- sr-users - kamailio.org

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

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[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

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[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

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[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!

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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

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[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.

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[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

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[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.

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[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.

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[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

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[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(); }; } #################################

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[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

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[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.

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[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

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[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"

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[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

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[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

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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. > > > >. > > > > > > >

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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

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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.

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[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 >> >> > > >

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[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

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[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/ .

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[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

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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 >>> >>> >>> >>> >> >> >> > > >. > > >

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

19 May '05

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[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

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[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

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[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>

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[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

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

19 May '05

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[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.

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[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

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[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

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

19 May '05

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[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

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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.

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[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?

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[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

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[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,

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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 > > > > > >

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[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

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[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!

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[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

[Serusers] allow_trusted
by Iqbal 18 May '05

18 May '05

Hi has anyone got this to work, it seems i cant get the pattern correct :-) I have if (!allow_trusted()) { log(1, loop1) } in allows trsuted I have a.b.c.d | any | sip:[0-9]*@a.b.c.d | but i dont see any calls being made to the db, to do a lookup or anything, how can I see what is being matched. Iqbal

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[Serusers] Multiple Registration Servers
by Darren Nay 18 May '05

18 May '05

Hey All, This is a long email, so please bare with me... Up until now we have used our primary SER serve as both registrar and call routing engine. However, we've recently hit a growth point where the registration load on our servers is high enough that it's affecting our call routing. We currently have about 4,000 IAD's all registering every 5 minutes. This works out to be almost 13.5 regs every second, which in itself is a lot .. If they were sequential. However, at times we will get hundreds of IADs all attempting to register simultaneously (within seconds of each other) and it will keep the server busy for a few moments and cause SIP timeouts on calls that are in the process of being setup. Currently I am using the auth_db module with MySQL for client registration authentication. I believe that the MySQL authentication is what is causing the high load on the server. What I want to do is try to offload the the registrations from our primary routing server, yet, I need the primary server to be stateful of the all registered clients. -or- Find a more efficient way to authenticate client registrations so that the load is not so high. Problem is that most of our customers are home-based VoIP telephone customers behind a Linksys cable/DSL router. So, in order to avoid NAT issues we must have all of our traffic which terminates to the IAD from our PSTN provider come from the same IP address that the the IAD registered from. So I can't use another server and then t_replicate the requests back to the routing server .. Because then the registrar / IP address that will be open on the customers router NAT gateway / firewall will differ from the IP that will terminate the traffic to the IAD from our system. So - my question is .. How do I take load off of the primary server?? Would implementing radius for authentication fix this problem? I am very familiar with radius (using Freeradius) and could easily set this up, but I would prefer not to go to the trouble if it's not going to help with the load. Has anyone else had an issue with client authentication via MySQL and auth_db causing a high load with SER? I am hoping that you fine folks might be able to point me in the right direction. Thanks so much! Darren Nay VoIP Network Development Ionosphere, Inc dnay(a)ionosphere.net

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[Serusers] [serunsers] Loose routing
by michael p 18 May '05

18 May '05

hi i didn't understand what is the loose routing someone can explain me clearly please? thanks M.

1 0

[Serusers] More on load balancing: LVS + SRVs, SER + Asterisk
by Pietro Ravasio 18 May '05

18 May '05

Hi all, I've read different approaches towards load balancing, expecially in these two threads: http://lists.iptel.org/pipermail/serusers/2005-May/019107.html http://lists.iptel.org/pipermail/serusers/2005-April/018058.html Unfortunately it seems that at the moment there's not a "state of the art" solution, so here is my "two cent" idea. Project goal: to develop a near-linearly scalable platform with NAT traversal support and call control. Proposed solution: multiple instances of SER will be reacheable via an unique virtual IP handled by LVS (with udp persistence), all SERs will use the same MySQL backend, an array of Asterisk servers will act as SBCs (Session Border Controller). Every SER will have a "third level" SIP domain associated to it, for instance UA_A will be registered as UA_A(a)ser1.domain.com and UA_B will be registered as UAC_B(a)ser2.domain.com. UAC_A will be able to call UA_B as "B" or as "B(a)domain.com"; SRV records for "domain.com" will be LVS virtual IP, SRV records for third level domain will be internal, not visible to Uas and associated to IP addresses:ports of every SER instance. To achieve this, we will need to store a domain for any UA in the "location" table, any instance of SER will do this, UAs don't need to know anything about "internal" domains. (Is this to be considered an "hack"?) ## Without call control, there would be only one call leg: When UA_A calls UA_B, its INVITE will reach the same SER it has previously registered to (SER_1, thanks to LVS persistence), SER_1 will then look for UA_B and will notice that it is registered as UA_B(a)ser2.domain.com, DNS will be queried and SER_2 will be contacted in order to handle INVITE and send it to UA_B through already opened UDP channel (resolving NAT). ## Instead, we want to be able to control the call process and to handle RTP nat too, so let's use Asterisk: -First Leg (left): UA_A calls UA_B, INVITE reaches SER_1 (LVS persistence), SER_1 finds out that UA_B is @ domain ser2.domain.com but its dialplan tells him to send the INVITE to one of the Asterisk servers (one of them). SER modifies the INVITE sent to * with Contact: (ip:port) of SER_2. -Second Leg (right): Asterisk (after obtaining authorization and call duration from AAA server) will look for UA_B at SER_2 and send the INVITE to SER_2 with SDP modified for the RTP, SER_2 will then forward INVITE (solving NAT problems) to UA_B ip:port (parameters taken from memory or DB). This kind of platform would permit call forking too: if UA_B were registered through two different proxies SER_1 would fork the call to two different "call instances" of Asterisk(s). Note that we want Asterisk to mangle SDP only when both UA_A and UA_B have private IP addresses Have you got any suggestion/comment? (expecially regarding: - more SERs using the same MySQL backend - binding a third level domain at the proxy Thanks and kind regards! Pietro Ravasio -- Ing. Pietro Ravasio - pietro.ravasio(a)abbeynet.it Enterprise Market Manager Abbeynet S.p.A. - http://www.abbeynet.it TEL: +390702109044 GSM: +393282172055 FAX: +390702109055

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[Serusers] SER -> Asterisk ==> Call failed: 403 Forbidden
by davor jovanovic 18 May '05

18 May '05

Hi all, I have a problem with forwarding calls from SER to Asterisk. XXX -> SER -> Asterisk -> YYY I have XXX UAC which is registered in SER and YYY UAC which is registered in Asterisk. When I forward call from SER to Asterisk I get "Call failed: 403 Forbidden" from Asterisk. Everything works fine if I register XXX UAC in Asterisk too (SER forwards call to Asterisk and it works). I think I don't understand this registering SER to Asterisk and other way and I would appreciate if someone could help with this. Best regards sip.conf [general] port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind SIP channel to context=ser autocreatepeer=yes register => 2000:1234@ser_ip_address/5678 [ser_ip_address] type=friend username=2000 secret=1234 fromdomain=srce.hr host= ser_ip_address [5678] type=friend username=5678 host=dynamic canreinvite=no extensions.conf [ser] exten => 5678,1,Dial(SIP/5678,20) exten => 5678,2,Voicemail2(u5678) exten => 5678,3,MusicOnHold() exten => 5678,102,Voicemail2(b5678) exten => 5678,103,Hangup

4 4

Re: [Serusers] Open Source SIP Phone
by Jay Manadat 18 May '05

18 May '05

Is it possible to use the sipXPhone from http://www.sipfoundry.org/sipXphone/ with SER? Does anybody on the list have an experience with this? Thanks -Jay --- Kofi Obiri-Yeboah <kofi(a)radiocomplex.com> wrote: > try sip-communicator at sip-communicator.org > -- > Worrying Is Like Paying Interest On A Loan You Are > Yet To Take > Be Happy! Don't Worry, > > > Quoting Vincent Verdot <vincent.verdot(a)int-evry.fr>: > > > Thanks, > > I used X-ten lite but it's not open source.. > > > > Regards, > > Vincent Verdot. > > ----- Original Message ----- > > From: Java Rockx > > To: Vincent Verdot > > Cc: serusers(a)lists.iptel.org > > Sent: Tuesday, May 17, 2005 2:24 PM > > Subject: Re: [Serusers] Open Source SIP Phone > > > > > > Kphone and LinPhone work on Linux. X-ten Lite is > free for Windows and Mac. > > > > You could even write your own by using a sip > stack such as osip. > > > > Regards, > > Paul > > > > > > On 5/17/05, Vincent Verdot > <vincent.verdot(a)int-evry.fr> wrote: > > Hi everybody, > > I'm looking for a good open source sip-phone, > can anyone help me ? > > > > > _______________________________________________ > > 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 > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

Fwd: RE: [Serusers] Multiple Registration Servers
by Corey S. McFadden 18 May '05

18 May '05

I'm hoping to clarify a couple questions on clustering SER in high-volume environments as well. Assuming authentication is RADIUS, wouldn't clustering something like what Darren is talking about be as simple as installing multiple SER boxes and just doing some DNS round-robin? Would that introduce any registration issues? -Corey >Darren Nay wrote: > > Senad, Thanks for your response. > > > > There are several reasons. > > > > The 2 biggest are.. > > > > 1 - We offer our customers the option to forward calls when an IAD is > > not registered .. In order to do that the registration interval needs > > to be relatively low. > >Fine... But still I am curious why u need to do in order to forward the >calls. > > > > > 2 - We've had some issues with setting registration intervals higher > > than 10 minutes. It seems to work about 99.5% of the time, but > > occaisionally an endpoint won't reregister before the reg interval on > > SER times out. I believe this is a bug in our endpoint and have been > > working with them on that. Setting the reg interval below 10 minutes > > has eliminated that problem for the time being .. However, even if > > this problem was fixed by the IAD manufacturer we still have the > > first reason (above) that would keep us from being able to increase > > the interval. > > >Well. apart from fixing the issue with your IAD, you could use DNS SRV >record. >Each time, IAD wants to re-register it will use DNS SRV record, hence >"hitting" different server. > >This way, u will need more than one SER server each getting its data from >central network database. >If you need cluster file system, then you could use GFS or similar. > >Regards, > >Senad > >PSS!!! > Which IAD are you using ? ********************************************* This message has been scanned for viruses and dangerous content, and is believed to be clean.

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[Serusers] Problems Compiling SER-0.9.0
by Felipe Martins 18 May '05

18 May '05

Hi Everybody, I'm experiencing problems trying to compile SER-0.9.0. The file I got from onsig.org is called ser-0.9.0.GettingStarted.1.1.tar.gz. I try to compile it exactly the howto orders to but it keeps giving me the same error on compiling RTPProxy. The compile command is # ./make_and_install It begins the compilation but at the end gives me the following error: ----- error ------------------------------ " THIS SCRIPT HAS BEEN PROVIDED FROM ONsip.org AS A SUPPORT FOR THE GETTING STARTED DOCUMENT PLEASE GO TO ONsip.org TO DOWNLOAD THE DOCUMENT AND THE CONFIGURATION FILES THAT GO WITH IT IPTEL.ORG OR THE DEVELOPERS OF SER HAVE NO SUPPORT ON THIS SCRIPT USAGE OF THE SCRIPT IS YOUR OWN RESPONSIBILITY ------------------------------------------------- YOU WILL FIND THE OUTPUT IN make_and_install_output.txt IN THE CURRENT DIRECTORY ---------------------- Cleaning up the source tree... ---------------------- ---------------------- Making ser and all modules. This may take some time... Press ENTER to start ---------------------- COMPILING...... ---------------------- Configuring rtpproxy... ---------------------- ---------------------- Making rtpproxy... ---------------------- FOUND ERRORS! See make_and_install_output.txt for full output" ----- end of error ----------------------- When I go to make_and_install_output.txt to take a look at the errors it gives me errors as if there were no directory from the configuration files, but I'm on the directory ... What can be that ? -- 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] allow_trusted
by Ricardo Martinez 17 May '05

17 May '05

By the way.... Can i do the same but with a flat file? Thanks! Ricardo Martinez.- > -----Mensaje original----- > De: Iqbal [mailto:iqbal@gigo.co.uk] > Enviado el: Martes, 17 de Mayo de 2005 16:27 > Para: serusers(a)lists.iptel.org > Asunto: [Serusers] allow_trusted > > > > Hi > > has anyone got this to work, it seems i cant get the pattern > correct :-) > > I have if (!allow_trusted()) { log(1, loop1) } > > in allows trsuted I have > > a.b.c.d | any | sip:[0-9]*@a.b.c.d | > > but i dont see any calls being made to the db, to do a lookup or > anything, how can I see what is being matched. > > Iqbal > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

2 1

[Serusers] register to serweb
by Llanos Serna García-Conde 17 May '05

17 May '05

1 0

[Serusers] Open Source SIP Phone
by Vincent Verdot 17 May '05

17 May '05

Hi everybody, I'm looking for a good open source sip-phone, can anyone help me ?

3 3

Re: [Serusers] Open Source SIP Phone
by Samuel Osorio Calvo 17 May '05

17 May '05

I have just seen that there is a linux "executable" for X-ten version 2 with several "highly experimental" warnings...I gave it a try and the only I can say is that it requires a lot of resources, much more than kphone and linphone although X-Ten has lots of configuration options that I have not tried....and for sure it works with SER.... Samuel. Unclassified. >>> Java Rockx <javarockx(a)gmail.com> 05/17/05 02:24PM >>> Kphone and LinPhone work on Linux. X-ten Lite is free for Windows and Mac. You could even write your own by using a sip stack such as osip. Regards, Paul On 5/17/05, Vincent Verdot <vincent.verdot(a)int-evry.fr> wrote: > > Hi everybody, > I'm looking for a good open source sip-phone, can anyone help me ? > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > >

1 0

[Serusers] Sip-Group problem
by Tiziano 17 May '05

17 May '05

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

Re: [Serusers] bugs on alias or save location
by Cesc Santasusana 17 May '05

17 May '05

comment inline >>> "Edgardo O. Gonzales II" <edgardo.g(a)pacific.net.ph> 05/17/05 03:19PM >>> > > >route[2] { > > log(1, "REGISTER: Authenticating user\n"); > if (!radius_www_authorize("")) { > www_challenge("","0"); > break; }; > > save("aliases"); consume_credentials(); break; Why save("aliases")? even if you did if( !save("aliases") ) { save(aliases); consume...; break;} it is not needed. You just need to update the locatioin table, not the aliases. Just remove the line. > > if (!save("location")) { > sl_reply_error(); > }; >} Unclassified

1 0

[Serusers] Help on running Ser.
by Simith Nambiar 17 May '05

17 May '05

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

1 0

Re: [Serusers] bugs on alias or save location
by Edgardo O. Gonzales II 17 May '05

17 May '05

but I did not use save location on my config file .. what I defined is save alias and lookup alias only.. below is my running config file .. please do some comment .. thanks .. # ------------------------- request routing logic ------------------- # main routing logic route{ setflag(2); 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; }; if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); sl_send_reply("403", "Forbidden Unauthorized Domain"); break; }; if (uri=~"zipper.pacific.net.ph") { if ((method=="BYE")||(method=="ACK")){ setflag(2); } if (method=="REGISTER") { route(2); break; }; if (method=="INVITE") { route(3); break; }; if (uri!=myself) { route(1); break; }; } append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; } route[2] { log(1, "REGISTER: Authenticating user\n"); if (!radius_www_authorize("")) { www_challenge("","0"); break; }; save("aliases"); consume_credentials(); break; if (!save("location")) { sl_reply_error(); }; } route[3] { if (!radius_www_authorize("domain.com")) { www_challenge("domain.com","1"); break; } consume_credentials(); lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { if (method=="INVITE" || method=="ACK") { sl_send_reply("404", "Extension number does not exist"); break; }; sl_send_reply("404", "Extension number does not exist"); break; } else { setflag(3); t_relay(); break; }; route(1); } =========================== At 07:46 PM 5/17/2005, Samuel Osorio Calvo wrote: >Hi, > >the "normal" mechanism is to have two table, aliases and location. In the >first one you have a "permament" binding between a an alias and the user >name (AoR). In the latter, you update the binding between the username >(AoR) and the current location treating the incoming REGISTER with >save("location"). > >In your attached config file there is save("location"), which will update >the location table: > > > if (method=="REGISTER") { > > > ># Uncomment this if you want to use digest > >authentication > > if > >(!www_authorize("justser.com", "subscriber")) { > > > >www_challenge("justser.com", "0"); > > break; > > }; > > > > save("location"); > > break; > > }; > >, so you would have: > ALIASES LOCATION >900--->test test--->UA'sIP > >When you do lookup("location") in > > > # native SIP destinations are handled > >using our USRLOC DB > > if (!lookup("location")) { > > sl_send_reply("404", "Not > >Found"); > > break; > > }; > > }; > >you will rewrite the Req-URI to the binding existing in the location >table, that's why they can reach the user with the username. And you >sohuld do it because the user's UA will recognise requests with test in >the Req-URI as targeted to itself but not requests with 900 in the Req-URI. >That's why you should make something like >lookup("aliases"); >lookup("location"); >in the config file to handle aliases. > >If you don't want other people to reach the user with the AoR (I'm >wondering you would like suche feature...): you should make somehting like >(it's just a possible approach from many differents and don't know if it >will work always....just experiment): > >if( lookup("aliases") ){ > if( ! lookup("location") ){ > sl_send_reply("404", "Not Found"); > break; > } >} > >Hope it's clearer..... > >Samuel > > > >Unclassified. > >>> "Edgardo O. Gonzales II" <edgardo.g(a)pacific.net.ph> 05/17/05 01:20PM >>> > > >Hi! > > >I have a little problem with regards to alias and my routing table because >as I understand the logic of logging and authorization, client can login using >a username and password assigned by the administrator. > >For example, I have the following information > >username : test >password : test123 >extension / alias : 900 > >By enabling radius support, for aaa, I was able to login using my username >and password. >Other parties can also call my extension / alias number which is 900 but I >wonder why they can >also call my username which is test. I have save("alias") on my config. > >why is this happening. > >thanks, >ed > > >At 03:24 AM 5/17/2005, Jose Bertuzzi wrote: > >Hello guys; I have everything in place to acc into > >mysql. I am already logging some BYEs, INVITEs and > >ACKs. > > > >Where do I have to place the setflag statement to > >properly log messages from and/to gw 66.166.166.66 > >only? > > > >My ser.cfg is as follows: > > > > > ># $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei > >Exp $ > ># > ># simple quick-start config script > ># > > > ># ----------- global configuration parameters > >------------------------ > > > >#debug=3 # debug level (cmd line: -dddddddddd) > >#fork=yes > >#log_stderror=no # (cmd line: -E) > > > >/* Uncomment these lines to enter debugging mode > >debug=7 > >fork=no > >log_stderror=yes > >*/ > > > >check_via=no # (cmd. line: -v) > >dns=no # (cmd. line: -r) > >rev_dns=no # (cmd. line: -R) > >#port=5060 > >#children=4 > >fifo="/tmp/ser_fifo" > >#fifo_mode=0666 > > > > > ># ------------------ module loading > >---------------------------------- > > > ># Uncomment this if you want to use SQL database > >loadmodule "/usr/local/lib/ser/modules/mysql.so" > > > >loadmodule "/usr/local/lib/ser/modules/sl.so" > >loadmodule "/usr/local/lib/ser/modules/tm.so" > >loadmodule "/usr/local/lib/ser/modules/rr.so" > >loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > >loadmodule "/usr/local/lib/ser/modules/usrloc.so" > >loadmodule "/usr/local/lib/ser/modules/registrar.so" > >loadmodule "/usr/local/lib/ser/modules/acc.so" > >#loadmodule "/usr/local/lib/ser/modules/group.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", "calculate_ha1", yes) > ># > ># If you set "calculate_ha1" parameter to yes (which > >true in this config), > ># uncomment also the following parameter) > ># > >modparam("auth_db", "password_column", "password") > > > ># -- rr params -- > ># add value to ;lr param to make some broken UAs happy > >modparam("rr", "enable_full_lr", 1) > > > > > ># -- acc params -- > > > >modparam("acc", "db_url", > >"mysql://ser:heslo@localhost/ser") > >#modparam("acc", "log_level", 1) > >#modparam("acc", "log_flag", 1) > >modparam("acc", "db_flag", 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=~"justser.com") { > > > > if (src_ip==66.166.166.66 and > >method=="INVITE") { ## Llamadas desde PSTN > > forward(uri:host, uri:port); > > ## > > break; > > ## > > }; > > ## > > > > > > if (method=="REGISTER") { > > > ># Uncomment this if you want to use digest > >authentication > > if > >(!www_authorize("justser.com", "subscriber")) { > > > >www_challenge("justser.com", "0"); > > break; > > }; > > > > save("location"); > > break; > > }; > > > > > > > > if > >(uri=~"^sip:1305[0-9]*@justser.com") { ## > > prefix("3000#"); > > ## > > > >rewritehostport("66.166.166.66:5060"); ## > >TERMINATION > > forward(uri:host, uri:port); > > ## AREA CODE 305 > > #setflag(1); ## > > break; > > ## > > }; > > ## > > > > > > if > >(uri=~"^sip:1786[0-9]*@justser.com") { ## > > prefix("3000#"); > > ## > > > >rewritehostport("66.166.166.66:5060"); ## > >TERMINATION > > forward(uri:host, uri:port); > > ## AREA CODE 786 > > 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 > > > > setflag(1); > > > > if (!t_relay()) { > > sl_reply_error(); > > }; > > > >} > > > > > > > > > >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

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[serusers] NAT
by michael p 17 May '05

17 May '05

hi how can i traverse a NAT ? i heard about nathelpser utility but i found few which one for what? thanks M.

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Re: [Serusers] acc module with mysql support
by Gorka Guridi 17 May '05

17 May '05

Before compiling. In acc makefile, uncomment: DEFS+=-DSQL_ACC In main Makefile exclude_modules?= cpl cpl-c extcmd \ snmp postgres \ im \ jabber mysql \ /*remove mysql*/ auth_radius group_radius uri_radius Be sure that mysql module is rightly compiled, and mysql.so file is created. In ser.cfg, the basic lines to enable accounting are: loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/acc.so" modparam("acc", "db_url", "sql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 1) //See if you put "log_flag", in this case you are logging in syslog. In route () { setflag(1); //All will be marked with flag one enabled. This flag must be the same that you are watching with "db_flag" parameter. In my case, only this is necessary to obtain an accounting. Best Regards

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Re: [Serusers] bugs on alias or save location
by Samuel Osorio Calvo 17 May '05

17 May '05

Hi, the "normal" mechanism is to have two table, aliases and location. In the first one you have a "permament" binding between a an alias and the user name (AoR). In the latter, you update the binding between the username (AoR) and the current location treating the incoming REGISTER with save("location"). In your attached config file there is save("location"), which will update the location table: > if (method=="REGISTER") { > ># Uncomment this if you want to use digest >authentication > if >(!www_authorize("justser.com", "subscriber")) { > >www_challenge("justser.com", "0"); > break; > }; > > save("location"); > break; > }; , so you would have: ALIASES LOCATION 900--->test test--->UA'sIP When you do lookup("location") in > # native SIP destinations are handled >using our USRLOC DB > if (!lookup("location")) { > sl_send_reply("404", "Not >Found"); > break; > }; > }; you will rewrite the Req-URI to the binding existing in the location table, that's why they can reach the user with the username. And you sohuld do it because the user's UA will recognise requests with test in the Req-URI as targeted to itself but not requests with 900 in the Req-URI. That's why you should make something like lookup("aliases"); lookup("location"); in the config file to handle aliases. If you don't want other people to reach the user with the AoR (I'm wondering you would like suche feature...): you should make somehting like (it's just a possible approach from many differents and don't know if it will work always....just experiment): if( lookup("aliases") ){ if( ! lookup("location") ){ sl_send_reply("404", "Not Found"); break; } } Hope it's clearer..... Samuel Unclassified. >>> "Edgardo O. Gonzales II" <edgardo.g(a)pacific.net.ph> 05/17/05 01:20PM >>> Hi! I have a little problem with regards to alias and my routing table because as I understand the logic of logging and authorization, client can login using a username and password assigned by the administrator. For example, I have the following information username : test password : test123 extension / alias : 900 By enabling radius support, for aaa, I was able to login using my username and password. Other parties can also call my extension / alias number which is 900 but I wonder why they can also call my username which is test. I have save("alias") on my config. why is this happening. thanks, ed At 03:24 AM 5/17/2005, Jose Bertuzzi wrote: >Hello guys; I have everything in place to acc into >mysql. I am already logging some BYEs, INVITEs and >ACKs. > >Where do I have to place the setflag statement to >properly log messages from and/to gw 66.166.166.66 >only? > >My ser.cfg is as follows: > > ># $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei >Exp $ ># ># simple quick-start config script ># > ># ----------- global configuration parameters >------------------------ > >#debug=3 # debug level (cmd line: -dddddddddd) >#fork=yes >#log_stderror=no # (cmd line: -E) > >/* Uncomment these lines to enter debugging mode >debug=7 >fork=no >log_stderror=yes >*/ > >check_via=no # (cmd. line: -v) >dns=no # (cmd. line: -r) >rev_dns=no # (cmd. line: -R) >#port=5060 >#children=4 >fifo="/tmp/ser_fifo" >#fifo_mode=0666 > > ># ------------------ module loading >---------------------------------- > ># Uncomment this if you want to use SQL database >loadmodule "/usr/local/lib/ser/modules/mysql.so" > >loadmodule "/usr/local/lib/ser/modules/sl.so" >loadmodule "/usr/local/lib/ser/modules/tm.so" >loadmodule "/usr/local/lib/ser/modules/rr.so" >loadmodule "/usr/local/lib/ser/modules/maxfwd.so" >loadmodule "/usr/local/lib/ser/modules/usrloc.so" >loadmodule "/usr/local/lib/ser/modules/registrar.so" >loadmodule "/usr/local/lib/ser/modules/acc.so" >#loadmodule "/usr/local/lib/ser/modules/group.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", "calculate_ha1", yes) ># ># If you set "calculate_ha1" parameter to yes (which >true in this config), ># uncomment also the following parameter) ># >modparam("auth_db", "password_column", "password") > ># -- rr params -- ># add value to ;lr param to make some broken UAs happy >modparam("rr", "enable_full_lr", 1) > > ># -- acc params -- > >modparam("acc", "db_url", >"mysql://ser:heslo@localhost/ser") >#modparam("acc", "log_level", 1) >#modparam("acc", "log_flag", 1) >modparam("acc", "db_flag", 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=~"justser.com") { > > if (src_ip==66.166.166.66 and >method=="INVITE") { ## Llamadas desde PSTN > forward(uri:host, uri:port); > ## > break; > ## > }; > ## > > > if (method=="REGISTER") { > ># Uncomment this if you want to use digest >authentication > if >(!www_authorize("justser.com", "subscriber")) { > >www_challenge("justser.com", "0"); > break; > }; > > save("location"); > break; > }; > > > > if >(uri=~"^sip:1305[0-9]*@justser.com") { ## > prefix("3000#"); > ## > >rewritehostport("66.166.166.66:5060"); ## >TERMINATION > forward(uri:host, uri:port); > ## AREA CODE 305 > #setflag(1); ## > break; > ## > }; > ## > > > if >(uri=~"^sip:1786[0-9]*@justser.com") { ## > prefix("3000#"); > ## > >rewritehostport("66.166.166.66:5060"); ## >TERMINATION > forward(uri:host, uri:port); > ## AREA CODE 786 > 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 > > setflag(1); > > if (!t_relay()) { > sl_reply_error(); > }; > >} > > > > >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

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[Serusers] ACC into MySQL: How to config ser to log gw messages to bill properly?
by Jose Bertuzzi 17 May '05

17 May '05

Hello guys; I have everything in place to acc into mysql. I am already logging some BYEs, INVITEs and ACKs. Where do I have to place the setflag statement to properly log messages from and/to gw 66.166.166.66 only? My ser.cfg is as follows: # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" #fifo_mode=0666 # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/acc.so" #loadmodule "/usr/local/lib/ser/modules/group.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", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- acc params -- modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") #modparam("acc", "log_level", 1) #modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 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=~"justser.com") { if (src_ip==66.166.166.66 and method=="INVITE") { ## Llamadas desde PSTN forward(uri:host, uri:port); ## break; ## }; ## if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("justser.com", "subscriber")) { www_challenge("justser.com", "0"); break; }; save("location"); break; }; if (uri=~"^sip:1305[0-9]*@justser.com") { ## prefix("3000#"); ## rewritehostport("66.166.166.66:5060"); ## TERMINATION forward(uri:host, uri:port); ## AREA CODE 305 #setflag(1); ## break; ## }; ## if (uri=~"^sip:1786[0-9]*@justser.com") { ## prefix("3000#"); ## rewritehostport("66.166.166.66:5060"); ## TERMINATION forward(uri:host, uri:port); ## AREA CODE 786 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 setflag(1); if (!t_relay()) { sl_reply_error(); }; } Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html

2 1

[Serusers] acc module with mysql support
by Grigory Fishilevich 17 May '05

17 May '05

dear serusers, the acc module on my ser make no inserts in the acc-table i change the acc-makefile for mysql support and add few lines from examp,config to my ser.cfg i see acc-output in /var/log/messages but nothing in the acc-table serversion: 0.10.99-dev5 i searching in this list for any help, but all the questions to acc from other users have been ignored i hope anybody can help me bg, Grigory

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[Serusers] dispatcher addition?
by Hendrik Scholz 17 May '05

17 May '05

Hi! I'm playing around with the dispatcher module and got it working but ran into some expected problems. Once I load-balance registrations I do run into the situation that I won't be able to know where to redirect an INVITE, i.e. due to aliases. If I use 'touri' to balance registers (let's pretend all registers have usernames not phone numbers) I can spread those (almost) evenly across multiple servers. Once I receive an invite for a username I can use the dispatcher function with the 'touri' hash to properly forward the request. What happens if I INVITE a phone number? A user might have an alias (aka his phone number) but since he registered with his username and all we know in the INVITE is the phone number the hashes might differ. I'm thinking about looking up the username for a phonenumber utilizing radius (same should work with the DB backend) and then use ds_select_dst() to get the proper SIP server. So in worst case it adds one hop to the whole thing but prevents me from looking up all aliases on the dispatching box. Any hints to get this done? Looking at the ds_select_dst() implementation I either have to replace the touri in ser routing in order to pass an already modified *sip_msg to it or add another function to lookup a dst for a given clear text string that I should be able to pull from Radius. Cheers, Hendrik -- freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 Kiel, Germany Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 Internet: http://www.freenet.de, eMail: hendrik.scholz(a)freenet-ag.de

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[Serusers] Cisco ATA 186, problem with SIP method BYE
by Che Sosa 17 May '05

17 May '05

Hello, I am using ser 0.8.14 This is a part of my routing logic: # send report to table acc, only INVITES and BYES if (method=="INVITE" || method=="BYE") acc_db_request("accounting", "acc"); All INVITES and BYES are logged in table acc for my serweb to access. Then I parse a succesful call by matching the INVITE and the BYE. So far, the method works in all ATAs and hard phones EXCEPT Cisco ATA 186. If the analog phone connected to Cisco ATA 186 sends the BYE, serweb is UNABLE to parse the BYE and send to acc. I traced the headers. This is the bye sent by Cisco ATA 186 (Version: v3.1.0 atasip (Build 040211A)). IP addresses to watch: 10.10.6.92 is the IP address of Cisco ATA 186 10.10.6.93 is the IP address of the SIP proxy server 10.10.6.96 is the phone being called (a Grandstream) U 10.10.6.92:5060 -> 10.10.6.93:5060 BYE sip:116@10.10.6.93 SIP/2.0..Route: <sip:116@10.10.6.96;user=phone>..Via: SIP/2.0/UDP 10.10.6.92:5060..From: <sip:1 12(a)10.10.6.93;user=phone>;tag=1264564150..To: <sip:116@10.10.6.93;user=phone>;tag=358dd5aaed8faa35..Call-ID: 783982044 @10.10.6.92..CSeq: 2 BYE..User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A)..Content-Length: 0.... # U 10.10.6.93:5060 -> 10.10.6.96:5060 BYE sip:116@10.10.6.96;user=phone SIP/2.0..Max-Forwards: 10..Via: SIP/2.0/UDP 10.10.6.93;branch=z9hG4bKfe41.1cf00d23.0 ..Via: SIP/2.0/UDP 10.10.6.92:5060..From: <sip:112@10.10.6.93;user=phone>;tag=1264564150..To: <sip:116@10.10.6.93;user =phone>;tag=358dd5aaed8faa35..Call-ID: 783982044@10.10.6.92..CSeq: 2 BYE..User-Agent: Cisco ATA 186 v3.1.0 atasip (04 0211A)..Content-Length: 0.... # U 10.10.6.96:5060 -> 10.10.6.93:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.10.6.93;branch=z9hG4bKfe41.1cf00d23.0..Via: SIP/2.0/UDP 10.10.6.92:5060..From: <si p:112@10.10.6.93;user=phone>;tag=1264564150..To: <sip:116@10.10.6.93;user=phone>;tag=358dd5aaed8faa35..Call-ID: 783982 044@10.10.6.92..CSeq: 2 BYE..User-Agent: Grandstream BT100 1.0.4.50..Contact: <sip:116@10.10.6.96;user=phone>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0.... # U 10.10.6.93:5060 -> 10.10.6.92:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.10.6.92:5060..From: <sip:112@10.10.6.93;user=phone>;tag=1264564150..To: <sip:116@1 0.10.6.93;user=phone>;tag=358dd5aaed8faa35..Call-ID: 783982044@10.10.6.92..CSeq: 2 BYE..User-Agent: Grandstream BT100 1.0.4.50..Contact: <sip:116@10.10.6.96;user=phone>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.. Content-Length: 0.... Observe the first block above (the instant when Cisco sent the BYE) and compare it to a working BYE sent by another phone. This is the working BYE (note of the first block when the phone sent the BYE): U 10.10.6.96:5060 -> 10.10.6.93:5060 BYE sip:112@10.10.6.92:5060;user=phone;transport=udp SIP/2.0..Via: SIP/2.0/UDP 10.10.6.96;branch=z9hG4bK28a66e39511d3e 5c..Route: <sip:116@10.10.6.93;ftag=1540304939;lr=on>..From: <sip:116@10.10.6.93;user=phone>;tag=46641aca1a1d8bac..To: <sip:112@10.10.6.93;user=phone>;tag=1540304939..Contact: <sip:116@10.10.6.96;user=phone>..Call-ID: 1958105398(a)10.10.6 .92..CSeq: 352 BYE..User-Agent: Grandstream BT100 1.0.4.50..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFE R,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0.... # U 10.10.6.93:5060 -> 10.10.6.92:5060 BYE sip:112@10.10.6.92:5060;user=phone;transport=udp SIP/2.0..Via: SIP/2.0/UDP 10.10.6.93;branch=z9hG4bKa401.62590761. 0..Via: SIP/2.0/UDP 10.10.6.96;branch=z9hG4bK28a66e39511d3e5c..From: <sip:116@10.10.6.93;user=phone>;tag=46641aca1a1d8 bac..To: <sip:112@10.10.6.93;user=phone>;tag=1540304939..Contact: <sip:116@10.10.6.96;user=phone>..Call-ID: 1958105398 @10.10.6.92..CSeq: 352 BYE..User-Agent: Grandstream BT100 1.0.4.50..Max-Forwards: 69..Allow: INVITE,ACK,CANCEL,BYE,NOT IFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0.... # U 10.10.6.92:5060 -> 10.10.6.93:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.10.6.93;branch=z9hG4bKa401.62590761.0..Via: SIP/2.0/UDP 10.10.6.96;branch=z9hG4bK2 8a66e39511d3e5c..From: <sip:116@10.10.6.93;user=phone>;tag=46641aca1a1d8bac..To: <sip:112@10.10.6.93;user=phone>;tag=1 540304939..Call-ID: 1958105398@10.10.6.92..CSeq: 352 BYE..Server: Cisco ATA 186 v3.1.0 atasip (040211A)..Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER..Content-Length: 0.... # U 10.10.6.93:5060 -> 10.10.6.96:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.10.6.96;branch=z9hG4bK28a66e39511d3e5c..From: <sip:116@10.10.6.93;user=phone>;tag= 46641aca1a1d8bac..To: <sip:112@10.10.6.93;user=phone>;tag=1540304939..Call-ID: 1958105398@10.10.6.92..CSeq: 352 BYE..S erver: Cisco ATA 186 v3.1.0 atasip (040211A)..Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER..Cont ent-Length: 0....

1 0

[Serusers] How to disconnect the loggedin user
by Manoj Ghag 16 May '05

16 May '05

Hi, All I have soem queries here How can i disconnect the logged in user ... is there any trigger by which .. i can disconnect him after say 5 min ? Secondly is there any way by which i can findout the distance at two ends ? How can i set the calling standard like G711 or G723 or G726 .. is there any setting i need to do in ser.cfg ? I mean use should stick to that standard only .. even if they try to change .. ( silly question ) I want to test the voice mail feature .. how can i test ? say if use is not accepting the call then voice mail should get active .......... etc etc . How can i add my contactes in to my MSN Messanger 5.1? because it gives error which say no contact found :( Regards Manoj , __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Serusers] billing in ser
by Joel n.solanki 16 May '05

16 May '05

Dear all, I want to setup billing system in ser. Is there any opensource software so that I can integrate with ser voip. How do I get CDR in ser voip ? Thanks for help in advance. Joel

1 0

[Serusers] what is a vendor id of SER
by Manoj Ghag 16 May '05

16 May '05

hi, I am trying to configure the ser with radius .. but what is a vendor id of SER regards Manoj --------------------------------- Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses.

1 0

[Serusers] 0.9.2 make problem
by Steve Blair 16 May '05

16 May '05

I just performed a cvs update to 0.9.2. When I try to make or make modules I get the following error for which I cannot find a solution. Has anyone else experienced this problem? If so any suggestions on how to fix it? Thanks,Steve [sip_router]$ sudo make /bin/sh: -c: line 1: syntax error near unexpected token `<<<' /bin/sh: -c: line 1: `set -e; gcc -MM -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -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 <<<<<<< Makefile.defs -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 cfg.tab.c | sed 's#$\(cfg.tabD)$\?cfg.tab\)\.o[ :]*#cfg.tab.o cfg.tab.d : #g' > cfg.tab.d; [ -s cfg.tab.d ] || rm -f cfg.tab.d' make: *** [cfg.tab.d] Error 2

2 1

[Serusers] ser+serweb+(Remote Radius+Mysql)
by a cairo 15 May '05

15 May '05

I have ser and remote radius server with mysql- I want to add serWEB module but how cai i do this? Is There a manual? Are there other module for ser that you prefer? Thanks

1 0

[Serusers] allow_trusted
by Iqbal 15 May '05

15 May '05

Hi When using this function in order to bypass proxy_auth checks, shouldnt it do a lookup in a DB, I have the following -------------------------------------- if (!allow_trusted()) { if (!proxy_authorize("sip.dom.com", "subscriber")) { proxy_challenge("sip.dom.com", "0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; } else if (!(is_from_local() || is_uri_host_local())) { sl_send_reply("403", "Please register to use our service"); break; }; }; --------------------------------- The problem is when I do this I always fall into this loop even if my pattern in my trusted table matches pattern is of the form sip:[0-9]*@a.b.c.d However I wanted to check to see if my entry in the DB was correct, but no DB query seems to be sent, or at least its not in my logs. IN debug also I cant seem to see the line which may show what it is checking. Iqbal

1 0

[Serusers] SER B2BUA
by mark.roddy＠lakecommunications.com 15 May '05

15 May '05

Is there a SER B2BUA available? thx Mark Roddy LAKE Communications http://www.lakecommunications.com ********************************************************************** This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify postmaster(a)lakecommunications.com This footnote also confirms that this email message has been scanned for the presence of computer viruses. www.lakecommunications.com **********************************************************************

3 2

[Serusers] UAC module BUG??
by Helge Waastad 15 May '05

15 May '05

Hi, from my debug after trying to use the uac_auth in a failure route, I get the following: Mar 31 16:37:46 sip ser[20394]: BUG:uac:uac_auth: empty reply on picked branch I've modified the cvs version to compile on 0.9.0 Anyone? Mvh, Helge Waastad

3 2

[Serusers] SER with internal/external interface
by Adrian A 14 May '05

14 May '05

I am trying to setup SER so it can be used both by internal and external users. I have tried to set it up with two interfaces, one internal and one external. There is a STUN server setup for external users as well. The problem with this is that internal clients will obviously add a VIA header with their internal IP so if they call someone on the outside and that someone sends a BYE for example, it never gets to the internal client because it tries to send it to the internal IP address. Putting SER completely on the outside does not work either unless all NAT-ed clients support ICE (because there is no sound coming through). The only client which I've seen to support ICE so far is Xten's Eyebeam softphone, but I'd like to be able to use hardphones as well. Does anyone have an idea how to accomplish what I need to do? Is there something I can do in ser.cfg so the VIA header of the internal client is stripped, or can I have 2 SER's, one listening on internal IP and one on the external IP and have them relay each other? I have tried the mhomed directive but that seems to mess things up even more.

1 0

[Serusers] voice problem
by Joel n.solanki 14 May '05

14 May '05

Dear all, i have configured ser-0.8.11-0 ser-mysql-0.8.11-0. i have configured using how to provided in iptel.org website. my Sjphone authenticated perfectly against mysql database. but while calling to other sjphone it is giving some unusual voice. it is giving strange voice. what could be the problem? is there any thing to configure for voice? While using Sjphone, On screen it show me (gsm). is that reason for unusual voice. if yes, then how do i solve this problem. Thanking you, Joel n.solanki

1 0

[Serusers] Registering SER - Solution
by Kaveh Aasaraai 14 May '05

14 May '05

Hi, Finally I found the solution! Accepting calls from Asterisk users by SER is simple, by adding special routes in ser.cfg. The problem is to route SER users' calls to Asterisk. By default, Asterisk replies with a "Not Found" error to SER if the "INVITE" message forwarded to it is not authorized. Making SER authorized to Asterisk is done by adding a user in sip.conf as a peer. This yields something like this: [un] type=peer username=un insecure=yes host=SER's ip address (host name) Whenever "insecure" is set to "yes", "host" provides security. This means packets are only allowed from the specified ip address. The "un" is the name of the calee! According to authentication procedure described in RFC-3261, the "TO" header field represents user agent being authorized. So, when a SER client is calling an Asterisk one, he/she is providing Asterisk with the "TO" field of calee URI. So, Asterisk tries to locate the authentication for that user (calee) and this is the point. This just worked! K. Aasaraai

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[Serusers] Re: Registering SER
by Kaveh Aasaraai 14 May '05

14 May '05

Hi again, As Casey said, I tried the Asterisk configuration (like I did before) and asterisk didn't reply correctly. My problem is that my SER's users can't make calls to Asterisk's users. In SER's configuration, I forward invite packets to Asterisk, and it really comes!. I dumped packets being sent, and found out that Asterisk replies with a "Not Found" error to SER, and this the thing that Client (SER's client) receives. I also captured an Asterisk's authorized user's invite packet for a specified calee, and the two "To" URIs are the same. (The one being sent from SER's user and the one from Asterisk's user). It seems registering Asterisk to SER is not sifficient to make SER authorized to Asterisk.Can anyone help me? Thank you. To Paul: I read the "Getting Started" document, but couldn't find anything related to registering SER to Asterisk. (Seems the section is still under construction) By the way, thanks to Casey and Paul. K. Aasaraai

3 2

[Serusers] PHP warnings
by Llanos Serna García-Conde 14 May '05

14 May '05

Hi all,when I try to open serweb's login screen, I get the following errors: Warning: Client does not support authentication protocol requested by server; consider upgrading MySQL client in /var/www/html/serweb/serweb_2004-07-27/phplib/db_mysql.inc on line 73 Database error: pconnect(localhost, ser, $Password) failed. MySQL Error: () Session halted. When I start APACHE, I get all these warnings: PHP Warning: Function registration failed - duplicate name - mysql_connect in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_pconnect in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_close in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_select_db in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_create_db in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_drop_db in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_query in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_unbuffered_query in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_db_query in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_list_dbs in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_list_tables in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_list_fields in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_error in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_errno in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_affected_rows in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_insert_id in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_result in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_num_rows in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_num_fields in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_fetch_row in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_fetch_array in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_fetch_assoc in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_fetch_object in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_data_seek in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_fetch_lengths in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_fetch_field in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_field_seek in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_free_result in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_field_name in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_field_table in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_field_len in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_field_type in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_field_flags in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_escape_string in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_get_client_info in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_get_host_info in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_get_proto_info in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_get_server_info in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_fieldname in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_fieldtable in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_fieldlen in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_fieldtype in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_fieldflags in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_selectdb in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_createdb in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_dropdb in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_freeresult in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_numfields in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_numrows in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_listdbs in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_listtables in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_listfields in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_db_name in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_dbname in Unknown on line 0 PHP Warning: Function registration failed - duplicate name - mysql_tablename in Unknown on line 0 PHP Warning: mysql: Unable to register functions, unable to load in Unknown on line 0 Do you know the reason for these warnings? Do they have any relation with the error I get when trying to connect? Can you help me please? Thank you Llanos

2 1

[Serusers] is there anyone who has configured ser 0.8.14 with remote radius server
by Shaikat Mahmud 14 May '05

14 May '05

Hi, I am trying to configure ser 0.8.14 with remote radius. I am using radiusclient 0.4.8 And freeradius.0.9.1 And my os is debian It is ok for local server. But when I am trying to do acc with remote radius, it is not sending any information. Now by seeing the debugging information ( and some code), I have found that the error has generated from the radiusclient. Now has anybody done it? I have found that many people have faced lots of problem with ser-0.8.14 and remote radius. It works for ser-0.8.12 Thank you, Shaikat

1 0

[Serusers] error opening radius configuration file
by Kamran Ahmad 14 May '05

14 May '05

hello # radiusclient -f test localhost auth testing123 error opening radius configuration file i m using radiusclient-0.3.3 Kamran __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

1 0

[Serusers] sip stack
by Andréa Giordanna 14 May '05

14 May '05

Hi all, I'm sorry if my questions are stupid, but I need to know: is SER considered a SIP stack? is SER development based on some stack? Thanks Andrea Giordanna --------------------------------- Yahoo! Mail: agora com 1GB de espaço grátis. Abra sua conta!

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[Serusers] Problem of accountig with Radius support
by lzf＠alternet-europe.com 14 May '05

14 May '05

Hi, I have some problem with radius support for the accounting. It's working well for authentification. But accounting doesn't work. I put SER in debug mode and I obtain the message: ERROR: acc_rad_request: radius-ing failed And when I look on the network (Ethereal), none frame get out of my SER. Has everyone some problems like that (and solutions :). Thanks. Have a good day.

2 1

[Serusers] session timers
by Jorge Crichigno 14 May '05

14 May '05

Hi All, I have the following question: I would like to know if there is an extension and/or and RFC related with timer sessions. The idea is to use timer sessions in final users, in order to shut down sessions at pre-defined times (at the begining of them) without necessity of deal with timer at servers (no necessity of dialog statefull servers). As far as I know, there is draft, "Session Timers in the Session Initiation Protocol (SIP)", draft-ietf-sip-session-timer-15 (july 2004), which is already implemented by some companies like Addpac. But, it is not directly applicable to our necessity (it is just a mechanism of keep-alive). Thanks in advance! Jorge

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Re: [Serusers] Re: sip load balancing with 0.9.0 - problems routing
by a c 13 May '05

13 May '05

hello Andrei, I got this working and it is load balancing ok between the 2 endpoints. I have noticed that SER does not switch over to a different endpoint in the dispatcher list if one of the endpoints is down. Do I need to enable this somehow or is this functionality not implemented? Could someone verify the routing logic below? Is this ok? thank you ac -- ser.cfg -- 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("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # 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 ds_select_dst("1", "0"); if (!t_relay()) { sl_reply_error(); }; } --- Andrei Pelinescu-Onciul <andrei(a)iptel.org> wrote: > On May 12, 2005 at 14:13, a c <lra101(a)yahoo.com> > wrote: > > ok. I got this working now..no core dumps. > > > > But still having problems with dispatcher. > > > > I have the following setup: > > > > sip client 1 -> SER -> sip client2 & sip client3 > > > > when client1 makes a call, it seems like SER does > not > > receive the request. I don't see any logs written > and > > the call timesout. Seems like SER is not receiving > the > > requests but when I do a netstat, I see it > listening > > on 5060. > > > > udp 0 0 castor:5060 *:* > > > > Normal routing without dispatcher works fine {thru > > client registrations}. Am I missing some configs > here? > > Your config looks ok, maybe some networks dumps > would help. > > > Andrei > __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

2 1

[Serusers] Does SER compile on other architectures?
by Wojciech Ziniewicz 13 May '05

13 May '05

Hello list, I would like to compile SER on other architecture then k7 or x86. I have some candidates: http://store.mini-box.com/ituner/viamo.html http://www.routerboard.com http://www.pcengines.ch/wrap.htm http://www.soekris.com Is SER rather Platform independent ? Or do i have to make up ports for every platform (I am not so keen on programming in C so it would be good to not to change the code :P) -- Pozdrawiam, Wojciech Ziniewicz Optocomp sp.z.o.o, www.optocomp.pl mailto: wojtekz(a)optocomp.pl +48(0)691031535

2 1

[Serusers] SER+SEMS+IVR+Voicemail
by Bahda Saxansaxo 13 May '05

13 May '05

Hello, I'm new to SER+SEMS, I'm having some problem: 1. I have SER running fine 0.9.0 from cvs. 2. sems installed (with TTS support) 3. needed to include IVR and voicemail. 4. need to run one ser instance (one ser.cfg and one sems.cfg) I read the docs both (ser+sems-howto and ivr_documnet) but after 2 weeks still no luck, so please if you got this working let me get steps to take and mor important how could I got simple tests. Thanks A Ahmed __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

1 0

Re: [serusers] NAT2
by michael p 13 May '05

13 May '05

sorry for the mailing i succeed to find the good Makefile (that was empty at the beginning of the download) and now what i need to do ? there are client and server .exe how configure them and use them? what i need to configure in my softphone to try that's working? thanks M.

1 0

Re: [Serusers] Does SER compile on other architectures?
by Samuel Osorio Calvo 13 May '05

13 May '05

Hoi! Maybe Andrei can help you more than I since he's "responsible" for porting SER to *ALL* platforms . Taking a look at Makefile.defs you find the next platforms, which can be assigned with the macro ARCH: *sparc64 *i386 *arm *ppc *mips Depending on the concrete board, you might have to tune the compilation with CC_EXTRA_OPTS or similar....just experiment and provide feedback!!! (I'm sure SER can be even ported to more platforms.............) For more info, check old posts to the developers mailing list. Thanks again to SER developers! Samuel. Unclassified. >> Wojciech Ziniewicz <WojtekZ(a)Optocomp.PL> 05/13/05 03:07PM >>> Hello list, I would like to compile SER on other architecture then k7 or x86. I have some candidates: http://store.mini-box.com/ituner/viamo.html http://www.routerboard.com http://www.pcengines.ch/wrap.htm http://www.soekris.com Is SER rather Platform independent ? Or do i have to make up ports for every platform (I am not so keen on programming in C so it would be good to not to change the code :P) -- Pozdrawiam, Wojciech Ziniewicz Optocomp sp.z.o.o, www.optocomp.pl mailto: wojtekz(a)optocomp.pl +48(0)691031535 _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] tcp_init: bind error
by Cesc Santasusana 13 May '05

13 May '05

Make sure that you killed all ser processes ... > ps aux | grep ser then >kill -9 pid ... sometimes, doing killall ser doesn't work cesc >>> Sönmez Güne (Bº <sonmezgunes(a)gmail.com> 05/13/05 11:26AM >>> Hello When I restart ser, taking this error. ERROR: tcp_init: bind(5, 0x80d08a0, 16) on 10.90.232.121: Address already in use What can be the problem? _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers unclassified

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[Serusers] SER Proxy with Redirect
by Eric Dean 13 May '05

13 May '05

I have a SER Proxy that is routing calls to a SIP Redirector which then informs SER as to which gateway to send call to. GW1 GW2 | | SERProxy------->SIP Redirector | IP Phone I would like for SER to take a 302 redirect and then proxy the call to the gateway per the redirect..however, SER is currently serving the redirect to the IP Phone..which is unauthorized to access the gateway. How do I configure SER to actually proxy the call per the redirect on behalf of the IP Phone?

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[Serusers] SERWEB and MySQL Connection ERROR Database error: pconnect(192.168.1.246, ser, $Password) failed.
by Manoj Ghag 13 May '05

13 May '05

hi, In case of MYSQL .. I am getting following ERROR when I try to open a SERWEB.... Warning: mysql_pconnect(): Access denied for user: 'ser@test' (Using password: YES) in /var/www/phplib/db_mysql.inc on line 73 Database error: pconnect(192.168.1.246, ser, $Password) failed. MySQL Error: () Session halted. ANY HELP ? --------------------------------- Yahoo! Mail Stay connected, organized, and protected. Take the tour

3 4

[Serusers] Error - libmysqlclient.so.10: cannot open shared object file
by Judy.Banta＠ser.com 13 May '05

13 May '05

OS: Red Hat Enterprise Linux, Version 3 (Linux tsp500_d2.ct.ser.com 2.4.21-27.0.1.ELsmp #1 SMP Mon Dec 20 18:47:45 EST 2004 i686 i686 i386 GNU/Linux) SER Release: ser-0.8.14_linux_i386.tar.gz SER Version: 0.8.14 (i386/linux) SER Config: <<ser.cfg>> SER Debug 4: <<ser.log>> No additional information found /var/log/messages MySQL Release: mysql-standard-4.1.11-pc-linux-gnu-i686.tar.gz Problem: Using MySQL for persistant data storage and receiving error when starting SER: "0(4099) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.10: cannot open shared object file: No such file or directory" MySQL contains the file "libmysqlclient.a" located in /usr/local/mysql-standard-4.1.11-pc-linux-gnu-i686/lib. Thanks, Judy Banta SER Solutions, Inc. 50 Washington St., 2nd Floor Norwalk, CT 06854 203-854-8903

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[Serusers] Can serweb run on Linksys WRT54GS
by Mikael Sköldemar 13 May '05

13 May '05

Hello! I wonder if the web interface serweb can run on a Linksys WRT54GS with openwrt. /Mikael

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Re: [Serusers] Mysql error () with SERWEB
by Llanos Serna García-Conde 13 May '05

13 May '05

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[Serusers] My questions is how can I trace the configure file error?
by zhaomin 13 May '05

13 May '05

My questions is how can I trace the configure file error? When I loading a ser.cfg ,sometime tell me "ERROR: bad config file (2 errors)" ,But I don't know where is error line in configure file .How can I trace the error in the configure file? Thanks advance ZhaoMin

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[Serusers] sems is dead?
by Grigory Fishilevich 13 May '05

13 May '05

hi guys, i read just "getting started" from onsip.org i seach for infos about pstn-gw, bit they speak only about cisco-hardware based solution. is the sems deprecated? bg, grigory fishilevich

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Re: [Serusers] Re: sip load balancing with 0.9.0 - problems routing
by a c 13 May '05

13 May '05

ok. I got this working now..no core dumps. But still having problems with dispatcher. I have the following setup: sip client 1 -> SER -> sip client2 & sip client3 when client1 makes a call, it seems like SER does not receive the request. I don't see any logs written and the call timesout. Seems like SER is not receiving the requests but when I do a netstat, I see it listening on 5060. udp 0 0 castor:5060 *:* Normal routing without dispatcher works fine {thru client registrations}. Am I missing some configs here? thank you ac ser.cfg file: # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # 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) # main routing logic 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 # gateways 1 sip:103@192.168.0.3:5060 1 sip:103@192.168.0.2:5060 from log : 0(0) DEBUG: init_mod: dispatcher 0(0) DISPATCHER: initializing ... 0(0) DISPATCHER:ds_load_list: dest [0/1/1] <sip:103@192.168.0.3:5060> 0(0) DISPATCHER:ds_load_list: dest [0/1/2] <sip:103@192.168.0.2:5060> 0(0) DISPATCHER:ds_load_list: found [1] dest sets 0(0) fixing /usr/local/lib/ser/modules/maxfwd.so mf_process_maxfwd_header 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/dispatcher.so ds_select_dst 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) DEBUG: udp_init: trying SO_RCVBUF: 131070 0(0) DEBUG: setting SO_RCVBUF; set=131070,verify=131070 0(0) DEBUG: udp_init: trying SO_RCVBUF: 262140 0(0) DEBUG: setting SO_RCVBUF; set=262140,verify=131070 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) DEBUG: udp_init: trying SO_RCVBUF: 133118 0(0) DEBUG: setting SO_RCVBUF; set=133118,verify=131070 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 0(0) WARNING: using only the first listen address (no fork) 0(0) DBG: open_fifo_server: no fifo will be opened 0(0) init_unixsock_socket: No unix domain socket will be opened 0(4567) DEBUG: init_mod_child (1): dispatcher 0(4567) DISPATCHER:init_child #1 / pid <4567> 1(4568) DEBUG: init_mod_child (-1): dispatcher 1(4568) DISPATCHER:init_child #-1 / pid <4568> --- Andrei Pelinescu-Onciul <andrei(a)iptel.org> wrote: > On May 11, 2005 at 20:18, a c <lra101(a)yahoo.com> > wrote: > > 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. > > Thanks, it's fixed on the latest cvs. > > (some minor on ser exit missing null pointer check) > > Andrei > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Serusers] Mediaproxy in LAN (?)
by bogdan aux 13 May '05

13 May '05

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

1 0

Re: [Serusers] Mysql error () with SERWEB
by Llanos Serna García-Conde 13 May '05

13 May '05

1 0

[Serusers] acc problem
by Sönmez Güneş 13 May '05

13 May '05

Hello I change Makefile like below. include ../../Makefile.defs auto_gen= NAME=acc.so LIBS= # uncomment the next line if you wish to enable SQL accounting DEFS+=-DSQL_ACC # uncomment the next two lines if you wish to enable RADIUS accounting #DEFS+=-DRAD_ACC -I$(LOCALBASE)/include #LIBS=-L$(LOCALBASE)/lib -lradiusclient include ../../Makefile.modules And compiled SER with below parameters. make all include_modules=mysql make prefix=/usr/local install include_modules=mysql Although I can talk between softphones. I could not get record in acc table. What can be the reason...

1 0

Re: [Serusers] Mysql error () with SERWEB
by Llanos Serna García-Conde 13 May '05

13 May '05

1 0

[Serusers] tcp_init: bind error
by Sönmez Güneş 13 May '05

13 May '05

Hello When I restart ser, taking this error. ERROR: tcp_init: bind(5, 0x80d08a0, 16) on 10.90.232.121: Address already in use What can be the problem?

1 0

[Serusers] Mysql error () with SERWEB
by Llanos Serna García-Conde 13 May '05

13 May '05

1 0

[Serusers] RE: NAT, ACK and BYEs
by Jorge Crichigno 13 May '05

13 May '05

Giovanni, As far as I know, you also need to edit on-reply route in your script. It seams that you don't correct the contact header of the called party, which is send in the 200 reply. The script would look like this: #################################################################### INVITE routing logic is something like this t_on_reply("1"); #in order to the replies pass to on reply block. You should add this! if (loose_route()) { if (has_totag() && method=="INVITE") { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); t_on_reply("1"); }; t_relay(); break; } if (method=="INVITE") { record_route(); if (nat_uac_test(19)) { fix_nated_contact(); force_rport(); setflag(6); } if ( !isflagset(6) || SAME_NAT) { t_relay(); } else { force_rtp_proxy(); t_on_reply("1"); t_relay(); } } # When NAT, the 200 (Ok) reply carries the contact header where the called party is reached. If it is # behind NAT, we need to correct his contact. onreply_route[1] { if (status=~"(183)|(2[0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; }

2 1

[Serusers] acc, gateway problem
by Sönmez Güneş 13 May '05

13 May '05

Hello everybody, I should explain what ever I do one by one. 1. I downloed ser files under "/usr/local/ser-0.8.14" directory. 2. I get in here and open "/usr/local/ser-0.8.14/modules/acc/Makefile" 3. Changed like this DEFS+=-DSQL_ACC 4. While in "/usr/local/ser-0.8.14" Run these commands : make all include_modules=mysql make prefix=/usr/local install include_modules=mysql setup seemed succesfull. 5. Install SER mysql database. 6. Change ser.cfg like this. # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=10.90.232.121 listen=127.0.0.1 port=5060 children=4 alias=10.90.232.121 fifo="/tmp/ser_fifo" fifo_mode=0666 # 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/acc.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/uri.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" modparam("acc|auth_db|domain|group|speeddial|uri_db|usrloc", "db_url", "mysql://ser:heslo@10.90.232.121/ser") #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) modparam("acc", "log_flag", 1) #modparam("acc", "log_missed_flag", 1) # -- 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) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "desc_time_order", 1) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_timer", 20) modparam("tm", "fr_inv_timer", 40) modparam("tm", "wt_timer", 5) # ------------------------- 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 if (loose_route()) { t_relay(); break; }; setflag(1); /*if (method=="INVITE" || method=="BYE") { setflag(1); };*/ /* record-route INVITEs -- all subsequent requests must visit us */ if (method=="INVITE") record_route(); # 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=="INVITE" || method== "CANCEL" || method== "ACK") { if (!proxy_authorize("10.90.232.121","subscriber")) { proxy_challenge("10.90.232.121","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; setflag(1); consume_credentials(); /*if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; };*/ } else if (method=="REGISTER") { if (!www_authorize("10.90.232.121", "subscriber")) { www_challenge("10.90.232.121", "0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); save("location"); break; }; lookup("aliases"); # 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 setflag(1); if (uri=~"^sip:00[0-9]*@10.90.232.121") { ## This assumes that the caller is log("Forwarding to PSTN\n"); ## registered in our realm rewritehostport("10.90.232.122:5060"); forward(uri:host, uri:port); #t_relay_to( "10.90.232.122", "5060"); break; }; if (!t_relay()) { sl_reply_error(); }; } Although I can speak between two softphone. There is no record in acc table. And I could not route calls which starts with "00" to SIP gateway. I would be great if you become guide...

1 0

[Serusers] acc, gateway problem
by Sönmez Güneş 13 May '05

13 May '05

Hello everybody, I should explain what ever I do one by one. 1. I downloed ser files under "/usr/local/ser-0.8.14" directory. 2. I get in here and open "/usr/local/ser-0.8.14/modules/acc/Makefile" 3. Changed like this DEFS+=-DSQL_ACC 4. While in "/usr/local/ser-0.8.14" Run these commands : make all include_modules=mysql make prefix=/usr/local install include_modules=mysql setup seemed succesfull. 5. Install SER mysql database. 6. Change ser.cfg like this. # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=10.90.232.121 listen=127.0.0.1 port=5060 children=4 #alias=sipserver.megatelekom.com.tr alias=10.90.232.121 fifo="/tmp/ser_fifo" fifo_mode=0666 # 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/acc.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/uri.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" modparam("acc|auth_db|domain|group|speeddial|uri_db|usrloc", "db_url", "mysql://ser:heslo@10.90.232.121/ser") #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) modparam("acc", "log_flag", 1) #modparam("acc", "log_missed_flag", 1) # -- 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) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "desc_time_order", 1) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_timer", 20) modparam("tm", "fr_inv_timer", 40) modparam("tm", "wt_timer", 5) # ------------------------- 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 if (loose_route()) { t_relay(); break; }; setflag(1); /*if (method=="INVITE" || method=="BYE") { setflag(1); };*/ /* record-route INVITEs -- all subsequent requests must visit us */ if (method=="INVITE") record_route(); # 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=="INVITE" || method== "CANCEL" || method== "ACK") { if (!proxy_authorize("10.90.232.121","subscriber")) { proxy_challenge("10.90.232.121","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; setflag(1); consume_credentials(); /*if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; };*/ } else if (method=="REGISTER") { if (!www_authorize("10.90.232.121", "subscriber")) { www_challenge("10.90.232.121", "0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); save("location"); break; }; lookup("aliases"); # 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 setflag(1); if (uri=~"^sip:00[0-9]*@10.90.232.121") { ## This assumes that the caller is log("Forwarding to PSTN\n"); ## registered in our realm rewritehostport("10.90.232.122:5060"); forward(uri:host, uri:port); #t_relay_to( "10.90.232.122", "5060"); break; }; if (!t_relay()) { sl_reply_error(); }; } Although I can speak between two softphone. There is no record in acc table. And I could not route calls which starts with "00" to SIP gateway. I would be great if you become guide...

1 0

[Serusers] SEMS_2004-7-27 + SER 0.8.14 + SERWEB 2004-7-27
by ooo ooo 13 May '05

13 May '05

my SER and SERWEB have a good run, but my SEMS have not speak. when I use sems -E , I have some error : (4939) WARNING: reloadModuleConfig (SemsConfiguration.cpp:66): no configuration found for module isdngw, maybe you want to specify config.isdngw in config file.(4939) ERROR: send (AmRtpPacket.cpp:119): while sending RTP packet: Invalid argument (4939) ERROR: send (AmRtpStream.cpp:179): while sending RTP packet.(4939) ERROR: send (AmRtpPacket.cpp:119): while sending RTP packet: Invalid argument (4939) ERROR: send (AmRtpStream.cpp:179): while sending RTP packet. and next is my conf file. # # $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_mode=0666 # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/group.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) modparam("usrloc|registrar", "use_domain", 1) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("group", "db_url", "mysql://ser:heslo@localhost/ser") # time to give up on ringing -- global timer, applies to # all transactions modparam("tm", "fr_inv_timer", 30) # ------------------------- request routing logic ------------------- # main routing logic route { if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483", "Alas Too Many Hops"); break; }; if (!(method=="REGISTER")) record_route(); if (loose_route()) { t_relay(); break; }; #if (!uri==myself) { # t_relay(); # break; #}; if (method == "REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; if (!save("location")) { sl_reply_error(); }; break; }; if (!uri==myself) { t_relay(); break; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail group?) -- determine it now and store it in # flag 4, before we rewrite the flag using UsrLoc if (is_user_in("Request-URI", "voicemail")) { setflag(4); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # handle user which was not found route(4); break; }; # if user is on-line and is in voicemail group, enable redirection if (method == "INVITE" && isflagset(4)) { t_on_failure("1"); }; t_relay(); } # ------------- handling of unavailable user ------------------ route[4] { # non-Voip -- just send "off-line" if (!(method == "INVITE" || method == "ACK" || method == "CANCEL")) { sl_send_reply("404", "Not Found"); break; }; # not voicemail subscriber if (!isflagset(4)) { sl_send_reply("404", "Not Found and no voicemail turned on"); break; }; # forward to voicemail now rewritehostport("sip800.8866.org:5090"); t_relay_to_udp("sip800.8866.org", "5090"); } # if forwarding downstream did not succeed, try voicemail running # at vm_host:5090 failure_route[1] { revert_uri(); rewritehostport("sip800.8866.org:5090"); append_branch(); t_relay_to_udp("sip800.8866.org", "5090"); } # # $Id: voicemail.cfg,v 1.2 2003/10/10 06:27:18 jiri Exp $ # # this script is configured for use as voicemail UAS; it processes # INVITEs and BYEs and asks SEMS to record media via "vm"; in this # script, all record-routing and other constructs known from proxy # scripts are not present -- it is a simple UAS # # ----------- global configuration parameters ------------------------ #debug= # debug level (cmd line: -dddddddddd) #fork=no #log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5090 children=4 fifo="/tmp/vm_ser_fifo" fifo_mode=0666 # fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/vm.so" # ----------------- setting module-specific parameters --------------- modparam("voicemail", "db_url","mysql://ser:heslo@localhost/ser") modparam("voicemail", "subscriber_table", "subscriber") modparam("voicemail", "email_column", "email_address") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; #if (len_gt( max_len )) { # sl_send_reply("513", "Message too big"); # break; #}; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; if (!uri==myself) { sl_send_reply("404", "not reponsible for host in r-uri"); break; }; # Voicemail specific configuration - begin if(method=="ACK" || method=="INVITE" || method=="BYE"){ if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ log("**************** vm start - begin ******************\n"); if (uri=~"sip:1000 at .*" || uri=~"sip:1001 at .*") { if (!vm("/tmp/am_fifo", "announcement")) { log("couldn't contact announcement server\n"); t_reply("500", "couldn not contact announcement server"); }; } else { if(!vm("/tmp/am_fifo","voicemail")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; }; log("**************** vm start - end ******************\n"); } else if(method=="BYE"){ log("**************** vm end - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log("**************** vm end - end ******************\n"); }; break; }; if (method=="CANCEL") { sl_send_reply("200", "cancels are junked here"); break; }; sl_send_reply("501", "method not understood here"); } # $Id: sems.conf.sample,v 1.11.2.5 2004/05/24 14:33:07 rco Exp $ # # sems.conf.sample # # Sip Express Media Server (sems) # # sample configuration file # # # whitespaces (spaces and tabs) are ignored # comments start with a "#" and may be used inline # # example: option=value1, value2 # i like this option # ################################## # global parameters # ################################## # optional parameter: fork={yes|no} # # - specifies if sems should run in daemon mode (background) # (fork=no is the same as -E) fork=yes # optional parameter: stderr={yes|no} # # - debug mode: do not fork and log to stderr # (stderr=yes is the same as -E) stderr=no # optional parameter: loglevel={0|1|2|3} # # - sets log level (error=0, warning=1, info=2, debug=3) # (same as -D) loglevel=1 # optional parameter: fifo_name=<filename> # # - path and file name of our fifo file (same as -i) fifo_name=/tmp/am_fifo # optional parameter: ser_fifo_name=<filename> # # - path and file name of Ser's fifo file (same as -o) # ser_fifo_name=/tmp/ser_fifo ser_fifo_name=/tmp/vm_ser_fifo # optional parameter: plugin_path=<path> # # - sets the path to the plug-ins # - may be absolute or relative to CWD plugin_path=/usr/local/lib/sems/plug-in/ # optional parameter: smtp_server=<hostname> # # - sets address of smtp server smtp_server=localhost # optional parameter: smtp_port=<port> # # - sets port of smtp server smtp_port=25 # optional parameter: rtp_low_port=<port> # # - sets port of rtp lowest server rtp_low_port=1024 # optional parameter: rtp_high_port=<port> # # - sets port of rtp highest server rtp_high_port=65535 ################################## # module specific parameters # ################################## # sample voicemail configuration (inline) config.voicemail=inline # optional parameter: announce_path=<path> # # - sets the path where announce files are searched for # - the file to be played is determined the following way: # <announce_path>/<domainname>/<username>.wav # if this file is not available <announce_path>/<default_anounce> is used announce_path=/usr/local/lib/sems/audio/ # parameter: default_announce=<filename> # # - sets the name of the default announce WAV file default_announce=default_en.wav # parameter: max_record_time=<seconds> # # - maximum record time max_record_time=30 # parameter: email_template=<filename> # # - email template file # # See the README file in <sems-src>/plug-in/voicemail # for more information on the syntax used. # email_template=/usr/local/lib/sems/plug-in/mail.template # end of configuration section for voicemail module config.voicemail=end # sample announcement configuration (inline) config.announcement=inline # optional parameter: announce_path=<path> # # - sets the path where announce files are searched for announce_path=/usr/local/lib/sems/audio/ # parameter: default_announce=<filename> # # - sets the name of the default announce WAV file default_announce=default_en.wav # end of configuration section for announcement module config.announcement=end # sample isdngw module configuration (external file) # config.isdngw=/etc/isdngw.conf # sample ivr module configuration (inline) config.ivr=inline #parameter: python_script_path=<full path> python_script_path=/etc/ivr #parameter: python_script_file=<filename> python_script_file=example.py # end of configuration section for ivr module config.ivr=end # sample conference configuration (inline) config.conference=inline # parameter: default_announce=<filename> # # - sets the full pathed name of the default announce WAV file. # Will be played to lonely users. default_announce=/usr/local/lib/sems/audio/first_participant.wav # parameter: join_sound=<filename> # # - sets the full pathed name of a WAV file # which will be played to conference users # when a new user joins the conference. join_sound=/usr/local/lib/sems/audio/beep.wav # parameter: drop_sound=<filename> # # - sets the full pathed name of a WAV file # which will be played to conference users # when a user drops the conference. drop_sound=/usr/local/lib/sems/audio/beep.wav # end of configuration section for conference module config.conference=end # example configuration for number reader config.number_reader=inline number_path=/usr/local/lib/sems/audio/ prolog_file=welcome_to_number_reader.wav epilog_file=thanks_calling_number_reader.wav # end of number_reader configuration config.number_reader=end # add more module configurations here (inline or external): # # config.mymodule=<filename> # or # config.mymodule=inline # ... # config.mymodule=end ------------------------------------------------- help me , thank you. B.R. Zhao Zigang. _________________________________________________________________ 与联机的朋友进行交流，请使用 MSN Messenger: http://messenger.msn.com/cn

1 0

[Serusers] UAC module (backport to 0.9.0)
by Michael Ulitskiy 12 May '05

12 May '05

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

1 1

[Serusers] sip - h323 gateway
by Lucas Aimaretto 12 May '05

12 May '05

Hi people, Do not know if this is the place, bue ... any body used Asterisk as a sip - h323 gateway ? I have ser working really fine and I'd like to use asterisk as a gateway ... anybody done this ??? Regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.9 - Release Date: 12/05/2005

1 0

[Serusers] How SER Call Flags Work
by reticent 12 May '05

12 May '05

I'm curious how SER associates flags to specific phone calls, i'm just trying to understand in what situations a flag would be associated to a SIP message. So far i've assumed that its message specific, as in when a SIP message comes into SER any flags set while it traverses through the SER script are only in effect during its traversal through the script, once the message is routed all flags are forgotten. Except in the special case of "lookup()" where a flag can be associated to a message when "save()" is called

1 0

Re: [Serusers] Re: sip load balancing with 0.9.0 + crash
by a c 12 May '05

12 May '05

hello Andrei, could you tell me how to get the latest stable version which has this fix? Should I be getting the stable version or dev version or a subset. What would be the command I would be using? cvs co -r ?? thank you ac --- Andrei Pelinescu-Onciul <andrei(a)iptel.org> wrote: > On May 11, 2005 at 20:18, a c <lra101(a)yahoo.com> > wrote: > > 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. > > Thanks, it's fixed on the latest cvs. > > (some minor on ser exit missing null pointer check) > > Andrei > Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html

2 1

[Serusers] UAC module (backport to 0.9.0)
by Michael Ulitskiy 12 May '05

12 May '05

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 occured"); } 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

1 0

[Serusers] PSTN call disconnect after 20 seconds
by José Ricardo Maia Moraes 12 May '05

12 May '05

I just install the SER on Sun Solaris 8 with a Cisco AS5300 gateway do PSTN. Between SIP phones all work's fine, but connections between SIP phones and the gateway (both directions) are dropped after 20 seconds. I'm newbie on SER, anyone can help me to fix this behavior. Regards, José Ricardo Maia Moraes

1 0

[Serusers] call routing
by Edgardo O. Gonzales II 12 May '05

12 May '05

Hi! Can somebody teach me how to enable call routing .. Meaning if the called number is not on my routing table, I will forward the request to another sip gatekeeper .. Need it so badly.. please help .. thanks, ed

2 1

[Serusers] How to check if a user is registered (in location table)
by PHAN, Quang-Minh (Docteur Ordinateur) 12 May '05

12 May '05

Hi List, Is there a function in some module which provides a way to count how many contacts a user has registered? Thanks. Regards, Minh

1 0

[Serusers] uri table?
by Gerd Feiner 12 May '05

12 May '05

Hi, since i introduced the uri-module to my configuration users get added to subscriber and uri. what's that table for and what's its purpose? Gerd

1 0

[Serusers] is there anyone who has configured ser 0.8.14 with remote radius server
by Shaikat Mahmud 12 May '05

12 May '05

Hi, I am trying to configure ser 0.8.14 with remote radius. I am using radiusclient 0.4.8 And freeradius.0.9.1 And my os is debian It is ok for local server. But when I am trying to do acc with remote radius, it is not sending any information. Now by seeing the debugging information ( and some code), I have found that the error has generated from the radiusclient. Now has anybody done it? I have found that many people have faced lots of problem with ser-0.8.14 and remote radius. It works for ser-0.8.12 Thank you, Shaikat Mahmud

1 0

[Serusers] How to test SER + FreeRadius
by Silent Man 12 May '05

12 May '05

Hi, I'm new to Ser, and I've just finished installing Ser + Freeradius. the problem is how to test my configuration? I've tried to register with softphones (X-lite), but I'm getting this error "Login Timed Out! contact the network administrator". SER and FreeRadius are launched in debug mode, but i don't see happens (no logs) when I try to register !!! Thanks

2 1

[Serusers] Registering SER
by Kaveh Aasaraai 12 May '05

12 May '05

Hi, I'm just a new SIPper! and have a difficulty finding how to register SER to someone else, like Asterisk. In the beginning, it seemed impossible to me, but now I feel that I can register SER to Asterisk, making SER's users able to establish calls to Asterisk's users. I appreciate any help if anyone could give. K. Aasaraai

2 1

[serusers]sipsack
by michael p 12 May '05

12 May '05

hi somebody can explain me how to use sipsack command? i need to find how to do a ping in SIP to know the latency between SER and the user's phone there is something linked to sipsack (or serctl)? thanks M.

2 2

[Serusers] help me about voicemail(sems)
by ooo ooo 12 May '05

12 May '05

when I call voicemail group user, I get error : [root@sip800 sbin]# ./sems -E Configuration: configuration file: /usr/local/etc/sems/sems.conf Ser's FIFO: /tmp/vm_ser_fifo our FIFO: /tmp/am_fifo plug-in path: /usr/local/lib/sems/plug-in/ daemon mode: 0 local IP: 192.168.0.3 (4939) WARNING: reloadModuleConfig (SemsConfiguration.cpp:66): no configuration found for module isdngw, maybe you want to specify config.isdngw in config file. (4939) ERROR: send (AmRtpPacket.cpp:119): while sending RTP packet: Invalid argu ment (4939) ERROR: send (AmRtpStream.cpp:179): while sending RTP packet.(4939) ERROR: send (AmRtpPacket.cpp:119): while sending RTP packet: Invalid argument (4939) ERROR: send (AmRtpStream.cpp:179): while sending RTP packet. who have same error , pls help me , thank you. B.R. Zhao Zigang _________________________________________________________________ 享用世界上最大的电子邮件系统― MSN Hotmail。 http://www.hotmail.com

1 0

RE: [Serusers] Radiator + SER
by Ricardo Martinez 12 May '05

12 May '05

You have to install SER with Radius support like any other Radius Server. There is no particular configuration to work with Radiator. There is a RADIUS_HOWTO guide in iptel site... that may help you. Regards Ricardo Martinez.- > -----Mensaje original----- > De: Edgardo O. Gonzales II [mailto:edgardo.g@pacific.net.ph] > Enviado el: Miércoles, 11 de Mayo de 2005 4:01 > Para: serusers(a)lists.iptel.org > Asunto: [Serusers] Radiator + SER > > > > Hi! > > Can somebody help us on how to install SER supporting radiator. > If possible provide us a scratch for us to easily understand the > process. > > Your immediate feedback is highly appreciated. > > thanks, > ed > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

3 2

[Serusers] socket="/tmp/ser_socket" - parse error
by davor jovanovic 12 May '05

12 May '05

Hi all, I have bad config file when I try to enable socket="/tmp/ser_socket" I'm using SER 0.8.14 from CVS. Can anyone help? Regards

1 0

[Serusers] SER 0.9.0 on Mandrake 10.0 with 3rd Party Radius AAA
by Ozan Blotter 12 May '05

12 May '05

Hi List, Is there anyone using "Advanced VoIP Billing Software", i have a few questions regarding to it : Both Billing Software and SER installed on same machine and LAN IP is 192.168.1.11 (test system) Linux Dist : Mandrake 10.0 (OS) Apache : Installed (base web server) Java : Installed (important for tomcat) Tomcat : Installed (important for web interface) Perl & PHP : Installed (and a few perl modules for various applications) MySql : Installed (advanced voip needs it and creates to database like "raddb" and "voipbilling") PostgreSql : Installed (no need for it at this time) Radiusclient-ng : Installed (it's needed for SER itself) I followed steps on http://www.iptel.org/ser/doc/ser_radius/ser_radius.html except freeradius server as normal. I've installed SER from RPM's for Mandrake: ser-0.9.0-1mdk.i586.rpm ser-jabber-0.9.0-1mdk.i586.rpm ser-mysql-0.9.0-1mdk.i586.rpm ser-radius-0.9.0-1mdk.i586.rpm <<<<< SER.CFG >>>>> [root@localhost ser]# cat ser.cfg # # $Id: ser.cfg,v 1.27 2005/03/10 14:16:25 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) #memlog=5 # memory debug log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" #user=ser #group=ser #fifo_user=ser # owner of the ser fifo #fifo_group=ser #fifo_mode=0660 # fifo's permissions #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/auth_radius.so" loadmodule "/usr/lib/ser/modules/xlog.so" # ----------------- setting module-specific parameters --------------- modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius", "service_type", 15) 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 if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("","0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } <<<<< SER --- When I run it >>>>> [root@localhost root]# ser -c 0(2178) WARNING: fix_socket_list: could not rev. resolve 192.168.1.11 0(2178) WARNING: fix_socket_list: could not rev. resolve 192.168.1.11 Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.1.11 [192.168.1.11]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.1.11 [192.168.1.11]:5060 Aliases: tcp: localhost:5060 udp: localhost:5060 config file ok, exiting... [root@localhost root] <<<<< RADIUS --- When I run it >>>>> [root@localhost advradius]# ./runserver.sh Wed, 11 May 2005 13:05:58 095: FINE: <GlobalConf> AuthenticationPort = 1812 AccountingPort = 1813 BindAddress = null DictionaryFile = /billing/advradius/conf/dictionary LogDirectory = /billing/advradius/log LogLevel = FINE ConsoleWrite = true FileWrite = true ConsiderSrcIPAsNASIP = false </GlobalConf> Wed, 11 May 2005 13:05:58 739: FINE: <ThreadPoolManager> NumThreads = 10 MaximumWorkTime = 5000 KeepAliveTime = -1 ExpiryThreadSleepInterval = 5000 </ThreadPoolManager> Wed, 11 May 2005 13:05:58 882: FINE: <Client DEFAULT> NASIdentifier = default Secret = mysecret Realm = db </Client> Wed, 11 May 2005 13:05:58 917: FINE: <Realm db> 18AuthBy = [db] AcctPolicy = BACK-UP Accounting = [db, file_acct] Session = db </Realm> Wed, 11 May 2005 13:05:59 708: FINE: <DBParams DEFAULT> DriverName = org.gjt.mm.mysql.Driver DBURL = jdbc:mysql://127.0.0.1:3306/voipbilling UserName = root UserPassword = </DBParams> Wed, 11 May 2005 13:05:59 967: FINE: <AuthBy db> CheckList = [(5, (260, 0)) , (6, (0, 0)) ] ReplyList = [(1, (101, 9)) , (2, (102, 9)) , (3, (103, 9)) , (4, (6, 0)) , (7, (0, 0)) ] BackStore = DB PwdEncType = 0 SimultaneousLogin = 0 BackStore = DB DBParams = null AuthQuery = Select CallAmnt, CallTime, ReturnCode, AuthServiceType, Password, OtherCheckItems, OtherReplyItems from TblActiveSessions where AuthConfID = $cisco-h323-conf-id and AuthType = @GetRequestNumber@ AttribMapping = ColMapped </AuthBy> Wed, 11 May 2005 13:05:59 990: FINE: <Session db> BackStore = DB InsertSession = Insert INTO TblActiveSessions ( UserName, TimeStart, AuthType, CalledNumber, AuthCallingNumber, AuthNASIPAddress, AuthNASPort, AuthNASPortType, AuthConfID, CallOrigin, CallType, AcctSessionID ) values ($User-Name, $Event-Time, 3, $Called-Station-Id, $Calling-Station-Id, $NAS-IP-Address, $NAS-Port, $NAS-Port-Type, $cisco-h323-conf-id, $cisco-h323-call-origin, $cisco-h323-call-type, $Acct-Session-Id) DeleteSession = Delete from TblActiveSessions where AuthConfID = $cisco-h323-conf-id CountSession = Select count(*) from TblActiveSessions where LoginName = $User-Name and AuthConfId=$cisco-h323-conf-id BackStore = DB </Session> Wed, 11 May 2005 13:06:00 039: FINE: <Accounting db> BackStore = DB InsertAcct = Insert INTO TblCallsRad ( TimeClose, UserName, ActualDuration, CallingNumber, CalledNumber, ConfID, CallOrigin, CallType, AcctSessionID, NASIPAddress, BytesIn, BytesOut, DelayTime, AcctStatusType, RemoteAddress, RemoteGatewayID, TerminationCause, ) values ( $Event-Time, $User-Name, $Acct-Session-Time:NUMBER, $Calling-Station-Id, $Called-Station-Id, $cisco-h323-conf-id, $cisco-h323-call-origin, $cisco-h323-call-type, $Acct-Session-Id, $NAS-IP-Address, $Acct-Input-Octets:NUMBER, $Acct-Output-Octets:NUMBER, $Acct-Delay-Time:NUMBER, $Acct-Status-Type, $cisco-h323-remote-address, $cisco-h323-gw-id, $cisco-h323-disconnect-cause, ) QueueLength = 5000 RecordAcctStart = FALSE DBParams = null InsertCheckpointAcct = null </Accounting> Wed, 11 May 2005 13:06:00 070: FINE: <Accounting file_acct> BackStore = FILE InsertAcct = null QueueLength = 5000 RecordAcctStart = TRUE FileNameFormat = yyyy-MM-dd-HH.csv AcctDirectory = AcctLog </Accounting> Your evaluation license will expire in 25 day(s). Wed, 11 May 2005 13:06:00 137: INFO: Listener started at: 1812 Wed, 11 May 2005 13:06:00 143: INFO: Listener started at: 1813 Started Successfully <<< Anyone have any idea how to implement this valuable software in SER because we're gonna pay for it.

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[Serusers] SER and NavisRadius not working
by davor jovanovic 12 May '05

12 May '05

Hi all, I'm trying to configure SER 0.8.14 to use authentication with NavisRadius 4.5.0. On my X-Lite (X-Lite v2.0 Build 1103m) client I'm getting "Login Failed! Contact Network Admin." I'm getting following message from NavisRadius. Why? => Reply-Message = "No check.password" I would be very grateful if someone could look at my ser.cfg and output from NavisRadius. Best regards **************************************************************************** * This is what I get on RADIUS server (different from SER machine) Client: Client-Class = "#default" Nas_Port_Normalization = off Radius_Remove_Trailing_Nul = TRUE Radius_Append_Trailing_Nul = FALSE Auto_Remove_Check_Items = TRUE Check_Authenticators = TRUE Session_Time_From_Time_Of_Day = FALSE Radius_Charset = "UTF8" Client-Class = "video" Client-Dictionary = "draft_ietf_radext_digest_auth_01" **** dictionary of radiusclient on SER machine is adjusted to dictionary of RADIUS, and RADIUS can recognize attributes - except if I MUST run sterman_aaa_sip_00 on NavisRadius???? I attached dictionary in radiusclient to this message **** Client-Secret = <hidden> Request: User-Name = "djovanov.srce" Digest-Username = "djovanov.srce" Digest-Realm = "srce.hr" Digest-Nonce = "427b5aa983df858da94c50d1f8132a69e3e703ad" Digest-URI = "sip:srce.hr" Digest-Method = "REGISTER" Digest-Response = "ca6202fe55c51501295a9bc6ab325420" Service-Type = IAPP-Register Anonymous = v0-a208-646A6F76616E6F7669632E73726365 **** RADIUS doesn't recognize code 208, which is Sip-Uri-User **** ****Am I missing Digest-Algorithm? Why SER doesn't send this attribute?**** NAS-IP-Address = 161.53.0.131 NAS-Port = 5060 Packet: Client-Name = "161.53.0.131" Packet-Type = Access-Request Packet-Identifier = 213 Packet-Length = 197 Packet-Authenticator = 2B25D6D4934B930EB21F8E1B6AEFFE50 Source-Address = 161.53.0.131 Source-Port = 33159 Destination-Address = 0.0.0.0 Destination-Port = 1812 Receipt-Time = "2005/05/06 13:44:32" Full-User-Name = "djovanov.srce" Base-User-Name = "djovanov.srce" Normalized-Nas-Port = 5060 19 <engine.worker.0> -> checkDigest[AuthHttpDigest] 19 <plugin.AuthHttpDigest.checkDigest> FAILURE -- No check.password 20 <engine.worker.0> Variable group trace Reply: Reply-Message = "No check.password" **** this is message is which I get from NavisRadius **** **************************************************************************** ************ This is my ser.cfg fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ 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/textops.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_radius.so" loadmodule "/usr/local/lib/ser/modules/group_radius.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 10) modparam("rr", "enable_full_lr", 1) modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius", "service_type", 15) modparam("group_radius", "use_domain", 0) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; };

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[Serusers] Ser Accepting Calls From Asterisk
by Mr Greg Plater 12 May '05

12 May '05

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 <mysql://ser:heslo@localhost/ser%22> ") 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

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[Serusers] VOIP analizer
by Ernesto Lucero 12 May '05

12 May '05

Hi I just finished a sip Server and i am making calls t i am trying to map the call on what it is doing so i can analize it Thanks Ernesto

2 1

[Serusers] Early media in 183 with nathelper/ rtpproxy.
by Schweizer Laurent 12 May '05

12 May '05

Hello, I use ser 0.9.0 and I try to fork a call to a sip ipPhone and to a mobile phone via a pstn gateway. The problem is that the gateways send a 183 Session Progress with early media and if I answer with the ipPhone the nathelper do not use the SDP from the 200 OK but continues to forward the rtp stream to the gateways ( SDP received in the 183). Thanks Laurent

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[Serusers] Call Forking and Mediaproxy
by reticent 12 May '05

12 May '05

Does anyone know of a way to use mediaproxy with forked phone calls? It seems to only allocates a single media session when use_media_proxy() is called on a SIP message with multiple branches, so when one session is established and the other is torn down, the media session with media proxy is also torn down.

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[Serusers] Installing SER 0.9.0 from source
by Felipe Martins 12 May '05

12 May '05

Hi everybody, I'm using Slackware 10.1, and I'm migrating SER 0.8.14 to 0.9.0, is there a good howto explaining how to install it with full resources ? I found the INSTALL file (the one that comes with ser-0.9.0.tar.gz) very confusing. Thanks a lot 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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[Serusers] udp_send failed and can't get bye in accounting (Second try)
by Chris St Denis 12 May '05

12 May '05

When I try to route calls through my SER using sipp, I get there errors May 11 17:17:34 proxy-ser ser[23159]: ERROR: udp_send: sendto(sock,0x283555b8,650,0,0x28355960,16): Can't assign requested address(49) May 11 17:17:34 proxy-ser ser[23159]: msg_send: ERROR: udp_send failed May 11 17:17:34 proxy-ser ser[23159]: ERROR: t_forward_nonack: sending request failed Google didn't turn up anything interesting on them. Anyone here know whats up with them? OS: FreeBSD 5.3 SER: version: 0.8.14 (i386/freebsd) Also, I have accounting enabled to a database but I can not get it to log Bye messages. I know they are going through my server from some debug messages but they are not making it into the database. partial ser.cfg shown below. Any other suggestions on it are also appreciated. # Accounting loadmodule "/usr/local/lib/ser/modules/acc.so" # -- acc params -- # set the reporting log level modparam("acc", "log_level", 1) modparam("acc", "db_flag", 1) #modparam("acc", "db_missed_flag", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 1) modparam("acc", "early_media", 1) route{ # xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> - %tu\n"); # 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 -- probebly a loop or something"); 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(); /* grant Route routing if route headers present */ # 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(!(src_ip==x.x.x.x || src_ip==127.0.0.1)) { if (method=="REGISTER") { if (!www_authorize("proxy-ser.voip.aebc.com", "subscriber")) { www_challenge("proxy-ser.voip.aebc.com", "1"); break; }; save("ser_location"); break; } else if (method=="INVITE"){ if (!proxy_authorize("proxy-ser.voip.aebc.com", "subscriber")) { proxy_challenge("proxy-ser.voip.aebc.com", "1"); break; }; if(!check_from()) { sl_send_reply("403", "You can't lie about who you are"); } } } #Accounting if a call is started or ended if (method=="INVITE" || method=="BYE" || method=="CANCEL") { # xlog("L_NOTICE", "Accounting INVITE/BYE/CANCLE\n"); setflag(1); acc_db_request("", "acc"); } # native SIP destinations are handled using our USRLOC DB if (lookup("ser_location")) { xlog("L_NOTICE", "Routing - Incommming call \n"); if (!t_relay()) { sl_reply_error(); }; break; }; } else { if(!(method=="ACK")) { sl_send_reply("403", "I only relay for my own customers"); } }; # authorize only for INVITEs . RR/Contact may result in weird # things showing up in d-uri that would break our logic; our if (method=="INVITE") { ## ## Conditional relaying using if(uri=~ and rewritehostport omitted for space and security ## } # 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(); };

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[Serusers] udp_send failed and can't get bye in accounting
by Chris St Denis 12 May '05

12 May '05

When I try to route calls through my SER using sipp, I get there errors May 11 17:17:34 proxy-ser ser[23159]: ERROR: udp_send: sendto(sock,0x283555b8,650,0,0x28355960,16): Can't assign requested address(49) May 11 17:17:34 proxy-ser ser[23159]: msg_send: ERROR: udp_send failed May 11 17:17:34 proxy-ser ser[23159]: ERROR: t_forward_nonack: sending request failed Google didn't turn up anything interesting on them. Anyone here know whats up wiht them? Also, I have accounting enabled to a database but I can not get it to log Bye messages. I know they are going through my server from some debug messages but they are not making it into the database. partial ser.cfg shown below. Any other suggestions on it are also appriciated. # Accounting loadmodule "/usr/local/lib/ser/modules/acc.so" # -- acc params -- # set the reporting log level modparam("acc", "log_level", 1) modparam("acc", "db_flag", 1) #modparam("acc", "db_missed_flag", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 1) modparam("acc", "early_media", 1) route{ # xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> - %tu\n"); # 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 -- probebly a loop or something"); 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(); /* grant Route routing if route headers present */ # 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(!(src_ip==x.x.x.x || src_ip==127.0.0.1)) { if (method=="REGISTER") { if (!www_authorize("proxy-ser.voip.aebc.com", "subscriber")) { www_challenge("proxy-ser.voip.aebc.com", "1"); break; }; save("ser_location"); break; } else if (method=="INVITE"){ if (!proxy_authorize("proxy-ser.voip.aebc.com", "subscriber")) { proxy_challenge("proxy-ser.voip.aebc.com", "1"); break; }; if(!check_from()) { sl_send_reply("403", "You can't lie about who you are"); } } } #Accounting if a call is started or ended if (method=="INVITE" || method=="BYE" || method=="CANCEL") { # xlog("L_NOTICE", "Accounting INVITE/BYE/CANCLE\n"); setflag(1); acc_db_request("", "acc"); } # native SIP destinations are handled using our USRLOC DB if (lookup("ser_location")) { xlog("L_NOTICE", "Routing - Incommming call \n"); if (!t_relay()) { sl_reply_error(); }; break; }; } else { if(!(method=="ACK")) { sl_send_reply("403", "I only relay for my own customers"); } }; # authorize only for INVITEs . RR/Contact may result in weird # things showing up in d-uri that would break our logic; our if (method=="INVITE") { ## ## Conditional relaying using if(uri=~ and rewritehostport omitted for space and security ## } # 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(); };

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[Serusers] auth_radius
by Mihaly Zachar 11 May '05

11 May '05

Hi everybody, I'm playing with SER andd I would like to add tha auth_radius module.. versions: ser: 0.8.14 radiusclient: 0.4.8 The compilation went well, but when I start SER, it says: 0(31749) DEBUG: init_mod: auth_radius 0(31749) auth_radius - Initializing 0(31749) auth_radius: No `Cisco' vendor in Radius dictionary 0(31749) find_export: found <pre_auth> in module auth [/usr/local/lib/ser/modules/auth.so] 0(31749) find_export: found <post_auth> in module auth [/usr/local/lib/ser/modules/auth.so] 0(31749) ERROR: auth_radius: can't get code for the User-Name attribute 0(31749) init_mod(): Error while initializing module auth_radius ERROR: error while initializing modules the relevant part of my ser.cfg: ----------------------------------------- CUT -------------------------------------------- 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/exec.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "sql://xxx:yyy@localhost/ser") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") ----------------------------------------- CUT -------------------------------------------- radclient.conf: auth_order radius,local login_tries 4 login_timeout 60 nologin /etc/nologin issue /usr/local/etc/radiusclient/issue authserver localhost acctserver localhost servers /usr/local/etc/radiusclient/servers dictionary /usr/local/etc/radiusclient/dictionary dictionary /usr/local/etc/radiusclient/dictionary.sip dictionary /usr/local/etc/radiusclient/dictionary.ascend dictionary /usr/local/etc/radiusclient/dictionary.compat dictionary /usr/local/etc/radiusclient/dictionary.merit login_radius /usr/local/sbin/login.radius seqfile /var/run/radius.seq mapfile /usr/local/etc/radiusclient/port-id-map default_realm radius_timeout 10 radius_retries 3 bindaddr localhost login_local /bin/login As you can see, I have added all of the dictionary files, but still does not work :( what can be the problem???? Thanks, Regards Misi

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[Serusers] Ser Accepting Calls From Asterisk
by Mr Greg Plater 11 May '05

11 May '05

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 <mysql://ser:heslo@localhost/ser%22> ") 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

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[Serusers] maxfwd.so and problems with "mf_process_maxfwd_header()" function
by reticent 11 May '05

11 May '05

I've configured call forwarding in my SER script but to catch a situation where a user forwards to another user that is forwarding back to the first user i've added a second "if ( !mf_process_maxfwd_header("10") )" stanza to the SER script. However, i'm finding that it will actually add additional "Max-Forwards" headers to the SIP message as opposed to decrimenting one that is already there: -- Max-Forwards: 9. Record-Route: *REMOVED* Max-Forwards: 10. Max-Forwards: 10. -- I am under the impression that when called the mf_process_maxfwd_header() function should search the message for the "Max-Forwards" header and only add additional headers if it is missing. tavis

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[Serusers] QoS with rtp
by Lucas Aimaretto 11 May '05

11 May '05

Hi all, I've used for long time the QoS capabilities of linux for shaping/prioritizing traffic. Now, i'm willing to prioritize rtp packets in my lan, over the rest ot packets. I've used the u32 filters, but I did not really found a nice way of "matching" a rtp packet. I read the rfc so as to look at the rtp-header and found that the only field wich does not change its value along time is the version-field. It is located at the 29th byte ( with byte 0 at the IP heder first byte ). But it is a really poor way of doing it, because there may be many other packets wich will "get caugth" by this rule ... any ideas ? any experience in the subject ... Regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.8 - Release Date: 10/05/2005

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[Serusers] Good Free SipSoftphones for Linux !!!
by Felipe Martins 11 May '05

11 May '05

Hi everybody, Do anybody know a good sip softphone for linux that works with the following codecs: - G 723.1 - G 729a - G 711 - and others I'm experiencing the following problem. I have 2 ser servers, a sip softphone connected to each one of them talking to each other with no problem (softphones: phoner (M$), kphone (Linux)), my problem is that When I try to do the same test but between a softphone (linux or ms) and a HardPhone (Grandstream Budge Tone 100) I can't hear a thing. No sound is passed in any direction. I believe this is a codec problem, because between my softphones the conversation goes OK. Any Hand Thanks in Advance 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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[Serusers] (no subject)
by Juan Carlos Correa 11 May '05

11 May '05

1 0

[Serusers] call analizer
by Ernesto Lucero 11 May '05

11 May '05

Hi Does anybody have a good call analizar for sip Thanks moyotlan

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[Serusers] Shared Line Appearance
by Steve Blair 11 May '05

11 May '05

Hello: I just received a request for what I am calling shared line appearance. To me this means two IP phones register to SER with the same number. When a call arrives for that number one party answers the call on phone #1, places the call on hold then picks up the same call on phone #2. I can think of a way to make this work with call park but I do not think this is the seamless way our users want the process to work. Has anyone had to address this issue? Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@upenn.edu

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[Serusers] pstn->ser->uac->pbx
by raid＠gmx.net 11 May '05

11 May '05

hi following example: uac1-->pbx / pstngw---->ser-----> \ uac2 uac1 is user1 with pstn headnumber 12345 extensions 0-100, rpid=12345,alias=12345 if uac2 (or anybody from the pstn) calls 1234577 (or any other extension of uac1), how can i forward only the extensions to uac1? regards raid -- +++ Lassen Sie Ihren Gedanken freien Lauf... z.B. per FreeSMS +++ GMX bietet bis zu 100 FreeSMS/Monat: http://www.gmx.net/de/go/mail

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[Serusers] Billing and Blind call transfer
by Schweizer Laurent 11 May '05

11 May '05

Hello, I have made some test with cisco ip phone & blind transfer and I have a question. Example: A: a(a)toto.com B: b(a)toto.com C: c(a)toto.com M: mobile(a)toto.com A calls B but B forward the call to C, C answer the call and transfer the call to a mobile phone M via a pstn gateway. When C sends the REFER request to A we have something like this: REFER. From: b(a)toto.com To: a(a)toto.com Refer-To: mobile(a)toto.com Referred-By: b(a)toto.com And then when A sends the Invite to the gateways: INVITE. From: a(a)toto.com To: mobile(a)toto.com Referred-By: b(a)toto.com But the call is transferred by C and not by B so the Referred-By: is not correct. Is that an error from the cisco ip phone (I think not) and how can I bill this call (I do the billing on the gateways). Thanks. Laurent

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[Serusers] 404 answer
by valter_tex＠libero.it 11 May '05

11 May '05

Hi, in ser.cfg.sample I see that the answer related to subscribe message for unregistered user is "404 -Not found", this makes the subscription session end and so the UA doesn't send automatically other subscribe messages. Is there any other possible answer that avoids this behavior? Best regards, tex ____________________________________________________________ Navighi a 4 MEGA e i primi 3 mesi sono GRATIS. Scegli Libero Adsl Flat senza limiti su http://www.libero.it

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[Serusers] SER _0_9_0 installation Without root's Privileges
by Alex 11 May '05

11 May '05

hi Ser-users, [intro] i'm sorry if my post is redundant but i really need to know!! [problem] I haven't root's privileges on a linux Fedora core 3 Server but i really would like install SER and Mediaproxy(for NATed client) on my server i tried by make prefix=/home/my_folder/ser_production make prefix=/home/my_folder/ser_production modules make prefix=/home/my_folder/ser_production install but i've a lot of troubles (while compiling and while running ./ser o ./serctl) [Question] is it possible to install SER on my linux Box without root's permissions? thanks in advance!! Alex

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Re: [Serusers] divert call to PSTN-number?
by Klaus Darilion 11 May '05

11 May '05

Samuel Osorio Calvo wrote: > Hi, > > >>I'm not sure, but I think you can handle it this way. You can add an >>alias sip:mobilenumber@gateway to the user, and an alias to itself. >>Then, after lookup(alias) you will have 2 brachnes, the original one and >>the mobilenumber@gateway. Then, lookup(location) will add the branches > >>from the location table. > > Unfortunately, I think it won't work because the first lookup (in aliases) will rewrite the first contact as Req-URI and the second binding will be added as an extra branch. > When you make the second lookup (in location), only the Req-URI is checked in the location table (which is the one rewritten by the first lookup), loosing the branch obtained in the first lookup (aliases). > > Please, correct me if I'm wrong... Yes, but if you have 2 aliases: 1. sip:a@a -> sip:a@a 2. sip:a@a -> sip:number@gateway this should work - once I had misconfiured my aliases in this way and behave like this. Nevertheless, using a permanent entry in the location table is a much cleaner approach. regards, klaus

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[Serusers] NAT, ACK and BYEs
by Giovanni Balasso 11 May '05

11 May '05

Hi serusers, another NAT issue from mine... when I have two parts calling each other inside the same NAT (behind a router with public IP different from ser's) I directly relay the call between the two, without use of rtpproxy. The good thing is that audio works perfectly both ways, the bad one is that SIP messages from caller reach ser but aren't relayed to the callee. So ACKs and BYEs reach too many hops and if the caller hangs up, the callee remains connected until he hangs up too. On the other side (BYE from callee to caller) everything works fine. By "ngrepping" on ser port I noticed this difference on the header of SIP BYE, which from callee to caller is BYE sip:caller@CALLER_PUBLIC_IP (=CALLEE_PUBLIC_IP) and from caller to callee is BYE sip:callee@SER_PUBLIC_IP and ser keeps relaying to himself... INVITE routing logic is something like this if (loose_route()) { if (has_totag() && method=="INVITE") { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); t_on_reply("1"); }; t_relay(); break; } if (method=="INVITE") { record_route(); if (nat_uac_test(19)) { fix_nated_contact(); force_rport(); setflag(6); } if ( !isflagset(6) || SAME_NAT) { t_relay(); } else { force_rtp_proxy(); t_on_reply("1"); t_relay(); } } If I ignore same NAT test and force the call thru rtpproxy, all works as it should. I thought the problem was in fix_nated_contact and force_rport, but removing the two lines does nothing. any ideas? thanks in advance -- Giovanni Balasso giaso(a)yahoo.it

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Re: [Serusers] divert call to PSTN-number?
by Samuel Osorio Calvo 11 May '05

11 May '05

Hi, >I'm not sure, but I think you can handle it this way. You can add an >alias sip:mobilenumber@gateway to the user, and an alias to itself. >Then, after lookup(alias) you will have 2 brachnes, the original one and >the mobilenumber@gateway. Then, lookup(location) will add the branches >from the location table. Unfortunately, I think it won't work because the first lookup (in aliases) will rewrite the first contact as Req-URI and the second binding will be added as an extra branch. When you make the second lookup (in location), only the Req-URI is checked in the location table (which is the one rewritten by the first lookup), loosing the branch obtained in the first lookup (aliases). Please, correct me if I'm wrong... Samuel. Unclassified.

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[Serusers] rpids ..again and cli
by Iqbal 11 May '05

11 May '05

Hi Have been trying to get get rpid , and cli working, dropped xlite to test with grandstream just in case xlite didnt like it. I have rpid=12345 in my subscriber db, against username 040600 In debug I get save_rpid(): rpid value is '12345' 4(29312) authorize(): set string AVP 'rpid = 12345' when I add append_rpid_hf("<sip:", ">;party=calling;id-type=subscriber;screen=yes") to my ser.cfg just before I t_relay, now in theory (someones :-)) this should add the header, but using ngrep I dont see any headers added. Iqbal

1 2

[Serusers] command cannot be used in block
by Iqbal 11 May '05

11 May '05

Hi I am using this block in my t_reply route, and it seems that it from here that the rpid isnt being added, whereas on the first INVITE I seem to be adding it. However I cannot do what Ihave written below... if (!search("^Content-Length:\ 0")) { append_rpid_hf("<sip:", ">;party=calling;id-type=subscriber;screen=yes"); use_media_proxy(); }; Iqbal

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[Serusers] divert call to PSTN-number?
by Gerd Feiner 11 May '05

11 May '05

hi there, when configuring multiple SIP-clients with the same user-account all of them ring when a call comes in - fine. is there a possibility for adding a PSTN-number (actually, a mobile-number) to some sort of alias to make the mobile ring, too? i mean simultaneously with the other clients? Brgds, Gerd

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[Serusers] Fwd: divert call to PSTN-number?
by Gerd Feiner 11 May '05

11 May '05

Anfang der weitergeleiteten E-Mail: > Von: Gerd Feiner <g.feiner(a)cablesurf.de> > Datum: 11. Mai 2005 14:52:48 MESZ > An: serusers(a)lists.iptel.org > Betreff: divert call to PSTN-number? > > hi there, > > when configuring multiple SIP-clients with the same user-account all > of them ring when a call comes in - fine. is there a possibility for > adding a PSTN-number (actually, a mobile-number) to some sort of alias > to make the mobile ring, too? i mean simultaneously with the other > clients? > > Brgds, > Gerd

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[Serusers] Fwd: divert call to PSTN-number?
by Gerd Feiner 11 May '05

11 May '05

Anfang der weitergeleiteten E-Mail: > Von: Gerd Feiner <g.feiner(a)cablesurf.de> > Datum: 11. Mai 2005 14:52:48 MESZ > An: serusers(a)lists.iptel.org > Betreff: divert call to PSTN-number? > > hi there, > > when configuring multiple SIP-clients with the same user-account all > of them ring when a call comes in - fine. is there a possibility for > adding a PSTN-number (actually, a mobile-number) to some sort of alias > to make the mobile ring, too? i mean simultaneously with the other > clients? > > Brgds, > Gerd

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[Serusers] loose routing
by Schweizer Laurent 11 May '05

11 May '05

Hello, I have a question about loose routing. I use this in my ser script: if (loose_route()) { route(1); break; }; And ReInvite or Refer request are usually detected by loose_route and forwarded to route(1) but this Notify request is not detected. U 10.192.73.153:5060 -> 193.x.x.x5:5060 NOTIFY sip:193.x.x.x5:5060;ftag=003094c3b47e0047500eefd5-6e8490f4;lr=on SIP/2.0.. Via: SIP/2.0/UDP 10.192.73.153:5060;branch=z9hG4bK61010d9d.. From: "0244202372" <sip:0244202372@toto.org>;tag=003094c3b47e0047500eefd5-6e8490f4.. To: <sip:0244230714@toto.org>;tag=59d6fbea2c7bb69bi0.. Call-ID: 003094c3-b47e0047-00691ac0-184e7aaf(a)10.192.73.153.. CSeq: 103 NOTIFY.. User-Agent: CSCO/7.. Contact: <sip:0244202372@10.192.73.153:5060>.. Route: <sip:193.x.x.x5:5060;ftag=003094c3b47e0047500eefd5-6e8490f4;lr=on>, <sip:0244230714@193.x.x.x5:5060>.. Event: refer.. Proxy-Authorization: Digestusername="0244202372",realm="toto.org",uri="sip:193.x.x.x3",response=" ae4eef9f30f749d568cd1668b8056256",nonce="4281f693783c979f5aacffe3555e0cd6cd9 d4443",cnonce="65f34551",qop=auth,nc=00000002,algorithm=md5. Content-Type: message/sipfrag.. Content-Length: 14.... SIP/2.0 200 OK

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[Serusers] sip load balancing
by a c 11 May '05

11 May '05

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

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Re: [Serusers] Problem compiling postgres module
by Gorka Guridi 11 May '05

11 May '05

I solved the problem. This file is in package postgresql-dev of debian. I installed it, delete the precompiled version of ser that i had (and didn't work), and recompile all another time. It successfully creates the postgres.so Sorry for time lost. Regards. Gorka.

1 0

[Serusers] php_mysql package
by Llanos Serna García-Conde 11 May '05

11 May '05

1 0

RE: [Serusers] php with mysql, where is APXS?
by Llanos Serna García-Conde 11 May '05

11 May '05

1 0

Re: [Serusers] serweb:apxs and php
by Karel Kozlik 11 May '05

11 May '05

Hello Llanos, first: please include mailing list in your replies in order to others can see answers Debian is one from several linux distributions. See http://www.debian.org/ for more informations. I can't tell you which distribution are you have installed. Try write command 'apt-get'. If this command exists, it is possible that you are useing debian. By the way I recommend to learn linux basics before you will try install any application to it. regards Karel Llanos Serna García-Conde napsal(a): > > > > Hello, thanks for your help. I'm just a student that's why I have so > many problems. To be honest, I'm not sure if I use debian, I mean, I > don't work with it but maybe it's running and the system does work with > it,- This is what I've found in my PC. > > /usr/share/doc/dialog-0.9b/samples/debian.rc > /usr/share/pixmaps/nautilus/Bluecurve/gnome-application-x-debian-package.png > /usr/share/pixmaps/gnome-debian.png > /usr/src/ser-0.8.14/debian > /usr/src/ser-0.8.14/debian/changelog > /usr/src/ser-0.8.14/debian/compat > /usr/src/ser-0.8.14/debian/control > /usr/src/ser-0.8.14/debian/copyright > /usr/src/ser-0.8.14/debian/cron.d.ex > /usr/src/ser-0.8.14/debian/ex.package.doc-base > /usr/src/ser-0.8.14/debian/manpage.1.ex > /usr/src/ser-0.8.14/debian/manpage.sgml.ex > /usr/src/ser-0.8.14/debian/postinst > /usr/src/ser-0.8.14/debian/postrm > /usr/src/ser-0.8.14/debian/preinst.ex > /usr/src/ser-0.8.14/debian/prerm.ex > /usr/src/ser-0.8.14/debian/rules > /usr/src/ser-0.8.14/debian/shlibs.local > /usr/src/ser-0.8.14/debian/watch.ex > > Do you think i only need to install de packages you told me ? Or am I > not using debian? Thanks. > > Llanos > > >Hello, > >apx should be part of af apache. But is long ago when I last > >compiled php. I don't remember it well. If you are useing debian, > >you only need install packages: libapache-mod-php4, php4-mysql and > >php4-pear > > > >Karel > > > >Llanos Serna García-Conde napsal(a): > >>Hello, I've got a serious problem with the isntallation of Serweb > >>and would really appreciate some help. > >>Before trying to install SERWEB, I had already apache running on my > >>pc: > >>Server version: Apache/2.0.40. > >>The error I found when trying to log to SERWEB was > >>*Fatal error*: Call to undefined function: mysql_pconnect() in > >>*/var/www/html/serweb/serweb_2004-07-27/phplib/db_mysql.inc* on > >>line *73* > >>and according to the INSTALL documente that was given with SERWEB I > >>found out I had to install php with mysql support. > >>I downloaded php-4.3.11 and tried to follow instructions for > >>installation, but I needed to include the option --with-apxs and > >>the configuration tool couldn't find apxs, neither could I > >>manually. I tried to install it including only the option > >>--with-mysql but the error I get is still the same one. > >> What should i do? Should I reinstall apache? Any other ideas? Is > >>it a php-mysql error for sure?Thank you!! > >>Llanos > >> > >> > >> > >>------------------------------------------------------------------------ > >> > >>_______________________________________________ > >>Serusers mailing list > >>serusers(a)lists.iptel.org > >>http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] serweb:apxs and php
by Llanos Serna García-Conde 11 May '05

11 May '05

2 2

[Serusers] Can't run Ser with Radius
by Silent Man 11 May '05

11 May '05

Hi, I'm new to SER and I really need your help. I'm trying to install SER 0.8.14 with a radius (Freeradius) authentification and accounting, but I Still have this error message when I "make" : authrad_mod.c:111: error: `DICT_VENDOR' undeclared (first use in this function) authrad_mod.c:111: error: (Each undeclared identifier is reported only once authrad_mod.c:111: error: for each function it appears in.) authrad_mod.c:111: error: `vend' undeclared (first use in this function) authrad_mod.c:135: attention : affectation transforme un entier en pointeur sans transtypage authrad_mod.c:140: error: too many arguments to function `rc_conf_str' authrad_mod.c:140: error: too many arguments to function `rc_read_dictionary' authrad_mod.c:145: attention : implicit declaration of function `rc_dict_findvend' authrad_mod.c:163: error: too many arguments to function `rc_dict_findattr' authrad_mod.c:163: error: too many arguments to function `rc_dict_findval' make: *** [authrad_mod.o] Erreur 1 I've followed all the steps in this tutorial http://www.iptel.org/ser/doc/ser_radius/ser_radius.html, installed two radius client : Radius Client Library (http://developer.berlios.de/projects/radiusclient-ng) and (http://freshmeat.net/projects/radiusclient/). $ cat /usr/local/include/radiusclient.h |grep ,v * $Id: radiusclient.h,v 1.9 1999/01/06 23:53:04 lf Exp $ $ cat /etc/ld.so.conf /usr/X11R6/lib /usr/local/lib My Environnement contain these two variables : INCLUDE_PATH=/usr/local/include LIB_PATH=/usr/local/lib/ Thanks in advance.

1 2

[Serusers] exported function : m_store(mode)
by Vincent Verdot 11 May '05

11 May '05

Could someone explain me what does the "m_store(mode)" exported function from the "msilo" module do ? I read the doc but I don't understand...

1 1

Re: [Serusers] Re: [Serdev] Patched free-TLS implementation
by Cesc Santasusana 11 May '05

11 May '05

Hi, Good you found out. The reason i turned session caching completely off was because it was not properly configured. So, putting it on a TODO list, i disable it completely, waiting to add it later. Now i found out that the TLS implementation of minisip, the phone i use, was kinda broken. In my working copy i have activated and configured ser-tls with server-side caching, and it works. The problem is that i read that the default session caching provided by openssl is rather simple, and even that it should not be used in any production environment. That was another reason to disable it. And also that some changes in the way reads and writes are handled should be changed ... I will send a patch to fix all this ... Another thing ... the tls code needs to be more "configurable", from the config file. What is the best way to do this? Adding variables to cfx.XX files? or maybe a tls-module could do that also plus several other features? Regards, Cesc Unclassified >>> Alex Mack <amack(a)fhm.edu> 05/10/05 03:10PM >>> Hi again! Funny thing I found out: snom 190 doesn't even USE Session Caching but it seems to need the server to be alel to. It doesn't request the old Session ID, every new TLS connection from the phone to SER is a new one without a session ID. From what I understand with session caching, the client, in order to refetch a session, has to provide the session ID given by the server in the first handshake. Alex Mack Alex Mack schrieb: > Hi! > > I've managed to find out what's bothering the snom 190: It's the > missing TLS Session ID when Session Chaching is turned off. > > The session ID is given to the client for being able to request that > session later on from the server. But > > SSL_CTX_set_session_cache_mode( _ctx, SSL_SESS_CACHE_OFF ); > > turned off Session Caching, so no Session ID is generated and > transfered and that seems to confuse the snom phones. So they answer > the SERVER HELLO immediately with an ALERT and break up the > connection. It seems like snom190 need to have session caching turned > on... > > The snom's firmware is: snom190-SIP 3.60b 6249 > > snom's softphone (emulating a snom 360 with firmware "snomSoft-SIP > 3.57q") seems to have the same problem. > > BTW, what was the problem with session caching in the first place? > Performance hit? > > Alex Mack > > Benny Ben-Ami schrieb: > >> Try checking the expiration date of the certificate and the time & >> date of >> the phone. I once had a phone where the time was not set and he >> rejected the server >> certificate due to date expiration. >> -----Original Message----- >> From: serdev-bounces(a)iptel.org [mailto:serdev-bounces@lists.iptel.org] On >> Behalf >> Of Alex Mack >> Sent: Wednesday, May 04, 2005 7:29 PM >> To: Cesc Santasusana >> Cc: serdev(a)lists.iptel.org; serusers(a)lists.iptel.org >> Subject: Re: [Serusers] Re: [Serdev] Patched free-TLS implementation >> >> Hi Cesc! >> >> I tried your previuos suggestions on changing cleint certificate >> check. I'm now running SSL_VERIFY_NONE. MS Messenger works again. But >> the snom >> still fails. >> Ethereal tells me the server doesn't ask for a client certificate any >> more. >> Ethereal also shows that "Cipher Suite: TLS_RSA_WITH_RC4_128_SHA" was >> selected. So far so good, but the snom still rejects the Server Hello >> with an Alert. >> >> What else have you changed? The original version worked with the snoms. >> >> BTW patch throws some warnings: >> >> # patch -i patch.core.cfg.files.diff >> patching file cfg.y >> Hunk #1 FAILED at 1. >> 1 out of 6 hunks FAILED -- saving rejects to file cfg.y.rej patching >> file >> cfg.lex Hunk #1 FAILED at 1. >> 1 out of 5 hunks FAILED -- saving rejects to file cfg.lex.rej >> >> cfg.y.rej reads: >> *************** >> *** 1,5 **** >> /* >> - * $Id: cfg.y,v 1.2 2005/01/06 14:35:10 sam Exp $ >> * >> * cfg grammar >> * >> --- 1,5 ---- >> /* >> + * $Id: cfg.y,v 1.4 2005/05/03 08:16:35 cesc Exp $ >> * >> * cfg grammar >> * >> >> cfg.lex.rej reads: >> *************** >> *** 1,5 **** >> /* >> - * $Id: cfg.lex,v 1.2 2005/01/06 14:35:10 sam Exp $ >> * >> * scanner for cfg files >> * >> --- 1,5 ---- >> /* >> + * $Id: cfg.lex,v 1.3 2005/04/11 08:18:31 cesc Exp $ >> * >> * scanner for cfg files >> * >> >> Seems to be a minor mismatch, it compiles well anyway. >> >> Alex Mack >> >> Cesc Santasusana schrieb: >> >> >> >>> Hi, >>> >>> Yeah ... I send it with my default config which is using client and >>> server >>> >> >> authentication. >> >> >>> I use this settings with minisip client (supports client side certs) >>> and >>> >> >> for tls between ser proxies. It works perfect. >> >> >>> To turn client authentication off, check: >>> tls/tls_init.c file >>> init_ssl_ctx_behavior function >>> >>> the line >>> SSL_CTX_set_verify( _ctx, SSL_VERIFY_PEER | >>> SSL_VERIFY_FAIL_IF_NO_PEER_CERT, 0); >>> >>> means that the server will request a certificate from the client and >>> if it >>> >> >> doesn't get one, it will fail. >> >> >>> Try changing it with: >>> SSL_CTX_set_verify( _ctx, SSL_VERIFY_PEER | SSL_VERIFY_CLIENT_ONCE, >>> 0); this way the server will request a cert, the client will not >>> provide, but >>> >> >> on the following renegotiation, the server (ser) will not ask for a >> cert. >> >> >>> And if you want to turn verification off ... completely ... >>> SSL_CTX_set_verify( _ctx, SSL_VERIFY_NONE, 0); this will also work >>> for you if only using tls for ser2phone ... it will not work if you >>> want tls >>> >> >> between proxies ... as the ser client will accept ANY certificate >> from the >> ser server. >> >> >>> And then, from the prompt: >>> >>> >>> >>>> make TLS=1 all && make TLS=1 install ;) >>>> >>>> >>> >>> This whole verification thing needs to be improved and probably the >>> >> >> parameters should be changeable directly from the config file. This >> and many >> other parameters should be exchangeable without the need to recompile. >> >> >>> Any volunteer for a ser-tls.README? :D >>> >>> Can you provide me with some extra info from the snom phones and the >>> >> >> messenger? Do it offline, so you can send me some ethereal captures >> and ser >> logs ... >> >>> Regards, >>> >>> Cesc >>> >>> >>> >>> >>> >>> >>>>>> Alex Mack <amack(a)fhm.edu> 05/04/05 02:01PM >>> >>>>>> >>>>>> >>>>> >>> Hi Cesc! >>> >>> I compiled in your patch. >>> >>> Now I'm facing a new problem: SER wants a client certificate from >>> the UA. Snom phones immediately reply with an ALERT and break up >>> connection upon the certificate requests. MS Messenger on the other >>> hand sends at least a reply - without certificate - and SER rejects >>> the Client Hello because of the missing client certificate: >>> >>> tls_accept: Error in SSL: >>> tls_error: error:140890C7:SSL routines:SSL3_GET_CLIENT_CERTIFICATE:peer >>> did not return a certificate >>> >>> Could you please provide some more documentation about the new TLS >>> options you added? It seems you've implemented support for client >>> certificates for a two-way certificate authorization - which would >>> be a good thing if supported by the UAs, which don't right now. So >>> how can I turn it off again and get back to server side certification? >>> >>> Alex Mack >>> >>> Cesc Santasusana schrieb: >>> >>> >>> >>> >>> >>>> Hi everybody, >>>> >>>> The last i sent is a replacement as a whole for the original code >>>> sent by >>>> >>> >> P. Griffiths. Sorry i forgot to mention that. >> >> >>>> The patches for cfg.y and cfg.lex are both in the same file >>>> (patch.core.cfg..files.diff) within the zip. I was lazy :) I resent >>>> it as >>>> >>> >> a whole, and not as a diff, because i indented all the code with tabs, >> instead of spaces (so a diff would be bigger than just sending all the >> files). >> >> >>>> As for the CVS thing ... i agree with Juha. Either gets into the >>>> >>> >> "official" cvs or we do something about it. The code i think is rather >> stable as it is (i only tested on my debian linux box, soon i will >> try on an >> ARM linux and i will report back on that too). For me, as long as it >> gets >> into a CVS, i don't care if it is mantained against HEAD or 0.9.0 (i use >> 0.9.0 .... so all my patches are against it). >> >>>> On a more philosophical level, i understand the "quietness" on iptel's >>>> >>> >> side ... they have their own version, and make money on it. But the >> thing is >> that this free version is here to stay ... it is the "problem" of >> opensource. >> >>>> Another option would be for them to release their proprietary >>>> >>> >> implementation if they feel that it is a better, more tested one. >> >>>> In any case, i think that this whole thing needs to be decided fast. >>>> >>>> Regards! >>>> >>>> Cesc >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>>>>> Alex Mack <amack(a)fhm.edu> 05/03/05 01:26PM >>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>> Hi Cesc! >>>> >>>> Nice to have those fixes in a package. >>>> >>>> Is your cfg.y-patch to be applied *after* cfg.y.patch was applied or >>>> *instead* of cfg.y.patch? >>>> >>>> Or is your version a patched one which replaces the original >>>> implementation as a whole? In that case where's cfg.lex.patch? >>>> >>>> Alex Mack >>>> >>>> Cesc Santasusana schrieb: >>>> >>>> >>>> >>>> >>>> >>>>> Hi, >>>>> >>>>> I really hate to be so pushy, but i dont understand how such an >>>>> important >>>>> >>>> >> piece of code as TLS is not moving on into CVS ... or anywhere else >> by this >> matter. I will keep sending patches till i get tired (soon). >> >> >>>>> Anyway ... i thought someone may be interested in a compilation >>>>> fix for >>>>> >>>> >> cfg.y introduced with the tls_domains (it would not compile if the cfg.y >> file had been patched but the tls-core files were not there); a bug >> fix for >> the session caching (fixed by turning session caching and resumption >> off); >> and an extension (the ability to choose the list of allowed ciphers >> from the >> config file). Oh, and all the files have been tabbed, instead of >> spaced (for >> indentation). >> >> >>>>> Enjoy! >>>>> >>>>> Cesc >>>>> >>>>> >>>>> >>>>> >>>> >>> >>> >>> >>> >>>>> --------------------------------------------------------------------- >>>>> --- >>>>> >>>>> _______________________________________________ >>>>> Serdev mailing list >>>>> serdev(a)lists.iptel.org >>>>> http://lists.iptel.org/mailman/listinfo/serdev >>>>> >>>>> >>>>> >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> 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 >>>> >>>> >>>> >>>> >>>> >>> >>> >>> >>> >>> >> >> >> _______________________________________________ >> Serdev mailing list >> serdev(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serdev >> >> >> >> >> > > _______________________________________________ > 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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[Serusers] Ser Accepting Calls From Asterisk
by Mr Greg Plater 11 May '05

11 May '05

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; };

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[Serusers] Radiator + SER
by Edgardo O. Gonzales II 11 May '05

11 May '05

Hi! Can somebody help us on how to install SER supporting radiator. If possible provide us a scratch for us to easily understand the process. Your immediate feedback is highly appreciated. thanks, ed

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[Serusers] RE: [Sems] How to absorb ACKs in UAS mode?
by Franz Edler 11 May '05

11 May '05

From: Stefan Sayer Sent: Tuesday, May 10, 2005 4:02 PM > in my opinion t_newtran() will release the transmission on processing of > an ACK so the 200 is not retransmitted; have a look at t_newtran (in > t_lookup.c about line 1094). Unfortunately I cannot verify this behaviour. See my code snippet of ser.cfg below. Without explicitly using t_relay() for ACK the 200 OKs are retransmitted by SER. Calling t_newtran() is obviously not enough to stop retransmissions. Any further opinions around? Regards Franz --- code snippet of ser.cfg follows -------------------------------------- if (method=="INVITE" || method=="BYE" || method=="CANCEL" || method=="ACK") { # switch to stateful mode: if (!t_newtran()) { sl_send_reply("500","could not create transaction\n"); break; }; if (method=="INVITE" && uri=~"sip:910@") { t_reply("100","Trying - just wait a minute !"); if(!t_write_req("/tmp/am_fifo","announcement")) { t_reply("500","error contacting sems"); }; break; }; if (method=="BYE" || method=="CANCEL") { if(!t_write_req("/tmp/am_fifo","bye")) { t_reply("500","error contacting sems"); }; break; }; if (method=="ACK") { t_relay(); }; };

1 0

[Serusers]Serweb Problem
by chin31kimo 11 May '05

11 May '05

try to modify the "output_buffering=on" in the php.ini and restart your apache server >Hi SerUser! >Can somebody help me with this problem.. >After I install and configure serweb as stated on >configuration guide.. I >got a blank page >accessing admin and user_interface page. >On logs of my apache, this is the feedback .. >210.23.237.236 - - [21/Apr/2005:23:51:12 +0800] "GET >/serweb/user_interface/index.php?phplib_Session=4026e0eaf3aa336098b7e1d2926f49ff&phplib_Session=dc200ec3587d1985cf9aaf706d3b248c&phplib_Session=8d66aaa5f4418cb75126beda37c2acde&phplib_Session=14d85207dd2ec6439325db9f78bea914&phplib_Session=3a23bcfc1ecae59dcefead744c64aaf3&phplib_Session=6ff4d142925c0574d1d3614bcb679578&phplib_Session=5d83ca96f291e0310ab4777442a547f9&phplib_Session=8ac2999cee413bde7ef4769b8f73b860&phplib_Session=96c5b130e01b2db96d73bc050248183e&phplib_Session=9ae3f9631ab331d0c0299cf394e6f691&phplib_Session=ddb01b22fdc4d988a9093a37cf4ee00b&phplib_Session=d48993a3af2167a562d151f2cb285ce4&phplib_Session=2492b77a05e07417c38d229d30f58d9a&phplib_Session=a4edc5415a40e4a922c4dbf721f25dc0&phplib_Session=5d41839c67fbc842ac27f59a558e982a&phplib_Session=7b5b95a63d36ffdf9b412264bf941691&phplib_Session=507571c99dc312b6e36ea364de2644f4&phplib_Session=0992981f5a11f248824edbb95f5db9f3&phplib_Session=1f25b8f300e72b404a54ab4c2c4060e8&phplib_Session=870611806d28cda0d0021f8422454! > 12a&ph >p >lib_Session=f5c422596455228e1749a081e1d9be1e&phplib_Session=2ecc3410c0ef13c4a66f12d51b02e7a3&phplib_Session=ff433b28f8596fe03518398fef080887&phplib_Session=aec18ca4299ec9009c236cde5d0bb733&phplib_Session=5937f583e214b20d67999da9947a9b7c&phplib_Session=854621f5353598bf6b6c9fd0f882766a&phplib_Session=0809e0fb3fddff21a14190e4240b814a&phplib_Session=22b80c053272e22c30c33a883182e014&phplib_Session=7c69db32533cd64b763462353c20d927&phplib_Session=c4eeba51a9991c5161269dafa9647d38&phplib_Session=472ca546c8ff6970ea9cae514940b86f&phplib_Session=c3c6871818a255d4eafe2d8a373228ef&phplib_Session=fded6f2f186c6d2dedd467a2b93f5c2a&phplib_Session=53033ca88aa7ec64b25d70f8ac424086&phplib_Session=62744ffaf3c7866e2f3324b1d6ca9f69&phplib_Session=d43a04d83057e27859e9a9db63782a4c&phplib_Session=bb309e7809c1341764212cab865e33ce&phplib_Session=002136a613c2bdee8d927d8a43be20ad&phplib_Session=82a678edfc54753212929c25f5fb5221&phplib_Session=2de93e7370b1de73041c6f4dc21a11df&phplib_Session=dfd10093b5f05cccd! >3178ad >f >cc3f8c62 HTTP/1.1" 302 - >Is there something I miss during the configuration .. >thanks, >ed _______________________________________________________________________ Yahoo!奇摩電子信箱免費容量250MB，信件在多也不怕 http://tw.promo.yahoo.com/mail_new/index.html

1 0

[Serusers] db_insert_ucontact(): Error while inserting contact
by Pawel Urawski 10 May '05

10 May '05

Hello, I would like to use my SER with dbtext for user location. I made everything that was written in the readmee file. Ser starts ok without any errors, but when I try to register a new user I get in syslog: May 10 22:07:53 qwe ser[2042]: db_insert_ucontact(): Error while inserting contact May 10 22:07:53 qwe ser[2042]: insert_ucontact(): Error while inserting in database Any idea ? Regards. Pawel.

1 0

[Serusers] BYE from Cisco gateway
by Torbjørn Lium 10 May '05

10 May '05

I'm using a cisco 1760 with a VIC2-4FXO card for my calls to PSTN. If a user on a softphone hangs up first the PSTN port on the cisco is released and new calls can be made on the same voice port. But when the user on the PSTN side hangs up first the voice port on the cisco stays open until the user on the softphone hangs up. Any ideas what I'm doing wrong?

2 2

[Serusers] migration 0.8.14 -> 0.9.0 and SQL broken
by Mik 10 May '05

10 May '05

Hello, I am started using SER from 0.8.14 version. All was fine. I use postgres SQL for registered location, stored cridentical username & password. But I needed some basic UAC functionalities like client authentication on other SIP-proxy. I take uac module. This module don't work with 0.8.14 SER. And I take 0.9.0. Compile and install new ser & modules. After that ALL my SQL shema is broken! SER don't work with SQL! Why? I nothing change in ser.cfg..... Old version 0.8.14 work, but new version 0.9.0 don't work. postgres - initializing stateless - initializing textops - initializing 0(0) Maxfwd module- initializing acc - initializing exec - initializing 0(0) bind_dbmod: Module sql does not export db_use_table function 0(0) ERROR: mod_init(): Can't bind database module 0(0) init_mod(): Error while initializing module usrloc ERROR: error while initializing modules If I am deleted 2 modules from SER.CFG: loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" I have error in other module postgres - initializing stateless - initializing textops - initializing 0(0) Maxfwd module- initializing acc - initializing exec - initializing 0(0) bind_dbmod: Module sql does not export db_use_table function 0(0) ERROR: auth_db_bind: unable to bind to the database module 0(0) init_mod(): Error while initializing module auth_db ERROR: error while initializing modules SER.CFG (very simple for debug): # ----------- global configuration parameters ------------------------ user=voip listen=213.24.21.2 log_stderror=yes debug=3 fifo="/var/run/ser.pipe" fifo_mode=0600 children=5 alias=tagan.ru alias=megalink.com.ru sip_warning=no # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/postgres.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/textops.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/registrar.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_url" ,"sql://pgsql@localhost/megalink") modparam("usrloc", "db_mode", 1) modparam("auth_db", "db_url" ,"sql://pgsql@localhost/megalink") modparam("auth_db", "password_column", "password") modparam("auth_db", "calculate_ha1", yes) modparam("acc", "log_level", 1) modparam("acc", "early_media", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "log_missed_flag", 3) modparam("acc", "report_cancels", 1) #modparam("acc", "db_url" ,"sql://pgsql@localhost/megalink") modparam("acc", "log_flag", 1) # ------------------------- request routing logic ------------------- route [0] { break; }

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[Serusers] SER ..... How to enable the Accounting with MYSQL
by Manoj Ghag 10 May '05

10 May '05

hi, All I have setup the SER with auth enable for MySQL ( and radius also .. at a time only one is used ) I am able to see that if user is not in database he is not able to login to SOFTPHONE .... Now I want to enable accounting also ,,, as in my case auth is working perfectly fine but accounting is not working ... I have tried all the things ,, but still I am missing somthing ...... please help me ...... as even inserting the acc module in ser.cfg file nothing is getting inserted into database .. in the form of accounting information.. At the same time where can i get the information about .. what is a information stored in each mysql table --------------------------------------------------------------------------------------------------------------------------------- my ser.cfg file is as follows ( where i have compiled ser with all the module enable) ------------------------------------------------------------------------------------------------------------------------------------ [root@test ser]# cat ser.cfg # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=5 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" # 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/acc.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" #loadmodule "/usr/local/lib/ser/modules/auth_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- # -- auth_radius params -- #modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") #modparam("usrloc", "db_mode", 0) # --domain module-- modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") #modparam("domain", "db_mode", 1) #modparam("domain", "domain_table", "domain") #modparam("domain", "domain_col", "domain") # --acc params-- modparam("acc", "log_level", 1) modparam("acc", "log_flag", 2) #modparam("acc", "log_missed_flag", 2) #modparam("acc", "log_fmt", "fimos") #modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") #modparam("acc", "db_missed_flag", 2) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 1) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- 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=='192.168.1.246') { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("192.168.1.246", "subscriber")) { www_challenge("192.168.1.246", "0"); 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(); }; } Please see teh table list [root@test ser]# mysql Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 376 to server version: 3.23.58 Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql> use ser Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Database changed mysql> show tables -> ; +-----------------------+ | Tables_in_ser | +-----------------------+ | acc | | active_sessions | | aliases | | config | | domain | | event | | grp | | location | | missed_calls | | pending | | phonebook | | preferences | | reserved | | server_monitoring | | server_monitoring_agg | | silo | | subscriber | | uri | | version | +-----------------------+ 19 rows in set (0.00 sec) --------------------------------- Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone.

3 2

[Serusers] SIP-PSTN Termination
by Matt morris 10 May '05

10 May '05

Hello List, This isn't really a technical question, but probably rather a question about SER's acceptance by the termination service provider out there. I have been looking around for a termination service provider out there that would work with SER and I could not find a decent one thus far. Most of them (at least six providers) I found seem to only work with Asterisk or UA like ATA186, and their approach with ppl using SER are "if it works, it works; if not, too bad, find another service provider, we are not going to look into why our system does not work with SER". Most of them said their gateways are SIP-compliant, then why would it be so hard for their gateways to work with SER? There are three providers listed in onsip.org and I tried one of them, but I pretty much get no response on my support questions. Do most of you build your own gateway(s) to use with your SER platform? If someone is working with reliable termination provider(s) and would like to recommend them to me, please email me. Thank you very much. _________________________________________________________________ Powerful Parental Controls Let your child discover the best the Internet has to offer. http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&DI=1034&SU=ht… Start enjoying all the benefits of MSN® Premium right now and get the first two months FREE*.

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RE: [Serusers] sip load balancing
by a c 10 May '05

10 May '05

thanks Mohammad. ac --- "info(a)beeplove.com" <info(a)beeplove.com> wrote: > It can be done by DNS SRV record. > > Mohammad > > > Original Message: > ----------------- > From: a c lra101(a)yahoo.com > Date: Tue, 10 May 2005 09:51:29 -0700 (PDT) > To: serusers(a)lists.iptel.org > Subject: [Serusers] sip load balancing > > > 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 > > -------------------------------------------------------------------- > mail2web - Check your email from the web at > http://mail2web.com/ . > > > _______________________________________________ > 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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[Serusers] UAC module problem with uac_auth()
by Mik 10 May '05

10 May '05

Hello, I am try uac_auth() client authentication to another SIP server (PSTN gateway) entity via Digest authentication. All my attempts failure with log messagess: ...... 3(98356) DEBUG:uac:uac_auth: picked reply is 0x0, code 401 3(98356) BUG:uac:uac_auth: empty reply on picked branch ...... I use example from http://www.voice-system.ro/docs/uac/ar01s06.html Give my good advice. I think, all needed information below. My SIP server: 213.24.21.2 My UAC: 213.24.21.30 PSTN gateway: 212.53.35.244 SIP messages send/recived with PSTN gateway: 213.24.21.2 -> 212.53.35.244 Session Initiation Protocol Request-Line: INVITE sip:70456@212.53.35.244:5060 SIP/2.0 Method: INVITE Resent Packet: False Message Header Record-Route: <sip:213.24.21.2;ftag=ef40b40ea4;lr> Via: SIP/2.0/UDP 213.24.21.2;branch=z9hG4bK2102.c205a46.0 Via: SIP/2.0/UDP 213.24.21.30:5060;branch=z9hG4bKef40b40ea497 From: <sip:2000@213.24.21.2>;tag=ef40b40ea4 To: <sip:70456@213.24.21.2> Call-ID: ef2efc40-3cdd-b417-810e-0002a401969c(a)213.24.21.30 CSeq: 97 INVITE Supported: timer, replaces Min-SE: 1800 Date: Mon, 19 Jul 2004 20:28:31 GMT User-Agent: AddPac SIP Gateway Contact: <sip:2000@213.24.21.30> Accept: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 249 Max-Forwards: 16 Message body Session Description Protocol 212.53.35.244 -> 213.24.21.2 Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Status-Code: 100 Resent Packet: False Message Header Via: SIP/2.0/UDP 213.24.21.2:5060;branch=z9hG4bK2102.c205a46.0 Via: SIP/2.0/UDP 213.24.21.30:5060;branch=z9hG4bKef40b40ea497 From: <sip:2000@213.24.21.2>;tag=ef40b40ea4 To: <sip:70456@213.24.21.2>;tag=3846972144 Call-ID: ef2efc40-3cdd-b417-810e-0002a401969c(a)213.24.21.30 CSeq: 97 INVITE Content-Length: 0 212.53.35.244 -> 213.24.21.2 Session Initiation Protocol Status-Line: SIP/2.0 401 Unauthorized Status-Code: 401 Resent Packet: False Message Header Via: SIP/2.0/UDP 213.24.21.2:5060;branch=z9hG4bK2102.c205a46.0 Via: SIP/2.0/UDP 213.24.21.30:5060;branch=z9hG4bKef40b40ea497 From: <sip:2000@213.24.21.2>;tag=ef40b40ea4 To: <sip:70456@213.24.21.2> Call-ID: ef2efc40-3cdd-b417-810e-0002a401969c(a)213.24.21.30 CSeq: 97 INVITE Contact: "SIP-H.323 IWF" <sip:TGK4@212.53.35.244:50666;transport=tcp> Record-Route: <sip:212.53.35.244:5060;lr> Record-Route: <sip:213.24.21.2;lr> WWW-Authenticate: Digest realm="etc.tario.ru",nonce="4280dfdc5f3d72476c787a60f587f741" Content-Length: 0 My ser.cfg: user=voip listen=213.24.21.2 log_stderror=yes debug=3 fifo="/var/run/ser.pipe" fifo_mode=0600 children=5 alias=tagan.ru alias=megalink.com.ru sip_warning=no # ------------------ module loading ---------------------------------- 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/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/uac.so" modparam("uac","credential","user:etc.tario.ru:password") route [0] { ..... skip...... if (method=="INVITE") { if (isflagset(2)) { # Our user xlog("L_INFO", "[%ci]: GO TO PSTN.\n"); t_on_failure("3"); t_relay_to_udp("212.53.35.244","5060"); break; } sl_send_reply("404", "Not found."); break; } ..... skip...... failure_route[3] { # authentication reply received? if ( t_check_status("401|407") ) { # have we already tried to authenticate? if (isflagset(7)) { t_reply("513","Authentication failed"); xlog("L_INFO", "[%ci]: Remote authentication failed\n"); break; } if (uac_auth()) { # mark that auth was performed setflag(7); # trigger again the failure route t_on_failure("3"); # repeat the request with auth response this time append_branch(); t_relay(); break; } t_reply("514","Unknow authentication peer"); xlog("L_INFO", "[%ci]: Unknown authentication peer.\n"); break; } }

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RE: [Serusers] sip load balancing
by info＠beeplove.com 10 May '05

10 May '05

It can be done by DNS SRV record. Mohammad Original Message: ----------------- From: a c lra101(a)yahoo.com Date: Tue, 10 May 2005 09:51:29 -0700 (PDT) To: serusers(a)iptel.org Subject: [Serusers] sip load balancing 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)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] what does it mean
by Vincent Verdot 10 May '05

10 May '05

I sent you a private message in french... I think it's better if evryone can understand advices in this mailing-list (if my english is not too bad ! ;-) ) so I prefer posting in english.. ----- Original Message ----- From: "michael p" <mikep3000(a)hotmail.com> To: <vincent.verdot(a)int-evry.fr> Sent: Tuesday, May 10, 2005 5:21 PM Subject: Re: [serusers] what does it mean > salut vincent > > si tu parles francais tu peux peut etre m'expliquer en francais ;) > > les credentials sont les comptes URI qui se rajoute ds les via a chaque > passage d'un proxy? > > j'ai pas trop compris > > Merci > > M. > > >From: "Vincent Verdot" <vincent.verdot(a)int-evry.fr> > >To: <serusers(a)lists.iptel.org> > >Subject: Re: [serusers] what does it mean > >Date: Tue, 10 May 2005 17:16:40 +0200 > > > >For example, if a user's SIP message must be forwarded by two proxies (from > >two distincts realms) and must be authenticated in the both.. then the sip > >message should contain 2 credentials : 1 for user-proxy1 authentication and > >an another one for proxy1-proxy2 authentication. > > > >Credential =~ hash(shared secret + nonce + ...) > > > >example (you can have many lines like this...): > >**************** > >Authorization: Digest > >username="Vincent",realm="int-evry.fr",nonce="xxxxxxxxxxxx",uri="sip:int-ev r > >y.fr",response="xxxxxxxxxxxxxxxx",algorithm=MD5 > >**************** > > > >----- Original Message ----- > >From: "michael p" <mikep3000(a)hotmail.com> > >To: <serusers(a)lists.iptel.org> > >Sent: Tuesday, May 10, 2005 4:42 PM > >Subject: [serusers] what does it mean > > > > > > > hi excuse me but it's a little difficult for me as a french this doc in > > > english about ser > > > > > > what does mean the word credentials in "A SIP message may contain > >several > > > credentials" ? > > > > > > can you give me an example please? > > > > > > thanks > > > > > > M. > > > > > > > > > _______________________________________________ > > > 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 > >

1 0

[Serusers] check user if authenticated
by Edgardo O. Gonzales II 10 May '05

10 May '05

Hi! How will I check if user is authenticated or not before I permit him to make a call. thanks, ed

2 2

[serusers] what does it mean
by michael p 10 May '05

10 May '05

hi excuse me but it's a little difficult for me as a french this doc in english about ser what does mean the word credentials in "A SIP message may contain several credentials" ? can you give me an example please? thanks M.

2 1

RE: [Serusers] check user if authenticated
by Matt Schulte 10 May '05

10 May '05

Something like this before your final route should work.. if (!proxy_authorize("blah.realm.blah", "subscriber")) { xlog("L_INFO", "Auth failed before %rm from <%fu>. sending challenge\n"); proxy_challenge("blah.realm.blah", "1"); break; }; if (!method == "CANCEL" && !method == "ACK" ) { if (!is_user_in("Credentials", "int")) { xlog("L_INFO", "User dialout but no privs or no ACL, dropping %fu URI %ru\n"); sl_send_reply("403", "Forbidden go away"); break; } } -----Original Message----- From: Edgardo O. Gonzales II [mailto:edgardo.g@pacific.net.ph] Sent: Tuesday, May 10, 2005 8:35 AM To: serusers(a)lists.iptel.org Subject: [Serusers] check user if authenticated Hi! How will I check if user is authenticated or not before I permit him to make a call. thanks, ed _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] sipath
by Iqbal 10 May '05

10 May '05

Hi has anyone used this to flash a router and get ser on a router like linksys Iqbal

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Re: [Serusers] Re: [Serdev] Patched free-TLS implementation
by Alex Mack 10 May '05

10 May '05

Hi! I've managed to find out what's bothering the snom 190: It's the missing TLS Session ID when Session Chaching is turned off. The session ID is given to the client for being able to request that session later on from the server. But SSL_CTX_set_session_cache_mode( _ctx, SSL_SESS_CACHE_OFF ); turned off Session Caching, so no Session ID is generated and transfered and that seems to confuse the snom phones. So they answer the SERVER HELLO immediately with an ALERT and break up the connection. It seems like snom190 need to have session caching turned on... The snom's firmware is: snom190-SIP 3.60b 6249 snom's softphone (emulating a snom 360 with firmware "snomSoft-SIP 3.57q") seems to have the same problem. BTW, what was the problem with session caching in the first place? Performance hit? Alex Mack Benny Ben-Ami schrieb: >Try checking the expiration date of the certificate and the time & date of >the phone. >I once had a phone where the time was not set and he rejected the server >certificate due to date expiration. > >-----Original Message----- >From: serdev-bounces(a)iptel.org [mailto:serdev-bounces@lists.iptel.org] On Behalf >Of Alex Mack >Sent: Wednesday, May 04, 2005 7:29 PM >To: Cesc Santasusana >Cc: serdev(a)lists.iptel.org; serusers(a)lists.iptel.org >Subject: Re: [Serusers] Re: [Serdev] Patched free-TLS implementation > >Hi Cesc! > >I tried your previuos suggestions on changing cleint certificate check. >I'm now running SSL_VERIFY_NONE. MS Messenger works again. But the snom >still fails. >Ethereal tells me the server doesn't ask for a client certificate any more. >Ethereal also shows that "Cipher Suite: TLS_RSA_WITH_RC4_128_SHA" >was selected. So far so good, but the snom still rejects the Server Hello >with an Alert. > >What else have you changed? The original version worked with the snoms. > >BTW patch throws some warnings: > ># patch -i patch.core.cfg.files.diff >patching file cfg.y >Hunk #1 FAILED at 1. >1 out of 6 hunks FAILED -- saving rejects to file cfg.y.rej patching file >cfg.lex Hunk #1 FAILED at 1. >1 out of 5 hunks FAILED -- saving rejects to file cfg.lex.rej > >cfg.y.rej reads: >*************** >*** 1,5 **** > /* >- * $Id: cfg.y,v 1.2 2005/01/06 14:35:10 sam Exp $ > * > * cfg grammar > * >--- 1,5 ---- > /* >+ * $Id: cfg.y,v 1.4 2005/05/03 08:16:35 cesc Exp $ > * > * cfg grammar > * > >cfg.lex.rej reads: >*************** >*** 1,5 **** > /* >- * $Id: cfg.lex,v 1.2 2005/01/06 14:35:10 sam Exp $ > * > * scanner for cfg files > * >--- 1,5 ---- > /* >+ * $Id: cfg.lex,v 1.3 2005/04/11 08:18:31 cesc Exp $ > * > * scanner for cfg files > * > >Seems to be a minor mismatch, it compiles well anyway. > >Alex Mack > >Cesc Santasusana schrieb: > > > >>Hi, >> >>Yeah ... I send it with my default config which is using client and server >> >> >authentication. > > >>I use this settings with minisip client (supports client side certs) and >> >> >for tls between ser proxies. It works perfect. > > >>To turn client authentication off, check: >>tls/tls_init.c file >>init_ssl_ctx_behavior function >> >>the line >>SSL_CTX_set_verify( _ctx, SSL_VERIFY_PEER | >>SSL_VERIFY_FAIL_IF_NO_PEER_CERT, 0); >> >>means that the server will request a certificate from the client and if it >> >> >doesn't get one, it will fail. > > >>Try changing it with: >>SSL_CTX_set_verify( _ctx, SSL_VERIFY_PEER | SSL_VERIFY_CLIENT_ONCE, 0); >>this way the server will request a cert, the client will not provide, but >> >> >on the following renegotiation, the server (ser) will not ask for a cert. > > >>And if you want to turn verification off ... completely ... >>SSL_CTX_set_verify( _ctx, SSL_VERIFY_NONE, 0); this will also work for >>you if only using tls for ser2phone ... it will not work if you want tls >> >> >between proxies ... as the ser client will accept ANY certificate from the >ser server. > > >>And then, from the prompt: >> >> >> >> >>>make TLS=1 all && make TLS=1 install ;) >>> >>> >>> >>> >>This whole verification thing needs to be improved and probably the >> >> >parameters should be changeable directly from the config file. This and many >other parameters should be exchangeable without the need to recompile. > > >>Any volunteer for a ser-tls.README? :D >> >>Can you provide me with some extra info from the snom phones and the >> >> >messenger? Do it offline, so you can send me some ethereal captures and ser >logs ... > > >>Regards, >> >>Cesc >> >> >> >> >> >> >>>>>Alex Mack <amack(a)fhm.edu> 05/04/05 02:01PM >>> >>>>> >>>>> >>>>> >>>>> >>Hi Cesc! >> >>I compiled in your patch. >> >>Now I'm facing a new problem: SER wants a client certificate from the >>UA. Snom phones immediately reply with an ALERT and break up connection >>upon the certificate requests. MS Messenger on the other hand sends at >>least a reply - without certificate - and SER rejects the Client Hello >>because of the missing client certificate: >> >>tls_accept: Error in SSL: >>tls_error: error:140890C7:SSL routines:SSL3_GET_CLIENT_CERTIFICATE:peer >>did not return a certificate >> >>Could you please provide some more documentation about the new TLS >>options you added? It seems you've implemented support for client >>certificates for a two-way certificate authorization - which would be a >>good thing if supported by the UAs, which don't right now. So how can I >>turn it off again and get back to server side certification? >> >>Alex Mack >> >>Cesc Santasusana schrieb: >> >> >> >> >> >>>Hi everybody, >>> >>>The last i sent is a replacement as a whole for the original code sent by >>> >>> >P. Griffiths. Sorry i forgot to mention that. > > >>>The patches for cfg.y and cfg.lex are both in the same file >>>(patch.core.cfg..files.diff) within the zip. I was lazy :) I resent it as >>> >>> >a whole, and not as a diff, because i indented all the code with tabs, >instead of spaces (so a diff would be bigger than just sending all the >files). > > >>>As for the CVS thing ... i agree with Juha. Either gets into the >>> >>> >"official" cvs or we do something about it. The code i think is rather >stable as it is (i only tested on my debian linux box, soon i will try on an >ARM linux and i will report back on that too). For me, as long as it gets >into a CVS, i don't care if it is mantained against HEAD or 0.9.0 (i use >0.9.0 .... so all my patches are against it). > > >>>On a more philosophical level, i understand the "quietness" on iptel's >>> >>> >side ... they have their own version, and make money on it. But the thing is >that this free version is here to stay ... it is the "problem" of >opensource. > > >>>Another option would be for them to release their proprietary >>> >>> >implementation if they feel that it is a better, more tested one. > > >>>In any case, i think that this whole thing needs to be decided fast. >>> >>>Regards! >>> >>>Cesc >>> >>> >>> >>> >>> >>> >>> >>> >>>>>>Alex Mack <amack(a)fhm.edu> 05/03/05 01:26PM >>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>Hi Cesc! >>> >>>Nice to have those fixes in a package. >>> >>>Is your cfg.y-patch to be applied *after* cfg.y.patch was applied or >>>*instead* of cfg.y.patch? >>> >>>Or is your version a patched one which replaces the original >>>implementation as a whole? In that case where's cfg.lex.patch? >>> >>>Alex Mack >>> >>>Cesc Santasusana schrieb: >>> >>> >>> >>> >>> >>> >>>>Hi, >>>> >>>>I really hate to be so pushy, but i dont understand how such an important >>>> >>>> >piece of code as TLS is not moving on into CVS ... or anywhere else by this >matter. I will keep sending patches till i get tired (soon). > > >>>>Anyway ... i thought someone may be interested in a compilation fix for >>>> >>>> >cfg.y introduced with the tls_domains (it would not compile if the cfg.y >file had been patched but the tls-core files were not there); a bug fix for >the session caching (fixed by turning session caching and resumption off); >and an extension (the ability to choose the list of allowed ciphers from the >config file). Oh, and all the files have been tabbed, instead of spaced (for >indentation). > > >>>>Enjoy! >>>> >>>>Cesc >>>> >>>> >>>> >>>> >>>> >>>> >>Unclassified >> >> >> >> >>>>--------------------------------------------------------------------- >>>>--- >>>> >>>>_______________________________________________ >>>>Serdev mailing list >>>>serdev(a)lists.iptel.org >>>>http://lists.iptel.org/mailman/listinfo/serdev >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>_______________________________________________ >>>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 >>> >>> >>> >>> >>> >>> >>> >> >> >> >> >> > >_______________________________________________ >Serdev mailing list >serdev(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serdev > > > > >

1 1

[Serusers] Problem compiling postgres module
by Gorka Guridi 10 May '05

10 May '05

Hi When I tried to compile this module, i obtain this error messages. In file con_postgres.h there is this include: #include "libpq-fe.h" I tried to find this file but it doesn't appear anywhere. What is the utility of this file? Thanks a lot. Best regards. Gorka. make[1]: Entering directory `/root/ser/ser-0.8.14/modules/postgres' ../../Makefile.rules:77: aug_alloc.d: No such file or directory ../../Makefile.rules:77: aug_util.d: No such file or directory ../../Makefile.rules:77: db_con.d: No such file or directory ../../Makefile.rules:77: db_mod.d: No such file or directory ../../Makefile.rules:77: db_res.d: No such file or directory ../../Makefile.rules:77: db_utils.d: No such file or directory ../../Makefile.rules:77: db_val.d: No such file or directory ../../Makefile.rules:77: dbase.d: No such file or directory In file included from dbase.c:49: con_postgres.h:43:22: libpq-fe.h: No such file or directory In file included from db_val.c:49: con_postgres.h:43:22: libpq-fe.h: No such file or directory In file included from db_res.c:44: con_postgres.h:43:22: libpq-fe.h: No such file or directory In file included from db_con.c:44: con_postgres.h:43:22: libpq-fe.h: No such file or directory make[1]: Leaving directory `/root/ser/ser-0.8.14/modules/postgres' make[1]: Entering directory `/root/ser/ser-0.8.14/modules/postgres' ../../Makefile.rules:77: db_con.d: No such file or directory ../../Makefile.rules:77: db_res.d: No such file or directory ../../Makefile.rules:77: db_val.d: No such file or directory ../../Makefile.rules:77: dbase.d: No such file or directory In file included from dbase.c:49: con_postgres.h:43:22: libpq-fe.h: No such file or directory In file included from db_val.c:49: con_postgres.h:43:22: libpq-fe.h: No such file or directory In file included from db_res.c:44: con_postgres.h:43:22: libpq-fe.h: No such file or directory In file included from db_con.c:44: con_postgres.h:43:22: libpq-fe.h: No such file or directory gcc-3.3 -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='""' -DVERSION='"0.8.14"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc-3.3 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_TCP -DDISABLE_NAGLE -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -I/usr/local/pgsql/include -I/usr/include/postgresql -c aug_alloc.c -o aug_alloc.o aug_alloc.c: In function `mem_bad': aug_alloc.c:118: warning: no return statement in function returning non-void aug_alloc.c: In function `mem_nomem': aug_alloc.c:161: warning: long unsigned int format, size_t arg (arg 4) aug_alloc.c: In function `mem_alloc': aug_alloc.c:228: warning: suggest parentheses around assignment used as truth value aug_alloc.c: In function `aug_set_nomem_func': etc.

1 0

[Serusers] t_write_unix
by TCCHAN 10 May '05

10 May '05

Hi ALL: Can anyone help! I try to call "t_write_unix" in script and found that it isn't implemented in tm modules. But documentation from Sems clearly show that tm is the module I should load. I am using Ser0.814. Do I need a newer version? Thanks. Regards, TC Chan

2 1

[Serusers] [ser-0.8.14] ERROR: Could NOT resolve hostname
by Star Bravo 10 May '05

10 May '05

Hi All, I am running ser-0.8.14 on a Public IP. Sometimes I see following errors while looking at /var/log/messages ser[20575]: ERROR: mk_proxy: could not resolve hostname: "serdomain.com" ser[20575]: ERROR: uri2proxy: bad host name in URI <sip:serdomain.com> ser[20575]: ERROR: t_forward_nonack: failure to add branches where, "serdomain.com" is the domain name of the machine running SER and is mapped to a Public IP. Any help in this regard will be much appreciated. Thks & Rgds, - Bravo

2 1

[Serusers] t_relay won't give up
by Brett's Asterisk Account 10 May '05

10 May '05

Hey All, I'm having a problem. I'm attempting a: if (!t_relay()) { route(2); break } When the destination ignores the request (perhaps the domain portion of the RURI points to a server that is down?), I see the INVITE processed like 10 times. Here is the trace.. 11.066729 192.168.20.10 -> 192.168.50.100 SIP/SDP Request: INVITE sip:2015551212;rn=2015550099;npdi=yes@192.168.50.100;dtg=HSTNSIPORIG02;user=phone, with session description 4(24793) ******** INCOMMING CALL FROM TRUSTED IP ADDRESS 4(24793) INFO Checking ANI from Trusted IP 4(24793) INFO: ANI Verification successful for trusted IP 4(24793) ****** INCOMING CALL PASSED SECURITY CHECKS 4(24793) INFO: Begining attempt 1 DID route for trusted ip customer 4(24793) ****** FOUND DID TABLE ROUTE 11.318093 192.168.50.100 -> 192.168.20.10 SIP Status: 100 trying -- your call is important to us 11.318223 192.168.50.100 -> 10.1.2.3 SIP/SDP Request: INVITE sip:2015551212@10.1.2.3, with session description 11.425865 192.168.50.100 -> 10.1.2.3 SIP/SDP Request: INVITE sip:2015551212@10.1.2.3, with session description 13.429423 192.168.50.100 -> 10.1.2.3 SIP/SDP Request: INVITE sip:2015551212@10.1.2.3, with session description 17.436544 192.168.50.100 -> 10.1.2.3 SIP/SDP Request: INVITE sip:2015551212@10.1.2.3, with session description 21.443657 192.168.50.100 -> 10.1.2.3 SIP/SDP Request: INVITE sip:2015551212@10.1.2.3, with session description 23.328599 192.168.20.10 -> 192.168.50.100 SIP Request: CANCEL sip:2015551212;rn=2015550099;npdi=yes@192.168.50.100;dtg=UTEXHSTNSIPORIG02;user=phone 3(24790) ******** INCOMMING CALL FROM TRUSTED IP ADDRESS 3(24790) INFO Checking ANI from Trusted IP 3(24790) INFO: ANI Verification successful for trusted IP 3(24790) ****** INCOMING CALL PASSED SECURITY CHECKS 3(24790) INFO: Begining attempt 1 DID route for trusted ip customer 3(24790) ****** FOUND DID TABLE ROUTE 23.581023 192.168.50.100 -> 192.168.20.10 SIP Status: 200 ok -- no more pending branches 25.451776 192.168.50.100 -> 10.1.2.3 SIP/SDP Request: INVITE sip:2015551212@10.1.2.3, with session description 29.458898 192.168.50.100 -> 10.1.2.3 SIP/SDP Request: INVITE sip:2015551212@10.1.2.3, with session description 33.466012 192.168.50.100 -> 10.1.2.3 SIP/SDP Request: INVITE sip:2015551212@10.1.2.3, with session description 37.473130 192.168.50.100 -> 10.1.2.3 SIP/SDP Request: INVITE sip:2015551212@10.1.2.3, with session description 40.479489 192.168.50.100 -> 192.168.20.10 SIP Status: 408 Request Timeout 40.487743 192.168.20.10 -> 192.168.50.100 SIP Request: ACK sip:2015551212;rn=2015550099;npdi=yes@192.168.50.100;dtg=HSTNSIPORIG02;user=phone 10.1.2.3 is not a real node.. but it really does attempt to send a call there.. I'd like for it to attempt, die, and process the next route. I've considered using t_on_failure, but it's not really what I want.. I want this failure to occur for any status >500 (ie: 5xx and 6xx) AND for dead SIP (ie, no SIP reply at all). t_on_failure will catch too much. Also, it takes too long to timeout. I could have sworn I had this working with t_relay the other day. Am I doing something wrong? What is making it retry over and over? Thanks, Brett

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[Serusers] writing modules
by Greg Martin 10 May '05

10 May '05

Is there anything more then serdev and the source code to aid with writing modules? Is this the correct list for programming questions relating to modules? Greg Martin Technology & Operations TELUS Communication Inc. office: 780-493-2786 cell: 780-718-4139

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[Serusers] RV: problems with digest
by Lucas Aimaretto 10 May '05

10 May '05

> Hi all, > > I'm having trouble at authentication using radius and digest. > Look at radius output. The rare thing is that some phones get > registered nicely, but others no. The ones who get registered > are X-Lite softphones and grandstream. The ones that not, are > the ATAs from voip solutions, MTA-V102. Any help would be > appreciated. The user is 1991106 and has NO PASSWORD assigned > ... ( but all of the users have NO PASSWORD ). > > rad_recv: Access-Request packet from host IP_SER:33483, > id=196, length=269 > User-Name = "1991106@IP_SER" > Digest-Attributes = 0x0a0931393931313036 > Digest-Attributes = 0x01103230382e3232312e3136392e3838 > Digest-Attributes = > 0x022a34323766656365613663303066636665343337623439613936343664 > 303666373363396635353639 > Digest-Attributes = 0x04147369703a3230382e3232312e3136392e3838 > Digest-Attributes = 0x030a5245474953544552 > Digest-Response = "9b256af89daa817caf568f682e1d15a6" > Service-Type = IAPP-Register > X-Ascend-PW-Lifetime = 0x31393931313036 > Cisco-AVPair = > "call-id=efbfcb25db042b56d47ddbe74e640d8f(a)10.0.0.5" > NAS-IP-Address = IP_SER > NAS-Port = 5060 > Processing the authorize section of radiusd.conf > modcall: entering group authorize for request 213 > modcall[authorize]: module "preprocess" returns ok for request 213 > modcall[authorize]: module "attr_filter" returns noop for > request 213 > modcall[authorize]: module "chap" returns noop for request 213 > rlm_digest: Converting Digest-Attributes to something sane... > Digest-User-Name = "1991106" > Digest-Realm = "IP_SER" > Digest-Nonce = "427fecea6c00fcfe437b49a9646d06f73c9f5569" > Digest-URI = "sip:IP_SER" > Digest-Method = "REGISTER" > rlm_digest: Adding Auth-Type = DIGEST > modcall[authorize]: module "digest" returns ok for request 213 > rlm_realm: Looking up realm "IP_SER" for User-Name = > "1991106@IP_SER" > rlm_realm: Found realm "IP_SER" > rlm_realm: Adding Stripped-User-Name = "1991106" > rlm_realm: Proxying request from user 1991106 to realm IP_SER > rlm_realm: Adding Realm = "IP_SER" > rlm_realm: Authentication realm is LOCAL. > modcall[authorize]: module "suffix" returns noop for request 213 > radius_xlat: '1991106' > rlm_sql (sql): sql_set_user escaped user --> '1991106' > radius_xlat: 'rad_authorize_check_query '1991106'' > rlm_sql (sql): Reserving sql socket id: 1 > radius_xlat: '' > radius_xlat: 'rad_authorize_reply_query '1991106',''' > radius_xlat: '' > rlm_sql (sql): Released sql socket id: 1 > modcall[authorize]: module "sql" returns ok for request 213 > modcall: group authorize returns ok for request 213 > rad_check_password: Found Auth-Type DIGEST > auth: type "digest" > Processing the authenticate section of radiusd.conf > modcall: entering group authenticate for request 213 > A1 = 1991106:IP_SER: > A2 = REGISTER:sip:IP_SER > KD = > b3b6936f2a09f4749902ff9f6e0f1b71:427fecea6c00fcfe437b49a9646d0 > 6f73c9f5569:1111962db7ab8b0547fc8fbaa6408dd6 > rlm_digest: FAILED authentication > modcall[authenticate]: module "digest" returns reject for > request 213 > modcall: group authenticate returns reject for request 213 > auth: Failed to validate the user. > Sending Access-Reject of id 196 to IP_SER:33483 > > ... any ideas ?? > > Look at this NGREP's ... > > U IP_UA:60975 -> IP_SER:5060 > REGISTER sip:IP_SER SIP/2.0. > Via: SIP/2.0/UDP 10.0.0.5:5070;branch=z9hG4bK2952116395. > From: <sip:1991106@IP_SER>;tag=2375800474. > To: <sip:1991106@IP_SER>. > Call-ID: efbfcb25db042b56d47ddbe74e640d8f(a)10.0.0.5. > CSeq: 15158 REGISTER. > Contact: sip:1991106@10.0.0.5:5070. > Expires: 120. > Max-Forwards: 70. > User-Agent: SIP-ICSG102-1.372-icablesystem/v2.0_enabled. > Content-Length: 0. > > U IP_SER:5060 -> IP_UA:60975 > SIP/2.0 401 Unauthorized. > Via: SIP/2.0/UDP > 10.0.0.5:5070;branch=z9hG4bK2952116395;rport=60975;received=64 > .32.92.159. > From: <sip:1991106@IP_SER>;tag=2375800474. > To: <sip:1991106@IP_SER>;tag=6f0d146d94c4cb042663ff3cf87e2e72.527a. > Call-ID: efbfcb25db042b56d47ddbe74e640d8f(a)10.0.0.5. > CSeq: 15158 REGISTER. > WWW-Authenticate: Digest realm="IP_SER", > nonce="427feab914e565fceccccccf1852a2b0ae3b69cb". > Content-Length: 0. > Warning: 392 IP_SER:5060 "Noisy feedback tells: pid=5366 > req_src_ip=IP_UA req_src_port=60975 in_uri=sip:IP_SER > out_uri=sip:IP_SER via_cnt==1". > > U IP_UA:60975 -> IP_SER:5060 > REGISTER sip:IP_SER SIP/2.0. > Via: SIP/2.0/UDP 10.0.0.5:5070;branch=z9hG4bK2608934381. > From: <sip:1991106@IP_SER>;tag=1079893788. > To: <sip:1991106@IP_SER>. > Call-ID: efbfcb25db042b56d47ddbe74e640d8f(a)10.0.0.5. > CSeq: 15159 REGISTER. > Contact: sip:1991106@10.0.0.5:5070. > Expires: 120. > Authorization: Digest username="1991106", realm="IP_SER", > nonce="427feab914e565fceccccccf1852a2b0ae3b69cb", > uri="sip:IP_SER", response="c7dc44af5d16f48c410813a7f4dc98f2". > Max-Forwards: 70. > User-Agent: SIP-ICSG102-1.372-icablesystem/v2.0_enabled. > Content-Length: 0. > > U IP_SER:5060 -> IP_UA:60975 > SIP/2.0 401 Unauthorized. > Via: SIP/2.0/UDP > 10.0.0.5:5070;branch=z9hG4bK2608934381;rport=60975;received=64 > .32.92.159. > From: <sip:1991106@IP_SER>;tag=1079893788. > To: <sip:1991106@IP_SER>;tag=6f0d146d94c4cb042663ff3cf87e2e72.16e1. > Call-ID: efbfcb25db042b56d47ddbe74e640d8f(a)10.0.0.5. > CSeq: 15159 REGISTER. > WWW-Authenticate: Digest realm="IP_SER", > nonce="427feab914e565fceccccccf1852a2b0ae3b69cb". > Content-Length: 0. > Warning: 392 IP_SER:5060 "Noisy feedback tells: pid=5366 > req_src_ip=IP_UA req_src_port=60975 in_uri=sip:IP_SER > out_uri=sip:IP_SER via_cnt==1". > > So, you can see that the UA wants to register. Ser tells him > to send nonce and digest data, but, once the UA resends the > info, it gets an 401 Unauthorized message. I do not know why > .... :( because it works with other phones ( xlite, grandstream ) ... Hi there again ... I've made another test and found out that if I assign a password to the UserAgent, it works perfectly well ... :( But only with this UserAgent only ( VoipSolutions' MTA-V102 ) ... With Xlite or Grandstream it does not matter if the UserAgent has a password or not. Any ideas why is the password bothering ... ??? Best Regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.5 - Release Date: 04/05/2005

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Re: [Serusers] pa module
by harry gaillac 09 May '05

09 May '05

First I would like to understand how to pa module works with database (pa.sql) because of i don't find docs. > The only thing currently shared between PA and the > XCAP server is > watcherinfo. PA and XCAP exchange information > through a table in the > database. The XCAP server sends a FIFO command or a > SIP message to PA > to tell it to refresh the watcherinfo for a user. I need help Harry --- Jamey Hicks <jamey.hicks(a)hp.com> a écrit: > harry gaillac wrote: > > >hello, > > > >Thanks for your reply. > > > >I try to understand drafts from : > >http://www.ietf.org/html.charters/simple-charter.html > > > >I looked at > >http://pals.internet2.edu/downloads/xcap/xcap-0.1.tar.gz > > > >however I don't find documentation at iptel.org in > >order to answer to my problem: > >Configuring Ser to handle ressource list with URI > > > > > SER does not handle resource lists. However, the > XCAP server will allow > a UA to GET/PUT resource lists. Do you have a > suggestion on how to form > the URIs for resource lists? > > The only thing currently shared between PA and the > XCAP server is > watcherinfo. PA and XCAP exchange information > through a table in the > database. The XCAP server sends a FIFO command or a > SIP message to PA > to tell it to refresh the watcherinfo for a user. I > imagine that a > similar scheme could be used for resource lists. > > It might actually be easier, especially in the case > of watcher lists, > for PA to just fetch the xml and parse it rather > than to force things > into a database. This would make it easier for the > XML schema to be > extended. > > What do you think? > > Jamey > > __________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/

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