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[Serusers] mysql.sock
by mjkane＠attbi.com 11 Mar '03

11 Mar '03

Hello all, what file directs where SER looks for the mysql.sock file. In my case mysql.sock is located in /tmp and as you can see SER is lookin for it in /var/lib/mysql Mar 10 16:46:33 ttalksvr /usr/sbin/ser[6592]: connect_db(): Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) Thanks Mike

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[Serusers] Admin Interface
by Erik Lagerway 07 Mar '03

07 Mar '03

We are contemplating building an Admin interface for SER that could run on a windows os. We like the fact that the server is running on Linux but we require a more intuitive interface for administration. A web interface is not acceptable, just not enough functionality. We would like to create an executable that runs on windows which interfaces with SER server running on Linux. I am curious if anyone else could make use of a GUI that would incorporate all of the SER features and functionality. … [View More]

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[Serusers] Accounting module mods with MySQL and SERWEB
by John N. French 07 Mar '03

07 Mar '03

Some interesting modifications I have made to improve acc process. 1) -------------- Some changes made to acc.c to strip down the user names for output and also to get missed calls logged with callee rather than caller. It was showing missed calls on the callers missed_calls.php which is incorrect. Here are the extracts, I can post the full prog if required: /* JF added following block to extract to user */ str_copy(&to_user, &rq->to->body); if ((ul_get_user(&… [View More]to_user) < 0) || !to_user.len) { LOG(L_ERR, "ERROR: acc_request: Error while extracting to_username\n"); return -1; } /* str_copy(&user, &rq->first_line.u.request.uri); JF modified this copy statement to create correct acc username (from not to!) */ str_copy(&user, &rq->from->body); 2) --------------------- I modified the $q in accounting.php to the below to include calls where BYE sent by callee as well as caller (before it was not displaying calls when closed by callee). Also elimination of double counting of calls due to multiple INVITEs being sent by UAC. max(time) is required as distinct rows are produced if the invites are > 1sec apart. "Group by" puts all the INVITEs together so that the latest one can be selected with max(). Assume that latest invite is correct for measuring call start time. The max(time) column has been aliased to c2 so that it can be used in the output $time=Substr($row->c2,0,16). Here it is: $q="select distinct t1.sip_callid, max(t1.time) c2, t1.sip_to, sec_to_time(unix_timestamp(t2.time)-unix_timestamp(max(t1.time))) as length ". "from ".$config->table_accounting." t1, ".$config->table_accounting." t2 ". "where (t1.user='".$auth->auth["uname"]."' and t1.sip_method='INVITE') and ((t2.user='".$auth->auth["uname"]."' or t2.sip_to='".$auth->auth["uname"]."') and t2.sip_method='BYE') and ". "t1.sip_callid=t2.sip_callid group by t1.sip_callid ". "order by t1.time desc"; 3) I modified the &q in missed_calls.php to the below to include calls to aliases as well. The query looks in the missed_calls table for the sip_to = logged-in user and also joins with aliases table to extract calls to alias of contact. Otherwise missed_calls.php only provided calls made directly to a user and ignored calls to aliases. Here it is: $q="select distinct t1.user, t1.sip_to, t1.time, t1.sip_status, t1.sip_callid from ".$config->table_missed_calls." t1,".$config->table_aliases." t2 where t1.sip_to='".$auth->auth["uname"]."' OR ('sip:".$auth->auth["uname"]."@".$config->default_domain."'=t2.contact AND t2.user=t1.sip_to) order by time desc"; echo $q; 4) ----------------- Any comments will be appreciated. __________________________________________________ Do you Yahoo!? Yahoo! Tax Center - forms, calculators, tips, more http://taxes.yahoo.com/ [View Less]

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RE: [Serusers] Admin Interface
by Dan Austin 06 Mar '03

06 Mar '03

How about a java applet for use with a web browser? I'd have no problem with a requirement for a Windows PC for management, but I can see it being an issue for others. And yes I would use it, in whatever form it took. Dan -----Original Message----- From: Erik Lagerway [mailto:sipdev@xten.com] Sent: Thursday, March 06, 2003 1:50 PM To: sip-implementors(a)cs.columbia.edu Cc: serusers(a)lists.iptel.org Subject: [Serusers] Admin Interface We are contemplating building an Admin … [View More]

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[Serusers] SER and SIPC (columbia uni)
by Mario Kolberg 06 Mar '03

06 Mar '03

Hi, I tried to use the SER SIP server together with a number of different UAs. I successfully used a Pingtel phone, pingtel softphone, the Ubiquity UA, and kphone on linux. However, when I tried to use SIPC from Columbia Uni I keep getting eror messages 400: Bad Request. As all the other UAs are working fine I assume this may be a problem with sipc. Have you any experiences with using sipc and ser? I attach an error log between the Pingtel phone (139.153.254.222) is trying to send an … [View More]

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[Serusers] serweb download
by nmacia＠info.unlp.edu.ar 06 Mar '03

06 Mar '03

I everybody. where can i download serweb???? at http://developer.berlios.de/projects/serweb/ there is not any release for download and in home page it says "Web-based user provisioning, serweb, available" thank's all ------------------------------------------------- This mail sent through IMP: http://mail.info.unlp.edu.ar/

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RE: [Serusers] multiple registration on one user login
by Ng, Soo Sim 06 Mar '03

06 Mar '03

Jiri, Scenario is providing IP Telephony to the household. I am more concern about the security of the Hardphone. I am thinking of auto-provisioned the hardphone (eg C7960, ATA186) without subsriber intervention. What the subscriber know is their phone # (Just like legacy phone system). Since the Hardphone is 'hard-coded', the phone can move round the vicinity of the redisential area and still able to make a call. Potentially this will lead to abuse, as someone may take the phone to a … [View More]

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RE: [Serusers] multiple registration on one user login
by Jiri Kuthan 05 Mar '03

05 Mar '03

Hello, I fear that such a case can't be avoided with allowing only a single registration. If I steal your phone away from your desk, you will not register with it anymore, but I will and we will have exactly one valid registration. Leaving SIP phones with hard-wired passwords on your desk has simply the same potential as leaving your credit-card or cell-phone there. What can be done about fraud? User education -- don't leave your money and phone unattended. Hotline -- report stolen phones … [View More]to lock the account. PIN Lock -- use phones which can log-off and log-on (I'm not aware of any now -- only 3com used to do that) -Jiri ps -- ability to move is a feature. I know people who are very glad to use Vonage's US phone number and move with their ATAs and the US phone number around in Europe. At 11:37 PM 3/5/2003, Ng, Soo Sim wrote: >Jiri, > >Scenario is providing IP Telephony to the household. >I am more concern about the security of the Hardphone. I am thinking of auto-provisioned the hardphone (eg C7960, ATA186) without subsriber intervention. What the subscriber know is their phone # (Just like legacy phone system). > >Since the Hardphone is 'hard-coded', the phone can move round the vicinity of the redisential area and still able to make a call. Potentially this will lead to abuse, as someone may take the phone to a different location when owner is not around and make a 'free' call, return back the phone and the billing still charge the original subsriber. > >Any other suggestion to counter this issue is much appreacited. > >SSng > >-----Original Message----- >From: Jiri Kuthan [mailto:jiri@iptel.org] >Sent: Wednesday, March 05, 2003 12:18 AM >To: Ng, Soo Sim; serusers(a)lists.iptel.org >Subject: Re: [Serusers] multiple registration on one user login > > >At 03:08 PM 3/4/2003, Ng, Soo Sim wrote: >>I have such requirements. In providing sip-based residential ip telephony, I would like to restrict each home subsriber is only allowed to register one UA per account. This would make easy for billing purposes and for security reasons. >> >>Is there a way to achieve this requirement with SER? > >If that is your desparate wish, it is little overhead to make you happy. >I'm still not sure though, it is a useful thing. > >Maybe an operator can make more revennues if my wife can accept calls at >any phone in my building and initiate calls in parallel with my doughter. > >What are exactly the billing/security reasons here? > >-Jiri -- Jiri Kuthan http://iptel.org/~jiri/ [View Less]

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RE: [Serusers] Messenger 4.7, CIsco and PSTN
by Dan Austin 05 Mar '03

05 Mar '03

I have to dig at it abit, but it may also be a codec issue on the Phone. I had a similar error before telling the Cisco which codec to use. I see you are using G711Alaw. Can you try G711Ulaw? The PSTN hand-off section of your config looks very familiar to me, so if you did pull it from the Howto, note that it is an example of what worked for me. I've seen quite a few much more sophisticated scripts on the list. Dan -----Original Message----- From: Jiri Kuthan [mailto:jiri@iptel.org] … [View More]Sent: Wednesday, March 05, 2003 2:29 AM To: Rikard Westlund; serusers(a)lists.iptel.org Subject: Re: [Serusers] Messenger 4.7, CIsco and PSTN I suspect what happens is that you forward the requests with your server's address in its r-uri to gateway "as is" and the Cisco gateway would like to see its IP address in the r-uri instead. Try rewriting r-uri -- see bellow. As for the Messenger problem, see our doc http://www.iptel.org/ser/doc/seruser-html/x878.html#AEN890 -Jiri At 11:04 AM 3/5/2003, Rikard Westlund wrote: >Hi all, > >I have a Ser 0.8.10-2 install on a Redhat 7.3 kernel 2.4.18-3. > >As clients I use Pingtel and messenger 4.7. I have followed the setup >guide on http://www.fitawi.com/ser-Howto.html > >I can register the pingtel phone with no problem. I can call from the >PSTN to the pingtel via a Cisco AS5300 with no problems. > >When i try toi call from pingtel to PSTN iget the following answer: > >1. from pingtel to ser - INVITE sip:<pstnnumber>@serserver_ip 2. from >ser to pingtel - Status: 100 trying 3. from ser to cisco - INVITE >sip:<pstnnumber>@serserver_ip 4. from cisco to ser - Status: 400 bad >request - ínvalid IP address' 5. from cisco to ser - Status: 400 bad >request - ínvalid IP address' > >This is my ser.cfg: > ># $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $ ># ># simple quick-start config script ># > ># ----------- global configuration parameters ------------------------ > >debug=4 # debug level (cmd line: -dddddddddd) >fork=yes >log_stderror=no # (cmd line: -E) >check_via=no # (cmd. line: -v) >dns=no # (cmd. line: -r) >rev_dns=no # (cmd. line: -R) >port=5060 >children=4 >fifo="/tmp/ser_fifo" > ># ------------------ module loading ---------------------------------- > ># Uncomment this if you want to use SQL database >loadmodule "//usr/lib/ser/modules/mysql.so" > >loadmodule "//usr/lib/ser/modules/sl.so" >loadmodule "//usr/lib/ser/modules/tm.so" >loadmodule "//usr/lib/ser/modules/rr.so" >loadmodule "//usr/lib/ser/modules/maxfwd.so" >loadmodule "//usr/lib/ser/modules/usrloc.so" >loadmodule "//usr/lib/ser/modules/registrar.so" > ># Uncomment this if you want digest authentication ># mysql.so must be loaded ! >loadmodule "//usr/lib/ser/modules/auth.so" > ># ----------------- setting module-specific parameters --------------- > ># -- usrloc params -- > >#modparam("usrloc", "db_mode", 0) > ># Uncomment this if you want to use SQL database ># for persistent storage and comment the previous line >modparam("usrloc", "db_mode", 2) > ># -- auth params -- ># Uncomment if you are using auth module ># >#modparam("auth", "secret", "alsdkhglaksdhfkloiwr") modparam("auth", >"calculate_ha1", yes) # ># If you set "calculate_ha1" parameter to yes (which true in this config), ># uncomment also the following parameter) ># >modparam("auth", "password_column", "password") > ># ------------------------- request routing logic ------------------- > ># main routing logic > >route{ > > # initial sanity checks -- messages with > # max_forwars==0, or excessively long requests > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > break; > }; > if (len_gt( max_len )) { > sl_send_reply("513", "Message too big"); > break; > }; > > > # Do strict routing if pre-loaded route headers present > rewriteFromRoute(); > > # if the request is for other domain use UsrLoc > # (in case, it does not work, use the following command > # with proper names and addresses in it) > if (uri==myself) { > > if (method=="REGISTER") { > ># Uncomment this if you want to use digest authentication > if (!www_authorize("norrtull.nexus.se", "subscriber")) { > www_challenge("norrtull.nexus.se", "0"); > break; > }; > > save("location"); > break; > }; > ># attempt handoff to PSTN > > if (uri=~"^sip:1[0-9]*@norrtull.nexus.se") { ## This assumes that the caller is > log("Forwarding to PSTN\n"); ## registered in our realm *** here *** rewrite uri prior to fwd-ing. rewritehostport("cisco_ip:5060"); > t_relay_to( "cisco_ip", "5060"); ## Our Cisco router > break; > }; > > > # native SIP destinations are handled using our USRLOC DB > if (!lookup("location")) { > sl_send_reply("404", "Not Found"); > break; > }; > }; > # forward to current uri now > if (!t_relay()) { > sl_reply_error(); > }; > >} > >--------------------------------- > >In the cisco I have the following config: > >! >dail-peer voice 25 voip >destination-pattern XXXX >session protocol sipv2 >codec g711alaw >no vad >session target ipv4:serserver_ip >! > >I have added 2 subscribers with the serctl command and registration is >working well from pingtel. In Messenger 4.7 it's not working at all. I >get 401 Unauthorized. > >Well I think thats about it.. > >Please feel free to contact me if you need more information > >Best regards > >Rikard Westlund > > > >_________________________________________________________________ >Tired of spam? Get advanced junk mail protection with MSN 8. >http://join.msn.com/?page=features/junkmail > >_______________________________________________ >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 [View Less]

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[Serusers] Messenger 4.7, CIsco and PSTN
by Rikard Westlund 05 Mar '03

05 Mar '03

Hi all, I have a Ser 0.8.10-2 install on a Redhat 7.3 kernel 2.4.18-3. As clients I use Pingtel and messenger 4.7. I have followed the setup guide on http://www.fitawi.com/ser-Howto.html I can register the pingtel phone with no problem. I can call from the PSTN to the pingtel via a Cisco AS5300 with no problems. When i try toi call from pingtel to PSTN iget the following answer: 1. from pingtel to ser - INVITE sip:<pstnnumber>@serserver_ip 2. from ser to pingtel - Status: 100 … [View More]trying 3. from ser to cisco - INVITE sip:<pstnnumber>@serserver_ip 4. from cisco to ser - Status: 400 bad request - ínvalid IP address' 5. from cisco to ser - Status: 400 bad request - ínvalid IP address' This is my ser.cfg: # $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "//usr/lib/ser/modules/mysql.so" loadmodule "//usr/lib/ser/modules/sl.so" loadmodule "//usr/lib/ser/modules/tm.so" loadmodule "//usr/lib/ser/modules/rr.so" loadmodule "//usr/lib/ser/modules/maxfwd.so" loadmodule "//usr/lib/ser/modules/usrloc.so" loadmodule "//usr/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "//usr/lib/ser/modules/auth.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth", "secret", "alsdkhglaksdhfkloiwr") modparam("auth", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth", "password_column", "password") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # Do strict routing if pre-loaded route headers present rewriteFromRoute(); # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("norrtull.nexus.se", "subscriber")) { www_challenge("norrtull.nexus.se", "0"); break; }; save("location"); break; }; # attempt handoff to PSTN if (uri=~"^sip:1[0-9]*@norrtull.nexus.se") { ## This assumes that the caller is log("Forwarding to PSTN\n"); ## registered in our realm t_relay_to( "cisco_ip", "5060"); ## Our Cisco router break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now if (!t_relay()) { sl_reply_error(); }; } --------------------------------- In the cisco I have the following config: ! dail-peer voice 25 voip destination-pattern XXXX session protocol sipv2 codec g711alaw no vad session target ipv4:serserver_ip ! I have added 2 subscribers with the serctl command and registration is working well from pingtel. In Messenger 4.7 it's not working at all. I get 401 Unauthorized. Well I think thats about it.. Please feel free to contact me if you need more information Best regards Rikard Westlund _________________________________________________________________ Tired of spam? Get advanced junk mail protection with MSN 8. http://join.msn.com/?page=features/junkmail [View Less]

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[Serusers] PRACK
by Greg Fausak 05 Mar '03

05 Mar '03

I copied a few SNOM phone messages to the list. I am having problems communicating with the SNOM phone. I guess it isn't working because the SNOM phone sends a PRACK after the 183, and the received OK doesn't have a 'Record-Route'. This doesn't make sense to me. I don't know why the phone is listening to the first OK (the OK to the PRACK) rather than the second OK (the OK to the INVITE). If that is the way it is then do I have the PROXY mis-configured? ----- if(method=="INVITE" | … [View More]method=="BYE" | method=="PRACK") { log(1,"TRACE: addRecordRoute()"); setflag(1); setflag(2); addRecordRoute(); }; append_hf("P-hint: ATEND\r\n"); if(!t_relay()) { sl_reply_error(); break; }; When I do a ngrep trace, I see the packet being relayed... ---- # U 2003/03/03 20:05:59.858473 216.87.145.22:5060 -> 216.87.144.203:5060 PRACK sip:2143357976@216.87.144.203;branch=0 SIP/2.0. Via: SIP/2.0/UDP 216.87.145.22:5060;branch=z9hG4bK-bec8z5qwsbqg. Max-Forwards: 70. RAck: 5620 2 INVITE. From: "snom man" <sip:4695461245@augustvoice.net>;tag=1r1bpkl72i. To: <sip:2143357976@augustvoice.net;user=phone>;tag=63631E34-197C. Call-ID: 3c26f8de39d9-7dyvkrekeha3(a)216.87.145.22. CSeq: 4 PRACK. Route: <sip:92143357976@216.87.144.196:5060;user=phone>. Contact: <sip:4695461245@216.87.145.22:5060;line=1>. Content-Length: 0. Authorization: Digest username="4695461245",realm="augustvoice.net",nonce="3e640b3300000000dbb edccd04f755cac7e3b6dd55d202c9",uri="sip:",response="3693b828395cd40fe1b1 ab5f9ad61308",algorithm=md5. . # U 2003/03/03 20:05:59.859540 216.87.144.203:5060 -> 216.87.144.196:5060 PRACK sip:92143357976@216.87.144.196:5060;user=phone SIP/2.0. Record-Route: <sip:2143357976@216.87.144.203;branch=0>. Via: SIP/2.0/UDP 216.87.144.203;branch=z9hG4bKe44a.ff0091c4.0. Via: SIP/2.0/UDP 216.87.145.22:5060;branch=z9hG4bK-bec8z5qwsbqg. Max-Forwards: 69. RAck: 5620 2 INVITE. From: "snom man" <sip:4695461245@augustvoice.net>;tag=1r1bpkl72i. To: <sip:2143357976@augustvoice.net;user=phone>;tag=63631E34-197C. Call-ID: 3c26f8de39d9-7dyvkrekeha3(a)216.87.145.22. CSeq: 4 PRACK. Contact: <sip:4695461245@216.87.145.22:5060;line=1>. Content-Length: 0. Authorization: Digest username="4695461245",realm="augustvoice.net",nonce="3e640b3300000000dbb edccd04f755cac7e3b6dd55d202c9",uri="sip:",response="3693b828395cd40fe1b1 ab5f9ad61308",algorithm=md5. P-hint: ATEND. . # U 2003/03/03 20:05:59.861477 216.87.144.196:5060 -> 216.87.144.203:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 216.87.144.203;branch=z9hG4bKe44a.ff0091c4.0,SIP/2.0/UDP 216.87.145.22:5060;branch=z9hG4bK-bec8z5qwsbqg. From: "snom man" <sip:4695461245@augustvoice.net>;tag=1r1bpkl72i. To: <sip:2143357976@augustvoice.net;user=phone>;tag=63631E34-197C. Date: Tue, 04 Mar 2003 02:05:59 GMT. Call-ID: 3c26f8de39d9-7dyvkrekeha3(a)216.87.145.22. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 4 PRACK. Content-Length: 0. . # U 2003/03/03 20:05:59.861705 216.87.144.203:5060 -> 216.87.145.22:5060 SIP/2.0 200 OK. Via:SIP/2.0/UDP 216.87.145.22:5060;branch=z9hG4bK-bec8z5qwsbqg. From: "snom man" <sip:4695461245@augustvoice.net>;tag=1r1bpkl72i. To: <sip:2143357976@augustvoice.net;user=phone>;tag=63631E34-197C. Date: Tue, 04 Mar 2003 02:05:59 GMT. Call-ID: 3c26f8de39d9-7dyvkrekeha3(a)216.87.145.22. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 4 PRACK. Content-Length: 0. . ----- There is a Record-route going out, but not one coming back. Does a Record-route need to be injected into the last OK back to the UA? This is too hard :-) ---greg [View Less]

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[Serusers] Debian ser wont start up
by Jaime Hemmett 05 Mar '03

05 Mar '03

I installed ser stable then unstable debs on my 2.2.20 kernel install of unstable debian. Errors: when init.d/ser start: Starting ser: serToo much shared memory demanded: 33554432 Terminated SYSLOG: Mar 5 10:09:19 webserver ser: ERROR: shm_mem_init: could not attach shared memo ry segment: Invalid argument Mar 5 10:09:19 webserver ser: could not initialize shared memory pool, exiting. a friend sent me: scmctl from shmctl(2): Various fields in a struct shmid_ds were shorts … [View More]

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[Serusers] multiple registration on one user login
by Benny Ho 04 Mar '03

04 Mar '03

Dear all, SIP is allow multiple ua to register under one uri? How could I disable this? I mean only one ua can login under one user account?

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Re: [Serusers] multiple registration on one user login
by Jiri Kuthan 04 Mar '03

04 Mar '03

At 03:08 PM 3/4/2003, Ng, Soo Sim wrote: >I have such requirements. In providing sip-based residential ip telephony, I would like to restrict each home subsriber is only allowed to register one UA per account. This would make easy for billing purposes and for security reasons. > >Is there a way to achieve this requirement with SER? If that is your desparate wish, it is little overhead to make you happy. I'm still not sure though, it is a useful thing. Maybe an operator can make more … [View More]

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[Serusers] multiple registration on one user login
by Ng, Soo Sim 04 Mar '03

04 Mar '03

Jan Janak writes: > Yes, several UAs can register with the same account and currently there > is no way in ser how to prevent it. Do you really need such a > restriction ? If you do that, users will be unable to register several > phones or user agents and utilize forking, for example. yes, i agree that allowing many uas to register the same uri is very useful. the only problem that i see in ser regarding it is that ser doesn't yet implement sequential forking based on the q value. … [View More]

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[Serusers] response function in a module
by Mario Kolberg 04 Mar '03

04 Mar '03

Hi, I'm writing a module which requires to see request messages and also responses. So I have declared a response_function. The function looks like: static int fi_response(struct sip_msg* msg, char* str1, char* str2); and I put it into the module_exports structure like this: #ifdef STATIC_FI struct module_exports fi_exports = { #else struct module_exports exports= { #endif "fi_module", (char*[]){ "fi_request", "fi_response" }, (cmd_function[]){ fi_request, fi_response }, (int[]){ 0, 0 }… [View More]

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[Serusers] SER error message in log
by Mario Kolberg 04 Mar '03

04 Mar '03

Hi, I'm using the CVS version of SER and I found a frequent error message in the SER log. I'm not sure what this means. Mar 4 12:06:08 DISCUS /usr/local/sbin/ser[1020]: ERROR: set_kr: kill_reason reset: from=1 to=2 Mar 4 12:06:08 DISCUS /usr/local/sbin/ser[1020]: ERROR: set_kr: kill_reason reset: from=1 to=2 The error usually occurs in pairs, but the values for 'from' and 'to' vary. How can I fix it? Thanks! Mario -- Mario Kolberg phone: +44 (0)1786 46 7440 … [View More]

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[Serusers] AW: Snom phone (fwd) more fromn greg fausak
by Christian Stredicke 04 Mar '03

04 Mar '03

Greg, ooppss that actually changes the scene. Maybe it was too late for me last night already You are right, the 200 Ok goes to the PRACK and the phone mixes that up. I changed it in the phone code and that will be part of the next release. Thanks, Christian -----Ursprüngliche Nachricht----- Von: Robert Messer [mailto:rmesser@abpintl.com] Gesendet: Dienstag, 4. März 2003 02:25 An: cs(a)snom.de Betreff: FW: Snom phone (fwd) more fromn greg fausak FYI More from greg Robert … [View More]Messer 972-745-1220 Direct ___________________ ABP International, Inc. snom technology - VoIP Phones www.abptech.com 972-745-1221 Phone 972-745-1226 Fax -----Original Message----- From: Greg Fausak [mailto:greg@august.net] Sent: Monday, March 03, 2003 6:13 PM To: 'Kevin' Cc: 'Robert Messer'; sip(a)august.net; serusers(a)lists.iptel.org Subject: RE: Snom phone (fwd) Kevin, I initially sent the packet: > > > This is that packet that came from the last 200 OK <- PROXY: > > > > > > # > > > U 2003/02/24 07:56:52.503535 216.87.144.203:5060 -> > > 216.87.145.22:5060 > > > SIP/2.0 200 OK. > > > Via: SIP/2.0/UDP 216.87.145.22:5060;branch=z9hG4bK-ng5tokyx448r. > > > From: "snom man" <sip:4695461245@augustvoice.net>;tag=8u6ju8wxuc. > > > To: <sip:2143357976@augustvoice.net;user=phone>;tag=3CBB0360-532. > > > Date: Mon, 24 Feb 2003 13:56:43 GMT. > > > Call-ID: 3c267202b6a8-lgseu8olovlp(a)216.87.145.22. > > > Server: Cisco-SIPGateway/IOS-12.x. > > > CSeq: 2 INVITE. > > > Session-Expires: 7200;refresher=uac. > > > Require: timer. > > > Allow-Events: telephone-event. > > > Contact: <sip:92143357976@216.87.144.196:5060;user=phone>. > > > Record-Route: <sip:2143357976@216.87.144.203;ftag=8u6ju8wxuc;lr>. > > > Content-Type: application/sdp. > > > Content-Length: 209. > > > . > > > v=0. > > > o=CiscoSystemsSIP-GW-UserAgent 7543 5694 IN IP4 216.87.144.196. > > > s=SIP Call. > > > c=IN IP4 216.87.144.196. > > > t=0 0. > > > m=audio 16632 RTP/AVP 0 100. > > > a=rtpmap:0 PCMU/8000. > > > a=rtpmap:100 X-NSE/8000. > > > a=fmtp:100 192-194. > > > This is the 200 OK (response to the INVITE) message as delivered to the phone. I couldn't figure out what you were saying, so I went back to the ethereal trace. After the snom phone receives a 183 status message, it sends a PRACK to the PROXY. This PRACK is OKed, without a Record-route. The next message is an OK responding to the original INVITE, which does indeed have a Record-Route. So, you are saying that the OK to your PRACK needs a record route? I can do that I think, because the OK to the INVITE does indeed have a Record-route. I don't even know what a PRACK is for... ---greg -----Original Message----- From: Kevin [mailto:kmoroz@abpintl.com] Sent: Monday, March 03, 2003 5:19 PM To: Greg Fausak Cc: 'Robert Messer' Subject: FW: Snom phone (fwd) Hi Greg, Sorry it took so long to get back to you. Normally it is faster but as I stated the SIP inter- operability was last week which caused the delay. Looks like the issue is with the SER proxy. If I knew the specification deeper I should have been able to answer it myself. Engineering ccs Jiri on their response to me so he is aware of the issue. Regards, Kevin -----Original Message----- The phone updates the route until it receives a 2xx code. The 200 Ok response does not contain such a route therefore the phone uses the last route it receives in the 200 which is empty. Therefore, the phone MUST send the ACK directly to the gateway. § 12.1.2 of the RFC3261. The dialog is NOT established by the 401-challened request. The proxy can very easily solve the problem by putting itself into the routing path of the 200 Ok. [View Less]

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[Serusers] AW: Snom phone (fwd)
by Christian Stredicke 04 Mar '03

04 Mar '03

Robert, the gateway mandates PRACK, that cant be turned off. The OPTIONS can be avoided if the phone is set to use REFER explicitly. CS -----Ursprüngliche Nachricht----- Von: Robert Messer [mailto:rmesser@abpintl.com] Gesendet: Dienstag, 4. März 2003 05:58 An: cs(a)snom.de Betreff: RE: Snom phone (fwd) Kevin, I will try to modify the code accordingly, however, I've been looking at the OPTIONS tag. Perhaps I can tell the snom phone not to use PRACK? I'll look into that. Thanks, … [View More] ---greg -----Original Message----- From: Kevin [mailto:kmoroz@abpintl.com] Sent: Monday, March 03, 2003 8:09 PM To: 'Greg Fausak' Cc: 'Robert Messer'; sip(a)august.net; serusers(a)lists.iptel.org Subject: RE: Snom phone (fwd) Greg, According to snom engineering you should put the routing information in the first 200 OK we receive. Can you do that and try it? You can test against my proxy if you like to see how it should work. Regards, Kevin -----Original Message----- From: Greg Fausak [mailto:greg@august.net] Sent: Monday, March 03, 2003 7:13 PM To: 'Kevin' Cc: 'Robert Messer'; sip(a)august.net; serusers(a)lists.iptel.org Subject: RE: Snom phone (fwd) Kevin, I initially sent the packet: > > > This is that packet that came from the last 200 OK <- PROXY: > > > > > > # > > > U 2003/02/24 07:56:52.503535 216.87.144.203:5060 -> > > 216.87.145.22:5060 > > > SIP/2.0 200 OK. > > > Via: SIP/2.0/UDP 216.87.145.22:5060;branch=z9hG4bK-ng5tokyx448r. > > > From: "snom man" <sip:4695461245@augustvoice.net>;tag=8u6ju8wxuc. > > > To: <sip:2143357976@augustvoice.net;user=phone>;tag=3CBB0360-532. > > > Date: Mon, 24 Feb 2003 13:56:43 GMT. > > > Call-ID: 3c267202b6a8-lgseu8olovlp(a)216.87.145.22. > > > Server: Cisco-SIPGateway/IOS-12.x. > > > CSeq: 2 INVITE. > > > Session-Expires: 7200;refresher=uac. > > > Require: timer. > > > Allow-Events: telephone-event. > > > Contact: <sip:92143357976@216.87.144.196:5060;user=phone>. > > > Record-Route: <sip:2143357976@216.87.144.203;ftag=8u6ju8wxuc;lr>. > > > Content-Type: application/sdp. > > > Content-Length: 209. > > > . > > > v=0. > > > o=CiscoSystemsSIP-GW-UserAgent 7543 5694 IN IP4 216.87.144.196. > > > s=SIP Call. > > > c=IN IP4 216.87.144.196. > > > t=0 0. > > > m=audio 16632 RTP/AVP 0 100. > > > a=rtpmap:0 PCMU/8000. > > > a=rtpmap:100 X-NSE/8000. > > > a=fmtp:100 192-194. > > > This is the 200 OK (response to the INVITE) message as delivered to the phone. I couldn't figure out what you were saying, so I went back to the ethereal trace. After the snom phone receives a 183 status message, it sends a PRACK to the PROXY. This PRACK is OKed, without a Record-route. The next message is an OK responding to the original INVITE, which does indeed have a Record-Route. So, you are saying that the OK to your PRACK needs a record route? I can do that I think, because the OK to the INVITE does indeed have a Record-route. I don't even know what a PRACK is for... ---greg -----Original Message----- From: Kevin [mailto:kmoroz@abpintl.com] Sent: Monday, March 03, 2003 5:19 PM To: Greg Fausak Cc: 'Robert Messer' Subject: FW: Snom phone (fwd) Hi Greg, Sorry it took so long to get back to you. Normally it is faster but as I stated the SIP inter- operability was last week which caused the delay. Looks like the issue is with the SER proxy. If I knew the specification deeper I should have been able to answer it myself. Engineering ccs Jiri on their response to me so he is aware of the issue. Regards, Kevin -----Original Message----- The phone updates the route until it receives a 2xx code. The 200 Ok response does not contain such a route therefore the phone uses the last route it receives in the 200 which is empty. Therefore, the phone MUST send the ACK directly to the gateway. § 12.1.2 of the RFC3261. The dialog is NOT established by the 401-challened request. The proxy can very easily solve the problem by putting itself into the routing path of the 200 Ok. [View Less]

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[Serusers] RE: Snom phone (fwd)
by Greg Fausak 04 Mar '03

04 Mar '03

Kevin, I initially sent the packet: > > > This is that packet that came from the last 200 OK <- PROXY: > > > > > > # > > > U 2003/02/24 07:56:52.503535 216.87.144.203:5060 -> > > 216.87.145.22:5060 > > > SIP/2.0 200 OK. > > > Via: SIP/2.0/UDP 216.87.145.22:5060;branch=z9hG4bK-ng5tokyx448r. > > > From: "snom man" <sip:4695461245@augustvoice.net>;tag=8u6ju8wxuc. > > > To: <sip:2143357976@augustvoice.… [View More]net;user=phone>;tag=3CBB0360-532. > > > Date: Mon, 24 Feb 2003 13:56:43 GMT. > > > Call-ID: 3c267202b6a8-lgseu8olovlp(a)216.87.145.22. > > > Server: Cisco-SIPGateway/IOS-12.x. > > > CSeq: 2 INVITE. > > > Session-Expires: 7200;refresher=uac. > > > Require: timer. > > > Allow-Events: telephone-event. > > > Contact: <sip:92143357976@216.87.144.196:5060;user=phone>. > > > Record-Route: <sip:2143357976@216.87.144.203;ftag=8u6ju8wxuc;lr>. > > > Content-Type: application/sdp. > > > Content-Length: 209. > > > . > > > v=0. > > > o=CiscoSystemsSIP-GW-UserAgent 7543 5694 IN IP4 216.87.144.196. > > > s=SIP Call. > > > c=IN IP4 216.87.144.196. > > > t=0 0. > > > m=audio 16632 RTP/AVP 0 100. > > > a=rtpmap:0 PCMU/8000. > > > a=rtpmap:100 X-NSE/8000. > > > a=fmtp:100 192-194. > > > This is the 200 OK (response to the INVITE) message as delivered to the phone. I couldn't figure out what you were saying, so I went back to the ethereal trace. After the snom phone receives a 183 status message, it sends a PRACK to the PROXY. This PRACK is OKed, without a Record-route. The next message is an OK responding to the original INVITE, which does indeed have a Record-Route. So, you are saying that the OK to your PRACK needs a record route? I can do that I think, because the OK to the INVITE does indeed have a Record-route. I don't even know what a PRACK is for... ---greg -----Original Message----- From: Kevin [mailto:kmoroz@abpintl.com] Sent: Monday, March 03, 2003 5:19 PM To: Greg Fausak Cc: 'Robert Messer' Subject: FW: Snom phone (fwd) Hi Greg, Sorry it took so long to get back to you. Normally it is faster but as I stated the SIP inter- operability was last week which caused the delay. Looks like the issue is with the SER proxy. If I knew the specification deeper I should have been able to answer it myself. Engineering ccs Jiri on their response to me so he is aware of the issue. Regards, Kevin -----Original Message----- The phone updates the route until it receives a 2xx code. The 200 Ok response does not contain such a route therefore the phone uses the last route it receives in the 200 which is empty. Therefore, the phone MUST send the ACK directly to the gateway. § 12.1.2 of the RFC3261. The dialog is NOT established by the 401-challened request. The proxy can very easily solve the problem by putting itself into the routing path of the 200 Ok. [View Less]

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[Serusers] New known problems with radius_auth module
by Bjoern Zuelch 03 Mar '03

03 Mar '03

Hi all, i have found some probs with the radius_auth module in ser. the ane in the function authorize() in file authorize.c : when calling the function check_nonce this function returns always that the two nonce's (given and new generated one are not the same. But if we compare the two nonce's manually both are the same. I replaced the memcmp function with the strcmp funtion,but nothings changes. So there is may a missing \0 at the end of the strings. Next problem is in the function … [View More]

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[Serusers] Problems with radius_auth(2)
by Bjoern Zuelch 03 Mar '03

03 Mar '03

Hi all, ich added following lines in avpair.c function rc_avpair_assign before "case PW_TYPE_STRING:": case PW_SIP_USER_ID: case PW_SIP_USER_REALM: case PW_SIP_USER_NONCE: case PW_SIP_USER_METHOD: case PW_SIP_USER_DIGEST_URI: case PW_SIP_USER_NONCE_COUNT: case PW_SIP_USER_QOP: case PW_SIP_USER_OPAQUE: case PW_SIP_USER_RESPONSE: case PW_SIP_USER_CNONCE: case PW_TYPE_STRING: this should help to build the radius-message. Bjoern

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[Serusers] threading weirdness?
by lgfausak＠august.net 02 Mar '03

02 Mar '03

Hi, I've been working on a problem which I thought was a postgres driver problem, but now I'm not so sure. I put a bunch of debug statements in the driver functions. I've grepped the debugs for db_init|db_close|connect_db|disconnect_db and here is what I have: -----------cut------------ Mar 1 23:59:49 rave ser[1803]: db_init(): sql://USER:PASSWORD@humpty.august.net:5432/THEDBNAME Mar 1 23:59:49 rave ser[1803]: connect_db(): here we go Mar 1 23:59:49 rave ser[1803]: connect_db(): we are … [View More]connected /tmp/debug.1803.out Mar 1 23:59:49 rave ser[1804]: db_close(): disconnect_db() Mar 1 23:59:49 rave ser[1804]: disconnect_db(): entry Mar 1 23:59:49 rave ser[1804]: disconnect_db(): PQfinish() Mar 1 23:59:49 rave ser[1804]: db_close(): free self Mar 1 23:59:49 rave ser[1804]: db_init(): sql://USER:PASSWORD@humpty.august.net:5432/THEDBNAME Mar 1 23:59:49 rave ser[1804]: connect_db(): here we go Mar 1 23:59:49 rave ser[1805]: db_close(): disconnect_db() Mar 1 23:59:49 rave ser[1805]: disconnect_db(): entry Mar 1 23:59:49 rave ser[1805]: disconnect_db(): PQfinish() Mar 1 23:59:49 rave ser[1805]: db_close(): free self Mar 1 23:59:49 rave ser[1805]: db_init(): sql://USER:PASSWORD@humpty.august.net:5432/THEDBNAME Mar 1 23:59:49 rave ser[1805]: connect_db(): here we go Mar 1 23:59:49 rave ser[1806]: db_close(): disconnect_db() Mar 1 23:59:49 rave ser[1806]: disconnect_db(): entry Mar 1 23:59:49 rave ser[1806]: disconnect_db(): PQfinish() Mar 1 23:59:49 rave ser[1806]: db_close(): free self Mar 1 23:59:49 rave ser[1804]: connect_db(): we are connected /tmp/debug.1804.out Mar 1 23:59:49 rave ser[1806]: db_init(): sql://USER:PASSWORD@humpty.august.net:5432/THEDBNAME Mar 1 23:59:49 rave ser[1804]: db_init(): sql://USER:PASSWORD@humpty.august.net:5432/THEDBNAME Mar 1 23:59:49 rave ser[1806]: connect_db(): here we go Mar 1 23:59:49 rave ser[1804]: connect_db(): here we go Mar 1 23:59:49 rave ser[1805]: connect_db(): we are connected /tmp/debug.1805.out Mar 1 23:59:49 rave ser[1805]: db_init(): sql://USER:PASSWORD@humpty.august.net:5432/THEDBNAME Mar 1 23:59:49 rave ser[1805]: connect_db(): here we go Mar 1 23:59:49 rave ser[1804]: connect_db(): we are connected /tmp/debug.1804.out Mar 1 23:59:50 rave ser[1805]: connect_db(): we are connected /tmp/debug.1805.out Mar 1 23:59:50 rave ser[1806]: connect_db(): we are connected /tmp/debug.1806.out Mar 1 23:59:50 rave ser[1806]: db_init(): sql://USER:PASSWORD@humpty.august.net:5432/THEDBNAME Mar 1 23:59:50 rave ser[1806]: connect_db(): here we go Mar 1 23:59:50 rave ser[1806]: connect_db(): we are connected /tmp/debug.1806.out Mar 2 00:00:50 rave ser[1807]: begin_transaction(): BEGIN error server closed the connection unexpectedly ^IThis probably means the server terminated abnormally ^Ibefore or while processing the request. -----------------cut-------------- The current processes that are running are: [root@rave sip_router]# ps -ax | grep ser 30060 ? S 0:00 /bin/sh bin/safe_mysqld --user=mysql 1803 ? S 0:00 ser -f /usr/local/etc/ser/serpost.cfg 1804 ? S 0:00 ser -f /usr/local/etc/ser/serpost.cfg 1805 ? S 0:00 ser -f /usr/local/etc/ser/serpost.cfg 1806 ? S 0:00 ser -f /usr/local/etc/ser/serpost.cfg 1807 ? S 0:00 ser -f /usr/local/etc/ser/serpost.cfg 1808 ? S 0:00 ser -f /usr/local/etc/ser/serpost.cfg -----------------cut------------------ Here is the top of my ser.cfg file: ---------------- debug=2 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=216.87.144.203 port=5060 children=2 fifo="/tmp/ser_fifo" alias="augustvoice.net" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/postgres.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("usrloc","db_mode",2) modparam("usrloc","db_url", "sql://USER:PASSWORD@humpty.august.net:5432/THEDBNAME") modparam("tm","retr_timer1p1", 2) # increase retry timeout modparam("tm","retr_timer1p2", 3) # increase retry timeout modparam("tm","retr_timer1p3", 6) # increase retry timeout modparam("acc", "log_level", 1) modparam("acc", "acc_flag", 1 ) modparam("acc", "report_ack", 0 ) modparam("auth","db_url", "sql://USER:PASSWORD@humpty.august.net:5432/THEDBNAME") # # this is the main routing block for augustvoice.net # this block can handle any routing request, local or foreign # route { # ... --------------------------- A couple of questions: 1) I state 'children=2', but I see 6 'processes'. I think linux does threads with processes, so am I seeing 2 'processes' with 3 threads per process??? 2) the debug output has the process ID and the action being performed. Either it is way out of order, or the postgres db driver is being asked to do things out of order. 3) Why is there all of the db_init(), db_close() stuff?? All of this stuff happens in the first second on startup. Surely that many opens and closes aren't really needed are they? 4) select statements seem to happen in one thread, and insert/delete statements seem to happen in another thread. What is that about? ---greg Greg Fausak [View Less]

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[Serusers] Some Module Questions
by Stephan Reiff-Marganiec 01 Mar '03

01 Mar '03

Hi, Context: I am currently writing a module that allows me to connect to a policy server -- not unlike CPL. It essentially routes the message out to the server and then receives a response. I get the message out ok, and will receive a complete sip message back. The module main function is called in the ser.cfg. Here are my questions: 1) how do I get a message that I have received -- say in char *msg_buf -- to be the message that is used for further routing? The policy server might have … [View More]

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[Serusers] Pkg_malloc
by Greg Fausak 01 Mar '03

01 Mar '03

Is pkg_malloc some sort of memory management thing? I mean, do I have to clean up, or does something come in and clean up behind me? Is there a parent/child memory mallocator available? It would make freeing easier. A single free of the parent rather than keeping track myself. I just don't want to be the source of a memory leak. ---greg

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