sr-users October 2008

sr-users@lists.kamailio.org

93 participants
170 discussions

[Kamailio-Users] presence module, events
by toly 14 Oct '08

14 Oct '08

Greetings, This is a question to the author: bogdan_iancu, but if anybody can answer, I'd appreciate it. Softphone client is Bria. Here is publish event from ngrep PUBLISH sip:tolyp@webley SIP/2.0. Via: SIP/2.0/TCP 10.50.1.101:45464;branch=z9hG4bK- T 2008/10/13 16:15:38.621949 10.50.1.101:1572 -> 10.50.4.40:5070 [AP] d8754z-697cf1172b7bf515-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:tolyp@10.50.1.101:1572;transport=TCP>. To: "tolyp"<sip:tolyp@webley>. From: "tolyp"<sip:tolyp@webley>;tag=ca65f37d. Call-ID: YmFlNjUxMzEwM2Q4Nzk0MWRhZGUxZmQwMDA5NmE4Nzc.. CSeq: 1 PUBLISH. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/pidf+xml. User-Agent: Bria release 2.4 stamp 49274. Event: presence. Content-Length: 441. . <?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' xmlns:lt='urn:ietf:params:xml:ns:location-type' entity='sip:tolyp@webley'><tuple id='t5a055764'><status><basic>open</basic></status></tuple><dm:person id='p5e347748'><dm:note>Available</dm:note></dm:person></presence> Here is the code from parser/parse_event.c if ((_e->text.len == PRES_STR_LEN) && !strncasecmp(PRES_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_PRESENCE; } else if ((_e->text.len == PRES_XCAP_CHANGE_STR_LEN) && !strncasecmp(PRES_XCAP_CHANGE_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_XCAP_CHANGE; } else if ((_e->text.len == PRES_WINFO_STR_LEN) && !strncasecmp(PRES_WINFO_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_PRESENCE_WINFO; } else if ((_e->text.len == PRES_SIP_PROFILE_STR_LEN) && !strncasecmp(PRES_SIP_PROFILE_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_SIP_PROFILE; } else if ((_e->text.len == DIALOG_STR_LEN) && !strncasecmp(DIALOG_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_DIALOG; } else if ((_e->text.len == MWI_STR_LEN) && !strncasecmp(MWI_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_MWI; } else { _e->parsed = EVENT_OTHER; } if( (*end)== ';') { str params_str; params_str.s= end+1; end= skip_token(params_str.s, tmp.len- _e->text.len- 1); params_str.len= end- params_str.s; if (parse_params(&params_str, CLASS_ANY, &phooks, &_e->params)<0) return -1; } else _e->params= NULL; It detects presence event correctly, but then if( (*end)== ';') does not find any semicolon and here is the result: 10/13 16:34:57 13030 err ERROR:presence:handle_publish: <publish.c: 508> Missing or unsupported event header field value 10/13 16:34:57 13030 err ERROR:presence:handle_publish: <publish.c: 511> event=[presence] What semicolon it is looking? If I do (*end)== ' ') it still can't parse. Is Bria message has incorrect format? What was the format of message the parser was written for? Thanks a lot, Toly -- View this message in context: http://www.nabble.com/presence-module%2C-events-tp19963494p19963494.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] How to handle DoS attack and OpenSER Pike module!
by Arif-Uz-Zaman 14 Oct '08

14 Oct '08

Hi all, I need to bother about crazy client by considering "Flood" detection technique. I can do it by using OpenSER Pike <http://kamailio.org/docs/modules/1.2.x/pike.html> module which helps to keep trace of all (or selected ones) incoming request's IP source and blocks the ones that exceeded some limit. In my case: If the number of SIP messages from a single IP address to my SIP Proxy exceeds 200 per minute. Recommended action: Block IP for 2 hours. I tried with the pike module but I'm little bit confused with sampling, density, and timeout value. Please help me with example configuration by considering my point. Thanks, ARIF

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[Serusers] ser-0.9.6: modules/postgres bugreport & patch
by Alexandre Snarskii 13 Oct '08

13 Oct '08

Hi! I'm running ser-0.9.6, on FreeBSD 6.1-stable, database backend is postgresql version 8.1.3. Today I got errors in logfiles, saying: messages.2.bz2:May 29 14:35:03 <XXX> /usr/local/sbin/ser[51448]: ERROR:avpops:dbrow2avp: dbrow contains NULL fields The similar problem reported in: http://lists.iptel.org/pipermail/serusers/2005-May/019681.html with much more detailed description of error, database contents and config samples. Patch is trivial, and looks more like a fix to copy'n'paste error: in mysql/val.c function str2val states: if (!_s) { memset(_v, 0, sizeof(db_val_t)); VAL_TYPE(_v) = _t; VAL_NULL(_v) = 1; return 0; } VAL_NULL(_v) = 0; and the last line mentions that 'well, that's value is not NULL'. In postgresql/db_val.c, line 182, function str2valp, the same statement is the: if (!_s) { DLOG("str2valp", "got a null value"); VAL_TYPE(_v) = _t; VAL_NULL(_v) = 1; return 0; } without explicit notification that this is not-NULL value. More than, nowhere else in this function VAL_NULL(_v) not set to 0. So, if a value _v.val contained anyting but 0 at the function start, resulting value will be threated as NULL despite the fact, that _s is NOT NULL. Patch is obvious, just add VAL_NULL(_v)=0; after cited block (line 188) and everyting will be ok. At least for me it's ok for some hours :)

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[Serusers] installing CDRTool on Freebsd
by Mohit Saxena 13 Oct '08

13 Oct '08

Hello Friends, Can any one help me installing the CDR TOOL to integrate with freeradius on freebsd? Any help will be highly appreciated. Thanks Mohit Mohit C. Saxena Sr. Network Administrator Starcomms Plc. www.starcomms.com <blocked::http://www.starcomms.com> Mob: +234-702-800-0709 DISCLAIMER: The information contained in this message (including any attachments) is confidential and may be privileged. If you have received it by mistake please notify the sender by return e-mail and permanently delete this message and any attachments from your system. Any form of dissemination, use, review, distribution, printing or copying of this message in whole or in part is strictly prohibited if you are not the intended recipient of this e-mail. Please note that e-mails are susceptible to change. STARCOMMS PLC shall not be liable for the improper or incomplete transmission of the information contained in this communication nor for any delay in its receipt or damage to your system. STARCOMMS PLC does not guarantee that the integrity of this communication has been maintained or that this communication is free of viruses, interceptions or interferences. STARCOMMS PLC reserves the right to monitor all e-mail communications, whether related to the business of STARCOMMS or not, through its internal or external networks.

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[Kamailio-Users] fr_anything_at_all_timer.
by Alex Balashov 13 Oct '08

13 Oct '08

It would be really nice to have a feature in TM for even faster failover that is turned off when any reply is received at all, provisional or otherwise. The problem with fr_timer and fr_inv_timer is that they throw on lack of final reply. But sometimes gateways have very lengthy, complicated asynchronous operations that must complete before they can return a final OK. If you are doing failover, the most important thing to know is that the gateway is there at all, which the provisional 1xx message accomplishes. What would be really nice is to have a timer that can be diffused as soon as a provisional reply arrives. I don't see how this can be accomplished with any of the existing timers. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Serusers] CDRTool error Table 'cdrtool.radacct200810' doesn't exist
by satish patel 13 Oct '08

13 Oct '08

Hi I have installed CDRTool and i got error on CDR search entry like this Table 'cdrtool.radacct200810' doesn't exist i have read on goolge about it and people tell the problem was if radacct200810 table not in radius database. i have also run rotatetable.php script to rotate database but i got error like this #php /var/www/CDRTool/scripts/SER/rotateTables.php rotateTable(radacct200810,200809,radacct200809) Error: cannot locate mysql creation file i am using 6.6.10 kindly suggest me.. $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

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[Kamailio-Users] problem with relaying invite
by Asim Riaz 13 Oct '08

13 Oct '08

Hi, I am relaying invites from Kamilio(1.2) to a third proxy, i am receivng Server error occured in SIP messages, I am seeing these messages in log file; Asterisk--------->Kamilio----------------->third-proxy route{ openser.cfg if(method=="INVITE"){ route(4); exit; } route[4] { if(!t_relay("udp:third_proxy_1:5060") || !t_relay("udp:third_proxy_2:5060")) { end_media_session(); sl_reply_error(); } } DEBUG: RFC3261 transaction matching failed DEBUG: t_lookup_request: no transaction found ERROR:tm:t_forward_nonack: no branch for forwarding ERROR:tm:w_t_relay: t_forward_nonack failed DEBUG:sl:sl_reply_error: error text is Server error occurred (1/SL) any idea ? Thanks Asim

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[Kamailio-Users] what is the best solution NAT problem
by Ricky Gutierrez 13 Oct '08

13 Oct '08

Hi list, I am trying to solve my nat problem with openser 1.3.2 and rtpproxy 1.1, I have my openser with 2 net cards. I detail my scenario: (192.168.10.1) LAN-eth1- Server Openser eth0-WAN (192.168.1.64)<-> NAT <-> ADSL dyndns <-> Internet <-> ADSL <-> NAT <-> UAC All my external clients are also behind an ADSL with address private ip and my server openser, I don't have ip it public but register a domain with dyndns and it configures it in my router adsl, I have access from out to my server through dyndns. All my external clients configure them so that they use of address proxy the dyndns domain, my external clients register but when they call to another UAC that this behind the server openser is not audio, and if a UAC that this behind the server openser calls an external client the call doesn't arrive he fails. I have open the ports UDP 5060:5065, 10000:20000, 35000:65000 TCP: 5060 some idea of like I can solve this problem?... best regards all list rickygm ### LOG SIP ### U +1.744617 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK2fc9.6c704556.0 Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-de71d299 Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3> From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 214 v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 18 101 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000481 192.168.1.64:5060 -> 190.184.35.4:5063 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-de71d299 Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3> From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 232 P-hint: onreply_route|force_rtp_proxy v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35006 RTP/AVP 18 101 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=nortpproxy:yes # U +0.219110 190.184.35.4:5063 -> 192.168.1.64:5060 ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.60:5063;branch=z9hG4bK-547056da From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 102 ACK Max-Forwards: 70 Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3>, <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Proxy-Authorization: Digest username="122",realm="gnuforever.homelinux.com",nonce="48f260b8529fd8e9d8f0247fa92f734a317f2da5",uri="sip:113@gnuforever.homelinux.com",algorithm=MD5,response="ec45ed24126b924160da73b8ba10e73d" Contact: <sip:122@192.168.0.60:5063> User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.000944 192.168.10.1:5060 -> 192.168.10.30:5062 ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK2fc9.6c704556.2 Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-547056da From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 102 ACK Max-Forwards: 69 Proxy-Authorization: Digest username="122",realm="gnuforever.homelinux.com",nonce="48f260b8529fd8e9d8f0247fa92f734a317f2da5",uri="sip:113@gnuforever.homelinux.com",algorithm=MD5,response="ec45ed24126b924160da73b8ba10e73d" Contact: <sip:122@190.184.35.4:5063;nat=yes> User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.140447 192.168.1.64:5060 -> 190.184.35.4:5063 OPTIONS sip:190.184.35.4:5063 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.64:5060;branch=0 From: sip:pinger@192.168.1.64;tag=f7508b75 To: sip:190.184.35.4:5063 Call-ID: 29b7fe71-06901b84-fa6(a)192.168.1.64 CSeq: 1 OPTIONS Content-Length: 0 # U +0.164881 190.184.35.4:5063 -> 192.168.1.64:5060 SIP/2.0 404 Not Found To: sip:190.184.35.4:5063;tag=a04ea06caeb3256i3 From: sip:pinger@192.168.1.64;tag=f7508b75 Call-ID: 29b7fe71-06901b84-fa6(a)192.168.1.64 CSeq: 1 OPTIONS Via: SIP/2.0/UDP 192.168.1.64:5060;branch=0 Server: Linksys/SPA942-5.2.8 Content-Length: 0 # U +1.195009 192.168.10.28:5060 -> 192.168.10.1:5060 # U +0.639899 192.168.10.1:5060 -> 192.168.10.27:5060 OPTIONS sip:192.168.10.27:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5060;branch=0 From: sip:pinger@192.168.1.64;tag=08508b75 To: sip:192.168.10.27:5060 Call-ID: 29b7fe71-16901b84-1b6(a)192.168.10.1 CSeq: 1 OPTIONS Content-Length: 0 # U +0.001412 192.168.10.27:5060 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1:5060;branch=0 From: sip:pinger@192.168.1.64;tag=08508b75 To: sip:192.168.10.27:5060;tag=b1e56101b3b09b53 Call-ID: 29b7fe71-16901b84-1b6(a)192.168.10.1 CSeq: 1 OPTIONS User-Agent: Grandstream GXV3000 1.1.3.14 Contact: <sip:120@192.168.10.27:5060> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel, path Content-Length: 0 # U +3.393457 190.184.35.4:5063 -> 192.168.1.64:5060 BYE sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.60:5063;branch=z9hG4bK-f7539b7 From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 103 BYE Max-Forwards: 70 Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3>, <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Proxy-Authorization: Digest username="122",realm="gnuforever.homelinux.com",nonce="48f260b8529fd8e9d8f0247fa92f734a317f2da5",uri="sip:113@192.168.10.30:5062",algorithm=MD5,response="4416adbfacebc76ed9cc3f002ff958a1" User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.000768 192.168.10.1:5060 -> 192.168.10.30:5062 BYE sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK3fc9.1cba8107.0 Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-f7539b7 From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 103 BYE Max-Forwards: 69 Proxy-Authorization: Digest username="122",realm="gnuforever.homelinux.com",nonce="48f260b8529fd8e9d8f0247fa92f734a317f2da5",uri="sip:113@192.168.10.30:5062",algorithm=MD5,response="4416adbfacebc76ed9cc3f002ff958a1" User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.055972 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK3fc9.1cba8107.0 Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-f7539b7 Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3> From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 103 BYE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0

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Re: [Kamailio-Users] No audio both ways
by luzango mfupe 13 Oct '08

13 Oct '08

Hi Castillo,I did post my config file showing how i used the Mediaproxy. However i would like to get your opinion on how to use/configure it properly. cheers, Lu. > Message: 3 > Date: Sun, 12 Oct 2008 23:45:35 +0200 > From: I?aki Baz Castillo <ibc(a)aliax.net> > Subject: Re: [Kamailio-Users] No audio both ways > To: users(a)lists.kamailio.org > Message-ID: <200810122345.35371.ibc(a)aliax.net> > Content-Type: text/plain; charset="utf-8" > > El Domingo, 12 de Octubre de 2008, luzango mfupe escribi?: > > I solved the problem by disabling use_mediaproxy() call and did put > nat=yes > > in the Asterisk side. > > This means that you didn't use MediaProxy correctly. If it's properly > configured and the proxy logic script calls use_mediaproxy() when needed, > then you don't need to enable the Comedia mode in Asterisk (nat=yes). > > -- > I?aki Baz Castillo > > > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > End of Users Digest, Vol 41, Issue 33 > ************************************* > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

2 1

Re: [Kamailio-Users] No audio both ways
by luzango mfupe 12 Oct '08

12 Oct '08

Hi mates, I solved the problem by disabling use_mediaproxy() call and did put nat=yesin the Asterisk side. Many thanks to Daniel and David. Cheers, Lu. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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sr-users October 2008