- sr-users - kamailio.org

[Users] AVP subst callerid example
by Norman Brandinger 30 Jun '05

30 Jun '05

At the risk of being flamed by the developers that argue messing with SIP headers can break interoperability, cause parse problems, is a bad programming practice, etc, etc. I'm posting a little example of using an AVP to set the callerid of a user. Disclaimer: I'm not suggesting this is what anyone should or should not do, I'm just showing an interesting example. In the usr_preferences table, there is the following row: Username: user1 Attribute: callerid Value: "My Name" <sip:19995551212@example.com Note that the ending > after .com in the value column is missing on purpose. I haven't spent the time tweaking the regular expression below to enable the trailing > to be included. #------------------------------------------------------------------------------------------- # Set the callerid for the user from an AVP #------------------------------------------------------------------------------------------- if (avp_db_load("$from/username", "s:callerid")) { subst('/^From: (.*)>(.*)$/From: $avp(callerid)>\2/ig'); }; Some things to consider would be to use one subst to change the "name" and another subst to change the "number". Obviously, two AVPs would be needed to implement this. Regards, Norman Brandinger norm at goes dot com

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[Serusers] ser start with problem
by Bruno Oliveira 30 Jun '05

30 Jun '05

hi. When I initiate the being the archive/var/log/syslog I show the following line: Jun 30 09:19:22 "localhost" ser: WARNING: fix_socket_list: could not rev. resolve 192.168.1.84 Jun 30 09:19:22 "localhost" ser: WARNING: fix_socket_list: could not rev. resolve 192.168.1.84 Jun 30 09:19:22 "localhost" /usr/sbin/ser[19382]: Maxfwd module- initializing Jun 30 09:19:22 "localhost" /usr/sbin/ser[19383]: WARNING: no fifo_db_url given - fifo DB commands disabled! Anyone could help me? -- []s, Bruno "Niggas" Oliveira Belo Horizonte - MG Msn: n1gg4s(a)gmail.com Icq: 176314647 "Todo o nosso descontentamento por aquilo que nos falta procede da nossa falta de gratidão por aquilo que temos."

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Re: [Serusers] Logging into SERWEB
by Karel Kozlik 30 Jun '05

30 Jun '05

Hi, > I changed config.php as follows: > > $this->realm=$this->domainnam=egrep_replace(...) CHANGE TO > > $this->realm=$this->domainname="sip.mydomain.net" > > I don't understand why I had to do this. serweb can be used to control multiple domains. Only thing which you need for it is configure virtual servers for other domains in apache. For that reason is domain obtained automaticaly from http headers. But this not working if you use url http://localhost/serweb... in your browser. You must use http://your-domain/serweb... and it may be little confusing for new users. By your change you set the domain to "sip.mydomain.net" by hand. Karel > > Leo P. > > -----Original Message----- > From: Karel Kozlik [mailto:karel@iptel.org] > Sent: Thursday, June 30, 2005 9:31 AM > To: Leo > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] Logging into SERWEB > > Hi Leo, > which version of serweb are you useing? I wrote about the latest (0.9.3) > > > > > Thank you for the quick response and the thorough answer, but I > cannot seem > > to follow your advice because: > > 1) I cannot find the file 'configure_data_layer.php'. I even did and > > updatedb and a locate. > > 2) There is nothing about username and password on my ser.cfg. > > Sorry, I'm not expert to configuration of ser. But you should have in > ser.cfg something like this: > > modparam("usrloc|avp_db", "db_url", "mysql://ser:heslo@localhost/ser") > > If you haven't probably the default (mysql://ser:heslo@localhost/ser) is > used. This db_url is in format: > mysql://username:password@host/db_name > > > > >>3) I don't know which .php file to change to add '$config->enable_loging = >>true'. I cannot find this parameter in the config.php file. I also don't >>know to where the logs would be stored. > > > config.php is the right file. If 'enable_loging' isn't here you probably > have some old serweb. > > Karel > > > >>The result of the sql query were as follows >>Username = admin >>Password = heslo >>Domain = sip.mydomain.net >> >>Leo P. >> >>-----Original Message----- >>From: Karel Kozlik [mailto:karel@iptel.org] >>Sent: Thursday, June 30, 2005 7:36 AM >>To: Leo >>Cc: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Logging into SERWEB >> >> Hi Leo, >>there are two kinds of user: >>1. system user for login into mysql. >>Default username=ser, password=heslo. This is used for security your >>database against unauthorized access. This values should be set in >>config_data_layer.php (and ser.cfg of course). How to change default >>password you can find in mysql manual. >> >>2. sip users >>Serweb admin is special sip user. Default username is >>admin@<your_domain> and password is heslo. Usernames and passwords of >>sip users are stored in table 'subscriber' in DB. To check your exact >>values you can login to mysql from command line: >> >>mysql -u ser -pheslo ser >> >>and type: >> >>select username, domain, password from subscriber; >> >> >> >>If you still can't login into serweb follow this FAQ: >> >> Q: Still can't login into serweb by default username and password >> A: Enable loging in serweb ($config->enable_loging = true) and >> see log file. For more verbosity set log level to PEAR_LOG_DEBUG >> >> >>Karel >> >> >>Leo napsal(a): >> >> >>>Dear serusers, >>> >>>It still does not work. Here is what I did so far to figure this out. >>> >>>BACKGROUND INFO: >>>1) When I log in to MySQL from the command line I use the following >> >>command: >> >> >>>"mysql -pMyPwd ser" >>>2) From conf.php: db_host=123.123.123.123, db_name=ser, db_user=ser, >>>db_pass=heslo >>>3) My domain is sip.leosser.net >>> >>>QUESTIONS/PROBLEMS : >>>1) Shouldn't db_user='admin' and NOT 'ser' since this is what the >> >>'username' >> >> >>>field is for the password 'heslo' in the MySQL database? However, if I >>>change this I don't even get to the login screen on serweb. >>>2) I have tried logging in as admin(a)sip.leosser.net and password heslo, >>>root(a)sip.leosser.net and mypwd. Nothing works when I do get to the serweb >>>login screen. >>> >>>I am stuck. Anyone have any suggestions? >>> >>>Leo P. >>>leo(a)ltcjp.com >>> >>>-----Original Message----- >>>From: li wei [mailto:augzbaby@yahoo.com.au] >>>Sent: Wednesday, June 29, 2005 10:14 PM >>>To: Leo >>>Cc: serusers(a)lists.iptel.org >>>Subject: Re: [Serusers] Logging into SERWEB >>> >>>Please make sure your username, password and domain >>>values are exactly matched the values of your >>>subscriber table. >>> >>>Wilson >>> >>>--- Leo <leo(a)ltcjp.com> wrote: >>> >>> >>> >>> >>>>Dear serusers, >>>> >>>> >>>> >>>>I successfully installed SERWEB.I think. >>>> >>>> >>>> >>>>However, I cannot seem to login at the login screen. >>>> >>>> >>>> >>>>I go to http://mydomain/admin/index.php and I get >>>>the login screen, but >>>>nothing I try there works. What could this be? >>>> >>>> >>>> >>>>I already tried: >>>> >>>>'root' and my passoword >>>> >>>>admin and heslo >>>> >>>> >>>> >>>>I don't know what else to do. Please help. >>>> >>>> >>>> >>>>Sincerely, >>>> >>>> >>>> >>>>Leo Papadopoulos >>>> >>>>leo(a)ltcjp.com >>>> >>>> >>>> >>>> >>>> >>>> >>>>>_______________________________________________ >>>> >>>>Serusers mailing list >>>>serusers(a)lists.iptel.org >>>>http://lists.iptel.org/mailman/listinfo/serusers >>>> >>> >>> >>> >>> >>> >>> >>> >>> >>>____________________________________________________ >>>Do you Yahoo!? >>>Try Yahoo! Photomail Beta: Send up to 300 photos in one email! >>>http://au.photomail.mail.yahoo.com >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >> >> >

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[Serusers] Re : PSTN - SER Basic
by gunawan 30 Jun '05

30 Jun '05

Hi... Thanks Iqbal, Thanks Steve (http://mit.edu/sip/sip.edu/ciscoGW.html) for ur answer.... Thanks for getting started document at www.onsip.org :) It's really help me to understand PSTN - Ser.. :) hhmm.. I have another question... :) 1. Could u tell me the way how I can know that my SIP server is already connected and can receive signal from Cisco Gateway when I dial from PSTN to SER Client... 2. do Cisco gateway have any command or syntax to detect signal from SIP client....????? thanks.. ---------------------------------------------------- Hi 1. If you wish to call PSTN, you dial the PSTN number direct, however ser needs to be told based on that number / or based on that prefix where to go, i.e the IP address of the gateway. EG lets say the pstn number is 001 445 667788, then inser you would have something like (syntax will be wrong cause I cant recall these things, so please check) if (uri=sip:001*@*) { rewritehost(a.b.c.d) t_relay } a.b.c.d is your pstn gateway, look at www.onsip.org, they have a getting started doc, which covers pstn connection 2. If the pstn needs to call xlite, you need to tell you cisco gateway that when someone dials +62(761)53803 it should goto the ser proxy, and ser should know that thatr number maps to your xlite username 3. No Iqbal gunawan wrote: >hi, I'm newbiie in using ser ..... >I am confused about how ser can connect to PSTN.... >in my experiment i have two client using xlite.. the >name are xlite1 and xlite2 with alias number 6276101 >and 6276102. and I have two PSTN number provided by >local communication provider one is +62(761)53803 and >another one is +62(761)53939.. >I use 53803 for my PSTN gateway... I use cisco gateway >2620XM and VIC2FXO... > >1. If I want to call to PSTN number, what number >should I dial from xlite1?????? Do I have to dial my >PSTN Gateway first before I dial my destination >number???? > >2. If I want to make a call from PSTN to xlite1, what >should I dial after I dial my PSTN gateway number??? > >3. Do I have to add my destination PSTN number at >subscriber table in my database????? > >Thanks before... > > > > >____________________________________________________ >Yahoo! Sports >Rekindle the Rivalries. Sign up for Fantasy Football >http://football.fantasysports.yahoo.com > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > >. > > > ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

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[serusers] Message Body
by michael p 30 Jun '05

30 Jun '05

hi someone knows how can i got (and put in a variable) the "FROM" , the "TO" and the "BODY" of a method "MESSAGE" ? i want to got it in ser.cfg my aim is to got this to use them to send sms after via another function... someone can help me please? thanks M.

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Re: [Serusers] Logging into SERWEB
by Karel Kozlik 30 Jun '05

30 Jun '05

Hi Leo, which version of serweb are you useing? I wrote about the latest (0.9.3) > > Thank you for the quick response and the thorough answer, but I cannot seem > to follow your advice because: > 1) I cannot find the file 'configure_data_layer.php'. I even did and > updatedb and a locate. > 2) There is nothing about username and password on my ser.cfg. Sorry, I'm not expert to configuration of ser. But you should have in ser.cfg something like this: modparam("usrloc|avp_db", "db_url", "mysql://ser:heslo@localhost/ser") If you haven't probably the default (mysql://ser:heslo@localhost/ser) is used. This db_url is in format: mysql://username:password@host/db_name > 3) I don't know which .php file to change to add '$config->enable_loging = > true'. I cannot find this parameter in the config.php file. I also don't > know to where the logs would be stored. config.php is the right file. If 'enable_loging' isn't here you probably have some old serweb. Karel > > The result of the sql query were as follows > Username = admin > Password = heslo > Domain = sip.mydomain.net > > Leo P. > > -----Original Message----- > From: Karel Kozlik [mailto:karel@iptel.org] > Sent: Thursday, June 30, 2005 7:36 AM > To: Leo > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] Logging into SERWEB > > Hi Leo, > there are two kinds of user: > 1. system user for login into mysql. > Default username=ser, password=heslo. This is used for security your > database against unauthorized access. This values should be set in > config_data_layer.php (and ser.cfg of course). How to change default > password you can find in mysql manual. > > 2. sip users > Serweb admin is special sip user. Default username is > admin@<your_domain> and password is heslo. Usernames and passwords of > sip users are stored in table 'subscriber' in DB. To check your exact > values you can login to mysql from command line: > > mysql -u ser -pheslo ser > > and type: > > select username, domain, password from subscriber; > > > > If you still can't login into serweb follow this FAQ: > > Q: Still can't login into serweb by default username and password > A: Enable loging in serweb ($config->enable_loging = true) and > see log file. For more verbosity set log level to PEAR_LOG_DEBUG > > > Karel > > > Leo napsal(a): > >>Dear serusers, >> >>It still does not work. Here is what I did so far to figure this out. >> >>BACKGROUND INFO: >>1) When I log in to MySQL from the command line I use the following > > command: > >>"mysql -pMyPwd ser" >>2) From conf.php: db_host=123.123.123.123, db_name=ser, db_user=ser, >>db_pass=heslo >>3) My domain is sip.leosser.net >> >>QUESTIONS/PROBLEMS : >>1) Shouldn't db_user='admin' and NOT 'ser' since this is what the > > 'username' > >>field is for the password 'heslo' in the MySQL database? However, if I >>change this I don't even get to the login screen on serweb. >>2) I have tried logging in as admin(a)sip.leosser.net and password heslo, >>root(a)sip.leosser.net and mypwd. Nothing works when I do get to the serweb >>login screen. >> >>I am stuck. Anyone have any suggestions? >> >>Leo P. >>leo(a)ltcjp.com >> >>-----Original Message----- >>From: li wei [mailto:augzbaby@yahoo.com.au] >>Sent: Wednesday, June 29, 2005 10:14 PM >>To: Leo >>Cc: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Logging into SERWEB >> >>Please make sure your username, password and domain >>values are exactly matched the values of your >>subscriber table. >> >>Wilson >> >>--- Leo <leo(a)ltcjp.com> wrote: >> >> >> >>>Dear serusers, >>> >>> >>> >>>I successfully installed SERWEB.I think. >>> >>> >>> >>>However, I cannot seem to login at the login screen. >>> >>> >>> >>>I go to http://mydomain/admin/index.php and I get >>>the login screen, but >>>nothing I try there works. What could this be? >>> >>> >>> >>>I already tried: >>> >>>'root' and my passoword >>> >>>admin and heslo >>> >>> >>> >>>I don't know what else to do. Please help. >>> >>> >>> >>>Sincerely, >>> >>> >>> >>>Leo Papadopoulos >>> >>>leo(a)ltcjp.com >>> >>> >>> >>> >>> >>>>_______________________________________________ >>> >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >> >> >> >> >> >> >> >> >>____________________________________________________ >>Do you Yahoo!? >>Try Yahoo! Photomail Beta: Send up to 300 photos in one email! >>http://au.photomail.mail.yahoo.com >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers > >

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RE: [Serusers] Logging into SERWEB
by li wei 30 Jun '05

30 Jun '05

Hi leo --- Leo <leo(a)ltcjp.com> wrote: > Dear serusers, > > It still does not work. Here is what I did so far to > figure this out. > > BACKGROUND INFO: > 1) When I log in to MySQL from the command line I > use the following command: > "mysql -pMyPwd ser" > 2) From conf.php: db_host=123.123.123.123, > db_name=ser, db_user=ser, > db_pass=heslo Open the file config_data_layer.php in config folder and make sure the following values to fit your real situation: $i=0; $config->data_sql->host[$i]['host']="localhost"; //database host $config->data_sql->host[$i]['port']=""; //database port - leave empty for default $config->data_sql->host[$i]['name']="ser"; //database name $config->data_sql->host[$i]['user']="ser"; //database conection user $config->data_sql->host[$i]['pass']="heslo"; //database conection password > 3) My domain is sip.leosser.net > > QUESTIONS/PROBLEMS : > 1) Shouldn't db_user='admin' and NOT 'ser' since > this is what the 'username' > field is for the password 'heslo' in the MySQL > database? However, if I > change this I don't even get to the login screen on > serweb. Serweb uses the username,password etc. information from file ser_mysql.sh which used to create your ser tables to connect the mysql database. So you should change the above config file what I mentioned. And the login username,password,domain values is the information which store in the subscriber table of ser in mysql database. > 2) I have tried logging in as admin(a)sip.leosser.net > and password heslo, > root(a)sip.leosser.net and mypwd. Nothing works when I > do get to the serweb > login screen. > > I am stuck. Anyone have any suggestions? > 1) check how your database daemon to run. Do you only use mysqld_safe or not? If yes, change your database hostname as localhost. 2) Use phpinfo() function to make sure how do you configure your php. Dont use the default setting like ./configure --with-mysql; please specify a location for it. 3)Turn on your php display error function setting in php.ini file and try to install pear log and change the $config->log_level = "PEAR_LOG_WARNING" to $config->log_level = "PEAR_LOG_DEBUG" of config.php to see more details. Hope these can help you to solve your problem. Wilson > Leo P. > leo(a)ltcjp.com > > -----Original Message----- > From: li wei [mailto:augzbaby@yahoo.com.au] > Sent: Wednesday, June 29, 2005 10:14 PM > To: Leo > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] Logging into SERWEB > > Please make sure your username, password and domain > values are exactly matched the values of your > subscriber table. > > Wilson > > --- Leo <leo(a)ltcjp.com> wrote: > > > Dear serusers, > > > > > > > > I successfully installed SERWEB.I think. > > > > > > > > However, I cannot seem to login at the login > screen. > > > > > > > > I go to http://mydomain/admin/index.php and I get > > the login screen, but > > nothing I try there works. What could this be? > > > > > > > > I already tried: > > > > 'root' and my passoword > > > > admin and heslo > > > > > > > > I don't know what else to do. Please help. > > > > > > > > Sincerely, > > > > > > > > Leo Papadopoulos > > > > leo(a)ltcjp.com > > > > > > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > ____________________________________________________ > > Do you Yahoo!? > Try Yahoo! Photomail Beta: Send up to 300 photos in > one email! > http://au.photomail.mail.yahoo.com > > Send instant messages to your online friends http://au.messenger.yahoo.com

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[Serusers] Re: SER not starting
by Abdul Lateef 30 Jun '05

30 Jun '05

Hello iqbal Following the result of your given commond. the problem is still i can not kill the ser. Is there anyother such as commond so i can kill him? [root@localhost root]# /usr/local/sbin/ser -ddddddd -E -f /usr/local/etc/ser/ser.cfg 0(5480) read 2221522790 from /dev/random 0(5480) seeding PRNG with 3341667162 0(5480) test random number 921703968 0(5480) shm_mem_init: success Listening on udp: 10.0.0.20 [10.0.0.20]:5060 tcp: 10.0.0.20 [10.0.0.20]:5060 Aliases: [root@localhost root]# Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html

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[Serusers] How do I enable logging in SERWEB so I can debug my log-in issue?
by Leo 30 Jun '05

30 Jun '05

Dear serusers: I am try to be self-sufficient, but I need to know how to debug first. I searched my whole config.php file and there is nothing about logging in there. Can anyone tell me how to enable logging for serweb? Also where will the logs go? I already have logging enabled for ser using ser.cfg and those logs go into /var/log/messages. Thanks, Leo Papadopoulos leo(a)ltcjp.com

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[Serusers] 2 sers on one machine
by Pavel Siderov 30 Jun '05

30 Jun '05

Hi Guys, Is it possible to run 2 separate sers ( on 2 different ips ) on one machine? I tried but with no success any idea how I can make it work ? Thanks in advance, Pavel

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[Serusers] SER listen on multiple ports
by smy＠addm.com 30 Jun '05

30 Jun '05

Hi all, I tried searching for a similar thread in the iptel archive, but can't find any related discussion, so please forgive me if some of you have discussed this issue previously for I did try searching for it!! :> I'm trying to have SER listen on other ports on top of its default (5060) without success. In my configuration, I tried.... listen=public-ip port=5080 port=5060 But users can only register on port 5060. Once I disable the line port=5060, users can now register on port 5080. I have even tried.... listen=public-ip:5060 listen=public-ip:5080 This resulted in SER failing to start. Therefore I had no choice but to move back to my 1st sample above in adding the port option. My question boils down to the following... Is SER capable of listening on multiple ports?? If so, how do I configure it?? Any help is appreciated! Best Regards, Samuel Yeung

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[Serusers] What is the status of this open relay issue that Jan, Maxim, and Juha discuessed on 2003-10-23?
by Java Rockx 30 Jun '05

30 Jun '05

Hi All. This link is to a discussion on how to keep SER from being an open relay due to the following logic that most everyone uses in their ser.cfg files. if (uri!=myself) { t_relay(); break; } http://lists.iptel.org/pipermail/serusers/2003-October/003264.html Some time has passed on this issue and the 2003 article described a few "in-progress" patches. Does anyone know if SER now has the ability to prevent malicious users from exploiting (uri!=myself) code blocks? Regards, Paul

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[Serusers] RFC supported.
by Sohil Thakkar 30 Jun '05

30 Jun '05

Hi Janak, I am using SER to as the server to test the functionalities we are developing. Some of the application uses RFCs for INFO, PRACK etc. So could I get the list of RFCs that are supported by SER. I would also appreciate if certain RFCs are partially support and/or would be supported in future releases. Thanks a lot in advance. Regards, Sohil.

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[Serusers] SER listen on multiple ports
by smy＠addm.com 30 Jun '05

30 Jun '05

Hi all, I tried searching for a similar thread in the iptel archive, but can't find any related discussion, so please forgive me if some of you have discussed this issue previously for I did try searching for it!! :> I'm trying to have SER listen on other ports on top of its default (5060) without success. In my configuration, I tried.... listen=public-ip port=5080 port=5060 But users can only register on port 5060. Once I disable the line port=5060, users can now register on port 5080. I have even tried.... listen=public-ip:5060 listen=public-ip:5080 This resulted in SER failing to start. Therefore I had no choice but to move back to my 1st sample above in adding the port option. My question boils down to the following... Is SER capable of listening on multiple ports?? If so, how do I configure it?? Any help is appreciated! Best Regards, Samuel Yeung

1 0

[Serusers] [Seruser] GettingStarted04 - RTPProxy Setting Problem !
by Freeman 30 Jun '05

30 Jun '05

Hi, I tried to follow the manual "GettingStarted 04 - 2005-05-26" to create ser.cfg ( for RTPProxy ), i found some error in "/var/log/messages", rtpproxy: Starting SER RTP Proxy server: rtpproxy rtpproxy: . rc: Starting rtpproxy: succeeded ser: WARNING: fix_socket_list: could not rev. resolve 192.168.10.1 ser: WARNING: fix_socket_list: could not rev. resolve 192.168.10.1 ser: Listening on ser: udp: 192.168.10.1 [192.168.10.1]:5060 ser: tcp: 192.168.10.1 [192.168.10.1]:5060 ser: Aliases: ser: ser: ser startup succeeded /usr/local/sbin/ser[2353]: Maxfwd module- initializing /usr/local/sbin/ser[2353]: INFO: udp_init: SO_RCVBUF is initially 65535 /usr/local/sbin/ser[2353]: INFO: udp_init: SO_RCVBUF is finally 131070 /usr/local/sbin/ser[2373]: INFO: fifo process starting: 2373 /usr/local/sbin/ser[2373]: rtpp_test: RTP proxy found, support for it enabled /usr/local/sbin/ser[2373]: SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... /usr/local/sbin/ser[2375]: rtpp_test: RTP proxy found, support for it enabled /usr/local/sbin/ser[2380]: rtpp_test: RTP proxy found, support for it enabled /usr/local/sbin/ser[2381]: rtpp_test: RTP proxy found, support for it enabled /usr/local/sbin/ser[2382]: rtpp_test: RTP proxy found, support for it enabled /usr/local/sbin/ser[2377]: rtpp_test: RTP proxy found, support for it enabled /usr/local/sbin/ser[2383]: rtpp_test: RTP proxy found, support for it enabled /usr/local/sbin/ser[2389]: rtpp_test: RTP proxy found, support for it enabled /usr/local/sbin/ser[2353]: rtpp_test: RTP proxy found, support for it enabled /usr/local/sbin/ser[2388]: rtpp_test: RTP proxy found, support for it enabled /usr/local/sbin/ser[2387]: rtpp_test: RTP proxy found, support for it enabled /usr/local/sbin/ser[2379]: rtpp_test: RTP proxy found, support for it enabled I tried to test the ser in some case : ( using Windows Messenger 5.1 and released all port for ser ) 1. "User A ( outside with NAT ) call User B ( same network with ser )" - User A can't call User B. 2. "User B ( same network with ser ) call User A ( outside with NAT )" - User B can call User B but No Voice and auto hangup. 3. "User A ( outside with NAT ) call User C ( outside with NAT )" - User A can't call User C. I found some information in MySQL ( Database : ser , Table : location ) username contact received user_a sip:192.168.1.1:7700 sip:234.234.234.234:1568 ( public ip ) user_b sip:192.168.10.11:7446 sip:192.168.10.1.3112 ( private ip ) user_c sip:192.168.2.1:15647 sip:123.123.123.123:21144 ( public ip ) I tried many times, but the problem is same ! Please help ! tks a lot ! Freeman _______________________________________ 想即時收到新 email 通知？下載 Yahoo! Messenger http://messenger.yahoo.com.hk

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標題： RE: [Serusers] [Seruser] GettingStarted04 - RTPProxy Setting Problem !
by Freeman 30 Jun '05

30 Jun '05

I tried to change the config in ser.cfg , listen=192.168.10.1 ( private ip ) to listen=234.234.234.234 ( public ip ), but i get some error in "/var/log/messages" as below : rtpproxy: Starting SER RTP Proxy server: rtpproxy rtpproxy: . rc: Starting rtpproxy: succeeded xfs: xfs startup succeeded xfs: ignoring font path element /usr/X11R6/lib/X11/fonts/cyrillic (unreadable) ser: WARNING: fix_socket_list: could not rev. resolve 234.234.234.234 ser: WARNING: fix_socket_list: could not rev. resolve 234.234.234.234 ser: Listening on ser: udp: 234.234.234.234 [234.234.234.234]:5060 ser: tcp: 234.234.234.234 [234.234.234.234]:5060 ser: Aliases: ser: Jun 30 15:41:03 localhost ser: ser startup succeeded Jun 30 15:41:03 localhost /usr/local/sbin/ser[2356]: Maxfwd module- initializing Jun 30 15:41:05 localhost /usr/local/sbin/ser[2356]: INFO: udp_init: SO_RCVBUF is initially 65535 Jun 30 15:41:05 localhost /usr/local/sbin/ser[2356]: INFO: udp_init: SO_RCVBUF is finally 131070 Jun 30 15:41:05 localhost /usr/local/sbin/ser[2356]: ERROR: udp_init: bind(5, 0x80fd674, 16) on 234.234.234.234: Cannot assign requested address tks a lot ! Freeman harry gaillac <gaillacharry(a)yahoo.fr> 說： Does your ser+rtpproxy listen on public and private interfaces ? try ser.cfg : listen=my public ip start ./rtpproxy -l public ip harry --- Freeman a 嶰rit : > Hi, > > I tried to follow the manual "GettingStarted > 04 - 2005-05-26" to create ser.cfg ( for RTPProxy ), > i found some error in "/var/log/messages", > > rtpproxy: Starting SER RTP Proxy server: rtpproxy > rtpproxy: . > rc: Starting rtpproxy: succeeded > ser: WARNING: fix_socket_list: could not rev. > resolve 192.168.10.1 > ser: WARNING: fix_socket_list: could not rev. > resolve 192.168.10.1 > ser: Listening on > ser: udp: 192.168.10.1 > [192.168.10.1]:5060 > ser: tcp: 192.168.10.1 > [192.168.10.1]:5060 > ser: Aliases: > ser: > ser: ser startup succeeded > /usr/local/sbin/ser[2353]: Maxfwd module- > initializing > /usr/local/sbin/ser[2353]: INFO: udp_init: SO_RCVBUF > is initially 65535 > /usr/local/sbin/ser[2353]: INFO: udp_init: SO_RCVBUF > is finally 131070 > /usr/local/sbin/ser[2373]: INFO: fifo process > starting: 2373 > /usr/local/sbin/ser[2373]: rtpp_test: RTP proxy > found, support for it enabled > /usr/local/sbin/ser[2373]: SER: open_uac_fifo: fifo > server up at /tmp/ser_fifo... > /usr/local/sbin/ser[2375]: rtpp_test: RTP proxy > found, support for it enabled > /usr/local/sbin/ser[2380]: rtpp_test: RTP proxy > found, support for it enabled > /usr/local/sbin/ser[2381]: rtpp_test: RTP proxy > found, support for it enabled > /usr/local/sbin/ser[2382]: rtpp_test: RTP proxy > found, support for it enabled > /usr/local/sbin/ser[2377]: rtpp_test: RTP proxy > found, support for it enabled > /usr/local/sbin/ser[2383]: rtpp_test: RTP proxy > found, support for it enabled > /usr/local/sbin/ser[2389]: rtpp_test: RTP proxy > found, support for it enabled > /usr/local/sbin/ser[2353]: rtpp_test: RTP proxy > found, support for it enabled > /usr/local/sbin/ser[2388]: rtpp_test: RTP proxy > found, support for it enabled > /usr/local/sbin/ser[2387]: rtpp_test: RTP proxy > found, support for it enabled > /usr/local/sbin/ser[2379]: rtpp_test: RTP proxy > found, support for it enabled > > I tried to test the ser in some case : ( using > Windows Messenger 5.1 and released all port for ser > ) > > 1. "User A ( outside with NAT ) call User B ( same > network with ser )" > - User A can't call User B. > > 2. "User B ( same network with ser ) call User A ( > outside with NAT )" > - User B can call User B but No Voice and auto > hangup. > > 3. "User A ( outside with NAT ) call User C ( > outside with NAT )" > - User A can't call User C. > > I found some information in MySQL ( Database : ser , > Table : location ) > > username contact > received > user_a sip:192.168.1.1:7700 > sip:234.234.234.234:1568 ( public ip ) > user_b sip:192.168.10.11:7446 > sip:192.168.10.1.3112 ( private ip ) > user_c sip:192.168.2.1:15647 > sip:123.123.123.123:21144 ( public ip ) > > > I tried many times, but the problem is same ! > > > Please help ! > > > tks a lot ! > Freeman > > > _______________________________________ > 想即時收到新 email 通知？ > 下載 Yahoo! Messenger http://messenger.yahoo.com.hk > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T幨嶰hargez cette version sur http://fr.messenger.yahoo.com 相片電郵測試版一個郵件就可以傳送過百張相片！

1 0

[Serusers] Serweb - Received Calls
by Daryl Sanders 30 Jun '05

30 Jun '05

Does serweb accounting have the ability to show received calls? If so how do I implement this? (note: I can see outgoing calls fine in accounting tab) Thanks! - Daryl

2 1

[Serusers] SER and IPv6
by Nguyen, Thuan 30 Jun '05

30 Jun '05

Hi, does anyone know how to configure SER to work with IPv6 ? I couldn't find any instruction on that at all. Thank you very much, Thuan Nguyen

2 1

[Serusers] Question about permissions module.
by Ricardo Martinez 30 Jun '05

30 Jun '05

Hello. I'm trying to make an "allow_trusted IP's" call without consult a mysql database, instead i want to check against a plaintext file. Can i do this through the permissions module?. In the README it says : 1.1. Overview 1.1.1. Call Routing The module can be used to determine if a call has appropriate permission to be established. Permission rules are stored in plaintext configuration files similar to hosts.allow and hosts.deny files used by tcpd. Thanks in advance. Ricardo Martinez.-

2 1

[Serusers] update ser on Debian
by harry gaillac 29 Jun '05

29 Jun '05

Hello asterisk users, I you run Debian try to setup mini-dinstall for updating SER from a local archive . Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] Docs to install SERWEB please..
by Leo 29 Jun '05

29 Jun '05

Dear SER Users, What is the best document to use to help me install SERWEB? Where might I find these documents? I already have SER running with MySQL. Leo Papadopoulos E-mail: leo(a)telecomCTO.com Web site: www.telecomCTO.com <http://www.telecomcto.com/>

1 1

[Serusers] multiple contact entries automatically....
by Iqbal 29 Jun '05

29 Jun '05

Hi I have a really strange problem, when I am registering a IP phone, which is connected to a simple 4 port router, it i submitting multiple contacts into the db. Even in the sip messages I can see multiple entries, the IP:port pairs only differ in the port numbers. Anyone else had this problem, Iqbal

3 9

[Users] mpath - new config parameter
by Daniel-Constantin Mierla 29 Jun '05

29 Jun '05

Hello, there is a new global parameter that can be used in config file: 'mpath'. The parameter specify the path from where to look for modules. The value will be prefixed to the parameter of any loadmodule when this is not absolute. For example: loadmodule "/usr/loca/lib/openser/modules/textops.so" loadmodule "/usr/loca/lib/openser/modules/xlog.so" is the same now with: mpath="/usr/loca/lib/openser/modules" loadmodule "textops.so" loadmodule "xlog.so" or with: mpath="/usr/loca/lib/openser" loadmodule "modules/textops.so" loadmodule "modules/xlog.so" This will ease the migration of the config when dealing with many instances installed with different prefixes. Cheers, Daniel

2 1

[Serusers] CANCEL request
by Iqbal 29 Jun '05

29 Jun '05

Hi I have pstn -->ser -->UA I also have asterisk hanging off ser for voicemail Now this is all working fine, voicemail and all triggers great on no answer, BUT to be sure I decided to look atthe sip dialogue, just to see if all was fine, and so that I could start to clean up my config file. When a user calls from pstn, then hit the switch, and drop into ser, which then maps the pstn number to a local alias. Ip phone rings, INVITE (pstn) to (ser) 100 trying (ser) to (pstn) INVITE (ser) to (ua) 100 trying (ua) to (ser) 180 ringing (ua) to (ser) 180 ringing (ser) to (pstn) so far so good, now if there is no answer, and I forward to asterisk should there be a cancel to the original INVITE, cause this is what I am getting: CANCEL (ser) to (ua) 200 OK (ua) to (ser) 487 request cancelled (ua) to (ser) then i get ACK (ser) to (ua) ------where this ACK comes fro I am not sure 200OK (ser) to (pstn) but useragent is no asterisk, hence this makes sense ACK (pstn) to (ser) so what I am not clear on is should the CANCEL be there, or not, it seems to make sense that it is, just want to confirm. Also since alot of people have the same setup, would it be a good idea alongside onsip.org and its startup config, if we could post/have a sip trace of common call scenarios, I know some of these are in the rfc etc, but someone they dont seem user friendly... Iqbal

3 3

[Users] test return code in config file
by Daniel-Constantin Mierla 29 Jun '05

29 Jun '05

Hello, there is a new variable in configuration file 'retcode' that can be used to test against an integer the value returned by last invoked function or route. In the case of routes that uses return, it is the returned value. The changes were done yesterday, so the pserver CVS head is now synchronized. It is similar to $? from bash ($? can be used also in configuration file, having the same meaning as 'retcode'). Take care when you are testing this value, it can be easily misleading. For example: route[1] { if(method=="INVITE") return(1); return(2); } route { route(1); if(retcode==2) { log("The request is a REGISTER\n"); }; if(retcode==1) { ... }; } In case of REGISTER, the 'log' function is executed and the value of retcode is changed. The proper scripting is: route { route(1); if(retcode==2) { log("The request is a REGISTER\n"); } else if(retcode==1) { ... }; } I have updated the docuwiki for 'return()' and added 'retcode' there, too. http://www.openser.org/dokuwiki/doku.php?id=openser_core_cookbook#return_int http://www.openser.org/dokuwiki/doku.php?id=openser_core_cookbook#retcode Regards, Daniel

3 5

[Serusers] Digits manipulation
by Daniel Mizrachi (net-uno.net) 29 Jun '05

29 Jun '05

Hi... Actually I have SER 0.8.14 up and running on Linux Redhat 9.0 and I´m trying to do something about digits manipulation. I have registered on it diferent users from diferent regions of the country and I need to process local calls thru PSTN just with 7 digits even though all the users are registered with 10 digits (3 for Area Code + 7 for phone number) For example, User from Region #1, that has 241 as area code, wants to make a call thru PSTN to a user with the same area code but only dialing 7 digits. Here my question: Can SER take the first 3 digits (Area Code) from the ANI (Ser user) and add them to the destination number (DNIS) before send to PSTN Gw ? Example: Ser user: 2417790000(a)ser.net-uno.net Local number dialed by SER User: 9850000 Number to send to PSTN: 2419850000 I have no idea how I can do something like that Best Regards, Daniel Mizrachi

2 1

[Users] AVP subst_uri example
by Norman Brandinger 29 Jun '05

29 Jun '05

I have pulled this from my working OpenSER configuration file, so I can verify that the logic works. The following example takes advantage of the recent marriage of AVPs with the TEXTOPS subst_uri() function. The problem is to dynamically create a call to 1 (area code) 555-1212 for a user that dials 411 on his/her phone. The area code (also known as an NPA) is assigned to a user in the usr_preferences table. A simple change to the value in the table will generate a call to a different "local directory assistance operator". The ability to dynamically associate an NPA with a customer can be used to give the customer a "virtual local calling area". If only 7 digits of a phone number are entered by a user, OpenSER can easily prepend a 1 plus the users dynamic NPA. I remember the days when a 1 + area code were optional for calls to your local calling area. This example is a first step in that direction. An important off-shoot of this will be to perhaps assist in one aspect of the FCC's 911 decision. If the NPA (and perhaps the NXX) can be pulled from a location database and associated to an end user in the usr_preferences table, (did someone say LERG) then when a call to 911 is made, the generated phone number could be routed to a specific PSAP operator. If the end user changes his/her location, a web interface can be used to copy the new NPA/NXX to the usr_preferences table. Database Entries: Table: usr_preferences Username: user1 Attribute: npa Value: 123 OpenSER.cfg route[1] { #--------------------------------------------------------------------------------------- # 411 Local Directory Assistance #--------------------------------------------------------------------------------------- # check the called number to see if it is 411 if (uri=~"^sip:411@.*") { # check the AVP database for a match with the "user", not "user@domain" part of the From: header # checking for user@domain should also work. # if the "user" has a database entry with an "npa" attribute, load the AVP. if (avp_db_load("$from/username","s:npa")) { # rewrite the uri by replacing the 411 with 1 plus the AVP "npa" plus 5551212 subst_uri('/^sip:411@(.*)$/sip:1$avp(npa)5551212@\1/'); # delete the AVP npa to release resources # the release of AVP resources is supposed to happen automatically so the delete # may not be strictly required. avp_delete("s:npa"); } else { # the AVP database didn't find a match with the "user" so we rewrite # the uri with a default call to "local directory assistance". rewriteuri("sip:1xxx5551212@example.com"); }; break; }; } Regards, Norman Brandinger norm at goes dot com

1 0

[Serusers] INFO: terminating due to SIGCHLD when using t_replicate
by Conrado Camacho - T.net S.p.A. 29 Jun '05

29 Jun '05

Hi, we have been running from several months a single server with SER 0.8.14 with no problems. Now, in order to have some kind of redundancy, we are setting up a backup SER server and we are replicating the information from the primary server to the secondary with the t_replicate command if (method=="REGISTER") { if (!radius_www_authorize("tnet.it")) { www_challenge("tnet.it", "0"); break; }; if (!save("location")) { sl_reply_error(); } else { if (!src_ip==backup.foo.bar) { t_replicate("backup.foo.bar", "5060"); }; }; break; }; It seems to work fine but after a few minutes running the server goes down and we get the following message on /var/log/messages: Jun 29 19:19:10 dns2 ser[5648]: child process 5649 exited by a signal 11 Jun 29 19:19:10 dns2 ser[5648]: core was not generated Jun 29 19:19:10 dns2 ser[5648]: INFO: terminating due to SIGCHLD Jun 29 19:19:10 dns2 ser[5652]: INFO: signal 15 received Jun 29 19:19:10 dns2 ser[5650]: INFO: signal 15 received Does anybody have any idea about how to solve this issue? Thanks in advance. Regards, Conrado Camacho ccamacho(a)tnet.it ___________________ T. NET_________________ Viale Africa, 84 - 95129 Catania - Italy Sede di Padova: Via Savelli, 50 - 35139 Padova - Italy Tel: +39 095 37 83 111 - Fax: +39 095 37 83 444 P. I.V.A.: 03979950874 www.tnet.it ________T. NET Telecommunication Company________ *************************************************************** Le informazioni in questa e-mail sono confidenziali e riservate esclusivamente al destinatario del messaggio. Information in this email is confidential and intended solely for the addressee; it may be legally privileged. ***************************************************************

1 0

[Serusers] Serweb Received Calls
by Daryl Sanders 29 Jun '05

29 Jun '05

Does serweb accounting have the ability to show received calls? If so how do I implement this? (note: I can see outgoing calls fine in accounting tab) Thanks! - Daryl

1 0

[Users] Accounting Failed Transaction
by Bogdan-Andrei Iancu 29 Jun '05

29 Jun '05

Hi folks, New changes in ACC module - the global flag "failed_transactions" replaced with the "failed_transaction_flag" per transaction flag. The idea is to allow you to mark which transaction should be accounted in case of failure (response>=300). The global flag was not enough. Imagine the following case: O want to account only positive INVITES (established calls) - I really don't want to oversize my logs with failed INVITEs -, but in the same time I want to account all BYEs (succeeded or failed). regards, bogdan PS: CVS head, pserver CVS and docs (cvs and web) already updated

1 0

[Serusers] RE: PSTN Configuration
by Abdul Lateef 29 Jun '05

29 Jun '05

Can any one explain me what does means # INSERT YOUR PSTN GATEWAY IP ADDRESS in the following routes. If i am right, is the IP for carier? route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("192.0.2.245"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); route(5); route(1); } Thank YOu Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

3 2

Re: [Serusers] Serweb start problem
by Karel Kozlik 29 Jun '05

29 Jun '05

Hi Wilson, please (CC) the list in your replies. Which is the domain you are useing for ser? For admin user? The 'ipaddress' appear strange for me. Check your database. Table subscriber and column domain. regards Karel li wei napsal(a): > Hi Karel: > > Glad to get your warm help. I have solved my problem > now. It is caused by the serweb didn't have the > privilege to write files into the /tmp/serweb/ folder. > > But I face a new problem now. My ser database use the > default username and password (ser:heslo). When I > enter the default username and password for serweb > admin (admin@ipaddress,heslo). It returns me the > following error: > > DB Error: connect failed > Bad username or password > > It's pretty strange. > > Can you do me a favor again? > > Thanks a lot. > > Wilson > > > > > > --- Karel Kozlik <karel(a)iptel.org> wrote: > > >>Hi, >>and what about user interface? Work it correctly? >> >>What about php log? Are there any messages? Have you >>enabled error >>reporting in php? I recommend to set these in your >>php.ini: >> >>error_reporting = E_ALL >>display_errors = On >> >> >>Karel >> >> >> >>li wei napsal(a): >> >>>Hi there: >>> >>>Have install the ser and serweb 0.93 official >> >>version. >> >>>When I try to open the admin page, It just returns >> >>me >> >>>the following message. >>> >>>IPaddress admin interface >>> >>>And I try to see the return html source code, it >> >>stops >> >>>at tag <div>. The detail as the follow: >>> >>><!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 >>>Strict//EN" >>> >> >>"http://www.w3.org/TR/xhtml1/DTD/xhtml1-strict.dtd"> >> >>><html> >>><head> >>><title>SIP Express Router - web interface</title> >>> <meta http-equiv="Content-Type" >> >>content="text/html; >> >>>charset=utf-8"> >>> <meta name="Author" content="Karel Kozlik <karel >> >>at >> >>>iptel dot org>"> >>> <meta http-equiv="PRAGMA" content="no-cache"> >>> <meta http-equiv="Cache-control" >> >>content="no-cache"> >> >>> <meta http-equiv="Expires" content="Wed, 29 Jun >> >>2005 >> >>>04:18:07 GMT"> >>> >>> <LINK REL="StyleSheet" >>>HREF="/serweb/html/domains/_default/styles.css" >>>TYPE="text/css"> >>> >>></head> >>> >>><script language="JavaScript" >>> >> > src="/serweb/html/js/login_completion.js.php"></script><body><h1> > >>>IPaddress admin interface</h1><hr> >>><div class="swMain"> >>> >>>Can anybody help me? >>> >>>Thanks a lot. >>> >>>Wilson >>> >>> >>> >>> >>> >>> >>> >> >>____________________________________________________ >> >> >>>Do you Yahoo!? >>>Try Yahoo! Photomail Beta: Send up to 300 photos >> >>in one email! >> >>>http://au.photomail.mail.yahoo.com >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >> > > > > > > > > ____________________________________________________ > Do you Yahoo!? > Try Yahoo! Photomail Beta: Send up to 300 photos in one email! > http://au.photomail.mail.yahoo.com

2 1

[Serusers] *** OFF LINE
by lucape＠inwind.it 29 Jun '05

29 Jun '05

I've configured ser like it's explained in the getting started document...ser.cfg with sql authentication... 1) i'm not able to insert new ser user, help me to do it? 2) i connect a user, that is present in ser data base, with messenger... if i look in the ser administration the user is not online...why? thanks ____________________________________________________________ 6X velocizzare la tua navigazione a 56k? 6X Web Accelerator di Libero! Scaricalo su INTERNET GRATIS 6X http://www.libero.it

1 0

[Serusers] RE: PSTN Configuration
by Abdul Lateef 29 Jun '05

29 Jun '05

hello friends, I did the following commonds, but no luck. [root@localhost root]# /usr/local/sbin/serctl stop Stopping SER : No PID file found! SER probably not running [root@localhost root]# ps -aux|grep ser rpcuser 849 0.0 0.1 1556 716 ? S 09:42 0:00 rpc.statd root 2275 0.0 1.3 19440 7184 ? S 15:31 0:00 kdeinit: dcopserv root 2296 0.0 0.0 1416 316 ? S 15:31 0:00 kwrapper ksmserve root 2298 0.0 1.7 20424 9188 ? S 15:31 0:00 kdeinit: ksmserve [root@localhost root]# /usr/local/sbin/serctl start Starting SER : started pid(3381) [root@localhost root]# ps -aux|grep ser rpcuser 849 0.0 0.1 1556 716 ? S 09:42 0:00 rpc.statd root 2275 0.0 1.3 19440 7184 ? S 15:31 0:00 kdeinit: dcopserv root 2296 0.0 0.0 1416 316 ? S 15:31 0:00 kwrapper ksmserve root 2298 0.0 1.7 20424 9188 ? S 15:31 0:00 kdeinit: ksmserve [root@localhost root]# Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Serusers] RE: PSTN Configuration
by Abdul Lateef 29 Jun '05

29 Jun '05

Hello dear Iqbal, Thanks for your great reply. I am really sorry to posting to much questions in this thread. I have one another issue, i did test the IP Phone with SER on the same local network and it is working fine. But when i changed the SER server with public IP address while chainging listen=212.77.213.69 port=5060 and still the IP Phone is in local network, the IP phone is not able to register using SER publick IP. here is a small log i capture it from message file. Jun 29 15:44:45 localhost sshd(pam_unix)[3040]: session opened for user root by (uid=0) Jun 29 15:44:50 localhost ser: WARNING: fix_socket_list: could not rev. resolve 212.77.213.69 Jun 29 15:44:50 localhost ser: WARNING: fix_socket_list: could not rev. resolve 212.77.213.69 Jun 29 15:44:50 localhost ser[3087]: INFO: udp_init: SO_RCVBUF is initially 65535 Jun 29 15:44:50 localhost ser[3087]: INFO: udp_init: SO_RCVBUF is finally 262142 Jun 29 15:44:50 localhost ser[3087]: ERROR: tcp_init: bind(6, 0x8114d24, 16) on 212.77.213.69: Address already in use Jun 29 15:44:50 localhost ser[3087]: INFO: signal 15 received Jun 29 15:44:59 localhost sshd(pam_unix)[3040]: session closed for user root Let me know what more configurations need? Thank You Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

1 0

Re: [Serusers] serweb and missed calls
by Klaus Darilion 29 Jun '05

29 Jun '05

Hi Sebastian! Please Cc: to the list. What is the error message? Does it work with the default log_fmt parameter? Posting your ser.cfg might help as well. klaus Sebastian Kühner wrote: > I have the error: I had to put following line > > modparam("acc", "log_fmt", "cftmsup") > > the "p" at the end didn't exist before... > > ----- Original Message ----- > From: "Klaus Darilion" <klaus.mailinglists(a)pernau.at> > To: "Sebastian Kühner" <skuehner(a)veraza.com> > Cc: <serusers(a)lists.iptel.org> > Sent: Tuesday, June 28, 2005 5:05 PM > Subject: Re: [Serusers] serweb and missed calls > > > >>Do you use log_missed_flag ? >> >>http://www.iptel.org/ser/doc/seruser/seruser.html#MISSEDCALLS >> >>regards, >>klaus >> >>Sebastian Kühner wrote: >> >>>Hello, >>> >>>Finally I have my serweb running (it had problems with apache2 and > > php5... > >>>many problems!). >>> >>>Now I have the problem that serweb doesn't show me the missed calls. I >>>looked at the mysql-database and noticed that ser doesn't save the > > domain > >>>and username - information in the table "missed_calls"... >>> >>>How can I make ser to save theese informations, too? >>> >>>Thanks! >>> >>>Sebastian >>> >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >> >> > > >

2 1

[Serusers] Cisco 3600 Config
by smy＠addm.com 29 Jun '05

29 Jun '05

To all the Cisco gurus, I'm having troubles with configuration a Cisco 3600 to work with SER. If my network (SER+Cisco) is running under port 5060, I can land calls as well as have calls passed from PSTN to SIP. (Not perfectly, some refining is needed) But my headaches starts when I switch the default 5060 port to some other port (let's say 5080) due to some network restrictions, thus I must change from port 5060 to something else. The problem I'm having is the following.... In Cisco 3600, I've setup session target to server-ip+5080.... but when SER receives a call coming from Cisco, SIP messages (such as trying) will be sent back to cisco via port 5060. I'm guessing this problem is caused by transport listen port is default to 5060. If so, how do I change this to other ports??? Better yet, does anyone know how I can configure a cisco to run on different ports under SIP??? I may be wrong with the transport and such, thus any inputs or suggestions is welcome (and desperately needed)!! :> Here?s a capture from my ngrep?. U cisco-ip:54215 -> SER-ip:5080 INVITE sip:6119203@SER-ip;user=phone;phone-context=local SIP/2.0..Via: SIP/2.0/UDP cisco-ip:5060..From: "6113640" <sip:6113640@cisco-ip>..To: <sip:6119203@SER-ip;user=phone;phone-context=local>....... U SER-ip:5080 -> cisco-ip:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP cisco-ip:5060..From: "6113640" <sip:6113640@cisco-ip>..To: <sip:6119203@SER-ip;user=phone;phone-context=local>........ U SER-ip:5080 -> cisco-ip:5080 INVITE sip:6119203@cisco-ip:5080;user=phone;phone-context=local SIP/2.0..Record-Route: <sip:6119203@SER-ip:5080;nat=yes;lr=on>..Via: SIP/2.0/UDP SER-ip:5080;branch=z9hG4bKea02.173ff176.0..Via: SIP/2.0/UDP cisco-ip:5060..From: "6113640" <sip:6113640@cisco-ip>..To: <sip:6119203@SER-ip;user=phone;phone-context=local>.............. Best Regards, Samuel Yeung

1 0

[Users] Call-Leg Accounting support
by Bogdan-Andrei Iancu 29 Jun '05

29 Jun '05

Hi folks, I just committed on CVS head support for call-leg accounting. The documentation was already updated both on CVS and site (http://www.openser.org/docs/modules/0.10.x/acc.html). this feature is quite useful when a call gets to have several legs ( for example, due call forwarding services). Having detailed accounting of the call leg will allow easy and proper CDR generation and billing. the feature is available for syslog, DB and RADIUS accounting (DIAMETER still missing). Any suggestions/comments regarding the way it works or it's implemented are welcomed. Also problem reports :) regards, Bogdan

2 4

[Users] AVP and Radius question
by Klaus Darilion 29 Jun '05

29 Jun '05

Hi! I'm quite new to avps - in fact I have not used them yet. Thus, please apologize if my questions are dumb. In my setup, (open)ser receives in the radius response two attributes: - RPID - prefix (for a call by call provider) In ser I have to add the RPID header using the RPID attribute, and prefix the number in the request URI with the prefix attribute. As far as I understand the auth_radius module in 0.9.4 supports receiving various radius attributes and storing them as AVPs. First, in ser.cfg I will add the RPID header. Is the function still append_rpid_hf() or do I have to use another function for generating the RPID header out of AVPs? Afterwards I have to prefix the request URI. Can this be done with avpops functions? I only found avp_pushto, which will replace the request URI - right? Can I store any arbitrary Radius attribute in AVPs or only predefined ones? regards, klaus

3 6

[Serusers] SER returns 400 Bad Request to SUBSCRIBE
by Victor Huertas Garcia 29 Jun '05

29 Jun '05

Hi all, I have configured the SER 0.8.14 to work with pa.so module and with the following lines in the ser.cfg: if (method==SUBSCRIBE){ if (t_newtran()){ handle_subscription("registrar"); break; }; }; then, when I register with Windows Messenger 4.7 and do "Add a Contact", it sends a SUBSCRIBE messages which is replied with a 400 Bad Request message. In the log file I see all the time: parse_hfs(): Error while parsing headers handle_subscription(): Error while parsing message header I don't understand why it doesn't work. Any idea? Thanks Victor PD: I have seen in other ser.cfg that pa.so module supports additional parameters (like db_url, etc.) but I cannot add these parameters at all. Why? Is there another pa.so module. If it is the case, would it work with SEr v0.8.14?

1 0

[Serusers] 30 sec. Timeout?
by Gerd Feiner 29 Jun '05

29 Jun '05

Hi, I got the following problem: When calling a (registered) user from PSTN (AS5350) to SIP (SER) I get a "Wrong CLI" error when the user does not answer within a few (say, 30) seconds. Is this an intended behaviour or do I get something wrong here? I followed the onsip.org guide, so my ser.conf should be known. If anyone needs anything in special - please let me know. Brgds, Gerd

1 0

Re: [Users] Problem starting Openser
by Andréa Giordanna 29 Jun '05

29 Jun '05

-- Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> escreveu: > regarding, the error generated by 'openserctl start' > - it mighr be > because it failes to find the config file (look into > the log for errors) > - if so, you should try to rename the config file ( > if from packages > /etc/openser/openser.cfg in /etc/openser/ser.cfg; I already have renamed this file. But the error is generated again. =/ Here is my ser.cfg: # # $Id: openser.cfg,v 1.1.1.1 2005/06/13 16:47:30 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=5 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" #loadmodule "/home/giordanna/Voip/Programas/openser-0.9.4/bin/lib/openser/modules/dbtext.so" loadmodule "/home/giordanna/VoIP/Programas/openser-0.9.4/bin/lib/openser/modules/sl.so" loadmodule "/home/giordanna/VoIP/Programas/openser-0.9.4/bin/lib/openser/modules/tm.so" loadmodule "/home/giordanna/VoIP/Programas/openser-0.9.4/bin/lib/openser/modules/rr.so" loadmodule "/home/giordanna/VoIP/Programas/openser-0.9.4/bin/lib/openser/modules/maxfwd.so" loadmodule "/home/giordanna/VoIP/Programas/openser-0.9.4/bin/lib/openser/modules/usrloc.so" loadmodule "/home/giordanna/VoIP/Programas/openser-0.9.4/bin/lib/openser/modules/registrar.so" loadmodule "/home/giordanna/VoIP/Programas/openser-0.9.4/bin/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/home/giordanna/VoIP/Programas/openser-0.9.4/bin/lib/openser/modules/auth.so" #loadmodule "/home/giordanna/VoIP/Programas/openser-0.9.4/bin/lib/openser/modules/auth_db.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_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } It seems everything is alright. Any idea? Thanks, Andrea Giordanna _______________________________________________________ Yahoo! Acesso Grátis - Internet rápida e grátis. Instale o discador agora! http://br.acesso.yahoo.com/

4 6

[Users] serweb and openser
by Klaus Darilion 29 Jun '05

29 Jun '05

Hi! Are there any experiences with openser and serweb? Is the FIFO compatible to ser? AFAIK the db structure has changed in openser, thus this need to be changed in serweb too. regards, klaus

2 1

[Serusers] Applying Patch to uac module
by Alberto Cruz 29 Jun '05

29 Jun '05

Sorry if somebody has made the same question. Could somebody send me the procedure to apply the "replace_from.patch" to the uac module in order I can compile it? I'm using ser 0.9.3 Regards Alberto Cruz

2 4

[Serusers] Suitable Distro for SER ?
by Ozan Blotter 29 Jun '05

29 Jun '05

Dear List, My Linux Box is so sticky so i have to renew it, before installing a new one i want to ask all of you a general question. I will do a scratch and full installation, without touching anything i want to install SER with MySql support for both Auth and Acc. without any problems in compiling which distro you offer me to install ? I have below ones: Mandrake 10.0 SuSe 9.3 Fedora Core 3 (Now i'm using this but so many problems) Debian Sarge Thanks, Ozan

2 1

[Serusers] Serweb start problem
by li wei 29 Jun '05

29 Jun '05

Hi there: Have install the ser and serweb 0.93 official version. When I try to open the admin page, It just returns me the following message. IPaddress admin interface And I try to see the return html source code, it stops at tag <div>. The detail as the follow: <!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Strict//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-strict.dtd"> <html> <head> <title>SIP Express Router - web interface</title> <meta http-equiv="Content-Type" content="text/html; charset=utf-8"> <meta name="Author" content="Karel Kozlik <karel at iptel dot org>"> <meta http-equiv="PRAGMA" content="no-cache"> <meta http-equiv="Cache-control" content="no-cache"> <meta http-equiv="Expires" content="Wed, 29 Jun 2005 04:18:07 GMT"> <LINK REL="StyleSheet" HREF="/serweb/html/domains/_default/styles.css" TYPE="text/css"> </head> <script language="JavaScript" src="/serweb/html/js/login_completion.js.php"></script><body><h1> IPaddress admin interface</h1><hr> <div class="swMain"> Can anybody help me? Thanks a lot. Wilson ____________________________________________________ Do you Yahoo!? Try Yahoo! Photomail Beta: Send up to 300 photos in one email! http://au.photomail.mail.yahoo.com

2 1

[Users] DB_Text module error
by Alexandre Passito 29 Jun '05

29 Jun '05

Hi all, I'm trying to use DB_text module with AVPOPS module and I got the following error. I think that it could be the format of the database (col and type). I tried to implement it according to AVPOPS manual, but I think it could not be complete. 24(10259) DBT:dbt_load_file: request for table [preferences] 24(10259) DBT:dbt_load_file: db is [/home/alexandre/Devel/openser-0.9.4/bin/dbtext] 24(10259) DBT:dbt_load_file: loading file [/home/alexandre/Devel/openser-0.9.4/bin/dbtext/preferences] 24(10259) DBT: column[0] is STR! 24(10259) DBT: column[1] is STR! 24(10259) DBT: column[2] is STR! 24(10259) DBT: column[3] is STR! 24(10259) DBT: column[4] is INT! 24(10259) DBT: column[5] is STR! 24(10259) DBT: column[6] is INT! 24(10259) DBT:dbt_table_check_row: NULL value not allowed - field 0 24(10259) DBT:dbt_load_file: error at row=1 col=8 c=ÿ 24(10259) DTB:dbt_db_get_table: preferences 24(10259) DBT:dbt_query: table does not exist! 24(10259) ERROR:avpops:load_avps: db_load failed 24(10259) DEBUG: t_newtran: msg id=25 , global msg id=4 , T on entrance=0xffffffff 24(10259) parse_headers: flags=-1 24(10259) parse_headers: flags=60 24(10259) t_lookup_request: start searching: hash=30008, isACK=0 24(10259) DEBUG: RFC3261 transaction matching failed 24(10259) DEBUG: t_lookup_request: no transaction found My table "preferences" looks like: uuid(str) username(str) domain(str) attribute(str) type(int) value(str) modified(int) :andrea:localhost:1:2:alexandre@localhost:111 Any help will be helpful :-) Best regards -- Alexandre Passito Universidade Federal do Amazonas Departamento de Ciência da Computação LabVoIP - GRSD

2 1

[Serusers] Help with caller_UUID and callee_UUID
by Daryl Sanders 29 Jun '05

29 Jun '05

Greetings, I've setup serweb from cvs with mysql acc. I'm now wondering what needs to be added to ser.cfg to get acc to populate the caller_UUID and callee_UUID fields. As I understand it these fields are necessary to show accounting for incoming calls in serweb accounting tab. I've set users_indexed_by= "uuid" in config_data_layer.php and I've also set set_opt('display_incoming', true) in accounting.php I have the following in my ser.cfg. if (!method=="REGISTER") record_route(); setflag(2); If I populate the caller_UUID and callee_UUID fields by hand, serweb accounting tab shows all the calls as I would like. My goal is to have accounting for incoming/outgoing & missed calls in serweb. Am I on the right path? - Daryl

3 7

[Serusers] Cisco3600 to SER
by smy＠addm.com 29 Jun '05

29 Jun '05

Hi all, I'm currently working with a cisco3600 with SER 0.9.2. Thanks to Onsp.org, I can land calls from SER to Cisco 3600 without any problems. Yet I'm experiencing some problems with Cisco3600 passing calls onto SER. My setup is as follow: Cisco 3600 ==> SER+Mediaproxy ===> IP Phone When a call come in via DID method through the Cisco3600, SER received the all SIP messages (I got it via ngrep), yet the call couldn't pass onto the IP Phone. Currently, all devices (including cisco) have been configured to use port 5080 instead of port 5060 due to port restrictions on the network, and that I have specified the ip in the trusted table under mysql. Yet I don't know why the call still can not get through. The following are the logs I received from ngrep and cisco..... Any help is welcome and is GREATLY APPRCIATED!! ------------------- ngrep from SER server U cisco-ip:55299 -> SER-ip:5080 INVITE sip:6119203@SER-ip;user=phone;phone-context=local SIP/2.0..Via: SIP/2.0/UDP cisco-ip:5060..From: "6113640" <sip:6113640@cisco-ip>..To: <sip:6119203@SER-ip;user=phone;phone-context=local>.............. # U SER-ip:5080 -> cisco-ip:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 222.50.103.247:5060..From: "6113640" <sip:6113640@cisco-ip>..To: <sip:6119203@SER-ip;user=phone;phone-context=local>........................... # U cisco-ip:55299 -> SER-ip:5080 CANCEL sip:6119203@SER-ip;user=phone;phone-context=local SIP/2.0..Via: SIP/2.0/UDP 222.50.103.247:5060..From: "6113640"<sip:6113640@cisco-ip>..To: <sip:6119203@SER-ip;user=phone;phone-context=local>................. ---------------- mysql trusted table +----------------+-------+--------------+ | src_ip | proto | from_pattern | +----------------+-------+--------------+ | cisco-ip | any | ^sip:.*$ | +----------------+-------+--------------+ ----------------- log from cisco3600 Jun 23 14:23:53: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: SER-ip:5080 Jun 23 14:23:57: Received: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP cisco-ip:5060 From: "6113640" <sip:6113640@cisco3600> To: <sip:6119203@SER-ip;user=phone;phone-context=local>;tag=bc78c497f1336138e83d95d77d963f1b-f81d Call-ID: 22B8BA3E-E32911D9-82A1EC22-7AD29AB0(a)222.50.103.247 CSeq

3 8

[Serusers] avpops and ip based auth
by Iqbal 29 Jun '05

29 Jun '05

Hi If I use avpops for IP based auth, and drop the normal username/password combo aside from spoofing what is the downside if any. Also if I do IP based auth, can I auth once, and be done with it, or is it auth once per call, I guess its once per call, if so is there any way to bypass auth completely for a particular IP address, again I am assuming no, since the IP will still need to be checked for each request. Iqbal

2 9

[Serusers] Serweb start problem
by wilson 29 Jun '05

29 Jun '05

Hi there: Have install the ser and serweb 0.93 official version. When I try to open the admin page, It just returns me the following message. IPaddress admin interface And I try to see the return html source code, it stops at tag <div>. The detail as the follow: <!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Strict//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-strict.dtd"> <html> <head> <title>SIP Express Router - web interface</title> <meta http-equiv="Content-Type" content="text/html;charset=utf-8"> <meta name="Author" content="Karel Kozlik <karel at iptel dot org>"> <meta http-equiv="PRAGMA" content="no-cache"> <meta http-equiv="Cache-control" content="no-cache"> <meta http-equiv="Expires" content="Wed, 29 Jun 2005 04:18:07 GMT"> <LINK REL="StyleSheet" HREF="/serweb/html/domains/_default/styles.css" TYPE="text/css"> </head> <script language="JavaScript" src="/serweb/html/js/login_completion.js.php"></script><body><h1> IPaddress admin interface</h1><hr> <div class="swMain"> Can anybody help me? Thanks a lot. Kind Regards, wilson 2005-06-29

1 0

[Serusers] Serweb start problem
by wilson 29 Jun '05

29 Jun '05

Hi there: Have install the ser and serweb 0.93 official version. When I try to open the admin page, It just returns me the following message. IPaddress admin interface And I try to see the return html source code, it stops at tag <div>. The detail as the follow: <!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Strict//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-strict.dtd"> <html> <head> <title>SIP Express Router - web interface</title> <meta http-equiv="Content-Type" content="text/html;charset=utf-8"> <meta name="Author" content="Karel Kozlik <karel at iptel dot org>"> <meta http-equiv="PRAGMA" content="no-cache"> <meta http-equiv="Cache-control" content="no-cache"> <meta http-equiv="Expires" content="Wed, 29 Jun 2005 04:18:07 GMT"> <LINK REL="StyleSheet" HREF="/serweb/html/domains/_default/styles.css" TYPE="text/css"> </head> <script language="JavaScript" src="/serweb/html/js/login_completion.js.php"></script><body><h1> IPaddress admin interface</h1><hr> <div class="swMain"> Can anybody help me? Thanks a lot. Kind Regards, wilson 2005-06-29

1 0

[Serusers] ERROR: module version mismatch for /usr/local/lib/ser/modules/acc.so
by Alberto Cruz 29 Jun '05

29 Jun '05

I'm updating my ser 0.9.2 to ser 0.9.3 Whe I run the ser command I'm getting the following error for the module acc: 0(32050) ERROR: module version mismatch for /usr/local/lib/ser/modules/acc.so; core: ser 0.9.3 (i386/linux); module: ser 0.9.2 (i386/linux) I compile ser in the following way: make clean make proper make all make install I already checked the creation date for the acc.so file and it has the same like the other modules. Does anybody know why this is happening? Regards Alberto Cruz

2 2

SV: [Serusers] serweb and missed calls
by mailliste＠renenielsen.net 28 Jun '05

28 Jun '05

> -----Org. message----- > Fra: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] På > vegne af Sebastian Kühner Finally I have my serweb running (it had > problems with apache2 and php5... many problems!). If i may ask, what problems were these...? I myself am having problem with array in my_account.php - you to? > Now I have the problem that serweb doesn't show me the missed > calls. I looked at the mysql-database and noticed that ser > doesn't save the domain and username - information in the > table "missed_calls"... For this problem, i also would like to know a solution.. René SYSTEM: BSD5.3, http 2.0.54, php 5.0.3, sql 4.0.24

1 0

[Serusers] radiusclient-ng
by Jose Bertuzzi 28 Jun '05

28 Jun '05

Hello everyone, I am testing ser 0.10.99 dev8, radiusclient-ng 0.5.1 with freeradius 1.0.2 on the same machine. It is not working. I am not sure whether it is a version or a configuration problem. The question is, do this versions work all together? Is there any radiusclient-ser compatibility matrix? I compile acc module with this options in Makefile: # uncomment the next line if you wish to enable SQL accounting #DEFS+=-DSQL_ACC # uncomment the next two lines if you wish to enable RADIUS accounting DEFS+=-DRAD_ACC -I$(LOCALBASE)/include LIBS=-L$(LOCALBASE)/lib -lradiusclient-ng # uncomment the next two lines if you wish to enable DIAMETER accounting #DEFS+=-DDIAM_ACC I don't have in my system any lradiusclient-ng, what I have is libradiusclient-ng.so. Is that OK?? Any clue is apreciated, Pablo. __________________________________ Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses. http://promotions.yahoo.com/new_mail

5 6

[Serusers] PSTN Configuration
by Abdul Lateef 28 Jun '05

28 Jun '05

Hello, I read lot of example ducumentations from net about the ser configuration. I found one example from voip-info.org. i configured the ser.cfg file accordidng to the guide. but when i am trying to restart the ser using /usr/local/sbin/serctl restart i am getting some pid file error messate like : ---------------------------------- [root@localhost root]# /usr/local/sbin/serctl stop Stopping SER : No PID file found! SER probably not running [root@localhost root]# /usr/local/sbin/serctl start Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed [root@localhost root]# -------------------------------- Here is my ser.cfg file configurations. I wanted to explain how i am going to use it. I have one VoIP Gatekeeper (MVTS) Which supports to register SER i want to register the SER in MVTS and the IP Phones i want to register in SER. So the call can be pass to destination using like: IP Phone > SER > MVTS > Destination PSTN. MVTS IP : 195.22.146.20 SER : 10.0.0.20 IP Phone : Should use the ser IP and uid/pws --------------- ser.cfg ---------------------- # # $Id: pstn.cfg,v 1.2 2003/06/03 03:18:12 jiri Exp $ # # # ------------------ module loading ---------------------------------- loadmodule "modules/sl/sl.so" loadmodule "modules/tm/tm.so" loadmodule "modules/acc/acc.so" loadmodule "modules/rr/rr.so" loadmodule "modules/maxfwd/maxfwd.so" loadmodule "modules/mysql/mysql.so" loadmodule "modules/auth/auth.so" loadmodule "modules/auth_db/auth_db.so" loadmodule "modules/group/group.so" loadmodule "modules/uri/uri.so" # ----------------- setting module-specific parameters --------------- modparam("auth_db", "db_url","sql://ser:heslo@10.0.0.26/ser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # acc params -- modparam("acc", "log_level", 1) # that is the flag for which we will account don't forget to # set the same one :-) modparam("acc", "log_flag", 1 ) # ------------------------- request routing logic ------------------- # main routing logic route{ /* ********* ROUTINE CHECKS ********************************** */ # filter too old messages if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Wow Message too large"); break; }; /* ********* RR ********************************** */ /* grant Route routing if route headers present */ if (loose_route()) { t_relay(); break; }; /* record-route INVITEs all subsequent requests must visit us */ if (method=="INVITE") { record_route(); }; # now check if it really is a PSTN destination which should be handled # by our gateway; if not, and the request is an invitation, drop it -- # we cannot terminate it in PSTN; relay non-INVITE requests it may # be for example BYEs sent by gateway to call originator if (!uri=~"sip:\+?[0-9]+@.*") { if (method=="INVITE") { sl_send_reply("403", "Call cannot be served here"); } else { forward(uri:host, uri:port); }; break; }; # account completed transactions via syslog setflag(1); # free call destinations ... no authentication needed if ( is_user_in("Request-URI", "free-pstn") /* free destinations */ | uri=~"sip:[79][0-9][0-9][0-9]@.*" /* local PBX */ | uri=~"sip:98[0-9][0-9][0-9][0-9]") { log("free call"); } else if (src_ip==192.168.0.10) { # our gateway doesn't support digest authentication; # verify that a request is coming from it by source # address log("gateway-originated request"); } else { # in all other cases, we need to check the request against # access control lists; first of all, verify request # originator's identity if (!proxy_authorize( "gateway" /* realm */, "subscriber" /* table name */)) { proxy_challenge( "gateway" /* realm */, "0" /* no qop */ ); break; }; # authorize only for INVITEs RR/Contact may result in weird # things showing up in d-uri that would break our logic; our # major concern is INVITE which causes PSTN costs if (method=="INVITE") { # does the authenticated user have a permission for local # calls (destinations beginning with a single zero)? # (i.e., is he in the "local" group?) if (uri=~"sip:0[1-9][0-9]+@.*") { if (!is_user_in("credentials", "local")) { sl_send_reply("403", "No permission for local calls"); break; }; # the same for long-distance (destinations begin with two zeros") } else if (uri=~"sip:00[1-9][0-9]+@.*") { if (!is_user_in("credentials", "ld")) { sl_send_reply("403", " no permission for LD "); break; }; # the same for international calls (three zeros) } else if (uri=~"sip:000[1-9][0-9]+@.*") { if (!is_user_in("credentials", "int")) { sl_send_reply("403", "International permissions needed"); break; }; # everything else (e.g., interplanetary calls) is denied } else { sl_send_reply("403", "Forbidden"); break; }; }; # INVITE to authorized PSTN }; # authorized PSTN # if you have passed through all the checks, let your call go to GW! rewritehostport("192.168.0.10:5060"); # forward the request now if (!t_relay()) { sl_reply_error(); break; }; } Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

2 1

[Serusers] Ser + radius compile error.
by Rosa De Santis 28 Jun '05

28 Jun '05

Hello all. I'm trying to install SER with Radius support for accounting, but it's not compiling right. I installed the freeradius server 0.9.1, radius client 0.3.2. and SER 0.9.0. The freeradius server and radius client appear to be right, I made the Radius server test just like the ser_radius.html manual, and the output is right. But...when I try to complie the ser I get some errors with the cpl, cpl_parser, pidf, db_con and acc.o modules... Please, can anybody help me? Thanks a lot Rosa

1 0

[Serusers] serweb and missed calls
by Sebastian Kühner 28 Jun '05

28 Jun '05

Hello, Finally I have my serweb running (it had problems with apache2 and php5... many problems!). Now I have the problem that serweb doesn't show me the missed calls. I looked at the mysql-database and noticed that ser doesn't save the domain and username - information in the table "missed_calls"... How can I make ser to save theese informations, too? Thanks! Sebastian

2 1

[Serusers] SER not starting.
by Abdul Lateef 28 Jun '05

28 Jun '05

Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________ Discover Yahoo! Find restaurants, movies, travel and more fun for the weekend. Check it out! http://discover.yahoo.com/weekend.html

1 0

[Serusers] Ser - Voice Problem
by Bruno Oliveira 28 Jun '05

28 Jun '05

oi I have being 0,9,0 and mysql being executed in a my serving when I make call enters two customers kphone the system of messages functions the voice more nao functions. you without none firewall you testing the being in the internal net. they can help me? -- []s, Bruno "Niggas" Oliveira Belo Horizonte - MG Msn: n1gg4s(a)gmail.com Icq: 176314647 "Todo o nosso descontentamento por aquilo que nos falta procede da nossa falta de gratidão por aquilo que temos."

1 0

[Serusers] clustering / failover
by Ingo Wolfsberger 28 Jun '05

28 Jun '05

Hi Has iptel or voice-sistem a cluster and failover solution? active-active systems? bye, Ingo -- Geschenkt: 3 Monate GMX ProMail gratis + 3 Ausgaben stern gratis ++ Jetzt anmelden & testen ++ http://www.gmx.net/de/go/promail ++

3 3

Fw: [Serusers] SER Windows messenger on-line problem
by Kissandrakis S. George 28 Jun '05

28 Jun '05

Hello again, I have used x-lite (www.xten.com) sip UA instead of windows messenger 5.1 and between the sip UAs i get the same message as below (while both are online) So i suppose tha its SER misconfiguration or bug and NOT UA's problem Any hints? 0(20323) build_res_from_sip_res: copied size: orig:88, new: 27, rest: 635 msg= SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP 147.52.3.244;branch=z9hG4bKebc6.ea874434.0 Via: SIP/2.0/UDP 147.52.250.40:5060;rport=5060;branch=z9hG4bK0C60B2DF5CBC44C1B8448703506A0A2C From: <sip:kissand2@ipv6.uoc.gr>;tag=2013581859 To: <sip:kissand@ipv6.uoc.gr>;tag=baeb3f9382bd3485fa08dc217d881d32.472a Call-ID: 34CB6351-DA6F-4AFE-BB07-02A3803F9483(a)147.52.250.40 CSeq: 25006 INVITE Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 147.52.3.244:5060 "Noisy feedback tells: pid=20323 req_src_ip=147.52.3.244 req_src_port=5060 in_uri=sip:kissand@ipv6.uoc.gr out_uri=sip:kissand@ipv6.uoc.gr via_cnt==71" Begin forwarded message: Date: Mon, 27 Jun 2005 17:22:14 +0300 From: "Kissandrakis S. George" <kissand(a)ucnet.uoc.gr> To: serusers(a)lists.iptel.org Subject: [Serusers] SER Windows messenger on-line problem Hello, I have installed SER 0.8.14 with radius-mysql authentication besides the default routing, alias=ipv6.uoc.gr ....... if (uri==~"sip:ipv6.uoc.gr") { i have if ( uri=~"sip:[0-9][0-9][0-9][0-9]") { forward (MY-GW-IP,5060); }; i have two accounts in radius kissand(a)ipv6.uoc.gr kissand2(a)ipv6.uoc.gr and with windows messenger 5.1 i go "online" and i can make calls with 4-digit numbers with both accounts If i go "online" with kissand(a)ipv6.uoc.gr and kissand2(a)ipv6.uoc.gr (registered normally) from different computers, kissand(a)ipv6.uoc.gr sees offline kissand2(a)ipv6.uoc.gr and vise versa with ngrep i get SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 147.52.250.40:7563..From: "kissand2(a)ipv6.uoc.gr" <sip:kissand2 @ipv6.uoc.gr>;tag=57126e7b464c4f4498de2e1882dd41ea;epid=3d3d4dcaec..To: <sip:kissand@ipv6.uoc.gr>;tag=baeb 3f9382bd3485fa08dc217d881d32.dbf8..Call-ID: d3e09061703b477a8e927df4c622ff8a..CSeq: 1 SUBSCRIBE..Server: S ip EXpress router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 147.52.3.244:5060 "Noisy feedback tells: pid=16582 req_src_ip=147.52.3.244 req_src_port=5060 in_uri=sip:kissand@ipv6.uoc.gr out_uri=sip:ki ssand(a)ipv6.uoc.gr via_cnt==71".... SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 147.52.250.62:13759..From: "kissand(a)ipv6.uoc.gr" <sip:kissand@ ipv6.uoc.gr>;tag=5c3c36c6e10640f28463ba634d78d5b0;epid=2fd193f505..To: <sip:kissand2@ipv6.uoc.gr>;tag=baeb 3f9382bd3485fa08dc217d881d32.170e..Call-ID: e0b1e57ea151409fbc29aaaeb6c83b4a..CSeq: 1 SUBSCRIBE..Server: S ip EXpress router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 147.52.3.244:5060 "Noisy feedback tells: pid=16582 req_src_ip=147.52.3.244 req_src_port=5060 in_uri=sip:kissand2@ipv6.uoc.gr out_uri=sip:k issand2(a)ipv6.uoc.gr via_cnt==71".... debbuging SER i get 0(16582) build_res_from_sip_res: copied size: orig:87, new: 27, rest: 599 msg=SIP/2.0 483 Too Many Hops^M Via: SIP/2.0/UDP 147.52.3.244;branch=z9hG4bK806e.6a590d4.0^M Via: SIP/2.0/UDP 147.52.250.62:13759^M From: "kissand(a)ipv6.uoc.gr" <sip:kissand@ipv6.uoc.gr>;tag=5c3c36c6e10640f28463ba634d78d5b0;epid=2fd193f505^M To: <sip:kissand2@ipv6.uoc.gr>;tag=baeb3f9382bd3485fa08dc217d881d32.170e^M Call-ID: e0b1e57ea151409fbc29aaaeb6c83b4a^M CSeq: 1 SUBSCRIBE^M Server: Sip EXpress router (0.8.14 (i386/linux))^M Content-Length: 0^M Warning: 392 147.52.3.244:5060 "Noisy feedback tells: pid=16582 req_src_ip=147.52.3.244 req_src_port=5060 in_uri=sip:kissand2@ipv6.uoc.gr out_uri=sip:kissand2@ipv6.uoc.gr via_cnt==71"^M sipsak gives the same results My sip server's ip is 147.52.3.244 resolves to edet.ucnet.uoc.gr SRV record fir _sip._upd.ipv6.uoc.gr resolves to 5060 edet.ucnet.uoc.gr Is it a problem that the server is not in ipv6.uoc.gr DNS domain? If not any hints about "Too Many Hops" between users? Thank you -- Kissandrakis S. George kissand(a)ucnet.uoc.gr NOC - System Administrator University of Crete _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers -- Kissandrakis S. George kissand(a)ucnet.uoc.gr NOC - System Administrator University of Crete

4 6

[Serusers] cisco 7970g
by Iqbal 28 Jun '05

28 Jun '05

Hi Has anyone had any experience with this phone, and any of its features Iqbal

1 0

Re: [Serusers] ser-mysql accounting
by Szabolcs Szasz 28 Jun '05

28 Jun '05

Hi guys! It looks like I have not compiled the acc module with database support. Can somebody confirm this? How can I do this? If I insert these rows : modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 1) this is the error: 0(29943) set_mod_param_regex: parameter <db_flag> not found in module <acc> 0(29943) parse error (87,29-30): Can't set module parameter 0(29943) set_mod_param_regex: parameter <db_missed_flag> not found in module <acc> 0(29943) parse error (88,36-37): Can't set module parameter ERROR: bad config file (2 errors) > Normaly it is ok. And it doesn't work? > > > Szabolcs Szasz a écrit : >> OK. Then, how can I define the acc module's parameter to register calls >> into the database? >> Now I have: >> >> modparam("acc", "log_level", 1) >> modparam("acc", "log_flag", 1 ) >> >> at the module parameters >> >> and >> >> setflag(1); >> >> What have I to do? >> >> >>>but you can have more as one accounting. You're flag numbre must be the >>>same as the number defined at the beggining (module's paramters) >>> >>>Szabolcs Szasz a écrit : >>> >>>> Yes, I do. I consider that this line of log is caused by the >>>>"setflag()" >>>>command : (ACC: transaction answered: method=INVITE,), or not??? >>>> >>>> >>>> >>>>>Do you use the command "setflag()" ? >>>>> >>>>>Szabolcs Szasz a écrit : >>>>> >>>>> >>>>>>Hello >>>>>> >>>>>>I have installed SER on my redhat box. I have a problem with >>>>>> ser-mysql >>>>>>accounting. In the log is apearing the acc information, but it >>>>>> doesn't >>>>>>write into the mysql database? It looks like this when I make a call >>>>>>from >>>>>>506029 to 506030 : >>>>>> >>>>>>0(24195) ------------------------------------------- >>>>>>0(24195) entering route[1] - relaying SIP message >>>>>>0(24195) relaying message ... >>>>>>0(24195) ACC: transaction answered: method=INVITE, >>>>>>i-uri=sip:506030@10.1.10.10:5060;user=phone, >>>>>>o-uri=sip:506030@10.1.10.105:5060, >>>>>>call_id=58dcd4-a010a68-13c4-119d-44d0f9-247(a)10.1.10.10, >>>>>>from=<sip:506029@10.1.10.10> ;tag=a010a68-13c4-119d-44d0f9-3b9d, >>>>>>code=200 >>>>>>0(24195) ------------------------------------------- >>>>>>0(24195) entering main loop >>>>>>0(24195) ACK message received >>>>>>0(24195) ------------------------------------------- >>>>>>0(24195) entering route[1] - relaying SIP message >>>>>>0(24195) relaying message ... >>>>>>0(24195) ACC: request acknowledged: method=ACK, >>>>>>i-uri=sip:506030@10.1.10.105:5060, o-uri=sip:506030@10.1.10.105:5060, >>>>>>call_id=58dcd4-a010a68-13c4-119d-44d0f9-247(a)10.1.10.10, >>>>>>from=<sip:506029@10.1.10.10> ;tag=a010a68-13c4-119d-44d0f9-3b9d, >>>>>>code=200 >>>>>>0(24195) ------------------------------------------- >>>>>>0(24195) entering main loop >>>>>>0(24195) BYE message received >>>>>>0(24195) ------------------------------------------- >>>>>>0(24195) entering route[1] - relaying SIP message >>>>>>0(24195) relaying message ... >>>>>>0(24195) ACC: transaction answered: method=BYE, >>>>>>i-uri=sip:506029@10.1.10.104:5060, o-uri=sip:506029@10.1.10.104:5060, >>>>>>call_id=58dcd4-a010a68-13c4-119d-44d0f9-247(a)10.1.10.10, >>>>>>from=<sip:506030@10.1.10.10:5060;user=phone> >>>>>>;tag=a010a69-13c4-11a0-44dbd5-740c, code=200 >>>>>> >>>>>>Can anyone help me ? >>>>>> >>>>>>Thanks >>>>>>Szabolcs >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>_______________________________________________ >>>>>>Serusers mailing list >>>>>>serusers(a)lists.iptel.org >>>>>>http://lists.iptel.org/mailman/listinfo/serusers >>>>>> >>>>> >>>> >>>>_____________________________ >>>>Szabolcs Szasz >>>>CCNA Network Engineer >>>> >>>>Digi Communication SRL >>>>H2/3 Petofi Sandor St. >>>>Sovata, >>>>Mures, Romania >>>>Tel/ Fax: +40 265 577888 >>>> +40 265 577785 >>>> +40 365 506000 >>>>sszasz(a)digicomm.ro >>>>www.digicomm.ro >>>>_____________________________ >>>> >>>> >>>> >>> >> >> >> _____________________________ >> Szabolcs Szasz >> CCNA Network Engineer >> >> Digi Communication SRL >> H2/3 Petofi Sandor St. >> Sovata, >> Mures, Romania >> Tel/ Fax: +40 265 577888 >> +40 265 577785 >> +40 365 506000 >> sszasz(a)digicomm.ro >> www.digicomm.ro >> _____________________________ >> >> >> > _____________________________ Szabolcs Szasz CCNA Network Engineer Digi Communication SRL H2/3 Petofi Sandor St. Sovata, Mures, Romania Tel/ Fax: +40 265 577888 +40 265 577785 +40 365 506000 sszasz(a)digicomm.ro www.digicomm.ro _____________________________

5 4

[Users] Suitable Distro for OpenSER ?
by Ozan Blotter 28 Jun '05

28 Jun '05

Dear List, My Linux Box is so sticky so i have to renew it, before installing a new one i want to ask all of you a general question. I will do a scratch and full installation, without touching anything i want to install OpenSER with MySql support for both Auth and Acc. without any problems in compiling which distro you offer me to install ? I have below ones: Mandrake 10.0 SuSe 9.3 Fedora Core 3 (Now i'm using this but so many problems) Debian Sarge Thanks, Ozan

3 2

[Users] Bug in OpenSER ?
by Ozan Blotter 28 Jun '05

28 Jun '05

Another boring may be interesting BUG or what else, SER, PSTN Gateway are on Public IP and and i've registered two ATA's with SER, both ATA's are connected through ADSL Gateways but Port Forwarding is done so they can call each other without problem and they can talk each other for a long time, when i do a route plan for number beginning with 0 which is directed to PSTN Gateway, the landline rings and when the callee party answers the phone the system locks up, the caller party (where ATA is) 's internet disconnects and i have to reset modem manually. Any Ideas ? Ozan 0(10163) DEBUG: init_mod_child (0): tm 0(10163) DEBUG: callid: '5ffafd9b-10163(a)127.0.0.1' 0(10163) DEBUG: init_mod_child (0): usrloc 6(10169) SIP Request: 6(10169) method: <REGISTER> 6(10169) uri: <sip:212.154.104.198:5060> 6(10169) version: <SIP/2.0> 6(10169) parse_headers: flags=1 6(10169) Found param type 232, <branch> = <z9hG4bK11732b745417decb>; state=16 6(10169) end of header reached, state=5 6(10169) parse_headers: Via found, flags=1 6(10169) parse_headers: this is the first via 6(10169) After parse_msg... 6(10169) preparing to run routing scripts... 6(10169) parse_headers: flags=128 6(10169) DEBUG:maxfwd:is_maxfwd_present: value = 70 6(10169) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 6(10169) parse_headers: flags=256 6(10169) end of header reached, state=9 6(10169) DEBUG: get_hdr_field: <To> [34]; uri=[sip:8334843600@212.154.104.198] 6(10169) DEBUG: to body [<sip:8334843600@212.154.104.198> ] 6(10169) get_hdr_field: cseq <CSeq>: <194> <REGISTER> 6(10169) DEBUG: get_hdr_body : content_length=0 6(10169) found end of header 6(10169) find_first_route: No Route headers found 6(10169) loose_route: There is no Route HF 6(10169) grep_sock_info - checking if host==us: 15==9 && [212.154.104.198] == [127.0.0.1] 6(10169) grep_sock_info - checking if port 5060 matches port 5060 6(10169) grep_sock_info - checking if host==us: 15==15 && [212.154.104.198] == [212.154.104.198] 6(10169) grep_sock_info - checking if port 5060 matches port 5060 6(10169) grep_sock_info - checking if host==us: 15==9 && [212.154.104.198] == [127.0.0.1] 6(10169) grep_sock_info - checking if port 5060 matches port 5060 6(10169) grep_sock_info - checking if host==us: 15==15 && [212.154.104.198] == [212.154.104.198] 6(10169) grep_sock_info - checking if port 5060 matches port 5060 6(10169) parse_headers: flags=-1 6(10169) parse_headers: flags=33554432 6(10169) build_contact(): Created Contact HF: Contact: <sip:8334843600@85.96.192.140:5060>;expires=60 6(10169) parse_headers: flags=-1 6(10169) check_via_address(85.96.192.140, 85.96.192.140, 0) 6(10169) DEBUG:destroy_avp_list: destroying list (nil) 6(10169) receive_msg: cleaning up 8(10171) SIP Request: 8(10171) method: <INVITE> 8(10171) uri: <sip:02324843600@212.154.104.198> 8(10171) version: <SIP/2.0> 8(10171) parse_headers: flags=1 8(10171) Found param type 232, <branch> = <z9hG4bK24ce735d3f448cd1>; state=16 8(10171) end of header reached, state=5 8(10171) parse_headers: Via found, flags=1 8(10171) parse_headers: this is the first via 8(10171) After parse_msg... 8(10171) preparing to run routing scripts... 8(10171) parse_headers: flags=128 8(10171) DEBUG:maxfwd:is_maxfwd_present: value = 70 8(10171) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 8(10171) parse_headers: flags=8 8(10171) end of header reached, state=9 8(10171) DEBUG: get_hdr_field: <To> [35]; uri=[sip:02324843600@212.154.104.198] 8(10171) DEBUG: to body [<sip:02324843600@212.154.104.198> ] 8(10171) DEBUG: add_param: tag=xIjNwUDMxID 8(10171) end of header reached, state=29 8(10171) parse_headers: flags=256 8(10171) get_hdr_field: cseq <CSeq>: <1> <INVITE> 8(10171) DEBUG: get_hdr_body : content_length=190 8(10171) found end of header 8(10171) find_first_route: No Route headers found 8(10171) loose_route: There is no Route HF 8(10171) grep_sock_info - checking if host==us: 15==9 && [212.154.104.198] == [127.0.0.1] 8(10171) grep_sock_info - checking if port 5060 matches port 5060 8(10171) grep_sock_info - checking if host==us: 15==15 && [212.154.104.198] == [212.154.104.198] 8(10171) grep_sock_info - checking if port 5060 matches port 5060 8(10171) grep_sock_info - checking if host==us: 15==9 && [212.154.104.198] == [127.0.0.1] 8(10171) grep_sock_info - checking if port 5060 matches port 5060 8(10171) grep_sock_info - checking if host==us: 15==15 && [212.154.104.198] == [212.154.104.198] 8(10171) grep_sock_info - checking if port 5060 matches port 5060 8(10171) lookup(): '02324843600' Not found in usrloc 8(10171) grep_sock_info - checking if host==us: 15==9 && [212.154.104.198] == [127.0.0.1] 8(10171) grep_sock_info - checking if port 5060 matches port 5060 8(10171) grep_sock_info - checking if host==us: 15==15 && [212.154.104.198] == [212.154.104.198] 8(10171) grep_sock_info - checking if port 5060 matches port 5060 8(10171) DEBUG: t_newtran: msg id=1 , global msg id=0 , T on entrance=0xffffffff 8(10171) parse_headers: flags=-1 8(10171) parse_headers: flags=60 8(10171) t_lookup_request: start searching: hash=15098, isACK=0 8(10171) DEBUG: RFC3261 transaction matching failed 8(10171) DEBUG: t_lookup_request: no transaction found 8(10171) SER: new INVITE 8(10171) parse_headers: flags=-1 8(10171) check_via_address(85.96.192.140, 85.96.192.140, 0) 8(10171) WARNING:vqm_resize: resize(0) called 8(10171) DEBUG: reply sent out. buf=0x80fbde0: SIP/2.0 1..., shmem=0xf50c6938: SIP/2.0 1 8(10171) DEBUG: _reply_light: finished 8(10171) DEBUG: mk_proxy: doing DNS lookup... 8(10171) check_via_address(85.96.192.140, 85.96.192.140, 0) 8(10171) DEBUG: add_to_tail_of_timer[4]: 0xf50c549c 8(10171) DEBUG: add_to_tail_of_timer[0]: 0xf50c54ac 8(10171) SER: new transaction fwd'ed 8(10171) DEBUG:destroy_avp_list: destroying list (nil) 8(10171) receive_msg: cleaning up 7(10170) SIP Reply (status): 7(10170) version: <SIP/2.0> 7(10170) status: <100> 7(10170) reason: <Trying> 7(10170) parse_headers: flags=1 7(10170) Found param type 232, <branch> = <z9hG4bKafa3.e5774111.0>; state=16 7(10170) end of header reached, state=5 7(10170) parse_headers: Via found, flags=1 7(10170) parse_headers: this is the first via 7(10170) After parse_msg... 7(10170) forward_reply: found module tm, passing reply to it 7(10170) DEBUG: t_check: msg id=1 global id=0 T start=0xffffffff 7(10170) parse_headers: flags=17 7(10170) Found param type 232, <branch> = <z9hG4bK24ce735d3f448cd1>; state=16 7(10170) end of header reached, state=5 7(10170) parse_headers: Via found, flags=17 7(10170) parse_headers: this is the second via 7(10170) DEBUG: add_param: tag=366843004674ff10ff0000ffffffff1e 7(10170) end of header reached, state=29 7(10170) DEBUG: get_hdr_field: <To> [72]; uri=[sip:02324843600@212.154.104.198] 7(10170) DEBUG: to body [<sip:02324843600@212.154.104.198>] 7(10170) get_hdr_field: cseq <CSeq>: <1> <INVITE> 7(10170) parse_headers: flags=4 7(10170) DEBUG: t_reply_matching: hash 15098 label 286553950 branch 0 7(10170) DEBUG: t_reply_matching: reply matched (T=0xf50c5380)! 7(10170) DEBUG: t_check: msg id=1 global id=1 T end=0xf50c5380 7(10170) DEBUG: reply_received: org. status uas=100, uac[0]=0 local=0 is_invite=1) 7(10170) ->>>>>>>>> T_code=100, new_code=100 7(10170) DEBUG: relay_reply: branch=0, save=0, relay=-1 7(10170) DEBUG: add_to_tail_of_timer[1]: 0xf50c54ac 7(10170) DEBUG:destroy_avp_list: destroying list (nil) 7(10170) receive_msg: cleaning up 10(10173) DEBUG: timer routine:4,tl=0xf50c549c next=(nil) 8(10171) SIP Reply (status): 8(10171) version: <SIP/2.0> 8(10171) status: <180> 8(10171) reason: <Ringing> 8(10171) parse_headers: flags=1 8(10171) Found param type 232, <branch> = <z9hG4bKafa3.e5774111.0>; state=16 8(10171) end of header reached, state=5 8(10171) parse_headers: Via found, flags=1 8(10171) parse_headers: this is the first via 8(10171) After parse_msg... 8(10171) forward_reply: found module tm, passing reply to it 8(10171) DEBUG: t_check: msg id=2 global id=1 T start=0xffffffff 8(10171) parse_headers: flags=17 8(10171) Found param type 232, <branch> = <z9hG4bK24ce735d3f448cd1>; state=16 8(10171) end of header reached, state=5 8(10171) parse_headers: Via found, flags=17 8(10171) parse_headers: this is the second via 8(10171) DEBUG: add_param: tag=366843004674ff10ff0000ffffffff1e 8(10171) end of header reached, state=29 8(10171) DEBUG: get_hdr_field: <To> [72]; uri=[sip:02324843600@212.154.104.198] 8(10171) DEBUG: to body [<sip:02324843600@212.154.104.198>] 8(10171) get_hdr_field: cseq <CSeq>: <1> <INVITE> 8(10171) parse_headers: flags=4 8(10171) DEBUG: t_reply_matching: hash 15098 label 286553950 branch 0 8(10171) DEBUG: t_reply_matching: reply matched (T=0xf50c5380)! 8(10171) DEBUG: t_check: msg id=2 global id=2 T end=0xf50c5380 8(10171) DEBUG: reply_received: org. status uas=100, uac[0]=100 local=0 is_invite=1) 8(10171) ->>>>>>>>> T_code=100, new_code=180 8(10171) DEBUG: relay_reply: branch=0, save=0, relay=0 8(10171) old size: 518, new size: 449 8(10171) build_res_from_sip_res: copied size: orig:90, new: 21, rest: 428 msg= SIP/2.0 180 Ringing Via: SIP/2.0/UDP 85.96.192.140:5060;branch=z9hG4bK24ce735d3f448cd1 Record-Route: <sip:212.154.104.198;ftag=xIjNwUDMxID;lr=on> From: <sip:8334843600@212.154.104.198;user=phone>;tag=xIjNwUDMxID To: <sip:02324843600@212.154.104.198>;tag=366843004674ff10ff0000ffffffff1e Call-ID: 709D1B912DA1212(a)192.168.1.201 CSeq: 1 INVITE Contact: <sip:02324843600@212.154.59.48;user=phone> Server: MERA MSIP v.1.0.1 Content-Length: 0 8(10171) DEBUG: reply relayed. buf=0x80fb9d8: SIP/2.0 1..., shmem=0xf50c6938: SIP/2.0 1 8(10171) DEBUG: add_to_tail_of_timer[1]: 0xf50c54ac 8(10171) DEBUG:destroy_avp_list: destroying list (nil) 8(10171) receive_msg: cleaning up >>>> internet connection get lost after above lines and below is SER.CFG <<<<< # # $Id: openser.cfg,v 1.1.1.1 2005/06/13 16:47:30 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) fork=yes #log_stderror=no # (cmd line: -E) alias=localhost alias=127.0.0.1 alias=212.154.104.198 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/openser/modules/auth.so" #loadmodule "/usr/lib/openser/modules/auth_db.so" # ----------------- 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_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) alias=212.154.104.198 alias=localhost alias=127.0.0.1 # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; if (uri=~"^sip:0[0-9]*@*") { rewritehost("212.154.59.48"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; }

2 1

[Serusers] SER + multiple domains
by gsalvucci＠ing.unrc.edu.ar 28 Jun '05

28 Jun '05

Hi, I would like to add new users at a table called subscriber_xxx. So I use the command serctl add user_name password email, but it adds the user in the table subscriber. Anyone knows how to do this without to rename the subscribers table? Thanks Gustavo Salvucci ---------------------------------------------------------------- Este mensaje ha sido enviado usando el servicio de Webmail de la Universidad Nacional de Río Cuarto. Basado en Horde/IMP

1 0

Re: [Serusers] SER FIFO and routing
by Samuel Osorio Calvo 28 Jun '05

28 Jun '05

I guess the admin guide is not popular..... You can explicitely say the next hop where SER has to send the message by adding a new line between the Req-URI and the first dot... :t_uac_dlg:cc_reply BYE sip:00111111111111@11.12.13.134:5060 ->11.12.13.135<- maybe with a sip: prefix... .. Via: SIP/2.0/UDP 31.32.33.34:25972;branch=z9hG4bKdb61c97f9cba17c1 Hope it helps, Samuel. Unclassified. >>> "Stuart Kirkwood - Jubilee Consutancy Limited" <stuart(a)jubileeconsultancy.com> 06/28/05 03:45PM >>> Hi I have a problem with the FIFO request below. The SIP request should route through 11.12.13.135 which guards the gateway at 11.12.13.134. The IP packet containing the SIP message that SER generates is good but SER sets the destination IP address of the packet as 11.12.13.134 and not 11.12.13.135. Is there something wrong with my request or is this a bug? Any help appreciated. :t_uac_dlg:cc_reply BYE sip:00111111111111@11.12.13.134:5060 .. Via: SIP/2.0/UDP 31.32.33.34:25972;branch=z9hG4bKdb61c97f9cba17c1 From: <sip:1000@xsipserver.com>;tag=c53a53516c552844 To: <sip:11111111111@xsipserver.com>;tag=3ADD9EA0-1EC0 Call-ID: 7c70b0b24a053e37(a)192.168.7.21 CSeq: 52779 BYE Route: <sip:11.12.13.135;ftag=c53a53516c552844;lr=on> Max-Forwards: 69 .. .. I'm running SER 0.8.14 on Fedora Core 3. Thank you. Stuart _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] SER FIFO and routing
by Stuart Kirkwood - Jubilee Consutancy Limited 28 Jun '05

28 Jun '05

Hi I have a problem with the FIFO request below. The SIP request should route through 11.12.13.135 which guards the gateway at 11.12.13.134. The IP packet containing the SIP message that SER generates is good but SER sets the destination IP address of the packet as 11.12.13.134 and not 11.12.13.135. Is there something wrong with my request or is this a bug? Any help appreciated. :t_uac_dlg:cc_reply BYE sip:00111111111111@11.12.13.134:5060 . Via: SIP/2.0/UDP 31.32.33.34:25972;branch=z9hG4bKdb61c97f9cba17c1 From: <sip:1000@xsipserver.com>;tag=c53a53516c552844 To: <sip:11111111111@xsipserver.com>;tag=3ADD9EA0-1EC0 Call-ID: 7c70b0b24a053e37(a)192.168.7.21 CSeq: 52779 BYE Route: <sip:11.12.13.135;ftag=c53a53516c552844;lr=on> Max-Forwards: 69 . . I'm running SER 0.8.14 on Fedora Core 3. Thank you. Stuart

1 0

[Serusers] ser_mysql accounting
by Szabolcs Szasz 28 Jun '05

28 Jun '05

Thank you for the help! It's working fine :) _____________________________ Szabolcs Szasz CCNA Network Engineer Digi Communication SRL H2/3 Petofi Sandor St. Sovata, Mures, Romania Tel/ Fax: +40 265 577888 +40 265 577785 +40 365 506000 sszasz(a)digicomm.ro www.digicomm.ro _____________________________

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[Users] pseudo-variables in textops (Was: Re: [Devel] unified pseudo-variables in avpops and xlog)
by Daniel-Constantin Mierla 28 Jun '05

28 Jun '05

Hello, the textops module has now support for pseudo-variables in subst(), subst_uri() and subst_user(). You can use any of the pseudo-variables that OpenSER supports (http://openser.org/docs/pseudo-variables.html) in the replacement part of the parameter given to subst*() -- the parameter for these functions has the following format "/regular_expression/replacement/flags". For more info, see the readme file: http://www.openser.org/docs/modules/0.10.x/textops.html Should be analyzed if worth to implement pseudo-variables support to regular_expression part because it will slow down a lot the processing since the regexp has to be compiled each time. Daniel On 06/24/05 01:11, Miguel Angel Villar wrote: >Hi Daniel > This is a really nice feature. I had been looking for something like >this in the textops module. Implementing pseudo-variables in subst_uri() and >subst() functions will give power tools for doing dynamic number/domain >translation of URI. Conbined with avpops module the transformation can be >based on external database, Great! > >Regards. >MaV.- > > > >>Hello, >> >>the avpops and xlog module are using now same format for specifiers to >>print formatted strings. avpops has a new function avp_printf(dest, >>format) which stores in the AVP 'dest' the string resulted after >>substitution of the specifiers from "format" parameter. >> >>The old marker for specifiers '%' has been replaced by '$' which is more >>common for variables. To specify a header or an AVP you have to use now >>$hdr(header_name[index]) or $avp(avp_id[index]). The list with all >>available pseudo-variables in OpenSER is posted at: >>http://openser.org/docs/pseudo-variables.html >> >>avp_printf() allow to compose a value from many parts of a SIP message, >>other avps and many other values taken from system. For example: >>avp_printf("i:20", "the time is $Tf, the request is a $rm and has the >> >> >>from header $hdr(from)"); > > >>The documentation for avpops and xlog module has been updated on CVS, >>too. Note that these features are available only on CVS head >>(development version - http://openser.org/index.php#download). >>Online documentation site for modules (http://openser.org/docs/) has >>been split in 0.9x (stable) and 0.10.x (devel) directories. >> >>Follows the integration of same format for pseudo-variables in textops, >>acc and tm. We will appreciate any feedback. >> >>Daniel >> >> > > >_______________________________________________ >Devel mailing list >Devel(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/devel > > >

1 0

[Serusers] Unable to register using linphone
by Mithun Hebbar 28 Jun '05

28 Jun '05

Hi, I downloaded the ser-0.8.14_linux_i386 tar zipped archive from the iptel.org website. Unzipped it in the / directory and then got the SIP server running from /usr/local/sbin/ directory. I am using the linphone version 1.0.0 pre-installed on SUSE linux 9.3. But I am running into configuration issues while trying to configure the SIP Express Router (ser) to run as a proxy/ registrar server. I made the following changes in the Preferences menu of linphone: Automatically guess the SIP Identity as say, user@192.40.40.15 Add proxy/ registrar server SIP Identity: sip:afila@192.40.40.15 and SIP Proxy server: sip:192.40.40.56 No changes were made to the ser.cfg file that comes with the distribution. My question is, should some changes be made to the ser.cfg file in order for a basic registration erquest to be received and accepted? If yes, could some one please suggest the changes? If no, what could I be doing wrong over here? Please help! My ser.cfg file is as follows: # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/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" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- 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_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (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; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } Regards. Mithun.

1 0

Re: [Serusers] Autostart SER on Boot up.
by Samuel Osorio Calvo 28 Jun '05

28 Jun '05

My memory is not good so check out the spelling....sorry but I am not on a linux machine and can not check the paths... Put the ser executable (the one usually in /usr/local/sbin) in /etc/init.d (a link would be enough, depending on what you want). Under /etc there are several directories that are used to start processes during startup, since I guess you are using runlevel 3, you should add a link in /etc/rc3.d/ to the ser application in /etc/init.d with special format: it has to start with the letter S, followed by two numbers and an arbitrary string: something like Sxxstring. The S stands for Start and the xx numbers indicate the order to start. The order is important!!! don't try to start ser before the networking or it will simply fail. To gracefully stop ser you should add in /etc/rc0.d and /etc/rc6.d the appropriate Kyystring link. Explore in the /rcM.d directrories to see how to make it working... About the variable, just add in your home .bash_profile export SIP_DOMAIN="myserver...." Hope it helps, Samuel. Unclassified. >>> "Sebastian Kigelman" <sebastian(a)kigelman.com.ar> 06/28/05 03:28AM >>> Hi!!.. Its my again!!!... As I tell you people, I am newy in the "SER World", and I have a existential question: How can I stat SER automatically during the machine startup?... Also, I need to know where I can put the environment variable (export SIP_DOMAIN="myserver.blah_blah.com") so I not need to type it on every remote session that I start... I am working in Debian Linux v2.4.27-2-386... Any help will be very usefull for my... Zeva... _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] RE: ser.cfg Configuration
by Abdul Lateef 28 Jun '05

28 Jun '05

Hello, Thank you friends, I got the Log file under /var/log/message Once again thanks for all friends. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________ Do you Yahoo!? Make Yahoo! your home page http://www.yahoo.com/r/hs

1 0

[Serusers] ser.cfg Configuration (Daryl Sanders)
by Abdul Lateef 28 Jun '05

28 Jun '05

Hello, Thank you for kind of information, I am reading the getting start. But i have a small question, Where i can see the SER log? I means in which directory is located? Thank You Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

3 2

[Serusers] Call expires after 30 seconds with SER + MEDIAPROXY + VOVIDA B2BUA
by Alberto Cruz 28 Jun '05

28 Jun '05

Does anybody have SER running with MEDIAPROXY and Vovida B2BUA at the same machine? I'm having a little trouble with the calls: They are finishing after 30 seconds. After I have reviewed the NGREP sniffing I can detect that the B2BUA is sending a BYE to the SER as you can see: U 2005/06/26 22:59:29.210454 200.67.33.247:61767 -> 200.94.75.6:5060 ACK sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-4413ed9d. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=d96bd682df5aa957o0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=71eb8754. Call-ID: 1a18e21e-eaa9b26b(a)172.31.253.127. CSeq: 101 ACK. Max-Forwards: 70. Route: <sip:018183324166@200.94.75.6:5065>. Contact: 8183651723 <sip:8183651723@172.31.253.127:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/26 22:59:29.213193 200.94.75.6:5060 -> 200.94.75.6:5065 ACK sip:018183324166@200.94.75.6:5065 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=d96bd682df5aa957o0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=0. Via: SIP/2.0/UDP 172.31.253.127:5060;rport=61767;received=200.67.33.247;branch=z9hG4bK-4413ed9d. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=d96bd682df5aa957o0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=71eb8754. Call-ID: 1a18e21e-eaa9b26b(a)172.31.253.127. CSeq: 101 ACK. Max-Forwards: 16. Contact: 8183651723 <sip:8183651723@172.31.253.127:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. P-hint: rr-enforced. . # U 2005/06/26 22:59:29.214421 200.94.75.6:5065 -> 200.94.75.6:5060 ACK sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 200.94.75.6:5065. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=33F1FB80-B96. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=d96bd682df5aa957o0. Call-ID: 8c9da6de70b56114b926f823b1173c3a(a)200.94.75.6. CSeq: 101 ACK. Max-Forwards: 70. Route: <sip:018183324166@200.94.75.1:5060>. Content-Length: 0. . # U 2005/06/26 22:59:29.218000 200.94.75.6:5060 -> 200.94.75.1:5060 ACK sip:018183324166@200.94.75.1:5060 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=d96bd682df5aa957o0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=0. Via: SIP/2.0/UDP 200.94.75.6:5065. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=33F1FB80-B96. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=d96bd682df5aa957o0. Call-ID: 8c9da6de70b56114b926f823b1173c3a(a)200.94.75.6. CSeq: 101 ACK. Max-Forwards: 16. Content-Length: 0. P-hint: rr-enforced. . ## U 2005/06/26 23:00:05.104387 200.94.75.6:5065 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060;lr=on SIP/2.0. Via: SIP/2.0/UDP 200.94.75.6:5065. To: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=d96bd682df5aa957o0. From: <sip:018183324166@sipsrv.conecta-te.com>;tag=71eb8754. Call-ID: 1a18e21e-eaa9b26b(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 16. Route: <sip:8183651723@200.94.75.6:61767>. Record-Route: <sip:200.94.75.6;ftag=d96bd682df5aa957o0;lr=on>. Contact: 8183651723 <sip:8183651723@172.31.253.127:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. P-hint: rr-enforced. Does anybody know if this is a problem with the VOVIDA B2BUA or the SER? Any idea? Thanks Alberto Cruz

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RE: [Serusers] *** SER - Asterisk
by harry gaillac 28 Jun '05

28 Jun '05

Luca, you may find help here: http://www.cs.colostate.edu/~somlo/CSU-SIP-notes/ http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large ask for help to asterisk-users(a)lists.digium.org Regards harry --- "lucape(a)inwind.it" <lucape(a)inwind.it> a écrit : > i'm not offended, but i speak a bad english, and > i've not found documents on ser + asterisk on > www.asterisk.org... > if you can help me to found it, give me the link... > > thanks > luca > > > ---------- Initial Header ----------- > > From : "harry gaillac" gaillacharry(a)yahoo.fr > To : "lucape(a)inwind.it" lucape(a)inwind.it > Cc : asterisk-users(a)lists.digium.com > Date : Mon, 27 Jun 2005 20:50:27 +0200 (CEST) > Subject : RE: [Serusers] *** SER - Asterisk > > > > > > > > > I don't want to offend you but you should have a > look > > to sems . > > > > You won't find docs to help you at asterisk.org > > > > --- "lucape(a)inwind.it" <lucape(a)inwind.it> a écrit > : > > > > > hello > > > > > > help me to configure ser + asterisk > > > how to do the configuration? > > > > > > Luca > > > > > > > > > > > > > > > ____________________________________________________________ > > > 6X velocizzare la tua navigazione a 56k? 6X Web > > > Accelerator di Libero! > > > Scaricalo su INTERNET GRATIS 6X > http://www.libero.it > > > > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > > > ____________________________________________________________ > Navighi a 4 MEGA e i primi 3 mesi sono GRATIS. > Scegli Libero Adsl Flat senza limiti su > http://www.libero.it > > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

RE: [Serusers] Use of fifo for registration :ul_add: command tutorial
by Samuel Osorio Calvo 28 Jun '05

28 Jun '05

Try the following..... ################################################# #!/bin/sh cat> /tmp/ser_fifo << :ul_add:return_fifo location sip:user@domain sip:user@IP {expires} {qvalue} {replicate} {flags} ################################################## return_fifo is the name of the fifo (with a /tmp prefix) that will use SER for answering to the fifo command.If you don't want to read it, just don't worry about the possible complaint of FIFO not existing, error opening return fifo or something like that in SER log. location is the table where to add/update the record sip:user@domain-->AoR you might have to look at SER's debug output because I remember there was something with sip: prefix in either AoR or contact. I think using sip: will work but just check if you have to remove the sip: prefix from the contact.... sip:user@IP-->contact expires-->record validity in seconds qvalue-->[0..1] replicate-->bool 0|1 flags-->not 100% sure if it corresponds to the flag vector of SER or just indicate natted contacts...I recommend to use a 0 value here.. Remember to add an End Of File character in the last line or there are some funny parsing things with subsequent messages isnerted in the fifo... Hope it helps, Samuel. Unclassified. >>> "Rodrigue ESTEVE" <rodrigue(a)mobius.fr> 06/28/05 09:13AM >>> Does anybody have an explanation to use the fifo for registration (using a script and not the serctl command) ? Thanks very much _____ De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de Rodrigue ESTEVE Envoyé : mardi 14 juin 2005 19:00 À : serusers(a)lists.iptel.org Objet : [Serusers] Use of fifo for registration :ul_add: command tutorial Hi all, I would like to use ser_fifo to register a user; I think the :ul_add: command is the clue, but I don't how to use the fifo to do this. My aim is to do the following: I have a Voip gateway on a customer with a lot of phone number (700) and I would like to authenticate each number. Unfortunately I have to add a small amount of configuration on the voip gateway each time I add a number (uri), moreover I'm not sure this gateway will work with so many numbers, and the bandwidth needed to register each number every 5 minutes will be wasted. The idea is to challenge only one time the gateway and then in ser.cfg if the challenge succeeds I execute a script to register by bulk all the numbers enclosed in a file on the proxy. To do this I have to pass arguments to the fifo using the :ul_add: command, but this is where I'm stuck. I've done this: echo :ul_add:receiving_fifo > /tmp/ser_fifo echo location > /tmp/ser_fifo echo 100 > /tmp/ser_fifo echo <sip:100@10.1.1.1:5060> > /tmp/ser_fifo echo 300 > /tmp/ser_fifo echo 0 > /tmp/ser_fifo echo 2 > /tmp/ser_fifo but it does not work, I think this is due to the receiving_fifo, I have to do something with it to process the response of SER, but what ?? I know I can use the following command to workaround the pbm, and I'm successful with this command: Serctl fifo ul_add location *.. But it requires one process each time I register a number, and it takes 1 minute on a Celeron 700 for 500 numbers (30 % CPU), and I think using the fifo is a more suitable solution (less CPU consuming). Thanks for your support, Rodrigue PS: I would like to register every 5minutes,cause the customer gateway is able to redirect to PSTN in case of failure.

1 0

[Serusers] cisco call manager 4.0 and ser
by Dinesh 28 Jun '05

28 Jun '05

Hi all, I tried to do a connection between ser and cisco call manager. As there was no documentation of how to do, I created a sip trunk on ccm 4.0 and I put it my sip servers ip address. # send out 2 prefix to if (uri=~"^sip:2[0-9].*@.*") { if (!is_user_in("From", "ld")) { sl_send_reply("403", "Not ld user"); break; }; setflag(1); strip(3); rewritehostport("ip_address_of_ccm4.0:5060"); if (!t_relay()) { sl_reply_error(); }; break; }; http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration When I dial on xten, I get a SIP/2.0 400 Bad Request - 'Malformed/Missing Record Route' Can anyone help, I attached the diagnostic log from xten. SEND TIME: 25265760 SEND >> 203.125.17.228:5060 INVITE sip:22265869804@imcb.a-star.edu.sg SIP/2.0 Via: SIP/2.0/UDP 137.132.43.121:5060;rport;branch=z9hG4bKA530CE5714B7497B83F484F151CEC3F5 From: Dinesh Birlasekaran <sip:65869804@imcb.a-star.edu.sg>;tag=279805281 To: <sip:22265869804@imcb.a-star.edu.sg> Contact: <sip:65869804@137.132.43.121:5060> Call-ID: B2D9356E-8F57-4657-8215-D22873F5A9FE(a)10.217.64.103 CSeq: 19691 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 301 v=0 o=65869804 25265730 25265760 IN IP4 137.132.43.121 s=X-Lite c=IN IP4 137.132.43.121 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 RECEIVE TIME: 25265810 RECEIVE << 203.125.17.228:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.217.64.103:5060;rport=5060;branch=z9hG4bKA530CE5714B7497B83F484F151CEC3F5 From: Dinesh Birlasekaran <sip:65869804@imcb.a-star.edu.sg>;tag=279805281 To: <sip:22265869804@imcb.a-star.edu.sg> Call-ID: B2D9356E-8F57-4657-8215-D22873F5A9FE(a)10.217.64.103 CSeq: 19691 INVITE Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 203.125.17.228:5060 "Noisy feedback tells: pid=18296 req_src_ip=137.132.43.121 req_src_port=5060 in_uri=sip:22265869804@imcb.a-star.edu.sg out_uri=sip:65869804@10.217.84.11:5060 via_cnt==1" RECEIVE TIME: 25265820 RECEIVE << 203.125.17.228:5060 SIP/2.0 400 Bad Request - 'Malformed/Missing Record Route' Via: SIP/2.0/UDP 10.217.64.103:5060;rport=5060;branch=z9hG4bKA530CE5714B7497B83F484F151CEC3F5 From: Dinesh Birlasekaran <sip:65869804@imcb.a-star.edu.sg>;tag=279805281 To: <sip:22265869804@imcb.a-star.edu.sg>;tag=16897853 Call-ID: B2D9356E-8F57-4657-8215-D22873F5A9FE(a)10.217.64.103 CSeq: 19691 INVITE Content-Length: 0 Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : dinesh(a)imcb.a-star.edu.sg WWW: www.imcb.a-star.edu.sg

1 0

[Users] MySql Module
by Ozan Blotter 28 Jun '05

28 Jun '05

Dear List, Is there a way to compile only mysql module, i've already installed OpenSER and it's working quite and now without recompiling i want to compile mysql.so only any chance ? Thanks, Ozan

2 1

[Serusers] Caller ID blocking thru SER
by Chris Tsai 28 Jun '05

28 Jun '05

Hi,ALL I have a problem when i use the caller id blocking at iptel.org through by SER. The Server will response the 403 forbidden as follows: Anybody knows what's the problem? BTW,the SER have implement as follow RFC-3324~3325 or draft-ietf-sip-privacy-04.txt ? and Our UAs how to send the INVITE message about Caller ID blocking? Thanks Chris Tsai ------------------------------------------- sip:8201@iptel.org SIP/2.0 Via: SIP/2.0/UDP 218.168.199.121:5060;branch=z9hG4bK-fTh-29195 From: Anonymous <sip:anonymous@iptel.org>;tag=Edi-3010 To: <sip:8201@iptel.org> Call-ID: rYQ-8380(a)218.168.199.121 CSeq: 4 INVITE Contact: <sip:christsai12@218.168.199.121> Max-Forwards: 70 Proxy-Require: privacy Privacy: id RPID-Privacy: full supported: replaces User-Agent: DrayTek UA-1.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY, SUBSCRIBE Content-Type: application/sdp Content-Length: 272 v=0 o=070500009 5972727 56415 IN IP4 218.168.199.121 s=SIP Call c=IN IP4 218.168.199.121 t=0 0 m=audio 10056 RTP/AVP 18 2 4 8 0 a=rtpmap:18 g729/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:4 g723/8000 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=ptime:10 a=sendrecv IP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 218.168.199.121:5060;branch=z9hG4bK-fTh-29195;rport=5060 From: Anonymous <sip:anonymous@iptel.org>;tag=Edi-3010 To: <sip:8201@iptel.org>;tag=b51ece8fd8c195776737473dd6552d43.bb01 Call-ID: rYQ-8380(a)218.168.199.121 CSeq: 4 INVITE Proxy-Authenticate: Digest realm="iptel.org", nonce="42c1008d49e435c10c729f0f86eb3a0f97dcc58d" Server: Sip EXpress router (0.9.0udpfifo (i386/linux)) Content-Length: 0 Warning: 392 195.37.77.99:5060 "Noisy feedback tells: pid=30605 req_src_ip=218.168.199.121 req_src_port=5060 in_uri=sip:8201@iptel.org out_uri=sip:8201@iptel.org via_cnt==1" ACK sip:8201@iptel.org SIP/2.0 Via: SIP/2.0/UDP 218.168.199.121:5060;branch=z9hG4bK-fTh-29195 From: Anonymous <sip:anonymous@iptel.org>;tag=Edi-3010 To: <sip:8201@iptel.org>;tag=b51ece8fd8c195776737473dd6552d43.bb01 Call-ID: rYQ-8380(a)218.168.199.121 CSeq: 4 ACK Content-Length: 0 INVITE sip:8201@iptel.org SIP/2.0 Via: SIP/2.0/UDP 218.168.199.121:5060;branch=z9hG4bK-jsZ-7968 From: Anonymous <sip:anonymous@iptel.org>;tag=Edi-3010 To: <sip:8201@iptel.org> Call-ID: rYQ-8380(a)218.168.199.121 CSeq: 5 INVITE Contact: <sip:christsai12@218.168.199.121> Proxy-Authorization: Digest username="christsai12", realm="iptel.org", nonce="42c1008d49e435c10c729f0f86eb3a0f97dcc58d", uri="sip:8201@iptel.org", response="d2b040d5f711da936f271319438880fa", algorithm=MD5 Max-Forwards: 70 Proxy-Require: privacy Privacy: id RPID-Privacy: full supported: replaces User-Agent: DrayTek UA-1.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY, SUBSCRIBE Content-Type: application/sdp Content-Length: 272 v=0 o=070500009 5972727 56415 IN IP4 218.168.199.121 s=SIP Call c=IN IP4 218.168.199.121 t=0 0 m=audio 10056 RTP/AVP 18 2 4 8 0 a=rtpmap:18 g729/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:4 g723/8000 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=ptime:10 a=sendrecv IP/2.0 403 That is ugly -- use From=id next time (OB) Via: SIP/2.0/UDP 218.168.199.121:5060;branch=z9hG4bK-jsZ-7968;rport=5060 From: Anonymous <sip:anonymous@iptel.org>;tag=Edi-3010 To: <sip:8201@iptel.org>;tag=b51ece8fd8c195776737473dd6552d43.b079 Call-ID: rYQ-8380(a)218.168.199.121 CSeq: 5 INVITE Server: Sip EXpress router (0.9.0udpfifo (i386/linux)) Content-Length: 0 Warning: 392 195.37.77.99:5060 "Noisy feedback tells: pid=30599 req_src_ip=218.168.199.121 req_src_port=5060 in_uri=sip:8201@iptel.org out_uri=sip:8201@iptel.org via_cnt==1" ACK sip:8201@iptel.org SIP/2.0 Via: SIP/2.0/UDP 218.168.199.121:5060;branch=z9hG4bK-jsZ-7968 From: Anonymous <sip:anonymous@iptel.org>;tag=Edi-3010 To: <sip:8201@iptel.org>;tag=b51ece8fd8c195776737473dd6552d43.b079 Call-ID: rYQ-8380(a)218.168.199.121 CSeq: 5 ACK Content-Length: 0

1 0

RE: [Serusers] Use of fifo for registration :ul_add: command tutorial
by Rodrigue ESTEVE 28 Jun '05

28 Jun '05

Does anybody have an explanation to use the fifo for registration (using a script and not the serctl command) ? Thanks very much _____ De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de Rodrigue ESTEVE Envoyé : mardi 14 juin 2005 19:00 À : serusers(a)lists.iptel.org Objet : [Serusers] Use of fifo for registration :ul_add: command tutorial Hi all, I would like to use ser_fifo to register a user; I think the :ul_add: command is the clue, but I dont how to use the fifo to do this. My aim is to do the following: I have a Voip gateway on a customer with a lot of phone number (700) and I would like to authenticate each number. Unfortunately I have to add a small amount of configuration on the voip gateway each time I add a number (uri), moreover Im not sure this gateway will work with so many numbers, and the bandwidth needed to register each number every 5 minutes will be wasted. The idea is to challenge only one time the gateway and then in ser.cfg if the challenge succeeds I execute a script to register by bulk all the numbers enclosed in a file on the proxy. To do this I have to pass arguments to the fifo using the :ul_add: command, but this is where Im stuck. Ive done this: echo :ul_add:receiving_fifo > /tmp/ser_fifo echo location > /tmp/ser_fifo echo 100 > /tmp/ser_fifo echo <sip:100@10.1.1.1:5060> > /tmp/ser_fifo echo 300 > /tmp/ser_fifo echo 0 > /tmp/ser_fifo echo 2 > /tmp/ser_fifo but it does not work, I think this is due to the receiving_fifo, I have to do something with it to process the response of SER, but what ?? I know I can use the following command to workaround the pbm, and Im successful with this command: Serctl fifo ul_add location .. But it requires one process each time I register a number, and it takes 1 minute on a Celeron 700 for 500 numbers (30 % CPU), and I think using the fifo is a more suitable solution (less CPU consuming). Thanks for your support, Rodrigue PS: I would like to register every 5minutes,cause the customer gateway is able to redirect to PSTN in case of failure.

1 0

[Serusers] 183 session progress
by Jorge Crichigno 28 Jun '05

28 Jun '05

I have a doubt with a scenario, where a call is done between two APX/MAX-TNT Lucent Gateways(GW): Gw A SER GW B | INVITE-> | | PSTN A | <-100 | | PSTN B side side | | INVITE-> | | | <-183 w/SDP | | <-183 w/SDP | | | | | | | | Then, the call downs by some reason as timeout. I just don't understand well the meaning of 183. AFAIK, when it happens, it should or could be an 'early' rtp flow from B that should be audible for A (for example, A could expect a play message from PSTN B side operator). Is that correct? Could someone clarify me this concept? Thank you! PS: does someone use ever APX/MAX-TNT Lucent gateways? I have some doubts about his configuration. Jorge

1 0

[Serusers] acc modparam radius_client issue
by Cameron Beattie 28 Jun '05

28 Jun '05

When I specify the modparam("acc", "radius_config", "\etc\radiusclient\radiusclient.conf"), I get an error "rc_read_config: can't open \etc^Gdiuscleint^Gdiuscleint.conf: No such file or directory" Is this a bug or am I doing something wrong? If I store the file in the default location /usr/local/etc/radiusclient/ then I have no such problems. Note this is v 0.10.99-dev7 Regards Cameron

1 1

[Serusers] Autostart SER on Boot up.
by Sebastian Kigelman 28 Jun '05

28 Jun '05

Hi!!.. Its my again!!!... As I tell you people, I am newy in the "SER World", and I have a existential question: How can I stat SER automatically during the machine startup?... Also, I need to know where I can put the environment variable (export SIP_DOMAIN="myserver.blah_blah.com") so I not need to type it on every remote session that I start... I am working in Debian Linux v2.4.27-2-386... Any help will be very usefull for my... Zeva...

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[Serusers] SER does not start with MySQL
by Sebastian Kigelman 28 Jun '05

28 Jun '05

Hi there!!!... Iam newy in SER mailing list, and I have a problem with my SER installation that I saw that other members have before, but I can't find a solution to the issue. I have installed the SER (v0.8.14), all works fine until I add MySQL support to get persistent users. after that, when I run "serctl start" I get this error: ---------------------------------------------------------------- Starting SER : cat: /var/run/ser.pid: No such file or directory started pid() ---------------------------------------------------------------- I have made some test to troubleshot this issue, for exsaplme: remplacing DIR=... with the full path to SER executable in "serctl"... and also I increase the debug level in "ser.cfg" to 7, and I get this output: ---------------------------------------------------------------- MyHost:/etc/ser# ser -c 0(2577) loading module /usr/lib/ser/modules/mysql.so 0(2577) loading module /usr/lib/ser/modules/sl.so 0(2577) loading module /usr/lib/ser/modules/tm.so 0(2577) loading module /usr/lib/ser/modules/rr.so 0(2577) loading module /usr/lib/ser/modules/maxfwd.so 0(2577) loading module /usr/lib/ser/modules/usrloc.so 0(2577) loading module /usr/lib/ser/modules/registrar.so 0(2577) loading module /usr/lib/ser/modules/auth.so 0(2577) loading module /usr/lib/ser/modules/auth_db.so 0(2577) set_mod_param_regex: usrloc matches module usrloc 0(2577) set_mod_param_regex: found <db_mode> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(2577) set_mod_param_regex: auth_db matches module auth_db 0(2577) set_mod_param_regex: found <calculate_ha1> in module auth_db [/usr/lib/ser/modules/auth_db.so] 0(2577) set_mod_param_regex: auth_db matches module auth_db 0(2577) set_mod_param_regex: found <password_column> in module auth_db [/usr/lib/ser/modules/auth_db.so] 0(2577) set_mod_param_regex: rr matches module rr 0(2577) set_mod_param_regex: found <enable_full_lr> in module rr [/usr/lib/ser/modules/rr.so] 0(2577) find_export: found <mf_process_maxfwd_header> in module maxfwd_module [/usr/lib/ser/modules/maxfwd.so] 0(2577) find_export: found <sl_send_reply> in module sl_module [/usr/lib/ser/modules/sl.so] 0(2577) find_export: found <sl_send_reply> in module sl_module [/usr/lib/ser/modules/sl.so] 0(2577) find_export: found <record_route> in module rr [/usr/lib/ser/modules/rr.so] 0(2577) find_export: found <loose_route> in module rr [/usr/lib/ser/modules/rr.so] 0(2577) find_export: found <t_relay> in module tm [/usr/lib/ser/modules/tm.so] 0(2577) find_export: found <www_authorize> in module auth_db [/usr/lib/ser/modules/auth_db.so] 0(2577) find_export: found <www_challenge> in module auth [/usr/lib/ser/modules/auth.so] 0(2577) find_export: found <save> in module registrar [/usr/lib/ser/modules/registrar.so] 0(2577) find_export: found <lookup> in module registrar [/usr/lib/ser/modules/registrar.so] 0(2577) find_export: found <sl_send_reply> in module sl_module [/usr/lib/ser/modules/sl.so] 0(2577) find_export: found <t_relay> in module tm [/usr/lib/ser/modules/tm.so] 0(2577) find_export: found <sl_reply_error> in module sl_module [/usr/lib/ser/modules/sl.so] 0(2577) routing table 0: ...blah ...blah ...blah ...blah WARNING: could not rev. resolve 192.168.0.4 Listening on 127.0.0.1 [127.0.0.1]:5060 192.168.0.4 [192.168.0.4]:5060 Aliases: MyHost:5060 localhost:5060 localhost.localdomain:5060 config file ok, exiting... 0(2577) DEBUG: tm_shutdown : start 0(2577) DEBUG: tm_shutdown : empting hash table 0(2577) DEBUG: tm_shutdown: releasing timers 0(2577) DEBUG: tm_shutdown : removing semaphores 0(2577) DEBUG: tm_shutdown : done 0(2577) shm_mem_destroy 0(2577) destroying the shared memory lock ---------------------------------------------------------------- I couldn't find what is wrong in my configuration... BTW, I can add, remove and edit users in MySQL Database using "serctl" subscribers commands (that means that my SER machine is talking with MySQL server as in expected)

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[Serusers] breaking down ser.cfg
by Iqbal 27 Jun '05

27 Jun '05

Hi My ser config seems to be growing, in fact its getting way too big, and I am beginning to wonder if there is a better way of creating the config. I have got the basic per METHOD routes Then I have got dialing plans covering alot of prefixes, which means there alot of checks be MESSAGE Then I have flags being set based on user type and what group he is in Add to this the authentication checks in each message dialogue, and slowly its getting out of control. Then we have natting to deal with also, Oh add voicemail also, and a few more customer types which need to be routed elsewhere. Everytime I change a route, or want to alter things, I have got to step through alot of the config, just to make sure I havent broken a route, or allowed someone to call they shouldn't. Has anyone come up with a cool way of doing this..or should I say a cleaner way. I was wondering if we could breal ser.cfg into lots of smaller files, eg one which die the auth, and then one per message type all of them could be included with each other, i.e I would include the auth file as a header to each of the METHOD (eg INVITE, REGISTER) files the I could have routing files, , which could also be included and have routes per pacjage type, or user type (i.e have as many as I want, each for its own cause). So the main ser.cfg would be like ser.cfg -------- include=>general.cfg include=>invite.cfg etc etc and invite.cfg would be invite.cfg ----------- include =>auth.cfg I just think then adding things like voicemail, and other features might be easier If anyone has done this, or works, would like to hv feedback. Or if anyone has a file in excess of 1600lines again setup layout ideas would be good Iqbal

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[Serusers] Duplicate SIP messages?
by Tim Pushor 27 Jun '05

27 Jun '05

Hi Friends, I am trying to learn ser and wrap my head around the routing logic. My first project is a simple outbound proxy to handle SIP/RTP from an SPA2000 behind a NAT. The ser server is on the public Internet, but I am having trouble making it work :( The spa2000 has been reset to default, and basically setup the same way that I'd set it up for FWD behind nat). I am using an example config from the Internet as as starting point (included below) and running on Ser 0.8.14 on FreeBSD (from a port). I am using the RTP proxy from ser cvs. This , It almost works, and capturing the traffic with ethereal looks to be mostly correct, but I am seeing duplicate sip messages (plz excuse formatting), which I think is causing me a big problem (even if it isn't, I'd like to know why this is happening). I didn't include frames 1 and 2, they are a SIP keepalive. 3 is the request from the spa2000 to the proxy 4 is the response from the proxy 5 is the request from the proxy to the itsp 6 is a dup! And then the problem compounds as the itsp tries to connect the same call twice. I would be very appreciative of any advice from you veterans ;-) Thanks, Tim ** Doctored trace - 207.46.199.15 is the address of the NAT - 207.46.199.14 is the address of the Proxy running ser - 69.16.138.164 is the address of my itsp's SIP proxy 3 4.563455 207.46.199.15 207.46.199.14 SIP/SDP Request: INVITE sip:6415551234@my.itsp.com, with session description 4 4.564809 207.46.199.14 207.46.199.15 SIP Status: 100 trying -- your call is important to us 5 4.566539 207.46.199.14 69.16.138.164 SIP/SDP Request: INVITE sip:6415551234@my.itsp.com, with session description 6 4.578979 207.46.199.14 69.16.138.164 SIP/SDP Request: INVITE sip:6415551234@my.itsp.com, with session description 7 4.589856 69.16.138.164 207.46.199.14 SIP Status: 100 Trying 8 4.602580 69.16.138.164 207.46.199.14 SIP Status: 407 Proxy Authorization Required 9 4.602733 207.46.199.14 69.16.138.164 SIP Request: ACK sip:6415551234@my.itsp.com 10 4.602808 207.46.199.14 207.46.199.15 SIP Status: 407 Proxy Authorization Required 11 4.611428 69.16.138.164 207.46.199.14 SIP Status: 407 Proxy Authorization Required 12 4.611574 207.46.199.14 69.16.138.164 SIP Request: ACK sip:6415551234@my.itsp.com 13 4.613772 207.46.199.15 207.46.199.14 SIP Request: ACK sip:6415551234@my.itsp.com 14 4.622070 207.46.199.15 207.46.199.14 SIP/SDP Request: INVITE sip:6415551234@my.itsp.com, with session description 15 4.623428 207.46.199.14 207.46.199.15 SIP Status: 100 trying -- your call is important to us 16 4.625164 207.46.199.14 69.16.138.164 SIP/SDP Request: INVITE sip:6415551234@my.itsp.com, with session description 17 4.648612 69.16.138.164 207.46.199.14 SIP Status: 100 Trying 18 7.101641 69.16.138.164 207.46.199.14 SIP Status: 180 Ringing 19 7.101844 207.46.199.14 207.46.199.15 SIP Status: 180 Ringing 20 7.601009 69.16.138.164 207.46.199.14 SIP Status: 180 Ringing 21 7.601206 207.46.199.14 207.46.199.15 SIP Status: 180 Ringing 22 8.859386 69.16.138.164 207.46.199.14 SIP Status: 180 Ringing 23 8.859577 207.46.199.14 207.46.199.15 SIP Status: 180 Ringing .... .... My config: # ----------- global configuration parameters ------------------------ fork=no log_stderror=yes debug=7 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=1 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # !! Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); if (!( search ("^Content-Length:\ 0") )) { force_rtp_proxy(); } # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; }

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[Serusers] ser.cfg Configuration
by Abdul Lateef 27 Jun '05

27 Jun '05

Hello All, I installed successfully SER on my Linux box. now I want to modify the ser.cfg file to make a test call using any SIP IP Phone. I have some little confusion, Where I should register the Gatekeeper and where I should register the IP Phone. because I am new in this field so first I want to work without any database. For the example my Gatekeeper information is: xxx.xxx.xxx.xxx uid = test pws = tsip Where I should make this entry, so if the IP Phone is going to call any PSTN the ser should be send the singnal to the gatekeeper. I will be really appreciate if any can give me a simple file of ser.cfg Thank You Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

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[Serusers] serweb
by Sebastian Kühner 27 Jun '05

27 Jun '05

Hello, Ich reinstalled my Linux Machine (all because of serweb) and I still have the problem that appears "The page cannot be displayed" when I call the admin/index.php - File. No one of you had the same problem? I don't find anything in the internet about that... Thanks. Sebastian

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[Serusers] natted sessions limited by hardware
by Iqbal 27 Jun '05

27 Jun '05

Hi Has anyone come across any limits set by routers on RP sessions, natted session etc, i seem to be hitting occasional problems at clienst when they hit more than 3 phones, the 4th will work, but eventually one of the group will drop out of the loop Iqbal

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[Serusers] *** SER - Asterisk
by lucape＠inwind.it 27 Jun '05

27 Jun '05

hello help me to configure ser + asterisk how to do the configuration? Luca ____________________________________________________________ 6X velocizzare la tua navigazione a 56k? 6X Web Accelerator di Libero! Scaricalo su INTERNET GRATIS 6X http://www.libero.it

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[Serusers] 3 seconds to connect?
by Chia Huey Lim 27 Jun '05

27 Jun '05

Just wondering, if party A dial to party B, it takes 3 seconds to hear the first ring. Is it normal or it's too long? Regards, Chia

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[Users] wich way to go?
by Jimmy Montano 27 Jun '05

27 Jun '05

Greetings all We'been using iptel's SER to offer voip services to our network users. Now with the advent of OpenSER, we are not sure wich is the best way to go. Any advantages/reasons to go with OpenSER and no Iptel's? thanks in advance

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[Serusers] NAT problem
by Bernd Froemel 27 Jun '05

27 Jun '05

Dear list, I have some wired problem between users behind the same NAT connected to a multihomed ser server (public&private IP). Ser is listening only on the public interface (ser.ip.address). The NATed clients have official.ip.address as their official IP address and 192.168.10.1 and 192.168.10.2 are their IP addresses behind NAT. Now after a register I get for the NATed clients the following entries into the location table: UA1: contact: sip:test@192.168.10.1:2050;transport=tcp;line=1hzwxw3z received: sip:official.ip.address:2050;transport=TCP flags: 1 UA2: contact: sip:test2@192.168.10.2:2050;transport=tcp;line=1hzwxw3z received: sip:official.ip.address:2050;transport=TCP flags: 1 which looks quite good (why is the flag 1 and not 6?), but then on INVITE/SUBSCRIBE/ (everthing which issues a t_relay) my logs get full of: ser[27588]: ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused ser[27588]: ERROR: tcpconn_connect: tcp_blocking_connect failed ser[27588]: ERROR: tcp_send: connect failed ser[27588]: msg_send: ERROR: tcp_send failed ser[27588]: ERROR: t_forward_nonack: sending request failed and the UA which issued the command receives a: 477 Unfortunately error on sending to next hop occurred message. I think it has something to do with the lookup of the target UA - at least I always get the uri back which contains the private IP. (debugging output: befor lookup 80.123.216.181 - - sip:test@domain.com after lookup 80.123.216.181 - - sip:test@192.168.10.1:2050;transport=tcp;line=lhzwxw3z ) Also I guess that the commented part which is uncommented in the default cfg shouldn't prevent all NAT calls, but only calls to real private IPs. ( found in route[1] beginning, the nat route) #if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && ! search("^Route:")) #{ # sl_send_reply("479", "forward request to private ip address denied"); # break; #}; ) But to due the lookup returning me a private IP uri this would result in a 479 error. My current test config is basically the one found in cvs/etc/ser.cfg.m4 with inserted values. I already tried other cfgs - even the example in modules/nathelper -- no success. Yes rtpproxy is running, I can even call the other UA and audio is working full duplex, but only at the very beginning. After a few minutes idleing or a server restart, without clearing the sql location table a call results always in the 477 error. Please help me - what am I missing here? Thanks in advance, Bernd ----- Here my cfg and some SIP messages during REGISTER and INVITE (I have removed IPs and unnecessary parts, Asterisk gw is 192.168.xx.xx): # # ----------- global configuration parameters ------------------------ [...] check_via=no # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=no # (cmd. line: -R) mhomed=1 [...] # ------------------ module loading ---------------------------------- [...] # ----------------- setting module-specific parameters --------------- [...] modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("acc", "report_ack", 1) modparam("acc", "log_level", 1) #if BYE fails (telephone is dead, record-routing broken, etc.), generate #a report nevertheless -- otherwise we would have no STOP event; => 1 modparam("acc", "failed_transactions", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) modparam("acc", "log_missed_flag", 3) modparam("acc", "db_missed_flag", 3) modparam("usrloc", "db_mode", 0) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("usrloc", "timer_interval", 10) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth", "nonce_expire", 300) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_timer", 20) modparam("tm", "fr_inv_timer", 90) modparam("tm", "wt_timer", 20) #modparam("enum", "domain_suffix", "e164.arpa.") modparam("msilo", "registrar", "sip:registrar@xxxxxxxx") alias=domain1.com alias=domain2.com /* flags: 1 ... ACCOUNT 3 ... MISSED CALLS 4 ... VOICEMAIL 6 ... NAT */ # ------------------------- request routing logic ------------------- # main routing logic route { # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","too many hops (loop?)"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "message too large"); break; }; if (nat_uac_test("3")) { if (method=="REGISTER" || !search("^Record-Route:")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; if (method=="INVITE") { log(1,"natted caller sent invite!\n"); fix_nated_sdp("1"); }; force_rport(); setflag(6); log(1, "natted caller detected\n"); append_to_reply("P-NATed-Caller: Yes\r\n"); } else { log(1, "something wrong here..\n"); }; }; #antispam if ( search("(From|F):.*@(domain1\.com)|(domain2\.com)|(192\.168\.xx \.xx)") ) { if ( (method=="INVITE" || method=="SUBSCRIBE") && !(src_ip == SER_IP || src_ip == 192.168.xx.xx) ) { if (!(proxy_authorize("","subscriber"))) { proxy_challenge("","0"); break; }; if (!check_from()) { log("LOG: From Cheating attempt in INVITE!\n"); sl_send_reply("403", "use From=id"); break; }; #consume_credentials(); }; #non-REGISTER from other domain } else if ((method=="INVITE" || method=="SUBSCRIBE" || method=="REGISTER") && !(uri==myself || uri=~"(@(192\.168\.xx \.xx)([;:].*)*)") ) { sl_send_reply("403", "no relaying"); break; }; if (!(method=="REGISTER")) { record_route(); }; #if (method=="BYE" || method=="CANCEL") #{ # unforce_rtp_proxy(); #}; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if ((method=="INVITE" || method=="ACK" || method=="CANCEL") && uri=~"(@(192\.168\.xx\.xx)([;:].*)*)") { route(4); # to asterisk } else { append_hf("P-hint: rr-enforced\r\n"); if (method=="BYE") { setflag(1); }; log(1, "and directly to nat..\n"); route(1); # to nat }; break; }; if (!(uri==myself || uri=~"(@(192\.168\.xx\.xx)([;:].*)*)")) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); log(1, "outbound\n"); route(1); # to nat break; }; # ---->request is for our domains!<---- # if (method=="REGISTER") { if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { log("LOG: To Cheating attempt\n"); sl_send_reply("403", "use From=id"); break; }; log(1, "(un)register successful\n"); if (!save("location")) { sl_reply_error(); }; m_dump(); break; }; if (uri=~"sip:daemon@") { sl_send_reply("410", "daemon is gone"); break; }; lookup("aliases"); if (!(uri==myself || uri=~"(@(192\.168\.xx\.xx)([;:].*)*)")) { append_hf("P-hint: ALIASED-OUTBOUND\r\n"); route(1); #to nat break; }; if (uri=~"^[a-zA-Z]+:\+[0-9]+@") { strip(1); prefix("00"); }; if (!does_uri_exist()) { if(uri=~"^[a-zA-Z]+:[0-9]+@") { route(4); #to pstn } else { sl_send_reply("604", "does not exist anywhere"); }; break; }; if (is_user_in("Request-URI", "voicemail")) { setflag(4); }; exec_msg("echo befor lookup $SIP_SRCIP - $SIP_ORURI - $SIP_RURI >> /tmp/ser.log"); if (!lookup("location")) { log(1, "lookup failed\n"); route(6); break; } else { log(1, "lookup successful\n"); }; exec_msg("echo after lookup $SIP_SRCIP - $SIP_ORURI - $SIP_RURI >> /tmp/ser.log"); if (uri=~"(@(192\.168\.xx\.xx)([;:].*)*)") { log(1, "LOG: Gateway address in UsrLoc!\n"); route(4); # to PSTN break; }; if (method=="INVITE" && isflagset(4)) { t_on_failure("1"); }; setflag(3); append_hf("P-hint: USRLOC\r\n"); log(1, "nearly at end and going to nat..\n"); exec_msg("echo $SIP_SRCIP - $SIP_ORURI - $SIP_RURI >> /tmp/ser.log"); route(1); # to nat } route[1] { #if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && ! search("^Route:")) #{ # sl_send_reply("479", "forward request to private ip address denied"); # break; #}; if (isflagset(6)) { if(!is_present_hf("P-RTP-Proxy")) { force_rtp_proxy(); append_hf("P-RTP-Proxy: Yes\r\n"); log(1, "rtp proxied\n"); }; log(1,"natted calee\n"); append_hf("P=NATed-Calee: Yes\r\n"); }; exec_msg("echo :: $SIP_SRCIP - $SIP_ORURI - $SIP_RURI >> /tmp/ser.log"); log(1, "1\n"); t_on_reply("1"); log(1, "2\n"); if (!t_relay()) { sl_reply_error(); break; }; log(1, "3\n"); } route[4] { [...] } onreply_route[1] { log(1, "taking onreply route\n"); if(isflagset(6) && status=~"(183)|2[0-9][0-9]" && !search("^Content-Length:\ +0")) { log(1,"onreply fixing nat\n"); fix_nated_contact(); force_rtp_proxy(); } else if (nat_uac_test("1")) { log (1, "onreply fixing nat alternate\n"); fix_nated_contact(); }; } route[4] { [...] } route[6] { [...] } ----------- REGISTER ----------- T official.ip.address:2062 -> ser.ip.address:5060 [A] REGISTER sip:domain.com SIP/2.0..Via: SIP/2.0/TCP 192.168.10.1:2062;br anch=z9hG4bK-wekq229vr3vn;rport..From: "Test" <sip:test@domain.com>;tag=3q2blh64wf..To: "Test" <sip:test@domain.com>..Call-ID: 3c26818830d4-ugwagbaz5vkz@snom360..CSeq: 22 REGISTER..Max- Forwards: 70..Contact: <sip:test@192.168.10.1:2062;transport=tcp;line= lhzwxw3z>;q=1.0; +sip.instance="<urn:uuid:0c541696-09aa-4d0c-b8ca-fb9889cc61ad>" ;audio;mobility="fixed";duplex="full";description="snom360";actor="principal"; events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY ,SUBSCRIBE,PRACK,MESSAGE,INFO"..User-Agent: snom360/3.60k..Supported: gruu. .Allow-Events: dialog..X-Real-IP: 192.168.0.191..WWW-Contact: <http://192.168.10.1:80>..WWW-Contact: <https://192.168.10.1:443>..Expires: 3600..Content-Length: 0 T ser.ip.address:5060 -> official.ip.address:2062 [AP] SIP/2.0 401 Unauthorized..Via: SIP/2.0/TCP 192.168.10.1:2062;branch=z9hG4b K-wekq229vr3vn;rport=2062;received=official.ip.address..From: "Test" <sip:test@domain.com>;tag=3q2blh64wf..To: "Test" <sip: test(a)domain.com>;tag=5431d75005d8ed216f7c100a44746400.19d5..Call-ID : 3c26818830d4-ugwagbaz5vkz@snom360..CSeq: 22 REGISTER..P-NATed-Caller: Yes ..WWW-Authenticate: Digest realm="domain.com", nonce="42baac51779c17ebe ec20a5ee2f9492821bd723e"..Server: Sip EXpress router (0.9.3 (i386/linux)).. Content-Length: 0..Warning: ser.ip.address:5060 "Noisy feedback tells: p id=27776 req_src_ip=official.ip.address req_src_port=2062 in_uri=sip:domain.com out_uri=sip:domain.com via_cnt==1" T official.ip.address:2062 -> ser.ip.address:5060 [A] REGISTER sip:domain.com SIP/2.0..Via: SIP/2.0/TCP 192.168.10.1:2062;br anch=z9hG4bK-6s4mq8dda681;rport..From: "Test" <sip:test@domain.com>;tag=3q2blh64wf..To: "Test" <sip:test@domain.com>..Call-ID: 3c26818830d4-ugwagbaz5vkz@snom360..CSeq: 23 REGISTER..Max- Forwards: 70..Contact: <sip:test@domain.com:2062;transport=tcp;line= lhzwxw3z>;q=1.0; +sip.instance="<urn:uuid:0c541696-09aa-4d0c-b8ca-fb9889cc61 ad>";audio;mobility="fixed";duplex="full";description="snom360"; actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY ,SUBSCRIBE,PRACK,MESSAGE,INFO"..User-Agent: snom360/3.60k..Supported: gruu. .Allow-Events: dialog..X-Real-IP: 192.168.10.1..WWW-Contact: <http://192.1 68.10.1:80>..WWW-Contact: <https://192.168.10.1:443>..Authorization: Digestusername="test",realm="domain.com",nonce="42baac51779c17ebeec20a 5ee2f9492821bd723e",uri="sip:domain.com",response="d2415424805014aac504 b643ea489795",algorithm=md5..Expires: 3600..Content-Length: 0.... T ser.ip.address:5060 -> official.ip.address:2062 [AP] SIP/2.0 200 OK..Via: SIP/2.0/TCP 192.168.10.1:2062;branch=z9hG4bK-6s4mq8dd a681;rport=2062;received=official.ip.address..From: "Test" <sip:test@domain.com>;tag=3q2blh64wf..To: "Test" <sip:test@domain.com>;tag=5431d75005d8ed216f7c100a44746400.d02e..Call-ID: 3c268188 30d4-ugwagbaz5vkz@snom360..CSeq: 23 REGISTER..P-NATed-Caller: Yes..Contact: <sip:test@192.168.10.1:2056;transport=tcp;line=lhzwxw3z>;q=1;expires= 2204;received="sip:official.ip.address:2056;transport=TCP", <sip:test@192.168.10.1:2055;transport=tcp;line=lhzwxw3z>;q=1;expires=1551; received="sip:official.ip.address:2055;transport=TCP", <sip:test@192.168.10.1:2062;transport=tcp;line=lhzwxw3z>;q=1;expires=3600; received="sip:official.ip.address:2062;transport=TCP"..Server: Sip EXpress router (0.9.3 (i386/linux))..Content-Length: 0.. Warning: 392 ser.ip.address:5060 "Noisy feedback tells: pid=27776 req_ src_ip=official.ip.address req_src_port=2062 in_uri=sip:domain.com out_uri=sip:domain.com via_cnt==1" ----------- INVITE ----------- T official.ip.address:33255 -> ser.ip.address:5060 [AP] INVITE sip:test@domain.com SIP/2.0..Via: SIP/2.0/TCP 192.168.10.2; branch=z9hG4bK3AA2CBD1;alias..CSeq: 2681 INVITE..To: <sip:test@domain .com>..Content-Type: application/sdp..From: "test2" <sip:test2@domain.com >;tag=67807379..Call-ID: 1766830276@192.168.10.2..Subject: sip:test @domain.com..Content-Length: 230..User-Agent: kphone/4.1.1..Contac t: "test" <sip:test@192.168.10.2;transport=tcp>....v=0..o=username 0 0 IN IP4 192.168.10.2..s=The Funky Flow..c=IN IP4 192.168.10.2..t=0 0..m=audio 32874 RTP/AVP 0 97 8 3..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpma p:8 PCMA/8000..a=rtpmap:97 iLBC/8000..a=fmtp:97 mode=30.. ## T ser.ip.address:5060 -> official.ip.address:33255 [AP] SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/TCP 192.168.10.2;b ranch=z9hG4bK3AA2CBD1;alias;rport=33255;received=official.ip.address..CSeq: 2681 INVITE..To: <sip:test@domain.com>;tag=5431d75005d8ed216f7c100a4474 6400.5110..From: "test" <sip:test2@domain.com>;tag=67807379..Call-I D: 1766830276@192.168.10.2..P-NATed-Caller: Yes..Proxy-Authenticate: Diges t realm="domain.com", nonce="42baaf784cd57486fa11fe10929ade10b8fc4 3ec"..Server: Sip EXpress router (0.9.3 (i386/linux))..Content-Length: 0..W arning: 392 ser.ip.address:5060 "Noisy feedback tells: pid=27776 req_src_ip= official.ip.address req_src_port=33255 in_uri=sip:test@domain.com out_ur i=sip:test@domain.com via_cnt==1".... ## T official.ip.address:33255 -> ser.ip.address:5060 [AP] ACK sip:test@domain.com SIP/2.0..Via: SIP/2.0/TCP 192.168.10.2;bra nch=z9hG4bK3AA2CBD1;alias..CSeq: 2681 ACK..To: <sip:test@domain.com >;tag=5431d75005d8ed216f7c100a44746400.5110..From: "test2" <sip:test2@ domain.com>;tag=67807379..Call-ID: 1766830276@192.168.10.2..Content-Leng th: 0..User-Agent: kphone/4.1.1..Contact: "test" <sip:test@192.168.10.2;tr ansport=tcp>.... ## T official.ip.address:33255 -> ser.ip.address:5060 [AP] INVITE sip:test@domain.com SIP/2.0..Via: SIP/2.0/TCP 192.168.10.2; branch=z9hG4bK36EBCF00;alias..CSeq: 2682 INVITE..To: <sip:test@domain.com >..Proxy-Authorization: Digest username="test2(a)domain.com", r ealm="domain.com", nonce="42baaf784cd57486fa11fe10929ade10b8fc43ec ", uri="sip:test@domain.com", cnonce="abcdefghi", nc=00000001, resp onse="2c088d59cb24c70c61d890245fe0f5ca", opaque="", algorithm="MD5"..Conten t-Type: application/sdp..From: "test2" <sip:test2@domain.com>;tag=67 807379..Call-ID: 1766830276@192.168.10.2..Subject: sip:test2@domain.com ..Content-Length: 230..User-Agent: kphone/4.1.1..Contact: "test" <sip: test(a)192.168.10.2;transport=tcp>....v=0..o=username 0 0 IN IP4 192.168.0.1 03..s=The Funky Flow..c=IN IP4 192.168.10.2..t=0 0..m=audio 32874 RTP/AVP 0 97 8 3..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8 PCMA/8000.. a=rtpmap:97 iLBC/8000..a=fmtp:97 mode=30.. ## T ser.ip.address:5060 -> official.ip.address:33255 [AP] SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/TCP 192.16 8.10.2;branch=z9hG4bK36EBCF00;alias;rport=33255;received=official.ip.address..C Seq: 2682 INVITE..To: <sip:test@domain.com>..From: "test" <sip:test @domain.com>;tag=67807379..Call-ID: 1766830276@192.168.10.2..P-NA Ted-Caller: Yes..Server: Sip EXpress router (0.9.3 (i386/linux))..Content-L ength: 0..Warning: 392 ser.ip.address:5060 "Noisy feedback tells: pid=27776 req_src_ip=official.ip.address req_src_port=33255 in_uri=sip:test@domain.com out_uri=sip:test@192.168.10.1:2056;transport=tcp;line=lhzwxw3z vi a_cnt==1".... ## T ser.ip.address:5060 -> official.ip.address:33255 [AP] SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM)..V ia: SIP/2.0/TCP 192.168.10.2;branch=z9hG4bK36EBCF00;alias;rport=33255;rece ived=official.ip.address..CSeq: 2682 INVITE..To: <sip:test@domain.com>;t ag=a0de3507a8823f96a254cc0a187acbf0-2573..From: "test2" <sip:test2@domain.com >;tag=67807379..Call-ID: 1766830276@192.168.10.2..P-NATed-Caller: Yes..Server: Sip EXpress router (0.9.3 (i386/linux))..Content-Length: 0..W arning: 392 ser.ip.address:5060 "Noisy feedback tells: pid=27776 req_src_ip= official.ip.address req_src_port=33255 in_uri=sip:test@domain.com out_ur i=sip:test@192.168.10.1:2056;transport=tcp;line=lhzwxw3z via_cnt==1".. .. ## T official.ip.address:33255 -> ser.ip.address:5060 [AP] ACK sip:test@domain.com SIP/2.0..Via: SIP/2.0/TCP 192.168.10.2;bra nch=z9hG4bK36EBCF00;alias..CSeq: 2682 ACK..To: <sip:test@domain.com >;tag=a0de3507a8823f96a254cc0a187acbf0-2573..From: "test2" <sip:test2@domain .com>;tag=67807379..Call-ID: 1766830276@192.168.10.2..Content-Leng th: 0..User-Agent: kphone/4.1.1..Contact: "test" <sip:test@192.168.10.2;tr ansport=tcp>.... #

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[Serusers] SER Windows messenger on-line problem
by Kissandrakis S. George 27 Jun '05

27 Jun '05

Hello, I have installed SER 0.8.14 with radius-mysql authentication besides the default routing, alias=ipv6.uoc.gr ....... if (uri==~"sip:ipv6.uoc.gr") { i have if ( uri=~"sip:[0-9][0-9][0-9][0-9]") { forward (MY-GW-IP,5060); }; i have two accounts in radius kissand(a)ipv6.uoc.gr kissand2(a)ipv6.uoc.gr and with windows messenger 5.1 i go "online" and i can make calls with 4-digit numbers with both accounts If i go "online" with kissand(a)ipv6.uoc.gr and kissand2(a)ipv6.uoc.gr (registered normally) from different computers, kissand(a)ipv6.uoc.gr sees offline kissand2(a)ipv6.uoc.gr and vise versa with ngrep i get SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 147.52.250.40:7563..From: "kissand2(a)ipv6.uoc.gr" <sip:kissand2 @ipv6.uoc.gr>;tag=57126e7b464c4f4498de2e1882dd41ea;epid=3d3d4dcaec..To: <sip:kissand@ipv6.uoc.gr>;tag=baeb 3f9382bd3485fa08dc217d881d32.dbf8..Call-ID: d3e09061703b477a8e927df4c622ff8a..CSeq: 1 SUBSCRIBE..Server: S ip EXpress router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 147.52.3.244:5060 "Noisy feedback tells: pid=16582 req_src_ip=147.52.3.244 req_src_port=5060 in_uri=sip:kissand@ipv6.uoc.gr out_uri=sip:ki ssand(a)ipv6.uoc.gr via_cnt==71".... SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 147.52.250.62:13759..From: "kissand(a)ipv6.uoc.gr" <sip:kissand@ ipv6.uoc.gr>;tag=5c3c36c6e10640f28463ba634d78d5b0;epid=2fd193f505..To: <sip:kissand2@ipv6.uoc.gr>;tag=baeb 3f9382bd3485fa08dc217d881d32.170e..Call-ID: e0b1e57ea151409fbc29aaaeb6c83b4a..CSeq: 1 SUBSCRIBE..Server: S ip EXpress router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 147.52.3.244:5060 "Noisy feedback tells: pid=16582 req_src_ip=147.52.3.244 req_src_port=5060 in_uri=sip:kissand2@ipv6.uoc.gr out_uri=sip:k issand2(a)ipv6.uoc.gr via_cnt==71".... debbuging SER i get 0(16582) build_res_from_sip_res: copied size: orig:87, new: 27, rest: 599 msg=SIP/2.0 483 Too Many Hops^M Via: SIP/2.0/UDP 147.52.3.244;branch=z9hG4bK806e.6a590d4.0^M Via: SIP/2.0/UDP 147.52.250.62:13759^M From: "kissand(a)ipv6.uoc.gr" <sip:kissand@ipv6.uoc.gr>;tag=5c3c36c6e10640f28463ba634d78d5b0;epid=2fd193f505^M To: <sip:kissand2@ipv6.uoc.gr>;tag=baeb3f9382bd3485fa08dc217d881d32.170e^M Call-ID: e0b1e57ea151409fbc29aaaeb6c83b4a^M CSeq: 1 SUBSCRIBE^M Server: Sip EXpress router (0.8.14 (i386/linux))^M Content-Length: 0^M Warning: 392 147.52.3.244:5060 "Noisy feedback tells: pid=16582 req_src_ip=147.52.3.244 req_src_port=5060 in_uri=sip:kissand2@ipv6.uoc.gr out_uri=sip:kissand2@ipv6.uoc.gr via_cnt==71"^M sipsak gives the same results My sip server's ip is 147.52.3.244 resolves to edet.ucnet.uoc.gr SRV record fir _sip._upd.ipv6.uoc.gr resolves to 5060 edet.ucnet.uoc.gr Is it a problem that the server is not in ipv6.uoc.gr DNS domain? If not any hints about "Too Many Hops" between users? Thank you -- Kissandrakis S. George kissand(a)ucnet.uoc.gr NOC - System Administrator University of Crete

1 0

[Serusers] "ser-0.9.0.GettingStarted.1.1" Setup Problem !!
by Freeman 27 Jun '05

27 Jun '05

Hi, I have some problem when i tried to setup "ser-0.9.0.GettingStarted.1.1.tar.gz" in Redhat 9, i get some error detail as below : how can i setup in RedHat 9 ? ========== make_and_install_output.txt ========= Makefile.rules:81: action.d: file or folder not found. Makefile.rules:81: crc.d: file or folder not found. Makefile.rules:81: daemonize.d: file or folder not found. Makefile.rules:81: data_lump.d: file or folder not found. Makefile.rules:81: data_lump_rpl.d: file or folder not found. Makefile.rules:81: dprint.d: file or folder not found. Makefile.rules:81: dset.d: file or folder not found. Makefile.rules:81: error.d: file or folder not found. Makefile.rules:81: fifo_server.d: file or folder not found. Makefile.rules:81: flags.d: file or folder not found. Makefile.rules:81: forward.d: file or folder not found. Makefile.rules:81: hash_func.d: file or folder not found. Makefile.rules:81: ip_addr.d: file or folder not found. Makefile.rules:81: main.d: file or folder not found. Makefile.rules:81: md5.d: file or folder not found. Makefile.rules:81: md5utils.d: file or folder not found. Makefile.rules:81: modparam.d: file or folder not found. Makefile.rules:81: msg_translator.d: file or folder not found. Makefile.rules:81: pass_fd.d: file or folder not found. Makefile.rules:81: proxy.d: file or folder not found. Makefile.rules:81: qvalue.d: file or folder not found. Makefile.rules:81: re.d: file or folder not found. Makefile.rules:81: receive.d: file or folder not found. Makefile.rules:81: resolve.d: file or folder not found. Makefile.rules:81: route.d: file or folder not found. Makefile.rules:81: route_struct.d: file or folder not found. Makefile.rules:81: script_cb.d: file or folder not found. Makefile.rules:81: socket_info.d: file or folder not found. Makefile.rules:81: sr_module.d: file or folder not found. Makefile.rules:81: stats.d: file or folder not found. Makefile.rules:81: tcp_main.d: file or folder not found. Makefile.rules:81: tcp_read.d: file or folder not found. Makefile.rules:81: timer.d: file or folder not found. Makefile.rules:81: tsend.d: file or folder not found. Makefile.rules:81: udp_server.d: file or folder not found. Makefile.rules:81: unixsock_server.d: file or folder not found. Makefile.rules:81: usr_avp.d: file or folder not found. Makefile.rules:81: mem/f_malloc.d: file or folder not found. Makefile.rules:81: mem/mem.d: file or folder not found. Makefile.rules:81: mem/memtest.d: file or folder not found. Makefile.rules:81: mem/q_malloc.d: file or folder not found. Makefile.rules:81: mem/shm_mem.d: file or folder not found. Makefile.rules:81: mem/vq_malloc.d: file or folder not found. Makefile.rules:81: parser/hf.d: file or folder not found. Makefile.rules:81: parser/msg_parser.d: file or folder not found. Makefile.rules:81: parser/parse_allow.d: file or folder not found. Makefile.rules:81: parser/parse_content.d: file or folder not found. Makefile.rules:81: parser/parse_cseq.d: file or folder not found. Makefile.rules:81: parser/parse_disposition.d: file or folder not found. Makefile.rules:81: parser/parse_diversion.d: file or folder not found. Makefile.rules:81: parser/parse_event.d: file or folder not found. Makefile.rules:81: parser/parse_expires.d: file or folder not found. Makefile.rules:81: parser/parse_fline.d: file or folder not found. Makefile.rules:81: parser/parse_from.d: file or folder not found. Makefile.rules:81: parser/parse_hname2.d: file or folder not found. Makefile.rules:81: parser/parse_hostport.d: file or folder not found. Makefile.rules:81: parser/parse_methods.d: file or folder not found. Makefile.rules:81: parser/parse_nameaddr.d: file or folder not found. Makefile.rules:81: parser/parse_param.d: file or folder not found. Makefile.rules:81: parser/parser_f.d: file or folder not found. Makefile.rules:81: parser/parse_rpid.d: file or folder not found. Makefile.rules:81: parser/parse_rr.d: file or folder not found. Makefile.rules:81: parser/parse_sipifmatch.d: file or folder not found. Makefile.rules:81: parser/parse_to.d: file or folder not found. Makefile.rules:81: parser/parse_uri.d: file or folder not found. Makefile.rules:81: parser/parse_via.d: file or folder not found. Makefile.rules:81: parser/digest/digest.d: file or folder not found. Makefile.rules:81: parser/digest/digest_parser.d: file or folder not found. Makefile.rules:81: parser/digest/param_parser.d: file or folder not found. Makefile.rules:81: parser/contact/contact.d: file or folder not found. Makefile.rules:81: parser/contact/parse_contact.d: file or folder not found. Makefile.rules:81: db/db.d: file or folder not found. Makefile.rules:81: db/db_fifo.d: file or folder not found. Makefile.rules:81: lex.yy.d: file or folder not found. Makefile.rules:81: cfg.tab.d: file or folder not found. bison -d -b cfg cfg.y cfg.y contains 1 shift/reduce conflict. flex cfg.lex ................... make: *** [usr_avp.o] Error 1 ................... ========== make_and_install_output.txt ========= Tks a Lot ! Freemane 芺笛戙菑穻

3 8

[Serusers] ser-mysql accounting
by Szabolcs Szasz 27 Jun '05

27 Jun '05

Hello I have installed SER on my redhat box. I have a problem with ser-mysql accounting. In the log is apearing the acc information, but it doesn't write into the mysql database? It looks like this when I make a call from 506029 to 506030 : 0(24195) ------------------------------------------- 0(24195) entering route[1] - relaying SIP message 0(24195) relaying message ... 0(24195) ACC: transaction answered: method=INVITE, i-uri=sip:506030@10.1.10.10:5060;user=phone, o-uri=sip:506030@10.1.10.105:5060, call_id=58dcd4-a010a68-13c4-119d-44d0f9-247(a)10.1.10.10, from=<sip:506029@10.1.10.10> ;tag=a010a68-13c4-119d-44d0f9-3b9d, code=200 0(24195) ------------------------------------------- 0(24195) entering main loop 0(24195) ACK message received 0(24195) ------------------------------------------- 0(24195) entering route[1] - relaying SIP message 0(24195) relaying message ... 0(24195) ACC: request acknowledged: method=ACK, i-uri=sip:506030@10.1.10.105:5060, o-uri=sip:506030@10.1.10.105:5060, call_id=58dcd4-a010a68-13c4-119d-44d0f9-247(a)10.1.10.10, from=<sip:506029@10.1.10.10> ;tag=a010a68-13c4-119d-44d0f9-3b9d, code=200 0(24195) ------------------------------------------- 0(24195) entering main loop 0(24195) BYE message received 0(24195) ------------------------------------------- 0(24195) entering route[1] - relaying SIP message 0(24195) relaying message ... 0(24195) ACC: transaction answered: method=BYE, i-uri=sip:506029@10.1.10.104:5060, o-uri=sip:506029@10.1.10.104:5060, call_id=58dcd4-a010a68-13c4-119d-44d0f9-247(a)10.1.10.10, from=<sip:506030@10.1.10.10:5060;user=phone> ;tag=a010a69-13c4-11a0-44dbd5-740c, code=200 Can anyone help me ? Thanks Szabolcs

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Re: [serusers] calling problem
by michael p 27 Jun '05

27 Jun '05

Paul, in ser.cfg i have : # Special flag marking contacts behind NAT modparam("registrar", "nat_flag", 6) and after: # Flag 5: NAT flag #=0: Request comes from user with public IP (caller has public IP addre$ #=1: Request comes from user behind NAT (caller is behind NAT) # Flag 6: NAT flag for contacts saved in the location table #=0: The callee has public IP address #=1: The callee is behind NAT why did you tell me to put the flag to 4? is the number of the flag is important or it's not ? do you know why after a specific time some SIP clients are deconnected alone? is there a variable that i can change for that? Best regards, Michael >From: Java Rockx <javarockx(a)gmail.com> >Reply-To: Java Rockx <javarockx(a)gmail.com> >To: michael p <mikep3000(a)hotmail.com>, serusers <serusers(a)lists.iptel.org> >Subject: Re: [serusers] calling problem >Date: Fri, 24 Jun 2005 07:24:19 -0400 > >Michael, > >I'm quite certain that if the value of the "flags" column in the location >table is set to 1 then the SER registrar module thinks that the REGISTER >message is from a NATed client. > >By default, SER uses flag 4 to indicate NAT status in the ser.cfg file. So >if you have called setflag(4) somewhere before save("location") then you >are >informing SER that the client is NATed. > >You can change the flag by using something like this > >modparam("registrar", "nat_flag", 2) > >The above line tells SER to use flag 2 as the NAT indicator. > >Regards, >Paul > >On 6/24/05, michael p <mikep3000(a)hotmail.com> wrote: > > > > it's impossible that it's that > > > > first all my client are behind a NAT and some flags are 1 and others 0 > > > > second i 'm already used a stun server. > > > > > > do you think that's something else? > > > > do you know why after a specific time some SIP clients are deconnected > > alone? > > is there a variable that i can change for that? > > > > thanks > > > > M. > > > > >From: Java Rockx <javarockx(a)gmail.com> > > >Reply-To: Java Rockx <javarockx(a)gmail.com> > > >To: michael p <mikep3000(a)hotmail.com> > > >CC: serusers(a)lists.iptel.org > > >Subject: Re: [serusers] calling problem > > >Date: Thu, 23 Jun 2005 09:02:29 -0400 > > > > > >Flags = 0 indicates that the SIP client is not behind a NAT device. > > > > > >Flags = 1 indicates that the SIP client is behind a NAT device. > > > > > >In order to get audio (ie, RTP streams) working one one or more SIP > > clients > > >is behind a NAT device you must use either rtpproxy or mediaproxy. > > > > > >The Getting Started document at http://onsip.org/ fully describes how >to > > >use > > >both. > > > > > >A third, but IMHO, less appealing option would be to use a STUN server, > > >such > > >as mystun which is availble at sourceforge.net <http://sourceforge.net> >< > > http://sourceforge.net>. > > > > > >Regards, > > >Paul > > > > > >On 6/23/05, michael p <mikep3000(a)hotmail.com> wrote: > > > > > > > > hi > > > > > > > > in my table location of SER i have a field flags, someone knows for > > what > > > > is? > > > > > > > > now i have three users registered and two can call and their flags >are > > >"0" > > > > and one can't call and is flags field is "1" > > > > > > > > is that the reason why it don't work? > > > > > > > > thanks > > > > > > > > M. > > > > > > > > > > > > _______________________________________________ > > > > Serusers mailing list > > > > serusers(a)lists.iptel.org > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > >

2 1

[Serusers] serweb session problem
by Sebastian Kühner 27 Jun '05

27 Jun '05

Hello, I have a big problem with serweb. I'm searching for the solution since hours... but I didn't find it!! I installed serweb according to in the installation guide and the HOWTOs that I found in the internet. The problem is that if I want to open the index.php-page (of admin), always appears the Explorer Window "The page cannot be displayed" and when I look at the access_log of apache2 there appears something like that: GET /serweb/index.php?phplib_Session=d69f77e7ccb0d3b6c7d684bcbc368063&phplib_Ses sion=4c8a0c40e09c1bfa01a91ef5832a026a&phplib_Session=5f354e9a405ade8796a54a3 88b0353db&phplib_Session=769c3fa2b669f1dcdd2291c24227ef5d&phplib_Session=4ca 670bdd9a1f4f20441eac5f96c7b8e&phplib_Session=da194f7b1855c60daf515de13b14dab 2&phplib_Session=9bba333123ff119f94f924aeb0bc4710&phplib_Session=f6c9842d7cf 429c632608ad4ee7645c7&phplib_Session=bf06f602d09a1d313297a00f4b293133&phplib _Session=6d798a6dcc84dac416415312debe47d3&phplib_Session=4d1991912b23d4a59ee 1f29f40830317&phplib_Session=31268d0242558745860dbacf41a1be15&phplib_Session =5255e631f1cde5c708cb985db5ebfcaa&phplib_Session=7020b63c06bd983fb5eb054ac4e f627f&phplib_Session=9651b78d2f391f94f23c8b46c603d0a5&phplib_Session=466c61f 55a87cb7ac4661e2816a3ec9d&phplib_Session=9515dbc324cc048f72fb7e1149166f18&ph plib_Session=e0940531f81730793c23a703d772e738&phplib_Session=8e67ddbe3e69eec 9a952b2a59eed48ea&phplib_Session=621fc501c7590d413a79c58278956677&phplib_Ses sion=a4f2af4b0d2e757e0568ee76f8563262&phplib_Session=70960d0cd7a32896b7fd80a 0a4c7f806&phplib_Session=5b78a5a53455042a992def2ab3012430&phplib_Session=0c8 fb10161429bca54375b4dda5aec51&phplib_Session=b1ddb14986db2740a239b5f3494ae08 3&phplib_Session=e281788f6b7533869817cce01a1ebaeb&phplib_Session=d16de271da5 e37b5de9f81268975330b&phplib_Session=45a453bdc8d3969f392c2a4e230c7e0a&phplib _Session=8cd36450b9f6d0592bfe9864bb4e3782&phplib_Session=db9b2e8ca0094fafc2b 923cd377c558c&phplib_Session=dcd92ee19095fba1c4a1475183b27352&phplib_Session =38e1ab2e6f6a65a1bce8d7602db10014&phplib_Session=b074fa09bae847e186be4c22ed8 280a5&phplib_Session=27e2d0c9e4172bed9752fe20a744195c&phplib_Session=d817e3e 3639c105042228e083f9081dd&phplib_Session=f9ac5312dcfcdf3bade2f80a7d6d98bf&ph plib_Session=5f15600e879327263d4d83f489ec6422&phplib_Session=dde588458177dde 8907b932a4ee8a0fe&phplib_Session=d55285a056e6e5a873735a673d80a7cc&phplib_Ses sion=1c51e86702ef7004638d9488cd967304&phplib_Session=bf5b51775d6208711b6904f eeddf5d7b&phplib_Session=a7940c6ed31b41c83b60174ae05b023d HTTP/1.1" 302 - ... that means that serweb wants to make "thousands" of sessions. I didn't find a solution for that in the internet and I hope that someone of you can help me! Thanks! Sebastian

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