sr-users January 2006

sr-users@lists.kamailio.org

134 participants
259 discussions

[Users] uac_replace_from function
by unplug 20 Jan '06

20 Jan '06

I am using uac_replace_from function as below. # replace only display and do not touch uri uac_replace_from("35418474",""); After running uac_replace_from function, I get the following message. U 203.193.4.242:5060 -> 203.193.4.226:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 203.193.4.226;branch=z9hG4bKaed7.46cc4101.0,SIP/2.0/UDP 10.0.0.52:5060;rport=5060;received=210.184.4.31;branch=z9hG4bKHNHpldVsKca6IYUS. From: "35418474"<sip:882754853589@ow01.owtel.com>;tag=sBl2gLH6O1myyqL5. To: "92588819" <sip:92588819@ow01.owtel.com>;tag=47C61DF4-26CD. Date: Fri, 06 Jan 2006 10:41:17 GMT. Call-ID: yTCQvS56ZCZXCHgt(a)10.0.0.52. Server: Cisco-SIPGateway/IOS-12.x. In the normal message, sip message, there is a space between display name and uri. ie. From: "35418474" <sip:882754853589@ow01.owtel.com> However, after I apply uac_replace_from function, there is a space there. How can I add a space such that it looks like a normal sip message? Does the space will affect the caller display?

3 10

[Users] centralized speed dial
by Reto Kortas 20 Jan '06

20 Jan '06

Hi! I am planning to set up a centralized speed dial service for a 2-layer proxy hierarchy. The design should look like these: PSTN | | UA --> LOCAL_SP --> MAIN-PROXY <-- LOCAL_SP <-- UA | | UA --> LOCAL_SP ------ | | PSTN ... and many many more local SIP Proxys (LOCAL_SP). Any branch has a local SIP-Proxy which functions also as a PSTN-Gateway and will route local calls, whether they are from inside or from the PSTN. The users are configured on the local SIP-Proxy, which will also register these user on a REGISTER-Request also on the centralized SIP-Proxy. The Speed dial database should contain internal as well as external (PSTN) numbers. The tricky thing is, that the users have to use their local PSTN Gateway, when they want to make off-net calls. The call flow must like these: UA -INVITE-> LOCAL_SP -INVITE-> MAIN-PROXY -> DB-Lookup for speed dial-alias MAIN-PROXY --301_RESPONSE--> LOCAL_SP --301_RESPONSE--> UA -> Setup new call with info from new contact header Does anybody now how to setup OpenSER, that he will execute a DB lookup for the user part of the R-URI and answers with a 301 response and the new address in the contact header? May I think the routing must look like this: Route[x] { NEW_URI = get_alias_for(1234); append_branch(NEW_URI); sl_send_reply("301" , "Number found"); } But how to set up and querie the database? Thx Reto

1 0

[Users] Question on re-writing From uri when ser forwards to Asterisk
by Barry Flanagan 20 Jan '06

20 Jan '06

Hi, I am using asterisk as PSTN gateway for users registered with OpenSER. Everything is working well for a single domain, but I will have problems with this when I add extra domains. I will have multiple domains using the OpenSER server, all using the same set of Asterisk servers for PSTN. My problem is that when user@domain1 gets sent to Asterisk, they are authenticated as just "user" (i.e. the domain portion is stripped off) and they get registered under the [user] sip name in Asterisk. Now, when I add the second domain, and user@domain2 goes to Asterisk, they also will appear to Asterisk to be [user] I think what I need to do is rewrite the From: URI to be user_domain1 and user_domain2 and then set them up in Asterisk as [user_domain1] and [user_domain2] so that they are unique So, my questions are: is this the correct/only way to do this, and if so, what do I need to do to achieve this? Many thanks. -- -Barry Flanagan

3 4

[Users] only one call from duplicate register
by Ibrahim TUNALI 20 Jan '06

20 Jan '06

i know i can multiple register on a sip proxy with same account at same time from different locations. And both of them can call different numbers. But i want to only one client can call, other one will block during first one's conversation and second one will recive "not acceptable" response. is it possible and can you give a configuration example. regards, itunali

1 0

[Users] NAT+uac_replace_from
by unplug 20 Jan '06

20 Jan '06

Actually, I am replacing the username of the uri with the alias that stored in the database. In my configuration file, it is using mediaproxy for NAT function (features-callfwd.5.0.cfg from getting started). I also add the following codes in the very first of the route routine for alias replacing purpose. route { ... if (!has_totag() && method=="INVITE") { if (avp_db_load("$from/uri","s:alias")) { xlog("L_INFO","sip408: have alias - [$avp(s:alias)]\n"); uac_replace_from("anonymous","sip:$avp(s:alias)@$si"); } else { xlog("L_INFO","sip411: no alias\n"); }; }; ... } When I make a call from a phone to the PSTN phone, the caller drops the ring when the callee rings. Callee hangs up and callee rings again few seconds after. You can find an error message "SIP/2.0 481 Call Leg/Transaction Does Not Exist" in line 180. I wonder if there is any thing wrong with the above code or my concept is wrong. Please help. Below is the sip message. interface: any filter: (ip) and ( port 5060 ) # U 10.0.0.46:5060 -> 10.0.0.27:5060 INVITE sip:34163634@10.0.0.27 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bKeKunMgjqbhggzOxi. Max-Forwards: 70. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-2d. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Contact: <sip:9000@10.0.0.46:5060>. CSeq: 1 INVITE. Supported: 100rel, replaces. Content-Type: application/sdp. Content-Length: 286. . v=0. o=9000 58685689 12166570 IN IP4 10.0.0.46. s=SIP CALL. c=IN IP4 10.0.0.46. t=0 0. m=audio 6000 RTP/AVP 18 3 0 8 4 101. a=rtpmap:18 G729/8000. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 10.0.0.27:5060 -> 10.0.0.46:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bKeKunMgjqbhggzOxi;rport=5060. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. CSeq: 1 INVITE. Server: OpenSer (1.0.0 (i386/linux)). Content-Length: 0. Warning: 392 10.0.0.27:5060 "Noisy feedback tells: pid=3353 req_src_ip=10.0.0.46 req_src_port=5060 in_uri=sip:34163634@10.0.0.27 out_uri=sip:934163634@10.30.0.248 via_cnt==1". . # U 10.0.0.27:5060 -> 10.30.0.248:5060 INVITE sip:934163634@10.30.0.248 SIP/2.0. Record-Route: <sip:34163634@10.0.0.27:5060;nat=yes;ftag=l7473818dCOs9ZMG;lr=on>. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bK0d2f.18572a04.0. Via: SIP/2.0/UDP 10.0.0.46:5060;rport=5060;branch=z9hG4bKeKunMgjqbhggzOxi. Max-Forwards: 69. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-2d. From: "anonymous"<sip:36418473@10.0.0.46>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Contact: <sip:9000@10.0.0.46:5060>. CSeq: 1 INVITE. Supported: 100rel, replaces. Content-Type: application/sdp. Content-Length: 287. . v=0. o=9000 58685689 12166570 IN IP4 10.0.0.46. s=SIP CALL. c=IN IP4 10.0.0.27. t=0 0. m=audio 35540 RTP/AVP 18 3 0 8 4 101. a=rtpmap:18 G729/8000. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 10.30.0.248:5060 -> 10.0.0.27:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bK0d2f.18572a04.0,SIP/2.0/UDP 10.0.0.46:5060;rport=5060;branch=z9hG4bKeKunMgjqbhggzOxi. From: "anonymous"<sip:36418473@10.0.0.46>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Date: Wed, 04 Jan 2006 09:15:35 GMT. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 10.30.0.248:5060 -> 10.0.0.27:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bK0d2f.18572a04.0,SIP/2.0/UDP 10.0.0.46:5060;rport=5060;branch=z9hG4bKeKunMgjqbhggzOxi. From: "anonymous"<sip:36418473@10.0.0.46>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Date: Wed, 04 Jan 2006 09:15:35 GMT. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Require: 100rel. RSeq: 1250. Allow-Events: telephone-event. Contact: <sip:934163634@10.30.0.248:5060>. Record-Route: <sip:34163634@10.0.0.27:5060;nat=yes;ftag=l7473818dCOs9ZMG;lr=on>. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 177. . v=0. o=CiscoSystemsSIP-GW-UserAgent 584 9780 IN IP4 10.30.0.248. s=SIP Call. c=IN IP4 10.30.0.248. t=0 0. m=audio 17340 RTP/AVP 18. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. # U 10.0.0.27:5060 -> 10.0.0.46:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 10.0.0.46:5060;rport=5060;branch=z9hG4bKeKunMgjqbhggzOxi. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Date: Wed, 04 Jan 2006 09:15:35 GMT. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Require: 100rel. RSeq: 1250. Allow-Events: telephone-event. Contact: <sip:934163634@10.30.0.248:5060>. Record-Route: <sip:34163634@10.0.0.27:5060;nat=yes;ftag=l7473818dCOs9ZMG;lr=on>. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 175. . v=0. o=CiscoSystemsSIP-GW-UserAgent 584 9780 IN IP4 10.30.0.248. s=SIP Call. c=IN IP4 10.0.0.27. t=0 0. m=audio 35540 RTP/AVP 18. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. # U 10.0.0.46:5060 -> 10.0.0.27:5060 PRACK sip:934163634@10.30.0.248:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bK8UeWxCfGrMJSO66f. Route: <sip:34163634@10.0.0.27:5060;nat=yes;ftag=l7473818dCOs9ZMG;lr=on>. Max-Forwards: 70. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-2d. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Contact: <sip:9000@10.0.0.46:5060>. CSeq: 2 PRACK. RAck: 1250 1 INVITE. Content-Length: 0. . # U 10.0.0.27:5060 -> 10.30.0.248:5060 PRACK sip:934163634@10.30.0.248:5060 SIP/2.0. Record-Route: <sip:10.0.0.27;ftag=l7473818dCOs9ZMG;lr=on>. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bKdc2f.dbee15d1.0. Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bK8UeWxCfGrMJSO66f. Max-Forwards: 69. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-2d. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Contact: <sip:9000@10.0.0.46:5060>. CSeq: 2 PRACK. RAck: 1250 1 INVITE. Content-Length: 0. . #----------------------------------line 180 -------------------------- U 10.30.0.248:5060 -> 10.0.0.27:5060 SIP/2.0 481 Call Leg/Transaction Does Not Exist. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bKdc2f.dbee15d1.0,SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bK8UeWxCfGrMJSO66f. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. CSeq: 2 PRACK. Content-Length: 0. . # U 10.0.0.27:5060 -> 10.0.0.46:5060 SIP/2.0 481 Call Leg/Transaction Does Not Exist. Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bK8UeWxCfGrMJSO66f. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. CSeq: 2 PRACK. Content-Length: 0. . # U 10.30.0.248:5060 -> 10.0.0.27:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bK0d2f.18572a04.0,SIP/2.0/UDP 10.0.0.46:5060;rport=5060;branch=z9hG4bKeKunMgjqbhggzOxi. From: "anonymous"<sip:36418473@10.0.0.46>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Date: Wed, 04 Jan 2006 09:15:35 GMT. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Require: 100rel. RSeq: 1250. Allow-Events: telephone-event. Contact: <sip:934163634@10.30.0.248:5060>. Record-Route: <sip:34163634@10.0.0.27:5060;nat=yes;ftag=l7473818dCOs9ZMG;lr=on>. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 177. . v=0. o=CiscoSystemsSIP-GW-UserAgent 584 9780 IN IP4 10.30.0.248. s=SIP Call. c=IN IP4 10.30.0.248. t=0 0. m=audio 17340 RTP/AVP 18. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes.

3 5

[Users] 481 - Call leg Transaction
by unplug 20 Jan '06

20 Jan '06

In what case will cause the following error response. Does it happen in call forwarding? 481 - Call Leg Transcation does Not Exist Do I need to handle it or just ignore it?

2 4

[Users] only one call from duplicate register
by Ibrahim TUNALI 19 Jan '06

19 Jan '06

1 0

[Users] textops regular expression matching = subst to replace from field
by Neill Wilkinson 18 Jan '06

18 Jan '06

Hi all, Looking to get a way to replace the From field in a REGISTER message to the forward on to another Registrar, as a third party registration I've tried something along the lines of: If (subs("/(^From:)([ ]*)/\1 sip:newaddress@new.server/I")) log(1,"Match!\n"); else log(1,"No match\n"); I basically need help with the Regex syntax and a good means of checking - is egrep or sed a good means of testing? Thanks..... Neill...;o) --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 0603-2, 17/01/2006 Tested on: 1/18/2006 11:16:18 avast! - copyright (c) 1988-2004 ALWIL Software. http://www.avast.com

2 4

[Users] call forward
by unplug 18 Jan '06

18 Jan '06

I am using the sample configuration file called "features-callfwd.5.0.cfg" with some modification. UA dials a 8 digits number and if it is not in the location table, prefix('9') will be appended in the number and route to the PSTN gateway. According to the configuration below, route 6 is responsible for call forward. With appropriate records in usr_preferences, openser can perform direct forward and forward on busy. However, in foward on no answer, one of the party can't hear any voice from the other party. I think the flow is something like this. UA1 openser UA5 UA6 ------INVITE----> --------INVITE------------> ... <-------time out/no answer- ------------------INVITE------------------> ... <--------------RTP established----------------------------------> result: UA6 can hear UA1, but UA1 can't get any sound from UA6 Anyone can help if there is any mistable in my configuration file to fix the above problem. Thanks! ------------start configuration------------- # $Id: features-callfwd.cfg 15 2005-09-15 17:37:10Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ debug=3 fork=yes log_stderror=no listen=203.193.16.226 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=yes rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" #fifo_mode=0666 #fifo permissions can be changed here log_facility=LOG_LOCAL0 alias=01.ol.com 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/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/uac.so" loadmodule "/usr/local/lib/openser/modules/group.so" modparam("auth_db|permissions|uri_db|usrloc|acc|group", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) #minimize write back window - default is 60 seconds modparam("usrloc", "timer_interval", 10) #use deomain information for user #modparam("usrloc", "use_domain", 1) modparam("registrar", "nat_flag", 6) #The maximum expires value of a Contact modparam("registrar", "max_expires", 120) #all contacts will be ordered in descending modification time order modparam("registrar", "desc_time_order", 1) #modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) #Timer which hits if no final reply for an INVITE modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("avpops", "avp_url", "mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) modparam("uac", "from_restore_mode", "auto") route { xlog("L_INFO","---[$fU-MR]---\n"); # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { xlog("L_INFO","---[$fU-main(91)]: 483 Too Many Hops---\n"); sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { xlog("L_INFO","---[$fU-main(97)]: 483 Message Overflow---\n"); sl_send_reply("513", "Message Overflow"); return; }; #----------------------------------------------------------------- # detect the anonymous incoming call and add anonymous to the uri #----------------------------------------------------------------- if (!has_totag() && (method=="INVITE"||method=="ACK"||method=="CANCEL"||method=="BYE")) { route(7); } #------------------------------------------------------------------ # Accoutnng Section #------------------------------------------------------------------ setflag(1); #t_relay(); # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("203.193.16.226:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { xlog("L_INFO","---[$fU-main(137)]: loose_route()---\n"); if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- #domian module if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { xlog("L_INFO","---[$fU-main(159)]: 403 Forbidden---\n"); sl_send_reply("403", "Forbidden"); }; return; }; if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; } else if (method=="ACK") { route(1); return; }; lookup("aliases"); #xlog("L_INFO","SIP line 191: [$fu] to [$tu]\n"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { xlog("L_INFO","---[$fU-main(188)]: 404 User Not Found---\n"); sl_send_reply("404", "User Not Found"); return; }; route(1); } route[1] { xlog("L_INFO","---[$fU-R1]---\n"); # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; #xlog("L_INFO","SIP Request line 220: [$rm] from [$fu] to [$tu]\n"); } route[2] { xlog("L_INFO","---[$fU-R2]---\n"); # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); return; }; if (!check_to()) { xlog("L_INFO","---[$fU-route2(237)]: 401 Unauthorized---\n"); sl_send_reply("401", "Unauthorized"); return; }; #------------check if the account is activated ---------- if (!is_user_in("From", "active")) { xlog("L_INFO","---[$fU-route2(244)]: 402 Payment Required---\n"); sl_send_reply("402", "Payment Required"); #append_hf("P-hint: PAYMANT REQUIRED\r\n"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { xlog("L_INFO","---[$fU-R3]---\n"); # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- #if (!allow_trusted()) { #if (!proxy_authorize("","subscriber")) { #proxy_challenge("","0"); #return; #} else if (!check_from()) { #sl_send_reply("403", "Use From=ID"); #return; #}; #consume_credentials(); #}; if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; #if (uri=~"^sip:1[0-9]{10}@") { #strip(1); #}; lookup("aliases"); #xlog("L_INFO","SIP line 292: [$fu] to [$tu]\n"); if (uri!=myself) { route(4); route(1); return; }; #if (uri=~"^sip:011[0-9]*@") { #route(4); #route(5); #return; #}; if (avp_db_load("$ruri/username", "s:callfwd")) { xlog("L_INFO","---[$fU-route3(305)]: destination uri: [$ru]---\n"); setflag(22); avp_pushto("$ruri", "s:callfwd"); xlog("L_INFO","---[$fU-route3(308)]: destination uri: [$ru]---\n"); route(6); return; }; if (!lookup("location")) { xlog("L_INFO","---[$fU]-(route3:315) destination uri: [$du]---\n"); if (uri=~"^sip:[0-9]*@") { xlog("L_INFO","---[$fU-route3(317)]: sip:(0-9*@)---\n"); prefix("9"); route(4); route(5); return; }; xlog("L_INFO","---[$fU-route3(324)]: 404 User Not Found---\n"); sl_send_reply("404", "User Not Found"); return; }; if (avp_db_load("$ruri/username", "s:fwdbusy")) { if (!avp_check("s:fwdbusy", "eq/$ruri/i")) { xlog("L_INFO","---[$fU-route3(331)]: setflag(26)---\n"); setflag(26); }; }; if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { if (!avp_check("s:fwdnoanswer", "eq/$ruri/i")) { xlog("L_INFO","---[$fU-route3(338)]: setflag(27)---\n"); setflag(27); }; }; t_on_failure("1"); route(4); route(1); } route[4] { xlog("L_INFO","---[$fU-R4]---\n"); # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { xlog("L_INFO","---[$fU-route4(357)]: $ru---\n"); if (!isflagset(8)) { xlog("L_INFO","---[$fU-route4(359)]: $ru---\n"); setflag(8); use_media_proxy(); }; }; } route[5] { xlog("L_INFO","---[$fU-R5]---\n"); # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("203.193.16.242"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); t_on_failure("1"); route(4); route(1); } route[6] { xlog("L_INFO","---[$fU-R6]---\n"); # ------------------------------------------------------------------------ # Call Forwarding Reply Route Handler # # This must be done as a route block because sl_send_reply() cannot be # called from the failure_route block # ------------------------------------------------------------------------ #if (uri=~"^sip:1[0-9]{10}@") { #strip(1); #}; lookup("aliases"); if (!isflagset(22)) { append_branch(); }; if (uri!=myself) { xlog("L_INFO","---[$fU-route6(405)]: (uri!=myyself)---\n"); route(4); route(1); return; }; #if (uri=~"^sip:011[0-9]*@") { } #if (uri=~"^sip:[0-9]*@") { #xlog("L_INFO","---[$fU-route6(415)]: (app prefix(9))---\n"); #prefix("9"); #route(4); #route(5); #return; #}; if (!lookup("location")) { if (uri=~"^sip:[0-9]*@") { xlog("L_INFO","---[$fU-route6(425)]: (sip:(0-9*@)---\n"); route(4); route(1); return; }; xlog("L_INFO","---[$fU-route6(432)]: 404 User Not Found---\n"); sl_send_reply("404", "User Not Found"); }; route(4); route(1); } route[7] { xlog("L_INFO","---[$fU-R7]---\n"); #if (method=="INVITE"||method=="ACK"||method=="CANCEL"||method=="BYE") { #if (is_user_in("from","")) { #xlog("L_INFO","SIP is_user_in=true\n"); #uac_replace_from("anonymous","sip:anonymous@s01.ol.com"); #} #} if (avp_db_load("$from/username","s:alias")) { xlog("L_INFO","---[$fU-route7(450)]: have alias from aliases- [$avp(s:alias)]---\n"); #uac_replace_from("$avp(s:alias)","sip:$avp(s:alias)@$si"); #uac_replace_from("$avp(s:alias) " ,""); } else { xlog("L_INFO","---[$fU-route7(454)]: no alias from aliases---\n"); }; } onreply_route[1] { xlog("L_INFO","---[$fU-OR1]---\n"); if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } failure_route[1] { xlog("L_INFO","---[$fU-FR1]---\n"); if (t_check_status("487")) { return; }; if (isflagset(26) && t_check_status("486")) { if (avp_pushto("$ruri", "s:fwdbusy")) { xlog("L_INFO","---[$fU-failure_route1(505)]: fwdbusy---\n"); avp_delete("s:fwdbusy"); resetflag(26); route(6); return; }; }; if (isflagset(27) && t_check_status("408")) { if (avp_pushto("$ruri", "s:fwdnoanswer")) { xlog("L_INFO","---[$fU-failure_route1(515)]: fwdnoanswer---\n"); avp_delete("s:fwdnoanswer"); resetflag(27); route(6); return; }; }; end_media_session(); } -----------end of configuration----------------

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[Users] How to check destination set after branch_route?
by Papadopoulos Georgios 18 Jan '06

18 Jan '06

Hello, I am trying to filter out requests going to some IPs. So, I have a branch_route block where I drop the branches that go to the IPs I want to block. After the t_on_branch command I call t_relay. This works fine if the callee has registered with at least one "good" IP. If the callee has registered with only one IP which I want to block then there is only one branch that is getting droped. The following t_relay results in a 500 Server error. Is there a way to check if all branches have been dropped? I tried using isdsturiset() but it did not work. Even if there is only one branch and this branch has been dropped, isdsturiset() returns true. Is this correct or am I wrong somewhere? thank you for any help George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

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sr-users January 2006