- sr-users - kamailio.org

[Serusers] fifo_db_url
by Llanos Serna García-Conde 24 May '05

24 May '05

1 0

[Serusers] How to generate ACD in redius?
by Bharat More 24 May '05

24 May '05

Hi... I want to generate ACD (Autometic Call Destribution ) using RADIUS. how can I do that? where to manipulate the code? Or Just give me the logical structure for the authentication and RADIUS accounting. Just as ( Request...sanity....forwarding.. checks for domains...etc.) Please explain in details coz i'm going to use SER First time... ________________________________________________________________________ Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony

1 0

[Serusers] cvs help
by Mohammad Khan 24 May '05

24 May '05

Hello, Where can I get instruction to download ser from cvs? What are the tags available to download? Thanks, Mohammad Khan

2 1

Re: [Serusers] javarockx (Paul) 's ser.cfg
by Dave 24 May '05

24 May '05

Hi, In the failure_route[1], an append_branch() function is missing after avp_pushto(), which causes an error during t_relay(). Am I correct or am I missing something? Thanks, Dave --- Java Rockx <javarockx(a)gmail.com> wrote: > Sorry all for the repost but the listserv scrubbed > my attachment. Here it is > again. > > P > > # ------------- Initial global variables > > debug=3 > fork=yes > log_stderror=no > > listen=10.3.0.221 <http://10.3.0.221> > port=5060 > children=4 > server_signature=no > > # hostname matching an alias will satisfy the > condition uri==myself" > alias=sip.mycompany.com <http://sip.mycompany.com> > > # dns - Uses dns to check if it is necessary to add > a "received=" field > # to a via. Default is no. > # rev_dns - Same as dns but use reverse DNS. > dns=no > rev_dns=no > > # check_via - Turn on or off Via host checking when > forwarding replies. > # Default is no. arcane. looks for discrepancy > between name and > # ip address when forwarding replies. > check_via=no > > # syn_branch - Shall the server use stateful synonym > branches? It is > # faster but not reboot-safe. Default is yes. > syn_branch=yes > > # memlog - Debugging level for final memory > statistics report. Default > # is L_DBG -- memory statistics are dumped only if > debug is set high. > memlog=1000 > > # sip_warning - Should replies include extensive > warnings? > # By default yes, it is good for trouble-shooting. > sip_warning=no > > # fifo - FIFO special file pathname > fifo_mode=0666 > fifo="/tmp/ser_fifo" > fifo_db_url="mysql://ser@localhost/ser" > > # reply_to_via - A hint to reply modules whether > they should send reply > # to IP advertised in Via. Turned off by default, > which means that > # replies are sent to IP address from which requests > came. > reply_to_via=no > > # user | uid - uid to be used by the server. 99 = > nobody. > #uid="nobody" > > # group | gid - gid to be used by the server. 99 = > nobody. > #gid="nobody" > > # mhomed -- enable calculation of outbound > interface. > # useful on multihomed servers. > mhomed=0 > > # ------------- external module loading > 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/acc.so" > loadmodule "/usr/local/lib/ser/modules/group.so" > loadmodule "/usr/local/lib/ser/modules/print.so" > loadmodule "/usr/local/lib/ser/modules/textops.so" > loadmodule > "/usr/local/lib/ser/modules/mediaproxy.so" > loadmodule "/usr/local/lib/ser/modules/nathelper.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/speeddial.so" > loadmodule "/usr/local/lib/ser/modules/options.so" > loadmodule "/usr/local/lib/ser/modules/avpops.so" > loadmodule > "/usr/local/lib/ser/modules/permissions.so" > loadmodule "/usr/local/lib/ser/modules/xdz_tools.so" > > # ------------ nathelper settings > modparam("nathelper", "rtpproxy_disable", 1) > modparam("nathelper", "natping_interval", 0) > > # ------------ media proxy settings > modparam("mediaproxy", "natping_interval", 30) > modparam("mediaproxy", "mediaproxy_socket", > "/var/run/mediaproxy.sock") > modparam("mediaproxy", "sip_asymmetrics", > "/usr/local/etc/ser/sip-asymmetric-clients") > modparam("mediaproxy", "rtp_asymmetrics", > "/usr/local/etc/ser/rtp-asymmetric-clients") > > # ------------- db_url setting > modparam("acc|auth_db|domain|group|lcr|permissions|speeddial|uri_db|usrloc|xdz_tools", > "db_url", "mysql://ser@localhost/ser") > > # ------------- use_domain setting > modparam("auth_db|group|registrar|speeddial|uri_db|usrloc", > "use_domain", 0) > > # ------------- permissions settings > modparam("permissions", "db_mode", 1) > modparam("permissions", "trusted_table", "trusted") > > # ------------- accounting parameters > modparam("acc", "db_flag", 1) > modparam("acc", "db_missed_flag", 1) > modparam("acc", "log_fmt", "cdfimorstup") > modparam("acc", "log_level", 1) > modparam("acc", "failed_transactions", 1) > modparam("acc", "report_cancels", 1) > modparam("acc", "report_ack", 0) > > # allows clear text passwords in the mysql database > modparam("auth_db", "calculate_ha1", 1) > modparam("auth_db", "password_column", "password") > modparam("auth_db", "avps_column_int", "domn") > modparam("auth_db", "avps_column_str", "uuid") > > # ------------- domain parameters > modparam("domain", "db_mode", 1) > > # ------------- registration parameters > modparam("registrar", "nat_flag", 6) > modparam("registrar", "min_expires", 300) > modparam("registrar", "max_expires", 3600) > modparam("registrar", "default_expires", 600) > modparam("registrar", "default_q", 100) > modparam("registrar", "desc_time_order", 1) > modparam("registrar", "append_branches", 0) > > # ------------- rr parameters > modparam("rr", "enable_full_lr", 0) > > # ------------- tm parameters > modparam("tm", "fr_timer", 22) > modparam("tm", "fr_inv_timer", 27) > modparam("tm", "wt_timer", 10) > modparam("tm", "fr_inv_timer_avp", "inv_timeout") > > # ------------- usrloc parameters > modparam("usrloc", "db_mode", 1) > modparam("usrloc", "timer_interval", 30) > modparam("usrloc", "desc_time_order", 1) > > # -------------- AVPOPS parameters > modparam("avpops", "avp_url", > "mysql://ser@localhost/ser") > modparam("avpops", "avp_table", "usr_preferences") > modparam("avpops", "avp_aliases", "voicemail=i:500") > > route { > > # > ------------------------------------------------------------------------ > # Sanity Check Section > # > ------------------------------------------------------------------------ > if (!mf_process_maxfwd_header("10")) { > log(1, "LOOP DETECTED"); > sl_send_reply("483", "Too Many Hops"); > break; > }; > > if (msg:len > max_len) { > log(1, "MESSAGE OVERFLOW"); > sl_send_reply("513", "Message Too Large"); > break; > }; > > # > ------------------------------------------------------------------------ > # Click2Dial Test > # > ------------------------------------------------------------------------ > if (src_ip==myself && (search("^From: <sip:700@") || > search("^To: === message truncated === __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new Resources site http://smallbusiness.yahoo.com/resources/

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[Serusers] SER CANCEL Processing
by a c 23 May '05

23 May '05

Seems like SER does not respond back to CANCEL Requests. Is this a issue / some config problem on my side? ngrep traces attached interface: eth0 (192.168.0.0/255.255.255.0) filter: ip and ( port 5060 ) # U 192.168.0.2:5062 -> 192.168.0.5:5060 INVITE sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.2:5062;r port;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voi p.com:5062>;tag=3282942385..To: <sip:9999999999@voip.com>..Contact: <sip:10 2@192.168.0.2:5062>..Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168. 0.2..CSeq: 940 INVITE..Max-Forwards: 70..Content-Type: application/sdp..Use r-Agent: X-Lite release 1103m..Content-Length: 258....v=0..o=102 7659613 76 59633 IN IP4 192.168.0.2..s=X-Lite..c=IN IP4 192.168.0.2..t=0 0..m=audio 80 00 RTP/AVP 0 8 3 98 96..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpma p:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:96 telephone-event/8000..a=fm tp:96 0-15.. # U 192.168.0.5:5060 -> 192.168.0.2:5062 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 192.16 8.0.2:5062;rport=5062;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voip.com:5062>;tag=3282942385..To: <sip:9999999999@voip.com>. .Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88@192.168.0.2..CSeq: 940 INVIT E..Server: Sip EXpress router (0.9.2 (i386/linux))..Content-Length: 0..Warn ing: 392 192.168.0.5:5060 "Noisy feedback tells: pid=2943 req_src_ip=192.1 68.0.2 req_src_port=5062 in_uri=sip:9999999999@voip.com out_uri=sip:9999999 999(a)voip.com via_cnt==1".... # U 192.168.0.5:5060 -> 192.168.0.3:5060 INVITE sip:9999999999@voip.com SIP/2.0..Record-Route: <sip:192.168.0.5;ftag =3282942385;lr=on>..Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bKe034.1018c5d 1.0..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;branch=z9hG4bKE29503D9D8C F4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voip.com:5062>;tag=3282942385..To : <sip:9999999999@voip.com>..Contact: <sip:102@192.168.0.2:5062>..Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88@192.168.0.2..CSeq: 940 INVITE..Max-For wards: 16..Content-Type: application/sdp..User-Agent: X-Lite release 1103m. .Content-Length: 258....v=0..o=102 7659613 7659633 IN IP4 192.168.0.2..s=X- Lite..c=IN IP4 192.168.0.2..t=0 0..m=audio 8000 RTP/AVP 0 8 3 98 96..a=rtpm ap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC /8000..a=rtpmap:96 telephone-event/8000..a=fmtp:96 0-15.. # U 192.168.0.3:5060 -> 192.168.0.5:5060 SIP/2.0 100 Trying..From: 102 <sip:102@voip.com:5062>;tag=3282942385..To: < sip:9999999999@voip.com>..Call-Id: 29469D19-CDF1-49ED-8245-A67E1B1DBB88@192 .168.0.2..Cseq: 940 INVITE..Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bKe034 .1018c5d1.0..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;branch=z9hG4bKE29 503D9D8CF4A479B6D9DDB17C5CFC1..Content-Length: 0.... # U 192.168.0.3:5060 -> 192.168.0.5:5060 SIP/2.0 180 Ringing..From: 102 <sip:102@voip.com:5062>;tag=3282942385..To: <sip:9999999999@voip.com>;tag=27378..Call-Id: 29469D19-CDF1-49ED-8245-A67E1 B1DBB88@192.168.0.2..Cseq: 940 INVITE..Via: SIP/2.0/UDP 192.168.0.5;branch= z9hG4bKe034.1018c5d1.0..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;branch =z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..Date: Mon, 23 May 2005 22:15:14 G MT..Contact: sip:192.168.0.3..User-Agent: sipX/2.5.2 (WinNT)..Accept-Langua ge: en..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE..Supported: sip-cc, sip-cc-01, timer, replaces..Content-Length: 0 .... # U 192.168.0.5:5060 -> 192.168.0.2:5062 SIP/2.0 180 Ringing..From: 102 <sip:102@voip.com:5062>;tag=3282942385..To: <sip:9999999999@voip.com>;tag=27378..Call-Id: 29469D19-CDF1-49ED-8245-A67E1 B1DBB88@192.168.0.2..Cseq: 940 INVITE..Via: SIP/2.0/UDP 192.168.0.2:5062;rp ort=5062;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..Date: Mon, 23 May 2005 22:15:14 GMT..Contact: sip:192.168.0.3..User-Agent: sipX/2.5.2 (WinNT) ..Accept-Language: en..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOT IFY, REGISTER, SUBSCRIBE..Supported: sip-cc, sip-cc-01, timer, replaces..Co ntent-Length: 0.... # U 192.168.0.2:5062 -> 192.168.0.5:5060 CANCEL sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.2:5062;r port;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voi p.com:5062>;tag=3282942385..To: <sip:9999999999@voip.com>..Contact: <sip:10 2@192.168.0.2:5062>..Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168. 0.2..CSeq: 940 CANCEL..Max-Forwards: 70..User-Agent: X-Lite release 1103m.. Content-Length: 0.... # U 192.168.0.2:5062 -> 192.168.0.5:5060 CANCEL sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.2:5062;r port;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voi p.com:5062>;tag=3282942385..To: <sip:9999999999@voip.com>..Contact: <sip:10 2@192.168.0.2:5062>..Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168. 0.2..CSeq: 940 CANCEL..Max-Forwards: 70..User-Agent: X-Lite release 1103m.. Content-Length: 0.... # U 192.168.0.2:5062 -> 192.168.0.5:5060 CANCEL sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.2:5062;r port;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voi p.com:5062>;tag=3282942385..To: <sip:9999999999@voip.com>..Contact: <sip:10 2@192.168.0.2:5062>..Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168. 0.2..CSeq: 940 CANCEL..Max-Forwards: 70..User-Agent: X-Lite release 1103m.. Content-Length: 0.... # U 192.168.0.5:5060 -> 192.168.0.2:5062 SIP/2.0 408 Request Timeout..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;b ranch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voip.com: 5062>;tag=3282942385..To: <sip:9999999999@voip.com>;tag=a6a1c5f60faecf035a1 ae5b6e96e979a-6e9f..Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168.0 .2..CSeq: 940 INVITE..Server: Sip EXpress router (0.9.2 (i386/linux))..Cont ent-Length: 0..Warning: 392 192.168.0.5:5060 "Noisy feedback tells: pid=29 51 req_src_ip=192.168.0.2 req_src_port=5062 in_uri=sip:9999999999@voip.com out_uri=sip:9999999999@voip.com via_cnt==0".... # U 192.168.0.5:5060 -> 192.168.0.3:5060 CANCEL sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.5;branch =z9hG4bKe034.1018c5d1.0..From: 102 <sip:102@voip.com:5062>;tag=3282942385.. Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88@192.168.0.2..To: <sip:9999999 999(a)voip.com>..CSeq: 940 CANCEL..User-Agent: Sip EXpress router(0.9.2 (i386 /linux))..Content-Length: 0.... # U 192.168.0.3:5060 -> 192.168.0.5:5060 SIP/2.0 200 OK..From: 102 <sip:102@voip.com:5062>;tag=3282942385..To: <sip: 9999999999(a)voip.com>..Call-Id: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168 .0.2..Cseq: 940 CANCEL..Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bKe034.101 8c5d1.0..Content-Length: 0.... # U 192.168.0.3:5060 -> 192.168.0.5:5060 SIP/2.0 487 Request Terminated..From: 102 <sip:102@voip.com:5062>;tag=32829 42385..To: <sip:9999999999@voip.com>;tag=27378..Call-Id: 29469D19-CDF1-49ED -8245-A67E1B1DBB88@192.168.0.2..Cseq: 940 INVITE..Via: SIP/2.0/UDP 192.168. 0.5;branch=z9hG4bKe034.1018c5d1.0..Via: SIP/2.0/UDP 192.168.0.2:5062;rport= 5062;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..Date: Mon, 23 May 2005 22:15:44 GMT..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REG ISTER, SUBSCRIBE..User-Agent: sipX/2.5.2 (WinNT)..Accept-Language: en..Supp orted: sip-cc, sip-cc-01, timer, replaces..Contact: sip:192.168.0.3..Conten t-Length: 0.... # U 192.168.0.5:5060 -> 192.168.0.3:5060 ACK sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.5;branch=z9 hG4bKe034.1018c5d1.0..From: 102 <sip:102@voip.com:5062>;tag=3282942385..Cal l-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88@192.168.0.2..To: <sip:9999999999 @voip.com>;tag=27378..CSeq: 940 ACK..User-Agent: Sip EXpress router(0.9.2 ( i386/linux))..Content-Length: 0.... # U 192.168.0.3:5060 -> 192.168.0.5:5060 SIP/2.0 200 OK..From: 102 <sip:102@voip.com:5062>;tag=3282942385..To: <sip: 9999999999(a)voip.com>..Call-Id: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168 .0.2..Cseq: 940 CANCEL..Contact: <sip:9999999999@192.168.0.3:5060>..Via: SI P/2.0/UDP 192.168.0.5;branch=z9hG4bKe034.1018c5d1.0..Date: Mon, 23 May 2005 22:15:44 GMT..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REG ISTER, SUBSCRIBE..User-Agent: sipX/2.5.2 (WinNT)..Accept-Language: en..Supp orted: sip-cc, sip-cc-01, timer, replaces..Content-Length: 0.... # U 192.168.0.5:5060 -> 192.168.0.2:5062 SIP/2.0 408 Request Timeout..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;b ranch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voip.com: 5062>;tag=3282942385..To: <sip:9999999999@voip.com>;tag=a6a1c5f60faecf035a1 ae5b6e96e979a-6e9f..Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168.0 .2..CSeq: 940 INVITE..Server: Sip EXpress router (0.9.2 (i386/linux))..Cont ent-Length: 0..Warning: 392 192.168.0.5:5060 "Noisy feedback tells: pid=29 51 req_src_ip=192.168.0.2 req_src_port=5062 in_uri=sip:9999999999@voip.com out_uri=sip:9999999999@voip.com via_cnt==0".... # U 192.168.0.5:5060 -> 192.168.0.2:5062 SIP/2.0 408 Request Timeout..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;b ranch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voip.com: 5062>;tag=3282942385..To: <sip:9999999999@voip.com>;tag=a6a1c5f60faecf035a1 ae5b6e96e979a-6e9f..Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168.0 .2..CSeq: 940 INVITE..Server: Sip EXpress router (0.9.2 (i386/linux))..Cont ent-Length: 0..Warning: 392 192.168.0.5:5060 "Noisy feedback tells: pid=29 51 req_src_ip=192.168.0.2 req_src_port=5062 in_uri=sip:9999999999@voip.com out_uri=sip:9999999999@voip.com via_cnt==0".... __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] sip responses
by a c 23 May '05

23 May '05

when SER gets a provisional response {100/180/..}, how do I figure out if the response is a 100 / 180 / ? What variable holds the response code? thank you ac __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] duplicate invite
by Ser Users 23 May '05

23 May '05

hello, I am using ser 0.8.14 with tm and acc modules and I have multiple "failure_route" blocks. when the call setup takes longer the client sends a second invite with the same call-id which I see in accounting that is not logged (it is absorbed by tm module as a retransmission I think) everything is OK, but the main route block is still executed for the retransmitted invite. is there a way I can configure ser not to execute the main routing block for the retransmitted invite ? I have tried: --------------------------------------- if (!t_newtran()) { break; } OR if(t_lookup_request) { break; } --------------------------------------- in the main route block but I get: "WARNING: script writer didn't release transaction" and a "server error" at client side. is 0.8.14 the latest "stable" version of ser ? thanks, Razvan Radu

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[Serusers] Re: [Serdev] Bug? Inconsistent return of loose_route function
by Jan Janak 23 May '05

23 May '05

Hello, comments inline. On 23-05-2005 18:57, Cesc Santasusana wrote: > Hi, > > I found what i think is a bug in the return value of loose_route() ... > bear with me because it is a rather long email. > > First, the scenario: > > phone1 ----- SER1 ----- SER2 ----- phone2 > > p1 to s1 and p2 to s2 use UDP > s1 to s2 uses tcp or tls > > The bug comes into place when there is a change of protocol in ser, > from UDP into TCP or TLS. > We do not use a preloaded route set from the phones. > What ser does is it adds 2 route headers, using the r2 parameter. > Route: <route1>, <route2> > > CASE A: -------------------------------------------- > In this scenario, say p1 places a call to p2, going through s1 and s2, > creating the route set with the record-route. > Now, let's say p2 generates the BYE message. > When the BYE gets to s1, the last hop, in the config we do: > if( loose_route () ) { > t_relay(); > } > But, and here is the inconsistency, loose_route (remember we are using > TCP/TLS, so there are 2 routes in one header field) returns FALSE. > > CASE B: -------------------------------------------- > If this experiment is repeated using ALL UDP, the route headers when they > get to s1 contain only one route, and the call to loose_route() returns .... TRUE! > > > Unless the config file does not allow blind sip message relaying, both cases end > up working ... the BYE gets to the final destination. But if the config file does not > accept relaying ... the request-uri in the BYE is not taken as a local domain ... and > thus the message is rejected ... (To Alex Mack: could this be the solution to your problem? > the BYEs not reaching the phone? add some log output and check it ... ) > > ================================================================ > So, this is not consistent. Now, what is the solution? > I think that both case A and B should return the same, but what, true or false? Yes, they should return the same value in this case and it should be true (see below). It looks like you discovered a bug. > As I see it, the solution causing less trouble is that both return true. So, the last proxy > considers this BYE as being loose_routed. > For this ... change the code in: > rr::loose_route.c:: static inline int after_loose(struct sip_msg* _m, int preloaded) > ..... > if (enable_double_rr && is_2rr(&puri.params)) { > .... > if (res > 0) { /* No next route found */ > DBG("after_loose: No next URI found\n"); > return (preloaded ? NOT_RR_DRIVEN : RR_DRIVEN); > or simply > return (preloaded RR_DRIVEN); > > > And now ... the other option is that both return false ... but this would make big changes. > Correct me if i am wrong, but wouldn't this be the compliant solution? > Considering that the r-uri is not part of the route set, when the last proxy processes > all route headers and there are no more left, there is no loose routing done. The > destination is taken from the r-uri, not the route set ... thus loose_route should return > false. > Or I am getting it all wrong? rfc and goal of loose_route() ? The meaning of the return value of loose_route is as follows (or should be): 1 - Record routing was used to route the message 0 - Record routing was not (usually requests establishing a dialog or transactional requests). So the function should return true in the mentioned case, because record routing was used although the message does not contain any Route header fields anymore. In other words the purpose of loose_route return value is to distinguish mid-dialog requests (such as ACK or BYE) from dialog-establishing requests (such as INVITE). The reason why we have to do this is that we typically do not apply all the logic in the script to mid-dialog requests (because you should not change the request-uri of the request anymore once the dialog has been established -- changes caused by loose_route do not count). You may also wonder why don't we use the contents of the Request-URI for this -- there should be the Contact value of phone1 in the Request-URI in mid-dialog requests and the value should not match uri==myself so all we need to do is to enclose all the processing logic within if (uri == myself) condition. This would work in a perfect world without the need to be backwards compatible with strict routers. In our world we have to be backwards compatible with strict routers and we have to deal with broken implementations too (that strip ;lr parameters). Due to backwards compatibility with strict routers it can happen that the Request-URI will contain the URI of the proxy that is currently processing the request and subsequent if (uri==myself) would be true. Jan.

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[Serusers] Bug? Inconsistent return of loose_route function
by Cesc Santasusana 23 May '05

23 May '05

Hi, I found what i think is a bug in the return value of loose_route() ... bear with me because it is a rather long email. First, the scenario: phone1 ----- SER1 ----- SER2 ----- phone2 p1 to s1 and p2 to s2 use UDP s1 to s2 uses tcp or tls The bug comes into place when there is a change of protocol in ser, from UDP into TCP or TLS. We do not use a preloaded route set from the phones. What ser does is it adds 2 route headers, using the r2 parameter. Route: <route1>, <route2> CASE A: -------------------------------------------- In this scenario, say p1 places a call to p2, going through s1 and s2, creating the route set with the record-route. Now, let's say p2 generates the BYE message. When the BYE gets to s1, the last hop, in the config we do: if( loose_route () ) { t_relay(); } But, and here is the inconsistency, loose_route (remember we are using TCP/TLS, so there are 2 routes in one header field) returns FALSE. CASE B: -------------------------------------------- If this experiment is repeated using ALL UDP, the route headers when they get to s1 contain only one route, and the call to loose_route() returns .... TRUE! Unless the config file does not allow blind sip message relaying, both cases end up working ... the BYE gets to the final destination. But if the config file does not accept relaying ... the request-uri in the BYE is not taken as a local domain ... and thus the message is rejected ... (To Alex Mack: could this be the solution to your problem? the BYEs not reaching the phone? add some log output and check it ... ) ================================================================ So, this is not consistent. Now, what is the solution? I think that both case A and B should return the same, but what, true or false? As I see it, the solution causing less trouble is that both return true. So, the last proxy considers this BYE as being loose_routed. For this ... change the code in: rr::loose_route.c:: static inline int after_loose(struct sip_msg* _m, int preloaded) ..... if (enable_double_rr && is_2rr(&puri.params)) { .... if (res > 0) { /* No next route found */ DBG("after_loose: No next URI found\n"); return (preloaded ? NOT_RR_DRIVEN : RR_DRIVEN); or simply return (preloaded RR_DRIVEN); And now ... the other option is that both return false ... but this would make big changes. Correct me if i am wrong, but wouldn't this be the compliant solution? Considering that the r-uri is not part of the route set, when the last proxy processes all route headers and there are no more left, there is no loose routing done. The destination is taken from the r-uri, not the route set ... thus loose_route should return false. Or I am getting it all wrong? rfc and goal of loose_route() ? Regards, Cesc Unclassified

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[Serusers] Audio Problems on SER !!!
by Felipe Martins 23 May '05

23 May '05

Hi everybody, I'm experiencing some problems with the audio of my server. My SER is configured with mysql auth and was working very well the whole friday, there were 6 people talking to each other within our private network, but today, I can't make a decent call, because the noise is very loud at every soft or hard phone I try internally. What May be that ?? Is there documented some codec audio problem about that ? Could it be a problem in my network ? Another question, I have ser worknig at a private network, with his Default Route to one of my internet gateway, My tests machine are connected via a hub just for tests, and this hub to my private network, which leaves to the internet. When I test it internallty there no problem occurs, but when I swich off the eth cable that leads to the internet (I mean the one that is my default route) SER stop authenticating my users, and I can't make a internal call. Is this a bug ?? Best Regards -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

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Re: [Serusers] UAC module (backport to 0.9.0)
by info＠beeplove.com 23 May '05

23 May '05

Does this module work on ser-0.9.0? Or, was you guys able to make it work on ser-0.9.0 If you were able to make it worked, would you please share the tricks, Thanks, Mohammad Original Message: ----------------- From: Michael Ulitskiy mdu113(a)acedsl.com Date: Mon, 23 May 2005 12:01:40 -0400 To: daniel(a)voice-system.ro, serusers(a)iptel.org Subject: Re: [Serusers] UAC module (backport to 0.9.0) On Sunday 22 May 2005 04:51 am, Daniel-Constantin Mierla wrote: > Hello, > > the issue with CSeq is known and it is stated in the documentation (see > the note there): > > http://www.voice-system.ro/docs/uac/ar01s04.html#uac_auth Thanks. Actually I read that docs, but somehow overlooked that note. Thanks for pointing it out. > Daniel Michael > > On 05/16/05 22:05, Michael Ulitskiy wrote: > > >Hello, > > > >I succeeded to make UAC module calculate credentials and > >resend message with the following simple patch on t_reply.c > >of tm module: > > > >--- t_reply.c.bak 2005-05-13 14:57:25.000000000 -0400 > >+++ t_reply.c 2005-05-13 15:16:45.000000000 -0400 > >@@ -761,6 +761,10 @@ > > a callback; save branch count to be able to determine > > later if new branches were initiated */ > > branch_cnt=Trans->nr_of_outgoings; > >+ /* also append the current reply to the transaction to > >+ * make it available in failure routes - a kind of "fake" > >+ * save of the final reply per branch */ > >+ Trans->uac[branch].reply = reply; > > > > /* run ON_FAILURE handlers ( route and callbacks) */ > > if ( has_tran_tmcbs( Trans, TMCB_ON_FAILURE_RO|TMCB_ON_FAILURE) > >@@ -769,6 +773,11 @@ > > picked_branch==branch?reply:Trans->uac[picked_branch].reply, > > picked_code); > > } > >+ > >+ /* now reset it; after the failure logic, the reply may > >+ * not be stored any more and we don't want to keep into > >+ * transaction some broken reference */ > >+ Trans->uac[branch].reply = 0; > > > > /* look if the callback perhaps replied transaction; it also > > covers the case in which a transaction is replied localy > > > >It's actually taken from cvs-head. > >Now ser.cfg written below works as expected with just one > >important issue. Authentication doesn't work anyway because > >ser doesn't increase CSeq number when resending message > >(INVITE) with credentials. I'm trying to authenticate > >to asterisk and in case of asterisk it immediately replies > >"503 Service Unavailable" if it receives INVITE with the same CSeq. > >I thought it could be asterisk-specific problem, but RFC3261 in > >section "8.1.3.5 Processing 4xx Responses" says: > >"new request constitutes a new transaction and SHOULD have the same > >value of the Call-ID, To, and From of the previous request, but the CSeq > >should contain a new sequence number that is one higher than the previous." > >So I guess this is not an asterisk-specific. > >It is my understanding that in order for this to work ser must increment > >CSeq when authenticating to UAS, but decrement CSeq in all subsequent > >in-dialog messages that will be sent to call originator including BYEs. > >This, in turn, requires for ser to be call-statefull which is not the case. > >Conclusion: uac authentication in ser is not possible. > >Please correct me if I'm wrong (I honestly want to be wrong on this matter:)) > >Thank you, > > > >Michael > > > >On Sunday 15 May 2005 12:36 am, you wrote: > > > > > >>Michael Ulitskiy wrote: > >> > >> > >> > >>>Hello, > >>> > >>>Has anyone succeeded in getting UAC authentication to work? > >>>I'm doing the following: > >>> > >>>route { > >>> resetflag(1); > >>> t_on_failure("1"); > >>> route(1); > >>>} > >>> > >>>route[1] > >>>{ > >>> if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)") { > >>> sl_send_reply("479", "We don't forward to private IP addresses"); > >>> break; > >>> }; > >>> if (!t_relay()) { > >>> sl_reply_error(); > >>> }; > >>>} > >>> > >>>failure_route[1] { > >>> # authentication reply received? > >>> if ( t_check_status("401|407") ) { > >>> if (!isflagset(1) && uac_auth()) { > >>> setflag(1); > >>> t_on_failure("1"); > >>> append_branch(); > >>> route(1); > >>> } else { > >>> t_reply("500","Error occurred"); > >>> } > >>> break; > >>> } > >>> > >>>} > >>> > >>>When uac_auth() is called I get the following in the: > >>>0(28973) DEBUG:uac:uac_auth: picked reply is (nil), code 407 > >>>0(28973) BUG:uac:uac_auth: empty reply on picked branch > >>> > >>>Any suggestions or ideas? > >>>Thank you, > >>> > >>>Michael > >>> > >>> > >>> > >>> > >>> > >>I am also getting same thing with different msg .. > >>BUG:uac:uac_auth: empty reply on picked branch > >> > >>Anybody know how to overcome this? > >> > >>Mohammad > >> > >> > >> > >> > > > >_______________________________________________ > >Serusers mailing list > >Serusers(a)iptel.org > >http://mail.iptel.org/mailman/listinfo/serusers > > > > > > > _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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Re: [Serusers] Advice needed
by Michael Ulitskiy 23 May '05

23 May '05

Yes, I guess it's another option. Not that I like it much though :) BTW I haven't seen providers that support it. 2all: Have you seen any? Thanks, Michael On Monday 23 May 2005 04:50 am, you wrote: > just an idea: > > does any PSTN provider supports TLS? If some does you have your auth > problem solved with the TLS support from SER. > > Samuel > > > Unclassified. > >>> "Greger V. Teigre" <greger(a)teigre.com> 05/22/05 08:57AM >>> > See inline. > > Michael Ulitskiy wrote: > > On Saturday 21 May 2005 02:31 am, you wrote: > >> I would say SER is what you need, except that you struggle with the > >> authentication. You have the following scenarios: > >> 1. PSTN termination with IP-based access control (easiest) > >> 2. PSTN termination with authentication of all INVITEs (yes, that's > >> the UAC module. You should contact the maintainer, Ramona-Elena > >> Modroiu about the status. I thought it was reported to work, but > >> haven't tried myself) > >> 3. PSTN termination with registration and authentication of > REGISTER > >> (but not INVITEs). Use sipsak to generate a REGISTER for your box. > >> > >> #2 requires that all INVITEs are sent twice and is not a very good > >> option. I would seek out PSTN providers who will give you #1. > >> g-) > > > > UAC module doesn't work and I think won't work unless ser is made > > call-statefull, 'cause it needs to adjust cseq within dialog. I > > posted my findings to this list > > several days ago (UAC module (backport to 0.9.0). Nobody replied so > I > > guess > > nobody knows the way to make it work. > > I saw your post on serusers, yes, but not on serdev. Because you cannot > make > a module work, doesn't mean it doesn't work for all, so as I said, if > you > have found a bug, post it to serdev (preferably) or directly to the > maintainer. That's the way open source software work... > > > As for ip auth I guess it's just not good enough. UDP invites don't > > require any handshake it's not hard at all to spoof ip address. I > > believe sending 2 invites worth the security it actually adds. > > Yes, but you can also do TCP. > > > Also I don't understand what you mean by #3. Taking ip address from > > authenticated REGISTER and then doing IP auth on that? > > No, using sipsak to actually do a REGISTER on behalf of your ser. No IP > > auth, basically it makes your ser a registered client of the GW. Of > course, > if INVITEs still must be authenticated, you are back to the UAC module > > problem. > > g-) > > > > Thanks, > > > > Michael > > > >> Michael Ulitskiy wrote: > >>> Hello, > >>> > >>> I'd like ask for advice on what is in your opinion the best > solution > >>> in the following scenario. > >>> I have a bunch of sip servers (asterisk boxes as my users need pbx > >>> functionality) that can make sip call to each other and my PSTN > >>> gateway. Now I want to purchase PSTN terminitaion in several > >>> different markets (and probably more in the future). All those > >>> terminations will require authentication. > >>> I want all my boxes when they see non-local call to send it to a > >>> central routing server that would determine where this call should > >>> be sent and authenticate to the appropriate provider so that I > don't > >>> have to configure all credentials on all asterisk boxes. Also I > want > >>> it not to deal with the media at all. All media streams should go > >>> directly from asterisk box to the PSTN termination provider. > >>> So basically it should be central SIP router that is able to > >>> authenticate calls if neccessary. > >>> I thought I could do it with SER and its UAC module, but it > appears > >>> UAC module doesn't work and probably won't work (see my previous > >>> post in this list about UAC backport to 0.9.0). > >>> Also I don't want to use asterisk in this place as asterisk always > >>> wants to stay in media path and I'd really like to avoid of > getting > >>> into hassle with re-invites. > >>> So the question is what are my options and what you would advice > >>> as a solution. Are there any software out there that can do it > >>> (preferably open-source, of course) or what else you could suggest > >>> to do to get desired results. > >>> Thanks a lot, > >> > >> > > > > -- > > See you later, > > Michael > > > > > > ------------------------------------------------------- > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Re: sip load balancing with 0.9.0 + crash
by a c 23 May '05

23 May '05

hello, I am getting a core dump when using the dispatcher module..Could someone help? I am using http://lists.iptel.org/pipermail/serdev/2004-August/002592.html to setup the config. thank you ac > > --config file-- > # $Id$ > # sample config file for dispatcher module > # > > debug=9 # debug level (cmd line: > -dddddddddd) > fork=no > log_stderror=yes # (cmd line: -E) > > children=2 > check_via=no # (cmd. line: -v) > dns=off # (cmd. line: -r) > rev_dns=off # (cmd. line: -R) > port=5060 > > # for more info: sip_router -h > > # ------------------ module loading > ---------------------------------- > > loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > loadmodule "/usr/local/lib/ser/modules/sl.so" > # loadmodule "/usr/local/lib/ser/modules/tm.so" > loadmodule > "/usr/local/lib/ser/modules/dispatcher.so" > > # ----------------- setting module-specific > parameters --------------- > # -- dispatcher params -- > > modparam("dispatcher", "list_file", > "/usr/local/etc/ser/dispatcher.list") > # modparam("dispatcher", "force_dst", 1) > > route{ > if ( !mf_process_maxfwd_header("10") ) > { > sl_send_reply("483","To Many Hops"); > drop(); > }; > > ds_select_dst("1", "0"); > > forward(uri:host, uri:port); > # t_relay(); > } > > --dispatcher.list-- > # $Id$ > # dispathcher destination sets > > #servers > 1 sip:192.168.0.2:5060 > 1 sip:192.168.0.3:5060 > > --error output-- > >ser -cf new2.cfg > > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x816402c frag. 0x8164014 (size=128) on 1 -th hit > 0(7246) loading module > /usr/local/lib/ser/modules/maxfwd.so > 0(7246) qm_malloc(0x812fc20, 16) called from > sr_module.c: register_module(136) > 0(7246) qm_malloc(0x812fc20, 16) returns address > 0x81640dc frag. 0x81640c4 (size=16) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x816411c frag. 0x8164104 (size=128) on 1 -th hit > 0(7246) loading module > /usr/local/lib/ser/modules/sl.so > 0(7246) qm_malloc(0x812fc20, 16) called from > sr_module.c: register_module(136) > 0(7246) qm_malloc(0x812fc20, 16) returns address > 0x81641cc frag. 0x81641b4 (size=16) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x816420c frag. 0x81641f4 (size=128) on 1 -th hit > 0(7246) loading module > /usr/local/lib/ser/modules/dispatcher.so > 0(7246) qm_malloc(0x812fc20, 16) called from > sr_module.c: register_module(136) > 0(7246) qm_malloc(0x812fc20, 16) returns address > 0x81642bc frag. 0x81642a4 (size=16) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x81642fc frag. 0x81642e4 (size=128) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x81643ac frag. 0x8164394 (size=128) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x816445c frag. 0x8164444 (size=128) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 13) called from > modparam.c: set_mod_param_regex(88) > 0(7246) qm_malloc(0x812fc20, 16) returns address > 0x816450c frag. 0x81644f4 (size=16) on 1 -th hit > 0(7246) set_mod_param_regex: dispatcher matches > module dispatcher > 0(7246) set_mod_param_regex: found <list_file> in > module dispatcher > [/usr/local/lib/ser/modules/dispatcher.so] > 0(7246) qm_free(0x812fc20, 0x816450c), called from > modparam.c: set_mod_param_regex(154) > 0(7246) qm_free: freeing frag. 0x81644f4 alloc'ed > from modparam.c: set_mod_param_regex(88) > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x816454c frag. 0x8164534 (size=128) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x81645fc frag. 0x81645e4 (size=128) on 1 -th hit > 0(7246) find_export: found > <mf_process_maxfwd_header> in module maxfwd > [/usr/local/lib/ser/modules/maxfwd.so] > 0(7246) qm_malloc(0x812fc20, 32) called from > route_struct.c: mk_action(89) > 0(7246) qm_malloc(0x812fc20, 32) returns address > 0x81646ac frag. 0x8164694 (size=32) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 20) called from > route_struct.c: mk_elem(70) > 0(7246) qm_malloc(0x812fc20, 20) returns address > 0x81646fc frag. 0x81646e4 (size=20) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 20) called from > route_struct.c: mk_exp(54) > 0(7246) qm_malloc(0x812fc20, 20) returns address > 0x8164740 frag. 0x8164728 (size=20) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x8164784 frag. 0x816476c (size=128) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x8164834 frag. 0x816481c (size=128) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x81648e4 frag. 0x81648cc (size=128) on 1 -th hit > 0(7246) find_export: found <sl_send_reply> in > module sl_module [/usr/local/lib/ser/modules/sl.so] > 0(7246) qm_malloc(0x812fc20, 32) called from > route_struct.c: mk_action(89) > 0(7246) qm_malloc(0x812fc20, 32) returns address > 0x8164994 frag. 0x816497c (size=32) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 32) called from > route_struct.c: mk_action(89) > 0(7246) qm_malloc(0x812fc20, 32) returns address > 0x81649e4 frag. 0x81649cc (size=32) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 32) called from > route_struct.c: mk_action(89) > 0(7246) qm_malloc(0x812fc20, 32) returns address > 0x8164a34 frag. 0x8164a1c (size=32) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x8164a84 frag. 0x8164a6c (size=128) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x8164b34 frag. 0x8164b1c (size=128) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x8164be4 frag. 0x8164bcc (size=128) on 1 -th hit > 0(7246) find_export: found <ds_select_dst> in > module dispatcher > [/usr/local/lib/ser/modules/dispatcher.so] > 0(7246) qm_malloc(0x812fc20, 32) called from > route_struct.c: mk_action(89) > 0(7246) qm_malloc(0x812fc20, 32) returns address > 0x8164c94 frag. 0x8164c7c (size=32) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 32) called from > route_struct.c: mk_action(89) > 0(7246) qm_malloc(0x812fc20, 32) returns address > 0x8164ce4 frag. 0x8164ccc (size=32) on 1 -th hit > 0(7246) routing table 0: > 0(7246) if ( 0(7246) NOT( 0(7246) > external_module_call( 0(7246) f_ptr<0x4001a060> > 0(7246) , "10" 0(7246) ); 0(7246) > external_module_call( 0(7246) f_ptr<0x4001a060> > 0(7246) , "10" 0(7246) ); 0(7246) ) 0(7246) ) { > 0(7246) external_module_call( 0(7246) > f_ptr<0x4016ba20> 0(7246) , "483" 0(7246) , "To Many > Hops" ... WARNING: no fork mode and more than one listen address found(will use only the the first one)^M config file ok, exiting...^M 0(7246) DISPATCHER: destroy module ...^M 0(7246) qm_free(0x812fc20, (nil)), called from dispatch.c: ds_destroy_list(347) ^M 0(7246) BUG: qm_free: bad pointer (nil) (out of memory block!) - aborting^M Aborted (core dumped)^M __________________________________ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail

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[Serusers] Re: SIP load balancing cont..
by a c 23 May '05

23 May '05

to clarify the question, how do you tell what messages SER is sending out {without using ngrep}? I would like to get a printout everytime it sends 'trying', 'ringing', / 200 OK. What would the config look like? Is this possible? alex --- a c <lra101(a)yahoo.com> wrote: > > this worked with t_check_status(). One thing I could > not figure out is how to get a log printout when I > receive a 200 OK. How do I know if I received a 200 > OK? > I can't put this logic in the main route block. > > thank you > ac > --- a c <lra101(a)yahoo.com> wrote: > > > > I need to route to a failure route when there is a > > certain kind of failure. For example, If I don't > > receive a 'Trying' back on the initial attempt{no > > response to Invite}, then I would route using the > > failure route. For all other failures {i.e. 404, > no > > answer, 486, etc..}, I don't try any other routes. > > How > > can this be configured? > > > > thank you > > ac > > > > > > > > Discover Yahoo! > > Have fun online with music videos, cool games, IM > > and more. Check it out! > > http://discover.yahoo.com/online.html > > > > > > __________________________________ > Do you Yahoo!? > Make Yahoo! your home page > http://www.yahoo.com/r/hs > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

Re: [Serusers] UAC module (backport to 0.9.0)
by Michael Ulitskiy 23 May '05

23 May '05

Hello, I succeeded to make UAC module calculate credentials and resend message with the following simple patch on t_reply.c of tm module: --- t_reply.c.bak 2005-05-13 14:57:25.000000000 -0400 +++ t_reply.c 2005-05-13 15:16:45.000000000 -0400 @@ -761,6 +761,10 @@ a callback; save branch count to be able to determine later if new branches were initiated */ branch_cnt=Trans->nr_of_outgoings; + /* also append the current reply to the transaction to + * make it available in failure routes - a kind of "fake" + * save of the final reply per branch */ + Trans->uac[branch].reply = reply; /* run ON_FAILURE handlers ( route and callbacks) */ if ( has_tran_tmcbs( Trans, TMCB_ON_FAILURE_RO|TMCB_ON_FAILURE) @@ -769,6 +773,11 @@ picked_branch==branch?reply:Trans->uac[picked_branch].reply, picked_code); } + + /* now reset it; after the failure logic, the reply may + * not be stored any more and we don't want to keep into + * transaction some broken reference */ + Trans->uac[branch].reply = 0; /* look if the callback perhaps replied transaction; it also covers the case in which a transaction is replied localy It's actually taken from cvs-head. Now ser.cfg written below works as expected with just one important issue. Authentication doesn't work anyway because ser doesn't increase CSeq number when resending message (INVITE) with credentials. I'm trying to authenticate to asterisk and in case of asterisk it immediately replies "503 Service Unavailable" if it receives INVITE with the same CSeq. I thought it could be asterisk-specific problem, but RFC3261 in section "8.1.3.5 Processing 4xx Responses" says: "new request constitutes a new transaction and SHOULD have the same value of the Call-ID, To, and From of the previous request, but the CSeq should contain a new sequence number that is one higher than the previous." So I guess this is not an asterisk-specific. It is my understanding that in order for this to work ser must increment CSeq when authenticating to UAS, but decrement CSeq in all subsequent in-dialog messages that will be sent to call originator including BYEs. This, in turn, requires for ser to be call-statefull which is not the case. Conclusion: uac authentication in ser is not possible. Please correct me if I'm wrong (I honestly want to be wrong on this matter:)) Thank you, Michael On Sunday 15 May 2005 12:36 am, you wrote: > Michael Ulitskiy wrote: > > >Hello, > > > >Has anyone succeeded in getting UAC authentication to work? > >I'm doing the following: > > > >route { > > resetflag(1); > > t_on_failure("1"); > > route(1); > >} > > > >route[1] > >{ > > if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)") { > > sl_send_reply("479", "We don't forward to private IP addresses"); > > break; > > }; > > if (!t_relay()) { > > sl_reply_error(); > > }; > >} > > > >failure_route[1] { > > # authentication reply received? > > if ( t_check_status("401|407") ) { > > if (!isflagset(1) && uac_auth()) { > > setflag(1); > > t_on_failure("1"); > > append_branch(); > > route(1); > > } else { > > t_reply("500","Error occurred"); > > } > > break; > > } > > > >} > > > >When uac_auth() is called I get the following in the: > > 0(28973) DEBUG:uac:uac_auth: picked reply is (nil), code 407 > > 0(28973) BUG:uac:uac_auth: empty reply on picked branch > > > >Any suggestions or ideas? > >Thank you, > > > >Michael > > > > > > > I am also getting same thing with different msg .. > BUG:uac:uac_auth: empty reply on picked branch > > Anybody know how to overcome this? > > Mohammad > >

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Re: [SPAM] [Serusers] Re: Module programming question
by Daniel Corbe 23 May '05

23 May '05

Actually those were incredibly useful. Thank you. Now, when I compile my module I am getting the following error message: 16:18:13 root#proxy2:/usr/local/src/ser-0.9.0/modules/radius_cache>gcc -c -o radius_cache.o radius_cache.c In file included from radius_cache.c:46: radius_cache.h:58: `NAME' undeclared here (not in a function) radius_cache.h:58: syntax error before string constant radius_cache.h:62: warning: initialization from incompatible pointer type I'm not sure what to do about that. It seems to be something internal and specific to MODULE_VERSION in the SER header. My module: http://www.rafb.net/paste/results/N0uQyJ91.html My module's header file: http://www.rafb.net/paste/results/vo3vG345.html Thanks for the help. -Daniel On 5/23/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > There's a simple module, called print that is meant to be mainly an > example. So I recommend you to take a look at its interface and to start > from there your module. > I attacht also several files that a little bit old but still they are > really useful. > > Luck! > > Samuel. > > > Unclassified. > >>> Daniel Corbe <daniel.junkmail(a)gmail.com> 05/22/05 10:41PM >>> > Here's another question: > > I got my module to compile by taking _init out and reworking the > initalization of the module into something else. > > Now when SER loads I get: > > May 22 21:34:00 proxy2 ser: ERROR: no version info in module > </usr/local/lib/ser/modules/radius_cache.so>: Undefined symbol > "module_version" > > What is module_version? How do I define it? > > Thanks. > > -Daniel > > > On 5/22/05, Daniel Corbe <daniel.junkmail(a)gmail.com> wrote: > > Hello, > > > > I have a SER module which I've written in C and I'm trying to > compile. > > When I compile it I get this error: > > > > 20:53:24 > root#proxy2:/usr/local/src/ser-0.9.0/modules/radius_cache>gcc > > -shared -o radius_cache.so radius_cache.o > > radius_cache.o: In function `_init': > > radius_cache.o(.text+0x0): multiple definition of `_init' > > /usr/lib/crti.o(.init+0x0): first defined here > > > > Is there something else in SER that exports _init already? > > > > Thanks. > > > > -Daniel > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > >

1 0

[Serusers] Module programming question
by Daniel Corbe 23 May '05

23 May '05

Hello, I have a SER module which I've written in C and I'm trying to compile. When I compile it I get this error: 20:53:24 root#proxy2:/usr/local/src/ser-0.9.0/modules/radius_cache>gcc -shared -o radius_cache.so radius_cache.o radius_cache.o: In function `_init': radius_cache.o(.text+0x0): multiple definition of `_init' /usr/lib/crti.o(.init+0x0): first defined here Is there something else in SER that exports _init already? Thanks. -Daniel

2 2

[Serusers] Re: SIP load balancing cont..
by a c 23 May '05

23 May '05

this worked with t_check_status(). One thing I could not figure out is how to get a log printout when I receive a 200 OK. How do I know if I received a 200 OK? I can't put this logic in the main route block. thank you ac --- a c <lra101(a)yahoo.com> wrote: > > I need to route to a failure route when there is a > certain kind of failure. For example, If I don't > receive a 'Trying' back on the initial attempt{no > response to Invite}, then I would route using the > failure route. For all other failures {i.e. 404, no > answer, 486, etc..}, I don't try any other routes. > How > can this be configured? > > thank you > ac > > > > Discover Yahoo! > Have fun online with music videos, cool games, IM > and more. Check it out! > http://discover.yahoo.com/online.html > __________________________________ Do you Yahoo!? Make Yahoo! your home page http://www.yahoo.com/r/hs

1 0

[Serusers] session Timers
by Jorge Crichigno 23 May '05

23 May '05

Hi all, I have a a doubt related with RFC 4028 - Session Timers in the Session Initiation Protocol (SIP). I am developing a new ser module that supports this RFC. While doing tests, I found the following situation: phone1(addpac) SER phone 2 (grandstream) Invite -> <-422 (Lower Min-SE) Ack -> Invite -> Invite -> <-481 (no such call) <-481 (no such call) What I saw was that, in the second Invite, the addpac sends the Invite with the the "to tag" that was attached by SER before (422 response). I belive that this is the reason of the 481 response from grandstream. I think it is a bug of Addpac. Is that correct? Thanks in advance!

2 2

[Serusers] Bug in ucontact.c?
by Francesco Bottà 23 May '05

23 May '05

Hi all, I've a question about removing alias from memory and DB. I've a SER instance in WRITE_BACK DB Mode (db_mode=2) and timer_interval param set to 10 seconds. When I should remove an alias from memory&DB (with 'serctl rm alias 'username'), in WRITE_BACK mode, the expire time of this cancelled alias go to something like 'expires=-1116854859' and increase in time but never disappear from memory neither from my DB...indeed, if I set WRITE-THROUGH mode (db_mode=1) all works ok, and the cancelled alias disappears immediately from mem and DB.... I've this SER ver installed: ser -V version: ser 0.9.2 (i386/linux) flags: STATS: Off, EXTRA_DEBUG, 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 11:28:47 May 6 2005 with gcc 3.3 Any hints? What I'm missing? Many thanx Verbal

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Re: [Serusers] error in compile ser-0.9.0
by Kamran Ahmad 23 May '05

23 May '05

i am using bison-2.0.tar.gz and flex-2.5.4a.tar.gz but unable to solve this issue #ps -A 8961 ? 00:00:00 ser <defunct> 8962 ? 00:00:00 ser <defunct> 8965 ? 00:00:00 ser <defunct> 8966 ? 00:00:00 ser <defunct> 8968 pts/1 00:00:00 ps # serctl moni Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config --- Grigory Fishilevich <g.fishilevich(a)gmail.com> wrote: > hi, > > these messages have nothing to say. i think you have > a permissions > problem, you should take a look at /var/log/messages > > ? have you ser running or not? > __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

1 0

[Serusers] SUBSCRIBE -> BAD REQEST (Instant Communication Application + PA Module)
by Jaroslaw Gawron 23 May '05

23 May '05

Hello Everybody I am trying to create Instant Communication Application based on MS RTC 1.2 library. I am also using SER 0.8.14 as a SIP server. Voice communication works perfect, messages works perfect but I have a big problem with sending user's status. When I tray to send SUBSCRIBE message I get BAD REQEST response. T 10.1.1.62:3749 -> 10.1.3.10:5060 [AP] SUBSCRIBE sip:pszymanek@sippbx.bb.wasko.pl SIP/2.0 Via: SIP/2.0/TCP 10.1.1.62:8718 Max-Forwards: 70 From: "yarek" <sip:yarek@sippbx.bb.wasko.pl>;tag=d290de95c2124ceca42a748963dd04e9;epid=885 a5b4691 To: <sip:pszymanek@sippbx.bb.wasko.pl> Call-ID: fb0ac597ee2b480f9315e88a9e6bcb06(a)10.1.1.62 CSeq: 2 SUBSCRIBE..Contact: <sip:yarek@sippbx.bb.wasko.pl:8718;maddr=10.1.1.62;transport=tcp> User-Agent: RTC/1.2..Expires: 0 Event: presence..Content-Length: 0 # T 10.1.3.10:5060 -> 10.1.1.62:3749 [AP] SIP/2.0 400 Bad Request Via: SIP/2.0/TCP 10.1.1.62:8718 From: "yarek" <sip:yarek@sippbx.bb.wasko.pl>;tag=d290de95c2124 ceca42a748963dd04e9;epid=885a5b4691 To: <sip:pszymanek@sippbx.bb.wasko.pl>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-3fbe Call-ID: fb0ac597ee2b480f9315e88a9e6bcb06(a)10.1.1.62 CSeq: 2 SUBSCRIBE..Error while parsing headersServer: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 10.1.3.10:5060 "Noisy feedback tells: pid=13753 req_src_ip= 10.1.1.62 req_src_port=3749 in_uri=sip:pszymanek@sippbx.bb.wasko.pl out_uri=sip:pszymanek@sippbx.bb.wasko.pl via_cnt==1" I configured the presence module, added the lines to ser.cfg if (method=="SUBSCRIBE") { log(1, "Subscribe\n"); if (t_newtran()) { log(1, "Registrar\n"); handle_subscription("registrar"); }; break; }; But it still doesn't work. Has anyone faced a similar problem ? Why it doesn't work ? Is the PA module compatible with MS RTC. Why this masseges are in bad format. Any ideas are very welcome. Best regards Jarosław Gawron email: J.Gawron(a)wasko.pl

1 0

[Serusers] Append a header field with dynamic data
by Michael Stier 23 May '05

23 May '05

Hi, I want to append a new header field on a 183 - session progress response with dynamic data: P-hint: dynamic data The string "dynamic data" e. g. could be generated by an external script. Is there any way to reach this inside ser.cfg? My imagination would be something like this: if (status=~"183") { exec_msg('TEMPSTRING=`cat /tmp/string.txt`'); append_hf("P-hint: $TEMPSTRING"); }; Many greetings Michael

2 1

Re: [Serusers] sip load balancing
by a c 23 May '05

23 May '05

hello Iqbal, just need the SIP messages load balance across servers. No RTP. thank you ac --- Iqbal <iqbal(a)gigo.co.uk> wrote: > mediaproxy can do it using dns. Am assuming you want > the rtp stream load > balanced, if its the sip messages, search throuygh > the archives there is > along discussion bout howto use LVS and load > balancing....in short > doesnt seem to work, cause SIP uses udp > > Iqbal > > a c wrote: > > >hello, > > > > got a question on SER. Could SER do load balance > >across SIP servers? > > > >for example: > > > >SIP carrier --> SER --> Media/App Servers > {multiple} > > > >Are there any Configuration examples I can look at? > > > >thank you > >ac > > > > > > > >__________________________________ > >Yahoo! Mail Mobile > >Take Yahoo! Mail with you! Check email on your > mobile phone. > >http://mobile.yahoo.com/learn/mail > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > > >. > > > > > > > __________________________________ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250

8 13

Re: [Serusers] Advice needed
by Samuel Osorio Calvo 23 May '05

23 May '05

just an idea: does any PSTN provider supports TLS? If some does you have your auth problem solved with the TLS support from SER. Samuel Unclassified. >>> "Greger V. Teigre" <greger(a)teigre.com> 05/22/05 08:57AM >>> See inline. Michael Ulitskiy wrote: > On Saturday 21 May 2005 02:31 am, you wrote: >> I would say SER is what you need, except that you struggle with the >> authentication. You have the following scenarios: >> 1. PSTN termination with IP-based access control (easiest) >> 2. PSTN termination with authentication of all INVITEs (yes, that's >> the UAC module. You should contact the maintainer, Ramona-Elena >> Modroiu about the status. I thought it was reported to work, but >> haven't tried myself) >> 3. PSTN termination with registration and authentication of REGISTER >> (but not INVITEs). Use sipsak to generate a REGISTER for your box. >> >> #2 requires that all INVITEs are sent twice and is not a very good >> option. I would seek out PSTN providers who will give you #1. >> g-) > > UAC module doesn't work and I think won't work unless ser is made > call-statefull, 'cause it needs to adjust cseq within dialog. I > posted my findings to this list > several days ago (UAC module (backport to 0.9.0). Nobody replied so I > guess > nobody knows the way to make it work. I saw your post on serusers, yes, but not on serdev. Because you cannot make a module work, doesn't mean it doesn't work for all, so as I said, if you have found a bug, post it to serdev (preferably) or directly to the maintainer. That's the way open source software work... > As for ip auth I guess it's just not good enough. UDP invites don't > require any handshake it's not hard at all to spoof ip address. I > believe sending 2 invites worth the security it actually adds. Yes, but you can also do TCP. > Also I don't understand what you mean by #3. Taking ip address from > authenticated REGISTER and then doing IP auth on that? No, using sipsak to actually do a REGISTER on behalf of your ser. No IP auth, basically it makes your ser a registered client of the GW. Of course, if INVITEs still must be authenticated, you are back to the UAC module problem. g-) > Thanks, > > Michael > >> Michael Ulitskiy wrote: >>> Hello, >>> >>> I'd like ask for advice on what is in your opinion the best solution >>> in the following scenario. >>> I have a bunch of sip servers (asterisk boxes as my users need pbx >>> functionality) that can make sip call to each other and my PSTN >>> gateway. Now I want to purchase PSTN terminitaion in several >>> different markets (and probably more in the future). All those >>> terminations will require authentication. >>> I want all my boxes when they see non-local call to send it to a >>> central routing server that would determine where this call should >>> be sent and authenticate to the appropriate provider so that I don't >>> have to configure all credentials on all asterisk boxes. Also I want >>> it not to deal with the media at all. All media streams should go >>> directly from asterisk box to the PSTN termination provider. >>> So basically it should be central SIP router that is able to >>> authenticate calls if neccessary. >>> I thought I could do it with SER and its UAC module, but it appears >>> UAC module doesn't work and probably won't work (see my previous >>> post in this list about UAC backport to 0.9.0). >>> Also I don't want to use asterisk in this place as asterisk always >>> wants to stay in media path and I'd really like to avoid of getting >>> into hassle with re-invites. >>> So the question is what are my options and what you would advice >>> as a solution. Are there any software out there that can do it >>> (preferably open-source, of course) or what else you could suggest >>> to do to get desired results. >>> Thanks a lot, >> >> > > -- > See you later, > Michael > > > ------------------------------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: [Serusers] exec module question
by Samuel Osorio Calvo 23 May '05

23 May '05

Hi, I am not sure if SER config file get the error code from the exit() call of your script but I just think so. Assuming it gets the return code, youn should be aware that only three possibilities can be checked from the config file: ret>0 --> config command was successfull ret<0 --> ERROR but: ret=0 --> drop message!!! So, if you return 0 from your script, the message will be dropped, just af it a break had occur... To allow this structure: if (exec_msg("/usr/local/bin/check_register.pl")) { break; } else { if (!radius_www_authorize("")) { You should return a positive value if the scrpit ended OK or a negative value if an error occurred. BTW, using exec|* commands can be a worse solution.......just take a look at older posts about this subject. Hope it helps, Samuel. Unclassified. >>> Daniel Corbe <daniel.junkmail(a)gmail.com> 05/20/05 05:31PM >>> Hello, I've noticed that 90% of my SIP traffic is REGISTER requests. I am trying to implement some sort of caching mechanism (which will inherently be faster than doing a RADIUS lookup on every REGISTER request) using exec and a perl script which does a lookup from a memory-cached file. I'm using exec_msg() to call the perl script; however it is not behaving as I would expect. Here's the code snippet from my SER config. if (exec_msg("/usr/local/bin/check_register.pl")) { break; } else { if (!radius_www_authorize("")) { I am just in preliminary testing stage so I have the perl script merely exiting with 0 status. exec documentation is not clear as to whether it uses the shell return codes to determine the exit status of exec_msg If I am exit(0)ing from the perl script I would expect the if( to fail, not break and do the radius_www_authorize. Is that not correct? Thanks for the help! Regards, Daniel Corbe _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] Why does the memory occupied by SER process not decrese?
by lemon 23 May '05

23 May '05

I am using ser 0.8.14. I wrote a program to test the stability of SER. The scenario like this caller----SER----callee. Caller program will send INVITE continuously.Callee program answer each call proxyed by SER. And then Caller program send BYE to finish the call. I run 'top' tool in linux to monitor the usage of memory and cup by SER. I found that with the call establishing speed increasing, the memory occupied by SER process increasing too.After I shut down the test program, the memory ocupied by SER process didn't decrease. Can anyone tell me why? And Does is mean memory leak? Best regards! Karman Wu ---------------------------------------------- 21CNÓÊÏäÈ«ÃæÉýG http://mail.21cn.com/huodong/0504/mail/ ²ÊÆÁÊÖ»ú°×ËÍÀ²!žÏ¿ìÀŽÄÃ! http://qipai.g.21cn.com ÖÇ»ÛŽóÌôÕœŸÍÔÚµÍŒÛ¶á±Š http://super.21cn.com/ ÌåÑéÁíÀàÔŒ»á£¬žÐÊÜ±ðÑùÀËÂþ http://y.21cn.com/club/ Ð¡ÁéÍš¶ÌÐÅÖÐÐÄ£¬¶ÌÐÅ÷ÈÁŠÎÞŒ« http://pas.21cn.com/ È«ÄÜÁÄ2005°æÉÁÁÁµÇ³¡£¡ http://callme.21cn.com

1 0

[Serusers] call PSTN -> SIP
by Kostas Marneris 23 May '05

23 May '05

Hello, I want to make a call from PSTN to a SIP phone. I use a cisco 5300 for answering/handling the PSTN number and then forward the call to a SER box (0.8.14) . POTS -> c5300 --> SER -> SIP softphone My conf on c5300: ! dial-peer voice 1003 voip destination-pattern 12345 session protocol sipv2 session target sip-server codec g711ulaw ! sip-ua sip-server ipv4:X.Y.Z.W ! What is the appropriate configuration on ser.cfg in order to handle this call and forward it to sip:userX@mydomain.com ?? Any hint about this ?? I've already done the opposite scenario (SIP->PSTN) : ser.cfg: if (uri =~ "^sip:2*") { rewritehostport("1.2.3.4:5060"); forward(uri:host, uri:port); break; }; thanks for any help, Kostas --

3 2

Re: [Serusers] error in compile ser-0.9.0
by Kamran Ahmad 23 May '05

23 May '05

how to solve this issue i am root ser is working but #ps -A 3793 ? 00:00:00 ser <defunct> --- Grigory Fishilevich <g.fishilevich(a)gmail.com> wrote: > hi, > > these messages have nothing to say. i think you have > a permissions > problem, you should take a look at /var/log/messages > > ? have you ser running or not? > __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

1 0

[Serusers] Basic Billing For SER 0.8.14 (MySql Based)
by Ozan Blotter 23 May '05

23 May '05

Does anyone know a free or lowcost billing solution which works with SER, before i've tried with some commercials ones but they do not worked with SER and they were Radius Supported ones so was creating its own database. All i need is a few simple PHP files so i can add/remove users from there and a add table for rating depending on prefix and it should use SER's database which is already created. And do sample calculations for me, enter user id/loginname/callerid or whatever and show me caller/callee/duration/time/date/cost and under all a grand total with tax space. So simple but i do not know how to do in PHP. Even some examples or good URL's to read are fine for me. I use SER 0.8.14 but may use 0.9.0 or newer if needed. Thanks, Ozan Blotter

4 3

[Serusers] admin errors in serweb
by Llanos Serna García-Conde 22 May '05

22 May '05

1 0

[Serusers] ser/mysql configuration
by Kim Culhan 22 May '05

22 May '05

After installing mysql 5.0.3 and running ser_mysql.sh then mysql> select * from user; Returns: ERROR 1146 (42S02): Table 'ser.user' doesn't exist Any help is grealty appreciated -kim -- w8hdkim(a)gmail.com

2 1

[Serusers] error in compile ser-0.9.0
by Kamran Ahmad 22 May '05

22 May '05

hi list any one know what could be the problem here is the trace i cannot use serctl moni Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config make_and_install_output.txt -------------------------------- Makefile.rules:81: action.d: No such file or directory Makefile.rules:81: crc.d: No such file or directory Makefile.rules:81: daemonize.d: No such file or directory Makefile.rules:81: data_lump.d: No such file or directory Makefile.rules:81: data_lump_rpl.d: No such file or directory Makefile.rules:81: dprint.d: No such file or directory Makefile.rules:81: dset.d: No such file or directory Makefile.rules:81: error.d: No such file or directory Makefile.rules:81: fifo_server.d: No such file or directory Makefile.rules:81: flags.d: No such file or directory Makefile.rules:81: forward.d: No such file or directory Makefile.rules:81: hash_func.d: No such file or directory Makefile.rules:81: ip_addr.d: No such file or directory Makefile.rules:81: main.d: No such file or directory Makefile.rules:81: md5.d: No such file or directory Makefile.rules:81: md5utils.d: No such file or directory Makefile.rules:81: modparam.d: No such file or directory Makefile.rules:81: msg_translator.d: No such file or directory Makefile.rules:81: pass_fd.d: No such file or directory Makefile.rules:81: proxy.d: No such file or directory Makefile.rules:81: qvalue.d: No such file or directory Makefile.rules:81: re.d: No such file or directory Makefile.rules:81: receive.d: No such file or directory Makefile.rules:81: resolve.d: No such file or directory Makefile.rules:81: route.d: No such file or directory Makefile.rules:81: route_struct.d: No such file or directory Makefile.rules:81: script_cb.d: No such file or directory Makefile.rules:81: socket_info.d: No such file or directory Discover Yahoo! Stay in touch with email, IM, photo sharing and more. Check it out! http://discover.yahoo.com/stayintouch.html

3 2

[Serusers] uac module for ser-0.9.0
by Mohammad Khan 22 May '05

22 May '05

Hello, would anybody please tell me where can I get right version of uac module for ser-0.9.0. I downoaded cvs snapshot of ser-0.10.99 can I use uac of this snapshot on ser-0.9.0? Mohammad

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Re: [Serusers] javarockx (Paul) 's ser.cfg
by Dave 22 May '05

22 May '05

Thank you much !! --- Java Rockx <javarockx(a)gmail.com> wrote: > Sorry all for the repost but the listserv scrubbed > my attachment. Here it is > again. > > P > > # ------------- Initial global variables > > debug=3 > fork=yes > log_stderror=no > > listen=10.3.0.221 <http://10.3.0.221> > port=5060 > children=4 > server_signature=no > > # hostname matching an alias will satisfy the > condition uri==myself" > alias=sip.mycompany.com <http://sip.mycompany.com> > > # dns - Uses dns to check if it is necessary to add > a "received=" field > # to a via. Default is no. > # rev_dns - Same as dns but use reverse DNS. > dns=no > rev_dns=no > > # check_via - Turn on or off Via host checking when > forwarding replies. > # Default is no. arcane. looks for discrepancy > between name and > # ip address when forwarding replies. > check_via=no > > # syn_branch - Shall the server use stateful synonym > branches? It is > # faster but not reboot-safe. Default is yes. > syn_branch=yes > > # memlog - Debugging level for final memory > statistics report. Default > # is L_DBG -- memory statistics are dumped only if > debug is set high. > memlog=1000 > > # sip_warning - Should replies include extensive > warnings? > # By default yes, it is good for trouble-shooting. > sip_warning=no > > # fifo - FIFO special file pathname > fifo_mode=0666 > fifo="/tmp/ser_fifo" > fifo_db_url="mysql://ser@localhost/ser" > > # reply_to_via - A hint to reply modules whether > they should send reply > # to IP advertised in Via. Turned off by default, > which means that > # replies are sent to IP address from which requests > came. > reply_to_via=no > > # user | uid - uid to be used by the server. 99 = > nobody. > #uid="nobody" > > # group | gid - gid to be used by the server. 99 = > nobody. > #gid="nobody" > > # mhomed -- enable calculation of outbound > interface. > # useful on multihomed servers. > mhomed=0 > > # ------------- external module loading > 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/acc.so" > loadmodule "/usr/local/lib/ser/modules/group.so" > loadmodule "/usr/local/lib/ser/modules/print.so" > loadmodule "/usr/local/lib/ser/modules/textops.so" > loadmodule > "/usr/local/lib/ser/modules/mediaproxy.so" > loadmodule "/usr/local/lib/ser/modules/nathelper.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/speeddial.so" > loadmodule "/usr/local/lib/ser/modules/options.so" > loadmodule "/usr/local/lib/ser/modules/avpops.so" > loadmodule > "/usr/local/lib/ser/modules/permissions.so" > loadmodule "/usr/local/lib/ser/modules/xdz_tools.so" > > # ------------ nathelper settings > modparam("nathelper", "rtpproxy_disable", 1) > modparam("nathelper", "natping_interval", 0) > > # ------------ media proxy settings > modparam("mediaproxy", "natping_interval", 30) > modparam("mediaproxy", "mediaproxy_socket", > "/var/run/mediaproxy.sock") > modparam("mediaproxy", "sip_asymmetrics", > "/usr/local/etc/ser/sip-asymmetric-clients") > modparam("mediaproxy", "rtp_asymmetrics", > "/usr/local/etc/ser/rtp-asymmetric-clients") > > # ------------- db_url setting > modparam("acc|auth_db|domain|group|lcr|permissions|speeddial|uri_db|usrloc|xdz_tools", > "db_url", "mysql://ser@localhost/ser") > > # ------------- use_domain setting > modparam("auth_db|group|registrar|speeddial|uri_db|usrloc", > "use_domain", 0) > > # ------------- permissions settings > modparam("permissions", "db_mode", 1) > modparam("permissions", "trusted_table", "trusted") > > # ------------- accounting parameters > modparam("acc", "db_flag", 1) > modparam("acc", "db_missed_flag", 1) > modparam("acc", "log_fmt", "cdfimorstup") > modparam("acc", "log_level", 1) > modparam("acc", "failed_transactions", 1) > modparam("acc", "report_cancels", 1) > modparam("acc", "report_ack", 0) > > # allows clear text passwords in the mysql database > modparam("auth_db", "calculate_ha1", 1) > modparam("auth_db", "password_column", "password") > modparam("auth_db", "avps_column_int", "domn") > modparam("auth_db", "avps_column_str", "uuid") > > # ------------- domain parameters > modparam("domain", "db_mode", 1) > > # ------------- registration parameters > modparam("registrar", "nat_flag", 6) > modparam("registrar", "min_expires", 300) > modparam("registrar", "max_expires", 3600) > modparam("registrar", "default_expires", 600) > modparam("registrar", "default_q", 100) > modparam("registrar", "desc_time_order", 1) > modparam("registrar", "append_branches", 0) > > # ------------- rr parameters > modparam("rr", "enable_full_lr", 0) > > # ------------- tm parameters > modparam("tm", "fr_timer", 22) > modparam("tm", "fr_inv_timer", 27) > modparam("tm", "wt_timer", 10) > modparam("tm", "fr_inv_timer_avp", "inv_timeout") > > # ------------- usrloc parameters > modparam("usrloc", "db_mode", 1) > modparam("usrloc", "timer_interval", 30) > modparam("usrloc", "desc_time_order", 1) > > # -------------- AVPOPS parameters > modparam("avpops", "avp_url", > "mysql://ser@localhost/ser") > modparam("avpops", "avp_table", "usr_preferences") > modparam("avpops", "avp_aliases", "voicemail=i:500") > > route { > > # > ------------------------------------------------------------------------ > # Sanity Check Section > # > ------------------------------------------------------------------------ > if (!mf_process_maxfwd_header("10")) { > log(1, "LOOP DETECTED"); > sl_send_reply("483", "Too Many Hops"); > break; > }; > > if (msg:len > max_len) { > log(1, "MESSAGE OVERFLOW"); > sl_send_reply("513", "Message Too Large"); > break; > }; > > # > ------------------------------------------------------------------------ > # Click2Dial Test > # > ------------------------------------------------------------------------ > if (src_ip==myself && (search("^From: <sip:700@") || > search("^To: === message truncated === __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] javarockx (Paul) 's ser.cfg
by Dave 22 May '05

22 May '05

Gentlemen: I saw a posting from Javarockx http://lists.iptel.org/pipermail/serusers/2005-February/015687.html mentioning that he wants to repost his reworked ser.cfg. My search did not give me any positive results. Can someone (Paul?) please email it to the list? Thanks, Dave __________________________________ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250

2 1

Re: [Serusers] Advice needed
by Michael Ulitskiy 21 May '05

21 May '05

Can it do what I described? Documentation isn't very good on that web site, but according to what I saw I doubt it. Also what does it mean SER based PBX? SER+voicemail? Thanks, Michael On Friday 20 May 2005 07:20 pm, you wrote: > > we have an all in one ser solution that might be usefull for you > http://www.wifi.com.ar/english/voip.html > > regards, > > ____________________________________________________________________________ > > Jaime Garcia Ghirelli http://www.brujula.net > jaime(a)fonosip.com http://www.fonosip.com > http://www.wifi.com.ar > > ---------- Forwarded message ---------- > Date: Fri, 20 May 2005 15:49:35 -0400 > From: Michael Ulitskiy <mdu113(a)acedsl.com> > To: serusers(a)lists.iptel.org > Subject: [Serusers] Advice needed > > Hello, > > I'd like ask for advice on what is in your opinion the best solution > in the following scenario. > I have a bunch of sip servers (asterisk boxes as my users need pbx > functionality) that can make sip call to each other and my PSTN > gateway. Now I want to purchase PSTN terminitaion in several > different markets (and probably more in the future). All those > terminations will require authentication. > I want all my boxes when they see non-local call to send it to a > central routing server that would determine where this call should > be sent and authenticate to the appropriate provider so that I don't > have to configure all credentials on all asterisk boxes. Also I want > it not to deal with the media at all. All media streams should go directly > from asterisk box to the PSTN termination provider. > So basically it should be central SIP router that is able to authenticate > calls if neccessary. > I thought I could do it with SER and its UAC module, but it appears > UAC module doesn't work and probably won't work (see my previous > post in this list about UAC backport to 0.9.0). > Also I don't want to use asterisk in this place as asterisk always wants to > stay in media path and I'd really like to avoid of getting into hassle with > re-invites. > So the question is what are my options and what you would advice > as a solution. Are there any software out there that can do it (preferably > open-source, of course) or what else you could suggest to do to get > desired results. > Thanks a lot, > -- > See you later, > Michael > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > -- See you later, Michael

2 3

[Serusers] is serweb mature enough ?
by Lucas Aimaretto 21 May '05

21 May '05

Hi all, Is anybody using serweb ? Is it mature ?? Do you guys recomend using it ? What can I do with serweb that I can not do with the console ? And what can I do with the console that i can not with serweb ? best regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.13 - Release Date: 19/05/2005

2 1

[Serusers] semget failed
by Jorge Crichigno 21 May '05

21 May '05

Hello, I have a problem when starting SER 0.9.0 an a linux ppc64. The trace is as follow: ... unixsock_register_cmd: New command (t_uac_cancel) registered 0(0) unixsock_register_cmd: New command (t_hash) registered 0(0) unixsock_register_cmd: New command (t_reply) registered 0(0) DEBUG: lock_initialize: lock initialization started 0(0) ERROR: lock_set_init (SYSV): semget (..., 251, 0700) failed: Invalid argument 0(0) DEBUG: lock_initialize: entry semaphore initialization failure: Invalid argument 0(0) DEBUG: lock_initialize: first time semaphore allocation failure 0(0) INFO: semaphore arrays of size 250 allocated ... This error was already reported on april, 2005 and Jan Janak suggested changing SEMMSL value in include files. I did that, but the error continues... Do someone know something about this problem? Thank you!

2 1

[Serusers] Solaris 10 box with SER 0.8.12
by ivan veizaga 21 May '05

21 May '05

Hello Can anybody please help me with the following problem: I have a Solaris 10 (x86) box with SER 0.8.12 preinstalled by default, when I run the SER with the default configuration file (ser.cfg.sample) everything works well (even with MySQL for authentication), but when I try to load the acc.so module appears the following error: # Mar 31 16:48:17 enterprise ser[14579]: ERROR: load_module: could not open module </usr/sfw/lib/ser/modules/acc.so>: ld.so.1: /usr/sfw/sbin/ser: fatal: relocation error: file /usr/sfw/lib/ser/modules/acc.so: symbol set_timer: referenced symbol not found Mar 31 16:48:17 enterprise ser[14579]: parse error (87,13-45): failed to load module Thank you very much for your time. Sorry if my english is bad. Regards. Ivan. _________________________________________________________ Do You Yahoo!? Información de Estados Unidos y América Latina, en Yahoo! Noticias. Visítanos en http://noticias.espanol.yahoo.com

2 1

[Serusers] exec module question
by Daniel Corbe 21 May '05

21 May '05

Hello, I've noticed that 90% of my SIP traffic is REGISTER requests. I am trying to implement some sort of caching mechanism (which will inherently be faster than doing a RADIUS lookup on every REGISTER request) using exec and a perl script which does a lookup from a memory-cached file. I'm using exec_msg() to call the perl script; however it is not behaving as I would expect. Here's the code snippet from my SER config. if (exec_msg("/usr/local/bin/check_register.pl")) { break; } else { if (!radius_www_authorize("")) { I am just in preliminary testing stage so I have the perl script merely exiting with 0 status. exec documentation is not clear as to whether it uses the shell return codes to determine the exit status of exec_msg If I am exit(0)ing from the perl script I would expect the if( to fail, not break and do the radius_www_authorize. Is that not correct? Thanks for the help! Regards, Daniel Corbe

3 2

[Serusers] VoIp Billing Solution SIP server compatible!!!
by Ollie Gejevski 21 May '05

21 May '05

Hey guys, I notice quite a few people are looking for a Billing solution. Please contact me as I can help. We already have few companies from this board use our solution with great success. We are Cisco as well as SIP server ( SER, SNOM ) compatible. Check out our website HYPERLINK "http://www.cyneric.com"www.cyneric.com Please contact me HYPERLINK "mailto:ollie.g@cyneric.com"ollie.g(a)cyneric.com for more information. All the best Ollie -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.13 - Release Date: 5/19/2005

2 1

[Serusers] Postgres does not compile with GCC 4.0.x
by Boris 21 May '05

21 May '05

After downloading the latest SER from CVS.. .It seems when trying to compile the postgres module with GCC 4.0.x and Glibc 2.3.5, I get the following error gcc -fPIC -DPIC -g -O -funroll-loops -Wcast-align -Wall -DNAME='"ser"' -DVERSION='"0.10.99-dev6"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.0"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_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/pgsql/include -I/usr/include/postgresql -c dbase.c -o dbase.o dbase.c: In function 'db_init': dbase.c:245: error: invalid lvalue in assignment make: *** [dbase.o] Error 1 Any ideas on how to fix this? I looked at the source on line 245 and it seems that one line is trying to do multiple things. Any ideas on how to fix this?

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Re: [Serusers] Advice needed
by Greger V. Teigre 21 May '05

21 May '05

BTW, the UAC doc is here (and contact info for maintainers): http://www.voice-system.ro/docs/uac/ g-) Greger V. Teigre wrote: > I would say SER is what you need, except that you struggle with the > authentication. You have the following scenarios: > 1. PSTN termination with IP-based access control (easiest) > 2. PSTN termination with authentication of all INVITEs (yes, that's > the UAC module. You should contact the maintainer, Ramona-Elena > Modroiu about the status. I thought it was reported to work, but > haven't tried myself) > 3. PSTN termination with registration and authentication of REGISTER > (but not INVITEs). Use sipsak to generate a REGISTER for your box. > > #2 requires that all INVITEs are sent twice and is not a very good > option. I would seek out PSTN providers who will give you #1. > g-) > > > Michael Ulitskiy wrote: >> Hello, >> >> I'd like ask for advice on what is in your opinion the best solution >> in the following scenario. >> I have a bunch of sip servers (asterisk boxes as my users need pbx >> functionality) that can make sip call to each other and my PSTN >> gateway. Now I want to purchase PSTN terminitaion in several >> different markets (and probably more in the future). All those >> terminations will require authentication. >> I want all my boxes when they see non-local call to send it to a >> central routing server that would determine where this call should >> be sent and authenticate to the appropriate provider so that I don't >> have to configure all credentials on all asterisk boxes. Also I want >> it not to deal with the media at all. All media streams should go >> directly from asterisk box to the PSTN termination provider. >> So basically it should be central SIP router that is able to >> authenticate calls if neccessary. >> I thought I could do it with SER and its UAC module, but it appears >> UAC module doesn't work and probably won't work (see my previous >> post in this list about UAC backport to 0.9.0). >> Also I don't want to use asterisk in this place as asterisk always >> wants to stay in media path and I'd really like to avoid of getting >> into hassle with re-invites. >> So the question is what are my options and what you would advice >> as a solution. Are there any software out there that can do it >> (preferably open-source, of course) or what else you could suggest to >> do to get desired results. >> Thanks a lot,

1 0

[Serusers] step by step ser 0.9.0 Please Help.... Urgent...
by Bharat More 21 May '05

21 May '05

Hi All ... I want to install ser 0.9.0 MYSQL and redius for AAA purpose.Can anybody tell me step by step workout for this. Also the media proxy and the useragent addition....and then how to make the call from mediaproxy to desktop sip phone.... Thanks in advance for any hint... Bharat. ________________________________________________________________________ Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony

2 1

RE: [Serusers] Mediaproxy in LAN (?)
by Giudice, Salvatore 21 May '05

21 May '05

Can anyone give an example of a ser.cfg that forces all connections through mediaproxy? -----Original Message----- From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Wednesday, May 18, 2005 12:52 AM To: Giudice, Salvatore; bogdan aux; serusers(a)lists.iptel.org Subject: Re: [Serusers] Mediaproxy in LAN (?) Have a look at the Getting Started document at http://onsip.org/ Chapter 1 explains the basics around NATing and the corresponding functions to use. g-) Giudice, Salvatore wrote: > I am trying to route a call from a non-routable 10.x.x.x network to a > phone on the internet using ser 0.9.0 and mediaproxy 1.31. Aka: PSTN > to SIP from my asterisk to an x-lite > > I found some references indicating that I should call: > > > if ( src_ip = IP_GATEWAY) > { > force_rport(); > fix_contact(); > use_media_proxy(); > }; > > Supposedly, you need the force_rport and the fix_contact to put media > proxy in the middle of the two rtp sessions, but I have et to see this > actually work. Has anyone been able to pass a call through media proxy > from a GW on a non-routable to a routable network, or even between two > ip's on the same network as the mediaproxy? > > If so, please share your configs. > > -----Original Message----- > From: bogdan aux [mailto:aux1d@yahoo.com] > Sent: Friday, May 13, 2005 6:16 AM > To: serusers(a)lists.iptel.org > Subject: [Serusers] Mediaproxy in LAN (?) > > I want to route RTP packets between 2 computers in the > same network through Mediaproxy. I know this may have > no sense, but it's just a test configuration. > The call use_media_proxy() seems to have no effect, > the RTP packets go directly from one computer to > another. > My guess is mediaproxy makes a test on the IPs (they > are 192.168.147.2 and 192.168.147.3) and refuses to > proxy the call. > Is mediaproxy working just between different networks > or it's an error in my configuration file? > > > > > Yahoo! Mail > Stay connected, organized, and protected. Take the tour: > http://tour.mail.yahoo.com/mailtour.html > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] problem with ngrep utility
by Timur Elzhov 21 May '05

21 May '05

Dear SER experts, I'd like to capture the 'REGISTER' messages with 'ngrep' utilty. I use ngrep from the Debian distribution, with version # ngrep -V ngrep: V1.43, $Revision: 1.66 $ So, I launch ngrep and then launch "X-Lite" client on the another Windows platform. X-Lite said that it is successfully logged in, however "ngrep" waiting meanwhile on SIP server .. # ngrep REGISTER port 5060 interface: eth0 (159.93.16.0/255.255.240.0) filter: ip and ( port 5060 ) match: REGISTER exit 258 received, 0 dropped .. trapped nothing. Where am I mistaken? Much thanks. -- WBR, Timur V. Elzhov

4 3

[Serusers] SIP load balancing cont..
by a c 20 May '05

20 May '05

I need to route to a failure route when there is a certain kind of failure. For example, If I don't receive a 'Trying' back on the initial attempt{no response to Invite}, then I would route using the failure route. For all other failures {i.e. 404, no answer, 486, etc..}, I don't try any other routes. How can this be configured? thank you ac Discover Yahoo! Have fun online with music videos, cool games, IM and more. Check it out! http://discover.yahoo.com/online.html

1 0

[Serusers] Pound Prefix
by Matt morris 20 May '05

20 May '05

Hello List, Got a question about the uri matching, I used a #0 prefix for calls but I could find the correct syntax for ser.cfg to recognize the "#" key. I have two different brands of hardware sip phones, one of them will send the # key as %23 (the syntax I used has no problem picking this). However, the other sends out the exact "#" Here are the syntax I have tried: 1) if (uri=~"^sip:#[0-9]+@") { 2) if (uri=~"^sip:\#[0-9]+@") { 3) if (uri=~"^sip:[#]{1}[0-9]+@") { Could someone inform me the correct syntax? TIA _________________________________________________________________ Don't just Search. Find! http://search.sympatico.msn.ca/default.aspx The new MSN Search! Check it out!

2 1

Re: [Serusers] where is my /etc/profile?
by Gerard Wagener 20 May '05

20 May '05

Hi, When you are working with BASH shell the profile is normally located in /etc/profile. When you can't find it and you don't want to read documentation you can start your shell with strace and look which files are read. For bash shell you type strace -e open bash --login. Regards, G�rard --- Bharat More <bharat_sharp(a)yahoo.co.in> wrote: > Hi all... > I has installed ser 8.12.0, ser is > running,and i exported my domain as novanet.co.in > its > gone successfully. but the next step is to export > the > domain in etc/profile file but where it is? Its > supposed to be in the folder etc/profile but there > is > nthing in this folder?... > Need urgent Help...Thanks for any hint in > Adv. > > > > ________________________________________________________________________ > Yahoo! India Matrimony: Find your life partner > online > Go to: http://yahoo.shaadi.com/india-matrimony > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html

1 0

[Serusers] Presence agent: not responding to update in usrloc
by Johan Bilien 20 May '05

20 May '05

Hi, I'm trying to setup the pa module in ser-0.9.0. I am using: modparam("pa", "pa_domain", "bilien.org") modparam("pa", "use_db", 1) modparam("pa", "db_url", "mysql://ser:heslo@localhost/ser") then in the route(): if (uri==myself) { if (method=="SUBSCRIBE") { if (t_newtran()) { handle_subscription("registrar"); break; }; }; This seems to work fine, as a SUBSCRIBE message results in May 20 21:09:08 coruscant /usr/sbin/ser[18496]: handle_subscription() entered May 20 21:09:08 coruscant /usr/sbin/ser[18496]: get_pres_uri: _puri=sip:foo@bar May 20 21:09:08 coruscant /usr/sbin/ser[18496]: new_watcher starting May 20 21:09:08 coruscant /usr/sbin/ser[18496]: new_watcher: _p->uri=johan-xten(a)bilien.org May 20 21:09:08 coruscant /usr/sbin/ser[18496]: db_new_watcher: watcher->uri=johan-kphone(a)bilien.org May 20 21:09:08 coruscant /usr/sbin/ser[18496]: new_watcher: watcher->package=presence May 20 21:09:08 coruscant /usr/sbin/ser[18496]: new_watcher: status=active in the logs. But when I then register with foo@bar, nothing happens. I don't get any NOTIFY, nor do I get something in the logs. I can recompile and add some printf, see if the pa callback is registered and called in usrloc, but I thought I'd first ask here if someone sees what the problem is. Thanks, Johan.

1 0

[Serusers] nagios monitoring remote rtpproxy
by Jev 20 May '05

20 May '05

Hi all, Here's a very quick and easy way to check if a rtpproxy is up, and responding on it's UDP command channel using the check_udp nagios utility: /usr/local/libexec/nagios/check_udp -e "123 " -s "123 V" -H your.rtp.host.com -p 22222 Note, this only works when you make rtpprxy listen on a UDP port, you have to find another method to monitor rtpproxy when it listens on a local socket. -Jev

1 0

[Serusers] SER module and pthreads
by Greg DuPertuis 20 May '05

20 May '05

I've written a module that uses a 3rd party library to retrieve information from a remote database. This library is multi-threaded using pthreads. SER hangs on startup when running in forked mode. Works fine is non-fork mode. Anyone experience such a problem? Thanks in advance.

2 1

[Serusers] MySQL system variables for ACC
by Mik Cheez 20 May '05

20 May '05

I'm wondering if anybody has an optimized configuration for mysql in an environment where several SER boxes are sending A LOT of traffic to one central server. The most notable thing I've done so far is increase the number of max-connections.

1 0

[Serusers] SIP notify via sipsak, too many hops loop
by Iqbal 20 May '05

20 May '05

Hi I am trying to send NOTIFY sip:040600@sip.domain.com SIP/2.0 Via: SIP/2.0/UDP a.b.c.d:5060 From: <sip:vmserver@sip.domain.com> To: <sip:040600@sip.domain.com> Contact: <sip:040600@a.b.c.d> Call-ID: 4d61cf9e505d40e905032a18329d61ec(a)a.b.c.d CSeq: 1 NOTIFY User-Agent: VoiceMail Event: message-summary Content-Type: application/simple-message-summary Content-Length: 40 Messages-Waiting: yes Voicemail: 23/2 using if (src_ip == a.b.c.d) { log(1, "INSIDE ROUTE 2"); if (!t_relay()) { sl_reply_error(); break; }; }; in my NOTIFY route however I keep getting too many hops. Should sipsak send the message to ser, or should it go direct to the IPphone, I am guessing at the former. Iqbal

2 6

[Serusers] Problem when Uncomment loadmodule "/usr/local/lib/ser/modules/auth_db.so" digest authentication
by mohammed irfan 20 May '05

20 May '05

Hello, we are using ser version 0.8.14 and the script ser.cfg is working fine without database. But after enabling write to database mode, and digest authentication we get one way request and no response from the server. We variefied this is happening when we uncomment loadmodule "/usr/local/lib/ser/modules/auth_db.so" I am sending U the complete script which we are using, please help us to solve our problem. #################################################### #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) # Uncomment these lines to enter debugging mode debug=10 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 #alias="localhost" 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" # 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_db.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", "db_url", "sql://ser:heslo@localhost:3306/ser") modparam("auth_db", "calculate_ha1", 1) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db","user_column", "username") modparam("auth_db", "password_column", "password") modparam("auth_db","domain_column","domain") # -- rr params -- # add value to ;lr param to make some broken UAs happy # 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 > max_len ) { 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); 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 (!www_authorize("192.168.1.101", "subscriber")) { www_challenge("192.168.1.101", "1"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } #################################

1 0

[Serusers] Filtering redirections
by MC 20 May '05

20 May '05

Hi, I would like to deny all 3xx messages coming from subscribers. I found following code in seusers mailing list but my ser give me: "unknown command, missing loadmodule? ". What is wrong? failure_route[1] { if (t_check_status("3[0-9][0-9]")) { t_reply("400","No redirect available"); break; } } I have following modules loaded: loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/acc.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_radius.so" SER version 0.8.12 Regards, MC

1 0

[Serusers] avpops - avp_pushto bug - maybe a do_action bug too.
by Marco Figus 20 May '05

20 May '05

Using avpops to change ruri/domani I found a strange behaviour: If you try to modify the ruri/domain of a previously received msg using avp_push to, you can't handle with the port. Suppose to receive a msg with ruri: name@domain:port and you wanna change it to name@newdomain:newport it seems to be impossible. if you do something like: avp_write("newdomain","s:_newdomain"); avp_pushto("$ruri/domain","s:_newdomain"); the result is: name@newdomain:port the do_action method just change the domain using the SET_HOST_T action type and leaves the final part of the string unchanged!! But I wanna change it too setting the newport value. A possible solution could be to recall the do_action wiht action SET_PORT_T with an empty value to reset the :port in the ruri field of the received domain, so I can set entire new domain using: avp_write("newdomain:newport","s:_newdomain"); avp_pushto("$ruri/domain","s:_newdomain"); having as a result: name@newdomain:newport and not name@newdomain:newport:port as happens now. Or (the best way I think) add a new value for the first parameters in method avp_pushto(destination, name) that allows to have the domainport as destination (invoking do_action whit SET_PORT_T action). I'm disposable to work on it, let me know. Marco.

1 0

[Serusers], [Semsusers] sems pstn gw with ser, bad voice stream quality
by Grigory Fishilevich 20 May '05

20 May '05

hi guys, i have running ser with sems as pstngw (eicon diva pri card). both product are cvs snapshots, downloaded few days ago. calls from sip to pstn have bad quality, some packets be dropped. has anybody experiences with such environment? sipcalls between serusers have a very good voice quality. i think it caused by sems. bg, Grigory Fishilevich

1 0

[Serusers] Password mysql changed
by Sandra Donaire 20 May '05

20 May '05

Hi!! I change the default password for the mysql database, an I change the SerWeb conf and runs ok, but wuehn I try restar the server, it's fail in db_connect. I can't find where its defined tha password for the module. Can somebody help me? Thanks. -- Sandra Donaire Arroyo Telefónica I+D División de "Tecnologías de acceso a redes IP"

3 3

[Serusers] where is my /etc/profile?
by Bharat More 20 May '05

20 May '05

Hi all... I has installed ser 8.12.0, ser is running,and i exported my domain as novanet.co.in its gone successfully. but the next step is to export the domain in etc/profile file but where it is? Its supposed to be in the folder etc/profile but there is nthing in this folder?... Need urgent Help...Thanks for any hint in Adv. ________________________________________________________________________ Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony

1 0

[Serusers] UA sending 302 Moved Temp message
by Lucas Aimaretto 19 May '05

19 May '05

Hi all, I'm getting a rare message from an UA when receivin an INVITE. The UA replies with a SIP/2.0 302 Moved Temporarily. Any ideas of what does it mean ?? How to solve it ?? This UA can make calls anywhere, but can not receive a call at all. Regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.12 - Release Date: 17/05/2005

2 1

RE: [Serusers] SER b2bua agent (+more)?
by Wojciech Ziniewicz 19 May '05

19 May '05

so ok.. it,s my phones case since they are rather old and cheap :) > -----Original Message----- > From: Iqbal [mailto:iqbal@gigo.co.uk] > Sent: Thursday, May 19, 2005 8:32 PM > To: Wojciech Ziniewicz > Cc: 'serusers(a)lists.iptel.org' > Subject: Re: [Serusers] SER b2bua agent (+more)? > > > ur sipphones should be deciding the RTP port, u can usually > set them to > pick a random port from a range. > > iqbal > > Wojciech Ziniewicz wrote: > > >questions inline > > > > > > > >>-----Original Message----- > >>From: Iqbal [mailto:iqbal@gigo.co.uk] > >>Sent: Thursday, May 19, 2005 12:34 PM > >>To: Wojciech Ziniewicz > >>Cc: 'serusers(a)lists.iptel.org' > >>Subject: Re: [Serusers] SER b2bua agent (+more)? > >> > >> > >> > >>>2. I've got simple pstn model > >>> > >>>LAN ---> SER > >>> \ | > >>> \ | > >>> PSTN > >>> > >>>I get simple disconnetion and loose rtp stream when calling > >>> > >>> > >>out to PSTN when > >> > >> > >>>making more than 5 calls. > >>> > >>>I don't actually know whether it's SER or my E1 Gateway > >>> > >>> > >>problem. Logs from > >> > >> > >>>Linksys PAP2 tells me that it's the "RTP PORT DUPLICATE" . > He's right > >>>because the sniffer tells the same . I see that two phones > >>> > >>> > >>connect their RTP > >> > >> > >>>stream on port 3000. > >>> > >>>Any ideas ? > >>> > >>> > >>> > >>> > >>Um..nope, it should not be a problem with SER, because SER > >>really does > >>not care about the RTP stream, and 5 connections to ser is really > >>nothing, I think its in linksys, I dont use the device > >>myself, but does > >>it have a limit on how many natted clients can sit behind it > >>(assuming u > >>r using NAT) or how many DHCP connections (I had this problem > >>once), or > >>even how many RTP streams it can handle > >> > >> > > > >I mean - my Sipphones connected to syslog say : > >"RTP PORT DUP:30000" > >but my E1 Gateway (Multitech Multivoip3010) does not say > anything. It looks > >like allocation of RTP is not random - every phone wants to > get on port > >30000. Any ideas (again) ? > > > >I'm quite sure it's not SER's case, but who knows. > > > >. > > > > > > >

1 0

Fwd: RES: [Serusers] Cisco 3640 X SER X PSTN
by Welesley Sibelson Dias 19 May '05

19 May '05

Many thks Jose Ricardo and "Ser Users" thks for your attetion more one time Jose Ricardo i have other problems now ahahahahahahah my call hang up and dont voice works now the caller rings e accept but cisco send me byes i send ethereal log to you into anex my IP is SER 200.184.153.54 Cisco 200.184.152.129 and Phone Voip 200.184.153.52 thks a lot this message below is in Portuguese: Muito Obrigado Jose Ricardo Falew pela forca desculpe meu pessimo ingles ---------- Mensagem reenviada ---------- Subject: RES: [Serusers] Cisco 3640 X SER X PSTN Date: Qui 19 Mai 2005 13:04 From: Jose Ricardo Maia Moraes <jrm(a)uol.com.br> To: serusers(a)lists.iptel.org Hi, I use R2 Digital on all of my AS5300 connected to PABX Nortel, Lucent and some carriers. Look the sample of my config: controller E1 0 framing NO-CRC4 clock source line primary ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani cas-custom 0 country brazil metering category 2 answer-signal group-b 1 voice-port 0:0 compand-type a-law cptone BR dial-peer voice 10 pots destination-pattern xxx direct-inward-dial port 0:0 Good luck. -----Mensagem original----- De: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] Em nome de Welesley Sibelson Dias Enviada em: quinta-feira, 19 de maio de 2005 06:33 Para: serusers(a)lists.iptel.org Assunto: [Serusers] Cisco 3640 X SER X PSTN Hi All I have try connect one E1 using SER and CISCO 3640 but i have some problems my PSTN Carrier Need Number for "A" but i don't send this , i need to complete calls in my carrier. Please have someone using cisco in R2 Digital send my one example of cfg . thks a lot

2 1

RE: [Serusers] SER b2bua agent (+more)?
by Wojciech Ziniewicz 19 May '05

19 May '05

questions inline > -----Original Message----- > From: Iqbal [mailto:iqbal@gigo.co.uk] > Sent: Thursday, May 19, 2005 12:34 PM > To: Wojciech Ziniewicz > Cc: 'serusers(a)lists.iptel.org' > Subject: Re: [Serusers] SER b2bua agent (+more)? > > >2. I've got simple pstn model > > > >LAN ---> SER > > \ | > > \ | > > PSTN > > > >I get simple disconnetion and loose rtp stream when calling > out to PSTN when > >making more than 5 calls. > > > >I don't actually know whether it's SER or my E1 Gateway > problem. Logs from > >Linksys PAP2 tells me that it's the "RTP PORT DUPLICATE" . He's right > >because the sniffer tells the same . I see that two phones > connect their RTP > >stream on port 3000. > > > >Any ideas ? > > > > > > Um..nope, it should not be a problem with SER, because SER > really does > not care about the RTP stream, and 5 connections to ser is really > nothing, I think its in linksys, I dont use the device > myself, but does > it have a limit on how many natted clients can sit behind it > (assuming u > r using NAT) or how many DHCP connections (I had this problem > once), or > even how many RTP streams it can handle I mean - my Sipphones connected to syslog say : "RTP PORT DUP:30000" but my E1 Gateway (Multitech Multivoip3010) does not say anything. It looks like allocation of RTP is not random - every phone wants to get on port 30000. Any ideas (again) ? I'm quite sure it's not SER's case, but who knows.

2 1

[Serusers] Re: [Serdev] [Fwd: Support for INFO method]
by Maxim Sobolev 19 May '05

19 May '05

Bayan Towfiq wrote: > Hi Maxim, > Do you know if anyone has RFC2833 support done for SEMS? Yes, we've done. It will be committed into the SEMS's cvs in few days from now. -Maxim > > Thanks, > Bayan > > Maxim Sobolev wrote: > >> Folks, >> >> Please approve attached patch, which adds support for INFO method in the >> parser and in the FIFO. This support is necessary for getting >> out-of-band DTMF support working in SEMS (yes, we do have this patch for >> SEMS already :). >> >> Thanks! >> >> -Maxim >> >> ------------------------------------------------------------------------ >> >> Index: stats.h >> =================================================================== >> RCS file: /cvsroot/ser/sip_router/stats.h,v >> retrieving revision 1.10 >> diff -d -u -d -u -r1.10 stats.h >> --- stats.h 24 Aug 2004 08:45:10 -0000 1.10 >> +++ stats.h 16 May 2005 22:49:34 -0000 >> @@ -47,6 +47,7 @@ >> case METHOD_ACK: stats->dir##_requests_ack++; >> break; \ >> case METHOD_CANCEL: stats->dir##_requests_cnc++; >> break; \ >> case METHOD_BYE: stats->dir##_requests_bye++; >> break; \ >> + case METHOD_INFO: stats->dir##_requests_info++; break; \ >> case METHOD_OTHER: stats->dir##_requests_other++; >> break; \ >> default: LOG(L_ERR, "ERROR: unknown method in rq stats >> (%s)\n", \ >> #dir); \ >> Index: modules/tm/t_fifo.c >> =================================================================== >> RCS file: /cvsroot/ser/sip_router/modules/tm/t_fifo.c,v >> retrieving revision 1.22 >> diff -d -u -d -u -r1.22 t_fifo.c >> --- modules/tm/t_fifo.c 25 Apr 2005 09:06:33 -0000 1.22 >> +++ modules/tm/t_fifo.c 16 May 2005 22:49:34 -0000 >> @@ -805,6 +805,7 @@ >> str_uri.len,str_uri.len ? str_uri.s : ""); >> >> if ( REQ_LINE(msg).method_value==METHOD_INVITE || + >> REQ_LINE(msg).method_value==METHOD_INFO || >> (twi->append && twi->append->add_body) ) { >> /* get body */ >> if( (body.s = get_body(msg)) == 0 ){ >> Index: parser/msg_parser.h >> =================================================================== >> RCS file: /cvsroot/ser/sip_router/parser/msg_parser.h,v >> retrieving revision 1.56 >> diff -d -u -d -u -r1.56 msg_parser.h >> --- parser/msg_parser.h 28 Feb 2005 15:28:24 -0000 1.56 >> +++ parser/msg_parser.h 16 May 2005 22:49:34 -0000 >> @@ -67,7 +67,7 @@ >> >> /* number methods as power of two to allow bitmap matching */ >> enum request_method { METHOD_UNDEF=0, METHOD_INVITE=1, >> METHOD_CANCEL=2, METHOD_ACK=4, - METHOD_BYE=8, METHOD_OTHER=16 }; >> + METHOD_BYE=8, METHOD_INFO=16, METHOD_OTHER=32 }; >> >> #define FL_FORCE_RPORT 1 /* force rport */ >> #define FL_FORCE_ACTIVE 2 /* force active SDP */ >> Index: parser/parse_fline.c >> =================================================================== >> RCS file: /cvsroot/ser/sip_router/parser/parse_fline.c,v >> retrieving revision 1.8 >> diff -d -u -d -u -r1.8 parse_fline.c >> --- parser/parse_fline.c 24 Aug 2004 09:01:25 -0000 1.8 >> +++ parser/parse_fline.c 16 May 2005 22:49:34 -0000 >> @@ -1192,6 +1192,7 @@ >> else IFISMETHOD( CANCEL, 'C') >> else IFISMETHOD( ACK, 'A' ) >> else IFISMETHOD( BYE, 'B' ) + else IFISMETHOD( INFO, 'I' ) >> /* if you want to add another method XXX, include METHOD_XXX in >> H-file (this is the value which you will take later in >> processing and define XXX_LEN as length of method name; >> Index: parser/parse_fline.h >> =================================================================== >> RCS file: /cvsroot/ser/sip_router/parser/parse_fline.h,v >> retrieving revision 1.6 >> diff -d -u -d -u -r1.6 parse_fline.h >> --- parser/parse_fline.h 24 Aug 2004 09:01:25 -0000 1.6 >> +++ parser/parse_fline.h 16 May 2005 22:49:34 -0000 >> @@ -51,6 +51,7 @@ >> #define CANCEL_LEN 6 >> #define ACK_LEN 3 >> #define BYE_LEN 3 >> +#define INFO_LEN 4 >> >> >> struct msg_start { >> >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serdev mailing list >> serdev(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serdev >> >> > > >

1 0

[Serusers] ser childs vs radius's RPID
by Mihaly Zachar 19 May '05

19 May '05

Hi, I'm trying to play with SER and auth_radius, with "Sip-RPId" attribute, because I would like to change something in every registered UA's number. When SER runs alone with "fork=no" everything is correct... If run ser with "fork=yes" and "children=4", in the sip packet's Remote-Party-ID header field contains the last registered user's Sip-RPId. Why??? That is why, the showed caller's number seems to be a random number... What can I do??? Thanks, Misi

1 1

[Serusers] RE: [Sipforum-discussion] - SIP servers and UA SDK
by info＠beeplove.com 19 May '05

19 May '05

If it would be an open source, we can help. :p Mohammad Original Message: ----------------- From: cpujol(a)navista.fr Date: Thu, 19 May 2005 18:04:52 +0200 (CEST) To: sipforum-discussion(a)lists.su.se Subject: [Sipforum-discussion] - SIP servers and UA SDK Hi all ! I'm searching for c++ or Java "SIP servers and UA" SDK because i have to develop my own SIP architecture for the company who employ me. Anyone could help me please ? Thanks. ------ To unsubscribe from the Sipforum-discussion mailing list visit https://mbox.su.se/mailman/listinfo/sipforum-discussion -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

2 1

[Serusers] SER stops responding
by Cameron Beattie 19 May '05

19 May '05

I have experienced (for the second time) a situation where SER (0.9.1) has stopped processing all register requests. Note I am also running mediaproxy 1.3.0. TOP shows 4 ser processes running each using between 20-30% CPU. There is one zombie process which is shown as [SER] <defunct> when I do a ps ax. There are only 5 running processes (4 SER + top). This is a test server that is not doing anything else and I am simply trying to register 1 UA against it. The server was running fine for a few days and then suddenly went AWOL. I looked back through the log and didn't find anything untoward. Any suggestions on what's the best way to try and find out what's causing it? Killing SER and starting again worked but I'd rather try to find out what went wrong. Maybe these are the pleasures of using a non-stable version? Regards Cameron

1 0

Re: [Serusers] call PSTN -> SIP
by Iqbal 19 May '05

19 May '05

not sure I follow, assume you have no src_ip check, what happens, does ser send a request for authroisation to the gateway, if so , then before u send that request that where you add a if statement to check its a request from ur src_ip... Iqbal Kostas Marneris wrote: >Thanks for your help Iqbal.. > >but I'm still experiencing problems : >I put the 'src_ip' in top of 'Call Processing System' > >my conf : > ... > # --------------------------------------- > # Record Route Section > # --------------------------------------- > ... > # --------------------------------------- > # Loose Route Section > # --------------------------------------- > ... > # ---------------------------------------- > # Call Type Processing System > # ---------------------------------------- > if (src_ip ==62.103.4.155) { > log(1,"PSTN-GW originated req..."); > break; > }; > > if (uri != myself) { > log(1,"URI is not myself"); > route(1); > break; > }; > ... > bla-bla-bla > > >I just expected SER logged the mesg "PSTN-GW originated req..." >to /var/log/messages... >But nothing is logged there... :( > >On the other hand I can see SIP mesgs on c5300 (debug ccsip all) >and I can verify these with tcpdump/5060 on SER box.. > > >thank you anyway, >Kostas > > > >Iqbal wrote: > > >>just accept the calls from that IP address, by detecting src_ip=a.b.cd >>or adding into trusted table, and checking for allow_trusted. >> >>The gateway should then send to ser, ser should then accept the call >>without calling any auth scenario >> >>Iqbal >> >>Kostas Marneris wrote: >> >> >> >> >>>Hello, >>> >>>I want to make a call from PSTN to a SIP phone. >>>I use a cisco 5300 for answering/handling the PSTN number >>>and then forward the call to a SER box (0.8.14) . >>> >>> POTS -> c5300 --> SER -> SIP softphone >>> >>>My conf on c5300: >>>! >>>dial-peer voice 1003 voip >>>destination-pattern 12345 >>>session protocol sipv2 >>>session target sip-server >>>codec g711ulaw >>>! >>>sip-ua >>>sip-server ipv4:X.Y.Z.W >>>! >>> >>> >>>What is the appropriate configuration on ser.cfg in order to >>>handle this call and forward it to sip:userX@mydomain.com ?? >>>Any hint about this ?? >>> >>> >>> >>>I've already done the opposite scenario (SIP->PSTN) : >>>ser.cfg: >>> if (uri =~ "^sip:2*") { >>> rewritehostport("1.2.3.4:5060"); >>> forward(uri:host, uri:port); >>> break; >>> }; >>> >>> >>> >>>thanks for any help, >>>Kostas >>> >>> >>> >>> >> >> >> > > >. > > >

1 0

[Serusers] another instance of ser
by Llanos Serna García-Conde 19 May '05

19 May '05

2 1

[Serusers] Ser 0.9.0 compiling mysql.so problem !!!
by Felipe Martins 19 May '05

19 May '05

Hi everybody ... I have compiled ser 0.9.0 and run it with the basic configuration without mysql ... it's working fine ... byt now I want to use mysql authentication. As I found at the manuals I needed to change a certain parameter at the Makefile to turn on the mysql support, til then, no problem, the problem is that when I compile it , it gives me inumerous errors, what am my doing wrong ?: ------------------------------------------------------------------------------ root@sipproxy1:/usr/src/ser-0.9.0/modules/mysql# make ../../Makefile.modules:21: "you should run make from the main ser directory" gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.0"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.3"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_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/pkg/include -I/usr/include/mysql -c dbase.c -o dbase.o dbase.c:34:25: mysql/mysql.h: No such file or directory dbase.c:35:26: mysql/errmsg.h: No such file or directory In file included from dbase.c:39: val.h:31:25: mysql/mysql.h: No such file or directory In file included from dbase.c:39: val.h:44: error: parse error before '*' token In file included from dbase.c:40: my_con.h:33:25: mysql/mysql.h: No such file or directory In file included from dbase.c:40: my_con.h:39: error: parse error before "MYSQL_RES" my_con.h:39: warning: no semicolon at end of struct or union my_con.h:40: warning: type defaults to `int' in declaration of `con' my_con.h:40: warning: data definition has no type or storage class my_con.h:41: error: parse error before "row" my_con.h:41: warning: type defaults to `int' in declaration of `row' my_con.h:41: warning: data definition has no type or storage class my_con.h:44: error: parse error before '}' token dbase.c: In function `submit_query': dbase.c:67: error: dereferencing pointer to incomplete type dbase.c:68: warning: implicit declaration of function `mysql_ping' dbase.c:68: error: dereferencing pointer to incomplete type dbase.c:72: error: dereferencing pointer to incomplete type dbase.c:90: warning: implicit declaration of function `mysql_query' dbase.c:90: error: dereferencing pointer to incomplete type dbase.c:93: warning: implicit declaration of function `mysql_errno' dbase.c:93: error: dereferencing pointer to incomplete type dbase.c:94: error: `CR_SERVER_GONE_ERROR' undeclared (first use in this function) dbase.c:94: error: (Each undeclared identifier is reported only once dbase.c:94: error: for each function it appears in.) dbase.c:94: error: `CR_SERVER_LOST' undeclared (first use in this function) dbase.c:98: warning: implicit declaration of function `mysql_error' dbase.c:98: error: dereferencing pointer to incomplete type dbase.c:98: error: dereferencing pointer to incomplete type dbase.c:98: warning: format argument is not a pointer (arg 3) dbase.c:98: error: dereferencing pointer to incomplete type dbase.c:98: warning: format argument is not a pointer (arg 3) dbase.c:98: error: dereferencing pointer to incomplete type dbase.c:98: warning: format argument is not a pointer (arg 3) dbase.c:98: error: dereferencing pointer to incomplete type dbase.c:98: warning: format argument is not a pointer (arg 3) dbase.c:98: error: dereferencing pointer to incomplete type dbase.c:98: warning: format argument is not a pointer (arg 3) dbase.c:98: error: dereferencing pointer to incomplete type dbase.c:98: warning: format argument is not a pointer (arg 3) dbase.c:98: error: dereferencing pointer to incomplete type dbase.c:98: warning: format argument is not a pointer (arg 3) dbase.c: At top level: dbase.c:138: error: parse error before '*' token dbase.c: In function `print_values': dbase.c:142: error: `_c' undeclared (first use in this function) dbase.c:142: error: `_b' undeclared (first use in this function) dbase.c:142: error: `_l' undeclared (first use in this function) dbase.c:142: error: `_v' undeclared (first use in this function) dbase.c:142: error: `_n' undeclared (first use in this function) dbase.c: At top level: dbase.c:166: error: parse error before '*' token dbase.c: In function `print_where': dbase.c:172: error: `_c' undeclared (first use in this function) dbase.c:172: error: `_b' undeclared (first use in this function) dbase.c:172: error: `_l' undeclared (first use in this function) dbase.c:172: error: `_k' undeclared (first use in this function) dbase.c:172: error: `_v' undeclared (first use in this function) dbase.c:172: error: `_n' undeclared (first use in this function) dbase.c:178: error: `_o' undeclared (first use in this function) dbase.c: At top level: dbase.c:207: error: parse error before '*' token dbase.c: In function `print_set': dbase.c:213: error: `_c' undeclared (first use in this function) dbase.c:213: error: `_b' undeclared (first use in this function) dbase.c:213: error: `_l' undeclared (first use in this function) dbase.c:213: error: `_k' undeclared (first use in this function) dbase.c:213: error: `_v' undeclared (first use in this function) dbase.c:213: error: `_n' undeclared (first use in this function) dbase.c: In function `store_result': dbase.c:304: error: dereferencing pointer to incomplete type dbase.c:304: warning: implicit declaration of function `mysql_store_result' dbase.c:304: error: dereferencing pointer to incomplete type dbase.c:305: error: dereferencing pointer to incomplete type dbase.c:306: warning: implicit declaration of function `mysql_field_count' dbase.c:306: error: dereferencing pointer to incomplete type dbase.c:311: error: dereferencing pointer to incomplete type dbase.c:311: error: dereferencing pointer to incomplete type dbase.c:311: warning: format argument is not a pointer (arg 3) dbase.c:311: error: dereferencing pointer to incomplete type dbase.c:311: warning: format argument is not a pointer (arg 3) dbase.c:311: error: dereferencing pointer to incomplete type dbase.c:311: warning: format argument is not a pointer (arg 3) dbase.c:311: error: dereferencing pointer to incomplete type dbase.c:311: warning: format argument is not a pointer (arg 3) dbase.c:311: error: dereferencing pointer to incomplete type dbase.c:311: warning: format argument is not a pointer (arg 3) dbase.c:311: error: dereferencing pointer to incomplete type dbase.c:311: warning: format argument is not a pointer (arg 3) dbase.c:311: error: dereferencing pointer to incomplete type dbase.c:311: warning: format argument is not a pointer (arg 3) dbase.c: In function `db_free_result': dbase.c:348: warning: implicit declaration of function `mysql_free_result' dbase.c:348: error: dereferencing pointer to incomplete type dbase.c:349: error: dereferencing pointer to incomplete type dbase.c: In function `db_query': dbase.c:398: error: dereferencing pointer to incomplete type dbase.c: In function `db_insert': dbase.c:471: error: dereferencing pointer to incomplete type dbase.c: In function `db_delete': dbase.c:516: error: dereferencing pointer to incomplete type dbase.c: In function `db_update': dbase.c:559: error: dereferencing pointer to incomplete type dbase.c:568: error: dereferencing pointer to incomplete type make: *** [dbase.o] Error 1 root@sipproxy1:/usr/src/ser-0.9.0/modules/mysql# ----------------------------------------------------------------------------- -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

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Fwd: [Serusers] Please tell me the steps...
by Bharat More 19 May '05

19 May '05

Note: forwarded message attached. ________________________________________________________________________ Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony

1 0

[Serusers] Cisco 3640 X SER X PSTN
by Welesley Sibelson Dias 19 May '05

19 May '05

Hi All I have try connect one E1 using SER and CISCO 3640 but i have some problems my PSTN Carrier Need Number for "A" but i don't send this , i need to complete calls in my carrier. Please have someone using cisco in R2 Digital send my one example of cfg . thks a lot

3 3

[Serusers] call stop
by Tiziano 19 May '05

19 May '05

Hi, i'd like to know if there is a way to stop sip call where communication is active with a command (for instance when credit=0) Thanks, Tiziano -- Tiziano <ml(a)asdasd.it>

4 8

[Serusers] Please tell me the steps...
by Bharat More 19 May '05

19 May '05

Hi All... I'm new buddy to SER.I want to use ser0.9.0 and redius for AAA purpose.And MySQL for database. can any one tell me the steps that what should be done ? as what to install first?SEr or MYSQL? then what about RADIUS? Little confused... Please help Thanks in advance... ________________________________________________________________________ Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony

1 0

[Serusers] serweb doesn't accept susbscribers
by Llanos Serna García-Conde 19 May '05

19 May '05

3 6

[SPAM] Re: [Serusers] Help on Running SER.
by Samuel Osorio Calvo 19 May '05

19 May '05

If you want to listen to more than one address, you should add fork=yes to the config file otherwise only the first address of the listen= a.b.c.d list will be used. With the children=X config paramter you can tune how many listener process you want to assign to each listening address. Samuel. Unclassified. >>> Iqbal <iqbal(a)gigo.co.uk> 05/19/05 12:58PM >>> well a copy of ser.cfg would help. Iqbal Simith Nambiar wrote: >Hi All, > I'am new to Ser , and hope this is the right list to ask >this Question. I get the following error on running Ser, what could be the >problem ? >and some hint on where to look for to fix my config file ? or something else >would be very much helpful ! > >============================================================================ >==== >WARNING: no fork mode and more than one listen address found(will use only >the the first one) >stateless - initializing > 0(0) Maxfwd module- initializing >textops - initializing >ERROR: error -478 while trying to fix configuration >============================================================================ >==== > >And where can i look for all the Error codes and thier corresponding >meanings ? > >Thank you. > >Cheers. >simith > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > >. > > > _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] Help on running Ser.
by Gerard Wagener 19 May '05

19 May '05

Hi, Before launching ser, check your config file with ser -c. If there are any errors the line and the column ,where is the error, are shown. Debug messages you can find in your kernel messages (e.g /var/log/messages) when you have enabled log_stderror=no in your config file. When you enable fork=yes, ser is listening on all interfaces. Regards, G�rard --- Simith Nambiar <simith(a)indtelesoft.com> wrote: > Hi All, > I'am new to Ser , and hope this is > the right list to ask. I > get the following error on running Ser, what could > be the problem ? > and some hint on where to look for to fix my config > file ? or something else > would be very much helpful ! > > ============================================================================ > ==== > WARNING: no fork mode and more than one listen > address found(will use only > the the first one) > stateless - initializing > 0(0) Maxfwd module- initializing > textops - initializing > ERROR: error -478 while trying to fix configuration > ============================================================================ > ==== > > > Thank you. > > Cheers. > simith > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

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RE: [Serusers] Help on running Ser.
by Simith Nambiar 19 May '05

19 May '05

Hi All, This Post is hitting the List late, so please ignore it, i had a problem with my config file, when i replaced it with the one in www.onsip.org, the issue did not appear anymore ! Cheers, Simith -----Original Message----- From: Simith Nambiar [mailto:simith@indtelesoft.com] Sent: Tuesday, May 17, 2005 7:36 PM To: 'serusers(a)lists.iptel.org' Subject: [Serusers] Help on running Ser. Hi All, I'am new to Ser , and hope this is the right list to ask. I get the following error on running Ser, what could be the problem ? and some hint on where to look for to fix my config file ? or something else would be very much helpful ! ============================================================================ ==== WARNING: no fork mode and more than one listen address found(will use only the the first one) stateless - initializing 0(0) Maxfwd module- initializing textops - initializing ERROR: error -478 while trying to fix configuration ============================================================================ ==== Thank you. Cheers. simith _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email ______________________________________________________________________ ______________________________________________________________________ This email has been scanned via a message filtering facility. If you have any issues with your email, please contact the IT Dept.

1 0

[Serusers] Help on Running SER.
by Simith Nambiar 19 May '05

19 May '05

Hi All, I'am new to Ser , and hope this is the right list to ask this Question. I get the following error on running Ser, what could be the problem ? and some hint on where to look for to fix my config file ? or something else would be very much helpful ! ============================================================================ ==== WARNING: no fork mode and more than one listen address found(will use only the the first one) stateless - initializing 0(0) Maxfwd module- initializing textops - initializing ERROR: error -478 while trying to fix configuration ============================================================================ ==== And where can i look for all the Error codes and thier corresponding meanings ? Thank you. Cheers. simith

2 1

[Serusers] SER b2bua agent (+more)?
by Wojciech Ziniewicz 19 May '05

19 May '05

Hello, I have two questions 1. I really don;t want to install Vocal (from vovida.org) or asterisk to use B2BUA (for prepaid users and "live"call management). So is there any contributed module for SER that I don;'t know about that is sane about all the sessoins and is not stateless? I was wondering also can SER send artificial BYE's to peers when he decides that the call should be ended but peers dont want to end the call. And of course SER would not let the RTP through himself because it's burdening his scalability (although being sane of sessions burdens it the same)... 2. I've got simple pstn model LAN ---> SER \ | \ | PSTN I get simple disconnetion and loose rtp stream when calling out to PSTN when making more than 5 calls. I don't actually know whether it's SER or my E1 Gateway problem. Logs from Linksys PAP2 tells me that it's the "RTP PORT DUPLICATE" . He's right because the sniffer tells the same . I see that two phones connect their RTP stream on port 3000. Any ideas ? Thanks in advance -- Pozdrawiam, Wojciech Ziniewicz Optocomp sp.z.o.o, www.optocomp.pl mailto: wojtekz(a)optocomp.pl +48(0)691031535

2 1

[Serusers] sems version
by Llanos Serna García-Conde 19 May '05

19 May '05

1 0

[Serusers] ser.cfg is write protected..How to make it Archive...
by Bharat More 19 May '05

19 May '05

Hi All.. I'm new buddy to Ser.I want to install SER, Bht the configuration file in the folder usr/local/etc/ser/ser.cfg is write protected and the owner of the file is root then How can I update this file? I am using Fadora and SER 0.9.0 for SER . Please Help me ...Urgent... Thanks in advance for all hints... ________________________________________________________________________ Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony

2 1

Re: [Serusers] Ser Accepting Calls From Asterisk
by Greger V. Teigre 19 May '05

19 May '05

Not much to say without the SIP messages giving the error. Looks like an empty Contact header. Could be a faulty user client or something completely different... g-) ---- Original Message ---- From: Mr Greg Plater To: 'Greger V. Teigre' Sent: Thursday, May 19, 2005 02:01 AM Subject: RE: [Serusers] Ser Accepting Calls From Asterisk > Hi Greger > > I note I have this content-length in the onreply route. > > Any Ideas > > Cheers > Greg > > BTW thanks for your help > > > > > From: Greger V. Teigre [mailto:greger@teigre.com] > Sent: Thursday, 12 May 2005 6:04 PM > To: Mr Greg Plater; serusers(a)lists.iptel.org > Subject: Re: [Serusers] Ser Accepting Calls From Asterisk > > Good to hear, Greg. Looks like a message without a Contact header > (or a non-conforming one). Could you find the SIP message that gives > this message? BTW, just to be sure: The onsip.org files have a test > for content-length in the onreply route (Empty body). Do you have it > in your ser.cfg? > g-) > > ---- Original Message ---- > From: Mr Greg Plater > To: serusers(a)lists.iptel.org > Sent: Thursday, May 12, 2005 02:26 AM > Subject: [Serusers] Ser Accepting Calls From Asterisk > >> Hi Calls work from Asterisk to SER now. >> >> I note in Debug output: >> >> 3(1814) contact_parser(): Empty body >> 3(1814) parse_contact(): Error while parsing >> 3(1814) error: mediaproxy/getContactURI(): cannot parse Contact >> header >> >> The call works fine, how do I fix this message or do I leave whats >> working alone. >> >> Cheers >> Greg >> >> _____________________________________________________________________ >> Hi Greger >> >> Thank you for the information I have found it just what I needed. >> >> Calls To SER from Asterisk: >> >> To add to trusted in MYSQL I would use this right: >> >> INSERT INTO trusted SET >> src_ip='***.***.***.***',proto='any',from_pattern='sip:[0-9]*@***.***.***.***'; >> #replace *** with the asterisk ip address >> >> With this in my ser.cfg: >> >> modparam("permissions", "db_url", "mysql://ser:heslo@localhost/ser") >> modparam("permissions", "db_mode", 1) >> modparam("permissions", "trusted_table", "trusted") >> >> >> and for routing to Asterisk From SER: >> >> if (method == "INVITE" && (uri=~"^sip:0")){ >> rewritehostport ("***.***.***.***:5060"); >> t_relay(); >> break(); >> }; >> >> >> >> Or do I just add: >> Asterisk to the trusted table and the following near the >> >> >> if(!allow_trusted) >> if (!proxy_authorize("","subscriber")) { >> proxy_challenge("","0"); >> break; >> } else if (!check_from()) { >> sl_send_reply("403", "Use From=ID"); >> break; >> >> >> >> >> Thank you very much for your help >> >> Cheers >> Greg >> >> >> >> >> >> From: Greger V. Teigre [mailto:greger@teigre.com] >> Sent: Wednesday, 11 May 2005 6:20 PM >> To: Mr Greg Plater; serusers(a)lists.iptel.org >> Subject: Re: [Serusers] Ser Accepting Calls From Asterisk >> >> Greg, >> Basically, you can do four things (first two for dialing >> asterisk-ser, the next two for dialing ser-asterisk): >> 1. Accept INVITEs from a specific src ip (asterisk) and add if(src_ip >> != asterisk_ip) around the INVITE authentication to avoid that part >> being triggered >> 2. Or you accomplish the same thing be adding asterisk to the trusted >> table and us if(!allow_trusted) instead. This will give you more >> control as you can filter on ruri as well. >> 3. You can create an account in asterisk and have asterisk register >> with SER like a regular user (who will be available in location for >> dialling into asterisk). This only makes sense if you have one main >> number for all users registered on asterisk (you don't need 1 or 2 if >> you do this) >> 4. Use some kind of routing, either through manual configuration >> (uri=~"some pattern", then forward to asterisk) or use the LCR (least >> cost routing) module to look up the ruris and forward to asterisk. >> This must be done in combination with 1 or 2. >> >> Hope this helps. >> g-) >> >> ---- Original Message ---- >> From: Mr Greg Plater >> To: serusers(a)lists.iptel.org >> Sent: Wednesday, May 11, 2005 06:10 AM >> Subject: [Serusers] Ser Accepting Calls From Asterisk >> >>> Hi >>> >>> I am using the onsip getting started doc with mediaproxy. >>> >>> Calls for registered users on SER work great but when I try to call >>> a registered ser user from my asterisk box. >>> I get a response in Asterisk Debug. Says "failed to auth on INVITE" >>> >>> If I wish to allow calls from my asterisk box to registered users on >>> SER what do I change. >>> >>> >>> IN SER.cfg I note these 2 snippets >>> >>> >>> if (!www_authorize("","subscriber")) { >>> www_challenge("","0"); >>> break; >>> }; >>> >>> >>> >>> if (!proxy_authorize("","subscriber")) { >>> proxy_challenge("","0"); >>> break; >>> } else if (!check_from()) { >>> sl_send_reply("403", "Use From=ID"); >>> break; >>> }; >>> >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] Post on behalf of Simith Nambiar: error -478 while trying to fix configuration
by Simith Nambiar 19 May '05

19 May '05

Hi All, Thanks for your responses, it was a problem with the ser.cfg file, when i used the one available on onsip.org as suggested, ser started flying :-) ! One more problem i faced was that, when the config file is downloaded from the internet, not in zipped format but one-by -one, there seems to be some html formatting that comes along with it, leave alone the HTML tags, once that is removed, there are some more problems like the > and < symbols in the Expressions of ser.cfg, will also be turned int thier HTML representation, with a > or something, once that was also removed and replaced with the actual symbol, things seemed to work for me. Thank you. Cheers. Simith -----Original Message----- From: Iqbal [mailto:iqbal@gigo.co.uk] Sent: Thursday, May 19, 2005 6:58 AM To: g.fishilevich(a)gmail.com; greger(a)teigre.com Cc: serusers(a)lists.iptel.org; Simith Nambiar Subject: Re: [Serusers] Post on behalf of Simith Nambiar: error -478 whiletrying to fix configuration Hi Check if u have any IP addresses in your config, it might be that it cant resolve them Iqbal On 5/18/2005, "Grigory Fishilevich" <g.fishilevich(a)gmail.com> wrote: >Simith: > >i had a problem alike yours. i'm sorry but i have no more exactly info. >i had an error in my ser.cfg und got the same strange error-message like yours. >you should analyze you ser.cfg and take a look to /var/log/messages > >bg, >Grigory > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > > ______________________________________________________________________ This email has been scanned via a message filtering facility. If you have any issues with your email, please contact the IT Dept.

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Re: [Serusers] ser.cfg is write protected..How to make it Archive...
by Gerard Wagener 19 May '05

19 May '05

Hi, 1. login as root 2. use the chmod command $chmod +w /usr/local/etc/ser/ser.cfg and you can edit the ser.cfg Regrads, G�rard --- Bharat More <bharat_sharp(a)yahoo.co.in> wrote: > Hi All.. > I'm new buddy to Ser.I want to install SER, > Bht the configuration file in the folder > usr/local/etc/ser/ser.cfg is write protected and the > owner of the file is root then How can I update this > file? I am using Fadora and SER 0.9.0 for SER . > Please Help me ...Urgent... > Thanks in advance for all hints... > > ________________________________________________________________________ > Yahoo! India Matrimony: Find your life partner > online > Go to: http://yahoo.shaadi.com/india-matrimony > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] Where do I find the LCR Module please?
by Stephen Kingham 19 May '05

19 May '05

Hi Juha, I have been watching the development of the LCR Module and I would like to use it. I have searched through IPTEL, Google, and the email list archives and failed to find any hint of where to find it. Where do I find documentation and the module it's self? Thanks Stephen Kingham -- Stephen Kingham, MIT, BSc, E&C Cert Project Manager and Consulting Engineer mailto:Stephen.Kingham@aarnet.edu.au Telephone +61 2 6222 3575 (office) +61 419 417 471 (mobile) sip:Stephen.Kingham@aarnet.edu.au Voice and Video over IP for The Australian Academic Research Network (AARNet) http://www.aarnet.edu.au

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[Serusers] Post on behalf of Simith Nambiar: error -478 while trying to fix configuration
by Greger V. Teigre 19 May '05

19 May '05

Simith, Not easy to tell what the problem can be. You haven't included your ser.cfg. If you are new to ser, I would recommend that you download the Getting Started document from http://onsip.org/ and the configuration file package with all the ser.cfg example files. Those should get you started. g-) Simith Nambiar wrote: > Hi Greger, > > Thanks for the response, I don't know you, but i have been seeing your > response > on the mailing list for Serusers, but my post is never landing there > , which is my problem. > > I subscribed yesterday, still my requests are pending "Moderator > approval" , i would have been happy > if someone could help me, in running SER, i have an Error coming from > route.c pasted below, > It is from fix_rls --> fix_action , i don't understand what they are > doing there in code, as i'am new to SER. > > Listening on > udp: 127.0.0.1 [127.0.0.1]:5070 > udp: 10.10.100.44 [10.10.100.44]:5070 > tcp: 127.0.0.1 [127.0.0.1]:5070 > tcp: 10.10.100.44 [10.10.100.44]:5070 > Aliases: > tcp: linuxdemo:5070 > tcp: linuxdemo.indts.com:5070 > tcp: localhost:5070 > tcp: localhost.localdomain:5070 > udp: linuxdemo:5070 > udp: linuxdemo.indts.com:5070 > udp: localhost:5070 > udp: localhost.localdomain:5070 > > WARNING: no fork mode and more than one listen address found(will > use only the the first one) > stateless - initializing > 0(0) Maxfwd module- initializing > textops - initializing > ERROR: error -478 while trying to fix configuration > > > > Something wrong with the config file ? > As my mails are not reaching the list, can you plz forward this or > offer any help that might fix my problem ! > Thanks for your time. > > > Cheers. > Simith > > > > > > -----Original Message----- > From: Greger V. Teigre [mailto:greger@teigre.com] > Sent: Wednesday, May 18, 2005 11:42 AM > To: Simith Nambiar > Subject: Re: [Serusers] More on load balancing: LVS + SRVs, SER + > Asterisk > > > I'm not sure if I know you and your problem? Posting on serusers is > your best hope to get help. Be specific and write short summary at > the top what the problem is, then add more info further below. Most > often a post that is > > not answered is because it's too much work to try to decipher what the > problem can be. > g-) > > Simith Nambiar wrote: >> Hi Greger, >> I feel you are busy , please can you give me some id, >> that i can write to to fix my problem, >> I mean the moderator or someone, i'am badly in ned of a solution, as >> i do not Ser very well, i'am not able to fix it on my >> own. >> >> Maybe once i study the code, i can fix it myself, until then i just >> wanted to see it up and running and test it for some >> funcionality, hope u understand. >> >> Please help ! >> >> cheers. >> Simith >> >> >> -----Original Message----- >> From: Greger V. Teigre [mailto:greger@teigre.com] >> Sent: Wednesday, May 18, 2005 11:12 AM >> To: Pietro Ravasio; serusers(a)lists.iptel.org >> Subject: Re: [Serusers] More on load balancing: LVS + SRVs, SER + >> Asterisk >> >> >> Pietro Ravasio wrote: >>> Anyway, at the moment "mixing Asterisk with SER" is not my major >>> concern, I'm pretty worried about trying to use a particular >>> outbound proxy for calls to a NATed UA: >>> >>>> I think that the dispatcher module of ser can be used >>>> cheaper instead of using LVS, which will not give you call-id >>>> persistence anyway. AFAIU, your scenario is quite similar to >>>> what I described in this thread: >>>> http://lists.iptel.org/pipermail/serusers/2005-May/019118.html >>>> Just a different implementation... >>> >>> Yes, I now agree with you in using this approach rather than >>> "multiple internal domains" one. The only things I'd like to modify >>> are: - storing location on every message, not only on REGISTER, in >>> case a ser dies and lvs udp persistence sends the next UA message to >>> another SER (for instance my Sipura phone sends a bunch of NOTIFY >>> messages in order to keep NAT open, if a ser dies and if phone is >>> already authenticated, I want to update "its SER") >> >> I'm not sure if I like the idea of doing an sql write for every >> single message. That's will be a massive performance hit. But the >> idea is good. What about doing LVS NAT? >> >>> - using only one table for every SER storing in a different column >>> the proxy (taken from dst_ip) who has handled the message. This way >>> I don't have to preprovision all SER instances in all ser.cfg >>> routing logics. At this point: >> >> But then you have to modify SER, which is back to where we started: >> 1. If you have to modify SER to do load balancing, what would be the >> best way to do it? >> 2. The answer is most likely: Do it in a module like dispatcher, but >> add call-id persistence >> 3. Doing level 7 load balancing can only be effective in the kernel >> >> So why not focus your attention on creating call-id persistence in >> LVS? ;-) >> >>> .) if I use a B2B user agent in order to handle NAT (for instance >>> asterisk) - I'd send INVITE to my B2B UA >>> (t_relay_to_udp("b2bua_lvs_virtual_ip")) with rewritten Contact: >>> field with the IP number of the (outbound) SER UAB has >>> registered/sent its latest message to (taken from DB) >>> >>> ..) otherwise, in a "pure SER" environment, >>> - I'd send INVITE directly to the SER proxy which IP:port is taken >>> from DB with the same logic as above >> >> Yes, but how do you know the server just went down and you still >> haven't updated the IP:port of the UA? I assume you will use >> t_replicate and save_memory to make sure that all servers have the >> updated registration info >> >> in memory? (as DB writethrough works fine, but other ser's will not >> load the >> >> new location upon receiving an INVITE) You probably have to do >> t_replicate for EVERY message to update the IP:port of the UA and >> you're into massive overload of SIP messages (used by t_replicate). >> >>> Using only one location table could be useful in order to create a >>> flexible routing logic: if the server taken from DB is not online >>> anymore, send message to another one (randomly chosen), hoping that >>> UAS is not behind nat. This logic could be inserted in B2BUA dialing >>> plans or in ser.cfg. >> >> I don't like the word "hoping"... >> >>>> If it's work, it's a workaround. No better or worse than other >>>> workarounds... g-) >>> >>> At the moment I start thinking that any technique that tries to bind >>> an outbound SER for all calls to a set of user agents, is going to >>> be a workaround... Am I wrong? :-/ >> >> No, I don't think so. That is trying to do something at the >> application level that people have long figured out is better done at >> lower levels... In fact, I believe the simplest solution would be to >> make sure that LVS can select ANY SER for load balancing and use LVS >> NAT (which is quite scalable and is the way Cisco does it). The >> simplest way to modify ser for this would be to make sure that all >> lookup("location") calls actually make an sql >> >> read. If each ser instance flushes each save to db and a shared sql >> cluster >> >> is used as backend, you are good to go. >> >> The other way is to implement call-id persistence in IPVS (LVS' load >> balancer). I like this better, because you do it in kernel mode and >> don't add a whole lot of new DB reads... >> >> g-) >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> ______________________________________________________________________ >> This email has been scanned via a message filtering facility. >> If you have any issues with your email, please contact the IT Dept. > > > ______________________________________________________________________ > This email has been scanned via a message filtering facility. > If you have any issues with your email, please contact the IT Dept.

3 2

[Serusers] SER & MYSQL data couple of emty fields
by Mr Greg Plater 19 May '05

19 May '05

Brain strain please help!!!!! I have this SER - MYSQL -SERWEB When I place setflag(3) or setflag(1) Example: setflag(3); acc_db_request("Missed Call", "missed_calls"); I note when I place a call to a user that is not online the missed_calls table is populated with: | "8838212361" <sip:8838212361@***.***.***.***>;tag=SxCH3s2YXJCUGNdq | "8838212349" <sip:8838212349@***.***.***.***> | Missed Call | INVITE | sip:8838212349@***.***.***.***| sip:8838212349@***.***.***.***| sip:8838212361@***.***.***.***| sip:8838212349@***.***.***.***| waLKvXL2KBr8oAuS(a)***.***.***.*** | n/a | n/a | SxCH3s2YXJCUGNdq | n/a | 2005-05-18 07:06:08 | 2005-05-18 17:06:08 | The two n/a are columns username & domain . When I log in as the user in ser web the missed the call the data does not display. I figure this is because of the fields n/a. Anyone have any ideas on what I have done wrong I have checked the table in mysql and noted the fields are set to "not null" this should mean that the fields are never empty, were does the n/a come from and why did the fields not populate with the correct data?

2 1

[Serusers] Splitting NAT traversal from proxy functionality
by Giuseppe Cardone 18 May '05

18 May '05

Hello, I was very interested in the outbound proxy functionality discussed last december on the serdev list: has anyone seen any implementations come out ? Thanks in advance, Giuseppe On 04-12 15:12, Adrian Georgescu wrote: > I think there should be no "sigle" outbound proxy out there, if it is > it becomes just another single point of failure. The ideal solution > would provide a shared user-location database between multiple servers > which would all act as outbound-proxy for each call they serve and > would enable NAT traversal only at the edges when and if necessary. I think this is exactly what Path is about, you can have different outbound proxy for each registered contact. > A first step in this direction would be to store more information in > the location table like the address of the server that handled the > registration and both NAT ed and not NAT-ed contacts. You mean the destination IP of the REGISTER request so that some other server elected when the original server fails can select which IP to use to get the INVITE through the NAT, right ? That's on my todo list (with quite high priority). The other part -- NATed and non-NATed contact -- is already implemented. If you call fix_nated_register then the original Contact value would be preserved and the source IP and port of the request would be stored in the received column of location table (or can be appended to the message as ;received parameter of the Contact if the registrar is not co-located with the NAT-enabled proxy). The lookup function would put the original Contact into the Request-URI but the request would be forwarded to the IP and port in received parameter. For the scenario with decoupled outbound proxies and registrar (and also if Path is used) we would need to communicate the IP and port of the NAT to the outbound proxy instead -- Route header field could be used for this, for example. Jan. > Based on this > information depending on the call flow the SIP farm members should > route the call to the member that serves the NAT-ed client and should > expire only locations served locally. > > I am somewhere between Jan and Klaus opinions working on a similar > concept, it will be nice to see some implementations available to > compare them. > > Adrian > > On Dec 4, 2004, at 2:58 PM, Jan Janak wrote: > > >On 04-12 13:24, Klaus Darilion wrote: > >> > >> > >>Jan Janak wrote: > >>>I like 1) more than 2). Support for Path header field could be also > >>>useful for other stuff then NAT traversal. We will definitely accept > >>>that into the main tree if you implement it. > >> > >>Uuh, you're right - someone has to implement it! :-) > > > > Yeah :-). > > > >>> And we could do something like: > >>> if (save("location")) { > >>> save_path(); > >>> copy_path_to_reply(); > >>> sl_send_reply("200", "OK"); > >>> }; > >>> > >>> That would allow us to keep this logic separated from registrar. > >>>- The path vector should be probably stored separately -- outside the > >>> location table and outside usrloc. In the script we could do > >>> something like: > >>> > >>> lookup("location"); > >>> lookup_path("path_table"); > >>> > >>> Function lookup_path would lookup the path vector and add it to the > >>> message as the pre-loaded route set. > >> > >>This would ensure to do not influence the registrar und location > >>module, > >>but I think implementation would be much more easier if we include it > >>into the registrar module and enable it with a configuration parameter > >>(usepath=0/1). Also from the database schema it makes sense to include > >>it into the location table. There is a 1:1 relation between the path > >>and > >>the location (and the path value may be NULL). > > > > I proposed separate table because, in my opinion, it is easier to > > implement. Moreover, if you keep it separately -- independent of > > usrloc, you can retrieve the path vector even if you do not use user > > location to get the next hop URI (PSTN gateways, static > >configuration, > > path vector configured by users, and so on). > > > >>>- One problem here might be the limitation of SER to execute the > >>>routing > >>> logic only for one branch only. That means it would not be possible > >>>to > >>> lookup different path vector for different contacts. I think that > >>>this > >>> problem can be ignored at the beginning, SER would be simply limited > >>> to using one path vector per AOR. > >> > >>That's a problem. In case for 3GPP this might be negligible (there is > >>always only 1 handset registered). But for typical VoIP this won't > >>work. > >>E.g. I have to phones registered: one at home, one at work, and both > >>are > >>behind NAT and use an outbound proxy. > >> > >>This limitation can be bypassed by using solution 2 - as the "route" > >>will be hidden in the contact uri. Furthermore, solution 2 would not > >>requrie changes to the main proxy and therefore testing and rollout > >>would be much easier. > >> > >>I also think solution 2 would be easier to implement, but still > >>solution > >>1 would be a more generic solution. > > > > Yes, that's true. I think, however, that 1) is the right approach. > > > > Jan. > > > >_______________________________________________ > >Serdev mailing list > >HYPERLINK "http://lists.iptel.org/mailman/listinfo/serdev"Serdev at iptel.org > >HYPERLINK "http://lists.iptel.org/mailman/listinfo/serdev"http://lists.iptel.org/mailman /listinfo/serdev -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.10 - Release Date: 13/05/2005

1 0

[Serusers] Error Code 392 Nosiy feedback Warning
by Kofi Obiri-Yeboah 18 May '05

18 May '05

In December of 2002, there was a question about this error Message: Warning 392: Nosiy feedback tells .. The following was Jiri's answer: "That is additional information on request processing: pid of answering process, source address from which a request came, incoming URI, outgoing URI and number of via's in replied request. Good for troubleshooting. Anyone in a SIP path sees better what is happening and can report it to SIP admin. Can be turned off using sip_warning option, but I don't recommend doing so." Can anybody please explain what exactly the noisy feedback is. It is obvious that the feedback is coming from the SER server. I am able to register successfully with the server. When I try dialing another registered user I receive this warning and in the details get a 404 error message. The strange thing is that the other registered person is able to dial and and am able to receive his call. Should the sequence of making a call through SER indicate dialing, ringing and connected? Has this sequence got anything to do with the Error code 392? Any "light" appreciated KOFI At 09:47 PM 12/9/2002, Greg Fausak wrote: >I am constantly seeing this warning: > >Warning: 392 216.87.128.67:5060 "Noisy feedback tells: pid=3961 req_src_ip=216.87.128.68 in_uri=sip:216.87.128.67 out_uri=sip:216.87.128.67 via_cnt==1". > >What does it mean??? > >---greg >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/ -- Worrying Is Like Paying Interest On A Loan You Are Yet To Take Be Happy! Don't Worry,

1 0

Re: [Serusers] SER Message waiting indicator light Cisco 7940/7960
by Java Rockx 18 May '05

18 May '05

Patrick, This is a bash script that externnotify executes #!/bin/ash # # This script is called by Asterisk every time a voice mail is # left for a subscriber. The CONTEXT, EXTENSION, and VM_COUNT # are passed by Asterisk. # # All we need to do is update a file in /var/spool/asterisk/mwi # to tell our cron job that we have to send a SIP NOTIFY message # to a particular UA # # This file is referenced in /etc/asterisk/voicemail.conf # # NOTE: The $2 argument is passed in to this script in the # following form: MAILBOX@DOMAIN # # example: 4075551212@default # MWI_DIR=/var/spool/asterisk/mwi/ if [ -n ${2} ] ; then if /bin/touch ${MWI_DIR}/${2} ; then exit 0 else exit 1 fi else exit 2 fi The following is executed in CRON every minute. This script actually notifies the subscribers by sending NOTIFY messages to SER. Please note that our asterisk is a modified version that stores mailboxes in a directory hash, so you can modify this to fit your needs. #!/bin/sh # This script send the SIP NOTIFY message to UA. The NOTIFY message # can either enable or disable the UA message indicator. The script # looks for any file in $VM_HOME and creates the NOTIFY message # # The actual NOTIFY message is sent to the SIP proxy by the sipsak # utilty. After the message is send we delete the processed file # from VM_HOME so we don't keep resending the same message. Most # UAs will periodically send a SUBSCRIBE message to the SIP proxy. # When this occurs the SIP proxy will place a file in the location # /var/spool/mwi/delay and a less frequent CRON job will process it. VM_ROOT=/var/spool/asterisk/voicemail VM_HOME=/var/spool/asterisk/mwi SIP_SERVER=71.110.20.33 SIP_PORT=5060 SIP_FROM=vmserver SIP_PASSWORD=heslo DB_NAME=ser DB_SERVER=10.255.237.18 DB_OPTIONS="--skip-column-names --connect_timeout=3" DB_USER=ser DB_PASSWORD=heslo # Allow glob expansions to return NULL shopt -s nullglob # Sanity check that VM_HOME exists, warn if it does not if [ ! -d $VM_HOME ] ; then echo "Cannot chdir to VM_HOME, exiting" exit 1 fi for file in $VM_HOME/* do file=$( basename $file @default ) # The file existance check may seem redundant, but it is theoretically possible # for this script to take longer than 1 cron cycle to run, or for it to be run # by hand while another copy is running, so we really do need this check. if [ -e ${VM_HOME}/${file}@default ] && [ ${#file} -eq 10 ]; then QUERY="SELECT subscriber.domain FROM subscriber, location WHERE subscriber.username='$file' AND subscriber.username = location.username LIMIT 1" DOMAIN=`mysql ${DB_OPTIONS} -h ${DB_SERVER} -u${DB_USER} -p${DB_PASSWORD} -e "$QUERY" ${DB_NAME}` # If the MySQL call fails, then $DOMAIN will hold the error message instead if [ $? -ne 0 ] ; then echo "Error connecting to MySQL: $DOMAIN" exit 2 fi if [ -n "${DOMAIN}" ]; then MAILBOX=$VM_ROOT/default/${file:0:3}/${file:3:3}/${file:6:4} # Sanity check that INBOX exists, warn if it does not if [ ! -d ${MAILBOX} ]; then echo "${MAILBOX} does not exist" continue fi # Delete all messages with a zero length. If the subscriber still # has voice mail files in their INBOX after this step then notify # them even if the total is zero, which will extinguish their MWI for message in ${MAILBOX}/INBOX/*.gsm do if [ ! -s $message ]; then GSM=`basename $message .gsm` rm -f ${MAILBOX}/INBOX/$GSM.* fi done TOTAL_MESSAGES=0 NEW_MESSAGES=0 OLD_MESSAGES=0 if [ -d ${MAILBOX}/INBOX ]; then tmp=( $( echo ${MAILBOX}/INBOX/*.txt ) ) NEW_MESSAGES=${#tmp[*]} else NEW_MESSAGES=0 fi if [ -d ${MAILBOX}/Old ]; then tmp=( $( echo ${MAILBOX}/Old/*.txt ) ) OLD_MESSAGES=${#tmp[*]} else OLD_MESSAGES=0 fi TOTAL_MESSAGES=$(($NEW_MESSAGES + $OLD_MESSAGES)) if [ "$NEW_MESSAGES" == "0" ]; then HAS_NEW="no" else HAS_NEW="yes" fi CONTENT_LENGTH=$(( 34 + ${#HAS_NEW} + ${#NEW_MESSAGES} + ${#OLD_MESSAGES} )) SEQUENCE=$( printf "%06d" $RANDOM ) ( cat <<-EOM NOTIFY sip:${file}@${DOMAIN} SIP/2.0 From: <sip:${SIP_FROM}@${DOMAIN}> To: <sip:${file}@${DOMAIN}> Contact: <sip:${file}@${SIP_SERVER}> Call-ID: ${SEQUENCE}@${SIP_SERVER} CSeq: ${SEQUENCE} NOTIFY Event: message-summary Content-Type: application/simple-message-summary Content-Length: ${CONTENT_LENGTH} Messages-Waiting: ${HAS_NEW} Voicemail: ${NEW_MESSAGES}/${OLD_MESSAGES} EOM ) | /usr/bin/sipsak shoot -a ${SIP_PASSWORD} -s sip:${file}@${DOMAIN} -f - if [ $? -ne 0 ] ; then echo "sipsak error for $file" fi else echo "DOMAIN is null for $file" fi else if [ ${#file} -ne 10 ] ; then echo "Extension was not 10 digits: '$file' (${#file})" else echo "Problem with file: ${VM_HOME}/${file}@default" fi fi rm ${VM_HOME}/${file}@default done exit 0 On Tue, 1 Mar 2005 10:06:56 -0500, Patrick Baker <patricksbaker(a)gmail.com> wrote: > could you show me an example of the sipsak command for the cisco phone > - I can then just make an agi that will call it after vm is left. > > Best regards, > > Patrick B > > > On Tue, 1 Mar 2005 10:03:35 -0500, Java Rockx <javarockx(a)gmail.com> wrote: > > Since the SIP UA is registered with SER you'll need to have Asterisk > > send a NOTIFY message to the user by means of SER. > > > > So you can send the NOTIFY to SER with a <From:> header as asterisk > > and the <To> / R-URI as the Cisco UA. > > > > FYI, we do this using sipsak to send the NOTIFY to ser. This is > > triggered either using the externnotify parameter in the Asterisk > > voicemail.conf file and by CRON for periodic MWI updates. > > > > Regards, > > Paul > > > > > > On Tue, 1 Mar 2005 09:53:43 -0500, Patrick Baker > > <patricksbaker(a)gmail.com> wrote: > > > Hello, > > > > > > I'm sure this may have been asked but, I just wanted clarify how this > > > would be accomplished. I'm looking to have the MWI light on my cisco > > > phone turn on when a message is left in VM. > > > > > > Here is my current situation: > > > > > > sip user -> ser -> asterisk > > > > > > Best regards, > > > > > > Patrick b. > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > >

2 1

[Serusers] SER advantages
by Felipe Louback 18 May '05

18 May '05

I am employing SER and Asterisk in my University and I was questioned about the reason I am using SER and not only Asterisk. What does SER do that Asterisk do not? What are the advantages of using SER? Thanx, Felipe Louback -- "Do not pray for easylives; pray to be stronger men! Do not pray for tasks equal to your powers; pray for powers equal to your tasks. Then the doing of your work shall be no miracle, but you shall be a miracle." Phillips Brook

2 1

[Serusers] Problem with avpops (bug?): dbrow contains NULL fields
by Michael Ulitskiy 18 May '05

18 May '05

Hello, I'm having problems with avpops module that looks to me as a bug. I'm using Postgres as database backend. Here's the table structure: users=# \d usr_preferences Table "ser.usr_preferences" Column | Type | Modifiers -----------+-----------------------------+---------------------------------------- uuid | character varying(64) | not null default ''::character varying username | character varying(100) | not null default ''::character varying domain | character varying(128) | not null default ''::character varying attribute | character varying(32) | not null default ''::character varying value | character varying(128) | not null default ''::character varying type | integer | not null default 0 modified | timestamp without time zone | not null default now() Indexes: "usr_preferences_pkey" primary key, btree (attribute, username, "domain") As you can see all fields defined as NOT NULL and there can't be any NULL values. Here's what I do in ser.cfg: # -- avpops params -- modparam("avpops", "avp_url", "postgres://SERDB_USER:SERDB_PW@DBHOST/users") modparam("avpops","avp_aliases","alias_p=I:100;alias_s=I:101") modparam("avpops","avp_table","usr_preferences") route { if (!avp_db_load("$ruri","$alias_p/usr_preferences")) { log(1,"avp not found"); } else { avp_print(); } sl_reply_error(); } When I'm sending the following INVITE: INVITE sip:12128121207@myserbox.com SIP/2.0 ... Here's the query that's sent to the database: select value,attribute,type from usr_preferences where username='12128121207' AND attribute='100' Here's the result of the query in psql: users=# select value,attribute,type from usr_preferences where username='12128121207' AND attribute='100'; value | attribute | type -------------------------------------+-----------+------ sip:12128121207@anotherbox.com | 100 | 2 (1 row) As you can see there's no NULL values and everything looks correct so far. But here's what I get in ser log: 0(12042) val2str(): converting 100, 3 0(12042) PG[217] str2valp got string sip:12128121207@anotherbox.com 0(12042) PG[217] str2valp got string 100 0(12042) PG[193] str2valp got int 2 0(12042) ERROR:avpops:dbrow2avp: dbrow contains NULL fields I tried it with ser 0.9.0, cvs stable and cvs head all with the same result. If this is not a bug, please point me to what I'm doing wrong. Otherwise please consider it a bug report. I assume that this is PostgreSQL support related as it seems people on the list were able to get it work with MySQL. Thank you, Michael

1 0

[Serusers] PSTN Provider
by Emmanuel Odoom 18 May '05

18 May '05

Hello, I am looking for a PSTN provider who I can pass International calls for mutual benefit. I provide PC to PC calls only but I figure that the real advantage is to make my clients be able to call to the family and friends in Europe and America at cheaper rates. Someone advice appropriately. Thank you in advance. Emmanuel. _____ I am using the free version of SPAMfighter for private users. It has removed 890 spam emails to date. Paying users do not have this message in their emails. Try SPAMfighter <http://www.spamfighter.com/Product_Info.asp?> for free now!

1 0

[Serusers] conference calls
by Aisling 18 May '05

18 May '05

Hello, I am not sure if someone has set up a SER/Asterisk platform for conference calls but if they have I would appreciate some help! - I was wondering how I would modify the ser.cfg given in the onsip document so that when users dial a particular number e.g. 2222, they are forwarded to asterisk where a conference call can be set up. I have set up asterisk as follows: [Meetme.conf] conf => 12345 [extensions.conf] exten => 222,1,Meetme,12345. If someone could give me some sort of example, it would help a lot. Thank you, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

1 0

[Serusers] [Fwd: Sip-Group problem]
by Tiziano 18 May '05

18 May '05

Hi, can sb help me? Thanks, Tiziano ------- Messaggio inoltrato ------- Da: Tiziano <ml(a)asdasd.it> A: serusers(a)lists.iptel.org Oggetto: Sip-Group problem Data: Tue, 17 May 2005 17:39:53 +0200 Hi, i'm trying to setup ser+radius groups... but when i do a: radius_is_user_in("Request-URI", "ld") i get: May 17 16:52:44 rad ser[648]: rc_avpair_new: unknown attribute 0 May 17 16:52:44 rad ser[648]: radius_is_user_in(): Error adding Sip-Group attribute which is the problem? i've Sip-Group in dictionary of both radiusclient and freeradius... Thanks for help, Tiziano -- Tiziano <ml(a)asdasd.it>

1 0

[Serusers] SERWEB error logs
by Llanos Serna García-Conde 18 May '05

18 May '05

2 1

[Serusers] phplib duplicated
by Llanos Serna García-Conde 18 May '05

18 May '05

2 1

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