sr-users February 2006

sr-users@lists.kamailio.org

189 participants
373 discussions

[Users] timeout problem with media gateway

by Matteo Piazza

HI, In my serial forking configuration I have this problem: if i use the a media gateway ( asterisk ) on the same machine ( but I see also on the another machine) of openser ( on the 5061 port) I have a problem with the timeout of a call. The call continues beyond the timeout and when by the PSTN telephone I reject the call appears the following log message: Warning: sl_send_reply: I won't send a reply for ACK!! Where I have mistake? Any idea? Another thing. I see that I redirect the call directly in the media gateway, (for example a location of a user is a PSTN number) whitout the traslation of IP number, all is ok. Matteo ====================================== # # $Id: openser.cfg$ # # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/lcr.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("registrar", "append_branches", 1) #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("usrloc","db_url","mysql://openser:heslo@127.0.0.1/openser") modparam("auth_db","db_url","mysql://openser:heslo@127.0.0.1/openser") modparam("lcr","db_url","mysql://openser:heslo@127.0.0.1/openser") modparam("avpops","avp_url","mysql://openser:heslo@127.0.0.1/openser") modparam("avpops","avp_table","usr_preferences") modparam("tm", "fr_timer", 7) modparam("tm", "fr_inv_timer", 10) modparam("tm", "wt_timer", 5) #modparam("avpops","avp_aliases","fwdbusy=i:665") # ------------------------- request routing logic ------------------- # main routing logic route{ # ============================================== # initial sanity checks -- messages with # max_forwards==0, or excessively long requests # ============================================== if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # ============================= # Notify Keep-Alive Section # ============================ if ((method=="NOTIFY") && search("Event: keep-alive")) { sl_send_reply("200","OK"); exit; }; if (method =="INVITE" && uri =~"^sip:0[0-9]*@*"){ log(1, "Check 1 Start PSTN Call\n"); rewritehostport("192.168.9.97:5061"); } # ======================================================== # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol #======================================================== if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if(method=="BYE"){ t_relay(); exit; }; # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("create-net.it", "subscriber")) { www_challenge("create-net.it", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; # ======================================= # PSTN Getaway # ======================================= if(method=="INVITE"){ if (avp_db_load("$ruri","s:fwdactive")) { if(avp_check("s:fwdactive","eq/y/i")){ log(1,"FWD ACTIVE\n!"); setflag(2); } }; if (avp_db_load("$ruri","s:mailactive")) { if(avp_check("s:mailactive","eq/y/i")){ log(1,"MAIL ACTIVE\n!"); setflag(4); } } if(load_contacts()){ xlog("L_ERR","LOAD CONTACTS!\n"); setflag(1); }; if(next_contacts()){ xlog("L_ERR","NEXT CONTACT!\n"); }; }; if(isflagset(2)){ t_on_failure("1"); t_relay(); exit; } else { if(isflagset(4)){ t_on_failure("2"); t_relay(); exit; } else { t_on_failure("3"); t_relay(); exit; } }; } route[1] { # log(1,"ROUTE 1 !\n"); if (!t_relay()) { sl_reply_error(); }; exit; } route[2] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP log(1,"ROUTE 2 !\n"); sl_send_reply("181","CALL IS BEING FORWARDED"); t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1]{ log(1,"FAILURE ROUTE FORWARD !\n"); if (t_check_status("408")){ if (next_contacts()) { log(1,"NUOVO CONTATTO !\n"); route(2); exit; } else { log(1,"FINE CHIAMATA - TIME OUT!\n"); if (isflagset(4)){ log(1,"ATTIVAZIONE VOICEMAIL!\n"); revert_uri(); xlog("L_ERR","<$ruri>"); rewritehostport("192.168.9.97:5061"); append_branch(); t_relay(); exit; } exit; }; } else { if ( t_check_status("603") || t_check_status("486") ){ log(1,"OCCUPATO!\n"); if (isflagset(4)){ log(1,"ATTIVAZIONE VOICEMAIL!\n"); revert_uri(); xlog("L_ERR","<$ruri>"); rewritehostport("192.168.9.97:5061"); append_branch(); t_relay(); exit; } exit; } }; } failure_route[2] { log(1,"FINE CHIAMATA 2 - SENZA FORWARD!\n"); revert_uri(); xlog("L_ERR","<$ruri>"); rewritehostport("192.168.9.97:5061"); append_branch(); t_relay(); exit; } failure_route[3] { log(1,"FINE CHIAMATA 3 - SENZA FORWARD!\n"); if ( t_check_status("603") || t_check_status("486") ){ t_relay(); exit; } exit; } ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

18 years, 3 months

[Serusers] avp_radius

by samuel

Hi all!! I'm testing stable ser version (0.9.6) with RADIUS authentication. I have properly setup the auth and next step was to get some extra attributes from the auth server using the avp_radius module. The thing is that I am a newby in the RADIUS world and don't know exactly what do I have to setup in both dictionaries. The main problem is that the attribute I have to read from ser script is a cisco-h323-whatever and don't even know if it is possible to read it as a caller_cisco-h323-whatever ...I wish I could use another attribute but I can not modify it :( I have set in both dictionaries cisco vendor attribute, the cisco-h323 attribute, SIP_AVP, SIP-caller-AVPs, and SIP_Callee-AVPs. I guess there something missing to glue the AVP with the cisco staff (if possible). ser log just says: avp_load_user: Failure DEBUG:avpops:check_avp: no avp found to check Any feedback is highly appreciated (even links to documentation,howtos....) Thank you, Sam.

18 years, 3 months

Re: [Users] Beer at wednesday 22 at 18:00 ?

by Helge Waastad

Hi, yet again a brilliant initiative, Atle. As much as I would like to join this time, I just cant. I hope we can have another go again soon. br hw -- Helge Waastad Senior Konsulent Systemavdelingen Smartnet

18 years, 3 months

[Users] how to use OpenSER

by Brijesh

hi all i am explaining my problem i have installed openser on one of machine. i have assigned SIP_DOMAIN in that machine as "test.com". then i also added users to the openser using *openserctl add user1(a)test.com test user1(a)test.com* command. so that user is also added to the myslql database. till now it is ok. now i am using Minisip as a sip end application. in that i have given account name as user1(a)test.com, sip uri as sip:user1@test.com, sip proxy address as machine's ip address on which proxy is installed and network port as 5060. but it is not working.. can someone help me out.. thanx in advance. Brijesh

18 years, 3 months

[Serusers] Help! DNS?

by ryan nalupa

hello! i was just wondering, since i'm still new here, where will i config this? i'm using debian sarge amd64 and ser 0.9.4. DNS SVR Resource Records It is important that your SIP clients can connect to your server for purposes of registration and call control. You might even want to have a redundant server to handle calls if your primary server is unavailable. These requirements can be meet by using DNS SVR Resource Records, available in BIND 8.X and up releases. The format for a SVR RR is this: _service._protocol SVR Priority Weight Port hostname In this case we want to establish an entry for our primary SIP server, gateway.mydomain.com, that will listen on UDP port 5060. The entry will look like this: _sip._udp SRV 0 0 5060 gateway.mydomain.com Placement of the new resource record is important. Here is a sample zone file: ; zone 'mydomain.com' last serial 1998071308 $ORIGIN com. mydomain 86400 IN SOA gateway.mydomain.com. postmaster.mydomain.com. ( 1998111908 ; Serial 36000 ; Refresh 900 ; Retry 36000 ; Expire 28800 ); Minimum IN NS gateway.mydomain.com. IN NS ns3.backupdomain.com. IN MX 1 gateway.mydomain.com. IN A 192.168.0.1 ;If we place the SRV record above the next line it fails to load $ORIGIN fitawi.com. _sip._udp SRV 0 0 5060 gateway.mydomain.com. gateway IN A 192.168.0.1 www IN CNAME gateway.mydomain.com. After reloading your zone file you can verify that the entry is working by using dig. dig -t SRV _sip._udp.mydomain.com The results should look something like this: ; <<>> DiG 9.1.0 <<>> -t SRV _sip._udp.mydomain.com ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 32654 ;; flags: qr aa rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 2, ADDITIONAL: 1 ;; QUESTION SECTION: ;_sip._udp.mydomain.com. IN SRV ;; ANSWER SECTION: _sip._udp.mydomain.com. 86400 IN SRV 0 0 5060 gateway.mydomain.com. ;; AUTHORITY SECTION: mydomain.com. 86400 IN NS ns3.elsewhere.com. mydomain..com 86400 IN NS gateway. mydomain.com. ;; ADDITIONAL SECTION: gateway. mydomain.com. 86400 IN A 192.168.0.150 ;; Query time: 6 msec ;; SERVER: 192.168.0.150#53(192.168.0.150) ;; WHEN: Tue Dec 3 08:34:17 2002 ;; MSG SIZE rcvd: 132 thanks in advance! ryan --------------------------------- Do you Yahoo!? Try the new Yahoo! Philippines Front Page!

18 years, 3 months

[Serusers] loose_route bug?!

by Ramin Nikaeen

Hi Everyone, I am using SER version Server: Sip EXpress router (0.9.2 (i386/linux)). I am initiating a call from a caller sip UA to a callee sip UA and I am receiving the ACK response to the INVITE from the callee on SER. Before loose_route processing it looks like this: ACK sip:2224440133@10.1.10.65:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.15.52:5060;rport;branch=z9hG4bK83e54f590401584628d473053d1ef8a8 From: <sip:2224440132@xxx.yyy.zzz:5060>;tag=1cac2190 To: <sip:2224440131@xxx.yyy.zzz:5060>;tag=d323038ba423a329i0 Call-ID: 66cb6ba85191e6025ebde2c552177442(a)172.16.15.52 CSeq: 1 ACK Contact: <sip:2224440132@172.16.15.52:5060> Max-Forwards: 70 Route: <sip:10.1.10.65;ftag=1cac2190;lr>,<sip:10.1.10.65;ftag=1cac2190;lr> Content-Length: 0 After loose_route processing and relaying, the ACK, instead of being routed to the callee (2224440133) User Agent, it is received back at SER looking like this: ACK sip:10.1.10.65;ftag=1cac2190;lr SIP/2.0 Record-Route: <sip:10.1.10.65;ftag=1cac2190;lr=on> Via: SIP/2.0/UDP 10.1.10.65;branch=0 Via: SIP/2.0/UDP 172.16.15.52:5060;rport=5060;branch=z9hG4bK83e54f590401584628d473053d1ef 8a8 From: <sip:2224440132@xxx.yyy.zzz:5060>;tag=1cac2190 To: <sip:2224440131@xxx.yyy.zzz:5060>;tag=d323038ba423a329i0 Call-ID: 66cb6ba85191e6025ebde2c552177442(a)172.16.15.52 CSeq: 1 ACK Contact: <sip:2224440132@172.16.15.52:5060> Max-Forwards: 16 Route: Content-Length: 0 Why is SER routing this ACK request incorrectly?! Is the syntax of the original ACK generated by the Callee's UA incorrect?! I would appreciate any help as I can't figure it whether there is something wrong with ACK's syntax or there is a bug in SER's loose_route processing?! Thanks a lot ramin

18 years, 3 months

RE: [Users] next_gw(): No ruri_user AVP

by Raymond Chen

Hi Bogdan, I solved problem by forcing avp_write("$ruri/username", "i:1402"). Thanks Ray -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Raymond Chen Sent: Thursday, February 16, 2006 12:27 AM To: 'Bogdan-Andrei Iancu' Cc: Users(a)openser.org Subject: RE: [Users] next_gw(): No ruri_user AVP Hi Bogdan, Both, after load_gw and before next_gw avps are visible. But not after next_gw. There is no where in the script delete any avp. Ray -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Wednesday, February 15, 2006 8:18 AM To: Raymond Chen Cc: daniel(a)voice-system.ro; Users(a)openser.org Subject: Re: [Users] next_gw(): No ruri_user AVP Ray, I think there is a problem in the script - if you see the gw_uri avps, it's quite impossible not to see them also before and after the next_gw() call in request route. Disregarding the ruri_avp, the gw_uri avps should be visible. are you sure you do not delete all the avps in your script. Try placing avp_print before and after each lcr function call. regards, bogdan Raymond Chen wrote: >Hi bogdan, > >I put avp_print after next_gw , no avp output. > > >1(1334) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.138:5060> > 1(1334) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.139:5060> > 1(1334) does_uri_exit(): User in request uri does not exist > 1(1334) is_user_in(): User is in group 'local' > 1(1334) db_flags=3, flags=12 > 1(1334) DEBUG:avpops:load_avps: loaded avps = 1 > 1(1334) parse_headers: flags=ffffffffffffffff > 1(1334) DEBUG:avpops:pushto_avps: 1 avps were processed >1(1334) next_gw(): No ruri_user AVP > > > > > > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Wednesday, February 15, 2006 4:39 AM >To: Raymond Chen >Cc: daniel(a)voice-system.ro; Users(a)openser.org >Subject: Re: [Users] next_gw(): No ruri_user AVP > >Ray, > >the ruri_avp is added by next_gw after its usage from the REQUEST route. >You may check this by placing an avp_print after you did next_gw() in >request route (after calling route 3, for example). Check if there is >any avp with ID 1402. > >regards, >bogdan > >Raymond Chen wrote: > > > >>Hi bogdan, >> >>We called Load_gw and next-gw from request route. We have no problem with >>next_gw if the if (avp_pushto("$ruri", "s:fwdnoanswer")) happens in the >> >> >main > > >>route. But when it does in the failure_route the next_gw can't find the >>ruri_user avp. >> >>Raymond >> >> >> >>Route { >> >> .......... >> >> Route(3); >> >> ........... >> >>} >> >>failure_route[1] { >> >> if (t_check_status("(480)|(408)")) { >> if (avp_pushto("$ruri", "s:fwdnoanswer")) { >> avp_delete("s:fwdnoanswer"); >> route(3); >> }; >> }; >> >>} >> >>Route[3] { >> >> if (!load_gws()) { >> sl_send_reply("500", "Server Internal Error - Cannot load >>gateways"); >> return; >> }; >> >> ............... >> >> Route(5); >> >>} >> >>Route[5] { >> >> if (!next_gw()) { >> rewriteuri("sip:userbusy@211.102.91.134:443"); >> t_relay(); >> return; >> }; >> >> .............. >> >>} >> >> >> >>-----Original Message----- >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >>Sent: Wednesday, February 15, 2006 2:25 AM >>To: Raymond Chen >>Cc: daniel(a)voice-system.ro; Users(a)openser.org >>Subject: Re: [Users] next_gw(): No ruri_user AVP >> >>Hi Ray, >> >>do you call load_gws() from failure route? if so, not that this is not >>supported. >>For LCR to work properly, you need to call load_gws() and next_gw() from >>the request route and later next_gw() from failure routes. >> >>regards, >>bogdan >> >>Raymond Chen wrote: >> >> >> >> >> >>>Hi Bogdan, >>> >>>Here is the debug >>> >>>1(2584) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.138:5060> >>>1(2584) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.139:5060> >>>1(2584) DEBUG:avpops:print_avp: p=0xf4f167c8, flags=2 >>>1(2584) DEBUG: id=<1400> >>>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.139:5060> >>>1(2584) DEBUG:avpops:print_avp: p=0xf4f16790, flags=2 >>>1(2584) DEBUG: id=<1400> >>>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.138:5060> >>>1(2584) does_uri_exit(): User in request uri does not exist >>>1(2584) is_user_in(): User is in group 'local' >>>1(2584) db_flags=3, flags=12 >>>1(2584) DEBUG:avpops:print_avp: p=0xf4f167f8, flags=B >>>1(2584) DEBUG: id=<1400> >>>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.139:5060> >>>1(2584) DEBUG:avpops:print_avp: p=0xf4f16790, flags=2 >>>1(2584) DEBUG: id=<1400> >>>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.xxx:5060> >>>1(2584) next_gw(): No ruri_user AVP >>> >>> >>> >>>Raymond >>> >>> >>>-----Original Message----- >>>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >>>Sent: Tuesday, February 14, 2006 10:55 AM >>>To: Raymond Chen >>>Cc: daniel(a)voice-system.ro; Users(a)openser.org >>>Subject: Re: [Users] next_gw(): No ruri_user AVP >>> >>>Hi Ray, >>> >>>use avp_print() after the load_gw() to see what avps were loaded and >>>again just before next_gw() to see the available avp. >>>this will help to see if it's a problem at the load or search part. >>> >>>regards, >>>bogdan >>> >>> _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

18 years, 3 months

[Serusers] Errors...

by ryan nalupa

hello! i'm following the ser howto that i got at iptel.org and i'm having errors when following these... 3.2. Adding an admin for your realm Now that we have a working database and ser is configured to use it, we need to add some users and at least one of them should have administrator privileges. The administrator role becomes important if you want to use a web management tool such as serweb. Basic account manipulation can be performed with the serctl script, located in /usr/sbin. To add a user use these commands serctl add JoeUser qwerty joe(a)mydomain.com The system notify for "Type MySQL Password", the default password is "heslo" To make JoeUser an administrator, we need to logon to MySQL and modify the database. mysql> connect ser; mysql> update subscriber set perms=?admin? where USER_ID=?JoeUser?; Query OK, 1 row affected (0.00 sec) Rows matched: 1 Changed: 1 Warnings: 0 mysql> select * from subscriber; | 4cefa7a4d3c8c2dbf6328520bd873a19 | JoeUser | qwerty | | | | joe(a)mydomain.com | 2002-12-02 19:20:41 | 2002-12-02 20:29:46 | 80e0f273b2067d40277b49ff842bb9e3 | o | | | c79a8f8f08596baa84bb02c88884426d | mydomain.com | f322c94b8b2fbe557d43ab3ac9e05b3a | admin | 1 | America/Los_Angeles | The third from last field shows that Joe has been assigned admin privileges. At this point Joe can logon to our server, but since he is the only user, there is not much he can do. We can now add additional users using the serctl script, or now is a good time to look at installing serweb, which will allow users to subscribe to our service. sql gives me an error saying check the manual that came with my mysql dist. and says an error from ?admin? where USER_ID=?JoeUser?. can anyone help me here..... :( ryan --------------------------------- Do you Yahoo!? Try the new Yahoo! Philippines Front Page!

18 years, 3 months

Re: [Serusers] Problem with time of call invite in Mysql

by mkumar＠mantragroup.com

Hi, Thanks for the reply. I can chage my TimeZone but many Users will be having many timezones and if calls of each and every user records in Mysql DB according to his timezone then how can we have a common timezone for all atleast for getting time records in a particular period for billing. Manoj. > Quoting Andrey Kouprianov <andrey.kouprianov(a)gmail.com>: > >> Change your timezone >> >> On 2/20/06, mkumar(a)mantragroup.com <mkumar(a)mantragroup.com> wrote: >>> Hi All, >>> >>> I have SER and mysql DB in USA and I am calling from India. For all >>> the calls I >>> am dialling from India the time of call invite is getting recorded >>> 5.5 hours >>> ahead the local time in USA. Suppose if time in USA is 6:30 and if >>> I call from >>> India, in mysql DB it is getting stored that I called at 12:00. So >>> for billing >>> if I calculate calls between NOW() and NOW()-30 days, all the calls >>> in this 5.5 >>> hours are missing. >>> Please tell me where I have to change my configuration. I have ser >>> -0.9.4 and >>> Mysql 4.0 on Linux. >>> >>> Thanks for your help and time, >>> Manoj. >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >> > > >

18 years, 3 months

[Serusers] modifying SDP of 180

by Ever Zalazar

Hi, I have a question. I need to modify the SDP of code 180 when my clients use my mediaproxy. How is the process between a SER and the media proxy? Best REgards Ever

18 years, 3 months

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sr-users February 2006