sr-users December 2006

sr-users@lists.kamailio.org

152 participants
258 discussions

[Serusers] mysql tables help
by Katty Xiong 22 Dec '06

22 Dec '06

I am tring to use Ottendorf. I have db tables for 0.9.6 on the machine, so I decide to drop the old tables and create new tables using ser_mysql.sh for Ottendorf, but the script keeps on giving me errors. Do I have to delete the entire old version of SER (0.9.6)? thanks, Joy __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Users] help needed on avpops
by Zoran Milic 22 Dec '06

22 Dec '06

Hi guys, I am trying to set up radius and avp for ages now, and this time i think i am close to it. I have a problem and I can't find any information about it. The thing is, my soft agent (set to uid 101) can't register, and it is getting "Forbidden IP" even with this in the cfg file (see below) I have set up freeradius and radiusclient-ng according to the manual on http://www.openser.org/docs/openser-radius-1.0.x.html. I can't find anything on the error i am getting in my log. If you need my complete ..cfg or log file, just ask, and I will send them promptly. thanks in advance, best regards, Zoran openser.cfg: .... xlog("L_ERR","rpid = $avp(s:rpid)"); xlog("L_ERR","adresa = $avp(s:adresa)"); xlog("L_ERR","adresa2 = $avp(s:adresa2)"); # check the src ip address # avp_load_radius("caller"); if(!avp_check("$adresa", "eq/$adresa/ig")) { sl_send_reply("403", "Forbidden IP"); exit; }; save("location"); exit; }; .... freeradius/users: .... 101 Auth-Type := Digest, User-Password == "101" Reply-Message = "Authenticated", Sip-Avp += "rpid:101", Sip-Avp += "adresa:10.0.0.157", Sip-Avp += "adresa2:10.0.0.157" .... last entries in the log after the rejected register. .... DEBUG:auth_radius:extract_avp: AVP val is <10.0.0.157> DEBUG:auth_radius:generate_avps: AVP 'adresa'/0='10.0.0.157'/0 has been added DEBUG:auth_radius:extract_avp: string is <adresa2:10.0.0.157> DEBUG:auth_radius:extract_avp: AVP name is <adresa2> DEBUG:auth_radius:extract_avp: AVP val is <10.0.0.157> DEBUG:auth_radius:generate_avps: AVP 'adresa2'/0='10.0.0.157'/0 has been added DEBUG:auth_radius:extract_avp: string is <rpid:101> DEBUG:auth_radius:extract_avp: AVP name is <rpid> DEBUG:auth_radius:extract_avp: AVP val is <101> DEBUG:auth_radius:generate_avps: AVP 'rpid'/0='101'/0 has been added DEBUG:auth_radius:extract_avp: string is <adresa:10.0.0.157> DEBUG:auth_radius:extract_avp: AVP name is <adresa> DEBUG:auth_radius:extract_avp: AVP val is <10.0.0.157> DEBUG:auth_radius:generate_avps: AVP 'adresa'/0='10.0.0.157'/0 has been added DEBUG:auth_radius:extract_avp: string is <adresa2:10.0.0.157> DEBUG:auth_radius:extract_avp: AVP name is <adresa2> DEBUG:auth_radius:extract_avp: AVP val is <10.0.0.157> DEBUG:auth_radius:generate_avps: AVP 'adresa2'/0='10.0.0.157'/0 has been added rpid = 101 adresa = 10.0.0.157 adresa2 = 10.0.0.157 xl_get_spec_value: error - null sp->itf ERROR:avpops:ops_check_avp: cannot get src value parse_headers: flags=ffffffffffffffff check_via_address(192.168.52.145, 192.168.52.145, 0) DEBUG:destroy_avp_list: destroying list 0x404d9238 receive_msg: cleaning up

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Re: [Users] OpenSER as load balancer for several Asterisk servers
by Edoardo Serra 22 Dec '06

22 Dec '06

Hi Users, I'm here again. After having read more docs I don't feel like using rtpproxy My Asterisks have both public IP Adresses so I don't need to proxy rtp stream I just need a configuration to randomly proxy SIP requests to Asterisk boxes behind SER (for load balancing purposes) and to handle registrations of user to forward incoming calls to them (calls are coming from Asterisks to OpenSER) I'm really stuck on it because I already have such a working setup, with same config files. I'm just trying to replicate it on another site Tnx in advance for help Edoardo My last mail to the ml follow for reference Tnx again Klaus, i'll read docs more in depth and try to understand each change to openser.cfg you suggested >>U 2006/12/20 19:15:35.025446 OOO.OOO.OOO.OOO:5060 -> CCC.CCC.CCC.CCC:21722 >>SIP/2.0 183 Session Progress. >>Via: SIP/2.0/UDP >>OOO.OOO.OOO.OOO:5060;branch=z9hG4bK-d87543-3e229802603d7c32-1--d87543-;rport=21722. >> > >^^^^^^^^^^^^^^ > >strange bug again - there must be CCC.CCC.CCC.CCC really strange, do you think it's an openser bug ? The strangest thing is that I just copied openser configs from a working system (Openser + Asterisks) changing just ip addresses OpenSER version is also the same... [...] >># !! Nathelper >>onreply_route[1] { >> # NATed transaction ? >> if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { >> fix_nated_contact(); >> #force_rtp_proxy(); > ^ >Are you sure it is commented? I do not believe it because the ngrep >shows that the SDP of 200 Ok is rewritten. Sure, I also tried to remove the line without solving the previous problem >Please read the Getting Started Turial from onsip.org carefully. It >describes how you handle NAT correctly and also describes how to do >NAT traversal for in-dialog messages, which is missing in your config. I will. Tnx again Regards Edoardo

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Re: [Users] Load-Balancing
by raviprakash sunkara 22 Dec '06

22 Dec '06

May i guess U Perform the Dispatcher modules ..... May this Hopes u to get On 12/22/06, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> wrote: > > > > > > On 12/22/06, Thomas Deillon < Thomas.Deillon(a)smart-telecom.ch> wrote: > > > > Hi, > > > > I'm trying to set a load balancing solution with SER and twos Asterisk. > > The Asterisks will be use as REGISTRAR, REDIRECT, GATEWAY .... > > > > The ser will just load balance the traffic on one Asterisk or an other. > > I use the dispatcher module for the SIP message and I use 2 public ip > > addresses for the asterisk and so, I'm able to directly send the RTP > > traffic to them (I didn't use mediaproxy or rtpproxy). > > > > On each asterisk, I set the SER as default GW to be sure that all > > message go through this computer. (Client are behind a FW/NAT and so, I > > would like to be sure that all message come from the same IP > > address:port else, the FW/NAT will DROP these messages). > > > > For the moment, with this solution, I have a problem. The message from > > client will go to SER which will resend it to the asterisk and then, the > > > > asterisk will try to answer responding on the same IP -> Loop detected > > !! > > > > On some device I saw that we can forward the request for the phone to > > the asterisk without changing the source ip address .... > > > > Do you know how can I do this ? > > > > Thanks a lot for your help, > > > > Thomas D > > > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com > M:+91 9985077535 > O:+91 40 23114549 > F:+91 40 40208727 > ravi.sunkara(a)hyperion-tech.com > www.hyperion-tech.com -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

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[Users] Routing calls to a sip terminator
by Xantek, Inc. 22 Dec '06

22 Dec '06

I am sure this must have been covered somewhere, but I cannot find it. I am trying to get openser to register with username and password and forward a call to that server. This is for outbound sip termination to the PSTN. If somebody could point me to the right place, I would be grateful. Bill

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[Users] Load-Balancing
by Thomas Deillon 21 Dec '06

21 Dec '06

Hi, I'm trying to set a load balancing solution with SER and twos Asterisk. The Asterisks will be use as REGISTRAR, REDIRECT, GATEWAY .... The ser will just load balance the traffic on one Asterisk or an other. I use the dispatcher module for the SIP message and I use 2 public ip addresses for the asterisk and so, I'm able to directly send the RTP traffic to them (I didn't use mediaproxy or rtpproxy). On each asterisk, I set the SER as default GW to be sure that all message go through this computer. (Client are behind a FW/NAT and so, I would like to be sure that all message come from the same IP address:port else, the FW/NAT will DROP these messages). For the moment, with this solution, I have a problem. The message from client will go to SER which will resend it to the asterisk and then, the asterisk will try to answer responding on the same IP -> Loop detected !! On some device I saw that we can forward the request for the phone to the asterisk without changing the source ip address .... Do you know how can I do this ? Thanks a lot for your help, Thomas D

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[Serusers] Re: How to change Caller ID in FROM field ?
by lawrence k.y. lin 21 Dec '06

21 Dec '06

Hi all, Appreciate if someone can give me a suggestion or even an answer. Since the call from SIP device to PSTN phone did not have Caller ID and our Telco carriers request for the Caller ID, we have to set a number before the calls are routed to PSTN gateway. I did it in Asterisk before with the function calls SetCallerID(CLID) or Set(CALLERID(number)=CLID) (depends on the Asterisk version) and these function calls worked. But because of the scalabliity issue, I changed to SER and would like to do the similar thing as in Asterisk. I read the SER document and awared SIP_HF_FROM has the data for the entired FROM field. Is there any other variables for Caller ID only ? I searched the mailing list for SER and did not find any related topic (only found one with title: caller-id with raius using sip-rpid, but it is irrelevant). Thanks in advance. Larry _________________________________________________________________ Try the next generation of search with Windows Live Search today! http://imagine-windowslive.com/minisites/searchlaunch/?locale=en-us&source=…

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[Users] Realtime Call Monitoring
by lists＠infoway.net 21 Dec '06

21 Dec '06

We are Asterisk-only users and currently use Asterisk Flash Operator Panel (http://www.asternic.org/) to get a real-time view of the different phones/extensions, channels, etc. We are planning on enhancing our setup by adding SER/OpenSER in front of our Asterisk systems and to expand into other venues as well. If we wanted to use something like SEMS instead of Asterisk for certain services, is there any tool similar to the Flash Operator Panel (FOP) available for SER/OpenSER + SEMS? I guess we could always continue using FOP with our Asterisk systems fronted by SER/OpenSER. However, for our other plans where we prefer to use SEMS, we would love to see if there is any solution available (free, sponsored, or commercial). Thanks, Daniel

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[Users] Re: any kind of statistics available
by Daniel-Constantin Mierla 21 Dec '06

21 Dec '06

Just to complete the thread, a new SNMP module is available for such purpose. In the announcement are listed several tools that can help to collect statistics and generate grphs. It is a matter of configuration and maybe small shell scripting: http://openser.org/pipermail/users/2006-December/008334.html http://cacti.net/ http://oss.oetiker.ch/mrtg/ At the first sight these tools can be used with FIFO or XMLRPC interface as well, if you write the shell script to collect the statistics. Cheers, Daniel Wienand, Joerg wrote: >> You can make a small script to pull periodically statistics via >> FiFO/XMLRPC into a database/file and from there generate graphics at >> your convenience. >> > > This is what I thought about. I expect that I'm not the only one who is > interested in statistics. So maybe anyone else has done this job before. ;-) > > regards, Jörg > >

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Re: [Users] Problem with avp_check
by Bogdan-Andrei Iancu 21 Dec '06

21 Dec '06

Hi Jose, I think your regexp is not correct - you need to force the end of string, otherwise only a substring (prefix) will be matched. try avp_check("$fU","re/^[0-9]{10}$/g") regards, bogdan Jose Gil Navarrete wrote: > Hi thanks Andrei, I tried with your recomendation but I obtain the > same result, if $fu is: 012345678901, then "To-username > -012345678901-" is inserted > > if(avp_check("$fU","re/^[0-9]{10}/g")) { > insert_hf("To-username: -$rU- \r\n", "Call-ID"); > } else { > sl_send_reply("400", "Peticion Erronea, numero > invalido"); > }; > > > */Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>/* escribió: > > Hi Jose, > > the FM operator in avp_check is not for regular expressions - FM is a > file matching- like operator (like "sip:*@domain). See "man 3 > fnmatch". > > if you want to check against an regexp, you need to use the RE > operator. > See: > http://www.openser.org/docs/modules/1.2.x/avpops.html#AEN384 > > as RE op does not support an AVP as right operand, you should try: > if(avp_check("$fU","re/^[0-9]{10}/g")) { > > regards, > bogdan > > Jose Gil Navarrete wrote: > > > Hi > > I try to check the username in the URI, this is, if the username do > > not have ten digits then send a message, I made: > > > > avp_write("^[0-9]{10}","$avp(s:fm_avp)"); > > if(avp_check("$fU","fm/$avp(s:fm_avp)/g")) { > > insert_hf("To-username: -$fU- \r\n", "Call-ID"); > > } else { > > sl_send_reply("400", "Bad Request"); > > exit; > > }; > > > > but I ever obtain 400 message. > > > > I try too with: > > if(avp_check("$fU","re/[0-9]{10}$/g")) { > > insert_hf("To-username: -$fU- \r\n", "Call-ID"); > > else{ > > sl_send_reply("400", "Bad request"); > > exit; > > } > > > > But I obtain the same message. Any idea? > > > > __________________________________________________ > > Correo Yahoo! > > Espacio para todos tus mensajes, antivirus y antispam ¡gratis! > > Regístrate ya - http://correo.yahoo.com.mx/ > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > __________________________________________________ > Correo Yahoo! > Espacio para todos tus mensajes, antivirus y antispam ¡gratis! > Regístrate ya - http://correo.yahoo.com.mx/ >

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sr-users December 2006