sr-users September 2005

sr-users@lists.kamailio.org

231 participants
507 discussions

by Richard Z

Hi, I am trying to figure out how to use branch route for nat traversal. If a subscriber registers from multiple locations, a branch route is called after lookup. For example, if locations include UAs behind same NAT and different NAT, i'd like to invoke any rtp/media-proxy accordingly in branch_route[], . However they can't be called inside a branch_route. What's the best way to do this? Thanks, Richard

18 years, 9 months

[Serusers] Try to do effective routing using sip server

by Prajyot

Hi all, Can anyone please tell me how to configure SIP ser server for effective routing based on DNIS. Right now its pretty static in nature, meaning that we have if queries based on the DNIS and the invite is forwarded to a server. What I am trying to achieve is load balancing in which I can specify more than one server for routing to a particular DNIS. Please guide me, thanks in advance. Regards, Prajyot Shirsolkar

18 years, 9 months

Re: [Serusers] ONSIP Script + Java Rockx ser 0.9.0 script

by Iqbal

Hi @Mark: I agree, but some people only really need ser, or should I say if your route to much to asterisk its pointless having ser, I keep my Ip__IP traffic in ser, in fact my billing there also, asterisk as you mentioned for features, and when my upstream gateway wont support REFER to do transfers, and cool things like call pickup. @Aisling: if (method=="INVITE"){ if(is_user_in("Request-URI", "voicemail")){ setflag(31); }; }; what this will do is to see if the username you have in RURI is in your grp table and has an entry of voicemail next to it, if so it will set a flag 31 if (method=="INVITE") { > avp_write("$from", "i:34"); > if (avp_check("i:34", "eq/$ruri/i")) { > > if (isflagset(31)) { > route(8); > break; > } else { > sl_send_reply("486", "Busy"); > break; > }; > }; > }; what this does is basically to see if your are calling yourself, i.e if the from and the RURI is the same, first it stores the from setting, and the compares that to your ruri.Now if they are the same, it will see if flag 31 is set, i.e if you have voicemail attached to the RURI in grp table. If both are true then goto route8, at which point it seems as if your are relaying to another server. You seem to be getting busy, which means that flag 31 is not being set, check your ngrep dump see what your RURI has, and see what entries your grp table has. The problem with matching like this (or at least the problem(s) I had) was that there are alot of different call scenarios which all need to be taken into account. EG my usernames are not e164 numbers, they are part of that number, which means my ruri, and from can be different to the e164 format. Now when a caller calls from 040600, to 0845040602, it should see if 0845xxx is busy, send to voicemail, but in grp ther is no 0845xxx, hence I need to make sure that the check/match is done after the alias lookup. I had to move lookup(aliases) near the to to make sure all my lookups had been done at the start, and then I can start matching. Your best friend will be ngrep, if you start doing the diff call scenarios, and seeing what needs to be matched, you can set 1 run settings, once it all works work out a tidier logic. If you throw asterisk into the loop, you add another headache (depending on what you do), cause asterisk was really designed for ip devices to register with it, but if you register in ser, you could lose some of that info (although latest beta, has some nice SIP header parsing, to pull info from). The other problem you may hit with voicemail, depends on how you set your username, initially my subscriber table had names for username...eg Iqbal, and the alias would be 040600, or whatever, problem with this was that voicemail in asterisk is better with number, especially when you have to dial the mailbox number from your phone :-), so I switched the subscriber table to hold the number, cause I think unless you will be using software dialers, no one will type in names. "voicemail=1:500;calltype=i:700;fwd_no_answer_type=1:701;fwd_busy_type >=1:702") All this does is to define aliases, you can use it, or not iqbal Mark Aiken wrote: > I see a lot of people trying to use SER and Asterisk together but > mixing the roles in strange and complex ways. > > Asterisk is an excellent SIP "feature server" and "b2bua" where SER is > an excellent SIP proxy, registrar, and far-end NAT solution. > > If you design your network based on those strengths you can create a > solution that rivals 7 figure commercial solutions. People that do > otherwise seem to just fall over themselves. > > Mark > > > On 9/21/05, *Aisling* <ashling.odriscoll(a)cit.ie > <mailto:ashling.odriscoll@cit.ie>> wrote: > > Many thanks Iqbal- That was the problem. > > I think the issue is that I don't even really understand what that > line > does or how the code that I took from Paul's config affected my config > file.....SER now displays no errors but is still not working as > expected. For example I expect that if a phone (e.g. 2092) dials > its own > extension, it should be caught by > > if (method=="INVITE"){ > if(is_user_in("Request-URI", "voicemail")){ > setflag(31); > }; > }; > > This then should be caught by the URI Compare code below and routed to > route[8].... > > However that's not happening and a 486 Busy is being returned to the > phone. I find this hard to troubleshoot because I think the code > should > be correct...given my current understanding anyhow... > > If Iqbal (or anyone else) has a clearer understanding of this and can > shed some light on it, that would be great. > > Kindest Regards, > Aisling. > > -----Original Message----- > From: Iqbal [mailto:iqbal@gigo.co.uk <mailto:iqbal@gigo.co.uk>] > Sent: 21 September 2005 15:20 > To: Aisling O'Driscoll > Cc: serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> > Subject: Re: [Serusers] ONSIP Script + Java Rockx ser 0.9.0 script > > "voicemail=1:500;calltype=i:700;fwd_no_answer_type=1:701;fwd_busy_type > =1:702") > > try a voicemail=i:500 not 1:500 > > Iqbal > > > > Aisling O'Driscoll wrote: > > >Hello everyone, > > > >I am trying to modify the Onsip call forward script to forward to > >asterisk voicemail given a users preference. I am using the script > >that Java RockX (Paul) posted in January... > >http://lists.iptel.org/pipermail/serusers/2005-January/014968.html > > > >I think I am getting very confused though with the logic...I want > >users to be able to load their preference using the avpops module of > >whether they want call forward on no answer or voicemail. They get > >directed to voicemail if they dial their own extension number....I > >have posted the modified script which is a combination of the Pauls > >script and the Onsip script...When I start SER I currently have these > >errors but I'd say these are a consequence of my messed up logic > >;).... > > > >0(20031) ERROR:parse_avp_name: unsupported type [1] > > 0(20031) ERROR:add_avp_galias_str: <500> not a valid AVP name > > 0(20031) parse error (70,20-21): Can't set module parameter > > > >The config is below.... > >If anyone can offer some insight, Id be very grateful. > >Many thanks, > >Aisling. > > > ># $Id: nat-rtpproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. > >Teigre/emailAddress= greger(a)onsip.org <mailto:greger@onsip.org> $ > >#debug=3 > >#fork=yes > >#log_stderror=no > > > >#listen=x.x.x.x # INSERT YOUR IP ADDRESS HERE > >#port=5060 > >#children=4 > > > >check_via=no > >dns=no > >rev_dns=no > >fifo="/tmp/ser_fifo" > >fifo_db_url="mysql://root:password@localhost/ser" > > > >alias="x.x.x.x:5060" > > > >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.so" > >loadmodule "/usr/local/lib/ser/modules/uri_db.so" > >loadmodule "/usr/local/lib/ser/modules/nathelper.so" > >loadmodule "/usr/local/lib/ser/modules/textops.so" > >loadmodule "/usr/local/lib/ser/modules/cpl-c.so" > >loadmodule "/usr/local/lib/ser/modules/avpops.so" > >loadmodule "/usr/local/lib/ser/modules/permissions.so" > >loadmodule "/usr/local/lib/ser/modules/speeddial.so" > >loadmodule "/usr/local/lib/ser/modules/group.so" > > > >modparam("auth_db|permissions|uri_db|usrloc", "db_url", > >"mysql://root:password@localhost/ser") > >modparam("auth_db", "calculate_ha1", 1) > >modparam("auth_db", "password_column", "password") > > > >modparam("nathelper", "natping_interval", 30) #ping every 30 seconds > >modparam("nathelper", "ping_nated_only", 1) > >modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") > > > >modparam("cpl-c", "cpl_db", "mysql://root:password@localhost/ser") > >modparam("cpl-c", "cpl_table", "cpl") > >modparam("cpl-c", "cpl_dtd_file", > >"/tmp/ser-0.9.3/modules/cpl-c/cpl-06.dtd") > >modparam("cpl-c", "lookup_domain", "location") > > > >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("usrloc", "db_mode", 2) > > > >modparam("registrar", "nat_flag", 6) > > > >modparam("rr", "enable_full_lr", 1) > > > >modparam("speeddial", "db_url", > "mysql://root:password@localhost/ser") > >modparam("speeddial", "user_column", "userid") > >modparam("speeddial", "sd_user_column", "short_user") > >modparam("speeddial", "sd_domain_column", "short_domain") > >modparam("speeddial", "new_uri_column", "real_uri") > > > >modparam("avpops", "avp_url", "mysql://root:password@localhost/ser") > >modparam("avpops", "avp_table", "usr_preferences") > >modparam("avpops", "avp_aliases", > >"voicemail=1:500;calltype=i:700;fwd_no_answer_type=1:701;fwd_busy_type > >=1:702") > > > >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; > > }; > > > > > #------------------------------------------------------------------ > > # NOTIFY Keep Alive Section > > > #------------------------------------------------------------------ > > if ((method=="NOTIFY") && search("^Event: keep-alive")){ > > sl_send_reply("200", "OK"); > > break; > > }; > > > > > #------------------------------------------------------------------ > > # Speed Dialing Section > > > #------------------------------------------------------------------ > > if ((method=="INVITE") && (uri=~"^sip:[0-9]{2}@.*")) { > > sd_lookup("speed_dial"); > > }; > > > > > #------------------------------------------------------------------ > > # Do not Disturb Section > > > #------------------------------------------------------------------ > > #if(avp_db_load("$ruri/username", "s:donotdisturb")){ > > # if(avp_check("s:donotdisturb", "eq/y/i")){ > > # route(x) #whereever asterisk voicemail is > > # break; > > # }; > > #}; > > > > # > ----------------------------------------------------------------- > > # Record Route Section > > # > ----------------------------------------------------------------- > > if (method!="REGISTER") { > > record_route(); > > }; > > > > if (method=="BYE" || method=="CANCEL") { > > unforce_rtp_proxy(); > > }; > > > > if (method=="INVITE"){ > > if(is_user_in("Request-URI", "voicemail")){ > > setflag(31); > > }; > > }; > > > > # > >---------------------------------------------------------------------- > >-- > > # URI Compare Section > > # > > # Here we compare the "from" and "to" to see if the > caller is > >dialing > > # their own extension. If so then we route to voicemail if > >needed > > # > >---------------------------------------------------------------------- > >-- > > if (method=="INVITE") { > > avp_write("$from", "i:34"); > > if (avp_check("i:34", "eq/$ruri/i")) { > > > > if (isflagset(31)) { > > route(8); > > break; > > } else { > > sl_send_reply("486", "Busy"); > > break; > > }; > > }; > > }; > > > > # > ----------------------------------------------------------------- > > # Loose Route Section > > # > ----------------------------------------------------------------- > > if (loose_route()) { > > > > if (has_totag() && (method=="INVITE" || > method=="ACK")) > { > > if (nat_uac_test("19")) { > > setflag(6); > > force_rport(); > > fix_nated_contact(); > > }; > > force_rtp_proxy("l"); > > }; > > route(1); > > break; > > }; > > > > # > ----------------------------------------------------------------- > > # Call Type Processing Section > > # > ----------------------------------------------------------------- > > > > if (uri!=myself) { > > route(4); > > route(1); > > break; > > }; > > > > if (method=="CANCEL") { > > route(1); > > break; > > } else if (method=="INVITE") { > > if(!cpl_run_script("incoming", "is_stateless")) > > { > > # script execution failed > > t_reply("500", "CPL script execution failed"); > > }; > > route(3); > > break; > > } else if (method=="REGISTER") { > > #handle REGISTER messages with CPL script > > cpl_process_register(); > > route(2); > > break; > > }; > > > > lookup("aliases"); > > if (uri!=myself) { > > route(4); > > route(1); > > break; > > }; > > > > if (!lookup("location")) { > > > > if(isflagset(31)){ > > setflag(19); > > }; > > > > #sl_send_reply("404", "User Not Found"); > > #break; > > }; > > > > route(1); > >} > > > >route[1] { > > > > # > ----------------------------------------------------------------- > > # Default Message Handler > > # > ----------------------------------------------------------------- > > > > t_on_reply("1"); > > > > if (!t_relay()) { > > if (method=="INVITE" && isflagset(6)) { > > unforce_rtp_proxy(); > > }; > > sl_reply_error(); > > }; > >} > > > >route[2] { > > > > # > ----------------------------------------------------------------- > > # REGISTER Message Handler > > # > ---------------------------------------------------------------- > > > > if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { > > setflag(6); > > fix_nated_register(); > > force_rport(); > > }; > > > > sl_send_reply("100", "Trying"); > > > > if (!www_authorize("","subscriber")) { > > www_challenge("","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("","subscriber")) { > > proxy_challenge("","0"); > > break; > > } else if (!check_from()) { > > sl_send_reply("403", "Use From=ID"); > > break; > > }; > > > > #consume_credentials(); > > > > if (nat_uac_test("19")) { > > setflag(6); > > } > > > > lookup("aliases"); > > if (uri!=myself) { > > route(4); > > route(1); > > break; > > }; > > > > #Blind Call Forwarding > > if (avp_db_load("$ruri/username", "s:callfwd")){ > > setflag(22); > > avp_pushto("$ruri", "s:callfwd"); > > route(6); > > break; > > }; > > > > if (!lookup("location")) { > > sl_send_reply("404", "User Not Found"); > > break; > > }; > > > > if (avp_db_load("$ruri/username", "s:fwdbusy")){ > > if(!avp_check("s:fwdbusy", "eq/$ruri/i")){ > > setflag(26); > > }; > > }; > > > > if (avp_db_load("$ruri/username", "s:fwdnoanswer")){ > > if(!avp_check("s:fwdnoanswer", "eq/$ruri/i")){ > > setflag(27); > > }; > > }; > > > > t_on_failure("1"); > > > > route(4); > > route(1); > >} > > > >route[4] { > > > > # > ----------------------------------------------------------------- > > # NAT Traversal Section > > # > ----------------------------------------------------------------- > > > > if (isflagset(6)) { > > force_rport(); > > fix_nated_contact(); > > force_rtp_proxy(); > > }; > >} > > > >route[5]{ > > > > > #----------------------------------------------------------------- > > # PSTN Handler > > > #----------------------------------------------------------------- > > > > revert_uri(); > > rewritehostport(" x.x.x.x:5064"); > > append_branch(); > > t_relay_to_udp("x.x.x.x", "5064"); > > break(); > > > > t_on_failure("1"); > > > > route(4); > > route(1); > >} > > > >route[6]{ > > > > > #------------------------------------------------------------------ > > # Call Forwarding 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(uri!=myself){ > > if(!isflagset(22)){ > > append_branch(); > > }; > > > > route(4); > > route(1); > > break; > > }; > > > > if(uri=~"^sip:011[0-9]*@"){ > > route(4); > > route(5); > > break; > > }; > > > > if(!lookup("location")){ > > if (uri=~"^sip:[0-9][10]@"){ > > route(4); > > route(1); > > break; > > }; > > sl_send_reply("404", "User Not Found"); > > }; > > > > route(4); > > route(1); > >} > > > >route[8] { > > > > # voicemail route #2 > > # > > # this path this executed during these conditions: > > # > > # cond 1) the called number is not in the location > table > > # cond 2) the from_uri == to_uri (ie, caller==callee) > > > > if (method == "INVITE" || method == "ACK"){ > > force_rtp_proxy(); > > }; > > > > rewritehostport("x.x.x.x:5064"); > > > > t_on_reply("1"); > > > > if (!t_relay()) { > > > > if(method == "INVITE" || method == "ACK"){ > > unforce_rtp_proxy(); > > }; > > > > sl_reply_error(); > > }; > >} > > > >route[9]{ > > > > #voicemail route 1 > > # > > #this path is executed during these conditions: > > # > > # cond 1) the called number is in the location table > > # but the callee did not answer the phone ( i.e. > failover > to > >voicemail) > > > > rewritehostport("x.x.x.x:5064"); > > append_branch(); > > > > t_on_reply("1"); > > > > if (!t_relay()){ > > if(method == "INVITE" || method=="ACK"){ > > unforce_rtp_proxy(); > > }; > > sl_reply_error(); > > }; > >} > > > >onreply_route[1] { > > > > if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { > > if (!search("^Content-Length:[ ]*0")) { > > force_rtp_proxy(); > > }; > > }; > > > > if (nat_uac_test("1")) { > > fix_nated_contact(); > > }; > >} > > > >failure_route[1]{ > > > > if(t_check_status("487")){ > > break; > > }; > > > > if(isflagset(26) && t_check_status("486")){ > > if(avp_pushto("$ruri", "s:fwdbusy")){ > > avp_delete("s:fwdbusy"); > > resetflag(26); > > append_branch(); > > route(6); > > break; > > }; > > }; > > > > #Here we can have either voicemail OR forward no answer > > #forward on no answer is flag 27 > > #voicemail is flag 31 > > > > if(isflagset(27) && t_check_status("408")){ > > if(avp_pushto("$ruri", "s:fwdnoanswer")){ > > avp_delete("s:fwdnoanswer"); > > resetflag(27); > > append_branch; > > route(6); > > break; > > }; > > } > > else if(isflagset(31) && avp_pushto("$ruri", "$voicemail")){ > > avp_delete("$voicemail"); > > route(9); > > break; > > }; > >} > > > > > > > >-------------------Legal > Disclaimer--------------------------------------- > > > >The above electronic mail transmission is confidential and intended > only for the person to whom it is addressed. Its contents may be > protected by legal and/or professional privilege. Should it be > received > by you in error please contact the sender at the above quoted email > address. Any unauthorised form of reproduction of this message is > strictly prohibited. The Institute does not guarantee the security of > any information electronically transmitted and is not liable if the > information contained in this communication is not a proper and > complete > record of the message as transmitted by the sender nor for any > delay in > its receipt. > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> > >http://lists.iptel.org/mailman/listinfo/serusers > > > >. > > > > > > > > -------------------Legal > Disclaimer--------------------------------------- > > The above electronic mail transmission is confidential and > intended only > for the person to whom it is addressed. Its contents may be > protected by > legal and/or professional privilege. Should it be received by you in > error please contact the sender at the above quoted email address. Any > unauthorised form of reproduction of this message is strictly > prohibited. The Institute does not guarantee the security of any > information electronically transmitted and is not liable if the > information contained in this communication is not a proper and > complete > record of the message as transmitted by the sender nor for any > delay in > its receipt. > > > -------------------Legal Disclaimer--------------------------------------- > > The above electronic mail transmission is confidential and > intended only for the person to whom it is addressed. Its contents > may be protected by legal and/or professional privilege. Should it > be received by you in error please contact the sender at the above > quoted email address. Any unauthorised form of reproduction of > this message is strictly prohibited. The Institute does not > guarantee the security of any information electronically > transmitted and is not liable if the information contained in this > communication is not a proper and complete record of the message > as transmitted by the sender nor for any delay in its receipt. > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers > <http://lists.iptel.org/mailman/listinfo/serusers> > >

18 years, 9 months

[Serusers] Does VOVIDA B2BUA works with SER+MEDIAPROXY?

by Thomas Tomter

Hello, We are now trying to integrate SER+Vovida B2BUA with Alepo RBS version 4.2 and I was wondering whether you succeeded in making it work? Would it be possible for you to share with us how you configured SER+ B2UBA in order to make this work? We are particularly interested in making SER+B2BUA act as a Cisco device towards the Alepo billing system. Thanks in advance. Best regards, Thomas Tomter

18 years, 9 months

[Serusers] Field STATE

by Alberto

Hi, Where can I get information by field STATE? When I execute the command "serctl alias show", I can see the field STATE and I have watched the values: CS_SYNC, CS_DIRTY and CS_NEW, which is the meaning of these values? Thanks, -- Alberto

18 years, 9 months

[Serusers] cisco 7905G

by Alberto

Hi, Somebody has experience with cisco7905G? The 7905G is registered OK and can receive calls but not send call. Thanks, -- Alberto

18 years, 9 months

RE: [Serusers] allow_trusted removed from permission module (HEAD)?

by Bjorn Asmul

The README should be updated then. Unless I'm really off here, the allow_trusted-section has been taken out. Bjorn -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Monday, September 26, 2005 2:03 AM To: Bjorn Asmul Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] allow_trusted removed from permission module (HEAD)? No, it is still there. Jan. On 25-09-2005 17:55, Bjorn Asmul wrote: > It looks like "allow_trusted" was removed from the permission module > in CVS HEAD. > Can someone confirm this? Or was it moved to a different module? > > Bjorn > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

18 years, 9 months

[Serusers] Alias routing section in http://lists.iptel.org/pipermail/serusers/2005-February/015700.html

by Ramin Nikaeen

I have looked at the sample ser.cfg: http://lists.iptel.org/pipermail/serusers/2005-February/015700.html Can someone tell me what the following section in this file is for: # ------------------------------------------------------------------------ # Alias Routing Section # ------------------------------------------------------------------------ lookup("aliases"); if (!uri==myself) { route(2); break; }; thanks ramin

18 years, 9 months

[Serusers] SER Vs. SIPsak = 483, incoerence!!!!

by Ángel Gómez

I´m trying to test my SER server, but... wen i use sipsak to test it, with this comand: sipsak -s sip:ninguna@10.22.1.249 -vvv -f mensage.txt I got this output with sipsak: fqdnhostname: local our Via-Line: Via: SIP/2.0/UDP local:2288;rport New message with Via-Line: MESSAGE sip:prueba@10.22.1.249 SIP/2.0 Via: SIP/2.0/UDP local:2288;rport From: <sip:prueba@10.22.1.249> To: <sip:ninguna@10.22.1.249> Call-ID: 917549908@local CSeq: 2 MESSAGE Content-Type: text/plain Content-Length: 48 Max-Forwards: 7 User-Agent: sipsak 0.8.9_pre usrloc test message from SIPsak for user ninguna0.sending message to ninguna... ** request ** MESSAGE sip:prueba@10.22.1.249 SIP/2.0 Via: SIP/2.0/UDP local:2288;rport From: <sip:prueba@10.22.1.249> To: <sip:ninguna@10.22.1.249> Call-ID: 917549908@local CSeq: 2 MESSAGE Content-Type: text/plain Content-Length: 48 Max-Forwards: 7 User-Agent: sipsak 0.8.9_pre usrloc test message from SIPsak for user ninguna0.sending message to ninguna... message received: SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP local:2288;received=10.22.1.5;rport=2288 From: <sip:prueba@10.22.1.249> To: <sip:ninguna@10.22.1.249>;tag=b27e1a1d33761e85846fc98f5f3a7e58.191f Call-ID: 917549908@local CSeq: 2 MESSAGE Server: Sip EXpress router (0.9.3 (i386/linux)) Content-Length: 0 Warning: 392 10.22.1.249:5060 "Noisy feedback tells: pid=1649 req_src_ip=10.22. 1.249 req_src_port=5060 in_uri=sip:prueba@10.22.1.249 out_uri=sip:prueba@10.22.1 .249 via_cnt==8" ** reply received after 0.000 ms ** SIP/2.0 483 Too Many Hops final received And the incoerence is because in ser.cfg I have THIS line (I think): if (!mf_process_maxfwd_header("16")) { sl_send_reply("483","Too Many Hops"); break; }; the mensage isn´t looping (or i thing so), i have alias in the config file and both user of the mensage (from and to) are registered in the server. Can some one help me???

18 years, 9 months

[Serusers] Seeking Advise With Caller ID Block

by Chris Tsai

hi, I have a question about blocking caller id at SER (public proxy) if I want to blocking my caller id to invite other user at IPtel.org How do I do ? How about the INVITE message format needs? I try it the messae as follows: but it's fail.........>< Request-Line: INVITE sip:8201@iptel.org SIP/2.0 Message Header Via: SIP/2.0/UDP 61.216.39.201:5060;branch=z9hG4bK-hrs-21193 From: Anonymous <sip:anonymous@iptel.org>;tag=DBb-18789 To: <sip:8201@iptel.org> Call-ID: gnA-11414(a)61.216.39.201 CSeq: 8 INVITE Contact: <sip:christsai12@61.216.39.201> Max-Forwards: 70 Remote-Party-ID: Chris <sip:christsai12@iptel.org>;privacy=full;party=calling supported: replaces User-Agent: DrayTek UA-1.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY, SUBSCRIBE Content-Type: application/sdp Content-Length: 270 Message body Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header Via: SIP/2.0/UDP 61.216.39.201:5060;branch=z9hG4bK-hrs-21193;rport=5060;received=61.216.35.30 From: Anonymous <sip:anonymous@iptel.org>;tag=DBb-18789 To: <sip:8201@iptel.org>;tag=b51ece8fd8c195776737473dd6552d43.8aa7 Call-ID: gnA-11414(a)61.216.39.201 CSeq: 8 INVITE P-Behind-NAT: Yes Proxy-Authenticate: Digest realm="iptel.org", nonce="42bf93c8d6bc77f7ef3e6e8f24513796fda5bccc" Server: Sip EXpress router (0.9.0udpfifo (i386/linux)) Content-Length: 0 Warning: 392 195.37.77.99:5060 "Noisy feedback tells: pid=30610 req_src_ip=61.216.35.30 req_src_port=5060 in_uri=sip:8201@iptel.org out_uri=sip:8201@iptel.org via_cnt==1" Session Initiation Protocol Request-Line: ACK sip:8201@iptel.org SIP/2.0 Message Header Via: SIP/2.0/UDP 61.216.39.201:5060;branch=z9hG4bK-hrs-21193 From: Anonymous <sip:anonymous@iptel.org>;tag=DBb-18789 To: <sip:8201@iptel.org>;tag=b51ece8fd8c195776737473dd6552d43.8aa7 Call-ID: gnA-11414(a)61.216.39.201 CSeq: 8 ACK Content-Length: 0 Session Initiation Protocol Request-Line: INVITE sip:8201@iptel.org SIP/2.0 Message Header Via: SIP/2.0/UDP 61.216.39.201:5060;branch=z9hG4bK-mWf-3372 From: Anonymous <sip:anonymous@iptel.org>;tag=DBb-18789 To: <sip:8201@iptel.org> Call-ID: gnA-11414(a)61.216.39.201 CSeq: 9 INVITE Contact: <sip:christsai12@61.216.39.201> Proxy-Authorization: Digest username="christsai12", realm="iptel.org", nonce="42bf93c8d6bc77f7ef3e6e8f24513796fda5bccc", uri="sip:8201@iptel.org", response="d1cb74bf6a92df9edf18e565ee2eac79", algorithm=MD5 Max-Forwards: 70 Remote-Party-ID: Chris <sip:christsai12@iptel.org>;privacy=full;party=calling supported: replaces User-Agent: DrayTek UA-1.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY, SUBSCRIBE Content-Type: application/sdp Content-Length: 270 Message body Session Initiation Protocol Status-Line: SIP/2.0 403 That is ugly -- use From=id next time (OB) Message Header Via: SIP/2.0/UDP 61.216.39.201:5060;branch=z9hG4bK-mWf-3372;rport=5060;received=61.216.35.30 From: Anonymous <sip:anonymous@iptel.org>;tag=DBb-18789 To: <sip:8201@iptel.org>;tag=b51ece8fd8c195776737473dd6552d43.b29b Call-ID: gnA-11414(a)61.216.39.201 CSeq: 9 INVITE P-Behind-NAT: Yes Server: Sip EXpress router (0.9.0udpfifo (i386/linux)) Content-Length: 0 Warning: 392 195.37.77.99:5060 "Noisy feedback tells: pid=30606 req_src_ip=61.216.35.30 req_src_port=5060 in_uri=sip:8201@iptel.org out_uri=sip:8201@iptel.org via_cnt==1"

18 years, 9 months

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sr-users September 2005