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37302 discussions

[Serusers] CVS version current, gives this error

by Lenny Tropiano

Default config (ser.cfg installed from gmake install), gives this error (I bumped up the debug level and had it log to stderr): p2:[root@enterprise.icus.com] 31 $ ser 0(62901) loading module /usr/local/lib/ser/modules/sl.so 0(62901) loading module /usr/local/lib/ser/modules/tm.so 0(62901) loading module /usr/local/lib/ser/modules/rr.so 0(62901) loading module /usr/local/lib/ser/modules/maxfwd.so 0(62901) loading module /usr/local/lib/ser/modules/usrloc.so 0(62901) loading module /usr/local/lib/ser/modules/registrar.so 0(62901) find_param_export: found <db_mode> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(62901) find_export: found <mf_process_maxfwd_header> in module maxfwd_module [/usr/local/lib/ser/modules/maxfwd.so] 0(62901) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(62901) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(62901) find_export: <rewriteFromRoute> not found 0(62901) parse error (80,19-20): unknown command, missing loadmodule? 0(62901) find_export: found <save> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(62901) find_export: found <lookup> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(62901) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(62901) find_export: found <t_relay> in module tm [/usr/local/lib/ser/modules/tm.so] 0(62901) find_export: found <sl_reply_error> in module sl_module [/usr/local/lib/ser/modules/sl.so] ERROR: bad config file (1 errors) 0(62901) INFO: signal 15 received 0(62901) Memory status (pkg): 0(62901) qm_status (0x80a1780): 0(62901) heap size= 1046128 0(62901) used= 0, used+overhead=2448, free=1043680 0(62901) max used (+overhead)= 2448 0(62901) dumping all allocked. fragments: 0(62901) dumping free list stats : 0(62901) hash= 137. fragments no.: 1 0(62901) ----------------------------- --- Lenny Tropiano E-mail: lenny(a)voiping.com Partner, Networking Specialist Pager: pager-lenny(a)voiping.com VoIPing, LLC URL: http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 Mobile: 512-698-VOIP [8647]

21 years, 9 months

[Serusers] Some compile time errors for FreeBSD of CVS Tree source (sip_router)

by Lenny Tropiano

pass_fd.c: added #include <sys/uio.h> tcp_main.c: moved #include <sys/select.h> underneath #include <sys/types.h.> Still getting this: $ gmake Makefile.defs:331: Old gcc detected (2.9x), use gcc >= 3.1 for better results gcc -g -O9 -funroll-loops -Wcast-align -Wall -m486 -malign-loops=4 -DNAME='"ser"' -DVERSION='"0.8.11pre6-tcp5-tm"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 2.95"' -D__CPU_i386 -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DDNS_IP_HACK -DUSE_IPV6 -DVOICE_MAIL -D_TOTAG -DUSE_TCP -DFAST_LOCK -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -c tcp_main.c -o tcp_main.o tcp_main.c: In function `tcp_send': tcp_main.c:351: `MSG_NOSIGNAL' undeclared (first use in this function) tcp_main.c:351: (Each undeclared identifier is reported only once tcp_main.c:351: for each function it appears in.) gmake: *** [tcp_main.o] Error 1 Cannot find proper #define for MSG_NOSIGNAL The flags, according to "send()" man page is: The flags parameter may include one or more of the following: #define MSG_OOB 0x1 /* process out-of-band data */ #define MSG_PEEK 0x2 /* peek at incoming message */ #define MSG_DONTROUTE 0x4 /* bypass routing, use direct interface */ #define MSG_EOR 0x8 /* data completes record */ #define MSG_EOF 0x100 /* data completes transaction */ The flag MSG_OOB is used to send ``out-of-band'' data on sockets that support this notion (e.g. SOCK_STREAM); the underlying protocol must also support ``out-of-band'' data. MSG_EOR is used to indicate a record mark for protocols which support the concept. MSG_EOF requests that the sender side of a socket be shut down, and that an appropriate indication be sent at the end of the specified data; this flag is only implemented for SOCK_STREAM sockets in the PF_INET protocol family, and is used to implement Transaction TCP (see ttcp(4)). MSG_DONTROUTE is usually used only by diagnostic or routing programs. --- Lenny Tropiano E-mail: lenny(a)voiping.com Partner, Networking Specialist Pager: pager-lenny(a)voiping.com VoIPing, LLC URL: http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 Mobile: 512-698-VOIP [8647]

21 years, 9 months

[Serusers] Via tags question...

by lgfausak＠august.net

This is a protocol question, I don't know if it is directly related to ser. Concerning the Via tags, should any proxy/gateway modify tags that don't belong to it, or should they simply be copied verbatum? For example, I have a situation where I send an INVITE message, and the TRYING response I receive modifies one of the Via: tags. I don't think this is right. Here are the two packets in question: # U 2003/02/19 18:36:20.276844 164.9.42.25:5060 -> 184.294.324.2:5060 INVITE sip:12143357976@184.294.324.2:5060 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:12143357976@164.9.42.25;branch=0>. From: <sip:8178863582@216.87.145.22> ;tag=d8579116-fdc-3e542303-23d34-4a84. To: <sip:12143357976@augustvoice.net> ;user=phone. Call-ID: 59fd88-d8579116-fdc-3e542303-23d2f-3897(a)216.87.145.22. CSeq: 2 INVITE. Via: SIP/2.0/UDP 164.9.42.25;branch=z9hG4bKcc26.d3c125d.0. Via: SIP/2.0/UDP 216.87.145.22:4060 ;branch=z9hG4bK-3e542304-23dfc-ea2. Contact: <sip:8178863582@216.87.145.22:4060>. User-Agent: FXS_GW (sip1028a.bin). Content-Type: application/SDP. Content-Length:291. P-hint: LD GATEWAY. . v=0. o=FXS_GW 12367 0 IN IP4 216.87.145.22. s=Audio Session. i=Audio Session. c=IN IP4 216.87.145.22. t=0 0. m=audio 16384 RTP/AVP 0 18 4 8 96. a=fmtp:96 0-11. a=rtpmap:0 PCMU/8000/1. a=rtpmap:18 G729/8000/1. a=rtpmap:4 G723/8000/1. a=rtpmap:8 PCMA/8000/1. a=rtpmap:96 telephone-event/8000. And your response is: # U 2003/02/19 18:36:20.902858 184.294.324.2:5060 -> 164.9.42.25:5060 SIP/2.0 100 Trying . Via: SIP/2.0/UDP 164.9.42.25;branch=z9hG4bKcc26.d3c125d.0. Via: SIP/2.0/UDP 216.87.145.22:4060. From: <sip:8178863582@216.87.145.22> ;tag=d8579116-fdc-3e542303-23d34-4a84. To: <sip:12143357976@augustvoice.net> ;user=phone;tag=27DC1DCC-F61. Date: Thu, 20 Feb 2003 00:36:20 GMT. Call-ID: 59fd88-d8579116-fdc-3e542303-23d2f-3897(a)216.87.145.22. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 2 INVITE. Allow-Events: telephone-event. Content-Length: 0. . See how the : Via: SIP/2.0/UDP 216.87.145.22:4060 ;branch=z9hG4bK-3e542304-23dfc-ea2. Has been changed to: Via: SIP/2.0/UDP 216.87.145.22:4060. Does anyone have any perl script to take a ngrep protocol dump and print it out in a pretty way? ---greg Greg Fausak August.Net Services, LLC

21 years, 9 months

[Serusers] Add a user to subscriber table.

by Ivan Vignola

Hi, I would add a user to subsriber table by sql command in my application (not using serctl script). What text strings I must use to calculate HA1 (=phplib_id) and ha1b values if USER_ID=ivan,PASSWORD=ivanpswd and REALM=192.168.11.237 ? Thanks in advance. Regards, Ivan Vignola. _________________________________________________________________ Vinci la nuova Nissan Micra con MSN Messenger! http://www.msn.it/messenger/

21 years, 9 months

[Serusers] serctl - tail problems on solaris8?

by Kamil Ince

bet i missed reading something, again :-/ but wasting too much time on this.. i get on solaris 8 the following: # echo $SIP_DOMAIN myser.my.net # echo $PW heslo # serctl add user1 hispw user1(a)my.net usage: tail [+/-[n][lbc][f]] [file] tail [+/-[n][l][r|f]] [file] new user added # serctl passwd user1 newpw usage: tail [+/-[n][lbc][f]] [file] tail [+/-[n][l][r|f]] [file] non-existent user of course the user has been added to the database... anyone came accross that before? tia -k

21 years, 9 months

[Serusers] [bug] TM module config

by Sebastien Pierre

Hi again, Here is how to get a segfault: <snip> #Load modules required by a stateful User Agent Service loadmodule "modules/tm.so" #SER timer parameters modparam("tm", "fr_inv_timer", "10"); modparam("tm", "fr_timer", "10"); </snip> This happened after quoting the numbers in the modparam. Tested with ser-0.8.10 on Linux RH 8.0 -- Sebastien

21 years, 9 months

[Serusers] Logging with SER

by Sebastien Pierre

Hi all, I finally found solutions to my problem: 1) It is easier when setting "fork=no" because then ser does not daemonise and stderr is then attached to the terminal in which I started ser 2) The configuration file should set a debugging level ("set debug=1"), and use a number equal or lower to the debug level in the log, like 'log(0, "Hello World")' So my HelloWorld.cfg is as follows: <snip> debug=0 log_stderror=yes fork=no route[0] { log(0, "Hello World!"); } </snip> And it works with "ser -f HelloWorld.cfg" Cheers, -- Sebastien

21 years, 9 months

Re: [Serusers] SER logging (and the mighty HelloWorld.cfg)

by Sebastien Pierre

Hi Nils! Le mar 18/02/2003 à 16:56, Nils Ohlmeier a écrit : > Hard to say without seeing the config file. But '-E' as command line > parameter should work and will overwrite any value in your config > file. Here it is (the HelloWorld.cfg): <snip> log_stderror=yes route[0] { log("Hello World!"); } </snip> Here is the SIP message I send to SER: <snip> INVITE sip:alice@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5070 To: Alice <sip:alice@localhost> From: Me <sip:me@localhost> Call-ID:123456789@127.0.0.1 CSeq: 1 INVITE Content-Length: 0 </snip> Here is the answer I get from SER: <snip> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 127.0.0.1:5070 To: Alice <sip:alice@localhost>;tag=d907c037823644515dfe0ede38ca9976.de25 From: Me <sip:me@localhost> Call-ID:123456789@127.0.0.1 CSeq: 1 INVITE Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 127.0.0.1:5060 "Noisy feedback tells: pid=9995 req_src_ip=127.0.0.1 in_uri=sip:alice@127.0.0.1 out_uri=sip:alice@127.0.0.1 via_cnt==1" </snip> And here is what I get from SER in stdout/stderr: <snip> $ ser -E HelloWorld.cfg Listening on 127.0.0.1 [127.0.0.1]::5060 172.25.49.134 [172.25.49.134]::5060 Aliases: XXX.alcatel.fr:5060 localhost:5060 localhost.localdomain:5060 XXX:5060 </snip> And from /var/log/messages <snip> Feb 18 17:33:21 frilld0030711 ser[9994]: INFO: udp_init: SO_RCVBUF is initially 65535 Feb 18 17:33:21 frilld0030711 ser[9994]: INFO: udp_init: SO_RCVBUF is finally 131070 Feb 18 17:33:21 frilld0030711 ser[9994]: INFO: udp_init: SO_RCVBUF is initially 65535 Feb 18 17:33:21 frilld0030711 ser[9994]: INFO: udp_init: SO_RCVBUF is finally 131070 Feb 18 17:33:21 frilld0030711 ser[10003]: INFO: fifo process starting: 10003 Feb 18 17:33:21 frilld0030711 ser[10003]: SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... </snip> So there is sadly no "HelloWorld!" somewhere... would it be because of the "404 not found" error I get ? TIA, -- Sebastien

21 years, 9 months

[Serusers] Simple SER Use Case

by Sebastien Pierre

Hi all, I am trying to implement a very simple call forwarding use case with SER. This use case is directly taken from a book entitled "Internet Applications with SIP". Here are the actors: - SIP caller, wants to call <alice@domain> - Proxy server of <domain> - SIP phone which is default terminal for <alice(a)domain.com> - Voicemail server Here is the scenario: 0) SIP caller INVITEs <alice@domain> 1) Proxy receives INVITE 2) Proxy forwards INVITE to alice default SIP phone 3) Proxy responses 100 TRYING to SIP caller 4) SIP phone responses 100 TRYING to proxy 5) SIP phone responses 180 RINGING to proxy 6) Proxy reponses 180 RINGING to caller 7) After timeout, proxy send CANCEL to default SIP phone 8) SIP phone responses OK to proxy... ... X) Proxy forward original INVITE to voice mail server... The fact is that the proxy first notifies the caller that it is trying to locate alice, listens for "180 RINGING" and forward it to the caller, then cancels on the SIP phone side because of a timeout -- then the proxy forwards the INVITE to the voice mail. I have tried some configurations, but I did not see how to listen for the 180 RINGING and forward them to the caller... is there a way to bind a "route" to a provisional response like: route[0] { ... #In case a 180 occur, we execute route[1] t_on_provisional("180", "1"); #But we're also waiting for an OK or a timer #event to arrive t_on_negative("2"); t_relay(); ... } route[1] { #Forwards the 180 Ringing from the SIP phone to #the caller t_relay(); } BTW, I was wondering if t_on_negative is a *blocking* action ? TIA, -- Sebastien

21 years, 9 months

[Serusers] ser-mysql for solaris

by Kamil Ince

hello, can someone help me with the database details for ser? I got only solaris available, and couldnt find ser-mysql as "non-rpm" tia -k

21 years, 9 months

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