sr-users October 2003

sr-users@lists.kamailio.org

105 participants
230 discussions

by Dovid

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

20 years, 7 months

Re: [Serusers] SER account for mySQL

by Jiri Kuthan

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/

20 years, 7 months

[Serusers] Problem strip and t_relay

by Tropico Network

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

20 years, 7 months

[Serusers] re: Problems with accounting - can't get stop records

by jaime.gil＠orange.co.uk

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. *******************************************************************************

20 years, 7 months

Re: [Serusers] Re: Problems with accounting - can't get stop records

by lgfausak＠august.net

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 >

20 years, 7 months

Re: [Serusers] RE: ser and enum (bind) issue

by Jiri Kuthan

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/

20 years, 7 months

RE: [Serusers] caller-id with raius using sip-rpid

by Steve Dolloff

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

20 years, 7 months

RE: [Serusers] caller-id with raius using sip-rpid

by Steve Dolloff

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

20 years, 7 months

RE: [Serusers] caller-id with raius using sip-rpid

by Steve Dolloff

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

20 years, 7 months

[Serusers] SER account for mySQL

by Harry Behrens

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

20 years, 7 months

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sr-users October 2003