sr-users April 2006

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188 participants
431 discussions

Re: [Users] nat_helper: multiple media IP address in SDP
by Dmitry Lyubimkov 12 Apr '06

12 Apr '06

We had the same problem with SDP. There are very many UA with this mistake error To bypass this restriction we have added after force_rtp_proxy(); Also such command subst("/^c=IN IP4 ([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/c=IN IP4 11.22.33.44\5/i"); Dmitry ------------------------------ Message: 6 Date: Tue, 11 Apr 2006 16:51:22 +0200 From: "Nicolas Olivier" <nolivier(a)alphalink.fr> Subject: Re: [Users] nat_helper: multiple media IP address in SDP To: "Bogdan-Andrei Iancu" <bogdan(a)voice-system.ro> Cc: users(a)openser.org Message-ID: <443BC26A.8040407(a)alphalink.fr> Content-Type: text/plain; format=flowed; charset="ISO-8859-1" Ok, I may have a look to the csv. Thanks for the help. regards, Nicolas Bogdan-Andrei Iancu wrote: > Hi, > > Nicolas Olivier wrote: > > > > > Hi Bogdan, > > > > Ok, I understand now. But I still encounter the problem because: > > - rtpproxy only rewrites the c= from media part (but it should be fine > > as you said) despite what a quick look in the rtpproxy code comments > > say ("We have to change ports in m-lines, and also change IP addresses > > in c-lines which can be placed either in session header (fallback for > > all medias) or media description.") > > yes, the nathelper will change the c= from session header only if it > finds a media section without a local c= (which means the default c= > from session hdr will be used). > > > - the centrex (which is an asterisk by the way) take only into account > > the c= from the session part, not the one from media part > > in the CVS devel there is a flag that force also changing of session c= : > http://openser.org/docs/modules/1.1.x/nathelper.html#AEN275 , the > "c" flag > > regards, > bogdan > > > > > > > > > regards, > > Nicolas > > > > Bogdan-Andrei Iancu wrote: > > > >> Hi Nicolas, > >> > >> it;s perfectly ok - see the SDP RFC : an SDP may contain a default > c= in > >> the session part; each media section (m=) may contain an ip (c=); if it > >> doesn't the session c= will be used. > >> > >> regards, > >> bogdan > >> > >> Nicolas Olivier wrote: > >> > >> > > >> > Hi, > >> > > >> > I've got a gateway which is only used for rounting and rtp proxying > >> > between providers and centrexes. > >> > > >> > On reply to an INVITE, one of our provider send back a "183 Session > >> > Progress". The problem is that in the SDP block, we've got 2 > media IP > >> > address and rtpproxy only rewrite one. > >> > > >> > Finally, the provider establish rtp session with our gateway, and > our > >> > centrex directly with the provider. > >> > > >> > provider gateway centrex > >> > 172.16.0.10 192.168.1.10 192.168.1.20 > >> > RTP -------------> RTP ------------> RTP > >> > ^-------------------------------------------------| > >> > > >> > So my questions are, is it possible to have multiple IP address in > >> SDP > >> > and if so, how can I tell rtpproxy to rewrite all of them. > >> > > >> > Coming from provider: > >> > > >> > SIP/2.0 183 Session Progress. > >> > Via: SIP/2.0/UDP > >> > 192.168.1.10;branch=z9hG4bKdd67.a4cc2c44.0,SIP/2.0/UDP > >> > 192.168.1.20:5062;branch=z9hG4bKdd67.08f45a33.0,SIP/2.0/UDP > >> > 192.168.1.20:5060;branch=z9hG4bK4af242b7. > >> > From: "02" <sip:0143132445@192.168.1.20>;tag=as226ce7b9. > >> > To: <sip:0123456789@192.168.1.20:5062>;tag=3123AAA8-20C5. > >> > Date: Tue, 11 Apr 2006 09:10:29 GMT. > >> > Call-ID: 079ab6663e403ff44a1107e5111b075f(a)192.168.1.20. > >> > Server: Cisco-SIPGateway/IOS-12.x. > >> > CSeq: 102 INVITE. > >> > Allow-Events: telephone-event. > >> > Contact: <sip:677238#0123456789@172.16.0.10:5060>. > >> > Record-Route: > >> > > >> > <sip:192.168.1.10;ftag=as226ce7b9;lr=on>,<sip:192.168.1.20:5062;ftag=as2 26ce7b9;lr=on>. > > >> > >> > > >> > Content-Disposition: session;handling=required. > >> > Content-Type: application/sdp. > >> > Content-Length: 261. > >> > . > >> > v=0. > >> > o=CiscoSystemsSIP-GW-UserAgent 3448 4768 IN IP4 172.16.0.10. > >> > s=SIP Call. > >> > c=IN IP4 172.16.0.10. > >> > t=0 0. > >> > m=audio 18322 RTP/AVP 18 101. > >> > c=IN IP4 172.16.0.10. > >> > a=rtpmap:18 G729/8000. > >> > a=fmtp:18 annexb=no. > >> > a=rtpmap:101 telephone-event/8000. > >> > a=fmtp:101 0-16. > >> > > >> > Forwarded to centrex: > >> > > >> > SIP/2.0 183 Session Progress. > >> > Via: SIP/2.0/UDP > >> > 192.168.1.20:5062;branch=z9hG4bK43a4.3e96aba3.0,SIP/2.0/UDP > >> > 192.168.1.20:5060;branch=z9hG4bK3213db83. > >> > From: "02" <sip:0143132445@192.168.1.20>;tag=as1a2f900d. > >> > To: <sip:0123456789@192.168.1.20:5062>;tag=3121D1B4-1BFD. > >> > Date: Tue, 11 Apr 2006 09:08:28 GMT. > >> > Call-ID: 08467c5e299ab833106517c63d3edc2e(a)192.168.1.20. > >> > Server: Cisco-SIPGateway/IOS-12.x. > >> > CSeq: 102 INVITE. > >> > Allow-Events: telephone-event. > >> > Contact: <sip:677238#0123456789@172.16.0.10:5060>. > >> > Record-Route: > >> > > >> > <sip:192.168.1.10;ftag=as1a2f900d;lr=on>,<sip:192.168.1.20:5062;ftag=as1 a2f900d;lr=on>. > > >> > >> > > >> > Content-Disposition: session;handling=required. > >> > Content-Type: application/sdp. > >> > Content-Length: 277. > >> > . > >> > v=0. > >> > o=CiscoSystemsSIP-GW-UserAgent 565 174 IN IP4 172.16.0.10. > >> > s=SIP Call. > >> > c=IN IP4 172.16.0.10. > >> > t=0 0. > >> > m=audio 36296 RTP/AVP 18 101. > >> > c=IN IP4 192.168.1.10. > >> > a=rtpmap:18 G729/8000. > >> > a=fmtp:18 annexb=no. > >> > a=rtpmap:101 telephone-event/8000. > >> > a=fmtp:101 0-16. > >> > a=nortpproxy:yes. > >> > > >> > > >> > openser.cfg > >> > > >> > (...) > >> > > >> > onreply_route[1] { > >> > if (status =~ "(180)|(183)|2[0-9][0-9]") { > >> > fix_nated_contact(); > >> > if (!search("^Content-Length:[ ]*0")) { > >> > force_rtp_proxy(); > >> > }; > >> > } else if (nat_uac_test("1")) { > >> > fix_nated_contact(); > >> > }; > >> > } > >> > > >> > (...) > >> > > >> > Best regards, > >> > Nicolas Olivier > >> > > >> > > >> > _______________________________________________ > >> > Users mailing list > >> > Users(a)openser.org > >> > http://openser.org/cgi-bin/mailman/listinfo/users > >> > > >> > > > ------------------------------ Message: 7 Date: Tue, 11 Apr 2006 16:52:14 +0200 From: Cesc <cesc.santa(a)gmail.com> Subject: Re: [Users] Allow only TLS connections To: "Thorsten.Haupt(a)t-systems.com" <Thorsten.Haupt(a)t-systems.com> Cc: users(a)openser.org Message-ID: <ce8208420604110752i733143d8k5b565ac45b0cfda2(a)mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 http://openser.org/dokuwiki/doku.php?id=openser_core_cookbook&DokuWiki=6 c17b007ea61fa37b86b391ce1b2a80f#tcp On 4/11/06, Thorsten.Haupt(a)t-systems.com <Thorsten.Haupt(a)t-systems.com> wrote: > I searched for this function, but I didn't found it :-( > Knows anyone the correct code, not only pseudo-code? > > Torsten > > -----Ursprьngliche Nachricht----- > Von: Cesc [mailto:cesc.santa@gmail.com] > Gesendet: Dienstag, 11. April 2006 14:03 > An: Haupt, Thorsten > Cc: users(a)openser.org > Betreff: Re: [Users] Allow only TLS connections > > I think in openser there is a function to check what transport the message came in ... you can do something like: > if ( transport != TLS ) { > send error to UA > break; > } > > Cesc > > On 4/11/06, Thorsten.Haupt(a)t-systems.com <Thorsten.Haupt(a)t-systems.com> wrote: > > > > > > Hello, > > > > I use OpenSER in a testing environment for VoIP security. My clients > > connect via TLS. If I deactivate UDP/5060 on the server, it doesn't work correct. > > Some Clients can't connect and others can't establish calls. I read in > > another thread, that UDP is mandatory for SIP and that the server need it. > > > > But how can I prevent users from connecting via UDP and force them to > > use TLS? I tried a firewall, blocking UDP and TCP on port 5060. But is > > this the correct way? Are there any parameters server-side to force > > users to connect via TLS? > > > > Thanks for response. > > Torsten > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > ------------------------------ Message: 8 Date: Tue, 11 Apr 2006 17:16:49 +0200 From: Andreas Granig <andreas.granig(a)inode.info> Subject: [Users] Overlapping AVPs To: users(a)openser.org Message-ID: <443BC861.6040303(a)inode.info> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi, Me again, sorry, but the docs aren't really noisy about AVP details... So if I have user preferences for both the caller and callee and load them from DB and print them using the following: avp_db_load("$avp($uuid_caller)", ""); avp_db_load("$avp($uuid_callee)", ""); avp_print(); then they may overlap because of the same ID (say "i:102" for toggling some specific feature on/off), but according to the debug output both are present: INFO:avpops:print_avp: p=0x4056db90, flags=100 INFO: id=<102> INFO: val_int=<1> INFO:avpops:print_avp: p=0x4056dc68, flags=100 INFO: id=<102> INFO: val_int=<0> So is it possible to selectively access the avp-value of both $uuid_caller and $uuid_callee? Something like $avp(i:102)[0] and $avp(i:102)[1] maybe? Thanks, Andy ------------------------------ Message: 9 Date: Tue, 11 Apr 2006 17:54:56 +0200 From: "D'Addelfio Davide" <Davide.D'Addelfio(a)italtel.it> Subject: R: [Users] load from db table To: "Bogdan-Andrei Iancu" <bogdan(a)voice-system.ro> Cc: users(a)openser.org Message-ID: <82C94EFCF026F74EB91A2048B1C963A504065277(a)BESONE.corp.dom> Content-Type: text/plain; charset="iso-8859-1" Hi Bogdan, I'm trying to setup my config file , did the same modparam("avpops", "db_scheme", "scheme0:username_col=from;value_col=timestamp;value_type=string;table=a cc") modparam("avpops","avp_aliases","timestamp=i:800") if (method=="INVITE") avp_db_load("$from","$timestamp/$scheme0"); I'm not sure that is correct... log gives me these errors Apr 11 08:42:59 localhost /usr/sbin/openser[13752]: submit_query: You have an error in your SQL syntax. Check the manual that corresponds to your MySQL server version for the right syntax to use near 'from='bob'' at line 1 Apr 11 08:42:59 localhost /usr/sbin/openser[13752]: db_query: Error while submitting query Apr 11 08:42:59 localhost /usr/sbin/openser[13752]: ERROR:avpops:load_avps: db_load failed Any thought? Thanks Davide -----Messaggio originale----- Da: users-bounces(a)openser.org [mailto:users-bounces@openser.org] Per conto di Bogdan-Andrei Iancu Inviato: martedм 11 aprile 2006 16.12 A: D'Addelfio Davide Cc: users(a)openser.org Oggetto: Re: [Users] load from db table Hi, see: http://www.voice-system.ro/docs/avpops/ar01s06.html#avp_db_load the "db_scheme" example. regards, bogdan D'Addelfio Davide wrote: >Hi Bogdan, > >i setup my openser.cfg to store SIP messages into acc tables of mysql, >using extra accounting to store also the body part of the message. >Now i need that openser read into that db's table, in particular in some >rows of db. >If I use avp_db_load it works only over usr_preference table, instead I >want it looks into acc table. > >How can I do? > >Thanks for help >Davide > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users ------------------------------ _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users End of Users Digest, Vol 11, Issue 27 *************************************

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[Serusers] About the Inter SIP Proxy Problem
by AR. 12 Apr '06

12 Apr '06

Dear ALL I have two SIP Proxy servers , One is SER, another one is CSPS (cisco) When I call a phone number that is belong to the CSPS routing, using the SER subscriber, the packet form SER to CSPS will get a problem with "400 Bad request" . (It's ok, If the UA in CSPS call to the UA in SER) My architecture is : UA1----SER----B2BUA---RADIUS | | Internet | | CSPS--UA2 Logs of B2BUA : 20060412-144443.276 [57396144] WARN bcWorker: uaState/UaStateRinging.cxx:133 UaStateRinging::Received 400 20060412-144443.276 [57396144] INFO bcWorker: controlState/StateCallReady.hxx:98 StateCallReady::bye was called, incomplete transaction Does anyone know how to solve it? thanks AR.

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[Serusers] chown'ing /tmp/ser_fifo
by Nick Hoffman 12 Apr '06

12 Apr '06

Hi guys, I need /tmp/ser_fifo to *always* be owned by root:foobar. What's the best way of accomplishing this? I was thinking of chown'ing the fifo at the end of ser's init script. However, I'm not sure when the fifo is created when starting up ser, or whether the fifo is ever recreated (thus requiring a new chown). Any suggestions? I'm all ears! -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

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[Users] avp_db_load problem
by Jayesh Nambiar 12 Apr '06

12 Apr '06

hi, It worked after I changed the "$fU" to "$fu". Now it properly does the avp_db_load(). jayesh --------------------------------- Jiyo cricket on Yahoo! India cricket Yahoo! Messenger Mobile Stay in touch with your buddies all the time.

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[Users] avp_db_load problem
by Jayesh Nambiar 11 Apr '06

11 Apr '06

hi, Sorry there was a typo. I get the message "entered condition" in the logs, which indicates it does not do the avp_db_load. It just enters the condition of the uri. thanks, jayesh --------------------------------- Jiyo cricket on Yahoo! India cricket Yahoo! Messenger Mobile Stay in touch with your buddies all the time.

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[Users] avp_db_load problem
by Jayesh Nambiar 11 Apr '06

11 Apr '06

Hi all, I was trying to do the following: Whenever a user presses *86 and then any number, first openser should check in the subscriber table, whether the value for features column is 1. (I've added an extra column features in the table subscriber with datatype int). If the value is 0, he should be replied with the message Forbidden. I tried doing the following: modparam("avpops","db_scheme", "scheme1:username_col=username;value_col=features;table=subscriber") if(uri=~"sip:\*86[0-9]*@") { log(1,"entered condition"); if(avp_db_load("$fU", "$avp(s:features)/$scheme1")) { log(1,"db load succeeded"); if(!avp_check("$avp(s:features)", "eq/1/g")) { sl_send_reply("403", "Forbidden"); exit; }; strip(3); log(1,"stripped"); avp_db_query("insert into usr_preferences values('', '$fU', '', 'callfwd', '', '$ruri', 'CURRENT_TIMESTAMP')"); sl_send_reply("200", "OK"); exit; }; }; But in the logs I only get "condition succeeded" which means it does not do the avp_db_load() properly. Can someone please help me in what should be the exact syntax to load it from the db and then check the value. thanks in advance, jayesh --------------------------------- Jiyo cricket on Yahoo! India cricket Yahoo! Messenger Mobile Stay in touch with your buddies all the time.

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[Users] Random error with NAT (error in 'c field')
by Bruno Machado 11 Apr '06

11 Apr '06

Hi Bogdan Do you want to see all the SIP messages from this call? Is it? Thanks for yor help. --------------------------------- Yahoo! Search Imposto de Renda 2006: o prazo está acabando. Faça já a sua declaração no site da Receita Federal.

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[Users] Forward response
by Benjamin Lawetz 11 Apr '06

11 Apr '06

Hello, Classic situation, overgrown asterisk and need to go openser for load-balancing and failover reasons. We want to do an incremental upgrade to openser, and for the moment we are trying to use openser as a fancy router in front of asterisk. I'm having a little problem with load-balancing the registration. Clients are mostly behind NAT sending registration to openser. I'm using the dispatcher module to load-balance it. The registration is going from the client to the openser, openser is forwarding it to asterisk. Asterisk replies to openser, and that's where I'm stuck. I want openser to forward the "TRYING" and "OK" response (and obviously all other responses) to the client, but can't seem to figure out how. Can anybody point me in the right direction? thanks

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[Users] AVPOPS manipulating header fields
by Martin Kjeldsen 11 Apr '06

11 Apr '06

Hi all I am in the need for a little assistance on how to best use some data taken from a invite request, and then create an appropriate AVP. My idea was to take the from header field, avp_write("$from", "$fromuri"); Then use it to extract the alias which belongs to this user, and create the AVP, avp_db_load("$fromuri","s:pai/$scheme0"); And then incorperate it into another AVP, in order to add a callerid, via P-Asserted-Identity, avp_printf("i:10","<sip:$avp(s:pai)@sip.domain.com>"); remove_hf("P-Asserted-Identity"); #remove org. hdr avp_pushto("$P-Asserted-Identity/request","i:10"); # set new hdr My question is, how do I best remove the port (eg. :5060) indication at the end of the from header field? Because the clients on a private IP range and my alias table does not carry the port numbers (eg. 88888888(a)sip.domain.com ). So I need to manipulate the $fromuri, by removing the port indication, before i do the avp_db_load! I was thinking of using the avp_subst function, but cant quite seem to get it to work. Can anyone help me with a smart way to solve this problem? All help greatly appreciated. Kind regards Martin Kjeldsen

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[Serusers] BYE
by Thomas Britis 11 Apr '06

11 Apr '06

Hi all, Is there a way to finish a call without a BYE? Here is my problem: I have my own billing system and I terminate a established call just when I get a BYE message from client (or gateway). So I was thinking if one could terminate the call without the BYE. Thank you. -- Thomas Storino Britis TCNet Informatica e Telecomunicacoes LTDA

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sr-users April 2006