- sr-users - kamailio.org

[Users] How to change Contact
by Taras Bendik 10 Nov '05

10 Nov '05

Hi I need something similar so fix_nated_contact but with parametars so I can set up my own values for host:port. Can I do this in OpenSer? Can avp_pushto do this? Regards, Taras

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[Users] Compile openser with tls error ?
by Nguyen Duc Phi 10 Nov '05

10 Nov '05

Hi all, I am compile openser-1.0.0-tls on Red Hat Linux 3.2.3-52 openssl-0.9.7a-33.17 openssl-devel-0.9.7a-33.17 It's show error : gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"openser"' -DVERSION='"1.0.0-tls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DUSE_TLS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/ssl/include -c main.c -o main.o In file included from /usr/include/openssl/ssl.h:179, from tls/tls_init.h:27, from main.c:112: /usr/include/openssl/kssl.h:72:18: krb5.h: No such file or directory In file included from /usr/include/openssl/ssl.h:179, from tls/tls_init.h:27, from main.c:112: /usr/include/openssl/kssl.h:134: syntax error before "krb5_enctype" /usr/include/openssl/kssl.h:136: syntax error before '*' token /usr/include/openssl/kssl.h:137: syntax error before '}' token /usr/include/openssl/kssl.h:149: syntax error before "kssl_ctx_setstring" /usr/include/openssl/kssl.h:149: syntax error before '*' token /usr/include/openssl/kssl.h:150: syntax error before '*' token /usr/include/openssl/kssl.h:151: syntax error before '*' token /usr/include/openssl/kssl.h:151: syntax error before '*' token /usr/include/openssl/kssl.h:152: syntax error before '*' token /usr/include/openssl/kssl.h:153: syntax error before "kssl_ctx_setprinc" /usr/include/openssl/kssl.h:153: syntax error before '*' token /usr/include/openssl/kssl.h:155: syntax error before "kssl_cget_tkt" /usr/include/openssl/kssl.h:155: syntax error before '*' token /usr/include/openssl/kssl.h:157: syntax error before "kssl_sget_tkt" /usr/include/openssl/kssl.h:157: syntax error before '*' token /usr/include/openssl/kssl.h:159: syntax error before "kssl_ctx_setkey" /usr/include/openssl/kssl.h:159: syntax error before '*' token /usr/include/openssl/kssl.h:161: syntax error before "context" /usr/include/openssl/kssl.h:162: syntax error before "kssl_build_principal_2" /usr/include/openssl/kssl.h:162: syntax error before "context" /usr/include/openssl/kssl.h:165: syntax error before "kssl_validate_times" /usr/include/openssl/kssl.h:165: syntax error before "atime" /usr/include/openssl/kssl.h:167: syntax error before "kssl_check_authent" /usr/include/openssl/kssl.h:167: syntax error before '*' token /usr/include/openssl/kssl.h:169: syntax error before "enctype" In file included from tls/tls_init.h:27, from main.c:112: /usr/include/openssl/ssl.h:909: syntax error before "KSSL_CTX" /usr/include/openssl/ssl.h:931: syntax error before '}' token make: *** [main.o] Error 1 Could you help me fix this problem ? Thank in advance! Best Regard.

2 1

[Users] Problem with OpenSER
by Krishna Chava 10 Nov '05

10 Nov '05

Hi, I am new to OpenSER or SER and am having problems registering users to OpenSER. i had the Default user Admin registered to my softphone but am unable to register my other users.. i am using mysql with openSER.. i saw the difference between the admin user that i had default in Subsriber table and the user 'abc' that serctl added when i sed the command serctl add 'abc' 'abc' 'a(a)a.com' <mailto:'a(a)a.com'> and i see the user added to the subscriber table with 1 difference.... The user abc is not having any value set for the confirmation field. Any help would be appreciated.. thanks krishna

2 1

[Serusers] "Click to call" with SER ?
by Christophe Irles 10 Nov '05

10 Nov '05

Hi, Is it possible to create a "click to call" button (from a contact web list for example) with SER since I know that the current user as as SIP device already registered ? Which module must be used ? Is there any examples / URL on this subject ? Thanks, Christophe

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[Users] accuracy of retransmission timer
by Klaus Darilion 10 Nov '05

10 Nov '05

Hi! I often see retransmission of of requests sent by openser after ~100ms. Is this normal? regards klaus

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Re: [Users] How to add Route headers?
by Bogdan-Andrei Iancu 10 Nov '05

10 Nov '05

Hi Gustavo, if you have fixed content, it's even simpler. Use the textops module to add the headers. See the append_hf() function: http://www.openser.org/docs/modules/1.1.x/textops.html#AEN189 regards, bogdan Gustavo García Bernardo wrote: >Hi Bogdan, thx for your response > >I need to put two fixed route headers, like these: > >Route: user(a)as.domain.com >Route: user(a)cscf.domain.com > >Can I do that with AVPops module? I think is not suitable for this task. > >BR, > >G. > >-----Mensaje original----- >De: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Enviado el: jueves, 10 de noviembre de 2005 0:28 >Para: Gustavo García Bernardo >CC: users(a)openser.org >Asunto: Re: [Users] How to add Route headers? > >Hi Gustavo, > >you can generate the body of the Route header(s) into AVP(s) be loading >them from database of by building them via avpops functions (like >avp_printf()). >then just set the AVP(s) into Route header(s) via the avp_pushto() function: >http://www.voice-system.ro/docs/avpops/ar01s06.html#avp_pushto > >mostly depends of the content you want to push into the headers. > >regards, >bogdan > >Gustavo García Bernardo wrote: > > > >>Hi, >> >>Is posible to add more route headers to a request before relaying it? >> >>I’m trying to implement the IMS mechanism before relaying a request to >>an application server, which consists in adding two route headers. >> >>Thx in advance. >> >>G. >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> > > > >

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[Users] How can i run openser ?
by messa＠innsof.com 10 Nov '05

10 Nov '05

Hi all I compile and install Openser-1.0.1-tls.tar.gz successfully. But in the documentation, i can't find how i can run and use it? Can you tell me? regards! Serge MESSA

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[Serusers] RV: [Serdev] Parse Referred-By
by Ing. Francisco Talavera 10 Nov '05

10 Nov '05

Hi all, I need to parse the header field Referred-By, can any one tell me. How can I do this? For example, which files I must to edit because the SER-0.9.0 can NOT parse this header field. Thanks in advance, Ing. Francisco Talavera Dpto. Investigación y Desarrollo Conexión S.A.

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[Users] compile error
by Nguyen Duc Phi 10 Nov '05

10 Nov '05

Hi all, I am compile openser-1.0.0-tls on Red Hat Linux 3.2.3-52 openssl-0.9.7a-33.17 openssl-devel-0.9.7a-33.17 It's show error : gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"openser"' -DVERSION='"1.0.0-tls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DUSE_TLS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/ssl/include -c main.c -o main.o In file included from /usr/include/openssl/ssl.h:179, from tls/tls_init.h:27, from main.c:112: /usr/include/openssl/kssl.h:72:18: krb5.h: No such file or directory In file included from /usr/include/openssl/ssl.h:179, from tls/tls_init.h:27, from main.c:112: /usr/include/openssl/kssl.h:134: syntax error before "krb5_enctype" /usr/include/openssl/kssl.h:136: syntax error before '*' token /usr/include/openssl/kssl.h:137: syntax error before '}' token /usr/include/openssl/kssl.h:149: syntax error before "kssl_ctx_setstring" /usr/include/openssl/kssl.h:149: syntax error before '*' token /usr/include/openssl/kssl.h:150: syntax error before '*' token /usr/include/openssl/kssl.h:151: syntax error before '*' token /usr/include/openssl/kssl.h:151: syntax error before '*' token /usr/include/openssl/kssl.h:152: syntax error before '*' token /usr/include/openssl/kssl.h:153: syntax error before "kssl_ctx_setprinc" /usr/include/openssl/kssl.h:153: syntax error before '*' token /usr/include/openssl/kssl.h:155: syntax error before "kssl_cget_tkt" /usr/include/openssl/kssl.h:155: syntax error before '*' token /usr/include/openssl/kssl.h:157: syntax error before "kssl_sget_tkt" /usr/include/openssl/kssl.h:157: syntax error before '*' token /usr/include/openssl/kssl.h:159: syntax error before "kssl_ctx_setkey" /usr/include/openssl/kssl.h:159: syntax error before '*' token /usr/include/openssl/kssl.h:161: syntax error before "context" /usr/include/openssl/kssl.h:162: syntax error before "kssl_build_principal_2" /usr/include/openssl/kssl.h:162: syntax error before "context" /usr/include/openssl/kssl.h:165: syntax error before "kssl_validate_times" /usr/include/openssl/kssl.h:165: syntax error before "atime" /usr/include/openssl/kssl.h:167: syntax error before "kssl_check_authent" /usr/include/openssl/kssl.h:167: syntax error before '*' token /usr/include/openssl/kssl.h:169: syntax error before "enctype" In file included from tls/tls_init.h:27, from main.c:112: /usr/include/openssl/ssl.h:909: syntax error before "KSSL_CTX" /usr/include/openssl/ssl.h:931: syntax error before '}' token make: *** [main.o] Error 1 Could you help me fix this problem ? Thank in advance! Best Regard.

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Re: [Serusers] problem with rtpproxy
by Greger V. Teigre 10 Nov '05

10 Nov '05

Please copy the list. If you use a vanilla onsip.org config, it's pretty thoroughly tested by now, so I would be tempted to say that there is something wrong with the client or the setup of the client. g-) ----- Original Message ----- From: rajesh volluru To: Greger V. Teigre Sent: Wednesday, November 09, 2005 2:04 AM Subject: Re: [Serusers] problem with rtpproxy hi Thanks for your reply. I did configure ser.cfg as said at onsip.org for rtp proxy. But, i still get below exception when I am behind NAT. net.java.sip.communicator.media.MediaException: Incorrect SDP data! at net.java.sip.communicator.media.MediaManager.openMediaStreams(MediaManager.java:307) at net.java.sip.communicator.SipCommunicator.callStateChanged(SipCommunicator.java:758) at net.java.sip.communicator.sip.Call.fireCallStatusChangedEvent(Call.java:250) at net.java.sip.communicator.sip.Call.setState(Call.java:156) at net.java.sip.communicator.sip.CallProcessing.processAck(CallProcessing.java:566) at net.java.sip.communicator.sip.SipManager.processRequest(SipManager.java:1532) at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:204) at gov.nist.javax.sip.EventScanner.run(EventScanner.java:352) at java.lang.Thread.run(Unknown Source) Caused by: javax.sdp.SdpParseException: Could not parse message at javax.sdp.SdpFactory.createSessionDescription(SdpFactory.java:114) at net.java.sip.communicator.media.MediaManager.openMediaStreams(MediaManager.java:302) ... 8 more Please let me know the possible reason for this error. When I start ser, it shows the message ' RTP proxy found. support for RTP enabled'. Thanks Rajesh "Greger V. Teigre" <greger(a)teigre.com> wrote: Are you using the configs from onsip.org? If not, you may find some help there ;-) g-) ----- Original Message ----- From: rajesh volluru To: Sip Users Sent: Tuesday, November 08, 2005 12:37 AM Subject: [Serusers] problem with rtpproxy hi I had installed ser with media and rtp proxy. Below is the output that i get when I start ser. However, iam not able receive any audio transmission when I connect through sip communicator when iam behind NAT. I am able to register with the server. 0(0) fixing /usr/local/lib/ser/modules/maxfwd.so mf_process_maxfwd_header 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so client_nat_test 0(0) fixing /usr/local/lib/ser/modules/textops.so search 0(0) module - fixing ^Record-Route: 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) fixing /usr/local/lib/ser/modules/nathelper.so fix_nated_sdp 0(0) fixing /usr/local/lib/ser/modules/rr.so record_route 0(0) fixing /usr/local/lib/ser/modules/rr.so loose_route 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/registrar.so lookup 0(0) get_connection(): Connection not found in the pool 0(0) preload_udomain():! Table is empty 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/auth_db.so www_authorize 0(0) get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/auth.so www_challenge 0(0) fixing /usr/local/lib/ser/modules/registrar.so save 0(0) get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/registrar.so lookup 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/textops.so search 0(0) module - fixing ^Route: 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so use_media_proxy 0(0) fixing /usr/local/lib/ser/modules/tm.so t_on_reply 0(0) fixing /usr/local/lib/ser/modules/tm.so t_relay 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_reply_error 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so use_media_proxy 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so client_nat_test 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) DEBUG: udp_init: trying SO_RCVBUF: 221184 0(0) DEBUG: setting SO_RCVBUF; set=221184,verify=221184 0(0) DEBUG: udp_init: trying SO_RCVBUF: 223232 0(0) DEBUG: setting SO_RCVBUF; set=223232,verify=221184 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) INFO: udp_init: SO_RCVBUF i! s finally 221184 0(0) DBG: open_uac_fifo: opening fifo... 0(0) DEBUG: FIFO created @ /tmp/ser_fifo 0(0) DEBUG: fifo /tmp/ser_fifo opened, mode=432 0(0) init_unixsock_socket: No unix domain socket will be opened 2(0) INFO: fifo process starting: 19946 2(0) DEBUG: init_mod_child (-2): tm 2(0) DEBUG: callid: '57b7f7f7-0(a)69.229.23.57' 2(0) DEBUG: init_mod_child (-2): usrloc 2(0) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): tm 0(19941) DEBUG: callid: '57b7f7f7-19941(a)69.229.23.57' 0(19941) DEBUG: init_mod_child (1): usrloc 0(19941) get_connection(): Connection not found in the pool 1(0) DEBUG: init_mod_child (-1): tm 1(19945) DEBUG: callid: '57b7f7f7-19945(a)69.229.23.57' 1(19945) DEBUG: init_mod_child (-1): usrloc 1(19945) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): auth_db 0(19941) get_connection(): Connection not found in the pool 1(19945) DEBUG: init_mod_child (-1): auth_db 1(19945) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): domain 0(19941) get_connection(): Connection found in the pool 0(19941) DEBUG: init_mod_child (1): nathelper 2(19946) DEBUG: init_mod_child (-2): auth_db 2(19946) get_connection(): Connection not found in the pool 1(19945) DEBUG: init_mod_child (-1): domain 1(19945) DEBUG: init_mod_child (-1): nathelper 0(19941) rtpp_test: RTP proxy found, support for it enabled 1(19945) rtpp_test: RTP proxy found, support for it enabled 2(19946) DEBUG: init_mod_child (-2): domain 2(19946) DEBUG: init_mod_child (-2): nathelper 2(19946) rtpp_test: RTP proxy found, support for it enabled 2(19946) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(19946) DEBUG: register_fifo_cmd: new command (print) registered 2(19946) DEBUG: register_fifo_cmd: new command (uptime) registered 2(19946) DEBUG: register_fifo_cmd: new command (version) registered 2(19946) DEBUG: register_fifo_cmd: new command (pwd) registered 2(19946) DEBUG: register_fifo_cmd: new command (arg) registered 2(19946) DEBUG: register_fifo_cmd: new command (which) registered 2(19946) DEBUG: register_fifo_cmd: new command (ps) registered 2(19946) DEBUG: register_fifo_cmd: new command (kill) registered 2(19946) WARNING: no fifo_db_url given - fifo DB commands disabled! Let me know if you find any configuration error in SER. Please let me if you need any more information. I am trying to configure SER from long time. Please help me. Thanks Rajesh -------------------------------------------------------------------------- Enjoy this Diwali with Y! India Click here -------------------------------------------------------------------------- Enjoy this Diwali with Y! India Click here -------------------------------------------------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ------------------------------------------------------------------------------ Enjoy this Diwali with Y! India Click here

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[Users] rport error
by Marco 10 Nov '05

10 Nov '05

Hi all. I've a problem with a call using mediaproxy. The call got established, but after a few seconds got disconnected and I obtain the following errors. The problem appears with several clients. Please can someone enlighten me on where I err? TIA Nov 8 23:43:20 fw /usr/sbin/openser[11625]: ERROR: parse_via: invalid port number <12797rport> Nov 8 23:43:20 fw /usr/sbin/openser[11625]: ERROR: parse_via on: <SIP/2.0/UDP 213.144.x.x:12797rport;branch=z9hG4bK510915090^M Route: <sip:213.215.x.x;ftag=274138964;lr=on>^M From: <sip:marco@domain>;tag=274138964^M To: <sip:000000@voip.domain>;tag=5E97045-71F^M Call-ID: 408923322(a)213.144.x.x^M CSeq: 21 ACK^M Contact: <sip:marco@213.144.x.x:12797^M Max-Forwards: 5^M User-Agent: Linphone-1.0.1/eXosip^M Content-Length: 0^M ^M > Nov 8 23:43:20 fw /usr/sbin/openser[11625]: ERROR: parse_via parse error, parsed so far:<SIP/2.0/UDP 213.144.80.48:12797rport;branch=z9hG4bK510915090^M > Nov 8 23:43:20 fw /usr/sbin/openser[11625]: ERROR: get_hdr_field: bad via Nov 8 23:43:20 fw /usr/sbin/openser[11625]: ERROR: bad header field Nov 8 23:43:20 fw /usr/sbin/openser[11625]: ERROR: parse_msg: message=<ACK sip:0000000@10.10.10.10:53429 SIP/2.0^M Via: SIP/2.0/UDP 213.144.x.x:12797rport;branch=z9hG4bK510915090^M Route: <sip:213.215.x.x;ftag=274138964;lr=on>^M From: <sip:marco@domain>;tag=274138964^M To: <sip:000000@voip.domain>;tag=5E97045-71F^M Call-ID: 408923322(a)213.144.x.x^M CSeq: 21 ACK^M Contact: <sip:marco@213.144.x.x:12797^M Max-Forwards: 5^M User-Agent: Linphone-1.0.1/eXosip^M Content-Length: 0^M ^M > -- In general, an implementation must be conservative in its sending behavior, and liberal in its receiving behavior. - Jon Postel (1943-1998)

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[Users] How to add Route headers?
by Gustavo García Bernardo 09 Nov '05

09 Nov '05

Hi, Is posible to add more route headers to a request before relaying it? Im trying to implement the IMS mechanism before relaying a request to an application server, which consists in adding two route headers. Thx in advance. G.

2 1

[Serusers] Calls from UAs behind NAT getting a 503 error...
by KM 09 Nov '05

09 Nov '05

Hi, calls from a UA that is on public ip go through the pstn gw and terminate ok. Calls from behind NAT get a 503 error from the GW when the INVITE message is sent from SER. The invite messages for both cases are the same except from the VIA header which for the latter one contains the private ip of the UA. Contact header is ok. Could that be the reason that im getting calls rejected? Is there a way to rewrite the VIA header? Im including the configuration file i have and some packet dumps of the calls.Thanks! debug=3 fork=no log_stderror=yes listen=213.207.***.*** # put your server IP address here port=5060 children=8 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:********@localhost/ser" #KM added module loadmodule "/usr/lib/ser/modules/exec.so" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/mediaproxy.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/avpops.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) #pstn part----------------------------------------- modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("213.207.***.***:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); #log(1,"route:NATED caller in loose route \n"); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- #Check if the target domain is one that the server is responsible for if (!is_uri_host_local()) { log(1,"Call type processing: Target domain is not in the domain pool \n"); #Check to see if the sender is in our own domain and allow messages from pstn,pgw if (is_from_local() || allow_trusted()) { log(1,"Call type processing: Caller is local or trusted \n"); route(4); route(1); } else { log(1,"Call type processing: 403 error \n"); sl_send_reply("403", "Forbidden"); }; break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- log (1,"------------route[3]:INVITE handler \n"); #if (allow_trusted()) {log(1,"------------route[3]:User is trusted\n");} ; #check if NATed caller if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { log (1,"------------route[3]:fixing NAT info \n"); setflag(6); force_rport(); fix_nated_contact(); }; if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { #log (1,"route[3]:User did not authorise! \n"); proxy_challenge("","1"); #log (1,"route[3]:Sent error 401"); break; } else if (!check_from()) { log (1,"route[3]: 403 error \n"); sl_send_reply("403", "Use From=ID"); break; }; log (1,"route[3]:Consuming credentials.....\n"); consume_credentials(); }; if (client_nat_test("3")) { setflag(7); force_rport(); }; #Strip prefix 00357 , 357 if (uri=~"^sip:00357[0-9]{8}@") { strip(5); } else if ( uri=~"^sip:357[0-9]{8}@") { strip(3); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (uri=~"^sip:00[1-9]*@") { # International PSTN 00* route(4); route(5); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{8}@") { # Domestic PSTN 2??????? route(4); route(5); break; }; sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- #Change the destination so it points to PGW rewritehost("213.207.***.***"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); route(4); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } ==================================================================================================== Packets for failed call behind NAT: ==================================================================================================== ΤΓ²΅?? ½qC~ΐ ζζώ-BΰLυEEΨDpΜΥΟ²ΥΟ³ΔΔΔ&INVITE sip:99389683@213.207.141.179 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bKfmqE9mAxO4AVb1rE Max-Forwards: 70 User-Agent: PA168S From: "22000002" <sip:22000002@213.207.141.179>;tag=vW5KTCrnEpud7yyE To: "99389683" <sip:99389683@213.207.141.179> Call-ID: Ra1xPN5naYXRZN67(a)192.168.1.6 Contact: <sip:22000002@192.168.1.6:5060> Proxy-Authorization: Digest username="22000002", realm="213.207.141.179", nonce="4371b5e37010200b061c47bdd2b9e553e3b7128a1", uri="sip:99389683@213.207.141.179", qop=auth, nc=00000001, cnonce="0a4f113b", response="805a849373492c421c9397d1621239b5" CSeq: 2 INVITE Content-Type: application/sdp Content-Length: 292 v=0 o=22000002 80268421 80523176 IN IP4 192.168.1.6 s=SIP CALL c=IN IP4 192.168.1.6 t=0 0 m=audio 11722 RTP/AVP 18 4 0 8 3 97 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 ½qCw zzLυEώ-BΰEl?@@pγΥΟ³ΥΟ²ΔΔXΙnSIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bKfmqE9mAxO4AVb1rE;rport=5060;received=213.207.141.178 From: "22000002" <sip:22000002@213.207.141.179>;tag=vW5KTCrnEpud7yyE To: "99389683" <sip:99389683@213.207.141.179> Call-ID: Ra1xPN5naYXRZN67(a)192.168.1.6 CSeq: 2 INVITE Server: Sip EXpress router (0.9.3 (i386/linux)) Content-Length: 0 Warning: 392 213.207.141.179:5060 "Noisy feedback tells: pid=8885 req_src_ip=213.207.141.178 req_src_port=5060 in_uri=sip:99389683@213.207.141.179 out_uri=sip:99389683@213.207.144.6 via_cnt==1" ½qCρ ΔΔΉR.ώ-BΰE?@@mDΥΟ³ΥΟΔΔΆΝ INVITE sip:99389683@213.207.144.6 SIP/2.0 Record-Route: <sip:99389683@213.207.141.179:5060;nat=yes;ftag=vW5KTCrnEpud7yyE;lr=on> Via: SIP/2.0/UDP 213.207.141.179;branch=z9hG4bK6f55.cba0c054.0 Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;received=213.207.141.178;branch=z9hG4bKfmqE9mAxO4AVb1rE Max-Forwards: 16 User-Agent: PA168S From: "22000002" <sip:22000002@213.207.141.179>;tag=vW5KTCrnEpud7yyE To: "99389683" <sip:99389683@213.207.141.179> Call-ID: Ra1xPN5naYXRZN67(a)192.168.1.6 Contact: <sip:22000002@213.207.141.178:5060> CSeq: 2 INVITE Content-Type: application/sdp Content-Length: 316 v=0 o=22000002 80268421 80523176 IN IP4 192.168.1.6 s=SIP CALL c=IN IP4 213.207.141.179213.207.141.179 t=0 0 m=audio 3506435064 RTP/AVP 18 4 0 8 3 97 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 ½qCy1 ΔΔΉR.ώ-BΰE?<@@mCΥΟ³ΥΟΔΔΆΝ INVITE sip:99389683@213.207.144.6 SIP/2.0 Record-Route: <sip:99389683@213.207.141.179:5060;nat=yes;ftag=vW5KTCrnEpud7yyE;lr=on> Via: SIP/2.0/UDP 213.207.141.179;branch=z9hG4bK6f55.cba0c054.0 Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;received=213.207.141.178;branch=z9hG4bKfmqE9mAxO4AVb1rE Max-Forwards: 16 User-Agent: PA168S From: "22000002" <sip:22000002@213.207.141.179>;tag=vW5KTCrnEpud7yyE To: "99389683" <sip:99389683@213.207.141.179> Call-ID: Ra1xPN5naYXRZN67(a)192.168.1.6 Contact: <sip:22000002@213.207.141.178:5060> CSeq: 2 INVITE Content-Type: application/sdp Content-Length: 316 v=0 o=22000002 80268421 80523176 IN IP4 192.168.1.6 s=SIP CALL c=IN IP4 213.207.141.179213.207.141.179 t=0 0 m=audio 3506435064 RTP/AVP 18 4 0 8 3 97 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 ½qCh ΄΄ώ-BΰΉR.E¦f~@ϋNoΥΟΥΟ³ΔΔ'^ SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.207.141.179;branch=z9hG4bK6f55.cba0c054.0 Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;received=213.207.141.178;branch=z9hG4bKfmqE9mAxO4AVb1rE From: "22000002" <sip:22000002@213.207.141.179>;tag=vW5KTCrnEpud7yyE To: "99389683" <sip:99389683@213.207.141.179>;tag=1367353431 Call-ID: Ra1xPN5naYXRZN67(a)192.168.1.6 CSeq: 2 INVITE Content-Length: 0 ½qCο΄΄ώ-BΰΉR.E¦f@ϋNnΥΟΥΟ³ΔΔ'^ SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.207.141.179;branch=z9hG4bK6f55.cba0c054.0 Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;received=213.207.141.178;branch=z9hG4bKfmqE9mAxO4AVb1rE From: "22000002" <sip:22000002@213.207.141.179>;tag=vW5KTCrnEpud7yyE To: "99389683" <sip:99389683@213.207.141.179>;tag=1367353431 Call-ID: Ra1xPN5naYXRZN67(a)192.168.1.6 CSeq: 2 INVITE Content-Length: 0 ½qCΎΊξξώ-BΰΉR.Eΰf?@ϋN3ΥΟΥΟ³ΔΔΜ"SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 213.207.141.179;branch=z9hG4bK6f55.cba0c054.0 Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;received=213.207.141.178;branch=z9hG4bKfmqE9mAxO4AVb1rE From: "22000002" <sip:22000002@213.207.141.179>;tag=vW5KTCrnEpud7yyE To: "99389683" <sip:99389683@213.207.141.179>;tag=1367353431 Call-ID: Ra1xPN5naYXRZN67(a)192.168.1.6 CSeq: 2 INVITE Contact: <sip:99389683@213.207.144.6:5060> Content-Length: 0 ½qCe»""ΉR.ώ-BΰE.@@osΥΟ³ΥΟΔΔqΚΫACK sip:99389683@213.207.144.6 SIP/2.0 Via: SIP/2.0/UDP 213.207.141.179;branch=z9hG4bK6f55.cba0c054.0 From: "22000002" <sip:22000002@213.207.141.179>;tag=vW5KTCrnEpud7yyE Call-ID: Ra1xPN5naYXRZN67(a)192.168.1.6 To: "99389683" <sip:99389683@213.207.141.179>;tag=1367353431 CSeq: 2 ACK User-Agent: Sip EXpress router(0.9.3 (i386/linux)) Content-Length: 0 ½qC>»RRLυEώ-BΰE @@q«ΥΟ³ΥΟ²ΔΔΘΆSIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;received=213.207.141.178;branch=z9hG4bKfmqE9mAxO4AVb1rE From: "22000002" <sip:22000002@213.207.141.179>;tag=vW5KTCrnEpud7yyE To: "99389683" <sip:99389683@213.207.141.179>;tag=1367353431 Call-ID: Ra1xPN5naYXRZN67(a)192.168.1.6 CSeq: 2 INVITE Contact: <sip:99389683@213.207.144.6:5060> Content-Length: 0 ½qCn??ώ-BΰLυEEEr ΥΟ²ΥΟ³ΔΔ"ACK sip:99389683@213.207.141.179 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bKfmqE9mAxO4AVb1rE Max-Forwards: 70 User-Agent: PA168S From: "22000002" <sip:22000002@213.207.141.179>;tag=vW5KTCrnEpud7yyE To: "99389683" <sip:99389683@213.207.141.179>;tag=1367353431 Call-ID: Ra1xPN5naYXRZN67(a)192.168.1.6 Contact: <sip:22000002@192.168.1.6:5060> Proxy-Authorization: Digest username="22000002", realm="213.207.141.179", nonce="4371be370510200b061c47bdd2b9e553e3b7128a1", uri="sip:99389683@213.207.141.179", qop=auth, nc=00000001, cnonce="0a4f113b", response="805a849373492c421c9397d1621239b5" CSeq: 2 ACK Content-Length: 0 ½qCΕ8 RRLυEώ-BΰE @@q?ΥΟ³ΥΟ²ΔΔΘΆSIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;received=213.207.141.178;branch=z9hG4bKfmqE9mAxO4AVb1rE From: "22000002" <sip:22000002@213.207.141.179>;tag=vW5KTCrnEpud7yyE To: "99389683" <sip:99389683@213.207.141.179>;tag=1367353431 Call-ID: Ra1xPN5naYXRZN67(a)192.168.1.6 CSeq: 2 INVITE Contact: <sip:99389683@213.207.144.6:5060> Content-Length: 0 ½qCγέ ??ώ-BΰLυEEFrΥΟ²ΥΟ³ΔΔ"ACK sip:99389683@213.207.141.179 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bKfmqE9mAxO4AVb1rE Max-Forwards: 70 User-Agent: PA168S From: "22000002" <sip:22000002@213.207.141.179>;tag=vW5KTCrnEpud7yyE To: "99389683" <sip:99389683@213.207.141.179>;tag=1367353431 Call-ID: Ra1xPN5naYXRZN67(a)192.168.1.6 Contact: <sip:22000002@192.168.1.6:5060> Proxy-Authorization: Digest username="22000002", realm="213.207.141.179", nonce="4371be370105200b061c47bdd2b9e553e3b7128a1", uri="sip:99389683@213.207.141.179", qop=auth, nc=00000001, cnonce="0a4f113b", response="805a849373492c421c9397d1621239b5" CSeq: 2 ACK Content-Length: 0 ½qC3G RRLυEώ-BΰE @@q©ΥΟ³ΥΟ²ΔΔΘΆSIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;received=213.207.141.178;branch=z9hG4bKfmqE9mAxO4AVb1rE From: "22000002" <sip:22000002@213.207.141.179>;tag=vW5KTCrnEpud7yyE To: "99389683" <sip:99389683@213.207.141.179>;tag=1367353431 Call-ID: Ra1xPN5naYXRZN67(a)192.168.1.6 CSeq: 2 INVITE Contact: <sip:99389683@213.207.144.6:5060> Content-Length: 0 ½qC.κ ??ώ-BΰLυEEGrΥΟ²ΥΟ³ΔΔ"ACK sip:99389683@213.207.141.179 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bKfmqE9mAxO4AVb1rE Max-Forwards: 70 User-Agent: PA168S From: "22000002" <sip:22000002@213.207.141.179>;tag=vW5KTCrnEpud7yyE To: "99389683" <sip:99389683@213.207.141.179>;tag=1367353431 Call-ID: Ra1xPN5naYXRZN67(a)192.168.1.6 Contact: <sip:22000002@192.168.1.6:5060> Proxy-Authorization: Digest username="22000002", realm="213.207.141.179", nonce="4371be370102005b061c47bdd2b9e553e3b7128a1", uri="sip:99389683@213.207.141.179", qop=auth, nc=00000001, cnonce="0a4f113b", response="805a849373492c421c9397d1621239b5" CSeq: 2 ACK Content-Length: 0 ½qCd RRLυEώ-BΰE @@q¨ΥΟ³ΥΟ²ΔΔΘΆSIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;received=213.207.141.178;branch=z9hG4bKfmqE9mAxO4AVb1rE From: "22000002" <sip:22000002@213.207.141.179>;tag=vW5KTCrnEpud7yyE To: "99389683" <sip:99389683@213.207.141.179>;tag=1367353431 Call-ID: Ra1xPN5naYXRZN67(a)192.168.1.6 CSeq: 2 INVITE Contact: <sip:99389683@213.207.144.6:5060> Content-Length: 0 ½qCA??ώ-BΰLυEEHrΥΟ²ΥΟ³ΔΔ"ACK sip:99389683@213.207.141.179 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bKfmqE9mAxO4AVb1rE Max-Forwards: 70 User-Agent: PA168S From: "22000002" <sip:22000002@213.207.141.179>;tag=vW5KTCrnEpud7yyE To: "99389683" <sip:99389683@213.207.141.179>;tag=1367353431 Call-ID: Ra1xPN5naYXRZN67(a)192.168.1.6 Contact: <sip:22000002@192.168.1.6:5060> Proxy-Authorization: Digest username="22000002", realm="213.207.141.179", nonce="4371be370102005b061c47bdd2b9e553e3b7128a1", uri="sip:99389683@213.207.141.179", qop=auth, nc=00000001, cnonce="0a4f113b", response="805a849373492c421c9397d1621239b5" CSeq: 2 ACK Content-Length: 0 ==================================================================================================== Packets for successful call from public IP: ==================================================================================================== ΤΓ²΅??½qCjlΏΏώ-BΰΪβΗEh±%*ϊΟ΅ΥΟ?ΥΟ³ΔΔίINVITE sip:22372673@213.207.141.179 SIP/2.0 Via: SIP/2.0/UDP 213.207.141.181:5060;branch=z9hG4bK-896914ce From: Kyriakos <sip:22000001@213.207.141.179>;tag=2f2e13766c08ee3ao0 To: <sip:22372673@213.207.141.179> Call-ID: fa843926-7b99844a(a)213.207.141.181 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="22000001",realm="213.207.141.179",nonce="4371be42f095d84938c2b65fb9b1d879749453b6e",uri="sip:22372673@213.207.141.179",algorithm=MD5,response="cf863bab65ffbaf6376746eb76cd127c",qop=auth,nc=00000001,cnonce="32d92d6b" Contact: Kyriakos <sip:22000001@213.207.141.181:5060> Expires: 240 User-Agent: Sipura/SPA841-3.1.1(a) Content-Length: 403 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 8038585 8038585 IN IP4 213.207.141.181 s=- c=IN IP4 213.207.141.181 t=0 0 m=audio 16460 RTP/AVP 0 2 4 8 18 96 97 98 101 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:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ½qCoOOΪβΗώ-BΰEA'@@qΥΟ³ΥΟ?ΔΔ-ΙFSIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 213.207.141.181:5060;branch=z9hG4bK-896914ce From: Kyriakos <sip:22000001@213.207.141.179>;tag=2f2e13766c08ee3ao0 To: <sip:22372673@213.207.141.179> Call-ID: fa843926-7b99844a(a)213.207.141.181 CSeq: 102 INVITE Server: Sip EXpress router (0.9.3 (i386/linux)) Content-Length: 0 Warning: 392 213.207.141.179:5060 "Noisy feedback tells: pid=8885 req_src_ip=213.207.141.181 req_src_port=5060 in_uri=sip:22372673@213.207.141.179 out_uri=sip:22372673@213.207.144.6 via_cnt==1" ½qC[oAAΉR.ώ-BΰE3"@@lΎΥΟ³ΥΟΔΔΝ?INVITE sip:22372673@213.207.144.6 SIP/2.0 Record-Route: <sip:213.207.141.179;ftag=2f2e13766c08ee3ao0;lr=on> Via: SIP/2.0/UDP 213.207.141.179;branch=z9hG4bKc56.673dbf01.0 Via: SIP/2.0/UDP 213.207.141.181:5060;branch=z9hG4bK-896914ce From: Kyriakos <sip:22000001@213.207.141.179>;tag=2f2e13766c08ee3ao0 To: <sip:22372673@213.207.141.179> Call-ID: fa843926-7b99844a(a)213.207.141.181 CSeq: 102 INVITE Max-Forwards: 16 Contact: Kyriakos <sip:22000001@213.207.141.181:5060> Expires: 240 User-Agent: Sipura/SPA841-3.1.1(a) Content-Length: 403 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 8038585 8038585 IN IP4 213.207.141.181 s=- c=IN IP4 213.207.141.181 t=0 0 m=audio 16460 RTP/AVP 0 2 4 8 18 96 97 98 101 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:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ½qCΏαώ-BΰΉR.Ezf@ϋNΥΟΥΟ³ΔΔf"QSIP/2.0 100 Trying Via: SIP/2.0/UDP 213.207.141.179;branch=z9hG4bKc56.673dbf01.0 Via: SIP/2.0/UDP 213.207.141.181:5060;branch=z9hG4bK-896914ce From: Kyriakos <sip:22000001@213.207.141.179>;tag=2f2e13766c08ee3ao0 To: <sip:22372673@213.207.141.179>;tag=1367353431 Call-ID: fa843926-7b99844a(a)213.207.141.181 CSeq: 102 INVITE Content-Length: 0

4 4

[Serusers] how to configure ser as stateful
by Santosh Kar 09 Nov '05

09 Nov '05

Hi all, I want to configure the SER as statefull proxy. What are the configuration required to do this.I really need this urgently. Thanks a lot in advance Regards, santosh

2 1

[Serusers] mediaproxy
by Richard C. Thompson 09 Nov '05

09 Nov '05

I hope someone can help. I'm was running mediaproxy just fine until the server was rebooted and now I get this message from ser Fixing NAT on reply with mediaproxy Nov 9 15:12:19 66 ser[5340]: error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy Nov 9 15:12:19 66 ser[5340]: ERROR: on_reply processing failed And I get one way audio Rick

1 0

[Serusers] Need help on Serweb installation and configuration: Parser error issue.
by Dhiman D Chowdhury 09 Nov '05

09 Nov '05

Hi all, I will appreciate it very much if you canhelp me configure the Serweb. I am having the following error message: Parse error: parse error, expecting `T_STRING' or `T_VARIABLE' or `'{'' or `'$'' in /var/www/html/serweb/config/config_paths.php on line 24 Warning: Cannot modify header information - headers already sent by (output started at /var/www/html/serweb/config/config_paths.php:24) in /var/www/html/serweb/html/index.php on line 13 This happens when I change the the following lines from root_path to anything: $config->img_src_path = $config->../."img/"; $config->js_src_path = $config->../."js/"; $config->style_src_path = $config->../."styles/"; $config->user_pages_path = $config->../."user/"; $config->admin_pages_path = $config->../."admin/"; $config->domains_path = $config->../."domains/"; If I do not change this, while browsing to the serweb, I get page not found. My serweb software is under /var/www/html/serweb and config.php or config_paths.php is under /var/www/html/serweb/config/ Please help. thanks Dhiman Deb Chowdhury

1 0

[Serusers] Adding route headers
by Gustavo García Bernardo 09 Nov '05

09 Nov '05

Hi, How could we add some top Route headers? Ive seen the append_hf function (textops module) but we need to put the new headers in top of the existing route headers instead that at the end. Im trying to implement with this the IMS mechanism for forwarding a request to an Application Server. It includes adding two route headers before forwarding (assuring that if Application Server proxies the request it will be received and forwarded to destination by the proxy/CSCF). Thnx in advance. G.

2 1

[Serusers] SER version
by Innocent Evil 09 Nov '05

09 Nov '05

Hello, What is the most recent stable version of SER. I saw there have 0.9.0 0.9.3 0.9.4 what each version means? Would anybody please write little about SER version? Thanks,

3 4

[Serusers] SIP client with IM and video
by Joao Pereira 09 Nov '05

09 Nov '05

Hello In my company people is using Lite and everybody is happy with it. But does someone knows a SIP client with more features (and free, because eyeBeam isnt) like Instant Messaging, video support, etc...? Thanks Joao Pereira

1 0

[Users] MediaProxy Questions
by Daryl Sanders 09 Nov '05

09 Nov '05

How does mediaproxy select the codec to be used when multiple are present? Does it try to use the lowest bandwidth codec first? Is there any way to manipulate this so if it finds 3-4 codecs available, it will use the one specified? - Daryl

3 4

[Serusers] Prepaid approach
by Ashutosh kumar 09 Nov '05

09 Nov '05

Hi, While trying to implement prepaid solution using SER, I decided to go be a intuitive approach which is as follows. -When the user registers, he is placed in a "voip" groups table of ser (or radius server database), i.e he can make only pc-to-pc calls. - Later , or otherwise, when the user registers, he is shifted to a "pstn" groups table of ser , i.e noew he can make only pc-to-pstn calls. -Henceforth, whenever the user logins and tries to make pstn call, the call is approved only if group_radius_is_user_in(username) succeds, else the call rejected by SER. -To restrict the user from making pstn calls when his credits are zero (or beyond a threshold) , a dedicated cron job is scripted to move users from "pstn" group to "voip" when their account_cerdit=0. Am I right in using this approach, or are there any foreseeable problems which I might be overlooking. Thanks. Regards, Ashutosh Kumar

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Re: [Serusers] FW: Call logs
by Steve Blair 09 Nov '05

09 Nov '05

Colin: You'll need the loadmodule "/usr/local/lib/ser/modules/acc.so" loaded but you have already done this. The log_flag is still used and must be set on INVITE and BYE messages. As a simple test place if (method=="INVITE"){setflag(-your log_flag-)}; at the beginning of route block 0. If everything else is working this should generate accounting records for INVITEs. These records appear in mysql>ser>acc table. The db_url parameter needs to point to your ser database instead of the flatstore file as mentioned earlier. Something like: modparam("acc", "db_url","mysql://ser:heslo@localhost/ser") should do it. Otherwise make sure you have modified the acc module Makefile, rebuilt the module and restart your server. That should do it. -Steve Colin Jordan wrote: >Hi Steve > >I am still having problems. How can I get it working on mysql? > >Kind regards, > >Colin > >-----Original Message----- >From: Steve Blair [mailto:blairs@isc.upenn.edu] >Sent: Thursday, October 27, 2005 2:09 PM >To: Colin Jordan >Subject: Re: [Serusers] FW: Call logs > >Colin: > > I use flatstore module for accounting data. MySQL tables filled up too >quickly >for my taste. Below is what I do: > >Load modules: > > loadmodule "/usr/local/lib/ser/modules/acc.so" # enables >accounting > loadmodule "/usr/local/lib/ser/modules/flatstore.so" # flatstore >acc module > >Define module parameters: > > > # ------------- accounting parameters > # > > modparam("acc", "db_url", "flatstore:/usr/local/var/ser") > modparam("acc", "db_flag", 1) > modparam("acc", "log_level", 1 ) > modparam("acc", "log_flag", 1 ) > modparam("acc", "log_fmt", "cdfsum" ) > modparam("acc", "early_media", 0 ) > modparam("acc", "report_ack", 0 ) # def=1 > modparam("acc", "report_cancels", 0 ) > modparam("acc", "failed_transactions", 0 ) > >Then use log_flag in your ser.cfg file to mark a particular transaction for >accounting. Be careful here because the placement of the setflag command >can cause extraneous (or missing) accounting records. > > if (method=="INVITE" || method=="BYE") > { > setflag(1); # account for only one instance of call > }; > >Then enable accounting by uncommenting the DEFS+=-DSQL_ACC in >sip_router/modules/acc Makefile and rebuild at least the acc module. >Rebuilding >the entire package will not hurt. > >If this all works (and you use flatstore) then in the db_url directory >you will >see multiple files of the format acc_6.log. Each ser child process >creates it's own >file to avoid delay due to record locking. To produce a billing report >you will >need to "munge" all of these files together. > >-Steve > >Colin Jordan wrote: > > > >>Hi there >> >>Will you kindly send me an example script of this? >> >>Kind regards, >> >>Colin >> >>-----Original Message----- >>From: Steve Blair <blairs(a)isc.upenn.edu> >>To: Colin Jordan <cjordan(a)econetwireless.com> >>Date: Wed, 26 Oct 2005 07:53:09 -0400 >>Subject: Re: [Serusers] FW: Call logs >> >> >> >> >> >>> You want to load and enable accounting then use the setflag command >>>to >>>set whichever flag is identified in the modparam("acc", "log_flag", X) >>>command >>>as your accounting flag. I happen to use flag #1 but the choice is >>>yours. >>>Also you want to uncomment DEFS+=-DSQL_ACC and rebuild the >>>accounting module if you want data to goto mysql. >>> >>> >>> >>>Colin Jordan wrote: >>> >>> >>> >>> >>> >>>>I can not find where the call logs are in the ser database. What can >>>> >>>> >>>> >>>> >>>I >>> >>> >>> >>> >>>>do to enable call information logging (time stamps, destination, exc) >>>> >>>> >>>> >>>> >>>? >>> >>> >>> >>> >>>>DISCLAIMER: The information in this message is confidential and is >>>> >>>> >>>> >>>> >>>legally >>> >>> >>> >>> >>>>privileged. It is intended solely for the addressee. Access to this >>>>message by >>>>anyone else is unauthorized.If receiving in error please accept our >>>>apologies >>>>and notify the sender immediately. You must also delete the original >>>>message from your machine. If you are not the intended recipient, any >>>> >>>> >>>> >>>> >>>use, >>> >>> >>> >>> >>>>disclosure, copying, distribution or action taken in reliance of it, >>>>is prohibited >>>>and maybe >>>>unlawful. >>>> >>>> >>>>---------------------------------------------------------------------- >>>> >>>> >>>> >>>> >>>-- >>> >>> >>> >>> >>>>_______________________________________________ >>>>Serusers mailing list >>>>serusers(a)lists.iptel.org >>>>http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>>> >>>> >>>> >>>> >>>-- >>> >>>ISC Network Engineering >>>The University of Pennsylvania >>>3401 Walnut Street, Suite 221A >>>Philadelphia, PA 19104 >>> >>> >>>voice: 215-573-8396 >>> >>> 215-746-8001 >>> >>>fax: 215-898-9348 >>> >>>sip:blairs@upenn.edu >>> >>> >>> >>> >>DISCLAIMER: The information in this message is confidential and is legally >>privileged. It is intended solely for the addressee. Access to this message >> >> >by > > >>anyone else is unauthorized.If receiving in error please accept our >> >> >apologies > > >>and notify the sender immediately. You must also delete the original >>message from your machine. If you are not the intended recipient, any use, >>disclosure, copying, distribution or action taken in reliance of it, is >> >> >prohibited > > >>and maybe >>unlawful. >> >> >> >> >> >> >> > >-- > >ISC Network Engineering >The University of Pennsylvania >3401 Walnut Street, Suite 221A >Philadelphia, PA 19104 > >voice: 215-573-8396 > > 215-746-8001 > >fax: 215-898-9348 > >sip:blairs@net.isc.upenn.edu >DISCLAIMER: The information in this message is confidential and is legally >privileged. It is intended solely for the addressee. Access to this message by >anyone else is unauthorized.If receiving in error please accept our apologies >and notify the sender immediately. You must also delete the original >message from your machine. If you are not the intended recipient, any use, >disclosure, copying, distribution or action taken in reliance of it, is prohibited >and maybe >unlawful. > > > > > -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@net.isc.upenn.edu

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[Serusers] Enable accounting for billing
by Colin Jordan 09 Nov '05

09 Nov '05

Hi there Can someone help me with what to do to enable accounting an SER. I need to generate logs so that I can use it for billing. I have mysql installed Best regards, Colin Jordan DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

2 1

[Serusers] rport
by han yibing 09 Nov '05

09 Nov '05

Hi all i have windows messenger to register with SER 0.9.3,but SER get the URI contain private IP address(8808@192.168.2.2:5060) so that 8808 can't as called party, i have rport config in ser.cfg and windows messenger have got rport and received information in 401 and 200 ok messages .ip phone have same problem. how to solve the problem? please help me ? thanks ___________________________________________________________ 雅虎免费G邮箱－No.1的防毒防垃圾超大邮箱 http://cn.mail.yahoo.com

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[Serusers] SER scaling issues
by Bruce Bauman 09 Nov '05

09 Nov '05

We are testing stress testing SER with location and subscriber databases containing 500K to 1 million entries through an artificially generated workload. With the stock SER configuration when we stop and restart SER and it runs out of memory almost immediately upon startup. It runs out of memory in convert_rows() at around row #2740 out of many (> 500000) rows. If we bump up PKG_MEM_POOL_SIZE from 1024*1024 to 100*1024*1024 ser starts up successfully. However, my concern is that the memory pool is allocated in each of the 20 or 30 children, sucking up a lot of resources. So, my question is this: why isn't this memory allocation truly dynamic rather than pre-allocated at startup? Or is there some better solution that I am missing completely? We also have similar issues with shared memory, and need to bump this allocation via the -m command line option to ser. FWIW, we are running SER 0.8.14 with a couple of local modifications. Any advice would be appreciated Thanks. Bruce Bauman Sr. Principal S/W Engineer WorldGate Communications, Inc. 3190 Tremont Avenue Trevose, PA 19053 Office: 215.354.5124 Cell: 215.768.8613

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[Serusers] Incorrect SDP data!
by rajesh volluru 08 Nov '05

08 Nov '05

hi I get following exception when I added code for forced rtp proxy is ser.cfg. I am using sip communicator as user agent. net.java.sip.communicator.media.MediaException: Incorrect SDP data! at net.java.sip.communicator.media.MediaManager.openMediaStreams(MediaManager.java:307) at net.java.sip.communicator.SipCommunicator.callStateChanged(SipCommunicator.java:758) at net.java.sip.communicator.sip.Call.fireCallStatusChangedEvent(Call.java:250) at net.java.sip.communicator.sip.Call.setState(Call.java:156) at net.java.sip.communicator.sip.CallProcessing.processAck(CallProcessing.java:566) at net.java.sip.communicator.sip.SipManager.processRequest(SipManager.java:1532) at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:204) at gov.nist.javax.sip.EventScanner.run(EventScanner.java:352) at java.lang.Thread.run(Unknown Source) Caused by: javax.sdp.SdpParseException: Could not parse message at javax.sdp.SdpFactory.createSessionDescription(SdpFactory.java:114) at net.java.sip.communicator.media.MediaManager.openMediaStreams(MediaManager.java:302) ... 8 more Please help me resolve this problem Thanks Rajesh --------------------------------- Enjoy this Diwali with Y! India Click here

1 0

[Users] TLS / SRTP implementation
by Christophe Irles 08 Nov '05

08 Nov '05

Hi, I'm a newbie in OpenSER usage and in SIP in general so perhaps it's very simple questions. I understand that TLS can be activated/disactivated from the configuration file but can not be used only according SIP number for example or call streams(internet), that's right ? What about SRTP ? In order to have SRTP working, I think I "just" need device phone/softphone with this feature. It's not dependant of the OpenSER configuration, is it ? Is it possible to force the use of SRTP when it's possible ? For this, perhaps, I need to configure "mediaproxy" or "rtpproxy" module or something else ( mediaproxy daemon for example) ? Thanks, Christophe.

2 1

[Serusers] memory configuration
by yserge 08 Nov '05

08 Nov '05

Hello, I am having some problems configuring memory for SER.. Here is what I have in my test instance: Server: Linux x64 Real Memory: 4GB Database: MySQL/NDB SER Shared memory: 256M Children: 16 Subscribers: 40000 In test environment: Registrations per sec: 120 Call per second: 120 When the SER process was started last week, it had only 4000 subscribers provisioned. Then another 36000 subscribers were provisioned. The SER was running fine. But when I decided to bounce SER, it would not restart. It seems from the log file that SER ran out of memory trying to load the LOCATION database into memory upon startup, failing around row #3500. LOCATION table had in excess of 10000 records at that point. The related piece of the logfile is below.. I wonder whether it is a matter of configuration or is it a bug.. The system definitely seem to have enough memory to store those few tens of thousands of records.. Even at 1K each, it amounts to nothing.. Especially given that at runtime it handles that amount of location info easily.. It's only a startup that seems to be a problem. I tried to play with giving it more or less of shared memory.. It does not seem to be relevant.. For now I just have to clean up the LOCATION table each time before restarting SER. Obviously, it is not a nice workaround.. Thanks, Sergei Mon Nov 7 12:29:59.281894 200: 0 (0):DEBUG:fixing /usr/local/ser/lib/ser/modules/registrar.so lookup Mon Nov 7 12:29:59.281964 200: 0(0):DEBUG:get_connection(): Connection not found in the pool Mon Nov 7 12:29:59.282329 200: 0(0):DEBUG:preload_udomain(): Table is empty Mon Nov 7 12:29:59.282342 200: 0(0):DEBUG:release_connection(): Removing connection from the pool Mon Nov 7 12:29:59.282360 200: 0 (0):DEBUG:fixing /usr/local/ser/lib/ser/modules/registrar.so save Mon Nov 7 12:29:59.282414 200: 0(0):DEBUG:get_connection(): Connection not found in the pool Mon Nov 7 12:29:59.357157 200: 0(0):WARNING:convert_row(): No memory left Mon Nov 7 12:29:59.357189 200: 0(0):WARNING:convert_rows(): Error while converting row #3445 Mon Nov 7 12:29:59.357413 200: 0(0):WARNING:convert_result(): Error while converting rows Mon Nov 7 12:29:59.357424 200: 0(0):WARNING:store_result(): Error while converting result Mon Nov 7 12:29:59.357432 200: 0(0):WARNING:preload_udomain(): Error while doing db_query Mon Nov 7 12:29:59.357438 200: 0(0):WARNING:register_udomain(): Error while preloading domain 'location' Mon Nov 7 12:29:59.357447 200: 0(0):DEBUG:release_connection(): Removing connection from the pool Mon Nov 7 12:29:59.357856 200: 0(0):WARNING:domain_fixup(): Error while registering domain ERROR: error -1 while trying to fix configuration Mon Nov 7 12:29:59.357901 200: 0(0)::DEBUG: tm_shutdown : start Mon Nov 7 12:29:59.357913 200: 0(0)::DEBUG: unlink_timer_lists : emptying DELETE list Mon Nov 7 12:29:59.357920 200: 0(0)::DEBUG: tm_shutdown : emptying hash table Mon Nov 7 12:29:59.359085 200: 0(0)::DEBUG: tm_shutdown : releasing timers Mon Nov 7 12:29:59.359108 200: 0(0)::DEBUG: tm_shutdown : removing semaphores Mon Nov 7 12:29:59.359114 200: 0(0)::DEBUG: tm_shutdown : destroying tmcb lists Mon Nov 7 12:29:59.359122 200: 0(0)::DEBUG: tm_shutdown : done Mon Nov 7 12:29:59.359143 200: 0(11662):DEBUG:shm_mem_destroy Mon Nov 7 12:29:59.359154 200: 0(11662):DEBUG:destroying the shared memory lock

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[Users] Help with pseudo-variable
by Daryl Sanders 08 Nov '05

08 Nov '05

First I want to say thank you to everyone for all the kind help I've received as I start to dig into OpenSER. I think I'm starting to get a grasp of avps, as I've worked with SER for a while, but I am seriously struggling to understand the proper use of pseudo variables in OpenSER. My thought was to try something like this... if (avp_check("s:myavp", "eq/$fU/i")) { # do something here... }; I was hoping to match against the From URI username with the $fU pseudo-variable, but this doesn't appear to work. I read the doc but it's still pretty gray to me. - Daryl

2 2

[Users] acc error
by Jayesh Nambiar 08 Nov '05

08 Nov '05

hi all, I compiled the latest version od openser from cvs head. I've set the acc. I've loaded module after tm and mysql module. I added the db_url parameter, db_flag, db_table_acc, db_extra, radius_config, service_type and radius_flag. I've compiled acc with both mysql and radius support. But while starting openserctl I get the following error: Oct 27 10:41:06 localhost /usr/local/sbin/openser[6575]: ACC - initializing Oct 27 10:41:06 localhost /usr/local/sbin/openser[6575]: ERROR: acc: can't get code for the Sip-Method attribute Oct 27 10:41:06 localhost /usr/local/sbin/openser[6575]: init_mod(): Error while initializing module acc can somebody pls help me with error as I need accounting in both radius and acc. I am eagerly waiting for the 1.0 release. Its tommorow right!!!! jayesh --------------------------------- Enjoy this Diwali with Y! India Click here

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[Serusers] Error with voicemail echo and conference modules, help pls.
by Antonio Membrives 08 Nov '05

08 Nov '05

Hi to all, when I try to contact with echo or conference module I received some errors in log, the next is error of conference call, the eco received same error, any idea. Nov 8 16:49:25 dns2 Sems[30911]: Error: (AmServer.cpp)(consume_request)(88): Consumed: 18989:863098006 Nov 8 16:49:25 dns2 Sems[30911]: Error: (AmServer.cpp)(consume_request)(88): Consumed: <sip:192.168.10.240;ftag=4478083b;lr=on>,<sip:192.168.10.240;ftag=4478083b;l r=on> Nov 8 16:49:25 dns2 Sems[30911]: Error: (AmServer.cpp)(consume_request)(88): Consumed: sip:192.168.10.240;ftag=4478083b;lr=on Nov 8 16:49:25 dns2 Sems[30911]: Error: (AmServer.cpp)(consume_request)(88): Consumed: P-MsgFlags: 0 Nov 8 16:49:25 dns2 Sems[30911]: Error: (AmServer.cpp)(consume_request)(88): Consumed: . Nov 8 16:49:25 dns2 Sems[30911]: Error: (AmServer.cpp)(consume_request)(88): Consumed: . Nov 8 16:50:04 dns2 ser[31283]: LOG: Contactando con el modulo de conferencia Nov 8 16:50:04 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: 0.3 Nov 8 16:50:04 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: sip_request.conference Nov 8 16:50:04 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: INVITE Nov 8 16:50:04 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: 100 Nov 8 16:50:04 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: telintcable.es Nov 8 16:50:04 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: 192.168.10.240 Nov 8 16:50:04 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: . Nov 8 16:50:04 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: sip:100@telintcable.es Nov 8 16:50:04 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: sip:amembrives@192.168.10.10:8394 Nov 8 16:50:04 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: Antonio Membrives<sip:amembrives@192.168.10.240> Nov 8 16:50:04 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: <sip:100@telintcable.es> Nov 8 16:50:04 dns2 ser[31283]: LOG: Saliendo del modulo de conferencia Nov 8 16:50:04 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: d93fa222d25a8104 Nov 8 16:50:04 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: 4d5c7852 Nov 8 16:50:04 dns2 Sems[29810]: Error: (AmServer.cpp)(run)(191): wrong FIFO Interface version:2 Nov 8 16:50:04 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: 47710:637038812 Nov 8 16:50:04 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: . Nov 8 16:50:05 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: . Nov 8 16:50:05 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: P-MsgFlags: 1 Nov 8 16:50:05 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: . Nov 8 16:50:05 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: v=0 Nov 8 16:50:05 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: o=- 111742627 111742720 IN IP4 192.168.10.10 Nov 8 16:50:05 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: s=eyeBeam Nov 8 16:50:05 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: c=IN IP4 192.168.10.10 Nov 8 16:50:05 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: t=0 0 Nov 8 16:50:05 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: m=audio 8486 RTP/AVP 100 6 0 8 3 18 5 101 Nov 8 16:50:05 dns2 ser[31284]: LOG: Contactando con el modulo de conferencia Nov 8 16:50:05 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: a=alt:1 1 : DDA3FD3D 000000C3 192.168.10.10 8486 Nov 8 16:50:05 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: a=fmtp:101 0-15 Nov 8 16:50:05 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: a=rtpmap:100 speex/16000 Nov 8 16:50:05 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: a=rtpmap:101 telephone-event/8000 Nov 8 16:50:05 dns2 Sems[29810]: Error: (AmServer.cpp)(consume_request)(88): Consumed: a=sendrecv Nov 8 16:50:07 dns2 ser[31281]: LOG: Contactando con el modulo de conferencia This is the ser version i have: ser -V version: ser 0.9.3 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_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 20:51:59 Jun 28 2005 with gcc 3.3 ------------------- ser.cfg ------------------------------ # $Id: nat-mediaproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ debug=3 fork=yes log_stderror=no listen=82.159.201.35 # INSERT YOUR IP ADDRESS HERE listen=192.168.10.240 alias=telintcable.es alias=dns2.telintcable.es alias=82.159.201.35 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_mode=0666 fifo_db_url="mysql://ser:heslo@localhost/ser" 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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" #loadmodule "/usr/local/lib/ser/modules/vm.so" modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db","use_domain",1) modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) modparam("msilo", "registrar", "sip:registar@telintcable.es") modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("tm", "fr_timer", 10) modparam("tm", "fr_inv_timer", 15) modparam("tm", "wt_timer", 10) ## modulos para la gestion de perdidas de llamadas y reflejarse ## despues en el serweb modparam("acc", "log_flag", 2) #1 modparam("acc", "log_level", 2) modparam("acc", "db_flag", 2) #1 modparam("acc", "db_missed_flag", 3) modparam("acc", "log_missed_flag", 3) modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") #modparam("acc", "report_ack", 0) modparam("acc", "log_fmt", "miocfsp") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("82.159.201.35:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; setflag(1); # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; } else if (method=="BYE" || method=="CANCEL"){ route(5); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { if (method == "MESSAGE") { if (!t_newtran()) { sl_reply_error(); break; }; if (m_store("0")) { t_reply("202", "Accepted for Later Delivery"); break; }; t_reply("503", "Service Unavailable"); break; }; # if (!t_newtran()){ sl_send_reply("500","could not create transaction"); break; }; # no service number, redirect to voicemail if(!t_write_req("/tmp/am_fifo","voicemail")){ t_reply("500","error contacting sems"); }; break; # }#; # setflag(3); #t_relay(); ## fin de registro de mensajes. route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("telintcable.es","subscriber")) { www_challenge("telintcable.es","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; m_dump(); #msilo.so } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("telintcable.es","subscriber")) { proxy_challenge("telintcable.es","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; ##modulo de eco #setflag(1); # funciona el eco if (uri=~"sip:echo.*@telintcable.es" || uri=~"sip:echo.*@82.159.201.35") { #switch to statefull mode and redirect to echo plugin log(1, "LOG: Contactando con el modulo de eco\n"); if (!t_newtran()){ sl_send_reply("500","could not create transaction"); break; }; if(!t_write_req("/tmp/am_fifo","echo")){ t_reply("500","error contacting sems"); }; log(1, "LOG: Saliendo del modulo de eco\n"); break; } if (uri=~"sip:100.*@telintcable.es" || uri=~"sip:100.*@82.159.201.35") { #switch to statefull mode and redirect to conferenze plugin log(1, "LOG: Contactando con el modulo de conferencia\n"); if (!t_newtran()){ sl_send_reply("500","could not create transaction"); break; }; if(!t_write_req("/tmp/am_fifo","conference")){ t_reply("500","error contacting sems"); }; log(1, "LOG: Saliendo del modulo de conferencia\n"); break; } # # Codigo para almacenar las llamadas no atendidas y mostrarlas por serweb # if (!lookup("location")) { if ((method=="INVITE" || method=="ACK") && t_newtran() ) { t_reply("404", "Usuario no conectado"); acc_db_request("Llamada perdida","missed_calls"); break; }; sl_send_reply("404", "Usuario no encontrado"); break; } # setflag(3); #t_relay(); #fin del codigo route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } route[5] { log(1,"LOG: vm end - inicio\n"); if(!t_write_req("/tmp/am_fifo","bye")){ log(1,"LOG: No se ha podido contactar con vm\n"); t_reply("500","Error contacting sems"); }; log(1,"LOG: vm end - final\n"); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } failure_route[1]{ if(t_check_status("487")){ break; } else{ # no service number, redirect to voicemail if(!t_write_req("/tmp/am_fifo","voicemail")){ t_reply("500","error contacting sems"); }; } break; }

1 0

[Users] Forwarding..
by BASTARD Sebastien 08 Nov '05

08 Nov '05

Hi, I would like to make run openser as a sip client when a client is connecting to it but with an unmanaged domain. I tried with the following configuration but with no result: 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=~"iptel.org <http://iptel.org>") { forward("iptel.org <http://iptel.org>",5060); exit; } #else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} route(1); }; Thanks for any help.

2 1

[Serusers] Strange problem - SIP-->PSTN - 40 sec calls duration
by deviator 08 Nov '05

08 Nov '05

Hello! When i'm trying to call from UA to PSTN i have a call failure after 40 sec. All my UA's (private and public ip) works fine when calling each other. Calls from PSTN ---> UA's also works fine. But when i'm calling to PSTN i may talk only 40 sec. I'm tried from NAT UA's and UA's with public ip, softphones and ipphones - same result. I don't know what to do, someone please help me. UA (Sjphone) - 192.168.97.88 (symmetric nat) - 211.211.211.211 (outside ip) Openser - 212.212.212.212 PSTN Gateway (Alterteks CallSwitch)- 213.213.213.213 U 2005/11/02 16:39:14.124767 211.211.211.211:60584 -> 212.212.212.212:5060 INVITE sip:989099037259@212.212.212.212 SIP/2.0. v: SIP/2.0/UDP 192.168.97.88;rport;branch=z9hG4bKc0a86158000000ce4368c1bc000013fc00000a32. l: 186. m: <sip:102@192.168.97.88:5060>. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. c: application/sdp. CSeq: 1 INVITE. f: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. Max-Forwards: 70. t: <sip:989099037259@212.212.212.212>. . v=0. o=- 3339927612 3339927612 IN IP4 192.168.97.88. s=SJphone. c=IN IP4 192.168.97.88. t=0 0. a=direction:active. m=audio 49212 RTP/AVP 8 0. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. # U 2005/11/02 16:39:14.125680 212.212.212.212:5060 -> 211.211.211.211:60584 SIP/2.0 407 Proxy Authentication Required. v: SIP/2.0/UDP 192.168.97.88;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc000013fc00000a32;received=211.211.211.211. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 1 INVITE. f: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. t: <sip:989099037259@212.212.212.212>;tag=32db283f0ca9dc011800edf1773bf961.9b08. Proxy-Authenticate: Digest realm="212.212.212.212", nonce="4368c2aeba95bc13805d7ec3dc6a876f41f57727", qop="auth". Content-Length: 0. . # U 2005/11/02 16:39:14.161313 211.211.211.211:60584 -> 212.212.212.212:5060 ACK sip:989099037259@212.212.212.212 SIP/2.0. v: SIP/2.0/UDP 192.168.97.88;rport;branch=z9hG4bKc0a86158000000ce4368c1bc000013fc00000a32. l: 0. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 1 ACK. f: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. Max-Forwards: 70. t: <sip:989099037259@212.212.212.212>;tag=32db283f0ca9dc011800edf1773bf961.9b08. . # U 2005/11/02 16:39:14.175284 211.211.211.211:60584 -> 212.212.212.212:5060 INVITE sip:989099037259@212.212.212.212 SIP/2.0. v: SIP/2.0/UDP 192.168.97.88;rport;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. l: 186. m: <sip:102@192.168.97.88:5060>. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. c: application/sdp. CSeq: 2 INVITE. f: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. Max-Forwards: 70. t: <sip:989099037259@212.212.212.212>. Proxy-Authorization: Digest username="102",realm="212.212.212.212",nonce="4368c2aeba95bc13805d7ec3dc6a876f41f57727",uri="sip:989099037259@212.212.212.212",response="e4235b528eb7d339d41e4cb7186d6f3d",cnonce="2245531214743",qop="auth",nc="00000001". . v=0. o=- 3339927612 3339927612 IN IP4 192.168.97.88. s=SJphone. c=IN IP4 192.168.97.88. t=0 0. a=direction:active. m=audio 49212 RTP/AVP 8 0. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. # U 2005/11/02 16:39:14.194971 212.212.212.212:5060 -> 211.211.211.211:60584 SIP/2.0 100 trying -- your call is important to us. v: SIP/2.0/UDP 192.168.97.88;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33;received=211.211.211.211. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. f: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. t: <sip:989099037259@212.212.212.212>. Content-Length: 0. . # U 2005/11/02 16:39:14.195599 212.212.212.212:5060 -> 213.213.213.213:5060 INVITE sip:989099037259@213.213.213.213 SIP/2.0. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe7b5.18fe27b3.0. v: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Content-Length: 187. m: <sip:102@211.211.211.211:60584>. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. c: application/sdp. CSeq: 2 INVITE. f: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. Max-Forwards: 16. t: <sip:989099037259@212.212.212.212>. P-hint: Local Destination. . v=0. o=- 3339927612 3339927612 IN IP4 192.168.97.88. s=SJphone. c=IN IP4 212.212.212.212. t=0 0. a=direction:active. m=audio 35252 RTP/AVP 8 0. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. # U 2005/11/02 16:39:14.226940 213.213.213.213:5060 -> 212.212.212.212:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe7b5.18fe27b3.0. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Length: 0. . # U 2005/11/02 16:39:14.254568 213.213.213.213:5060 -> 212.212.212.212:5060 SIP/2.0 183 Call Proceeding. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe7b5.18fe27b3.0. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 153. . v=0. o=213.213.213.213 15435 15435 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:14.257017 212.212.212.212:5060 -> 211.211.211.211:60584 SIP/2.0 183 Call Proceeding. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 153. P-hint: NATed Reply. . v=0. o=213.213.213.213 15435 15435 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 212.212.212.212. t=0 0. m=audio 35252 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:16.420642 213.213.213.213:5060 -> 212.212.212.212:5060 SIP/2.0 183 Progress. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe7b5.18fe27b3.0. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9796 9796 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:16.422563 212.212.212.212:5060 -> 211.211.211.211:60584 SIP/2.0 183 Progress. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. P-hint: NATed Reply. . v=0. o=213.213.213.213 9796 9796 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 212.212.212.212. t=0 0. m=audio 35252 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:20.940301 213.213.213.213:5060 -> 212.212.212.212:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe7b5.18fe27b3.0. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 153. . v=0. o=213.213.213.213 13550 13550 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:20.942433 212.212.212.212:5060 -> 211.211.211.211:60584 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 153. P-hint: NATed Reply. . v=0. o=213.213.213.213 13550 13550 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 212.212.212.212. t=0 0. m=audio 35252 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:21.229331 213.213.213.213:5060 -> 212.212.212.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe7b5.18fe27b3.0. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:21.231643 212.212.212.212:5060 -> 211.211.211.211:60584 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. P-hint: NATed Reply. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 212.212.212.212. t=0 0. m=audio 35252 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:21.282286 211.211.211.211:60584 -> 212.212.212.212:5060 ACK sip:989099037259@212.212.212.212:5060;nat=yes SIP/2.0. v: SIP/2.0/UDP 192.168.97.88;rport;branch=z9hG4bKc0a86158000000ce4368c1c300007a4400000a39. l: 0. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 ACK. f: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. Max-Forwards: 70. Route: <sip:989099037259@212.212.212.212>. t: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. . # U 2005/11/02 16:39:21.852393 213.213.213.213:5060 -> 212.212.212.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe7b5.18fe27b3.0. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:21.854375 212.212.212.212:5060 -> 211.211.211.211:60584 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. P-hint: NATed Reply. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 212.212.212.212. t=0 0. m=audio 35252 RTP/AVP 8. a=ptime:20. # ACK sip:989099037259@212.212.212.212:5060;nat=yes SIP/2.0. v: SIP/2.0/UDP 192.168.97.88;rport;branch=z9hG4bKc0a86158000000ce4368c1c400000e6c00000a3b. l: 0. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 ACK. f: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. Max-Forwards: 70. Route: <sip:989099037259@212.212.212.212>. t: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. . # U 2005/11/02 16:39:23.100422 213.213.213.213:5060 -> 212.212.212.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe7b5.18fe27b3.0. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:23.102447 212.212.212.212:5060 -> 211.211.211.211:60584 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. P-hint: NATed Reply. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 212.212.212.212. t=0 0. m=audio 35252 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:23.143039 211.211.211.211:60584 -> 212.212.212.212:5060 ACK sip:989099037259@212.212.212.212:5060;nat=yes SIP/2.0. v: SIP/2.0/UDP 192.168.97.88;rport;branch=z9hG4bKc0a86158000000ce4368c1c5000046cd00000a3d. l: 0. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 ACK. f: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. Max-Forwards: 70. Route: <sip:989099037259@212.212.212.212>. t: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. . SIP/2.0 200 OK. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe7b5.18fe27b3.0. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:25.601457 212.212.212.212:5060 -> 211.211.211.211:60584 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. P-hint: NATed Reply. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 212.212.212.212. t=0 0. m=audio 35252 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:25.701745 211.211.211.211:60584 -> 212.212.212.212:5060 ACK sip:989099037259@212.212.212.212:5060;nat=yes SIP/2.0. v: SIP/2.0/UDP 192.168.97.88;rport;branch=z9hG4bKc0a86158000000ce4368c1c800003e8d00000a3f. l: 0. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 ACK. f: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. Max-Forwards: 70. Route: <sip:989099037259@212.212.212.212>. t: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. . # U 2005/11/02 16:39:30.602673 213.213.213.213:5060 -> 212.212.212.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe7b5.18fe27b3.0. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:30.602992 212.212.212.212:5060 -> 211.211.211.211:60584 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:30.643020 211.211.211.211:60584 -> 212.212.212.212:5060 ACK sip:989099037259@212.212.212.212:5060;nat=yes SIP/2.0. v: SIP/2.0/UDP 192.168.97.88;rport;branch=z9hG4bKc0a86158000000ce4368c1cd0000365000000a41. l: 0. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 ACK. f: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. Max-Forwards: 70. Route: <sip:989099037259@212.212.212.212>. t: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. . # U 2005/11/02 16:39:35.601857 213.213.213.213:5060 -> 212.212.212.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe7b5.18fe27b3.0. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:35.602200 212.212.212.212:5060 -> 211.211.211.211:60584 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:35.640413 211.211.211.211:60584 -> 212.212.212.212:5060 ACK sip:989099037259@212.212.212.212:5060;nat=yes SIP/2.0. v: SIP/2.0/UDP 192.168.97.88;rport;branch=z9hG4bKc0a86158000000ce4368c1d200002cf000000a43. l: 0. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 ACK. f: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. Max-Forwards: 70. Route: <sip:989099037259@212.212.212.212>. t: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. . U 2005/11/02 16:39:40.601018 213.213.213.213:5060 -> 212.212.212.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe7b5.18fe27b3.0. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:40.601360 212.212.212.212:5060 -> 211.211.211.211:60584 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:40.643358 211.211.211.211:60584 -> 212.212.212.212:5060 ACK sip:989099037259@212.212.212.212:5060;nat=yes SIP/2.0. v: SIP/2.0/UDP 192.168.97.88;rport;branch=z9hG4bKc0a86158000000ce4368c1d700007e8300000a45. l: 0. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 ACK. f: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. Max-Forwards: 70. Route: <sip:989099037259@212.212.212.212>. t: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. . # U 2005/11/02 16:39:45.600242 213.213.213.213:5060 -> 212.212.212.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe7b5.18fe27b3.0. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:45.600587 212.212.212.212:5060 -> 211.211.211.211:60584 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:45.640985 211.211.211.211:60584 -> 212.212.212.212:5060 ACK sip:989099037259@212.212.212.212:5060;nat=yes SIP/2.0. v: SIP/2.0/UDP 192.168.97.88;rport;branch=z9hG4bKc0a86158000000ce4368c1dc00003be400000a47. l: 0. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 ACK. f: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. Max-Forwards: 70. Route: <sip:989099037259@212.212.212.212>. t: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. . # U 2005/11/02 16:39:50.600485 213.213.213.213:5060 -> 212.212.212.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe7b5.18fe27b3.0. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:50.600831 212.212.212.212:5060 -> 211.211.211.211:60584 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:50.643167 211.211.211.211:60584 -> 212.212.212.212:5060 ACK sip:989099037259@212.212.212.212:5060;nat=yes SIP/2.0. v: SIP/2.0/UDP 192.168.97.88;rport;branch=z9hG4bKc0a86158000000ce4368c1e10000362100000a49. l: 0. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 ACK. f: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. Max-Forwards: 70. Route: <sip:989099037259@212.212.212.212>. t: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. . # U 2005/11/02 16:39:52.900945 211.211.211.211:5060 -> 212.212.212.212:5060 REGISTER sip:212.212.212.212 SIP/2.0. Via: SIP/2.0/UDP 192.168.97.234:5060;branch=z9hG4bKeU7FGO95Q. Max-Forwards: 70. User-Agent: PA168S. From: "105" <sip:105@212.212.212.212>;tag=6Bf55ba2e0h1TmPV. To: "105" <sip:105@212.212.212.212>. Call-ID: yc75i6rZUpdaIHhb(a)192.168.97.234. CSeq: 23653 REGISTER. Contact: <sip:105@192.168.97.234:5060>. Expires: 300. Content-Length: 0. . # U 2005/11/02 16:39:52.901377 212.212.212.212:5060 -> 211.211.211.211:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.97.234:5060;branch=z9hG4bKeU7FGO95Q;rport=5060;received=211.211.211.211. From: "105" <sip:105@212.212.212.212>;tag=6Bf55ba2e0h1TmPV. To: "105" <sip:105@212.212.212.212>;tag=32db283f0ca9dc011800edf1773bf961.f2c5. Call-ID: yc75i6rZUpdaIHhb(a)192.168.97.234. CSeq: 23653 REGISTER. WWW-Authenticate: Digest realm="212.212.212.212", nonce="4368c2d4935659bb5c3b365613f4d0e88ad5b13f", qop="auth". Content-Length: 0. . # U 2005/11/02 16:39:53.118213 211.211.211.211:5060 -> 212.212.212.212:5060 REGISTER sip:212.212.212.212 SIP/2.0. Via: SIP/2.0/UDP 192.168.97.234:5060;branch=z9hG4bKOhHh4mXia. Max-Forwards: 70. User-Agent: PA168S. From: "105" <sip:105@212.212.212.212>;tag=pWYCPrNjthWpPEsS. To: "105" <sip:105@212.212.212.212>. Call-ID: yc75i6rZUpdaIHhb(a)192.168.97.234. CSeq: 23654 REGISTER. Contact: <sip:105@192.168.97.234:5060>. Expires: 300. Authorization: Digest username="105", realm="212.212.212.212", nonce="4368c2d4935659bb5c3b365613f4d0e88ad5b13f", uri="sip:212.212.212.212", qop=auth, nc=00000001, cnonce="0a4f113b", response="37fd096cde5f418acbe914551ef79842". Content-Length: 0. . # U 2005/11/02 16:39:53.120270 212.212.212.212:5060 -> 211.211.211.211:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.97.234:5060;branch=z9hG4bKOhHh4mXia;rport=5060;received=211.211.211.211. From: "105" <sip:105@212.212.212.212>;tag=pWYCPrNjthWpPEsS. To: "105" <sip:105@212.212.212.212>;tag=32db283f0ca9dc011800edf1773bf961.6560. Call-ID: yc75i6rZUpdaIHhb(a)192.168.97.234. CSeq: 23654 REGISTER. Date: Wed, 02 Nov 2005 13:39:53 GMT. Contact: <sip:105@192.168.97.234:5060>;q=0.1;expires=300;received="sip:211.211.211.211:5060". Content-Length: 0. . # U 2005/11/02 16:39:55.602545 213.213.213.213:5060 -> 212.212.212.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe7b5.18fe27b3.0. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:55.602872 212.212.212.212:5060 -> 211.211.211.211:60584 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:39:55.641383 211.211.211.211:60584 -> 212.212.212.212:5060 ACK sip:989099037259@212.212.212.212:5060;nat=yes SIP/2.0. v: SIP/2.0/UDP 192.168.97.88;rport;branch=z9hG4bKc0a86158000000ce4368c1e60000163800000a4b. l: 0. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 ACK. f: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. Max-Forwards: 70. Route: <sip:989099037259@212.212.212.212>. t: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. . # U 2005/11/02 16:40:00.601769 213.213.213.213:5060 -> 212.212.212.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe7b5.18fe27b3.0. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:40:00.602053 212.212.212.212:5060 -> 211.211.211.211:60584 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.97.88;received=211.211.211.211;rport=60584;branch=z9hG4bKc0a86158000000ce4368c1bc0000318000000a33. Record-Route: <sip:989099037259@212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>. Contact: outbound <sip:989099037259@212.212.212.212>. From: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. To: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 INVITE. Content-Type: application/sdp. Content-Length: 151. . v=0. o=213.213.213.213 9442 9442 IN IP4 213.213.213.213. s= Alter Proxy Softswitch. c=IN IP4 213.213.213.213. t=0 0. m=audio 30340 RTP/AVP 8. a=ptime:20. # U 2005/11/02 16:40:00.644051 211.211.211.211:60584 -> 212.212.212.212:5060 ACK sip:989099037259@212.212.212.212:5060;nat=yes SIP/2.0. v: SIP/2.0/UDP 192.168.97.88;rport;branch=z9hG4bKc0a86158000000ce4368c1eb000017ba00000a4d. l: 0. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 2 ACK. f: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. Max-Forwards: 70. Route: <sip:989099037259@212.212.212.212>. t: <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. . # U 2005/11/02 16:40:01.288291 213.213.213.213:5060 -> 212.212.212.212:5060 BYE sip:102@212.212.212.212 SIP/2.0. Via: SIP/2.0/UDP 213.213.213.213. Route: <sip:102@211.211.211.211:60584>. From: outbound <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. To: Anonymous <sip:102@212.212.212.212>;tag=2245526517777. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 20391 BYE. Content-Length: 0. Date: Wed, 02 Nov 2005 13:42:16 GMT. . # U 2005/11/02 16:40:01.290540 212.212.212.212:5060 -> 211.211.211.211:60584 BYE sip:102@211.211.211.211:60584 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:102@212.212.212.212:5060;ftag=9796-15435-32134;lr=on>. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bK0c1c.e5e9e194.0. Via: SIP/2.0/UDP 213.213.213.213. From: outbound <sip:989099037259@212.212.212.212>;tag=9796-15435-32134. To: Anonymous <sip:102@212.212.212.212>;tag=2245526517777. Call-ID: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 20391 BYE. Content-Length: 0. Date: Wed, 02 Nov 2005 13:42:16 GMT. P-hint: Loose Routed. . # U 2005/11/02 16:40:01.332573 211.211.211.211:60584 -> 212.212.212.212:5060 SIP/2.0 200 OK. v: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bK0c1c.e5e9e194.0,SIP/2.0/UDP 213.213.213.213. l: 0. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 20391 BYE. f: outbound<sip:989099037259@212.212.212.212>;tag=9796-15435-32134. t: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. . # U 2005/11/02 16:40:01.333074 212.212.212.212:5060 -> 213.213.213.213:5060 SIP/2.0 200 OK. v: SIP/2.0/UDP 213.213.213.213. l: 0. i: C406C981-D6DB-4FF3-B716-D046FC871087(a)192.168.97.88. CSeq: 20391 BYE. f: outbound<sip:989099037259@212.212.212.212>;tag=9796-15435-32134. t: "Anonymous"<sip:102@212.212.212.212>;tag=2245526517777. P-hint: NATed Reply. .

2 6

[Users] AVP Distinctive-Ring question
by Daryl Sanders 08 Nov '05

08 Nov '05

I am using AVPs for distinctive ring preferences. I allows users to store up to 5 phone numbers that can be assigned a custom ring. The method below works fine but as you can see it requires 5 db queries. There must be a better way to do this. Any ideas? - Daryl if (avp_db_load("$ruri/username", "s:ringone")) { if (avp_check("s:ringone", "eq/$from/i")) { append_hf("Alert-Info: Bellcore-r1\r\n"); }; }; if (avp_db_load("$ruri/username", "s:ringtwo")) { if (avp_check("s:ringtwo", "eq/$from/i")) { append_hf("Alert-Info: Bellcore-r2\r\n"); }; }; if (avp_db_load("$ruri/username", "s:ringthree")) { if (avp_check("s:ringthree", "eq/$from/i")) { append_hf("Alert-Info: Bellcore-r3\r\n"); }; }; if (avp_db_load("$ruri/username", "s:ringfour")) { if (avp_check("s:ringfour", "eq/$from/i")) { append_hf("Alert-Info: Bellcore-r4\r\n"); }; }; if (avp_db_load("$ruri/username", "s:ringfive")) { if (avp_check("s:ringfive", "eq/$from/i")) { append_hf("Alert-Info: Bellcore-r5\r\n"); }; };

4 4

[Users] Routing to NATed gateways
by Leonid Fainshtein 08 Nov '05

08 Nov '05

Hi, I'd like to use OpenSER for routing calls to NATed gateways. I also want to support the LCR feature. Unfortunately, the current LCR module doesn't support NAT. I mean that the gateway IP addresses must be defined explicitly in the "gtw" table. Now I'm looking for a way to force OpenSer to use information from the "location" table. I want to write a module that will have the similar functionality like the current LCR module has but resolving procedure will be more complicated (not based on prefix and From only). The module should build avp list where each gateway will have symbolic user name. For example, g1, g2 etc. The gateways will also be registered on the proxy with those names. Then I think it will be possible to call lookup("location") after successful call the next_gtw() from the script. Is it feasible? Is there another way to solve my problem? Thank you in advance, Leonid Fainshtein

2 1

[Serusers] setting prefix according to $SIP_HF_FROM?
by Gerd Feiner 08 Nov '05

08 Nov '05

hi, is there some way, to prefix a given uri (=~[1-9]\d*, for example) based on the From-header field? suppose i want to extract a specific part (the area-code) from the From-header (like in bla(a)04711.sip...), i want to add prefix(echo $SIP_HF_FROM | sed 's/.*\@$[^.]*$\..*/\1/') or something like that in my SER config, but (even with the exec- module, which isn't at all that flexible in my opinion) there seems to be no way to accomplish this. background is my previous mail regarding aliases based on SIP_DOMAIN. if i could just extract that area-code from the From-header, then i could easily prefix it to the To-header, so i would have no problems with aliases at all. any clues? brgds, g. feiner

3 4

Re: [Users] Auth_Radius, AVP_Radius, AVOPOS Hex Value Bug
by Bogdan-Andrei Iancu 08 Nov '05

08 Nov '05

Hi Lenir, Hexa values are not supported, so set "#102#8". regards, bogdan Lenir wrote: >Hello list, > >I'm using openser 1.0 and I'm using auth_radius and avp_radius to retrieve >all my user's preferences via AVPs. Everything works very well except, that >I cannot retrieve AVPs with value of i:0xhex. I'm trying to implement the >ACL example in http://www.voice-system.ro/docs/avpops/ar01s08.html > >In my radius config I have the following: > >UserName Attribute op Value >Sipuser SIP-AVP = #102#0x08 > >In my openser.cfg I have the following: >... >if (uri=~"^sip:[0-9]{10}@") { > # Send to PSTN Gateway > xlog("L_N", " DOMESTIC CALL: from $fU to $rU\n"); > if (avp_check("i:102", "and/i:0x01")) { > route(4); #NAT Test > route(5); #PSTN Route > > } else { > xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please >check $fU's ACL\n"); > sl_send_reply("503", "Service Unavailable"); > }; > return; >}; >... > >This is the debug message that I get from openser: >0(25052) ERROR:auth_radius:extract_avp: invalid AVP numrical value '0x08' > ># ------------------------------------------------------------------------ > >So I tried to cheat, by copying i:102 to a temporary AVP i:302 with the N >flag to convert the string to numerical value, like so: >UserName Attribute op Value >Sipuser SIP-AVP = #102:0x08 > >Openser.cfg: >if (uri=~"^sip:[0-9]{10}@") { > # Send to PSTN Gateway > if (avp_copy("i:102","i:302/n") && avp_check("i:302", "and/i:0x01")) >{ > route(4); #NAT Test > route(5); #PSTN Route > > } else { > xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please >check $fU's ACL\n"); > sl_send_reply("503", "Service Unavailable"); > }; > return; >}; > >This is the debug message that I get from openser: >0(25135) ERROR:avpops:copy_avp: cannot convert str to int ># ------------------------------------------------------------------------ > >And finally, I tried to force it by having the value as string, but try to >use the ANDing op, like so: >UserName Attribute op Value >Sipuser SIP-AVP = #102:0x08 > >Openser.cfg: >... >if (uri=~"^sip:[0-9]{10}@") { > # Send to PSTN Gateway > if (avp_check("i:102", "and/i:0x01")) { > route(4); #NAT Test > route(5); #PSTN Route > > } else { > xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please >check $fU's ACL\n"); > sl_send_reply("503", "Service Unavailable"); > }; > return; >}; >... > >This is the debug message that I get from openser: >0(25140) ERROR:avpops:check_avp: value types don't match ># ------------------------------------------------------------------------ > >It seems to me that the problem lies in auth_radius by not allowing >extraction of AVPs with numerical values in hex. If we can get that work >work, I think the rest should work fine without any alterations. > >Thanks for your help in advance! > > >Lenir > > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

1 0

[Serusers] Logging problem
by Zac Tolley 08 Nov '05

08 Nov '05

I have a sip server running purely as a proxy, all it does is forward requests onto a remote sip box. I want to log all the requests that go through it, and use this to create billing records, but I can't get anything out of the server. I have setup the logging level to 8, but only get a few messages at startup. I enabled the acc.so module, in hope of creating accounting information that way, and also nothing. I know this must be because all my requests are forwarded, but I should get something surely... Any ideas what I'm doing wrong? debug=8 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" ... loadmodule "/usr/local/lib/ser/modules/acc.so" ... # -- acc params -- modparam("acc", "log_level", 8) modparam("acc", "log_flag", 1) modparam("acc", "log_fmt", "mfctios") alias="acme.co.uk" route[0]{ # 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; }; # record transaction using accounting module setflag(1); # Record route all messages to make sure that subsequent messages # go through sip proxy server. record_route(); # forward all requsts to --- forward (192.168.1.2, 5060 ); }

1 0

[Users] OpenSER now in FreeBSD ports tree
by Jesus Rodriguez 08 Nov '05

08 Nov '05

Hello, OpenSER port has been committed in FreeBSD ports tree. The port will also appear as a package in the next packages build. There are two ways to install a port in FreeBSD. Both methods require to update the ports tree (with cvsup or tarball). Once the port tree is updated: If portupgrade tool is installed just do: portupgrade -N openser If portupgrade is not used or you prefer the usual/old method: cd /usr/ports/net/openser make install clean The port allows to select some compile options like TLS support and database support for MySQL and/or Postgres. Some OpenSER packages have been generated for different FreeBSD versions (4.x, 5.x and 6.0). To install the package version just do: pkg_add http://openser.org/pub/openser/latest/packages/FreeBSD/xxxx where xxx is the package version you want to install. Use that URL to know the existing packages. When the package appears in the next official build, you will only need to do: pkg_add -r openser to install the package... easy, right? ;) Saludos JesusR. ------------------------------- Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 -------------------------------

1 0

[Serusers] ser 0.9.4 radius_auth problem
by Pham Ngoc Khanh 08 Nov '05

08 Nov '05

Dear All, I have just followed the ser radius instructions to configure and install ser-0.9.4 from source in my redhat 9 machine. Whenever I enable the auth_radius module by inserting the following lines in my ser.cfg loadmodule "/usr/local/lib/ser/modules/auth_radius.so" modparam("auth_db", "password_column", "password") modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") I got the error "bad config file (3 errors)" in the process of starting ser. But ser will start smoothly when I commented these lines. It takes me a lot of time. Pls help me to resolve this problem. Here is my full ser.cfg # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=9 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/ser_fifo" # ------------------ 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" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- 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") modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius", "service_type", 15) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- 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"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # 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()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # 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 (!radius_www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } Thank you. Best regards, Pham Ngoc Khanh

2 1

[Serusers] "copying" the SIP traffic
by Ángel Gómez 08 Nov '05

08 Nov '05

Hi every one, i interested in making SER to send a "copy" of al the trafic than comes to another SIP server. I want SER to act like ever, but in adition send the messages to another server without waitting for a response. Sorry for my english :)

2 1

[Serusers] Not able to download SER from IPTEL site
by rajesh.rajguru＠wipro.com 08 Nov '05

08 Nov '05

Hi, I am trying to access ftp://ftp.berlios.de/pub/ser/latest/ to download SER. After trying for many times also, I am not able to reach this site Has anybody faced this problem ? If yes, what is the solution ? >From where to download latest SER ? Please let me know at your earliest Regards, Rajesh Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

1 0

[Users] Call executing multiple blocks of code
by Brandon Price 08 Nov '05

08 Nov '05

I am having a problem with calls matching a certain pattern executing more than one block of code. I have a block of code that looks like if (method =~ "INVITE" && uri =~ "sip:\*12.*") { ... do some stuff }; and another like if (method=="INVITE" && uri =~ "sip:1[0-9]{10}@.*"){ ... do some stuff }; see full block here . The problem is that whenever *12 is dialed it executes both blocks, I thought the pattern matching is just wrong but when a number matching the second pattern is dialed then it doesn't execute the 1st block. Can someone please explain this behavior to me. --Sorry if this is a repost--

6 7

[Serusers] SER + dispatcher + mediaproxy, and mediaproxy could not start.
by zhangshuai 08 Nov '05

08 Nov '05

Dear all, When we use only one mediaproxy which is on the same machine with SER and disable the dispatcher, this system works perfectly. But now we want to use the dispatcher on the machine A with SER(ip:218.97.252.38), and install a mediaproxy on a separater machine B(ip:218.97.252.40), then mediaproxy just can not start. Call signaling is all OK, but there is no voice. SDP section of invite message is not changed by the mediaproxy. The log file on machine A shows that: Nov 8 16:36:10 b2bua proxydispatcher[5188]: command request 8029CEE9-92A3-4E09-91B5-0F1EE955B343(a)218.97.252.39 218.97.252.39:8000:audio 218.97.252.39 sensates.com remote 203.86.79.190 remote X-Lite=20release=201103a info=from:38001@sensates.com,to:3801@sensates.com,fromtag:614201966,totag: Nov 8 16:36:11 b2bua proxydispatcher[5188]: error: couldn't forward command to 218.97.252.40:25060 (Connection refused) The log file on machine B shows that: Nov 8 11:46:10 localhost mediaproxy[4082]: mediaproxy started. Nov 8 11:46:10 localhost mediaproxy[4082]: Listening for commands on local socket is disabled Nov 8 11:46:10 localhost mediaproxy[4082]: Mediaproxy error: couldn't create command socket (Cannot assign requested address). Nov 8 11:46:10 localhost mediaproxy[4082]: exiting. Nov 8 11:46:10 localhost proxydispatcher[4085]: proxydispatcher started. Nov 8 11:46:10 localhost proxydispatcher[4085]: disabled from configuration Nov 8 11:46:10 localhost proxydispatcher[4085]: exiting. We set mediaproxy.ini (on machine B) as follow: [Dispatcher] start = no socket = /var/run/proxydispatcher.sock group = ser [MediaProxy] start = yes socket = None group = ser listen = 218.97.252.38:25060 allow = 218.97.252.38 proxyIP = 218.97.252.40 portRange = 35000:65000 TOS = 0xb8 idleTimeout = 60 holdTimeout = 3600 forceClose = 0 accounting = off Thanks a lot.

1 0

[Serusers] Auth_Radius, AVP_Radius, AVOPOS Hex Value Bug
by Lenir 08 Nov '05

08 Nov '05

Hello list, I'm using openser 1.0 and I'm using auth_radius and avp_radius to retrieve all my user's preferences via AVPs. Everything works very well except, that I cannot retrieve AVPs with value of i:0xhex. I'm trying to implement the ACL example in http://www.voice-system.ro/docs/avpops/ar01s08.html In my radius config I have the following: UserName Attribute op Value Sipuser SIP-AVP = #102#0x08 In my openser.cfg I have the following: ... if (uri=~"^sip:[0-9]{10}@") { # Send to PSTN Gateway xlog("L_N", " DOMESTIC CALL: from $fU to $rU\n"); if (avp_check("i:102", "and/i:0x01")) { route(4); #NAT Test route(5); #PSTN Route } else { xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please check $fU's ACL\n"); sl_send_reply("503", "Service Unavailable"); }; return; }; ... This is the debug message that I get from openser: 0(25052) ERROR:auth_radius:extract_avp: invalid AVP numrical value '0x08' # ------------------------------------------------------------------------ So I tried to cheat, by copying i:102 to a temporary AVP i:302 with the N flag to convert the string to numerical value, like so: UserName Attribute op Value Sipuser SIP-AVP = #102:0x08 Openser.cfg: if (uri=~"^sip:[0-9]{10}@") { # Send to PSTN Gateway if (avp_copy("i:102","i:302/n") && avp_check("i:302", "and/i:0x01")) { route(4); #NAT Test route(5); #PSTN Route } else { xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please check $fU's ACL\n"); sl_send_reply("503", "Service Unavailable"); }; return; }; This is the debug message that I get from openser: 0(25135) ERROR:avpops:copy_avp: cannot convert str to int # ------------------------------------------------------------------------ And finally, I tried to force it by having the value as string, but try to use the ANDing op, like so: UserName Attribute op Value Sipuser SIP-AVP = #102:0x08 Openser.cfg: ... if (uri=~"^sip:[0-9]{10}@") { # Send to PSTN Gateway if (avp_check("i:102", "and/i:0x01")) { route(4); #NAT Test route(5); #PSTN Route } else { xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please check $fU's ACL\n"); sl_send_reply("503", "Service Unavailable"); }; return; }; ... This is the debug message that I get from openser: 0(25140) ERROR:avpops:check_avp: value types don't match # ------------------------------------------------------------------------ It seems to me that the problem lies in auth_radius by not allowing extraction of AVPs with numerical values in hex. If we can get that work work, I think the rest should work fine without any alterations. Thanks for your help in advance! Lenir

1 0

[Users] Auth_Radius, AVP_Radius, AVOPOS Hex Value Bug
by Lenir 08 Nov '05

08 Nov '05

Hello list, I'm using openser 1.0 and I'm using auth_radius and avp_radius to retrieve all my user's preferences via AVPs. Everything works very well except, that I cannot retrieve AVPs with value of i:0xhex. I'm trying to implement the ACL example in http://www.voice-system.ro/docs/avpops/ar01s08.html In my radius config I have the following: UserName Attribute op Value Sipuser SIP-AVP = #102#0x08 In my openser.cfg I have the following: ... if (uri=~"^sip:[0-9]{10}@") { # Send to PSTN Gateway xlog("L_N", " DOMESTIC CALL: from $fU to $rU\n"); if (avp_check("i:102", "and/i:0x01")) { route(4); #NAT Test route(5); #PSTN Route } else { xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please check $fU's ACL\n"); sl_send_reply("503", "Service Unavailable"); }; return; }; ... This is the debug message that I get from openser: 0(25052) ERROR:auth_radius:extract_avp: invalid AVP numrical value '0x08' # ------------------------------------------------------------------------ So I tried to cheat, by copying i:102 to a temporary AVP i:302 with the N flag to convert the string to numerical value, like so: UserName Attribute op Value Sipuser SIP-AVP = #102:0x08 Openser.cfg: if (uri=~"^sip:[0-9]{10}@") { # Send to PSTN Gateway if (avp_copy("i:102","i:302/n") && avp_check("i:302", "and/i:0x01")) { route(4); #NAT Test route(5); #PSTN Route } else { xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please check $fU's ACL\n"); sl_send_reply("503", "Service Unavailable"); }; return; }; This is the debug message that I get from openser: 0(25135) ERROR:avpops:copy_avp: cannot convert str to int # ------------------------------------------------------------------------ And finally, I tried to force it by having the value as string, but try to use the ANDing op, like so: UserName Attribute op Value Sipuser SIP-AVP = #102:0x08 Openser.cfg: ... if (uri=~"^sip:[0-9]{10}@") { # Send to PSTN Gateway if (avp_check("i:102", "and/i:0x01")) { route(4); #NAT Test route(5); #PSTN Route } else { xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please check $fU's ACL\n"); sl_send_reply("503", "Service Unavailable"); }; return; }; ... This is the debug message that I get from openser: 0(25140) ERROR:avpops:check_avp: value types don't match # ------------------------------------------------------------------------ It seems to me that the problem lies in auth_radius by not allowing extraction of AVPs with numerical values in hex. If we can get that work work, I think the rest should work fine without any alterations. Thanks for your help in advance! Lenir

1 0

[Serusers] Multiple INVITE's in acc table when putting caller on hold
by Bjorn Asmul 08 Nov '05

08 Nov '05

Does anyone have a solution for eliminating multiple INVITE's in the accounting table (acc) as a result of the caller putting someone on hold? Here's my accounting section: ... modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_ack", 0) ... route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (method=="INVITE" || method=="ACK" || method=="BYE" || method=="CANCEL") { setflag(1); }; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; }

2 1

[Serusers] [OTAnn] Feedback
by shenanigans 08 Nov '05

08 Nov '05

I was interested in getting feedback from current mail group users. We have mirrored your mail list in a new application that provides a more aggregated and safe environment which utilizes the power of broadband. Roomity.com v 1.5 is a web 2.01 community webapp. Our newest version adds broadcast video and social networking such as favorite authors and an html editor. It?s free to join and any feedback would be appreciated. S. -------------------------------------------------------------------------------------------------------------------------------------- Broadband interface (RIA) + mail box saftey = <a href="http://SIP_Router_Users_List.roomity.com">SIP_Router_Users_List.roomity.com</a> *Your* clubs, no sign up to read, ad supported; try broadband internet. ~~1131397872002~~ --------------------------------------------------------------------------------------------------------------------------------------

2 1

[Serusers] problem with rtpproxy
by rajesh volluru 08 Nov '05

08 Nov '05

hi I had installed ser with media and rtp proxy. Below is the output that i get when I start ser. However, iam not able receive any audio transmission when I connect through sip communicator when iam behind NAT. I am able to register with the server. 0(0) fixing /usr/local/lib/ser/modules/maxfwd.so mf_process_maxfwd_header 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so client_nat_test 0(0) fixing /usr/local/lib/ser/modules/textops.so search 0(0) module - fixing ^Record-Route: 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) fixing /usr/local/lib/ser/modules/nathelper.so fix_nated_sdp 0(0) fixing /usr/local/lib/ser/modules/rr.so record_route 0(0) fixing /usr/local/lib/ser/modules/rr.so loose_route 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/registrar.so lookup 0(0) get_connection(): Connection not found in the pool 0(0) preload_udomain(): Table is empty 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/auth_db.so www_authorize 0(0) get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/auth.so www_challenge 0(0) fixing /usr/local/lib/ser/modules/registrar.so save 0(0) get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/registrar.so lookup 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/textops.so search 0(0) module - fixing ^Route: 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so use_media_proxy 0(0) fixing /usr/local/lib/ser/modules/tm.so t_on_reply 0(0) fixing /usr/local/lib/ser/modules/tm.so t_relay 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_reply_error 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so use_media_proxy 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so client_nat_test 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) DEBUG: udp_init: trying SO_RCVBUF: 221184 0(0) DEBUG: setting SO_RCVBUF; set=221184,verify=221184 0(0) DEBUG: udp_init: trying SO_RCVBUF: 223232 0(0) DEBUG: setting SO_RCVBUF; set=223232,verify=221184 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) DBG: open_uac_fifo: opening fifo... 0(0) DEBUG: FIFO created @ /tmp/ser_fifo 0(0) DEBUG: fifo /tmp/ser_fifo opened, mode=432 0(0) init_unixsock_socket: No unix domain socket will be opened 2(0) INFO: fifo process starting: 19946 2(0) DEBUG: init_mod_child (-2): tm 2(0) DEBUG: callid: '57b7f7f7-0(a)69.229.23.57' 2(0) DEBUG: init_mod_child (-2): usrloc 2(0) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): tm 0(19941) DEBUG: callid: '57b7f7f7-19941(a)69.229.23.57' 0(19941) DEBUG: init_mod_child (1): usrloc 0(19941) get_connection(): Connection not found in the pool 1(0) DEBUG: init_mod_child (-1): tm 1(19945) DEBUG: callid: '57b7f7f7-19945(a)69.229.23.57' 1(19945) DEBUG: init_mod_child (-1): usrloc 1(19945) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): auth_db 0(19941) get_connection(): Connection not found in the pool 1(19945) DEBUG: init_mod_child (-1): auth_db 1(19945) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): domain 0(19941) get_connection(): Connection found in the pool 0(19941) DEBUG: init_mod_child (1): nathelper 2(19946) DEBUG: init_mod_child (-2): auth_db 2(19946) get_connection(): Connection not found in the pool 1(19945) DEBUG: init_mod_child (-1): domain 1(19945) DEBUG: init_mod_child (-1): nathelper 0(19941) rtpp_test: RTP proxy found, support for it enabled 1(19945) rtpp_test: RTP proxy found, support for it enabled 2(19946) DEBUG: init_mod_child (-2): domain 2(19946) DEBUG: init_mod_child (-2): nathelper 2(19946) rtpp_test: RTP proxy found, support for it enabled 2(19946) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(19946) DEBUG: register_fifo_cmd: new command (print) registered 2(19946) DEBUG: register_fifo_cmd: new command (uptime) registered 2(19946) DEBUG: register_fifo_cmd: new command (version) registered 2(19946) DEBUG: register_fifo_cmd: new command (pwd) registered 2(19946) DEBUG: register_fifo_cmd: new command (arg) registered 2(19946) DEBUG: register_fifo_cmd: new command (which) registered 2(19946) DEBUG: register_fifo_cmd: new command (ps) registered 2(19946) DEBUG: register_fifo_cmd: new command (kill) registered 2(19946) WARNING: no fifo_db_url given - fifo DB commands disabled! Let me know if you find any configuration error in SER. Please let me if you need any more information. I am trying to configure SER from long time. Please help me. Thanks Rajesh --------------------------------- Enjoy this Diwali with Y! India Click here --------------------------------- Enjoy this Diwali with Y! India Click here

2 1

[Serusers] avp_db_load fails with avp as source
by brett-ser-list 08 Nov '05

08 Nov '05

Hello All, I'm trying (desperately) to use avp_db_load with an avp: version: ser 0.9.3 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_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 12:28:21 Aug 16 2005 with gcc 3.3 modparam("auth_db","avps_column_str","callscope") modparam("avpops","avp_aliases","callgroups=i:200") modparam("avpops","avp_aliases","callscope=i:500") avp_db_load("$callscope","$callgroups/avps") ; In watching the debug, I see the avp loaded just fine from the database from the auth_db parameter: 3(1735) DEBUG:avpops:print_avp: p=0xf51c20d8, flags=3 3(1735) DEBUG: name=<callscope> 3(1735) DEBUG: val_str=<200> but the above avp_db_load line is resulting in: 3(1735) DEBUG:avpops:get_val_as_str: no avp found 3(1735) ERROR:avpops:load_avps: failed to get uuid I also notice that it doesn't even bother to hit the database after the auth_db hit. So unless the avp table has been cached (Which I have no idea if it has..) how would it know? Seems like error #1: no avp found? Seems like it doesn't like my syntax but according to the avpops documentation, it should be right: source = (sip_uri|avp_alias|str_value) ['/'('username'|'domain'|'uri'|'uuid')] note: avp_db_load("$callscope/uuid","$callgroups/avps") ; gives me: 0(0) ERROR:avpops:fixup_db_avp: source/flags "callscope" unknown! Look, what I really want to do is use the authentication credentials to load an avp. Isn't that was avp is for? I don't want the FROM or the TO or the [R]URI, but the actual AUTHentication username. Any ideas?? Thanks!! BTW, I've tried the newest SER DEVEL CVS, and OPENSER 1.0 with basically the same results. Some custom modules I have would also make me want to stick with 0.9.0. -Brett

7 13

[Serusers] New Bee Information
by Toygun Mavinil 07 Nov '05

07 Nov '05

Hi, I want to use voip for a medium size subscriber, i suggest 10.000 i want to; subscribers talk each other with no charge some one can get 2 or 4 subscribing for faks or something else i want voicemail option for people who can not answer calls maybe conference call Call logs for incoming and outgoing Phonebook maybe first i thougt that ser can do this but i dont know if it can i use suse 9.3 /mysql/php on an intel desktop pc (not a server) with p.4 3ghz 1 Gbit DDR2-533 there are some programs i did not understand their roles; Radius? Asteriks? Sim? Which programs i should use to do them? Please give information, Thanks Toygun

1 0

[Serusers] Alias per SIP_DOMAIN? something broken here?
by Gerd Feiner 07 Nov '05

07 Nov '05

hi, i run an SER which supports multiple realms. all works fine, authentication, calls etc. except one thing: we are in the rare sitiuation of having the same number-block in two different cities. i have aliases for my users (username is areacode +number) which is only the number (without area-code) to support dialing without area-code. but in the case with the two same blocks in different area-codes the alias handling gets a bit confused: # serctl alias show 88999998 <sip:0471288999998@04712.sip.domain.de>;q=1;expires=459311264 <sip:0471188999998@04711.sip.domain.de>;q=1;expires=459311264 so i have two aliases for the same number. any of the numbers is in different SIP_DOMAINS, namely 04712.sip... and 04711.sip..., but even when i state the SIP_DOMAIN to use for the lookup stuff i get: # SIP_DOMAIN=04712.sip.domain.de serctl alias show 88999998 <sip:0471288999998@04712.sip.domain.de>;q=1;expires=459311207 <sip:0471188999998@04711.sip.domain.de>;q=1;expires=459311207 and of course the other way round with 04711. the realm/domain is stored within the alias, so there are two records: 88999998 04711.sip.domain.de sip:0471188999998@04711.sip.domain.de ... 88999998 04712.sip.domain.de sip:0471288999998@04712.sip.domain.de ... with different domains. now, what happens is: when a user from the 04711 area-code calls another user (for him in that same area), he/ she might end up reaching a user with the 04712 area-code. even though the INVITE messages and all state, that the particular (any) user is from the 04711 domain, or 04712 for that matter ... i also tried to name the alias 88999998(a)04712.sip... and so on, but then, it doesn't even find an alias at all. what did i miss? oh, and yes, both realms are defined via an alias-directive in SER.cfg so he treats both as local (or at least it should). any hint would be appreciated. brgds, g. feiner

2 2

[Users] Buffer overflow with Xlog
by Jose Maria Gonzalez 07 Nov '05

07 Nov '05

I have installed openser v1.0 and it is working perfectly. But I have a problem with the xlog module. Many times I've got the following error: Nov 7 11:28:34 localhost openser[1736]: xl_printf: buffer overflow -- increase the buffer size... The line that produces this error is this (it is inside a case stament): xlog("L_ERR","Has been selected the 1 index"); I have tried to increase the buffer size in the following form: modparam("xlog","buf_size",8192) No matters the size that I put on the line buf_size, I always get the same error. I thing that a buffer size of 8kb should be enough for this log message, isn't it? Thanks. -- Salu2: *************************************** José María González ***************************************

3 3

[Serusers] Incoming calls from behind NAT
by George Lambson 07 Nov '05

07 Nov '05

Is it possible for SER to accept inbound calls from a user that is not registered and still use the mediaproxy plugin?

1 0

[Users] Are pseudo variables supported in replace()?
by Federico Giannici 07 Nov '05

07 Nov '05

Are pseudo variables sopposed to work in replace() function? The pseudo variables documentation web page says that they are supported only by avpops and xlog, but the OpenSER 1.0.0 announcement web page says that they are supported by textops too... Who is right? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

3 6

[Serusers] core dump in SER 0.9.4
by Giuseppe Tricarico 07 Nov '05

07 Nov '05

Hi all I installed SER 0.9.4 on Fedora core 4, it works well but if I use include auth_radius module (use radiusclient-ng-0.5.1) it gives a "core dump" ... Does anyone have a similar problem? I tryed to solve the problem by compiling source distribution but nothing changes.... Best Regards Giuseppe

2 3

[Serusers] allow_trusted
by OT geomatics 07 Nov '05

07 Nov '05

I need a way to trust calls coming from a dynamic IP but with a domain-name. In fact, it's an asterisk on a dynamic IP with a dynamic dns. allow_trusted just permit to consider an IP as trustable, is there another way to do that with a domain name? Olivier

3 2

RE: [Users] Cannot compile MySQL in 1.0.0
by Papadopoulos Georgios 07 Nov '05

07 Nov '05

Bogdan, It also works with -I$(LOCALBASE)/mysql/include/mysql George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Monday, November 07, 2005 3:56 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] Cannot compile MySQL in 1.0.0 > > George, > > my mistake - I meant "mysql" module, not "acc" - is any > difference between the Makefiles for this module? what was > the path you had to add? > > regards, > bogdan > Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

RE: [Users] Cannot compile MySQL in 1.0.0
by Papadopoulos Georgios 07 Nov '05

07 Nov '05

The Makefile for mysql module is identical in 0.9.5 and 1.0.0. I just had to add: -I/usr/src/mysql-4.1.10/include in DEFS, because this is where I found MySQL header files. George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Monday, November 07, 2005 3:56 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] Cannot compile MySQL in 1.0.0 > > George, > > my mistake - I meant "mysql" module, not "acc" - is any > difference between the Makefiles for this module? what was > the path you had to add? > > regards, > bogdan > > Papadopoulos Georgios wrote: > > >The Makefile for mysql module is identical in 0.9.5 and 1.0.0. > >What does the acc module have to do with this? Anyway, the > Makefile for > >acc module in 1.0.0 has > >DEFS+=-DSQL_ACC > >whereas in 0.9.5 it is commented out. > > > >I updated the -I line in modules/mysql and works fine. So I > guess the > >mysql module did not use the MySQL header files in 0.9.5. > > > >thanks a lot > > > >George > > > > > > > > > >>-----Original Message----- > >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >>Sent: Monday, November 07, 2005 2:45 PM > >>To: Papadopoulos Georgios > >>Cc: users(a)openser.org > >>Subject: Re: [Users] Cannot compile MySQL in 1.0.0 > >> > >>Hi George, > >> > >>it seams that the mysql hdr files (devel package) are not found. > >>Please check with your package manager which is the correct path to > >>these files and update the "-I....." in modules/acc/Makefile > >> > >>BTW, is any difference between the makefile for acc module between > >>your > >>0.9.5 and 1.0.0 versions? > >> > >>regards, > >>bogdan > >> > >>Papadopoulos Georgios wrote: > >> > >> > >> > >>>Hello, > >>> > >>>I am trying to upgrade from 0.9.5 to 1.0.0 and I am having trouble > >>>compiling the MySQL module (worked fine with 0.9.5). > >>>I tried both openser-1.0.0_src.tar.gz and > >>> > >>> > >>openser-1.0.0-tls_src.tar.gz, > >> > >> > >>>but the error was the same. > >>>Am I missing something? > >>> > >>>gcc -fPIC -DPIC -g -O9 -funroll-loops -Wall -mcpu=ultrasparc > >>>-mtune=ultrasparc -DNAME='"openser"' -DVERSION='"1.0.0"' > >>>-DARCH='"sparc64"' -DOS='"linux"' -DCOMPILER='"gcc 3.3"' > >>>-D__CPU_sparc64 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' > >>>-DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 > >>> > >>> > >>-DUSE_MCAST > >> > >> > >>>-DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC > -DFAST_LOCK > >>>-DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 > >>>-DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL > >>>-DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include > >>>-I/usr/local/include/mysql -I/usr/local/mysql/include > >>>-I/usr/include/mysql -c db_mod.c -o db_mod.o > >>>db_mod.c:35:19: mysql.h: No such file or directory > >>>db_mod.c: In function `mysql_mod_init': > >>>db_mod.c:87: warning: implicit declaration of function > >>>`mysql_get_client_info' > >>>db_mod.c:87: warning: format argument is not a pointer (arg 3) > >>>db_mod.c:87: warning: format argument is not a pointer (arg 3) > >>>db_mod.c:87: warning: format argument is not a pointer (arg 3) > >>>db_mod.c:87: warning: format argument is not a pointer (arg 3) > >>>db_mod.c:87: warning: format argument is not a pointer (arg 3) > >>>db_mod.c:87: warning: format argument is not a pointer (arg 3) > >>>db_mod.c:87: warning: format argument is not a pointer (arg 3) > >>>make: *** [db_mod.o] Error 1 > >>> > >>>thank you in advance > >>> > >>>George > >>> > >>> > > Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

1 0

RE: [Users] Cannot compile MySQL in 1.0.0
by Papadopoulos Georgios 07 Nov '05

07 Nov '05

The Makefile for mysql module is identical in 0.9.5 and 1.0.0. What does the acc module have to do with this? Anyway, the Makefile for acc module in 1.0.0 has DEFS+=-DSQL_ACC whereas in 0.9.5 it is commented out. I updated the -I line in modules/mysql and works fine. So I guess the mysql module did not use the MySQL header files in 0.9.5. thanks a lot George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Monday, November 07, 2005 2:45 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] Cannot compile MySQL in 1.0.0 > > Hi George, > > it seams that the mysql hdr files (devel package) are not > found. Please check with your package manager which is the > correct path to these files and update the "-I....." in > modules/acc/Makefile > > BTW, is any difference between the makefile for acc module > between your > 0.9.5 and 1.0.0 versions? > > regards, > bogdan > > Papadopoulos Georgios wrote: > > >Hello, > > > >I am trying to upgrade from 0.9.5 to 1.0.0 and I am having trouble > >compiling the MySQL module (worked fine with 0.9.5). > >I tried both openser-1.0.0_src.tar.gz and > openser-1.0.0-tls_src.tar.gz, > >but the error was the same. > >Am I missing something? > > > >gcc -fPIC -DPIC -g -O9 -funroll-loops -Wall -mcpu=ultrasparc > >-mtune=ultrasparc -DNAME='"openser"' -DVERSION='"1.0.0"' > >-DARCH='"sparc64"' -DOS='"linux"' -DCOMPILER='"gcc 3.3"' > >-D__CPU_sparc64 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' > >-DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 > -DUSE_MCAST > >-DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DFAST_LOCK > >-DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 > >-DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL > >-DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include > >-I/usr/local/include/mysql -I/usr/local/mysql/include > >-I/usr/include/mysql -c db_mod.c -o db_mod.o > >db_mod.c:35:19: mysql.h: No such file or directory > >db_mod.c: In function `mysql_mod_init': > >db_mod.c:87: warning: implicit declaration of function > >`mysql_get_client_info' > >db_mod.c:87: warning: format argument is not a pointer (arg 3) > >db_mod.c:87: warning: format argument is not a pointer (arg 3) > >db_mod.c:87: warning: format argument is not a pointer (arg 3) > >db_mod.c:87: warning: format argument is not a pointer (arg 3) > >db_mod.c:87: warning: format argument is not a pointer (arg 3) > >db_mod.c:87: warning: format argument is not a pointer (arg 3) > >db_mod.c:87: warning: format argument is not a pointer (arg 3) > >make: *** [db_mod.o] Error 1 > > > >thank you in advance > > > >George > > > >Disclaimer > >The information in this e-mail and any attachments is > confidential. It is intended solely for the attention and use > of the named addressee(s). If you are not the intended > recipient, or person responsible for delivering this > information to the intended recipient, please notify the > sender immediately. Unless you are the intended recipient or > his/her representative you are not authorized to, and must > not, read, copy, distribute, use or retain this message or > any part of it. E-mail transmission cannot be guaranteed to > be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or > contain viruses. > > > > > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[Serusers] documentation on nat_uac_test
by Maria Yndefors 07 Nov '05

07 Nov '05

Hi, Where can I find documentation on what the flags to nat_uac_test means? I have checked the admin guide and the nathelper module docs but I could not find anything. Regards /Maria

3 2

[Users] Cannot compile MySQL in 1.0.0
by Papadopoulos Georgios 07 Nov '05

07 Nov '05

Hello, I am trying to upgrade from 0.9.5 to 1.0.0 and I am having trouble compiling the MySQL module (worked fine with 0.9.5). I tried both openser-1.0.0_src.tar.gz and openser-1.0.0-tls_src.tar.gz, but the error was the same. Am I missing something? gcc -fPIC -DPIC -g -O9 -funroll-loops -Wall -mcpu=ultrasparc -mtune=ultrasparc -DNAME='"openser"' -DVERSION='"1.0.0"' -DARCH='"sparc64"' -DOS='"linux"' -DCOMPILER='"gcc 3.3"' -D__CPU_sparc64 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include -I/usr/local/include/mysql -I/usr/local/mysql/include -I/usr/include/mysql -c db_mod.c -o db_mod.o db_mod.c:35:19: mysql.h: No such file or directory db_mod.c: In function `mysql_mod_init': db_mod.c:87: warning: implicit declaration of function `mysql_get_client_info' db_mod.c:87: warning: format argument is not a pointer (arg 3) db_mod.c:87: warning: format argument is not a pointer (arg 3) db_mod.c:87: warning: format argument is not a pointer (arg 3) db_mod.c:87: warning: format argument is not a pointer (arg 3) db_mod.c:87: warning: format argument is not a pointer (arg 3) db_mod.c:87: warning: format argument is not a pointer (arg 3) db_mod.c:87: warning: format argument is not a pointer (arg 3) make: *** [db_mod.o] Error 1 thank you in advance George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[Users] Access to pseudo variables from within core
by Brandon Price 07 Nov '05

07 Nov '05

I have some questions about access to certain variables from within openser. The documentation states that I can access more of the pseudo variables like $fU and $fu from within more modules and the openser core. I am trying to use $hdr(From[0]) something that I can apparently still only do from within avpops or xlog. I need to have openser perform a certain action only if no SIP Display info exists if it does exist I do not want this action performed. For example.... if ("$hdr(From[0])") =~ "^\<sip:.*") {.... do something } else { do something else... }; Or if there is SIP Display info do A else do B. Any help greatly appreciated.

3 2

[Serusers] gflags
by Iqbal 07 Nov '05

07 Nov '05

Hi Can I just replace flag usage with gflag usage within my config, or are there any caveats Iqbal

3 5

[Serusers] ser+nat+siproxd+asterisk
by jeff kwong 07 Nov '05

07 Nov '05

Hi Guys! I just would like to share that I was able to get a working setup using SER as Softswitch, Asterisk as PSTN gateway and SIPROXD on my NAT Router. SIPROXD is an open source ALG and it effectively handles sip nat traversals. With it I dont have to run a seperate mediaproxy. When making calls from SIP UA to PSTN, RTP is as below: UA---NAT/SIPROXD---ASTERISK for 2 UA behind the same NAT: UA1--NAT---UA2 and for 2 UA behind different NATs: UA1--NAT1----NAT2---UA2 Thus there is less latency on signals and less traffic on SER. My question is, from the experience of other guys here, what do you think is the drawback or advantages of using SIPROXD together with SER to solve SIP NAT issues compared to other methods like using mediaproxy and rtpproxy?Will I still be able to do other SER features like accounting? Thanks! _jeff

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[Serusers] Call logs
by Colin Jordan 07 Nov '05

07 Nov '05

Hi there I can not find where the call logs are in the ser database. What can I do to enable call information logging (time stamps, destination, exc)? I am using mysql. Best regards, Colin

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[Serusers] Is SER a SIP PBX?
by gorand＠dvvti.com 07 Nov '05

07 Nov '05

Hey all you SER Guru's here is a noobie question that may seem dum, yet I am still going to ask it. I know SER is a SIP Proxy and a SIP Registrar and it works great. My question is can you use SER as a full blown SIP PROXY as ASTERISK? Does it handle call waiting? Call Park? Call Transfer? 3 way Conferencing? If so what are the key strokes that activates this in SER? Does SER Support Voicemail? Inquiring minds want to know? Also what is the username and password for Serweb? I tried admin and heslo and it's telling bad username and password on the screen. Thanks. Goran

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[Users] Re: [Fwd: [Serusers] Vertical Service Codes]
by Daryl Sanders 06 Nov '05

06 Nov '05

Ok, so *98 seems to the standard for at least Bell South and Qwest and also works with all my ATAs by default, so now I'm off to tackle Caller-ID Blocking, Message Waiting Indicator (w/asterisk), and Distinctive Ring. Please feel free to share you comment/experiences If you've been down this road already. - Daryl On 11/6/05, Patrick J. Menjor <jmenjor(a)comcast.net> wrote: > *98 > > -------- Original Message -------- > Subject: [Serusers] Vertical Service Codes > Date: Sun, 6 Nov 2005 08:09:13 -0700 > From: Daryl Sanders <daryl.sanders(a)gmail.com> > To: SER Users <serusers(a)iptel.org>, users(a)openser.org > > > > Can SER detect star codes? I see that there are many codes already > defined in most ATAs. How would I make these work with SER? > > Is there a standard code for allowing users to check their voicemail > from their SIP-phone/ATA? I was considering also using 00 as the code > my users dial to access their voicemail? Would there any be any > conflict with 00 and other functionality such as international > calling? There are no DID/DDI that begin with 00 correct? > > Thanks! > - Daryl > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers > > >

3 2

[Serusers] SER Business List
by Rehan Ahmed 06 Nov '05

06 Nov '05

Hello All, I am wanting to buy some did's from Japan, i wanted to know if there is a Business Mailing list of SER, and if some one can point me to it. Thank You, Rehan

1 0

[Users] Vertical Service Codes
by Daryl Sanders 06 Nov '05

06 Nov '05

Can SER detect star codes? I see that there are many codes already defined in most ATAs. How would I make these work with SER? Is there a standard code for allowing users to check their voicemail from their SIP-phone/ATA? I was considering also using 00 as the code my users dial to access their voicemail? Would there any be any conflict with 00 and other functionality such as international calling? There are no DID/DDI that begin with 00 correct? Thanks! - Daryl

1 0

[Serusers] Clearing fr_inv_timer_avp after failure route
by Evan Borgstrom 06 Nov '05

06 Nov '05

Hey All, I have fr_inv_timer_avp pointing to an AVP named inv_timeout which users are allowed to set and everything is working fine except upon call forwarding from a failure route. Here's the situation... - Call comes in for user(a)domain.com - User at domain.com has inv_timeout set to 10s - User at domain.com also has forward on unavailable to a PSTN number - 10s of ringing to the user elapses - The failure route kicks in and pushes the PSTN URI to a new branch - avp_delete("inv_timeout"); is called - avp_write("i:3600", "inv_timeout"); is called - The PSTN number begins to ring - 10s later SER sends a CANCEL to the PSTN gateway How do I properly clear the users preset inv_timeout and reset it to 1 hour for the PSTN destination? Thanks, Evan

2 2

Re: [Serusers] avp_db_load fails with avp as source
by brett-ser-list 06 Nov '05

06 Nov '05

I keep coming back to that. However, a user can be a member of multiple calling scopes. for example 1)Houston 2)LongDistance 3)Information And I can't figure out how to do it nicely that way.. seems like maybe I'd have to create a seperate column for each callscope? Surely there sure be a different way.. I simply want to load a *list* of avps based on login credentials.. seems simple enough, right? -Brett ---------- Original Message ---------------------------------- From: "Greger V. Teigre" <greger(a)teigre.com> Date: Fri, 4 Nov 2005 08:10:04 +0100 >Brett, >I have never used avpairs with sql (I use radius), but I believe the >avp_db_load statement should be removed alltogether. The avps_column_str >instructs the auth module to load the value in the callscope column ON >SUCCESSFUL AUTHENTICATION, i.e. when the user is authenticated, the avp is >already loaded and ready to be used. >g-) > >----- Original Message ----- >From: "brett-ser-list" <brett-ser-list(a)worldcall.net> >To: <serusers(a)lists.iptel.org> >Sent: Friday, November 04, 2005 2:30 AM >Subject: [Serusers] avp_db_load fails with avp as source > > >> Hello All, >> I'm trying (desperately) to use avp_db_load with an avp: >> version: ser 0.9.3 (i386/linux) >> flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, >> DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_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 12:28:21 Aug 16 2005 with gcc 3.3 >> >> modparam("auth_db","avps_column_str","callscope") >> modparam("avpops","avp_aliases","callgroups=i:200") >> modparam("avpops","avp_aliases","callscope=i:500") >> avp_db_load("$callscope","$callgroups/avps") ; >> >> In watching the debug, I see the avp loaded just fine from the database >> from the auth_db parameter: >> 3(1735) DEBUG:avpops:print_avp: p=0xf51c20d8, flags=3 >> 3(1735) DEBUG: name=<callscope> >> 3(1735) DEBUG: val_str=<200> >> >> but the above avp_db_load line is resulting in: >> 3(1735) DEBUG:avpops:get_val_as_str: no avp found >> 3(1735) ERROR:avpops:load_avps: failed to get uuid >> >> I also notice that it doesn't even bother to hit the database after the >> auth_db hit. So unless the avp table has been cached (Which I have no idea >> if it has..) how would it know? >> >> Seems like error #1: no avp found? Seems like it doesn't like my syntax >> but according to the avpops documentation, it should be right: >> source = (sip_uri|avp_alias|str_value) >> ['/'('username'|'domain'|'uri'|'uuid')] >> >> >> note: >> avp_db_load("$callscope/uuid","$callgroups/avps") ; >> gives me: >> 0(0) ERROR:avpops:fixup_db_avp: source/flags "callscope" unknown! >> >> Look, what I really want to do is use the authentication credentials to >> load an avp. Isn't that was avp is for? I don't want the FROM or the TO or >> the [R]URI, but the actual AUTHentication username. Any ideas?? >> >> Thanks!! >> BTW, I've tried the newest SER DEVEL CVS, and OPENSER 1.0 with basically >> the same results. Some custom modules I have would also make me want to >> stick with 0.9.0. >> >> -Brett >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> > >

2 1

[Serusers] ser+asterisk example
by julien bos 05 Nov '05

05 Nov '05

Hi all, I am working with asterisk to active voicemail for ser. I am newbie for it. So can you give me an example of ser.cfg for do that? Thank you very much for your instruction.

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Re: [Serusers] How to always remove auth credentials?
by brett-ser-list 05 Nov '05

05 Nov '05

I;m not entirely sure, but consume_credentials might take care of this.. Heh, or just get rid of the grandstreams! ick! ;) -Brett ---------- Original Message ---------------------------------- From: Federico Giannici <giannici(a)neomedia.it> Date: Sat, 05 Nov 2005 16:50:29 +0100 >I have found that some UAs (e.g Grandstream's ones) send the >Proxy-Authorization header even in ACKs messages. > >As we don't explicitly try to authenticate ACKs, we cannot use the >consume_credentials() function. In fact, it seems that we have to call >proxy_authorize() before we can use the consume_credentials(). But it's >contrary to RFC to call proxy_authorize() for ACKs and CANCELs methods! > >So, how can we prevent the Proxy-Authorization information to be >forwarded to subsequent hops? > > >Thanks. > >-- >___________________________________________________ > __ > |- giannici(a)neomedia.it > |ederico Giannici http://www.neomedia.it >___________________________________________________ > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Nat Problem
by harry gaillac 05 Nov '05

05 Nov '05

Hello, Is it possible to set host:port in the contact field when registration is sent ? Does nathelper allow this ? Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] How to always remove auth credentials?
by Federico Giannici 05 Nov '05

05 Nov '05

I have found that some UAs (e.g Grandstream's ones) send the Proxy-Authorization header even in ACKs messages. As we don't explicitly try to authenticate ACKs, we cannot use the consume_credentials() function. In fact, it seems that we have to call proxy_authorize() before we can use the consume_credentials(). But it's contrary to RFC to call proxy_authorize() for ACKs and CANCELs methods! So, how can we prevent the Proxy-Authorization information to be forwarded to subsequent hops? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

1 0

[Serusers] Value of max_len
by Federico Giannici 05 Nov '05

05 Nov '05

Is there a way to know what is the value assigned to the "max_len" core value? I wasn't able to find it looking at the source code... Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

1 0

[Serusers] where is the remove_hf function?
by Dale J. Robertson 05 Nov '05

05 Nov '05

I used and tested the remove_hf function in ser 8.14 on a test box running netbsd. After confirming that it operated to my satisfaction, I loaded the latest package on my soekris single board i386 computer running openbsd 3.7. When ser started up and parsed my config, It complained that it couldn't find the remove_hf function. Was this function removed from ser 9.4? or has it been relocated to an external module. Thanks, Dale

1 0

[Serusers] Windows Messenger charges user fees?
by Michael Kendall 04 Nov '05

04 Nov '05

Mark, I cannot tell you how much I appreciate your rare and insightful comments about Windows Messenger (WM). Valuable warnings of this kind are understandably difficult to find on the Microsoft Website, or anywhere else for that matter. The restrictions you describe are probably the reason that WM gets so little mention on the serusers list. I visited the Microsoft website describing WM, and all of the WM documents I find are dated 2001, as if Microsoft has abandoned the WM product, which I don't think is really the case. The documents did not mention any plan for me to pay a fee for WM use, with or without whiteboard and IM service. The site makes WM appear 100% free. Can you please direct me to a website or a phone number where I can find out the amount and conditions of such WM charges? During my startup phase, while I am evaluating the viability of my tutoring business model, it is probably acceptable for me to pay WM fees to Microsoft, and to restrict myself to the uPnP routers required by WM. But over the longer term, both of those limitations will present a serious problem, just as you have suggested. Microsoft's failure to include STUN support in WM is a serious error which limits its deployability. thanks for your help, Michael

2 2

[Serusers] Asterisk Sending SER 407
by Brett Nemeroff 04 Nov '05

04 Nov '05

Hey All, For the life of me, I can't figure this out. And I know it should be really simple. I can't get Asterisk to accept a call from SER. It appears that it's trying to authenticate the ORIGINAL caller and not the proxy itself. I've set asterisk up to use insecure=yes, but I still get a 407. I know this may be an Asterisk question, but I'm not sure if I need to be doing something different (like a consume credentials) on the SER side. Any ideas? -Brett

1 0

[Serusers] Line transfer and call waiting
by Christophe Irles 04 Nov '05

04 Nov '05

Hi, I wondering how to implement line transfer and call waiting. For the line transfer on the user side, it's necessary to create a special "ser.cfg" ? Or it's only a feature on the SIP phone/device ? How about the call waiting ? Need only a specific SIP phone/device ? Thanks, Christophe

2 1

[Serusers] serctl moni
by Richard C. Thompson 04 Nov '05

04 Nov '05

Can anyone tell me if serctl moni tells you the number of calls in progress? Transaction Statistics Current: 0 (38 waiting) Total: 2918 (0 local) Replied localy: 7014 Completion status 6xx: 0, 5xx: 26, 4xx: 546, 3xx: 0,2xx: 3410 Is it the number of waiting transactions? Thanks Richard C. Thompson

2 1

[Serusers] Checking Authorization Username == From: Username
by Russell James Bate 04 Nov '05

04 Nov '05

I have a requirement to ensure that Authorization Username == From: Username before a call is allowed. I would appreciate any advice, guidance or reference to other resources that may enable me to do this. Kind regards, Russell Bate Livewire International -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.6/152 - Release Date: 31/10/2005 The information transmitted including any attachments is intended only for the exclusive use of the person(s) or entity to which it is addressed and may contain confidential and privileged material that may be subject to legal privilege. Any perusal, review, retransmission, dissemination, distribution, reproduction or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient without the sender's prior consent is unauthorised and strictly prohibited. If you have received this message in error, please notify the sender immediately and delete the message from your computer.

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[Serusers] SER performance when enabling presence & IM
by Vanlandschoote Michel 04 Nov '05

04 Nov '05

Hi again, yet another query about SER performance figures (somewhat related to my previous post): how many users can you put on your machine running SER stateful proxy when enabling SIP presence & IM? And which hardware (CPU & memory) are you using? Many thanks, Michel ************************************************************* Dit e-mail bericht inclusief eventuele ingesloten bestanden kan informatie bevatten die vertrouwelijk is en/of beschermd door intellectuele eigendomsrechten. Dit bericht is uitsluitend bestemd voor de geadresseerde(n). Elk gebruik van de informatie vervat in dit bericht (waaronder de volledige of gedeeltelijke reproductie of verspreiding onder elke vorm) door andere personen dan de geadresseerde(n) is verboden. Indien u dit bericht per vergissing heeft ontvangen, gelieve de afzender hiervan te verwittigen en dit bericht te verwijderen. This e-mail and any attachment thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the sole use of the addressees. Any use of the information contained herein (including but not limited to total or partial reproduction or distribution in any form) by other persons than the addressees is prohibited. If you have received this e-mail in error, please notify the sender and delete its contents. *************************************************************

1 0

[Serusers] SER & RTPProxy performance figures
by Vanlandschoote Michel 04 Nov '05

04 Nov '05

Hi all, I'd like to know some real life performance figures for both SER and RTPProxy (with both applications running on separate machines): SER (as stateful SIP proxy & registrar): - how many calls-per-second and how many concurrent sessions do you handle per machine? - and of course, which hardware are you using (CPU & memory) RTPProxy: - how many streams do you handle per machine? - most important of all; which hardware (CPU & memory) - would be interesting if you also know: - which voice/video codecs are used? - what's the overall throughput in Mbps? Any feedback is highly appreciated. Michel ************************************************************* Dit e-mail bericht inclusief eventuele ingesloten bestanden kan informatie bevatten die vertrouwelijk is en/of beschermd door intellectuele eigendomsrechten. Dit bericht is uitsluitend bestemd voor de geadresseerde(n). Elk gebruik van de informatie vervat in dit bericht (waaronder de volledige of gedeeltelijke reproductie of verspreiding onder elke vorm) door andere personen dan de geadresseerde(n) is verboden. Indien u dit bericht per vergissing heeft ontvangen, gelieve de afzender hiervan te verwittigen en dit bericht te verwijderen. This e-mail and any attachment thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the sole use of the addressees. Any use of the information contained herein (including but not limited to total or partial reproduction or distribution in any form) by other persons than the addressees is prohibited. If you have received this e-mail in error, please notify the sender and delete its contents. *************************************************************

1 0

[Serusers] SER+ASTERISK
by harry gaillac 04 Nov '05

04 Nov '05

Hello, I wish to setup this scheme: ser-0.9.4 asterisk-1.2-bêta polycom ip300 phones asterisk 5050-- |firewall+nat|-----192.168. ser 5060------- My ip phones use ser as outbound sip proxy and asterisk as sip registrar server. Ser Forward REGISTER requests to asterisk however when a phone try to send an invite message then asterisk send icmp to private ip (host=dynamic in sip.conf) Is it possible to solve this problem ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] Mediaproxy Clarification
by Daryl Sanders 04 Nov '05

04 Nov '05

I would really appreciate any feedback or clarification on my understanding of Mediaproxy. It appears that Mediaproxy can be setup in such a way to allow load balancing based on domain. This seems to be the only native mechanism for getting any sort of pseudo geo-location out of it. My understanding is as follows... My users ATAs will query DNS which will direct them to a SIP proxy based on my SRV records. The SIP proxy (SER) machines will be running proxydispatcher.py and have the required mediaproxy module loaded. In addition, the SER machines may also have mediaproxy.py running locally, or mediaproxy.py may be on separate machines altogether. When a request for routing comes into the SIP proxy it is determined then if rtp proxying is necessary based on source IP. If rtp proxying is required, proxydispatcher.py send a DNS query to locate a suitable mediaproxy server. This is where SRV is used to decide (based on domain) which mediaproxy server is to be used. Given this, a separate domain would need to be setup for each geographic region that is be served by one or more mediaproxy servers. Then these domains are published to SRV to facilitate the lookups performed by each proxydispatcher.py instance. Please let me know if I'm understanding all of this correctly. Thanks! - Daryl

2 1

[Serusers] Ser and Asterisk
by Aster 04 Nov '05

04 Nov '05

SerUser, I am trying to use SER and Asterisk on the same host/system. When i try to make ser talk to asterisk, it failed and didn't want to talk to it. But if i put asterisk on different system then ser has no problem talking to asterisk. Here is flow of the traffic: UA ----------- SER ----------------<if PSTN> ------------ Asterisk ------------- Gateway. <if another user> ---------- UA Could someone please give me an idea how to make ser talk to asterisk on the same host. TIA

3 4

[Serusers] Failure route and CANCEL
by H Quintana 04 Nov '05

04 Nov '05

Hi, I'm doing some tests using the failure route feature. I send a call to an offline gateway and after the timeout, I'm sending the call to an online Gateway. It works fine if the call is not canceled, but when I did so before the offline route timeout, SER is not stopping and is sending the call to the online route. It seems that in this case the Reply status is NULL because I'm getting: "Entering failure_route. STATUS: <null>" in the log. Any ideas? Thanks for your help, Humberto ================================ failure_route[2] { xlog("L_INFO", "Entering failure_route. STATUS: %rs\n"); if (t_check_status("487")) { xlog("L_INFO", "Call cancelled\n"); break; } if (t_check_status("408") ) { xlog("L_INFO", "Time out. Gateway offline\n"); break; } rewritehost("a.b.c.d"); append_branch(); if (!t_relay()) { sl_reply_error(); break; } break; } __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

1 0

[Serusers] sipc unavailable for commercial users
by Michael Kendall 03 Nov '05

03 Nov '05

Mark, You suggested that I try sipc as an alternative to Windows Messenger. Sipc sounds like a great program since it has add-ons for voip, video, and whiteboard, as well as support for STUN. Sipc is also free for educational and research users. But I am not sure that my for-profit tutoring service would qualify as an educational organization. So I contacted Sipquest.com about pricing for a commercial license. They sent me to 3Com, who is supposed to be using sipc in their VCX product for "medium to very large businesses". The 3Com saleguy said he had never sold sipc to anyone, but he would pass my name to a company called Parallel Technologies which is considering creating a SIP server and might be planning to use sipc as a client. In summary, it appears that the commercial version of sipc is not available right now. I will try to get a free educational copy of sipc, just to evaluate it. thanks Michael

1 0

[Serusers] SRC Port change
by Ritesh Jalan 03 Nov '05

03 Nov '05

Hi Pls. help us in resolving the problem that the SIP proxy server changes the Source and Destination port while replying for message. Pls. find the below message for the Reference In below message A cisco ata device (which is behind NAT with IP 192.168.10.52) is trying to register it self with SIP proxy server IP 172.16.101.10 with SRC port 5060 and DST port 5060, but when SIP proxy server replies back to the message the port number gets changed i.e SRC port 53492 and DST port 5060. Is there any way by which we can fix this port num. No. Time Source Destination Protocol Info 532 152.672945 172.16.101.251 172.16.101.10 SIP Request: REGISTER sip:172.16.101.10 Frame 532 (469 bytes on wire, 469 bytes captured) Arrival Time: Nov 3, 2005 11:32:02.280623000 Time delta from previous packet: 0.007492000 seconds Time since reference or first frame: 152.672945000 seconds Frame Number: 532 Packet Length: 469 bytes Capture Length: 469 bytes Protocols in frame: eth:ip:udp:sip Ethernet II, Src: Cisco_b5:15:f0 (00:04:c1:b5:15:f0), Dst: Dell_17:ea:82 (00:14:22:17:ea:82) Destination: Dell_17:ea:82 (00:14:22:17:ea:82) Source: Cisco_b5:15:f0 (00:04:c1:b5:15:f0) Type: IP (0x0800) Internet Protocol, Src: 172.16.101.251 (172.16.101.251), Dst: 172.16.101.10 (172.16.101.10) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x68 (DSCP 0x1a: Assured Forwarding 31; ECN: 0x00) 0110 10.. = Differentiated Services Codepoint: Assured Forwarding 31 (0x1a) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 455 Identification: 0x01f8 (504) Flags: 0x00 0... = Reserved bit: Not set .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 249 Protocol: UDP (0x11) Header checksum: 0x1e63 [correct] Source: 172.16.101.251 (172.16.101.251) Destination: 172.16.101.10 (172.16.101.10) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 435 Checksum: 0x6c94 [correct] Session Initiation Protocol Request-Line: REGISTER sip:172.16.101.10 SIP/2.0 Method: REGISTER Resent Packet: True Suspected resend of frame: 34 Message Header Via: SIP/2.0/UDP 192.168.10.52:5060;branch=z9hG4bK9749dfca14c0ffb2 From: 23456 <sip:23456@172.16.101.10;user=phone>;tag=2971075767 SIP Display info: 23456 SIP from address: sip:23456@172.16.101.10 SIP tag: 2971075767 To: 23456 <sip:23456@172.16.101.10;user=phone> SIP Display info: 23456 SIP to address: sip:23456@172.16.101.10 Call-ID: 3431569068(a)192.168.10.52 CSeq: 1 REGISTER Contact: 23456 <sip:23456@192.168.10.52:5060;user=phone;transport=udp>;expires=300 Contact Binding: 23456 <sip:23456@192.168.10.52:5060;user=phone;transport=udp>;expires=300 URI: 23456 <sip:23456@192.168.10.52:5060;user=phone;transport=udp> SIP Display info: 23456 SIP contact address: sip:23456@192.168.10.52:5060 User-Agent: Cisco ATA 186 v3.2.1 atasip (050616A) Content-Length: 0 No. Time Source Destination Protocol Info 533 152.674292 172.16.101.10 172.16.101.251 SIP Status: 401 Unauthorized (0 bindings) Frame 533 (432 bytes on wire, 432 bytes captured) Arrival Time: Nov 3, 2005 11:32:02.281970000 Time delta from previous packet: 0.001347000 seconds Time since reference or first frame: 152.674292000 seconds Frame Number: 533 Packet Length: 432 bytes Capture Length: 432 bytes Protocols in frame: eth:ip:udp:sip Ethernet II, Src: Dell_17:ea:82 (00:14:22:17:ea:82), Dst: Cisco_b5:15:f0 (00:04:c1:b5:15:f0) Destination: Cisco_b5:15:f0 (00:04:c1:b5:15:f0) Source: Dell_17:ea:82 (00:14:22:17:ea:82) Type: IP (0x0800) Internet Protocol, Src: 172.16.101.10 (172.16.101.10), Dst: 172.16.101.251 (172.16.101.251) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x60 (DSCP 0x18: Class Selector 3; ECN: 0x00) 0110 00.. = Differentiated Services Codepoint: Class Selector 3 (0x18) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 418 Identification: 0x0000 (0) Flags: 0x04 (Don't Fragment) 0... = Reserved bit: Not set .1.. = Don't fragment: Set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 64 Protocol: UDP (0x11) Header checksum: 0x9988 [correct] Source: 172.16.101.10 (172.16.101.10) Destination: 172.16.101.251 (172.16.101.251) User Datagram Protocol, Src Port: 53492 (53492), Dst Port: 5060 (5060) Source port: 53492 (53492) Destination port: 5060 (5060) Length: 398 Checksum: 0x0ad1 [correct] Session Initiation Protocol Status-Line: SIP/2.0 401 Unauthorized Status-Code: 401 Resent Packet: True Suspected resend of frame: 480 Message Header Via: SIP/2.0/UDP 192.168.10.52:5060;received=172.16.101.251;branch=z9hG4bK9749dfca14c0ffb2 Call-ID: 3431569068(a)192.168.10.52 From: 23456 <sip:23456@172.16.101.10;user=phone>;tag=2971075767 SIP Display info: 23456 SIP from address: sip:23456@172.16.101.10 SIP tag: 2971075767 To: 23456 <sip:23456@172.16.101.10;user=phone> SIP Display info: 23456 SIP to address: sip:23456@172.16.101.10 CSeq: 1 REGISTER WWW-Authenticate: DIGEST realm="CISCO", nonce="4369a82d", qop="auth", algorithm=MD5 Content-Length: 0 Ritesh Jalan Senior Engineer - Business Solution Net4India Ltd. D-25 Secto3 Noida - 201301 India INDIA Tel: 91-120-5323500 Fax: 91-120-5323520 URL: http://www.net4.in _____ This message may contain confidential and/or privileged information. If you are not the addressee or authorized to receive this for the addressee, you must not use, copy, disclose or take any action based on this message or any information herein. If you have received this message in error, please advise the sender immediately by reply e-mail and delete this message. Thank you for your cooperation. _____

2 1

[Serusers] Packets fragmentation and message reduction
by Federico Giannici 03 Nov '05

03 Nov '05

I found a situation where some SIP phones worked and some don't. After a while I found that, because of the added bytes of the "Proxy-Authorization:" header, the messages generated by some sip phones were larger than 1480 bytes and this caused the connection with the PSTN gateway to not work. I tried to forwad the message with the t_relay_to_tcp() function, but by the ngrep output it seems that it still uses UDP (but I may be wrong). Anyway, what can be done in this situations? What about a SER module that rewrite all header field names with their compact form? Often, the message is only a few bytes more than 1480, so this could be a solution... What do you think? Bye. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

5 7

[Serusers] SERWEB Username/password
by gorand＠dvvti.com 03 Nov '05

03 Nov '05

I finally got Serweb to work. Thanks to all who contributed to that! Now I am running into a very stupid problem. I can't log in! What is the username and password? Or do i have to create it or edit certain files? Please let me know. Thanks Goran

1 0

[Serusers] FW: EmbeddedObj.o Sipstack Make error
by administrator 03 Nov '05

03 Nov '05

_____ From: administrator [mailto:administrator@hellasfon.com] Sent: Wednesday, October 26, 2005 11:49 AM To: 'b2bua(a)vovida.org' Subject: EmbeddedObj.o Make error Hello everybody, I am trying to install b2bus from cvs and I get the following error: make[1]: Leaving directory `/root/vocal/sip/sipstack' make[1]: Entering directory `/root/vocal/sip/sipstack' g++ -Wall -fPIC -Wno-deprecated -D_REENTRANT -DUSE_PTHREADS -g -I../../build -I../../build/../sdp/sdp2 -I../../build/../util -I../../build/../util/threads -I../../build/../util/logging -I../../build/../util/crypto -I../../build/../util/statistics -I../../build/../util/snmp -I../../build/../util/signals -I../../build/../util/behavior -I../../build/../util/io -I../../build/../util/services -I../../build/../util/transport -I../../build/../util/config -I../../build/../util/dnssrv -I../../build/../util/deprecated -I../../build/../util/adt -I../../build/../contrib/libxml2.Linux.i686 -I../../build/../contrib/libxml2.Linux.i686/include/libxml -I../../build/../contrib/libxml2.Linux.i686/include -DVOCAL_USE_DEPRECATED -DVOCAL_USING_PENTIUM -DOLD_PROV -c -o obj.debug.Linux.i686/EmbeddedObj.o EmbeddedObj.cxx BaseUrl.hxx:146: error: specialization of βtemplate<class _Key> struct __gnu_cxx::hashβ in different namespace /usr/lib/gcc/i386-redhat-linux/4.0.1/../../../../include/c++/4.0.1/ext/hash_ fun.h:71: error: from definition of βtemplate<class _Key> struct __gnu_cxx::hashβ EmbeddedObj.cxx: In member function βData Vocal::EmbeddedObj::doReverseEscape(const std::string&)β: EmbeddedObj.cxx:213: warning: comparison between signed and unsigned integer expressions make[1]: *** [obj.debug.Linux.i686/EmbeddedObj.o] Error 1 make[1]: Leaving directory `/root/vocal/sip/sipstack' make: *** [sip] Error 2 I user SER(0.9.4) and freeRadius (latest) on Fedora Core 4. Please Help ASAP. Thank you in advance Yiannis Marios

3 4

[Serusers] conference server question
by Jorge C 03 Nov '05

03 Nov '05

Hello everybody, I have a question about the conference server. I wanted how the mixer part is done. Specifically, how it could be implemented. For example, should the input streams (and so the input samples) from different participants be interleaved to build the output stream? or should they be added in linear way? are there other alternatives? Thanks in advance!

1 0

[Serusers] multiple avp tables
by Iqbal 03 Nov '05

03 Nov '05

Hi Is it possible to have multiple avp tables, if so how is it done, is it via mulitple modparam entries as and when needed Iqbal

2 1

[Users] Max-Forward - new functionalities
by Bogdan-Andrei Iancu 03 Nov '05

03 Nov '05

Hi all, I just committed some new functionalities in maxfwd modules. Shortly: - more detailed reporting via the return code of the functions - success cases and error cases - new function - is_maxfwd_lt(x) - checks if the value of the MF is less then x. Is useful when you forward via a path with a known number of hops. Ex: avoid fwd to another proxy requests with MF=0 (which will be rejected for sure). for more details, please see http://www.openser.org/docs/modules/1.0.x/maxfwd.html regards, bogdan

1 1

[Users] problem with uac_replace_from
by Brandon Price 03 Nov '05

03 Nov '05

I am using uac_replace_from to set the Sip display info. The problem is that it only replaces the display when one exists and is passed. It doesn't create one when one doesn't exist. So for example if I use ... uac_replace_from("batman",""); then place a call from one of my registered user agents to another then the callerid name value is displayed properly as "batman", however if I place a call from the pstn to one of my user agents which comes in with no Sip display into set then it is not generated as batman and added. I need sip display info to be replaced if it does exist, and generated if it doesn't. Any help will be greatly appreciated.

3 4

[Serusers] Need help to implement into my ser.cfg compatibility with sems or voicemail
by Antonio Membrives 03 Nov '05

03 Nov '05

Hi to all, sorry for this mail, but, I need some help to implement some voicemail compatibility, above I put my ser.cfg, this config work in Fedora 3 + MySQL + Ser 0.9.3 and work fine, the server have 2 NIC and can talk with user into my lan and out my land. But I read some documents of SEMS and try some configs and don't work. If anyone can waste some time to see this code and help me a gratefully her with some beer if visit my city for tourist :D Sems are 0.9.0 and the config are above. Sorry for my horrible English and thanks for the help. Antonio Membrives ####---------------------- ser.cfg ---------------------------------### debug=3 fork=yes log_stderror=no listen=82.159.201.35 listen=192.168.10.240 alias=telintcable.es alias=82.159.201.35 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_mode=0666 fifo_db_url="mysql://ser:heslo@localhost/ser" 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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db","use_domain",1) modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) modparam("msilo", "registrar", "sip:registar@telintcable.es") modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("tm", "fr_timer", 10) modparam("tm", "fr_inv_timer", 15) modparam("tm", "wt_timer", 10) modparam("acc", "log_flag", 2) #1 modparam("acc", "log_level", 2) modparam("acc", "db_flag", 2) #1 modparam("acc", "db_missed_flag", 3) modparam("acc", "log_missed_flag", 3) modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") #modparam("acc", "report_ack", 0) modparam("acc", "log_fmt", "miocfsp") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("82.159.201.35:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; setflag(1); # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { if (method == "MESSAGE") { if (!t_newtran()) { sl_reply_error(); break; }; if (m_store("0")) { t_reply("202", "Accepted for Later Delivery"); break; }; t_reply("503", "Service Unavailable"); break; }; if ((method=="INVITE" || method=="ACK") && t_newtran() ) { t_reply("404", "Not Found"); acc_db_request("Llamada perdida","missed_calls"); break; }; sl_send_reply("404", "Usuario Not Found"); break; }; setflag(3); route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("telintcable.es","subscriber")) { www_challenge("telintcable.es","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; m_dump(); #msilo.so } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("telintcable.es","subscriber")) { proxy_challenge("telintcable.es","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if(uri=~"sip:100.*@telintcable.es" || uri=~"sip:100.*@82.159.201.35"){ log(1, "LOG: Contactando con el modulo de conferencia\n"); if(!t_write_req("/tmp/am_fifo","conference")){ t_reply("500","Error contacting sems conference"); }; log(1, "LOG: Cerrando conexion con el modulo de conferencia\n"); break; }; if(method=="BYE" || method=="CANCEL"){ log(1,"LOG: vm end - inicio\n"); if(!t_write_req("/tmp/am_fifo","bye")){ log(1,"LOG: No se ha podido contactar con vm\n"); t_reply("500","Error contacting sems"); }; log(1,"LOG: vm end - final\n"); }; if (!lookup("location")) { if ((method=="INVITE" || method=="ACK") && t_newtran() ) { t_reply("404", "Usuario no conectado"); acc_db_request("Llamada perdida","missed_calls"); break; }; sl_send_reply("404", "Usuario no encontrado"); break; }; setflag(3); route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } ### ------------------- sems.config ---------------------------------#### # $Id: sems.conf.sample,v 1.21 2005/01/05 16:48:04 sayer Exp $ # # sems.conf.sample # # Sip Express Media Server (sems) # # sample configuration file # # # whitespaces (spaces and tabs) are ignored # comments start with a "#" and may be used inline # # example: option=value1, value2 # i like this option # ################################## # global parameters # ################################## # optional parameter: fork={yes|no} # # - specifies if sems should run in daemon mode (background) # (fork=no is the same as -E) fork=yes # optional parameter: stderr={yes|no} # # - debug mode: do not fork and log to stderr # (stderr=yes is the same as -E) stderr=no # optional parameter: loglevel={0|1|2|3} # # - sets log level (error=0, warning=1, info=2, debug=3) # (same as -D) loglevel=1 # optional parameter: fifo_name=<filename> # # - path and file name of our fifo file (same as -i) fifo_name=/tmp/am_fifo # optional parameter: ser_fifo_name=<filename> # # - path and file name of Ser's fifo file (same as -o) ser_fifo_name=/tmp/ser_fifo # optional parameter: socket_name=<filename> # # - path and file name of our unix socket #socket_name=/tmp/am_sock # optional parameter: ser_socket_name=<filename> # # - path and file name of Ser's unix socket #ser_socket_name=/tmp/ser_sock # optional parameter: send_method=<filename> # # - sets which method will be used to contact Ser: # FIFO or unix socket server. # - values: fifo (default), unix_socket # # - PLEASE NOTE THAT: if you use 'fifo', make sure that # fifo_name and ser_fifo_name have been set. if you use 'unix_socket', # make sure that socket_name and ser_socket_name have been set. # #send_method=unix_socket send_method=fifo # optional parameter: plugin_path=<path> # # - sets the path to the plug-ins # - may be absolute or relative to CWD plugin_path=/usr/local/lib/sems/plug-in/ # optional parameter: smtp_server=<hostname> # # - sets address of smtp server smtp_server=82.159.201.34 # optional parameter: smtp_port=<port> # # - sets port of smtp server smtp_port=25 # optional parameter: rtp_low_port=<port> # # - sets port of rtp lowest server #rtp_low_port=1024 # optional parameter: rtp_high_port=<port> # # - sets port of rtp highest server #rtp_high_port=65535 ################################## # module specific parameters # ################################## # sample voicemail configuration (inline) config.voicemail=inline # optional parameter: rec_file_extension=<ext> # # - sets the file extension which will be used # to record messages. # - example 1: wav # - example 2: mp3 # # WARNING: you must compile the mp3 plug-in # in order to use 'mp3' as default # extension. You will need the lame # encoder for that. # See plug-in/mp3/ReadmeMP3.txt for # more information. # rec_file_extension=wav # optional parameter: announce_path=<path> # # - sets the path where announce files are searched for # - the file to be played is searched in the following order: # <announce_path>/<domainname>/<username>.wav # <announce_path>/<domainname>/<language>/<default_announce> # <announce_path>/<domainname>/<default_announce> # <announce_path>/<language>/<default_announce> # <announce_path>/<default_announce> # where <language> is taken from the body of P-Language header # of the request (if any). announce_path=/usr/local/lib/sems/audio/ # parameter: default_announce=<filename> # # - sets the name of the default announce WAV file default_announce=default_en.wav # parameter: max_record_time=<seconds> # # - maximum record time max_record_time=30 # parameter: accept_delay=<x> # # - delays accepting of the call for x seconds # - default value is 0 accept_delay=0 # parameter: email_template_path=<filename> # # - email templates path # # See the README file in <sems-src>/plug-in/voicemail # for more information on the syntax used. # - template to be used is selected in the following order: # <email_template_path><domain>_<language>.template # <email_template_path><domain>.template # <email_template_path>default_<language>.template # <email_template_path>default.template # where <language> is taken from the body of P-Language header # of the request (if any). # email_template_path=/usr/local/etc/sems/ # end of configuration section for voicemail module config.voicemail=end # sample announcement configuration (inline) config.announcement=inline # optional parameter: announce_path=<path> # # - sets the path where announce files are searched for announce_path=/usr/local/lib/sems/audio/ # parameter: default_announce=<filename> # # - sets the name of the default announce WAV file default_announce=default_en.wav # end of configuration section for announcement module config.announcement=end # sample isdngw module configuration (external file) #config.isdngw=/etc/isdngw.conf # sample ivr module configuration (inline) config.ivr=inline # # # The IVR checks for a script with the named of the callee # (<local part in r-uri>.py for python, <local part in r-uri>.pl for perl) # in the directory <ivr_script_path><domain>, then for # <ivr_script_path><domain><ivr_script_file>. If this is not found, # <ivr_script_path>/<local part in r-uri>.py if searched, # and if this is not found, <ivr_script_path>/<ivr_script_file> is used. # # So with a call to sayer(a)iptel.org and #ivr_script_path=/etc/ivr and #ivr_script_file=ivr.py # these files are checked: #/etc/ivr/iptel.org/sayer.py #/etc/ivr/iptel.org/ivr.py #/etc/ivr/sayer.py #/etc/ivr/ivr.py # #parameter: ivr_script_path: # path to ivr scripts # ivr_script_path=/etc/ivr/ # default script file: this will be executed if <user>.py does not exist. # ivr_script_file=ivr.py # parameter : tts_caching # y or n # text will be read from waves already synthesized and # cached in cache_path tts_caching=y # parameter : tts_cache_path # path to cache waves # path must exist! tts_cache_path=/tmp/wavs # end of configuration section for ivr module config.ivr=end # sample conference configuration (inline) config.conference=inline # parameter: default_announce=<filename> # # - sets the full pathed name of the default announce WAV file. # Will be played to lonely users. default_announce=/usr/local/lib/sems/audio/first_participant.wav join_sound=/usr/local/lib/sems/audio/beep.wav drop_sound=/usr/local/lib/sems/audio/beep.wav # end of configuration section for conference module config.conference=end # example configuration for number reader config.number_reader=inline number_path=/usr/local/lib/sems/audio/ prolog_file=welcome_to_number_reader.wav epilog_file=thanks_calling_number_reader.wav # end of number_reader configuration config.number_reader=end # add more module configurations here (inline or external): # # config.mymodule=<filename> # or # config.mymodule=inline # ... # config.mymodule=end

1 0

[Users] MySql accounting in OpenSer 1.0
by Hilco de Lathouder 03 Nov '05

03 Nov '05

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sr-users