sr-users

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37456 discussions

[Serusers] Number registered
by Teran Jesus 20 Jul '06

20 Jul '06

Hello I want to validate and get the 2 first digits of the telephone number registered, if i have a ip phone registered with a number like this 5556000000, when i recieve a INVITE from this number, i want to get 55 and validate the number, are there something to do this? Can you help me?

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[Serusers] Re: [Serweb-users] newbie: fifo issue on user login page
by Andrey Kouprianov 20 Jul '06

20 Jul '06

Hi, Add fifo_mode=766 to your ser.cfg and restart ser to make change permanent. If you want to make a temporary change (i.e. for a single time ser is started), just chmod 766 /tmp/ser_fifo Andrey. On 7/20/06, Sprechenzi English <sprechensenglish(a)yahoo.co.uk> wrote: > > > hi all! im stil new to this list. actually i just started configuring serweb > yesterday. and find it quite interesting though i lack enough knowledge in > troubleshooting it. i just would like to seek some assistance from this list > on my little problem: > > when logging into the user page, i can enter but it displays a red warning > or some sort of alert like this: > > sorry -- cannot open write fifo > > can someone help me to get rid of this? i've read the previous digests from > the list and i didnt seem to find the solution to my problem. by the way did > find the solution on the issue on missed calls! thanks to this list. i'm > really hoping someone kind enough will teach or just direct me to some > helpful link to solve my problem! thanks again and more power! > > sprechenzi > _______________________________________________ > Serweb-users mailing list > Serweb-users(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serweb-users > > >

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RE: [Serusers] Remote sip and rtp analyzing
by İlker Aktuna (Koç.net) 19 Jul '06

19 Jul '06

Hi, These are all very nice tools but none of them allow remote capturing of RTP. I am looking for something which would also capture RTP and let us analyze it. Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Wednesday, July 19, 2006 9:52 AM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] Remote sip and rtp analyzing Have you looked at these? http://www.pernau.at/kd/voip/bookmarks-sip-test.html g-) İlker Aktuna (Koç.net) wrote: Hi, I still couldn't find a nice way of remote sip & rtp packet capturing and analyzing. Is there any tool to do this ? Do I need to use rcap protocol ? How can I use it on SER ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Friday, July 07, 2006 2:39 PM To: serusers(a)iptel.org Subject: [Serusers] Remote sip and rtp analyzing Hi, I am looking for a solution to analyze sip and rtp packets on a Ser + Mediaproxy server. SIPAlanyzerV6 seems to receive remote capturing, but there is no information on its manuals for remote capture. What is the best solution to analyze sip and rtp packets remotely from a SER server ? I also would like to have some statistical output, if possible. Thanks for your comments. ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, I still couldn't find a nice way of remote sip & rtp packet capturing and analyzing. Is there any tool to do this ? Do I need to use rcap protocol ? How can I use it on SER ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Friday, July 07, 2006 2:39 PM To: serusers(a)iptel.org Subject: [Serusers] Remote sip and rtp analyzing Hi, I am looking for a solution to analyze sip and rtp packets on a Ser + Mediaproxy server. SIPAlanyzerV6 seems to receive remote capturing, but there is no information on its manuals for remote capture. What is the best solution to analyze sip and rtp packets remotely from a SER server ? I also would like to have some statistical output, if possible. Thanks for your comments. ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, These are all very nice tools but none of them allow remote capturing of RTP. I am looking for something which would also capture RTP and let us analyze it. Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Wednesday, July 19, 2006 9:52 AM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] Remote sip and rtp analyzing Have you looked at these? http://www.pernau.at/kd/voip/bookmarks-sip-test.html g-) İlker Aktuna (Koç.net) wrote: Hi, I still couldn't find a nice way of remote sip & rtp packet capturing and analyzing. Is there any tool to do this ? Do I need to use rcap protocol ? How can I use it on SER ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Friday, July 07, 2006 2:39 PM To: serusers(a)iptel.org Subject: [Serusers] Remote sip and rtp analyzing Hi, I am looking for a solution to analyze sip and rtp packets on a Ser + Mediaproxy server. SIPAlanyzerV6 seems to receive remote capturing, but there is no information on its manuals for remote capture. What is the best solution to analyze sip and rtp packets remotely from a SER server ? I also would like to have some statistical output, if possible. Thanks for your comments. ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, I still couldn't find a nice way of remote sip & rtp packet capturing and analyzing. Is there any tool to do this ? Do I need to use rcap protocol ? How can I use it on SER ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Friday, July 07, 2006 2:39 PM To: serusers(a)iptel.org Subject: [Serusers] Remote sip and rtp analyzing Hi, I am looking for a solution to analyze sip and rtp packets on a Ser + Mediaproxy server. SIPAlanyzerV6 seems to receive remote capturing, but there is no information on its manuals for remote capture. What is the best solution to analyze sip and rtp packets remotely from a SER server ? I also would like to have some statistical output, if possible. Thanks for your comments. ilker <http://387555.sigclick.mailinfo.com/sigclick/000A0602/040D4903/0004054B/032…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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[Serusers] SER/OpenSER + Asterisk Voicemail
by Marc Lewis 19 Jul '06

19 Jul '06

I've got OpenSER 1.0.1 working very well taking care of SIP UA's and have them going into asterisk voicemail correctly using Asterisk Realtime and MySQL 5.0 views. To ease our customers into this and give them a signle sign-on for all of the services we offer, we're using their username/email address on our main system. This gives them a single username/password combination for their POP email, DSL connection, webmail access, Jabber services and (soon) SIP services. My problem is this: Its no problem getting Asterisk to accept an alphanumeric mailbox for storing the user's voicemail. The problem is when they want to access it. If they're on a Linksys or Grandstream ATA and want to dial into their voicemail, it prompts them for their mailbox number. If their mailbox number is alphanumeric there is no way for them to enter that and they won't be able to get at their voicemail. What I would like to do is assign a mailbox number that is equal to their "primary" phone number. This can be stored in the usr_preferences table as an AVP. What I haven't figured out yet is how to rewrite that RURI before it sends it off to the asterisk server for voicemail. By doing this, we could use alphanumeric SIP logins, and then phone numbers for voicemail boxes. Given this configuration, how would I rewrite the user? I've tried: modparam("avpops", "avp_aliases", "mailboxuser=s:801") And then: avp_write("$ruri/username", "s:mailboxuser/g"); rewritehostport("10.0.0.2:5060"); append_branch(); But the user never changes. I've tried various things with rewriteuser and AVP's but haven't had any success so far. The usr_preferences row looks like this: +------+----------+--------+-----------+------+------------+ | uuid | username | domain | attribute | type | value | +------+----------+--------+-----------+------+------------+ | | sipuser | | 801 | 0 | 4255551212 | +------+----------+--------+-----------+------+------------+ Anyone have any suggestions on how to solve this problem? Thanks. - Marc -- Marc Lewis Avvanta Communications Corporation

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[Users] Parallel ringing
by Andreas M. 19 Jul '06

19 Jul '06

Hello, how could it be done through AVP´s and database lookup, to do a parallel ringing. Following setup: If uri eg example(a)test.dom is called, this call should be forwarded in parallel to a pstn-gateway. That means, receive the AVP with the telephonnumber associated to uri example(a)test.dom, rewrite uri, rewrite host/port and forward in parallel to gateway. I tried "append_branch", but was not able to put in the AVP (phonenumber), rewrite host/port points directly to the gateway. I have no troubles to load the AVP, only to make is possible, that the SIP Client and PSTN Phone are ringing in parallel. Some suggestions ? regards, Andreas

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Re: [Users] Redirect and fork
by Andrew And 19 Jul '06

19 Jul '06

But my server destination has dynamic IP because I am using ENUM (DNS). I could to use "rewrite" and "forward" if my servers were static. 2006/7/18, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com>: > > use rewirtehost('ip','port'); or forward('ip:port'); > > On 7/19/06, Andrew And <andcop2006(a)gmail.com> wrote: > > > > > How can I do Redirect in my Openser (server)? And if I have 2 or more > > destination I need to do redirect + fork. > > > > Does someone could help my? > > > > > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > -- > Thanks and Regards with cheers > Sunkara Ravi Prakash (Voip Developer) > Hyperion Technology > Kondapur, Hi-tech city, > Hyderabad. > www.hyperion-tech.com > +91-9985077535 >

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[Users] UAC Problem with Sipura
by ram 19 Jul '06

19 Jul '06

Hi all I have read the UAC Module and added to my openser.cfg modparam("uac","credential","99999:domain:99999") rewritehostport("domain:5060"); Registered use using my X-lite and iam able to make calls out another account i have configured in sipura, and try to call from Local X-lite to Sipura working Fine >From sipura iam trying to call out, call not establishing when i check the log its follow as : Jul 18 09:11:05 sert /usr/local/sbin/openser[14230]: CALL: PSTN gateway Jul 18 09:11:05 sert /usr/local/sbin/openser[14227]: INFO: CALL TO VOICEMAIL Jul 18 09:11:05 sert /usr/local/sbin/openser[14227]: ACC: transaction answered: from=22222 <sip:22222@mydomain.com>;tag=30bd 8a0d1c12d72o1, i-uri=sip:17322180369@mydomain.com, code=200, uid=22222, method=INVITE, totag=as297d95de, userpart=1732218036 9, fromtag=30bd8a0d1c12d72o1 Jul 18 09:11:05 sert /usr/local/sbin/openser[14221]: ACC: request acknowledged: from=22222 <sip:22222@mydomain.com>;tag=30bd 8a0d1c12d72o1, i-uri=sip:17322180369@myserver-ipaddress:5090, code=200, uid=n/a, method=ACK, totag=as297d95de, userpart=17322180369 , fromtag=30bd8a0d1c12d72o1 Jul 18 09:11:05 sert /usr/local/sbin/openser[14230]: ERROR: parse_uri: bad uri, state 0 parsed: <:173> (4) / <:17322180369@myserver-ipaddress:5090> (31) Jul 18 09:11:05 sert /usr/local/sbin/openser[14230]: ERROR: parse_sip_msg_uri: bad uri <:17322180369@myserver-ipaddress:5090> Jul 18 09:11:05 sert /usr/local/sbin/openser[14230]: loose_route: Error while parsing Request URI Jul 18 09:11:05 sert /usr/local/sbin/openser[14230]: ERROR: parse_uri: bad uri, state 0 parsed: <:173> (4) / <:17322180369@myserver-ipaddress:5090> (31) Jul 18 09:11:05 sert /usr/local/sbin/openser[14230]: ERROR: parse_sip_msg_uri: bad uri <:17322180369@myserver-ipaddress:5090> Jul 18 09:11:05 sert /usr/local/sbin/openser[14230]: does_uri_exist(): Error while parsing URI Jul 18 09:11:05 sert /usr/local/sbin/openser[14230]: CALL: PSTN gateway Jul 18 09:11:05 sert /usr/local/sbin/openser[14230]: ERROR: parse_uri: bad uri, state 0 parsed: <:173> (4) / <:17322180369@myserver-ipaddress:5090> (31) From: <sip::myserver-ipaddress;ftag=30bd8a0d1c12d72o1;lr=on>a2b;rporttion: bad uri <:17322180369@myserver-ipaddress:5090 SIP/2.0 Content-Length: 0isk PBXc6@sipura-ip:5090>8a0d1c12d72o1 >, dropping packet Jul 18 09:11:05 sert /usr/local/sbin/openser[14230]: ERROR: parse_uri: bad uri, state 0 parsed: <:173> (4) / <:17322180369@myserver-ipaddress:5090> (31) Jul 18 09:11:05 sert /usr/local/sbin/openser[14230]: ERROR: parse_sip_msg_uri: bad uri <:17322180369@myserver-ipaddress:5090> Jul 18 09:11:05 sert /usr/local/sbin/openser[14230]: ERROR: new_t: uri invalid Jul 18 09:11:05 sert /usr/local/sbin/openser[14230]: ERROR: t_newtran: new_t failed Jul 18 09:11:05 sert /usr/local/sbin/openser[14230]: ERROR: t_relay failed Jul 18 09:11:05 sert /usr/local/sbin/openser[14230]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/ SL) where iam doing wrong ? could some one Guide me ram

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[Users] BYE details
by tele 19 Jul '06

19 Jul '06

Hi, look at this scenario: UA --> SBC --> SIPproxy | | mediaproxy (openser+mediaproxy)sbc use the mediaproxy for all calls. (mediaproxy) some external mediaproxy controlled by the proxydispatcher (openser)sipproxy do the accounting for the calls. I have modified mediaproxy for sending e fake BYE message in case of timeout of media to the SIPproxy. call UA<->UA, if during conference i disconnect the ethernet cable of both UA.. i need to bill anyway the call when media timeout. this is a fake BYE generated from mediaproxy: SBC - 82.215.130.50 SIPproxy - 82.215.130.51 U 82.215.130.50:60083 -> 82.215.130.51:5060 BYE sip:396006660097@82.215.130.51 SIP/2.0. Record-Route: <sip:82.215.130.51;lr=on;ftag=2d8a17eda169fb3e>. Max-Forwards: 69. Via: SIP/2.0/UDP 82.215.130.51. To: <sip:0104491009@sbc.plexia.com>;tag=dc6ec3b669dd8b16. From: <sip:396006660097@sbc.plexia.com>;tag=2d8a17eda169fb3e. Call-ID: d3ac49c2ba1793cd(a)82.215.128.140. Content-Length: 0. the unique info i had for generate the fake BYE are: from_uri,to_uri,to_tag,from_tag,callid the BYE is processed by openser without error but no radius accounting is sent. 3(8754) receive_msg: cleaning up what i need to know is if i missing something essential in the BYE for the accounting. 3(8754) SIP Request: 3(8754) method: <BYE> 3(8754) uri: <sip:396006660097@82.215.130.51> 3(8754) version: <SIP/2.0> 3(8754) parse_headers: flags=2 3(8754) end of header reached, state=2 3(8754) parse_headers: Via found, flags=2 3(8754) parse_headers: this is the first via 3(8754) After parse_msg... 3(8754) preparing to run routing scripts... 3(8754) DEBUG:maxfwd:is_maxfwd_present: value = 69 3(8754) parse_headers: flags=10 3(8754) DEBUG: add_param: tag=dc6ec3b669dd8b16 3(8754) DEBUG:parse_to:end of header reached, state=29 3(8754) DBUG:parse_to: display={}, ruri={sip:0104491009@sbc.plexia.com} 3(8754) DEBUG: get_hdr_field: <To> [54]; uri=[sip:0104491009@sbc.plexia.com] 3(8754) DEBUG: to body [<sip:0104491009@sbc.plexia.com>] 3(8754) DEBUG: add_param: tag=2d8a17eda169fb3e 3(8754) DEBUG:parse_to:end of header reached, state=29 3(8754) DBUG:parse_to: display={}, ruri={sip:396006660097@sbc.plexia.com} 3(8754) parse_headers: flags=40 3(8754) DEBUG:avpops:dbstore_avps: 1 avps were stored 3(8754) DEBUG:avpops:dbstore_avps: 1 avps were stored 3(8754) avpops:ops_dbquery_avps: query [select value from usr_preferences where uuid='d3ac49c2ba1793cd(a)82.215.128.140' and attribute='setup_time'] 3(8754) avpops:ops_dbquery_avps: query [select value from usr_preferences where uuid='d3ac49c2ba1793cd(a)82.215.128.140' and attribute='connect_time'] 3(8754) avpops:ops_dbquery_avps: query [select value from usr_preferences where uuid='d3ac49c2ba1793cd(a)82.215.128.140' and attribute='disconnect_time'] 3(8754) avpops:ops_dbquery_avps: query [select value from usr_preferences where uuid='d3ac49c2ba1793cd(a)82.215.128.140' and attribute='valid_call'] 3(8754) db_flags=3, flags=4 3(8754) DEBUG:avpops:load_avps: loaded avps = 1 3(8754) DEBUG:avpops:check_avp: check <yes> against <yes> as str 3(8754) [ACCT] do the radius accounting 3(8754) parse_headers: flags=200 3(8754) DEBUG: get_hdr_body : content_length=0 3(8754) found end of header 3(8754) find_first_route: No Route headers found 3(8754) loose_route: There is no Route HF 3(8754) DEBUG:avpops:load_avps: loaded avps = 1 3(8754) [LID/CLI] Find association: 396006660097 390104491097 3(8754) subst_run: running. r=0 3(8754) subst_run: matched (0, 12): [390104491097] 3(8754) avpops:ops_subst: subst to 1 avps 3(8754) DEBUG:destroy_avp_list: destroying list 0x285287a0 3(8754) receive_msg: cleaning up regards, :tele

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[Users] show active calls
by Mark Anthony C. Delfin 19 Jul '06

19 Jul '06

Hi List, Is there a module for .openser/openserctl where I can view active calls and/or disconnect them. thanks Mark

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[Users] AVP problem ,Require help
by Tanzeel serfaraz 19 Jul '06

19 Jul '06

Hi users , i am new to use openser,i have to implement trusted ip example(http://www.voice-system.ro/docs/avpops/ar01s08.html#ex_trusting) i configure openser.cfg acording to this example.when i run command "openser -c -f path to openser.cfg" i got message "config file ok, exiting...".But when i stated openser by "/usr/local/sbin/openserctl start" i am geeting error like: "Starting OpenSER : PID file /var/run/openser.pid does not exist -- OpenSER start failed " here is my configuration file .Please help which statement is wrong so that my SER gets started . Thanks Regards Tanzeel --------------------------------- Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less. # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode 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" listen=192.168.1.37:5060 # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/alias_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" #loadmodule "/usr/local/lib/openser/modules/alias_db.so" loadmodule "/usr/local/lib/openser/modules/auth_diameter.so" loadmodule "/usr/local/lib/openser/modules/diversion.so" loadmodule "/usr/local/lib/openser/modules/enum.so" loadmodule "/usr/local/lib/openser/modules/exec.so" # ----------------- setting module-specific parameters --------------- # Database Configuration modparam("usrloc", "db_url", "mysql://openser:tanzeel@localhost/openser") # -- 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) ############################## AVP parameters ########### modparam("avpops","avp_url","mysql://root:tanzeel@tanzeel/openser") modparam("avpops","avp_table","usr_preferences") modparam("avpops","uuid_column","uuid") modparam("avpops","username_column","username") modparam("avpops","domain_column","localhost") modparam("avpops","attribute_column","attribute") modparam("avpops","value_column","value") modparam("avpops","type_column","type") #modparam("avpops","use_domain","1") modparam("avpops","avp_aliases","uuid=I:2;email=s:raza@localhost;fwd=i:53") modparam("avpops","avp_url","mysql://root:tanzeel@tanzeel/openser") # ------------------------- 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # 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(); ####################################### AVP EXAMPLE ################ # if the request pretends to belong to a local domain if (is_from_local()) { #authenticate only INVITE and MESSAGES if (method=="INVITE" || method=="MESSAGE" { # is it a trusted IP address? - first load the trusted IPs (avp # NAME 't_ips" from DB table "ips") for the target domain (domain part # of RURI); then check if at least one value of 't_ips'AVPs equals # the source IP of the request if (!(avp_db_load("$ruri/localhost","s:t_ips/usr_preferences") && avp_check("s:t_ips","eq/$src_ip/gi"))) { sl_send_reply("404", "not allowed"); return; } log("**************** AVP PRINT START ******************\n"); avp_print(); log("**************** AVP PRINT END ******************\n"); } } # 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); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} route(1); }; # 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("localhost", "subscriber")) { www_challenge("localhost", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; 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(); }; exit; }

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