- sr-users - kamailio.org

[Users] Polycom soundpoint ip 500
by Cesc 12 Oct '05

12 Oct '05

Hi, We have a polycom sound point ip 500, sip app ver 1.5 (from voip-info.org<http://voip-info.org>). It turns out that it won't display MESSAGE ... but it does receive them and it replies with a 200 OK. But no message on the screen or anywhere within the phone. The configuration has not been changed, it is pretty much the standard stuff that comes from voip-info.org <http://voip-info.org>. Anyone has ever came across the same problem? Regards, Cesc PS - Getting the sip software from polycom resellers is a mess ... well, i would say almost impossible. Next time we'll go for a costumer frendlier phone manufacturer i guess.

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Re: [Serusers] Have some INVITE problems
by Samuel Osorio Calvo 12 Oct '05

12 Oct '05

I'm sorry but I have never played with these models so I can not help you much..... In order to extend the INVITE retransmission, there must be somewhere in the web interface of these phones (if they have) something like T1 and T2, these values determine the retransmission period. I guess you should check the Internet for information about your phones. Hope somebody else with more experience can help you, Samuel. Unclassified. >>> Alex <alexandergav(a)gmail.com> 10/11/05 05:45PM >>> I am using cisco 7960 and grandstream. 286,486 On 10/11/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > > Which UA are you using??? It looks like all your issues comes from the > implementation you are using as end-point and has nothing to do with SER. > > Samuel. > > > Unclassified. > >>> Alex <alexandergav(a)gmail.com> 10/11/05 11:06AM >>> > Hi all > I would like to explain some problem. > When my UAC initiates INVITE the server response with 100 trying > After receiving 100 trying the UAC sending again INVITE. ( HERE is the > problem). > RFC - 3261 ("After receiving a 1xx response, any retransmissions cease > altogether "); > Any help on that will be appreciated. > In addition if there any way to extend the timeout between the INVITES. > Thanks > >

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[Users] CDRTool call rating "Cannot instantiate non-existent class"
by Cameron Beattie 12 Oct '05

12 Oct '05

I have been testing CDRTool and get the following message in the log files when browsing to www.example.com/scripts/ratingEngine.php: Cannot instantiate non-existent class: ratingtables in /var/www/CDRTool/scripts/ratingEngine.php Is this because the non commercial version of CDRTool does not provide rating or is it something I have done wrong in the setup? Any advice would be appreciated. Regards Cameron

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[Serusers] rewriting domain
by Maciej Świniarski 12 Oct '05

12 Oct '05

i have stupid hardphone, proxy, registrer server is always domain :/ I have configured DNS and ser to use some domain.com as sip domain, but proxy have sip.mydomain.com addres. Registering is no problem because i can tell what domain ser should search but problem is for examplke with aliases i don't want write every time `serctl alias add user(a)domain.com` `serctl alias add user(a)sip.domain.com` if (uri=~".*sip\.my\.domain\.com") { rewritehost("my.somain.com"; } don't work becaus it rewrites requested uri i hope you get my point. Any one know how to solve this problem?

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[Serusers] acc+nat
by Maciej Świniarski 12 Oct '05

12 Oct '05

hi i have little problem with acc and nat becaus mysql records created with acc module contains ip behind nat. And yes, i use fix_nated_contact befor setting flag for acc module. Any one have some experience with this issue?

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Re: [Users] Constructing sip uri from Dialed number and userloc
by Daniel-Constantin Mierla 12 Oct '05

12 Oct '05

On 10/12/05 02:54, Gene Willingham wrote: > > > > Not sure if I have all of the terminology right, but I would like to > have Asterisk register with openser. Then when a call comes in I > would like to construct the sip uri using the Dialed number and the > user location. The final uri should be: > > > > Sip:240744XXXX@25.25.25.25. Where the IP address is the user location > of the Asterisk box. > You can play with avpops to get this. You have there avp_write() to backup the original dialed number, or avp_subst() to extract parts of the original dialed number. Then use avp_printf() to concatenate the values, after usrloc lookup (see http://www.voice-system.ro/docs/avpops/) and avp_pushto() to rewrite the r-uri. Cheers, Daniel > > > Any suggestions. > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[Serusers] BYE method not handled properly
by Sam Lee 12 Oct '05

12 Oct '05

I would like to know why does my BYE method are always replied with a 'Call Leg/Transaction does not exist' . How do they compare whether the transaction in the BYE method exist or not ? ( tag? ftag ? ) Are there any thing in the config that might cause this kind of problem ? Just want to highlight that all the calls are made in a good condition, everything except when the call is ending. I have checked that the phones have not received a prior BYE. Any other idea what is wrong ? Here's a more detailed situation :- Caller (PSTN) --> Voice Gateway --> OPENSER --> Callee (UA) When Callee (UA) tried to end the call , OPENSER will forward a copy of the BYE to Voice Gateway to inform him of the BYE. The Gateway , somehow , replied with a 'Call Leg/Transaction Does Not Exist' . The strange thing is, the Caller (PSTN) was somehow informed of the BYE method and terminate the session . Anyone has any idea how to handle these errors ? I will be glad to provide a ngrep for more reference. Please let me know if you dont understand. Regards, Sam

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RE: [Users] BYE method accompanied by error
by Sam Lee 12 Oct '05

12 Oct '05

I have checked that the phones have not received a prior BYE. Any other idea what is wrong ? Here's a more detailed situation :- Caller (PSTN) --> Voice Gateway --> OPENSER --> Callee (UA) When Callee (UA) tried to end the call , OPENSER will forward a copy of the BYE to Voice Gateway to inform him of the BYE. The Gateway , somehow , replied with a 'Call Leg/Transaction Does Not Exist' . The strange thing is, the Caller (PSTN) was somehow informed of the BYE method and terminate the session . Anyone has any idea how to handle these errors ? I will be glad to provide a ngrep for more reference. Regards, Sam -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Iqbal Sent: Tuesday, October 11, 2005 7:35 PM To: Sam Lee Cc: users(a)openser.org Subject: Re: [Users] BYE method accompanied by error Can you check to see if you have already received a BYE for that call, some phones I had were sending there own Bye's after the GW had Iqbal Sam Lee wrote: > Hi all, > > I would like to know why does my BYE method are always replied with a > 'Call Leg/Transaction does not exist' . How do they compare whether > the transaction in the BYE method exist or not ? ( tag? ftag ? ) Are > there any thing in the config that might cause this kind of problem ? > Just want to highlight that all the calls are made in a good > condition, everything except when the call is ending. > > Please let me know if you dont understand. > > Regards, > Sam > >----------------------------------------------------------------------- >- > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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Re: [Users] Constructing sip uri from Dialed number and userloc
by Klaus Darilion 12 Oct '05

12 Oct '05

Check out the admin guide! http://www.iptel.org/ser/doc/seruser/seruser.html#URIREWRITINGEXAMPLES You want to use rewritehostport(). You could also use AVPs and avpops module. klaus Gene Willingham wrote: > > > Not sure if I have all of the terminology right, but I would like to > have Asterisk register with openser. Then when a call comes in I would > like to construct the sip uri using the Dialed number and the user > location. The final uri should be: > > > > Sip:240744XXXX@25.25.25.25. Where the IP address is the user location > of the Asterisk box. > > > > Any suggestions. > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users

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[Serusers] Forward to PSTN gateway question
by Alejandro G 12 Oct '05

12 Oct '05

Hi, I have the following configuration: 192.168.0.101 192.168.0.1 Public IP <---> Asterisk <------------------------------> SER <---> Public IP <-----> SIP Phone as PSTN GW I use mediaproxy for dealing with NAT, everything works fine. We have 2 internet connections: one over Asterisk with low bandwith, and the other over SER with more bandwidth. The SIP phone is Sipura 2100. The routing block I use in INVITE is something liske this: route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("192.168.0.1"); # GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); append_hf("P-hint: GATEWAY\r\n"); # ----------------------------------------------------------------- # RTP Proxy Enabler # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { # flags set by client_nat_test() use_media_proxy(); }; route(1); } When the user with the SIP phone terminates in PSTN via Asterisk, SER forwards to Asterisk but Asterisk then * contacts rtp streams directly to SIP phone bypassing SER and using the low bandwidth connection. My knowledge is poor about SER and SIP issues but I believe that * tries to contact to the UA specified in the Contact header. Is this correct? Is anyway SIP phone could terminate in PSTN connecting using SER and private LAN to connect to asterisk instead of using asterisk's public IP? Sorry, I'm not sure that my questions is clear enough. Thanks. Alejandro Ghergherian

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Re: [Serusers] Please help on terminating active sessions!!!
by Ryan Pagquil 12 Oct '05

12 Oct '05

Hi Samuel, I'm getting these (To: From: Call-ID) informations from the database, since I'm accounting those messages in mysql database, will it help? Does this method will terminate active sessions for both side? I mean the caller and the callee will be terminated when I do this? Is there any other way to terminate calls instead of sipsak? I want to try all available option just to do this. Please help.... Thanks, Ryan Samuel Osorio Calvo wrote: >I guess you change To:, From: and Call-ID headers every time you want to tear down a concrete session.....these headers change for every session and you will have to take the values from the message exchange creating the session. (INVITE/OK/ACK). >Moreover, it looks weird that the To,From, and Contact represents the same identity while the req-uri is another...... > >Hope you can use these hints to improve the so-wanted pre-paid stuff for SER ;) > >Samuel. > > >Unclassified. > > >>>><rpagquil(a)philonline.com> 10/11/05 12:52PM >>> >>>> >>>> >Hi guys, >I'm having a big problem on how can I terminate active sessions... I'm currently using sipsak to terminate it but all tests are not successfull. > >Here is what I do: >my bye file is like this: > >BYE sip:rpagquil@202.84.24.126:5060 SIP/2.0 >To: <sip:acjeff@sip.philonline.com>;tag=3986313335 >From: <sip:acjeff@sip.philonline.com>;tag=1362465910 >Contact: <sip:acjeff@202.84.24.126:5060> >Call-ID: 4D161E9C-3A79-11DA-BFB7-00E04CAB4AB4(a)10.0.1.144 >CSeq: 7 BYE >Max-Forwards: 16 >User-Agent: X-Lite release 1103m >Content-Length: 0 > >here im trying to terminate these two calls. >and here is what my command in sipsak: > >sipsak -vv -f byemes -s sip:sip.philonline.com > >But those things aren't working with me... any suggestions or any other methods are applicable with what i'm doing? > >I'm doing this so that those users that have no credits left are disconnected automatically...I'm trying to do a prepaid service for SER. > >Please help on this matter.... > >I really need your help... > >Thanks, > >Ryan Pagquil >Infodyne Inc. (www.philonline.com) >Tel. (632)-6870715 > > > > > -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

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[Users] Constructing sip uri from Dialed number and userloc
by Gene Willingham 11 Oct '05

11 Oct '05

Not sure if I have all of the terminology right, but I would like to have Asterisk register with openser. Then when a call comes in I would like to construct the sip uri using the Dialed number and the user location. The final uri should be: Sip:240744XXXX@25.25.25.25. Where the IP address is the user location of the Asterisk box. Any suggestions.

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[Users] BYE method accompanied by error
by Sam Lee 11 Oct '05

11 Oct '05

Hi all, I would like to know why does my BYE method are always replied with a 'Call Leg/Transaction does not exist' . How do they compare whether the transaction in the BYE method exist or not ? ( tag? ftag ? ) Are there any thing in the config that might cause this kind of problem ? Just want to highlight that all the calls are made in a good condition, everything except when the call is ending. Please let me know if you dont understand. Regards, Sam

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RE: [Serusers] logging of SIP dialog
by Ricardo Martinez 11 Oct '05

11 Oct '05

ngrep -W byline [calling-number] port 5060 -q It works to me.. Regards, Ricardo Martinez.- > -----Mensaje original----- > De: Iqbal [mailto:iqbal@gigo.co.uk] > Enviado el: Martes, 11 de Octubre de 2005 14:46 > Para: Klaus Darilion > CC: serusers(a)lists.iptel.org > Asunto: Re: [Serusers] logging of SIP dialog > > > Hi > > Is there any easy way of matching all messages belonging to a call in > ngrep, aside from pulling out the call-id and the grepping > for all those > which match that call-id...I am guessing not... > > iqbal > > Klaus Darilion wrote: > > > We use the low-level approach: > > > > > crontab -l > > # purge SIP packet sniffs > > 35 5 * * * find /export/sipsniff -name pack\* -mtime +60 > -exec rm {} \; > > > > > cat /usr/local/bin/sipsniff > > #!/bin/sh > > FILES=/export/sipsniff/packets > > while true; do > > FILESUFFIX=`date +%Y%m%d-%H%M%S` > > /usr/bin/tethereal -i any -a filesize:256 -n -w $FILES.$FILESUFFIX > > port 5060 or port 5061 > > done; > > > > This is not very convenient for daily debugging, but > sufficient I you > > debug once a month. > > > > regards > > klaus > > > > Iqbal wrote: > > > >> Hi > >> > >> Has anyone come up with a scalable method to log the complete SIP > >> trace of ALL calls, i.e to separate files etc > >> Ngrep is greate for a few, but I want to be able to track > all calls, > >> for various purposes > >> > >> Iqbal > >> > >> _______________________________________________ > >> 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] logging of SIP dialog
by Iqbal 11 Oct '05

11 Oct '05

tks, but I need to pull all data for all calls, and then disect them later for graphing (oh I wish). Iqbal Ricardo Martinez wrote: >ngrep -W byline [calling-number] port 5060 -q > >It works to me.. >Regards, > >Ricardo Martinez.- > > > > >>-----Mensaje original----- >>De: Iqbal [mailto:iqbal@gigo.co.uk] >>Enviado el: Martes, 11 de Octubre de 2005 14:46 >>Para: Klaus Darilion >>CC: serusers(a)lists.iptel.org >>Asunto: Re: [Serusers] logging of SIP dialog >> >> >>Hi >> >>Is there any easy way of matching all messages belonging to a call in >>ngrep, aside from pulling out the call-id and the grepping >>for all those >>which match that call-id...I am guessing not... >> >>iqbal >> >>Klaus Darilion wrote: >> >> >> >>>We use the low-level approach: >>> >>> >>> >>>>crontab -l >>>> >>>> >>># purge SIP packet sniffs >>>35 5 * * * find /export/sipsniff -name pack\* -mtime +60 >>> >>> >>-exec rm {} \; >> >> >>>>cat /usr/local/bin/sipsniff >>>> >>>> >>>#!/bin/sh >>>FILES=/export/sipsniff/packets >>>while true; do >>> FILESUFFIX=`date +%Y%m%d-%H%M%S` >>> /usr/bin/tethereal -i any -a filesize:256 -n -w $FILES.$FILESUFFIX >>>port 5060 or port 5061 >>>done; >>> >>>This is not very convenient for daily debugging, but >>> >>> >>sufficient I you >> >> >>>debug once a month. >>> >>>regards >>>klaus >>> >>>Iqbal wrote: >>> >>> >>> >>>>Hi >>>> >>>>Has anyone come up with a scalable method to log the complete SIP >>>>trace of ALL calls, i.e to separate files etc >>>>Ngrep is greate for a few, but I want to be able to track >>>> >>>> >>all calls, >> >> >>>>for various purposes >>>> >>>>Iqbal >>>> >>>>_______________________________________________ >>>>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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[Users] Question about one tables's layout
by Juan Carlos Castro y Castro 11 Oct '05

11 Oct '05

Why UNIQUE KEY? Wouldn't PRIMARY KEY be better? CREATE TABLE dbaliases ( alias_username varchar(64) default '' NOT NULL, alias_domain varchar(128) default '' NOT NULL, username varchar(64) default '' NOT NULL, domain varchar(128) default '' NOT NULL, UNIQUE KEY alias_key (alias_username,alias_domain), INDEX alias_user (username, domain) );

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[Users] Problem with Radius authentication
by Matthew Williams 11 Oct '05

11 Oct '05

Hello, I'm trying to set up OpenSER to use Radius authentication, but am having problems. I can't get the OpenSER server to actually send any Radius packets out over the network to my Radius server. I've been sniffing the network with Ethereal and no packets are sent out, even though I get messages in the syslog saying; Oct 10 16:59:24 ibwok-2mrt71j /usr/sbin/openser[9057]: rc_send_server: no reply from RADIUS server boxster:1812 Oct 10 16:59:24 ibwok-2mrt71j /usr/sbin/openser[9058]: rc_send_server: no reply from RADIUS server boxster:1812 Oct 10 16:59:25 ibwok-2mrt71j /usr/sbin/openser[9059]: rc_send_server: no reply from RADIUS server boxster:1812 and so on. My openser.cfg file (see below) is pretty much unchanged from the default config file, except adding Radius options as according to this HOWTO http://www.iptel.org/ser/doc/ser_radius/ser_radius.html I've seen this problem both with the pre-complied Debian packages on a Debian Sarge installation and also when compiled by hand on a Sun Solaris 9 server. I must be missing something from my configuration files, but I can't work out what it is. Does anyone have any ideas? Thanks in advance, Matt Williams ------------------------------------------------------------------ openser.cfg ------------------------------------------------------ # # $Id: openser.cfg,v 1.1.1.1 2005/06/13 16:47:30 bogdan_iancu 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 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/openser_fifo" alias=testdomain # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/openser/modules/auth.so" #loadmodule "/usr/lib/openser/modules/auth_db.so" loadmodule "/usr/lib/openser/modules/auth_radius.so" # ----------------- setting module-specific parameters --------------- modparam("auth_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- 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) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("testdomain")) { www_challenge("testdomain", "1"); 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(); }; } ------------------------------------------------------------------------ radiusclient.conf ------------------------------------------------------------------------ General settings # specify which authentication comes first respectively which # authentication is used. possible values are: "radius" and "local". # if you specify "radius,local" then the RADIUS server is asked # first then the local one. if only one keyword is specified only # this server is asked. auth_order radius # maximum login tries a user has login_tries 4 # timeout for all login tries # if this time is exceeded the user is kicked out login_timeout 60 # name of the nologin file which when it exists disables logins. # it may be extended by the ttyname which will result in # a terminal specific lock (e.g. /etc/nologin.ttyS2 will disable # logins on /dev/ttyS2) nologin /etc/nologin # name of the issue file. it's only display when no username is passed # on the radlogin command line issue /etc/radiusclient-ng/issue # RADIUS settings # RADIUS server to use for authentication requests. this config # item can appear more then one time. if multiple servers are # defined they are tried in a round robin fashion if one # server is not answering. # optionally you can specify a the port number on which is remote # RADIUS listens separated by a colon from the hostname. if # no port is specified /etc/services is consulted of the radius # service. if this fails also a compiled in default is used. authserver boxster # RADIUS server to use for accouting requests. All that I # said for authserver applies, too. # acctserver boxster # file holding shared secrets used for the communication # between the RADIUS client and server servers /etc/radiusclient-ng/servers # dictionary of allowed attributes and values # just like in the normal RADIUS distributions dictionary /etc/radiusclient-ng/dictionary # program to call for a RADIUS authenticated login login_radius /usr/sbin/login.radius # file which holds sequence number for communication with the # RADIUS server seqfile /var/run/radius.seq # file which specifies mapping between ttyname and NAS-Port attribute mapfile /etc/radiusclient-ng/port-id-map # default authentication realm to append to all usernames if no # realm was explicitly specified by the user # the radiusd directly form Livingston doesnt use any realms, so leave # it blank then default_realm # time to wait for a reply from the RADIUS server radius_timeout 10 # resend request this many times before trying the next server radius_retries 3 # local address from which radius packets have to be sent bindaddr localhost # LOCAL settings # program to execute for local login # it must support the -f flag for preauthenticated login login_local /bin/login This e-mail and any attachments are confidential and may also be legally privileged and/or copyright material of Intec Telecom Systems PLC (or its affiliated companies). If you are not an intended or authorised recipient of this e-mail or have received it in error, please delete it immediately and notify the sender by e-mail. In such a case, reading, reproducing, printing or further dissemination of this e-mail or its contents is strictly prohibited and may be unlawful. Intec Telecom Systems PLC does not represent or warrant that an attachment hereto is free from computer viruses or other defects. The opinions expressed in this e-mail and any attachments may be those of the author and are not necessarily those of Intec Telecom Systems PLC.

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Re: [Serusers] Have some INVITE problems
by Alex 11 Oct '05

11 Oct '05

I am using cisco 7960 and grandstream. 286,486 On 10/11/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > > Which UA are you using??? It looks like all your issues comes from the > implementation you are using as end-point and has nothing to do with SER. > > Samuel. > > > Unclassified. > >>> Alex <alexandergav(a)gmail.com> 10/11/05 11:06AM >>> > Hi all > I would like to explain some problem. > When my UAC initiates INVITE the server response with 100 trying > After receiving 100 trying the UAC sending again INVITE. ( HERE is the > problem). > RFC - 3261 ("After receiving a 1xx response, any retransmissions cease > altogether "); > Any help on that will be appreciated. > In addition if there any way to extend the timeout between the INVITES. > Thanks > >

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Re: [Serusers] Please help on terminating active sessions!!!
by Samuel Osorio Calvo 11 Oct '05

11 Oct '05

I guess you change To:, From: and Call-ID headers every time you want to tear down a concrete session.....these headers change for every session and you will have to take the values from the message exchange creating the session. (INVITE/OK/ACK). Moreover, it looks weird that the To,From, and Contact represents the same identity while the req-uri is another...... Hope you can use these hints to improve the so-wanted pre-paid stuff for SER ;) Samuel. Unclassified. >>> <rpagquil(a)philonline.com> 10/11/05 12:52PM >>> Hi guys, I'm having a big problem on how can I terminate active sessions... I'm currently using sipsak to terminate it but all tests are not successfull. Here is what I do: my bye file is like this: BYE sip:rpagquil@202.84.24.126:5060 SIP/2.0 To: <sip:acjeff@sip.philonline.com>;tag=3986313335 From: <sip:acjeff@sip.philonline.com>;tag=1362465910 Contact: <sip:acjeff@202.84.24.126:5060> Call-ID: 4D161E9C-3A79-11DA-BFB7-00E04CAB4AB4(a)10.0.1.144 CSeq: 7 BYE Max-Forwards: 16 User-Agent: X-Lite release 1103m Content-Length: 0 here im trying to terminate these two calls. and here is what my command in sipsak: sipsak -vv -f byemes -s sip:sip.philonline.com But those things aren't working with me... any suggestions or any other methods are applicable with what i'm doing? I'm doing this so that those users that have no credits left are disconnected automatically...I'm trying to do a prepaid service for SER. Please help on this matter.... I really need your help... Thanks, Ryan Pagquil Infodyne Inc. (www.philonline.com) Tel. (632)-6870715

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Re: [Serusers] Have some INVITE problems
by Samuel Osorio Calvo 11 Oct '05

11 Oct '05

Which UA are you using??? It looks like all your issues comes from the implementation you are using as end-point and has nothing to do with SER. Samuel. Unclassified. >>> Alex <alexandergav(a)gmail.com> 10/11/05 11:06AM >>> Hi all I would like to explain some problem. When my UAC initiates INVITE the server response with 100 trying After receiving 100 trying the UAC sending again INVITE. ( HERE is the problem). RFC - 3261 ("After receiving a 1xx response, any retransmissions cease altogether "); Any help on that will be appreciated. In addition if there any way to extend the timeout between the INVITES. Thanks

1 0

[Serusers] Writing a module
by Alan 11 Oct '05

11 Oct '05

Hi, I'm in the process of writing a module and was wondering if it was possible to perform SQL JOINS with the db query method? So instead of having the one huge table of usernames/domains have the two with username/domain_id and domain_id/domain. This is to perform number lookup and routing functions so will be quite big. If not is it possible to tell the db module what SQL it should execute?? Many Thanks, Alan

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Re: [Users] (no subject)
by Bogdan-Andrei Iancu 11 Oct '05

11 Oct '05

Hi Nelson, try: modparam("acc","db_table_acc","my_acc_table") this param is not documented unfortunately - the docs will be updated on release. regards, bogdan Nelson Silva wrote: > Anyone knows how to change the table for the acc logging ? > > Have been searching the accouting module but can't seem to find a way > to change the table. > > The default table used is 'acc' i would like to change to another one. > > if anyone knows how to, a help would be apreciate it. > > > Nelson Silva > ----------------- > email: nelson.silva(a)neuvex.com > website: http://www.neuvex.com > > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

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Re: [Users] (no subject)
by Klaus Darilion 11 Oct '05

11 Oct '05

It is not documented. But lookint at the source code (modules/acc/acc_mod.c) shows that the parameter is called "db_table_acc". regards klaus Nelson Silva wrote: > Anyone knows how to change the table for the acc logging ? > > Have been searching the accouting module but can't seem to find a way to > change the table. > > The default table used is 'acc' i would like to change to another one. > > if anyone knows how to, a help would be apreciate it. > > > Nelson Silva > ----------------- > email: nelson.silva(a)neuvex.com > website: http://www.neuvex.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] (no subject)
by Nelson Silva 11 Oct '05

11 Oct '05

1 0

[Users] Multiple interfaces and mediaproxy
by Fernando Rodriguez 11 Oct '05

11 Oct '05

Is there any way to run mediaproxy.py an make it listen on various ip adresses of a machine Thanks Fernando Rodriguez V. frod(a)aitelecom.net AITelecom S.A. de C.V. http://www.aitelecom.net

2 2

[Users] Calls where both endpoints are registered users fail.
by Brandon Price 11 Oct '05

11 Oct '05

Can someone explain what is happening here. On the dialing end @216.120.224.171 I get immediate congestion after the INVITE, on the receiving end @216.120.224.16 I get ringing even after I hang up. Relevant tethereal and openser.cfg below... 0.205952 216.120.224.171 -> 8.11.1.7 SIP/SDP Request: INVITE sip:15184782411@ser1.manhattan.ourdomain.net, with session description 0.252619 8.11.1.7 -> 216.120.224.171 SIP Status: 100 trying -- your call is important to us 0.252801 8.11.1.7 -> 216.120.224.163 SIP/SDP Request: INVITE sip:15184782411@216.120.224.163:5060, with session description 0.253078 8.11.1.7 -> 216.120.224.171 SIP Status: 500 I'm terribly sorry, server error occurred (1/SL) 0.279899 216.120.224.171 -> 8.11.1.7 SIP Request: ACK sip: 15184782411(a)ser1.manhattan.ourdomain.net 0.283146 216.120.224.163 -> 8.11.1.7 SIP Status: 100 Trying 0.296137 216.120.224.163 -> 8.11.1.7 SIP Status: 180 Ringing 0.311522 8.11.1.7 -> 216.120.224.171 SIP Status: 180 Ringing 7.032764 216.120.224.163 -> 8.11.1.7 SIP/SDP Status: 200 OK, with session description 7.033527 8.11.1.7 -> 216.120.224.171 SIP/SDP Status: 200 OK, with session description 7.054373 216.120.224.171 -> 8.11.1.7 SIP Request: ACK sip: 15184782411(a)ser1.manhattan.ourdomain.net { if (method=="INVITE" && uri =~ "sip:1[0-9]{10}@.*"){ if (is_user_in("From", "ld")){ if (!www_authorize("", "subscriber")){ www_challenge("", "1"); break; append_rpid_hf(); }; setflag(1); consume_credentials(); if(lookup("location")) { # if found, forward there... t_relay(); } else { prefix("+"); rewritehost("ourpstngateway"); }; };

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Re: Re: [Serusers] help:I can't build auth_radius module
by shikun.wang 11 Oct '05

11 Oct '05

Dear Zhangshuai, I'm sure there is no such file under my path.in fact it seems that it didn't create that file at all. Is it the problem of radiusclient library?I have installed radiusclient-0.4.9 and it has some errors when compling: authrad_mod.c:51:31: radiusclient-ng.h: 没有那个文件或目录(no such file or directory) Maybe I need use radiusclient-ng? -----原始邮件----- 发件人:"zhangshuai" 发送时间:2005-10-11 18:06:35 收件人:"serusers" 抄送:(无) 主题:Re: [Serusers] help:I can't build auth_radius module Dear shikun.wang, There are no such module file under your path: /usr/local/lib/ser/modules/ ?　　 hi, I have install radiusclient-0.3.2,and comment the related exclude modules in the main Makefile.But when I build the whole SER,it outputs something as follows: ... make[1]: Entering directory `/usr/local/src/ser-0.9.3/modules/auth_radius' make[1]: Leaving directory `/usr/local/src/ser-0.9.3/modules/auth_radius' make[1]: Entering directory `/usr/local/src/ser-0.9.3/modules/auth_radius' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.3"' -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 -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 -c authorize.c -o authorize.o gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.3"' -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 -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 -c authrad_mod.c -o authrad_mod.o make[1]: Leaving directory `/usr/local/src/ser-0.9.3/modules/auth_radius' ... It does NOT create an auth_radius.so file. Can anyone help me with some instructions?Thank you.

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RE: [Serusers] how to add the auth_radius module into SER
by Ricardo Martinez 11 Oct '05

11 Oct '05

Hi, Did you include the modules into the SER Makefile? Check for : exclude_modules?= and remove the auth_radius, acc_radius from it.. Regards, Ricardo.- -----Mensaje original----- De: shikun.wang [mailto:shikun.wang@163.com] Enviado el: Lunes, 10 de Octubre de 2005 23:55 Para: serusers(a)lists.iptel.org Asunto: [Serusers] how to add the auth_radius module into SER Hi,all I want to make my ser support radius accounting,so I add: loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/acc.so" into my ser.cfg,and uncomment DEFS+=-DRAD_ACC LIBS=-L$(LOCALBASE)/lib -lradiusclient in modules/acc/Makefile. But after I make the whole SER and find no auth_radius.so file under /usr/local/lib/ser/modules,neither can I start the SER for it can't load this module. Is there any thing I have not set? Any ideas?Thank you.:) 需要一个 <http://www.126.com> 2000 兆的免费邮箱吗？网易免费邮箱是中国最多人使用的电子邮箱。

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[Serusers] save_memory behavior for NATed clients
by George Perantinos 11 Oct '05

11 Oct '05

Dear list, I'm trying to implement registration replication between 2 SERs (ver. 0.9.4, they are AA behind an F5 BigIP). Both SERs are using the same mysql database and mediaproxy is used for RTP relaying (whenever needed). The basis for my configuration is the mediaproxy config from onsip.org (excellent work, many thanx guys). My servers are 10.21.128.232 and 10.21.128.233. I'm using forward_tcp and save_memory for replication purposes, as can be seen in the following registration route block: route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if(src_ip==10.21.128.233) { # If it is a forward from our brother then... if (client_nat_test("3")) { # check if the user is NATed setflag(6); fix_nated_register(); force_rport(); }; save_memory("location"); break; } else { if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!radius_www_authorize("domain.com")) { www_challenge("domain.com","0"); xlog("L_NOTICE","Could not authorize user %fu, IP %is \n"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); xlog("L_NOTICE","From: does not match URI:, user %fu, IP %is \n"); break; }; consume_credentials(); if (!save("location")) { xlog("L_WARN","MYSQL down? Could not save location\n"); sl_reply_error(); }; xlog("L_INFO","Updating registration for user %fu, IP %is \n"); add_rcv_param(); if (!forward_tcp("10.21.128.233", 5060)) { xlog("L_WARN","Cannot replicate user %fu at server 10.21.128.233\n"); }; }; } When performing a "serctl ul show" on the servers, the results are as follows: 1) For UAs coming from real IPs, serctl shows identical results on both servers. 2) For NATed UAs there is difference in the "Flags:" field. On the server that initialy proccessed the request, the value of the "Flags" is 1 (OK!), while on the server that the registration was forwarded, the value of the "Flags" field is 257(??). My impression is that the Flags field should be 1 if a UA is NATed and 0 if not. Am I wrong? If I'm right on the above, then shouldn't both servers show the same result in Flags (i.e. 1) for NATed clients? On the other hand, if I restart SER (and the contacts are read from the database), then every "weird" 257 Flag is reset to 1, until the next UA re-registration (where it returns again to 257). Thanx for any help, George.

2 3

Re: [Serusers] msilo configuration
by Teemu Harju 11 Oct '05

11 Oct '05

Hi Nhadie, Yes... you can use it for instant messaging. eyeBeam is in my opinion the best featured SIP client ever. It can do voice, video, instant messaging, presence, conferencing etc... and all standard SIP. Note that the eyeBeam is the commercial version and Xlite is free. And with Xlite you cannot do anything else than voice. - Teemu 2005/10/11, Nhadie <nhadie(a)cbcpworld.com>: > > > Hi Teemu, > > Are you saying you can use eyebeam afor instant messagin? i thought it > was just for calling. > > Regards, > Nhadie > > > Hi, > > > > I've been configuring msilo for offline messaging and I've faced few > > problems. > > > > First... I use Xten (or now Counterpath) Eyebeam and each time user > > starts to type an instant message it sends a SIP MESSAGE with > > Content-Type: application/im-iscomposing+xml and msilo naturally > > stores this message also. I tried to prevent this with the following > > configuration but it didn't work.. > > > > if (search("Content-Type: application/im-iscomposing+xml")) { > > > > # > > # Do not store composing MESSAGEs > > # > > } else { > > > > # > > # Store offline message > > # > > } > > > > Or course this is done only for method=="MESSAGE"... Can you see why > > this is not working? > > > > Second... Is there a way to make the reply to the offline message > > appear to be coming from the user that it was sent to? I mean you can > > configure msilo to reply for example with user > > registrar(a)myserproxy.com <mailto:registrar@myserproxy.com>, but that > > opens a new IM window for the user so it's not that good. > > > > Thanks in advance... > > -- > > Teemu Harju > > http://www.teemuharju.net > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > -- > ---------------------------- ------ | > -- Computers are like Air Conditioners, | > ------ It doesn't work when Windows are open! | > ---------------------------- ------ | > > > -- Teemu Harju http://www.teemuharju.net

2 5

[Serusers] Please help on terminating active sessions!!!
by rpagquil＠philonline.com 11 Oct '05

11 Oct '05

1 0

Re: [Serusers] help:I can't build auth_radius module
by zhangshuai 11 Oct '05

11 Oct '05

Dear shikun.wang, There are no such module file under your path: /usr/local/lib/ser/modules/ ?　　 hi, I have install radiusclient-0.3.2,and comment the related exclude modules in the main Makefile.But when I build the whole SER,it outputs something as follows: ... make[1]: Entering directory `/usr/local/src/ser-0.9.3/modules/auth_radius' make[1]: Leaving directory `/usr/local/src/ser-0.9.3/modules/auth_radius' make[1]: Entering directory `/usr/local/src/ser-0.9.3/modules/auth_radius' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.3"' -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 -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 -c authorize.c -o authorize.o gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.3"' -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 -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 -c authrad_mod.c -o authrad_mod.o make[1]: Leaving directory `/usr/local/src/ser-0.9.3/modules/auth_radius' ... It does NOT create an auth_radius.so file. Can anyone help me with some instructions?Thank you.

1 0

[Serusers] Need some help
by Alex 11 Oct '05

11 Oct '05

Hi all I would like to explain some problem. When my UAC initiates INVITE the server response with 100 trying After receiving 100 trying the UAC sending again INVITE. ( HERE is the problem). RFC - 3261 ("After receiving a 1xx response, any retransmissions cease altogether "); Any help on that will be appreciated. In addition if there any way to extend the timeout between the INVITES. Thanks

1 0

[Serusers] Have some INVITE problems
by Alex 11 Oct '05

11 Oct '05

Hi all I would like to explain some problem. When my UAC initiates INVITE the server response with 100 trying After receiving 100 trying the UAC sending again INVITE. ( HERE is the problem). RFC - 3261 ("After receiving a 1xx response, any retransmissions cease altogether "); Any help on that will be appreciated. In addition if there any way to extend the timeout between the INVITES. Thanks

1 0

[Users] [Fwd: [Sip-implementors] TLS certificate question]
by Klaus Darilion 11 Oct '05

11 Oct '05

Hi! I asked a question on the sip-implementors list. As nobody answered I post it here again, as it is also related to ser/openser. thanks klaus -------- Original Message -------- Subject: [Sip-implementors] TLS certificate question Date: Wed, 05 Oct 2005 09:10:05 +0200 From: Klaus Darilion <klaus.mailinglists(a)pernau.at> To: SIP Implementors <sip-implementors(a)cs.columbia.edu> Hi! I'm trying to figure out how to make a certificate for a SIP proxy. RFC3262, section 4.1 states: For NAPTR records with SIPS protocol fields, (if the server is using a site certificate), the domain name in the query and the domain name in the replacement field MUST both be valid based on the site certificate handed out by the server in the TLS exchange. Similarly, the domain name in the SRV query and the domain name in the target in the SRV record MUST both be valid based on the same site certificate. Otherwise, an attacker could modify the DNS records to contain replacement values in a different domain, and the client could not validate that this was the desired behavior or the result of an attack. I'm not sure what the phrase "...MUST both be valid based on the site certificate" means. Does it mean that all possible domains must be present in the certificate? now imagine the follwing DNS lookups: sip:user@example.com ; order pref flags service regexp replacement IN NAPTR 50 50 "s" "SIPS+D2T" "" _sips._tcp.example.com. IN NAPTR 90 50 "s" "SIP+D2T" "" _sip._tcp.example.com IN NAPTR 100 50 "s" "SIP+D2U" "" _sip._udp.example.com. _sips._tcp.example.com. That lookup would return: ;; Priority Weight Port Target IN SRV 0 1 5061 server1.example.com IN SRV 0 2 5061 server2.example.com _sip._tcp.example.com. That lookup would return: ;; Priority Weight Port Target IN SRV 0 1 5060 server1.example.com IN SRV 0 2 5060 server2.example.com _sip._udp.example.com. That lookup would return: ;; Priority Weight Port Target IN SRV 0 1 5060 server1.example.com IN SRV 0 2 5060 server2.example.com Finally, a TLS connection is made with server1.example.com. IF I understand RFC3263 correct, all of the above domains must be present in the certificate, but how to do this? 1. Should I put CN=example.com into the Subject and all other domains into the Subject Alternative Name? DNS=_sips._tcp.example.com. DNS=_sip._tcp.example.com DNS=_sip._udp.example.com. DNS=server1.example.com DNS=server2.example.com 2. Should I leave the Subject empty and put all domains into the Subject Alternative Name? 3. Why is it not sufficient to use only the domain "example.com" in the certificate (putting it into the subject field)? 4. Which SIP URIs should be used to check against the domains in the certificate (mutual proxy-proxy scenario)? Is it correct to check the domain in the request URI against the certificate of the receiving proxy, and check the domain in the From: URI against the certificate of the originating proxy? Thanks for any clarifications Klaus _______________________________________________ Sip-implementors mailing list Sip-implementors(a)cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

6 11

[Serusers] help:I can't build auth_radius module
by shikun.wang 11 Oct '05

11 Oct '05

hi, I have install radiusclient-0.3.2,and comment the related exclude modules in the main Makefile.But when I build the whole SER,it outputs something as follows: ... make[1]: Entering directory `/usr/local/src/ser-0.9.3/modules/auth_radius' make[1]: Leaving directory `/usr/local/src/ser-0.9.3/modules/auth_radius' make[1]: Entering directory `/usr/local/src/ser-0.9.3/modules/auth_radius' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.3"' -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 -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 -c authorize.c -o authorize.o gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.3"' -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 -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 -c authrad_mod.c -o authrad_mod.o make[1]: Leaving directory `/usr/local/src/ser-0.9.3/modules/auth_radius' ... It does NOT create an auth_radius.so file. Can anyone help me with some instructions?Thank you.

1 0

[Serusers] msilo configuration
by Teemu Harju 11 Oct '05

11 Oct '05

Hi, I've been configuring msilo for offline messaging and I've faced few problems. First... I use Xten (or now Counterpath) Eyebeam and each time user starts to type an instant message it sends a SIP MESSAGE with Content-Type: application/im-iscomposing+xml and msilo naturally stores this message also. I tried to prevent this with the following configuration but it didn't work.. if (search("Content-Type: application/im-iscomposing+xml")) { # # Do not store composing MESSAGEs # } else { # # Store offline message # } Or course this is done only for method=="MESSAGE"... Can you see why this is not working? Second... Is there a way to make the reply to the offline message appear to be coming from the user that it was sent to? I mean you can configure msilo to reply for example with user registrar(a)myserproxy.com, but that opens a new IM window for the user so it's not that good. Thanks in advance... -- Teemu Harju http://www.teemuharju.net

1 0

[Serusers] Radius authentication and accounting issues
by deepak.chandrasekaran＠wipro.com 11 Oct '05

11 Oct '05

Hi all, I am trying to provide radius authentication and accounting in ser. I am using freeradius. Everytime I try to start ser, it crashes. Can anybody suggest some possible reasons as to why this happens? I followed radius how-to doc to configure radius server and radiusclient-ng library. I hv attached herewith my ser.cfg file. Any help would be greatly appreciated. Thanks in advance. Regards, Deepak Ser.cfg: # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=no #log_stderror=yes # (cmd line: -E) /* 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" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # RADIUS support #loadmodule "/usr/local/lib/ser/modules/acc.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" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- acc params -- #modparam("acc", "radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") #modparam("acc", "radius_missed_flag", 2) #modparam("acc", "radius_flag", 1) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- auth_radius params -- modparam("auth_radius", "radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius", "service_type", 15) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # log(1, "REGISTER: Authenticating user\n"); # Uncomment this if you want to use digest authentication # if (!www_authorize("nerd.vngncoe.wipro.com", "subscriber")) { if (!radius_www_authorize("")) { log(1, "REGISTER: challenging user\n"); 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(); }; } Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

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[Serusers] how to add the auth_radius module into SER
by shikun.wang 11 Oct '05

11 Oct '05

Hi,all I want to make my ser support radius accounting,so I add: loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/acc.so" into my ser.cfg,and uncomment DEFS+=-DRAD_ACC LIBS=-L$(LOCALBASE)/lib -lradiusclient in modules/acc/Makefile. But after I make the whole SER and find no auth_radius.so file under /usr/local/lib/ser/modules,neither can I start the SER for it can't load this module. Is there any thing I have not set? Any ideas?Thank you.:)

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[Serusers] CDRTool
by Cameron Beattie 11 Oct '05

11 Oct '05

I have been testing CDRTool and get the following message in the log files when browsing to www.example.com/scripts/ratingEngine.php: Cannot instantiate non-existent class: ratingtables in /var/www/CDRTool/scripts/ratingEngine.php Is this because the non commercial version of CDRTool does not provide rating or is it something I have done wrong in the setup? Any advice would be appreciated. Regards Cameron

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Re: [Serusers] denying call based on country codes
by Nhadie 10 Oct '05

10 Oct '05

Thank you for your reply. Unfortunately I send all pstn traffic to a single gateway provider. So what I really need is for ser to check if this user is in the 36countries group, and it dialled a number outside of the 36 countries then denied. I was thinking of using multiple if URI but i don't want to define a lot IF statements. Hope someone here can share something that they were able to do. > > > > > > > > > > > > > > > > > Sure... it's possible and actually pretty easy, dependent on how things > are set up. Assuming, for the sake of argument, that you have two servers > that handle PSTN calls and one handles calls to anywhere and one handles > calls to only 36 countries: > > > > I'd use the grp table for this sort of thing (although some people would > use AVPops). I'd just add the users to a particular kind of group for all > access.... > > > > In MySQL, you'd want to do something like: > > > > insert into grp > values('&lt;username&gt;','&lt;domainname&gt;','fullaccess',now()); > > > > > > > > Then, set a flag if the user is in that group... > > > > if(is_user_in(&quot;Request-URI&quot;, &quot;fullaccess&quot;)) > > { > > setflag(19); > > }; > > > > > > Then... somewhere in an INVITE check block (or wherever, really), check > for the URI and the flag. > > > > if (uri =~ &quot;^sip:00*&quot;) > > { > > if(isflagset(19)) > > { > > log(1, &quot;request for alternate 800 provider > receivedn&quot;); > > rewritehostport(&quot;10.1.1.1:5060&quot;); # PSTN for ALL > numbers > > forward(uri:host, uri:port); > > route(1); > > break; > > } > > else > > { > > log(1, &quot;request for alternate 800 provider > receivedn&quot;); > > > > rewritehostport(&quot;10.1.1.2:5060&quot;); # PSTN for only > 36 countries > > > > forward(uri:host, uri:port); > > > > route(1); > > > > break; > > }; > > > > }; > > > > > > > > Now, the trick comes when you DON'T have PSTN server separate for > different countries... > > > > THEN you might have to do some very in-depth rules based on the format of > the URI to the 36 countries in question which really isn't THAT bad, but > dependent on their numeric format for numbers, it could get annoying to > specify each one (especially if they have differing formats for land line > and mobile numbers). It's not undoable, though. > > > > N. > > > > > > On Mon, 10 Oct 2005 20:17:50 +0800, Nhadie wrote > > &gt; Hi All, > > &gt; > > &gt; I'd like to setup to groups, one that can call only around 36 > > countries and one that can call anywhere. Currently, when I see that the > URI > > starts with 00 i'll check if it belongs to the world-pstn group then it > can > > dial-out if not it's blocked. Is what I'm thinking > > possible? > > &gt; > > &gt; TIA > > &gt; > > &gt; Regards, > > &gt; Nhadie > > > > > > > > > > > > > > > > > > > > > > > > >

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RE: [Serusers] SER Calls Per Second
by Martin, Peter 10 Oct '05

10 Oct '05

Let's say, just sip proxy... Regards, Peter Martin -----Original Message----- From: Daryl Sanders [mailto:daryl.sanders@gmail.com] Sent: Monday, October 10, 2005 3:35 PM To: Martin, Peter Subject: Re: [Serusers] SER Calls Per Second This depends on if you will be proxying the media stream or if you are just handling call setup with SER. If you plan to carry the media you will need to use rtpproxy or mediaproxy along with SER. It will also matter if you are using SER as the registrar on the same box or just routing calls. In addition, the specs of the box/boxes you'll be using will have a huge effect on the number of calls. - Daryl On 10/10/05, Martin, Peter <Peter.Martin(a)ipc.com> wrote: > > > Anyone know what is the most calls per second that has been tested > > and verified using SER? > > > > Peter Martin > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > >

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[Users] TLS setup
by Bogdan-Andrei Iancu 10 Oct '05

10 Oct '05

Hi everybody, finally the TLS proxy is up and running - just to remember what was about -> http://www.openser.org/pipermail/users/2005-September/000953.html The configuration is quite a simple one - it accepts exclusively requests over TLS and does relay based on RURI (keeping the TLS protocol). server info: openser.org port 5061 - SRV record is also present. The idea is to interconnect several domains via this TLS relay. If one of the interconnected servers has to deliver a request to another domain via TLS, it has just to relay it to the TLS proxy. Note that the proxy denies any local requests. How to do the redirect from your script? Before doing t_relay(), you may add something like: if ( uri=~".*(a)domain1.com" || uri=~".*(a)domain2.com") { avp_write("sip:openser.org;transport=tls","i:11"); avp_pushto("$duri","i:11"); } via DST_URI, the TLS protocol will be forced without affecting the RURI. Normally this logic should be totally transparent via NAPTR lookup, but we do not have it in OpenSER yet. Now, about the TLS part. The TLS proxy has both tls_verify and tls_require_certificate enabled. For certificates there are two ways: 1) generate your own root CA and certificate and set the CA to me to added to the proxy (tools available on CVS in tls/tools) 2) ask and I will provide a certificate signed with the proxy root CA. TLS configuration on your side: tls_certificate="path/cert.pem" tls_private_key="path/privkey.pem" tls_ca_list="path/calist.pem" tls_verify=on tls_require_certificate=on listen=tls:xxx.xxx.xxx.xxx tls_port_no=5061 For any other additional hints about TLS, please see the tutorial http://www.openser.org/docs/tls People interested in interconnection, please join (just let me know which way you prefer for certificates). First domain connected to the proxy is siphub.net - it's a local platform we are using. regards, bogdan PS: if there are people interested in testing SIP UA with TLS, let me know and I can add registrar support. Also, if somebody has ideas how to enhance the proxy config, feel free....

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RE: [Serusers] RPM's of recent SER versions
by Elad Dotan 10 Oct '05

10 Oct '05

Try from this site: http://yum.sip-router.org/fedora-4/ Elad. -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Yair Hakak Sent: Monday, October 10, 2005 10:27 PM To: serusers(a)lists.iptel.org Subject: [Serusers] RPM's of recent SER versions hello ser users list, where can i get RPM's for recent versions of SER? on berlios the last SER to have RPM's is 0.8.12. thanks, yair

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[Serusers] RPM's of recent SER versions
by Yair Hakak 10 Oct '05

10 Oct '05

hello ser users list, where can i get RPM's for recent versions of SER? on berlios the last SER to have RPM's is 0.8.12. thanks, yair

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[Serusers] SER Calls Per Second
by Martin, Peter 10 Oct '05

10 Oct '05

Anyone know what is the most calls per second that has been tested and verified using SER? Peter Martin

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[Users] tls_require_certificate
by Alexander Ph. Lintenhofer 10 Oct '05

10 Oct '05

Hi there, I have a question concerning TLS in openser: By switching tls_require_certificate to "on", the peer is forced to send his certificate for means of mutual authentication. My problem is, that the peer may be another proxy server whom I want to authenticate with its cert - but the peer might also be an user agent. In my situation I use a Snom 360 which has not the possibility to import an own user-certificate (only a CA-cert for verifying server-certs). ----------- ---------- --------- | snom 360 | <------ TLS -------> | outbound | <----- TLS -----> | inbound | ----------- server sends cert ---------- mutual AUTH --------- But when I activate tls_require_certificate=on in the openser.cfg of the outbound proxy, the snom360 can't register, because it has no user-cert. On the other hand, when I disable tls_require_certificate, the snom can register, but the security between the proxies is weak. Is there an appropriate solution for this problem ?? Maybe I didn't understand the sample configuration at all.... Thanks in advance and regards, Philipp

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[Serusers] Calling other SIP users
by Voipers Portugal 10 Oct '05

10 Oct '05

Hi, I have my SER working when i try to talk to other SIP servers. However, i cannot make calls inside my own network. I am using LDAP to authenticate my users, and I have sipldap script that circulates in the web. Everything works fine until there, i get the following output when the user exists: sip:username@host isn´t this what was supposed to happen? When i then try to (!lookup("location")) { if (!exec_dset("/etc/ser/sipldap")) { sl_send_reply("404", "Not Found"); break; } else { log(1," sipldap call"); }; }; #!lookup I get the Not Found message. But if i call the following way: username@username_IP the call works fine.. any ideia? Jose Simoes

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[Serusers] check_from
by Jason Mohler 10 Oct '05

10 Oct '05

I have a spoofed call that keeps coming in with the ANI of 000000000. How can I use check_from() and is it comes in as 000000000 to just drop the call so it does not take up my resources. Is below a reliable synatx for this? if (method==INVITE) && (!check_from()){ break; };

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[Users] Have openser make changes based on message header
by Brandon Price 10 Oct '05

10 Oct '05

First I want to thank everyone thats helped me. Here is what I am working on today. I have a set of sipura 2002, and linkys pap2 sip ata's. When you dial *67 (cid block) it tells the adapter to return a dialtone and modify the sip message header in such a way the SIP Display info is changed to "Anonymous". This doesn't set the privacy=full flag in the rpid and thus the cid is passed to the terminating end. I have tried having openser perform an action based on if from_uri =~ Anonymous but that apparently only matches to the SIP from address: not, SIP Display info. Below I have a sample bit of sip messaging.... My question is, is there any way to have openser look inside the packet and do something based on "SIP Display info"? Can I do this w / AVP's have it pull that var form the sip messaging so I can work with it later? Or heck does anyone have a solution to make the ata's pass the privacy=full flag? Message Header Via: SIP/2.0/UDP 216.120.224.163:5060;branch=z9hG4bK-f4bd2420 From: Anonymous <sip: 15184782406(a)ser1.manhattan.vtnoc.net>;tag=d5d92a6937d9f838o0 SIP Display info: Anonymous SIP from address: sip:15184782406@ser1.manhattan.vtnoc.net SIP tag: d5d92a6937d9f838o0 To: <sip:13143212222@ser1.manhattan.vtnoc.net> SIP to address: sip:13143212222@ser1.manhattan.vtnoc.net Call-ID: bae3e6fd-a92244b4@localhost CSeq: 101 INVITE Max-Forwards: 70

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[Serusers] ser replying at udp source port instead of via-port
by Atif Rasheed 10 Oct '05

10 Oct '05

Hello all, I am using ser 0.9.4 + rtpproxy. now when our SSW INVITEs SER, SER sends 100-trying to the udp-source-port where SER should reply with connection information in via or contact headers. (probably via header) Please comment. Thank you -- Atif

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[Serusers] SIp auto dialer
by Iqbal 10 Oct '05

10 Oct '05

Hi Does anyone have any recommendations, linux one is better, and would sipsak do this. All I want is a number to be dialled, and then handup once answered, no voice required. This is not for spam, just to allow me to dial up random numbers worldwide, from which I want to plot a time to completion graph, from where I can then hopfully automatically change my GW in lcr, for best route. Iqbal

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[Serusers] SER 0.9.4/ postgresql Registration problems
by H Quintana 10 Oct '05

10 Oct '05

Hi, I just updated SER from 0.9.3 to 0.9.4. In both cases I'm using Postgresql. With SER 0.9.3 the REGISTER process works fine, the REGISTER is accepted and then the record is inserted in DB. With SER 0.9.4, I've found that when the "location" is inserted or updated in the DB, SER is generating core. However when SER starts the locations list is read from DB without any problem. Any ideas? Thanks for your help, Humberto __________________________________ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs

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[Users] REGISTER expires and replication
by Helge Waastad 10 Oct '05

10 Oct '05

Hi, I have a scenario in which I have a little bit of confusion.. I have a redundant setup, two OpenSER, a MySQL cluster and replication of REGISTERs. The UAC sends expires=3600 Both OpenSER's has a max_expires=1800, min_expires=300,default_expires=300 My Sipura-1001 has default expires=3600. In the OK message sent from the OpenSER handeling the REGISTER, expires=1800 is sent back to the UAC which is OK (However the Sipura does not obey this, which also is struggeling, but that is another story....) The replicated REGISTER, hovever, uses the original expiery from the UAC...expires=3600 Do I have to rewrite the expires in the replicated message? -- mvh/best regards Helge Waastad System Engineer Smartnet

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[Serusers] SER, and the two firewall problem
by Richard Southwick 10 Oct '05

10 Oct '05

Hi all -- We're using SER as a registrar and proxy, and need to deal with the following NAT problem: * SER is proxying to asterisk, which is behind a firewall. * Our users are registering with SER, but their UAs are behind another firewall. We're looking at nathelper and rtpproxy. Two questions: * Will this combo (SER, nathelper, rtpproxy) work in this scenario to solve the NAT and pinhole issues? * Does SER have to live outside the firewall, or is there a way it can be behind the same firewall as Asterisk? Thanks for any insight, Richard

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[Serusers] [Fwd: [Sip-implementors] TLS certificate question]
by Adrian Georgescu 10 Oct '05

10 Oct '05

Juha, I host for example multiple domains for multiple users. This means that users rely on my services entirely to provide them with identity services. Ideally I would like to present the Certificate of domain X from user in domain X when I forward his calls through my proxy. Alternatively I could say I put certificate with CN=ag-projects.com in all request going outside my domains but then my white-label customers will become yellow. So something to match business with the technicalities is required, don't you think? Adrian > But then, the whole authorization thing would be nonsens. > > Just imagine a host named "sip.badguy.com". This host has a valid > certificate for its hostname. Then, this SIP proxy sends a SIP request > with the header: > From: "Klaus Darilion" <sip:klaus at darilion.com> > > Now, what is the receiving proxy interested in? Does it want to validate > the host or the sender (From header)? there are other ietf means to validate the sender. usually they involve signing of from uri with the certificate of its domain. see for example draft-rosenberg-sip-identity-privacy-00 for a good summary of the issues and problems involved. in proxy-to-proxy case, all that needs to be done is to validate the remote proxy. > IMO, I want to authenticate the sender in the From header. Thus, the > certificate would have to match the SIP domain, and not the host > name. see above. -- juha

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[Serusers] Conferencing...
by Voipers Portugal 10 Oct '05

10 Oct '05

As far as I have seen, SER itself cannot offer conferencing services, right? Therefore I have also noticed that you recommend Asterisk, SEMS or others. My doubt is, how does SER contacts with those servers? Does SER supports MGCP? I tought not, but if this is true, how do they communicate? If anyone knows where i can see more info on SER conferencing, please let me know. Regards, Jose Simoes

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[Serusers] time stamping each SIP message in a dialogue
by Iqbal 10 Oct '05

10 Oct '05

Hi I just realised that the date: parameters is not set in each SIP message, am sure this is correct, but wouldnt this be a good idea, I am trying to work out which part of my call is being delayed, and the date command would make it easier. Just compare times between each message in ngrep. Something like what mailservers do, so that you can see if the mail was on a queue for a long time. Iqbal

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[Serusers] denying call based on country codes
by Nhadie 10 Oct '05

10 Oct '05

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[Serusers] lcr module based on timeout
by Iqbal 10 Oct '05

10 Oct '05

Hi Can the lcr module be deployed in such as way where, if call takes x secs to complete it will go elsewhere..or perhaps even before the invite is sent, a RTT to the gateway is done, and it picks the nearest/best gateway for that prefix Iqbal

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[Serusers] Testing
by Pavol Segeč 10 Oct '05

10 Oct '05

Hi all, Is there some methodology and test tools for doing load testing (performance capabilities) of SIP Servers? Thanks pavol Sorry, for this off topic.

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[Users] caller id authentication
by jayesh nambiar 10 Oct '05

10 Oct '05

hi all, I am trying to implement this feature in my openser 0.10. It is as follows: I allow users in pstn network to dial my gateway number, wherein they get a dial tone and the gateway then forwards all the calls to openser and are processed accordingly. But now I want to authenticate these caller ids. I want to give this privilege only to my subscribers. I had thought of adding the ids in the trusted table as my gateway is in trusted table, but this does not seem to be a good solution. Can I add an extra column in the subscriber table wherin I mention the caller-id for individual users (I give only one caller_id to one user), so that if the user is authenticated then the caller- id for that user should be authenticated. Can it be possible. I was thinkin of using AVPs for this, but I dont get the idea of retreiving the caller-ids from the table to check if the caller-id is valid. Can someone help me in this regard. thanks jayesh --------------------------------- Yahoo! Music Unlimited - Access over 1 million songs. Try it free.

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RE: [Serusers] ser web
by Elad Dotan 10 Oct '05

10 Oct '05

Hi, Thank you for your response. I'm now able to see the serweb login screen. When I put the username (admin) and password (heslo) I get a "bad username or password" error Message. Could you please assist. Thanks, Elad. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Friday, September 30, 2005 1:02 PM To: Jan Janak Cc: Elad Dotan; serusers(a)lists.iptel.org Subject: Re: [Serusers] ser web And watch the apache log files. There you will see the http requests and why apache can't find them. regards klaus Jan Janak wrote: > SERWeb should be accessible as http://<ip_of_the_host>/serweb. > > If that returns 404 then you need to create /serweb alias in Apache > configuration. You can do so by including /etc/serweb/apache.conf file > in the main Apache configuration file. > > Jan. > > On 29-09-2005 16:40, Elad Dotan wrote: > >>Hi, >>I've installed 4 RPM files on a Fedora CORE 3. >>The RPM are: >> >>ser-0.9.3-0.i386.rpm >>php-pecl-classkit-0.4-2.i386.rpm >>ser-mysql-0.9.3-0.i386.rpm >>serweb-0.9.3-0.i386.rpm >>I enabled the httpd and mysql service. >>I'm not sure what to do next. >>All the Documentation I have is related to source flies and not RPM. >>Ser is working but I don't know how to configure the serweb in order >>for it to work. >> >>I tried: http:// >>but I get an error. >> >>where can I find documentation about ser web? >>Thanks, >>Elad. >> > > >>_______________________________________________ >>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] PSTN through CISCO
by share phone 10 Oct '05

10 Oct '05

Hi All, I have a question how to use PSTN through cisco through openser.I am intrested in the security any help is appreciated. Do any one have the CDR Report through User Interface? Regards, karun. --------------------------------- Yahoo! Music Unlimited - Access over 1 million songs. Try it free.

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[Serusers] how to keep passwords encrypted in radchek table ??
by tulay sazak 10 Oct '05

10 Oct '05

Hello EveryBody, I am using SER version 0.9.3 and freeRADIUS version 1.0.4. I am trying to keep the user passwords as encrypted in freeRadius DB which is mySql table radcheck. When I run the Radius Request I got: auth: No User-Password or CHAP-Password attribute in the request auth: Failed to validate the user. What should I do to solve this problem? 1) in radiusd.conf ? 2) What should be the value of "Auth-Type" parameter in radcheck or radgroupchek tables ? 3) What should be the attribute for password in radcheck table ? Is it "User-Password" or "Chap-Password" or "Crypt-Password" ?? I'll appreciate very much if anyone can help. Regards Tulay Sazak __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

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[Serusers] registration problems...
by Olivier Taylor 10 Oct '05

10 Oct '05

hello, I have a registration problem. I use many different uacs(Zyxel,Grandstream ata486, budgettone100,sipura 3000, alcatel st280, eyebeam,....) Ser version is 0.9.4 on Freebsd 5.4 On ser.cfg, I have : modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 1200) modparam("registrar", "max_expires", 1800) and if (nat_uac_test("19")) { if (method == "REGISTER" || !search("^Record-Route:")) { fix_nated_contact(); force_rport(); etflag(6); }; }; The problem is that after a few ReRegistration, ser stop updates od the dababase even if ReRegistration continues to come... Ok, min_expires seems not Rfc compliant, but even without that I have the same behavior. I have seen many discussions about that and tried many solutions... The 2 greps shows register messages for 2 differents user agents Any good idea? ps: Greger, isn't it to telegraphic this time? ;) U 2005/10/09 14:37:37.460238 81.241.192.55:6938 -> 82.146.123.252:5060 REGISTER sip:finalcut.be SIP/2.0..To: 3227470340<sip:3227470340@finalcut.be>..From: 3227470340<sip:3227470340@finalcut.be>;tag=ab7b0c68..Via: SIP/2.0/UDP 192.168.2.101:6938 ;branch=z9hG4bK-d87543-505047371-1--d87543-;rport..Call-ID: 5e6f6777312d2659..CSeq: 13 REGISTER..Contact: <sip:3227470340@192.168.2.101:6938>..Expires: 3600..Max-Forwards: 70..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: eyeBeam release 3007n stamp 17816..Authorization: Digest username="322747 0340",realm="finalcut.be",nonce="4349103d8b8ae8fadbc161b5bb1a93a299fb4f4a",u ri="sip:finalcut.be",response="e6735467abe564030707db572d6f9f1f",algorithm=M D5..Content-Length: 0.... # U 2005/10/09 14:37:37.480571 82.146.123.252:5060 -> 81.241.192.55:6938 SIP/2.0 200 OK..To: 3227470340<sip:3227470340@finalcut.be>;tag=59f1565d6938a98ef3fc4707bf9b43aa. 1234..From: 3227470340<sip:3227470340@finalcut.be>;tag=ab7b0c68..Via: SIP/2. 0/UDP 192.168.2.101:6938;branch=z9hG4bK-d87543-505047371-1--d87543-;rport=6938;rec eived=81.241.192.55..Call-ID: 5e6f6777312d2659..CSeq: 13 REGISTER..Contact: <sip:322747034 ;expires=1800"> <mailto:0@81.241.192.55:6938> 0@81.241.192.55:6938>;expires=1800, <sip:3227470340@81.241.192.55:1048>;expires=119..Server: Sip EXpress router (0.9.4 (i386/freebsd))..Content-Length: 0..Warning: 392 82.1 46.123.252:5060 "Noisy feedback tells: pid=43104 req_src_ip=81.241.192.55 req_src_port=6938 in_uri=sip:finalcut.be out_uri=sip:finalcut.be via_cnt==1".... U 217.136.0.70:53284 -> 82.146.123.252:5060 REGISTER sip:82.146.123.252;user=phone SIP/2.0..Via: SIP/2.0/UDP 192.168.1.13:5060..From: 1300 <sip:1300@82.146.123.252;user=phone>..To: 1300 <sip:1300@82.146.123.252;user= phone>..Call-ID: <mailto:43478@192.168.1.13..Contact> 43478@192.168.1.13..Contact: <sip:1300@192.168.1.13:5060;user=phone>..CSeq: 18961 REGISTER..Max-Forwards: 70..Content-Length: 0..User-Agent: ST280 (REV-A2+ 2.439-1)..Expires: 1800..Authorization: Digest username="1300",realm="finalcut.be",nonce="43490c49911fa90e11538f52e07c19509 087b505",uri="sip:82.146.123.252",algorithm=MD5,r esponse="7ac5e0f3719872d4624a86c14dc677c8".... # U 82.146.123.252:5060 -> 217.136.0.70:53284 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.13:5060;rport=53284;received=217.136.0.70..From: 1300 <sip:1300@82.146.123.252;user=phone>..To: 1300 <sip:1300@82.146.123.252;use r=phone>;tag=59f1565d6938a98ef3fc4707bf9b43aa.1e42..Call-ID: <mailto:43478@192.168.1.13..CSeq> 43478@192.168.1.13..CSeq: 18961 REGISTER..Contact: <sip:1300@217.136.0.70:53284;user=phone>;expires=1800..Serve r: Sip EXpress router (0.9.4 (i386/freebsd))..Content-Length: 0..Warning: 392 82.146.123.252:5060 "Noisy feedback tells: pid=43111 req_src_ip=217.136.0.70 req_src_port=532 84 in_uri=sip:82.146.123.252;user=phone out_uri=sip:82.146.123.252;user=phone via_cnt==1".... #

2 1

[Serusers] test mail
by Sam Lee 10 Oct '05

10 Oct '05

please ignore

1 0

[Serusers] testmail
by Sam Lee 10 Oct '05

10 Oct '05

please ignore

1 0

[Serusers] Mediaproxy must be with SER?
by Ryan Pagquil 10 Oct '05

10 Oct '05

Hi, Should I also install SER for each mediaproxy installation for balancing media traffic? I'm planning to have multiple installation of media proxy located in some part of the world to divert traffic and load balance the media traffic. Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

1 0

[Users] Billing System
by harry gaillac 09 Oct '05

09 Oct '05

Hello, Somebody has tested freeSide billing system !? Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

5 13

[Serusers] SER Codecs/Web Admin
by gorand＠dvvti.com 09 Oct '05

09 Oct '05

Can anyone answer these two questions for me. Is there a web admin which similar to pingtel or Asterisk that allows for easier admin for SER Server? Question 2. What codec formats is supported by ser?

2 1

[Users] Dumb question - can I use Oracle instead of MySQL
by Juan Carlos Castro y Castro 09 Oct '05

09 Oct '05

for the "openser" database? Perhaps via unixODBC? Thx, Juan

2 3

[Serusers] Asigning a Subscriber to a Specific rtpproxy
by Andres 09 Oct '05

09 Oct '05

Hi, Can anybody think of a way to assign a subscriber's call to a specific rtpproxy? The idea is to have one main SER server but multiple rtpproxy servers handling the media stream close to where the subs are. The SER server can be a continent away but the rtpproxy is in the same city as the users. I was thinking of using ACL (groups) somehow but not sure if it is at all possible. Any ideas? Thanks, -- Andres Network Admin http://www.telesip.net

3 4

[Users] tls_require_certificate - sorry!
by Alexander Ph. Lintenhofer 08 Oct '05

08 Oct '05

----------- ---------- --------- | snom 360 | <------ TLS -------> | outbound | <----- TLS -----> | inbound | ----------- server sends cert ---------- mutual AUTH ---------

1 0

[Users] tls_require_certificate - 2nd try
by Alexander Ph. Lintenhofer 08 Oct '05

08 Oct '05

Sorry for the bad ASCII-graphic... Another try.... ----------- ---------- --------- | snom 360 | <------ TLS -------> | outbound | <----- TLS -----> | inbound | ----------- server sends cert ---------- mutual AUTH --------- Philipp

1 0

[Serusers] Multiple SER concept
by Ryan Pagquil 08 Oct '05

08 Oct '05

Hi Guys, How does multiple SER deployed strategically in some part of the world works and what does it do with users that are registering with the same domain for all those SER's? How does calls look like when this was implemented? Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

1 0

[Serusers] Shoot my logic full of holes....
by sip 08 Oct '05

08 Oct '05

Okay... I'm wondering if this would work as a quick hack for checking call blocking (and making it rational). In order to use AVPOps, and not rewrite a module, I've created a table called callblock, which looks exactly like the usr_preferences table (basically an AVP table), but has no primary keys (as I want to be able to add multiple values for the same username/domain/attribute). In my code, I then have a block that looks like: if(avp_db_load("$ruri/username","s:callblkna/callblock")) { if(avp_check("s:callblkna","eq/$from/gi")) { sl_send_reply("603", "Call denied"); break; }; }; if(avp_db_load("$ruri/username","s:callblkvm/callblock")) { if(avp_check("s:callblkvm","eq/$from/gi")) { route(4); break; }; }; If it's a callblkna, it then sends an error reply. If it's a callblkvm, it does a route(4) (which is straight to voicemail). Anyone see a glaring hole in this logic here? It seems to work, but I'd hate to get down the road and discover that I overlooked something glaringly obvious (which I often do). N.

2 1

[Serusers] Can 'avp_check' check whether sourceIP and destinationIP are the same subnet segment?
by Zen Kato 08 Oct '05

08 Oct '05

If 2 UAs exist on the same network segment, e.g., 192.168.0.1 and 192.168.0.2, how to check whether sourceIP and destinationIP are the same subnet segment using avpops module? If 'ser' sits on 192.168.100.1, UA1 can send 'INVITE' then UA2 rings, but when we pick up UA2's handset, these two UAs goes to a panic. I have to reboot two UAs. So, I would like to 'rewritehostport' INVITE packets to 'man-in-the-middle-sip-server', 'asterisk' when UA1 INVITEs UA2. ser asterisk |.1 |.2 192.168.100.0/24 -------------------------------- |.254 router |.254 192.168.0.0/24 --------------------------------- |.1 |.2 UA1 UA2 Regards, Zen

1 0

[Serusers] Convert SIP-T to SIP
by Jorge Cuevas 07 Oct '05

07 Oct '05

Hi, I just found out about SER, and I cannot find info on SIP-T, just that it can proxy SIP-T calls (in an older message). I want to go beyond that and convert from SIP-T to SIP, so I can use it with asterisk or directly to cisco equipment. Does anyone know if SER can be used to receive calls in SIP-T and forward them as SIP ? Can it be done with no coding, little coding, a lot of coding? Any enlightment would be really apreciated. Thanks, Jorge

1 0

[Serusers] OT - VoIP TestDay
by Joao Pereira 07 Oct '05

07 Oct '05

Hello to all, We are installing SER in the Portuguese Universities, and next Monday (17:00 GMT) we will test the connectivity between the Universities, but also from the Universities to the rest of the world. Does someone knows lines around the world for us to call and test the communication? I googled it a lot and I just found one: The weather report in Australia: sip:+61262223590@aarnet.edu.au If someone knows of more lines like this one, or want to help us in the tests by just answering the calls, please contact me (maybe out of the list because this is a bit out of topic ). Thanks Joao Pereira www.fccn.pt

3 2

[Serusers] T38 Support
by Atif Rasheed 07 Oct '05

07 Oct '05

Hello all, can SER with RTP-proxy or Media-Proxy relay T38. If supported, do we need to do some extra configuration or it will work in standard configuation. Thank you, Atif

3 3

[Serusers] SIPATH Internet setup
by gorand＠dvvti.com 07 Oct '05

07 Oct '05

I am trying to set setup SER on SIPATH. I have a WRT54g v 3.0. It Flashed fine, and followed all the instructions up to where I need to wgetthe packages. The problem is that I don't have any connectivity with this device to the internet. I don't know the proper ip configurations. My network is a 172.16.0.0/16 Network This is the ifconfig from the router. What is the proper commands to get this guy to talk to the internet. Your help would be greatly appreciated. Thanks. br0 Link encap:Ethernet HWaddr 00:13:10:8B:D4:5C inet addr:192.168.1.1 Bcast:192.168.1.255 Mask:255.255.255.0 UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:1381 errors:0 dropped:0 overruns:0 frame:0 TX packets:511 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:103850 (101.4 KiB) TX bytes:59305 (57.9 KiB) eth0 Link encap:Ethernet HWaddr 00:13:10:8B:D4:5C UP BROADCAST RUNNING PROMISC MULTICAST MTU:1500 Metric:1 RX packets:5403 errors:0 dropped:0 overruns:0 frame:0 TX packets:378 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:100 RX bytes:756228 (738.5 KiB) TX bytes:86750 (84.7 KiB) Interrupt:5 Base address:0x2000 eth1 Link encap:Ethernet HWaddr 00:13:10:8B:D4:5E UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:1379 errors:0 dropped:0 overruns:0 frame:4094 TX packets:785 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:100 RX bytes:123144 (120.2 KiB) TX bytes:97726 (95.4 KiB) Interrupt:4 Base address:0x1000 lo Link encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 UP LOOPBACK RUNNING MTU:16436 Metric:1 RX packets:1 errors:0 dropped:0 overruns:0 frame:0 TX packets:1 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:76 (76.0 B) TX bytes:76 (76.0 B) vlan0 Link encap:Ethernet HWaddr 00:13:10:8B:D4:5C UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:291 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:0 (0.0 B) TX bytes:35072 (34.2 KiB) vlan1 Link encap:Ethernet HWaddr 00:13:10:8B:D4:5C UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:87 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:0 (0.0 B) TX bytes:51678 (50.4 KiB)

1 0

[Users] RPID and how it relates to CallerID
by Brandon Price 07 Oct '05

07 Oct '05

Can someone please explain to me how RPID works under openser? I have some experience with SIP and Asterisk, but I am having some trouble manipulating the RPID variables under openser. How does RPID relate to Callerid or simple services like private calling, or anonymous call block. How do I tell openser to send all calls from a certain user with the privacy=full flag, I'm assuming this tells the sip proxy or UA on the other end not to present the callerid? Any help greatly appreciated. -Brandon Price-

3 8

[Serusers] Where to set T1 timer (RFC 3261)?
by John Paul Morrison 07 Oct '05

07 Oct '05

Where is the T1 timer set in Ser? The RFC suggests a default of 500 ms for round-trip time. I have an ethereal trace showing Ser re-transmitting after less than 300 ms. Anyway, because of our environment I'd like to set T1 to be 1000 ms. I'm using 0.8.14. thanks

2 3

[Users] Openser not consuming the credentials
by Brandon Price 07 Oct '05

07 Oct '05

I have openser set up to remove previously authorized credentials from the message being processed by the server, using consume_credentials. Yet the authorization credentials are still being passed. Any help is as always appreciated. Thank you. The related openser.cfg portions are posted below... if (!www_authorize("", "subscriber")) { www_challenge("", "1"); break; }; consume_credentials(); save("location"); break; if I then restart openser, make a call and start tethereal I get.... tethereal -V port 5060 | grep username Authorization: Digest username="theusernamewashere",realm="ser1.manhattan.whatever.net",nonce= "43457703fa69c4dd5321a7229ea41102f2200390",uri="sip: 13143212222(a)ser1.manhattan.whatever.net",algorithm=MD5,response="fc57c2b 4897d3d7b8ad887a788bf07f6",qop=aut

3 3

[Serusers] variables in ser.cfg
by Müller Thorsten 07 Oct '05

07 Oct '05

Hi all, is there a possibility to use variables in ser.cfg (ser0.9.0) like ... set myip=192.168.1.11 set myaccount=testuser set mypasswd=testpasswd ... modparam("uac","credential","$myaccount:$myip:$mypasswd"); ... rewritehost($myip); ... Perhaps a solution would be with avpops, but I didn't get it to work in conjunction with dbtext. Has someone an example how to read variables with avpops out of a dbtext table which are not connected to a SIP-message? (I read the README for avpops!!) Any other solutions would be a help!! Would this be possible with openser's pseudo variables or another ser-version? Thanks Thorsten

2 1

[Serusers] Blocking User Agent for a call?
by Ryan Pagquil 07 Oct '05

07 Oct '05

Hi Guys, Is there a way for SER to block calls coming from a specific user-agent? I want to block calls so that uniform useragaent will be used by my users. Any modules that can do this? Thanks, Ryan

1 0

[Serusers] ACK loop in SER - Route missing problem ?
by Lambert, Micheline (Satnet) 07 Oct '05

07 Oct '05

Hi, I'm using SER with Sipura and VoIP GW. When there is no line available from the VoIP GW, the VoIP GW replies with 486 Busy Here. The Sipura receives this message and sends an ACK. The ACK has the IP addr. Of the SER. If it is ok, the ACK does not include the "Route" parameter, is it a problem from the Sipura? This cause the SIP to loop the ACK because there is no Route and with the following code, the packet becomes too big!!! ............ if (!method=="REGISTER") record_route(); # ------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------ # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); t_relay(); break; }; ...................... Same configuration and same trace 192.168.20.155 is the SIP Proxy 192.168.20.105 is the VoIP GW 192.168.20.90 is the Sipura 312 54.858597 192.168.20.90 192.168.20.155 SIP/SDP Request: INVITE sip:3353@192.168.20.155, with session description 313 54.860243 192.168.20.155 192.168.20.90 SIP Status: 407 Proxy Authentication Required 314 54.877006 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 315 54.886801 192.168.20.90 192.168.20.155 SIP/SDP Request: INVITE sip:3353@192.168.20.155, with session description 316 54.889720 192.168.20.155 192.168.20.105 SIP/SDP Request: INVITE sip:3353@192.168.20.105:5060, with session description 318 54.924666 192.168.20.105 192.168.20.155 SIP Status: 100 Trying 319 54.925866 192.168.20.155 192.168.20.90 SIP Status: 100 Trying 320 54.950257 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 321 54.951442 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 323 54.968942 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 440 55.527290 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 441 55.528528 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 442 55.547237 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 627 56.624649 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 639 56.626924 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 649 56.647321 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 912 58.719683 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 913 58.720967 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 916 58.746851 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 1192 62.806510 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 1193 62.807741 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 1194 62.827148 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 1417 66.892115 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 1418 66.893350 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 1419 66.907704 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 1628 68.143481 192.168.20.101 192.168.20.55 SIP Thanks. Micheline _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] Sems IVR Module
by Nicky 07 Oct '05

07 Oct '05

Please help. ----- Original Message ----- From: "Nicky" <nicky(a)caliber.com.sg> To: <serusers(a)lists.iptel.org>; <sems(a)lists.iptel.org> Sent: Wednesday, September 28, 2005 10:38 PM Subject: Sems IVR Module > Hi all, > > I am testing the sems version from the Head and when i uses a SIP Phone to > call into > the IVR module for recording when I am in the middle of recording, the > program detected DTMF > signal and jump out of the script. Why is it so? as I never pressed on any > key? How can I resolve this > problem? Why is the voice signal become the DTMF signal? Is there any > adjustment that > i can do to solve this problem , as my voice goes louder the more DTMF > signal is being detected. > > Please help. > > regards, > nicky >

5 7

[Serusers] mediaproxy and private ip
by Olivier Taylor 07 Oct '05

07 Oct '05

Using the last cfg from onsip.org, i just see that now Uas are registered with their private Ip when using mediaproxy. Mediaproxy seems the best solution for scalability, that's why I want to test it. Any Idea?

2 5

[Serusers] ACK loop in SER
by Lambert, Micheline (Satnet) 07 Oct '05

07 Oct '05

Hi, I'm using SER with Sipura and VoIP GW. When there is no line available from the VoIP GW, the VoIP GW replies with 486 Busy Here. The Sipura receives this message and sends an ACK. The ACK has the IP addr. Of the SER. Is it ok, or it should be the IP address of the VoIP GW. When SER receives the ACK with its IP address 192.168.20.155, there is a loop of ACK in SER......... I attached a trace and this is my configuration. 192.168.20.155 is the SIP Proxy 192.168.20.105 is the VoIP GW 192.168.20.90 is the Sipura 312 54.858597 192.168.20.90 192.168.20.155 SIP/SDP Request: INVITE sip:3353@192.168.20.155, with session description 313 54.860243 192.168.20.155 192.168.20.90 SIP Status: 407 Proxy Authentication Required 314 54.877006 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 315 54.886801 192.168.20.90 192.168.20.155 SIP/SDP Request: INVITE sip:3353@192.168.20.155, with session description 316 54.889720 192.168.20.155 192.168.20.105 SIP/SDP Request: INVITE sip:3353@192.168.20.105:5060, with session description 318 54.924666 192.168.20.105 192.168.20.155 SIP Status: 100 Trying 319 54.925866 192.168.20.155 192.168.20.90 SIP Status: 100 Trying 320 54.950257 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 321 54.951442 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 323 54.968942 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 440 55.527290 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 441 55.528528 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 442 55.547237 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 627 56.624649 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 639 56.626924 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 649 56.647321 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 912 58.719683 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 913 58.720967 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 916 58.746851 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 1192 62.806510 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 1193 62.807741 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 1194 62.827148 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 1417 66.892115 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 1418 66.893350 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 1419 66.907704 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 1628 68.143481 192.168.20.101 192.168.20.55 SIP Request: BYE sip:06@192.168.20.105:5060;maddr=192.168.20.105 1630 68.147489 192.168.20.55 192.168.20.105 SIP Request: BYE sip:06@192.168.20.105:5060;maddr=192.168.20.105 1631 68.174246 192.168.20.105 192.168.20.55 SIP Status: 200 OK 1632 68.176946 192.168.20.55 192.168.20.101 SIP Status: 200 OK Micheline

2 1

[Users] possibility to access dst_uri
by Daniel-Constantin Mierla 06 Oct '05

06 Oct '05

Hello, latest openser version from cvs includes a set of new functions (core and avpops) and pseudo-variable which allow to access and manage the value of dst_uri. The dst_uri field contains the address of the next hop, when the routing does not follow R-URI address. Such cases are Route header driven routing (dst_uri is set after loose_route() to the value of next Route header), contacts behind nat (dst_uri is set to the address of nat after lookup("location")) or dispatcher usage. The dst_uri has higher priority in routing over R-URI but less than explicit parameters in relaying functions (e.g., t_relay_to*()). The field was invisible from the script but it proved in the discussions on the mailing list to be important to access it. For example, to detect whether the caller and callee are behind same nat. Also, it can be checked in case of preloaded Route header to secure relaying to special resources (e.g., pstn gateway). With the new branch_route the value of dst_uri can be checked for each branch. A short summary of what was added: * core functions: - setdsturi("uri") - set the value of dst_uri - resetdsturi() - reset the value of dst_uri - isdsturiset() - test if the value of dst_uri is null * pseudo-variables - $du - value of dst_uri (added long time ago) - $dd - domain of dst_uri - $dp - port of dst_uri - $dP - transport protocol of dst_uri * avpops changes - avp_write() can read the value of dst_uri and write it in an avp - avp_pushto() can write the value of an avp in dst_uri Tutorials and dokuwiki will be updated soon. Daniel

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[Users] openser to only offer ulaw
by Brandon Price 06 Oct '05

06 Oct '05

How can I specify only to offer one codec when negotiating an INVITE. Particularly G-711-ulaw?

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[Serusers] Simple Proxy Server
by Zac Tolley 06 Oct '05

06 Oct '05

I want to build a server which will act purely as a SIP proxy. The idea is I will put the proxy into my DMZ, and on my internal firewall I already have a Cisco 2600, which will convert the sip traffic into something the pbx understands. Can I do this with Ser? Do I need to have MySQL installed, after all it's just a pure proxy? Can I use it to create CDR's? Any help appreciated. Zac Tolley

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[Serusers] prefix
by Rick Thompson 06 Oct '05

06 Oct '05

Hi All Does anyone know how to include a # in the following command Prefix("123#4"); Because the above doesn't work. SER doesn't recognize the # and in fact errors out on it. I've also tried Prefix("123\#\4"); Prefix("123'#'4"); Prefix("123[#]4"); Thanks Richard C. Thompson VP of Technology, CCNA Virtual Communications, LLC o 954-334-4000 x 420 f 954-217-7322

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[Serusers] MOH
by Xavier TRENTIN 06 Oct '05

06 Oct '05

Hi all, does anybody knows if there's a way to do something like music on hold with sems, or if somebody is currently programing a plugin for it? Thanks. B.R. Xavier

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Re: [Serusers] xmlrpc calls
by zkeatts 06 Oct '05

06 Oct '05

Jan, Looking the logs hasn't revealed much telling info, but I'll go ahead and send you what I have anyways. I took the liberty of extracting this from my /var/logs/everything.log I was hoping there would be something more helpful regarding that memory allocation error. I do know, after looking at the source, that the "Can't allocate x-byte block" error is coming from my particular xmlrpc-c library. Interestingly enough I have a generic xmlrpc-c.tar.gz which does indicate very well what version it is on. Looking at the sourceforge page all of them have a particular schema which has left me wondering where I found this package. I am going to assume that it is ver 1.03.02, but I am going to go ahead and update it to see what happens, since there was a new release yesterday. I will let you know if that error persists after the update. Logs: Sep 6 09:23:00 myhost ./ser[5674]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5675]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5676]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5677]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5678]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5679]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5680]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5681]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5682]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5683]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5684]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5685]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5686]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5687]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5688]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5689]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5690]: INFO: signal 15 received Sep 6 09:23:22 myhost ser: read 936833521 from /dev/urandom Sep 6 09:23:22 myhost ser: seeding PRNG with 2062863074 Sep 6 09:23:22 myhost ser: test random number 923067521 Sep 6 09:23:22 myhost ser: WARNING: fix_socket_list: could not rev. resolve 63.77.68.19 Sep 6 09:23:22 myhost ./ser[5753]: Maxfwd module- initializing Sep 6 09:23:22 myhost ./ser[5753]: INFO: udp_init: SO_RCVBUF is initially 110592 Sep 6 09:23:22 myhost ./ser[5753]: INFO: udp_init: SO_RCVBUF is finally 221184 Sep 6 09:23:22 myhost ./ser[5753]: INFO: udp_init: SO_RCVBUF is initially 110592 Sep 6 09:23:22 myhost ./ser[5753]: INFO: udp_init: SO_RCVBUF is finally 221184 Sep 6 09:25:22 myhost ./ser[5770]: Binding 'testuser','sip:testUser@localhost' has expired Sep 6 09:25:48 myhost syslog-ng[1812]: STATS: dropped 0 Sep 6 09:27:23 myhost ./ser[5770]: Binding 'testuser','sip:testUser@localhost' has expired Jan Janak wrote: >This is interesting. Your server sends "Can't allocate x-byte memory >block" but this message does not come from SER, neither can I find it in >my version of xmlrpc-c library. > >What version of xmlrpc-c library do you have ? What is the >OS/distribution ? Could you also send me the log from SER ? > > thanks for helping me to debug this. > > Jan. > > >

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[Serusers] Theory behind a media proxy and NATed UA
by Felipe Louback 06 Oct '05

06 Oct '05

The problem with NAT is because the IP address of the SDP body is a non routeable IP. So how does a media proxy works? I read that a media proxy put his address at the sdp body and both UA talks with the media proxy instead of each other. At the time a NAT user sends an INVITE, there is no corresponding port on the NAT. So How does a media proxy contact the NATed UA? Isn't this the same problem the called part will have to contact the NATed party? I read a couple of papers about it, but in all of them it is said that both ends talk to the media proxy and that is all. No explanation about how things work. I know it works because I am using mediaproxy. If anyone could point me how this works or where I can find a document explaining.... Thanks, Felipe -- Master Student - Electrical Engineering Department Computer Engineering and Telecommunications Research Group Universidade Federal de Minas Gerais - Brazil "For God so loved the world that he gave his one and only Son, that whoever believes in him shall not perish but have eternal life." John 3:16

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[Serusers] to integrate Server SIP
by Ederson 06 Oct '05

06 Oct '05

as to integrate servers SIP My name is Ederson My language is Portuguesse --------------------------------- Novo Yahoo! Messenger com voz: ligações, Yahoo! Avatars, novos emoticons e muito mais. Instale agora!

1 0

[Users] rtpproxy failover
by harry gaillac 06 Oct '05

06 Oct '05

Hello, How can we use ser and rtpproxy for load-balancing and failover ? if ser is able to handle n calls per second how can we calculate the load for rtpproxy ( memory, cpu, ...) in order to send all traffic to one rtpproxy? If this one is down nathelper module is able to send traffic to an other rtpproxy . SER----------------------------- | | | rtpproxy1 rtpproxy2 rtpproxy3 via unix socket What's the best way for SER failover ? ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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