October 2003 - sr-users

[Serusers] loose_route
by Dovid 02 Oct '03

02 Oct '03

My understanding is that loose_route() checks for a Route header field. The standard cfg entry will then relay the SIP request. Let's take the following scenario. I have a PSTN gateway which listens on port 5060 and accepts to that port only from my SIP proxy. Normally the proxy takes care of authentication and authorization and if approved, forwards to the gateway. What if someone were to send an INVITE to my proxy with a header: Route: <sip:phone-number@pstn-gateway>? Would the proxy forward it, and the gateway then set up the call? Thanks in advance. Dovid

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Re: [Serusers] SER account for mySQL
by Jiri Kuthan 02 Oct '03

02 Oct '03

you need to use the ser mysql password. By default, it is 'heslo'. -jiri At 05:42 PM 10/2/2003, Harry Behrens wrote: >Hi everybody, > >we have started using the SER a few weeks ago and have now "upgraded" to authenticating. >We are therefore using mySQL and the ser database. > >The situation we have is: > >- ser is installed and works fine >- the default user (admin/heslo) is installed and can be authenticated and registered >- serctl ul show works fine > >However when I try adding a user with > >serctl add ..... >I am prompted for the MySQL password. I tried entering the password for the user mysql, however the operation failed and told me that it failed to authenticate >ser@localhost. >I have not explicitly set up this user nor did I find any reference to this user in any of the SER files. I obviously do not therefor know the password. I am sure this must be a simple configuration issue I am missing. >Can any of you help? > >Thanx in advance, > > Harry > > >Dr. Harry Behrens >Projektleitung BIB3R >DAI Labor - Technische Universität Berlin >Sekretariat GOR 1-1, Franklinstrasse 28/29, 10587 Berlin > >Fon: +49 30 314 23383 >Fax: +49 30 314 21799 > >Email: harry.behrens(a)dai-labor.de > >http://www.dai-labor.de > > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] Problem strip and t_relay
by Tropico Network 02 Oct '03

02 Oct '03

Dear Ladies and Gentlemen, after installing the version 0.8.11 I am facing a strange problem: In my dialing plan, all numbers starting with a 0 should go to a PSTN-gateway (172.16.0.101). So I have in my ser.cfg the following lines: if (uri=~"^sip:0[0-9]*@") then { strip(1); rewritehostport(172.16.0.101:5060); if (!t_relay()) { bla; bla; }; ); The router is sending the packet right to the GW, but including the leading 0, so that the call cannot be completed. With version 0.8.10-2 it was already working. What do I have to change, or is there maybe a small glitch in the software? with kind regards Juergen Peters Tropico Network General Manager Tel. ++504 440-1461 fax. ++504 443-0660

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[Serusers] re: Problems with accounting - can't get stop records
by jaime.gil＠orange.co.uk 02 Oct '03

02 Oct '03

Steven, This was the config used: # # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ # # simple quick-start config script # # ----------- 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 debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "./modules/mysql/mysql.so" loadmodule "./modules/sl/sl.so" loadmodule "./modules/tm/tm.so" loadmodule "./modules/rr/rr.so" loadmodule "./modules/maxfwd/maxfwd.so" loadmodule "./modules/usrloc/usrloc.so" loadmodule "./modules/registrar/registrar.so" loadmodule "./modules/uri/uri.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "./modules/auth/auth.so" loadmodule "./modules/auth_db/auth_db.so" loadmodule "modules/acc/acc.so" loadmodule "./modules/auth_radius/auth_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("acc", "log_level", 1) modparam("acc", "radius_flag", 1) # ------------------------- request routing logic ------------------- alias=xxxxx.orange.co.uk # 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 (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing # if (loose_route()) { # t_relay(); # break; # }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (method=="REGISTER") { log(1, "REGISTER: Authenticating user\n"); if (!radius_www_authorize("")) { log(1, "REGISTER: challenging user\n"); www_challenge("", "0"); break; }; save("location"); break; }; if (method=="INVITE") { log(1, "INVITE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="MESSAGE") { log(1, "MESSAGE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="BYE" || method=="CANCEL") { log (1, "BYE or CANCEL\n"); setflag(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; if (!t_relay()) { sl_reply_error(); break; }; } "Steven R. Bunin" <steve(a)solaas.com> on 29/09/2003 19:01:55 To: Jaime GIL/EN/HTLUK@HTLUK cc: serusers(a)lists.iptel.org Subject: re: Problems with accounting - can't get stop records Jaime, Thanks for the resposne.. I have my loose_route shut off. I think the problem may have to do with an error message I am receiving which is "ERROR: reply cannot be parsed" from my stderr. Do you have a working version of a script that works with 0.8.11 and gets stop records I could look at? It should give me a hand.. I know I am so close to solving this but it is a struggle. Steve jaime.gil(a)orange.co.uk wrote: > Steven, > > I remember having the same problem. In my case, the problem seemed to be in the > loose_route processing. I don't know what it has to do with the accounting > module, but commenting out this part made the trick in my case. > > Jaime > > "Steven R. Bunin" <steve(a)solaas.com> on 26/09/2003 17:11:41 > > To: serusers(a)lists.iptel.org > cc: (bcc: Jaime GIL/EN/HTLUK) > > Subject: [Serusers] Re: Serusers Digest, Vol 5, Issue 63 > > Hi all, > > I have successfully gotten radius authentication working and I started getting > Radius Start records for accounting but I am not sure what I am doing wrong in > regards to getting radius Stop records. > > Below is the area I believe has the most affect on Radius Acccounting from my > Log File. Any suggestions would be appreciated and if it would help to see the > full Config file I will send it as well. > > record_route(); > > if (method=="REGISTER") { > > # Uncomment this if you want to use digest authentication > if (!radius_www_authorize("")) { > www_challenge("", "0"); > break; > }; > > save("location"); > break; > }; > > if (method =="INVITE") > { > log(1,"INVITE\n"); > setflag(1); > }; > if (method=="MESSAGE") { > log(1,"MESSAGE\n"); > setflag(1); > }; > if (method=="BYE"){ > log (1, "BYE or CANCEL\n"); > setflag(1); > }; > if (method=="CANCEL"){ > log (1, "BYE or CANCEL\n"); > setflag(1); > }; > > Thanks in advance, > > Steve > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > ------------------------------------------------------------------------ > Name: att1.eml > att1.eml Type: Internet E-Mail Message (message/rfc822) > Encoding: base64 > > ------------------------------------------------------------------------ > > ******************************************************************************* > Important. > Confidentiality: This communication is intended for the above-named person and > may be confidential and/or legally privileged. Any opinions expressed in this > communication are not necessarily those of the company. If it has come to you > in error you must take no action based on it, nor must you copy or show it to > anyone; please delete/destroy and inform the sender immediately. > > Monitoring/Viruses > Orange may monitor all incoming and outgoing emails in line with current > legislation. Although we have taken steps to ensure that this email and > attachments are free from any virus, we advise that in keeping with good > computing practice the recipient should ensure they are actually virus free. > > Orange PCS Limited is a subsidiary of Orange SA and is registered in England No > 2178917, with its address at St James Court, Great Park Road, Almondsbury Park, > Bradley Stoke, Bristol BS32 4QJ. > ******************************************************************************* -- Steven R. Bunin - Managing Partner SOLAAS LLC 10 East 39th Street Suite 1125 New York, NY 10016 (+001) 212-532-6700 Cellular: 646-739-7000 Fax (+001) 212-532-6776 http://www.solaas.com -- This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. ******************************************************************************* Important. Confidentiality: This communication is intended for the above-named person and may be confidential and/or legally privileged. Any opinions expressed in this communication are not necessarily those of the company. If it has come to you in error you must take no action based on it, nor must you copy or show it to anyone; please delete/destroy and inform the sender immediately. Monitoring/Viruses Orange may monitor all incoming and outgoing emails in line with current legislation. Although we have taken steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. Orange PCS Limited is a subsidiary of Orange SA and is registered in England No 2178917, with its address at St James Court, Great Park Road, Almondsbury Park, Bradley Stoke, Bristol BS32 4QJ. *******************************************************************************

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Re: [Serusers] Re: Problems with accounting - can't get stop records
by lgfausak＠august.net 02 Oct '03

02 Oct '03

Wow, I have been stung by this as well. Very good call, thanks :-) ---greg > >Indeed, Daniel hit the nail, thanks. > >Turning off record-routing as someone proposed previously is not advicable. >Not only subsequent requests, such as BYE, skip the proxy server. Getting >the server out of loop may cause for example failed signaling if both >parties use different signaling protocols and can't speak directly to >each other. > >-jiri > >At 04:43 PM 9/30/2003, Daniel-Constantin Mierla wrote: >>Hello, >>the BYEs were not accounted because the next piece of configuration file: >> >> if (loose_route()) { >> t_relay(); >> break; >> }; >> >> >>BYEs are subsequent requests within a dialog and they include at least a Route header and loose_route() returns true if no error occurs during processing the top most. In the above piece of configuration after the loose routing processing the request is forwarded to the next hope -- the one from Route header -- without setting any flag for accounting. You have to set it before t_relay(), e.g.,: >> >> if (loose_route()) { >> if (method=="BYE" | ...){ >> log (1, "Accounting flag on\n"); >> setflag(1); >> }; >> t_relay(); >> break; >> }; >> >> >>Daniel >> >> >> >>On 9/30/2003 4:19 PM, Steven R. Bunin wrote: >> >>>Jaime, >>> >>>Thanks for the config. I managed to get it working yesterday by playing with it >>>quite a bit. What I ended up doing was moving my if (method=="bye") {} statements >>>to be the first thing after route { . I don't know what this worked but it works >>>even with loose_route() enabled. If you want a copy of my config just let me know >>>and I will send it out. >>> >>>Steve >>> >>>jaime.gil(a)orange.co.uk wrote: >>> >>> >>> >>>>Steven, >>>> >>>>This was the config used: >>>> >>>># >>>># $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ >>>># >>>># simple quick-start config script >>>># >>>> >>>># ----------- 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 >>>>debug=7 >>>>fork=no >>>>log_stderror=yes >>>>*/ >>>> >>>>check_via=no # (cmd. line: -v) >>>>dns=no # (cmd. line: -r) >>>>rev_dns=no # (cmd. line: -R) >>>>port=5060 >>>>children=4 >>>>fifo="/tmp/ser_fifo" >>>> >>>># ------------------ module loading ---------------------------------- >>>> >>>># Uncomment this if you want to use SQL database >>>>loadmodule "./modules/mysql/mysql.so" >>>> >>>>loadmodule "./modules/sl/sl.so" >>>>loadmodule "./modules/tm/tm.so" >>>>loadmodule "./modules/rr/rr.so" >>>>loadmodule "./modules/maxfwd/maxfwd.so" >>>>loadmodule "./modules/usrloc/usrloc.so" >>>>loadmodule "./modules/registrar/registrar.so" >>>>loadmodule "./modules/uri/uri.so" >>>> >>>># Uncomment this if you want digest authentication >>>># mysql.so must be loaded ! >>>>loadmodule "./modules/auth/auth.so" >>>>loadmodule "./modules/auth_db/auth_db.so" >>>> >>>>loadmodule "modules/acc/acc.so" >>>> >>>>loadmodule "./modules/auth_radius/auth_radius.so" >>>> >>>># ----------------- setting module-specific parameters --------------- >>>> >>>># -- usrloc params -- >>>> >>>>modparam("usrloc", "db_mode", 2) >>>> >>>>modparam("auth_radius", "radius_config", >>>>"/usr/local/etc/radiusclient/radiusclient.conf") >>>> >>>># -- rr params -- >>>># add value to ;lr param to make some broken UAs happy >>>>modparam("rr", "enable_full_lr", 1) >>>> >>>>modparam("acc", "log_level", 1) >>>>modparam("acc", "radius_flag", 1) >>>> >>>># ------------------------- request routing logic ------------------- >>>> >>>>alias=xxxxx.orange.co.uk >>>> >>>># 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 (len_gt( max_len )) { >>>> sl_send_reply("513", "Message too big"); >>>> break; >>>> }; >>>> >>>> # we record-route all messages -- to make sure that >>>> # subsequent messages will go through our proxy; that's >>>> # particularly good if upstream and downstream entities >>>> # use different transport protocol >>>> record_route(); >>>> # loose-route processing >>>># if (loose_route()) { >>>># t_relay(); >>>># break; >>>># }; >>>> >>>> # if the request is for other domain use UsrLoc >>>> # (in case, it does not work, use the following command >>>> # with proper names and addresses in it) >>>> >>>> if (method=="REGISTER") { >>>> >>>> log(1, "REGISTER: Authenticating user\n"); >>>> >>>> if (!radius_www_authorize("")) { >>>> log(1, "REGISTER: challenging user\n"); >>>> www_challenge("", "0"); >>>> break; >>>> }; >>>> save("location"); >>>> break; >>>> }; >>>> >>>> if (method=="INVITE") { >>>> >>>> log(1, "INVITE\n"); >>>> >>>> setflag(1); /* set for accounting (the same value as in log_flag!) */ >>>> }; >>>> >>>> if (method=="MESSAGE") { >>>> log(1, "MESSAGE\n"); >>>> setflag(1); /* set for accounting (the same value as in log_flag!) */ >>>> }; >>>> >>>> if (method=="BYE" || method=="CANCEL") { >>>> log (1, "BYE or CANCEL\n"); >>>> setflag(1); >>>> }; >>>> >>>> # native SIP destinations are handled using our USRLOC DB >>>> if (!lookup("location")) { >>>> sl_send_reply("404", "Not Found"); >>>> break; >>>> }; >>>> if (!t_relay()) { >>>> sl_reply_error(); >>>> break; >>>> }; >>>> >>>>} >>>> >>>>"Steven R. Bunin" <steve(a)solaas.com> on 29/09/2003 19:01:55 >>>> >>>>To: Jaime GIL/EN/HTLUK@HTLUK >>>>cc: serusers(a)lists.iptel.org >>>> >>>>Subject: re: Problems with accounting - can't get stop records >>>> >>>>Jaime, >>>> >>>>Thanks for the resposne.. I have my loose_route shut off. I think the problem >>>>may >>>>have to do with an error message I am receiving which is "ERROR: reply cannot be >>>>parsed" from my stderr. >>>> >>>>Do you have a working version of a script that works with 0.8.11 and gets stop >>>>records I could look at? It should give me a hand.. I know I am so close to >>>>solving >>>>this but it is a struggle. >>>> >>>>Steve >>>> >>>>jaime.gil(a)orange.co.uk wrote: >>>> >>>> >>>> >>>>>Steven, >>>>> >>>>>I remember having the same problem. In my case, the problem seemed to be in >>>>> >>>>the >>>> >>>> >>>>>loose_route processing. I don't know what it has to do with the accounting >>>>>module, but commenting out this part made the trick in my case. >>>>> >>>>>Jaime >>>>> >>>>>"Steven R. Bunin" <steve(a)solaas.com> on 26/09/2003 17:11:41 >>>>> >>>>>To: serusers(a)lists.iptel.org >>>>>cc: (bcc: Jaime GIL/EN/HTLUK) >>>>> >>>>>Subject: [Serusers] Re: Serusers Digest, Vol 5, Issue 63 >>>>> >>>>>Hi all, >>>>> >>>>>I have successfully gotten radius authentication working and I started getting >>>>>Radius Start records for accounting but I am not sure what I am doing wrong in >>>>>regards to getting radius Stop records. >>>>> >>>>>Below is the area I believe has the most affect on Radius Acccounting from my >>>>>Log File. Any suggestions would be appreciated and if it would help to see the >>>>>full Config file I will send it as well. >>>>> >>>>>record_route(); >>>>> >>>>> if (method=="REGISTER") { >>>>> >>>>># Uncomment this if you want to use digest authentication >>>>> if (!radius_www_authorize("")) { >>>>> www_challenge("", "0"); >>>>> break; >>>>> }; >>>>> >>>>> save("location"); >>>>> break; >>>>> }; >>>>> >>>>>if (method =="INVITE") >>>>>{ >>>>>log(1,"INVITE\n"); >>>>>setflag(1); >>>>>}; >>>>>if (method=="MESSAGE") { >>>>>log(1,"MESSAGE\n"); >>>>>setflag(1); >>>>>}; >>>>>if (method=="BYE"){ >>>>>log (1, "BYE or CANCEL\n"); >>>>>setflag(1); >>>>>}; >>>>>if (method=="CANCEL"){ >>>>>log (1, "BYE or CANCEL\n"); >>>>>setflag(1); >>>>>}; >>>>> >>>>>Thanks in advance, >>>>> >>>>>Steve >>>>> >>>>>_______________________________________________ >>>>>Serusers mailing list >>>>>serusers(a)lists.iptel.org >>>>>http://lists.iptel.org/mailman/listinfo/serusers >>>>> >>>>> ------------------------------------------------------------------------ >>>>> Name: att1.eml >>>>> att1.eml Type: Internet E-Mail Message (message/rfc822) >>>>> Encoding: base64 >>>>> >>>>> ------------------------------------------------------------------------ >>>>> >>>>> >>>>> >>>>******************************************************************************* >>>> >>>> >>>>>Important. >>>>>Confidentiality: This communication is intended for the above-named person and >>>>>may be confidential and/or legally privileged. Any opinions expressed in this >>>>>communication are not necessarily those of the company. If it has come to you >>>>>in error you must take no action based on it, nor must you copy or show it to >>>>>anyone; please delete/destroy and inform the sender immediately. >>>>> >>>>>Monitoring/Viruses >>>>>Orange may monitor all incoming and outgoing emails in line with current >>>>>legislation. Although we have taken steps to ensure that this email and >>>>>attachments are free from any virus, we advise that in keeping with good >>>>>computing practice the recipient should ensure they are actually virus free. >>>>> >>>>>Orange PCS Limited is a subsidiary of Orange SA and is registered in England >>>>> >>>>No >>>> >>>> >>>>>2178917, with its address at St James Court, Great Park Road, Almondsbury >>>>> >>>>Park, >>>> >>>> >>>>>Bradley Stoke, Bristol BS32 4QJ. >>>>> >>>>> >>>>******************************************************************************* >>>> >>>>-- >>>>Steven R. Bunin - Managing Partner >>>> >>>>SOLAAS LLC >>>>10 East 39th Street >>>>Suite 1125 >>>>New York, NY 10016 >>>>(+001) 212-532-6700 >>>>Cellular: 646-739-7000 >>>>Fax (+001) 212-532-6776 >>>> >>>>http://www.solaas.com >>>> >>>>-- >>>> >>>>This e-mail may contain confidential and/or privileged information. If you are >>>>not >>>>the intended recipient (or have received this e-mail in error) please notify the >>>>sender immediately and destroy this e-mail. Any unauthorized copying, disclosure >>>>or >>>>distribution of the material in this e-mail is strictly forbidden. >>>> >>>> ------------------------------------------------------------------------ >>>> Name: att1.eml >>>> att1.eml Type: Internet E-Mail Message (message/rfc822) >>>> Encoding: base64 >>>> >>>> ------------------------------------------------------------------------ >>>> >>>>******************************************************************************* >>>>Important. >>>>Confidentiality: This communication is intended for the above-named person and >>>>may be confidential and/or legally privileged. Any opinions expressed in this >>>>communication are not necessarily those of the company. If it has come to you >>>>in error you must take no action based on it, nor must you copy or show it to >>>>anyone; please delete/destroy and inform the sender immediately. >>>> >>>>Monitoring/Viruses >>>>Orange may monitor all incoming and outgoing emails in line with current >>>>legislation. Although we have taken steps to ensure that this email and >>>>attachments are free from any virus, we advise that in keeping with good >>>>computing practice the recipient should ensure they are actually virus free. >>>> >>>>Orange PCS Limited is a subsidiary of Orange SA and is registered in England No >>>>2178917, with its address at St James Court, Great Park Road, Almondsbury Park, >>>>Bradley Stoke, Bristol BS32 4QJ. >>>>******************************************************************************* >>>> >>> >>>-- >>>Steven R. Bunin - Managing Partner >>> >>>SOLAAS LLC >>>10 East 39th Street >>>Suite 1125 >>>New York, NY 10016 >>>(+001) 212-532-6700 >>>Cellular: 646-739-7000 >>>Fax (+001) 212-532-6776 >>> >>>http://www.solaas.com >>> >>>-- >>> >>>This e-mail may contain confidential and/or privileged information. If you are not >>>the intended recipient (or have received this e-mail in error) please notify the >>>sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or >>>distribution of the material in this e-mail is strictly forbidden. >>> >>> >>>_______________________________________________ >>>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 > >-- >Jiri Kuthan http://iptel.org/~jiri/ > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers >

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Re: [Serusers] RE: ser and enum (bind) issue
by Jiri Kuthan 02 Oct '03

02 Oct '03

See more examples at our FAQ: http://iptel.org/~faqomatic/fom-serve/cache/83.html -jiri At 11:00 PM 9/30/2003, Marc Archer wrote: >My issue got sorted on the bind mailing list. FYI, I needed a zone file like >the following to get the query to work: > >localhost A 127.0.0.1 >$ORIGIN 8.6.4.e164.arpa. -- Jiri Kuthan http://iptel.org/~jiri/

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RE: [Serusers] caller-id with raius using sip-rpid
by Steve Dolloff 02 Oct '03

02 Oct '03

I want to retrieve the caller-id of a user from radius when they register and use it when I redirect an invite request to the voicemail system so that the voicemail system can route the call based on caller-id. I also want to be able to send it to the sip gateway when I am routing calls to the pstn. I have this in my config currently and it doesn't appear to set the caller-id in the invite message. rewritehostport("209.242.10.153:5061"); append_rpid_hf(); t_relay(); Subject: Re: [Serusers] caller-id with raius using sip-rpid append_rpid_hf has no parameters. Version with 2 parameters is in unstable branch in the CVS only. What exactly do you want to do ? Jan. On 02-10 11:26, Steve Dolloff wrote: > I found this example in a previous Serusers message. > > append_rpid_hf("<sip:+", > > "@zettou.net>;party=calling;id-type=subscriber;privacy=off;screen=yes"); > > I tried calling append_rpid_hf();, but it does nothing. > > I also tried > append_rpid_hf("party=calling;id-type=subscriber;privacy=off;screen=yes" > ); but I get an unknown command presumably because I am not using the > right number of parameters. > > I only want to modify the calling-id info. > > Can anyone provide an example? > > -----Original Message----- > From: Jan Janak [mailto:jan@iptel.org] > Sent: Thursday, October 02, 2003 11:09 AM > To: Steve Dolloff > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] caller-id with raius using sip-rpid > > If you want to add Remote-Party-ID header field then you have to call > append_rpid_hf function in your script. > > Jan. > > On 02-10 11:07, Steve Dolloff wrote: > > OK, I am setting up a Voicemail system (using asterisk) and I am > > currently doing a rewritehostport(ip:port) and then trelay() to send > it > > to the voicemail system if an invite fails. > > > > Should I change something? See my ser.cfg and output from the call to > > the vm. > > > > Here is the code from ser.cfg > > > > > rewritehostport("219.242.10.153:5061"); > > t_relay(); > > > > Here is the sip info from asterisk. > > > > INVITE sip:200@219.242.10.153:5061;user=phone SIP/2.0 > > Max-Forwards: 10 > > Record-Route: <sip:200@219.242.10.153;ftag=2236658534;lr=on> > > Via: SIP/2.0/UDP 219.242.10.153;branch=z9hG4bK13de.31901234.0 > > Via: SIP/2.0/UDP 226.145.234.113:5060;rport=5060 > > From: sip:test@voip2.test.net;tag=2236658534 > > To: <sip:200@voip2.test.net;user=phone> > > Call-ID: 2218108971(a)226.145.234.113 > > CSeq: 1 INVITE > > Contact: <sip:test@226.145.234.113:5060;transport=udp> > > User-Agent: Cisco ATA 186 v2.16.2 ata18x (030909a) > > Expires: 300 > > Content-Length: 271 > > Content-Type: application/sdp > > > > > > > > Subject: Re: [Serusers] caller-id with raius using sip-rpid > > > > Remote-Party-ID header field is inserted into forwarded requests, not > > responses. > > > > Jan. > > > > On 02-10 10:53, Steve Dolloff wrote: > > > Ser doesn't appear to be passing the Caller-id to the ata at auth or > I > > > am doing something wrong. Can anyone point me in the right > direction? > > > > > > Thanks, > > > > > > Stephen > > > > > > I have the following entry in my freeradius users file. > > > > > > test(a)voip2.test.net Auth-Type := Digest, User-Password == "test" > > > Reply-Message = "Hello, test with digest", Sip-Rpid = > > > "8472222222" > > > > > > When I run a radclient test I get the correct info.. > > > > > > radclient -f digest.test 219.242.10.153:1812 auth testing > > > Received response ID 134, code 2, length = 57 > > > Reply-Message = "Hello, test with digest" > > > Sip-Rpid = "8472222222" > > > > > > This is the output from ngrep port 5060 > > > > > > U 216.222.234.113:5060 -> 219.242.10.153:5060 > > > REGISTER sip:voip2.test.net SIP/2.0..Via: SIP/2.0/UDP > > > 216.222.234.113:5060.. > > > From: sip:test@voip2.test.net;tag=277486986..To: > > > sip:test@voip2.test.net..Cal > > > l-ID: 2687235586@216.222.234.113..CSeq: 2 REGISTER..Contact: > > > <sip:test@216. > > > 222.234.113:5060;transport=udp>;expires=120..User-Agent: Cisco ATA > > 186 > > > v2. > > > 16.2 ata18x (030909a)..Authorization: Digest > > > > > > username="test",realm="voip2.test.net",nonce="3f7c49b40e81572eff05bdf50c > > > 867a85bbb0da3c",uri="sip:voip2.test > > > .net",response="1684410c130d6faa9a3c573365f36ab6"..Content-Length: > > > 0.... > > > # > > > U 219.242.10.153:5060 -> 216.222.234.113:5060 > > > SIP/2.0 200 OK..Via: SIP/2.0/UDP > > > 216.222.234.113:5060;rport=5060..From: sip > > > :test@voip2.test.net;tag=277486986..To: > > > sip:test@voip2.test.net;tag=b27e1a1d3 > > > 3761e85846fc98f5f3a7e58.13c2..Call-ID: > > > 2687235586@216.222.234.113..CSeq: 2 > > > REGISTER..Contact: > > > <sip:test@216.222.234.113:5060;transport=udp>;q=0.00;exp > > > ires=120..Server: Sip EXpress router (0.8.12dev-t16 > > > (i386/linux))..Content- > > > Length: 0..Warning: 392 219.242.10.153:5060 "Noisy feedback tells: > > > pid=121 > > > 59 req_src_ip=216.222.234.113 req_src_port=5060 > > > in_uri=sip:voip2.test.net ou > > > t_uri=sip:voip2.test.net via_cnt==1".... > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] caller-id with raius using sip-rpid
by Steve Dolloff 02 Oct '03

02 Oct '03

I found this example in a previous Serusers message. append_rpid_hf("<sip:+", "@zettou.net>;party=calling;id-type=subscriber;privacy=off;screen=yes"); I tried calling append_rpid_hf();, but it does nothing. I also tried append_rpid_hf("party=calling;id-type=subscriber;privacy=off;screen=yes" ); but I get an unknown command presumably because I am not using the right number of parameters. I only want to modify the calling-id info. Can anyone provide an example? -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Thursday, October 02, 2003 11:09 AM To: Steve Dolloff Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] caller-id with raius using sip-rpid If you want to add Remote-Party-ID header field then you have to call append_rpid_hf function in your script. Jan. On 02-10 11:07, Steve Dolloff wrote: > OK, I am setting up a Voicemail system (using asterisk) and I am > currently doing a rewritehostport(ip:port) and then trelay() to send it > to the voicemail system if an invite fails. > > Should I change something? See my ser.cfg and output from the call to > the vm. > > Here is the code from ser.cfg > > rewritehostport("219.242.10.153:5061"); > t_relay(); > > Here is the sip info from asterisk. > > INVITE sip:200@219.242.10.153:5061;user=phone SIP/2.0 > Max-Forwards: 10 > Record-Route: <sip:200@219.242.10.153;ftag=2236658534;lr=on> > Via: SIP/2.0/UDP 219.242.10.153;branch=z9hG4bK13de.31901234.0 > Via: SIP/2.0/UDP 226.145.234.113:5060;rport=5060 > From: sip:test@voip2.test.net;tag=2236658534 > To: <sip:200@voip2.test.net;user=phone> > Call-ID: 2218108971(a)226.145.234.113 > CSeq: 1 INVITE > Contact: <sip:test@226.145.234.113:5060;transport=udp> > User-Agent: Cisco ATA 186 v2.16.2 ata18x (030909a) > Expires: 300 > Content-Length: 271 > Content-Type: application/sdp > > > > Subject: Re: [Serusers] caller-id with raius using sip-rpid > > Remote-Party-ID header field is inserted into forwarded requests, not > responses. > > Jan. > > On 02-10 10:53, Steve Dolloff wrote: > > Ser doesn't appear to be passing the Caller-id to the ata at auth or I > > am doing something wrong. Can anyone point me in the right direction? > > > > Thanks, > > > > Stephen > > > > I have the following entry in my freeradius users file. > > > > test(a)voip2.test.net Auth-Type := Digest, User-Password == "test" > > Reply-Message = "Hello, test with digest", Sip-Rpid = > > "8472222222" > > > > When I run a radclient test I get the correct info.. > > > > radclient -f digest.test 219.242.10.153:1812 auth testing > > Received response ID 134, code 2, length = 57 > > Reply-Message = "Hello, test with digest" > > Sip-Rpid = "8472222222" > > > > This is the output from ngrep port 5060 > > > > U 216.222.234.113:5060 -> 219.242.10.153:5060 > > REGISTER sip:voip2.test.net SIP/2.0..Via: SIP/2.0/UDP > > 216.222.234.113:5060.. > > From: sip:test@voip2.test.net;tag=277486986..To: > > sip:test@voip2.test.net..Cal > > l-ID: 2687235586@216.222.234.113..CSeq: 2 REGISTER..Contact: > > <sip:test@216. > > 222.234.113:5060;transport=udp>;expires=120..User-Agent: Cisco ATA > 186 > > v2. > > 16.2 ata18x (030909a)..Authorization: Digest > > > username="test",realm="voip2.test.net",nonce="3f7c49b40e81572eff05bdf50c > > 867a85bbb0da3c",uri="sip:voip2.test > > .net",response="1684410c130d6faa9a3c573365f36ab6"..Content-Length: > > 0.... > > # > > U 219.242.10.153:5060 -> 216.222.234.113:5060 > > SIP/2.0 200 OK..Via: SIP/2.0/UDP > > 216.222.234.113:5060;rport=5060..From: sip > > :test@voip2.test.net;tag=277486986..To: > > sip:test@voip2.test.net;tag=b27e1a1d3 > > 3761e85846fc98f5f3a7e58.13c2..Call-ID: > > 2687235586@216.222.234.113..CSeq: 2 > > REGISTER..Contact: > > <sip:test@216.222.234.113:5060;transport=udp>;q=0.00;exp > > ires=120..Server: Sip EXpress router (0.8.12dev-t16 > > (i386/linux))..Content- > > Length: 0..Warning: 392 219.242.10.153:5060 "Noisy feedback tells: > > pid=121 > > 59 req_src_ip=216.222.234.113 req_src_port=5060 > > in_uri=sip:voip2.test.net ou > > t_uri=sip:voip2.test.net via_cnt==1".... > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] caller-id with raius using sip-rpid
by Steve Dolloff 02 Oct '03

02 Oct '03

OK, I am setting up a Voicemail system (using asterisk) and I am currently doing a rewritehostport(ip:port) and then trelay() to send it to the voicemail system if an invite fails. Should I change something? See my ser.cfg and output from the call to the vm. Here is the code from ser.cfg rewritehostport("219.242.10.153:5061"); t_relay(); Here is the sip info from asterisk. INVITE sip:200@219.242.10.153:5061;user=phone SIP/2.0 Max-Forwards: 10 Record-Route: <sip:200@219.242.10.153;ftag=2236658534;lr=on> Via: SIP/2.0/UDP 219.242.10.153;branch=z9hG4bK13de.31901234.0 Via: SIP/2.0/UDP 226.145.234.113:5060;rport=5060 From: sip:test@voip2.test.net;tag=2236658534 To: <sip:200@voip2.test.net;user=phone> Call-ID: 2218108971(a)226.145.234.113 CSeq: 1 INVITE Contact: <sip:test@226.145.234.113:5060;transport=udp> User-Agent: Cisco ATA 186 v2.16.2 ata18x (030909a) Expires: 300 Content-Length: 271 Content-Type: application/sdp Subject: Re: [Serusers] caller-id with raius using sip-rpid Remote-Party-ID header field is inserted into forwarded requests, not responses. Jan. On 02-10 10:53, Steve Dolloff wrote: > Ser doesn't appear to be passing the Caller-id to the ata at auth or I > am doing something wrong. Can anyone point me in the right direction? > > Thanks, > > Stephen > > I have the following entry in my freeradius users file. > > test(a)voip2.test.net Auth-Type := Digest, User-Password == "test" > Reply-Message = "Hello, test with digest", Sip-Rpid = > "8472222222" > > When I run a radclient test I get the correct info.. > > radclient -f digest.test 219.242.10.153:1812 auth testing > Received response ID 134, code 2, length = 57 > Reply-Message = "Hello, test with digest" > Sip-Rpid = "8472222222" > > This is the output from ngrep port 5060 > > U 216.222.234.113:5060 -> 219.242.10.153:5060 > REGISTER sip:voip2.test.net SIP/2.0..Via: SIP/2.0/UDP > 216.222.234.113:5060.. > From: sip:test@voip2.test.net;tag=277486986..To: > sip:test@voip2.test.net..Cal > l-ID: 2687235586@216.222.234.113..CSeq: 2 REGISTER..Contact: > <sip:test@216. > 222.234.113:5060;transport=udp>;expires=120..User-Agent: Cisco ATA 186 > v2. > 16.2 ata18x (030909a)..Authorization: Digest > username="test",realm="voip2.test.net",nonce="3f7c49b40e81572eff05bdf50c > 867a85bbb0da3c",uri="sip:voip2.test > .net",response="1684410c130d6faa9a3c573365f36ab6"..Content-Length: > 0.... > # > U 219.242.10.153:5060 -> 216.222.234.113:5060 > SIP/2.0 200 OK..Via: SIP/2.0/UDP > 216.222.234.113:5060;rport=5060..From: sip > :test@voip2.test.net;tag=277486986..To: > sip:test@voip2.test.net;tag=b27e1a1d3 > 3761e85846fc98f5f3a7e58.13c2..Call-ID: > 2687235586@216.222.234.113..CSeq: 2 > REGISTER..Contact: > <sip:test@216.222.234.113:5060;transport=udp>;q=0.00;exp > ires=120..Server: Sip EXpress router (0.8.12dev-t16 > (i386/linux))..Content- > Length: 0..Warning: 392 219.242.10.153:5060 "Noisy feedback tells: > pid=121 > 59 req_src_ip=216.222.234.113 req_src_port=5060 > in_uri=sip:voip2.test.net ou > t_uri=sip:voip2.test.net via_cnt==1".... > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] SER account for mySQL
by Harry Behrens 02 Oct '03

02 Oct '03

Hi everybody, we have started using the SER a few weeks ago and have now "upgraded" to authenticating. We are therefore using mySQL and the ser database. The situation we have is: - ser is installed and works fine - the default user (admin/heslo) is installed and can be authenticated and registered - serctl ul show works fine However when I try adding a user with serctl add ..... I am prompted for the MySQL password. I tried entering the password for the user mysql, however the operation failed and told me that it failed to authenticate ser@localhost. I have not explicitly set up this user nor did I find any reference to this user in any of the SER files. I obviously do not therefor know the password. I am sure this must be a simple configuration issue I am missing. Can any of you help? Thanx in advance, Harry Dr. Harry Behrens Projektleitung BIB3R DAI Labor - Technische Universität Berlin Sekretariat GOR 1-1, Franklinstrasse 28/29, 10587 Berlin Fon: +49 30 314 23383 Fax: +49 30 314 21799 Email: harry.behrens(a)dai-labor.de http://www.dai-labor.de

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[Serusers] caller-id with raius using sip-rpid
by Steve Dolloff 02 Oct '03

02 Oct '03

Ser doesn't appear to be passing the Caller-id to the ata at auth or I am doing something wrong. Can anyone point me in the right direction? Thanks, Stephen I have the following entry in my freeradius users file. test(a)voip2.test.net Auth-Type := Digest, User-Password == "test" Reply-Message = "Hello, test with digest", Sip-Rpid = "8472222222" When I run a radclient test I get the correct info.. radclient -f digest.test 219.242.10.153:1812 auth testing Received response ID 134, code 2, length = 57 Reply-Message = "Hello, test with digest" Sip-Rpid = "8472222222" This is the output from ngrep port 5060 U 216.222.234.113:5060 -> 219.242.10.153:5060 REGISTER sip:voip2.test.net SIP/2.0..Via: SIP/2.0/UDP 216.222.234.113:5060.. From: sip:test@voip2.test.net;tag=277486986..To: sip:test@voip2.test.net..Cal l-ID: 2687235586@216.222.234.113..CSeq: 2 REGISTER..Contact: <sip:test@216. 222.234.113:5060;transport=udp>;expires=120..User-Agent: Cisco ATA 186 v2. 16.2 ata18x (030909a)..Authorization: Digest username="test",realm="voip2.test.net",nonce="3f7c49b40e81572eff05bdf50c 867a85bbb0da3c",uri="sip:voip2.test .net",response="1684410c130d6faa9a3c573365f36ab6"..Content-Length: 0.... # U 219.242.10.153:5060 -> 216.222.234.113:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 216.222.234.113:5060;rport=5060..From: sip :test@voip2.test.net;tag=277486986..To: sip:test@voip2.test.net;tag=b27e1a1d3 3761e85846fc98f5f3a7e58.13c2..Call-ID: 2687235586@216.222.234.113..CSeq: 2 REGISTER..Contact: <sip:test@216.222.234.113:5060;transport=udp>;q=0.00;exp ires=120..Server: Sip EXpress router (0.8.12dev-t16 (i386/linux))..Content- Length: 0..Warning: 392 219.242.10.153:5060 "Noisy feedback tells: pid=121 59 req_src_ip=216.222.234.113 req_src_port=5060 in_uri=sip:voip2.test.net ou t_uri=sip:voip2.test.net via_cnt==1"....

2 1

[Serusers] Windows Messenger
by Annie Sasidar 02 Oct '03

02 Oct '03

Daniel, Yes the windows messenger (5.0) that i am using supports SIP. So now how do i go about it. I did configure the messenger to include SIP address. I want to check this user record on the server. How do i do that? Should i send my ser.cfg to you just to check my settings. Please let me know if it is feasible. Thanks, Annie

2 1

[Serusers] (no subject)
by Annie Sasidar 02 Oct '03

02 Oct '03

Thanks Daniel, I want to allow any user in the SIP domain to be able to register since i have control over that. What i cant figure out is when i log into Windows Messenger( which right now is not supporting SIP temporarily ) with my SIP user name shouldnt i be registered in the SIP server? How do i check this cause i dont seem to find any record of any user. Since the registrar modules are loaded by default i assume that the server is working as a registrar server too. Am i wrong or am i missing some info. Please guide. Thanks for the support. Annie

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[Serusers] rtpproxy
by Josh Newman 02 Oct '03

02 Oct '03

Maxim, I am still having a problem accessing this site. Is it up yet? Can you send me the latest tarball please? -Josh > > Dovid wrote: > Does anyone know of a working site to download rtpproxy? > https://demo.portaone.com/~sobomax/PortaSIP is not responding. > (Does anyone have a copy to send me?) Yes, I know, we are moving our server to the new location, so that it is off-line now. It should be available later today, if not, drop me a note I'll mail you a tarball. -Maxim

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Re: [Serusers] SIP Requests getting matched on my Transparent Web Proxy.
by Daniel-Constantin Mierla 02 Oct '03

02 Oct '03

Hello, I got the debug messages -- if the mail is too large, send it to serteam(a)iptel.org instead of serusers(a)lists.iptel.org -- this one has size limitation. The INVITE is processed correctly, the Request-URI is replaced with the contact address of the destination user (2222) and then the request is forwarded but the client does not react. What kind of SIP client do you use? Both clients are registered from same IP address and port (202.133.79.3:5060). }Daniel On 10/2/2003 11:50 AM, John Foster wrote: > Previous mail stuck for moderator, due to larger size of mail.. > > > Hi All n Daniel, > > Here are network Dumps.. > Sorry for too lengthy dumps:) These include first portion of starting > up ser -d -d -E, then both soft phone logged in, lastly there are > traces of call initiating attempts from user 12345(a)cooking.com.pk > <mailto:12345@cooking.com.pk> to user 22222(a)cooking.com.pk > <mailto:22222@cooking.com.pk> > > > Regards, > JF >

2 1

[Serusers] SIP Requests getting matched on my Transparent Web Proxy.
by John Foster 02 Oct '03

02 Oct '03

Hi All, A new user here, I m using 0.8.11, with default conf file shipped with package.without auth. While my softphone registers with the proxy, it gets listed in "serctl ul show", but another user while joins through same proxy with all other params same except username. The two cannt dial eachother.their addresses URIs are 12345(a)xxx.yyy.com and 22222(a)xxx.yyy.com, while caller,callee and proxy are in same IP subnet. They gett Timeout message. Can anyone give any idea? Thanks in Adv. JF --------------------------------- Do you Yahoo!? The New Yahoo! Shopping - with improved product search

3 3

[Serusers] initial setup requirement
by Annie Sasidar 02 Oct '03

02 Oct '03

Thanks Daniel, I want to allow any user in the SIP domain to be able to register since i have control over that. What i cant figure out is when i log into Windows Messenger( which right now is not supporting SIP temporarily ) with my SIP user name shouldnt i be registered in the SIP server? How do i check this cause i dont seem to find any record of any user. Since the registrar modules are loaded by default i assume that the server is working as a registrar server too. Am i wrong or am i missing some info. Please guide. Thanks for the support. Annie

2 1

[Serusers] exec_dset() in failure_route
by Alexander Mayrhofer 02 Oct '03

02 Oct '03

Hi, I'm trying to change the destination set of an request in a failure route using exec_dset(). My routing looks like route { ... t_on_failure(1); ... } failure_route[1] { revert_uri(); exec_dset("find-voicemail.pl"); t_relay(); break; } unfortunately, that does not seem to work, i get the following error from SER when a request hits the failure_route[]: Oct 1 14:33:17 graham /usr/sbin/ser[4814]: ERROR: t_forward_nonack: no branched for fwding Oct 1 14:33:17 graham /usr/sbin/ser[4814]: ERROR: w_t_relay (failure mode): forwarding failed Simply adding a static destination by append_branch() works fine, but that's only the solution for an announcement... I'd need something like exec_append_branch() ... So, is there another way to dynamically add branches? Any hints appreciated! Alexander Mayrhofer nic.at

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[Serusers] упрощенная система в 2004 г.: за и против
by Обучение 02 Oct '03

02 Oct '03

СЕМИНАР «Упрощенная система налогообложения, учета и отчетности для малых предприятий в 2004 году» 7 октября 2003 1. Упрощенная система налогообложения (УСН). Налоги и сборы, исчисляемые при УСН; в каких случаях организации, перешедшие на УСН, являются налоговыми агентами по НДС, налогу на прибыль и другим налогам; преимущества и недостатки новой упрощенной системы налогообложения. Новое в законодательстве по УСН. 2. Переход на упрощенную систему налогообложения и отказ от нее. Порядок извещения налоговых органов о переходе на иную систему налогообложения. Особенности исчисления налогов при переходе с общей системы налогообложения на УСН и наоборот. 3. Организация бухгалтерского и налогового учета предприятия при переходе на УСН. Налоговые регистры, порядок расчета налога по бюджетам и фондам, льготы при УСН. 4. Порядок признания доходов и расходов в переходный период. Учет стоимости основных средств и сумм начисленной амортизации в переходный период. 5. Единый налог по результатам хозяйственной деятельности. Два варианта исчисления единого налога: объекты налогообложения, налоговая база и налоговые ставки; налоговый период, дата признания и порядок определения доходов и расходов при УСН; расходы, учитываемые и не учитываемые при определении налоговой базы; особенности учета отдельных видов доходов и расходов (расходов на приобретение основных средств и доходов от их реализации; доходов и расходов, выраженных в иностранной валюте; доходов, полученных в натуральной форме); учет убытков предыдущих периодов. 6. Учетная политика субъектов, применяющих упрощенную систему в 2004 г. 7. Отчетность организации, применяющей УСН. Порядок заполнения и сроки подачи налоговой отчетности 8. Особенности трудовых отношений с работниками в организации, применяющей УСН. Стоимость семинара - 3900 рублей, с учетом НДС. Наши телефоны: 789 81 90, 207 26 21 Отписка - abuse(a)intermarketing.ru

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[Serusers] rtpproxy
by Josh Newman 01 Oct '03

01 Oct '03

Maxim, I am still having a problem accessing this site. Is it up yet? Can you send me the latest tarball please? -Josh > > Dovid wrote: > Does anyone know of a working site to download rtpproxy? > https://demo.portaone.com/~sobomax/PortaSIP is not responding. > (Does anyone have a copy to send me?) Yes, I know, we are moving our server to the new location, so that it is off-line now. It should be available later today, if not, drop me a note I'll mail you a tarball. -Maxim

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Fw: [Serusers] sems voice mail server capacity
by Raphael Coeffic 01 Oct '03

01 Oct '03

sorry, i forgot to CC the list.... -Raphael. ----- Original Message ----- From: "Raphael Coeffic" <rco(a)iptel.org> To: "Steven R. Bunin" <steve(a)solaas.com> Sent: Wednesday, October 01, 2003 11:32 PM Subject: Re: [Serusers] sems voice mail server capacity > > Does anyone know of any capcity limits for the SEMS Voice Mail module or > have any other voice mail experience with large SER deployments? > > Sems/voicemail has been tested internally with 8 simultanous calls. It > didn't seem to have reach its maximal load capability. > > > Would you suggest running SEMS on the same server as SER or on a different > box and how much hard drive space would you set aside? > > It strongly depends on your needs. If you plan to use it at a large scale, i > would suggest to use a seperate server for voicemail. You don't need so much > space on disk, as voicemail immediatly mails the messages. All the disk > space you need, is for message buffering, let's say < 10 Mb for 25 > simultanous calls. > If you want handle many calls, you should have a strong processor and enough > RAM, that's all. > > -Raphael. >

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[Serusers] Re: Serusers Digest, Vol 5, Issue 63
by Steven R. Bunin 01 Oct '03

01 Oct '03

Hi all, I have successfully gotten radius authentication working and I started getting Radius Start records for accounting but I am not sure what I am doing wrong in regards to getting radius Stop records. Below is the area I believe has the most affect on Radius Acccounting from my Log File. Any suggestions would be appreciated and if it would help to see the full Config file I will send it as well. record_route(); if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save("location"); break; }; if (method =="INVITE") { log(1,"INVITE\n"); setflag(1); }; if (method=="MESSAGE") { log(1,"MESSAGE\n"); setflag(1); }; if (method=="BYE"){ log (1, "BYE or CANCEL\n"); setflag(1); }; if (method=="CANCEL"){ log (1, "BYE or CANCEL\n"); setflag(1); }; Thanks in advance, Steve

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Re: [Serusers] serweb not working?
by Daniel-Constantin Mierla 01 Oct '03

01 Oct '03

I am sorry, too. I am not a MySQL expert. I would try to add another row with IP address (I guess 192.168.10.121) instead of 'sip.sm6tpn.mine.nu'. If doesn't work, post the question on the mysql mailing lists. }Daniel On 10/1/2003 9:12 PM, Håkan wrote: >Hello. > >Tanks for youre answer. I am sorry. I can not get it right. >I can logg in like this. > >"mysql -u root -p" but not with "mysql -u root -h sip.sm6tpn.mine.nu -p" >Even if I kan se > >| Host | User | Pass..................... >| localhost | root | 123...... | Y | Y | >Y |........... >| sip.sm6tpn.mine.nu | root | 123...... | Y | Y | Y >|......... > >I do not understand why I can't log in. Localhost and sip.sm6tpn.mine.nu is >the very same computer... > >Regards Håkan > >----- Original Message ----- >From: "Daniel-Constantin Mierla" ><Daniel-Constantin.Mierla(a)fokus.fraunhofer.de> >To: "Håkan" <hakan(a)sm6tpn.mine.nu> >Cc: <serusers(a)lists.iptel.org> >Sent: Wednesday, October 01, 2003 10:32 AM >Subject: Re: [Serusers] serweb not working? > > > > >>Hello, >>the user ser has no access right from host sip.sm6tpn.mine.nu to the >>mysql server. See the 'Databse administration' chapter of mysql manual >>to learn how to do it. >>Useful links: >>http://www.mysql.com/documentation/mysql/bychapter/index.html >> >> >> >http://www.mysql.com/documentation/mysql/bychapter/manual_MySQL_Database_Ad… > > >>Best regards, >>Daniel >> >>On 9/30/2003 10:16 PM, Håkan wrote: >> >> >> >>>My problem is when I go to http://my.domain/admin/index.php I get this >>>information:- >>> >>>Warning: Access denied for user: 'ser(a)sip.sm6tpn.mine.nu' (Using >>> >>> >password: > > >>>YES) in /var/www/phplib/db_mysql.inc on line 73 >>>Database error: pconnect(192.168.10.121, ser, $Password) failed. >>>MySQL Error: () >>>Session halted. >>> >>>I have installd >>> >>>ser-0.8.11-0.i386.rpm >>>ser-jabber-0.8.11-0.i386.rpm >>>ser-mysql-0.8.11-0.i386.rpm >>>php-4.1.2-7.i386.rpm >>>php-mysql-4.1.2-7.i386.rpm >>>mysql-server-3.23.49-3.i386.rpm >>>apache-1.3.27-5.i386.rpm >>> >>>I run RedHat 7.3 >>>kernel 2.4.20-20.7 >>> >>>I can logging in as ser and serro in mysql. the ser db is there too. >>> >>>I have folowed Dan Austins SER HowTo but no luck :-( >>> >>>Please help me. My head is hurting ;-) >>> >>>regardes Hakan. >>> >>> >>> >>>_______________________________________________ >>>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] SER/SIP & RADIUS/Auth-Type = Digest
by Steve Dolloff 01 Oct '03

01 Oct '03

OK, I figured out that I had messed up the digest file that I was using with radclient. I now get a correct response using radclient to test against the freeradius server. When I try to auth from ser though, I am getting a failure. Group authenticate returns reject. I'm not intending to do any group authentication. I tried loading the group_radius module instead of the group module and I also tried loading no group modules, but I still get the same error. Please see my radiusd-x output... modcall: group authorize returns ok rad_check_password: Found Auth-Type Digest auth: type "digest" modcall: entering group authenticate A1 = sdolloff:voip2.test.net:test A2 = REGISTER:sip:voip2.test.net KD = ad3c99a75e03ad3ead8254ce95a59a3b:3f7b05a030240eba31ec566b2d783170e9c9830 0:797c155d7796a9cb0be4154d07e88417 rlm_digest: FAILED authentication modcall[authenticate]: module "digest" returns reject modcall: group authenticate returns reject auth: Failed to validate the user. Stephen -----Original Message----- From: Daniel-Constantin Mierla [mailto:Daniel-Constantin.Mierla@fokus.fraunhofer.de] Sent: Wednesday, October 01, 2003 3:39 AM To: Steve Dolloff Cc: Serusers Subject: Re: [Serusers] SER/SIP & RADIUS/Auth-Type = Digest Hello, comments inline. On 9/30/2003 10:32 PM, Steve Dolloff wrote: >I have installed freeradius according to the "HOW TO" for radius and now >I am seeing the following error. I assume that since I am seeing errors >on both servers that it is a problem with either the dictionary or the >client. Here are the new error logs... any ideas? > >rad_recv: Access-Request packet from host 209.242.100.153:33612, id=103, >length=148 > User-Name = "sdolloff" > Digest-Response = "631d6d73147add2f9e437f59bbc3aeb7" > Digest-Attributes = "\001\017voip2.test.net" > Digest-Attributes = "\002\006test" > Digest-Attributes = "\003\010INVITE" > Digest-Attributes = "\004\034sip:5555551212@example.com" > Digest-Attributes = "\006\005MD5" > Digest-Attributes = "\n\nsdolloff" >modcall: entering group authorize > modcall[authorize]: module "preprocess" returns ok > modcall[authorize]: module "chap" returns noop >rlm_eap: EAP-Message not found > modcall[authorize]: module "eap" returns noop > rlm_digest: Converting Digest-Attributes to something sane... > Digest-Realm = "voip2.test.net" > Digest-Nonce = "test" > Digest-Method = "INVITE" > Digest-Uri = "sip:5555551212@example.com" > Digest-Algorithm = "MD5" > Digest-User-Name = "sdolloff" >rlm_digest: Adding Auth-Type = DIGEST > modcall[authorize]: module "digest" returns ok > rlm_realm: No '@' in User-Name = "sdolloff", looking up realm NULL > rlm_realm: No such realm "NULL" > modcall[authorize]: module "suffix" returns noop > users: Matched DEFAULT at 152 > modcall[authorize]: module "files" returns ok > modcall[authorize]: module "mschap" returns noop >modcall: group authorize returns ok > rad_check_password: Found Auth-Type DIGEST >auth: type "digest" >modcall: entering group authenticate >rlm_digest: Configuration item "User-Password" is required for >authentication. > It seems that the "User-Password" attribute is missing for user "sdolloff" in radius users file. It should look like the example from Radius HOW-TO: http://iptel.org/ser/doc/ser_radius/ser_radius.html#AEN139. Daniel > modcall[authenticate]: module "digest" returns invalid >modcall: group authenticate returns invalid >auth: Failed to validate the user. >Delaying request 6 for 1 seconds >Finished request 6 >Going to the next request >--- Walking the entire request list --- >Waking up in 1 seconds... >--- Walking the entire request list --- >Waking up in 1 seconds... >--- Walking the entire request list --- >Sending Access-Reject of id 103 to 209.242.100.153:33612 >Waking up in 4 seconds... >--- Walking the entire request list --- >Cleaning up request 6 ID 103 with timestamp 3f79e7dc >Nothing to do. Sleeping until we see a request. >Subject: Re: [Serusers] SER/SIP & RADIUS/Auth-Type = Digest > >On (30.09.03 13:54), Steve Dolloff wrote: > > >>209.242.100.153 for 'sdolloff(a)voip2.test.net' is ignored;no password >> or CHAP password is used >> >> > >Your RADIUS server has to support Digest Authentication, and the line >above seems to indicate that it does not do that. > >If you can change your Radius server software, give Freeradius or >Radiator (commercial, but excellent) a try. If you can not, try to >educate your existing server to do CHAP-Type authentication. > >hope that helps. > >Alex Mayrhofer >nic.at > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > > >

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[Serusers] Problem with mySql (new thread)
by giAndrea 01 Oct '03

01 Oct '03

I've created a new server and i've installed ser 0.8.11 (ser and ser-mysql-0.8.11-0) with rpm. When i use the command ./ser_mysql.sh create and it return this: ./ser_mysql.sh: line 1: gen_ha1: command not found HA1 calculation failed any idea? Bye, Andrea

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sr-users October 2003