sr-users April 2005

sr-users@lists.kamailio.org

250 participants
697 discussions

[Serusers] SER & Mediaproxy. Sipura vs x-lite (again)

by Cameron Beattie

An earlier post discussed issues I had calling between Sipura and x-lite. After receiving a lot of help that issue was resolved ... sort of. For those interested, calls between two x-lite UAs through an incorrectly configured mediaproxy had two-way audio but calls between x-lite and sipura did not. I now get one-way audio between the Sipura and x-lite when the call goes over the internet and back into the same NAT, but get no audio when the calls goes out and back into another NAT. I am using: ser 0.9.1 mediaproxy 1.3.0 ser.cfg from onsip.org I have posted the relevant Invites from 4 scenarios below and would appreciate any insight: 1. Sipura to x-Lite on same NAT (one-way audio) 2. Sipura to x-Lite on different NAT (no audio) 3. x-Lite to x-Lite on same NAT (audio OK) 4. x-Lite to x-Lite on different NAT (audio OK) The problem may be that the c record is the local IP address for the Sipura with the Sipura calls whereas with the x-lite calls it is the public IP address. I've checked the settings on the sipura and the NAT settings = no (I think this means leave NAT to mediaproxy) whilst the Via settings = yes. *Scenario 1* The following invite is on a call from Sipura (beattiek) to x-Lite (beattiec) on the same NAT. Audio is heard in one direction (from x-lite to Sipura). Sipura -> Nat 1 -> Proxy -> Nat 1 -> x-lite U 147.202.44.XXX:5060 -> 60.234.199.XXX:1028 INVITE sip:beattiea@60.234.199.XXX:5060 SIP/2.0..Record-Route: <sip:alias1@147.202.44.XXX:5060;nat=yes;ftag=1df09ec95856c84do0;lr=on> ..Via: SIP/2.0/UDP 147.202.44.XXX;branch=z9hG4bK6799.73035531.0..Via: SIP/2.0/UDP 192.168.0.11:5060;received=60.234.199.XXX;branch=z9hG4b K-d4727cc1;rport=5060..From: Cameron's sipura 2000 <sip:beattiek@beta.mydomain.co.nz>;tag=1df09ec95856c84do0..To: <sip:alias1@beta. mydomain.co.nz>..Call-ID: 91508d79-f1c0295d@192.168.0.11..CSeq: 102 INVITE..Max-Forwards: 16..Contact: Cameron's sipura 2000 <sip:beatt iek@60.234.199.XXX:5060>..Expires: 240..User-Agent: Sipura/SPA2000-2.0.13(g)..Content-Length: 424..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: x-sipura..Content-Type: application/sdp....v=0..o=- 8343043 8343043 IN IP4 192.168.0.11..s=-..c=IN IP 4 192.168.0.11..t=0 0..m=audio 16454 RTP/AVP 0 2 4 8 18 96 97 98 100 101..a=rtpmap:0 PCMU/8000..a=rtpmap:2 G726-32/8000..a=rtpmap:4 G723/ 8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729a/8000..a=rtpmap:96 G726-40/8000..a=rtpmap:97 G726-24/8000..a=rtpmap:98 G726-16/8000..a=rtpma p:100 NSE/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. *Scenario 2* The following invite is on a call from Sipura (beattiek) to x-Lite (pearcej) on a separate NAT. No audio is heard in either direction. Sipura -> Nat 1 -> Proxy -> Nat 2 -> x-lite U 147.202.44.XXX:5060 -> 219.88.100.XX:5060 INVITE sip:pearcej@219.88.100.XX:5060 SIP/2.0..Record-Route: <sip:alias2@147.202.44.XXX:5060;nat=yes;ftag=3c35fa19eb73610fo0;lr=on>..Via : SIP/2.0/UDP 147.202.44.XXX;branch=z9hG4bK810a.ea51b996.0..Via: SIP/2.0/UDP 192.168.0.11:5060;received=60.234.199.XXX;branch=z9hG4bK-ca5 e9437;rport=5060..From: Cameron's sipura 2000 <sip:beattiek@beta.mydomain.co.nz>;tag=3c35fa19eb73610fo0..To: <sip:alias2@beta.mydomain .co.nz>..Call-ID: 92d1ee61-22aab967@192.168.0.11..CSeq: 102 INVITE..Max-Forwards: 16..Contact: Cameron's sipura 2000 <sip:beattiek@60.2 34.199.XXX:5060>..Expires: 240..User-Agent: Sipura/SPA2000-2.0.13(g)..Content-Length: 424..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: x-sipura..Content-Type: application/sdp....v=0..o=- 7997909 7997909 IN IP4 192.168.0.11..s=-..c=IN IP4 192.16 8.0.11..t=0 0..m=audio 16442 RTP/AVP 0 2 4 8 18 96 97 98 100 101..a=rtpmap:0 PCMU/8000..a=rtpmap:2 G726-32/8000..a=rtpmap:4 G723/8000..a= rtpmap:8 PCMA/8000..a=rtpmap:18 G729a/8000..a=rtpmap:96 G726-40/8000..a=rtpmap:97 G726-24/8000..a=rtpmap:98 G726-16/8000..a=rtpmap:100 NS E/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. *Scenario 3* The following invite is on a call from x-Lite (beattiec) to x-Lite (beattiea) on the same NAT. x-lite-> Nat 1 -> Proxy -> Nat 1 -> x-lite U 147.202.44.XXX:5060 -> 60.234.199.XXX:1028 INVITE sip:beattiea@60.234.199.XXX:5060 SIP/2.0..Record-Route: <sip:beattiea@147.202.44.XXX:5060;nat=yes;ftag=251677925;lr=on>..Via: SIP/ 2.0/UDP 147.202.44.XXX;branch=z9hG4bKd72b.189eef55.0..Via: SIP/2.0/UDP 60.234.199.XXX:5060;rport=1027;branch=z9hG4bK52D2A416DB7849C4BC88D 0A3B2B4B1D1..From: Cameron's laptop <sip:beattiec@beta.mydomain.co.nz>;tag=251677925..To: <sip:beattiea@beta.mydomain.co.nz>..Contact : <sip:beattiec@60.234.199.XXX:1027>..Call-ID: 0E31A4AE-E92F-46D1-8ACB-7CB7E2FF1641@192.168.0.15..CSeq: 18590 INVITE..Max-Forwards: 16..C ontent-Type: application/sdp..User-Agent: X-Lite release 1103m..Content-Length: 303....v=0..o=beattiec 169182050 169182181 IN IP4 60.234. 199.XXX..s=X-Lite..c=IN IP4 60.234.199.XXX..t=0 0..m=audio 8000 RTP/AVP 0 8 3 98 97 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rt pmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15.. *Scenario 4* The following invite is on a call from x-Lite (beattiec) to x-Lite (pearcej) on a separate NAT. x-lite -> Nat 1 -> Proxy -> Nat 2 -> x-lite U 147.202.44.XXX:5060 -> 219.88.100.XX:5060 INVITE sip:pearcej@219.88.100.XX:5060 SIP/2.0..Record-Route: <sip:alias2@147.202.44.XXX:5060;nat=yes;ftag=3274571863;lr=on>..Via: SIP/2. 0/UDP 147.202.44.XXX;branch=z9hG4bKa447.eadaf8e4.0..Via: SIP/2.0/UDP 60.234.199.XXX:5060;rport=1027;branch=z9hG4bKED04E2DC4F0846A5B616A13 47747365E..From: Cameron's laptop <sip:beattiec@beta.mydomain.co.nz>;tag=3274571863..To: <sip:alias2@beta.mydomain.co.nz>..Contact: <sip:beattiec@60.234.199.XXX:1027>..Call-ID: 60F12C88-B4A4-44D9-A207-FF7F6A97C3C8@192.168.0.15..CSeq: 7003 INVITE..Max-Forwards: 16..Cont ent-Type: application/sdp..User-Agent: X-Lite release 1103m..Content-Length: 303....v=0..o=beattiec 165963913 165964043 IN IP4 60.234.199 .XXX..s=X-Lite..c=IN IP4 60.234.199.XXX..t=0 0..m=audio 8000 RTP/AVP 0 8 3 98 97 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpma p:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15.. Many thanks for any help received. Regards Cameron Beattie

19 years, 5 months

[Serusers] Window Messenger and SER

by Hiep Trung Tran

Hi, After install SER on Solaris successfully then I use Window Messenger on 2 workstation to connect to the SER but neither sees the other is online. Could you please let me know how to fix this problem? Do I need to install mysql before using Window Messenger? Thanks a lot, Hiep

19 years, 5 months

[Serusers] need help please, v0.9.0 problems

by Noel Paul Evangelista

hi guys, i need help please. very new on SER. i just installed v0.9.0(ser-0.9.0_src.tar) using "make prefix=/usr/local all" then "make prefix=/usr/local install". looks like everything was installed. ser.cfg is /usr/local/etc/ser and executables are in /usr/local/sbin. the modules are in their correct directory as shown in the ser.cfg. my problem is when i do a "serctl start" i get this problem: [root@SIPserver sbin]# serctl start Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed secondly, if i do a "serctl moni" i get this message: [root@SIPserver sbin]# serctl moni Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config i am sure that the default ser.cfg has the line fifo=/tmp/ser_fifo. thirdly, if i do a "ser start" i get this message but it looks like ser is not still running: [root@SIPserver sbin]# ser start Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 146.82.219.99 [146.82.219.99]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 146.82.219.99 [146.82.219.99]:5060 Aliases: tcp: SIPserver.islandtel.com.ph:5060 tcp: localhost:5060 tcp: localhost.localdomain:5060 tcp: SIPserver:5060 udp: SIPserver.islandtel.com.ph:5060 udp: localhost:5060 udp: localhost.localdomain:5060 udp: SIPserver:5060 [root@SIPserver sbin]# other info about the server are the following: [root@SIPserver sbin]# ser -V version: ser 0.9.0 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ main.c compiled on 14:40:36 Apr 19 2005 with gcc 3.2 [root@SIPserver sbin]# uname -a Linux SIPserver 2.4.20-8 #1 Thu Mar 13 17:54:28 EST 2003 i686 i686 i386 GNU/Linux [root@SIPserver sbin]# any help regarding my problem will be highly appreciated. best regards, Noel __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

19 years, 5 months

[Serusers] Permissions module, database configuration.

by Rod Bacon

In the latest CVS, and in everything else I have read, I can see no mention of the permissions module using SQL as it's data source. I have added the options to my ser.cfg (as per other people's configuration), but SER insists on attempting to read the permissions.allow and permissions.deny files on startup. Is this normal? -- ========================================== Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 ==========================================

19 years, 5 months

RE: [Serusers] Serious Problems with SER - Not responding anythin g.

by Ricardo Martinez

Thanks Jiri. I remember have been checked the typical services when the problem was happening. My Radius (i'm not use mysql) for example was working without problems... How can i log information of SER failures? Regards, Ricardo.- > -----Mensaje original----- > De: Jiri Kuthan [mailto:jiri@iptel.org] > Enviado el: Miércoles, 20 de Abril de 2005 13:12 > Para: Ricardo Martinez; 'serusers(a)lists.iptel.org' > Asunto: Re: [Serusers] Serious Problems with SER - Not responding > anything. > > > a typical reason is SER is blocked waiting for some external service. > It may be possibly unavailable mysql, DNS, or RADIUS... > > -jiri > > At 07:09 PM 4/20/2005, Ricardo Martinez wrote: > >Hello list. > > I'm experiencing problems with my SER. As i > mentioned in a past > >post > (http://lists.iptel.org/pipermail/serusers/2005-April/018324.html), > >from time to time (it is getting pretty often) my SER not > answering any > >request, when a REGISTER reach the server it is not > challenged, this happens > >too with the INVITE's messages. About two hours ago this > happened again. > >I asked in my last post how to LOG the internal errors or > messages from SER > >to see if i can get some idea about what's going on, but i > have no answer > >about it. > >This is the infornmation i was able to capture when this was > happening : > > > >the serctl moni command shows a lot of "failures". > > > >[cycle #: 11; if constant make sure server lives and fifo is on] > >Server: Sip EXpress router (0.8.14-3 (i386/linux)) > >Now: Wed Apr 20 12:56:29 2005 > >Up Since: Thu Apr 14 21:32:18 2005 > >Up time: 487451 [sec] > > > >Transaction Statistics > >Current: 6 (2612 waiting) Total: 38194 (0 local) > >Replied localy: 80752 > >Completion status 6xx: 4643, 5xx: 768, 4xx: 10664, 3xx: > 0,2xx: 22552 > > > >Stateless Server Statistics > >200: 5975004 202: 0 2xx: 0 > >300: 0 301: 0 302: 0 3xx: 0 > >400: 0 401: 205980 403: 0 404: 0 407: 22918 408: 0 483: 0 > 4xx: 19 > >500: 0 5xx: 0 > >6xx: 0 > >xxx: 0 > >failures: 24180 > > > >UsrLoc Stats > >Domain Registered Expired > >'location' 219 4552 > >-------------------------------------------------------------- > >[cycle #: 5; if constant make sure server lives and fifo is on] > >Server: Sip EXpress router (0.8.14-3 (i386/linux)) > >Now: Wed Apr 20 12:56:47 2005 > >Up Since: Thu Apr 14 21:32:18 2005 > >Up time: 487469 [sec] > > > >Transaction Statistics > >Current: 4 (2619 waiting) Total: 38219 (0 local) > >Replied localy: 80762 > >Completion status 6xx: 4643, 5xx: 770, 4xx: 10667, 3xx: > 0,2xx: 22552 > > > >Stateless Server Statistics > >200: 5975030 202: 0 2xx: 0 > >300: 0 301: 0 302: 0 3xx: 0 > >400: 0 401: 205995 403: 0 404: 0 407: 22922 408: 0 483: 0 > 4xx: 19 > >500: 0 5xx: 0 > >6xx: 0 > >xxx: 0 > >failures: 24380 > > > >UsrLoc Stats > >Domain Registered Expired > >'location' 220 4552 > >----------------------------------------------------------------- > >[cycle #: 2; if constant make sure server lives and fifo is on] > >Server: Sip EXpress router (0.8.14-3 (i386/linux)) > >Now: Wed Apr 20 12:56:52 2005 > >Up Since: Thu Apr 14 21:32:18 2005 > >Up time: 487474 [sec] > > > >Transaction Statistics > >Current: 4 (2620 waiting) Total: 38224 (0 local) > >Replied localy: 80762 > >Completion status 6xx: 4643, 5xx: 770, 4xx: 10667, 3xx: > 0,2xx: 22552 > > > >Stateless Server Statistics > >200: 5975034 202: 0 2xx: 0 > >300: 0 301: 0 302: 0 3xx: 0 > >400: 0 401: 206000 403: 0 404: 0 407: 22923 408: 0 483: 0 > 4xx: 19 > >500: 0 5xx: 0 > >6xx: 0 > >xxx: 0 > >failures: 24445 > > > >UsrLoc Stats > >Domain Registered Expired > >'location' 215 4557 > > > >What is happening to cause the "failures" counter grows that much?. > > > >The free command shows this : > >[root@root]# free > > total used free shared > buffers cached > >Mem: 1030888 626464 404424 0 > 86692 369848 > >-/+ buffers/cache: 169924 860964 > >Swap: 2040244 4864 2035380 > > > >The top command shows not much use of CPU. The proccess > were running (the > >SER and mediaproxy) > >Despite of all this, the SER was not answering anything. > >Could this be a bug? or maybe a problem with my ser.cfg file?. I'm > >attaching my ser.cfg file too to see if someone can tell me > if there is > >something wrong about it. > >I really hope that someone could help me, this is getting > really complicated > >every time it happens. > > > >Thanks > >Ricardo Martinez.- > > > >************************************************** > >SER.CFG > >************************************************** > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++++++++++++++++++++++++ > > > > > > > > > ># ----------- global configuration parameters > ------------------------ > > > >debug=0 # 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 > > > >listen=ip.of.my.ser > > > >alias=sip.mydomain.com > >alias=sip2.mydomain.net > >alias=ip.of.my.ser > > > > > >check_via=no # (cmd. line: -v) > >dns=no # (cmd. line: -r) > >rev_dns=no # (cmd. line: -R) > >port=5060 > >#children=4 > >fifo="/tmp/ser_fifo" > >fifo_mode=0666 > > > > > ># ------------------ module loading > ---------------------------------- > > > ># Uncomment this if you want to use SQL database > >#loadmodule "/usr/local//lib/ser/modules/mysql.so" > > > >loadmodule "/usr/local//lib/ser/modules/sl.so" > >loadmodule "/usr/local//lib/ser/modules/tm.so" > >loadmodule "/usr/local//lib/ser/modules/rr.so" > >loadmodule "/usr/local//lib/ser/modules/maxfwd.so" > >loadmodule "/usr/local//lib/ser/modules/usrloc.so" > >loadmodule "/usr/local//lib/ser/modules/registrar.so" > >loadmodule "/usr/local//lib/ser/modules/textops.so" > >loadmodule "/usr/local//lib/ser/modules/exec.so" > > > ># ++++++++++ Para trabajar con NAT > > > >loadmodule "/usr/local//lib/ser/modules/mediaproxy.so" > >loadmodule "/usr/local//lib/ser/modules/domain.so" > >loadmodule "/usr/local//lib/ser/modules/dbtext.so" > >loadmodule "/usr/local//lib/ser/modules/nathelper.so" > > > ># Uncomment this if you want digest authentication > ># mysql.so must be loaded ! > >loadmodule "/usr/local/lib/ser/modules/auth.so" > > > ># ++++++++++ Modulos de Accounting y Autorizacion via Radius > > > >loadmodule "/usr/local/lib/ser/modules/auth_radius.so" > >loadmodule "/usr/local/lib/ser/modules/group_radius.so" > >loadmodule "/usr/local/lib/ser/modules/acc.so" > > > ># ----------------- setting module-specific parameters > --------------- > ># -- rr params -- > ># add value to ;lr param to make some broken UAs happy > >modparam("rr", "enable_full_lr", 1) > > > ># +++++ module authorization > >modparam("auth_radius","radius_config","/usr/local/etc/radius > client/radiuscl > >ient.conf") > >modparam("auth_radius","service_type",15) > > > > > ># +++++ module accounting > >modparam("acc","radius_config","/usr/local/etc/radiusclient/r > adiusclient.con > >f") > >modparam("acc","log_level",1) > >modparam("acc", "service_type", 15) > >modparam("acc", "radius_flag", 1) > >modparam("acc", "radius_missed_flag", 3) > >modparam("acc", "failed_transactions", 1) > >modparam("acc", "report_ack", 0) > > > > > ># +++++ group radius > >modparam("group_radius","radius_config","/usr/local/etc/radiu > sclient/radiusc > >lient.conf") > >modparam("group_radius", "use_domain", 1) > > > > > ># +++++ module mediaproxy > >modparam("mediaproxy", "natping_interval", 60) > >modparam("mediaproxy", "mediaproxy_socket", > "/var/run/mediaproxy.sock") > >#modparam("mediaproxy", "sip_asymmetrics", > >"/usr/local/etc/ser/sip-asymmetrics-clients") > >#modparam("mediaproxy", "rtp_asymmetrics", > >"/usr/local/etc/ser/rtp-asymmetrics-clients") > >modparam("registrar", "nat_flag", 5) > > > ># +++++ module domain > >modparam("domain", "db_url", "/usr/local/etc/ser/domaintables") > >modparam("domain", "domain_table", "domain") > >modparam("domain", "domain_col", "domain") > > > ># +++++ module tm > >modparam("tm", "fr_timer", 15) > >modparam("tm", "fr_inv_timer", 22) > >modparam("tm", "wt_timer", 5) > > > ># +++++ module registrar (Con estos parametro haces que el > location lookup > >priorize solo los mas recientes contacts) > >modparam("registrar", "append_branches", 0) > >modparam("registrar", "desc_time_order", 1) > > > > > ># ------------------------- request routing logic > ------------------- > > > ># main routing logic > > > >route { > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > ++++++++++++++ > > # Llamada a Rutina Externa "translate" para cambio de numero > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > ++++++++++++++ > > > > if (method=="INVITE" || method=="CANCEL") { > > exec_dset("/usr/local/etc/ser/translate"); > > }; > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > ++++++++++++++ > > # 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"); > > break; > > }; > > > > if ( msg:len > max_len ) { > > sl_send_reply("513", "Message too big"); > > break; > > }; > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > > # Marcamos todos los mensajes para que pasen por > nuestro servidor > > # a traves del record route. > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > > > > record_route(); > > > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > > # Permitimos el uso del campo Route en el ruteo de la llamada > > # > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > > > > if (method=="BYE" || method=="CANCEL") { > > log(1, "NAT: BYE o CANCEL recibido --> > terminando la sesion > >de media\n"); > > end_media_session(); > > setflag(1); > > }; > > > > if (loose_route()) { > > t_relay(); > > break; > > }; > > > > > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+ > > # Manejo de los mensajes NOTIFY para Keep Alive de NAT > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+ > > > > if ((method=="NOTIFY") && search("^Event: keep-alive")) { > > sl_send_reply("200", "OK"); > > break; > > }; > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+ > > # Marcamos las llamadas para accounting > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+ > > > > setflag(1); > > > > > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+ > > # Test para clientes NATeados > > # > > # 1 : Chequea si el cliente tiene IP privada en el > campo Contact. > > # 2 : Chequea si el cliente se contacto con una IP > distinta a la del > >campo VIA. > > # 3 : Chequea si el cliente tiene IP privada en el > ultimo campo VIA. > > # > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+ > > > > > > if (client_nat_test("3")) { > > log(1, "NAT: Requerimiento de IP privada --> > fixed contact > >(en rutina principal)\n"); > > setflag(5); > > force_rport(); > > > > if (method=="REGISTER") { > > fix_nated_contact(); > > } else { > > fix_contact(); > > }; > > append_hf("P-hint: fixed NAT contact for > request\r\n"); > > }; > > > > > > if (uri==myself) { > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++ > > # Chequeo de REGISTRO para equipos del dominio 1 > > # Dominio Voiss : sip.mydomain.com > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++ > > > > if (method=="REGISTER" && > uri=~"^sip:.*sip.mydomain.com") { > > # Challenge/Response > > if ( > !radius_www_authorize("sip.mydomain.com")) { > > > www_challenge("sip.mydomain.com", "1"); > > break; > > }; > > > > # Mandamos un mensaje de Log si es > que un cliente > >NATEADO se registro. > > if (isflagset(5)) { > > log(1, "NAT: Cliente > NAT'eado registrado > >(dentro del chequeo de REGISTER)\n"); > > }; > > > > save("location"); > > break; > > }; # Fin chequeo de Registro para Dominio : > sip.mydomain.com > > > > > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++ > > # Chequeo de REGISTRO para equipos del dominio 2 > > # Dominio Mayoristas : sip2.mydomain.net > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++ > > > > if (method=="REGISTER" && > uri=~"^sip:.*sip2.mydomain.net") { > > # Challenge/Response > > if ( > !radius_www_authorize("sip2.mydomain.net")) { > > > www_challenge("sip2.mydomain.net", "1"); > > break; > > }; > > > > # Mandamos un mensaje de Log si es > que un cliente > >NATEADO se registro. > > if (isflagset(5)) { > > log(1, "NAT: Cliente > NAT'eado registrado > >(dentro del chequeo de REGISTER)\n"); > > }; > > > > save("location"); > > break; > > }; # Fin chequeo de Registro para Dominio : > >sip2.mydomain.net > > > > > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++ > > # CHEQUEO DEL INVITE PARA TODAS LAS LLAMADAS > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++ > > > > if (method=="INVITE" && src_ip!=xx.xx.xx.xx) { > > if ( !radius_proxy_authorize("")) { > > proxy_challenge("", "1"); > > log (1, "LOG: Llamada no > AUTORIZADA\n"); > > break; > > }; > > }; > > > > > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++++ > > # Llamadas hacia dominio H.323 > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++++ > > > > if ( (uri=~"^sip:717.*@.*") | > (uri=~"^sip:707.*@.*") | > >(uri=~"^sip:777.*@.*") | (uri=~"^sip:333.*@.*") | (uri=~"^sip > >:0.*@.*") ){ > > log(1,"FORWARDING : Llamada hacia plataforma > >H.323\n"); > > route(1); > > break; > > }; > > > > > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++ > > # Llamadas SIP locales > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++ > > > > if (!lookup("location")) { #Usuarios que no estan en > >"location" database. > > route(4); > > break; > > }; > > > > > > setflag(3); > > > > > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++ > > # Manejo para llamadas NAT'das > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++ > > > > if (method=="INVITE") { > > log(1, "ROUTING: Reply processing > (online user) > >enabled to handle NAT'd resonses\n"); > > t_on_reply("1"); > > if ( (isflagset(5)) && > (method=="INVITE" || > >method=="ACK") ) { > > log(1, "NAT: Invite > received --> > >enabling media proxy (else del lookup 2do if)\n"); > > use_media_proxy(); > > append_hf("P-hint: > request forced to > >media proxy\r\n"); > > }; > > }; > > > > # Do it ? (Ricardo Martinez) > > }; > > > > append_hf("P-hint: USRLOC\r\n"); > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+++++ > > # Efectuamos la llamada > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+++++ > > > > if (!t_relay()) { > > sl_reply_error(); > > break; > > }; > > > >} /* end of initial routing logic */ > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++++++++++++ > ># COMIENZO DE LA SECCION DE RUTAS > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++++++++++++ > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++++++++++++ > ># Route 1 : Llamadas con destino plataforma > >H.323#+++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+++++++++++++++++++++++ > >route[1] > >{ > > # send it out now; use stateful forwarding as it > works reliably > > # even for UDP2TCP > > > > # Esta NATEADO? > > if (isflagset(5)) { > > log(1, "NAT: At least one participant NAT'd > --> enable reply > >processing (hacia GW)\n"); > > t_on_reply("1"); > > if (method=="INVITE") { > > log(1, "NAT: Invite received --> > enabling proxied > >media session (hacia GW)\n"); > > setflag(1); > > use_media_proxy(); > > }; > > }; > > > > rewritehostport("mygw.mydomain.com:5060"); > > append_hf("P-hint: GATEWAY\r\n"); > > > > if (!t_relay()) { > > sl_reply_error(); > > break; > > }; > > > >} > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+++++++++++++++++ > ># Route 4 : Llamadas a usuarios Off-line > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+++++++++++++++++ > > > >route[4] { > > > > if (!t_newtran()) { > > sl_reply_error(); > > }; > > > > if (!t_reply("404", "Not Found IT!")) { > > sl_reply_error(); > > }; > > break; > >} > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+++++++++++++++++ > ># On Reply Route 1 : Equipos NATeados!! > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+++++++++++++++++ > > > >onreply_route[1] { > > if ((isflagset(5) || client_nat_test("3")) && > >(status=~"(180)|(183)|2[0-9][0-9]")) { > > if (!search("^Content-Length:\ 0")) { > > use_media_proxy(); > > }; > > }; > > > > if (client_nat_test("1")) { > > fix_nated_contact(); > > }; > > > > > >} > > > > > > > > > >Ricardo Martinez.- > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > -- > Jiri Kuthan http://iptel.org/~jiri/ >

19 years, 5 months

[Serusers] Precall Speech with SER and SEMS

by matt morris

Hello List, I would like to implement a precall speech feature with SER with possibly SEMS. Basically for a certain of group of our user we would like them to hear an audio before establishing their calls with the callee. However, I don't have much clues on how to do this. I have tried the Asterisk B2BUA by Mr. Mike Tkachuk (which worked) but I don't want the media stream to be handled by the B2BUA. Unfortunately, I do not know how to configure a signalling-only B2BUA with the playback - seems whenever I used the playback function in Asterisk and subsequent call media will be have to be handled by Asterisk. That is why I want to explore the possibility of another approach with SEMS. Would it be possible for the call to be first forwarded to the SER+SEMS box to play the announcement AND THEN forward the call to the actual called number? Please shed some light on how I could make this happen. Thank you very much. _________________________________________________________________ Scan and help eliminate destructive viruses from your inbound and outbound e-mail and attachments. http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&DI=1034&SU=ht… Start enjoying all the benefits of MSN® Premium right now and get the first two months FREE*.

19 years, 5 months

[Serusers] resolv problem ser 0.8.14

by Bruno Oliveira

hi, when beginning the server to be appears the message in log: localhost ser: WARNING: could not rev. resolve 192.168.1.100 anyone knows because? -- []s, Bruno "Niggas" Oliveira Belo Horizonte - MG Msn: n1gg4s(a)gmail.com Icq: 176314647 "Todo o nosso descontentamento por aquilo que nos falta procede da nossa falta de gratidão por aquilo que temos."

19 years, 5 months

[Serusers] Re: Strange error on registration - Still a Problem in a Certain Scenario

by reticent

I was looking back through the mailing list archives and i found this email regarding failed inserts into the location table of a database due to duplicate entries and there is still a situation where this problem will occur. For the full original Thread, this is where is starts http://lists.iptel.org/pipermail/serusers/2004-April/007151.html http://lists.iptel.org/pipermail/serusers/2004-April/thread.html#7173 Basically the issue is that the database has an entry for a user that SER does not have in memory so when a registration comes in SER uses an INSERT statement instead of an UPDATE which fails because a key (username, contact, domain) already exists for the user in the database. The situation where it still occurs is when the database goes away and during the hiatus a registration expires, SER wipes the registration from memory and fails on the database delete. SER will recover from this situation only when it is stopped and started, this complicates load balanced situations where it is expected that SER can recover from certain failures (database failures in this case) without any administrive intervention

19 years, 5 months

[Serusers] mediaproxy timeout problem

by Paul van Schagen

Hi I am testing mediaproxy with ser. I have a config with a Grandstream handytone 486 as a UA and a CISCO 5350 as a gateway. My sip server is SER 0.9.0 . I find that when setting up a phone call and disconnecting the Grandstream ( power off or disconnecting LAN cable ) the mediaproxy doesn't timeout and disconnect the call. This is what I have done ... Grandstream 486 -------- SER 0.9.0 with mediaproxy ------------CISCO 5350 I setup a call from a Grandstream Handytone 486 to the PSTN. and see this on the proxy server. root# ./sessions.py Caller Via Called Status Duration Codec Type Traffic ------------------------------------------------------------------------ --------------------------------- ------------ 212.4.174.186:18400 - 217.194.194.181:35008 - 83.157.4.99:56364 active 0'34" G723 Audio 35.81k/53.59k/18.71k Total traffic: 4.09kbps/352bps/4.44kbps (in1/in2/out) Session count: 1 Now I disconnect the grandstream from the LAN cable in the middle of the conversation to test the timeout function of the mediaproxy. now I get this. root # ./sessions.py Caller Via Called Status Duration Codec Type Traffic ------------------------------------------------------------------------ --------------------------------------------- 212.4.174.186:16652 - 217.194.194.181:35010 - 83.157.4.99:56364 active 0'36" G723 Audio 62.92k/74.53k/12.77k Total traffic: 16.53kbps/0bps/15.47kbps (in1/in2/out) Session count: 1 The disconnect is reflected in in2 (0bps ), the connection stays active because I keep on receiving UDP packets ( background noise maybe or somebody talking ) from the PSTN gateway. Is it correct to assume there is no end to end communication anymore as soon as one of both RTP streams is in 0bps ? or could it be that in that case there is silence from one end ? If it is correct to assume the connection to the UA has been lost as soon as there is no UDP data coming from it , how can I modify mediaproxy in a way it will disconnect on either one of the inputs being in 0bps? any help highly appreciated . Paul van Schagen

19 years, 5 months

[Serusers] SEMS conference module (no sound)

by Oscar Patricio

Hi! I'm having trouble setting up SEMS conference calls! When i call the conference number everything goes ok. As soon as the second user joins, we enter in conference, but there is no sound!! I see udp trafic to and from both clients, but i hear no sound! As anyone ever had a similar problem? Thanx! Óscar

19 years, 5 months

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sr-users April 2005