do you have any suggestion on what is the easier way to do it or you want me to attach everything from the debug log? I can't reproduce this problem on my testing environment and it only happen on production. The debug log and siptrace fill up very quickly. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/ERROR-mediaproxy-tm-request-in-could-not-create-new-dialog-tp6817274p6820998.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:mediaproxy:__tm_request_in: could not create new dialog
I don't see any error before. Will do the debug level 6 when i get to work in few hour. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/ERROR-mediaproxy-tm-request-in-could-not-create-new-dialog-tp6817274p6820435.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:mediaproxy:__tm_request_in: could not create new dialog
There got to be something I am missing for 1.7. on top of this error i have no audio both way. IP Phone -> Opensips -> PSTN have no problem PSTN -> OpenSIPS -> IP Phone with no audio. The funny part is Linksys IP Phone, Linksys PAP2, Pangolin Dialer all have the same problem no audio but using Zoiper Dialer i have two way audio. Interestingly this only happen on 1.7 and it happen on both engage_media_proxy and use_media_proxy but not on 1.6. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/ERROR-mediaproxy-tm-request-in-could-not-create-new-dialog-tp6817274p6818392.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:mediaproxy:__tm_request_in: could not create new dialog
I think I figured out what is the problem. IP Phone -> Opensips -> PSTN have no problem PSTN -> OpenSIPS -> IP Phone will produce this error -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/ERROR-mediaproxy-tm-request-in-could-not-create-new-dialog-tp6817274p6817852.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:mediaproxy:__tm_request_in: could not create new dialog
hi Saul, thanks for your response. I am not calling create_dialog in the opensips.cfg. I am using engage_mdeia_proxy() in the opensips.cfg. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/ERROR-mediaproxy-tm-request-in-could-not-create-new-dialog-tp6817274p6817418.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] ERROR:mediaproxy:__tm_request_in: could not create new dialog
I recently upgraded from OpenSIPS 1.6 to 1.7. Everything tested good in my staging environment. When I upgrade the production OpenSIPS from 1.6 to 1.7 I start getting this error once a while ERROR:mediaproxy:__tm_request_in: could not create new dialog. I tried to google around to for this message but i am unable to find anything. Anyone know this error is coming OpenSIPS configuration or Mediaproxy? Thank You -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/ERROR-mediaproxy-tm-request-in-could-not-create-new-dialog-tp6817274p6817274.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Out of mem all of the sudden
Thank you very much for all your response. The problem is fixed for me. It was a configuration issue that someone sending lot of calls to my gateway with a # at the end of the number caused a loop and used up all of my memory. I sent a address incomplete now if none of the URI match. Thank You -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Out-of-mem-all-of-the-sudden-tp5534338p5539236.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Out of mem all of the sudden
I even increased the SHM to 1024 still haven't the same problem. I believe identified the problem now and will share with everyone once I confirm this is the fix. Thank You -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Out-of-mem-all-of-the-sudden-tp5534338p5535297.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Out of mem all of the sudden
I am using opensips 1.6.3 with very only few modules loaded. The OpenSIPS is up for over a months with up any issue until now. I also tried to increase the PKG to 4mb and SHM to 256 mb still crash. loadmodule "db_mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "uri.so" loadmodule "dispatcher.so" loadmodule "mediaproxy.so" loadmodule "nathelper.so" loadmodule "dialog.so" loadmodule "mi_datagram.so" loadmodule "signaling.so" loadmodule "localcache.so" loadmodule "sst.so" loadmodule "avpops.so" loadmodule "load_balancer.so" -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Out-of-mem-all-of-the-sudden-tp5534338p5534792.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Out of mem all of the sudden
Now I have the time to look into the log file more. The the memlog is very big. What specific should I look into it? WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation ERROR:tm:new_t: out of mem ERROR:tm:t_newtran: new_t failed ERROR:tm:store_reply: failed to alloc' clone memory ERROR:tm:insert_tmcb: no more shared memory ERROR:dialog:dlg_create_dialog: failed to register TMCB INFO:core:handle_sigs: child process 10078 exited by a signal 11 INFO:core:handle_sigs: core was not generated INFO:core:handle_sigs: terminating due to SIGCHLD -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Out-of-mem-all-of-the-sudden-tp5534338p5534772.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Out of mem all of the sudden
All of the sudden yesterday and today my OpenSIPS kept crashing every every few hours before of out of mem messages. I tried to increase the PKG 4*1024*1024 and SHM to 256 and recompile and still having the same problem. There is no core dump generated for the crash and there is out of memory for almost everything when it happen Memlog=1 TOTAL: 75693 free fragments = 266358504 free bytes TOTAL: 136436832 large bytes TOTAL: 24 overhead -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Out-of-mem-all-of-the-sudden-tp5534338p5534338.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.6.2 to 1.6.3 SST module question
Thank you very much for your response. I will look into the re-invite problem now. There is no BYE in the SIPTrace from the SIPTrace module associated to this. The first Invite is received at 17:15:19 and the last ACK is at 17:16:05 from SIPTrace module. The dialog module send BYE at 17:19:05 -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-1-6-2-to-1-6-3-SST-module-question-tp5441423p5470345.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.6.2 to 1.6.3 SST module question
The call is established but terminated after some time. Here is the SIP trace from siptrace module and debug 5 from Opensips. There is no BYE in the SIPTrace. Debug 5 from Opensips did show BYE sent to caller and to callee from dialog module. INVITE sip:1510495x...@74.x.x.x. SIP/2.0 Record-Route: Via: SIP/2.0/UDP 74.x.x.x.;branch=z9hG4bKd659.0ab95ac3.0 Via: SIP/2.0/UDP 192.168.8.222:5068;received=173.x.x.x;branch=z9hG4bK-d87543-21170d43ec319d0c-1--d87543-;rport=5068 Max-Forwards: 69 Contact: To: From: "";tag=5145635c Call-ID: OThhNWNjOWNjZWFmZDkwMmFlNTExYmVmNTM1ZTE2NjM. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Type: application/sdp User-Agent: Idefisk Content-Length: 220 Session-Expires: 180 v=0 o=Idefisk_user 6056184806875838134 13270 IN IP4 192.168.8.222 s=Idefisk_user c=IN IP4 74.x.x.x. t=0 0 m=audio 1306 RTP/AVP 97 101 a=fmtp:101 0-15 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 SIP/2.0 100 Trying Via: SIP/2.0/UDP 74.x.x.x.;branch=z9hG4bKd659.0ab95ac3.0;received=74.x.x.x. Via: SIP/2.0/UDP 192.168.8.222:5068;received=173.x.x.x;branch=z9hG4bK-d87543-21170d43ec319d0c-1--d87543-;rport=5068 Record-Route: From: "";tag=5145635c To: Call-ID: OThhNWNjOWNjZWFmZDkwMmFlNTExYmVmNTM1ZTE2NjM. CSeq: 2 INVITE User-Agent: iWorld Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.8.222:5068;received=173.x.x.x;branch=z9hG4bK-d87543-21170d43ec319d0c-1--d87543-;rport=5068 Record-Route: From: "";tag=5145635c To: ;tag=as3154366e Call-ID: OThhNWNjOWNjZWFmZDkwMmFlNTExYmVmNTM1ZTE2NjM. CSeq: 2 INVITE User-Agent: iWorld Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 74.x.x.x.;branch=z9hG4bKd659.0ab95ac3.0;received=74.x.x.x. Via: SIP/2.0/UDP 192.168.8.222:5068;received=173.x.x.x;branch=z9hG4bK-d87543-21170d43ec319d0c-1--d87543-;rport=5068 Record-Route: From: "";tag=5145635c To: ;tag=as3154366e Call-ID: OThhNWNjOWNjZWFmZDkwMmFlNTExYmVmNTM1ZTE2NjM. CSeq: 2 INVITE User-Agent: iWorld Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.8.222:5068;received=173.x.x.x;branch=z9hG4bK-d87543-21170d43ec319d0c-1--d87543-;rport=5068 Record-Route: From: "";tag=5145635c To: ;tag=as3154366e Call-ID: OThhNWNjOWNjZWFmZDkwMmFlNTExYmVmNTM1ZTE2NjM. CSeq: 2 INVITE User-Agent: iWorld Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 259 v=0 o=root 7723 7723 IN IP4 64.x.x.x s=session c=IN IP4 74.x.x.x. t=0 0 m=audio 1304 RTP/AVP 97 101 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30 a=sendrecv SIP/2.0 200 OK Via: SIP/2.0/UDP 74.x.x.x.;branch=z9hG4bKd659.0ab95ac3.0;received=74.x.x.x. Via: SIP/2.0/UDP 192.168.8.222:5068;received=173.x.x.x;branch=z9hG4bK-d87543-21170d43ec319d0c-1--d87543-;rport=5068 Record-Route: From: "";tag=5145635c To: ;tag=as3154366e Call-ID: OThhNWNjOWNjZWFmZDkwMmFlNTExYmVmNTM1ZTE2NjM. CSeq: 2 INVITE User-Agent: iWorld Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 259 v=0 o=root 7723 7724 IN IP4 64.x.x.x s=session c=IN IP4 64.x.x.x t=0 0 m=audio 50450 RTP/AVP 97 101 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30 a=sendrecv SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.8.222:5068;received=173.x.x.x;branch=z9hG4bK-d87543-21170d43ec319d0c-1--d87543-;rport=5068 Record-Route: From: "";tag=5145635c To: ;tag=as3154366e Call-ID: OThhNWNjOWNjZWFmZDkwMmFlNTExYmVmNTM1ZTE2NjM. CSeq: 2 INVITE User-Agent: iWorld Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 259 Session-Expires: 180;refresher=uac v=0 o=root 7723 7724 IN IP4 64.x.x.x s=session c=IN IP4 74.x.x.x. t=0 0 m=audio 1304 RTP/AVP 97 101 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30 a=sendrecv ACK sip:1510495x...@64.x.x.x SIP/2.0 Via: SIP/2.0/UDP 192.168.8.222:5068;branch=z9hG4bK-d87543-fc30fd0917758e54-1--d87543-;rport Max-Forwards: 70 Route: Contact: To: ;tag=as3154366e From: "";tag=5145635c Call-ID: OThhNWNjOWNjZWFmZDkwMmFlNTExYmVmNTM1ZTE2NjM. CSeq: 2 ACK Proxy-Authorization: Digest username="",realm="74.x.x.x.",nonce="4c754fc50003cc350d97ea9ab177ce47146b80f9b7a5",uri="sip:1510495x...@74.x.x.x.",response="8037cc4b5b1bc6c822f7742c02f3e7e0",algorithm=MD5 User-Agent: Idefisk Content-Length:
Re: [OpenSIPS-Users] Opensips 1.6.2 to 1.6.3 SST module question
I apologized that I wasn't clear enough. The call is established but terminated after some time. I can provide the SIP trace later today. Thank You -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-1-6-2-to-1-6-3-SST-module-question-tp5441423p5462213.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Mediaproxy conntrack timeout
I finally figured out the problem by changing the nat setting on asterisk from nat=yes always assume NAT to nat=never. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Mediaproxy-conntrack-timeout-tp5327717p5441886.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Load balancer retcode question on OpenSIPS 1.6.2
Thanks for your recommendation it work great :). -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Load-balancer-retcode-question-on-OpenSIPS-1-6-2-tp5346088p5441869.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips 1.6.2 to 1.6.3 SST module question
I am not a expert on this but would like to get some understand what is the
problem with my configuration.
I am getting call drop caused by the STT module on 1.6.3 but not on 1.6.2
every 180 seconds if I set modparam("sst", "min_se", 180).
This is my configuration
# - Dialog params -
modparam("dialog", "dlg_flag", 4)
modparam("dialog", "timeout_avp", "$avp(i:10)")
modparam("dialog", "bye_on_timeout_flag", 14)
modparam("dialog", "db_mode", 1)
# - SST params -
modparam("sst", "timeout_avp", "$avp(i:10)")
modparam("sst", "sst_flag", 5)
modparam("sst", "min_se", 180) # Must be >= 90
if (is_method("INVITE")) {
if (sstCheckMin("1")) {
xlog("L_ERR", "Session Timer Too Small.\n\n");
exit;
}
setflag(4); #Dialog module Flag
setflag(5); #SST module Flag
setflag(14); #SST Dialog Timeout Flag
route(2);
}
--
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Re: [OpenSIPS-Users] Mediaproxy closes ports
I have the same problem that I still haven't yet figured out. Media go both direction but media relay unable to detect the RTP and conntrack time out and disconnect the call for me. This only happen to me if use a http Tunnel server. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Mediaproxy-closes-ports-tp5380645p5381098.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP UAs Authentication based on a combination of username, password and IP address of the UA
Using the permission module to check the source address and username first before www_authorize should work. http://www.opensips.org/html/docs/modules/1.6.x/permissions#id233458 -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/SIP-UAs-Authentication-based-on-a-combination-of-username-password-and-IP-address-of-the-UA-tp5377841p5378203.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Mediaproxy2/OpenSIPS 1.6 use_media_proxy onreply no audio
I spend two day on this finally figured out the problem. I will post up more detail later. Thank You -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Mediaproxy2-OpenSIPS-1-6-use-media-proxy-onreply-no-audio-tp5373247p5373877.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Multiple contact entries
Try to use max_contacts http://www.opensips.org/html/docs/modules/1.6.x/registrar.html#id228388. Or use save f flag http://www.opensips.org/html/docs/modules/1.6.x/registrar.html#id228388 -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Multiple-contact-entries-tp5373513p5373592.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Mediaproxy2/OpenSIPS 1.6 use_media_proxy onreply no audio
I also tried to upgrade from 1.6.2 to 1.6.3 same problem. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Mediaproxy2-OpenSIPS-1-6-use-media-proxy-onreply-no-audio-tp5373247p5373358.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Mediaproxy2/OpenSIPS 1.6 use_media_proxy onreply no audio
There is two way audio from UA to OpenSIPs to Asterisk, but there is no Audio
from Asterisk to OpenSIPs to UA after upgrading to OpenSIPS 1.6.2 with
Mediaproxy 2 from OpenSIPS 1.3 with Mediaproxy 1. Can someone please help me
out? I tried everything I can think of. I can see the Media session created
in Mediaproxy and Audio is pass from Mediaproxy to Asterisk but nothing from
Mediaproxy to UA. MediaRelay debug so unknow RTP address for the UA.
I tired to use engage_media_proxy on request route and use_media_proxy on
onreply_route both are not working. I also tried to use fix_nated_sdp as
well as fix_contact via nat_travesal module instead of nathelper module.
This worked before I did the upgrade with the same onreply_route. Thank You
onreply_route[3] {
if(nat_uac_test("1") ) {
xlog("L_INFO", "INFO: M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci n\n");
fix_nated_contact();
}
if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") {
if(!search("^Content-Length:[ ]*0")) {
use_media_proxy();
}
}
exit;
}
INVITE sip:5...@173.8.136.75:41136 SIP/2.0
Record-Route:
Via: SIP/2.0/UDP 7x.2x.6x.2x;branch=z9hG4bKbb2c.3f48cee.0
Via: SIP/2.0/UDP
6x.7x.1xx.5x:5060;received=6x.7x.1xx.5x;branch=z9hG4bK32b365a5;rport=5060
From: "151055" ;tag=as42c14947
To:
Contact:
Call-ID: 5d1d3f31229f52e80a59a5f67e8ec...@6x.7x.1xx.5x
CSeq: 102 INVITE
User-Agent: iWorld
Max-Forwards: 69
Date: Tue, 03 Aug 2010 23:14:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 399
Session-Expires: 90
v=0
o=root 12815 12815 IN IP4 6x.7x.1xx.5x
s=session
c=IN IP4 6x.7x.1xx.5x
t=0 0
m=audio 34852 RTP/AVP 18 4 97 3 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
SIP/2.0 100 Trying
To:
From: "151055" ;tag=as42c14947
Call-ID: 5d1d3f31229f52e80a59a5f67e8ec...@6x.7x.1xx.5x
CSeq: 102 INVITE
Via: SIP/2.0/UDP 7x.2x.6x.2x;branch=z9hG4bKbb2c.3f48cee.0
Via: SIP/2.0/UDP
6x.7x.1xx.5x:5060;received=6x.7x.1xx.5x;branch=z9hG4bK32b365a5;rport=5060
Record-Route:
Server: Linksys/PAP2T-5.1.6(LS)
Content-Length: 0
SIP/2.0 180 Ringing
To: ;tag=fcc3c088aabd3fcfi0
From: "151055" ;tag=as42c14947
Call-ID: 5d1d3f31229f52e80a59a5f67e8ec...@6x.7x.1xx.5x
CSeq: 102 INVITE
Via: SIP/2.0/UDP 7x.2x.6x.2x;branch=z9hG4bKbb2c.3f48cee.0
Via: SIP/2.0/UDP
6x.7x.1xx.5x:5060;received=6x.7x.1xx.5x;branch=z9hG4bK32b365a5;rport=5060
Record-Route:
Server: Linksys/PAP2T-5.1.6(LS)
Content-Length: 0
SIP/2.0 200 OK
To: ;tag=fcc3c088aabd3fcfi0
From: "151055" ;tag=as42c14947
Call-ID: 5d1d3f31229f52e80a59a5f67e8ec...@6x.7x.1xx.5x
CSeq: 102 INVITE
Via: SIP/2.0/UDP 7x.2x.6x.2x;branch=z9hG4bKbb2c.3f48cee.0
Via: SIP/2.0/UDP
6x.7x.1xx.5x:5060;received=6x.7x.1xx.5x;branch=z9hG4bK32b365a5;rport=5060
Record-Route:
Contact: 164699
Server: Linksys/PAP2T-5.1.6(LS)
Content-Length: 275
Allow: ACK, BYE, CANCEL, INFO, INVITE, not1510555...@6x.7x.1xx.5xify,
OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 5176 5176 IN IP4 192.168.8.201
s=-
c=IN IP4 192.168.8.201
t=0 0
m=audio 16436 RTP/AVP 18 100 101
a=rtpmap:18 G729a/8000
a=fmtp:18 annexb=no
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
SIP/2.0 200 OK
To: ;tag=fcc3c088aabd3fcfi0
From: "151055" ;tag=as42c14947
Call-ID: 5d1d3f31229f52e80a59a5f67e8ec...@6x.7x.1xx.5x
CSeq: 102 INVITE
Via: SIP/2.0/UDP 7x.2x.6x.2x;branch=z9hG4bKbb2c.3f48cee.0
Via: SIP/2.0/UDP
6x.7x.1xx.5x:5060;received=6x.7x.1xx.5x;branch=z9hG4bK32b365a5;rport=5060
Record-Route:
Contact: 164699
Server: Linksys/PAP2T-5.1.6(LS)
Content-Length: 275
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 5176 5176 IN IP4 192.168.8.201
s=-
c=IN IP4 192.168.8.201
t=0 0
m=audio 16436 RTP/AVP 18 100 101
a=rtpmap:18 G729a/8000
a=fmtp:18 annexb=no
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
SIP/2.0 200 OK
To: ;tag=fcc3c088aabd3fcfi0
From: "151055" ;tag=as42c14947
Call-ID: 5d1d3f31229f52e80a59a5f67e8ec...@6x.7x.1xx.5x
CSeq: 102 INVITE
Via: SIP/2.0/UDP 7x.2x.6x.2x;branch=z9hG4bKbb2c.3f48cee.0
Via: SIP/2.0/UDP
6x.7x.1xx.5x:5060;received=6x.7x.1xx.5x;branch=z9hG4bK32b365a5;rport=5060
Record-Route:
Contact: 164699
Server: Linksys/PAP2T-5.1.6(LS)
Content-Length: 275
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: appl
Re: [OpenSIPS-Users] Load balancer retcode question on OpenSIPS 1.6.2
I figured it out.
This work all the time
if ( uri=~"sip:92[1-9][0-...@.*" ) {
load_balance("27","white");
} else if ( uri=~"sip:3392[1-9][0-...@.*" ) {
load_balance("27","grey"); #
}
if ( $retcode < 0 ) {
sl_send_reply("500","Service full");
exit;
}
This work sometime
if ( uri=~"sip:92[1-9][0-...@.*" ) {
load_balance("27","white");
}
if ( uri=~"sip:3392[1-9][0-...@.*" ) {
load_balance("27","grey"); #
}
if ( $retcode < 0 ) {
sl_send_reply("500","Service full");
exit;
}
--
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[OpenSIPS-Users] Load balancer retcode question on OpenSIPS 1.6.2
I am using OpenSIPS 1.6.2 and followed the tutorial
http://www.opensips.org/Resources/DocsTutLoadbalancing to use the load
balancer module.
The Tutorial use $retcode<0 for Service Full reply. I get $rectcode = 1
instead of 0. What is the correct retcode load_balance("id","resource") when
resource is full?
This work for me
if ( $retcode=1 ) {
sl_send_reply("500","Service full");
exit;
}
--
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Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Check the radiusclient.conf to make sure that your dictionary is mapped to the right path. It be a good idea to check your radius log as well. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/FW-Error-when-setting-OpenSips-with-Radius-tp5342015p5345781.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Mediaproxy conntrack timeout
k1028 wrote: > > I find the problem after sniffing the packet today. The problem is that > the somehow the tunnel server sent the RTP to different port and that > Mediaproxy 2 doesn't support Asymmetric client anymore. > Please ignore my previous response. I triple looked into the sniffing and I am not a expert in this. What i can see is that the Tunnel server sending out of order sequence number on the same SSRC caused mediaproxy to change port. 64.x.x.x is tunnel sever, 74.x.x.x is mediaproxy, 54.x.x.x is asterisk. 64.x.x.x tunnel server send seq 4 and 5 to Mediaproxy on the same 40061 src and 1136 dst port 74.x.x.x mediaproxy send seq 4 to asterisk on the same 1138 src and 13746 dst port 74.x.x.x mediaproxy send seq 5 to asterisk on a different src port 1024 and 13746 dst port 54.x.x.x asterisk sent seq 53534 to mediaproxy on src port 13746 and dst port 1024 41558.34536164.x.x.x 74.x.x.x RTP PT=ITU-T G.729, SSRC=0x4C493D32, Seq=4, Time=4262923962 41568.34537364.x.x.x 74.x.x.x RTP PT=ITU-T G.729, SSRC=0x4C493D32, Seq=5, Time=4262924122 41578.34556674.x.x.x 54.x.x.x RTP PT=ITU-T G.729, SSRC=0x4C493D32, Seq=4, Time=4262923962 41598.34592274.x.x.x 54.x.x.x RTP PT=ITU-T G.729, SSRC=0x4C493D32, Seq=5, Time=4262924122 41658.35876954.x.x.x 74.x.x.x RTP PT=ITU-T G.729, SSRC=0x6004B08D, Seq=53534, Time=36800 41668.35887774.x.x.x 54.x.x.x RTP PT=ITU-T G.729, SSRC=0x6004B08D, Seq=53534, Time=36800 41838.39084764.x.x.x 74.x.x.x RTP PT=ITU-T G.729, SSRC=0x4C493D32, Seq=6, Time=4262924282 41848.39087264.x.x.x 74.x.x.x RTP PT=ITU-T G.729, SSRC=0x4C493D32, Seq=7, Time=4262924442 41858.39095774.x.x.x 54.x.x.x RTP PT=ITU-T G.729, SSRC=0x4C493D32, Seq=6, Time=4262924282 41868.39101674.x.x.x 54.x.x.x RTP PT=ITU-T G.729, SSRC=0x4C493D32, Seq=7, Time=4262924442 -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Mediaproxy-conntrack-timeout-tp5327717p5331652.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Mediaproxy conntrack timeout
I find the problem after sniffing the packet today. The problem is that the somehow the tunnel server sent the RTP to different port and that Mediaproxy 2 doesn't support Asymmetric client anymore. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Mediaproxy-conntrack-timeout-tp5327717p5331054.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Mediaproxy conntrack timeout
Sorry I forgot to mention I upgraded to OpenSIPS 1.6.2 with Mediaproxy 2.4.3. Hope someone can give me a hint thank you -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Mediaproxy-conntrack-timeout-tp5327717p5327730.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Mediaproxy conntrack timeout
I upgraded OpenSIPS 1.3 with Mediaprxoy 1 to OpenSIPS 1.6 with Mediaproxy 2.
ATA->OpenSIPS is working well. ATA->Tunnel Server->OpenSIPS get conntrack
timeout via engage_media_proxy and use_media_proxy even call is connected
with 2 way audio. I spend two days looking into this and still can't figure
out the problem.
The dialog is created in OpenSIPS, SIP Trace looked good, Media-Relay see
the SDP and the updated SDP. The stream is started but the conntrack rule is
not inserted even with audio passing both direction and mediaproxy show no
input or output octet
Debug from Media-Relay
debug: Added new stream: (audio) x.x.x.x:40325 (RTP: Unknown, RTCP: Unknown)
<-> x.x.x.x:1112 <-> x.x.x.x:1114 <-> Unknown (RTP: Unknown, RTCP: Unknown)
debug: created new session 0...@192.168.8.220: 0...@x.x.x.x:5060 (111467078)
--> 1510...@x.x.x.x:5060
debug: updating existing session 0...@192.168.8.220: 0...@x.x.x.x:5060
(111467078) --> 1510...@x.x.x.x:5060
debug: Received updated SDP answer
debug: Got initial answer from callee for stream: (audio) x.x.x.x:40325
(RTP: Unknown, RTCP: Unknown) <-> x.x.x.x:1112 <-> x.x.x.x:1114 <->
x.x.x.x:1792 (RTP: Unknown, RTCP: Unknown)
debug: Got traffic information for stream: (audio) x.x.x.x:40325 (RTP:
Unknown, RTCP: Unknown) <-> x.x.x.x:1112 <-> x.x.x.x:1114 <-> x.x.x.x:1792
(RTP: x.x.x.x:1792, RTCP: Unknown)
debug: Got traffic information for stream: (audio) x.x.x.x:40325 (RTP:
x.x.x.x:40325, RTCP: Unknown) <-> x.x.x.x:1112 <-> x.x.x.x:1114 <->
x.x.x.x:1792 (RTP: x.x.x.x:1792, RTCP: Unknown)
debug: updating existing session 0...@192.168.8.220: 0...@x.x.x.x:5060
(111467078) --> 1510...@x.x.x.x:5060
debug: Received updated SDP answer
debug: Unchanged stream: (audio) x.x.x.x:40325 (RTP: x.x.x.x:40325, RTCP:
Unknown) <-> x.x.x.x:1112 <-> x.x.x.x:1114 <-> x.x.x.x:1792 (RTP:
x.x.x.x:1792, RTCP: Unknown)
debug: expired session 0...@192.168.8.220: 0...@x.x.x.x:5060 (111467078) -->
1510...@x.x.x.x:5060
Debug from Media-Dispatcher
debug: Got statistics: {'all_streams_ice': False, 'from_tag':
'111467076719582485', 'dialog_id': '484:1906238777', 'start_time':
1279841102.099, 'timed_out': True, 'call_id': '0...@192.168.8.220',
'to_tag': 'as4b168bde', 'streams': [{'status': 'conntrack timeout',
'caller_codec': 'G729', 'post_dial_delay': 5.26600909233, 'callee_codec':
'G729', 'start_time': 0, 'caller_bytes': 0, 'callee_bytes': 0,
'caller_packets': 0, 'end_time': 60, 'callee_remote': 'x.x.x.x:1792',
'caller_remote': 'x.x.x.x:40325', 'media_type': 'audio', 'callee_local':
'x.x.x.x:1114', 'timeout_wait': 0, 'caller_local': 'x.x.x.x:1112',
'callee_packets': 0}], 'duration': 60, 'to_uri': '1510...@x.x.x.x:5060',
'from_uri': '0...@x.x.x.x:5060', 'callee_ua': 'world', 'caller_ua':
'MobileDialer'}
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Re: [OpenSIPS-Users] permission module problem
This happen to me before when i have a space at the end of the ip address inserted into the database. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/permission-module-problem-tp5273775p5276518.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] auth_db unable to connect to mysql
I did. all other module work that require mysql but auth_db. I triple checked
the db_url for auth_db..
Saúl Ibarra wrote:
>
> Did you specify the MySQL password? Maybe the option is not parsed
> correctly...
>
>
> On Wed, Jun 10, 2009 at 11:03 PM, k1028 wrote:
>>
>> I discovered that my openser givign me a
>> ERROR:mysql:db_mysql_submit_query:
>> driver error: there is no ''@'192.168.x.x' registered. This happen with
>> multiple servers with the same error message and source IP. Some reason
>> the
>> db_mysql sending no user name and wrong source IP ''@'192.168.x.x' to
>> authenticate with mysql server.
>>
>> Jun 10 20:53:22 STGPROXY01 /usr/local/sbin/openser[10335]:
>> ERROR:mysql:db_mysql_submit_query: driver error: There is no
>> ''@'192.168.x.x' registered
>> Jun 10 20:53:22 STGPROXY01 /usr/local/sbin/openser[10335]:
>> ERROR:mysql:db_mysql_query: error while submitting query
>> Jun 10 20:53:22 STGPROXY01 /usr/local/sbin/openser[10335]:
>> ERROR:auth_db:get_ha1: failed to query database
>>
>> version: openser 1.3.2-notls (x86_64/linux)
>>
>> loadmodule "auth_db.so"
>> modparam("auth_db", "db_url", "mysql://x...@192.168.x.x/")
>>
>>
>> --
>> View this message in context:
>> http://n2.nabble.com/auth_db-unable-to-connect-to-mysql-tp3058344p3058344.html
>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>>
>>
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>>
>
>
>
> --
> Saúl -- "Nunca subestimes el ancho de banda de un camión lleno de
> disketes."
>
> http://www.saghul.net/
>
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>
>
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[OpenSIPS-Users] auth_db unable to connect to mysql
I discovered that my openser givign me a ERROR:mysql:db_mysql_submit_query:
driver error: there is no ''@'192.168.x.x' registered. This happen with
multiple servers with the same error message and source IP. Some reason the
db_mysql sending no user name and wrong source IP ''@'192.168.x.x' to
authenticate with mysql server.
Jun 10 20:53:22 STGPROXY01 /usr/local/sbin/openser[10335]:
ERROR:mysql:db_mysql_submit_query: driver error: There is no
''@'192.168.x.x' registered
Jun 10 20:53:22 STGPROXY01 /usr/local/sbin/openser[10335]:
ERROR:mysql:db_mysql_query: error while submitting query
Jun 10 20:53:22 STGPROXY01 /usr/local/sbin/openser[10335]:
ERROR:auth_db:get_ha1: failed to query database
version: openser 1.3.2-notls (x86_64/linux)
loadmodule "auth_db.so"
modparam("auth_db", "db_url", "mysql://x...@192.168.x.x/")
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Re: [OpenSIPS-Users] load_balancer module retcode
I got it to work using
if ( !load_balance("40","pstn") {
sl_send_reply("500","Service FUll");
xlog("L_INFO","Service Full");
exit;
}
instead of
load_balance("40","pstn") {
if ($retcode<0 ) {
sl_send_reply("500","Service full");
exit;
}
k1028 wrote:
>
> I tried everything possible and couldn't get the return code to return a
> negative value when it reach the limiation. I have the same problem with
> the sample script from Opensips tutorial for load_balancer.so. The retcode
> is always return back as 18446744073709551614 and instead of negative
> value.
>
>
> Bogdan-Andrei Iancu wrote:
>>
>> Hi,
>>
>> I think there is a error in your scriptthe $retcode returns the
>> return code of the last used function, but your LB function is much,
>> much above the retcode testing
>>
>> Regards,
>> Bogdan
>>
>> k1028 wrote:
>>> I am playing with the Load_balancer module at this time. The retcode
>>> does not
>>> return a negative value for me instead it return 18446744073709551614
>>> when
>>> it reach the pstn limit
>>>
>>> I tried with pstn=1 and pstn=2 using 1 peer and 2 peer. All come back
>>> with
>>> the same retcode.
>>>
>>> I also tried my route script as well as the one from opensips tutorial.
>>> Also
>>> tried google, search forum and looked up tracker not able to find
>>> anything.
>>> Any help would be greatly appreciated
>>>
>>> version: opensips 1.5.1-notls (x86_64/linux)
>>>
>>> this is my route script
>>> # - Dialog params -
>>> modparam("dialog", "dlg_flag", 5)
>>> modparam("dialog", "timeout_avp", "$avp(i:4242)")
>>> #Set
>>> AVP timeout variable
>>>
>>> # - SST params -
>>> modparam("sst", "sst_flag", 6)
>>> #Set
>>> SST flag
>>> modparam("sst", "timeout_avp", "$avp(i:4242)")
>>> modparam("sst", "min_se", 10800)
>>> #Min
>>> Session Timer
>>>
>>> # - QOS params -
>>> modparam("qos", "qos_flag", 7)
>>> #Set
>>> QoS falg
>>>
>>>
>>> route{
>>>
>>> if(msg:len > max_len)
>>> {
>>> sl_send_reply("513", "Message Too Big");
>>> exit;
>>> }
>>>
>>> if (!mf_process_maxfwd_header("3")) {
>>> sl_send_reply("483","Too Many Hops");
>>> exit;
>>> }
>>>
>>> # record routing
>>> if (!has_totag()) {
>>> # initial request
>>> record_route();
>>> } else {
>>> # sequential request -> obey Route indication
>>> loose_route();
>>> t_relay();
>>> exit;
>>> }
>>>
>>> if ( is_method("INVITE") ) {
>>> if (sstCheckMin("1")) {
>>> xlog("L_ERR", "422 Session Timer Too Small reply
>>> sent.\n");
>>> exit;
>>> }
>>> # track the session timers via the dialog module
>>> setflag(5);
>>> setflag(6);
>>> setflag(7);
>>> }
>>>
>>> if ( uri=~"sip:[0-9][0-...@.*" ) {
>>> load_balance("40","pstn");
>>> xlog("L_INFO","Selected destination is: $du = $du AND
>>> retcode = $retcode \n\n");
>>> route(3);
>>> }
>>>
>>> route[3] {
>>>
>>> t_on_reply("1");
>>>
>>> # LB function returns negative if no suitable destination (for
>>> requested resources) is found,
>>> # or if all destinations are
Re: [OpenSIPS-Users] Acc table
You need to use db_extra in order to capture the username and callednumber to
log extra value that are not default. You also need to add the field in to
the database.
Look in acc module db_extra and Pseudo Variables
http://www.opensips.org/Resources/DocsCoreVar15#varpv to log extra
variables
This is from my modparam for db_extra.
modparam("acc", "db_extra", "from_uri=$fU; to_uri=$tU; ip=$si;
ua=$hdr(User-Agent)")
$fU is from username variables and from_uri is the databse table field
$tu is to URI variable and to_uri is the databse table field
$si is source Ip and ip is the database table field
$hdr(Use-Agent) is user agent headr informaiton and ua is the database table
field.
hope this help
Nhadie wrote:
>
> Hi Guys,
>
> Newbie on opensips, i was able to mysql accounting, but when i looked at
> the table logs:
>
> +++++--+--+--+-+
> | id | method | from_tag | to_tag | callid
> | sip_code | sip_reason | time
> |
> +++++--+--+--+-+
> | 1 | INVITE | 8d9f7e51 | as619fe46d |
> NDAwYzQ4NDMxMGU2Y2UxYzg3Njk0OWJhYzYwMjhlMTg. | 183 | Session
> Progress | 2009-05-09 00:11:51 |
> | 2 | INVITE | 8d9f7e51 | as619fe46d |
> NDAwYzQ4NDMxMGU2Y2UxYzg3Njk0OWJhYzYwMjhlMTg. | 200 | OK
> | 2009-05-09 00:11:57 |
>
>
> I noticed that username of the caller and the number called are not on
> that table, is there another table i should link to the acc? i looked at
> dialog table but there's nothing on it.
>
> anything i missed? thanks in advanced.
>
> regards,
> ron
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
--
View this message in context:
http://n2.nabble.com/Acc-table-tp2846361p2846421.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
___
Users mailing list
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Re: [OpenSIPS-Users] load_balancer module retcode
I tried everything possible and couldn't get the return code to return a
negative value when it reach the limiation. I have the same problem with the
sample script from Opensips tutorial for load_balancer.so. The retcode is
always return back as 18446744073709551614 and instead of negative value.
Bogdan-Andrei Iancu wrote:
>
> Hi,
>
> I think there is a error in your scriptthe $retcode returns the
> return code of the last used function, but your LB function is much,
> much above the retcode testing
>
> Regards,
> Bogdan
>
> k1028 wrote:
>> I am playing with the Load_balancer module at this time. The retcode does
>> not
>> return a negative value for me instead it return 18446744073709551614
>> when
>> it reach the pstn limit
>>
>> I tried with pstn=1 and pstn=2 using 1 peer and 2 peer. All come back
>> with
>> the same retcode.
>>
>> I also tried my route script as well as the one from opensips tutorial.
>> Also
>> tried google, search forum and looked up tracker not able to find
>> anything.
>> Any help would be greatly appreciated
>>
>> version: opensips 1.5.1-notls (x86_64/linux)
>>
>> this is my route script
>> # - Dialog params -
>> modparam("dialog", "dlg_flag", 5)
>> modparam("dialog", "timeout_avp", "$avp(i:4242)")
>> #Set
>> AVP timeout variable
>>
>> # - SST params -
>> modparam("sst", "sst_flag", 6)
>> #Set
>> SST flag
>> modparam("sst", "timeout_avp", "$avp(i:4242)")
>> modparam("sst", "min_se", 10800)
>> #Min
>> Session Timer
>>
>> # - QOS params -
>> modparam("qos", "qos_flag", 7)
>> #Set
>> QoS falg
>>
>>
>> route{
>>
>> if(msg:len > max_len)
>> {
>> sl_send_reply("513", "Message Too Big");
>> exit;
>> }
>>
>> if (!mf_process_maxfwd_header("3")) {
>> sl_send_reply("483","Too Many Hops");
>> exit;
>> }
>>
>> # record routing
>> if (!has_totag()) {
>> # initial request
>> record_route();
>> } else {
>> # sequential request -> obey Route indication
>> loose_route();
>> t_relay();
>> exit;
>> }
>>
>> if ( is_method("INVITE") ) {
>> if (sstCheckMin("1")) {
>> xlog("L_ERR", "422 Session Timer Too Small reply
>> sent.\n");
>> exit;
>> }
>> # track the session timers via the dialog module
>> setflag(5);
>> setflag(6);
>> setflag(7);
>> }
>>
>> if ( uri=~"sip:[0-9][0-...@.*" ) {
>> load_balance("40","pstn");
>> xlog("L_INFO","Selected destination is: $du = $du AND
>> retcode = $retcode \n\n");
>> route(3);
>> }
>>
>> route[3] {
>>
>> t_on_reply("1");
>>
>> # LB function returns negative if no suitable destination (for
>> requested resources) is found,
>> # or if all destinations are full
>> if ($retcode<0 ) {
>> sl_send_reply("500","Service full");
>> exit;
>> }
>>
>> # send it out
>> if (!t_relay()) {
>> sl_reply_error();
>> }
>>
>> onreply_route[1]
>> {
>> xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si
>> ID=$ci\n\n");
>> exit;
>>
>> }
>>
>>
>> exit;
>> }
>>
>>
>> Level 6 debug message
>> May 7 19:59:19 [30633] DBG:load_balancer:do_load_balance: found
>> requested
>> (0) resource pstn
>> May 7 19:59:19 [30633] DBG:dialog:build_new_dlg: new dialog
>> 0x7f77ae5740a0
>> (c=2b56f9b707a
Re: [OpenSIPS-Users] QoS Module
First thought it my my mind is QoS module will help do something to help improve the quality by keep tracking of the dialog SDP session. Now I understand and will play with it more ;). Thank you very much for the explanation. Ovidiu Sas wrote: > > Hello k1028, > > For now, the qos modules just keeps track of the SDP sessions > established inside a dialog. > You can inspect the established SDP sessions via the mi interface. > There are no modules using the qos API. You will need to write your > own module and do whatever you want to do on that module. > If something is unclear on the doc file, let me know and I will improve > it. > > > Regards, > Ovidiu Sas > > On Thu, May 7, 2009 at 3:33 PM, k1028 wrote: >> >> I tried to google around and look through all the documentation but I am >> still unclear how the QoS module work to test it on my staging network. >> According to the module documenation it keep track of the SDP session and >> provide API to be use by other module. Which other module use the QoS >> module? >> -- >> View this message in context: >> http://n2.nabble.com/QoS-Module-tp2835356p2835356.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 > > -- View this message in context: http://n2.nabble.com/QoS-Module-tp2835356p2838335.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] load_balancer module retcode
I am playing with the Load_balancer module at this time. The retcode does not
return a negative value for me instead it return 18446744073709551614 when
it reach the pstn limit
I tried with pstn=1 and pstn=2 using 1 peer and 2 peer. All come back with
the same retcode.
I also tried my route script as well as the one from opensips tutorial. Also
tried google, search forum and looked up tracker not able to find anything.
Any help would be greatly appreciated
version: opensips 1.5.1-notls (x86_64/linux)
this is my route script
# - Dialog params -
modparam("dialog", "dlg_flag", 5)
modparam("dialog", "timeout_avp", "$avp(i:4242)") #Set
AVP timeout variable
# - SST params -
modparam("sst", "sst_flag", 6) #Set
SST flag
modparam("sst", "timeout_avp", "$avp(i:4242)")
modparam("sst", "min_se", 10800)#Min
Session Timer
# - QOS params -
modparam("qos", "qos_flag", 7) #Set
QoS falg
route{
if(msg:len > max_len)
{
sl_send_reply("513", "Message Too Big");
exit;
}
if (!mf_process_maxfwd_header("3")) {
sl_send_reply("483","Too Many Hops");
exit;
}
# record routing
if (!has_totag()) {
# initial request
record_route();
} else {
# sequential request -> obey Route indication
loose_route();
t_relay();
exit;
}
if ( is_method("INVITE") ) {
if (sstCheckMin("1")) {
xlog("L_ERR", "422 Session Timer Too Small reply
sent.\n");
exit;
}
# track the session timers via the dialog module
setflag(5);
setflag(6);
setflag(7);
}
if ( uri=~"sip:[0-9][0-...@.*" ) {
load_balance("40","pstn");
xlog("L_INFO","Selected destination is: $du = $du AND
retcode = $retcode \n\n");
route(3);
}
route[3] {
t_on_reply("1");
# LB function returns negative if no suitable destination (for
requested resources) is found,
# or if all destinations are full
if ($retcode<0 ) {
sl_send_reply("500","Service full");
exit;
}
# send it out
if (!t_relay()) {
sl_reply_error();
}
onreply_route[1]
{
xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n\n");
exit;
}
exit;
}
Level 6 debug message
May 7 19:59:19 [30633] DBG:load_balancer:do_load_balance: found requested
(0) resource pstn
May 7 19:59:19 [30633] DBG:dialog:build_new_dlg: new dialog 0x7f77ae5740a0
(c=2b56f9b707a0f7bb7585ab1655349...@xxx,f=sip:x...@xx,t=sip:xx...@xxx,ft=as4634cbd6)
on hash 2403
May 7 19:59:19 [30633] DBG:dialog:populate_leg_info: route_set , contact
sip:x...@x, cseq 102 and bind_addr udp:x:5060
May 7 19:59:19 [30633] DBG:dialog:dlg_set_leg_info: set leg 0 for
0x7f77ae5740a0: tag= rr=<> ct= cseq=<102>
May 7 19:59:19 [30633] DBG:load_balancer:do_load_balance: destination
selected for LB set with free=1 (max=1)
xlog Selected destination is: $du = sip: AND retcode =1
May 7 19:59:31 [30633] DBG:dialog:build_new_dlg: new dialog 0x7f77ae578410
(c=291ea90b4956416b47e7932f06753...@xxx,f=sip:x...@x,t=sip:xx...@xx,ft=as718571da)
on hash 2865
May 7 19:59:31 [30633] DBG:core:parse_headers: flags=400
May 7 19:59:31 [30633] DBG:core:get_hdr_field: content_length=357
May 7 19:59:31 [30633] DBG:core:get_hdr_field: found end of header
May 7 19:59:31 [30633] DBG:dialog:populate_leg_info: route_set , contact
sip:xx...@xxx, cseq 102 and bind_addr udp:xxx:5060
May 7 19:59:31 [30633] DBG:dialog:dlg_set_leg_info: set leg 0 for
0x7f77ae578410: tag= rr=<> ct= cseq=<102>
May 7 19:59:31 [30633] DBG:dialog:link_dlg: ref dlg 0x7f77ae578410 with 3
-> 3
May 7 19:59:31 [30633] DBG:rr:add_rr_param: adding (;did=13b.f0a11e75)
0x780150
May 7 19:59:31 [30633] DBG:load_balancer:
d_balance: destination selected for LB set with free=0
(max=0)
May 7 19:59:31 [30633] DBG:load_balancer:do_load_balance: no destination
found
May 7 19:59:31 [30633] DBG:core:pv_get_dsturi: no destination URI
Selected destination is: $du = AND retcode = 18446744073709551614
--
View this message in context:
http://n2.nabble.com/load_balancer-module-retcode-tp2838151p2838151.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] QoS Module
I tried to google around and look through all the documentation but I am still unclear how the QoS module work to test it on my staging network. According to the module documenation it keep track of the SDP session and provide API to be use by other module. Which other module use the QoS module? -- View this message in context: http://n2.nabble.com/QoS-Module-tp2835356p2835356.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
