- sr-users - kamailio.org

[Serusers] error with setup
by Wolfe, Craig A 11 Mar '03

11 Mar '03

When I tryto go the page localhost/admin/index.php I get the following error. Warning: mysql_pconnect() [function.mysql-pconnect <http://www.php.net/function.mysql-pconnect>]: Can't connect to local MySQL server through socket '/tmp/mysql.sock' (2) in /usr/local/serweb/phplib/db_mysql.inc on line 73 Database error: pconnect(localhost, ser, $Password) failed. MySQL Error: () Session halted. Any help would ne greatly appreciated. Thanks Craig Wolfe

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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. What do you think? Should we attempt it? Does anyone have any preference on the language we use, e.g. JAVA, C++ ? Erik Lagerway Xten Networks Inc. 7170-515 West Hastings St Vancouver, BC V6B5K3 Ph. 1.604.878.0440 ext.5

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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(&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/

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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 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. What do you think? Should we attempt it? Does anyone have any preference on the language we use, e.g. JAVA, C++ ? Erik Lagerway Xten Networks Inc. 7170-515 West Hastings St Vancouver, BC V6B5K3 Ph. 1.604.878.0440 ext.5

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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 INVITE to sipc (139.153.254.34). SER (and ngrep) are running on 139.153.254.50. (registrations are going through fine and also invites the other way (from sipc to pingtel) go through ok). Thanks very much for your help! Regards, Mario -- Mario Kolberg phone: +44 (0)1786 46 7440 Lecturer in Computing Science fax : +44 (0)1786 46 4551 email: mko(a)cs.stir.ac.uk Department of Computing Science and Mathematics University of Stirling Stirling FK9 4LA Scotland, UK -- The University of Stirling is a university established in Scotland by charter at Stirling, FK9 4LA. Privileged/Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not disclose, copy or deliver this message to anyone and any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer do not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of the University of Stirling shall be understood as neither given nor endorsed by it.

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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 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

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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 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/

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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] 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

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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 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

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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" | 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

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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 under Linux 2.2 and have become longs under Linux 2.4. To take advantage of this, a recompilation under glibc-2.1.91 or later should suffice. (The kernel dis-tinguishes old and new calls by a IPC_64 flag in cmd.) Do I need kernel 2.4? Any hints? Have I missed some documentation? Jaime Hemmett

2 2

[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 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

1 0

[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. it would we great if first my sip ua would ring and it that times out, then my mobile phone would ring. -- juha ============ 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?

3 2

[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 }, (fixup_function[]){ 0 }, 2, /* Number of exported functions*/ NULL, /* Module parameter names */ NULL, /* Module parameter types */ NULL, /* Module parameter variable pointers */ 0, /* Number of module paramers */ mod_init, fi_response, (destroy_function) 0, 0, 0 }; I'm wondering if I correctly declared the function as I'm getting a warning during compilation about incompatible pointer types (4th last line in structure where the function is declared as response function). I really only need the message as parameter. What should the parameters be? This is for the CVS version of SER. Thanks very much for your help! Regards, Mario -- Mario Kolberg phone: +44 (0)1786 46 7440 Lecturer in Computing Science fax : +44 (0)1786 46 4551 email: mko(a)cs.stir.ac.uk Department of Computing Science and Mathematics University of Stirling Stirling FK9 4LA Scotland, UK -- The University of Stirling is a university established in Scotland by charter at Stirling, FK9 4LA. Privileged/Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not disclose, copy or deliver this message to anyone and any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer do not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of the University of Stirling shall be understood as neither given nor endorsed by it.

2 1

[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 Lecturer in Computing Science fax : +44 (0)1786 46 4551 email: mko(a)cs.stir.ac.uk Department of Computing Science and Mathematics University of Stirling Stirling FK9 4LA Scotland, UK -- The University of Stirling is a university established in Scotland by charter at Stirling, FK9 4LA. Privileged/Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not disclose, copy or deliver this message to anyone and any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer do not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of the University of Stirling shall be understood as neither given nor endorsed by it.

2 1

[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 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.

1 0

[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, ---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.

1 0

[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.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.

2 2

[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 check_response: On Solaris we have no freeradiusd ,so I replaced it through radius_authorize(...). If the function radius_authorize is called, the radiusmessage will be created through calling the function rc_avpair_add(...). All is fine if it will be standard radius-attributes. when the first sip-specific attribute is comming the rc_avpair_add funtions returns -1. I followed this problem and find out that in the radiusclient-lib. in function rc_avpair_assign(avpair.c)/rc_avpair_new(avpair.c) only attributes of the types PW_TYPE_STRING, PW_TYPE_DATE, PW_TYPE_INTEGER, PW_TYPE_IPADDR are reccognised. If a PW_SIP_USER_ID attribute or other specific Sip attributes will be added to the radiusmessage, this function returns with error message "UNKNOWN ATTIBUTE TYPE" and no radius-authentification is done.... .... RADIUS: rc_avpair_new MALLOC erfolgreich RADIUS: rc_avpair_assign type=0 RADIUS rc_avpair_assign result=0 RADIUS: rc_avpair_new unknown attribute 110 RADIUS: VALUEPAIR insert fehgeschlagen.!!! check_cred(): returnvalue of radius_authorize()=-1 .... One possibility is that, I add the PW_SIP_* attributes to the radiusclient and look, if it run. The next Problem may be, that the radius server does not understand the sip specific attributes, if we not use freeradius. Gruß Bjoern

2 1

[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

1 0

[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 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

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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 changed any header fields or even created a (or several) completely new message(s). 2) can the module send back intermediate messages, such as 1xx info, while it is still continuing its work? This is to provide the user with feedback should things take while ... 3) I also want the module to react to SIP response messages (4xx and so). I assumed providing a response_function would do this, but apparently not (I had a simple one which was just LOGging some text to the system log, but it never did create any entries). So how can I do this? thanks, Stephan -- Dr Stephan Reiff-Marganiec Research Fellow Department of Computing Science; University of Stirling email: srm(a)cs.stir.ac.uk tel: 01786 46 7448 -- The University of Stirling is a university established in Scotland by charter at Stirling, FK9 4LA. Privileged/Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not disclose, copy or deliver this message to anyone and any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer do not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of the University of Stirling shall be understood as neither given nor endorsed by it.

2 4

[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

2 3

[Serusers] no mysql authentication on solaris8
by Kamil Ince 01 Mar '03

01 Mar '03

thx for the help so far, but the adventure continues: compiled latest ser from CVS on sol8 from scratch.. and still using the same ser.cfg but can't start ser anymore, that's what the log shows: 0(15264) loading module /usr/local/lib/ser/modules/mysql.so 0(15264) loading module /usr/local/lib/ser/modules/sl.so 0(15264) loading module /usr/local/lib/ser/modules/tm.so 0(15264) loading module /usr/local/lib/ser/modules/rr.so 0(15264) loading module /usr/local/lib/ser/modules/maxfwd.so 0(15264) loading module /usr/local/lib/ser/modules/usrloc.so 0(15264) loading module /usr/local/lib/ser/modules/registrar.so 0(15264) loading module /usr/local/lib/ser/modules/auth.so 0(15264) find_param_export: found <db_mode> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(15264) find_param_export: found <calculate_ha1> in module auth [/usr/local/lib/ser/modules/auth.so] 0(15264) find_param_export: found <password_column> in module auth [/usr/local/lib/ser/modules/auth.so] 0(15264) find_export: found <mf_process_maxfwd_header> in module maxfwd_module [/usr/local/lib/ser/modules/maxfwd.so] 0(15264) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(15264) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(15264) find_export: <rewriteFromRoute> not found 0(15264) parse error (79,19-20): unknown command, missing loadmodule? 0(15264) find_export: found <www_authorize> in module auth [/usr/local/lib/ser/modules/auth.so] 0(15264) find_export: found <www_challenge> in module auth [/usr/local/lib/ser/modules/auth.so] 0(15264) find_export: found <save> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(15264) find_export: found <lookup> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(15264) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(15264) find_export: found <t_relay> in module tm [/usr/local/lib/ser/modules/tm.so] 0(15264) find_export: found <sl_reply_error> in module sl_module [/usr/local/lib/ser/modules/sl.so] ERROR: bad config file (1 errors) 0(15264) INFO: signal 15 received 0(15264) Memory status (pkg): 0(15264) qm_status (72d38): 0(15264) heap size= 1022872 0(15264) used= 0, used+overhead=25704, free=997168 0(15264) max used (+overhead)= 25704 0(15264) dumping all allocked. fragments: 0(15264) dumping free list stats : 0(15264) hash= 521. fragments no.: 1 0(15264) ----------------------------- anyone a pointer to what i have missed? tia again -k

2 2

[Serusers] ser.cfg overwrite
by lgfausak＠august.net 28 Feb '03

28 Feb '03

i was messing around with the postgres build over the last couple of days. I've rebuilt it several times. A few make installs later I went back to edit my ser.cfg file, it was overwritten by the new installation. ouch... ---greg Greg Fausak

2 1

[Serusers] IPv6 SIP UA
by jaime.gill＠orange.co.uk 28 Feb '03

28 Feb '03

Hi, Is anyone aware of any IPv6 compliant SIP softphone? When will sipsak implement this feature? Jaime ******************************************************************************* 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. *******************************************************************************

3 2

[Serusers] auth.so modules and digest auth
by Benny Ho 28 Feb '03

28 Feb '03

I load the auth.so and the default config. Uncomment those to use auth.so But when I call up two user agents not register in the database. The only difference is there a warning for registration. They are still calling each other. Why?

2 1

[Serusers] postgres db support
by lgfausak＠august.net 27 Feb '03

27 Feb '03

Has anybody started on support for Postgres? I've checked out the layering and it seems to be a fairly modular project. I haven't coded in C for quite some time, but I think I can handle it. Any hints? Should I even try? The only thing I can see having problems with is the build/configure system. How do I hook in a new module? How are include directories and new libraries added? I was pretty good with make, gmake, imake in my day but I can't seem to find the head of the Make tree! ---greg Greg Fausak

4 4

[Serusers] what makes a transaction?
by lgfausak＠august.net 27 Feb '03

27 Feb '03

I'm trying to understand stateful vs. stateless transactions. A stateless 'transaction' simply lasts as long as it takes to process the current sip request. It is then forgotten. However, a stateful transaction I'm having a little problem grasping. For instance, if I t_relay() an INVITE, logically the transaction is started....so when the response comes it isn't delivered to the main route {} block, but instead it is delivered to the awaiting stateful t_relay(), probably keyed by callid. So, for instance: INVITE <- 401 Unauthorized -> Is this entire sequence a transaction? Right after this the UA sends an ACK, this is a transaction unto itself? ACK <- The UA then sends : INVITE <- 100 Trying -> 180 Ringing -> 200 OK -> All of these messages are a 'transaction' handled by t_relay()? The next transaction is: ACK <- Finally, the last transaction would be (as an example): BYE <- 200 OK -> So the ser route{} block would need to recognize and handle INVITE ACK BYE in this context. Is that correct? If I was running 'statelessly' I would need to handle and send every single message listed? ---greg

3 2

[Serusers] Segmentation fault: fifo_server fgets failed: Illegal seek
by jaime.gill＠orange.co.uk 27 Feb '03

27 Feb '03

Hi,

1 0

[Serusers] ACC's ACK
by lgfausak＠august.net 26 Feb '03

26 Feb '03

I realize the accounting logging of ACK is only an attempt, but ... It does seem fairly important in determining when the actual call started. Here is some filtered output from the syslog accounting output file: --- -------- -------- -------- --- ---------- ---------- Call INVITE ACC BYE Length From To --- -------- -------- -------- --- ---------- ---------- 1 18:50:39 18:50:39 18:50:48 9 4695461243 -> 9723236598 2 18:53:19 18:53:19 19:08:46 927 4695461243 -> 9723937976 3 20:14:00 0 4695461246 -> 4695461235 4 20:16:41 20:16:48 7 4695461246 -> 4695461235 5 20:19:09 20:19:17 8 4695461246 -> 4695461235 6 20:30:18 20:30:23 5 4695461235 -> 4695461235 7 20:30:51 20:31:01 10 4695461246 -> 4695461235 8 20:37:14 20:37:17 3 4695461246 -> 4695461235 9 21:02:08 21:03:28 80 4695461235 -> 4695461240 10 21:21:26 21:21:26 21:36:33 907 4695461235 -> 4695461243 11 22:42:03 22:42:03 22:52:23 620 4695461235 -> 4695461235 12 22:59:22 22:59:33 11 4695461246 -> 4695461235 13 23:00:46 23:01:00 14 4695461246 -> 4695461235 14 23:08:11 23:08:21 10 4695461246 -> 4695461235 15 23:19:37 23:19:37 23:19:41 4 4695461235 -> 4695461235 16 23:22:49 23:22:56 7 4695461235 -> 4695461235 17 23:28:55 23:28:55 23:29:07 12 4695461235 -> 4695461243 18 23:29:18 23:29:18 23:42:53 815 4695461235 -> 4695461243 19 09:08:11 09:08:11 09:08:52 41 9725678129 -> 4695461236 20 09:09:50 09:10:38 48 4695461236 -> 99727139968 21 09:11:15 09:12:00 45 4695461236 -> 99727139968 22 09:12:35 09:12:35 09:13:39 64 2143357976 -> 4695461240 23 09:18:57 09:18:57 09:19:29 32 4695461235 -> 9724620762 24 09:23:17 09:24:25 68 4695461235 -> 4695461240 --- -------- -------- -------- --- ---------- ---------- I note the ACK times are all the same as the INVITE times. Isn't the wrong ACK being recorded? Is this a problem with my .cfg? A much more useful ACK would be the second one (when the far end ACKs after the 180 Ringing and 200 OK messages. A typical call, from UA perspective: INVITE -> 401 unauth.. <- ACK -> (evidently, the one being accounted) INVITE (with cred) -> 100 Trying <- 180 Ringing <- 200 OK <- ACK -> (the ACK that SHOULD be accounted, IMHO) ... ---greg

3 3

Re: [Serusers] no mysql authentication on solaris8
by Jiri Kuthan 26 Feb '03

26 Feb '03

Thanks a lot. It seems that the log line > 5(391) authorize(): Invalid nonce value received, very suspicious ! is the key -- Jan will probably handle that better than myself. Are you sure you applied the solaris/digest patch? (See nonce validation in ISSUES on SER's website. See also the topmost solaris-specific issue -- better take SER sources and recompile them without DBG_QM_MALLOC in Makefile.defs) If things don't work then, send me please captured SIP messages too. (Use tools like "ngrep port 5060" to record them.) Thanks, -Jiri

2 1

[Serusers] OT: sip ua question
by Tomas Björklund 26 Feb '03

26 Feb '03

Hi, A bit offtopic perhaps, but are there any good win32 sip ua's out there? I'm interested in both commercial and opensource software (my friends doesn't seem to whant to buy Cisco 7960's ... wonder why :) / Tomas

3 2

[Serusers] Redundant forwarding
by Jamin W. Collins 25 Feb '03

25 Feb '03

This is probably a very simple question. I'm trying to create a very simplistic configuration that simply forwards the call to one of two fixed destinations. Essentially, it forwards all calls to server A and if that fails, forwards the call to server B. Forwarding to server A is no problem. Within route, I simply have: 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; }; rewriteFromRoute(); if (!forward("192.168.1.1")) { sl_reply_error(); }; This works perfectly, but only forwards to one server. I tried changing the last bit to: if (!forward("192.168.1.1")) { if (!forward("192.168.0.1")) { sl_reply_error(); }; }; However, this results in "ERROR: bad config file (5 errors)". Any suggestions? Is something like this possible? -- Jamin W. Collins

3 13

RE: [Serusers] problem in using acc.so and radius_acc.so and auth.so
by Benny Ho 25 Feb '03

25 Feb '03

Dear Jiri, I have successfully compile the acc.so with SQL support. And I run a number of testing on it. It seems works fine but I encounter the following error which crach the ser. =========================================================== [root@md benny]# 0(3545) mod_init(): Database connection opened successfuly Maxfwd module- initializing rr - initializing acc - initializing stateless - initializing mysql - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 9(3565) INFO: fifo process starting: 3565 9(3565) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 10(3574) ACC: transaction answered: method=INVITE, i-uri=sip:1000@md.com;user=ph one, o-uri=sip:1000@192.168.10.18:5000;user=phone, call_id= 426a69eafe94e7923666 1278d86d5024(a)192.168.10.18, from= UserAgent-1<sip:1000@192.168.10.18:5000;user=p hone>, code=408 10(3574) use_table(): Invalid parameter value 0(3545) child process 3574 exited by a signal 11 0(3545) core was not generated 0(3545) INFO: terminating due to SIGCHLD ========================================================================= It seems to be a crash on use_table() when code=408, I tried to unanswer to call or call to itself, and this problem generated. Benny -----Original Message----- From: Jiri Kuthan [mailto:jiri@iptel.org] Sent: Monday, February 24, 2003 6:30 PM To: Benny Ho Subject: RE: [Serusers] problem in using acc.so and radius_acc.so and auth.so First of all, note well: http://lists.iptel.org/pipermail/serusers/2003-February/000493.html At 04:36 AM 2/24/2003, Benny Ho wrote: >Could you tell me how to enable SQL_ACC def? and setting teh db_url >parameter? uncomment the "#DEFS+=-DSQL_ACC" line in modules/acc/Makefile >I cannot find any doc for mysql modules. mysql module is not used from ser scripts directly -- modules which rely on mysql use it internally. acc w/mysql has not been released yet, it is work in progress on CVS and it has not been documented. -jiri

1 0

Re: [Serusers] Redundant forwarding
by lgfausak＠august.net 24 Feb '03

24 Feb '03

I'm currently working with SNOM to get this issue fixed. I'll report when I know something. It might not hurt if someone else reported it to them! ---greg Greg Fausak > >At 09:34 PM 2/24/2003, Jamin W. Collins wrote: >>On Mon, Feb 24, 2003 at 08:54:00PM +0100, Jiri Kuthan wrote: >>> What is CIC? >> >>A PBX made by I3 (www.inin.com) >> >>> Send us network dumps please -- that will make it easier to understand >>> for us. You can for example use "ngrep port 5060". >> >>Network dump from the SER server is attached. > >snom phone is a common suspect again. > >The first signaling transaction looks ok. The only unusual thing is that >for some reason, the ININ server advertises 192.168.0.9 as media receiver >whereas its signaling lives at 192.168.1.16. Is media decomposed? > >The really suspicious is the second transaction. The snom phone sitting >at .36 advertises contacts with .34 in its replies. That makes the >caller (ININ) to send an ACK to .34. Snom does not received it and >keeps soliciting it by retransmitting 200s. > >-Jiri > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[Serusers] Some thoughts and questions
by Tomas Björklund 24 Feb '03

24 Feb '03

Hi, I've been looking at ser for a couple of days now and there some things I can't find in the documentation. .1. Logging funktionality I'm used to being able to log more detail than what I can get out ser. I'm problably missing something in the documentation, but what I want is something like the following timestamp ser[pid] Incoming packet from ip:port [dump of packet (only the sip header/body)] timestamp ser[pid] Invite from [] to [] timestamp ser[pid] After rewrite: from [] to [] timestamp ser[pid] Relaying packet to [pstn-gw/other sip server] / timestamp ser[pid] Handling request for local user [] Can you send arguments to the log() function in the configuration file for example log(1,"Call from %s", uri:from); ?? .2. Rewrite rules Is it possible to rewrite the from uri? For example when I [sip:tomas@krixor.xy.org] want to call a pstn number I must rewrite the from address to [sip:myphonenumber@krixor.xy.org] before I send the request to the pstn gateway And when a incoming call comes from the pstn gateway I want to rewrite the from address from [<sip:number@ip-of-pstn-gateway>] to {"PSTN Luser" <sip:number@pstn-gw.mydomain.com>] I haven't looked at the source in the CVS yet, but I aim to do so soon, I saw in the list archive that you have lots of nice new features planned for the next release. ENUM support is one of the better ones :) Anyway, looking forward to trying to help with the development of this software. Best regards, Thomas Björklund

3 5

[Serusers] Rewriting URI in the Contact field
by Maxim Sobolev 24 Feb '03

24 Feb '03

Folks, I need an advise on how to better implement one feature, which isn't currently present in SER. We need to allow UAs behind NAT properly register with the registrar - by "properly" I mean that host:port portion of URI in Contact field should not be used, but host:port the request came from should be used instead. By definition we know that those UAs will support symmetric SIP signalling, so that this scheme will work just fine. In my opinion there are two ways to do it: either add new rewritecontact* family of functions similar to rewritehost ones. or add a new flag for the save() function. This is where I need your help - which implementation looks better for you (or maybe you have even some better idea), since we are really interested in inclusion of our changes into the mainline to reduce our local hacks. Regards, Maxim

8 44

[Serusers] Some patches for integration
by Maxim Sobolev 24 Feb '03

24 Feb '03

Folks, Attached please find several patches for ser 0.8.10. They do the following: 1. Removes hardcoded `/usr/local' from Makefiles and replaces it with LOCALBASE variable setable in the top Makefile.defs. This should make life of packagers easier. 2. Allows to override PREFIX variable with something else from environment. 3. Adds support for FreeBSD doc/man directories layout. 4. Adds some missed -I's into radius modules. 5. Fixes radius modules to use the same symbolic avp names as in dictionary. 6. Adds support for recording source port in msg structure and adds it in "Noisy feedback". This should be extremely useful for debugging various SIP&NAT scenarios and later could be re-used to implement `rport' support described in my previous message. It would be nice to have those patches integrated into the next release. Thanks! -Maxim

4 5

[Serusers] Record-route:/Contact: conflict?
by Greg Fausak 24 Feb '03

24 Feb '03

I've got a packet that looks like: # 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. The main thing to note is the packet is sent to a SNOM phone from my PROXY server. Curious, when the SNOM phone responds it is responding to 216.87.144.196, which is my GATEWAY. My GATEWAY is configured to only listen to the PROXY server, so it drops the packet. In the packet above, the Record-route line indicates that 216.87.144.203 should be responded to, which is my PROXY. Also in the packet above, there is a 'Contact:' line that indicates the GATEWAY at 216.87.144.196 should be contacted. Obviously the SNOM phone is getting the address from the Contact: line (or maybe the SDP) rather than the Record-route: line. Is it doing so incorrectly, or do I have a bug in my GATEWAY, or does the PROXY need to strip out the Contact: line?

2 1

[Serusers] SER parser benchmarks
by Sebastien Pierre 24 Feb '03

24 Feb '03

Hi! I was just wondering if you had some benchmark for the SER parser... as SER does not seem to offer a libser, it makes it more difficult to write benchmarking programs :/ TIA, -- Sebastien

2 1

Re: [Serusers] FCP support in SER: Modifying SDP
by jaime.gill＠orange.co.uk 24 Feb '03

24 Feb '03

Thanks Jiri, I'll have a look at the CVS version. I'm sure there will be more hacking to do only to make it compile.... :( However, got the replacement of the "c=.." part working. I'll test this as soon as I can. Jaime Jiri Kuthan <jiri(a)iptel.org> on 24/02/2003 13:20:00 To: Jaime GILL/EN/HTLUK@HTLUK Nils Ohlmeier <nils(a)ohlmeier.de> cc: Jan Janak <J.Janak(a)sh.cvut.cz> serusers(a)lists.iptel.org Subject: Re: [Serusers] FCP support in SER: Modifying SDP At 12:50 PM 2/24/2003, jaime.gill(a)orange.co.uk wrote: [...] >It think the problem is in the replacement of the SDP information. The first >occurrence of the IP address in "v= " and the port in "m= " in the SDP get >replaced, but the second IP in "c=" is not. nit: it's not the "v=" line, but "o=" ('owner') line which you are replacing. However, you are not probably worried so much about this one -- it maintains primarily a (not widely utilizied) identification purpose. All "c=" occurences do matter. (In addition to port numbers in "m=" lines.) [...] >I have been trying to understand how the proxy builds the forwarded message from >the old one, and realised that for the Via replacement (or adding of more >params), I need to be using a string called add_to_branch_s and >add_to_branch_len (so ignore the replace_via implementation in the current >tar.gz). I suggest you used the mhomed option (available only on CVS). The issue is you need to print the correct IP address in Via on multihomed host. With mhomed enabled, IP routing is utilized to determine the right IP address. Let me know if you need something more for getting Via right. >But for the SDP, whenever I work with get_body, it does not modify it >appropriately. So currently, I'm using msg->orig to get to the initial message, >search for certain IP4 and audio strings and replace them with the information >provided by the fcp server. That means, in the case of the SDP, 2 IP address >replacements (in v=.. and c=..) and 1 port replacement (in m=..). As I mentioned >before, I only manged to change the v=.. and m=... Whenever I try to replace >more than one appearance, strange things happen, like strings in non expected >places, like Via, and cannot work out why. So my question is an open one:): what >is the best way to change the SDP part? I suggest here too -- use the CVS version. It has departed from the use of the buffers (orig and buf) -- we have now just one buffer (buf) without any zero termination. Previously, the two buffers and 0-termination caused lot of issues, some of them possibly annoying you right now. Look at the textops/replace_all action (only on CVS too) to see how to replace multiple occurences of a string in SIP messages. (Caution: you will eventually need to calculate new SDP body size and change content-length too.) >The other of my questions is whether all this mess with NAT's will get solved >when the proxy supports TCP, The major problem is media, which will keep using UDP. >and whether this is the best approach to solve the >SIP through NAT/FW problem. As all NAT traversal methods -- none of them is perfect, each has cons and pros. The benefit of FCP is that once fcpd works, maintenance of the SIP code is easier in user space. Also, you can better couple your pinhole policy with SER's SIP-layer policy. >For example, how about a nathelper module for >netfilter/iptables that gets this working, in the same manner as IRC or ftp >currently? Does anybody know about any work progressing this for linux/FreeBSD? I'm not aware of such. There is Billy Biggs masquerading module, but it is pretty old and no longer maintained. -Jiri _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ******************************************************************************* 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. *******************************************************************************

1 0

Re: [Serusers] FCP support in SER: Modifying SDP
by jaime.gill＠orange.co.uk 24 Feb '03

24 Feb '03

Hi Nils, Once again I'm writing to ask about more things and to give you an update on what is happening with the FCP tests... We have set up a simple scenario with 2 proxies separated by a firewall/NAT, and 2 UA, one within (UA1) and another outside the firewall (UA2). One UA registers with the natted proxy (UA1), the other with the "public" proxy (UA2). At the moment, SIP messages go forwards and backwards without problems, but media is not flowing across the firewall. It think the problem is in the replacement of the SDP information. The first occurrence of the IP address in "v= " and the port in "m= " in the SDP get replaced, but the second IP in "c=" is not. I have been trying all sorts of things, but no joy :( . Here are the INVITE messages in more detail. I also include the latest fcp module for you to play with it. ---------- UA1 to proxy message ------------------ U 172.21.68.78:1129 -> 192.168.6.153:5060 INVITE sip:jaime@asereje.orange.co.uk SIP/2.0 Call-ID: 5812832001907970791(a)172.21.68.78 Content-Length: 121 Content-Type: application/sdp To: sip:jaime@asereje.orange.co.uk From: sip:pepe@asereje.orange.co.uk;tag=-779729009 Contact: sip:pepe@172.21.68.78:5061 CSeq: 1 INVITE Via: SIP/2.0/UDP 172.21.68.78:5061;branch=AC15444E13C5000000F38F819DE5-2*0 v=0..o=- 1046084768435 1046084768465 IN IP4 172.21.68.78 s=- c=IN IP4 172.21.68.78 t=0 0 m=audio 5006 RTP/AVP 8 3 0 ---------- End of UA1 to proxy message ----------------- -------- Proxy to UA2 message --------------------- U 192.168.6.153:5060 -> 172.21.68.78:15592 INVITE sip:172.21.68.78:15592 SIP/2.0 Call-ID: 5812832001907970791(a)172.21.68.78 Content-Length: 121 Content-Type: application/sdp To: sip:jaime@asereje.orange.co.uk From: sip:pepe@asereje.orange.co.uk;tag=-779729009 Contact:<sip:192.168.0.1:33240> CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.6.153;branch=z9hG4bK1019.21c52996.0 Via: SIP/2.0/UDP 172.21.68.78:5061;branch=AC15444E13C5000000F38F819DE5-2*0 v=0..o=- 1046084768435 1046084768465 IN IP4 192.168.0.1 s=- c=IN IP4 172.21.68.78 <--- Need to change this as well!!! t=0 0 m=audio 33240 RTP/AVP 8 3 0 ------------ End of Proxy to UA2 message ------------------- (See attached file: fcp-module210203.tar.gz) I have been trying to understand how the proxy builds the forwarded message from the old one, and realised that for the Via replacement (or adding of more params), I need to be using a string called add_to_branch_s and add_to_branch_len (so ignore the replace_via implementation in the current tar.gz). But for the SDP, whenever I work with get_body, it does not modify it appropriately. So currently, I'm using msg->orig to get to the initial message, search for certain IP4 and audio strings and replace them with the information provided by the fcp server. That means, in the case of the SDP, 2 IP address replacements (in v=.. and c=..) and 1 port replacement (in m=..). As I mentioned before, I only manged to change the v=.. and m=... Whenever I try to replace more than one appearance, strange things happen, like strings in non expected places, like Via, and cannot work out why. So my question is an open one:): what is the best way to change the SDP part? The other of my questions is whether all this mess with NAT's will get solved when the proxy supports TCP, and whether this is the best approach to solve the SIP through NAT/FW problem. For example, how about a nathelper module for netfilter/iptables that gets this working, in the same manner as IRC or ftp currently? Does anybody know about any work progressing this for linux/FreeBSD? Greetings, Jaime Nils Ohlmeier <nils(a)ohlmeier.de> on 18/02/2003 02:58:47 To: Jaime GILL/EN/HTLUK@HTLUK cc: Jan Janak <J.Janak(a)sh.cvut.cz> Jiri Kuthan <jiri(a)iptel.org> Subject: Re: [Serusers] FCP support in SER: Modifying SDP Hi Jaime, debugging without the code is really hard :-) But maybe your problme with SDP is correlated to a bug in the Via header which i marked below. Greetings Nils On Monday 17 February 2003 11:58, jaime.gill(a)orange.co.uk wrote: > U 192.168.6.153:5060 -> 172.21.68.78:5061 > INVITE sip:pepe@172.21.68.78:5061 SIP/2.0..Via: SIP/2.0/UDP > 192.168.6.153;b ranch=z9hG4bKb848.8a014f84.0..Via: SIP/2.0/UDP > 192.168.0.1192.168.0.1:9439. .From: "jaime" ^^^^^^^^^^^^^^^^^^^^^^ Here you inserted the external IP twice. Maybe your SDP replacer did this? > <sip:jaime@asereje.orange.co.uk>;tag=8c20540f-4259-11d7-9cc5 > -00065b4c11cb..To: <sip:pepe@asereje.orange.co.uk>..Call-ID: 8c205410-4259- > 11d7-9cc5-00065b4c11cb@172.21.68.78..CSeq: 1 INVITE..Contact:<sip:192.168.0 > .1:33186>.User-Agent: Windows RTC/1.0..Content-Type: application/sdp..Conte > nt-Length: 211....v=0..o=gill_j 0 0 IN IP4 172.21.68.78..s=session..c=IN IP > 4 172.21.68.78..b=CT:1000..t=0 0..m=audio 33186 RTP/AVP 97 0 8 4..a=rtpmap: > 97 red/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:4 G723/80 > 00.. -- gpg-key: http://www.ohlmeier.org/public_key.asc ******************************************************************************* 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. *******************************************************************************

2 1

[Serusers] Cannot add "alias" mysql DB (using default passwords)
by Lenny Tropiano 24 Feb '03

24 Feb '03

I have turned on mysql authentication, and was able to add users, but aliases complain that "table" doesn't exist: $ serctl add lenny abcdef lenny(a)voiping.com MySql Password: new user added $ serctl alias add 1234 sip:lenny@voiping.com sip:lenny@voiping.com 400 Table 'aliases' Not Found $ mysql -p Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 243 to server version: 3.23.53 Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql> connect ser; Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Connection id: 244 Current database: ser mysql> show tables; +-----------------+ | Tables_in_ser | +-----------------+ | acc | | active_sessions | | aliases | | config | | event | | grp | | location | | missed_calls | | pending | | phonebook | | reserved | | silo | | subscriber | | version | +-----------------+ 14 rows in set (0.00 sec) mysql> --- Lenny Tropiano E-mail: lenny(a)voiping.com Partner, Networking Specialist Pager: pager-lenny(a)voiping.com VoIPing, LLC URL: http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 Mobile: 512-698-VOIP [8647]

5 6

[Serusers] On CVS, Radius and Generally SER Status (2003/II)
by Jiri Kuthan 23 Feb '03

23 Feb '03

I'm observing increasing number of downloads from CVS and would like to issue few disclaimers to prevent any disappointments. The CVS code is work in progress -- it has not been integrated, tested, it is not complete and we cannot provide help on incomplete software. Also, folks should know that compilation of one of the most frequently asked features, Radius support, is disabled on purpose. It is an external contribution which deserves a clean-up, we are working on it but we are not there yet. My previous forecast for the upcoming release is turning out to be too optimistic. We decided to carry out some more clean-ups and the work takes time. Anyway, we think it is better to clean up now than later -- interest in ser is growing, and we need to keep the codebase sane despite all the new features. -Jiri -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Serusers] problem in using acc.so and radius_acc.so and auth.so
by Benny Ho 23 Feb '03

23 Feb '03

Dear all, after days of trial, I finally load radius_acc.so in my ser. But when I testing the modules with the new routing config. Nothing happen. I follow the sample config in the CVS as follow. I check the radius server with radius -x, no packet seem to send to radius server. I also check the Mysql database, no entry is wrote to acc table. Why? Can anyone tell me what's wrong? ================== [benny@md benny]$ more /etc/ser/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=3 # 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" [benny@md benny]$ cat /etc/ser/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=3 # 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" # Uncomment this to use radius for accounting loadmodule "/usr/lib/ser/modules/radius_acc.so" loadmodule "//usr/lib/ser/modules/acc.so" #--- radius_ acc.so params --- modparam("radius_acc","log_level",1) modparam("radius_acc","acc_flag",1) modparam("radius_acc","report_ack",1) modparam("radius_acc","early_media",0) modparam("radius_acc","failed_transactions",0) #--- acc.so params --- modparam("acc","log_level",1) modparam("acc","acc_flag",1) modparam("acc","report_ack",1) modparam("acc","early_media",1) modparam("acc","failed_transactions",1) 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) #modparam("auth", "calculate_ha1", no) # # 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 (method=="INVITE") { addRecordRoute(); #} else { 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("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; #labeled all transaction for accouting setflag(1); #record-route INVITES to make sure BYEs will visit our server too #if (method=="INVITE") addRecordRoute(); # forward to current uri now if (!t_relay()) { sl_reply_error(); }; } [benny@md benny]$ ==================================

2 1

[Serusers] NAT Traversal Technique Survey
by Jiri Kuthan 23 Feb '03

23 Feb '03

Folks, please help me -- share with me techniques for NAT traversal you use and have hands-on experience with. People repeatedly ask about it, and I'd like to create an FAQ that reflects deployment experience and as wide user feed-back as possible. Just tell me the technique you use, its requirements, limitations, the devices it is known (not) to work with, why you prefer one method over the other, etc. I'll then try to compile it in an FAQ. So please send me an e-mail, an example is attached. I will appreciate any practical details. Thank you, -Jiri ---------------------------------------------------------------- technique: using symmetric communication requirements: phone devices that support symmetric communication; existing species: Cisco's ATA configuration practice: ATAs need to be configured to advertise public address in signaling, or learn it from REGISTER replies; alternatively, one can rewrite signaling using ser's nethelper module; one needs to rewrite SIP anyway because ATAs don't advertise their symmetricity; see www.foo.bar for info on configuring ATA... limitations: non-symmetric devices, like Messenger don't work; misc: ATA has no display, that's why I am anxiously waiting for more vendors to support symmetric signaling ---------------------------------------------------------------- technique: UPnP requirements: NATs and phones with UPnP support; Messenger and snom are known to support UPnP; there is linux support for it configuration practice: of course, upnp requires by definition no configuration ;-) (I'm not serious -- anyone actually tried it?) ---------------------------------------------------------------- technique: geek tweaks: set-up port forwading manually configuration practice: you need to configure NATs to split its public-side port numbers accross your private-side phones, and configure the phones (if they allows so) to use these port numbers; also, phones need to be configured to use publicly reachable address in their payloads requriements: configurable NATs (many residental NATs are configurable) and configurable phones (ATAs do that, I heard pingtel did it too) ---------------------------------------------------------------- technique: ALG requirements: SIP-capable NAT (like Intertex or Cisco/PIX) issues: intertex freezes my ssh connections after some time on-line and elderly models don't like all Ethernet devices; when things don't work, the red-button off-on helps sometimes ---------------------------------------------------------------- technique: STUN requirements: STUN-enabled phone (like k-phone, snom) limitations: doesn't work over symmetric NATs (words-of-mouth propaganda has been telling me that many residential NATs are fortunately not symmetric, but I don't know how objective this information really is) ---------------------------------------------------------------- -- Jiri Kuthan http://iptel.org/~jiri/

4 7

[Serusers] Windows- Sms gateway
by ant buckland 21 Feb '03

21 Feb '03

Hi- I am a very Junior designer, and as part of my efforts to impress my boss, I am trying to create a simple email to sms application. Sadly, here in South Africa, the cellular service providers do not provide an sms gateway to the public, and so I am researching the iptel.org SIP Express Router. I need to know the following: after reading the iptel.org website, I would like to know whether it is possible to install the application on a Windows 2000 server, and run the gateway. From what I read, apparently only the core application can be installed... and this does not appear to include the gateway? Obviously, I will pass the project on to a programmer who knows more, so you will not be plagued by too-simple questions from an inexperienced designer... but I would appreciate any assistance/info that will allow me to convince my boss. Thank you! ph (021) 685 6499 anthony_buckland(a)yahoo.co.uk 70 Strubens Rd, Observatory, Cape Town --------------------------------- With Yahoo! Mail you can get a bigger mailbox -- choose a size that fits your needs

2 1

[Serusers] Absorbing provisional responses
by Sebastien Pierre 21 Feb '03

21 Feb '03

Hi all, This message is somewhat related to a preceding one (Simple SER Use Case), because I realised I did not ask the right question. Here it is: I want to make a proxy that will try to locate the callee by sequentially trying n different UA, by sending an INVITE and waiting for either OK or timeout to occur. On timeout, the proxy will try the next UA. The detail here is that the proxy sends a "100 Trying" to the caller *before* contacting the various UA, and that *no more* "100 Trying" are forwarded by the proxy, which absorbs those responses from contacted UA. Jiri indicated that the configuration presented in <http://www.iptel.org/ser/doc/seruser-html/x510.html#AEN616> realised the scenario I was mentioning... but it seems to me that the proxy does not send and then absorb "100" responses, but rather relays this responses to the caller. This would lead to the caller receiving as many "100" responses as contacted UAs. Is it possible to do this with SER ? TIA, -- Sebastien

3 4

[Serusers] Crash in SER
by Maxim Sobolev 21 Feb '03

21 Feb '03

Hi, We've observed the following crash in ser. Do you have any ideas what the cause could be? I'm keeping corefile and binary compiled with debug symbols, so that don't hesitate to contact me if any additional information is necessary. Thank you in advance! -Maxim Script started on Fri Feb 21 05:43:09 2003 bash-2.05a$ gdb ~/sip/ser/work/ser-0.8.10/ser ser.core GNU gdb 4.18 (FreeBSD) Copyright 1998 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type "show copying" to see the conditions. There is absolutely no warranty for GDB. Type "show warranty" for details. This GDB was configured as "i386-unknown-freebsd"...Deprecated bfd_read called at /usr/src/gnu/usr.bin/binutils/gdb/../../../../contrib/gdb/gdb/dbxread.c line 2627 in elfstab_build_psymtabs Deprecated bfd_read called at /usr/src/gnu/usr.bin/binutils/gdb/../../../../contrib/gdb/gdb/dbxread.c line 933 in fill_symbuf Core was generated by `ser'. Program terminated with signal 11, Segmentation fault. Reading symbols from /usr/lib/libc.so.4...done. Reading symbols from /usr/local/lib/ser/modules/sl.so...done. Reading symbols from /usr/local/lib/ser/modules/tm.so...done. Reading symbols from /usr/local/lib/ser/modules/rr.so...done. Reading symbols from /usr/local/lib/ser/modules/maxfwd.so...done. Reading symbols from /usr/local/lib/ser/modules/usrloc.so...done. Reading symbols from /usr/local/lib/ser/modules/registrar.so...done. Reading symbols from /usr/local/lib/ser/modules/nathelper.so...done. Reading symbols from /usr/local/lib/ser/modules/textops.so...done. Reading symbols from /usr/local/lib/ser/modules/radius_auth.so...done. Reading symbols from /usr/local/lib/libradiusclient.so.0...done. Reading symbols from /usr/lib/libcrypt.so.2...done. Reading symbols from /usr/libexec/ld-elf.so.1...done. #0 0x807bdd3 in qm_status (qm=0x2816d000) at mem/q_malloc.c:447 447 f!=&(qm->free_hash[h].head); f=f->u.nxt_free, i++, j++); (gdb) bt #0 0x807bdd3 in qm_status (qm=0x2816d000) at mem/q_malloc.c:447 #1 0x8079a70 in qm_debug_frag (qm=0x2816d000, f=0x282fd0a4) at mem/q_malloc.c:109 #2 0x807ab37 in qm_free (qm=0x2816d000, p=0x282fd0bc, file=0x2a197a2d "t_reply.c", func=0x2a197e28 "t_on_reply", line=826) at mem/q_malloc.c:359 #3 0x2a188000 in t_on_reply (p_msg=0x80dfe30) at t_reply.c:826 #4 0x805933a in forward_reply (msg=0x80dfe30) at forward.c:346 #5 0x806af1c in receive_msg (buf=0x80c3340 "SIP/2.0", len=360, src_su=0x81dafa0) at receive.c:161 #6 0x8077ff0 in udp_rcv_loop () at udp_server.c:336 #7 0x805eaa5 in main_loop () at main.c:618 #8 0x806197d in main (argc=1, argv=0xbfbffc04) at main.c:1383 #9 0x804c5bd in _start () (gdb) up #1 0x8079a70 in qm_debug_frag (qm=0x2816d000, f=0x282fd0a4) at mem/q_malloc.c:109 109 qm_status(qm); (gdb) l 104 }; 105 if ((FRAG_END(f)->check1!=END_CHECK_PATTERN1)|| 106 (FRAG_END(f)->check2!=END_CHECK_PATTERN2)){ 107 LOG(L_CRIT, "BUG: qm_*: fragm. %p end overwritten(%x, %x)!\n", 108 f, FRAG_END(f)->check1, FRAG_END(f)->check2); 109 qm_status(qm); 110 abort(); 111 } 112 if ((f>qm->first_frag)&& 113 ((PREV_FRAG_END(f)->check1!=END_CHECK_PATTERN1) || (gdb) up #2 0x807ab37 in qm_free (qm=0x2816d000, p=0x282fd0bc, file=0x2a197a2d "t_reply.c", func=0x2a197e28 "t_on_reply", line=826) at mem/q_malloc.c:359 359 qm_debug_frag(qm, f); (gdb) l 354 return; 355 } 356 prev=next=0; 357 f=(struct qm_frag*) ((char*)p-sizeof(struct qm_frag)); 358 #ifdef DBG_QM_MALLOC 359 qm_debug_frag(qm, f); 360 if (f->u.is_free){ 361 LOG(L_CRIT, "BUG: qm_free: freeing already freed pointer," 362 " first free: %s: %s(%d) - aborting\n", 363 f->file, f->func, f->line); (gdb) up #3 0x2a188000 in t_on_reply (p_msg=0x80dfe30) at t_reply.c:826 826 shm_free(ack); (gdb) l 821 /* acknowledge negative INVITE replies */ 822 if (t->is_invite && (msg_status>=300 || (t->local && msg_status>=200))) { 823 ack = build_ack( p_msg, t, branch , &ack_len); 824 if (ack) { 825 SEND_PR_BUFFER( &uac->request, ack, ack_len ); 826 shm_free(ack); 827 } 828 } /* ack-ing negative INVITE replies */ 829 830 /* clean-up the transaction when transaction completed */ (gdb) up #4 0x805933a in forward_reply (msg=0x80dfe30) at forward.c:346 346 if (mod->exports->response_f(msg)==0) goto skip; (gdb) l 341 /* quick hack, slower for mutliple modules*/ 342 for (mod=modules;mod;mod=mod->next){ 343 if ((mod->exports) && (mod->exports->response_f)){ 344 DBG("forward_reply: found module %s, passing reply to it\n", 345 mod->exports->name); 346 if (mod->exports->response_f(msg)==0) goto skip; 347 } 348 } 349 350 /* we have to forward the reply stateless, so we need second via -bogdan*/ (gdb) print msg $1 = (struct sip_msg *) 0x80dfe30 (gdb) print msg [1@* $2 = {id = 1104, first_line = {type = 2, u = {request = {method = {s = 0x80c3340 "SIP/2.0", len = 7}, uri = {s = 0x80c3348 "488", len = 3}, version = { s = 0x80c334c "Not Acceptable Here", len = 19}, method_value = 488}, reply = {version = { s = 0x80c3340 "SIP/2.0", len = 7}, status = {s = 0x80c3348 "488", len = 3}, reason = { s = 0x80c334c "Not Acceptable Here", len = 19}, statuscode = 488}}}, via1 = 0x80e27fc, via2 = 0x80e2d1c, headers = 0x80e620c, last_header = 0x80e0b08, parsed_flag = 63, h_via1 = 0x80e620c, h_via2 = 0x80e0b98, callid = 0x80e0008, to = 0x80e10ac, cseq = 0x80e0b08, from = 0x80e0f2c, contact = 0x0, maxforwards = 0x0, route = 0x0, record_route = 0x0, content_type = 0x0, content_length = 0x0, authorization = 0x0, expires = 0x0, proxy_auth = 0x0, www_auth = 0x0, supported = 0x0, require = 0x0, proxy_require = 0x0, unsupported = 0x0, allow = 0x0, event = 0x0, eoh = 0x0, unparsed = 0x80c3493 "Content-Length: 0\r\n\r\n", src_ip = { af = 2, len = 4, u = {addr32 = {1214690368, 0, 0, 0}, addr16 = {46144, 18534, 0, 0, 0, 0, 0, 0}, addr = "@╢fH", '\000' <repeats 11 times>}}, src_port_no = 5061, dst_ip = {af = 2, len = 4, u = {addr32 = {1214690368, 0, 0, 0}, addr16 = {46144, 18534, 0, 0, 0, 0, 0, 0}, addr = "@╢fH", '\000' <repeats 11 times>}}, orig = 0x80e2a48 "SIP/2.0 488 Not Acceptable Here\r\nVia: SIP/2.0/UDP 64.180.102.72:5060;branch=z9hG4bKfa9f.17ad3774445fef6b70f18b74de26e0b2.0\r\nVia: SIP/2.0/UDP 172.17.1.127:5060;rport=5060\r\nTo: <sip:151@64.180.102.72;us"..., buf = 0x80c3340 "SIP/2.0", len = 360, new_uri = {s = 0x0, len = 0}, parsed_uri_ok = 0, parsed_uri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0x0, len = 0}, port = {s = 0x0, len = 0}, params = {s = 0x0, len = 0}, headers = {s = 0x0, len = 0}, port_no = 0}, add_rm = 0x0, repl_add_rm = 0x80e0514, reply_lump = 0x0, add_to_branch_s = '\000' <repeats 54 times>, add_to_branch_len = 0, hash_index = 0, flags = 0} (gdb) up #5 0x806af1c in receive_msg (buf=0x80c3340 "SIP/2.0", len=360, src_su=0x81dafa0) at receive.c:161 161 forward_reply(msg); (gdb) l 156 gettimeofday( & tvb, &tz ); 157 STATS_RX_RESPONSE ( msg->first_line.u.reply.statuscode / 100 ); 158 #endif 159 160 /* send the msg */ 161 forward_reply(msg); 162 163 #ifdef STATS 164 gettimeofday( & tve, &tz ); 165 diff = (tve.tv_sec-tvb.tv_sec)*1000000+(tve.tv_usec-tvb.tv_usec); (gdb) q bash-2.05a$ exit Script done on Fri Feb 21 05:44:36 2003

2 1

[Serusers] problem with loading compiled radius modules
by Benny Ho 21 Feb '03

21 Feb '03

I had the same error msgs when I load the radius_acc modules the following is the error msgs [root@md benny]# /usr/sbin/ser restart 0(3699) ERROR: load_module: could not open module </usr/lib/ser/modules/radius_ acc.so>: libradiusclient.so.0: cannot open shared object file: No such file or d irectory 0(3699) parse error (25,13-48): failed to load module ERROR: bad config file (1 errors) [root@md benny]# and I check that the radius_acc.so is in /usr/lib/ser/modules and the libradiusclient.so.0 is also in /usr/local/lib as it is stated in the Makefile. Where is the error came from? Have any one successfully implement it?

1 3

[Serusers] RE: RE: problem with loading compiled radius modules
by Yang Xiang 21 Feb '03

21 Feb '03

Hi Benny, I had the same problem as you. After I statically linked the radiusclient library in the both radius modules this problem seems to be resolved. http://lists.iptel.org/pipermail/serusers/2003-February/000419.html However radius authentication still doesn't work and the developer said: "the path is not staticaly linked, it should be resolved by you dynamic linker, I don't know where is the problem, it seems to me that there is some problem with radiusclient libraries on your system, " I use Solaris 8. Now I am trying to recompile and reinstall radiusclient library. If I find out something, I'll tell you. regards, yang

1 0

[Serusers] CVS version current, gives this error
by Lenny Tropiano 20 Feb '03

20 Feb '03

Default config (ser.cfg installed from gmake install), gives this error (I bumped up the debug level and had it log to stderr): p2:[root@enterprise.icus.com] 31 $ ser 0(62901) loading module /usr/local/lib/ser/modules/sl.so 0(62901) loading module /usr/local/lib/ser/modules/tm.so 0(62901) loading module /usr/local/lib/ser/modules/rr.so 0(62901) loading module /usr/local/lib/ser/modules/maxfwd.so 0(62901) loading module /usr/local/lib/ser/modules/usrloc.so 0(62901) loading module /usr/local/lib/ser/modules/registrar.so 0(62901) find_param_export: found <db_mode> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(62901) find_export: found <mf_process_maxfwd_header> in module maxfwd_module [/usr/local/lib/ser/modules/maxfwd.so] 0(62901) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(62901) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(62901) find_export: <rewriteFromRoute> not found 0(62901) parse error (80,19-20): unknown command, missing loadmodule? 0(62901) find_export: found <save> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(62901) find_export: found <lookup> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(62901) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(62901) find_export: found <t_relay> in module tm [/usr/local/lib/ser/modules/tm.so] 0(62901) find_export: found <sl_reply_error> in module sl_module [/usr/local/lib/ser/modules/sl.so] ERROR: bad config file (1 errors) 0(62901) INFO: signal 15 received 0(62901) Memory status (pkg): 0(62901) qm_status (0x80a1780): 0(62901) heap size= 1046128 0(62901) used= 0, used+overhead=2448, free=1043680 0(62901) max used (+overhead)= 2448 0(62901) dumping all allocked. fragments: 0(62901) dumping free list stats : 0(62901) hash= 137. fragments no.: 1 0(62901) ----------------------------- --- Lenny Tropiano E-mail: lenny(a)voiping.com Partner, Networking Specialist Pager: pager-lenny(a)voiping.com VoIPing, LLC URL: http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 Mobile: 512-698-VOIP [8647]

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[Serusers] Some compile time errors for FreeBSD of CVS Tree source (sip_router)
by Lenny Tropiano 20 Feb '03

20 Feb '03

pass_fd.c: added #include <sys/uio.h> tcp_main.c: moved #include <sys/select.h> underneath #include <sys/types.h.> Still getting this: $ gmake Makefile.defs:331: Old gcc detected (2.9x), use gcc >= 3.1 for better results gcc -g -O9 -funroll-loops -Wcast-align -Wall -m486 -malign-loops=4 -DNAME='"ser"' -DVERSION='"0.8.11pre6-tcp5-tm"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 2.95"' -D__CPU_i386 -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DDNS_IP_HACK -DUSE_IPV6 -DVOICE_MAIL -D_TOTAG -DUSE_TCP -DFAST_LOCK -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -c tcp_main.c -o tcp_main.o tcp_main.c: In function `tcp_send': tcp_main.c:351: `MSG_NOSIGNAL' undeclared (first use in this function) tcp_main.c:351: (Each undeclared identifier is reported only once tcp_main.c:351: for each function it appears in.) gmake: *** [tcp_main.o] Error 1 Cannot find proper #define for MSG_NOSIGNAL The flags, according to "send()" man page is: The flags parameter may include one or more of the following: #define MSG_OOB 0x1 /* process out-of-band data */ #define MSG_PEEK 0x2 /* peek at incoming message */ #define MSG_DONTROUTE 0x4 /* bypass routing, use direct interface */ #define MSG_EOR 0x8 /* data completes record */ #define MSG_EOF 0x100 /* data completes transaction */ The flag MSG_OOB is used to send ``out-of-band'' data on sockets that support this notion (e.g. SOCK_STREAM); the underlying protocol must also support ``out-of-band'' data. MSG_EOR is used to indicate a record mark for protocols which support the concept. MSG_EOF requests that the sender side of a socket be shut down, and that an appropriate indication be sent at the end of the specified data; this flag is only implemented for SOCK_STREAM sockets in the PF_INET protocol family, and is used to implement Transaction TCP (see ttcp(4)). MSG_DONTROUTE is usually used only by diagnostic or routing programs. --- Lenny Tropiano E-mail: lenny(a)voiping.com Partner, Networking Specialist Pager: pager-lenny(a)voiping.com VoIPing, LLC URL: http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 Mobile: 512-698-VOIP [8647]

2 1

[Serusers] Via tags question...
by lgfausak＠august.net 20 Feb '03

20 Feb '03

This is a protocol question, I don't know if it is directly related to ser. Concerning the Via tags, should any proxy/gateway modify tags that don't belong to it, or should they simply be copied verbatum? For example, I have a situation where I send an INVITE message, and the TRYING response I receive modifies one of the Via: tags. I don't think this is right. Here are the two packets in question: # U 2003/02/19 18:36:20.276844 164.9.42.25:5060 -> 184.294.324.2:5060 INVITE sip:12143357976@184.294.324.2:5060 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:12143357976@164.9.42.25;branch=0>. From: <sip:8178863582@216.87.145.22> ;tag=d8579116-fdc-3e542303-23d34-4a84. To: <sip:12143357976@augustvoice.net> ;user=phone. Call-ID: 59fd88-d8579116-fdc-3e542303-23d2f-3897(a)216.87.145.22. CSeq: 2 INVITE. Via: SIP/2.0/UDP 164.9.42.25;branch=z9hG4bKcc26.d3c125d.0. Via: SIP/2.0/UDP 216.87.145.22:4060 ;branch=z9hG4bK-3e542304-23dfc-ea2. Contact: <sip:8178863582@216.87.145.22:4060>. User-Agent: FXS_GW (sip1028a.bin). Content-Type: application/SDP. Content-Length:291. P-hint: LD GATEWAY. . v=0. o=FXS_GW 12367 0 IN IP4 216.87.145.22. s=Audio Session. i=Audio Session. c=IN IP4 216.87.145.22. t=0 0. m=audio 16384 RTP/AVP 0 18 4 8 96. a=fmtp:96 0-11. a=rtpmap:0 PCMU/8000/1. a=rtpmap:18 G729/8000/1. a=rtpmap:4 G723/8000/1. a=rtpmap:8 PCMA/8000/1. a=rtpmap:96 telephone-event/8000. And your response is: # U 2003/02/19 18:36:20.902858 184.294.324.2:5060 -> 164.9.42.25:5060 SIP/2.0 100 Trying . Via: SIP/2.0/UDP 164.9.42.25;branch=z9hG4bKcc26.d3c125d.0. Via: SIP/2.0/UDP 216.87.145.22:4060. From: <sip:8178863582@216.87.145.22> ;tag=d8579116-fdc-3e542303-23d34-4a84. To: <sip:12143357976@augustvoice.net> ;user=phone;tag=27DC1DCC-F61. Date: Thu, 20 Feb 2003 00:36:20 GMT. Call-ID: 59fd88-d8579116-fdc-3e542303-23d2f-3897(a)216.87.145.22. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 2 INVITE. Allow-Events: telephone-event. Content-Length: 0. . See how the : Via: SIP/2.0/UDP 216.87.145.22:4060 ;branch=z9hG4bK-3e542304-23dfc-ea2. Has been changed to: Via: SIP/2.0/UDP 216.87.145.22:4060. Does anyone have any perl script to take a ngrep protocol dump and print it out in a pretty way? ---greg Greg Fausak August.Net Services, LLC

2 1

[Serusers] Add a user to subscriber table.
by Ivan Vignola 20 Feb '03

20 Feb '03

Hi, I would add a user to subsriber table by sql command in my application (not using serctl script). What text strings I must use to calculate HA1 (=phplib_id) and ha1b values if USER_ID=ivan,PASSWORD=ivanpswd and REALM=192.168.11.237 ? Thanks in advance. Regards, Ivan Vignola. _________________________________________________________________ Vinci la nuova Nissan Micra con MSN Messenger! http://www.msn.it/messenger/

3 2

[Serusers] serctl - tail problems on solaris8?
by Kamil Ince 20 Feb '03

20 Feb '03

bet i missed reading something, again :-/ but wasting too much time on this.. i get on solaris 8 the following: # echo $SIP_DOMAIN myser.my.net # echo $PW heslo # serctl add user1 hispw user1(a)my.net usage: tail [+/-[n][lbc][f]] [file] tail [+/-[n][l][r|f]] [file] new user added # serctl passwd user1 newpw usage: tail [+/-[n][lbc][f]] [file] tail [+/-[n][l][r|f]] [file] non-existent user of course the user has been added to the database... anyone came accross that before? tia -k

3 5

[Serusers] [bug] TM module config
by Sebastien Pierre 19 Feb '03

19 Feb '03

Hi again, Here is how to get a segfault: <snip> #Load modules required by a stateful User Agent Service loadmodule "modules/tm.so" #SER timer parameters modparam("tm", "fr_inv_timer", "10"); modparam("tm", "fr_timer", "10"); </snip> This happened after quoting the numbers in the modparam. Tested with ser-0.8.10 on Linux RH 8.0 -- Sebastien

2 1

[Serusers] Logging with SER
by Sebastien Pierre 19 Feb '03

19 Feb '03

Hi all, I finally found solutions to my problem: 1) It is easier when setting "fork=no" because then ser does not daemonise and stderr is then attached to the terminal in which I started ser 2) The configuration file should set a debugging level ("set debug=1"), and use a number equal or lower to the debug level in the log, like 'log(0, "Hello World")' So my HelloWorld.cfg is as follows: <snip> debug=0 log_stderror=yes fork=no route[0] { log(0, "Hello World!"); } </snip> And it works with "ser -f HelloWorld.cfg" Cheers, -- Sebastien

1 0

Re: [Serusers] SER logging (and the mighty HelloWorld.cfg)
by Sebastien Pierre 18 Feb '03

18 Feb '03

Hi Nils! Le mar 18/02/2003 à 16:56, Nils Ohlmeier a écrit : > Hard to say without seeing the config file. But '-E' as command line > parameter should work and will overwrite any value in your config > file. Here it is (the HelloWorld.cfg): <snip> log_stderror=yes route[0] { log("Hello World!"); } </snip> Here is the SIP message I send to SER: <snip> INVITE sip:alice@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5070 To: Alice <sip:alice@localhost> From: Me <sip:me@localhost> Call-ID:123456789@127.0.0.1 CSeq: 1 INVITE Content-Length: 0 </snip> Here is the answer I get from SER: <snip> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 127.0.0.1:5070 To: Alice <sip:alice@localhost>;tag=d907c037823644515dfe0ede38ca9976.de25 From: Me <sip:me@localhost> Call-ID:123456789@127.0.0.1 CSeq: 1 INVITE Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 127.0.0.1:5060 "Noisy feedback tells: pid=9995 req_src_ip=127.0.0.1 in_uri=sip:alice@127.0.0.1 out_uri=sip:alice@127.0.0.1 via_cnt==1" </snip> And here is what I get from SER in stdout/stderr: <snip> $ ser -E HelloWorld.cfg Listening on 127.0.0.1 [127.0.0.1]::5060 172.25.49.134 [172.25.49.134]::5060 Aliases: XXX.alcatel.fr:5060 localhost:5060 localhost.localdomain:5060 XXX:5060 </snip> And from /var/log/messages <snip> Feb 18 17:33:21 frilld0030711 ser[9994]: INFO: udp_init: SO_RCVBUF is initially 65535 Feb 18 17:33:21 frilld0030711 ser[9994]: INFO: udp_init: SO_RCVBUF is finally 131070 Feb 18 17:33:21 frilld0030711 ser[9994]: INFO: udp_init: SO_RCVBUF is initially 65535 Feb 18 17:33:21 frilld0030711 ser[9994]: INFO: udp_init: SO_RCVBUF is finally 131070 Feb 18 17:33:21 frilld0030711 ser[10003]: INFO: fifo process starting: 10003 Feb 18 17:33:21 frilld0030711 ser[10003]: SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... </snip> So there is sadly no "HelloWorld!" somewhere... would it be because of the "404 not found" error I get ? TIA, -- Sebastien

3 2

[Serusers] Simple SER Use Case
by Sebastien Pierre 18 Feb '03

18 Feb '03

Hi all, I am trying to implement a very simple call forwarding use case with SER. This use case is directly taken from a book entitled "Internet Applications with SIP". Here are the actors: - SIP caller, wants to call <alice@domain> - Proxy server of <domain> - SIP phone which is default terminal for <alice(a)domain.com> - Voicemail server Here is the scenario: 0) SIP caller INVITEs <alice@domain> 1) Proxy receives INVITE 2) Proxy forwards INVITE to alice default SIP phone 3) Proxy responses 100 TRYING to SIP caller 4) SIP phone responses 100 TRYING to proxy 5) SIP phone responses 180 RINGING to proxy 6) Proxy reponses 180 RINGING to caller 7) After timeout, proxy send CANCEL to default SIP phone 8) SIP phone responses OK to proxy... ... X) Proxy forward original INVITE to voice mail server... The fact is that the proxy first notifies the caller that it is trying to locate alice, listens for "180 RINGING" and forward it to the caller, then cancels on the SIP phone side because of a timeout -- then the proxy forwards the INVITE to the voice mail. I have tried some configurations, but I did not see how to listen for the 180 RINGING and forward them to the caller... is there a way to bind a "route" to a provisional response like: route[0] { ... #In case a 180 occur, we execute route[1] t_on_provisional("180", "1"); #But we're also waiting for an OK or a timer #event to arrive t_on_negative("2"); t_relay(); ... } route[1] { #Forwards the 180 Ringing from the SIP phone to #the caller t_relay(); } BTW, I was wondering if t_on_negative is a *blocking* action ? TIA, -- Sebastien

3 2

[Serusers] ser-mysql for solaris
by Kamil Ince 18 Feb '03

18 Feb '03

hello, can someone help me with the database details for ser? I got only solaris available, and couldnt find ser-mysql as "non-rpm" tia -k

3 4

[Serusers] Re: [Serhelp] SER logging
by Sebastien Pierre 18 Feb '03

18 Feb '03

Hi Nils! Le mar 18/02/2003 à 16:03, Nils Ohlmeier a écrit : > On Tuesday 18 February 2003 15:57, Sebastien Pierre wrote: > > I was just wondering (did not find in User or Dev guide) where to get > > the result of SER "log" action... I have written a fake SIP UA that > syslog (probably /var/log/messages) if log_stderror=no, otherwise stderr (your > console). I thought I did not install syslog with root privileges, so I did not look in the *obvious* file! Anyway, I put a "log_sterror=yes" line at the beginning of my SER configuration script, but SER keeps outputting on /var/log/messages and not on STDERR... what did I do wrong ? > > sends SIP messages written in a file to test custom SER configurations, > > but I don't know if SER properly receives my messages or executes my > > routing program. Any advices ? > Congratulation for reinventing the wheel -> http://sipsak.berlios.de :-) I know about it but... >> sipsak -s sip:alice@localhost error: this FQDN or IP is not valid: frwed00304 I'm still puzzled about this FQDN and my hostname... and I could not resist making a 20 lines Python script to get SER tested ;) -- Seb

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Re: [Serusers] SerWeb
by Karel Kozlik 16 Feb '03

16 Feb '03

Hello, check your php.ini file if you have: register_globals = On Karel > > >Hi, > > > >I' ve installed SERWEB.....But when I try to register or login...nothing happens... the page is reload. The Apache server is working. Anyone had this same problem? > > > >Thanks, > > > >Adriana Simizo > >Telefonica Brasil - Telecomunicaçőes de Săo Paulo S. A. > > > > -- > Jiri Kuthan http://iptel.org/~jiri/

1 0

Re: [Serusers] problem with loading compiled radius modules
by Xiang 15 Feb '03

15 Feb '03

I have resolved this problem. The radiusclient library was not included in the both radius modules, somehow. After I statically linked the radiusclient library with the option -lradiusclient when compiling, it works. regards yang ------------------- > The modules need radiusclient library but linker is unable to find > symbols from the library. Do you have the radiusclient library installed > correctly ? > > regards, Jan. > > On 14-02 14:35, Yang Xiang wrote: > > Hi, > > > > I get the same error. > > > > I have compiled the whole server and try to load modules on the host where > > the server was compiled. Unfortunatelly it doesn't work. I get the same > > error message as before. > > > > What should I do???? > > > > Regards > > > > Yang > > > > > > ----- Original Message ----- > > From: "Jan Janak" <J.Janak(a)sh.cvut.cz> > > To: "Yang Xiang" <yang.xiang(a)iitb.fraunhofer.de> > > Sent: Friday, February 14, 2003 2:03 PM > > Subject: Re: [Serusers] problem with loading compiled radius modules > > > > > > > Recompile the whole server if you can. As you said you use precomplied > > > binaries and it is possible that it was compiled using a different > > > compiler. > > > > > > regards, Jan. > > > > > > On 14-02 13:56, Yang Xiang wrote: > > > > Hi, > > > > > > > > Now I get more error messages: > > > > > > > > 0(22528) ERROR: load_module: could not open module > > > > </usr/local/lib/ser/modules/radius_acc.so>: ld.so.1: ser: fatal: > > relocation > > > > error: file /usr/local/lib/ser/modules/radius_acc.so: symbol > > rc_avpair_add: > > > > referenced symbol not found > > > > 0(22528) parse error (30,13-54): failed to load module > > > > 0(22528) ERROR: load_module: could not open module > > > > </usr/local/lib/ser/modules/radius_auth.so>: ld.so.1: ser: fatal: > > relocation > > > > error: file /usr/local/lib/ser/modules/radius_auth.so: symbol > > > > rc_read_config: referenced symbol not found > > > > 0(22528) parse error (31,13-55): failed to load module > > > > ERROR: bad config file (2 errors) > > > > > > > > What is wrong? In acc_mod.c and auth_mod.c I changed the definition of > > > > CONFIG_FILE and let it point to an absolute path > > > > "/usr/local/etc/radiusclient/radiusclient.conf". By the way, I compiled > > the > > > > modiles on another mashine and copied them to the mashine where ser > > runs. Is > > > > that right? > > > > > > > > regards, > > > > > > > > yang > > > > > > > > > > > > > >

1 0

[Serusers] Is there a tool for benchmarking SER?
by Maxim Sobolev 15 Feb '03

15 Feb '03

Folks, Our company mostly finished development of the SIP VoIP system that consists of the SER for registration, de-NATifying, and routing and B2BUA for debit applications. Now we would like to test performance of the whole system under the load. In the SER's documentation I saw very impressive performance numbers and wonder how they were obtained. Are there any freely-available tools for creating huge number of SIP sessions simulateneously and if yes, how they can be obtained? Any pointers are greatly appreciated. Thanks! -Maxim

2 2

[Serusers] calling a module
by Mario Kolberg 15 Feb '03

15 Feb '03

Hi, > simply use fi_check() in the configuration script. That will call your > function. Thats what I thought also, but unfortunately, SER does not run when. If I run it with a call fi_check() in the ser.cfg file it logs a parse error: Feb 14 14:15:00 DISCUS ser: parse error (39,1-9): parse error Feb 14 14:15:00 DISCUS ser: parse error (39,1-9): I have attached the complete cfg file. But if I remove that fi_check() call, ser runs fine. Have I messed something up with parameters???? Thanks for your help! Mario -- Mario Kolberg phone: +44 (0)1786 46 7440 Lecturer in Computing Science fax : +44 (0)1786 46 4551 email: mko(a)cs.stir.ac.uk Department of Computing Science and Mathematics University of Stirling Stirling FK9 4LA Scotland, UK -- The University of Stirling is a university established in Scotland by charter at Stirling, FK9 4LA. Privileged/Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not disclose, copy or deliver this message to anyone and any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer do not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of the University of Stirling shall be understood as neither given nor endorsed by it. # # $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # 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" alias="cs.stir.ac.uk" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #mario 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" loadmodule "//usr/lib/ser/modules/fi.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #mario loadmodule "//usr/lib/ser/modules/auth.so" fi_check(); # ----------------- 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 #mario modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth", "secret", "alsdkhglaksdhfkloiwr") #mario modparam("auth", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #mario 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 #mario if (!www_authorize("cs.stir.ac.uk", "subscriber")) { www_challenge("cs.stir.ac.uk", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now if (!t_relay()) { sl_reply_error(); }; }

3 2

[Serusers] writing a module
by Mario Kolberg 15 Feb '03

15 Feb '03

Hi, I am trying to write a module for SER. I managed to write a tiny test module ("hello world") and it compiles fine (the source is attached). However, I'm not quite sure I fully understood how to execute the functions defined in the module. I included the module to sr_modules.c and also a line 'loadmodule' in the ser.cfg. But how do I actually execute the function in the module every time a (INVITE) message is received by the server? The docs couldn't really help ... Thanks very much! Regards, Mario -- Mario Kolberg phone: +44 (0)1786 46 7440 Lecturer in Computing Science fax : +44 (0)1786 46 4551 email: mko(a)cs.stir.ac.uk Department of Computing Science and Mathematics University of Stirling Stirling FK9 4LA Scotland, UK -- The University of Stirling is a university established in Scotland by charter at Stirling, FK9 4LA. Privileged/Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not disclose, copy or deliver this message to anyone and any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer do not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of the University of Stirling shall be understood as neither given nor endorsed by it. #include <stdio.h> #include <string.h> #include <stdlib.h> #include "../../sr_module.h" #include "../../dprint.h" #include "../../error.h" #include "../../ut.h" static int fi_check(struct sip_msg* msg,char* str,char* str2); static int mod_init(void); #ifdef STATIC_FI struct module_exports fi_exports = { #else struct module_exports exports= { #endif "fi_module", (char*[]){ "fi_check" }, (cmd_function[]){ fi_check }, (int[]){ 0 }, (fixup_function[]){ 0 }, 1, NULL, NULL, NULL, 0, mod_init, (response_function) 0, (destroy_function) 0, 0, 0 }; static int mod_init(void) { printf("FI module initializing\n"); return 0; } static int fi_check(struct sip_msg* msg, char* str1,char* str2) { printf("FI module executing\n"); return 1; }

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Re: [Serusers] Routing with ser / Problems with linphone
by Jiri Kuthan 14 Feb '03

14 Feb '03

/* digging through the archives to make sure I haven't missed anything while I was off-line last weeks ... */ I actually think that is a backwards compatibility issue in SIP. Loose-routing works correctly if sender knows whether the next-hop is loose or strict router. That is a correct assumption as long as the routing information collected automatically in record-route tells you that. If you use pre-configured Route along with an outbound proxy, you configure this strict/loose on your own and you need to get it right. Hence, when you configure loose routing for use with a strict outbound router, things will fail. Anyway, with lr support in next ser release, it should not matter any more. We've had the same issue with kphone previouly, it works since an option has been introduce to enforce use of strict routing. see http://www.ietf.org/mail-archive/working-groups/sipping/current/msg03451.ht… -Jiri At 11:18 AM 2/5/2003, Klaus Darilion wrote: >Hello! > >I've installed ser (sip:obelix.ict.tuwien.ac.at) und want'ed to use it >with linphone, so I configured linphone to use ser as outbound proxy and >registrar. The registration works fine but not the INVITEs. ser answers >to an INVITE from linphone with a 404 Not Found and I found out that the >problem is the "Route" header in the INVITE from linphone. > >Route: <sip:obelix.ict.tuwien.ac.at;lr> > >When I remove the header from the invite and send it manually (sipsak) >the ser proxy accepts the invite and forwards it. I think ser reacts >wrong because if an RFC3261 proxy gets an request with a route header >which points to itself it should remove the header and forward the >request. > >Than I tried the same with an different proxy in the route header: > >Route: <sip:iptel.org;lr> > >Now, my ser proxy (sip:obelix.ict.tuwien.ac.at) accepts the request and >forwards it to iptel.org, but it rewrites the invite to: >INVITE sip:iptel.org;lr SIP/2.0 > >Thats what an RFC2543 proxy would do, but not an RFC3261 proxy, which >only is allowed to do that if the route header has no "lr" parameter or >if he is responsible for the domain in the request URI. If there is an >"lr" parameter the proxy must not change the request URI. > >If I'm wrong please let me know. > >I use ser 0.8.10 with the standard config file and linphone 0.9.1 & >0.10.0 > >Regards, >Klaus >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Serusers] Testing SER
by Jaime Hemmett 14 Feb '03

14 Feb '03

Hello, I installed SER, I have started the services. I haven't set up the mysql part yet, I wanted to test it working at all. With bone phone I have tried connecting but gotten: "java.net.BindException: Address already in use" The SER documentation makes out that SER can be used without the mysql parts to make calls with microsoft messenger > version 4.6 ut I've been unable to connect with this either. What now? Jaime.

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[Serusers] Small patch for sip_dictionary
by Maxim Sobolev 14 Feb '03

14 Feb '03

Folks, sip_dictionary buindled with ser doesn't include Vendor-Specific attribute. Attached patch should fix it, it would be nice to have it included into the next release. Thanks! -Maxim $FreeBSD$ --- etc/sip_dictionary 2003/01/27 14:13:18 1.1 +++ etc/sip_dictionary 2003/01/27 14:13:39 @@ -48,6 +48,7 @@ ATTRIBUTE Framed-Route 22 string ATTRIBUTE Framed-IPX-Network 23 ipaddr ATTRIBUTE State 24 string +ATTRIBUTE Vendor-Specific 26 string ATTRIBUTE Session-Timeout 27 integer ATTRIBUTE Idle-Timeout 28 integer ATTRIBUTE Termination-Action 29 integer

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Re: [Serusers] User permission question.
by Ivan Vignola 14 Feb '03

14 Feb '03

Hi, thank you for your replies. I need be sure that user 'ivan', with password 'ivanpswd', registers himself typing 'sip:ivan.vignola@192.168.11.237' and not typing 'silvio.berlusconi(a)192.168.11.237'. Regards. Ivan. >From: Jiri Kuthan <jiri(a)iptel.org> >To: "Ivan Vignola" <ivanvignola(a)hotmail.com>, serusers(a)lists.iptel.org >Subject: Re: [Serusers] User permission question. >Date: Fri, 14 Feb 2003 11:45:37 +0100 > >I'm not sure what it means "authenticate by his uri". There are quite >many URIs in each request, which BTW have very low security value. >Can you give more details? > >Thanks, > >-Jiri > >At 10:29 AM 2/14/2003, Ivan Vignola wrote: > >Hi, > > > >in my tests user can authenticate himself using user_id, password and any >sip uri. I need user authenticating himself only by his associated uri. How >can I do that? > > > >Thanks in advance. > > > >Ivan Vignola. > > > > > > > > > > > >_________________________________________________________________ > >Vinci la nuova Nissan Micra con MSN Messenger! >http://www.msn.it/messenger/ > > > >_______________________________________________ > >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 _________________________________________________________________ MSN Foto: condividi, ritocca e stampa le tue foto online http://photos.msn.it

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[Serusers] User permission question.
by Ivan Vignola 14 Feb '03

14 Feb '03

Hi, in my tests user can authenticate himself using user_id, password and any sip uri. I need user authenticating himself only by his associated uri. How can I do that? Thanks in advance. Ivan Vignola. _________________________________________________________________ Vinci la nuova Nissan Micra con MSN Messenger! http://www.msn.it/messenger/

3 2

[Serusers] problem with loading compiled radius modules
by Yang Xiang 14 Feb '03

14 Feb '03

Hi Jan, I have successfully compiled the radius modules radius_acc.so and radius_auth.so. But when I try to load the both modules ser tells me: ERROR: bad config file (2 errors) I use the solaris binary package from iptel and have compiled myself only the both radius modules. Is there any incompability? How can I get mor debug information? Regards, Yang ----------------------------------- My config file: debug=7 # 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/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/radius_acc.so" loadmodule "/usr/local/lib/ser/modules/radius_auth.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/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 alias="pix-demo.fhg.de" 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("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now if (!t_relay()) { sl_reply_error(); }; }

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[Serusers] SIP2H323 and H3232SIP
by Benny Ho 13 Feb '03

13 Feb '03

Hi, How could I integrate SIP and H323 together? I would like to h323 endpoint call sip endpoint. Is it possible using SER? My config is Cisco AS5300, gnu gatekeeper and SER.

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[Serusers] SerWeb
by Adriana Simizo 13 Feb '03

13 Feb '03

Hi, I' ve installed SERWEB.....But when I try to register or login...nothing happens... the page is reload. The Apache server is working. Anyone had this same problem? Thanks, Adriana Simizo Telefonica Brasil - Telecomunicações de São Paulo S. A.

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[Serusers] and errors in digest.c ?
by Yang Xiang 13 Feb '03

13 Feb '03

Hi, and in file digest.c of the module radius_auth: warning: digest.c: In function `radius_authorize': digest.c:79: `PW_SIP_NONCE' undeclared (first use in this function) digest.c:79: (Each undeclared identifier is reported only once digest.c:79: for each function it appears in.) digest.c:83: `PW_SIP_NONCE_COUNT' undeclared (first use in this function) Changed from PW_SIP_NONCE to PW_SIP_USER_NONCE and from PW_SIP_NONCE_COUNT to PW_SIP_USER_NONCE_COUNT which are defined in your new header file "radiusclient.h". Hope this is ok. Regards Yang

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[Serusers] Error in acc.c ?
by Yang Xiang 13 Feb '03

13 Feb '03

Hi, While trying to compile the module radius_acc I get some warnings as follows: acc.c: In function `radius_log_reply': acc.c:142: warning: passing arg 1 of `parse_from_header' from incompatible pointer type acc.c: In function `radius_log_ack': acc.c:426: warning: passing arg 1 of `parse_from_header' from incompatible pointer type acc.c: In function `rad_acc_request': acc.c:708: warning: passing arg 1 of `parse_from_header' from incompatible pointer type So I looked into the file acc.c and found such statements like if ( parse_from_header( msg )==-1 ) Because the function parse_from_header needs a parameter of type hdr_field, I changed the statements above to if ( parse_from_header( msg->headers )==-1 ) after this the warnings don't appear any longer. Is this change ok? Best Regards Yang

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Re: [Serusers] SIP/H.323 Gateway
by Giorgos Tsiouris 12 Feb '03

12 Feb '03

No it is not very stable and it produces a huge executable but it is the only open source that I am aware off. I also think that the development of sip2h323gw has stop since the programer that did the development has left long ago from Vovida. You can also complile this with the new h323 libraries available from www.h323.org or so I think. Giorgos Tsiouris ----- Original Message ----- From: "Vlasis Hatzistavrou" <vhatz(a)auth.gr> To: "Giorgos Tsiouris" <gtsiouris(a)telecom.ntua.gr> Sent: Wednesday, February 12, 2003 5:11 PM Subject: Re: [Serusers] SIP/H.323 Gateway > Hello Giorgo, > > Have you worked with the Vovida sip2h323gw? Is it stable or compatible with > other major vendor H323 GW's like Cisco for example? > > Best regards, > Vlasis. > > Giorgos Tsiouris wrote: > > > There is one SIP to H323 gateway from www.vovida.org . You can download > > vocal and you can compile only the gateway. I also think that there is one > > from siemens (I do not remeber their site) but this is not open source. > > > > Giorgos Tsiouris > > > > ----- Original Message ----- > > From: "Ng, Soo Sim" <SooSim.Ng(a)hp.com> > > To: <serusers(a)lists.iptel.org> > > Sent: Wednesday, February 12, 2003 1:09 PM > > Subject: [Serusers] SIP/H.323 Gateway > > > > > Is there testing being done SIP end-point to H.323 end-point call and > > vice-versa. What's the SIP/H.323 gateway being tested? > > > > > > Thanks > > > > > > SS > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] SIP/H.323 Gateway
by Ng, Soo Sim 12 Feb '03

12 Feb '03

Is there testing being done SIP end-point to H.323 end-point call and vice-versa. What's the SIP/H.323 gateway being tested? Thanks SS

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[Serusers] Problem compiling module radius_auth
by Bjoern Zuelch 12 Feb '03

12 Feb '03

Hi, we try to compile ser 0.8.10 with radius support under Solaris 8. First we have a problem with the radius_acc module. We found s solution in the mailinglist archive. Now we have a problem with the radius_auth module. If we compile it we get the following error message: digest.c -o digest.o digest.c: In function `radius_authorize': digest.c:79: `PW_SIP_NONCE' undeclared (first use in this function) digest.c:79: (Each undeclared identifier is reported only once digest.c:79: for each function it appears in.) digest.c:83: `PW_SIP_NONCE_COUNT' undeclared (first use in this function) digest.c: In function `radius_authorize_freeradius': digest.c:160: `PW_DIGEST_USER_NAME' undeclared (first use in this function) digest.c:164: `PW_DIGEST_REALM' undeclared (first use in this function) digest.c:168: `PW_DIGEST_NONCE' undeclared (first use in this function) digest.c:172: `PW_DIGEST_URI' undeclared (first use in this function) digest.c:176: `PW_DIGEST_METHOD' undeclared (first use in this function) digest.c:184: `PW_DIGEST_QOP' undeclared (first use in this function) digest.c:192: `PW_DIGEST_CNONCE' undeclared (first use in this function) digest.c:201: `PW_DIGEST_NONCE_COUNT' undeclared (first use in this function) digest.c:211: `PW_DIGEST_BODY_DIGEST' undeclared (first use in this function) digest.c:241: `PW_DIGEST_ALGORITHM' undeclared (first use in this function) digest.c:252: `PW_DIGEST_ATTRIBUTES' undeclared (first use in this function) digest.c:258: `PW_DIGEST_RESPONSE' undeclared (first use in this function) gmake: *** [digest.o] Error 1 we think that there some definitions not defined in an header-file. we included the dict.h file from the radius_acc module(CVS version), so most of the definitions PW_SIP_* are present. Is there a possibility to get the missing definitions? greetings bjoern

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Re: [Serusers] New Setup of SER
by chid＠eglobalcall.com 11 Feb '03

11 Feb '03

Hi, Can I just cut and paste the script directly into the ser.cfg file? -Chi Original Message: ----------------- From: serusers (Jiri) serusers(a)lists.iptel.org Date: Tue, 11 Feb 2003 17:30:52 +0100 To: chid(a)eglobalcall.com, serusers(a)lists.iptel.org Subject: Re: [Serusers] New Setup of SER At 01:22 PM 2/11/2003, Chi Dang wrote: >Hi, > >Sorry to be greedy but I have two questions. I've just finished >installed the ser 0.8 version on Solaris 2.8. > >1. When I ran ser, it said it is listening to only 3 servers. I have >added 2 more into my /etc/hosts file but make no difference. Where does >ser daemon get the ip of servers that it is listening to? /etc/hosts configures you resolver and does not affect the interfaces configured on your host. ser determines existing interfaces but does not care about your resolver's configuration. >2. I plan to use ser with my cisco voip gateways. Is there anything >special that I need to configure ser to make it work with Cisco >gateways? Maybe access control http://www.iptel.org/ser/doc/seruser-html/x740.html#AEN786 Note that Cisco gateways have no digest support, so you are left only with source IP address "authentication". -Jiri _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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[Serusers] New Setup of SER
by Chi Dang 11 Feb '03

11 Feb '03

Hi, Sorry to be greedy but I have two questions. I've just finished installed the ser 0.8 version on Solaris 2.8. 1. When I ran ser, it said it is listening to only 3 servers. I have added 2 more into my /etc/hosts file but make no difference. Where does ser daemon get the ip of servers that it is listening to? 2. I plan to use ser with my cisco voip gateways. Is there anything special that I need to configure ser to make it work with Cisco gateways? Thanks, Chi Dang

2 1

[Serusers] 3PCC-Szenario
by Karsten Knuettel 11 Feb '03

11 Feb '03

Hi, I am working on a 3rd Party Call Control implementation. Is it possible to get notifyed as a 3rd Party (SER owner) when Sip Party A connects SIP Party B? Thank you! karsten

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RE: [Serusers] user permission problem on ser-jabber gw
by Toni Barata (EPS) 11 Feb '03

11 Feb '03

Hi Daniel, Ok, I found a configuration error on ser.cfg. On the Jaddress parameter the Ip address had a sintax error (one of the dots was replaceded by a comma). But now there is another problem: - When the sip client (messenger) sends a IM to the Jabber client, the jabber client do not receive the message but he is "notified" that the sip-client is on-line. - When the jabber client send an IM to the sip client the messages is received by the sip client. Any idea of what is going on? Best regards, Toni -----Original Message----- From: Daniel-Constantin MIERLA [mailto:mierla@fokus.fraunhofer.de] Sent: sexta-feira, 31 de Janeiro de 2003 18:30 To: Toni Barata (EPS) Subject: Re: [Serusers] user permission problem on ser-jabber gw You have to set debug=9 and log_stderror=yes in ser's config file. Debug messages will be displayed on console and you can copy/paste in a file. If you got that error message, means that ser-jabber gw was connected to Jabber server, but something was happening and the connection was broken. So some messages should have been exchanged between ser and Jabber. Did you check all your network interfaces (I told you in last mail you have to sniff loopback interface if Jabber server and ser is on the same machine)? Regards, Daniel Toni Barata (EPS) wrote: Hi, I forgot to tell you that I put the ethereal sniffing the network and no packet was sent by the ser-jabber gw to the port 5222 on the jabber server. I seems that this is an internal problem on the ser-jabber. Is there a way to send the logs to a file? Best regards, Toni

2 1

[Serusers] Simple methods
by Anuradha Kalyan 11 Feb '03

11 Feb '03

hello, we are using ser 0.8.10. it worked fine with SIP methods. we would like to know whether we can use the ser to test the simple methods (simple, subscribe and notify methods) with thanks and regards, Anuradha -- ************************************************************** eRiva provides IT Solutions & Services to companies worldwide. Our State of the Art Research & Development Center provides Innovative Solutions to global customers. eRiva's unique value proposition is further delivered through our Offshore Outsourcing Model and stringent Quality Processes of SEI / ISO and Six Sigma. [www.erivasystems.com] ************************************************************** --

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[Serusers] Bug in new_hash2() function [patch]
by Maxim Sobolev 09 Feb '03

09 Feb '03

Folks, There is a bug in ser, which results in different hashes to be generated if some of the fields participating in hash generation has extra spaces before value in reply that were not present in request, i.e.: request: From: <vasia_pupkin(a)1.2.3.4> reply: From: <vasia_pupkin(a)1.2.3.4> This leads to matching failure and as a result causes useless retransmition, if tm module is used. For example, Vovida's SIP stack is found to be guilty in inserting extra spaces into reply's headers (which isn't really prohibited by the RFC). Attached little patch should fix the problem. It would be nice to have it integrated into the next release. Thanks! -Maxim

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[Serusers] SER and SIP-CGI
by Stephan Reiff-Marganiec 07 Feb '03

07 Feb '03

Hi, I was wondering whether there is a ser module for SIP-CGI. Or whether there are attempts at creating one? If so, could anyone point me in the right direction? thanks, Stephan -- Dr Stephan Reiff-Marganiec Research Fellow Department of Computing Science; University of Stirling email: srm(a)cs.stir.ac.uk tel: 01786 46 7448 -- The University of Stirling is a university established in Scotland by charter at Stirling, FK9 4LA. Privileged/Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not disclose, copy or deliver this message to anyone and any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer do not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of the University of Stirling shall be understood as neither given nor endorsed by it.

2 1

[Serusers] core when running the rad_acc module
by Claudio Thorell dos Santos 07 Feb '03

07 Feb '03

Hello, I'm trying to setup the radius accounting module but SER craches wherever I make a call. The call does complete, but the core is generated a couple seconds later. Without rad_acc moudule SER works fine, so I assume this have something to do with the module. I'm attaching some resumed debugs for the crash. I also have the complete debug (log level 10), wich is quite big so I decided not so send it to the list unless the following debugs are not enough. Any ideas on what's the problem? Thanks and regards, Claudio Thorell ########### ... 6(24956) qm_free(0x80adf80, 0x80b5174), called from parser/msg_parser.c: free_sip_msg(487) 6(24956) qm_free: freeing block alloc'ed from action.c: do_action(346) 6(24956) qm_free(0x80adf80, 0x80b5090), called from parser/parse_via.c: free_via_param_list(1816) 6(24956) qm_free: freeing block alloc'ed from parser/parse_via.c: parse_via(1642) 6(24956) qm_free(0x80adf80, 0x80b52ec), called from parser/parse_via.c: free_via_list(1828) 6(24956) qm_free: freeing block alloc'ed from parser/msg_parser.c: get_hdr_field(85) 6(24956) qm_free(0x80adf80, 0x80b5128), called from parser/hf.c: free_hdr_field_lst(103) 6(24956) qm_free: freeing block alloc'ed from parser/msg_parser.c: parse_headers(229) 6(24956) qm_free(0x80adf80, 0x80b4ff0), called from parser/hf.c: free_hdr_field_lst(103) 6(24956) qm_free: freeing block alloc'ed from parser/msg_parser.c: parse_headers(229) 6(24956) qm_free(0x80adf80, 0x80b4fa4), called from parser/hf.c: free_hdr_field_lst(103) 6(24956) qm_free: freeing block alloc'ed from parser/msg_parser.c: parse_headers(229) 6(24956) qm_free(0x80adf80, 0x80b46cc), called from parser/parse_to.c: free_to(741) 6(24956) qm_free: freeing block alloc'ed from parser/parse_to.c: parse_to_param(265) 6(24956) qm_free(0x80adf80, 0x80b482c), called from parser/parse_to.c: free_to(744) 6(24956) qm_free: freeing block alloc'ed from parser/msg_parser.c: get_hdr_field(123) 6(24956) qm_free(0x80adf80, 0x80b479c), called from parser/hf.c: free_hdr_field_lst(103) 6(24956) qm_free: freeing block alloc'ed from parser/msg_parser.c: parse_headers(229) 6(24956) qm_free(0x80adf80, 0x80b51dc), called from parser/hf.c: free_hdr_field_lst(103) 6(24956) qm_free: freeing block alloc'ed from parser/msg_parser.c: parse_headers(229) 6(24956) qm_free(0x80adf80, 0x80b47e8), called from parser/parse_cseq.c: free_cseq(94) 6(24956) qm_free: freeing block alloc'ed from parser/msg_parser.c: get_hdr_field(104) 6(24956) qm_free(0x80adf80, 0x80b5228), called from parser/hf.c: free_hdr_field_lst(103) 6(24956) qm_free: freeing block alloc'ed from parser/msg_parser.c: parse_headers(229) 6(24956) qm_free(0x80adf80, 0x80b4f58), called from parser/hf.c: free_hdr_field_lst(103) 6(24956) qm_free: freeing block alloc'ed from parser/msg_parser.c: parse_headers(229) 6(24956) qm_free(0x80adf80, 0x80b50dc), called from parser/hf.c: free_hdr_field_lst(103) 6(24956) qm_free: freeing block alloc'ed from parser/msg_parser.c: parse_headers(229) 6(24956) qm_free(0x80adf80, 0x80b4c70), called from parser/hf.c: free_hdr_field_lst(103) 6(24956) qm_free: freeing block alloc'ed from parser/msg_parser.c: parse_headers(229) 6(24956) qm_free(0x80adf80, 0x80b4d08), called from data_lump.c: free_lump_list(248) 6(24956) qm_free: freeing block alloc'ed from data_lump.c: del_lump(153) 6(24956) qm_free(0x80adf80, 0x80b475c), called from data_lump.c: free_lump(212) 6(24956) qm_free: freeing block alloc'ed from textops.c: append_hf(231) 6(24956) qm_free(0x80adf80, 0x80b4418), called from data_lump.c: free_lump_list(237) 6(24956) qm_free: freeing block alloc'ed from data_lump.c: insert_new_lump_before(129) 6(24956) qm_free(0x80adf80, 0x80b4cbc), called from data_lump.c: free_lump_list(248) 6(24956) qm_free: freeing block alloc'ed from data_lump.c: anchor_lump(183) 6(24956) qm_free(0x80adf80, 0x80b54bc), called from parser/msg_parser.c: free_sip_msg(493) 6(24956) qm_free: freeing block alloc'ed from receive.c: receive_msg(83) 6(24956) qm_free(0x80adf80, 0x80b4adc), called from receive.c: receive_msg(177) 6(24956) qm_free: freeing block alloc'ed from receive.c: receive_msg(63) 6(24956) qm_malloc(0x80adf80, 356) called from receive.c: receive_msg(63) 6(24956) qm_malloc(0x80adf80, 356) returns address 0x80b4adc on 0 -th hit 6(24956) qm_malloc(0x80adf80, 415) called from receive.c: receive_msg(83) 6(24956) qm_malloc(0x80adf80, 416) returns address 0x80b54bc on 0 -th hit 6(24956) SIP Request: 6(24956) method: <ACK> 6(24956) uri: <sip:1005@200.219.173.141;branch=0> 6(24956) version: <SIP/2.0> 6(24956) parse_headers: flags=1 6(24956) qm_malloc(0x80adf80, 28) called from parser/msg_parser.c: parse_headers(229) 6(24956) qm_malloc(0x80adf80, 28) returns address 0x80b4cbc on 0 -th hit 6(24956) qm_malloc(0x80adf80, 104) called from parser/msg_parser.c: get_hdr_field(85) 6(24956) qm_malloc(0x80adf80, 104) returns address 0x80b52ec on 0 -th hit 0(24946) child process 24961 exited by a signal 11 0(24946) core was generated 0(24946) INFO: terminating due to SIGCHLD 4(24954) Memory status (pkg): 4(24954) qm_status (0x80adf80): 4(24954) heap size= 1022872 4(24954) used= 1040, used+overhead=27272, free=995600 4(24954) max used (+overhead)= 27340 4(24954) dumping all allocked. fragments: ...

1 0

Re: [Serusers] How to configure Serweb
by Jiri Kuthan 06 Feb '03

06 Feb '03

Dan Austin's howto (see SER webpage) is probably the currently most extensive documentation available. -Jiri At 05:16 PM 2/6/2003, Jerry Yin wrote: >Hi All, > >Can somebody tell me how to configure the serweb? Which directory should I >put those files and which file and parameters should I modify. Or maybe this >question was asked previously by somebody already, then please provide me a >like to those answers. > >Thanks, > >Jerry Yin >Mitel Networks. >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Serusers] How to configure Serweb
by Jerry Yin 06 Feb '03

06 Feb '03

Hi All, Can somebody tell me how to configure the serweb? Which directory should I put those files and which file and parameters should I modify. Or maybe this question was asked previously by somebody already, then please provide me a like to those answers. Thanks, Jerry Yin Mitel Networks.

1 0

RE: [Serusers] ser crashes
by Klaus Darilion 06 Feb '03

06 Feb '03

Hi! I'm really very sorry that I have wasted your time - I've made a really stupid mistake. Now I fixed the proper line (103) and everything works fine. No messages in the log file, no crashes. Thanks a lot! Klaus > -----Original Message----- > From: Jan Janak [mailto:J.Janak@sh.cvut.cz] > Sent: Thursday, February 06, 2003 4:45 PM > To: Klaus Darilion > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] ser crashes > > > On 06-02 16:25, Klaus Darilion wrote: > > My experience is the following: > > > > Line 109 in t_msgbuilder.c: > > > > len += USER_AGENT_LEN + CRLF_LEN; > > ==> ser crashes when I send the special invite! > > > > *len += USER_AGENT_LEN + CRLF_LEN; > > ==> ser doesn't crash but logs this in /val/log/messages: > > Feb 6 15:48:24 obelix /usr/local/sbin/ser[6985]: ERROR: > > t_build_and_send_CANCEL: cannot allocate memory > > > > > > hope my explanation is a little bit clearer now: > > i also downloaded > > ftp://ftp.berlios.de/pub/ser/0.8.10/src/ser-0.8.10_src.tar.gz > > and this tarball contains: *len += USER_AGENT_LEN + CRLF_LEN; > > > > That's another line, look at line 103: > > len+=((hdr->body.s+hdr->body.len ) - hdr->name.s ) + > CRLF_LEN; > > This should be changed to: > > *len+=((hdr->body.s+hdr->body.len ) - hdr->name.s ) + > CRLF_LEN; > > regards, Jan. > >

1 0

RE: [Serusers] ser crashes
by Klaus Darilion 06 Feb '03

06 Feb '03

My experience is the following: Line 109 in t_msgbuilder.c: len += USER_AGENT_LEN + CRLF_LEN; ==> ser crashes when I send the special invite! *len += USER_AGENT_LEN + CRLF_LEN; ==> ser doesn't crash but logs this in /val/log/messages: Feb 6 15:48:24 obelix /usr/local/sbin/ser[6985]: ERROR: t_build_and_send_CANCEL: cannot allocate memory hope my explanation is a little bit clearer now: i also downloaded ftp://ftp.berlios.de/pub/ser/0.8.10/src/ser-0.8.10_src.tar.gz and this tarball contains: *len += USER_AGENT_LEN + CRLF_LEN; regards, klaus > -----Original Message----- > From: Jan Janak [mailto:J.Janak@sh.cvut.cz] > Sent: Thursday, February 06, 2003 4:19 PM > To: Klaus Darilion > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] ser crashes > > > Hello, > > On 06-02 16:02, Klaus Darilion wrote: > > Thank you, the path to the modules was the problem! > > > > Now ser runs also as self compiled version and doesnt > crash. I didn't > > had to change the source because the source code available > to download > > is already corrected. To proof your guess I inserted the > error again ( > > len+=) and than it crashes again. > > ser-0.8.10_src.tar.gz has this bug too. > > > Conclusion: with the source code version from 0.8.10 ser > doesn't crash > > anymore. Nevertheless, /var/log/messages shows an error > message when I > > send my special invite: > > > > Feb 6 15:48:24 obelix /usr/local/sbin/ser[6985]: ERROR: > > t_build_and_send_CANCEL: cannot allocate memory > > That's the bug I told you about, AFAIK it is caused by the missing > dereference operator in t_msgbuilder.c. See my previous email for > description how to fix it. (rewrite len+= to *len+=, recompile and > reinstall). > > regards, Jan. > >

2 1

RE: [Serusers] ser crashes
by Klaus Darilion 06 Feb '03

06 Feb '03

Thank you, the path to the modules was the problem! Now ser runs also as self compiled version and doesnt crash. I didn't had to change the source because the source code available to download is already corrected. To proof your guess I inserted the error again ( len+=) and than it crashes again. Conclusion: with the source code version from 0.8.10 ser doesn't crash anymore. Nevertheless, /var/log/messages shows an error message when I send my special invite: Feb 6 15:48:24 obelix /usr/local/sbin/ser[6985]: ERROR: t_build_and_send_CANCEL: cannot allocate memory thanks and regards, Klaus > -----Original Message----- > From: Jan Janak [mailto:J.Janak@sh.cvut.cz] > Sent: Thursday, February 06, 2003 3:36 PM > To: Klaus Darilion > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] ser crashes > > > Double check path of the modules. Probably you are using > config file from the rpm package, after make install the > modules will be in /usr/local/lib/ser/modules. The rpm > package installs them in /usr/lib/ser/modules so you will > have to change it. > > There is no uninstall in the makefiles, you can uninstall > only by hand right now. > > regards, Jan. > > On 06-02 15:19, Klaus Darilion wrote: > > Hello! > > > > I have problems running the self compiled ser version (without any > > sourcecode changes). > > > > I uninstalled the rpms ("rpm -e ser-mysql" and "rpm -e ser" - I use > > the mysql authentication). Then I compiled ser with "make > all" without > > errors. I installed ser with "make install". Then I tried > to start ser > > with: > > # /usr/local/sbin/ser -f /etc/ser/ser.cfg > > ERROR: bad config file (25 errors) > > > > Why does my config file doesn't work anymore (it worked > with the rpm > > version)? When I start ser with -f, than it starts (but than it > > doesn't use the mysql database). > > > > Have I done something wrong during the installation/compiling? > > > > How can I delete the files that were installed by "make > install"? make > > uninstall? > > > > regards, > > klaus > > > > > -----Original Message----- > > > From: Jan Janak [mailto:J.Janak@sh.cvut.cz] > > > Sent: Thursday, February 06, 2003 12:12 AM > > > To: Klaus Darilion > > > Cc: serusers(a)lists.iptel.org > > > Subject: Re: [Serusers] ser crashes > > > > > > > > > Hello, > > > > > > I wasn't able to reproduce it, but I think I know where the > > > problem is. There is a bug in 0.8.10 which shows up when an > > > INVITE contains Route header field. The bug has been fixed > > > in the CVS already. > > > > > > If you can compile ser from sources, you can easily fix > it in your > > > release: > > > > > > 1) Edit ser-0.8.10/modules/tm/t_msgbuilder.c > > > 2) Jump to line 103 > > > 3) replace len+= with *len+= > > > 4) Recompile ser > > > > > > I can send you a patch if it is more convinient for you. > > > > > > regards, Jan. > > > > > > On 05-02 12:13, Klaus Darilion wrote: > > > > Hello! > > > > > > > > During playing around with route headers I discovered that > > > ser crashes > > > > reproduceable when I send the following request to it with > > > sipsak, I > > > > think the invite is a valid request with strict routing. > > > > > > > > sipsak.exe -f invite3.txt -s sip:obelix.ict.tuwien.ac.at -vvv > > > > > > > > invite3.txt: > > > > INVITE sip:obelix.ict.tuwien.ac.at SIP/2.0 > > > > Via: SIP/2.0/UDP 128.131.80.191:5070;branch=z9hG4bK2024693217 > > > > Route: <sip:iptel.org> > > > > Route: <sip:klaus3000@klausix.ict.tuwien.ac.at> > > > > From: <sip:brutus@obelix.ict.tuwien.ac.at>;tag=262992618 > > > > To: <sip:klaus3000@iptel.org> > > > > Call-ID: 744056460(a)128.131.80.191 > > > > CSeq: 20 INVITE > > > > Contact: <sip:brutus@128.131.80.191:5070> > > > > max-forwards: 10 > > > > user-agent: oSIP/Linphone-0.9.1 > > > > Content-Type: application/sdp > > > > Content-Length: 335 > > > > > > > > .. > > > > sdp > > > > .. > > > > > > > > > > > > then ser stops to respond: > > > > > > > > >[root@obelix darilion]# /etc/init.d/ser status > > > > >ser dead but subsys locked > > > > > > > > and in /var/log/messages > > > > Feb 5 11:57:21 obelix ser: Listening on > > > > Feb 5 11:57:21 obelix ser: 128.131.80.136 > > > > [128.131.80.136]::5060 > > > > Feb 5 11:57:21 obelix ser: Aliases: > obelix.ict.tuwien.ac.at:5060 > > > > Feb 5 11:57:21 obelix ser: ser startup succeeded Feb > 5 11:57:21 > > > > obelix /usr/sbin/ser[21426]: mod_init(): Database connection > > > > opened successfuly Feb 5 11:57:21 obelix /usr/sbin/ser[21426]: > > > > INFO: > > > udp_init: SO_RCVBUF > > > > is initially 65535 > > > > Feb 5 11:57:21 obelix /usr/sbin/ser[21426]: INFO: > > > udp_init: SO_RCVBUF > > > > is finally 131070 > > > > Feb 5 11:57:21 obelix /usr/sbin/ser[21432]: INFO: fifo process > > > > starting: 21432 > > > > Feb 5 11:57:21 obelix /usr/sbin/ser[21432]: SER: > > > open_uac_fifo: fifo > > > > server up at /tmp/ser_fifo... > > > > > > > > ..up to here everything works fine, than I send the request: > > > > > > > > Feb 5 11:57:30 obelix /usr/sbin/ser[21428]: BUG: qm_*: fragm. > > > > 0x422b9ad0 end overwritten(35322e64, 39306665)! Feb 5 > > > 11:57:30 obelix > > > > /usr/sbin/ser[21426]: child process 21428 exited by a signal 11 > > > > Feb 5 11:57:30 obelix /usr/sbin/ser[21426]: core was not > > > > generated Feb 5 11:57:30 obelix /usr/sbin/ser[21426]: INFO: > > > terminating due to > > > > SIGCHLD > > > > > > > > Here is the ngrep output: > > > > > > > > U 128.131.80.101:1558 -> 128.131.80.136:5060 > > > > INVITE sip:obelix.ict.tuwien.ac.at SIP/2.0..Via: SIP/2.0/UDP > > > > klausix.ict.tuwien.ac.at:1557..Via: SIP/2.0/UDP > > > > 128.131.80.191:5070;branch=z9hG4bK2024693217..Route: <sip:iptel. > > > > org>..Route: <sip:klaus3000@klausix.ict.tuwien.ac.at>..From: > > > > <sip:brutus@obelix.ict.tuwien.ac.at>;tag=262992618..To: > > > > <sip:klaus3000@iptel.org>..Call-ID: 744056460(a)128.131.80 > > > > .191..CSeq: 20 INVITE..Contact: > > > > <sip:brutus@128.131.80.191:5070>..max-forwards: 10..user-agent: > > > > oSIP/Linphone-0.9.1..Content-Type: > > > > application/sdp..Content-Length: > > > > 335 .... > > > > v=0..o=brutus 123456 654321 IN IP4 128.131.80.191..s=A > > > > c..onversation..c=IN IP4 128.131.80.191..t=0 0..m=audio > > > 7078 RTP/AVP 3 > > > > 115 8 110 111 101..a=rtpmap:3 gsm/8000/1..a=rtp > > > > map:115 lpc10-1.5/8000/1..a=rtpmap:8 PCMA/8000/1..a=rtpmap:110 > > > > speex-4/8000/1..a=rtpmap:111 speex_lbr-4/8000/1..a=rtpmap:101 > > > > telephone-event/8000..a=fmtp:101 0-11.. > > > > # > > > > U 128.131.80.136:5060 -> 128.131.80.101:1557 > > > > SIP/2.0 100 trying -- your call is important to us..Via: > > > SIP/2.0/UDP > > > > klausix.ict.tuwien.ac.at:1557..Via: SIP/2.0/UDP > > > > 128.131.80.191:5070;branch=z9hG4bK2024693217..From: <sip > > > > :brutus@obelix.ict.tuwien.ac.at>;tag=262992618..To: > > > > <sip:klaus3000@iptel.org>..Call-ID: > > > 744056460@128.131.80.191..CSeq: 20 > > > > INVITE..Server: Sip EXpress router (0.8.10 (i386/l > > > > inux))..Content-Length: 0..Warning: 392 > > > 128.131.80.136:5060 "Noisy > > > > feedback tells: pid=21552 req_src_ip=128.131.80.101 > > > > in_uri=sip:obelix.ict.tuwien.ac.at out_uri=sip:iptel.o > > > > rg via_cnt==2".... > > > > > > > > # > > > > U 128.131.80.136:5060 -> 195.37.77.101:5060 > > > > INVITE sip:iptel.org SIP/2.0..Via: SIP/2.0/UDP > > > > 128.131.80.136;branch=z9hG4bK3a6d.25ef09d1.0..Via: SIP/2.0/UDP > > > > klausix.ict.tuwien.ac.at:1557..Via: SIP/2.0/UDP 128.131.80.191: > > > > 5070;branch=z9hG4bK2024693217..Route: > > > > <sip:klaus3000@klausix.ict.tuwien.ac.at>..From: > > > > <sip:brutus@obelix.ict.tuwien.ac.at>;tag=262992618..To: > > > > <sip:klaus3000@iptel.org>..Call > > > > -ID: 744056460@128.131.80.191..CSeq: 20 INVITE..Contact: > > > > <sip:brutus@128.131.80.191:5070>..max-forwards: 9 ..user-agent: > > > > oSIP/Linphone-0.9.1..Content-Type: application/sdp.. > > > > Content-Length: 335 ....v=0..o=brutus 123456 654321 IN IP4 > > > > 128.131.80.191..s=A c..onversation..c=IN IP4 > 128.131.80.191..t=0 > > > > 0..m=audio 7078 RTP/AVP 3 115 8 110 111 101..a=r > > > > tpmap:3 gsm/8000/1..a=rtpmap:115 lpc10-1.5/8000/1..a=rtpmap:8 > > > > PCMA/8000/1..a=rtpmap:110 speex-4/8000/1..a=rtpmap:111 > > > > speex_lbr-4/8000/1..a=rtpmap:101 telephone-event/8000..a > > > > =fmtp:101 0-11.. > > > > > > > > # > > > > U 195.37.77.101:5060 -> 128.131.80.136:5060 > > > > SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP > > > > 128.131.80.136;branch=z9hG4bK3a6d.25ef09d1.0..Via: SIP/2.0/UDP > > > > klausix.ict.tuwien.ac.at:1557..Via: SIP/2.0/UDP 12 > > > > 8.131.80.191:5070;branch=z9hG4bK2024693217..From: > > > > <sip:brutus@obelix.ict.tuwien.ac.at>;tag=262992618..To: > > > > > <sip:klaus3000@iptel.org>;tag=fb86ad2694115d75c77dce61523c9f07.3f3c > > > > ..Call-ID: 744056460@128.131.80.191..CSeq: 20 > > > > INVITE..Proxy-Authenticate: Digest realm="iptel.org", > > > > nonce="3e40f29900000000d6e1790446755f1b4b96f5f5bd97b283", > > > > algorithm=MD5.. > > > > Server: Sip EXpress router (0.8.9 > (i386/linux))..Content-Length: > > > > 0..Warning: 392 195.37.77.101:5060 "Noisy feedback > tells: pid=7651 > > > > req_src_ip=128.131.80.136 in_uri=sip:ipte > > > > l.org out_uri=sip:klaus3000@klausix.ict.tuwien.ac.at > > > via_cnt==3".... > > > > > > > > # > > > > U 128.131.80.136:5060 -> 128.131.80.101:1557 > > > > SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP > > > > klausix.ict.tuwien.ac.at:1557..Via: SIP/2.0/UDP > > > > 128.131.80.191:5070;branch=z9hG4bK2024693217..From: > <sip:brutus@o > > > > belix.ict.tuwien.ac.at>;tag=262992618..To: > > > > > > > > <sip:klaus3000@iptel.org>;tag=fb86ad2694115d75c77dce61523c9f07.3f3c. > > > .C > > > > al > > > > l-ID: 744056460@128.131.80.191..CSeq: 20 INVITE..Proxy-Aut > > > > henticate: Digest realm="iptel.org", > > > > nonce="3e40f29900000000d6e1790446755f1b4b96f5f5bd97b283", > > > > algorithm=MD5..Server: Sip EXpress router (0.8.9 > > > > (i386/linux))..Content-Length > > > > : 0..Warning: 392 195.37.77.101:5060 "Noisy feedback > > > tells: pid=7651 > > > > req_src_ip=128.131.80.136 in_uri=sip:iptel.org > > > > out_uri=sip:klaus3000@klausix.ict.tuwien.ac.at via_cnt==3 > > > > ".... > > > > > > > > # > > > > U 128.131.80.136:5060 -> 195.37.77.101:5060 > > > > ACK sip:iptel.org SIP/2.0..Via: SIP/2.0/UDP > > > > 128.131.80.136;branch=z9hG4bK3a6d.25ef09d1.0..From: > > > > <sip:brutus@obelix.ict.tuwien.ac.at>;tag=262992618..Call-ID: > > > > 744056460(a)128.13 > > > > 1.80.191..To: > > > > > > > <sip:klaus3000@iptel.org>;tag=fb86ad2694115d75c77dce61523c9f07 > > > .3f3c..CSe > > > > q: 20 ACK.. > > > > > > > > > > > > > > > > If you need more details please let me know. > > > > > > > > regards, > > > > Klaus > > > > _______________________________________________ > > > > Serusers mailing list > > > > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > >

2 1

RE: [Serusers] ser crashes
by Klaus Darilion 06 Feb '03

06 Feb '03

Hello! I have problems running the self compiled ser version (without any sourcecode changes). I uninstalled the rpms ("rpm -e ser-mysql" and "rpm -e ser" - I use the mysql authentication). Then I compiled ser with "make all" without errors. I installed ser with "make install". Then I tried to start ser with: # /usr/local/sbin/ser -f /etc/ser/ser.cfg ERROR: bad config file (25 errors) Why does my config file doesn't work anymore (it worked with the rpm version)? When I start ser with -f, than it starts (but than it doesn't use the mysql database). Have I done something wrong during the installation/compiling? How can I delete the files that were installed by "make install"? make uninstall? regards, klaus > -----Original Message----- > From: Jan Janak [mailto:J.Janak@sh.cvut.cz] > Sent: Thursday, February 06, 2003 12:12 AM > To: Klaus Darilion > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] ser crashes > > > Hello, > > I wasn't able to reproduce it, but I think I know where the > problem is. There is a bug in 0.8.10 which shows up when an > INVITE contains Route header field. The bug has been fixed > in the CVS already. > > If you can compile ser from sources, you can easily fix it in your > release: > > 1) Edit ser-0.8.10/modules/tm/t_msgbuilder.c > 2) Jump to line 103 > 3) replace len+= with *len+= > 4) Recompile ser > > I can send you a patch if it is more convinient for you. > > regards, Jan. > > On 05-02 12:13, Klaus Darilion wrote: > > Hello! > > > > During playing around with route headers I discovered that > ser crashes > > reproduceable when I send the following request to it with > sipsak, I > > think the invite is a valid request with strict routing. > > > > sipsak.exe -f invite3.txt -s sip:obelix.ict.tuwien.ac.at -vvv > > > > invite3.txt: > > INVITE sip:obelix.ict.tuwien.ac.at SIP/2.0 > > Via: SIP/2.0/UDP 128.131.80.191:5070;branch=z9hG4bK2024693217 > > Route: <sip:iptel.org> > > Route: <sip:klaus3000@klausix.ict.tuwien.ac.at> > > From: <sip:brutus@obelix.ict.tuwien.ac.at>;tag=262992618 > > To: <sip:klaus3000@iptel.org> > > Call-ID: 744056460(a)128.131.80.191 > > CSeq: 20 INVITE > > Contact: <sip:brutus@128.131.80.191:5070> > > max-forwards: 10 > > user-agent: oSIP/Linphone-0.9.1 > > Content-Type: application/sdp > > Content-Length: 335 > > > > .. > > sdp > > .. > > > > > > then ser stops to respond: > > > > >[root@obelix darilion]# /etc/init.d/ser status > > >ser dead but subsys locked > > > > and in /var/log/messages > > Feb 5 11:57:21 obelix ser: Listening on > > Feb 5 11:57:21 obelix ser: 128.131.80.136 > > [128.131.80.136]::5060 > > Feb 5 11:57:21 obelix ser: Aliases: obelix.ict.tuwien.ac.at:5060 > > Feb 5 11:57:21 obelix ser: ser startup succeeded > > Feb 5 11:57:21 obelix /usr/sbin/ser[21426]: mod_init(): Database > > connection opened successfuly > > Feb 5 11:57:21 obelix /usr/sbin/ser[21426]: INFO: > udp_init: SO_RCVBUF > > is initially 65535 > > Feb 5 11:57:21 obelix /usr/sbin/ser[21426]: INFO: > udp_init: SO_RCVBUF > > is finally 131070 > > Feb 5 11:57:21 obelix /usr/sbin/ser[21432]: INFO: fifo process > > starting: 21432 > > Feb 5 11:57:21 obelix /usr/sbin/ser[21432]: SER: > open_uac_fifo: fifo > > server up at /tmp/ser_fifo... > > > > ..up to here everything works fine, than I send the request: > > > > Feb 5 11:57:30 obelix /usr/sbin/ser[21428]: BUG: qm_*: fragm. > > 0x422b9ad0 end overwritten(35322e64, 39306665)! Feb 5 > 11:57:30 obelix > > /usr/sbin/ser[21426]: child process 21428 exited by a signal 11 > > Feb 5 11:57:30 obelix /usr/sbin/ser[21426]: core was not generated > > Feb 5 11:57:30 obelix /usr/sbin/ser[21426]: INFO: > terminating due to > > SIGCHLD > > > > Here is the ngrep output: > > > > U 128.131.80.101:1558 -> 128.131.80.136:5060 > > INVITE sip:obelix.ict.tuwien.ac.at SIP/2.0..Via: SIP/2.0/UDP > > klausix.ict.tuwien.ac.at:1557..Via: SIP/2.0/UDP > > 128.131.80.191:5070;branch=z9hG4bK2024693217..Route: <sip:iptel. > > org>..Route: <sip:klaus3000@klausix.ict.tuwien.ac.at>..From: > > <sip:brutus@obelix.ict.tuwien.ac.at>;tag=262992618..To: > > <sip:klaus3000@iptel.org>..Call-ID: 744056460(a)128.131.80 > > .191..CSeq: 20 INVITE..Contact: > > <sip:brutus@128.131.80.191:5070>..max-forwards: 10..user-agent: > > oSIP/Linphone-0.9.1..Content-Type: application/sdp..Content-Length: > > 335 .... > > v=0..o=brutus 123456 654321 IN IP4 128.131.80.191..s=A > > c..onversation..c=IN IP4 128.131.80.191..t=0 0..m=audio > 7078 RTP/AVP 3 > > 115 8 110 111 101..a=rtpmap:3 gsm/8000/1..a=rtp > > map:115 lpc10-1.5/8000/1..a=rtpmap:8 PCMA/8000/1..a=rtpmap:110 > > speex-4/8000/1..a=rtpmap:111 speex_lbr-4/8000/1..a=rtpmap:101 > > telephone-event/8000..a=fmtp:101 0-11.. > > # > > U 128.131.80.136:5060 -> 128.131.80.101:1557 > > SIP/2.0 100 trying -- your call is important to us..Via: > SIP/2.0/UDP > > klausix.ict.tuwien.ac.at:1557..Via: SIP/2.0/UDP > > 128.131.80.191:5070;branch=z9hG4bK2024693217..From: <sip > > :brutus@obelix.ict.tuwien.ac.at>;tag=262992618..To: > > <sip:klaus3000@iptel.org>..Call-ID: > 744056460@128.131.80.191..CSeq: 20 > > INVITE..Server: Sip EXpress router (0.8.10 (i386/l > > inux))..Content-Length: 0..Warning: 392 > 128.131.80.136:5060 "Noisy > > feedback tells: pid=21552 req_src_ip=128.131.80.101 > > in_uri=sip:obelix.ict.tuwien.ac.at out_uri=sip:iptel.o > > rg via_cnt==2".... > > > > # > > U 128.131.80.136:5060 -> 195.37.77.101:5060 > > INVITE sip:iptel.org SIP/2.0..Via: SIP/2.0/UDP > > 128.131.80.136;branch=z9hG4bK3a6d.25ef09d1.0..Via: SIP/2.0/UDP > > klausix.ict.tuwien.ac.at:1557..Via: SIP/2.0/UDP 128.131.80.191: > > 5070;branch=z9hG4bK2024693217..Route: > > <sip:klaus3000@klausix.ict.tuwien.ac.at>..From: > > <sip:brutus@obelix.ict.tuwien.ac.at>;tag=262992618..To: > > <sip:klaus3000@iptel.org>..Call > > -ID: 744056460@128.131.80.191..CSeq: 20 INVITE..Contact: > > <sip:brutus@128.131.80.191:5070>..max-forwards: 9 ..user-agent: > > oSIP/Linphone-0.9.1..Content-Type: application/sdp.. > > Content-Length: 335 ....v=0..o=brutus 123456 654321 IN IP4 > > 128.131.80.191..s=A c..onversation..c=IN IP4 128.131.80.191..t=0 > > 0..m=audio 7078 RTP/AVP 3 115 8 110 111 101..a=r > > tpmap:3 gsm/8000/1..a=rtpmap:115 lpc10-1.5/8000/1..a=rtpmap:8 > > PCMA/8000/1..a=rtpmap:110 speex-4/8000/1..a=rtpmap:111 > > speex_lbr-4/8000/1..a=rtpmap:101 telephone-event/8000..a > > =fmtp:101 0-11.. > > > > # > > U 195.37.77.101:5060 -> 128.131.80.136:5060 > > SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP > > 128.131.80.136;branch=z9hG4bK3a6d.25ef09d1.0..Via: SIP/2.0/UDP > > klausix.ict.tuwien.ac.at:1557..Via: SIP/2.0/UDP 12 > > 8.131.80.191:5070;branch=z9hG4bK2024693217..From: > > <sip:brutus@obelix.ict.tuwien.ac.at>;tag=262992618..To: > > <sip:klaus3000@iptel.org>;tag=fb86ad2694115d75c77dce61523c9f07.3f3c > > ..Call-ID: 744056460@128.131.80.191..CSeq: 20 > > INVITE..Proxy-Authenticate: Digest realm="iptel.org", > > nonce="3e40f29900000000d6e1790446755f1b4b96f5f5bd97b283", > > algorithm=MD5.. > > Server: Sip EXpress router (0.8.9 (i386/linux))..Content-Length: > > 0..Warning: 392 195.37.77.101:5060 "Noisy feedback tells: pid=7651 > > req_src_ip=128.131.80.136 in_uri=sip:ipte > > l.org out_uri=sip:klaus3000@klausix.ict.tuwien.ac.at > via_cnt==3".... > > > > # > > U 128.131.80.136:5060 -> 128.131.80.101:1557 > > SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP > > klausix.ict.tuwien.ac.at:1557..Via: SIP/2.0/UDP > > 128.131.80.191:5070;branch=z9hG4bK2024693217..From: <sip:brutus@o > > belix.ict.tuwien.ac.at>;tag=262992618..To: > > > <sip:klaus3000@iptel.org>;tag=fb86ad2694115d75c77dce61523c9f07.3f3c..C > > al > > l-ID: 744056460@128.131.80.191..CSeq: 20 INVITE..Proxy-Aut > > henticate: Digest realm="iptel.org", > > nonce="3e40f29900000000d6e1790446755f1b4b96f5f5bd97b283", > > algorithm=MD5..Server: Sip EXpress router (0.8.9 > > (i386/linux))..Content-Length > > : 0..Warning: 392 195.37.77.101:5060 "Noisy feedback > tells: pid=7651 > > req_src_ip=128.131.80.136 in_uri=sip:iptel.org > > out_uri=sip:klaus3000@klausix.ict.tuwien.ac.at via_cnt==3 > > ".... > > > > # > > U 128.131.80.136:5060 -> 195.37.77.101:5060 > > ACK sip:iptel.org SIP/2.0..Via: SIP/2.0/UDP > > 128.131.80.136;branch=z9hG4bK3a6d.25ef09d1.0..From: > > <sip:brutus@obelix.ict.tuwien.ac.at>;tag=262992618..Call-ID: > > 744056460(a)128.13 > > 1.80.191..To: > > > <sip:klaus3000@iptel.org>;tag=fb86ad2694115d75c77dce61523c9f07 > .3f3c..CSe > > q: 20 ACK.. > > > > > > > > If you need more details please let me know. > > > > regards, > > Klaus > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > > >

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RE: [Serusers] ser crashes
by Jiri Kuthan 06 Feb '03

06 Feb '03

Looking at log files may help you to discover the issue. It is very likely that the module paths are different for the compiled than for the installed modules. -jiri At 03:19 PM 2/6/2003, Klaus Darilion wrote: >Hello! > >I have problems running the self compiled ser version (without any >sourcecode changes). > >I uninstalled the rpms ("rpm -e ser-mysql" and "rpm -e ser" - I use the >mysql authentication). Then I compiled ser with "make all" without >errors. I installed ser with "make install". Then I tried to start ser >with: ># /usr/local/sbin/ser -f /etc/ser/ser.cfg >ERROR: bad config file (25 errors) > >Why does my config file doesn't work anymore (it worked with the rpm >version)?

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Re: [Serusers] Sip server
by Geetha Shree 06 Feb '03

06 Feb '03

Hi, Thanks for the Reply. I am using the SER server with the default configuration. Infact I have not installed the SQL database also. I am using the SER 0.8.10 verison. With Linux 7.3 as OS. I registered with the following SIP URI to SER server from two different machines. sip:anuradha@there.com sip:LittleGuy@there.com It registered Successfully. But When I sent an INVITE request from sip:anuradha@there.com to LittleGuy(a)there.com it gives the server error. Thanks Geetha ---------- Original Message ---------------------------------- From: Jiri Kuthan <serusers(a)lists.iptel.org> Date: Wed, 05 Feb 2003 19:13:48 +0100 >Please send a complete description of the case: >full network dumps, your configuration file, your >logs and information about version of ser and OS >you are using. > >Thank you, > >-Jiri > >At 02:02 PM 2/5/2003, Geetha Shree wrote: >>Hi, >> >>I am new to SIP. >>I have installed the SER server and my REGISTER method is working ok. >>But when I send the INVITE method I am getting the following error from SER server. >> >>The invite packet which has been sent by me is: >> >>INVITE sip:anuradha@there.com SIP/2.0 >>From: <sip:LittleGuy@there.com>;tag=1044365919410 >>To: <sip:anuradha@there.com > >>Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1159de0c6cca18f9131c8ff342527668 >>CSeq: 1 INVITE >>Call-Id: a1371c790546f9a2df76be1f59f6a7dc(a)192.168.1.9 >>Contact: <sip:192.168.1.9:5060;transport=UDP> >>Content-Length: 0 >> >>192.168.1.9 is my SER server's address >> >>The response got from the SER server is: >> >>SIP/2.0 500 I'm terribly sorry, server error occured (1/SL) >>From: <sip:LittleGuy@there.com>;tag=1044365919410 >>To: <sip:anuradha@there.com>;tag=d907c037823644515dfe0ede38ca9976.3592 >>Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1159de0c6cca18f9131c8ff342527668;received=192.168.1.21 >>CSeq: 1 INVITE >>Call-Id: a1371c790546f9a2df76be1f59f6a7dc(a)192.168.1.9 >>Server: Sip EXpress router (0.8.10 (i386/linux)) >>Content-Length: 0 >>Warning: 392 192.168.1.9:5060 "Noisy feedback tells: pid=1587 req_src_ip=192.168.1.21 in_uri=sip:LittleGuy@there.com out_uri=sip:LittleGuy@there.com via_cnt==1" >> >>Can you help on this please >> >>Thanks in advance >>Geetha >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers > >

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