[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? -

[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 scr

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

Re: [OpenSIPS-Users] load_balancer module retcode

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 > >

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 T

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 ) {

[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 auth

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: >>

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 Na

[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 figu

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 maili

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

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 look

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

[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

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-

[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 crea

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. _

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-entr

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 - U

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

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-serv

[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 - m

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. ___

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 mailin

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

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-Rout

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 --

[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

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' clon

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"

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-tp5534

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 --

[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

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-tp681727

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-tp68172

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 Zo

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

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:

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. N

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 m

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

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/libmys

[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(la

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 OpenS