sr-users August 2006

sr-users@lists.kamailio.org

184 participants
360 discussions

by Michel Zenone

Hi! I wonder if there is only a unique bitmap of flags or if there is a bitmap of flags for each SIP dialog? Thanks! Michel

17 years, 10 months

[Users] Using subst() of Textops module

by Leonid Fainshtein

Hi, In the textops module doc (http://openser.org/docs/modules/1.0.x/textops.html#AEN124) there is an example that demonstrates how to replace URI in the header "To" with the message URI: # replace the uri in to: with the message uri (just an example) if ( subst('/^To:(.*)sip:[^@]*@[a-zA-Z0-9.]+(.*)$/t:\1\u\2/ig') ) {}; After I've fixed a small mistake (instead of /t: I put /To:) it worked. But I don't understand how does it work. It looks like "u" is substituted by URI from R-URI but I didn't find such core variable. Could somebody explain me? Thank you in advance, Leonid Fainshtein

17 years, 10 months

[Serusers] Serweb: HTTP 403 Forbidden error

by Roberto Lopes

Dear SER Users, I've a problem with Serweb and I hope somebody could help me out I know this problem was already talked about but i never saw a solution to it, so I'll post it again hoping somebody will know how to solve this mistake. So the problem is that when i try to put serweb working in my firefox browser the following error is shown: Forbidden You don't have permission to access /serweb/index.html on this server. Additionally, a 403 Forbidden error was encountered while trying to use an ErrorDocument to handle the request. ------------------------------ Apache/2.2.0 (Fedora) Server at xxx.xxx.xxx.xxx Port 80 I think i installed everything correctly. I'll show some of the configurations I've made #1 - In "/etc/php.ini" I changed the memory limit from 8 to 12M because of the Serweb extension classkit.so … memory_limit = 12M #2 - I used pear to install classkit.so pear install <path to classkit-0.4.tgz> #3 - In "php.ini" … ;;;;;;;;;;;;;;;;;;;;;; ; Dynamic Extensions ; ;;;;;;;;;;;;;;;;;;;;;; … extension=classkit.so #4 - In "/serweb-0.4/config/config_data_layer.php": … $config->data_sql->host[$i]['host']="<IP MySQL Server>"; … $config->data_sql->host[$i]['user']="<DB User>"; $config->data_sql->host[$i]['pass']="<User password>"; .. #5 - In "/root/instalacao/serweb-0.9.4/config/config_paths.php" $config->root_path="/root/instalacao/serweb- 0.9.4/html/"; #6 - I created a log file touch /var/log/serweb chown apache /var/log/serweb #7 - I created a folder for SIP domain mkdir /root/installation/serweb-0.4/html/domains/xxx.xxx.xxx.xxx(IP of SER) #8 - I copied the files and folder from the default folder ("/root/installation/serweb-0.4/html/domains/_default/") to the one i created. #9 - I specified the SIP domain in the file "/root/installation/serweb-0.4/config/set_domain.php": … $config->domain = "xxx.xxx.xxx.xxx(IP of SER)"; … #10 - I enabled logging in file "/root/installation/serweb-0.4 /config/config.php": … $config->enable_loging = true; $config->log_file = "/var/log/serweb; #11 - In /etc/httpd/conf/httpd.conf I added an alias … <Directory "/var/www/html"> Options Indexes FollowSymLinks AllowOverride None Order deny,allow Allow from all </Directory> <IfModule mod_userdir.c> UserDir disable root #UserDir public_html </IfModule> Alias /serweb/ "/root/instalacao/serweb-0.9.4/html" <Directory "/root/instalacao/serweb-0.9.4/html"> Options Indexes MultiViews Options +FollowSymlinks AllowOverride None Order Deny,Allow Allow from all </Directory> ... #12 - I then put some commands for Apache to begin when computer would start chkconfig httpd on service httpd start #13 - I introduced the following URL http://<IP of SER>/<alias i wrote in httpd.conf>/ http://xxx.xxx.xxx.xxx/serweb/ And then that HTTP 403 error shows up I really hope somebody can help me because I'm doing a VoIP project for my school and I'd really like to have everything done in time. Thank you for your attention, Roberto Lopes

17 years, 10 months

Re: [Serusers] Proxy Authentication

by Andrey Kouprianov

Hi, Why are you selecting 313814? The problem is with 313812 isnt it? Your trace log shows the conflict between 313812 and 313813 (not 313814)... :) Here's the examle (from RFC document) of authorizationm response from the client back to the server with a 2nd INVITE... Authorization: Digest username="bob", realm="biloxi.com", nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093", uri="sip:bob@biloxi.com", qop=auth, nc=00000001, cnonce="0a4f113b", response="6629fae49393a05397450978507c4ef1", opaque="5ccc069c403ebaf9f0171e9517f40e41" As you can see uri is "sip:bob@bioxy.com" as well as the username is "bob" as well (i.e. they are same)... That gives me an impression there's something wrong with ur client-side setup.. Regards, Andrey. On 8/29/06, Javier Oviedo <joviedo(a)plcendesa.com> wrote: > Andrey Kouprianov escribió: > > Hi, > > > > First of all it is GREAT that you've sent your logs and ser.cfg!! I > > wish everyone would do it (however, this is rarely the case). > > > > You SJPhone cannot call, because for some reason your URI and username > > differ. Take a closer look at this part: > > > > Proxy-Authorization: Digest > > username="313813",realm="localhost",nonce="44f307373cf9fb29da3028442e23f976513fc91d",uri="sip:313812@sipserver.es",response="bfd77c8cdc1066487b170b19e1e0fe61",algorithm=MD5 > > > > > > The username is 313813, but the URI is sip:313812@sipserver.es. Notice > > any difference? Plz, take a look at your account setup in SJPhone > > closely. I think the problem should be there. > > > > Regards, > > > > Andrey. > > > > On 8/28/06, Javier Oviedo <joviedo(a)plcendesa.com> wrote: > >> > >> Hi all > >> > >> In my network schema I have a Sip Express Router to authenticate my > >> outbound > >> calls to a PSTN. I add a new equipment to a ser with "serctl add user > >> pass > >> e-mail" command. My softphone is registered in proxy but when I make > >> a call > >> to a PSTN I obtain the following error: Call rejected: 407 Proxy > >> Authentication Rquired : > >> > >> I use the SJPhone software. > >> > >> Any idea about where is the problem > >> > >> Thanks in advance > >> Joe > >> > >> My ser configuration is: > >> > >> debug=3 > >> fork=no > >> log_stderror=yes > >> > >> port=5060 > >> children=4 > >> > >> dns=no > >> rev_dns=no > >> fifo="/tmp/ser_fifo" > >> fifo_db_url="mysql://ser:heslo@localhost/ser" > >> > >> loadmodule "/usr/local/lib/ser/modules/mysql.so" > >> loadmodule "/usr/local/lib/ser/modules/sl.so" > >> loadmodule "/usr/local/lib/ser/modules/tm.so" > >> loadmodule "/usr/local/lib/ser/modules/rr.so" > >> loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > >> loadmodule "/usr/local/lib/ser/modules/usrloc.so" > >> loadmodule "/usr/local/lib/ser/modules/registrar.so" > >> loadmodule "/usr/local/lib/ser/modules/auth.so" > >> loadmodule "/usr/local/lib/ser/modules/auth_db.so" > >> loadmodule "/usr/local/lib/ser/modules/uri_db.so" > >> > >> modparam("auth_db|uri_db|usrloc", "db_url", > >> "mysql://ser:heslo@localhost/ser") > >> modparam("auth_db", "calculate_ha1", 1) > >> modparam("auth_db", "password_column", "password") > >> modparam("usrloc", "db_mode", 2) > >> modparam("rr", "enable_full_lr", 1) > >> > >> route { > >> > >> # > >> ----------------------------------------------------------------- > >> # Sanity Check Section > >> # > >> ----------------------------------------------------------------- > >> if (!mf_process_maxfwd_header("10")) { > >> sl_send_reply("483", "Too Many Hops"); > >> break; > >> }; > >> > >> if (msg:len > max_len) { > >> sl_send_reply("513", "Message Overflow"); > >> break; > >> }; > >> > >> # > >> ----------------------------------------------------------------- > >> # Record Route Section > >> # > >> ----------------------------------------------------------------- > >> if (method!="REGISTER") { > >> record_route(); > >> }; > >> > >> # > >> ----------------------------------------------------------------- > >> # Loose Route Section > >> # > >> ----------------------------------------------------------------- > >> if (loose_route()) { > >> route(1); > >> break; > >> }; > >> > >> # > >> ----------------------------------------------------------------- > >> # Call Type Processing Section > >> # > >> ----------------------------------------------------------------- > >> if (uri!=myself) { > >> route(1); > >> break; > >> }; > >> > >> if (method=="ACK") { > >> route(1); > >> break; > >> } if (method=="INVITE") { > >> route(3); > >> break; > >> } else if (method=="REGISTER") { > >> route(2); > >> break; > >> }; > >> > >> lookup("aliases"); > >> > >> if (uri!=myself) { > >> route(1); > >> break; > >> }; > >> > >> if (!lookup("location")) { > >> sl_send_reply("404", "User Not Found"); > >> break; > >> }; > >> > >> route(1); > >> } > >> > >> route[1] { > >> > >> # > >> ----------------------------------------------------------------- > >> # Default Message Handler > >> # > >> ----------------------------------------------------------------- > >> if (!t_relay()) { > >> sl_reply_error(); > >> }; > >> } > >> > >> route[2] { > >> > >> # > >> ----------------------------------------------------------------- > >> # REGISTER Message Handler > >> # > >> ---------------------------------------------------------------- > >> sl_send_reply("100", "Trying"); > >> > >> if (!www_authorize("localhost","subscriber")) { > >> www_challenge("localhost","0"); > >> break; > >> }; > >> > >> if (!check_to()) { > >> sl_send_reply("401", "Unauthorized"); > >> break; > >> }; > >> > >> consume_credentials(); > >> > >> if (!save("location")) { > >> sl_reply_error(); > >> }; > >> } > >> > >> route[3] { > >> # > >> ----------------------------------------------------------------- > >> # INVITE Message Handler > >> # > >> ----------------------------------------------------------------- > >> if (!proxy_authorize("localhost","subscriber")) { > >> proxy_challenge("localhost","0"); > >> break; > >> } else if (!check_from()) { > >> sl_send_reply("403", "Use From=ID"); > >> break; > >> }; > >> > >> consume_credentials(); > >> > >> lookup("aliases"); > >> if (uri!=myself) { > >> route(1); > >> break; > >> }; > >> > >> #if (!lookup("location")) { > >> # sl_send_reply("404", "User Not Found"); > >> # break; > >> #}; > >> > >> route(1); > >> } > >> > >> > >> > >> SoftPhone LOG: > >> > >> 17:04:43.1 > >> SENDING TO: 10.110.0.127:5060 > >> INVITE sip:313812@sipserver.es SIP/2.0 > >> To: <sip:313812@sipserver.es> > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport > >> Call-ID: fc7b726c6545d126 > >> CSeq: 1 INVITE > >> Contact: <sip:313813@172.25.97.138:9528> > >> Max-Forwards: 70 > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > >> SUBSCRIBE, > >> INFO > >> Content-Type: application/sdp > >> User-Agent: eyeBeam release 3004t stamp 16741 > >> Content-Length: 274 > >> > >> v=0 > >> o=- 32759240 32759303 IN IP4 172.25.97.138 > >> s=eyeBeam > >> c=IN IP4 172.25.97.138 > >> t=0 0 > >> m=audio 7262 RTP/AVP 100 6 0 8 3 18 5 101 > >> a=alt:1 1 : 8013E837 DD8642E8 172.25.97.138 7262 > >> a=fmtp:101 0-15 > >> a=rtpmap:100 speex/16000 > >> a=rtpmap:101 telephone-event/8000 > >> a=sendrecv > >> > >> 17:04:43.1 > >> RECEIVING FROM: 10.110.0.127:5060 > >> SIP/2.0 407 Proxy Authentication Required > >> To: > >> <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.3909 > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport=9528 > >> Call-ID: fc7b726c6545d126 > >> CSeq: 1 INVITE > >> Proxy-Authenticate: Digest realm="localhost", > >> nonce="44f307373cf9fb29da3028442e23f976513fc91d" > >> Server: Sip EXpress router (0.9.6 (i386/linux)) > >> Content-Length: 0 > >> Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 > >> req_src_ip=172.25.97.138 req_src_port=9528 > >> in_uri=sip:313812@sipserver.es > >> out_uri=sip:313812@sipserver.es via_cnt==1" > >> > >> > >> 17:04:43.1 > >> SENDING TO: 10.110.0.127:5060 > >> ACK sip:313812@sipserver.es SIP/2.0 > >> To: > >> <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.3909 > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport > >> Call-ID: fc7b726c6545d126 > >> CSeq: 1 ACK > >> Content-Length: 0 > >> > >> > >> 17:04:43.1 > >> SENDING TO: 10.110.0.127:5060 > >> INVITE sip:313812@sipserver.es SIP/2.0 > >> To: <sip:313812@sipserver.es> > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport > >> Call-ID: fc7b726c6545d126 > >> CSeq: 2 INVITE > >> Contact: <sip:313813@172.25.97.138:9528> > >> Max-Forwards: 70 > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > >> SUBSCRIBE, > >> INFO > >> Content-Type: application/sdp > >> Proxy-Authorization: Digest > >> username="313813",realm="localhost",nonce="44f307373cf9fb29da3028442e23f976513fc91d",uri="sip:313812@sipserver.es",response="bfd77c8cdc1066487b170b19e1e0fe61",algorithm=MD5 > >> > >> User-Agent: eyeBeam release 3004t stamp 16741 > >> Content-Length: 274 > >> > >> v=0 > >> o=- 32759240 32759303 IN IP4 172.25.97.138 > >> s=eyeBeam > >> c=IN IP4 172.25.97.138 > >> t=0 0 > >> m=audio 7262 RTP/AVP 100 6 0 8 3 18 5 101 > >> a=alt:1 1 : 8013E837 DD8642E8 172.25.97.138 7262 > >> a=fmtp:101 0-15 > >> a=rtpmap:100 speex/16000 > >> a=rtpmap:101 telephone-event/8000 > >> a=sendrecv > >> > >> 17:04:43.1 > >> RECEIVING FROM: 10.110.0.127:5060 > >> SIP/2.0 100 trying -- your call is important to us > >> To: <sip:313812@sipserver.es> > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport=9528 > >> Call-ID: fc7b726c6545d126 > >> CSeq: 2 INVITE > >> Server: Sip EXpress router (0.9.6 (i386/linux)) > >> Content-Length: 0 > >> Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 > >> req_src_ip=172.25.97.138 req_src_port=9528 > >> in_uri=sip:313812@sipserver.es > >> out_uri=sip:313812@sipserver.es via_cnt==1" > >> > >> > >> 17:04:43.1 > >> RECEIVING FROM: 10.110.0.127:5060 > >> SIP/2.0 407 Proxy Authentication Required > >> To: > >> <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.fcd1 > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport=9528 > >> Call-ID: fc7b726c6545d126 > >> CSeq: 2 INVITE > >> Proxy-Authenticate: Digest realm="localhost", > >> nonce="44f307373cf9fb29da3028442e23f976513fc91d" > >> Server: Sip EXpress router (0.9.6 (i386/linux)) > >> Content-Length: 0 > >> Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 > >> req_src_ip=10.110.0.127 req_src_port=5060 in_uri=sip:313812@sipserver.es > >> out_uri=sip:313812@sipserver.es via_cnt==2" > >> > >> > >> 17:04:43.1 > >> SENDING TO: 10.110.0.127:5060 > >> ACK sip:313812@sipserver.es SIP/2.0 > >> To: > >> <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.fcd1 > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport > >> Call-ID: fc7b726c6545d126 > >> CSeq: 2 ACK > >> Content-Length: 0 > >> > >> > >> 17:04:43.2 Call (l:'Joe' r:'sip:313812@sipserver.es') - Call being > >> terminated. Reasons: "Proxy Authentication Required", (code: 407) > >> > >> > >> _______________________________________________ > >> Serusers mailing list > >> Serusers(a)lists.iptel.org > >> http://lists.iptel.org/mailman/listinfo/serusers > >> > >> > >> > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > Hi Andrey > > I think tha tihs part of log is correct: > > Proxy-Authorization: Digest > username="313813",realm="localhost",nonce="44f307373cf9fb29da3028442e23f976513fc91d",uri="sip:313812@sipserver.es",response="bfd77c8cdc1066487b170b19e1e0fe61",algorithm=MD5 > > > The username is the caller id and the uri is de destination number. My > SJphone configuration is simply: The user, pass and account name are > 313813. I attach my mysql subscriber table: > > > mysql> select * from subscriber where username=313814; > +----------------------------------+----------+----------------------+----------+------------+-----------+-------+-------------------+---------------------+---------------------+--------------+------+------------------+----------+----------------------------------+----------------------------------+------------+----------+------+------+------+ > | phplib_id | username | domain | > password | first_name | last_name | phone | email_address | > datetime_created | datetime_modified | confirmation | flag | > sendnotification | greeting | ha1 | > ha1b | allow_find | timezone | rpid | domn | > uuid | > +----------------------------------+----------+----------------------+----------+------------+-----------+-------+-------------------+---------------------+---------------------+--------------+------+------------------+----------+----------------------------------+----------------------------------+------------+----------+------+------+------+ > | 3fbf3938ba07fa0f280ffe44926e4cd3 | 313814 | sipserver.es | 313814 > | | | | joe(a)plcendesa.com | 2006-08-28 > 18:23:22 | 0000-00-00 00:00:00 | | o | > | | 3fbf3938ba07fa0f280ffe44926e4cd3 | > 147098ce3cd77298d89982cd37b5813a | 0 | NULL | NULL | NULL | > NULL | > +----------------------------------+----------+----------------------+----------+------------+-----------+-------+-------------------+---------------------+---------------------+--------------+------+------------------+----------+----------------------------------+----------------------------------+------------+----------+------+------+------+ > 1 row in set (0.00 sec) > > > and my uri table: > > mysql> select * from uri; > +----------+----------------------+----------+---------------------+ > | username | domain | uri_user | last_modified | > +----------+----------------------+----------+---------------------+ > | 313813 | sipserver.createc.es | 313813 | 2006-08-28 17:52:48 | > | 313814 | sipserver.createc.es | 313814 | 2006-08-28 18:23:22 | > +----------+----------------------+----------+---------------------+ > 2 rows in set (0.00 sec) > > > With the X-Lite softphone the same thing happens. > > Thanks in advace! > Regards > Joe >

17 years, 10 months

[Users] Understanding the Call Flow

by Max Clark

Hi all, I am trying to get a better understanding of the call flow - specifically for distant users. In our current configuration we have all of the UAs registering to our OpenSER box which manages the authentication, and passes the call forward if allowed. Like so: UA -> SER -> PSTN GW -> PSTN I have been experimenting with using a mediaproxy/rtp proxy to improve the quality of service to users beyond the network edge - effectively: UA -> Proxy -> SER -> PSTN GW -> PSTN While this works great, we would like to place some PSTN Gateways closer to the edge of the network (for more than one obvious reason). In this case we would still have a central SER instance for authentication and accounting, but I would like for the edge proxies to be able to send the traffic to the closest applicable PSTN GW. How does one go about deploying something like this? What does the call flow look like? What software components do I need in what places? What is the glue that holds all of this together? Thanks, Max -- Max Clark http://www.clarksys.com

17 years, 10 months

[Users] Please help me in openser administrator...Please

by Ferianto siregar

Dear all, Thanks to all of you for your kind-hearted to give me solution in solving my problem. Thank you very much. Thank you very much to Andrey Kouprianov for replying my message. Thanks I have some questions about the openser admin. I do anybody can help me to understand what the openser admin use for and how to install it. These are my question: 1. Is openser adminstrator a web application? I mean, does it have same function like "serweb" in ser? 2. How to install the openser administrator in openser server? I use openser1.1.0-tls and what packets does it need? 3.Is the openser administrator just used by the administrator to manage his/her server? or The openser administrator can be used like a serweb?(the client also can use it) because as I read from the reply message in this forum, the serweb can not use for openser1.1.0-tls 4. If it is not like serweb, can anybody tell me what is the suitable web application for openser ? so the client can register to in communication session by using a web application like serweb That is all my questions. I do hope anybody can help me. Please.... Thank you very much Regards, Ferianto --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.

17 years, 10 months

R: R: [Users] using AVPs to write into database

by D'Addelfio Davide

Thanks, I've looked into but it works only with 1.1.x version...!!! What's the different with avp_db_store command, may I use it to store something in my SQL DB? I'll try to do but I have some problems, he doesn't register... I have another question: I need to work over sdp messages, is possible to do a parsing, always using the scripts language...? Any thought? Thanks Davide -----Messaggio originale----- Da: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Inviato: mercoledì 5 aprile 2006 19.15 A: D'Addelfio Davide Cc: Jayesh Nambiar; Bogdan-Andrei Iancu; openser Oggetto: Re: R: [Users] using AVPs to write into database Hello, On 04/05/06 18:01, D'Addelfio Davide wrote: > > Hi guys, I have a similar problem, maybe easier. I'm new in openser so > I'd like to have some help. > > I'd like openser create a table in MySQL db when receive an INVITE, a > row with from uri, method and something else, I've look into modules > docs but I've not understanding at all...could you help me, please? > if you are using openser development version, then take a look at avp_db_query() in avpops module http://openser.org/docs/modules/1.1.x/avpops.html#AEN239 Cheers, Daniel > Thanks for reply > > Davide > > ------------------------------------------------------------------------ > > *Da:* users-bounces(a)openser.org [mailto:users-bounces@openser.org] > *Per conto di *Jayesh Nambiar > *Inviato:* mercoledì 5 aprile 2006 15.53 > *A:* Bogdan-Andrei Iancu > *Cc:* openser > *Oggetto:* Re: [Users] using AVPs to write into database > > Hi Bogdan, > > Thanks for the reply and sorry for sending the earlier mail 4 times as > yahoo was not responding peoperly. > > When I do an avp_write(), why does the avp name get stored in the uuid > column? > > I am not using any uuid as of now. I also tried to add the username of > the person who is calling but it again creates two rows, one with the > value as > > username and other row with the value as forwarded number. > > I thought if there is a way to insert the username also, it wud be > easier to check, if the incoming call for that user is to be forwarded > or not. > > I get the following in my usr_preferences table after I dial 86 > followed by any number. > > +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ > | uuid | username | domain | attribute | type | value | modified | > +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ > | s:callfwd | | | callfwd | 0 | sip:5515551478@202.80.61.10 | > 2006-04-05 19:10:49 | > +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ > > The script snippet is as follows: > > if(uri=~"^sip:86[0-9]*@") { > > > strip(2); > avp_write("$ruri", "s:callfwd"); > #avp_write("$from/username", "i:999"); > #avp_db_store("i:999", "i:/usr_preferences"); > avp_db_store("s:callfwd", "s:callfwd/usr_preferences"); > sl_send_reply("200", "OK"); > exit; > }; > > > */Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>/* wrote: > > Hi, > > you need two steps: > 1) write the ruri (after strip) into an AVP - use avp_write() > 2) write the avp into the db - use avp_db_store() > > regards, > bogdan > > Jayesh Nambiar wrote: > > > Hi all, > > I am trying to implement the call forwarding feature in openser. The > > forwarding part is fine. But I have the follwing scenario: > > Whenever a user needs to change his forwarding number, he should be > > able to do so from his end device only. > > For eg: He first presses 86 and then the 10 digit number to be > > forwarded. Here the 86 has to be stripped and the 10 digit number > > should be inserted into the call forward value column of the > > preferences table. > > Also if he wants to remove the forwarded number, he can do so by > > pressing say for eg: 87. on receiving this number openser should > > delete that value from the > > table. > > Is this possible. I tried to do the following, but somehow it > does not > > change the > > value. > > if(uri=~"^sip:86[0-9]*@") { > > if(avp_db_load("$from/username", "s:callfwd")) { #check > > if call-fwd feature is enabled for the user > > log(1,"AVP condition returned true"); > > strip(2); > > avp_write("$ruri", "s:callfwd"); > > avp_print(); > > log(1,"AVP written"); > > sl_send_reply("200", "OK"); > > exit; > > }; > > }; > > Is avp_write the proper method or I guess avp_db_store can also help > > me. The avp_print() function also does not show me anything in the > > log. Are there any > > logical mistakes or I have mis-interpreted the syntax of avpops > functions. > > Please help me in thsi regard. > > Thanks a lot in advance. > > Jayesh. > > > > ------------------------------------------------------------------------ > > Jiyo cricket on Yahoo! India cricket > > > > Yahoo! Messenger Mobile > > > > Stay in touch with your buddies all the time. > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > ------------------------------------------------------------------------ > > Jiyo cricket on Yahoo! India cricket > <http://us.rd.yahoo.com/mail/in/mailcricket/*http:/in.sports.yahoo.com/crick…> > Yahoo! Messenger Mobile > <http://us.rd.yahoo.com/mail/in/mailmobilemessenger/*http:/in.mobile.yahoo.c…> > Stay in touch with your buddies all the time. > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

17 years, 10 months

[Serusers] SER with VoiceMail Problem

by ram

Hi i am new to ser, iam going through the list, i found some of the similar queries http://lists.iptel.org/pipermail/serusers/2003-October/003132.html I have used the config file changed according to my setup and iam able to pick up messages from Asterisk also But i have some problem here is 1. i have registered user 8888 in my SER, when i dial 8888 its going to voice mail, how can i set the caller==calee Busy even i have added this if (!uri==myself) { t_relay(); break; }; But still no use. 2. when i dial to PSTN number, when the PSTN also go to voice mail after certain rings I suppose to get the voice Menu of other Side Number, but the voice go Blank ( instead of going to VoiceMenu) when checked in my Log, i get this error in my syslog Aug 28 18:33:55 router ser[757]: no location or no user Aug 28 18:34:01 router ser[751]: 10 digit exp match w/leading 1 Aug 28 18:34:01 router ser[751]: route[2]:SIP-to-PSTN call routed Aug 28 18:34:10 router ser[757]: no location or no user Aug 28 18:34:25 router ser[758]: no location or no user Aug 28 18:34:28 router ser[791]: ACC: call missed: from=88888 < sip:88888@mydomain.com>;tag=1674409670, i-uri= sip:1xxxxx80369@mydomain.com, method=INVITE, o-uri=sip:1xxxxx80369@myprovider:5060, code=408 Request Timeout Aug 28 18:34:28 router ser[791]: failure_route[5] Aug 28 18:34:28 router ser[791]: route[5]: re-relay to PSTN Aug 28 18:34:28 router ser[751]: ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487 Aug 28 18:34:37 router ser[758]: 10 digit exp match w/leading 1 Aug 28 18:34:37 router ser[758]: route[2]:SIP-to-PSTN call routed Aug 28 18:34:37 router ser[758]: 10 digit exp match w/leading 1 Aug 28 18:34:37 router ser[758]: route[2]:SIP-to-PSTN call routed Aug 28 18:34:40 router ser[751]: no location or no user any suggestions Ram

17 years, 10 months

[Serusers] Proxy Authentication

by Javier Oviedo

Hi all In my network schema I have a Sip Express Router to authenticate my outbound calls to a PSTN. I add a new equipment to a ser with "serctl add user pass e-mail" command. My softphone is registered in proxy but when I make a call to a PSTN I obtain the following error: *Call rejected: 407 Proxy Authentication Rquired *: I use the SJPhone software. Any idea about where is the problem Thanks in advance Joe My ser configuration is: debug=3 fork=no log_stderror=yes port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (method=="ACK") { route(1); break; } if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("localhost","subscriber")) { www_challenge("localhost","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("localhost","subscriber")) { proxy_challenge("localhost","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(1); break; }; #if (!lookup("location")) { # sl_send_reply("404", "User Not Found"); # break; #}; route(1); } SoftPhone LOG: 17:04:43.1 SENDING TO: 10.110.0.127:5060 INVITE sip:313812@sipserver.es SIP/2.0 To: <sip:313812@sipserver.es> From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport Call-ID: fc7b726c6545d126 CSeq: 1 INVITE Contact: <sip:313813@172.25.97.138:9528> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 274 v=0 o=- 32759240 32759303 IN IP4 172.25.97.138 s=eyeBeam c=IN IP4 172.25.97.138 t=0 0 m=audio 7262 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 8013E837 DD8642E8 172.25.97.138 7262 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv 17:04:43.1 RECEIVING FROM: 10.110.0.127:5060 SIP/2.0 407 Proxy Authentication Required To: <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.3909 From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport=9528 Call-ID: fc7b726c6545d126 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="localhost", nonce="44f307373cf9fb29da3028442e23f976513fc91d" Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 req_src_ip=172.25.97.138 req_src_port=9528 in_uri=sip:313812@sipserver.es out_uri=sip:313812@sipserver.es via_cnt==1" 17:04:43.1 SENDING TO: 10.110.0.127:5060 ACK sip:313812@sipserver.es SIP/2.0 To: <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.3909 From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport Call-ID: fc7b726c6545d126 CSeq: 1 ACK Content-Length: 0 17:04:43.1 SENDING TO: 10.110.0.127:5060 INVITE sip:313812@sipserver.es SIP/2.0 To: <sip:313812@sipserver.es> From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport Call-ID: fc7b726c6545d126 CSeq: 2 INVITE Contact: <sip:313813@172.25.97.138:9528> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="313813",realm="localhost",nonce="44f307373cf9fb29da3028442e23f976513fc91d",uri="sip:313812@sipserver.es",response="bfd77c8cdc1066487b170b19e1e0fe61",algorithm=MD5 User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 274 v=0 o=- 32759240 32759303 IN IP4 172.25.97.138 s=eyeBeam c=IN IP4 172.25.97.138 t=0 0 m=audio 7262 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 8013E837 DD8642E8 172.25.97.138 7262 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv 17:04:43.1 RECEIVING FROM: 10.110.0.127:5060 SIP/2.0 100 trying -- your call is important to us To: <sip:313812@sipserver.es> From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport=9528 Call-ID: fc7b726c6545d126 CSeq: 2 INVITE Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 req_src_ip=172.25.97.138 req_src_port=9528 in_uri=sip:313812@sipserver.es out_uri=sip:313812@sipserver.es via_cnt==1" 17:04:43.1 RECEIVING FROM: 10.110.0.127:5060 SIP/2.0 407 Proxy Authentication Required To: <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.fcd1 From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport=9528 Call-ID: fc7b726c6545d126 CSeq: 2 INVITE Proxy-Authenticate: Digest realm="localhost", nonce="44f307373cf9fb29da3028442e23f976513fc91d" Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 req_src_ip=10.110.0.127 req_src_port=5060 in_uri=sip:313812@sipserver.es out_uri=sip:313812@sipserver.es via_cnt==2" 17:04:43.1 SENDING TO: 10.110.0.127:5060 ACK sip:313812@sipserver.es SIP/2.0 To: <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.fcd1 From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport Call-ID: fc7b726c6545d126 CSeq: 2 ACK Content-Length: 0 17:04:43.2 Call (l:'Joe' r:'sip:313812@sipserver.es') - Call being terminated. Reasons: "Proxy Authentication Required", (code: 407)

17 years, 10 months

Re: [Serusers] rewrite the From part of SIP INVITE messages

by rcarvalho＠iric.up.pt

I'm sceptic about the use of functions that replace parts of SIP messages, because that way, will Ser still keep record of transactions? Wont those replacements affect Ser to recognise to whom route subsequent messages? Regards, Ricardo. Alexandr Dubovikov wrote: > On Fri, Aug 25, 2006 at 05:33:52PM +0100, Ricardo Carvalho wrote: >> I need to rewrite the From part of SIP INVITE messages before sending >> them to my IP Telco, or else they wont accept my calls! >> Anyone knows how can I do that? > > replace or subst > > the sample you can find in README file of textops module. > > >> Thanks, >> >> Ricardo. >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >

17 years, 10 months

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sr-users August 2006