- sr-users - kamailio.org

[Serusers] use of SRV records for failover
by Markus Storm 27 Sep '05

27 Sep '05

Dear all, we would like to do loadbalancing and failover with SRV in our multi-SER setup. I.e. ser-1 gets a call and can forward the call to either ser-2 or ser-3. There's a DNS SRV record that points to both, ser-2 and ser-3 (same weight, same priority). On ser-1, I use the SRV name to forward calls to. Now I'd like ser-1 to distribute calls according to weight and priority, i.e. 50% to ser-2 and 50% to ser-3. But it doesn't, it keeps sending to ser-2 (the first entry in the SRV record's list). Also, if say ser-2 goes down, I'd like ser-1 to notice that and only ever try to send subsequent requests to ser-2 until <something> happens (e.g. until the SRV record TTL expires). I found a one-year old message in the archives stating that neither of this is possible (unless I use a DNS A record that points to multiple IP addresses, but that's not what I want). Is it meanwhile possible with the current SER software ? thanks Markus

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[Serusers] SER 0.9.3 core dumps
by Darren Nay 27 Sep '05

27 Sep '05

Hey All, I installed SER 0.9.3 about 3 weeks ago and since then have been getting core dumps about every 7-8 days consistently. Acts almost like a progressive memory leak. So .. I've noticed that release 0.9.4 has also been released about 2 weeks ago with a few bug fixes (including a memory leak or two). However, the CVS head is also available with 0.9.5-pre1. So .. What I really need to know is.. Which version should I use? :) Thanks, Darren Nay

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[Serusers] lcr
by Sebastian Kühner 27 Sep '05

27 Sep '05

Hello, I'm using the lcr module with prefix capability to make failover. When I use a "big" IP, the lcr module gives back a wrong IP. It was very hard for me to compile lcr with prefix capability and ser-0.9.3... has anyone of you a lcr module with prefix-support for ser-0.9.3 and without this bug? Thanks! Sebastian

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[Users] Route, Record Route and quintum
by Iqbal 27 Sep '05

27 Sep '05

Hi I have a strange looking dialogue, after sending a 200 OK to a quintum box, and get back an ACK, but the Route field seems to be wrong, can someone confirm this. -------- U 2005/09/06 15:23:16.613003 111.222.333.444:5060 -> 10.20.30.40:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.20.30.40;branch=z9hG4bK-tenor-c3ac-7b24-0018. From: <sip:046677@10.20.30.40>;tag=c3ac7b24-a. To: <sip:123456789@111.222.333.444>;tag=1A4297F4-2C9. Date: Tue, 06 Sep 2005 14:07:59 gmt. Call-ID: call-00CC9E9D-4FBE-D311-0207-A(a)10.20.30.40. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:0044123456789@212.156.5.134:5060>. Record-Route: <sip:212.156.5.135;ftag=c3ac7b24-a;lr=on>,<sip:111.222.333.444;ftag=c3ac7b24-a;lr=on>. Content-Type: application/sdp. Content-Length: 206. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7438 1091 IN IP4 212.156.5.134. s=SIP Call. c=IN IP4 212.156.5.134. t=0 0. m=audio 18590 RTP/AVP 18. c=IN IP4 212.156.5.134. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. # U 2005/09/06 15:23:16.726095 10.20.30.40:5060 -> 111.222.333.444:5060 ACK sip:111.222.333.444;ftag=c3ac7b24-a;lr=on SIP/2.0. CSeq: 1 ACK. Call-ID: call-00CC9E9D-4FBE-D311-0207-A(a)10.20.30.40. Contact: <sip:046677@10.20.30.40>. From: <sip:046677@10.20.30.40>;tag=c3ac7b24-a. Route: <sip:0044123456789@212.156.5.134:5060>. Session-GUID: 859321956-876164920-926430820-875782963. To: <sip:123456789@111.222.333.444>;tag=1A4297F4-2C9. Via: SIP/2.0/UDP 10.20.30.40;branch=z9hG4bK-tenor-c3ac-7b24-0018. User-Agent: Quintum/1.0.0. Quintum: 0c01030b0232360501000715000000000000000d006c0a828180803034363637371301000f0b413031322d313033353941. Max-Forwards: 70. ---------------- I thought the Route field should have the data which it pulls from the Record Route headers tks iqbal

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[Serusers] SER callerid/ANI removal
by Jason Mohler 27 Sep '05

27 Sep '05

I am trying to find a way to set my SER caller ID to unknown. What I trying to accomplish is to not have calls coming with an ANI. It can be unknown , anonymous, etc. But I do not want the ANI/caller ID number showing. Can anyone help me answer this question? I think it would be an rpid setting but not certain.

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[Users] use_domain not found in avpops
by Franz Edler 27 Sep '05

27 Sep '05

Hi, I just wanted to use the parameter use_domain in avpops, but I got the error "<use_domain> not found in module <avpops>". A quick check in module avpops.c shows that the variable declaration for use_domain is not declared as a pointer. Might this be the cause? Regards Franz

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[Serusers] is SER ( 0.9.4) 100% 3GPP SIP call flow compliant ?
by ashish gilotra 27 Sep '05

27 Sep '05

Hi We are evaluating SER for our purpose. We need to use 3GPP SIP call flows. Is SER compliant ? Thanks Ash __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] SER Vs. SIPsak = 483, incoerence!!!!
by Ángel Gómez 27 Sep '05

27 Sep '05

Nils, thank for making me se this. Despite of this i thin this is starnger, because i maque a REGISTER or INVITE and this doesn´t happen. Now I´m sending my ser.cfg files. To see wath is wrong. # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode */ #fork=no #log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 children=4 fifo="/tmp/ser_fifo" listen=10.22.1.249 alias="10.22.1.249" alias="213.194.156.186" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 1) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("16")) { sl_send_reply("483","Too Many Hops"); }; if (msg:len > max_len) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } Thack for all.

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[Serusers] Windows Messenger 5.1 -> SER authentication fails
by davor jovanovic 27 Sep '05

27 Sep '05

Hi all, I have a problem with using Windows Messenger 5.1, that is, I can't log in with SER 0.9.3. Everything is working OK with Xlite client. I try to set up my realm value to equal server name as it says in Admin guide for SER, but it doesn't help. If someone had similar problem and solve it, I would be very thankful if you can share solution with me. Thanks This is output from SER when trying to authenticate with WM. 0(22883) parse_headers: flags=-1 0(22883) check_via_address(192.168.0.101, 192.168.0.101, 0) 0(22883) DEBUG:destroy_avp_list: destroying list (nil) 0(22883) receive_msg: cleaning up 0(22883) SIP Request: 0(22883) method: <REGISTER> 0(22883) uri: <sip:debtest.stu.du> 0(22883) version: <SIP/2.0> 0(22883) parse_headers: flags=1 0(22883) end of header reached, state=5 0(22883) parse_headers: Via found, flags=1 0(22883) parse_headers: this is the first via 0(22883) After parse_msg... 0(22883) preparing to run routing scripts... 0(22883) parse_headers: flags=128 0(22883) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(22883) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(22883) parse_headers: flags=256 0(22883) end of header reached, state=9 0(22883) DEBUG: get_hdr_field: <To> [21]; uri=[sip:admin@debtest.stu.du] 0(22883) DEBUG: to body [<sip:admin@debtest.stu.du> ] 0(22883) get_hdr_field: cseq <CSeq>: <1> <REGISTER> 0(22883) DEBUG: get_hdr_body : content_length=0 0(22883) found end of header 0(22883) find_first_route: No Route headers found 0(22883) loose_route: There is no Route HF 0(22883) grep_sock_info - checking if host==us: 7==12 && [debtest.stu.du] == [161.53.0.131] 0(22883) grep_sock_info - checking if port 5060 matches port 5060 0(22883) grep_sock_info - checking if host==us: 7==12 && [debtest.stu.du] == [161.53.0.131] 0(22883) grep_sock_info - checking if port 5060 matches port 5060 0(22883) parse_headers: flags=-1 0(22883) check_via_address(192.168.0.101, 192.168.0.101, 0) 0(22883) parse_headers: flags=4096 0(22883) pre_auth(): Credentials with given realm not found 0(22883) build_auth_hf(): 'WWW-Authenticate: Digest realm="debtest.stu.du", nonce="43390f58ff776ef2b6700bd7e5cb53ea5b79453b" ' 0(22883) parse_headers: flags=-1 0(22883) check_via_address(192.168.0.101, 192.168.0.101, 0) 0(22883) DEBUG:destroy_avp_list: destroying list (nil) 0(22883) receive_msg: cleaning up

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[Serusers] Register problem
by Sebastian Kühner 26 Sep '05

26 Sep '05

Hi, I installed SER on a new system and now I can't REGISTER anymore. With a little bit luck, the users can REGISTER after a restart of the system, but the register expires and the user is unregistered for ever. SER sends back a Unauthorized, although the user is registerin correctly. It's a phenomenon that I don't understand. May be the database? Here's a log of the database: Sep 26 18:00:22 pbx5 postgres[286]: [13-1] LOG: 00000: statement: BEGIN Sep 26 18:00:22 pbx5 postgres[286]: [13-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:00:22 pbx5 postgres[286]: [14-1] LOG: 00000: statement: select userpass,rpid from subscriber where username='43750' Sep 26 18:00:22 pbx5 postgres[286]: [14-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:00:22 pbx5 postgres[286]: [15-1] LOG: 00000: statement: COMMIT Sep 26 18:00:22 pbx5 postgres[286]: [15-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:00:22 pbx5 postgres[307]: [25-1] LOG: 00000: statement: BEGIN Sep 26 18:00:22 pbx5 postgres[307]: [25-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:00:22 pbx5 postgres[307]: [26-1] LOG: 00000: statement: select userpass,rpid from subscriber where username='43750' Sep 26 18:00:22 pbx5 postgres[307]: [26-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:00:22 pbx5 postgres[307]: [27-1] LOG: 00000: statement: COMMIT Sep 26 18:00:22 pbx5 postgres[307]: [27-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:00:22 pbx5 postgres[282]: [10-1] LOG: 00000: statement: BEGIN Sep 26 18:00:22 pbx5 postgres[282]: [10-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:00:22 pbx5 postgres[282]: [11-1] LOG: 00000: statement: select userpass,rpid from subscriber where username='43750' Sep 26 18:00:22 pbx5 postgres[282]: [11-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:00:22 pbx5 postgres[282]: [12-1] LOG: 00000: statement: COMMIT Sep 26 18:00:22 pbx5 postgres[282]: [12-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:01:00 pbx5 postgres[332]: [19-1] LOG: 00000: statement: BEGIN Sep 26 18:01:00 pbx5 postgres[332]: [19-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:01:00 pbx5 postgres[332]: [20-1] LOG: 00000: statement: delete from location where username='43750' AND contact='si p:43750@xxx.xxx.xxx.xxx:63713' AND Can anyone of you help me?? Thanks!!! Sebastian

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[Users] branch route
by Richard Z 26 Sep '05

26 Sep '05

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

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[Serusers] Try to do effective routing using sip server
by Prajyot 26 Sep '05

26 Sep '05

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

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Re: [Serusers] ONSIP Script + Java Rockx ser 0.9.0 script
by Iqbal 26 Sep '05

26 Sep '05

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> > >

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[Serusers] Does VOVIDA B2BUA works with SER+MEDIAPROXY?
by Thomas Tomter 26 Sep '05

26 Sep '05

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

2 1

[Serusers] Field STATE
by Alberto 26 Sep '05

26 Sep '05

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

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[Serusers] cisco 7905G
by Alberto 26 Sep '05

26 Sep '05

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

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RE: [Serusers] allow_trusted removed from permission module (HEAD)?
by Bjorn Asmul 26 Sep '05

26 Sep '05

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

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[Serusers] Alias routing section in http://lists.iptel.org/pipermail/serusers/2005-February/015700.html
by Ramin Nikaeen 26 Sep '05

26 Sep '05

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

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[Serusers] SER Vs. SIPsak = 483, incoerence!!!!
by Ángel Gómez 26 Sep '05

26 Sep '05

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???

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[Serusers] Seeking Advise With Caller ID Block
by Chris Tsai 26 Sep '05

26 Sep '05

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"

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Re: [Users] rewriteuser
by Nelson Silva 26 Sep '05

26 Sep '05

1 0

[Serusers] is SER ( 0.9.4) 100% 3GPP SIP call flow compliant ?
by Ashish Gilotra 26 Sep '05

26 Sep '05

Hi We are evaluating SER for our purpose. We need to use 3GPP SIP call flows. Is SER compliant ? Thanks Ash

1 0

Re: [Users] rewriteuser
by Daniel-Constantin Mierla 26 Sep '05

26 Sep '05

Hello, you can implement this with avpops module (http://www.voice-system.ro/docs/avpops/). Functions of interest for you are avp_write() and avp_pushto(). ... avp_write("1234", "s:client_n"); alias_db_lookup("dbaliases"); lookup("location); avp_pushto("$ruri/username", "s:client_n"); t_relay(); ... For optimization reasons, you may consider replacing the name of avp "s:client_n" to an integer id, e.g., "i:500" (faster operations). Cheers, Daniel On 09/26/05 14:14, Nelson Silva wrote: > Have been using openser for a couple of months and i run into a problem. > > Currently i'm using alias_db_lookup, location and t_relay to do the > routing. > > Since many clients have many numbers, i need to to rewrite the contact > to the right client. It's doing it but only for a client with one > number. Need for clients with more numbers. > > This is what i want: > > > $client_n = 1234 > > alias_db_lookup("db_aliases"); -> client_number(a)provider.com > > location("location"); -> client_number@client_ip > > rewriteuser($client_n); > > t_relay(); > > This is what i have: > > alias_db_lookup("dbaliases"); > > lookup("location); > > t_relay(); > > > > Anyone have this implemented? > > > Nelson Silva > ----------------- > email: nelson.silva(a)neuvex.com > website: http://www.neuvex.com > > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

1 0

[Users] rewriteuser
by Nelson Silva 26 Sep '05

26 Sep '05

1 0

[Users] nathelper force_rtp_proxy onreply_route
by OpenSer Users 26 Sep '05

26 Sep '05

hello, in SDP body sent out by the proxy the original media IP address is still present, the rtpproxy ip is added as a another line. is this the correct behavior ? should I add some flags to remove the original media IP ? for the invite this does not happen, the c= field is REWRITTEN with the ip received from the rtpproxy. message received by openser: ============================================ v=0. o=CiscoSystemsSIP-GW-UserAgent 4176 788 IN IP4 GW-IP-ADDRESS. s=SIP Call. c=IN IP4 GW-IP-ADDRESS. t=0 0. m=audio 17644 RTP/AVP 18. c=IN IP4 GW-IP-ADDRESS. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=direction:passive. ============================================ message sent out by openser: ============================================ v=0. o=CiscoSystemsSIP-GW-UserAgent 4176 788 IN IP4 GW-IP-ADDRESS. s=SIP Call. c=IN IP4 GW-IP-ADDRESS. t=0 0. m=audio 35784 RTP/AVP 18. c=IN IP4 RTPPROXY-IP-ADDRESS. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=direction:passive. a=nortpproxy:yes. ============================================ onreply_route block ============================================ if(status =~ "(183)|2[0-9][0-9]") { if(!force_rtp_proxy()) { log(3, "force_rtp_proxy FAILED"); }; }; ============================================ thanks Razvan Radu

2 1

[Users] one way audio problem with stun
by Diego Valencia 26 Sep '05

26 Sep '05

Hi everybody, I have a "one way audio" problem. My enviroment is: xx.xx.63.75/27 = SER xx.xx.63.68/27 = Natbox (linux iptables masquerade) xx.xx.70.2/25 = STUN server xx.xx.63.78/27 = PAP1 linksys AOR 2000 10.1.1.233/24 = PAP2 linksys AOR 2001 (behind NATbox with stun enable) I have no audio from PAP1 to PAP2 when the call is made from PAP1 to PAP2, but when the call is made from PAP2 to PAP1 it works correctly. I see on the natbox that the RTP packet is droped with unreachable cause (icmp messages). This is the trace (ngrep) for the call from PAP1 to PAP2 on external interface of the natbox, where I can see STUN packets and the number udp port that the PAP willl be use (16444 to 16447). The problem is natbox use other port (1173). I don't know if this problem is on natbox or stun server or PAP. Can you helpme? Thanks!! U xx.xx.63.75:5060 -> xx.xx.63.68:5060 INVITE sip:2001@xx.xx.63.68:5060 SIP/2.0..Record-Route: <sip:xx.xx.63.75;ftag=d1ccd3db9e31eb1fo0;lr=on>..Via: SIP/2.0/UDP xx.xx.63.75;branch=z9hG4bK52fb.41985863.0..Via: SIP/2.0/UDP xx.xx.63.78:5060;branch=z9hG4bK-57b1577f..From: 2000 <sip:5 890000(a)xx.xx.63.75>;tag=d1ccd3db9e31eb1fo0..To: <sip:2001@xx.xx.63.75>..Call-ID: d6e2df8b-5193d32f@xx.xx.63.78..CSeq: 101 INVITE..Max-Forwards: 16..Contact: 2000 <sip:2000@xx.xx.63.78:5060>..Expires: 240..User-Agent: Linksys/PAP2-2.0.10(L Sc)..Content-Length: 420..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: x-sipura..Content-Type: app lication/sdp....v=0..o=- 29375 29375 IN IP4 xx.xx.63.78..s=-..c=IN IP4 xx.xx.63.78..t=0 0..m=audio 16472 RTP/AVP 18 0 2 4 8 9 6 97 98 100 101..a=rtpmap:18 G729a/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:2 G726-32/8000..a=rtpmap:4 G723/8000..a=rtpmap:8 PCMA/8 000..a=rtpmap:96 G726-40/8000..a=rtpmap:97 G726-24/8000..a=rtpmap:98 G726-16/8000..a=rtpmap:100 NSE/8000..a=rtpmap:101 telephon e-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. # U xx.xx.63.68:5060 -> xx.xx.63.75:5060 SIP/2.0 100 Trying..To: <sip:2001@xx.xx.63.75>..From: 2000 <sip:2000@xx.xx.63.75>;tag=d1ccd3db9e31eb1fo0..Call-ID : d6e2df8b-5193d32f@xx.xx.63.78..CSeq: 101 INVITE..Via: SIP/2.0/UDP xx.xx.63.75;branch=z9hG4bK52fb.41985863.0..Via: SIP/2.0/U DP xx.xx.63.78:5060;branch=z9hG4bK-57b1577f..Record-Route: <sip:xx.xx.63.75;ftag=d1ccd3db9e31eb1fo0;lr=on>..Server: Linksys/P AP2-2.0.10(LSc)..Content-Length: 0.... # U xx.xx.63.68:5060 -> xx.xx.70.2:3478 .....*f..*f..*f..*f. # U xx.xx.63.68:16444 -> xx.xx.70.2:3478 .....OB..OB..OB..OB. # U xx.xx.63.68:16445 -> xx.xx.70.2:3478 .................... # U xx.xx.63.68:16446 -> xx.xx.70.2:3478 .......&...&...&...& # U xx.xx.63.68:16447 -> xx.xx.70.2:3478 ....#...#...#...#... # U xx.xx.70.2:3478 -> xx.xx.63.68:16447 ...$#...#...#...#.........@?.+?D.........+F..........+F. # U xx.xx.63.68:5060 -> xx.xx.70.2:3478 .....*f..*f..*f..*f. # U xx.xx.63.68:16444 -> xx.xx.70.2:3478 .....OB..OB..OB..OB. # U xx.xx.70.2:3478 -> xx.xx.63.68:5060 ...$.*f..*f..*f..*f..........+?D.........+F..........+F. # U xx.xx.63.68:16445 -> xx.xx.70.2:3478 .................... # U xx.xx.63.68:16446 -> xx.xx.70.2:3478 .......&...&...&...& # U xx.xx.70.2:3478 -> xx.xx.63.68:16444 ...$.OB..OB..OB..OB.......@<.+?D.........+F..........+F. # U xx.xx.70.2:3478 -> xx.xx.63.68:16446 ...$...&...&...&...&......@>.+?D.........+F..........+F. # U xx.xx.63.68:5060 -> xx.xx.63.75:5060 SIP/2.0 180 Ringing..To: <sip:2001@xx.xx.63.75>;tag=156298945b7d27i0..From: 2000 <sip:2000@xx.xx.63.75>;tag=d1ccd 3db9e31eb1fo0..Call-ID: d6e2df8b-5193d32f@xx.xx.63.78..CSeq: 101 INVITE..Via: SIP/2.0/UDP xx.xx.63.75;branch=z9hG4bK52fb.4198 5863.0..Via: SIP/2.0/UDP xx.xx.63.78:5060;branch=z9hG4bK-57b1577f..Record-Route: <sip:xx.xx.63.75;ftag=d1ccd3db9e31eb1fo0;lr= on>..Server: Linksys/PAP2-2.0.10(LSc)..Content-Length: 0.... # U xx.xx.70.2:3478 -> xx.xx.63.68:16444 ...$.OB..OB..OB..OB.......@<.+?D.........+F..........+F. # U xx.xx.63.68:5060 -> xx.xx.63.75:5060 SIP/2.0 200 OK..To: <sip:2001@xx.xx.63.75>;tag=156298945b7d27i0..From: 2000 <sip:2000@xx.xx.63.75>;tag=d1ccd3db9e 31eb1fo0..Call-ID: d6e2df8b-5193d32f@xx.xx.63.78..CSeq: 101 INVITE..Via: SIP/2.0/UDP xx.xx.63.75;branch=z9hG4bK52fb.41985863. 0..Via: SIP/2.0/UDP xx.xx.63.78:5060;branch=z9hG4bK-57b1577f..Record-Route: <sip:xx.xx.63.75;ftag=d1ccd3db9e31eb1fo0;lr=on>.. Contact: 2001 <sip:2001@xx.xx.63.68:5060>..Server: Linksys/PAP2-2.0.10(LSc)..Content-Length: 236..Allow: ACK, BYE, CANCE L, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: x-sipura..Content-Type: application/sdp....v=0..o=- 659101 659101 IN IP4 20 0.43.63.68..s=-..c=IN IP4 xx.xx.63.68..t=0 0..m=audio 16444 RTP/AVP 18 100 101..a=rtpmap:18 G729a/8000..a=rtpmap:100 NSE/8000. .a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. # U xx.xx.63.75:5060 -> xx.xx.63.68:5060 ACK sip:2001@xx.xx.63.68:5060 SIP/2.0..Record-Route: <sip:xx.xx.63.75;ftag=d1ccd3db9e31eb1fo0;lr=on>..Via: SIP/2.0/UDP 200 .43.63.75;branch=0..Via: SIP/2.0/UDP xx.xx.63.78:5060;branch=z9hG4bK-5a4c9c84..From: 2000 <sip:2000@xx.xx.63.75>;tag =d1ccd3db9e31eb1fo0..To: <sip:2001@xx.xx.63.75>;tag=156298945b7d27i0..Call-ID: d6e2df8b-5193d32f@xx.xx.63.78..CSeq: 101 AC K..Max-Forwards: 16..Contact: 2000 <sip:2000@xx.xx.63.78:5060>..User-Agent: Linksys/PAP2-2.0.10(LSc)..Content-Length: 0.... # U xx.xx.63.78:16472 -> xx.xx.63.68:16444 ..8...-..L....@.....tF.34..V.|B..n..a.k(.R # I xx.xx.63.68 -> xx.xx.63.78 3:3 ....E..F.........+?N.+?D@X@<.2....8...-..L....@.....tF.34..V.|B..n..a.k(.R # U xx.xx.63.68:1173 -> xx.xx.63.78:16472 ..!o..vy.F.!...|...&.T.BP..Z..n...:.3..d.. # U xx.xx.63.78:16472 -> xx.xx.63.68:16444 ..8......L... .s..._exR..Kc.....+.O<.RZ.O{ # I xx.xx.63.68 -> xx.xx.63.78 3:3 ....E..F.........+?N.+?D@X@<.2....8......L... .s..._exR..Kc.....+.O<.RZ.O{ # U xx.xx.63.68:1173 -> xx.xx.63.78:16472 ..!p..wi.F.!.'D.F...K....@C..e.Z.[~P...Et, # U xx.xx.63.78:16472 -> xx.xx.63.68:16444 ..8.../..L......@H..._...,[.P...rU.A..P... # I xx.xx.63.68 -> xx.xx.63.78 3:3 ....E..F.........+?N.+?D@X@<.2....8.../..L......@H..._...,[.P...rU.A..P... # U xx.xx.63.68:1173 -> xx.xx.63.78:16472 ..!q..xY.F.!....Mf.....Fu.Q.....~.}....v.. # U xx.xx.63.78:16472 -> xx.xx.63.68:16444 ..8...0..L..\.~..h..+..|S.C.M..~6W}....6sR # I xx.xx.63.68 -> xx.xx.63.78 3:3 ....E..F.........+?N.+?D@X@<.2....8...0..L..\.~..h..+..|S.C.M..~6W}....6sR # U xx.xx.63.68:1173 -> xx.xx.63.78:16472 ..!r..yI.F.!x|..Eo..._.1...;..GRz....*.x.Z # U xx.xx.63.78:16472 -> xx.xx.63.68:16444 ..8...1..L..x.....n."..6.v....S.D.......S. # I xx.xx.63.68 -> xx.xx.63.78 3:3 ....E..F.........+?N.+?D@X@<.2.!..8...1..L..x.....n."..6.v....S.D.......S. # U xx.xx.63.68:1173 -> xx.xx.63.78:16472 ..!s..z9.F.!0...ooPE..F. .5...E....v...... # U xx.xx.63.78:16472 -> xx.xx.63.68:16444 ..8...2..L..z....+.b..r?...r....|..$.x...^ # I xx.xx.63.68 -> xx.xx.63.78 3:3 ....E..F.........+?N.+?D@X@<.2....8...2..L..z....+.b..r?...r....|..$.x...^ # U xx.xx.63.68:1173 -> xx.xx.63.78:16472 ..!t..{).F.!.o...{..r..3....F.$.r.. V..by. #

2 1

[Serusers] Problem/Question regarding mediaproxy
by Ronald Voermans 26 Sep '05

26 Sep '05

Hello, I'm using Ser in combination with Mediaproxy. The situation is as follow: My SER has a private ip address (10.254.254.1), and my UAs also have private IP addresses, they can all see SER, and SER can see the UAs. Next, our company has a Voice Interconnect with a telco. (it's using Cirpack as softswitch). SER can see Cirpack and vice versa. However, the UAs cannot see the Cirpack. So I want to use Mediaproxy to overcome this. I can make outbound calls, but when calling inbound, I sometimes have one-way-voice of no voice at all. I think the problem is related to Cirpack, but I have a question thouh: - When calling mediaproxy, how does mediaproxy decide which one is the caller and which one is the called? Is this based on the From and To header? Because, if i call from PSTN -> SIP, i get serveral thousands (!) of messages saying that the called person signed in (no caller signs in in mediaproxy). This could be due to the fact that the from and to headers are incorrect. My ser.cfg is the one from onsip.org (pstn with nat). I can post configs/traces if someone needs them! Regards, Ronald

1 0

[Serusers] allow_trusted removed from permission module (HEAD)?
by Bjorn Asmul 26 Sep '05

26 Sep '05

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

2 1

[Serusers] SER + Mysql.so problem
by smy＠addm.com 26 Sep '05

26 Sep '05

Hi all, I've been working with SER + Mysql for over a month without any problems, yet all these (including the DB) are installed on the same server (LocalHost). I'm currently trying to centralize and securing all Data, thus would like to move all data to a centralized DB offsite. Now, I have SER DB migrated to this new DB server, and have changed the SER.cfg so that all mysql related loadmodule would be pointing to the proper server (ip address) instead of localhost. This is where the nightmare begins. The SER server can not seem to be connecting to the Mysql Server properly. Thus it took me over 3 days to debug and see if there were any auth problems from a remote site to the Mysql server. Apparently, remote auth & connect to the DB is working perfectly since I've installed other application which can connect to this DB server. So I've checked with the SER log + ngrep, of which to my amazement that the SER never sent any signals to the DB server. As soon as I moved the DB back to localhost, everything worked! My question is, is there something I need to configure in the mysql module so that it can work with a remote server???? Any insight is greatly appreciated!!! Best Regards, Samuel Yeung

3 7

[Serusers] accounting and radius/sql
by Bjorn Asmul 25 Sep '05

25 Sep '05

The best solution is to use two different tables for start and stop accounting. Have a look in the sql.conf file in FreeRADIUS Problem is that SER replaces the Username in the stop-message, with whoever hung up... Therefore the default queries in FreeRADIUS won't find the corresponding record. To get the CDR just join the two tables on AcctSessionId. I have experienced problems joining on the AcctSessionId in MySQL versions prior to 4.1, for some reason. Bjorn

1 0

[Serusers] More random avpops questions....
by sip 25 Sep '05

25 Sep '05

Okay... I think with Steve's help, I've been getting somewhere with my avpops stuff, but I have a couple of questions. 1) What is the format of the $from and $to headers as far as avpops is concerned? How can I find that out? 2) How does avp_print() work (i.e. to where does it print) ? Is there any way I can get intermediate values of things so I can see just what it is my little scripts are doing? N.

3 4

[Serusers] [about SER]networking among three SIP servers.....
by 송 미진 25 Sep '05

25 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear........ I want to networking among three SIP servers. But now I am trying using two servers likes three servers. That is, - source : computer A ip address ; 192.168.178.194 sipuri : sip:mj@192.168.178.194 - via : computer B ip address : 192.168.178.193 sipuri : sip:hj@192.168.178.193 - destination : computer A ip address :192.168.178.194 sipuri : sip:mj@192.168.178.194 And I am using sipp to test (I modified the "via" part based on original uac module). ===================================================== << computer B's test.cfg >> # # $Id: welcome.cfg,v 1.3 2003/11/03 13:03:56 janakj Exp $ # # welcome message for new subscribers; based on exec # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) # must be yes since REGISTER processing causes an INVITE to be sent, # which needs to be processed by another process fork=yes children=4 alias="kims.myserver.sampro.cse" alias="myserver.sampro.cse" alias="192.168.178.193" # debugging log_stderror=yes # (cmd line: -E) mhomed=yes fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/exec.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", "mysql://ser:heslo@192.168.2.16/ser" ) modparam("usrloc", "db_url", "mysql://root:heslo@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ log(1, "\n\n"); log(1, "START!!!\n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len>=204822) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol # if (method=="INVITE") record_route(); # 1=loose routing if(method=="INVITE"){ log(1, "receive INVITE\n"); record_route(); sl_send_reply("200", "ok"); }; # loose-route processing if (loose_route()) { t_relay(); break; }; # if(method=="ACK"){ # log(1, "receive ACK\n"); # }; log(1, "record-route processing completed\n"); # 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(uri=="sip:mijin@sam.cse"){ # forward(192.168.178.194, 5060); # }; if (uri=~"sip:192.168.178.193"){ if (method=="REGISTER") { log(1, "receive REGISTER\n"); # save location before initiating welcome save("location"); log(1, "save ì„±ê³µ\n"); record_route(); sl_send_reply("200", "ok"); # welcome message log(1, "route3ìœ¼ë¡œ ì�´ë�™ \n"); route(3); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); log(1, "Usrloc DBì—�ì„œ ì°¾ ì�„ ìˆ˜ ì—†ì�Œ \n"); break; }; }; if(method=="BYE"){ log(1, "receive BYE\n"); record_route(); sl_send_reply("200", "ok"); break; }; t_relay(); } # welcome message -- if a REGISTER succeeded, look if it is the first-time; # if so, initiate click-to-dial transaction to set up call to an announcement # server; edit the config values first to make it work route[3] { if (!exec_msg(' # config: # --announcement server URI ANS="sip:hjkim@myserver.sampro.cse" # --SIP domain # DOMAIN="192.168.178.193" DOMAIN="192.168.178.193" # ctd CTD=${HOME}/sip_router/examples/ctd.sh # ------------------------------------ # check if first time ... SIP_UID=`echo $SIP_HF_TO | sed -e "s/^.*sip:$[a-zA-Z0-9_\.]*$@.*$/\1/g"` QUERY="select flag from subscriber where username=\"$SIP_UID\"; update subscriber set flag=\"x\" where username=\"$SIP_UID\" "; mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null # ... if so, c-t-d to announcement server if [ "$?" -ne 0 ] ; then # flag was not set to x yet -- first-time registration; # initiate a call from telephone of the user to an announcement server $CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1 fi ')) { log(1, "welcome exec failed\n"); } log(1, "-------route3 ë�� \n"); } As you see, I added modparam("rr", "enable_full_lr", no) if(uri=="sip:mijin@192.168.178.194"){ forward(192.168.178.194, 5060); append_branch("sip:hjkim@192.168.178.193:5060"); log(1, "next hop...\n"); break; }; but, I received error message in computer B(via): 8(12253) ERROR: forward_reply: no 2nd via found in reply What does it mean? What can I do for doing what I want ??? Is there any problem in my test.cfg file ????? I would like to know in detail . please tell me what to do............................... sipp module can't support that???? And Please tell me about forward funtion. I don't know whether I understand forward function usage well....... sorry ............ Song _________________________________________________________________ 고.. 감.. 도.. 사.. 랑.. 만.. 들.. 기.. MSN 러브 http://www.msn.co.kr/love/

1 0

[Serusers] Radius Authorization and AVP Retrieval
by Tavis P 24 Sep '05

24 Sep '05

I'm having some trouble trying to integrate both Radius authentication and Radius AVP storage. I'm using freeradius and there doesn't seem to be much documentation regarding configuration semantics and such. Could anyone share some information on how they were able to use a Radius server to authenticate requests and (with avp_radius) as an AVP retrieval mechanism? I'm uncertain as to how I can stack these two uses of data on the freeradius server tavis

3 4

[Serusers] [about SER]networking among three SIP servers.....
by 송 미진 24 Sep '05

24 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear........ I want to networking among three SIP servers. But now I am trying using two servers likes three servers. That is, - source : computer A ip address ; 192.168.178.194 sipuri : sip:mj@192.168.178.194 - via : computer B ip address : 192.168.178.193 sipuri : sip:hj@192.168.178.193 - destination : computer A ip address :192.168.178.194 sipuri : sip:mj@192.168.178.194 And I am using sipp to test (I modified the "via" part based on original uac module). ===================================================== << computer B's test.cfg >> # # $Id: welcome.cfg,v 1.3 2003/11/03 13:03:56 janakj Exp $ # # welcome message for new subscribers; based on exec # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) # must be yes since REGISTER processing causes an INVITE to be sent, # which needs to be processed by another process fork=yes children=4 alias="kims.myserver.sampro.cse" alias="myserver.sampro.cse" alias="192.168.178.193" # debugging log_stderror=yes # (cmd line: -E) mhomed=yes fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/exec.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", "mysql://ser:heslo@192.168.2.16/ser" ) modparam("usrloc", "db_url", "mysql://root:heslo@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ log(1, "\n\n"); log(1, "START!!!\n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len>=204822) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol # if (method=="INVITE") record_route(); # 1=loose routing if(method=="INVITE"){ log(1, "receive INVITE\n"); record_route(); sl_send_reply("200", "ok"); }; # loose-route processing if (loose_route()) { t_relay(); break; }; # if(method=="ACK"){ # log(1, "receive ACK\n"); # }; log(1, "record-route processing completed\n"); # 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(uri=="sip:mijin@sam.cse"){ # forward(192.168.178.194, 5060); # }; if (uri=~"sip:192.168.178.193"){ if (method=="REGISTER") { log(1, "receive REGISTER\n"); # save location before initiating welcome save("location"); log(1, "save ì„±ê³µ\n"); record_route(); sl_send_reply("200", "ok"); # welcome message log(1, "route3ìœ¼ë¡œ ì´ë™ \n"); route(3); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); log(1, "Usrloc DBì—ì„œ ì°¾ ì„ ìˆ˜ ì—†ìŒ \n"); break; }; }; if(method=="BYE"){ log(1, "receive BYE\n"); record_route(); sl_send_reply("200", "ok"); break; }; t_relay(); } # welcome message -- if a REGISTER succeeded, look if it is the first-time; # if so, initiate click-to-dial transaction to set up call to an announcement # server; edit the config values first to make it work route[3] { if (!exec_msg(' # config: # --announcement server URI ANS="sip:hjkim@myserver.sampro.cse" # --SIP domain # DOMAIN="192.168.178.193" DOMAIN="192.168.178.193" # ctd CTD=${HOME}/sip_router/examples/ctd.sh # ------------------------------------ # check if first time ... SIP_UID=`echo $SIP_HF_TO | sed -e "s/^.*sip:$[a-zA-Z0-9_\.]*$@.*$/\1/g"` QUERY="select flag from subscriber where username=\"$SIP_UID\"; update subscriber set flag=\"x\" where username=\"$SIP_UID\" "; mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null # ... if so, c-t-d to announcement server if [ "$?" -ne 0 ] ; then # flag was not set to x yet -- first-time registration; # initiate a call from telephone of the user to an announcement server $CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1 fi ')) { log(1, "welcome exec failed\n"); } log(1, "-------route3 ë \n"); } As you see, I added modparam("rr", "enable_full_lr", no) if(uri=="sip:mijin@192.168.178.194"){ forward(192.168.178.194, 5060); append_branch("sip:hjkim@192.168.178.193:5060"); log(1, "next hop...\n"); break; }; but, I received error message in computer B(via): 8(12253) ERROR: forward_reply: no 2nd via found in reply What does it mean? What can I do for doing what I want ??? Is there any problem in my test.cfg file ????? I would like to know in detail . please tell me what to do............................... Song _________________________________________________________________ 싸게 싸게 MSN공동구매 http://www.waawaa.com/cobuy/cobuy_default.asp?siteid=10160

1 0

[Serusers] about SER
by 송 미진 24 Sep '05

24 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear. All Hi~ I was adviced many times by "seruser" but my problems has not been solved. I,m afraid that I have many questions. So I intend to specify my present situation. Operating System: Linux Fedora 3 ================================================================================ [root@cse3 ~]# serctl start Starting SER : started pid(8370) [root@cse3 ~]# serctl ps 0 8370 attendant 1 8375 fifo server 2 8377 receiver child=0 sock= 127.0.0.1:5060 3 8379 receiver child=1 sock= 127.0.0.1:5060 4 8384 receiver child=2 sock= 127.0.0.1:5060 5 8386 receiver child=3 sock= 127.0.0.1:5060 6 8387 receiver child=0 sock= 192.168.178.193:5060 7 8390 receiver child=1 sock= 192.168.178.193:5060 8 8393 receiver child=2 sock= 192.168.178.193:5060 9 8397 receiver child=3 sock= 192.168.178.193:5060 10 8401 timer 11 8403 tcp receiver 12 8406 tcp receiver 13 8410 tcp receiver 14 8412 tcp receiver 15 8415 tcp main process ============================================================================== my_ip: 192.168.178.193 sip_uri: sip:hjkim@192.168.178.193 I added my hostname and my ip address in /etc/host file. And I added alias="hostname.myserver.sampro.cse" alias="192.168.178.193" in /usr/local/etc/ser/ser.cfg file. ==================================== 1 ============================================ [root@cse3 ~]# serctl ping sip:hjkim@192.168.178.193 [1]+ Done cat <$path | filter_fl 483 =========================================================================================== ==================================== 2 =========================================== I received "SIP/2.0 483 Too Many Hops" . but I don't know the reason. Well...My computer is connected other computers including computer has 192.168.178.194 as ip address. And my /etc/host file has 127.0.0.1 cse4 localhost.localdomain localhost 192.168.178.194 kims myserver.sampro.cse Is there reason in here??? I don't know . =========================================================================================== [root@cse3 ~]# sipsak -T -s sip:hjkim@192.168.178.193 warning: IP extract from warning activated to be more informational 0: 192.168.178.193 (0.355 ms) SIP/2.0 483 Too Many Hops 1: 192.168.178.193 (0.982 ms) SIP/2.0 483 Too Many Hops 2: 192.168.178.193 (0.459 ms) SIP/2.0 483 Too Many Hops 3: 192.168.178.193 (0.520 ms) SIP/2.0 483 Too Many Hops .................... 177: timeout after 500 ms 177: timeout after 1000 ms 177: timeout after 2000 ms 177: 192.168.178.193 (3306.067 ms) SIP/2.0 483 Too Many Hops 178: 192.168.178.193 (3.073 ms) SIP/2.0 483 Too Many Hops 179: 192.168.178.193 (3.104 ms) SIP/2.0 483 Too Many Hops .................. 313: 192.168.178.193 (3.345 ms) SIP/2.0 483 Too Many Hops 314: 192.168.178.193 (3.494 ms) SIP/2.0 483 Too Many Hops 315: 192.168.178.193 (3.038 ms) SIP/2.0 483 Too Many Hops 316: 192.168.178.193 (3.518 ms) SIP/2.0 483 Too Many Hops 317: 192.168.178.193 (5.048 ms) SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) without Contact header ============================================================================================ ======================================== 3 ====================================== In response part, I found the message "SIP/2.0 401 Unauthorized" and "authorization failed request already contains (Proxy-) Authorization, but received 40[1|7], see above" What happened?? ============================================================================================ [root@cse3 ~]# sipsak -U -vv -s sip:hjkim@192.168.178.193 warning: ignoring -i option when in usrloc mode registering user hjkim... authorizing registering user hjkim... request: REGISTER sip:192.168.178.193 SIP/2.0 Authorization: Digest username="hjkim", uri="sip:192.168.178.193", algorithm=MD5, realm="myserver.sampro.cse", nonce="4333e4b8b0199c2d9c0f8bd249bedc3b8b20bf87", response="f58e11aff01e5123bcf5010405e91278" Via: SIP/2.0/UDP 127.0.0.1:32860;branch=z9hG4bK.4f28e7fb;rport From: sip:hjkim@192.168.178.193;tag=4de59540 To: sip:hjkim@192.168.178.193 Call-ID: 1306891584(a)127.0.0.1 CSeq: 2 REGISTER Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.9.1 Expires: 15 Contact: sip:hjkim@127.0.0.1:32860 response: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 127.0.0.1:32860;branch=z9hG4bK.4f28e7fb;rport=32860;received=192.168.178.193 From: sip:hjkim@192.168.178.193;tag=4de59540 To: sip:hjkim@192.168.178.193;tag=b27e1a1d33761e85846fc98f5f3a7e58.ad3a Call-ID: 1306891584(a)127.0.0.1 CSeq: 2 REGISTER WWW-Authenticate: Digest realm="myserver.sampro.cse", nonce="4333e4b8b0199c2d9c0f8bd249bedc3b8b20bf87" Server: Sip EXpress router (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 192.168.178.193:5060 "Noisy feedback tells: pid=8397 req_src_ip=192.168.178.193 req_src_port=32860 in_uri=sip:192.168.178.193 out_uri=sip:192.168.178.193 via_cnt==1" error: authorization failed request already contains (Proxy-) Authorization, but received 40[1|7], see above ============================================================================================ [root@cse3 ~]# serctl ul show Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y ===Domain list=== ---Domain--- name : 'aliases' size : 512 table: 0xf50b44c8 d_ll { n : 0 first: (nil) last : (nil) } ---/Domain--- ---Domain--- name : 'location' size : 512 table: 0xf50b2218 d_ll { n : 2 first: 0xf50b4220 last : 0xf50b4348 } ...Record(0xf50b4220)... domain: 'location' aor : 'kim' ~~~Contact(0xf50b4260)~~~ domain : 'location' aor : 'kim' Contact : 'sip:hjkim@myserver.sampro.cse' Expires : Permanent q : 1 Call-ID : 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything' CSeq : 42 User-Agent: 'SIP Express Router FIFO' received : '' State : CS_SYNC Flags : 128 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ...Record(0xf50b4348)... domain: 'location' aor : 'mj' ~~~Contact(0xf50b4388)~~~ domain : 'location' aor : 'mj' Contact : 'sip:mj@myserver.sampro.cse' Expires : Permanent q : 1 Call-ID : 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything' CSeq : 42 User-Agent: 'SIP Express Router FIFO' received : '' State : CS_SYNC Flags : 128 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ---/Domain--- ===/Domain list=== ==================================================================================================================================== 4 ===================================== I received the message "received ICMP packet (type: 3, code: 10): from 192.168.178.194" What does means this message?? ============================================================================================ [root@cse3 ~]# sipsak -T -vv -s sip:iptel@192.168.178.194 warning: IP extract from warning activated to be more informational Max-Forwards set to 0 received ICMP packet (type: 3, code: 10): from 192.168.178.194 ======================================================= ======================================================= << my ser.cfg >> # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) alias="192.168.178.193" alias="myserver.sampro.cse" alias="kims.myserver.sampro.cse" /* 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/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2000048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri=="sip:192.168.178.193") { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri=="sip:192.168.178.193") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("myserver.sampro.cse", "subscriber")) { www_challenge("myserver.sampro.cse", "0"); break; }; save("location"); log(1, "save \n"); break; }; lookup("aliases"); if (!uri=="sip:192.168.178.193") { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } ###################################################################################### I’d appreciate it if you would give me some solutions. Thank you in advance..... mi-jin Song _________________________________________________________________ 행운의 주인공이 이번엔 나일꺼야, 진짜루... 인터넷 복권 http://www.msn.co.kr/money/interlotto/

2 1

[Serusers] cannot receive
by rajesh volluru 24 Sep '05

24 Sep '05

hi I am using Sip communicator. I cant receive any voice from the other calling party but the calling party can hear whatever I talk. The ReceiveStreamEvent is not getting generated. Would anyone help me reg this please? Thanks Rajesh --------------------------------- Yahoo! India Matrimony: Find your partner now.

2 1

RE: [Serusers] Permanent aliases
by Igor Balk 23 Sep '05

23 Sep '05

Thank you guys for the clarification. If you do not mind I would ask one more question. Let say I have aliases 1234 of the account user@server and I'm getting call to the 1234 alias but there is no one registered at the moment with URI name@server. I also have SER configured in a way that if call will come to name@server URI and no user is availeble the call will be forwarded to voice mail box. Will same scenario work in case of call to the alias if I do lookup("aliases") before forwarding or I should do something more sofisticated to make it work? Thank you in advance, Igor -----Original Message----- From: Atle Samuelsen [mailto:clona@cyberhouse.no] Sent: Friday, September 23, 2005 1:14 PM To: Igor Balk Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Permanent aliases Hi Igor, I think your wrong there. AS far as you register with your box to your sipserver, and have a alias pointing to the username@yoursipserver then you should'nt have a problem. -Atle * Igor Balk <IBalk(a)intermedia.net> [050923 19:01]: > > Sorry if I'm asking stupid question but I'm trying to set up alias for > example 1234<->john.doe and make it permanent. As far as I understand > if I use serctl add alias command and then contact for the person will > change (dynamic ip, login from other coputer etc) my alias would not > work anymore. Please tell me if I'm worng and if there is a way to fix > this problem. > > Thank you, > Igor > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

2 1

[Serusers] Question regarding UDP TCP translation
by Weiler, Frank 23 Sep '05

23 Sep '05

Hi all, I'm new to SER and I started to play around with it as we need a solution to connect UPD UA's connected to Asterisk to a Microsoft LCS and we tough to use SER for this purpose. I've read so far through the Admin guide and it seems that the TCP support is natively embedded into SER. However I've added to the ser.cfg file the following lines (within the route[0] block just before branching to route [1]): #Forward all traffic with the uri*(a)172.16.0.201 <mailto:uri*@172.16.0.201> If (uri=~"sip:*@172.16.0.201) { forward_tcp(172.16.0.201,5060); Break; }; However while sniffing the packets it still appears that the INVITE requests are sent to the LCS using UDP and I receive an ICMP port unreachable. The original request for the UA is sent in UDP. I assume that I configured something wrong and I've checked the mailing lists and I didn't find any related documents. I'm using the latest version of SER available. Needless to say that UDP SIP calls between ASTERISK and SER work fine whatever clients I use. Any clue or help would be highly appreciated. Thank you very much in forecast for your help. Cheers, Frank

2 1

[Serusers] Getting starting with ser...
by IS2 - Florent BARBIER 23 Sep '05

23 Sep '05

Hello all, Not really easy ton understand how to setup... If somebody can help, I'll really appreciate. My config : redhat 7.3 (SME E/Smith edition) Asterisk 1.0.9 ser serweb IP phone 1 IP phone 2 IP public 1 IP public 2 ! ! ! Internet ! --------------------------------------------- ! ! ! IP public 3 ! ! Lan Asterisk ! ------------------------------------------------- ! ! IP Phone 3 IP Phone 4 IP Local 3 IP Local 4 Every IP Phone are registered in the AsterisK IP Phone 3 comunique with IP Phone 4 whithout PB IP Phone 1 comunique with IP Phone 3 or 4 whithout PB IP Phone 1 comunique with IP Phone 2 whith delay ! So audio stream doesn't go directly from IP Public 1 to IP Public 2. As I understand S.E.R. should help me to redirect audio stream... Well I setup S.E.R. and SERWEB under the Asterisk box (wich is the gateway & the firewall) I add some user whith serweb I add register in my sip.conf to register my asterisk to ser I try to do an echo test and nothing append. Well, how to setup SER to build this config... I'm sure this question certainly basic for most of you, but I've good Knowledge of AsterisK and I start with SIP proxies. I'm really not find F.A.Q. or userguide to build this... Thank you for your help, Florent.

2 1

[Serusers] about SER
by 송 미진 23 Sep '05

23 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear. All Hi~ I was adviced many times by "seruser" but my problems has not been solved. I,m afraid that I have many questions. So I intend to specify my present situation. Operating System: Linux Fedora 3 ================================================================================ [root@cse3 ~]# serctl start Starting SER : started pid(8370) [root@cse3 ~]# serctl ps 0 8370 attendant 1 8375 fifo server 2 8377 receiver child=0 sock= 127.0.0.1:5060 3 8379 receiver child=1 sock= 127.0.0.1:5060 4 8384 receiver child=2 sock= 127.0.0.1:5060 5 8386 receiver child=3 sock= 127.0.0.1:5060 6 8387 receiver child=0 sock= 192.168.178.193:5060 7 8390 receiver child=1 sock= 192.168.178.193:5060 8 8393 receiver child=2 sock= 192.168.178.193:5060 9 8397 receiver child=3 sock= 192.168.178.193:5060 10 8401 timer 11 8403 tcp receiver 12 8406 tcp receiver 13 8410 tcp receiver 14 8412 tcp receiver 15 8415 tcp main process ============================================================================== my_ip: 192.168.178.193 sip_uri: sip:hjkim@192.168.178.193 I added my hostname and my ip address in /etc/host file. And I added alias="hostname.myserver.sampro.cse" alias="192.168.178.193" in /usr/local/etc/ser/ser.cfg file. ==================================== 1 ============================================ [root@cse3 ~]# serctl ping sip:hjkim@192.168.178.193 [1]+ Done cat <$path | filter_fl 483 =========================================================================================== ==================================== 2 =========================================== I received "SIP/2.0 483 Too Many Hops" . but I don't know the reason. Well...My computer is connected other computers including computer has 192.168.178.194 as ip address. And my /etc/host file has 127.0.0.1 cse4 localhost.localdomain localhost 192.168.178.194 kims myserver.sampro.cse Is there reason in here??? I don't know . =========================================================================================== [root@cse3 ~]# sipsak -T -s sip:hjkim@192.168.178.193 warning: IP extract from warning activated to be more informational 0: 192.168.178.193 (0.355 ms) SIP/2.0 483 Too Many Hops 1: 192.168.178.193 (0.982 ms) SIP/2.0 483 Too Many Hops 2: 192.168.178.193 (0.459 ms) SIP/2.0 483 Too Many Hops 3: 192.168.178.193 (0.520 ms) SIP/2.0 483 Too Many Hops .................... 177: timeout after 500 ms 177: timeout after 1000 ms 177: timeout after 2000 ms 177: 192.168.178.193 (3306.067 ms) SIP/2.0 483 Too Many Hops 178: 192.168.178.193 (3.073 ms) SIP/2.0 483 Too Many Hops 179: 192.168.178.193 (3.104 ms) SIP/2.0 483 Too Many Hops .................. 313: 192.168.178.193 (3.345 ms) SIP/2.0 483 Too Many Hops 314: 192.168.178.193 (3.494 ms) SIP/2.0 483 Too Many Hops 315: 192.168.178.193 (3.038 ms) SIP/2.0 483 Too Many Hops 316: 192.168.178.193 (3.518 ms) SIP/2.0 483 Too Many Hops 317: 192.168.178.193 (5.048 ms) SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) without Contact header ============================================================================================ ======================================== 3 ====================================== In response part, I found the message "SIP/2.0 401 Unauthorized" and "authorization failed request already contains (Proxy-) Authorization, but received 40[1|7], see above" What happened?? ============================================================================================ [root@cse3 ~]# sipsak -U -vv -s sip:hjkim@192.168.178.193 warning: ignoring -i option when in usrloc mode registering user hjkim... authorizing registering user hjkim... request: REGISTER sip:192.168.178.193 SIP/2.0 Authorization: Digest username="hjkim", uri="sip:192.168.178.193", algorithm=MD5, realm="myserver.sampro.cse", nonce="4333e4b8b0199c2d9c0f8bd249bedc3b8b20bf87", response="f58e11aff01e5123bcf5010405e91278" Via: SIP/2.0/UDP 127.0.0.1:32860;branch=z9hG4bK.4f28e7fb;rport From: sip:hjkim@192.168.178.193;tag=4de59540 To: sip:hjkim@192.168.178.193 Call-ID: 1306891584(a)127.0.0.1 CSeq: 2 REGISTER Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.9.1 Expires: 15 Contact: sip:hjkim@127.0.0.1:32860 response: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 127.0.0.1:32860;branch=z9hG4bK.4f28e7fb;rport=32860;received=192.168.178.193 From: sip:hjkim@192.168.178.193;tag=4de59540 To: sip:hjkim@192.168.178.193;tag=b27e1a1d33761e85846fc98f5f3a7e58.ad3a Call-ID: 1306891584(a)127.0.0.1 CSeq: 2 REGISTER WWW-Authenticate: Digest realm="myserver.sampro.cse", nonce="4333e4b8b0199c2d9c0f8bd249bedc3b8b20bf87" Server: Sip EXpress router (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 192.168.178.193:5060 "Noisy feedback tells: pid=8397 req_src_ip=192.168.178.193 req_src_port=32860 in_uri=sip:192.168.178.193 out_uri=sip:192.168.178.193 via_cnt==1" error: authorization failed request already contains (Proxy-) Authorization, but received 40[1|7], see above ============================================================================================ [root@cse3 ~]# serctl ul show Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y ===Domain list=== ---Domain--- name : 'aliases' size : 512 table: 0xf50b44c8 d_ll { n : 0 first: (nil) last : (nil) } ---/Domain--- ---Domain--- name : 'location' size : 512 table: 0xf50b2218 d_ll { n : 2 first: 0xf50b4220 last : 0xf50b4348 } ...Record(0xf50b4220)... domain: 'location' aor : 'kim' ~~~Contact(0xf50b4260)~~~ domain : 'location' aor : 'kim' Contact : 'sip:hjkim@myserver.sampro.cse' Expires : Permanent q : 1 Call-ID : 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything' CSeq : 42 User-Agent: 'SIP Express Router FIFO' received : '' State : CS_SYNC Flags : 128 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ...Record(0xf50b4348)... domain: 'location' aor : 'mj' ~~~Contact(0xf50b4388)~~~ domain : 'location' aor : 'mj' Contact : 'sip:mj@myserver.sampro.cse' Expires : Permanent q : 1 Call-ID : 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything' CSeq : 42 User-Agent: 'SIP Express Router FIFO' received : '' State : CS_SYNC Flags : 128 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ---/Domain--- ===/Domain list=== ==================================================================================================================================== 4 ===================================== I received the message "received ICMP packet (type: 3, code: 10): from 192.168.178.194" What does means this message?? ============================================================================================ [root@cse3 ~]# sipsak -T -vv -s sip:iptel@192.168.178.194 warning: IP extract from warning activated to be more informational Max-Forwards set to 0 received ICMP packet (type: 3, code: 10): from 192.168.178.194 ###################################################################################### I’d appreciate it if you would give me some solutions. Thank you in advance..... mi-jin Song _________________________________________________________________ 행운의 주인공이 이번엔 나일꺼야, 진짜루... 인터넷 복권 http://www.msn.co.kr/money/interlotto/

2 1

[Serusers] Permanent aliases
by Igor Balk 23 Sep '05

23 Sep '05

Sorry if I'm asking stupid question but I'm trying to set up alias for example 1234<->john.doe and make it permanent. As far as I understand if I use serctl add alias command and then contact for the person will change (dynamic ip, login from other coputer etc) my alias would not work anymore. Please tell me if I'm worng and if there is a way to fix this problem. Thank you, Igor

3 2

[Serusers] MediaProxy Not Showing The Traffic
by Felipe Louback 23 Sep '05

23 Sep '05

I installed ser 8.14 and using mediaproxy 1.4.2. I am trying to use X-lite to make a call to the PSTN through a PSTN gateway (asterisk). The PSTN phone rings and get voice, but X-lite only pass voice. I can hear nothing. When I use the tool sessions.py available with mediaproxy, it shows me the call but shows that it is inactive, although the PSTN phone can hear audio. And after 60 seconds it drops the session, but the PSTN end still can hear the audio sent from X-lite. And sessions.py does not show the IP of asterisk.... [root@someplace mediaproxy]# ./sessions.py Caller Via Called Status Duration Codec Type Traffic ---------------------------------------------------------------------------------------------- 200.150.x.x:8000 - 127.0.0.1:35008 - ?.?.?.?:? inactive 0'10" Unknown Audio 0/0/0 If I run the rtpgenerator.py it shows the test rtp with traffic flowing. Here it is what I do in ser.cfg: if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; if (!lookup("location")) { if ( ( uri=~"^sip:[0-9]{8}@.*" ) ) { t_relay_to_udp("X.X.X.X", "5060"); break; } sl_send_reply("404", "User not found"); break; }; If I turn off mediaproxy and disable the use_media_proxy, everything works fine. Any ideas? I read every possible document and found nothing about this. Thanks Felipe -- Master Student - Electrical Engineering Department Computer Engineering and Telecommunications Research Group Universidade Federal de Minas Gerais - Brazil "For God so loved the world that he gave his one and only Son, that whoever believes in him shall not perish but have eternal life." John 3:16

2 2

RE: [Serusers] SIP/2.0 483 Too Many Hops
by Mike Williams 23 Sep '05

23 Sep '05

Here, try this script. I couldn't test it, but apart from minor errors it should work. # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) # must be yes since REGISTER processing causes an INVITE to be sent, # which needs to be processed by another process fork=yes children=4 alias=server.sam.cse alias=sam.cse # debugging log_stderror=yes # (cmd line: -E) #mhomed=yes fifo="/tmp/ser_fifo" fifo_db_url="mysql://root:vlftmd@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/exec.so" # ----------------- setting module-specific parameters --------------- modparam("rr", "enable_full_lr", 1) # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://root:vlftmd@localhost/ser") # ------------------------- 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")) { log(1, "ERROR: Too Many Hops\n"); sl_send_reply("483","Too Many Hops"); break; }; #keep this low to prevent possible DoS attacks and misconfigurations from overloading the server if (msg:len>=2048) { log(1, "ERROR: Message Too Big\n"); sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (loose_route()) { route(1); break; }; if (method=="REGISTER") { log(1, "INFO: User Registered\n"); # save location before initiating welcome save("location"); record_route(); sl_send_reply("200", "ok"); # welcome message log(1, "INFO: Initiating Welcome Message \n"); route(3); break; }; log(1, "record-route processing completed\n"); # 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=="INVITE") { log(1, "INFO: SIP Invite"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); log(1, "ERROR: User Not Found\n"); break; }; }; record_route(); }; route(1); } route[1] { #send the call outward if (!t_relay()) { xlog("L_WARN", "ERROR: t_relay failed"); sl_reply_error(); }; } # welcome message -- if a REGISTER succeeded, look if it is the first-time; # if so, initiate click-to-dial transaction to set up call to an announcement # server; edit the config values first to make it work route[3] { if (!exec_msg(' # config: # --announcement server URI ANS="sip:mijin@sam.cse" # --SIP domain # DOMAIN="192.168.178.193" DOMAIN="192.168.178.194" # ctd CTD=${HOME}/sip_router/examples/ctd.sh # ------------------------------------ # check if first time ... SIP_UID=`echo $SIP_HF_TO | sed -e "s/^.*sip:$[a-zA-Z0-9_\.]*$@.*$/\1/g"` QUERY="select flag from subscriber where username=\"$SIP_UID\"; update subscriber set flag=\"x\" where username=\"$SIP_UID\" "; mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null # ... if so, c-t-d to announcement server if [ "$?" -ne 0 ] ; then # flag was not set to x yet -- first-time registration; # initiate a call from telephone of the user to an announcement server $CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1 fi ')) { log(1, "ERROR: Welcome Message Script Failed\n"); } log(1, "INFO: Welcome Message Sent\n"); } #----------------End Script--------------------- ---Mike -----Original Message----- From: serusers-bounces(a)lists.iptel.org on behalf of ? ?? Sent: Fri 9/23/2005 6:30 AM To: serusers(a)lists.iptel.org Subject: [Serusers] SIP/2.0 483 Too Many Hops confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear.... I inserted command like this. What I want is that when I loaded test.cfg file, what I check SIP messages using sipp client mode I want 200 OK. But, as you see, I received "SIP/2.0 483 Too Many Hops" error message. So I was advanced about this. - alias="sam.cse"//SIP_ DOMAIN alias="server.sam.cse"//my_computer.SIP_DOMAIN alias="192.168.178.194"//my_ip - add my hostname in /etc/host - change to : if (!mf_process_maxfwd_header("100") but, I couldn't solve this problem. I would really appreciate for giving some advise. Thank you in advance............................. mi-jin Song ===================================================== [root@cse4 ~]# ser -f test.cfg Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.178.194 [192.168.178.194]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.178.194 [192.168.178.194]:5060 Aliases: tcp: server:5060 tcp: localhost:5060 tcp: localhost.localdomain:5060 tcp: cse4:5060 udp: server:5060 udp: localhost:5060 udp: localhost.localdomain:5060 udp: cse4:5060 *: 192.168.178.194:* *: sam.cse:* *: server.sam.cse:* stateless - initializing 0(0) Maxfwd module- initializing exec - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) ERROR: tcp_init: bind(6, 0x810dafc, 16) on 127.0.0.1: Address already in use ===================================================== [root@cse4 sipp]# ./sipp -sn uac 192.168.178.194 Resolving 192.168.178.194... ------------------------------ Scenario Screen -------- [1-4]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 10.0(0 ms)/1.000s 5061 2.54 s 25 192.168.178.194:5060(UDP) 5 new calls during 0.538 s period 2 ms scheduler resolution 0 concurrent calls (limit 30) Peak was 1 calls, after 0 s 0 out-of-call msg (discarded) 1 open sockets Messages Retrans Timeout Unexpected-Msg INVITE ----------> 25 0 0 100 <---------- 25 0 25 180 <---------- 0 0 0 200 <---------- E-RTD 0 0 0 ACK ----------> 0 0 [ 0 ms] BYE ----------> 0 0 0 200 <---------- 0 0 0 ------------------------------ Test Terminated --------------------------------- ----------------------------- Statistics Screen ------- [1-4]: Change Screen -- Start Time | 2005-09-23 20:03:13 Last Reset Time | 2005-09-23 20:03:15 Current Time | 2005-09-23 20:03:16 -------------------------+---------------------------+-------------------------- Counter Name | Periodic value | Cumulative value -------------------------+---------------------------+-------------------------- Elapsed Time | 00:00:00:536 | 00:00:02:543 Call Rate | 9.328 cps | 9.831 cps -------------------------+---------------------------+-------------------------- Incoming call created | 0 | 0 OutGoing call created | 5 | 25 Total Call created | | 25 Current Call | 0 | -------------------------+---------------------------+-------------------------- Successful call | 0 | 0 Failed call | 6 | 25 -------------------------+---------------------------+-------------------------- Response Time | 00:00:00:000 | 00:00:00:000 Call Length | 00:00:00:004 | 00:00:00:004 ------------------------------ Test Terminated -------------------------------- 2005-09-23 20:03:16: Unexpected message for Call-ID '25.4743.127.0.0.1(a)sipp.call.id': while expecting '100' response, received 'SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP 127.0.0.1:5061;received=192.168.178.194 From: sipp <sip:sipp@127.0.0.1:5061>;tag=25 To: sut <sip:service@192.168.178.194:5060>;tag=b27e1a1d33761e85846fc98f5f3a7e58.6e68 Call-ID: 25.4743.127.0.0.1(a)sipp.call.id CSeq: 1 INVITE Server: Sip EXpress router (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 192.168.178.194:5060 "Noisy feedback tells: pid=4688 req_src_ip=192.168.178.194 req_src_port=5060 in_uri=sip:service@192.168.178.194:5060 out_uri=sip:service@192.168.178.194:5060 via_cnt==18" ' . sipp: There were more errors, enable -trace_err to log them. ======================================================= << test.cfg >> # # $Id: welcome.cfg,v 1.3 2003/11/03 13:03:56 janakj Exp $ # # welcome message for new subscribers; based on exec # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) # must be yes since REGISTER processing causes an INVITE to be sent, # which needs to be processed by another process fork=yes children=4 alias="server.sam.cse" alias="sam.cse" alias="192.168.178.194" # debugging log_stderror=yes # (cmd line: -E) mhomed=yes fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/exec.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", "mysql://ser:heslo@192.168.2.16/ser" ) modparam("usrloc", "db_url", "mysql://root:vlftmd@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ log(1, "\n\n"); log(1, "START!!!\n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("15")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len>=204822) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol # if (method=="INVITE") record_route(); # 1=loose routing if(method=="INVITE"){ log(1, "receive INVITE\n"); record_route(); sl_send_reply("200", "ok"); }; # loose-route processing if (loose_route()) { t_relay(); break; }; # if(method=="ACK"){ # log(1, "receive ACK\n"); # }; log(1, "record-route processing completed\n"); # 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") { log(1, "receive REGISTER\n"); # save location before initiating welcome save("location"); log(1, "save ??\n"); record_route(); sl_send_reply("200", "ok"); # welcome message log(1, "route3?? ?? \n"); route(3); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); log(1, "Usrloc DB?? ?? ? ?? \n"); break; }; }; if(method=="BYE"){ log(1, "receive BYE\n"); record_route(); sl_send_reply("200", "ok"); break; }; t_relay(); } # welcome message -- if a REGISTER succeeded, look if it is the first-time; # if so, initiate click-to-dial transaction to set up call to an announcement # server; edit the config values first to make it work route[3] { if (!exec_msg(' # config: # --announcement server URI ANS="sip:mijin@sam.cse" # --SIP domain # DOMAIN="192.168.178.193" DOMAIN="192.168.178.194" # ctd CTD=${HOME}/sip_router/examples/ctd.sh # ------------------------------------ # check if first time ... SIP_UID=`echo $SIP_HF_TO | sed -e "s/^.*sip:$[a-zA-Z0-9_\.]*$@.*$/\1/g"` QUERY="select flag from subscriber where username=\"$SIP_UID\"; update subscriber set flag=\"x\" where username=\"$SIP_UID\" "; mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null # ... if so, c-t-d to announcement server if [ "$?" -ne 0 ] ; then # flag was not set to x yet -- first-time registration; # initiate a call from telephone of the user to an announcement server $CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1 fi ')) { log(1, "welcome exec failed\n"); } log(1, "-------route3 ? \n"); } _________________________________________________________________ ???? ?? ??? ???? ???... MSN ??? ??? http://vod.msn.co.kr _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] $from in avpops different?
by sip 23 Sep '05

23 Sep '05

I have this avpops test going... avp_db_load("$ruri/username", "$urire/$urire_scheme"); # Load the urire column for this and check to see if # it matches the $from address if(avp_check("$urire", "eq/$from/I")) { do stuff } I'm calling from the number number(a)domain.com. If I use the above, the avp_check test fails. If I put if(avp_check("$urire", "eq/number(a)domain.com/I")) instead, the check works exactly the way it's supposed to. It's almost as if the $from address isn't right. Is there some way I can check what it's doing at this point other than this painful trial and error? N.

1 0

[Serusers] need help
by chenhao 23 Sep '05

23 Sep '05

Hi! I realy curious about the details of how midiaproxy system works.I mean you know there three parts proxy sever, proxy dispather and mediaproxy module as i know.I got some general ideas from the readme file.But I dont think thats enough.Please send me some detail files or maybe one example or something.Thanks very much!

2 1

[Serusers] SIP/2.0 483 Too Many Hops
by 송 미진 23 Sep '05

23 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear.... I inserted command like this. What I want is that when I loaded test.cfg file, what I check SIP messages using sipp client mode I want 200 OK. But, as you see, I received "SIP/2.0 483 Too Many Hops" error message. So I was advanced about this. - alias="sam.cse"//SIP_ DOMAIN alias="server.sam.cse"//my_computer.SIP_DOMAIN alias="192.168.178.194"//my_ip - add my hostname in /etc/host - change to : if (!mf_process_maxfwd_header("100") but, I couldn't solve this problem. I would really appreciate for giving some advise. Thank you in advance............................. mi-jin Song ===================================================== [root@cse4 ~]# ser -f test.cfg Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.178.194 [192.168.178.194]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.178.194 [192.168.178.194]:5060 Aliases: tcp: server:5060 tcp: localhost:5060 tcp: localhost.localdomain:5060 tcp: cse4:5060 udp: server:5060 udp: localhost:5060 udp: localhost.localdomain:5060 udp: cse4:5060 *: 192.168.178.194:* *: sam.cse:* *: server.sam.cse:* stateless - initializing 0(0) Maxfwd module- initializing exec - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) ERROR: tcp_init: bind(6, 0x810dafc, 16) on 127.0.0.1: Address already in use ===================================================== [root@cse4 sipp]# ./sipp -sn uac 192.168.178.194 Resolving 192.168.178.194... ------------------------------ Scenario Screen -------- [1-4]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 10.0(0 ms)/1.000s 5061 2.54 s 25 192.168.178.194:5060(UDP) 5 new calls during 0.538 s period 2 ms scheduler resolution 0 concurrent calls (limit 30) Peak was 1 calls, after 0 s 0 out-of-call msg (discarded) 1 open sockets Messages Retrans Timeout Unexpected-Msg INVITE ----------> 25 0 0 100 <---------- 25 0 25 180 <---------- 0 0 0 200 <---------- E-RTD 0 0 0 ACK ----------> 0 0 [ 0 ms] BYE ----------> 0 0 0 200 <---------- 0 0 0 ------------------------------ Test Terminated --------------------------------- ----------------------------- Statistics Screen ------- [1-4]: Change Screen -- Start Time | 2005-09-23 20:03:13 Last Reset Time | 2005-09-23 20:03:15 Current Time | 2005-09-23 20:03:16 -------------------------+---------------------------+-------------------------- Counter Name | Periodic value | Cumulative value -------------------------+---------------------------+-------------------------- Elapsed Time | 00:00:00:536 | 00:00:02:543 Call Rate | 9.328 cps | 9.831 cps -------------------------+---------------------------+-------------------------- Incoming call created | 0 | 0 OutGoing call created | 5 | 25 Total Call created | | 25 Current Call | 0 | -------------------------+---------------------------+-------------------------- Successful call | 0 | 0 Failed call | 6 | 25 -------------------------+---------------------------+-------------------------- Response Time | 00:00:00:000 | 00:00:00:000 Call Length | 00:00:00:004 | 00:00:00:004 ------------------------------ Test Terminated -------------------------------- 2005-09-23 20:03:16: Unexpected message for Call-ID '25.4743.127.0.0.1(a)sipp.call.id': while expecting '100' response, received 'SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP 127.0.0.1:5061;received=192.168.178.194 From: sipp <sip:sipp@127.0.0.1:5061>;tag=25 To: sut <sip:service@192.168.178.194:5060>;tag=b27e1a1d33761e85846fc98f5f3a7e58.6e68 Call-ID: 25.4743.127.0.0.1(a)sipp.call.id CSeq: 1 INVITE Server: Sip EXpress router (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 192.168.178.194:5060 "Noisy feedback tells: pid=4688 req_src_ip=192.168.178.194 req_src_port=5060 in_uri=sip:service@192.168.178.194:5060 out_uri=sip:service@192.168.178.194:5060 via_cnt==18" ' . sipp: There were more errors, enable -trace_err to log them. ======================================================= << test.cfg >> # # $Id: welcome.cfg,v 1.3 2003/11/03 13:03:56 janakj Exp $ # # welcome message for new subscribers; based on exec # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) # must be yes since REGISTER processing causes an INVITE to be sent, # which needs to be processed by another process fork=yes children=4 alias="server.sam.cse" alias="sam.cse" alias="192.168.178.194" # debugging log_stderror=yes # (cmd line: -E) mhomed=yes fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/exec.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", "mysql://ser:heslo@192.168.2.16/ser" ) modparam("usrloc", "db_url", "mysql://root:vlftmd@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ log(1, "\n\n"); log(1, "START!!!\n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("15")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len>=204822) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol # if (method=="INVITE") record_route(); # 1=loose routing if(method=="INVITE"){ log(1, "receive INVITE\n"); record_route(); sl_send_reply("200", "ok"); }; # loose-route processing if (loose_route()) { t_relay(); break; }; # if(method=="ACK"){ # log(1, "receive ACK\n"); # }; log(1, "record-route processing completed\n"); # 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") { log(1, "receive REGISTER\n"); # save location before initiating welcome save("location"); log(1, "save 성공\n"); record_route(); sl_send_reply("200", "ok"); # welcome message log(1, "route3으로 이동 \n"); route(3); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); log(1, "Usrloc DB에서 찾을 수 없음 \n"); break; }; }; if(method=="BYE"){ log(1, "receive BYE\n"); record_route(); sl_send_reply("200", "ok"); break; }; t_relay(); } # welcome message -- if a REGISTER succeeded, look if it is the first-time; # if so, initiate click-to-dial transaction to set up call to an announcement # server; edit the config values first to make it work route[3] { if (!exec_msg(' # config: # --announcement server URI ANS="sip:mijin@sam.cse" # --SIP domain # DOMAIN="192.168.178.193" DOMAIN="192.168.178.194" # ctd CTD=${HOME}/sip_router/examples/ctd.sh # ------------------------------------ # check if first time ... SIP_UID=`echo $SIP_HF_TO | sed -e "s/^.*sip:$[a-zA-Z0-9_\.]*$@.*$/\1/g"` QUERY="select flag from subscriber where username=\"$SIP_UID\"; update subscriber set flag=\"x\" where username=\"$SIP_UID\" "; mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null # ... if so, c-t-d to announcement server if [ "$?" -ne 0 ] ; then # flag was not set to x yet -- first-time registration; # initiate a call from telephone of the user to an announcement server $CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1 fi ')) { log(1, "welcome exec failed\n"); } log(1, "-------route3 끝 \n"); } _________________________________________________________________ 책상위에 다리 올리고 느긋하게 즐긴다... MSN 온라인 상영관 http://vod.msn.co.kr

1 0

[Serusers] error 513 when sendin a message
by Ángel Gómez 23 Sep '05

23 Sep '05

Hi, I has send a MESSAGE to my SER server and it allways send me the response "513 message to big". I send it with sipsak -s sip:ninguna@10.22.1.249 -vvv -l 5060 -M, when I send a REGISTER or INVITE it works whell. I have alias for various domains, an it can´t be te error, have somo one had this error ever? How can I get RIP of it? please help.

1 0

[Serusers] What is a difference between Class4 and Class 5 Softswitch ( VOIP)
by Manoj Ghag 23 Sep '05

23 Sep '05

hi, All Good morning Can anybody tell me what is a exact difference between Class4 and Class5 softswitch ? Because many people or company have there own defination Regards --Manoj --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

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[Serusers] SER and sipsak problem;;;;;;;
by 송 미진 23 Sep '05

23 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear. Alll Hi, I have some questions............. My study object is "communication among many computers(three computers)based SIP". So I have used SER with sipsak. (operating system : Linux Fedora 3) but SER was not runned well. In frankly, I don't know what to do. Following is my try, my SIP_DOMAIN is "sam.cse" and ip is 192.168.178.194. [root@cse4 ~]# sipsak -s sip:mijin@sam.cse -M warning: ignoring -i option when in usrloc mode received: SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP 127.0.0.1:32778;received=192.168.178.194;branch=z9hG4bK.01b7ca3a;rport=32778 From: sip:sipsak@127.0.0.1:32778;tag=141698a0 To: sip:mijin@sam.cse;tag=b27e1a1d33761e85846fc98f5f3a7e58.5de2 Call-ID: 337025184(a)127.0.0.1 CSeq: 1 MESSAGE Server: Sip EXpress router (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 192.168.178.194:5060 "Noisy feedback tells: pid=3626 req_src_ip=192.168.178.194 req_src_port=5060 in_uri=sip:mijin@sam.cse out_uri=sip:mijin@sam.cse via_cnt==18" error: did not received the 'MESSAGE' that was sent (see above). aborting =============================================== I want to communicate another computer has ip:192.168.178.193. [root@cse4 ~]# ser -f /usr/local/etc/ser/ser.cfg -l 192.168.178.193 0(3965) WARNING: fix_socket_list: could not rev. resolve 192.168.178.193 0(3965) WARNING: fix_socket_list: could not rev. resolve 192.168.178.193 Listening on udp: 192.168.178.193 [192.168.178.193]:5060 tcp: 192.168.178.193 [192.168.178.193]:5060 Aliases: stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) ERROR: udp_init: bind(4, 0x810ab6c, 16) on 192.168.178.193: Cannot assign requested address =============================================== I setted my hostname and ip in /etc/host. and inserted alias="sam.cse" alias="server.sam.cse" alias="192.168.178.194" but, as you see, I received "483 Too Many Hops" message;;;;; I don't know what to do................ I appreciate if you give me some solutions, please ^^ Thank you in advance.... mi-jin Song _________________________________________________________________ 증권 정보 가장 빠르고 편하게 보실 수 있습니다. MSN 증권/투자 http://www.msn.co.kr/stock/

1 0

[Serusers] LCR usage (How to enter data in gw table?)
by Amos Nungu 23 Sep '05

23 Sep '05

Hi, I have a question on how to enter values specifically ip_address in the gw table? I would like to give a go this LCR module. I'm facing a problem where by: - 1. I created the tables (gw, lcr, gw_grp) 2. Inserting values into the tables ready for use. The gw:ip_addr field is int data type, not able to store the decimal IP format!! How to manage that? Thanks in advance.

6 14

[Serusers] One path of RTP traffic
by Alberto 23 Sep '05

23 Sep '05

Hi, I have a SER + Mediaproxy. I have not any problem the call between SIP clients (behind or not the NATs) but when I try to call to PSTN (via cisco) I only have RTP traffic from SIP client to PSTN. Summarizing, the path of rtp traffic would have to be from: up: SIP Client ----> SER ---> GW-PSTN down: SIP Client <---- SER <--- GW-PSTN but, really is: up: SIP Client ----> SER ---> GW-PSTN down: SIP Client <------------- GW-PSTN I use the command: rewritehostport("212.xxx.xxx.xxx:5060"); when I match a geographic number. The complete scheme is: SIP Client ---- NAT --------- SER+Mediaproxy -------- SIP Server --- GWPSTN Some idea? Thanks, -- Alberto

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[Serusers] serweb configuration
by Elad Dotan 23 Sep '05

23 Sep '05

Hi, I've installed 4 RPM files on a Fedora CORE 3. The RPM are: ser-0.9.3-0.i386.rpm php-pecl-classkit-0.4-2.i386.rpm ser-mysql-0.9.3-0.i386.rpm serweb-0.9.3-0.i386.rpm I enabled the httpd and mysql service. I'm not sure what to do next. All the Documentation I have is related to source flies and not RPM. Ser is working but I don't know how to configure the serweb in order for it to work. Could you Please help? Thank you, Elad.

1 0

[Serusers] Problems with radius_proxy_authorize and others
by Bruno Machado 23 Sep '05

23 Sep '05

Hello We are running SER 0.8.14 and FreeRadius 1.0.4 on Slackware 10 kernel 2.6.12.3. Yesterday, we had problems with some functions of SER. radius_www_authorize & www_challenge radius_proxy_authorize & proxy_challenge if(method=="REGISTER") { Make sure that user's dont register infinite loops # (note: does not match with folded lines) if(search("^(Contact|m):.*@(domain\.com\.br)")) { log(1, "LOG\n"); sl_send_reply("476","No Server Address in Contacts Allowed" ); break; }; #if (!radius_www_authorize("")) #{ #www_challenge("","0"); # break; #}; #if (!radius_proxy_authorize( "")) { #proxy_challenge("sipproxy.taho.com.br", "0" ); #break; #}; save("location"); #if (!save("location")) { # sl_reply_error(); #}; #m_dump(); break; }; This block ( if(method=="REGISTER"){block} ) has these functions. When "radius_www_authorize & www_challenge" or "radius_proxy_authorize & proxy_challenge" are uncommented, the SER becomes too slow: it receives around 5 consecutive invites in 7 seconds. After the fifth INVITE, SER answers with a 487 to the User Agent. The call to PSTN is established after around 30 seconds . When we comment these functions, the system runs perfectly, fast and fast. But is strange, because these functions are called normally in the routes we the INVITE is received: route[5] ............. if (!radius_proxy_authorize("")) { proxy_challenge( "domain.com.br", "0" ); break; }; ............. In the route, they work perfectly. I'd like to know if I'm doing wrong things in this scripts. Thanks a lot, friends. Bruno Machado __________________________________________________ Faça ligações para outros computadores com o novo Yahoo! Messenger http://br.beta.messenger.yahoo.com/

2 1

[Serusers] How to use "serctl file <file_name" command?
by Bhaskar Bhumkar 23 Sep '05

23 Sep '05

Hi, I am planning to use SER to send some SIP requests. Is it possible to initiate requests like INVITE from SER? If yes, how can I do that? I saw the description of "serctl file <file_name" and it is supposed to be used to read commands from a file. Does it refer to rest of the serctl commands or SIP requests like INVITE, REGISTER? Appreciate your help, Regards, Bhaskar...

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[Serusers] JABBER and SER
by Saleh, Saleh 22 Sep '05

22 Sep '05

Hello Everyone, I am new to this list. I would greatly appreciate anyone who can tell me what I need to do next in order for SER version 0.9.3 to use the Jabber Module correctly? This is what I have done so far (on a Red Hat Linux 9 machine): Installed Jabberd 1.4.4 Installed GNU pth-2.0.4 Installed Expat 1.95.8 Installed Libidn -I have Modified the config file for SER to load the "Jabber.so" module after the "pa.so" and "tm.so" modules have been loaded -(I'm sure this is where I am stuck, what to do with parameters in config file for SER but I have started to place those in as well, I am just confused about making the documentation match my settings/variable names) -I have partially created a database in MySQL but I have not put tables or data in it because I have never done this before. This is what I can/cannot do: -I can run the jabber server -I cannot run the SER with the config file which includes the Jabber.so module load command. What other files do I need to modify or download? What information do I need? I'm not very familiar with linux, so please assume I know basic commands/knowledge... Any comments/suggestions/websites/help is GREATLY appreciated! Thanks, Saleh

1 0

[Serusers] Avpops info....?
by sip 22 Sep '05

22 Sep '05

Is there any good, DETAILED information on AVPops usage that, say, doesn't read like a developers API reference page? Currently, I'm cleaning up portions of my ser.cfg which have horrid, ugly hacks in them that I put in to handle this case or that case... For instance, I use the calls_forwarding table to handle call blocking and call forwarding info (since it seems a little more reasonable than usr_preferences as it simply has more relevant tables). For call blocking, I can say give a purpose of 'blocking' and an action of relay or reply... with parameters set to what kind of reply or where to relay (voicemail server? Alternate server? etc). For forwarding, I can give an action of forwarding with a parameter of where to forward and another to see if he has permission to forward to PSTN numbers. Seems like a very handy table for it. Now, my current method of doing this is with this horrid mixture of external scripts, stripping portions off the SIP_HF_FROM header (since it comes with additional info like tags and brackets and comments), and feeding the new stuff back into an SQL query to determine whether or not the call needs to be blocked, forwarded, and if either, where it goes. It's a mess, I'll admit, but it's because I've been poring over AVPops stuff and have yet to figure out what I'm actually DOING with any of it. I get that it uses little i:XX constructs as variables (which is bizarre, but whatever -- I imagine I could alias them to something human-readable). And... well... that's about as far as I've gotten deciphering the API docs and making them into something usable. I imagine I would what... set the avp_table to be calls_forwarding, and then create some schemes for it (?)... but after that, actually doing things like taking just the sip:bob@fred.com portion of an incoming uri (which would look along the lines of "Ima Doofus" <sip:bob@fred.com:5060;user=tel>tag=something;tag2=something else ) and then trying to determine if that user is on the list of blocked users and, if so, what should be the response action... or taking the callee info, and if the callee isn't available, checking to see if he has a forwarding number, checking to see where it goes, etc, etc. It all SEEMS like it should be elementary stuff, but I'm simply not getting it. Anyone have any good references, well-detailed documentation, long, in-depth books on the topic they could send me toward? I'd be your friend for, like, ever. :) N.

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Re: [Serusers] Avpops info....?
by Paul Hazlett 22 Sep '05

22 Sep '05

You could look at the Getting Started document on www.onsip.org because we use the AVPOPS module in several examples and we give a much deeper description of what the functions do. However, we only use a subset of the AVPOPS library. Regards, Paul On 9/22/05, sip <sip(a)arcdiv.com> wrote: > Yeah... the problem with this documentation is that it was clearly written by > someone who doesn't WRITE documentation as a matter of course, but needed > something to be out there just to say there's documentation. > > It's a little like a developer writing docs for his own code or API. VERY > understandable to the developer himself, but not so crystal to anyone else. > This is one of the primary reasons I never let my developers write their own > API docs. They get with a tech writer, and the tech writer asks questions > about what the API does, and then writes documentation in such a way that > monkeys (such as myself) can understand. > > I was looking for something a little more... lucid. > > If it doesn't exist, cool. I'll keep muddling away and perhaps I'll hit that > eureka point of understanding somewhere, but I was hoping. > > N. > > > On Thu, 22 Sep 2005 09:29:31 -0400, Paul Hazlett wrote > > The offical documentation for AVPOPS (for SER-0.9.x) can be found here: > > > > http://www.voice-system.ro/docs/avpops/0.9.0/ > > > > Regards, > > Paul > > > > On 9/22/05, sip <sip(a)arcdiv.com> wrote: > > > Is there any good, DETAILED information on AVPops usage that, say, doesn't > > > read like a developers API reference page? Currently, I'm cleaning up portions > > > of my ser.cfg which have horrid, ugly hacks in them that I put in to handle > > > this case or that case... > > > > > > For instance, I use the calls_forwarding table to handle call blocking and > > > call forwarding info (since it seems a little more reasonable than > > > usr_preferences as it simply has more relevant tables). For call blocking, I > > > can say give a purpose of 'blocking' and an action of relay or reply... with > > > parameters set to what kind of reply or where to relay (voicemail server? > > > Alternate server? etc). For forwarding, I can give an action of forwarding > > > with a parameter of where to forward and another to see if he has permission > > > to forward to PSTN numbers. Seems like a very handy table for it. > > > > > > Now, my current method of doing this is with this horrid mixture of external > > > scripts, stripping portions off the SIP_HF_FROM header (since it comes with > > > additional info like tags and brackets and comments), and feeding the new > > > stuff back into an SQL query to determine whether or not the call needs to be > > > blocked, forwarded, and if either, where it goes. > > > > > > It's a mess, I'll admit, but it's because I've been poring over AVPops stuff > > > and have yet to figure out what I'm actually DOING with any of it. I get that > > > it uses little i:XX constructs as variables (which is bizarre, but whatever -- > > > I imagine I could alias them to something human-readable). And... well... > > > that's about as far as I've gotten deciphering the API docs and making them > > > into something usable. > > > > > > I imagine I would what... set the avp_table to be calls_forwarding, and then > > > create some schemes for it (?)... but after that, actually doing things like > > > taking just the sip:bob@fred.com portion of an incoming uri (which would look > > > along the lines of "Ima Doofus" > > > <sip:bob@fred.com:5060;user=tel>tag=something;tag2=something else ) and then > > > trying to determine if that user is on the list of blocked users and, if so, > > > what should be the response action... or taking the callee info, and if the > > > callee isn't available, checking to see if he has a forwarding number, > > > checking to see where it goes, etc, etc. > > > > > > It all SEEMS like it should be elementary stuff, but I'm simply not getting > > > it. Anyone have any good references, well-detailed documentation, long, > > > in-depth books on the topic they could send me toward? > > > > > > I'd be your friend for, like, ever. :) > > > > > > N. > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > >

1 0

[Serusers] Audiocodes + SER
by GT 22 Sep '05

22 Sep '05

Hello All, Immediately I start SER, I get the following errors using SER with Audiocodes. Is it SER working with Audiocodes, if not which devices are recommended ? MP-104 errors Version ID: 4.60A.016.003 1d:23h:23m:11s ( sip_stack)(653 ) !! [ERROR] SIPStackMngr: Message was not Parsed correctly ! 1d:23h:23m:11s ( sip_stack)(654 ) !! [ERROR] Last parsed line was : 1d:23h:23m:11s ( sip_stack)(655 ) !! [ERROR] The error was = Unexpected input 1d:23h:23m:11s ( sip_stack)(656 ) !! [ERROR] The Line of the error was = 1 1d:23h:23m:11s ( sip_stack)(657 ) !! [ERROR] The Column of the error was = 2 Thanks, Ggtel

1 0

[Serusers] Remote Party ID
by Sebastian Kühner 22 Sep '05

22 Sep '05

Hi, I'm using the append_rpid_hf() command to add the remote party id. I've implemented it like this: modparam("auth", "rpid_prefix", "<sip:") modparam("auth", "rpid_suffix", "@xxx.xxx.xxx.xxx;user=phone>;party=calling;id-type=subscriber;screen=yes;privacy=off") [...] # first the caller needs to be authenticated if (uri=~"^sip:[0-9]*@(xxx.xxx.xxx.xxx|(voip1\.)?test\.com)") { if (!(src_ip==xxx.xxx.xxx.xxx | method==ACK | method=="CANCEL" | method=="BYE")) { if (!proxy_authorize("xxx.xxx.xxx.xxx", "subscriber")) { proxy_challenge( "xxx.xxx.xxx.xxx","0"); break; } else if (method=="INVITE" & !check_from()) { log(1, "LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; }; }; append_rpid_hf(); What I noticed is that sometimes ser sends it and sometimes not. I have a value in the rpid column of the user... Has anyone of you an idea why that happens?? Thanks! Sebastian

1 0

[Serusers] Can I put network or subnet in trusted table?
by Ryan Pagquil 22 Sep '05

22 Sep '05

Hi, Will SER able to determine if I put a whole network or subnet in the trusted table? So that I will not need to put all the ip addresses of my network in the trusted table. Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

2 1

[Serusers] dbtext module.
by Skimmer 22 Sep '05

22 Sep '05

Hello! Could you help me? I try to use ser with dbtext module. In ser.cfg I have... ------------------------------------------------ ... loadmodule "/opt/sfw/lib/ser/modules/dbtext.so" ... modparam("usrloc", "db_mode", 2) modparam("usrloc|auth_db", "db_url", "/var/sertdb" ... Where /var/sertdb - the path to text files (the tables) I have the following tables: ----------------------------------------------- root@Solaris>ls -l /var/sertdb/ total 6 -rw-rw-rw- 1 root other 90 Sep 22 13:04 location -rw-rw-rw- 1 root other 59 Sep 22 12:26 subscriber -rw-rw-rw- 1 root other 59 Sep 22 13:07 version The "Location" file ----------------------------------------------- username(str) domain(str,null) contact(str) expires(int) q(double) callid(str) cseq(int) But when I start ser I got the following messages in log: ----------------------------------------------- Sep 22 13:32:19 Solaris ser[8611]: [ID 767465 daemon.warning] Warning: uac does not spread accross the whole hash table Sep 22 13:32:19 Solaris ser[8611]: [ID 736066 daemon.error] preload_udomain(): Error while doing db_query Sep 22 13:32:19 Solaris ser[8611]: [ID 247783 daemon.error] register_udomain(): Error while preloading domain 'location' Sep 22 13:32:19 Solaris ser[8611]: [ID 551419 daemon.error] domain_fixup(): Error while registering domain What is the problem? What I do wrong? Thank you. Best regards. Simon. -----------------ser.cfg----------------------------- ----------------------------------------------------- check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/opt/sfw/lib/ser/modules/mysql.so" loadmodule "/opt/sfw/lib/ser/modules/dbtext.so" loadmodule "/opt/sfw/lib/ser/modules/sl.so" loadmodule "/opt/sfw/lib/ser/modules/tm.so" loadmodule "/opt/sfw/lib/ser/modules/rr.so" loadmodule "/opt/sfw/lib/ser/modules/maxfwd.so" loadmodule "/opt/sfw/lib/ser/modules/usrloc.so" loadmodule "/opt/sfw/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/opt/sfw/lib/ser/modules/auth.so" #loadmodule "/opt/sfw/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) modparam("usrloc|auth_db", "db_url", "/var/sertdb" modparam("usrloc", "user_column", "username" # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) #

2 1

[Serusers] 513 message too big
by Ángel Gómez 22 Sep '05

22 Sep '05

hi people, I have instaled SIP Express Router in al PC with debian sarge and trying to test it. I have tested the REGISTER and INVITE request, and it works, but when I send a MESSAGE with sipsak (sispak - M) the server gave me a response of "513 message too big". I have search the web for a solution but I did got it. The message arive to te server one time (it don`t enter into a bucle). Thank for helping, and sorry for mi bad english.

1 0

[Serusers] how to log in softphone?
by zhangshuai 22 Sep '05

22 Sep '05

Dear all, I'm new to SER. I wish to use softphone X-lite on SER and radius in db_mode. I have installed SER 0.8.14 (by tarball) and Mysql 3.23.54 in Redhat 9. I have added some subscribers into DB "ser". And after I added some users in the file: /usr/local/etc/raddb/users as the document "iptel.org SIP Express Router v0.11.0 -- Admin's Guide"(http://www.iptel.org/ser/doc/seruser/seruser.html#DEFAULTSCRIPT) guided in Chap 3.3: joe(a)iptel.org Auth-Type := Digest, User-Password == "heslo" Reply-Message = "Authenticated", Sip-Rpid = "1234" I could log in X-lite and make calls. Now, how could I log in X-lite by using the users that DB contains rather than those in the file: /usr/local/etc/raddb/users? Many thanks. Shuai http://www.godentek.biz

1 0

Re: RE: [Serusers] need help
by zhangshuai 22 Sep '05

22 Sep '05

hi, Rodrigue ESTEVE�� I think the counter will increase every 2 or 3 seconds. Hi, Is the counter increasing when you start ?nbsp;serctl monitor ?nbsp;? Are there any errors in your /var/log/syslog for ser ? Do you still receive icmp port unreachable error ? Can you do the following: # nc -u your_sip_ip 5060 Type something, if you receive an error message like this : read(net): Connection refused, your sip port is not reachable. De : julien [mailto:julientoser@gmail.com] Envoy?nbsp;: jeudi 22 septembre 2005 07:31 ?nbsp;: Iqbal Cc : Rodrigue ESTEVE; serusers(a)iptel.org Objet : Re: [Serusers] need help sorry... i want to say that: i can not see port 5060 when i try netstat -an. Cos i test wit fedora, so i am using ser version 0.8.12 On 9/22/05, julien <julientoser(a)gmail.com> wrote: Now, ser is running. But i can see port 5060 when i try netstat -an. I force the line listen=10.0.0.6 (ip of server linux) in ser.cfg something is wrong? On 9/22/05, julien <julientoser(a)gmail.com > wrote: Hi all, At least i see in the console the line error in ser.cfg, so i am trying to correct it. On 9/22/05, julien bossart <julientoser(a)gmail.com > wrote: Thank you for your instruction. To:Rodrigue Actually, i don't see that my server linux open port 5060 when i use netstat -an To:iqbal I can ping to other IP address. Today, i make the same example, but with a line ADSL so my server now have ip 10.0.0.6 (ip private). Then i use ip 10.0.0.5 for sip client. I try to connect to server but i get the same IMCP. Today, i tried to do: /etc/init.d/ser start -c -E to check the config file, it notice that: ERROR: bad config file (3 errors). (I don't understand cos i use ser.cfg default and uncomment only some line for mysql) What will you do when you meet this error? how can i active mode debug to find the error in ser.cfg? Thank you so much for your instruction. Julien On 9/21/05, Iqbal <iqbal(a)gigo.co.uk> wrote: and find the IP address of the client, do a traceroute to it, although this will fail it might let you see how far it is getting. Also can you ping any other IP addresses like ping www.yahoo.com iqbal Rodrigue ESTEVE wrote: > Hi Julien, > > Can you do this in command line on your linux : > > Netstat �an > > And check that your ser is listening on udp port 5060. > > rodrigue > > ------------------------------------------------------------------------ > > *De :* serusers-bounces(a)iptel.org [mailto:serusers-bounces@iptel.org] > *De la part de* julien bossart > *Envoy?:* mercredi 21 septembre 2005 12:02 > *?:* serusers(a)iptel.org > *Objet :* [Serusers] need help > > Hi all, > > I am new to this forum. > > Today, i download ser package, then i install in my fedora core 4. > > Everything is ok. > > Next, i start ser and use serctl add to add account to ser server. > > Then i use sjphone to connect to ser server. > > Sjphone try to send REGISTER to ser server. I use ethereal to get > > the package. I see that ser server responds ICMP host unreacheable > > all the time. So, REGISTER is failed. > > Can you give me some suggestion? > > Thank you for that. > >------------------------------------------------------------------------ > >_______________________________________________ >Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers > > = = = = = = = = = = = = = = = = = = = = = = �� Shuai ��zhangshuai(a)goldentek.biz ��2005-09-22

1 0

[Serusers] ser + cpl-script timeuot question!
by Oskar Hartikainen 22 Sep '05

22 Sep '05

Hi. Can some one explain or have a hint on, why my ser 0.9.3 will not use timeout specified in a cpl-script (<proxy timeout="15">)? ser uses the modparam("tm", "fr_inv_timer", 20) specified in ser.cfg. how do i get ser to use the timeout from the cpl-script proxy tag. the script is working properly otherwise. Thanks, Oskar _________________________________________________________________________ <?xml version="1.0" encoding="UTF-8"?> <!DOCTYPE cpl SYSTEM "cpl-06.dtd"> <cpl> <subaction id="totest"> <location url="sip:test@7017ws12.haninge.kth.se"> <proxy/> </location> </subaction> <incoming> <location url="sip:oskar@7017ws07.haninge.kth.se"> <proxy timeout="15"> <busy> <sub ref="totest"/> </busy> <noanswer> <sub ref="totest"/> </noanswer> <failure> <sub ref="totest"/> </failure> </proxy> </location> </incoming> </cpl> ________________________________________________________________________

1 0

RE: [Serusers] need help
by Rodrigue ESTEVE 22 Sep '05

22 Sep '05

Hi, Is the counter increasing when you start « serctl monitor » ? Are there any errors in your /var/log/syslog for ser ? Do you still receive icmp port unreachable error ? Can you do the following: # nc -u your_sip_ip 5060 Type something, if you receive an error message like this : read(net): Connection refused, your sip port is not reachable. ________________________________ De : julien [mailto:julientoser@gmail.com] Envoyé : jeudi 22 septembre 2005 07:31 À : Iqbal Cc : Rodrigue ESTEVE; serusers(a)lists.iptel.org Objet : Re: [Serusers] need help sorry... i want to say that: i can not see port 5060 when i try netstat -an. Cos i test wit fedora, so i am using ser version 0.8.12 On 9/22/05, julien <julientoser(a)gmail.com> wrote: Now, ser is running. But i can see port 5060 when i try netstat -an. I force the line listen=10.0.0.6 <http://10.0.0.6/> (ip of server linux) in ser.cfg something is wrong? On 9/22/05, julien <julientoser(a)gmail.com > wrote: Hi all, At least i see in the console the line error in ser.cfg, so i am trying to correct it. On 9/22/05, julien bossart <julientoser(a)gmail.com > wrote: Thank you for your instruction. To:Rodrigue Actually, i don't see that my server linux open port 5060 when i use netstat -an To:iqbal I can ping to other IP address. Today, i make the same example, but with a line ADSL so my server now have ip 10.0.0.6 <http://10.0.0.6/> (ip private). Then i use ip 10.0.0.5 <http://10.0.0.5/> for sip client. I try to connect to server but i get the same IMCP. Today, i tried to do: /etc/init.d/ser start -c -E to check the config file, it notice that: ERROR: bad config file (3 errors). (I don't understand cos i use ser.cfg default and uncomment only some line for mysql) What will you do when you meet this error? how can i active mode debug to find the error in ser.cfg? Thank you so much for your instruction. Julien On 9/21/05, Iqbal <iqbal(a)gigo.co.uk> wrote: and find the IP address of the client, do a traceroute to it, although this will fail it might let you see how far it is getting. Also can you ping any other IP addresses like ping www.yahoo.com <http://www.yahoo.com/> iqbal Rodrigue ESTEVE wrote: > Hi Julien, > > Can you do this in command line on your linux : > > Netstat -an > > And check that your ser is listening on udp port 5060. > > rodrigue > > ------------------------------------------------------------------------ > > *De :* serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] > *De la part de* julien bossart > *Envoyé :* mercredi 21 septembre 2005 12:02 > *À :* serusers(a)lists.iptel.org > *Objet :* [Serusers] need help > > Hi all, > > I am new to this forum. > > Today, i download ser package, then i install in my fedora core 4. > > Everything is ok. > > Next, i start ser and use serctl add to add account to ser server. > > Then i use sjphone to connect to ser server. > > Sjphone try to send REGISTER to ser server. I use ethereal to get > > the package. I see that ser server responds ICMP host unreacheable > > all the time. So, REGISTER is failed. > > Can you give me some suggestion? > > Thank you for that. > >------------------------------------------------------------------------ > >_______________________________________________ >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> > >

1 0

[Serusers] iptel.org setup
by Ryan Pagquil 22 Sep '05

22 Sep '05

Hi, I'm interested in iptel.org ser configuration. The security setup is nice, where those iptel.org users that are not registered will not be able to call anywhere, and also those users that are not belong to iptel.org will also not permitted to make a call. But registered users from other domains can call registered iptel.org users. What should be the ser.cfg for me to achieve this? Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

1 0

RE: [Serusers] need help
by Rodrigue ESTEVE 22 Sep '05

22 Sep '05

Hi Julien, Can you do this in command line on your linux : Netstat -an And check that your ser is listening on udp port 5060. rodrigue ________________________________ De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de julien bossart Envoyé : mercredi 21 septembre 2005 12:02 À : serusers(a)lists.iptel.org Objet : [Serusers] need help Hi all, I am new to this forum. Today, i download ser package, then i install in my fedora core 4. Everything is ok. Next, i start ser and use serctl add to add account to ser server. Then i use sjphone to connect to ser server. Sjphone try to send REGISTER to ser server. I use ethereal to get the package. I see that ser server responds ICMP host unreacheable all the time. So, REGISTER is failed. Can you give me some suggestion? Thank you for that.

4 5

[Serusers] npdi strip from uri
by Jason Mohler 22 Sep '05

22 Sep '05

Please help. I have ndpi=yes in my uri and i need to get it stripped out. I have tried replace and it did nto work. Does anyone have any suggestions? Thanks. example: INVITE sip:3302471580;npdi=yes@xxx.xxx.xxx.xxx;user=phone SIP/2.0..Via: SIP/ 2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3accrq304g50udkp01s1.1sr..From: <sip:4432316574@xxx.xxx.xxx.xxx:5060;isup-oli=0;GX=GXDIDGOLD-dkdc1o22 11mf4>;pstn-params=808481808882;tag=SD5gi2b01-121..To: <sip:3302471580@xxx.xxx.xxx.xxx :5060>..Call-ID: SD5gi2b01-74753aacf8cebc27afa2abeba01ef67b-v3000 i1..CSeq: 31346 INVITE..Allow: OPTIONS, SUBSCRIBE, INVITE, CANCEL, ACK, PRA CK, INFO, REFER, NOTIFY, BYE..Accept: multipart/mixed, application/sdp, app lication/isup, application/dtmf, application/dtmf-relay..Contact: <sip:4432316574@xxx.xxx.xxx.xxx:5060;GX=GXAPTELADIDGOLD-dkdc1o2211mf4;transport=udp> ..Remote-Party-ID: <sip:4432316574@xxx.xxx.xxx.xxx:4000>;privacy=off..Content- Length: 161..Content-Disposition: session;handling=required..Content-Type: application/sdp..Max-Forwards: 70....v=0..o=Sonus_UAC 31345 16101 IN IP4 xxx.xxx.xxx.xxx. .s=SIP Media Capabilities..c=IN IP4 xxx.xxx.xxx.xxx..t=0 0..m =audio 50050 RTP/AVP 0..a=sendrecv..a=ptime:20..

1 0

[Serusers] How to enable aliases without expiration?
by Ryan Pagquil 21 Sep '05

21 Sep '05

Hi guys, How can I make an alias that will never expire? I want my users to use their username to register in SER but I want them to make calls using numbers or aliases? is this possible? I checked the alises table and it has an expire field, does it really expire on that time? Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

2 4

[Users] RE: AW: [Serusers] CDRtool-3.3.3
by harry gaillac 21 Sep '05

21 Sep '05

Hello, > That's what i find in > http://cdrtool.ag-projects.com/EVALUATION.txt I read it but can we set the billing application for non commercial use ? I just can't install the basic gui which one read the radacct table . I can't set rates tables, ... Does ag-projects.com provide a very limited of CDRTool ? Thanks for help Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] re: SER, rtpproxy, and zyxel wifi phone
by Yair Hakak 21 Sep '05

21 Sep '05

hi all, i have a zyxel wifi phone running with SER. it registers fine, and i can dial out. However, i can't hear anything. configuration is as follows: zyxel behind nat (d-link home router) <->SER on public IP:5060 <->asterisk on same public IP:5070 i set up an extension on asterisk to play music, called it from the zyxel. no music. i am using rtpproxy. and the zyxel is set up with NAT=outbound proxy. if i do the same from an ATA behind the same nat, i get the music. does anyone have a zyxel working behind a nat, and have a config to share? or any other help? all help is much appreciated, yair

1 0

[Users] Radius Authorization and AVP Retrieval
by Tavis P 21 Sep '05

21 Sep '05

I'm having some trouble trying to integrate both Radius authentication and Radius AVP storage. I'm using freeradius and there doesn't seem to be much documentation regarding configuration semantics and such. Could anyone share some information on how they were able to use a Radius server to authenticate requests and (with avp_radius) as an AVP retrieval mechanism? I'm uncertain as to how I can stack these two uses of data on the freeradius server, the problem is that when a client authenticates (either INVITE proxy-authorization or REGISTER) the radius transaction is successfull and several SIP-AVP attributes are returned. Later on in the script i'm trying to use the avp_radius module (avp_load_radius function) to load SIP-AVPs for the callee which fails on the radius server due to the lack of digest authentication credentials in the radius requst. Is there some technique i can employ on the freeradius server to allow this second transaction? I'm not a freeradius expert and so i have not yet found a way to do this tavis

1 0

[Serusers] Stun Exception
by rajesh volluru 21 Sep '05

21 Sep '05

hi Would anyone please help me handling below exception while using sip communicator 2594 [NetworkDiagnosticsThread] ERROR common.NetworkDiagnostics - The network diagnostics process has failed with the following exception net.java.stun4j.StunException: An Exception occurred while sending message bytes through a network socket! at net.java.stun4j.stack.NetAccessManager.sendMessage(NetAccessManager.java:397) at net.java.stun4j.stack.StunClientTransaction.sendRequest(StunClientTransaction.java:205) at net.java.stun4j.stack.StunProvider.sendRequest(StunProvider.java:95) at net.java.stun4j.client.BlockingRequestSender.sendRequestAndWaitForResponse(BlockingRequestSender.java:79) at net.java.stun4j.client.NetworkConfigurationDiscoveryProcess.doTestI(NetworkConfigurationDiscoveryProcess.java:284) at net.java.stun4j.client.NetworkConfigurationDiscoveryProcess.determineAddress(NetworkConfigurationDiscoveryProcess.java:186) at net.java.sip.communicator.common.NetworkDiagnostics.run(NetworkDiagnostics.java:94) Caused by: java.lang.NullPointerException: null address || null buffer at java.net.PlainDatagramSocketImpl.send(Native Method) at java.net.DatagramSocket.send(DatagramSocket.java:611) at net.java.stun4j.stack.NetAccessPoint.sendMessage(NetAccessPoint.java:200) at net.java.stun4j.stack.NetAccessManager.sendMessage(NetAccessManager.java:393) ... 6 more Thanks Rajesh --------------------------------- Yahoo! India Matrimony: Find your partner now.

1 0

[Users] CDRtool-3.3.3
by harry gaillac 21 Sep '05

21 Sep '05

Hello, I wish to setup CDRTool-3.3.3 in order to test a billing system. I can't configure neither rating engine nor SOAP/XML server in non commercial release ! Does anybody know this billing system ? Regards harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] | operator
by Federico Giannici 21 Sep '05

21 Sep '05

The "|" operator (in expressions like method == "REGISTER|INVITE|MESSAGE") seems to DON'T work when used more than once in the string (with more than two choices). Is this a known limitation? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

2 1

[Serusers] using postgresql
by Nhadie 21 Sep '05

21 Sep '05

Hi All, I'd like to use postgres instead of mysql. Is there a script like ser_mysql.sh to create the database/tables or should I add it manually? Regard Nhadie

1 0

[Serusers] reach a radius attribute
by Mihaly Zachar 21 Sep '05

21 Sep '05

Hi everybody, I would like to build a condition, based on an attribute (h323-preferred-lang), what I get while the registration process. How can I reach this value in ser.cfg ?? What should I read about it ? Thanks, regards, Misi

2 3

[Serusers] ONSIP Script + Java Rockx ser 0.9.0 script
by Aisling O'Driscoll 21 Sep '05

21 Sep '05

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://mail.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 $ #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.

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RE: [Serusers] how to load radius-related modules in ser 0.8.14?
by Ricardo Martinez 21 Sep '05

21 Sep '05

Hi Shuai You need to edit the ser "Makefile" to include the radius related modules and then recompile SER. There is a HOWTO for Radius somewhere in the website. Regards, Ricardo Martinez > -----Mensaje original----- > De: zhangshuai [mailto:zhangshuai@goldentek.biz] > Enviado el: Miércoles, 21 de Septiembre de 2005 7:42 > Para: serusers > Asunto: [Serusers] how to load radius-related modules in ser 0.8.14? > > > Dear all, > > > I'm new to SER. I wish to use softphone X-lite on SER and > radius in db_mode. > > I have installed SER 0.8.14 (by tarball) and Mysql 3.23.54 in > Redhat 9. Also, I have added some subscribers into DB "ser". > After setting /usr/local/etc/ser/ser.cfg as the document > "iptel.org SIP Express Router v0.11.0 -- Admin's > Guide"(http://www.iptel.org/ser/doc/seruser/seruser.html#DEFAU > LTSCRIPT) guided, I could load the modules "auth" and > "auth_db" and successfully log any subscriber that Mysql > contains in X-lite. > > And now, I want to load radius-related modules: "acc", > "auth_radius", "group_radius", "uri_radius" in SER. But when > I put only one line: "loadmodule > "/usr/local/lib/ser/modules/auth_radius.so"" in ser.cfg and > then run ser, there is an error reply: "ERROR: bad config > file (1 errors)". > > How come? How to load those modules properly? > > Many thanks. > > > > Shuai > http://www.godentek.biz > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[Serusers] db mode radius
by Nhadie 21 Sep '05

21 Sep '05

Hi All, I was able to authenticate via radius using the file /etc/raddb/users 1001(a)my.domain.com Auth-Type := Digest, User-Password == "mypassword" Reply-Message = "Authenticated" Next thing I want to do is authenticate via radius using postgres, I've already created the tables but I don't know what things i should on there. Anyone can give me a hint? I would like to do accounting via radius/postgres. If there are any documents you know I can read kindly send it. Thank You. Regards, Nhadie

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Re: [Serusers] Handoff to PSTN
by Iqbal 21 Sep '05

21 Sep '05

do you have ngrep installed, also to see if you are in the loop, see if you LOG line is being displayed in /var/log/messages or wherever else you are dumping ser's output Colin Jordan wrote: >It is my first time with this, so sorry for the stupid question. I run >ngrep, it does not see it as a command. How can I see if I am falling into >the loop? > >-----Original Message----- >From: Iqbal [mailto:iqbal@gigo.co.uk] >Sent: Wednesday, September 21, 2005 3:11 PM >To: Colin Jordan >Cc: 'Leonid Fainshtein'; serusers(a)lists.iptel.org >Subject: Re: [Serusers] Handoff to PSTN > >Can u post a ngrep of the call, to see what is actually being sent to >and from where. > >Also have you checked to see if you are actually falling into the loop, >i.e the if statement is matching > >Iqbal > >Colin Jordan wrote: > > > >>Hi there >> >>Yes I have just tried it but still do not work. >> >>Best regards, >> >>Colin >> >>-----Original Message----- >>From: Leonid Fainshtein [mailto:LFAIN@boscom.com] >>Sent: Wednesday, September 21, 2005 3:27 PM >>To: cjordan(a)econetwireless.com >>Cc: serusers(a)lists.iptel.org >>Subject: RE: [Serusers] Handoff to PSTN >> >>Hi Colin, >>First of all, according to RFC3261 the pound sign can't be used in the >>user name. Secondly, it may be considered by SER as start of the >>comment. >>Have you tried to put prefix 666 without '#'? >> >>Best regards, >>Leonid Fainshtein >>DISCLAIMER: The information in this message is confidential and is legally >>privileged. It is intended solely for the addressee. Access to this message >> >> >by > > >>anyone else is unauthorized.If receiving in error please accept our >> >> >apologies > > >>and notify the sender immediately. You must also delete the original >>message from your machine. If you are not the intended recipient, any use, >>disclosure, copying, distribution or action taken in reliance of it, is >> >> >prohibited > > >>and maybe >>unlawful. >> >> >> >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >>. >> >> >> >> >> >DISCLAIMER: The information in this message is confidential and is legally >privileged. It is intended solely for the addressee. Access to this message by >anyone else is unauthorized.If receiving in error please accept our apologies >and notify the sender immediately. You must also delete the original >message from your machine. If you are not the intended recipient, any use, >disclosure, copying, distribution or action taken in reliance of it, is prohibited >and maybe >unlawful. > > > > > >. > > >

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RE: [Serusers] Handoff to PSTN
by Leonid Fainshtein 21 Sep '05

21 Sep '05

Hi Colin, First of all, according to RFC3261 the pound sign can't be used in the user name. Secondly, it may be considered by SER as start of the comment. Have you tried to put prefix 666 without '#'? Best regards, Leonid Fainshtein

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[Users] Freebsd segmentation fault problem
by Diego Valencia 21 Sep '05

21 Sep '05

I compiled the openser 0.9.5 with acc module, and radiusclient-ng 0.5.1 on freebsd 5.4. The compilation is fine, but when I configure the config file to load acc.so module, I see this error: blah..blah..blah WARNING: no fork mode and more than one listen address found(will use only the the first one) 0(28499) shm_mem_init: success 0(0) core dump limits set to 134596096 0(0) DEBUG: init_mod: acc 0(0) ACC - initializing 0(0) find_export: found <load_tm> in module tm [/home/dvalencia/usr/local/lib/openser/modules/tm.so] Segmentation fault (core dumped) Anybody can help me about this? Thanks! ____________________ Diego Valencia Network Support PowerVT SA Tel: 0351-4257748 Movil: 0351-155194286

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[Serusers] how to load radius-related modules in ser 0.8.14?
by zhangshuai 21 Sep '05

21 Sep '05

Dear all, I'm new to SER. I wish to use softphone X-lite on SER and radius in db_mode. I have installed SER 0.8.14 (by tarball) and Mysql 3.23.54 in Redhat 9. Also, I have added some subscribers into DB "ser". After setting /usr/local/etc/ser/ser.cfg as the document "iptel.org SIP Express Router v0.11.0 -- Admin's Guide"(http://www.iptel.org/ser/doc/seruser/seruser.html#DEFAULTSCRIPT) guided, I could load the modules "auth" and "auth_db" and successfully log any subscriber that Mysql contains in X-lite. And now, I want to load radius-related modules: "acc", "auth_radius", "group_radius", "uri_radius" in SER. But when I put only one line: "loadmodule "/usr/local/lib/ser/modules/auth_radius.so"" in ser.cfg and then run ser, there is an error reply: "ERROR: bad config file (1 errors)". How come? How to load those modules properly? Many thanks. Shuai http://www.godentek.biz

1 0

[Serusers] failure route
by Jason Mohler 21 Sep '05

21 Sep '05

Okay everyone...I have an easy one for you. I am trying to do a simple fail over from one ip to the other if I get a 4040 and it fails. Can one of you type the correct syntax for me. Below is what I am trying to accomplish. if (uri=~"sip:^1234567890.*@.*"){ prefix("1234"); rewritehostport("xxx.xxx.xxx.xxx:5060"); I need the hostport to be able to be rolled to another hostport ip if this one comes back with a 404 or any other failure. Can on eof you please help?

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[Serusers] Handoff to PSTN
by Colin Jordan 21 Sep '05

21 Sep '05

I have a big problem handing my call of to the PSTN, It give me a 404 error (not connected). What can I do to fix this? My handoff script looks like this: if (uri=~"^sip:9[0-9] *(a)gaco.co.za") { log ("Forwarding to PSTN\n"); prefix("666#"); t_relay_to_udp("217.15.11.87","5060"); break; }; The PSTN interconnect description: 1) I need to send a prefix of 666# followed by full intl number without leading 0's. eg. 666#442071016211 2) The PSTN are connecting me to a Cisco SIP Proxy server, which uses Tech-prefixes eg 123# to separate and authenticate the calls into different domains. What can I do to get it working? DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

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[Serusers] Problem with Notify
by Sam Lee 21 Sep '05

21 Sep '05

Dear all, I am getting 404 User Not Found whenever i received a NOTIFY method. Don't know what is wrong with it. All i did in the ser.cfg was a simple lookup("location") but it seems that they cannot find the record in the location table. I have check and confirm that the user location record is in the table. What is going on ? What are they suppose to compare or what is the lookup suppose to do ? What i cannot understand is why are there no if else to take in method==NOTIFY . Please help. Regards, Sam

1 0

[Serusers] warning(4349) Warning: could not rev.resolve 192.168.0.1
by julien bossart 21 Sep '05

21 Sep '05

Hi all experts I install ser in server linux 192.168.0.1 <http://192.168.0.1> fedora. After start ser, i see in the console: Warning:could not rev.resolve 192.168.0.1 <http://192.168.0.1> So that when i send REGISTER to server SER. SER always responds ICMP host unreachable. What do you think? Many thank for your help

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[Serusers] ser.cfg
by julien bossart 21 Sep '05

21 Sep '05

Hi all, I use ser.cfg defaul, and uncomment some line to support mysql. After start ser, i use: /usr/sbin/serctl ps i see in the console that: stand-alone receiver @ 127.0.0.1:5060 <http://127.0.0.1:5060> so that i always got ICMP host unreachable when i use SIP client to send REGISTER. How can i add IP server for ser? Thank you

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RE: [Serusers] SER with asterisk as voicemail problem... please help!
by Quang Minh PHAN 21 Sep '05

21 Sep '05

Hi, It seems to be a problem in Asterisk configuration, not in SER's. Showing your asterisk's extensions.conf file may help. Regards, Minh -----Message d'origine----- De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de Ryan Pagquil Envoyé : mardi 20 septembre 2005 09:50 À : serusers(a)lists.iptel.org Objet : [Serusers] SER with asterisk as voicemail problem... please help! Hi, I configured SER to forward certain calls to asterisk for the voicemail functionality. But when I forward the SIP request to Asterisk my xlite softphone prompts "403", "Forbidden". This is my ser.cfg part where i forward the certain uri to the asterisk. if (uri=~"^sip:1[0-9]{3}@sip.philonline.com.*$") { strip(1); rewritehostport("202.84.24.47:5060"); t_relay(); break; }; Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] need help
by julien bossart 21 Sep '05

21 Sep '05

Hi all, I am new to this forum. Today, i download ser package, then i install in my fedora core 4. Everything is ok. Next, i start ser and use serctl add to add account to ser server. Then i use sjphone to connect to ser server. Sjphone try to send REGISTER to ser server. I use ethereal to get the package. I see that ser server responds ICMP host unreacheable all the time. So, REGISTER is failed. Can you give me some suggestion? Thank you for that.

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[Users] Compilation error on Mac OS
by Amos Nungu 21 Sep '05

21 Sep '05

I'm installing SEMS-0.9 on Mac OS (Darwin 8.2.0). upon running the make all command, I got an error as shown below ------------------------------------------------- # make all make[1]: Nothing to be done for `deps'. g++ -o sems AmApi.o AmAudio.o AmCmd.o AmConfig.o AmEventQueue.o AmIcmpWatcher.o AmMail.o AmPlugIn.o AmRequest.o AmRtpPacket.o AmRtpPacketTracer.o AmRtpScheduler.o AmRtpStream.o AmSdp.o AmServer.o AmSession.o AmSmtpClient.o AmThread.o AmUtils.o EmailTemplate.o SemsConfiguration.o SerClient.o SerDBQuery.o log.o sems.o -lm -ldl -lsocket -lnsl -lpthread /usr/bin/ld: can't locate file for: -lsocket collect2: ld returned 1 exit status make[1]: *** [sems] Error 1 make: [all] Error 2 (ignored) ------------------------------------------------------- Any idea on how to go about it will be great. Regards

2 1

[Users] SER + MediaProxy or rtpproxy
by Frank Kostin 21 Sep '05

21 Sep '05

Hi All, Using Onsip SER with Mediaproxy --no-fork I can see caller/called/relayed. This is OK for local calls only. With or without mediaproxy I have audio both directions. When I call to /from another network the phone rigning but no audio. Can I relay /forward by IP address using mediaproxy and eventually redirect media streams throught a 3rd party ? Thanks, Frank --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

2 1

[Serusers] Problem redirecting to voicemail
by Chris St Denis 20 Sep '05

20 Sep '05

I'm having a problem redirecting to voicemail. This may be an asterisk bug I'm not sure, can somebody confirm? Network layout GATEWAY - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h connected to a PRI line. (Additionally patched with http://bugs.digium.com/view.php?id=2687) PROXY - Ser version: ser 0.9.3 (i386/freebsd) FEATURE - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h handling voicemail. GATEWAY---PROXY---FEATURE | | UA For simplicity, hostnames and IPs replaced with the above names. USERNAME, DSTNUM and SRCNUM also used to replace the UA's username, the source number of the call, and the destination number of the call. The basic SIP dialog goes: Gateway invites proxy Gateway invites UA UA replies 180 Ringing. Transaction times out and drops to failure route PROXY invites FEATURE server (INVITE sip:*voicemail-busy-2002006@ FEATURE:5060 SIP/2.0.) PROXY cancels UA FEATURE replies 200 ok to PROXY PROXY replies 200 ok to GATEWAY FEATURE replies 200 ok to PROXY PROXY replies 200 ok to GATEWAY UA tells PROXY '487 Request Terminated.' FEATURE replies 200 ok to PROXY PROXY replies 200 ok to GATEWAY ... For some reason, the asterisk gateway doesn't seem to be ACKing the 200 ok. I don't see any new invite or reinvite going to the gateway which I think may be what is confusing it. Ser.cfg excerpt ## # User did not answer phone, or could not connect, or is on the phone and does not use call waiting. ## failure_route[3] { if(isflagset(10)) { if(t_check_status("486")) { if (!subst_user('/^/*voicemail-busy-/')){ log(1,"Err in subst_user\n"); } xlog("L_ERR", "Relaying to voicemail Busy\n"); } else { if (!subst_user('/^/*voicemail-noanswer-/')){ log(1,"Err in subst_user\n"); } xlog("L_ERR", "Relaying to voicemail No answer\n"); } rewritehostport("FEATURE:5060"); append_branch(); t_relay(); } } Ngrep from PROXY's point of view http://pastebin.ca/23469 Ngrep from GATEWAY's point of view http://pastebin.ca/23470 Ngrep from FEATURE's point of view http://pastebin.ca/23471

1 0

[Serusers] Can SER determine NAT type?
by Chris St Denis 20 Sep '05

20 Sep '05

I want to use STUN as a primary nat solution, with the nathelper or mediaproxy module doing rewrites as a secondary/backup solution. However, this will not work for symmetric NATs. I want to proxt the rtp data from these (and only these) kind of nat. Can SER determine that? My client adds a Warning: 399 spa "Restricted Cone NAT Dectected". Header to the register message (determined by stun), but that looks proprietary and not something I can expect to be the same from other clients so I'm not sure I want to header match on that.

1 0

[Serusers] isntalling SER on Red hat linux
by rajesh volluru 20 Sep '05

20 Sep '05

hi Can we install SER on Red Hat linux. If yes, then please send me the instructions. Also, I get Transaction unavailable exception when I try to communicate through Sip Communicator. Does installing SER solve that problem? Thanks Rajesh --------------------------------- Yahoo! India Matrimony: Find your partner now.

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Re: [Serusers] COnfiguring a PSTN number with SER
by Iqbal 20 Sep '05

20 Sep '05

in ur messages file, i have u traced to see where in ur config its getting stuck, i mean when it gos through with the INVITE when it does alias tabel lookup is it matching anything, also look in mysql log file to see what info it is selecting upon. what is the format of your entry in aliases, it will map your username to alias, wheras you want number mapped to username i think iqbal Ashutosh Kumar wrote: >Hi, > I assigned the DID to the user but serctl add alias, and the entry >got updated in ser.aliases table. > But when I call the number from pstn, the registered user does not get >the call. The nextone contacts the SER, but SER responsds to it with a "478 >unresolvable desintation". > Following is the ngrep excerpt. > >U PSTN_GW_IP:5060 -> SER_IP:5060 >INVITE sip:@SER_IP SIP/2.0. >Max-Forwards: 70. >Session-Expires: 3600;Refresher=uac. >Supported: timer. >. >v=0. >o=msw1 1234 1929 IN IP4 x.y.z.101. >s=sip call. >c=IN IP4 x.y.z.101. >t=0 0. >m=audio 27468 RTP/AVP 0 8 3 101. >a=rtpmap:0 PCMU/8000. >a=rtpmap:8 PCMA/8000. >a=rtpmap:3 GSM/8000. >a=rtpmap:101 telephone-event/8000. >a=fmtp:101 0-16. >a=silenceSupp:off - - - -. > >## >U SER_IP:5060 -> PSTN_GW_IP:5060 >SIP/2.0 100 trying -- your call is important to us. > > >############# >U SER_IP:5060 -> PSTN_GW_IP:5060 >SIP/2.0 478 Unresolvable destination (478/TM). > >Relevant code from ser.cfg is as follows: > >... >... >modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") >modparam("usrloc", "db_mode", 2) > > > if (uri==myself) { > > if (method=="ACK") { > route(7); > break; > } else if (method=="CANCEL") { > route(3); > break; > } else if (method=="INVITE") { > route(3); > break; > } else if (method=="MESSAGE"){ > route(4); > break; > } else if (method=="REGISTER") { > route(2); > break; > }; > > lookup("aliases"); > if (uri!=myself) { > route(5); > route(1); > break; > }; > > > if (!lookup("location")) { > sl_send_reply("404", "User Not Found"); > break; > }; > > >I don't know where I am wrong. > >Regards, >Ashutosh > > > > > >-----Original Message----- >From: Iqbal [mailto:iqbal@gigo.co.uk] >Sent: Tuesday, September 20, 2005 10:35 PM >To: Ashutosh Kumar >Cc: 'Andreas Granig'; serusers(a)lists.iptel.org >Subject: Re: [Serusers] COnfiguring a PSTN number with SER > >just alias that number to a voip username which is bound to that user, >so when 123456789 is dialed nextone sends it to ser, ser looks up in >aliases, sees the match and sends it to that user > >Iqbal > >Ashutosh Kumar wrote: > > > >>Hi Andreas, >> Thanks for the reply, I was thinking of another approach to do this, >>tell me if I am wrong. >> May I use the 'phone' field in subscriber table to run an external >>sql script to map the DID (phone) with a username, and then rewrite the >> >> >uri, > > >>and then forward the call to that user? >> >>Regards, >>Ashutosh >> >> >>-----Original Message----- >>From: Andreas Granig [mailto:andreas.granig@inode.info] >>Sent: Tuesday, September 20, 2005 6:22 PM >>To: Ashutosh Kumar >>Cc: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] COnfiguring a PSTN number with SER >> >>Ashutosh Kumar wrote: >> >> >> >> >>> I have a DID number to which I want to assign to a user in >>>SER, the call will come through a NExtone. How / what changes I have to >>>make in ser.cfg to arrange that whenever a call comes on that number on >>>nextone, that call should be forwarded to the sip-user bound to that >>> >>> >>> >>> >>number. >> >> >> >> >>>How can I use aliases in this scenario. >>> >>> >>> >>> >>Check the PSTN chapter of the Getting Started document at http://onsip.org >> >>Andy >> >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >>. >> >> >> >> >> > > > >. > > >

1 0

[Serusers] COnfiguring a PSTN number with SER
by Ashutosh Kumar 20 Sep '05

20 Sep '05

Hi, I have a DID number to which I want to assign to a user in SER, the call will come through a NExtone. How / what changes I have to make in ser.cfg to arrange that whenever a call comes on that number on nextone, that call should be forwarded to the sip-user bound to that number. How can I use aliases in this scenario. Please guide. Regards, Ashutosh Kumar

3 4

[Serusers] obsfuctaing ser.cfg
by Iqbal 20 Sep '05

20 Sep '05

is this possible other than just removing the spaces and all. Iqbal

1 0

[Serusers] accounting and radius/sql
by harry gaillac 20 Sep '05

20 Sep '05

Hello ser users, I set accounting with freeradius 1.0.2 and mysql for testing billing (CDRTool). I configured radius to work with mysql. I load acc module with radius support it's ok. I have a problem with accouting and radacct table. In fact "AcctStopTime" and related to time fields are not updated . I would need experienced users with freeradius server and ser to debug accounting . Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] asterisk vm and ser
by Sebastian Kühner 20 Sep '05

20 Sep '05

Hello, I'm using ser with asterisk as voicemail server. I send a notify message to a user that has a new message. If the user deletes his messages, asterisk (and sipsak) sends the notify message correctly and the VM-Indicator turns off... but after a while, my SIPURA turns on the vm indication without receiving a new notify. Do I have to send this notify message periodically? Thanks! Sebastian

1 0

[Serusers] Possible bug
by Rubén Mazariegos del Rey 20 Sep '05

20 Sep '05

Hi! I am using version 0.9.3 of SER and I have observed the following behaviour: When a user sends a SIP message to another user through SER, and the body of the SIP message ends with the byte "0x00", this byte is removed by SER from the message before forwarding it to the destination. Is this a known behaviour? I suppose it is a bug, isn't it? Is there a newer version of SER fixing this bug? Thanks in advance, Rubén

1 0

[Serusers] sdp and rtp errors
by Rick Thompson 20 Sep '05

20 Sep '05

Hi All I hope someone can help me with an issue I"ve had since installing ser. For some reason, whenever I call fix_nated_sdp or force_rtp_proxy I get the following errors Sep 19 10:31:06 localhost ser[3921]: Mon Sep 19 10:31:06 2005 18 a.b.c.d 62141 ****** Fixing NAT on reply with rtp Sep 19 10:31:06 localhost ser[3921]: ERROR: extract_body: message body has length zero Sep 19 10:31:06 localhost ser[3921]: ERROR: fix_nated_sdp: cannot extract body from msg! Sep 19 10:31:06 localhost ser[3921]: ERROR: extract_body: message body has length zero Sep 19 10:31:06 localhost ser[3921]: ERROR: force_rtp_proxy2: can't extract body from the message Sep 19 10:31:06 localhost ser[3921]: ERROR: on_reply processing failed Here is the part of the script that is generating the error onreply_route[1] { xlog("L_ALERT", "%Tf %mf ****** On Reply From: <%si> <%tu> <%fu> <%rm> <%rs> <%rr>\n"); if(status=~"[12][0-9][0-9]" && nat_uac_test("19")) { fix_nated_contact(); force_rport(); }; if(status=~"[12][0-9][0-9]" && nat_uac_test("19") && src_port>=10000) { xlog("L_ALERT", "%Tf %mf %si %sp ****** Fixing NAT on reply with rtp\n"); fix_nated_sdp("3"); force_rtp_proxy(); } else if(status=~"[12][0-9][0-9]" && nat_uac_test("19")) { xlog("L_ALERT", "%Tf %mf %si %sp ****** Fixing NAT on reply with sdp\n"); fix_nated_sdp("3"); }; } Any help would be appreciated. Thanks Rick

2 3

[Serusers] UltraEdit
by Rick Thompson 20 Sep '05

20 Sep '05

Hi all I was wondering if anybody has written a ser script wordlist.txt for ultraEdit and would mind sharing it. Just so I don't have to re-invent the wheel. Thanks Rick

1 0

[Serusers] Solaris8/SPARC compilation
by Ryan Pagquil 20 Sep '05

20 Sep '05

Hi guys, I'm having a problem with SER-0.9.3 installation in Solaris8/SPARC i've got these errors: bash-2.03# make all Makefile.rules:84: action.d: No such file or directory Makefile.rules:84: crc.d: No such file or directory Makefile.rules:84: daemonize.d: No such file or directory Makefile.rules:84: data_lump.d: No such file or directory Makefile.rules:84: data_lump_rpl.d: No such file or directory Makefile.rules:84: dprint.d: No such file or directory Makefile.rules:84: dset.d: No such file or directory Makefile.rules:84: error.d: No such file or directory Makefile.rules:84: fifo_server.d: No such file or directory Makefile.rules:84: flags.d: No such file or directory Makefile.rules:84: forward.d: No such file or directory Makefile.rules:84: hash_func.d: No such file or directory Makefile.rules:84: ip_addr.d: No such file or directory Makefile.rules:84: main.d: No such file or directory Makefile.rules:84: md5.d: No such file or directory Makefile.rules:84: md5utils.d: No such file or directory Makefile.rules:84: modparam.d: No such file or directory Makefile.rules:84: msg_translator.d: No such file or directory Makefile.rules:84: pass_fd.d: No such file or directory Makefile.rules:84: proxy.d: No such file or directory Makefile.rules:84: qvalue.d: No such file or directory Makefile.rules:84: re.d: No such file or directory Makefile.rules:84: receive.d: No such file or directory Makefile.rules:84: resolve.d: No such file or directory Makefile.rules:84: route.d: No such file or directory Makefile.rules:84: route_struct.d: No such file or directory Makefile.rules:84: script_cb.d: No such file or directory Makefile.rules:84: socket_info.d: No such file or directory Makefile.rules:84: sr_module.d: No such file or directory Makefile.rules:84: stats.d: No such file or directory Makefile.rules:84: tcp_main.d: No such file or directory Makefile.rules:84: tcp_read.d: No such file or directory Makefile.rules:84: timer.d: No such file or directory Makefile.rules:84: tsend.d: No such file or directory Makefile.rules:84: udp_server.d: No such file or directory Makefile.rules:84: unixsock_server.d: No such file or directory Makefile.rules:84: usr_avp.d: No such file or directory Makefile.rules:84: mem/f_malloc.d: No such file or directory Makefile.rules:84: mem/mem.d: No such file or directory Makefile.rules:84: mem/memtest.d: No such file or directory Makefile.rules:84: mem/q_malloc.d: No such file or directory Makefile.rules:84: mem/shm_mem.d: No such file or directory Makefile.rules:84: mem/vq_malloc.d: No such file or directory Makefile.rules:84: parser/hf.d: No such file or directory Makefile.rules:84: parser/msg_parser.d: No such file or directory Makefile.rules:84: parser/parse_allow.d: No such file or directory Makefile.rules:84: parser/parse_content.d: No such file or directory Makefile.rules:84: parser/parse_cseq.d: No such file or directory Makefile.rules:84: parser/parse_disposition.d: No such file or directory Makefile.rules:84: parser/parse_diversion.d: No such file or directory Makefile.rules:84: parser/parse_event.d: No such file or directory Makefile.rules:84: parser/parse_expires.d: No such file or directory Makefile.rules:84: parser/parse_fline.d: No such file or directory Makefile.rules:84: parser/parse_from.d: No such file or directory Makefile.rules:84: parser/parse_hname2.d: No such file or directory Makefile.rules:84: parser/parse_hostport.d: No such file or directory Makefile.rules:84: parser/parse_methods.d: No such file or directory Makefile.rules:84: parser/parse_nameaddr.d: No such file or directory Makefile.rules:84: parser/parse_param.d: No such file or directory Makefile.rules:84: parser/parse_rpid.d: No such file or directory Makefile.rules:84: parser/parse_rr.d: No such file or directory Makefile.rules:84: parser/parse_sipifmatch.d: No such file or directory Makefile.rules:84: parser/parse_to.d: No such file or directory Makefile.rules:84: parser/parse_uri.d: No such file or directory Makefile.rules:84: parser/parse_via.d: No such file or directory Makefile.rules:84: parser/parser_f.d: No such file or directory Makefile.rules:84: parser/digest/digest.d: No such file or directory Makefile.rules:84: parser/digest/digest_parser.d: No such file or directory Makefile.rules:84: parser/digest/param_parser.d: No such file or directory Makefile.rules:84: parser/contact/contact.d: No such file or directory Makefile.rules:84: parser/contact/parse_contact.d: No such file or directory Makefile.rules:84: db/db.d: No such file or directory Makefile.rules:84: db/db_fifo.d: No such file or directory Makefile.rules:84: lex.yy.d: No such file or directory Makefile.rules:84: cfg.tab.d: No such file or directory yacc -d -b cfg cfg.y conflicts: 1 shift/reduce make: *** [cfg.tab.c] Broken Pipe what could be the possible cause of these errors? How will i deal with it? Another question, does the ser-0.9.3 package for solaris support mysql, radius accounting and authentication? Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

4 4

[Serusers] busy
by Sebastian Kühner 20 Sep '05

20 Sep '05

Hello, Does anybody know how I can detect if a phone is busy. I make accounting with ser and sometimes the BYE messages don't reach my server, so I have to test if a phone is busy without using INVITE messages... Thanks! Sebastian

1 0

Re: [Serusers] LDAP authentication (without Radius)
by Arek Bekiersz 20 Sep '05

20 Sep '05

Yes, there is a nice module, it was created by Marcel Baur, ETH Zurich. You can find it under: http://www.ethworld.ethz.ch/technologies/sipeth/ser_modules/ldaps It should be helpful to take a look at the code and just get idea how it works. Greetings, Arek ----- Original Message ----- From: "Samuel Osorio Calvo" <samuel.osorio(a)nl.thalesgroup.com> To: <clona(a)camaro.no>; <sip(a)perceval.net> Cc: <serusers(a)lists.iptel.org> Sent: Monday, September 19, 2005 10:32 AM Subject: Re: [Serusers] LDAP authentication (without Radius) > If I don't recall wrong, there was an LDAP module for SER developed in > some university/research center from Switzerland, and I am pretty sure you > could dopwnload it....just try to google, because I haven't got internet > access right now... > > Samuel. >

2 1

[Serusers] LCR Module on MaC OS (Reverse IP)
by Amos Nungu 20 Sep '05

20 Sep '05

> > I managed to get LCR running on both Debian-Sarge and MacOS-Darwin, > thanks. > I'm running ser 0.9.3 from source on both machines. > However there exists some di-similarities between in operations between > the two installations dispite of using the same installation source and same > ser.cfg logic. > 1.I convert my IP to decimal using www.dnsstuff.com<http://www.dnsstuff.com/>( > 192.168.0.101 <http://192.168.0.101>) to get 1694542016. > 2.Add that value into the lcr table. > 3.Making a PSTN call, using xlog to capture %ru > > Result: > Debian configuration will show abc(a)192.168.0.101 while MacOS will show > abc(a)101.0.168.192 > > I can get the desired effect if I convert the IP address in reverse order > for MacOS (192.168.0.101 <http://192.168.0.101>) to (3232235621). But, I > think this is not how it was supposed to be. > > Any idea? > > Regards. >

1 0

[Serusers] Acc table fields
by Federico Giannici 20 Sep '05

20 Sep '05

I cannot understad the meaning and the purpose of the two fields "caller_deleted" and "callee_deleted" in the "acc" SQL table. I found no reference to them in the entire SER sources (0.9.4)... Anybody knows what is their purpose? As we are talking about accounting... I'd like if it was possible to register the source IP of the request messages and the session media info (if present): used codec, media type (audio, video...), etc... Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

2 2

RE: [Serusers] Handoff to PSTN with a set prefix
by Colin Jordan 20 Sep '05

20 Sep '05

I need to send a prefix of 666# followed by full intl number without leading 0's. eg. 666#442071016211 The PSTN are connecting me to a Cisco SIP Proxy server, which uses Tech-prefixes eg 123# to separate and authenticate the calls into different domains. Can you please help me? -----Original Message----- From: Andreas Granig [mailto:andreas.granig@inode.info] Sent: Tuesday, September 20, 2005 10:50 AM To: Colin Jordan Cc: serusers(a)lists.iptel.org; serusers-request(a)lists.iptel.org Subject: Re: [Serusers] Handoff to PSTN with a set prefix Colin Jordan wrote: > # attemt handoff to PSTN > if (uri=~"^sip:9[0-9] *(a)gaco.co.za") { > log ("Forwarding to PSTN\n"); > t_relay_to_udp("217.15.11.87","5060"); > t_relay_to_tcp("217.15.11.87","5060"); > break; > }; Hm? Both UDP and TCP? Try: if (uri=~"^sip:9[0-9] *(a)gaco.co.za") { log ("Forwarding to PSTN\n"); prefix("666#"); t_relay_to_udp("217.15.11.87","5060"); break; }; I'm not sure about the "#" and if it's parsed correctly or if it's treated as start of a comment, haven't ever tried that. Andy

1 0

[Serusers] Handoff to PSTN with a set prefix
by Colin Jordan 20 Sep '05

20 Sep '05

I need to set a fixed prefix of, 666#, to enable me to break out to the PSTN that I am sending my traffic. Will you please help me with the needed commands and where to place it? Please find below the ser.cfg file: ________________________________ # # $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) # new insterted statement listen=66.8.91.200 listen=127.0.0.1 # hostname matching an allias will satify the condition uri-myself alias=gaco.co.za alias=66.8.91.200 /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) # this was commented port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # database location - NEW #modparam("auth_db, "db_uri, "sql://ser:>heslo>@localhost/ser") # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("gaco.co.za", "subscriber")) { www_challenge("gaco.co.za", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; # attemt handoff to PSTN if (uri=~"^sip:9[0-9] *(a)gaco.co.za") { log ("Forwarding to PSTN\n"); t_relay_to_udp("217.15.11.87","5060"); t_relay_to_tcp("217.15.11.87","5060"); break; }; } ________________________________ DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

2 2

[Serusers] Enabling Radius communication between SER and Alepo RBS
by Thomas Tomter 20 Sep '05

20 Sep '05

Hello, We are working on establishing Radius-communication between SER and the billing system Alepo RBS version 4.2, but we are experiencing some trouble with this. We think the problem might be caused by the way we have defined SER as a network component in the Alepo RBS configuration file called 'NasCfgDbs' and maybe how the SER dictionary is built up. Does anyone have any experience with integration of SER with Alepo RBS? We would appreciate any kind of help or tip. Thanks in advance. Best regards, Thomas

1 0

[Serusers] SER with asterisk as voicemail problem... please help!
by Ryan Pagquil 20 Sep '05

20 Sep '05

Hi, I configured SER to forward certain calls to asterisk for the voicemail functionality. But when I forward the SIP request to Asterisk my xlite softphone prompts "403", "Forbidden". This is my ser.cfg part where i forward the certain uri to the asterisk. if (uri=~"^sip:1[0-9]{3}@sip.philonline.com.*$") { strip(1); rewritehostport("202.84.24.47:5060"); t_relay(); break; }; Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

1 0

[Serusers] 408 to Caller UA when CANCEL to Callee
by Tulika Pradhan 20 Sep '05

20 Sep '05

hi, i am facing the following situation. UA1 calls a user(UA2) who does not answer. the control comes to failure_route where i try another UA (UA3). but as UA3 rings, SER sends 408 Request timeout to UA1 and call gets disconnected. this is the SIP message flow. UA1 SER UA2 UA3 INVITE----------------> INVITE--------------> <----------------TRYING <----------------RINGING <------------------RINGING CANCEL--------------> <---------------------408 INVITE----------------------------------------> <---------------------487 ACK-------------------> <-----------------------OK <-------------------------------------------TRYING <--------------------------------------------RINGING (but UA already has got the busy tone) and does not hear this ringing. if 408 was not sent to UA1, then the call could have been established. what is going wrong, regards, tulika

4 10

[Serusers] SER+MEDIAPROXY+ B2BUA
by Amit.Chawre＠relianceinfo.com 20 Sep '05

20 Sep '05

Hello I want to test a scenario with SER, MEDIAPROXY and Vovida B2BUA, but I'm having some problems. If someone has tested this before, I would like to ask you the config files to help me. When NATedUA1 calls NATedUA2, invite reaches NATedUA2. NATedUA2 sends responses back to SER (100 trying 180 ringing) but SER doesn’t forward it further to NATedUA1.I'm having problems when NatedUA2 send '100 Trying' and '180 ringing' to SER, it forwards it to B2BUA, but then there is no flow of messages. NATedUA2 ^ | 4.INVITE | 1.INVITE 2.INVITE NATedUA1 ----> SER+MEDIAPROXY ------> B2BUA <----- 3. INVITE ser.cfg is: # Example ser.cfg for mediaproxy functionality check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=72.21.40.107 port=5060 children=4 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/textops.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/domain.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" modparam("usrloc", "db_mode", 0) #modparam("usrloc", "db_url","mysql://ser:heslo@localhost/ser") modparam("mediaproxy", "natping_interval", 60) modparam("registrar", "nat_flag", 2) route{ if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483", "Too many hops"); }; break; }; if (msg:len >= max_len) { if (method!="ACK") { sl_send_reply("513", "Message too big"); }; break; }; if (method=="REGISTER") { if (is_from_local()) { # Mark as NAT'ed if (client_nat_test("3")) { setflag(2); force_rport(); fix_contact(); }; if (!radius_www_authorize("72.21.40.107")) { www_challenge("72.21.40.107", "0"); break; } else if (!check_to()) { sl_send_reply("403", "Username!=To not allowed"); break; }; if (!save("location")) { sl_reply_error(); }; } else { sl_send_reply("403", "This domain is not served here"); }; break; }; if (method=="INVITE") { if (!(is_from_local() || is_uri_host_local())) { sl_send_reply("403", "Relaying is forbidden"); break; }; t_on_failure("1"); } else if (method == "BYE" || method == "CANCEL") { end_media_session(); }; # Force subsequent messages to pass trough this proxy if (method == "INVITE") { record_route(); }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed force_rport(); fix_contact(); }; #if (method=="INVITE") { t_on_reply("1"); #}; if (is_uri_host_local()) { if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; if(!(src_port==5065 && src_ip==72.21.40.107)){ if (method=="INVITE" || method=="BYE") { log("amit: Forwarding to B2bua\n"); forward(72.21.40.107,5065); break; }; }; if (loose_route()) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here t_relay(); break; }; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } failure_route[1] { end_media_session(); } onreply_route[1] { #if (status=~"(183)|(2[0-9][0-9])") { if (status=~"[1-2][0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; } Thanks for your help amit

1 0

[Serusers] SER + MediaProxy or rtpproxy
by Frank Kostin 20 Sep '05

20 Sep '05

Hi All, Using Onsip SER with Mediaproxy --no-fork I can see caller/called/relayed. This is OK for local calls only. With or without mediaproxy I have audio both directions. When I call to /from another network the phone rigning but no audio. Can I relay /forward by IP address using mediaproxy and eventually redirect media streams throught a 3rd party ? Thanks, Frank --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

1 0

[Users] request service to openser
by Loide Mara 20 Sep '05

20 Sep '05

How the UA requests a service for proxy (openser)? Is possible a direct communication between the UA and proxy? For example... How the UA requests the blacklist's service? Somebody has some suggestion? -------------------------------------------------- Loide Mara Verdes de Jesus 6º período - Ciência da Computação / UFAM Laboratório VoIP - UFAM

2 1

[Users] Bug on pseudevariables in failure_route
by Franz Edler 20 Sep '05

20 Sep '05

Hi, I am actually testing with openser 0.10.0-dev12-tls and detected the following problem: When I use pseudo-variables $rs and $rr in a failure_route I get <null> instead of reply status and reply reason in case of a failure response. I think this is a bug. Regards Franz

2 1

[Serusers] Instant Messaging cannot work with Windows Messenger 5.0
by kokmeng＠moij.biz 20 Sep '05

20 Sep '05

Hi, I have two PCs running Windows Messenger 5.0 and SER as registrar. I am able to get Voice, Video, App Sharing, Whiteboard, Presence (peer to peer only) working. However, I cannot get Instant Messaging to work. When I look at my Ethereal capture, I see that with 5.0, INVITE is used to start an IM session. The other client responds with TRYING, then OK. However, it also sends a BYE immediately after sending the OK. Subsequently when the originating client sends the MESSAGE, the other client responds with an Error because "Call Leg does not exist". Has anyone encountered this before? I've seen this problem being mentioned in Google searches, but no answer so far. Regards, KokMeng Loh

3 3

[Serusers] DNS SRV for load balancing
by Peter 19 Sep '05

19 Sep '05

We are trying to use DNS SRV records to load balance outbound transactions across 2 or more servers. Something like this: UA => SER => (Some processing logic) => Back to SER after a 300 Redirect which contains a Contact phone number and IP/DNS at which to contact the contact => DNS SRV lookup for the contact IP and send it to one of the two servers returned by the DNS lookup. So this last leg is where we have to load balance multiple contact IP servers and we were trying to use DNS SRV for that. We haven't seen any success in this, it does load balance when both are up. But as soon as only one server is up, SER keeps trying to reach the first one which is down and never tries to reach the second one. Ultimately the transaction times out. Has anybody had success in implementing this kind of scenario for load balancing with DNS SRV? Is anything else besides using DNS SRV an option? Thanks in advance!! __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

1 0

RE: [Serusers] SER Load balancing
by Rose, John 19 Sep '05

19 Sep '05

Marcello, Here is one method that I have been looking at - has anyone tried something similar? # --- Rewrite 50/50 -------------------------------- if (uri=~"test") { if (is_gflag("1")) { reset_gflag("1"); rewriteuri("sip:test@10.1.1.1"); } else { set_gflag("1"); rewriteuri("sip:vsd@10.1.1.2"); }; t_on_failure("1"); } # -------------------------------------------------- Thanks, John Rose -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Marcello Lupo Sent: Saturday, September 17, 2005 7:38 AM To: serusers(a)lists.iptel.org Subject: [Serusers] SER Load balancing Hi to all, did you have suggestions how to accomplish to load balancing load of SER between 2 machine running both SER and DB to check routing entries?? I was thinking to use LVM but i don't know how to let the same call messages to go to the same machine. There should be something line an LVM module able to check the CallID field and keep track of it. May be some other ways?? Using DNS records or other? Another question is if there is a searchable archive of this Mailing list. Thanks, Bye, Marcello _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] MySQL socket file
by John, Paul 19 Sep '05

19 Sep '05

How can SER be made to look for the MySQL socket file in /var/run/mysql/mysql.sock instead of /var/lib/mysql/mysql.sock? Paul

1 0

[Serusers] 5060
by Sebastian Kühner 19 Sep '05

19 Sep '05

Hi, I notice that ser ignores all "REGISTER"s that come from a special server with the port 5060. All other ports are accepted. What could be the problem? Sebastian

1 0

[Serusers] MySQL 4.1.10a
by John, Paul 19 Sep '05

19 Sep '05

I'm having trouble with MySQL starting when the socket file is located in /var/run/mysql/mysql.sock instead of the default location /var/lib/mysql/mysql.sock. I have changed the owner and permission of the new directory, but I still get the following error in the log. I don't have another mysqld server running on this machine. 050919 12:24:11 [ERROR] Can't start server : Bind on unix socket: Permission denied 050919 12:24:11 [ERROR] Do you already have another mysqld server running on socket: /var/run/mysql/mysql.sock ? 050919 12:24:11 [ERROR] Aborting Paul

1 0

[Serusers] ForwardNoAnswer
by Ramin Nikaeen 19 Sep '05

19 Sep '05

Dear Friends, Has anyone attempted implementing the "Forward On No Answer" service with SER who can give me some pointers. I greatly appreciate your help Thanks Ramin rnikaeen(a)goldline.net

2 1

[Serusers] avpops and register
by Rodrigue ESTEVE 19 Sep '05

19 Sep '05

Hi, I'm trying to do the following things using the avpops: When I receive a Register Message from uri 666@.., I'd like to register the uri 1000@... and 2000@... at the same time. I'd like to do this also for uri 777, to register 778 and 779 when 777 triggers a Register Message. So I decided to use the SQL Table below to achieve this: Initnumber: Varchar(64) Attribute: Varchar(64) Type: INT (default to 0) Associated: Varchar(64) Initnumber attribute type associated 666 register 0 1000 666 register 0 2000 777 register 0 778 777 register 0 779 In Ser.cfg, I put this: modparam("avpops","avp_url","mysql://ser:heslo@IP/telephone") modparam("avpops", "avp_table", "associate") modparam("avpops","username_column","initnumber") modparam("avpops","attribute_column","attribute") modparam("avpops","type_column","type") modparam("avpops","value_column","associated") if (method=="REGISTER") { # Authentication Challenge Challenge #Test of AVPs log(1, "LOG: AVP REGISTRATION"); avp_db_load("$from ","s:register"); #to display avp in memory avp_print(); #fork registration using new avp value to register avp_pushto("$from","s:register/g"); save("location"); break; }; What I expect this to do is to get the values "associated" with "Initnumber", using avp_db_load and to fork the registration process using avp_pushto to register the numbers in the associated column. I use avp_print to see the avp loaded in memory, but I see nothing in the syslog. Moreover nothing happens, ie only the uri which initiates the Register message is registered on SER :'( Can somebody help me please, Rodrigue

1 0

R: [Serusers] fields To/From
by Piccari Gianluca 19 Sep '05

19 Sep '05

ok ): i'll performed that ugly thing named "changing the source code". Good for me (; -----Messaggio originale----- Da: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]Per conto di Piccari Gianluca Inviato: venerdì 16 settembre 2005 13.10 A: serusers(a)lists.iptel.org Oggetto: [Serusers] fields To/From Hi all, i have a big problem... i need to parser information in Header field From() and eventually add or delete part of info in Header field TO. For istance: if i(n From field there is '55') add ('55' in To field). I don't have familiar to SER (i have started to study it last week (: ) and i likr e to know if it is possible to do it using SER's method without change the source code .... thanks for your help _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] cisco dial-peer
by Nhadie 19 Sep '05

19 Sep '05

Hi List, Don't know if this is the right list for this but I hope someone can help me: I have a SIP server when it starts on a certain string e.g. 123 it will forward it to gateway else it will forward to gateway 2. Gateway 1 will be a cisco 1760 with and FXO port, so whe I dial 12345678 it should forwar it to the Cisco 1760, using ngep I can see that it's sending the invite to the CISCO 1760 but I don't know what typre of dial-pper to define in order for th 1760 to respond, also what typre of dial-peer would forward it to the FXO port and eventually call PSTN. TIA Regards, nhadie

1 0

[Serusers] Error when make mysql in ser-0.9.3
by Alberto 19 Sep '05

19 Sep '05

Have you resolve your problem?

1 0

Re: [Serusers] Speeddial
by Aisling O'Driscoll 19 Sep '05

19 Sep '05

Thanks Ramona. > >---- Original Message ---- >From: ramona(a)voice-system.ro >To: ashling.odriscoll(a)cit.ie >Subject: Re: [Serusers] Speeddial >Date: Thu, 15 Sep 2005 20:10:56 +0200 > >>Aisling wrote: >> >>> Hello. >>> >>> I would really appreciate if someone could show me an example of >how a >>> speed-dialing entry should be added to the ser mysql database? >>> >>> E.g. If user 2092 dials 3 it rings user 3500. >>> >>> I tried : insert into usr_preferences (username, domain, >sd_username, >>> sd_domain, new_uri) values (2092, x.x.x.x, 3500, x.x.x.x, >3) >>> >>Hello Aisling, >> >>the entry should be (2092, x.x.x.x, 3, x.x.x.x, >sip:3500@x.x.x.x). >> >>Regards, >>Ramona >> >>-------------------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.

4 4

[Serusers] PSTN Gateway and caller-id
by Luca Corti 19 Sep '05

19 Sep '05

Hello, I've setup a ser proxy/registrar which forwards PSTN calls to an AS5350. The AS5350 also redirects PSTN calls to the ser box, which associates incoming called numbers with sip accounts through associations in the SQL DB. My problem is that the AS doesn't know about these associations when forwarding calls from sip to PSTN. I'd like to set the callerid on the outgoing calls to be the PSTN number associated with the sip account for incoming calls. thanks -- Luca Corti PGP Key ID 1F38C091

3 2

[Serusers] serweb. DB Error: connect failed
by Alberto 19 Sep '05

19 Sep '05

Hello, When I try to connect with admin user or other user account I get follow error: DB Error: connect failed Bad username or password I can connect with MySQL Administrator (windows) to server (freeBSD). The server is a freeBSD 5.4 + MySQL 4.1.14 + apache 2.0.54 + PHP 4.4.0 + ser-0.9.3 + serweb-0.9.3 For example, when I try to connect with administrator account I have tried to enter: username: admin(a)192.168.0.2 or admin or admin@sip or admin(a)teching.net password: heslo I do not understand what she is happening. In the 'subscriber' table: +----------+-----------+-------------+ | username | password | domain | +----------+-----------+-------------+ | admin | heslo | sip | | 1000 | 123454321 | teching.net | | 2000 | 12345 | teching.net | | 3000 | 12345 | teching.net | | 4000 | 12345 | teching.net | | 5000 | 12345 | teching.net | | 4000 | 12345 | 192.168.0.2 | +----------+-----------+-------------+ 7 rows in set (0.00 sec) and 'domain' table: +-------------+---------------------+ | domain | last_modified | +-------------+---------------------+ | teching.net | 2005-09-16 11:01:21 | | 192.168.0.2 | 2005-09-16 11:01:21 | | sip | 2005-09-19 13:55:21 | +-------------+---------------------+ 3 rows in set (0.00 sec) thanks, -- Alberto

1 0

[Serusers] send receive sms via http
by saiful bahri 19 Sep '05

19 Sep '05

hi all, is there any plan or work carry out to implement send/receive sms via http interface instead only via gsm modem? regards, saifulbahri.

1 0

Re: [Serusers] SER Load balancing
by Samuel Osorio Calvo 19 Sep '05

19 Sep '05

www.google.com might help you....;) site:iptel.org LVM "load balancing" Samuel. Unclassified. >>> "Marcello Lupo" <lupo(a)itspecialist.it> 09/17/05 03:37PM >>> Hi to all, did you have suggestions how to accomplish to load balancing load of SER between 2 machine running both SER and DB to check routing entries?? I was thinking to use LVM but i don't know how to let the same call messages to go to the same machine. There should be something line an LVM module able to check the CallID field and keep track of it. May be some other ways?? Using DNS records or other? Another question is if there is a searchable archive of this Mailing list. Thanks, Bye, Marcello _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: [Serusers] LDAP authentication (without Radius)
by Samuel Osorio Calvo 19 Sep '05

19 Sep '05

If I don't recall wrong, there was an LDAP module for SER developed in some university/research center from Switzerland, and I am pretty sure you could dopwnload it....just try to google, because I haven't got internet access right now... Samuel. Unclassified. >>> "Arek Bekiersz" <sip(a)perceval.net> 09/19/05 10:17AM >>> Hi Atle, all, It is not only my decision, as I did it as part of my normal job. I will ask management, let you know a.s.a.p. Arek, ----- Original Message ----- From: "Atle Samuelsen" <clona(a)camaro.no> To: "Arek Bekiersz" <sip(a)perceval.net> Cc: <serusers(a)lists.iptel.org> Sent: Monday, September 19, 2005 10:03 AM Subject: Re: [Serusers] LDAP authentication (without Radius) > Hi Arek, > > Why dont you give the LDAP module back to the comunity, and get it into > the exprimental branch. > > -Atle > > * Arek Bekiersz <sip(a)perceval.net> [050919 09:51]: >> Hello, >> >> >> I went through the same problem. Used Radius + LDAP for a while. >> >> Then only option for me was to write my own SER module, to perform digest >> authentication, lookup, group membership checking and so on, in LDAP. It >> is >> not so complicated though, can be really done in few days (ser 0.8.12) >> >> Works fine since then. Now I'm building more and more solutions for SER >> (including lcr and sdp manipulation, according to user profiles) based on >> LDAP. I use MySQL only for ser location register - that's all. >> >> >> Arek, >> >> >> >> ----- Original Message ----- >> From: "Felipe Louback" <louback(a)gmail.com> >> To: <serusers(a)lists.iptel.org> >> Sent: Thursday, September 15, 2005 7:59 PM >> Subject: [Serusers] LDAP authentication (without Radius) >> >> >> >I'm working in a VoIP solution here in my university and we have a >> >wide LDAP base and I am thinking about using it for >> > >> >I found on the internet Ldap/Ldaps modules for SER >> >(http://www.ethworld.ethz.ch/technologies/sipeth/ser_modules/ldaps), >> >but I am afraid that I could only use them for lookups and not direct >> >authentication. >> > >> >I guess I could use radius for authentication but I dont know if >> >installing a radius server "just" for this would be worth the effort. >> > >> >Does anyone know if there is a way to authenticate users in a LDAP >> >server without using Radius? I mean, could I use SER directly with >> >LDAP without using RADIUS? >> > >> >Regards, >> > >> >Felipe _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: [Serusers] Handoff to PSTN
by Iqbal 19 Sep '05

19 Sep '05

Hi I need to see the ser.cfg, or to make life easier use one from onsip.org, which is a good little get started guide, and will connect to pstn, u just pretty much cut and paste the config make a few changes and away you go iqbal Colin Jordan wrote: >I do not too much about this, can please let me know where and in what part >of the ser.cfg it will be? > > >-----Original Message----- >From: Iqbal [mailto:iqbal@gigo.co.uk] >Sent: Friday, September 16, 2005 12:07 PM >To: cjordan(a)econetwireless.com; olivier.taylor(a)gmail.com >Cc: serusers(a)lists.iptel.org >Subject: RE: [Serusers] Handoff to PSTN > > >where u route your INVITE > >Iqbal > >On 9/16/2005, "Colin Jordan" <cjordan(a)econetwireless.com> wrote: > > > >>Where should I insert it in the code below? >> >> >> >># attempt handoff to PSTN >> >>if (uri=~“^sip:9[0-9]*@mydomain.com”) { >> >>log(“Forwarding to PSTN\n”); >> >>t_relay_to( “192.168.0.2”, “5060”); >> >>break; >> >>}; >> >> >> >> >> >> _____ >> >>From: Olivier Taylor [mailto:olivier.taylor@gmail.com] >>Sent: Thursday, September 15, 2005 4:00 PM >>To: 'Colin Jordan' >>Cc: serusers(a)lists.iptel.org >>Subject: RE : [Serusers] Handoff to PSTN >> >> >> >>prefix("666") >> >>-----Message d'origine----- >>De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la >>part de Colin Jordan >>Envoyé : jeudi 15 septembre 2005 15:19 >>À : 'Olivier Taylor' >>Cc : serusers(a)lists.iptel.org >>Objet : RE: [Serusers] Handoff to PSTN >> >>The PSTN has given me a prefix of 666# to dial before the number, where can >>I insert this so that it will allow me to make a call? >> >> >> >> >> _____ >> >> >>From: Olivier Taylor [mailto:olivier.taylor@gmail.com] >>Sent: Thursday, September 15, 2005 2:58 PM >>To: 'Colin Jordan' >>Cc: serusers(a)lists.iptel.org >>Subject: RE : [Serusers] Handoff to PSTN >> >> >> >>module tm >> >>-----Message d'origine----- >>De : Colin Jordan [mailto:cjordan@econetwireless.com] >>Envoyé : jeudi 15 septembre 2005 14:55 >>À : 'Olivier Taylor' >>Cc : serusers(a)lists.iptel.org >>Objet : RE: [Serusers] Handoff to PSTN >> >>t_relay_to( “192.168.0.2”, “5060”); >> >> >> >> >> >> >> _____ >> >> >>From: Olivier Taylor [mailto:olivier.taylor@gmail.com] >>Sent: Thursday, September 15, 2005 2:50 PM >>To: 'Colin Jordan' >>Cc: serusers(a)lists.iptel.org >>Subject: RE : [Serusers] Handoff to PSTN >> >> >> >>what's on the line 143??? >> >>-----Message d'origine----- >>De : Colin Jordan [mailto:cjordan@econetwireless.com] >>Envoyé : jeudi 15 septembre 2005 14:46 >>À : 'Olivier Taylor' >>Cc : serusers(a)lists.iptel.org >>Objet : RE: [Serusers] Handoff to PSTN >> >>The line 143 points to the script below. What modules will it be looking >>for? >> >> >> >> >> _____ >> >> >>From: Olivier Taylor [mailto:olivier.taylor@gmail.com] >>Sent: Thursday, September 15, 2005 2:33 PM >>To: 'Colin Jordan' >>Cc: serusers(a)lists.iptel.org >>Subject: RE : [Serusers] Handoff to PSTN >> >> >> >>please reply to all >> >> >> >>u have a module not loaded, the one involved on line 143 >> >> >> >> >> >>Olivier >> >> >> >>-----Message d'origine----- >>De : Colin Jordan [mailto:cjordan@econetwireless.com] >>Envoyé : jeudi 15 septembre 2005 14:28 >>À : 'Olivier Taylor' >>Objet : RE: [Serusers] Handoff to PSTN >> >>It gives me the following message: >> >>_______________________________________________________ >> >> >> >>Parse error (143,40-41): unknown command missing loadmodule? >> >>_______________________________________________________ >> >> >> _____ >> >> >>From: Olivier Taylor [mailto:olivier.taylor@gmail.com] >>Sent: Thursday, September 15, 2005 1:21 PM >>To: 'Colin Jordan'; serusers-request(a)lists.iptel.org >>Cc: serusers(a)lists.iptel.org >>Subject: RE : [Serusers] Handoff to PSTN >> >> >> >>just have a look at /var/log/messages u will find the line number where >> >> >the > > >>error occurs >> >> >> >>Olivier >> >>-----Message d'origine----- >>De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la >>part de Colin Jordan >>Envoyé : jeudi 15 septembre 2005 12:33 >>À : serusers-request(a)lists.iptel.org >>Cc : serusers(a)lists.iptel.org >>Objet : [Serusers] Handoff to PSTN >> >> >> >>I am not able to restart ser if I put he code below in the ser.cfg to >>handoff calls to PSTN. >> >> >> >># attempt handoff to PSTN >> >>if (uri=~“^sip:9[0-9]*@mydomain.com”) { >> >>log(“Forwarding to PSTN\n”); >> >>t_relay_to( “192.168.0.2”, “5060”); >> >>break; >> >>}; >> >> >> >> >> >>Please help!!!! >> >>DISCLAIMER: The information in this message is confidential and is legally >>privileged. It is intended solely for the addressee. Access to this message >>by >>anyone else is unauthorized.If receiving in error please accept our >>apologies >>and notify the sender immediately. You must also delete the original >>message from your machine. If you are not the intended recipient, any use, >>disclosure, copying, distribution or action taken in reliance of it, is >>prohibited >>and maybe >>unlawful. >> >>DISCLAIMER: The information in this message is confidential and is legally >>privileged. It is intended solely for the addressee. Access to this message >>by >>anyone else is unauthorized.If receiving in error please accept our >>apologies >>and notify the sender immediately. You must also delete the original >>message from your machine. If you are not the intended recipient, any use, >>disclosure, copying, distribution or action taken in reliance of it, is >>prohibited >>and maybe >>unlawful. >> >>DISCLAIMER: The information in this message is confidential and is legally >>privileged. It is intended solely for the addressee. Access to this message >>by >>anyone else is unauthorized.If receiving in error please accept our >>apologies >>and notify the sender immediately. You must also delete the original >>message from your machine. If you are not the intended recipient, any use, >>disclosure, copying, distribution or action taken in reliance of it, is >>prohibited >>and maybe >>unlawful. >> >>DISCLAIMER: The information in this message is confidential and is legally >>privileged. It is intended solely for the addressee. Access to this message >>by >>anyone else is unauthorized.If receiving in error please accept our >>apologies >>and notify the sender immediately. You must also delete the original >>message from your machine. If you are not the intended recipient, any use, >>disclosure, copying, distribution or action taken in reliance of it, is >>prohibited >>and maybe >>unlawful. >> >>DISCLAIMER: The information in this message is confidential and is legally >>privileged. It is intended solely for the addressee. Access to this message >>by >>anyone else is unauthorized.If receiving in error please accept our >>apologies >>and notify the sender immediately. You must also delete the original >>message from your machine. If you are not the intended recipient, any use, >>disclosure, copying, distribution or action taken in reliance of it, is >>prohibited >>and maybe >>unlawful. >> >> >> >> >> >> >> > > >. > > >

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Re: [Serusers] No VM module and IVR plugin
by Nicky 19 Sep '05

19 Sep '05

Hi, thank you for your reply. as I need to have multiple ppl to access this sems and i understand from the readme that it will be better to use socket over fifo for more users. The ser.cfg file does not take the variable "socket", I have tested on the unix_socket, it works. Is this the correct vairable to assign? Do you know how can I alllow sems to have multiple users to access as I am having this error:~ Error : stop (AmThread.cpp:99) : pthread_detach failed with code ENINVAL : thread already in detached state. Please help. regards, nicky ----- Original Message ----- From: Gareth C. Fowler To: Nicky Sent: Friday, September 16, 2005 5:46 PM Subject: RE: [Serusers] No VM module and IVR plugin It should auto-create this file, check the socket file location in the ser configuration. You could also try the command: # touch /tmp/ser_socket Any particular reason for using the socket over fifo? Hope this helps. Rgds. G. -----Original Message----- From: Nicky [mailto:nicky@caliber.com.sg] Sent: 16 September 2005 09:34 To: Gareth C. Fowler Subject: Re: [Serusers] No VM module and IVR plugin Hi G. Thank you so much for your helps. I have another problem which I would like to use the ser_socket but in my /tmp there isnt any related ser_socket file. Do you know how can i get this file? Please help...thanks... Regards, Nicky ----- Original Message ----- From: Gareth C. Fowler To: sems(a)lists.iptel.org ; serusers(a)lists.iptel.org Sent: Thursday, September 15, 2005 10:24 PM Subject: RE: [Serusers] No VM module and IVR plugin The IVR module is only available in the SEMS development version, and can be downloaded @ http://www.iptel.org/sems/sems_devel_snapshot_190505.tar.bz2 Sadly they took out the VM module in version 0.9.0 onwards, and it's now in the tm module. Instead of using vm you have to use t_write_req (fifo), and t_write_unix (socket). Hope this helps! G. -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org]On Behalf Of Nicky Sent: 15 September 2005 13:19 To: sems(a)lists.iptel.org; serusers(a)lists.iptel.org Subject: [Serusers] No VM module and IVR plugin Dear all, Please help. I am trying to download Sip_router_rel_0_8_12 and Sems version ser_rel_0_8_12 from the CVS but in the sems there is missing module for IVR. Then I try to download 0.9.0 version for both but there is missing VM module in Ser which is required to link to Sems. Please advice if I would like to use 0.9.0 version, without VM module how can link to SEMS and why in the sems version ser_rel_0_8_12 the IVR plugin is missing and where can I find it. Thanks in advance.... Regards, Nicky -------------------------------------------------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Can connect to server but can't call
by Alessio Rosi 18 Sep '05

18 Sep '05

Hi, I'm stting up ser for my home (local network). I'm using MySql registration I can login whit a softphone (Xlite) but I'm not able to call. The answer is : Call failed: 404 Not Found This is my ser.cfg # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ alias=sip.eraclito.it alias=eraclito.it alias=10.10.10.7 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("sip.eraclito.it", "subscriber")) { # www_challenge("sip.eraclito.it", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } ********************************** This is wath ngrep catch (note that 81.208.34.102 is the public IP of my router/firewall) U 10.10.10.5:5060 -> 10.10.10.7:5060 INVITE sip:admin@sip.eraclito.it SIP/2.0..Via: SIP/2.0/UDP 81.208.34.102:50 60..From: arfio <sip:arfiodefarro@sip.eraclito.it>;tag=139597290..To: <sip: admin(a)sip.eraclito.it>..Contact: <sip:arfiodefarro@81.208.34.102:5060>..Cal l-ID: 89ABA3C9-01C8-4B06-8847-4C322A9E27FA@10.10.10.5..CSeq: 11480 INVITE.. Content-Type: application/sdp..Content-Length: 272....v=0..o=arfiodefarro 7 787500 7787500 IN IP4 81.208.34.102..s=X-Lite..c=IN IP4 81.208.34.102..t=0 0..m=audio 8000 RTP/AVP 4 0 8 3 101..a=rtpmap:4 G723/8000..a=rtpmap:0 pcmu/ 8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:101 telephone-eve nt/8000..a=fmtp:101 0-15.. # U 10.10.10.7:5060 -> 10.10.10.5:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 81.208.34.102:5060;received=10.1 0.10.5..From: arfio <sip:arfiodefarro@sip.eraclito.it>;tag=139597290..To: < sip:admin@sip.eraclito.it>;tag=b27e1a1d33761e85846fc98f5f3a7e58.05fe..Ca ll- ID: 89ABA3C9-01C8-4B06-8847-4C322A9E27FA@10.10.10.5..CSeq: 11480 INVITE..Se rver: Sip EXpress router (0.9.3 (i386/linux))..Content-Length: 0..Warning: 392 10.10.10.7:5060 "Noisy feedback tells: pid=23852 req_src_ip=10.10.10.5 req_src_port=5060 in_uri=sip:admin@sip.eraclito.it out_uri=sip:admin@sip.e raclito.it via_cnt==1".... # U 10.10.10.5:5060 -> 10.10.10.7:5060 ACK sip:admin@sip.eraclito.it SIP/2.0..Via: SIP/2.0/UDP 81.208.34.102:5060; received=10.10.10.5..From: arfio <sip:arfiodefarro@sip.eraclito.it>;tag=139 597290..To: <sip:admin@sip.eraclito.it>;tag=b27e1a1d33761e85846fc98f5f3a7e5 8.05fe..Contact: <sip:arfiodefarro@81.208.34.102:5060>..Call-ID: 89ABA3C9-0 1C8-4B06-8847-4C322A9E27FA@10.10.10.5..CSeq: 11480 ACK..Max-Forwards: 70..C ontent-Length: 0....

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[Serusers] From header
by Iqbal 18 Sep '05

18 Sep '05

Hi The From request in REGISTER and INVITE, what should be in the domain segment of it, should it be, the IP address of the UA, the proxy or what, I seem to be getting conflicting results with a quintum, and wanted to double check Iqbal

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Re: [Serusers] Re: Does SER works on TWO network interfaces with public and private IP addresses?
by Charles Wang 18 Sep '05

18 Sep '05

Mike: Before I test these two interfaces(Private and Public), I have only a public interface on my SER proxy. My NATed clients are XLite or any SIP IP phones or SIP gateways. They work fine with SER and each other. When I try to make these clients register from private interface, the problem happens. So I don't think it is the problem of NAT functions at XLite or Gateways. On 9/18/05, Mike Williams <mwilliams(a)etc1.net> wrote: > I was having problems with NAT myself, and found this. I thought it sounded like your problem. > > http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.h… > > ---Mike > > 5.1. SIP with NAT or Firewalls ( Back to Tutorials Page ) > > 1.1. Problem Description: > > > Most conventional voip protocols (SIP, h323, …) are not programmed with NAT in mind, on itself they only carry call signaling (call setup, teardown,… and use RTP to carry the audio samples. > > The signaling usually uses fixed and standardized ports, but the RTP uses random ports to exchange both call legs (incoming and outgoing audio). > > Most firewalls/NATs are unable to link the signalling protocol packets with the audio packets and are in some cases unable to tell where to send the audio to. > > When making a call, everything will seem to go normal, caller id will get passed, ringing will start, you can pick up and hangup the call, but no audio in one or both directions. > > > > -----Original Message----- > From: Charles Wang [mailto:lazy.charles@gmail.com] > Sent: Sat 9/17/2005 2:40 AM > To: Mike Williams; serusers(a)iptel.org > Subject: Re: [Serusers] Re: Does SER works on TWO network interfaces with public and private IP addresses? > > Hi, Mike: > the default gateway is 192.168.11.254 not 192.168.11.1. > So i dont think the flow is 192.168.11.2 to 192.168.11.1. > > On 9/17/05, Mike Williams <mwilliams(a)etc1.net> wrote: > > I believe your problem is simple. With the SIP protocol, you are sending > > the streams like this: > > > > 192.168.11.2 -> 192.162.11.1 -> 221.21.X.X > > > > After you answer, the clients negotiate for RTP traffic, and try to send > > data directly from 192.168.11.2 to 221.21.X.X, not using the SIP server. > > You are probably having problems actually routing the data (trying > > pinging the 221.21.X.X box from your 192.168.11.2 client) or you're > > having NAT issues. Is far as I know, you must have a direct route from > > the caller to the callee to pass RTP streams; you can't proxy them > > through the SIP server. > > > > Good luck, and let me know if you have any more questions. > > > > Mike Williams (mwilliams(a)etc1.net) > > > > Charles Wang wrote: > > > > >On 9/13/05, Charles Wang <lazy.charles(a)gmail.com> wrote: > > > > > > > > >>Hi, ALL: > > >> > > >>I use ser + mediaproxy + PSTN support, and my ser with two interfaces. > > >>One is public IP address such as 211.21.xxx.xxx. > > >>Another one is private IP address such as 192.168.11.1. > > >> > > >>And I use XLite (192.168.11.2) register to SER's private interface(via > > >>HUB only). > > >>It can register sucessfully. > > >> > > >>But when I make a call to PSTN with this XLite, the callee rings and I > > >>answer it. > > >>I can not hear any sounds from each side. > > >> > > >>I try to register another XLite(192.168.11.3), and make a call to > > >>another private XLite(192.168.11.2). I can hear rings but it is still > > >>no any sounds from each side. > > >> > > >>Can anyone tell me what it happens? > > >> > > >>Best Regard > > >>Charles > > >> > > >> > > >> > > > > > > > > > > > > > > > > > -- > > Best Regards > Charles > > > -- Best Regards Charles -- Best Regards Charles

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[Serusers] SER Load balancing
by Marcello Lupo 17 Sep '05

17 Sep '05

Hi to all, did you have suggestions how to accomplish to load balancing load of SER between 2 machine running both SER and DB to check routing entries?? I was thinking to use LVM but i don't know how to let the same call messages to go to the same machine. There should be something line an LVM module able to check the CallID field and keep track of it. May be some other ways?? Using DNS records or other? Another question is if there is a searchable archive of this Mailing list. Thanks, Bye, Marcello

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Re: [Serusers] Re: Does SER works on TWO network interfaces with public and private IP addresses?
by Charles Wang 17 Sep '05

17 Sep '05

Hi, Mike: the default gateway is 192.168.11.254 not 192.168.11.1. So i dont think the flow is 192.168.11.2 to 192.168.11.1. On 9/17/05, Mike Williams <mwilliams(a)etc1.net> wrote: > I believe your problem is simple. With the SIP protocol, you are sending > the streams like this: > > 192.168.11.2 -> 192.162.11.1 -> 221.21.X.X > > After you answer, the clients negotiate for RTP traffic, and try to send > data directly from 192.168.11.2 to 221.21.X.X, not using the SIP server. > You are probably having problems actually routing the data (trying > pinging the 221.21.X.X box from your 192.168.11.2 client) or you're > having NAT issues. Is far as I know, you must have a direct route from > the caller to the callee to pass RTP streams; you can't proxy them > through the SIP server. > > Good luck, and let me know if you have any more questions. > > Mike Williams (mwilliams(a)etc1.net) > > Charles Wang wrote: > > >On 9/13/05, Charles Wang <lazy.charles(a)gmail.com> wrote: > > > > > >>Hi, ALL: > >> > >>I use ser + mediaproxy + PSTN support, and my ser with two interfaces. > >>One is public IP address such as 211.21.xxx.xxx. > >>Another one is private IP address such as 192.168.11.1. > >> > >>And I use XLite (192.168.11.2) register to SER's private interface(via > >>HUB only). > >>It can register sucessfully. > >> > >>But when I make a call to PSTN with this XLite, the callee rings and I > >>answer it. > >>I can not hear any sounds from each side. > >> > >>I try to register another XLite(192.168.11.3), and make a call to > >>another private XLite(192.168.11.2). I can hear rings but it is still > >>no any sounds from each side. > >> > >>Can anyone tell me what it happens? > >> > >>Best Regard > >>Charles > >> > >> > >> > > > > > > > > > -- Best Regards Charles

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[Serusers] SER proxy and SER+SEMS media server
by Teemu Harju 17 Sep '05

17 Sep '05

Hi, I'm implementing following kind of system. I have two separate machines where I've installed SER on. The other machine has also SEMS installed. Like you probably already figured out other is used as a plain proxy/registrar and the other as a sort of UAS serving voicemail, conference etc with the help of SEMS. Now my question is, what is the best way to forward messages from proxy/registrar to the media server? I've done the forwarding according to the examples I've found and it uses rewritehostport() to forward messages to voicemail. Voicemail needs to get the email address from mysql database located in proxy. This is no problem connecting to the database, but isn't it so that when I've rewritten the Request-URI that the voicemail cannot find the email from database beacause it's rewritten and has the media servers host part in the URI. How have you guys implemented this? I would also need other SEMS plug-ins like conference, so I would also need to differentiate those requests also. Could I rewerite the Request-URI like voicemail@mediaserver? Can the voicemail still find out who was I actually trying to call to since the URI is rewritten? Is there difference in implementing the media server with asterisk? Wouldn't I need then two different databases? One in asterisk and one in SER. Thanks... -- Teemu Harju http://www.teemuharju.net

2 2

[Serusers] Building SEMS on Mac OS..share your experience..
by Amos Nungu 16 Sep '05

16 Sep '05

Hi, I am Installing SER/SEMS on Mac. SER is running fine Building SEMS give some error but it does build to the end, and I can run "make Install" BUT can't start SEMS.. error below ser:~ root# sems bash: /usr/local/sbin/sems: cannot execute binary file and compilation error is.. #make all cd gsm-1.0-pl10; make ./lib/libgsm.a make[5]: `lib/libgsm.a' is up to date. cc -o gsm.so gsm.o gsm-1.0-pl10/lib/libgsm.a -shared powerpc-apple-darwin8-gcc-4.0.0: unrecognized option `-shared' /usr/bin/ld: gsm.o bad magic number (not a Mach-O file) /usr/bin/ld: warning empty table of contents: gsm-1.0-pl10/lib/libgsm.a (can't load from it) collect2: ld returned 1 exit status make[4]: *** [gsm.so] Error 1 make[3]: [all] Error 2 (ignored) make[4]: Nothing to be done for `deps'. make[5]: Nothing to be done for `all'. make[4]: Nothing to be done for `deps'. make[4]: `../../lib/apps/isdngw.so' is up to date. make[4]: Nothing to be done for `deps'. g++ -o ivr.so Ivr.o IvrDtmfDetector.o IvrEvents.o IvrMediaHandler.o IvrPython.o -lm -ldl -lsocket -lnsl -lpthread -shared -ldl -lpthread -lutil -lm -Xlinker --export-dynamic -L/usr/lib/python2.3/config -lpython2.3 /usr/lib/python2.3/lib-dynload/time.so powerpc-apple-darwin8-g++-4.0.0: unrecognized option `-shared' /usr/bin/ld: unknown flag: --export-dynamic collect2: ld returned 1 exit status make[4]: *** [ivr.so] Error 1 make[3]: [all] Error 2 (ignored) make[4]: Nothing to be done for `deps'. cc -fPIC -I ../../amci -Wall -I./lame-3.96/include -c mp3.c -o mp3.o mp3.c:37:18: error: lame.h: No such file or directory mp3.c: In function 'MP3_create': mp3.c:88: error: 'lame_global_flags' undeclared (first use in this function) mp3.c:88: error: (Each undeclared identifier is reported only once mp3.c:88: error: for each function it appears in.) mp3.c:88: error: 'gfp' undeclared (first use in this function) mp3.c:93: warning: implicit declaration of function 'lame_init' mp3.c:98: warning: implicit declaration of function 'lame_set_errorf' mp3.c:99: warning: implicit declaration of function 'lame_set_debugf' mp3.c:100: warning: implicit declaration of function 'lame_set_msgf' mp3.c:102: warning: implicit declaration of function 'lame_set_num_channels' mp3.c:103: warning: implicit declaration of function 'lame_set_in_samplerate' mp3.c:104: warning: implicit declaration of function 'lame_set_brate' mp3.c:105: warning: implicit declaration of function 'lame_set_mode' mp3.c:106: warning: implicit declaration of function 'lame_set_quality' mp3.c:108: warning: implicit declaration of function 'id3tag_init' mp3.c:109: warning: implicit declaration of function 'id3tag_set_title' mp3.c:110: warning: implicit declaration of function 'lame_init_params' mp3.c:118: warning: control reaches end of non-void function mp3.c: In function 'MP3_destroy': mp3.c:122: error: 'lame_global_flags' undeclared (first use in this function) mp3.c:122: error: parse error before ')' token mp3.c: In function 'Pcm16_2_MP3': mp3.c:138: warning: implicit declaration of function 'lame_encode_buffer' mp3.c:138: error: 'lame_global_flags' undeclared (first use in this function) mp3.c:138: error: parse error before ')' token mp3.c:143: warning: control reaches end of non-void function mp3.c: In function 'MP3_close': mp3.c:224: warning: implicit declaration of function 'lame_encode_flush' mp3.c:224: error: 'lame_global_flags' undeclared (first use in this function) mp3.c:224: error: parse error before ')' token mp3.c:228: warning: implicit declaration of function 'lame_mp3_tags_fid' mp3.c:228: error: parse error before ')' token mp3.c:221: warning: unused variable 'mp3buffer' mp3.c:229: warning: control reaches end of non-void function make[4]: *** [mp3.o] Error 1 make[3]: [all] Error 2 (ignored) . . . make[4]: Nothing to be done for `deps'. make[4]: `../../lib/apps/voicemail.so' is up to date. make[4]: Nothing to be done for `deps'. make[4]: `../../lib/audio/wav.so' is up to date. Regards

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[Serusers] Authentication
by Sebastian Kühner 16 Sep '05

16 Sep '05

Hello, I want to authenticate the users that wants to make a PSTN call on every INVITE. I have one user that can't make PSTN calls, because the message "407 Proxy Authentication Required" goes back to the wrong port (the IP is correct). I checked it with ngrep. Has anyone of you an idea how could I fix that problem? Thanks! Sebastian

2 1

[Serusers] Re: 400 Bad Request
by Alan Chiu 16 Sep '05

16 Sep '05

This INVITE came from a client device but not sure why SER 0.8.14 (i386/linux) would reject it with 400 Bad Request and there's no reason given. Any insights would help. ***************************************************** Session Initiation Protocol Request line: INVITE sip:4519@10.100.0.111:5060;transport=udp SIP/2.0 Message Header To: <sip:4519@10.100.0.222> From: "PHONE TEST"<sip:1111@172.31.1.1>;tag=5b4ae231 Via: SIP/2.0/UDP 172.31.1.20:5060;branch=z9hG4bK-d87543-ffe1a8337da8f24e89ef-1-cHA5ZGU5Mz E0ZWU3YTE3YjUzNjgxYg..-d87543- Call-ID: 4132a9661e815dd517bec43512d032db CSeq: 267297881 INVITE Contact: <sip:4AmIBQqlz52H8OKUlL_xUG_l--zuAAngOdmZbN4wa1rlMGPDB0hIkU7H3mat-JyuSGS Vy4.@172.31.1.20:5060> Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Date: Thu, 15 Sep 2005 16:34:32 GMT User-Agent: Server Authorization: Digest username="testuser(a)domains.com",realm="companyX.com",algorithm="MD5",uri ="sip:4519@10.100.0.222",no nce="432993119128c2b1a32ca43b9436e2fe0bca62f0",response="e42bc4a17da0282 552265396c0eb461e",opaque="" Content-Length: 243 Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 27663 1963582334 IN IP4 172.31.1.20 Owner Username: root Session ID: 27663 Session Version: 1963582334 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 172.31.1.20 Session Name (s): session Connection Information (c): IN IP4 172.31.1.20 Connection Network Type: IN Connection Address Type: IP4 Connection Address: 172.31.1.20 Time Description, active time (t): 0 0 Session Start Time: 0 Session Start Time: 0 Media Description, name and address (m): audio 10492 RTP/AVP 0 8 101 Media Type: audio Media Port: 10492 Media Proto: RTP/AVP Media Format: 0 Media Format: 8 Media Format: 101 Media Attribute (a): fmtp:101 0-16 Media Attribute Fieldname: fmtp Media Attribute Value: 101 0-16 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 101 telephone-event/8000 Media Attribute (a): silenceSupp:off - - - - Media Attribute Fieldname: silenceSupp Media Attribute Value: off - - - - Session Initiation Protocol Status line: SIP/2.0 400 Bad Request Message Header To: <sip:4519@10.100.0.222>;tag=b27e1a1d33761e85846fc98f5f3a7e58.6a43 From: "PHONE TEST"<sip:1111@172.31.1.1>;tag=5b4ae231 Via: SIP/2.0/UDP 172.31.1.20:5060;branch=z9hG4bK-d87543-ffe1a8337da8f24e89ef-1-cHA5ZGU5Mz E0ZWU3YTE3YjUzNjgxYg..-d87543- Call-ID: 4132a9661e815dd517bec43512d032db CSeq: 267297881 INVITE Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 10.100.0.111:5060 "Noisy feedback tells: pid=6024 req_src_ip=172.31.1.20 req_src_port=5060 in_uri=sip:4519@10. 100.0.111:5060;transport=udp out_uri=sip:4519@10.100.0.111:5060;transport=udp via_cnt==1" *****************************************************

1 0

[Serusers] Does SER works on TWO network interfaces with public and private IP addresses?
by Charles Wang 16 Sep '05

16 Sep '05

Hi, ALL: I use ser + mediaproxy + PSTN support, and my ser with two interfaces. One is public IP address such as 211.21.xxx.xxx. Another one is private IP address such as 192.168.11.1. And I use XLite (192.168.11.2) register to SER's private interface(via HUB only). It can register sucessfully. But when I make a call to PSTN with this XLite, the callee rings and I answer it. I can not hear any sounds from each side. I try to register another XLite(192.168.11.3), and make a call to another private XLite(192.168.11.2). I can hear rings but it is still no any sounds from each side. Can anyone tell me what it happens? Best Regard Charles

1 1

[Serusers] SIP VXML
by Abdul Lateef 16 Sep '05

16 Sep '05

Hi all, Can any suggest me to developing IVR using vxml for SIP Gatekeeper, Is there any SIP Vxml open source available? Thank You Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

FW: [Serusers] Call forwarding Question (following issue 5)
by Aisling 16 Sep '05

16 Sep '05

Sorry I never attached the messages... -----Original Message----- From: Aisling [mailto:ashling.odriscoll@cit.ie] Sent: 16 September 2005 11:31 To: 'Greger V. Teigre'; 'serusers(a)lists.iptel.org' Subject: RE: [Serusers] Call forwarding Question (following issue 5) I have attached a complete ngrep of the messages. There were no errors in the var/log/messages file. Originally 3500 and 5000 were on the same machine so I changed the scenario so that 3500 (Windows messenger) was calling 2092 (BT100 - which is busy) and should be forwarded to 2009 (KPhone, same pc as SER). However it still doesn't forward the call.....Its the last two messages confuse me...I don't understand why a 407 Proxy Authentication required would be sent back to original caller.... My rule in the usr_preferences table of the mysql database is: Username (2092) Attribute (fwdbusy) Value (sip:2009@serveraddress) Any ideas? Many thanks, Aisling. -----Original Message----- From: Greger V. Teigre [mailto:greger@teigre.com] Sent: 15 September 2005 18:43 To: Aisling; serusers(a)lists.iptel.org Subject: Re: [Serusers] Call forwarding Question (following issue 5) Yes, we have verified that the configs lack the append_branch you mention. The fix has been done (https://siprouter.onsip.org/trac/changeset/15) and new configs are available through the regular downloads at onsip.org. To your other problem: No clue, Aisling. Sounds very strange indeed. The config should only trigger proxy authentication on an INVITE (not REGISTER). Maybe what you are experiencing is Messenger sending a new INVITE. (It should not, as it should receive a 100 Trying from ser). Only a *complete* ngrep trace will help in understanding what's happening (and any error messages in /var/log/messages with the timestamp for matching) g-) Aisling wrote: > Hi, > > I am testing the call forwarding features demonstrated in Issue 5 of > the > onsip getting started document. I found that blind call transfer > worked > perfectly but fwdbusy & fwdnoanswer gave me errors: > > ERROR: t_forward_nonack: non branched for forwarding > ERROR: w_t_relay(failure mode): forwarding failed > ERROR: sl_reply_error used: I'm terribly sorry, server error occurred. > > On the onsip site I noted that someone else had this problem and it > was > solved by putting append_branch in the fwdbusy and fwdnoanswer > sections in > the failure route. > > Thankfully that fixed those errors. However when I went to test > fwdbusy > again, it doesn't give any errors but still doesn't work. The call > scenario > was as follows: > > Windows Messenger client 3500 ring BT100 2092. 2092 is off the hook > (thereby > sending a 486 busy message) and the call should be forwarded to xlite > client > 5000. > > i.e. 3500 -> 2092(busy) -> 5000 > > The message sequence showed that everything was correct up to 2092 > sending > the 486 busy to SER and then SER sending an ACK back to 2092. But > then SER > sends a 407 proxy authentication required to 3500 which replies with > an > ACK....and that's it... > > Can someone explain why SER would send a 407 Proxy authentication to > the > original caller?...I thought this should only be in response to a > register > message?.... > > Any help appreciated, > Thanks, > Aisling. > > > > -------------------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 >> 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.

2 2

Re: [Serusers] SER proxy and SER+SEMS media server
by Samuel Osorio Calvo 16 Sep '05

16 Sep '05

You might try to just forward without modifying the Req-URI with the config file options: t_relay_to_udp(ip, port); OR t_relay_to_tcp(ip, port OR forward(ip,port); where the tuple ip,port corresponds to the SER instance acting as voicemail/conference server. The last option does not require the tm module and does not involve any transaction, Depending on your performance requirements, it should be better to use one or another... Hope it helps, Samuel. Unclassified. >>> Teemu Harju <teemu.harju(a)gmail.com> 09/16/05 03:11PM >>> Hi, I'm implementing following kind of system. I have two separate machines where I've installed SER on. The other machine has also SEMS installed. Like you probably already figured out other is used as a plain proxy/registrar and the other as a sort of UAS serving voicemail, conference etc with the help of SEMS. Now my question is, what is the best way to forward messages from proxy/registrar to the media server? I've done the forwarding according to the examples I've found and it uses rewritehostport() to forward messages to voicemail. Voicemail needs to get the email address from mysql database located in proxy. This is no problem connecting to the database, but isn't it so that when I've rewritten the Request-URI that the voicemail cannot find the email from database beacause it's rewritten and has the media servers host part in the URI. How have you guys implemented this? I would also need other SEMS plug-ins like conference, so I would also need to differentiate those requests also. Could I rewerite the Request-URI like voicemail@mediaserver? Can the voicemail still find out who was I actually trying to call to since the URI is rewritten? Is there difference in implementing the media server with asterisk? Wouldn't I need then two different databases? One in asterisk and one in SER. Thanks... -- Teemu Harju http://www.teemuharju.net

1 0

[Serusers] DTMF
by Iqbal 16 Sep '05

16 Sep '05

Hi I am guessing the ser does nothing with dtmf coming from the pstn side, since it's all in the rtp stream, if is there anyway of "seeing" what ser is doing with it, or what is being sent, I can see the dtmf in asterisk being passed from ser, but if I use another remote gateway, I wanted to be able to see if possible, even using ethreal or whatever Iqbal

1 0

[Serusers] [about SER] 483 too many hops problem
by 송 미진 16 Sep '05

16 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 dear, I'm working on a project and i'm using ser as a sip server. actually i have this problem: every time i send an instant message to a subscriber i get this error message: 483 too many hops i tried to fix it but i couldn't and i don't know what to do ! (My computer connect many computers. Is it reason?!) if any one can help me i'll be very grateful thank you in advance i'm using ser 0.9.3 & here's my code: # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://root:vlftmd@localhost/ser" alias=sam.cse # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc","db_url","mysql://root:vlftmd@localhost/ser") #modparam("usrloc", "db_mode", 0) modparam("registrar","use_domain",1) # 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","secret","123123123") #modparam("auth_db","db_url","mysql://root:vlftmd@localhost/ser") #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("sam.cse", "subscriber")) { www_challenge("sam.cse", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } _________________________________________________________________ 상큼한 만남과 따뜻한 공동체 생활... 지금 MSN 커뮤니티에서 시작하세요! http://groups.msn.com/?pgmarket=ko-kr

2 1

[Serusers] fields To/From
by Piccari Gianluca 16 Sep '05

16 Sep '05

Hi all, i have a big problem... i need to parser information in Header field From() and eventually add or delete part of info in Header field TO. For istance: if i(n From field there is '55') add ('55' in To field). I don't have familiar to SER (i have started to study it last week (: ) and i likr e to know if it is possible to do it using SER's method without change the source code .... thanks for your help

1 0

RE : [Serusers] Handoff to PSTN
by Olivier Taylor 16 Sep '05

16 Sep '05

module tm -----Message d'origine----- De : Colin Jordan [mailto:cjordan@econetwireless.com] Envoyé : jeudi 15 septembre 2005 14:55 À : 'Olivier Taylor' Cc : serusers(a)lists.iptel.org Objet : RE: [Serusers] Handoff to PSTN t_relay_to( 192.168.0.2, 5060); _____ From: Olivier Taylor [mailto:olivier.taylor@gmail.com] Sent: Thursday, September 15, 2005 2:50 PM To: 'Colin Jordan' Cc: serusers(a)lists.iptel.org Subject: RE : [Serusers] Handoff to PSTN what's on the line 143??? -----Message d'origine----- De : Colin Jordan [mailto:cjordan@econetwireless.com] Envoyé : jeudi 15 septembre 2005 14:46 À : 'Olivier Taylor' Cc : serusers(a)lists.iptel.org Objet : RE: [Serusers] Handoff to PSTN The line 143 points to the script below. What modules will it be looking for? _____ From: Olivier Taylor [mailto:olivier.taylor@gmail.com] Sent: Thursday, September 15, 2005 2:33 PM To: 'Colin Jordan' Cc: serusers(a)lists.iptel.org Subject: RE : [Serusers] Handoff to PSTN please reply to all u have a module not loaded, the one involved on line 143 Olivier -----Message d'origine----- De : Colin Jordan [mailto:cjordan@econetwireless.com] Envoyé : jeudi 15 septembre 2005 14:28 À : 'Olivier Taylor' Objet : RE: [Serusers] Handoff to PSTN It gives me the following message: _______________________________________________________ Parse error (143,40-41): unknown command missing loadmodule? _______________________________________________________ _____ From: Olivier Taylor [mailto:olivier.taylor@gmail.com] Sent: Thursday, September 15, 2005 1:21 PM To: 'Colin Jordan'; serusers-request(a)lists.iptel.org Cc: serusers(a)lists.iptel.org Subject: RE : [Serusers] Handoff to PSTN just have a look at /var/log/messages u will find the line number where the error occurs Olivier -----Message d'origine----- De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de Colin Jordan Envoyé : jeudi 15 septembre 2005 12:33 À : serusers-request(a)lists.iptel.org Cc : serusers(a)lists.iptel.org Objet : [Serusers] Handoff to PSTN I am not able to restart ser if I put he code below in the ser.cfg to handoff calls to PSTN. # attempt handoff to PSTN if (uri=~^sip:9[0-9]*@mydomain.com) { log(Forwarding to PSTN\n); t_relay_to( 192.168.0.2, 5060); break; }; Please help!!!! DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful. DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful. DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful. DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

3 6

[Serusers] Call forwarding Question (following issue 5)
by Aisling 16 Sep '05

16 Sep '05

Hi, I am testing the call forwarding features demonstrated in Issue 5 of the onsip getting started document. I found that blind call transfer worked perfectly but fwdbusy & fwdnoanswer gave me errors: ERROR: t_forward_nonack: non branched for forwarding ERROR: w_t_relay(failure mode): forwarding failed ERROR: sl_reply_error used: I'm terribly sorry, server error occurred. On the onsip site I noted that someone else had this problem and it was solved by putting append_branch in the fwdbusy and fwdnoanswer sections in the failure route. Thankfully that fixed those errors. However when I went to test fwdbusy again, it doesn't give any errors but still doesn't work. The call scenario was as follows: Windows Messenger client 3500 ring BT100 2092. 2092 is off the hook (thereby sending a 486 busy message) and the call should be forwarded to xlite client 5000. i.e. 3500 -> 2092(busy) -> 5000 The message sequence showed that everything was correct up to 2092 sending the 486 busy to SER and then SER sending an ACK back to 2092. But then SER sends a 407 proxy authentication required to 3500 which replies with an ACK....and that's it... Can someone explain why SER would send a 407 Proxy authentication to the original caller?...I thought this should only be in response to a register message?.... Any help appreciated, Thanks, Aisling. -------------------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.

2 2

[Serusers] ceratins users to one gateway others to another
by Nhadie 16 Sep '05

16 Sep '05

Hi All, Can I have some extensions use gateway 1 and some can use gateway 2? I want to do this for clients that pay premium they have unlimited calls for free on a certain onuntry and others will still go to the per minute basis. Can I set this up on groups? Hope anyone can help me. Thank You Regards, Nhadie

3 4

[Serusers] ser.cfg with PSTN gateway connectivity
by Alberto 16 Sep '05

16 Sep '05

Hi, I've read the SER Getting Started form http://www.onsip.org . When I configure the ser with PSTN Gateway (subject 8 of the guide) I can't register any client (403, Forbidden). I think, the problem is the next line of ser.cfg: if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403","Forbidden"); }; break; }; I'd copied the permissions.allow and permissions.deny to /usr/local/etc/ser and in the .allow file I'd uncomment the last line (ALL:ALL) Any idea? Thanks. -- Alberto

3 8

Re: [Serusers] Distributing "OSP enable SER" binary file w/ source
by Dmitry Isakbayev 16 Sep '05

16 Sep '05

Hi Jan, I am following up on the conversation that you and Vikrant had a short while ago. The OSP interface was implemented as a SER module. It does not require any changes to the core SER or any of its modules. We have been hosting the project @ http://developer.berlios.de/projects/osp-module We would like to move the project to the experimental branch in your CVS. How should we proceed? > > In what form does the come ? Did you implement it as a SER module > > ? If so then we can put it in the experimental branch in CVS. If you > > made some changes to other modules or the core then it would be great if > > you could generate patches against the latest development version. Best Regards, Dmitry > > ----- Original Message ----- > > From: "Jan Janak" <jan(a)iptel.org> > > To: <vmathur(a)transnexus.com> > > Cc: <serusers(a)lists.iptel.org> > > Sent: Friday, August 12, 2005 2:35 PM > > Subject: Re: [Serusers] Distributing "OSP enable SER" binary file w/ > > source > > code > > > > > >> On 12-08-2005 18:18, vmathur(a)transnexus.com wrote: > >> > Hi Jan, > >> > > >> > The code has been written as a SER module. The details of the code are > > available at: > >> > http://developer.berlios.de/projects/osp-module > >> > > >> > Right now the process of compilation requires that you download the OSP > > toolkit source code and complie it. This would build the OSP client > > library > > that the SER would use. You should then download the SER and the OSP > > modules > > and build them. The process is rather lengthy and we were wondering if we > > come up with an easier way of doing this. > >> > > >> > What exactly is the experimental branch in CVS ? > > > >> SER CVS repository contains experimental directory. This is the > >> directory where anyone can put their contribution. It stays there for > >> a while and if it proves to be useful and supported by the author then > >> we move it to the main source tree, giving the author write access to > >> the main tree. > >> > >> Jan. > >> > >> > > > > In what form does the come come ? Did you implement it as a SER module > > ? If so then we can put it in the experimental branch in CVS. If you > > made some changes to other modules or the core then it would be great if > > you could generate patches against the latest development version. > > > > Jan.

2 1

[Serusers] Making SER implement Jabber
by Saleh, Saleh 15 Sep '05

15 Sep '05

Hello Everyone, Can anyone tell me what I need to do next in order for SER version 0.9.3 to use the Jabber Module correctly? This is what I have done so far (on a Red Hat Linux 9 machine): Installed Jabberd 1.4.4 Installed GNU pth-2.0.4 Installed Expat 1.95.8 Installed Libidn ? I have Modified the config file for SER to load the Jabber.so module after the pa.so and tm.so (I'm sure this is where I am stuck, what to do with parameters in config file for SER) This is what I can/cannot do: I can run the jabber server I cannot run the SER with the config file which includes the Jabber.so module load command but I can run it without this command in the config file. I am using MySQL for SER What other files do I need to modify? What information do I need? I'm not very familiar with linux, I am a beginner... Any comments/suggestions/websites/help is GREATLY appreciated! Thanks, Saleh

1 0

[Serusers] ser 0.9.4 crash
by Tulika Pradhan 15 Sep '05

15 Sep '05

i am trying the 0.9.4 release and sometimes ser starts up ok. other times it crashes. here is the 'bt' results. Loaded symbols for /lib/libnss_dns.so.2 #0 0x00b4eec3 in preload_udomain (_c=0x810c898, _d=0xb6178ce8) at udomain.c:302 302 ua.len = strlen(ua.s); (gdb) bt #0 0x00b4eec3 in preload_udomain (_c=0x810c898, _d=0xb6178ce8) at udomain.c:302 #1 0x00b4ab76 in register_udomain (_n=0x0, _d=0xbfedf6ec) at dlist.c:249 #2 0x00133795 in domain_fixup (param=0x810d834, param_no=1) at reg_mod.c:207 #3 0x08070723 in fix_actions (a=0x0) at route.c:238 #4 0x08070845 in fix_actions (a=0x0) at route.c:225 #5 0x080744e4 in fix_rls () at route.c:726 #6 0x0805e33d in main (argc=2, argv=0xbfedf9c4) at main.c:1582 (gdb) any help would be greatly appreciated. regards, tulika

2 1

[Serusers] SER and RFC3325
by Gummadidala, Ravi 15 Sep '05

15 Sep '05

Does SER support RFC3325? Does it understand headers like "Privacy", "P-Preferred-Identity" and "P-Asserted-Identity"? If so, how can we specify the Spec(T) to control SER's behavior? Thanks, -Ravi.

2 1

[Users] how to modify record route header with advertised_address ?
by Gustavo García 15 Sep '05

15 Sep '05

Hi, Ive configured an address with advertised_address. I can see that its used in Via address but not in record route. What could be the reason? Thanks a lot

3 6

[Users] cvs is frozen
by Daniel-Constantin Mierla 15 Sep '05

15 Sep '05

Hello everybody, the cvs head is frozen in preparation for the new release. Please commit only fixes to the C code. Documentation updates are very welcome as well as adjacent tools upgrades. In about two weeks (more or less), a new branch will be created the the cvs head will be unfrozen and open for new features. Cheers, Daniel

3 6

[Serusers] call hunt
by Tulika Pradhan 15 Sep '05

15 Sep '05

i am tring to implement call hunt in ser 0.8.12 (no avpops). there is one problem. i am using exec_dset in the failure route as follows. failure_route[1] { exec_dset("/etc/ser/getnextnumber1 $SIP_OUSER; echo>/dev/null;"); append_branch(); t_on_failure("1"); xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> <%tu>\n"); t_relay(); } after the first failure, the SER sends 408 to the UA (caller) and it gets a busy tone. at this point, the second number of the call hunt group is ringing. but no voice can go through as the caller UA is showing 408 Request Timeout. please help, regards, tulika

1 0

[Serusers] Speeddial
by Aisling 15 Sep '05

15 Sep '05

Hello. I would really appreciate if someone could show me an example of how a speed-dialing entry should be added to the ser mysql database? E.g. If user 2092 dials 3 it rings user 3500. I tried : insert into usr_preferences (username, domain, sd_username, sd_domain, new_uri) values ("2092", "x.x.x.x", "3500", "x.x.x.x", "3") where x.x.x.x = ser address I have the following in my ser.cfg: loadmodule "/usr/local/lib/ser/modules/speeddial.so" 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") if ((method=="INVITE") && (uri=~"^sip:[0-9]{2}@.*")){ sd_lookup("speed dial"); }; I got a 404 not found, but its probably because I didn't create the entry properly in the mysql database. If someone could show me an example that would be great. Many thanks. -------------------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.

2 1

[Serusers] SER and rtpproxy in a DMZ
by Holger Moskopp 15 Sep '05

15 Sep '05

Hello, my name is Holger Moskopp and i'm Student at the FH Cologne. At the moment i'm working on my Thesis. I have to build a Firewall with DMZ and a SIP Expressrouter with RTPPROXY. This should look like this: -------------------------------------------------------- http://www.ganeymed.de/pixx/fw_ids/ser1.jpg -------------------------------------------------------- I'm in a Subnet of the FH and got on the I' in a Subnetz of the school and have installed on the computer xxx.22 fwclient a SER Registrar with RTPproxy and a Kphone softphone. On the internal SER there is a Kphone registered. (holleinnen). In the DMZ is a SER with rtpproxy. In the FH-Net there is a SER with Radius authentification and two softphones. Phil(a)xxx.73 <mailto:Phil@xxx.73> is registerd at that SER. holleaussen is registerd on another registrar that is not on the picture. If I want to call from holleinnen to phil, everything functions marvelously. The SIP signaling and the RTP-Traffic runs throuhg the DMZ. -------------------------------------------------------- http://www.ganeymed.de/pixx/fw_ids/ser2.jpg -------------------------------------------------------- Now to the problem: If I start a call from holleaussen to holleinnen the SIP phase , works perfectly thruh the DMZ. It rings inside and I can assume. After that nothing more happens. With tetereal and etheral I saw that the RTP traffic „wants“ to take the way directly from end to end. -------------------------------------------------------- http://www.ganeymed.de/pixx/fw_ids/ser4.jpg -------------------------------------------------------- Do you have an idea what is going wrong? I attached the two ser.cfg files because i think it is a mistake there. I tryed to fix that since 3 days now - but with no success. -------------------------------------------------------------- http://www.ganeymed.de/pixx/fw_ids/ser-dmz.txt http://www.ganeymed.de/pixx/fw_ids/ser-innen.txt --------------------------------------------------------------- Here is the relevant Firewallpart: $IPTABLES -N SIPLOG $IPTABLES -I FORWARD -p udp -i $DMZ_ETH --sport 1024:65535 -o $EXTERN_ETH --dport 1024:65535 -m state --state NEW,ESTABLISHED,RELATED -j SIPLOG $IPTABLES -I FORWARD -p udp -i $EXTERN_ETH --sport 1024:65535 -o $DMZ_ETH --dport 1024:65535 -m state --state NEW,ESTABLISHED,RELATED -j SIPLOG $IPTABLES -I FORWARD -p udp -i $INTERN_ETH --sport 1024:65535 -o $DMZ_ETH --dport 1024:65535 -m state --state NEW,ESTABLISHED,RELATED -j SIPLOG $IPTABLES -I FORWARD -p udp -i $DMZ_ETH --sport 1024:65535 -o $INTERN_ETH --dport 1024:65535 -m state --state NEW,ESTABLISHED,RELATED -j SIPLOG $IPTABLES -I SIPLOG -j LOG --log-prefix "SIPLOG: " $IPTABLES -A SIPLOG -j ACCEPT $IPTABLES -t nat -I PREROUTING -p udp -i $EXTERN_ETH --dport 5060:5062 -j DNAT --to $prox /(That changing only send the packets to prox but prox don#t take them: $IPTABLES -t nat -I PREROUTING -p udp -i $EXTERN_ETH --dport 1024:65535/ /-j DNAT --to prox) / I think a soulution could be to use the statefull iptabels filtering, but I don't like that solution. Thank you and best regards Holger Moskopp

1 0

[Serusers] SER install on RH ES4
by Martin, Peter 15 Sep '05

15 Sep '05

Has anyone successfully installed SER on RH ES4? I am getting MySQL errors when trying to start the service or ser.pid errors. my.cnf: ################################################## # The following options will be passed to all MySQL clients [client] #password = your_password port = 3306 socket = /var/run/mysql/mysql.sock [mysqld] datadir=/var/lib/mysql socket=/var/run/mysql/mysql.sock [mysql.server] user=mysql basedir=/var/lib [safe_mysqld] err-log=/var/log/mysqld.log pid-file=/var/run/mysqld/mysqld.pid ################################################### SQL wants to see mysql.sock in /var/lib SER wants to see mysql.sock in /var/run BUT when I move mysql.sock to /var/lib by changing the paths in my.cnf then SER gives me grief: [root@localhost mysql]# serctl start Starting SER : cat: /var/run/ser.pid: No such file or directory started pid() Regards, Pete Martin

3 4

[Serusers] serweb php session
by Iqbal 15 Sep '05

15 Sep '05

Hi On one of my servers I'm using serweb with php session ID's is anyone else having a problem where sometimes the session is not created, hence login, or new user registrations cannot happen, cause it gives a 302 error Iqbal

2 5

[Serusers] No VM module and IVR plugin
by Nicky 15 Sep '05

15 Sep '05

Dear all, Please help. I am trying to download Sip_router_rel_0_8_12 and Sems version ser_rel_0_8_12 from the CVS but in the sems there is missing module for IVR. Then I try to download 0.9.0 version for both but there is missing VM module in Ser which is required to link to Sems. Please advice if I would like to use 0.9.0 version, without VM module how can link to SEMS and why in the sems version ser_rel_0_8_12 the IVR plugin is missing and where can I find it. Thanks in advance.... Regards, Nicky

2 1

[Serusers] Load balancer
by Kapil Dhawan 15 Sep '05

15 Sep '05

Dear List How can we implement a load balancer in SER. Regards _________________________________________________________________ Amitabh fan? Heres a chance to chat with the Big B himself. http://server1.msn.co.in/sp05/DJbk/index.asp Click here now!

2 1

RE : [Serusers] Handoff to PSTN
by Olivier Taylor 15 Sep '05

15 Sep '05

what's on the line 143??? -----Message d'origine----- De : Colin Jordan [mailto:cjordan@econetwireless.com] Envoyé : jeudi 15 septembre 2005 14:46 À : 'Olivier Taylor' Cc : serusers(a)lists.iptel.org Objet : RE: [Serusers] Handoff to PSTN The line 143 points to the script below. What modules will it be looking for? _____ From: Olivier Taylor [mailto:olivier.taylor@gmail.com] Sent: Thursday, September 15, 2005 2:33 PM To: 'Colin Jordan' Cc: serusers(a)lists.iptel.org Subject: RE : [Serusers] Handoff to PSTN please reply to all u have a module not loaded, the one involved on line 143 Olivier -----Message d'origine----- De : Colin Jordan [mailto:cjordan@econetwireless.com] Envoyé : jeudi 15 septembre 2005 14:28 À : 'Olivier Taylor' Objet : RE: [Serusers] Handoff to PSTN It gives me the following message: _______________________________________________________ Parse error (143,40-41): unknown command missing loadmodule? _______________________________________________________ _____ From: Olivier Taylor [mailto:olivier.taylor@gmail.com] Sent: Thursday, September 15, 2005 1:21 PM To: 'Colin Jordan'; serusers-request(a)lists.iptel.org Cc: serusers(a)lists.iptel.org Subject: RE : [Serusers] Handoff to PSTN just have a look at /var/log/messages u will find the line number where the error occurs Olivier -----Message d'origine----- De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de Colin Jordan Envoyé : jeudi 15 septembre 2005 12:33 À : serusers-request(a)lists.iptel.org Cc : serusers(a)lists.iptel.org Objet : [Serusers] Handoff to PSTN I am not able to restart ser if I put he code below in the ser.cfg to handoff calls to PSTN. # attempt handoff to PSTN if (uri=~^sip:9[0-9]*@mydomain.com) { log(Forwarding to PSTN\n); t_relay_to( 192.168.0.2, 5060); break; }; Please help!!!! DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful. DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful. DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

3 3

RE : [Serusers] Handoff to PSTN
by Olivier Taylor 15 Sep '05

15 Sep '05

please reply to all u have a module not loaded, the one involved on line 143 Olivier -----Message d'origine----- De : Colin Jordan [mailto:cjordan@econetwireless.com] Envoyé : jeudi 15 septembre 2005 14:28 À : 'Olivier Taylor' Objet : RE: [Serusers] Handoff to PSTN It gives me the following message: _______________________________________________________ Parse error (143,40-41): unknown command missing loadmodule? _______________________________________________________ _____ From: Olivier Taylor [mailto:olivier.taylor@gmail.com] Sent: Thursday, September 15, 2005 1:21 PM To: 'Colin Jordan'; serusers-request(a)lists.iptel.org Cc: serusers(a)lists.iptel.org Subject: RE : [Serusers] Handoff to PSTN just have a look at /var/log/messages u will find the line number where the error occurs Olivier -----Message d'origine----- De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de Colin Jordan Envoyé : jeudi 15 septembre 2005 12:33 À : serusers-request(a)lists.iptel.org Cc : serusers(a)lists.iptel.org Objet : [Serusers] Handoff to PSTN I am not able to restart ser if I put he code below in the ser.cfg to handoff calls to PSTN. # attempt handoff to PSTN if (uri=~^sip:9[0-9]*@mydomain.com) { log(Forwarding to PSTN\n); t_relay_to( 192.168.0.2, 5060); break; }; Please help!!!! DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful. DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

2 1

[Serusers] Handoff to PSTN
by Colin Jordan 15 Sep '05

15 Sep '05

I am not able to restart ser if I put he code below in the ser.cfg to handoff calls to PSTN. # attempt handoff to PSTN if (uri=~"^sip:9[0-9]*@mydomain.com") { log("Forwarding to PSTN\n"); t_relay_to( "192.168.0.2", "5060"); break; }; Please help!!!! DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

2 1

[Serusers] Handoff to PSTN
by Colin Jordan 15 Sep '05

15 Sep '05

I am not able to restart ser if I put he code below in the ser.cfg to handoff calls to PSTN. # attempt handoff to PSTN if (uri=~"^sip:9[0-9]*@mydomain.com") { log("Forwarding to PSTN\n"); t_relay_to( "192.168.0.2", "5060"); break; }; Please help!!!! DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

1 0

[Users] openser + osip
by Loide Mara 15 Sep '05

15 Sep '05

Hi, Did somebody implement a module to Openser that use libosip? I'm trying to implement it... Does somebody have some idea? Best regards -------------------------------------------------- Loide Mara Verdes de Jesus 6º período - Ciência da Computação / UFAM Laboratório VoIP - UFAM

3 4

[Serusers] SER with SEMS with Asterisk
by tdeillon＠cri74.org 15 Sep '05

15 Sep '05

Hello, I send this mail just to know if a configuration's documentation exist to configure them. Thanks, Thomas

1 0

[Serusers] codec restrict mediaproxy.
by Vikash Khatuwala 15 Sep '05

15 Sep '05

Hi, Does anyone know if its possible to restrict the codecs in mediaproxy. ie. to force only lower bandwidth codecs to reduce the load. I couldnt find any info on this on the search engines. Thanks. Vikash.

3 3

[Serusers] [about SER] mydomain is unresolveable
by 송 미진 15 Sep '05

15 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear Hi, serhelper. It’s my great pleasure to e-mail you. My name is Mi-Jin Song. I am a senior majoring in CS(Computer Science) at the Ewha Univ. in Korea. I just have a few questions about sip. First of all, I would like to explain about my situation. I am planning to make ‘testbed’ using 2 UA (User Agent)s and 3 sip servers, and I use linux fedora (version 3) as an operating system. In doing so, I have been using SER(Sip Express Router), sip open source provided by iptel.org, in order to use sip (session initiation protocol). When I was using sipsak, SER test tool made by your company, to check the network condition, but my computer did not seem to understand uri (sip:mi@sam.cse). For example, when I put root]#sipsak -T -s sip:mi@sam.cse I got an error message which is ‘sam.cse’ is unresolvable’ I made sure I put root]#export SIP_DOMAIN=”sam.cse” root]#serctl add mi 111 mi@sam. cse’ before. What I am guessing is that only using the order, export SIP _DOMAIN=“same. cse” does not do the uri setting. It’s been very hard since I could not find enough data and information regarding SER. I would like to know if you would be able to tell me what I should do to make the server recognize uri. If I need to fix something or use other methods additionally, could you please let me know? I will appreciate it very much if you could give an answer as soon as possible since the deadline for my project is coming very soon. Thank you so much. Sincerely, Mi-Jin Song. _________________________________________________________________ 보다 빠르고 보기 편한 뉴스. 오늘의 화제는 MSN 뉴스에서 확인하세요. http://www.msn.co.kr/news/

2 1

[Serusers] Cisco 2600
by Ryan Pagquil 15 Sep '05

15 Sep '05

Hi Guys, Have anyone on the list tried setting up Cisco 2600 as SIP gateway together with SER? Could you explain what will be the function of the SIP gateway and it's relation with SER. Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

3 5

[Serusers] Need sample incoming call acc data
by Eric Haskins 15 Sep '05

15 Sep '05

Hello, We are working on a billing solution for use with SER. We have completed a great PrePaid Billing solution for a client V1010.com but they currently only do outbound and arent planning to do inbound for a time. We are looking for some sample inbound call data from PSTN if anyone has some we would appreciate it! We can handle all Country codes and Area Code rates as well as custom rates as of now. So hope to have monthly billing completed soon. Thanks for any help Eric Haskins Lead Software Developer eric(a)rackspeed.net

1 0

[Serusers] Re: Iqbal maybe you can help
by Ali Zaidi 14 Sep '05

14 Sep '05

Hi, Thanks for reply, problem is fixed i reinstalled rtp from source, it seems that every element in place, only thing left is to put ser on public NOT in DMZ. I tried Mediaproxy before doing rtpproxy and i gave up. i will come back to mediaproxy some time in future. Thanks, Ali On 9/14/05, Iqbal <iqbal(a)gigo.co.uk> wrote: > > Hi > > Unfortunately I dont usertpproxy, I use mediaproxy, which I find to be > better. > > Iqbal > > Ali Zaidi wrote: > > > Hi, > > > > I have trouble with RTPPRoxy. > > I downloaded a everything.GettingStarted.tar.gz package from Onsip, > > hello-world.3.04.cfg and auth-mysql.3.05.cfg work but when i use > > nat-rtpproxy.3.04.cfg i get following errors. > > > > [root@localhost root]# /etc/init.d/rtpproxy start > > [root@localhost root]# /etc/init.d/ser restart > > Stopping ser: [FAILED] > > Starting ser: [ OK ] > > [root@localhost root]# serctl ul show > > Dumping all contacts may take long: are you sure you want to proceed? > > [Y|N] y > > Error opening ser's FIFO /tmp/ser_fifo > > Make sure you have line fifo=/tmp/ser_fifo in your config > > [root@localhost root]# > > > > > > By the way i don't see rtpproxy.sock under \var\run\ > > > > Questions: > > > > If i use rtpptoxy, how we start it? i do "/etc/init.d/rtpproxy start" > > then it doesn't show if that is started or not unless i run "ps -A" > > Can we put SER into DMZ and do port forwarding OR SER must have Puplic > IP? > > If we can put SER on DMZ then what other ports need to be open beside > > 5060? > > Do i need new version of RTPProxy? > > > > > > > > # =============================================== > > > > debug=3 > > fork=yes > > log_stderror=no > > > > listen= x.x.x.x # put your server IP address here That is MY WAN IP. > > 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/nathelper.so" > > loadmodule "/usr/local/lib/ser/modules/textops.so" > > loadmodule "/usr/local/lib/ser/modules/uri_db.so" > > loadmodule "/usr/local/lib/ser/modules/uri.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("nathelper", "natping_interval", 30) > > modparam("nathelper", "ping_nated_only", 1) > > modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") > > > > modparam("usrloc", "db_mode", 2) > > > > modparam("registrar", "nat_flag", 6) > > > > 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(); > > }; > > > > if (method=="BYE" || method=="CANCEL") { > > unforce_rtp_proxy(); > > } > > > > # > > ----------------------------------------------------------------- > > # 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(1); > > break; > > }; > > > > if (uri==myself) { > > > > if (method=="CANCEL") { > > route(3); > > break; > > } else 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 > > # > > ----------------------------------------------------------------- > > > > 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] { > > > > # > > ----------------------------------------------------------------- > > # CANCEL and INVITE Message Handler > > # > > ----------------------------------------------------------------- > > > > if (nat_uac_test("19")) { > > setflag(6); > > } > > > > lookup("aliases"); > > if (uri!=myself) { > > route(1); > > break; > > }; > > > > if (!lookup("location")) { > > sl_send_reply("404", "User Not Found"); > > break; > > }; > > > > if (!proxy_authorize("","subscriber")) { > > proxy_challenge("","0"); > > break; > > } else if (!check_from()) { > > sl_send_reply("403", "Use From=ID"); > > break; > > }; > > > > consume_credentials(); > > > > if (isflagset(6)) { > > force_rport(); > > fix_nated_contact(); > > force_rtp_proxy(); > > }; > > > > t_on_reply("1"); > > > > if (!t_relay()) { > > if(isflagset(6)) { > > 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(); > > }; > > } > > > > > > > > > > Thanks, > > > > > > Ali G. >

1 0

[Serusers] append_rpid_hf
by Sebastian Kühner 14 Sep '05

14 Sep '05

Hello, I tried to use the append_rpid_hf command. I read in the internet that the value is saved in the subscriber table in the column rpid. Is that all I have to do? I noticed that there's no information in the header (with ngrep). Thanks! Sebastian

2 5

RE: [Serusers] UAC module
by Pavol Segeč 14 Sep '05

14 Sep '05

For which OS platform is it compiled? ps > -----Original Message----- > From: Chris St Denis [mailto:chris@aebc.com] > Sent: Tuesday, September 13, 2005 7:18 PM > To: 'Pavol Segec'; serusers(a)lists.iptel.org > Cc: chris(a)aebc.com > Subject: RE: [Serusers] UAC module > > Somebody wrote a backport to 0.9.x. I don't remember where I > got it from, but I've attached it here. > > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Pavol Segec > Sent: Tuesday, September 13, 2005 8:39 AM > To: serusers(a)lists.iptel.org > Subject: [Serusers] UAC module > > Hi, > > I see that 0.9.3 and 0.9.4 do not contain UAC module (src > package neither), just as I see last CVS snapshot have it. > How can I add UAC module for 0.9.x server? > Is there some way? > > thanks > > pavol > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

2 1

[Serusers] serctl for postgres
by Joao Pereira 14 Sep '05

14 Sep '05

Hi I have a version of serctl ($Id: serpgctl,v 1.1 2003/04/08 01:25:35 lgfausak Exp $) that is getting the "400 ul_add: flags expected" error. Iqbal said that maybe thats not the serctl version for my SER. I m using ser-0.9.0 and I dont have a postgres version of serctl for this version of SER. Does someone has the serctl_postgres for ser-0.9.0 ? Thanks João

2 2

[Serusers] avpops
by Sebastian Kühner 14 Sep '05

14 Sep '05

Hello, Can anyone of you give me an example how to get data from a database with the avpops-module? I want to get a value from a special table and a special column... I didn't find any examples in the internet and the commands only don't help me very much... Many thanks! Sebastian

4 8

[Serusers] nathelper RTP Proxy config
by Daniel Corbe 14 Sep '05

14 Sep '05

Hello, Seeing as how SER can handle a hell of alot more traffic than rtpproxy, is there a way to set up multiple rtpproxy servers and then configure SER to use them in a load balancing/failover fashion? Here's my settings for nathelper currently: # -- nathelper params -- modparam("nathelper", "natping_interval", 10) modparam("registrar", "nat_flag", 6) modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("nathelper", "rtpproxy_sock", "udp:10.10.17.5:22222") Thanks! -Daniel

3 2

[Serusers] acc+mysql ser-0.9.4
by Szabolcs Szasz 14 Sep '05

14 Sep '05

Hi guys! I would like to upgrade my SER from 0.8.14 to 0.9.3 or 0.9.4 on my FC3 box. I have a problem, I can't start SER with acc module (segmentation fault). I have this problem in both 0.9.3 and 0.9.4 versions. This is the end of the debug messages (debug level:4) when I start my SER: ........... WARNING: no fork mode 0(32449) shm_mem_init: success 0(0) core dump limits set to 4294967295 0(0) DEBUG: init_mod: acc acc - initializing 0(0) find_export: found <load_tm> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <register_tmcb> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_newtran> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_relay_to_tcp> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_relay_to_udp> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_relay> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_reply> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_reply_with_body> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_is_local> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_get_trans_ident> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_lookup_ident> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_add_blind> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_forward_nonack> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_request_within> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_request_outside> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_request> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <new_dlg_uac> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <dlg_response_uac> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <new_dlg_uas> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <dlg_request_uas> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <free_dlg> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <print_dlg> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) find_export: found <t_gett> in module tm [/usr/local/lib/ser/modules/tm.so] Segmentation fault (core dumped)

2 2

[Serusers] Recommendations for a fast/flexible B2BUA
by Daniel Corbe 14 Sep '05

14 Sep '05

Anyone know of a good/fast/flexible B2BUA? I'm looking to do some LCR, Billing (prepaid, postpaid, credit accounts, etc), and just some general call supervision. I see that Vovida's b2bua hasn't seen a fresh release in over 2 years now and I'm wondering if anyone still has this deployed and if it still offers the kind of flexibility that would be required to build an LCR/Billing engine. Does anyone have asterisk deployed as a B2BUA? If so, is it fast and does it provide a decent amount of flexibility? Have you had to throw a session border controller in front of it? Any other recommendations on a B2BUA? Thanks. -Daniel

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Re: [Serusers] SIP Express Router now is commercial????
by Greger V. Teigre 14 Sep '05

14 Sep '05

[new post of previous summary of Tekelec acquisition after input from Daniel-Constantin Mierla on this list earlier today. This post can also be found at http://onsip.org/ ] AFAIK, Tekelec bought iptelorg.com with its intellectual property (commercial extensions/extras/related software to SER) and its employees. Iptel.org was NOT part of the deal and is still sponsored by FOKUS Fraunhofer (as stated on the iptel.org website). SER is an independent GPL project and Tekelec cannot do anything about that. However, iptelorg.com sponsored development resources on the open source SER. Tekelec is free to decide to stop this sponsoring. I cannot see how this can be in their interest, but if the focus on the development they want to do with SER is too far away the open source project, the project may not give Tekelec enough value. If this happens, the developers must decide whether to continue in their spare time, quit SER development, or look for employment with another company willing to sponsor hours on open source SER development. In addition, SER has MANY other developers who were not involved at all with iptelorg.com. Of course, as some of the most active coders were with iptelorg.com, such a decision may impact SER's development. Also, note that the OpenSER project has branched off with many main contributions from Voice System. The Voice System developers are not affected by the Tekelec deal. However, as OpenSER since openser has a published policy to stay compatible with ser and thus continously incorporate SER code into OpenSER, OpenSER will be equally affected (in terms of reduced code contribution) if those contributing this SER code are prevented from working on the code in their day-time job. Again, both SER and OpenSER have many non-iptelorg.com contributors to the code bases and the development will not halt.

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[Serusers] Mediaproxy
by Alberto 14 Sep '05

14 Sep '05

I have a doubt about Mediaproxy. I've read, if one o more SIP clients are behind NAT the traffic RTP, all the traffic, it will cross the server (ser and mediaproxy same server). I have verified it. Is There some way that is not thus? Or it is the only form to work the Mediaproxy (always to be and mediaproxy same server). Thanks, -- Alberto

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[Serusers] SIP Express Router now is commercial????
by carlos ivan grajales 14 Sep '05

14 Sep '05

Hi Guys, And now what Tekelec bought to iptelorg that it will be of SER??? your users list??? and GPL software ??? Sincerely, CArlos Ivan Grajales M. Colombia, South America --- serusers-request(a)lists.iptel.org wrote: > Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > > To subscribe or unsubscribe via the World Wide Web, > visit > http://lists.iptel.org/mailman/listinfo/serusers > or, via email, send a message with subject or body > 'help' to > serusers-request(a)lists.iptel.org > > You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > > When replying, please edit your Subject line so it > is more specific > than "Re: Contents of Serusers digest..." > > > Today's Topics: > > 1. I cannot compile "check_from()" method > although I load module > "uri" (tulay sazak) > 2. Compilation Error on Mac OS (Darwin) - SEMS.. > (Amos Nungu) > 3. call me (harry gaillac) > 4. SER + Mediaproxy sl_send_reply: I won't send a > reply for > ACK!! (GT) > 5. HELP! duplicate accounting (torres lloyd) > 6. Re: SER + Mediaproxy sl_send_reply: I won't > send a reply for > ACK!! (Greger V. Teigre) > 7. Re: Re: SER + Mediaproxy sl_send_reply: I > won't send a reply > for ACK!! (GT) > 8. Re: t_check_status in failure_route (Klaus > Darilion) > 9. Media proxy idle calls. (Vikash Khatuwala) > 10. Re: t_check_status in failure_route (Juha > Heinanen) > 11. Re: t_check_status in failure_route (Andreas > Granig) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun, 11 Sep 2005 04:12:08 -0700 (PDT) > From: tulay sazak <tulaysazak(a)yahoo.com> > Subject: [Serusers] I cannot compile "check_from()" > method although I > load module "uri" > To: serusers(a)lists.iptel.org > Message-ID: > <20050911111208.51148.qmail(a)web52108.mail.yahoo.com> > Content-Type: text/plain; charset=iso-8859-1 > > Hello everbody, > > I loaded module "uri" and my ser.cfg is as below: > > ..... > loadmodule "/usr/local/lib/ser/modules/uri.so" > ..... > if (method=="INVITE") { > if (!radius_proxy_authorize("iptel.org")) { > proxy_challenge("iptel.org", "1"); > break; > } > > if (!check_from()) { > sl_send_reply("403", "Forbidden"); > break; > } > } > ........ > > When I run ser, it says: > > 0(10457) find_export: <check_from> not found > 0(10457) find_export: <check_from> not found > 0(10457) parse error (183,35-36): unknown command, > missing loadmodule? > > Can you see what am I doing wrong ?? > > Thanks in advance > Regards > > Tulay > > > > > > > > ______________________________________________________ > > Yahoo! for Good > Watch the Hurricane Katrina Shelter From The Storm > concert > http://advision.webevents.yahoo.com/shelter > > > > ------------------------------ > > Message: 2 > Date: Sun, 11 Sep 2005 15:05:54 +0200 > From: Amos Nungu <amos.nungu(a)gmail.com> > Subject: [Serusers] Compilation Error on Mac OS > (Darwin) - SEMS.. > To: serusers(a)lists.iptel.org > Message-ID: > <86caff8805091106053263560b(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > I'm installing SEMS-0.9 on Mac OS (Darwin 8.2.0). > upon running the make > all command, I got an error as shown below > ------------------------------------------------- > # make all > make[1]: Nothing to be done for `deps'. > g++ -o sems AmApi.o AmAudio.o AmCmd.o AmConfig.o > AmEventQueue.o > AmIcmpWatcher.o AmMail.o AmPlugIn.o AmRequest.o > AmRtpPacket.o > AmRtpPacketTracer.o AmRtpScheduler.o AmRtpStream.o > AmSdp.o AmServer.o > AmSession.o AmSmtpClient.o AmThread.o AmUtils.o > EmailTemplate.o > SemsConfiguration.o SerClient.o SerDBQuery.o log.o > sems.o -lm -ldl > -lsocket -lnsl -lpthread > /usr/bin/ld: can't locate file for: -lsocket > collect2: ld returned 1 exit status > make[1]: *** [sems] Error 1 > make: [all] Error 2 (ignored) > ------------------------------------------------------- > > Any idea on how to go about it will be great. > > Regards > > > > ------------------------------ > > Message: 3 > Date: Sun, 11 Sep 2005 19:19:03 +0200 (CEST) > From: harry gaillac <gaillacharry(a)yahoo.fr> > Subject: [Serusers] call me > To: serusers(a)lists.iptel.org > Message-ID: > <20050911171904.17161.qmail(a)web25502.mail.ukl.yahoo.com> > Content-Type: text/plain; charset=iso-8859-1 > > hello, > > Please to call me at for tests. > > sip:84@nxs;yi.org > or > sip:85@nxs;yi.org > or > sip:86@nxs;yi.org > or > sip:87@nxs;yi.org > > Regards > Harry > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > ------------------------------ > > Message: 4 > Date: Sun, 11 Sep 2005 20:11:26 -0800 > From: "GT" <geotelcom(a)yahoo.fr> > Subject: [Serusers] SER + Mediaproxy sl_send_reply: > I won't send a > reply for ACK!! > To: serusers(a)lists.iptel.org <serusers(a)lists.iptel.org> > Message-ID: <20050911181312.D169319C6(a)fox.iptel.org> > Content-Type: text/plain; charset="iso-8859-1" > > Hello, > === message truncated === __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

5 5

[Serusers] Problem with command "serctl add"
by Alberto 14 Sep '05

14 Sep '05

Hi, After that I install SER and I run the command: serctl add 1000 12345 mil(a)domain.net I haven't any problem. But, when I restart the server and I run the command: serctl add 2000 54321 dosmil(a)domain.net I get the next problem: ERROR 1045 (28000): Access denied for user 'serro'@'localhost' (using password: YES) error: SER/FIFO not accessible: 2 I have tried the next: (but, I haven't resolved the problem) shell# mysql -p mysql> used ser; mysql> SET PASSWORD FOR 'serro'@'localhost' = PASSWORD('thepassword'); mysql> SET PASSWORD FOR 'ser'@'localhost' = PASSWORD('thepassword'); Some idea? Thanks, Alberto.

2 1

[Serusers] SER <==> SEMS
by Pavol Segeč 14 Sep '05

14 Sep '05

Hi, Ser, sems is leaving following error message in my suyslog ep 14 12:12:32 mail /usr/sbin/ser[23536]: Warning: run_failure_handlers: no UAC support (0, 256) Sep 14 12:12:32 mail Sems[23635]: Error: (AmRequest.cpp)(bye)(256): AmRequest::b ye: 481 Call/Transaction Does Not Exist What does it means? pavol

1 0

[Serusers] handle_subscription
by Bin Wang 14 Sep '05

14 Sep '05

hello, I'm using the methode handle_subscribtion for testing purposes. It's working, but the to-tag and from-tag in notify don't come up to our testing expectations: - to will be put in from and a tag will be generated - from will be put in to including tag but we need: the to- and from-tags in subscribe should be the from- and to-tags in notify, without generating a new tag or so - like shown above: Request-Line: SUBSCRIBE .... Message Header Via: ... ... To: sip:xyz@blabla.com; tag=b27e1a1d33761e... From: JP <sip:abc@blabla.com; tag=3c85d92779ea08e ... and notify should look like: Request-Line: NOTIFY .... Message Header via: ... ... To: JP <sip:abc@blabla.com; tag=3c85d92779ea08e From: sip:xyz@blabla.com; tag=b27e1a1d33761e... Is it possible to change the code and where do I have to change it? thanks, Bin -- GMX DSL = Maximale Leistung zum minimalen Preis! 2000 MB nur 2,99, Flatrate ab 4,99 Euro/Monat: http://www.gmx.net/de/go/dsl

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[Serusers] Sip EXpress router (0.9.4 (i386/linux))
by fedora＠voice.dyndns.tv 14 Sep '05

14 Sep '05

[cycle #: 13; if constant make sure server lives and fifo is on] Server: Sip EXpress router (0.9.4 (i386/linux)) Now: Wed Sep 14 17:26:25 2005 Up Since: Wed Sep 14 16:46:13 2005 Up time: 2412 [sec] Transaction Statistics Current: 0 (0 waiting) Total: 0 (0 local) Replied localy: 0 Completion status 6xx: 0, 5xx: 0, 4xx: 0, 3xx: 0,2xx: 0 Stateless Server Statistics 200: 97 202: 0 2xx: 0 300: 0 301: 0 302: 0 3xx: 0 400: 0 401: 35 403: 0 404: 0 407: 0 408: 0 483: 0 4xx: 0 500: 0 5xx: 0 6xx: 0 xxx: 132 failures: 0 UsrLoc Stats Domain Registered Expired 'location' 5 0 'aliases' 52 0 BYE sip:8991@221.186.150.67:5062 SIP/2.0 Via: SIP/2.0/UDP 221.186.150.68:5060;rport;branch=z9hG4bK350F835C78BE44D5AF06441318FB65AE From: 8000 <sip:8000@voice.dyndns.tv>;tag=4202623510 To: <sip:8991@voice.dyndns.tv>;tag=as4431c5d7 Contact: <sip:8000@221.186.150.68:5060> Route: <sip:221.186.150.67;ftag=4202623510;lr=on> Call-ID: 973AC72D-4AFD-4C5A-96BF-CFF6E27063CB(a)221.186.150.68 CSeq: 59653 BYE User-Agent: X-PRO build 1082 Content-Length: 0 BYE sip:8000@221.186.150.68:5060 SIP/2.0 Via: SIP/2.0/UDP 221.186.150.67:5062;branch=z9hG4bK7b866277;rport Route: <sip:8000@221.186.150.68:5060> From: <sip:8991@voice.dyndns.tv>;tag=as4431c5d7 To: 8000 <sip:8000@voice.dyndns.tv>;tag=4202623510 Contact: <sip:8991@221.186.150.67:5062> Call-ID: 973AC72D-4AFD-4C5A-96BF-CFF6E27063CB(a)221.186.150.68 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0

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[Serusers] First ENUM/SIP conference in the Netherlands
by Adrian Georgescu 14 Sep '05

14 Sep '05

For those interested: http://www.isoc.nl/activ/2005-ENUMSIP/ taking place in the Hague on the 14-th of October. High profile speakers at a fraction of cost compared to other similar conferences. Same week as RIPE 51 in Amsterdam, for those around it would be nice to meet there as well. Best regards, Adrian Georgescu

1 0

Re: [Serusers] No OK reply for REGISTER request ?
by Samuel Osorio Calvo 14 Sep '05

14 Sep '05

serctl ul show Samuel Unclassified. >>> Bhaskar Bhumkar <Bhaskar.Bhumkar(a)Sun.COM> 09/14/05 04:07AM >>> Thanks for the tip. It works - I received the 200 OK response for the REGISTER request. However, I can't see the new user name from the serweb tool (User Management screen) Is there any visual way to see the new user either by serweb or serctl? Chris St Denis wrote: > Probably the uri==myself isn't matching. > > You need to define aliases for yourself in the top of the config file. > alias="registrar.mydomain.com" > > -----Original Message----- > From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On > Behalf Of Bhaskar Bhumkar > Sent: Monday, September 12, 2005 6:12 PM > To: serusers(a)lists.iptel.org > Subject: [Serusers] No OK reply for REGISTER request ? > > Hi, > > I have set up a SER server and am testing it for REGISTER request. > I am sending following REGISTER request to the server : > > REGISTER sip:registrar.mydomain.com SIP/2.0 > Via: SIP/2.0/UDP bobspc.mydomain.com:5060;branch=z9hG4bKnashds7 > Max-Forwards: 70 > To: Bob <sip:bob@mydomain.com> > From: Bob <sip:bob@mydomain.com>;tag=456248 > Call-ID: 843817637684230@998sdasdh09 > CSeq: 1826 REGISTER > Contact: <sip:bob@192.0.2.4> > Expires: 7200 > Content-Length: 0 > > However, I am not getting OK reply from the server. How can I verify > user's registration at the SER? I tried to modify the ser.cfg; but > no success. Following is part of my ser.cfg : > > > ........ > if (uri==myself) { > if (method=="REGISTER") { > > > # Uncomment this if you want to use digest authentication and set > domainname > # if (!www_authorize("domainname", > "subscriber")) { > # www_challenge("domainname", "0"); > # break; > # }; > > save("location"); > break; > }; > > ........... > > > Any help would be appreciated on this. > > Thanks in advance. > > > > Regards, > Bhaskar... > > > _______________________________________________ > 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 _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] new stable release: ser-0.9.4-fc1.0 SOLVED !
by fedora＠voice.dyndns.tv 14 Sep '05

14 Sep '05

+ /usr/lib/rpm/find-debuginfo.sh /usr/src/redhat/BUILD/ser-0.9.4 extracting debug info from /var/tmp/ser-0.9.4-root/usr/sbin/ser extracting debug info from /var/tmp/ser-0.9.4-root/usr/sbin/gen_ha1 extracting debug info from /var/tmp/ser-0.9.4-root/usr/sbin/serunix extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/acc.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/auth.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/auth_db.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/auth_diameter.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/avp.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/avp_db.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/avpops.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/dbtext.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/dispatcher.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/diversion.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/domain.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/enum.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/exec.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/flatstore.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/gflags.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/group.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/lcr.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/mangler.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/maxfwd.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/mediaproxy.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/msilo.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/nathelper.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/options.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/pdt.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/permissions.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/pike.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/print.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/registrar.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/rr.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/sl.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/sms.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/speeddial.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/textops.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/tm.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/uac.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/uri.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/uri_db.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/usrloc.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/xlog.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/mysql.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/jabber.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/auth_radius.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/group_radius.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/uri_radius.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/cpl-c.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/pa.so extracting debug info from /var/tmp/ser-0.9.4-root/usr/lib/ser/modules/postgres.so -rw-r--r-- 1 root root 747155 9月 14 15:02 ser-0.9.4-fc1.0.i386.rpm -rw-r--r-- 1 root root 1921633 9月 14 15:02 ser-0.9.4-fc1.0.src.rpm -rw-r--r-- 1 root root 1914146 9月 14 15:10 ser-0.9.4_src.tar.gz -rw-r--r-- 1 root root 70821 9月 14 15:02 ser-cpl-0.9.4-fc1.0.i386.rpm -rw-r--r-- 1 root root 3611704 9月 14 15:03 ser-debuginfo-0.9.4-fc1.0.i38 6.rpm -rw-r--r-- 1 root root 64844 9月 14 15:02 ser-jabber-0.9.4-fc1.0.i386.r pm -rw-r--r-- 1 root root 114645 9月 14 15:02 ser-mysql-0.9.4-fc1.0.i386.rpm -rw-r--r-- 1 root root 53330 9月 14 15:02 ser-pa-0.9.4-fc1.0.i386.rpm -rw-r--r-- 1 root root 20923 9月 14 15:02 ser-postgres-0.9.4-fc1.0.i386 .rpm -rw-r--r-- 1 root root 21675 9月 14 15:02 ser-radius-0.9.4-fc1.0.i386.r pm -rw-r--r-- 1 root root 9356 9月 14 15:09 ser.spec.fedora-0.9.4

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[Serusers] No OK reply for REGISTER request ?
by Bhaskar Bhumkar 14 Sep '05

14 Sep '05

Hi, I have set up a SER server and am testing it for REGISTER request. I am sending following REGISTER request to the server : REGISTER sip:registrar.mydomain.com SIP/2.0 Via: SIP/2.0/UDP bobspc.mydomain.com:5060;branch=z9hG4bKnashds7 Max-Forwards: 70 To: Bob <sip:bob@mydomain.com> From: Bob <sip:bob@mydomain.com>;tag=456248 Call-ID: 843817637684230@998sdasdh09 CSeq: 1826 REGISTER Contact: <sip:bob@192.0.2.4> Expires: 7200 Content-Length: 0 However, I am not getting OK reply from the server. How can I verify user's registration at the SER? I tried to modify the ser.cfg; but no success. Following is part of my ser.cfg : ........ if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication and set domainname # if (!www_authorize("domainname", "subscriber")) { # www_challenge("domainname", "0"); # break; # }; save("location"); break; }; ........... Any help would be appreciated on this. Thanks in advance. Regards, Bhaskar...

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