- sr-users - kamailio.org

Re: [Serusers] Xlite not working with SER+SEMS
by Pat wang 08 Aug '05

08 Aug '05

Hi Samuel, Thanks for the reply. This is exactly the problem. Stefen also point this out. I got it running now. Cheers, Pat >From: "Samuel Osorio Calvo" <samuel.osorio(a)nl.thalesgroup.com> >To: <wangyu39(a)hotmail.com>, <serusers(a)lists.iptel.org> >Subject: Re: [Serusers] Xlite not working with SER+SEMS >Date: Mon, 08 Aug 2005 10:59:10 +0200 > >Try to start SEMS with a user different from root. This way, the ICMP >watcher does not run and the RTP session will not be closed due to >non-compliant end-points. > >Samuel. > > >Unclassified. > >>> "yu wang" <wangyu39(a)hotmail.com> 08/02/05 07:19AM >>> >Hi, > >I've been working on conference with Xlite but it is not work. The problem >is SEMS starts sending media (default wav) back to Xlite once SEMS receives >the INVITE/SDP from Xlite. It seems a bug on Xlite not listening on the >media port after sending the INVITE thus it replys with ICMP port >unreachable. But I don't see anyone complain this on this list. Is >something >wrong with my configuration? > >Please please help me.... Thanks, > >Pat > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers >

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Re: Tr: Re: [Serusers] hardphones MOH
by harry gaillac 08 Aug '05

08 Aug '05

hello Maxim , I'm waiting your reply Harry --- Maxim Sobolev <sobomax(a)portaone.com> a écrit : > Yes, it will. I have this functionality in my > private repo, bit > unfortunately due to lack of free time I have not > merged support for > this feature into the public version of nathelper > yet. We have some > heavy changes that aren't applicable for the public > version, which has > to be filtered out. > > I am planning to do it sooner or later, maybe if you > are really > interested and can allocate some small amount of > money to this project I > can subcontract some of my friends/colleagues to do > it for me. > > -Maxim > > harry gaillac wrote: > > Hello, > > > > Does nathelper module + rtpproxy will be able to > ac as > > a 3cpp to provide MOH > > > > caller--------ser---------callee > > nathelper > > rtpproxy > > Regards > > Harry > > Remarque : message transféré en pièce jointe. > > > > > > > > > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > > > > ------------------------------------------------------------------------ > > > > Subject: > > Re: [Serusers] hardphones MOH > > From: > > harry gaillac <gaillacharry(a)yahoo.fr> > > Date: > > Thu, 30 Jun 2005 23:01:32 +0200 (CEST) > > To: > > Nils Ohlmeier <lists(a)ohlmeier.org> > > > > To: > > Nils Ohlmeier <lists(a)ohlmeier.org> > > CC: > > serusers(a)lists.iptel.org > > > > > > What do you advise me ? > > > > I wish to use SER+Sems(ivr) to provide MOH. > > I've got two polycom ip300 however we just can > place > > on hold caller without music ! > > > > What can I do? what's the way? > > Does Ser or sems must act as 3cpp ? > > > > Regards > > Harry > > --- Nils Ohlmeier <lists(a)ohlmeier.org> a écrit : > > > > > >>snom UA's do, except that they spare the empty > offer > >>to the MOH server, but > >>send directly the SDP of the holdee to the MOH > >>server and expecting that the > >>server supports the chosen codec. > >>Beware that this 3pcc MOH alternative completely > >>breaks if there is any NAT > >>involved. > >> > >> Nils > >> > >>On Thursday 30 June 2005 22:29, harry gaillac > wrote: > >> > >>>hello, > >>> > >>>Anybody would be informed about ip phones able to > >>>provide moh according to > >>>http://www.tech-invite.com/Ti-sip-service-3.html > >>> > >>>regards > >>>harry > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >> > > > ___________________________________________________________________________ > > > >>>Appel audio GRATUIT partout dans le monde avec le > >> > >>nouveau Yahoo! Messenger > >> > >>>Téléchargez cette version sur > >> > >>http://fr.messenger.yahoo.com > >> > >>>_______________________________________________ > >>>Serusers mailing list > >>>serusers(a)lists.iptel.org > >>>http://lists.iptel.org/mailman/listinfo/serusers > >> > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] Issue with call forwarding
by Dave 08 Aug '05

08 Aug '05

Hello,I have implemented CFNR (Call forward on No Response) and CFU(Call forwardinf unconditional) using avp and I am running into the following problem: When CFU or CFNR are detected, the control is passed to route[2] (see below). CFU works fine and the call is forwarded to it's new destination. But CFNR does not work and the UA making the call gets a "Proxy Authentication Required" message from SER. Upon receiving this message, my UA gives me a congestion tone. This is strange since CFU works fine but in the case of CFNR, I see using ngrep that SER makes DNS queries to resolve my SIP domain (dns=no has been configured) The code is: route[2] { if (is_uri_host_local()) { if (!lookup("location")) { if (does_uri_exist()) { sl_send_reply("480", "Temporarily Unavailable"); break; } else { sl_send_reply("404", "User Not Found"); break; } ; } else { route(1); <-- this simply does t_relay() break; }; } else { sl_send_reply("302", "Temporarily Redirected"); break; }; } CFNR code is: ------------------------------------------- failure_route[1] { if (isflagset(22) && t_check_status("408")) { if (avp_pushto("$ruri", "s:cfnr")) { append_branch(); avp_delete("s:cfnr"); resetflag(22); route(2); break; }; }; } -------------------------------------------- Any pointers will be highly appreciated. Thanks ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

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[Serusers] Config Errors
by zkeatts 08 Aug '05

08 Aug '05

I am getting the following config file error when I try to run the xml-rpc branch version of SER on a Red Hat 3.4.3-9 box. 0(20635) parse error (99,14-15): unknown command, missing loadmodule? 0(20635) parse error (100,16-17): unknown command, missing loadmodule? Can anyone indicate to me what might be the problem? The following are the steps I took to compile the package. 1) Compile and Install xmlrpc-c libraries. 2) $ make proper include_modules="mysql" $ make $ make modules include_modules="mysql" $ make install include_modules="mysql" 3) $ /usr/local/sbin/ser the ser.cfg file I am using is the quickstart ser.cfg Can anyone give me some insight into these errors? Thanks. Zach Keatts Software Engineer Nuvio.com

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[Serusers] serweb
by marc doucen 08 Aug '05

08 Aug '05

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[Serusers] adding groups
by harry gaillac 08 Aug '05

08 Aug '05

Hello, I want to add groups however for users however if i want to add new group not voicemail ld or local i receive: "Invalid privilege: mailbox ignored" So I have to add my group by hand in table grp. How can i solve this problem? Harry serveur1:~/asterisk# serctl acl grant bob mailbox MySql password: Invalid privilege: mailbox ignored +----------+------------+-----------+---------------------+ | username | domain | grp | last_modified | +----------+------------+-----------+---------------------+ | bob | nxs.yi.org | cellphone | 2005-08-08 14:11:00 | | bob | nxs.yi.org | local | 2005-08-08 14:14:52 | | bob | nxs.yi.org | voicemail | 2005-08-08 14:16:54 | +----------+------------+-----------+---------------------+ ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] ser radius accounting
by Dimitar Mojsovski 08 Aug '05

08 Aug '05

Hi all, I'm using ser and freeradius, and so far i have managed to set up the authentication, but I'm still having problems with the radius accounting. I've disabled sql and enabled radius acc module Makefile and compiled it. It loads with no problems, here's how i do it in my config : loadmodule "//lib/ser/modules/acc.so" modparam("acc", "log_level", 2) modparam("acc", "log_flag", 1) modparam("acc", "radius_config","/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "service_type", 15) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 3) and then I have : if (uri==myself) { if (method=="INVITE") { # mark the INVITEs for accounting setflag(1); # enforce record-routing so the BYEs will come through this server record_route(); }; if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { log(1, "REGISTER: challenging user\n"); www_challenge("", "0"); break; }; setflag(1); 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; }; }; Ser starts and runs with no errors, however when my sip users make calls I see no acc log in radacct table on the radius server. Am I looking at the right place or am I doing something wrong? -- Dimitar Mojsovski Macedonia On-Line Service Provider www.mol.com.mk

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[Users] avpops tutorial updated
by Daniel-Constantin Mierla 08 Aug '05

08 Aug '05

Hello, the AVPOPS tutorial has been updated to the latest development version: http://www.voice-system.ro/docs/avpops/ Among presentations of the new methods (avp_op(), avp_subst(), avp_printf() ...), it includes a new example of how the newest AVPOPS capabilities allow to store and use bitmap ACL per user. The old version, valid for 0.9.x, is now located at: http://www.voice-system.ro/docs/avpops/0.9.0/ Daniel

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[Serusers] radius auth, can't login with a linksys pap2-na
by Simone Cittadini 08 Aug '05

08 Aug '05

I can't have the lines of a linksys pap authenticate on ser, but only if the backend is radius, seems it doesn't like the uri: <sip:172.18.1.12> part of the SIP request, with Xlite that part reads uri: <sip:2klab.net> and radius authenticates ok. with a mysql backend for auth both uri authenticates well, so I think I'm missing something in radius configuration. (I've followed the RADIUS_howto on iptel.org) from ser.cfg : if (uri==myself) { if (method=="REGISTER") { if (!radius_www_authorize("2klab.net")) { www_challenge("2klab.net", "0"); break; }; save("location"); break; }; lookup("aliases"); # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; This is the log for a succesful register with mysql auth backend : 6(1885) SIP Request: 6(1885) method: <REGISTER> 6(1885) uri: <sip:172.18.1.12> 6(1885) version: <SIP/2.0> 6(1885) parse_headers: flags=1 6(1885) Found param type 232, <branch> = <z9hG4bK-1741b415>; state=16 6(1885) end of header reached, state=5 6(1885) parse_headers: Via found, flags=1 6(1885) parse_headers: this is the first via 6(1885) After parse_msg... 6(1885) preparing to run routing scripts... 6(1885) parse_headers: flags=128 6(1885) end of header reached, state=9 6(1885) DEBUG: get_hdr_field: <To> [29]; uri=[sip:2002@172.18.1.12] 6(1885) DEBUG: to body [2002 <sip:2002@172.18.1.12> 6(1885) get_hdr_field: cseq <CSeq>: <2> <REGISTER> 6(1885) DEBUG:maxfwd:is_maxfwd_present: value = 70 6(1885) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 6(1885) grep_sock_info - checking if host==us: 11==11 && [172.18.1.12] == [172.18.1.12] 6(1885) grep_sock_info - checking if port 5060 matches port 5060 6(1885) parse_headers: flags=4096 6(1885) check_nonce(): comparing [42f6ef654b9c6efc04d7e468746d818f8e1633ab] and [42f6ef654b9c6efc04d7e468746d818f8e1633ab] 6(1885) HA1 string calculated: 9b7a9697157ad673ed341c1aca7c12c5 6(1885) check_response(): Our result = '1f77523c69a80ae921d5e05d8a5e0fcd' 6(1885) check_response(): Authorization is OK 6(1885) save_rpid(): rpid value is '' 6(1885) parse_headers: flags=-1 6(1885) DEBUG: get_hdr_body : content_length=0 6(1885) found end of header 6(1885) parse_headers: flags=33554432 6(1885) build_contact(): Created Contact HF: Contact: <sip:2002@172.18.1.13:5061>;expires=3600 6(1885) parse_headers: flags=-1 6(1885) check_via_address(172.18.1.13, 172.18.1.13, 0) 6(1885) DEBUG:destroy_avp_list: destroying list (nil) 6(1885) receive_msg: cleaning up And this one for an unsuccesful register from the same pap client, but radius backend : 9(2042) SIP Request: 9(2042) method: <REGISTER> 9(2042) uri: <sip:172.18.1.12> 9(2042) version: <SIP/2.0> 9(2042) parse_headers: flags=1 9(2042) Found param type 232, <branch> = <z9hG4bK-80868dab>; state=16 9(2042) end of header reached, state=5 9(2042) parse_headers: Via found, flags=1 9(2042) parse_headers: this is the first via 9(2042) After parse_msg... 9(2042) preparing to run routing scripts... 9(2042) parse_headers: flags=128 9(2042) end of header reached, state=9 9(2042) DEBUG: get_hdr_field: <To> [29]; uri=[sip:2002@172.18.1.12] 9(2042) DEBUG: to body [2002 <sip:2002@172.18.1.12> 9(2042) get_hdr_field: cseq <CSeq>: <3> <REGISTER> 9(2042) DEBUG:maxfwd:is_maxfwd_present: value = 70 9(2042) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 9(2042) grep_sock_info - checking if host==us: 11==11 && [172.18.1.12] == [172.18.1.12] 9(2042) grep_sock_info - checking if port 5060 matches port 5060 9(2042) parse_headers: flags=4096 9(2042) check_nonce(): comparing [42f6f2f3e9dd51cad68be2927d5863727cd3798a] and [42f6f2f3e9dd51cad68be2927d5863727cd3798a] 9(2042) authorize(): Credentials realm and URI host do not match 9(2042) build_auth_hf(): 'WWW-Authenticate: Digest realm="2klab.net", nonce="42f6f2f3e9dd51cad68be2927d5863727cd3798a" 9(2042) parse_headers: flags=-1 9(2042) DEBUG: get_hdr_body : content_length=0 9(2042) found end of header 9(2042) check_via_address(172.18.1.13, 172.18.1.13, 0) 9(2042) DEBUG:destroy_avp_list: destroying list (nil) 9(2042) receive_msg: cleaning up seems Xlite sends my domain as uri, and radius likes it : 4(2034) SIP Request: 4(2034) method: <REGISTER> 4(2034) uri: <sip:2klab.net> 4(2034) version: <SIP/2.0> 4(2034) parse_headers: flags=1 4(2034) Found param type 235, <rport> = <n/a>; state=6 4(2034) Found param type 232, <branch> = <z9hG4bK461FD1B046D3D5FB1FE0012188DF1BB8>; state=16 4(2034) end of header reached, state=5 4(2034) parse_headers: Via found, flags=1 4(2034) parse_headers: this is the first via 4(2034) After parse_msg... 4(2034) preparing to run routing scripts... 4(2034) parse_headers: flags=128 4(2034) end of header reached, state=9 4(2034) DEBUG: get_hdr_field: <To> [29]; uri=[sip:user1@2klab.net] 4(2034) DEBUG: to body [user1 <sip:user1@2klab.net> 4(2034) get_hdr_field: cseq <CSeq>: <2352> <REGISTER> 4(2034) DEBUG:maxfwd:is_maxfwd_present: value = 70 4(2034) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 4(2034) grep_sock_info - checking if host==us: 9==11 && [2klab.net] == [172.18.1.12] 4(2034) grep_sock_info - checking if port 5060 matches port 5060 4(2034) grep_sock_info - checking if host==us: 9==11 && [2klab.net] == [172.18.1.12] 4(2034) grep_sock_info - checking if port 5060 matches port 5060 4(2034) check_nonce(): comparing [42f6f204483c60785f0ffd667859ed55b34fad23] and [42f6f204483c60785f0ffd667859ed55b34fad23] 4(2034) radius_authorize_sterman(): Success 4(2034) save_rpid(): rpid value is '' 4(2034) parse_headers: flags=-1 4(2034) DEBUG: get_hdr_body : content_length=0 4(2034) found end of header 4(2034) parse_headers: flags=33554432 4(2034) build_contact(): Created Contact HF: Contact: <sip:user1@172.18.1.10:5060>;expires=1800 4(2034) parse_headers: flags=-1 4(2034) check_via_address(172.18.1.10, 172.18.1.10, 0) 4(2034) DEBUG:destroy_avp_list: destroying list (nil) 4(2034) receive_msg: cleaning up

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Re: [Serusers] Sending FQDN in VIA Header
by Samuel Osorio Calvo 08 Aug '05

08 Aug '05

I think Juha already answered you...I'll try to expand his answer, see inline Hope it helps, Samuel. Unclassified. >>> ranveer kunal <ranveerkunal(a)gmail.com> 08/08/05 08:07AM >>> Hi frnds, Is there a way to configure SER so that it sends the domainname in via header ? [SAM] I don't know....take a look at advertised_ip and advertised_port (check spelling because I'm not sure about it..) and they might do the trick. Why is SER not responding to OPTIONS sent to it, it throws an error "404 not found" ? [SAM] Probably you do not handle SIP OPTIONS message properly in your config file, and it is instead treated as an INVITE. You have to add something like if (method==OPTIONS){ options_reply(); break; } There are several parameters to configure what SER is sending back with options_reply(). Take a look at options module. modparam("options", "support", "100rel") modparam("options", "accept_language", "de") modparam("options", "accept_encoding", "gzip") modparam("options", "accept", "application/*") Regards, Ranveer. -- Memories : They bring Diamonds and Rust _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] acc_request
by Pavol Segeč 08 Aug '05

08 Aug '05

Hi all! I tried to set up missed call reporting through the Admin guide v0.11.0, but for action acc_request ser -c reports parse error. Is the action part of the the acc.so module anymore? palo

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[Serusers] Setting tm module parameter "fr_inv_timer_avp" more than once during a call
by reticent 08 Aug '05

08 Aug '05

It seems that setting the "fr_inv_timer_avp" parameter more than once during a call is not possible. However there is a situation where this is necessary: 1. VoIP ATA to VoIP ATA call with call forwarding where call forwarded number is a PSTN destination To facilitate Call Forward No Answer, the fr_inv_timer_avp is used and set to a value (say 12 seconds), after this timer hits a failure_block is run, in this failure block i rewrite the destination and check if this destination is a PSTN number. If it is than I would like to change the fr_inv_timer_avp parameter to somthing higher due to the nature of PSTN termination but it does not seem to have any effect. Has anyone experienced this problem and possible found a workaround? Or maybe this is a known issue tavis

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[Serusers] sms module/ gsm_auth
by harry gaillac 08 Aug '05

08 Aug '05

hello, How can we send/received sms and calls from gsm gateway see http://www.2n.cz/products/gsm_gateways/voip/voiceblue_enterprise.html Does sms module and gsm_auth are out of date ? harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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Re: [Serusers] Xlite not working with SER+SEMS
by Samuel Osorio Calvo 08 Aug '05

08 Aug '05

Try to start SEMS with a user different from root. This way, the ICMP watcher does not run and the RTP session will not be closed due to non-compliant end-points. Samuel. Unclassified. >>> "yu wang" <wangyu39(a)hotmail.com> 08/02/05 07:19AM >>> Hi, I've been working on conference with Xlite but it is not work. The problem is SEMS starts sending media (default wav) back to Xlite once SEMS receives the INVITE/SDP from Xlite. It seems a bug on Xlite not listening on the media port after sending the INVITE thus it replys with ICMP port unreachable. But I don't see anyone complain this on this list. Is something wrong with my configuration? Please please help me.... Thanks, Pat _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] voicemail module
by harry gaillac 08 Aug '05

08 Aug '05

Hello, If we receive a call from any users who are not registred in subscriber table does voicemail module is able to set caller address in mail header ? For example : caller ----PSTN--------ser+sems------callee unavailable 123456 DB sip:callee@mydomain Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] delay in dialing or registration
by Chia Huey Lim 08 Aug '05

08 Aug '05

Hi list, I face a serious problem where sip packets are not processed immediately, as in when I dial a number, it takes a while to ring to the other party, on the system log, I can see the packets coming in a lot later than it should. However sometimes it is immediate, but it's not been consistent. Registration also same thing, sometimes fast, sometimes have to wait for long time. I am running postgres db, sip registration and sip proxy in the same server. It's has single XEON proc, 2 GB RAM, with 2MBps burstable to 10MBps to the Internet. In my ser.cfg file, I m using db mode 2 for usrloc, with children not set, which should be 8 children if according to serctl ps status I currently have around 600 registered user only, with expiry values of 1200s or 3600s. Do I need to readjust the priority for the ser processes? Will that help? Any other area I should look into? Thanks in advance for your advice.. Regards, Chia

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[Serusers] problem in starting mysql module in ser !
by zhu 08 Aug '05

08 Aug '05

hello, guys: I try to use freeradius and ser. when i start ser, the following messages came out and reload mysql three times and finally stop at mydql module: get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) find_mod_export: found <db_use_table> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_init> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_close> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_query> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_raw_query> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_free_result> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_insert> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_delete> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(0) find_mod_export: found <db_update> in module mysql [/usr/local/lib/ser/modules/mysql.so] any firend knows how to solve it. thanks lot! zhu

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[Serusers] missed_calls shows IP for domain
by John Clements 07 Aug '05

07 Aug '05

I am running ser 0.9.3 and serweb. When I go to Missed Calls, it shows that I have none, but the missed_calls table shows a bunch of entries. In all of them, domain is set to the IP of the SER server, instead of the domain that I used to add the user. Anyone know why and how to get it to save the domain in missed_calls instead of the IP? -John

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[Serusers] Proxy
by Anderson Alves de Albuquerque 07 Aug '05

07 Aug '05

How cani I could use SER as proxy?

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Re: Tr: Re: [Serusers] hardphones MOH
by harry gaillac 07 Aug '05

07 Aug '05

Hello, Sorry I should not this post here. I 've been waiting for a reply for a long time . I agree to pay for it if you can you explain me how nathelper/rtpproxy provide MOH ? Regards Harry --- Maxim Sobolev <sobomax(a)portaone.com> a écrit : > Yes, it will. I have this functionality in my > private repo, bit > unfortunately due to lack of free time I have not > merged support for > this feature into the public version of nathelper > yet. We have some > heavy changes that aren't applicable for the public > version, which has > to be filtered out. > > I am planning to do it sooner or later, maybe if you > are really > interested and can allocate some small amount of > money to this project I > can subcontract some of my friends/colleagues to do > it for me. > > -Maxim > > harry gaillac wrote: > > Hello, > > > > Does nathelper module + rtpproxy will be able to > ac as > > a 3cpp to provide MOH > > > > caller--------ser---------callee > > nathelper > > rtpproxy > > Regards > > Harry > > Remarque : message transféré en pièce jointe. > > > > > > > > > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > > > > ------------------------------------------------------------------------ > > > > Subject: > > Re: [Serusers] hardphones MOH > > From: > > harry gaillac <gaillacharry(a)yahoo.fr> > > Date: > > Thu, 30 Jun 2005 23:01:32 +0200 (CEST) > > To: > > Nils Ohlmeier <lists(a)ohlmeier.org> > > > > To: > > Nils Ohlmeier <lists(a)ohlmeier.org> > > CC: > > serusers(a)lists.iptel.org > > > > > > What do you advise me ? > > > > I wish to use SER+Sems(ivr) to provide MOH. > > I've got two polycom ip300 however we just can > place > > on hold caller without music ! > > > > What can I do? what's the way? > > Does Ser or sems must act as 3cpp ? > > > > Regards > > Harry > > --- Nils Ohlmeier <lists(a)ohlmeier.org> a écrit : > > > > > >>snom UA's do, except that they spare the empty > offer > >>to the MOH server, but > >>send directly the SDP of the holdee to the MOH > >>server and expecting that the > >>server supports the chosen codec. > >>Beware that this 3pcc MOH alternative completely > >>breaks if there is any NAT > >>involved. > >> > >> Nils > >> > >>On Thursday 30 June 2005 22:29, harry gaillac > wrote: > >> > >>>hello, > >>> > >>>Anybody would be informed about ip phones able to > >>>provide moh according to > >>>http://www.tech-invite.com/Ti-sip-service-3.html > >>> > >>>regards > >>>harry > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >> > > > ___________________________________________________________________________ > > > >>>Appel audio GRATUIT partout dans le monde avec le > >> > >>nouveau Yahoo! Messenger > >> > >>>Téléchargez cette version sur > >> > >>http://fr.messenger.yahoo.com > >> > >>>_______________________________________________ > >>>Serusers mailing list > >>>serusers(a)lists.iptel.org > >>>http://lists.iptel.org/mailman/listinfo/serusers > >> > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] accounting
by harry gaillac 06 Aug '05

06 Aug '05

Hello, I use ser/serweb-0.9.3 I added tow accounts with numerical aliases. When I dial user1@domain to user2@domain it's ok. if i wish to forward to voicemail sems send me back voicemessages. However if i dial alias1@domain to alias2@domain I get the message below. How can I solve this problem ? Harry /////////////////////////////////////////////////////// Jul 24 14:58:15 serveur1 Sems[2535]: Error: (AmSession.cpp)(startSession)(497): 404 voicemail: no email address for user <84> Jul 24 14:58:15 serveur1 /usr/sbin/ser[7495]: ACC: call missed: method=INVITE, i-uri=sip:84@nxs.yi.org:5060;user=phone, o-uri=sip:h.gaillac@192.168.0.20, call_id=66219bb6-f9eef79c-bc2479d3(a)192.168.0.21, from="test" <sip:test@nxs.yi.org>;tag=CB323432-D0A509E1, code=404 voicemail: no email address for user <84>, userpart=84 /////////////////////////////////////////////////////// ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

5 9

[Serusers] Xten With SER
by Marcelo Vasconcelos 06 Aug '05

06 Aug '05

Hi all, i´m a new user of ser and i belive that ser is working properly, but i can´t login with X-LITE i only have unauthorized message. Please Help me. Tanks Marcelo Vasconcelos

1 0

[Users] Changing 'expires' value in response to REGISTER
by Kanakatti Mahesh Subramanya 06 Aug '05

06 Aug '05

And the beat goes on... I 've got a bunch of Snom's that have a minimum register interval of 1 minute. I *know* that I can get them to register every 30 seconds if I set the 'expires' value in the Contact header of the '200 OK' response to the REGISTER request appropriately (whew) e.g., instead of returning Contact: <sip:jake.foobaz@1.2.3.4:49154>;expires=60 I return Contact: <sip:jake.foobaz@1.2.3.4:49154>;expires=30 The question is - how do I change the 'expires' value? append_to_reply() isn't relevant - i don't want another header subst() on the request's 'Expires' header isn't useful since build_contact() seems to extract it's values from the original request. Any changes I made to it seem to be ignored Any ideas? cheers

4 6

RE: [Serusers] Open source Legal Issue
by Mr Greg Plater 06 Aug '05

06 Aug '05

I say because alterations and configuring was done from many peoples works on lists and forums, the work is already published and is covered by the GPL and if any is not published it must be publish under the GPL. I don't see how they can deny me or any person working under the GPL the right to use alter or distribute the applications again from the original source code. As a contractor I believe if any source code is developed its owned by the contractor and he is obligated under the license he developed it the GPL for example. If the person is an Employee then the code developed would be owned by the company Employing the staff member and the company would be obligated under the license that applies example GPL. I believe the outcome of this injunction places developers in the position that once they develop something under the GPL then the company it was developed for then has the right to say they can never use it again. I don't believe the GPL was intended to allow this. -----Original Message----- From: Iqbal [mailto:iqbal@gigo.co.uk] Sent: Saturday, 6 August 2005 9:55 AM To: Mr Greg Plater Subject: Re: [Serusers] Open source Legal Issue Also On one hand they dont want you to disclose, however on the othe hand they accepted GPL, if they argue GPL, and that you must handover, then the handover is to the public in general, hence you will be disclosing, which they do not want, they are contradicting themselves. Iqbal Mr Greg Plater wrote: > 5th Aug 2005 an order was handed down from a supreme court judge in > Brisbane Australia at 10am. > > Injunction restraining the defendant his servants or agents from > directly using, disclosing, selling, transferring or in any other > manner dealing with the software written or developed or configured by > the defendant relating to the VOIP network proposed by the plaintiff > including any correspondence, notes , records or other documentation > relating to such software, and any variations on the same. > > The complete solution was based on open source applications and they > received a copy of the GPL before the project commenced and had been > given access to the source code used. I am unable to disclose the open > source applications used. > > I note i was required to shutdown the servers and the platform used > yesterday including private projects because the same base linux > version was used. > > The issue started when i was contracted to create a network based on > open source applications, the party contracting myself decided to drop > the project and items in the agreement between them and i are still > outstanding including payment and this is an on going battle in the > courts. > > Until that is all settled I'm not required to hand over the hardware > or applications, source code, notes etc... but under the license GPL i > am required to provide it so i have provided the source code etc > including all alterations made and there source code... This allowing > myself to stay within the license GPL. > > I don't believe they are able to give the injunction and need advice > on what or if anything i can do. > > My understanding of Open source. It is free to use alter or distribute > under the GPL and its conditions. > >------------------------------------------------------------------------ > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

[Serusers] Open source Legal Issue
by Mr Greg Plater 05 Aug '05

05 Aug '05

5th Aug 2005 an order was handed down from a supreme court judge in Brisbane Australia at 10am. Injunction restraining the defendant his servants or agents from directly using, disclosing, selling, transferring or in any other manner dealing with the software written or developed or configured by the defendant relating to the VOIP network proposed by the plaintiff including any correspondence, notes , records or other documentation relating to such software, and any variations on the same. The complete solution was based on open source applications and they received a copy of the GPL before the project commenced and had been given access to the source code used. I am unable to disclose the open source applications used. I note i was required to shutdown the servers and the platform used yesterday including private projects because the same base linux version was used. The issue started when i was contracted to create a network based on open source applications, the party contracting myself decided to drop the project and items in the agreement between them and i are still outstanding including payment and this is an on going battle in the courts. Until that is all settled I'm not required to hand over the hardware or applications, source code, notes etc... but under the license GPL i am required to provide it so i have provided the source code etc including all alterations made and there source code... This allowing myself to stay within the license GPL. I don't believe they are able to give the injunction and need advice on what or if anything i can do. My understanding of Open source. It is free to use alter or distribute under the GPL and its conditions.

1 0

[Serusers] SIP messages timeout
by Sebastian Kühner 05 Aug '05

05 Aug '05

Hello, I'm with accounting in ser and I noticed, that in some calls the BYE message doesn't reach my server. In the most cases yes, but in 5% of the cases not. Is there any way to force the caller/callee to send the BYE message back? Thanks!! Sebastian

4 4

[Serusers] How to test SIP with TCP?
by Pat wang 05 Aug '05

05 Aug '05

Hi, I understand both Xlite (or any other SIP client such as a Cisco 7960) and SER support TCP but I am not clear how to force a UAC starts a call via TCP. I've been searching over internet but couldn't find anything yet (Forgive my bad searching skills). I am really appreciate if anyone can show me some information of testing SER and UAC with TCP. Thanks, Have a nice weekend, Pat

2 2

[Users] feature request documentation
by Klaus Darilion 05 Aug '05

05 Aug '05

Hi! It would be nice to have a list of all functions inside openser (core + modules). Is it possible to autmatically generate this? e.g. for the core parsing the lex files, for the module functions parsing the sgml files regards, klaus

2 3

[Serusers] Routing
by Sebastian Kühner 05 Aug '05

05 Aug '05

Hi! Has anyone of you tried to use the database to make routing? I need to get information of the database and then rewritehost with the result... Thanks! Sebastian

2 2

[Serusers] Mediaproxy doesnt resolve private IP to public IP
by Ashutosh Kumar 05 Aug '05

05 Aug '05

Hi, I m using mediaproxy for NAT traversal for client phones. I m using the following code straight out of book from ser-0.9.0 source package, but still the natted clients are showing their internal IPs in their AOR when I run a "serctl ul show ". Can anyone tell me where is the problem- 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 (client_nat_test("3")) {setflag(2);force_rport();fix_contact();}; if (!www_authorize("", "subscriber")) {www_challenge("", "0");break;} if (!save("location")) {sl_reply_error();}; 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();}; if (loose_route()) { if (method=="INVITE" || method=="ACK") {use_media_proxy();}; t_relay(); break; }; # Force subsequent messages to pass trough this proxy if (method == "INVITE") {record_route();}; if (client_nat_test("3") && !search("^Record-Route:")) {force_rport();fix_contact(); }; if (method=="INVITE") { t_on_reply("1"); }; if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; 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 (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; } Regards, Ashutosh Kumar Chetu, Inc.

1 0

Re: [Serusers] Logs in mysql
by Iqbal 05 Aug '05

05 Aug '05

Hi Just to get this straight, you want to add logging to Mysql, if yes, have to edited the Makefile, in the /modules/acc/ directory, by default it allows logging with Radius, you need to comment out that line, and uncomment the mysql line. If this is done, then if you follow the docs in onsip.org, you will get logging into mysql. Also post a copy of ur ser.cfg file, so that we can take a look Iqbal Hernando Maisano wrote: > Thank you very much, it lowers east file to me: > http://www.onsip.org/modules/mydownloads/singlefile.php?cid=7&lid=26 > and raises the "SER" but it gives errors me, and I had q to remove a > pair from line. But what I want is to add log to base MYSQL.... and > there is no case. I am using RED HAT 7,3 with SER 0.8.12 > HELPP! > > Nando > ----- Original Message ----- From: "Iqbal" <iqbal(a)gigo.co.uk> > To: "Hernando Maisano" <nando(a)cps.com.ar> > Cc: <serusers(a)lists.iptel.org> > Sent: Friday, August 05, 2005 12:52 PM > Subject: Re: [Serusers] Logs in mysql > > >> goto www.onsip.org, there is a gettng started guide with mysql and >> all, it will be of use >> >> Iqbal >> >> Hernando Maisano wrote: >> >>> Hello, they recien I am beginning in the "SER" and the truth I am >>> lost. What I need to do now is to leave a registry of the calls, but >>> in MYSQL? Somebody can help me..... Thanks >>> PS: pardon my pesimo Ingles but I am of Latinoamerica >>> Nando >>> >>> ------------------------------------------------------------------------ >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> > > > . >

2 3

[Serusers] Logs in mysql
by Hernando Maisano 05 Aug '05

05 Aug '05

Hello, they recien I am beginning in the "SER" and the truth I am lost. What I need to do now is to leave a registry of the calls, but in MYSQL? Somebody can help me..... Thanks PS: pardon my pesimo Ingles but I am of Latinoamerica Nando

2 2

[Users] PSTN Interconnect Advice
by Adam Sherman 05 Aug '05

05 Aug '05

I'd like some advice. What are people doing for PSTN connectivity when using third-party gateways? - For outbound, lcr with the uac module for authentication seems to be the way. - For inbound, either getting the providers to route DIDs directly using SRV or using sipsak to REGISTER seems to be a possibility. For DIDs, just using aliases to push the call to a user? Thanks, A. -- Adam Sherman Technologist

3 6

[Serusers] ERROR: fifo_server fgets failed: Illegal seek
by Ashutosh Kumar 05 Aug '05

05 Aug '05

Hi, I am a newbie to this user group and would like to have some urgent assistance regarding the following error. I am running ser-0.8.14 with nathelper and mediaproxy. The problem I am facing is that after I disconnect my first call, or cancel a RING, or upon a BYE, the following message is encountered and SER crashes. --------- 0(3970) ERROR: build_res_buf_from_sip_req: out of memory ; needs 745 0(3971) ERROR: sl_reply_error used: I'm terribly sorry, server error occured (1/SL) 0(3972) check_via_address(61.16.236.34, 192.168.11.37, 0) 2(3975) ERROR: fifo_server fgets failed: Illegal seek -------------- Any help will be highly appreciated Regards, Ashutosh

2 1

[Serusers] [Partially OffTopic] - Tekelec acquisition of Iptelorg...
by Pietro Ravasio 05 Aug '05

05 Aug '05

Hi all, Sorry for posting an offtopic message, but, considering the discussion that went on during the openser/ser development forking, what should we (ser users) expect after Tekelc acquisition of iptelorg? Cfr.: http://www.tekelec.com/news/prdetail.asp?prnum=543 Regards, Pietro -- Ing. Pietro Ravasio - pietro.ravasio(a)abbeynet.it Enterprise Market Manager Abbeynet S.p.A. - http://www.abbeynet.it TEL: +390702109044 GSM: +393282172055 FAX: +390702109055

3 2

[Users] Determining if User Exists
by Adam Sherman 05 Aug '05

05 Aug '05

In order to forward calls to voicemail when the user is not registered, I need to check if the user exists in the subscriber database. How can I do that? Thanks, A. -- Adam Sherman Technologist

2 5

[Serusers] SER installation on OpenBSD 3.7
by Fulvio Picecchi 05 Aug '05

05 Aug '05

Hi all, I'm a newbie and I'm having trouble with the installation of SER 0.8.10 package on OpenBSD 3.7. The package is the one available here: http://www.openbsd.org/3.7_packages/i386.html After a pkg_add ftp://ftp.openbsd.org/pub/OpenBSD/3.7/packages/i386/ser-0.8.10.tgz I've copied the example ser.cfg to /etc/ser.cfg and then I've tried a simple: /usr/local/sbin/ser But this is the output: /usr/local/sbin/ser:/usr/local/lib/ser/modules/sl.so: undefined symbol 'mem_lock' /usr/local/sbin/ser: /usr/local/lib/ser/modules/sl.so: can't resolve reference 'mem_lock' /usr/local/sbin/ser:/usr/local/lib/ser/modules/sl.so: undefined symbol 'process_no' /usr/local/sbin/ser: /usr/local/lib/ser/modules/sl.so: can't resolve reference 'process_no' /usr/local/sbin/ser:/usr/local/lib/ser/modules/sl.so: undefined symbol 'mem_block' ... ... ... Aug 5 17:16:33 zeus ser: parse error (95,36-37): unknown command, missing loadmodule? Aug 5 17:16:33 zeus ser: parse error (95,36-37): unknown command, missing loadmodule? Aug 5 17:16:33 zeus ser: parse error (100,15-16): unknown command, missing loadmodule? Aug 5 17:16:33 zeus ser: parse error (100,15-16): unknown command, missing loadmodule? Aug 5 17:16:33 zeus ser: parse error (101,18-19): unknown command, missing loadmodule? Aug 5 17:16:33 zeus ser: parse error (101,18-19): unknown command, missing loadmodule? ERROR: bad config file (16 errors) Thanks for any suggestion Fulvio

1 0

[Serusers] CLIR/CLIP
by Anders Hansson 05 Aug '05

05 Aug '05

Hi, Does anyone know if is possible to use CLIR/CLIP using SER connected to PSTN without help from other instance? To present the incoming number on the phone connected to SER is no problem, neither when calling internal numbers. But when calling eternal PSTN numbers the IP telephone number is not showing on the terminating side. Anyone got any idea or explanation? Thanks, Anders

2 1

Re: [Serusers] RTPProxy issues with G729 at different frame lengths
by Noel Sharpe 05 Aug '05

05 Aug '05

Andres wrote: > Noel Sharpe wrote: > > By the way, if you are using Asterisk as your gateway, there is no way > to modify G729 frame size via a config file. It is fixed at 20ms. > You should use 20ms on all your UAs. If you need to hardcode it to > something else, I can lookup my notes and let you know. > > Cheers > I'm using Asterisk as a b2bua, so I guess that is where the problem is. It does however work when the frames are > 20ms - infact I get the best quality when I use 40ms. I will post to the Asterisk list and let the list know what I find out. Thanks.

1 0

RE: [Serusers] ser start up
by Sam Lee 05 Aug '05

05 Aug '05

Are u using Digest Auth ? ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Chia Huey Lim Sent: Friday, August 05, 2005 4:13 PM To: serusers(a)lists.iptel.org Subject: [Serusers] ser start up Hi, I have a situation where the SER is only doing registration and proxy, where else ASTERISK is handling all the call switching behind (UA to UA as well as UA to pstn) I am combining registration and proxy to be handled by one SER, only handling around 650 users at this stage. However when I reboot the server and start SER, it takes around 10 minutes just for binding user and then handle registration, during that period, no UA can make any outgoing call though they are already registered. After the registration is settling down, only then UA can make outgoing call. And after that, periodically, there will be a time where the SER is very busy with registration and no outgoing call can be made again... Most of the UA expiry value is 20 minutes. Is that normal? Or which area I shall look into? Is it advisable to separate registration and proxy into 2 SER? Thanks in advance for the advice. Regards, Chia

2 1

[Users] Stability, and -DF_MALLOC
by Kanakatti M. Subramanya 05 Aug '05

05 Aug '05

I'm running 0.9.5, on my system, and *have* to comment out -DF_MALLOC (and uncomment -DDBG_QM_MALLOC) to have the system be stable. It pretty much always crashes when trying to do save("location") Anyone else seen this? List of modules is loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/exec.so" loadmodule "/usr/local/lib/openser/modules/group.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/options.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/lcr.so"

3 5

[Serusers] ser start up
by Chia Huey Lim 05 Aug '05

05 Aug '05

Hi, I have a situation where the SER is only doing registration and proxy, where else ASTERISK is handling all the call switching behind (UA to UA as well as UA to pstn) I am combining registration and proxy to be handled by one SER, only handling around 650 users at this stage. However when I reboot the server and start SER, it takes around 10 minutes just for binding user and then handle registration, during that period, no UA can make any outgoing call though they are already registered. After the registration is settling down, only then UA can make outgoing call. And after that, periodically, there will be a time where the SER is very busy with registration and no outgoing call can be made again. Most of the UA expiry value is 20 minutes. Is that normal? Or which area I shall look into? Is it advisable to separate registration and proxy into 2 SER? Thanks in advance for the advice. Regards, Chia

1 0

[Serusers] Why BYE doesn't accounted?
by Ryan Pagquil 05 Aug '05

05 Aug '05

Guys, Iv'e configured my SER to authenticate with freeradius, well thanks everybody. Yesterday I tried to make mysql accounting and it worked but the problem is when I hungup I can't see the BYE on the accounting, I only see INVITE and ACK. What should be the problem with my ser.cfg? Here is the configuration specifically with mysql accounting: modparam("acc", "log_flag", 1) modparam("acc", "log_level", 1) modparam("acc", "report_ack", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "failed_transactions", 1) modparam("acc", "report_cancels", 1) modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) <<other stuff>> if (method=="INVITE" || method=="BYE") record_route(); setflag(1); <<other stuff>> Please help. 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] Problem with SER and PAP2, I get a 407 message when trying a call
by Sam Lee 05 Aug '05

05 Aug '05

What exactly is the problem? I got a similar set up.... Regards, Sam -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Yiannis Martis Sent: Thursday, August 04, 2005 4:11 PM To: jiri(a)iptel.org; sisalem(a)iptel.org; serusers(a)lists.iptel.org Subject: [Serusers] Problem with SER and PAP2,I get a 407 message when trying a call Hello there, I am Network Administrator working for a company called Hellasfon Network S.A. I use SER 0.9.3 with Vovida's STUN server. PAP2 UAis NATed and gets registered using my ADSL modem's IP and everything looks OK. The problem is when I try to make a call to another UA) of my service I get the following output from my server (ngrep):. Please help me if u can!!! U 2005/08/03 13:55:57.380953 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 INVITE sip:2810318308@XXXXXXXXXX.com SIP/2.0. Record-Route: <sip:XX.XX.XX.XX;ftag=7ab32f2bbf2689feo0;lr=on>. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport=5060. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 INVITE. Max-Forwards: 16. Contact: <sip:2810300308@XX.XX.XX.XX:5060>. Expires: 240. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 420. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 13486 13486 IN IP4 XX.XX.XX.XX. s=-. c=IN IP4 XX.XX.XX.XX. t=0 0. m=audio 16420 RTP/AVP 18 0 2 4 8 96 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/8000. 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 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2005/08/03 13:55:57.381212 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport=5060. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a17 9. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="XXXXXXXXXX.com", nonce="42f0a3e9ea677d46038bdbeccaf8d95c18bce58e". Server: Sip EXpress router (0.9.3 (i386/linux)). Content-Length: 0. Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=4360 req_src_ip=XX.XX.XX.XX req_src_port=5060 in_uri=sip:2810318308@XXXXXXXXXX.com out_uri=sip:2810318308@XXXXXXXXXX.com via_cnt==2". . # U 2005/08/03 13:55:57.381353 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 ACK sip:2810318308@XXXXXXXXXX.com SIP/2.0. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. Call-ID: b521746f-5119658a(a)192.168.1.2. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a17 9. CSeq: 102 ACK. User-Agent: Sip EXpress router(0.9.3 (i386/linux)). Content-Length: 0. . # U 2005/08/03 13:55:57.664069 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 ACK sip:2810318308@XXXXXXXXXX.com SIP/2.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a17 9. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 ACK. Max-Forwards: 70. Proxy-Authorization: Digest username="2810300308",realm="XXXXXXXXXX.com",nonce="42f0a3e8f78926d70014 0a7c490f06dd965ea748",uri="sip:2810318308@XXXXXXXXXX.com",algorithm=MD5, response="ac73134a0de530d527cfbe4fe1903707". Contact: <sip:2810300308@XX.XX.XX.XX:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] lookup 2 uri
by Richard Z 05 Aug '05

05 Aug '05

The issue is that the locationis not always the same. I will try to fool around the register message and save function to see if there is a way to make it work. Richard On 8/3/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > If you know in advance the IP of 1002 and it is static, you can use serctl to add it's contact. This way doing a lookup("location") for 1003 will return both contacts...it's not a nice solution but it whould work. > > Samuel. > > Unclassified. > >>> Richard Z <rzheng(a)gmail.com> 08/03/05 08:55PM >>> > Hi Samuel, > > That's a very good detailed explanation! That's exactly what happened. > In my existing config, I have the logic to deal with the second ring > to 1003(a)test.com. It forks to itself and creates another ser > transaction. I'd like to make it simpler by doing it in a single > transaction. > > One way I am thinking about, but haven't tried yet, is when 1003 > registers, ser adds its location to both 1002 and 1003. So when ser > does lookup("location") for 1002, the real location for 1003 shows up. > Not sure that it is feasible. > > Thanks, > Richard > > > > On 8/2/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > > Just one remark: only one of the two URI will be translated to the contact. > > > > I'll try to explain... > > > > imagine the incoming Req-URI is 1001(a)test.com. When it passes the first lookup, > > lookup("aliases"); > > SER will retrieve the two aliases and put one of them as the new req-uri and the other as branch. > > Req-URI: 1002(a)test.com > > branch: 1003(a)test.com > > Doing the second lookup, > > lookup("location"); > > will put the contact of 1002(a)test.com as Req-URI (if more than one contact is present, the subsequent bindings will be added as branches) and the 1003(a)test.com will be left as branch untouched. > > Req-URI: 1002(a)10.1.1.102 > > branch: 1003(a)test.com > > This happens because lookup() only takes as input the req-URI. > > > > The result is that SER makes a DNS query for test.com, to send the request to 1003(a)test.com, instead of the end-point. If the DNS is properly configured, test.com should point to SER itself, making SER sending the message to itself via the network. Depending on the config file, it can work but you should check the logs to see what is exactly happening. > > User 1002 will get the message without any problem. > > > > NOTE: depending on the order in the database, 1003 and 1002 can exchange roles in upper explanation. > > > > As SER is now coded, above can not be changed (as far as I know).... > > > > Hope it helps, > > > > Samuel. > > > > Unclassified. > > > > >>> "Zeus Ng" <zeus.ng(a)isquare.com.au> 08/03/05 01:51AM >>> > > You would need two steps here, looup("aliases"), then lookup("location"). > > > > Zeus > > > > > -----Original Message----- > > > From: serusers-bounces(a)lists.iptel.org > > > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard Z > > > Sent: Wednesday, 3 August 2005 3:33 AM > > > To: Atle Samuelsen > > > Cc: serusers(a)lists.iptel.org > > > Subject: Re: [Serusers] lookup 2 uri > > > > > > > > > Hi Atle, > > > > > > Alias doesn't replace uri with its real location. I'd like to use > > > lookup("location") to find out its real location. For > > > example, if 102(a)test.com is at 102(a)10.1.1.102 and > > > 103(a)test.com is at 103(a)10.1.1.103. I'd like to replace all > > > uris with 102(a)10.1.1.102 and 103(a)10.1.1.103 in a single ser > > > transaction logic. Would it be possible? > > > > > > Thanks, > > > Richard > > > > > > > > > On 8/1/05, Atle Samuelsen <clona(a)camaro.no> wrote: > > > > > > > > Hi, > > > > > > > > you could do something like : > > > > > > > > serctl alias add 101 sip:102@test.com > > > > serctl alias add 101 sip:103@test.com > > > > > > > > This way the invite would be forked to both 102 and 103, > > > > > > > > just remember to have both lookup("aliases"); and > > > save("aliases"); in > > > > your config at the right spot :-) > > > > > > > > -Atle > > > > > > > > > > > > > > > > * Richard Z <rzheng(a)gmail.com> [050802 04:34]: > > > > > Hi, > > > > > > > > > > I am trying to implement a feature to ring two ip phones > > > > > simultaneously. For example, when I call 101(a)test.com, it > > > rings both > > > > > 101(a)test.com and 102(a)test.com. In this case, the ruri is > > > > > 101(a)test.com. I can use append_branch 102(a)test.com, but > > > is there a > > > > > way to lookup the locations of both 101 and 102? > > > > > > > > > > Thanks, > > > > > Richard > > > > > > > > > > _______________________________________________ > > > > > 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] SER and Cisco 1760V /w VIC2 2BRI-NT/TE
by Antanas Masevicius 04 Aug '05

04 Aug '05

Hello, does enyone had any experiance trying to run cisco router with 2BRI-NT/TE voip NIC with SER? I wonder if it is possible at least. I also didn't found how cisco should send sip-password to SER. Antanas

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[Serusers] What is unique to a call in acc database table??
by Eric Haskins 04 Aug '05

04 Aug '05

here are a few sip_callid's from the ACC table they are all from the same user. This is being used in a prepaid outgoing only. 855d2882-32ae277b(a)192.168.1.100 7cd4ca18-5abe85b0(a)192.168.1.100 We are using same ATA adapter for these calls. I just wanted to confirm this is unique to each call or I may have to write something to export info Eric

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[Users] force_rtp_proxy() problem.
by Ray Van Dolson 04 Aug '05

04 Aug '05

Thanks for the help earlier Bogdan. I've gotten things working for the most part, but am having a problem with the force_rtp_proxy function. Once again, here is my setup: SIP ATA - 10.0.201.196 OpenSER - eth0: 208.208.53.5/24 - eth1: 10.0.201.5/24 Asterisk - eth0: 65.234.34.5/24 - eth1: 10.0.201.6/24 Yes, Asterisk and OpenSER are on the same Private network currently. This is just for testing as clients were originally connecting directly to Asterisk. It will not always be the case. I have registration working correctly. If my SIP ATA makes an outbound call (INVITE) it appears to inform Asterisk in its SDP payload that media traffic (RTP) should be sent back to 10.0.201.5 (on eth1). Asterisk has no problem with this since it also has an IP on the 10.0.201.0/24 network. However this obviously will not always be the case. I want media (RTP) to travel back via the eth0 IP addresses. I've tried the following: force_rtp_proxy("", "208.208.53.5") To hopefully force my media address to be the IP on eth0 on OpenSER. However, calls no longer succeed and I see the following in the SDP payload now: v=0 o=- 1298429 1298429 IN IP4 10.0.201.196 s=- c=IN IP4 208.208.53.510.0.201.5 t=0 0 m=audio 3508835088 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 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 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=direction:active a=nortpproxy:yes a=nortpproxy:yes As you can see, the info in c= has both IP's and they are not seperate by a space. This same phenomenon occurs even if I use force_rtp_proxy by itself or with any of the flags. Normally there is just one IP address in this field it seems. Is this a bug in force_rtp_proxy or am I doing something incorrectly? Thanks! -- Ray Van Dolson Linux/Unix Systems Administrator Digital Path, Inc.

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[Serusers] What is unique to a call in acc database table??
by Eric Haskins 04 Aug '05

04 Aug '05

We are currently working on our Billing system and I was looking over our SER database and noticed sip_callid appeared to be unique to each call but when I looked back in the database I saw a bunch that appeared to be the same??? They were notify messages is this common or do I have to devise a different method? We have the hardest part done that calculates call price by Country Code , City Code, customer assigned rate or default Wholesale/Retail rate. We had it completed when running SanSay Equipment and we were told to make it work with SER. Eric Haskins Lead Software Developer v1010 (954) 272-4961 ext. 206 (727) 239-3823 Cell eric(a)merchant2020.com

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[Users] PostgreSQL Version of "sc"
by Adam Sherman 04 Aug '05

04 Aug '05

There is a nice little script in sip-server/scripts/sc that allows for managing many of the tables. Has anyone done the necessary tweaking to port it to PostgreSQL? Thanks, A. -- Adam Sherman Technologist

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[Serusers] route loop quandary
by Daniel Poulsen 04 Aug '05

04 Aug '05

I am having a problem with calls that are parallel forked. The problem happens when a call comes in, gets forked using multiple aliases and one UA is unavailable. The question is, how do I handle the offline call leg? I can't do if (!lookup("location")) { break; (or sl_send_reply("404") } because then the whole call dies since each branch of the call uses the same callid. If I t_relay them both, then the fork works but I the offline leg gets repeatedly relayed unltil I get a 483 "Too many hops". This scenario would be OK [althought certainly not ideal], except it does not allow me to add a condition to hunt to the next number upon a failed lookup("location"). If I do, the online UA will ring once and the offline UA will get routed immediately to voicemail which picks up after just one ring. How are other handling these problems? I think the answer lies in letting failure_route handle everything, but I don't know how to get that to work because if I don't handle the offline users in the (!lookup("location"), I end up with routing loops and eventualluy a 483. Anybody that has no online contacts whatsoever never makes it to failure_route (where the forward to VM is handled). Below is my ser.cfg. I've been playing around with things like if (isflagset(6)) { route(x) } but I still need some way to handle the failed portion of the fork. Is there a way to "null route" the failed portion or something? route { /* Registration and other logic snipped for brevity */ # Translate local address according to aliases table xlog("L_INFO", "%ci: Preparing to look up alias for %ru\n"); if(lookup("aliases")) { xlog("L_NOTICE", "%ci: alias lookup changed uri to %ru\n"); if (!(uri==myself)) { xlog("L_NOTICE", "%ci: Outbound aliases %ru\n"); route(4); break; }; }; if ( (uri=~"^sip:411@.*") | (uri=~"^sip:911@.*") | (uri=~"^sip:011[0-9]+@.*") | (uri=~"^sip:1[0-9]{10}@.*") ) { route(3); # 3: PSTN with authentication break; }; # Anything left is a local call. xlog("L_INFO", "%ci: Preparing to look up location for %ru\n"); if (!lookup("location")) { xlog("L_NOTICE", "%ci: no location for %ru\n"); setflag(5); } else { # Had a sucessful lookup setflag(6); }; route(4); } route[4] { # If an invitation, we want to hunt on failure if(method == "INVITE") { t_on_failure("1"); # first hunt }; append_hf("P-hint: relay\r\n"); xlog("L_NOTICE", "%ci: r4: t_relay %ru\n"); if (!t_relay()) { xlog("L_NOTICE", "%ci: %ru r4: Relay Failed\n"); sl_reply_error(); break; }; } failure_route[1] { revert_uri(); route(7); } ### Handle hunt sequences.

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[Users] Trouble with freshly compiled OpenSer0.9.5
by Martin Kjeldsen 04 Aug '05

04 Aug '05

Hi all I was wondering if any of you might have a solution to my problem! The thing is that i've compiled and installed OpenSer0.9.5 from CVS, on a SUSE 9.2, added MySQL support and created a DB for OpenSer. And now I'm stuck at executing openserctl start and only ending up with the PID filed getting created and ultimately deleted, when OpenSer terminates with the message "PID file /home/ser/var/run/openser.pid does not exist -- OpenSER start failed", and path to the PID has been altered in openserctl. I'm using debug mode 5 but cant realy figure out what might be causing this error! All help will be greatly appreciated. I have attached the output: (Sorry for the big post) ser@site1:~/running/openser/sbin> [21Popenserctl start Starting OpenSER : 0(14441) read 2733431526 from /dev/urandom 0(14441) seeding PRNG with 3856599290 0(14441) test random number 930545650 0(14441) loading module /home/ser/running/openser/lib/openser/modules/mysql.so 0(14441) loading module /home/ser/running/openser/lib/openser/modules/sl.so 0(14441) loading module /home/ser/running/openser/lib/openser/modules/tm.so 0(14441) loading module /home/ser/running/openser/lib/openser/modules/rr.so 0(14441) loading module /home/ser/running/openser/lib/openser/modules/maxfwd.so 0(14441) loading module /home/ser/running/openser/lib/openser/modules/usrloc.so 0(14441) loading module /home/ser/running/openser/lib/openser/modules/registrar.so 0(14441) loading module /home/ser/running/openser/lib/openser/modules/textops.so 0(14441) set_mod_param_regex: usrloc matches module usrloc 0(14441) set_mod_param_regex: found <db_url> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(14441) set_mod_param_regex: usrloc matches module usrloc 0(14441) set_mod_param_regex: found <db_mode> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(14441) set_mod_param_regex: rr matches module rr 0(14441) set_mod_param_regex: found <enable_full_lr> in module rr [/home/ser/running/openser/lib/openser/modules/rr.so] 0(14441) find_export: found <mf_process_maxfwd_header> in module maxfwd [/home/ser/running/openser/lib/openser/modules/maxfwd.so] 0(14441) find_export: found <sl_send_reply> in module sl_module [/home/ser/running/openser/lib/openser/modules/sl.so] 0(14441) find_export: found <sl_send_reply> in module sl_module [/home/ser/running/openser/lib/openser/modules/sl.so] 0(14441) find_export: found <record_route> in module rr [/home/ser/running/openser/lib/openser/modules/rr.so] 0(14441) find_export: found <loose_route> in module rr [/home/ser/running/openser/lib/openser/modules/rr.so] 0(14441) find_export: found <append_hf> in module textops [/home/ser/running/openser/lib/openser/modules/textops.so] 0(14441) find_export: found <append_hf> in module textops [/home/ser/running/openser/lib/openser/modules/textops.so] 0(14441) find_export: found <save> in module registrar [/home/ser/running/openser/lib/openser/modules/registrar.so] 0(14441) find_export: found <lookup> in module registrar [/home/ser/running/openser/lib/openser/modules/registrar.so] 0(14441) find_export: found <append_hf> in module textops [/home/ser/running/openser/lib/openser/modules/textops.so] 0(14441) find_export: found <lookup> in module registrar [/home/ser/running/openser/lib/openser/modules/registrar.so] 0(14441) find_export: found <sl_send_reply> in module sl_module [/home/ser/running/openser/lib/openser/modules/sl.so] 0(14441) find_export: found <append_hf> in module textops [/home/ser/running/openser/lib/openser/modules/textops.so] 0(14441) find_export: found <t_relay> in module tm [/home/ser/running/openser/lib/openser/modules/tm.so] 0(14441) find_export: found <sl_reply_error> in module sl_module [/home/ser/running/openser/lib/openser/modules/sl.so] 0(14441) routing table 0: 0(14441) if ( 0(14441) NOT( 0(14441) external_module_call( 0(14441) f_ptr<0x4027ffd0> 0(14441) , "10" 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x4027ffd0> 0(14441) , "10" 0(14441) ); 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "483" 0(14441) , "Too Many Hops" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) UNKNOWN 0(14441) <UNKNOWN> 0(14441) 2048 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "513" 0(14441) , "Message too big" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4027c7c0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) external_module_call( 0(14441) f_ptr<0x4027a1c0> 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x4027a1c0> 0(14441) ); 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: rr-enforced " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) { 0(14441) if ( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4029bb30> 0(14441) , "location" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "aliases" 0(14441) ); 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound alias " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "404" 0(14441) , "Not Found" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: usrloc applied " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) if ( 0(14441) UNKNOWN 0(14441) <UNKNOWN> 0(14441) 2048 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "513" 0(14441) , "Message too big" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4027c7c0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) external_module_call( 0(14441) f_ptr<0x4027a1c0> 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x4027a1c0> 0(14441) ); 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: rr-enforced " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) { 0(14441) if ( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4029bb30> 0(14441) , "location" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "aliases" 0(14441) ); 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound alias " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "404" 0(14441) , "Not Found" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: usrloc applied " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) if ( 0(14441) NOT( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4027c7c0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) external_module_call( 0(14441) f_ptr<0x4027a1c0> 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x4027a1c0> 0(14441) ); 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: rr-enforced " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) { 0(14441) if ( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4029bb30> 0(14441) , "location" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "aliases" 0(14441) ); 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound alias " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "404" 0(14441) , "Not Found" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: usrloc applied " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) if ( 0(14441) external_module_call( 0(14441) f_ptr<0x4027a1c0> 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x4027a1c0> 0(14441) ); 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: rr-enforced " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) { 0(14441) if ( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4029bb30> 0(14441) , "location" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "aliases" 0(14441) ); 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound alias " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "404" 0(14441) , "Not Found" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: usrloc applied " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) { 0(14441) if ( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4029bb30> 0(14441) , "location" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "aliases" 0(14441) ); 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound alias " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "404" 0(14441) , "Not Found" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: usrloc applied " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) if ( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) { 0(14441) if ( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4029bb30> 0(14441) , "location" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "aliases" 0(14441) ); 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound alias " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "404" 0(14441) , "Not Found" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: usrloc applied " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: usrloc applied " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) 0(14441) routing table 1: 0(14441) if ( 0(14441) NOT( 0(14441) external_module_call( 0(14441) f_ptr<0x4024c740> 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x4024c740> 0(14441) ); 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170580> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) Listening on udp: 212.97.202.73 [212.97.202.73]:5060 tcp: 212.97.202.73 [212.97.202.73]:5060 Aliases: tcp: osite1:5060 tcp: osite1.cuatro.dk:5060 udp: osite1:5060 udp: osite1.cuatro.dk:5060 WARNING: no fork mode 0(14441) shm_mem_init: success 0(0) core dump limits set to 4294967295 0(0) DEBUG: init_mod: sl_module stateless - initializing 0(0) DEBUG: register_fifo_cmd: new command (sl_stats) registered 0(0) unixsock_register_cmd: New command (sl_stats) registered 0(0) DEBUG: MD5 calculated: ba950484b7cb9e0b78ff4f84c7272d40 0(0) DEBUG: init_mod: tm 0(0) TM - (size of cell=2780, sip_msg=760) initializing... 0(0) Call-ID initialization: '773ffe9e' 0(0) DEBUG: register_fifo_cmd: new command (t_uac_dlg) registered 0(0) DEBUG: register_fifo_cmd: new command (t_uac_cancel) registered 0(0) DEBUG: register_fifo_cmd: new command (t_hash) registered 0(0) DEBUG: register_fifo_cmd: new command (t_reply) registered 0(0) unixsock_register_cmd: New command (t_uac_dlg) registered 0(0) unixsock_register_cmd: New command (t_uac_cancel) registered 0(0) unixsock_register_cmd: New command (t_hash) registered 0(0) unixsock_register_cmd: New command (t_reply) registered 0(0) DEBUG: lock_initialize: lock initialization started 0(0) DEBUG: register_fifo_cmd: new command (t_stats) registered 0(0) unixsock_register_cmd: New command (t_stats) registered 0(0) DEBUG: MD5 calculated: 5818e8718167640aef9d06ecea0d94fe 0(0) DEBUG: MD5 calculated: 55629ad9d6de7376eb3d3c6f0e1dea9c 0(0) DEBUG: init_mod: rr 0(0) rr - initializing 0(0) DEBUG: init_mod: maxfwd 0(0) Maxfwd module- initializing 0(0) DEBUG: init_mod: usrloc 0(0) usrloc - initializing 0(0) DEBUG: register_fifo_cmd: new command (ul_stats) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_rm) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_rm_contact) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_dump) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_flush) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_add) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_show_contact) registered 0(0) unixsock_register_cmd: New command (ul_stats) registered 0(0) unixsock_register_cmd: New command (ul_rm) registered 0(0) unixsock_register_cmd: New command (ul_rm_contact) registered 0(0) unixsock_register_cmd: New command (ul_dump) registered 0(0) unixsock_register_cmd: New command (ul_flush) registered 0(0) unixsock_register_cmd: New command (ul_add) registered 0(0) unixsock_register_cmd: New command (ul_show_contact) registered 0(0) find_mod_export: found <db_use_table> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) find_mod_export: found <db_init> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) find_mod_export: found <db_close> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) find_mod_export: found <db_query> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) find_mod_export: found <db_raw_query> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) find_mod_export: found <db_free_result> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) find_mod_export: found <db_insert> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) find_mod_export: found <db_delete> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) find_mod_export: found <db_update> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) DEBUG: init_mod: registrar 0(0) registrar - initializing 0(0) find_export: found <sl_send_reply> in module sl_module [/home/ser/running/openser/lib/openser/modules/sl.so] 0(0) find_export: found <ul_bind_usrloc> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_register_udomain> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_get_all_ucontacts> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_insert_urecord> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_delete_urecord> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_get_urecord> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_lock_udomain> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_unlock_udomain> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_release_urecord> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_insert_ucontact> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_delete_ucontact> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_get_ucontact> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_update_ucontact> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_register_watcher> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_unregister_watcher> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_register_ulcb> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) DEBUG: init_mod: textops textops - initializing 0(0) fixing /home/ser/running/openser/lib/openser/modules/maxfwd.so mf_process_maxfwd_header 0(0) fixing /home/ser/running/openser/lib/openser/modules/sl.so sl_send_reply 0(0) fixing /home/ser/running/openser/lib/openser/modules/sl.so sl_send_reply 0(0) fixing /home/ser/running/openser/lib/openser/modules/rr.so record_route 0(0) fixing /home/ser/running/openser/lib/openser/modules/rr.so loose_route 0(0) fixing /home/ser/running/openser/lib/openser/modules/textops.so append_hf 0(0) fixing /home/ser/running/openser/lib/openser/modules/textops.so append_hf 0(0) fixing /home/ser/running/openser/lib/openser/modules/registrar.so save 0(0) get_connection(): Connection not found in the pool 0(0) preload_udomain(): Table is empty 0(0) release_connection(): Removing connection from the pool 0(0) fixing /home/ser/running/openser/lib/openser/modules/registrar.so lookup 0(0) get_connection(): Connection not found in the pool 0(0) preload_udomain(): Table is empty 0(0) release_connection(): Removing connection from the pool 0(0) fixing /home/ser/running/openser/lib/openser/modules/textops.so append_hf 0(0) fixing /home/ser/running/openser/lib/openser/modules/registrar.so lookup 0(0) fixing /home/ser/running/openser/lib/openser/modules/sl.so sl_send_reply 0(0) fixing /home/ser/running/openser/lib/openser/modules/textops.so append_hf 0(0) fixing /home/ser/running/openser/lib/openser/modules/tm.so t_relay 0(0) fixing /home/ser/running/openser/lib/openser/modules/sl.so sl_reply_error 0(0) INFO: udp_init: SO_RCVBUF is initially 111616 0(0) DEBUG: udp_init: trying SO_RCVBUF: 223232 0(0) DEBUG: setting SO_RCVBUF; set=223232,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 225280 0(0) DEBUG: setting SO_RCVBUF; set=225280,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 227328 0(0) DEBUG: setting SO_RCVBUF; set=227328,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 229376 0(0) DEBUG: setting SO_RCVBUF; set=229376,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 231424 0(0) DEBUG: setting SO_RCVBUF; set=231424,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 233472 0(0) DEBUG: setting SO_RCVBUF; set=233472,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 235520 0(0) DEBUG: setting SO_RCVBUF; set=235520,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 237568 0(0) DEBUG: setting SO_RCVBUF; set=237568,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 239616 0(0) DEBUG: setting SO_RCVBUF; set=239616,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 241664 0(0) DEBUG: setting SO_RCVBUF; set=241664,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 243712 0(0) DEBUG: setting SO_RCVBUF; set=243712,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 245760 0(0) DEBUG: setting SO_RCVBUF; set=245760,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 247808 0(0) DEBUG: setting SO_RCVBUF; set=247808,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 249856 0(0) DEBUG: setting SO_RCVBUF; set=249856,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 251904 0(0) DEBUG: setting SO_RCVBUF; set=251904,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 253952 0(0) DEBUG: setting SO_RCVBUF; set=253952,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 256000 0(0) DEBUG: setting SO_RCVBUF; set=256000,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 258048 0(0) DEBUG: setting SO_RCVBUF; set=258048,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 260096 0(0) DEBUG: setting SO_RCVBUF; set=260096,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 262144 0(0) DEBUG: setting SO_RCVBUF; set=262144,verify=262142 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 0(0) DBG: open_uac_fifo: opening fifo... 0(0) DEBUG: FIFO created @ /tmp/openser_fifo 0(0) DEBUG: fifo /tmp/openser_fifo opened, mode=777 0(0) ERROR: init_fifo_server: fifo_write did not open: Permission denied 0(0) initializing fifo server failed 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(14441) shm_mem_destroy 0(14441) destroying the shared memory lock PID created! total 0 PID file /home/ser/var/run/openser.pid does not exist -- OpenSER start failed Kind regards Martin

2 3

[Users] regexp problem
by Klaus Darilion 04 Aug '05

04 Aug '05

Hi! I have a problem with search(); I try to match a contact header with IP address: if !(search("Contact:.*@[0-9]+\.[0-9]+\.[0-9]+\.[0-9]+:")) { works fine, whereas if !(search("Contact:.*@[0-9]+\.[0-9]+\.[0-9]+\.\d+:")) { does not work. ^^ Looks like search() does not like \d Are there any known limitations for the regular expressions? regards klaus

2 2

RE: SV: [Serusers] SER core dump on SuSE
by Pat wang 04 Aug '05

04 Aug '05

Thanks Roger, You are right. I reinstalled SuSE with all the available development packages cause I am not sure what else is needed if there are always something missing. SER works fine with authentication now. Thanks, Pat >From: "Roger Lewau" <roger.lewau(a)serverhallen.com> >To: "'Pat wang'" <wangyu39(a)hotmail.com>,<serusers(a)lists.iptel.org> >Subject: SV: [Serusers] SER core dump on SuSE >Date: Wed, 3 Aug 2005 19:01:08 +0200 > >Hi, > >Did you install a precompiled version of SER and modules? I had the exact >same problem on FreeBSD but solved it by downloading the source and compile >the mysql.so module by hand and then replacing the precompiled module. > >Dont know the exact reason for the crash I had, but the above step solved >it. Could be woth a try. > >Kind regards >Roger > > >-----Ursprungligt meddelande----- >Från: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] För >Pat >wang >Skickat: den 3 augusti 2005 17:53 >Till: serusers(a)lists.iptel.org >Ämne: [Serusers] SER core dump on SuSE > >Hi, > >I am having trouble running SER on SuSE Linux. SER crashes when >authentication module is used. Whenever a REGISTER arrives to SER, it gives >a core dump. If I turn authentication off then everything is fine. I have >tried this on different versions of SuSE including: 9.0 (2.4 Kernel) and >9.1 >and 9.3 (2.6 Kernel). They all give this core dump: > >Core was generated by `ser -E -w .'. >Program terminated with signal 11, Segmentation fault. > >warning: current_sos: Can't read pathname for load map: Input/output error > >Reading symbols from /lib/libdl.so.2...done. >Loaded symbols for /lib/libdl.so.2 >Reading symbols from /lib/libresolv.so.2...done. >Loaded symbols for /lib/libresolv.so.2 >Reading symbols from /lib/tls/libc.so.6...done. >Loaded symbols for /lib/tls/libc.so.6 >Reading symbols from /lib/ld-linux.so.2...done. >Loaded symbols for /lib/ld-linux.so.2 >Reading symbols from /usr/local/lib/ser/modules/mysql.so...done. >Loaded symbols for /usr/local/lib/ser/modules/mysql.so >Reading symbols from /usr/lib/libmysqlclient.so.12...done. >Loaded symbols for /usr/lib/libmysqlclient.so.12 Reading symbols from >/lib/libz.so.1...done. >Loaded symbols for /lib/libz.so.1 >Reading symbols from /lib/libcrypt.so.1...done. >Loaded symbols for /lib/libcrypt.so.1 >Reading symbols from /lib/libnsl.so.1...done. >Loaded symbols for /lib/libnsl.so.1 >Reading symbols from /lib/tls/libm.so.6...done. >Loaded symbols for /lib/tls/libm.so.6 >Reading symbols from /usr/local/lib/ser/modules/sl.so...done. >Loaded symbols for /usr/local/lib/ser/modules/sl.so Reading symbols from >/usr/local/lib/ser/modules/tm.so...done. >Loaded symbols for /usr/local/lib/ser/modules/tm.so Reading symbols from >/usr/local/lib/ser/modules/rr.so...done. >Loaded symbols for /usr/local/lib/ser/modules/rr.so Reading symbols from >/usr/local/lib/ser/modules/maxfwd.so...done. >Loaded symbols for /usr/local/lib/ser/modules/maxfwd.so >Reading symbols from /usr/local/lib/ser/modules/usrloc.so...done. >Loaded symbols for /usr/local/lib/ser/modules/usrloc.so >Reading symbols from /usr/local/lib/ser/modules/registrar.so...done. >Loaded symbols for /usr/local/lib/ser/modules/registrar.so >Reading symbols from /usr/local/lib/ser/modules/textops.so...done. >Loaded symbols for /usr/local/lib/ser/modules/textops.so >Reading symbols from /usr/local/lib/ser/modules/auth.so...done. >Loaded symbols for /usr/local/lib/ser/modules/auth.so >Reading symbols from /usr/local/lib/ser/modules/auth_db.so...done. >Loaded symbols for /usr/local/lib/ser/modules/auth_db.so >Reading symbols from /lib/libnss_files.so.2...done. >Loaded symbols for /lib/libnss_files.so.2 #0 qm_malloc (qm=0x80ffac0, >size=20) at q_malloc.c:290 >290 q_malloc.c: No such file or directory. > in q_malloc.c >(gdb) >(gdb) >(gdb) >(gdb) >(gdb) bt >#0 qm_malloc (qm=0x80ffac0, size=20) at q_malloc.c:290 >#1 0x4016322c in new_result () at res.c:199 >#2 0x40162035 in store_result (_h=0x8106138, _r=0xbfffee4c) at dbase.c:298 >#3 0x080998ec in table_version (dbf=0x4039b320, connection=0x8106138, >table=0xbfffeeb0) > at db.c:176 >#4 0x40398eaf in auth_db_ver (db_url=0x0, name=0x0) at authorize.c:337 >#5 0x40397fba in str_fixup (param=0x8106c30, param_no=2) at >authdb_mod.c:238 >#6 0x0806ec55 in fix_actions (a=<value optimized out>) at route.c:243 >#7 0x0806f136 in fix_expr (exp=0x8106c48) at route.c:134 >#8 0x0806f17e in fix_expr (exp=0x8106c78) at route.c:99 >#9 0x0806ed1b in fix_actions (a=<value optimized out>) at route.c:221 #10 >0x0806ed3e in fix_actions (a=<value optimized out>) at route.c:225 >#11 0x0806ed3e in fix_actions (a=<value optimized out>) at route.c:225 >#12 0x0806f026 in fix_rls () at route.c:726 >#13 0x0805df35 in main (argc=4, argv=0xbffff1b4) at main.c:1558 >(gdb) > >I have to use Redhat playing SER now. I am really appreciate if someone can >take a look at the cause. By the way, I tried to install SER from binary >package as well as compile from the source code, result is the same. > >Regards, > >Pat > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

[Serusers] Segmentation fault-fifo_server fgets failed: Illegal seek
by Ashutosh Kumar 04 Aug '05

04 Aug '05

Hi , I am getting a segmentation fault when my sip server runs. It happens mostly when a client tries to call another client phone. I think that this has got something to do with ser_fifo file permissions. For that reason, the relevant code from my ser.cfg file is as belows; --------------- fifo="/tmp/ser_fifo" fifo_mode=0666 --------------- I am running SER-0.8.14.The debug trace is as produced below: 0(3972) check_self - checking if host==us: 15==15 && [sip.myownphone.com] == [sip. myownphone.com] 0(3972) check_self - checking if port 5060 matches port 5060 0(3972) rwrite(): Rewriting Request-URI with 'sip:satya@210.7.75.189:5060' 0(3972) DEBUG: t_addifnew: msg id=3 , global msg id=1 , T on entrance=0xffffffff 0(3972) parse_headers: flags=-1 0(3972) parse_headers: flags=60 0(3972) t_lookup_request: start searching: hash=13553, isACK=0 0(3972) DEBUG: RFC3261 transaction matching failed 0(3972) DEBUG: t_lookup_request: no transaction found 0(3972) DBG: callback type 2, id 3 entered 0(3972) DEBUG: t_lookupOriginalT: searching on hash entry 13553 0(3972) DEBUG: RFC3261 transaction matching failed 0(3972) DEBUG: t_lookupOriginalT: no CANCEL maching found! 0(3972) DEBUG: t_lookupOriginalT completed 0(3972) DEBUG: mk_proxy: doing DNS lookup... 0(3972) check_via_address(61.16.236.34, 192.168.11.37, 0) 2(3975) ERROR: fifo_server fgets failed: Illegal seek 2(3975) ERROR: fifo_server fgets failed: Illegal seek 2(3975) ERROR: fifo_server fgets failed: Illegal seek 2(3975) ERROR: fifo_server fgets failed: Illegal seek 2(3975) INFO: signal 15 received 2(3975) Memory status (pkg): 2(3975) fm_status (0x80c2c80): 2(3975) heap size= 1047440 2(3975) dumping free list: 2(3975) hash = 1 fragments no.: 11, bucket size: 8 - 8 (first 8) 2(3975) hash = 2 fragments no.: 2, bucket size: 16 - 16 (first 16) 2(3975) hash = 4 fragments no.: 1, bucket size: 32 - 32 (first 32) 2(3975) hash = 15 fragments no.: 1, bucket size: 120 - 120 (first 120) 2(3975) hash = 20 fragments no.: 3, bucket size: 160 - 160 (first 160) 2(3975) hash = 265 fragments no.: 1, bucket size: 524288 - 1048576 (first 1030944) 2(3975) TOTAL: 19 free fragments = 1031696 free bytes 2(3975) ----------------------------- 1(3974) INFO: signal 15 received 1(3974) Memory status (pkg): 1(3974) fm_status (0x80c2c80): 1(3974) heap size= 1047440 1(3974) dumping free list: 1(3974) hash = 1 fragments no.: 11, bucket size: 8 - 8 (first 8) 1(3974) hash = 2 fragments no.: 1, bucket size: 16 - 16 (first 16) 1(3974) hash = 4 fragments no.: 1, bucket size: 32 - 32 (first 32) 1(3974) hash = 6 fragments no.: 1, bucket size: 48 - 48 (first 48) 1(3974) hash = 8 fragments no.: 1, bucket size: 64 - 64 (first 64) 1(3974) hash = 20 fragments no.: 3, bucket size: 160 - 160 (first 160) 1(3974) hash = 32 fragments no.: 1, bucket size: 256 - 256 (first 256) 1(3974) hash = 78 fragments no.: 1, bucket size: 624 - 624 (first 624) 1(3974) hash = 265 fragments no.: 1, bucket size: 524288 - 1048576 (first 1030048) 1(3974) TOTAL: 21 free fragments = 1031656 free bytes 1(3974) ----------------------------- Segmentation fault Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

1 0

[Serusers] serweb 0.9.3 config on RedHat ES4
by Amit Rastogi 04 Aug '05

04 Aug '05

Hello, I am trying to install serweb 0.9.3 release. I already have ser 0.9.3 running with mysql support. When i try to access the http:<host>/serweb/admin.php, it displays a blank page and nothing else.. Here are the steps i have done so far.. untar package to <root>/serweb directory copy ./html to /var/www/html/serweb directory copy ./phplib to /var/www/phplib directory modified php.ini to have short_open_tag to on Added alias in httpd.conf as Alias /serweb/ "/var/www/html/serweb/" Quick install guide mentions modifying config_data_later.php, i am using the default username pass so i skipped it. changed config_paths.php -> root path is /serweb/ Can someone please help ? Thanks, Akr

3 2

Re: [Serusers] Implementing Codec Translator with Asterisk
by Noel Sharpe 04 Aug '05

04 Aug '05

Frank Kostin wrote: > > BTW, I heard that someone might need a licence for G.729 !? It's a proprietory codec, so yes you need a licence for each and every channel you transcode. If you want to support the transcoding of 100 simultaneous conversations, you need 100 licences. You don't need a licence to just use G729; you only need licences when you need to encode or decode using G729. If you use asterisk, you can purchase a licence from Digium.

1 0

Re: [Serusers] lookup 2 uri
by Samuel Osorio Calvo 04 Aug '05

04 Aug '05

If you know in advance the IP of 1002 and it is static, you can use serctl to add it's contact. This way doing a lookup("location") for 1003 will return both contacts...it's not a nice solution but it whould work. Samuel. Unclassified. >>> Richard Z <rzheng(a)gmail.com> 08/03/05 08:55PM >>> Hi Samuel, That's a very good detailed explanation! That's exactly what happened. In my existing config, I have the logic to deal with the second ring to 1003(a)test.com. It forks to itself and creates another ser transaction. I'd like to make it simpler by doing it in a single transaction. One way I am thinking about, but haven't tried yet, is when 1003 registers, ser adds its location to both 1002 and 1003. So when ser does lookup("location") for 1002, the real location for 1003 shows up. Not sure that it is feasible. Thanks, Richard On 8/2/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > Just one remark: only one of the two URI will be translated to the contact. > > I'll try to explain... > > imagine the incoming Req-URI is 1001(a)test.com. When it passes the first lookup, > lookup("aliases"); > SER will retrieve the two aliases and put one of them as the new req-uri and the other as branch. > Req-URI: 1002(a)test.com > branch: 1003(a)test.com > Doing the second lookup, > lookup("location"); > will put the contact of 1002(a)test.com as Req-URI (if more than one contact is present, the subsequent bindings will be added as branches) and the 1003(a)test.com will be left as branch untouched. > Req-URI: 1002(a)10.1.1.102 > branch: 1003(a)test.com > This happens because lookup() only takes as input the req-URI. > > The result is that SER makes a DNS query for test.com, to send the request to 1003(a)test.com, instead of the end-point. If the DNS is properly configured, test.com should point to SER itself, making SER sending the message to itself via the network. Depending on the config file, it can work but you should check the logs to see what is exactly happening. > User 1002 will get the message without any problem. > > NOTE: depending on the order in the database, 1003 and 1002 can exchange roles in upper explanation. > > As SER is now coded, above can not be changed (as far as I know).... > > Hope it helps, > > Samuel. > > Unclassified. > > >>> "Zeus Ng" <zeus.ng(a)isquare.com.au> 08/03/05 01:51AM >>> > You would need two steps here, looup("aliases"), then lookup("location"). > > Zeus > > > -----Original Message----- > > From: serusers-bounces(a)lists.iptel.org > > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard Z > > Sent: Wednesday, 3 August 2005 3:33 AM > > To: Atle Samuelsen > > Cc: serusers(a)lists.iptel.org > > Subject: Re: [Serusers] lookup 2 uri > > > > > > Hi Atle, > > > > Alias doesn't replace uri with its real location. I'd like to use > > lookup("location") to find out its real location. For > > example, if 102(a)test.com is at 102(a)10.1.1.102 and > > 103(a)test.com is at 103(a)10.1.1.103. I'd like to replace all > > uris with 102(a)10.1.1.102 and 103(a)10.1.1.103 in a single ser > > transaction logic. Would it be possible? > > > > Thanks, > > Richard > > > > > > On 8/1/05, Atle Samuelsen <clona(a)camaro.no> wrote: > > > > > > Hi, > > > > > > you could do something like : > > > > > > serctl alias add 101 sip:102@test.com > > > serctl alias add 101 sip:103@test.com > > > > > > This way the invite would be forked to both 102 and 103, > > > > > > just remember to have both lookup("aliases"); and > > save("aliases"); in > > > your config at the right spot :-) > > > > > > -Atle > > > > > > > > > > > > * Richard Z <rzheng(a)gmail.com> [050802 04:34]: > > > > Hi, > > > > > > > > I am trying to implement a feature to ring two ip phones > > > > simultaneously. For example, when I call 101(a)test.com, it > > rings both > > > > 101(a)test.com and 102(a)test.com. In this case, the ruri is > > > > 101(a)test.com. I can use append_branch 102(a)test.com, but > > is there a > > > > way to lookup the locations of both 101 and 102? > > > > > > > > Thanks, > > > > Richard > > > > > > > > _______________________________________________ > > > > 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 > >

1 0

[Serusers] t_uac_dlg pb
by mma 04 Aug '05

04 Aug '05

I want to send an invite request to ser's fifo using t_uac_dlg command and exec_msg, my pb is that i can't construct a valid invite header , when i check the log of ser, i see that there is errors in parsing headers. i'm trying to send something like this : cat > /tmp/ser_fifo << EOF :t_uac_dlg:hh INVITE sip: user2(a)sip.server.net . From: sip:user1@sip.server.net To: sip:user2@sip.server.net . Content-Type: application/sdp EOF please can someone correct it. Regards PS : the valid header is the following : Session Initiation Protocol Request-Line: INVITE sip:5001@sip.softiphone.net SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK52ffa6c1cda5b285 From: "5003" <sip:5003@sip.softiphone.net>;tag=81ee9e99dc30cc34 SIP Display info: "5003" SIP from address: sip:5003@sip.softiphone.net SIP tag: 81ee9e99dc30cc34 To: <sip:5001@sip.softiphone.net> SIP to address: sip:5001@sip.softiphone.net Contact: <sip:5003@192.168.1.100> Supported: replaces, timer Call-ID: b2d776f6e99851a4(a)192.168.1.100 CSeq: 44559 INVITE User-Agent: Grandstream GXP2000 1.0.1.9 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 310 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 5003 8000 8000 IN IP4 192.168.1.100 Session Name (s): SIP Call Connection Information (c): IN IP4 192.168.1.100 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 5004 RTP/AVP 0 8 18 3 4 101 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:4 G723/8000 Media Attribute (a): ptime:20 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11

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[Serusers] No stop record for pstn calls
by Anders Hansson 04 Aug '05

04 Aug '05

Hi all, I have a problem accounting transactions to the PSTN network. I do only get the Stop record - no Start record... Anyone knows what could be the reason? I´m using radiusserver for the accounting records. This is part of my ser.cfg file. In the log the messages INVITE accounted and BYE or CANCEL accounted can be seen, but nothing in the radius file, except for the stop record. To make it worse, not all of the stop records can be seen in radius - it looks like only some destinations creates stop records - other dont. The only thing about that is that when it´s a voicemachine in the other end, the stop record is not occuring... # Flag for accounting if (method=="INVITE") { log("INVITE accounted"); setflag(1); }; if (method=="BYE" || method=="CANCEL") { log("BYE or CANCEL accounted"); setflag(1); }; setflag(3); Thanks, Anders

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[Serusers] Problem with SER and PAP2, I get a 407 message when trying a call
by Yiannis Martis 04 Aug '05

04 Aug '05

Hello there, I am Network Administrator working for a company called Hellasfon Network S.A. I use SER 0.9.3 with Vovidas STUN server. PAP2 UAis NATed and gets registered using my ADSL modems IP and everything looks OK. The problem is when I try to make a call to another UA) of my service I get the following output from my server (ngrep):. Please help me if u can!!! U 2005/08/03 13:55:57.380953 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 INVITE sip:2810318308@XXXXXXXXXX.com SIP/2.0. Record-Route: <sip:XX.XX.XX.XX;ftag=7ab32f2bbf2689feo0;lr=on>. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport=5060. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 INVITE. Max-Forwards: 16. Contact: <sip:2810300308@XX.XX.XX.XX:5060>. Expires: 240. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 420. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 13486 13486 IN IP4 XX.XX.XX.XX. s=-. c=IN IP4 XX.XX.XX.XX. t=0 0. m=audio 16420 RTP/AVP 18 0 2 4 8 96 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/8000. 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 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2005/08/03 13:55:57.381212 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport=5060. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a179. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="XXXXXXXXXX.com", nonce="42f0a3e9ea677d46038bdbeccaf8d95c18bce58e". Server: Sip EXpress router (0.9.3 (i386/linux)). Content-Length: 0. Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=4360 req_src_ip=XX.XX.XX.XX req_src_port=5060 in_uri=sip:2810318308@XXXXXXXXXX.com out_uri=sip:2810318308@XXXXXXXXXX.com via_cnt==2". . # U 2005/08/03 13:55:57.381353 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 ACK sip:2810318308@XXXXXXXXXX.com SIP/2.0. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. Call-ID: b521746f-5119658a(a)192.168.1.2. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a179. CSeq: 102 ACK. User-Agent: Sip EXpress router(0.9.3 (i386/linux)). Content-Length: 0. . # U 2005/08/03 13:55:57.664069 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 ACK sip:2810318308@XXXXXXXXXX.com SIP/2.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a179. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 ACK. Max-Forwards: 70. Proxy-Authorization: Digest username="2810300308",realm="XXXXXXXXXX.com",nonce="42f0a3e8f78926d700140a7c490f06dd965ea748",uri="sip:2810318308@XXXXXXXXXX.com",algorithm=MD5,response="ac73134a0de530d527cfbe4fe1903707". Contact: <sip:2810300308@XX.XX.XX.XX:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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RE: RE : [Serusers] is SER Dependable for prepaid System ? Anyone usingitfor prepaid system?
by Sam Lee 04 Aug '05

04 Aug '05

Dear all, How does the b2bua and the Jradius work ? It's the first time I've seen it. Does the billing system rely on radius ? I already have a SER running on digest authentication. How do I integrate them with this prepaid system? Regards, Sam -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Olivier Taylor Sent: Thursday, August 04, 2005 12:00 AM To: 'Iqbal'; 'Bruno Lopes F. Cabral' Cc: serusers(a)lists.iptel.org Subject: RE : [Serusers] is SER Dependable for prepaid System ? Anyone usingitfor prepaid system? With Jradius, we disconnect the user when credit limit is reached. Olivier -----Message d'origine----- De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de Iqbal Envoyé : mercredi 3 août 2005 14:58 À : Bruno Lopes F. Cabral Cc : serusers(a)lists.iptel.org Objet : Re: [Serusers] is SER Dependable for prepaid System ? Anyone usingit for prepaid system? but it dont do prepay, it is a rating engine I think, and shows you how much each call cost, i dont think it disconnects a call if they go above their credit limit Iqbal Bruno Lopes F. Cabral wrote: > Hello > > is CDRTool acurate? it seems not to need a B2BUA system > > Cheers > !3runo > > Olivier Taylor wrote: > >> I think the only clean way to do prepaid with ser is to use b2bua > > > as third party for billing, the only annoyance is that you have to > > manage the bandwidth for signaling. > >> Our side, we use B2bua, radius and Jradius for accounting. All the >> accounting logic is managed thru Jradius. > > > _______________________________________________ > 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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RE: RE : [Serusers] is SER Dependable for prepaid System ? Anyone usingitfor prepaid system?
by Sam Lee 04 Aug '05

04 Aug '05

-----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Olivier Taylor Sent: Thursday, August 04, 2005 12:00 AM To: 'Iqbal'; 'Bruno Lopes F. Cabral' Cc: serusers(a)lists.iptel.org Subject: RE : [Serusers] is SER Dependable for prepaid System ? Anyone usingitfor prepaid system? With Jradius, we disconnect the user when credit limit is reached. Olivier -----Message d'origine----- De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de Iqbal Envoyé : mercredi 3 août 2005 14:58 À : Bruno Lopes F. Cabral Cc : serusers(a)lists.iptel.org Objet : Re: [Serusers] is SER Dependable for prepaid System ? Anyone usingit for prepaid system? but it dont do prepay, it is a rating engine I think, and shows you how much each call cost, i dont think it disconnects a call if they go above their credit limit Iqbal Bruno Lopes F. Cabral wrote: > Hello > > is CDRTool acurate? it seems not to need a B2BUA system > > Cheers > !3runo > > Olivier Taylor wrote: > >> I think the only clean way to do prepaid with ser is to use b2bua > > > as third party for billing, the only annoyance is that you have to > > manage the bandwidth for signaling. > >> Our side, we use B2bua, radius and Jradius for accounting. All the >> accounting logic is managed thru Jradius. > > > _______________________________________________ > 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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[Users] Mediaproxy, fix_contact, and ip:port checks
by Kanakatti Mahesh Subramanya 03 Aug '05

03 Aug '05

Currently, fix_contact immediately returns if src_ip == contact_ip However, there are a bunch-a situations where src_ip==contact_ip, but src_port != contact_port (Dont even ask. Just blame the *really* bizarre cable routers that Comcast deploys every now and then) I *could* go about modifying FixContact in functions.c to check both the ip and port. Question is, should I? Is there something I am missing as to why we check only the IP? Should I be explicitly *not* checking the port? Cheers

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[Serusers] pstn -> ser -> asterisk voicemail
by Jack Wei 03 Aug '05

03 Aug '05

i'm having a minor problem with pstn to ser forwarding to asterisk voicemail. when ser forwards the call to asterisk, the destination ua keeps ringing while the regular phone hears the voicemail. this doesn't happen when it's a ip-to-ip call. anyone knows why this is happening? ser.cfg ================= route { ... if ( lookup( "location" ) ) # Destination UA is online { xlog("L_INFO", "%ru is online (in usrloc)\n\n"); setflag( 10 ); } else # Checking database to check if destination UA is online { route(4); }; if ( !isflagset( 10 ) ) # Destination UA is not online...sending to Asterisk voicemail { xlog("L_INFO", "%ru is not online...sending to voicemail\n\n"); route(3); } else { resetflag( 10 ); append_hf( "P-hint: VOIP OUTBOUND\r\n" ); t_on_failure("1"); }; ... } route[3] { xlog("L_INFO", "Sending URI (%ru) to Asterisk voicemail\n\n"); append_hf( "P-hint: VOICEMAIL\r\n" ); rewritehostport("x.x.x.x:5060"); } failure_route[1] { xlog("L_INFO", "Call timed out\n\n"); route(3); append_branch(); t_relay_to_udp("x.x.x.x","5060"); break; } Jack Wei __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] UDP
by Sebastian Kühner 03 Aug '05

03 Aug '05

Hi! I'm getting INVITES from another Cisco server, but my ser isn't catching it. The difference between a "normal message" (from one of our phones) is, that a dump shows us "UDP" like protocol (Cisco)... and SIP from our phones. Do I have to respect different SIP signallings?? I made a log on "INVITES"... but ser isn't catching it :-( Thanks! Sebastian

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[Serusers] RTPProxy issues with G729 at different frame lengths
by Noel Sharpe 03 Aug '05

03 Aug '05

Hi I am have some difficulties using the latest RTPProxy. Everything seems to work fine unless you use G729 codec with 10ms frame lengths. Has anyone else experienced this issue, and does anyone have any suggestions to work around the issue? Kind Regards Noel

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Re: [Serusers] lookup 2 uri
by Richard Z 03 Aug '05

03 Aug '05

Hi Samuel, That's a very good detailed explanation! That's exactly what happened. In my existing config, I have the logic to deal with the second ring to 1003(a)test.com. It forks to itself and creates another ser transaction. I'd like to make it simpler by doing it in a single transaction. One way I am thinking about, but haven't tried yet, is when 1003 registers, ser adds its location to both 1002 and 1003. So when ser does lookup("location") for 1002, the real location for 1003 shows up. Not sure that it is feasible. Thanks, Richard On 8/2/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > Just one remark: only one of the two URI will be translated to the contact. > > I'll try to explain... > > imagine the incoming Req-URI is 1001(a)test.com. When it passes the first lookup, > lookup("aliases"); > SER will retrieve the two aliases and put one of them as the new req-uri and the other as branch. > Req-URI: 1002(a)test.com > branch: 1003(a)test.com > Doing the second lookup, > lookup("location"); > will put the contact of 1002(a)test.com as Req-URI (if more than one contact is present, the subsequent bindings will be added as branches) and the 1003(a)test.com will be left as branch untouched. > Req-URI: 1002(a)10.1.1.102 > branch: 1003(a)test.com > This happens because lookup() only takes as input the req-URI. > > The result is that SER makes a DNS query for test.com, to send the request to 1003(a)test.com, instead of the end-point. If the DNS is properly configured, test.com should point to SER itself, making SER sending the message to itself via the network. Depending on the config file, it can work but you should check the logs to see what is exactly happening. > User 1002 will get the message without any problem. > > NOTE: depending on the order in the database, 1003 and 1002 can exchange roles in upper explanation. > > As SER is now coded, above can not be changed (as far as I know).... > > Hope it helps, > > Samuel. > > Unclassified. > > >>> "Zeus Ng" <zeus.ng(a)isquare.com.au> 08/03/05 01:51AM >>> > You would need two steps here, looup("aliases"), then lookup("location"). > > Zeus > > > -----Original Message----- > > From: serusers-bounces(a)lists.iptel.org > > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard Z > > Sent: Wednesday, 3 August 2005 3:33 AM > > To: Atle Samuelsen > > Cc: serusers(a)lists.iptel.org > > Subject: Re: [Serusers] lookup 2 uri > > > > > > Hi Atle, > > > > Alias doesn't replace uri with its real location. I'd like to use > > lookup("location") to find out its real location. For > > example, if 102(a)test.com is at 102(a)10.1.1.102 and > > 103(a)test.com is at 103(a)10.1.1.103. I'd like to replace all > > uris with 102(a)10.1.1.102 and 103(a)10.1.1.103 in a single ser > > transaction logic. Would it be possible? > > > > Thanks, > > Richard > > > > > > On 8/1/05, Atle Samuelsen <clona(a)camaro.no> wrote: > > > > > > Hi, > > > > > > you could do something like : > > > > > > serctl alias add 101 sip:102@test.com > > > serctl alias add 101 sip:103@test.com > > > > > > This way the invite would be forked to both 102 and 103, > > > > > > just remember to have both lookup("aliases"); and > > save("aliases"); in > > > your config at the right spot :-) > > > > > > -Atle > > > > > > > > > > > > * Richard Z <rzheng(a)gmail.com> [050802 04:34]: > > > > Hi, > > > > > > > > I am trying to implement a feature to ring two ip phones > > > > simultaneously. For example, when I call 101(a)test.com, it > > rings both > > > > 101(a)test.com and 102(a)test.com. In this case, the ruri is > > > > 101(a)test.com. I can use append_branch 102(a)test.com, but > > is there a > > > > way to lookup the locations of both 101 and 102? > > > > > > > > Thanks, > > > > Richard > > > > > > > > _______________________________________________ > > > > 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] Need to do load balancing with two ser servers.
by Prajyot 03 Aug '05

03 Aug '05

Hi, We are using ser v 0.8.14, we are looking for a solution in which I can use two or more SER servers to do load balancing like a cluster design. Any ideas how can I achieve this? Thanks in advance. Regards, Prajyot Shirsolkar

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[Serusers] Record-Route/Via headers not added when protocol=TCP
by Erich Nahum 03 Aug '05

03 Aug '05

Howdy, I'm playing with SER 0.9.3 with sipp as the workload generator on a bunch of linux boxes. When the transport is UDP, openser adds the record-route and via headers like it is supposed to as a proxy. But when I use TCP these headers are not added. This is a UAC->proxy->UAS scenario. I'm attaching ethereal dumps of the UDP-based UAC, TCP-based UAC, UDP-based UAS, and TCP-based UAS for reference. Any suggestions appreciated. Thanks, -Erich -- Erich M. Nahum IBM T.J. Watson Research Center Research Staff Member P.O. Box 704 nahum(a)watson.ibm.com Yorktown Heights NY 10598

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RE: [Serusers] is SER Dependable for prepaid System ? Anyone using itfor prepaid system?
by Sam Lee 03 Aug '05

03 Aug '05

I have been working for quite some time on deploying SER as a prepaid system. Currently it is not done yet as there's still quite a number of things to be done. Probably we can share a few ideas down the road. Regards, Sam -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Sanjim Sanjim Sent: Tuesday, August 02, 2005 7:02 PM To: serusers(a)lists.iptel.org Subject: [Serusers] is SER Dependable for prepaid System ? Anyone using itfor prepaid system? is SER Dependable for prepaid System ? Anyone using it for prepaid system? I very much like the SER performance in the test environment. Only drawback(maybe advantage if we provide unlimited plan) ) is stateless operation. But I want to know anyone using it in realtime prepaid business? Thanks, JiM _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Call Quality
by John Clements 03 Aug '05

03 Aug '05

I am using SER 0.9.3, MediaProxy 1.4.2 and sometimes the call quality gets pretty bad. Parts of the conversation drops and sometimes there will be a delay of 10 or 15 seconds. The SER server has a public IP and is not behind a firewall. The UAs are spread out at different locations behind different firewalls. Sometimes it works perfect though, with a 45min conversation having no problems whatsoever. Does anyone have any suggestions on what I might be able to do to track down/fix this? -John

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[Users] check_from
by Timur Irmatov 03 Aug '05

03 Aug '05

Hi! I am new to SER, and have following question. How can I provide that: 1) Users can register only their sip addresses; 2) Users can use their username in sip From: field. In my installation sip username and authentication usernames differ a little. For example, sip username could be 112233, but auth username is sip112233. So, I want to be sure that user that can successfully autheticate as sip112233 will not use 345678 username, and always will be using 112233 in From? I need something like check_from, but this function requires identical sip and auth usernames. Is this thing doable? If so, I will be glad to hear how I should do this in openser.cfg or any hints/links that could point me in right direction. Thanks! -- Timur Irmatov, JID: thor(a)jabber.org.au.

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[Serusers] Routing
by Sebastian Kühner 03 Aug '05

03 Aug '05

Hi! Has anyone of you tried to use the database to make routing? I need to get information of the database and then rewritehost with the result... Thanks! Sebastian

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Re: [Serusers] UDP
by Sebastian Kühner 03 Aug '05

03 Aug '05

Hello, Yes, the INVITE goes to PORT 5060. Here is my ngrep: U 2005/08/02 15:43:56.584366 xxx.xxx.xxx.xxx:51403 -> xxx.xxx.xxx.xxx:5060 INVITE sip:10099999999999@xxx.xxx.xxx.xxx:5060 SIP/2.0. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;x-route-tag="tgrp:OutTrunkGroup". From: "anonymous" <sip:54@xxx.xxx.xxx.xxx>;tag=CEA94420-184A. To: <sip:10099999999999@xxx.xxx.xxx.xxx>. Date: Tue, 02 Aug 2005 18:43:56 GMT. Call-ID: 375AFBE2-2BC11DA-B615D293-1A0569D1(a)xxx.xxx.xxx.xxx. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 927669769-45879770-3054686867-436562385. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 6. Remote-Party-ID: <sip:54@xxx.xxx.xxx.xxx>;party=calling;screen=yes;privacy=full. Timestamp: 1123008236. Contact: <sip:54@xxx.xxx.xxx.xxx:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 382. Thanks! Sebastian ----- Original Message ----- From: "Miguel Angel Villar" <mavilla(a)redvoiss.net> To: "'Sebastian Kühner'" <skuehner(a)veraza.com> Sent: Tuesday, August 02, 2005 7:38 PM Subject: RE: [Serusers] UDP Did you chaeck that the INVITE messages from the Cisco server are send to UDP port 5060?. MaV.- > -----Mensaje original----- > De: Sebastian Kühner [SMTP:skuehner@veraza.com] > Enviado el: Martes, 02 de Agosto de 2005 04:11 p.m. > Para: serusers(a)lists.iptel.org > Asunto: [Serusers] UDP > > Hi! > > I'm getting INVITES from another Cisco server, but my ser isn't catching > it. > The difference between a "normal message" (from one of our phones) is, > that > a dump shows us "UDP" like protocol (Cisco)... and SIP from our phones. > > Do I have to respect different SIP signallings?? I made a log on > "INVITES"... but ser isn't catching it :-( > > Thanks! > > Sebastian > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Implementing Codec Translator with Asterisk
by Frank Kostin 03 Aug '05

03 Aug '05

Hi everybody, Looking to implement Codec Translator with Asterisk - loading codec modules API in Asterisk to support transcoding. Has anyone experienced this issue, and does anyone have any suggestions or hint, simple scripts, whatever ? Thanks in advance and kind Regards, Frank --------------------------------- Yahoo! Mail for Mobile Take Yahoo! Mail with you! Check email on your mobile phone.

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[Users] Radius or SQL
by Jayesh Nambiar 03 Aug '05

03 Aug '05

Hi Bogdan, Thanx for the information about using allow_trusted. It works fine. I just wanted to clarify one thing, is using SQL more efficient or Radius is better. I wanted to know performance wise. What would be the overhead in both cases. one more thing, whenever we add a new IP address to the trusted table openser needs to be restarted. Is there any alternative to this. Which type of calls would be affected when we restart openser. please let me know some information about this. Thanx a lot. Jayesh

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[Users] more TCP problems
by Erich Nahum 03 Aug '05

03 Aug '05

Thanks to Bogdan, who pointed out that my UAC was not behaving properly. Having solved that problem in sipp, I now have a new issue. The UAC is opening a TCP connection to the proxy, but the proxy then forwards the INVITE to the UAS over UDP. I'm using the stock openser.cfg file. Any suggestions? Thanks, -Erich -- Erich M. Nahum IBM T.J. Watson Research Center Research Staff Member P.O. Box 704 nahum(a)watson.ibm.com Yorktown Heights NY 10598

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[Serusers] 407 Proxy Authentication Required
by Yiannis Martis 03 Aug '05

03 Aug '05

Hello there, I have installed ser 0.9.3 with vovida stun server and I use Linksys PAP2 UAs. This is what I get with ngrep when I try to make an IP-to-IP call to another UA: Please Help me... U 2005/08/03 13:55:57.380953 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 INVITE sip:2810318308@XXXXXXXXXX.com SIP/2.0. Record-Route: <sip:XX.XX.XX.XX;ftag=7ab32f2bbf2689feo0;lr=on>. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport=5060. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 INVITE. Max-Forwards: 16. Contact: <sip:2810300308@XX.XX.XX.XX:5060>. Expires: 240. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 420. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 13486 13486 IN IP4 XX.XX.XX.XX. s=-. c=IN IP4 XX.XX.XX.XX. t=0 0. m=audio 16420 RTP/AVP 18 0 2 4 8 96 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/8000. 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 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2005/08/03 13:55:57.381212 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport=5060. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a179. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="XXXXXXXXXX.com", nonce="42f0a3e9ea677d46038bdbeccaf8d95c18bce58e". Server: Sip EXpress router (0.9.3 (i386/linux)). Content-Length: 0. Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=4360 req_src_ip=XX.XX.XX.XX req_src_port=5060 in_uri=sip:2810318308@XXXXXXXXXX.com out_uri=sip:2810318308@XXXXXXXXXX.com via_cnt==2". . # U 2005/08/03 13:55:57.381353 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 ACK sip:2810318308@XXXXXXXXXX.com SIP/2.0. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. Call-ID: b521746f-5119658a(a)192.168.1.2. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a179. CSeq: 102 ACK. User-Agent: Sip EXpress router(0.9.3 (i386/linux)). Content-Length: 0. . # U 2005/08/03 13:55:57.664069 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 ACK sip:2810318308@XXXXXXXXXX.com SIP/2.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a179. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 ACK. Max-Forwards: 70. Proxy-Authorization: Digest username="2810300308",realm="XXXXXXXXXX.com",nonce="42f0a3e8f78926d700140a7c490f06dd965ea748",uri="sip:2810318308@XXXXXXXXXX.com",algorithm=MD5,response="ac73134a0de530d527cfbe4fe1903707". Contact: <sip:2810300308@XX.XX.XX.XX:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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[Users] ovpops for serial forking and its cures
by Francesco Bottà 03 Aug '05

03 Aug '05

Hi all, I'm playing around serial forking with avpops module. My goal is to have one PSTN number mapped to N voip number; so, when the first voip number is busy, OPENSER catch 486 reply and then try with the second AOR and so on. But here raise the problem: the OPENSER forward the reply upstream with a code 407 Proxy Auth Required and the PSTN GW send his ACK and all stop here. Below ther's a snippet of mi openser.cfg: modparam("avpops", "avp_aliases", "serial_fork=i:665") [..skipping..] #if RURI is to a voip user: avp_db_load("$ruri", "$serial_fork"); avp_print(); avp_pushto("$ruri","$serial_fork"); lookup("aliases"); if lookup("location") { t_on_reply("1"); t_on_failure("2"); t_relay(); break; }; onreply_route[1] { if (t_check_status("486")) { t_on_failure("2"); break; }; } failure_route[2] { if (t_check_status("486")) { # delete the first element of the list (if any) and pass to second from list avp_delete("$serial_fork"); if (avp_pushto("$ruri", "$serial_fork")) { append_branch(); avp_delete("$serial_fork"); t_on_failure("2"); t_relay(); } } } My DB entries are: +------+------------+------------------------+-----------+------+---------------------------------------+---------------------+ | uuid | username | domain | attribute | type | value | modified | +------+------------+------------------------+-----------+------+---------------------------------------+---------------------+ | | 0662293703 | sipexp.mydomain.org | 665 | 2 | sip:0662293701@sipexp.mydomain.org | 2005-08-02 15:03:27 | | | 0662293703 | sipexp.mydomain.org | 665 | 2 | sip:0662293702@sipexp.mydomain.org | 2005-08-02 15:03:33 | +------+------------+------------------------+-----------+------+---------------------------------------+---------------------+ Do you have a suggestion to resolve this problem? Why OpenSer send upstream 407 after received 486 Busy Here and not try to send a new INVITE to voip user with new R-URI? Many thanx, Verbal

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RE: [Serusers] is SER Dependable for prepaid System ? Anyone using itfor prepaid system?
by Sam Lee 03 Aug '05

03 Aug '05

Jim, Your prepaid system concept is a little different from mine. For mine, the SER is just a middleman. I route all my calls to Cisco CallManager which will then route accordingly. So I just get all my accounting info from the Callmanager CDR. I then run ASP scripts to check the users balances and if it ever finds a low-in-credit user, it will go into the mysql db by odbc and insert the user into a grp (acl is preconfigured to block this grp from accessing the PSTN). For your case, probably the only thing u can do is to match 2 things, the dest. Country code and the different local charges for each country. Do comment about my set up. Regards, Sam -----Original Message----- From: Sanjim Sanjim [mailto:sanjim007@gmail.com] Sent: Wednesday, August 03, 2005 5:33 PM To: Sam Lee Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] is SER Dependable for prepaid System ? Anyone using itfor prepaid system? Sure Sam, I am currently developing a script which will check the credit balance of the user from mysql and accordingly rewrite the uri. If the balance is low... it will forward to SEMS announcement. Like that when SER receives a BYE, the Script should check the destination and total mins and calculate the amount and reduce it from the user credits. Currently the difficult part is the destination country code. I have stored all the codes in the mysql prefix table. But each country have sub codes for states. For example UK proper is only .02 cents, but mobile is .40 cents We have to differentiate the code properly. If you have any other suggestions kindly let me know. Thanks, JiM On 8/3/05, Sam Lee <sam.lee(a)super.net.sg> wrote: > I have been working for quite some time on deploying SER as a prepaid > system. Currently it is not done yet as there's still quite a number > of things to be done. > Probably we can share a few ideas down the road. > > Regards, > Sam > > -----Original Message----- > From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] > On Behalf Of Sanjim Sanjim > Sent: Tuesday, August 02, 2005 7:02 PM > To: serusers(a)lists.iptel.org > Subject: [Serusers] is SER Dependable for prepaid System ? Anyone > using itfor prepaid system? > > is SER Dependable for prepaid System ? Anyone using it for prepaid > system? > > I very much like the SER performance in the test environment. Only > drawback(maybe advantage if we provide unlimited plan) ) is stateless > operation. > > But I want to know anyone using it in realtime prepaid business? > > > > Thanks, > JiM > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Can I know Call ID of UAC from Proxy server?
by Abhijit A.Mahajani 03 Aug '05

03 Aug '05

Hi, Can I know the Call ID of a UAC / UAS who has been registered on th eproxy server? I need to send the MESSAGE to the Particular "UAC" but I do not know the Call ID ? There is no INVITE nor Call Established.But I need to send IM to that particular UAC. As Far As I Know, Call id is mandatory , Hence I need to know the Call id from the prosy server. How should I proceed? Thanks and Regards. Abhijit

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RE: [Users] Radius or SQL
by Jayesh Nambiar 03 Aug '05

03 Aug '05

hi bogdan, Thanx for the help. I am sorry to ask again but how do i use the "trusted_reload" command in the script. I tried different techniques but i could not get the results. please help!! Thanx again

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[Serusers] SER got error during the start due to MySQL connection 2
by ser＠ipcb.net 03 Aug '05

03 Aug '05

Hi all. I have configured ProxySER for my test clients and connected they via SER, but... I can't configure SER - MySQL interoperability. I've read "SIP.edu Cookbook" by ProxySER afterward I made the changes into ser.cfg and db configuration. After SER startup I've got the next log: Aug 2 19:21:21 fwd ser[11846]: Maxfwd module- initializing Aug 2 19:21:21 fwd ser[11849]: new_connection(): Access denied for user 'serro'@'localhost' (using password: YES) Aug 2 19:21:21 fwd ser[11849]: db_init(): Could not create a connection Aug 2 19:21:21 fwd ser[11849]: ERROR: auth_db_bind: unable to connect to the database Aug 2 19:21:21 fwd ser[11849]: init_mod_child(): Error while initializing module auth_db Aug 2 19:21:21 fwd ser[11849]: ERROR: open_uac_fifo: init_child failed Aug 2 19:21:21 fwd ser[11849]: starting fifo server failed I've had the next config: # # $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=7 # debug level (cmd line: -dddddddddd) # fork=yes # log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "//lib/ser/modules/mysql.so" loadmodule "//lib/ser/modules/sl.so" loadmodule "//lib/ser/modules/tm.so" loadmodule "//lib/ser/modules/rr.so" loadmodule "//lib/ser/modules/maxfwd.so" loadmodule "//lib/ser/modules/usrloc.so" loadmodule "//lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "//lib/ser/modules/auth.so" loadmodule "//lib/ser/modules/auth_db.so" loadmodule "//lib/ser/modules/acc.so" loadmodule "//lib/ser/modules/exec.so" # loadmodule "//lib/ser/modules/group.so" # loadmodule "//lib/ser/modules/msilo.so" loadmodule "//lib/ser/modules/print.so" loadmodule "//lib/ser/modules/textops.so" loadmodule "//lib/ser/modules/uri.so" # loadmodule "//lib/ser/modules/uri_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", "db_url", "mysql://ser:mypass@fwd.serout.com/myserdb") modparam("auth", "secret", "myphrase") # -- 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 ------------------- skipped... Where I can dig for solving this trouble? Thank you in advance. Alexander Naumov.

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[Serusers] Ask Billing Server for SER
by Asep Saifudin 03 Aug '05

03 Aug '05

Hi All, Does anyone knows Billing Server for SIP based on SER ? whether for prepaid or postpaid system. I would be glad if there's Opensource SER Billing system but Licensed Billing also ok for me. Thanks & Regards, Asep

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RE: [Serusers] lookup 2 uri
by Samuel Osorio Calvo 03 Aug '05

03 Aug '05

Just one remark: only one of the two URI will be translated to the contact. I'll try to explain... imagine the incoming Req-URI is 1001(a)test.com. When it passes the first lookup, lookup("aliases"); SER will retrieve the two aliases and put one of them as the new req-uri and the other as branch. Req-URI: 1002(a)test.com branch: 1003(a)test.com Doing the second lookup, lookup("location"); will put the contact of 1002(a)test.com as Req-URI (if more than one contact is present, the subsequent bindings will be added as branches) and the 1003(a)test.com will be left as branch untouched. Req-URI: 1002(a)10.1.1.102 branch: 1003(a)test.com This happens because lookup() only takes as input the req-URI. The result is that SER makes a DNS query for test.com, to send the request to 1003(a)test.com, instead of the end-point. If the DNS is properly configured, test.com should point to SER itself, making SER sending the message to itself via the network. Depending on the config file, it can work but you should check the logs to see what is exactly happening. User 1002 will get the message without any problem. NOTE: depending on the order in the database, 1003 and 1002 can exchange roles in upper explanation. As SER is now coded, above can not be changed (as far as I know).... Hope it helps, Samuel. Unclassified. >>> "Zeus Ng" <zeus.ng(a)isquare.com.au> 08/03/05 01:51AM >>> You would need two steps here, looup("aliases"), then lookup("location"). Zeus > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard Z > Sent: Wednesday, 3 August 2005 3:33 AM > To: Atle Samuelsen > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] lookup 2 uri > > > Hi Atle, > > Alias doesn't replace uri with its real location. I'd like to use > lookup("location") to find out its real location. For > example, if 102(a)test.com is at 102(a)10.1.1.102 and > 103(a)test.com is at 103(a)10.1.1.103. I'd like to replace all > uris with 102(a)10.1.1.102 and 103(a)10.1.1.103 in a single ser > transaction logic. Would it be possible? > > Thanks, > Richard > > > On 8/1/05, Atle Samuelsen <clona(a)camaro.no> wrote: > > > > Hi, > > > > you could do something like : > > > > serctl alias add 101 sip:102@test.com > > serctl alias add 101 sip:103@test.com > > > > This way the invite would be forked to both 102 and 103, > > > > just remember to have both lookup("aliases"); and > save("aliases"); in > > your config at the right spot :-) > > > > -Atle > > > > > > > > * Richard Z <rzheng(a)gmail.com> [050802 04:34]: > > > Hi, > > > > > > I am trying to implement a feature to ring two ip phones > > > simultaneously. For example, when I call 101(a)test.com, it > rings both > > > 101(a)test.com and 102(a)test.com. In this case, the ruri is > > > 101(a)test.com. I can use append_branch 102(a)test.com, but > is there a > > > way to lookup the locations of both 101 and 102? > > > > > > Thanks, > > > Richard > > > > > > _______________________________________________ > > > 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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[Users] Looking for openser's gentoo ebuild
by Cheng Zhang 03 Aug '05

03 Aug '05

Hi, there In INSTALL doc [http://www.openser.org/docs/INSTALL] section 3-C, it seems to me that a gentoo's ebuild is already available in the official portage tree for OpenSER. Unfortunately I couldn't find it in my daily-rsync'ed Gentoo box, nor I can find it in http://packages.gentoo.org. Searching in both users and devel mailing lists and Google search couldn't find anything useful either. Am I missing something obvious here? Could someone give me a hint please? Thanks. -- Cheng

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[Serusers] How to rewrite source IP -- SIP packet forwarding.
by Ray Van Dolson 03 Aug '05

03 Aug '05

I'm using nathelper with SER. Here's a quickie diagram of my setup: [SIP ATA] <-------------> [SER] <------------------> [Asterisk] 10.0.201.5 eth1 - 10.0.201.7 eth0 - <public_ip1> eth0 - <public_ip0> I'm using the basic nathelper example configuration, except for one modification. I removed the call to t_relay and replaced it with: t_relay_to("<public_ip1>", "5060"); Basically I want a SIP packet to come in to eth1 on SER server, get rewritten as needed by nathelper, then get sent out to public_ip1 on my Asterisk PBX. This appears to be working except a tcpdump on eth0 on SER shows: 14:39:08.654908 IP 10.0.201.7.5060 > <public_ip1>.5060: UDP, length 629 Obviously the packets will never get back with that source IP address. What command should I be looking at to rewrite this IP so it shows up as <public_ip0> automatically? I'm trying to get SIP ATA's to REGISTER on my Asterisk box but proxy through SER ... (and eventually make use of the rtp proxy). Can post more info if requested, I am new to SER. :-) -- Ray Van Dolson Linux/Unix Systems Administrator Digital Path, Inc.

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[Serusers] How to catch re-invite and signal rtpproxy ??
by jimmy 03 Aug '05

03 Aug '05

Hi all: I am using ser 0.8.14 with rtpproxy and kphone 0.4.1.1, When kphone holds the call, then unholds later, sometimes kphone will change the rtp port in SDP differ from previously used. Therefore, the rtp packet can't be forward correctly via rtpproxy if ser doesn't signal rtpproxy to change the forwarding port mapping I have asked before about this problem, and Zeus Ng have give me some information that the key factor is to catch the re-invite, but i don't know how to verify if a INVITE is really a INVITE or r reINVITE... Is there any method can let ser catch or verify the re-invite message in ser.cfg ? Thanks and regards Jimmy

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[Serusers] Re: Problem on pthread and dtmf
by Nicky 03 Aug '05

03 Aug '05

Please help..... ----- Original Message ----- From: Nicky To: sems(a)lists.iptel.org ; serusers(a)lists.iptel.org Sent: Tuesday, July 05, 2005 6:56 PM Subject: Problem on pthread and dtmf Dear all, Anyone encountered the problem below,please help. Thank you. ----- Original Message ----- From: Nicky To: sems(a)lists.iptel.org ; serusers(a)lists.iptel.org Sent: Thursday, June 16, 2005 3:51 PM Subject: Problem on pthread and dtmf Dear all, 1. Please help, when i run sems -E i experienced this error, what can be the caused of it:~ ~~11352) ERROR: start (AmThread.cpp:73): pthread create failed with code 12 2. There is signal of DTMF which is not pressed but is being send to the server, please see below for the log. Please advice what can be the caused and problem. Thank you in advance. ~~(4395) DEBUG: isdn_audio_eval_dtmf_relative (IvrDtmfDetector.cpp:349): Posting D TMF 0: 5 (5) (4395) DEBUG: postEvent (AmEventQueue.cpp:47): AmEventQueue: trying to post even t (4395) DEBUG: postEvent (AmEventQueue.cpp:55): AmEventQueue: event posted (4395) DEBUG: processEvents (AmEventQueue.cpp:68): before processing event (4395) DEBUG: process (IvrPython.cpp:988): IvrPython processing event... (4395) DEBUG: process (IvrPython.cpp:991): IvrDialog processing event... (4395) DEBUG: onDTMFEvent (IvrPython.cpp:1130): IvrPython::onDTMFEvent(): callin g onDTMFCallback key is 5... (4395) DEBUG: ivrEmptyMediaQueue (IvrPython.cpp:102): IVR: emptying media queue . (4395) DEBUG: IvrMediaEvent (IvrEvents.cpp:31): New Media Event: 1, (4395) DEBUG: postEvent (AmEventQueue.cpp:47): AmEventQueue: trying to post even t (4395) DEBUG: postEvent (AmEventQueue.cpp:55): AmEventQueue: event posted (4395) DEBUG: ivrStopRecording (IvrPython.cpp:132): IVR: stop recording. best regards, nicky

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RE: [Serusers] Start and fnish call
by Sam Lee 03 Aug '05

03 Aug '05

Method==Bye ? Try a ngrep during a call transaction to see what are the header sent. -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Anderson Alves de Albuquerque Sent: Wednesday, August 03, 2005 12:20 AM To: serusers(a)lists.iptel.org Subject: [Serusers] Start and fnish call With SER how can I know when the client SIP finish a call? I need that SER can start a script when the client SIP finish a call. _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] serctl add user error
by Sam Lee 03 Aug '05

03 Aug '05

Try adding an alias in it. Eg. Alias=192.168.2.2 -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of han yibing Sent: Wednesday, August 03, 2005 10:08 AM To: serusers(a)lists.iptel.org Subject: [Serusers] serctl add user error hi serctl add 8800 8800 8800(a)192.168.2.2 using above command to add user ,but get following error ERROR:400;check if you use aliases in SER,in fact ,i don't use alisses in ser.cfg file. please help me. thanks ___________________________________________________________ 雅虎邮箱超强增值服务－2G超大空间、pop3收信、无限量邮件提醒 http://cn.mail.yahoo.com/mail_alert/promo1.html _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] serctl add user error
by han yibing 03 Aug '05

03 Aug '05

hi serctl add 8800 8800 8800(a)192.168.2.2 using above command to add user ,but get following error ERROR:400;check if you use aliases in SER,in fact ,i don't use alisses in ser.cfg file. please help me. thanks ___________________________________________________________ 雅虎邮箱超强增值服务－2G超大空间、pop3收信、无限量邮件提醒 http://cn.mail.yahoo.com/mail_alert/promo1.html

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[Serusers] rtpproxy problem.
by Ray Van Dolson 03 Aug '05

03 Aug '05

I'm running into a problem with SER & rtpproxy trying to get a NAT scenario to work correctly. I'm still learning about SER, so bear with me... :) Here is my network layout in a nutshell: SIP ATA - 10.0.201.196 (private only, no NAT to internet) SER - eth0: 208.208.53.5/24 - eth1: 10.0.201.5/24 Asterisk - eth0: 65.234.34.5/24 *Everything* is configured to proxy through SER to Asterisk currently. My configuration file is here: http://webdev.digitalpath.net/~rayvd/voice/ser.cfg Here's the problem I'm running into: 1. SIP ATA places call (INVITE sent) 2. SER routes INVITE to Asterisk (and modifies c= in the SDP portion to point back to 208.208.53.5 as I understand it). 3. Ring, ring, other side answers. 4. Asterisk sends back a 200 OK message with c= 65.234.34.5 in its SDP message body. 5. My configuration changes this to 208.208.53.5 6. Voice packets now are outbound on SER eth1 with src ip of 208.208.53.5 and dst ip of 10.0.201.196. Obviously this will not work. >From what I understand however, rtpproxy appears to only use one IP address regardless of the direction traffic is going. If I bind rtpproxy to 10.0.201.5 on SER, then the IP in step 6 above will be correct, but the Asterisk server will be trying to send RTP traffic to 10.0.201.5 instead of 208.208.53.5. What am I doing wrong here? Can I run two instances of rtpproxy depending on the direction of the audio? I've tried different parameters to force_rtp_proxy() to no avail. Is there a function to manually change the c= line in the SDP body before it gets sent out? Thanks for any help! -- Ray Van Dolson Linux/Unix Systems Administrator Digital Path, Inc.

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[Serusers] failure_route problem
by davor jovanovic 03 Aug '05

03 Aug '05

Hi all, I have a problem with sending a call to PSTN gateway when user is offline. I always get "404 Not found" what is normal because user is not in location database, but I don't understand why failure_route[1] doesn't work. Below is my ser.cfg so if someone could help. I'm using ser 0.9.3. Any help would be appreciated. Thanks __________________________________ 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 >= max_len ) { sl_send_reply("513", "Message too big"); break; }; if (method=="REGISTER") record_route(); loose_route(); if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("srce.hr", "subscriber")) { www_challenge("srce.hr", "0"); break; }; save("location"); break; }; lookup("aliases"); # if (!uri==myself) { # append_hf("P-hint: outbound alias\r\n"); # route(1); # break; # }; if (method=="INVITE"){ t_on_failure("1"); }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); route(2); break; }; }; if (!t_relay()) { sl_reply_error(); }; } ############ route[2] { # non-Voip -- just send "off-line" if (!(method == "INVITE" || method == "ACK" || method == "CANCEL")) { sl_send_reply("404", "Not Found"); break; }; # forward to gateway rewritehostport("161.53.2.235:5060"); t_relay_to_udp("161.53.2.235", "5060"); } failure_route[1] { revert_uri(); rewritehostport("161.53.2.235:5060"); append_branch(); t_relay_to_udp("161.53.2.235", "5060"); }

3 2

[Serusers] lookup 2 uri
by Richard Z 02 Aug '05

02 Aug '05

Hi, I am trying to implement a feature to ring two ip phones simultaneously. For example, when I call 101(a)test.com, it rings both 101(a)test.com and 102(a)test.com. In this case, the ruri is 101(a)test.com. I can use append_branch 102(a)test.com, but is there a way to lookup the locations of both 101 and 102? Thanks, Richard

3 3

[Users] avp questions
by Klaus Darilion 02 Aug '05

02 Aug '05

Hi! I'm playing with AVPs stored in a database and wonder about the following example found in the doc: avp_db_load("$from","i:678"); This loads all AVPs with attribute==678, regardless of the from URI. Is there a usecase for this? If not, why not use the specifier /uri as default if no specifier is selected. Also, in avp_db_load the source may be an avp-alias. Thus, how do you differ between the From: header URI "$from" and the avp-alias "$from"? regards, Klaus PS: Maybe I'm a little bit too academic ;-)

2 1

[Users] about the next release
by Daniel-Constantin Mierla 02 Aug '05

02 Aug '05

The features we proposed (and many other) for next release are almost done. As you can see in the roadmap (http://openser.org/roadmap.php), mandatory items were achieved, a few optional are still to be worked on and other that would be good to have in next release popped up on the mailing lists. The major item which is still pending is the authentication via uac module. Another thing that should be there is the clean up of serctl + support for the other database types. Other topics discussed on mailing list: - error handling - to be done gradually - IP filters - some basic privacy support - user ACL (group membership improvements) All these are nice features, and they should be included in OpenSER as soon as possible. Anyhow, all the time will show up new good ideas, so at a moment we have to make a new release, and as OpenSER proposed to be, the releases should not make users to wait too long. Since there were many new features in development version, the proposed way to go is to start testing the development version as much as possible from now, even if we do not freeze the CVS right now. After just a few weeks freeze the CVS so we can have a fresh release somewhere in September. Meanwhile new features (some from listed above, and not only) may make their way into release, if they are fast to achieve. The more complex ones, will go probably in the second one. Details about what is new in the development version: http://openser.org/features-0.10.x.php http://cvs.sourceforge.net/viewcvs.py/openser/sip-server/ChangeLog?view=mar… Any comment and suggestions are welcome. Let's start preparing a new release!

1 0

[Serusers] Start and fnish call
by Anderson Alves de Albuquerque 02 Aug '05

02 Aug '05

With SER how can I know when the client SIP finish a call? I need that SER can start a script when the client SIP finish a call.

5 8

[Serusers] URI rewrite utils
by Pavol Segeč 02 Aug '05

02 Aug '05

Hi, Ser has a few cmd which allows rewrite request uri or some part of it. Does SER have some cmd which allow rewrite caller uri or some part of it? For example, my uri is 1767(a)mydomain.com but when I'm calling outside of my VoIP world to PSTN (for example through Cisco call manager) I would like to identify myself by E.164 address format, i.e. +4214151341767(a)mydomain.com. It should be helpful for me change my URI. Thanks, palo

3 2

[Serusers] MARK RODDY/LAKE is out of the office.
by mark.roddy＠lakecommunications.com 02 Aug '05

02 Aug '05

I will be out of the office starting 08/02/2005 and will not return until 08/18/2005. I will not be checking my mail. For Daidalos related issues please contact Stephen Butler. *********************************************************************************** This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify postmaster(a)lakecommunications.com This footnote also confirms that this email message has been scanned for the presence of computer viruses and other security threats. ***********************************************************************************

1 0

[Serusers] is SER Dependable for prepaid System ? Anyone using it for prepaid system?
by Sanjim Sanjim 02 Aug '05

02 Aug '05

is SER Dependable for prepaid System ? Anyone using it for prepaid system? I very much like the SER performance in the test environment. Only drawback(maybe advantage if we provide unlimited plan) ) is stateless operation. But I want to know anyone using it in realtime prepaid business? Thanks, JiM

2 1

[Users] via header not added when transport=TCP?
by Erich Nahum 02 Aug '05

02 Aug '05

Howdy, I'm playing with OPENSER 0.9.5 with sipp as the workload generator on a bunch of linux boxes. When the transport is UDP, openser adds a via header like it is supposed to as a proxy. But when I use TCP no via header is added. This is a UAC->proxy->UAS scenario. Here's the header as seen at the UAS: | UDP message received [633] bytes : | | INVITE sip:service@10.0.1.41:5060 SIP/2.0 | Record-Route: <sip:10.0.1.42;ftag=1;lr=on> | Via: SIP/2.0/UDP 10.0.1.42;branch=z9hG4bKe9cc.cc24a79.0 | Via: SIP/2.0/UDP bronxville1:5060;received=10.0.1.40;branch=z9hG4bK-1-0 | From: sipp <sip:sipp@bronxville1:5060>;tag=1 | To: sut <sip:service@10.0.1.41:5060> | Call-ID: 1-31891@bronxville1 | CSeq: 1 INVITE | Contact: sip:sipp@bronxville1:5060 | Max-Forwards: 16 | Subject: Performance Test | Content-Type: application/sdp | Content-Length: 132 | P-hint: outbound And the corresponding message received at the UAS when using TCP: | TCP message received [495] bytes : | | INVITE sip:service@10.0.1.41:5060 SIP/2.0 | Via: SIP/2.0/TCP bronxville1:5060;branch=z9hG4bK-1-0 | From: sipp <sip:sipp@bronxville1:5060>;tag=1 | To: sut <sip:service@10.0.1.41:5060> | Call-ID: 1-31865@bronxville1 | CSeq: 1 INVITE | Contact: sip:sipp@bronxville1:5060 | Max-Forwards: 70 | Subject: Performance Test | Content-Type: application/sdp | Content-Length: 132 Note the absence of the 2nd Via header. Openser complains about this on the way back to the UAC. Any suggestions appreciated. Thanks, -Erich -- Erich M. Nahum IBM T.J. Watson Research Center Research Staff Member P.O. Box 704 nahum(a)watson.ibm.com Yorktown Heights NY 10598

2 5

[Users] lcr from_gw in reply_route
by Klaus Darilion 02 Aug '05

02 Aug '05

Hi! Currently it is not possible to use the from_gw function in a reply_route. I'm no code expert, but to me it looks like it would be safe also to use it in a reply_route. Can please someone review this? regards, klaus

3 4

[Serusers] Please help me with NAT...
by Yiannis Martis 02 Aug '05

02 Aug '05

1 0

[Users] current calls
by Ozan Blotter 02 Aug '05

02 Aug '05

dear list, i'm still working on the billing system... there'r a few things left like active calls and active registers... from openser database, in "location" table i can read current registered users, and it's easy to make a PHP from there to show "active registers" from a web page. but in addition to it, i wan to see active calls, is there a specific way to do it from ser.cfg or database ? debug=3 fork=yes log_stderror=no listen=212.XXX.104.XXX port=5060 children=4 alias=212.XXX.104.XXX dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/acc.so" modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", 1) modparam("domain", "db_mode", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 180) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-asymmetric-clients") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("registrar", "default_expires", 60) modparam("registrar", "min_expires", 30) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) modparam("acc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc", "failed_transactions", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 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 and Acc section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("212.XXX.104.XXX:5060;nat=yes"); } else if (method!="REGISTER") { if!(uri=~"^sip:833[0-9]*@") { record_route(); setflag(1); } }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; 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" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && client_nat_test("7")) { 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 (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (uri=~"^sip:0[0-9]*@*") { rewritehost("195.XXX.122.XXX"); # this is our PSTN gateway route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); 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) || isflagset(7)) { use_media_proxy(); }; route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

3 2

[Users] Problems with digest authentication
by aimable 02 Aug '05

02 Aug '05

I have been using various versions of SER from last year without any problem but recently I made a new installation of OpenSER 0.9.5. Since then I am having problems with digest authentication from some of the phones. I have a bunch of 186 ATAs and Cisco 7940 phones but they cannot register to the server, while all the soft phones can register successfully. The server says Credentials with given realm not found. I tried to change the realm to localhost and to the the IP address of the server, with no luck. And below is the result of ngrep I tried to grep the messages from the phones and here below is one message from a Cisco 186 ATA which has failed to register ########### Beginning of the capture ################## U PHONEIP:5060 -> SERVERIP:5060 REGISTER sip:SERVERIP SIP/2.0. Via: SIP/2.0/UDP PHONEIP:5060. From: <sip:06090003@SERVERIP;user=phone>;tag=500808430. To: <sip:06090003@SERVERIP;user=phone>. Call-ID: 704382462@PHONEIP. CSeq: 1 REGISTER. Contact: <sip:06090003@PHONEIP:5060;user=phone;transport=udp>;expires=3600. User-Agent: Cisco ATA 186 v2.16.2 ata18x (030829a). Content-Length: 0. # U SERVERIP:5060 -> PHONEIP:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP PHONEIP:5060. From: <sip:06090003@SERVERIP;user=phone>;tag=500808430. To: <sip:06090003@SERVERIP;user=phone>. Call-ID: 704382462@PHONEIP. CSeq: 1 REGISTER. Server: OpenSer (0.9.5 (i386/linux)). Content-Length: 0. Warning: 392 SERVERIP:5060 "Noisy feedback tells: pid=4490 req_src_ip=PHONEIP req_src_port=5060 in_uri=sip:SERVERIP out_uri=sip:SERVERIP via_cnt==1". # U SERVERIP:5060 -> PHONEIP:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP PHONEIP:5060. From: <sip:06090003@SERVERIP;user=phone>;tag=500808430. To: <sip:06090003@SERVERIP;user=phone>;tag=329cfeaa6ded039da25ff8cbb8668bd2.8af0 . Call-ID: 704382462@PHONEIP. CSeq: 1 REGISTER. WWW-Authenticate: Digest realm="talk.artel.rw", nonce="42edb29e1dbcc6fa814dd3396634ed7be68eea56". Server: OpenSer (0.9.5 (i386/linux)). Content-Length: 0. Warning: 392 SERVERIP:5060 "Noisy feedback tells: pid=4490 req_src_ip=PHONEIP req_src_port=5060 in_uri=sip:SERVERIP out_uri=sip:SERVERIP via_cnt==1". Any idea? Aimable

2 3

[Serusers] Xlite not working with SER+SEMS
by yu wang 02 Aug '05

02 Aug '05

Hi, I've been working on conference with Xlite but it is not work. The problem is SEMS starts sending media (default wav) back to Xlite once SEMS receives the INVITE/SDP from Xlite. It seems a bug on Xlite not listening on the media port after sending the INVITE thus it replys with ICMP port unreachable. But I don't see anyone complain this on this list. Is something wrong with my configuration? Please please help me.... Thanks, Pat

3 3

Re: [Serusers] URI rewrite utils
by Klaus Darilion 02 Aug '05

02 Aug '05

Pavol Segeč wrote: > Hi, > > But as I see in rfc3261 such header as Remote-Party-ID is not there, or I'm > wrong? What this headre means? In which rfc it is described? I suppose that It never reached the RFC state, it was only a draft, but is supported by Cisco, Asterisk, ser, Inalp, ... http://www.ietf.org/proceedings/02mar/I-D/draft-ietf-sip-privacy-04.txt > for wished functionality should we need rewrite From and Contact headers Do not rewrite From: as it is forbidden (used for dialog matching in RFC2543) Do not rewrite Contact:, user rpid. regards, klaus PS: cc to the mailing list > > pavol > > >>-----Original Message----- >>From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] >>Sent: Tuesday, August 02, 2005 1:31 PM >>To: Pavol Segeč >>Cc: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] URI rewrite utils >> >>This can be achieved by using the Remote-Party-Id: header (rpid). >> >>Take a look at append_rpid_hf from auth module: >>http://openser.org/docs/modules/0.10.x/auth.html#APPEND-RPID-H >>F-NO-PARAMS >> >>regards, >>klaus >> >>Pavol Segeč wrote: >> >>>Hi, >>> >>>Ser has a few cmd which allows rewrite request uri or some >> >>part of it. >> >>>Does SER have some cmd which allow rewrite caller uri or >> >>some part of it? >> >>>For example, my uri is 1767(a)mydomain.com but when I'm >> >>calling outside >> >>>of my VoIP world to PSTN (for example through Cisco call manager) I >>>would like to identify myself by E.164 address format, i.e. >> >>+4214151341767(a)mydomain.com. >> >>>It should be helpful for me change my URI. >>> >>>Thanks, >>> >>>palo >>> >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >> > >

1 0

[Serusers] Radius Authentication
by Ryan Pagquil 02 Aug '05

02 Aug '05

Guys, I'm testing SER to authenticate with radius. But when I start authenticating I see this on the radius log: rad_recv: Access-Request packet from host 127.0.0.1:1131, id=222, length=262 User-Name = "rpagquil@server4all" Digest-Attributes = "\n\nrpagquil" Digest-Attributes = "\001\014server4all" Digest-Attributes = "\002*42ef48a123c4e75c2d998852eaa5d4fb14bc9917" Digest-Attributes = "\004\020sip:server4all" Digest-Attributes = "\003\nREGISTER" Digest-Response = "1df283adcf333605c0007d8a86a2e332" Service-Type = Sip-Session Sip-URI-User = "rpagquil" Cisco-AVPair = "call-id=CE373D63037311DABFB500E04CAB4AB4@server4all" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 modcall: entering group authorize for request 150 modcall[authorize]: module "preprocess" returns ok for request 150 modcall[authorize]: module "chap" returns noop for request 150 modcall[authorize]: module "eap" returns noop for request 150 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "rpagquil" Digest-Realm = "server4all" Digest-Nonce = "42ef48a123c4e75c2d998852eaa5d4fb14bc9917" Digest-URI = "sip:server4all" Digest-Method = "REGISTER" rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 150 rlm_realm: Looking up realm "server4all" for User-Name = "rpagquil@server4all" rlm_realm: No such realm "server4all" modcall[authorize]: module "suffix" returns noop for request 150 users: Matched rpagquil@server4all at 138 modcall[authorize]: module "files" returns ok for request 150 modcall[authorize]: module "mschap" returns noop for request 150 modcall: group authorize returns ok for request 150 rad_check_password: Found Auth-Type Digest auth: type "Digest" modcall: entering group Auth-Type for request 150 A1 = rpagquil:server4all:test A2 = REGISTER:sip:server4all KD = 94b43b69398cb3ca2eef355e9875d36f:42ef48a123c4e75c2d998852eaa5d4fb14bc9917:33f62c1688cd77a13c84b07b3877bb1c *rlm_digest: FAILED authentication modcall[authenticate]: module "digest" returns reject for request 150 modcall: group Auth-Type returns reject for request 150 auth: Failed to validate the user. Login incorrect: [rpagquil@server4all/<no User-Password attribute>] (from client me2 port 5060)* It says that there is no User-Password attribute contained in my authentication request. What could be the problem? 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

[Users] How to rewrite source IP -- SIP packet forwarding.
by Ray Van Dolson 02 Aug '05

02 Aug '05

I'm using nathelper with OpenSER 0.10.x. Here's a quickie diagram of my setup: [SIP ATA] <-------------> [OpenSER] <--------------> [Asterisk] 10.0.201.5 eth1 - 10.0.201.7 eth0 - <public_ip1> eth0 - <public_ip0> I'm using the basic nathelper example configuration, except for one modification. I removed the call to t_relay and replaced it with: t_relay_to_udp("<public_ip1>", "5060"); Basically I want a SIP packet to come in to eth1 on OpenSER server, get rewritten as needed by nathelper, then get sent out to public_ip1 on my Asterisk PBX. This appears to be working except a tcpdump on eth0 on OpenSER shows: 14:39:08.654908 IP 10.0.201.7.5060 > <public_ip1>.5060: UDP, length 629 Obviously the packets will never get back with that source IP address. What command should I be looking at to rewrite this IP so it shows up as <public_ip0> automatically? I'm trying to get SIP ATA's to REGISTER on my Asterisk box but proxy through OpenSER ... (and eventually make use of the rtp proxy). Can post more info if requested, I am new to SER. :-) -- Ray Van Dolson Linux/Unix Systems Administrator Digital Path, Inc.

2 1

[Serusers] SER <==> SEMS fifo permisions problem
by Pavol Segeč 02 Aug '05

02 Aug '05

Hi All ! I'm implementing SER, SEMS server, I've configured fifo as communication method between them (folowing install manual). In ser.cfg I've enabled conference part in INVITE message route handling as follows : if (uri=~"sip:100.*@") { if (!t_newtran()){ sl_send_reply("500","Could not create transaction"); break; }; t_reply("100","Trying - just wait a minute !"); # assumes that Sems configuration parameter 'socket_name=' # has been set to /tmp/am_sock if(!t_write_req("/tmp/am_fifo","conference")) { t_reply("500","error contacting sems - in uri check 100"); }; break; }; I can make a call on sip:100.*, but I don't have received any announcement and in /var/log/syslog I see following problems with SEMS and more oftenly with fifo permisions: Aug 1 14:30:09 mail Sems[25848]: Error: (SerClient.cpp)(read_from_fifo)(286): no more retries! Aug 1 14:30:09 mail Sems[25848]: Error: (SerClient.cpp)(read_from_fifo)(287): last error: Success Aug 1 14:30:09 mail Sems[25848]: Error: (SerClient.cpp)(send)(137): while reading Ser's response. Aug 1 14:30:09 mail Sems[25848]: Error: (AmRequest.cpp)(send)(219): while sending request to Ser Aug 1 14:30:15 mail Sems[25852]: Error: (SerClient.cpp)(read_from_fifo)(286): no more retries! Aug 1 14:30:15 mail Sems[25852]: Error: (SerClient.cpp)(read_from_fifo)(287): last error: Success Aug 1 14:30:15 mail Sems[25852]: Error: (SerClient.cpp)(send)(137): while reading Ser's response. Aug 1 14:30:15 mail Sems[25852]: Error: (AmRequest.cpp)(send)(219): while sending request to Ser Aug 1 14:33:28 mail /usr/sbin/ser[25731]: ERROR: open_reply_pipe: open error (/tmp/0000660D1443C4FC): Permission denied Aug 1 14:33:28 mail /usr/sbin/ser[25731]: ERROR: fifo_reply: no reply pipe /tmp/0000660D1443C4FC Thanks palo

2 4

[Serusers] can not open dictionary from freeradiusd!
by zhu lizhong 02 Aug '05

02 Aug '05

hello, guys: I installed freeradiusd and SER. now, i want to connect the two together. the problme is that when i include dictionary.ser in raddb/dictionary, and run radiusd -X, then the messages came out as follow: ------------------------------------- read_config_files: reading dictionary Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary[14]: Couldn't open dictionary " /usr/local/share/freeradius/dictionary": No such file or directory Errors reading radiusd.conf -------------------------------------- any one knows how to solve it, please give me hints. thanks lot! zhu

1 0

[Serusers] Problem with digest authentication
by aimable 02 Aug '05

02 Aug '05

Hi serusers, I have been using various versions of SER from last year without any problem but recently I made a new installation of OpenSER 0.9.5. Since then I am having problems with digest authentication from some of the phones. I have a bunch of 186 ATAs and Cisco 7940 phones but they cannot register to the server, while all the soft phones can register successfully. The server says Credentials with given realm not found. I tried to change the realm to localhost, the IP address, with no luck. And below is the result of ngrep I tried to grep the messages from the phones and here below is one message from a Cisco 186 ATA which has failed to register ########### Beginning of the capture ################## U PHONEIP:5060 -> SERVERIP:5060 REGISTER sip:SERVERIP SIP/2.0. Via: SIP/2.0/UDP PHONEIP:5060. From: <sip:06090003@SERVERIP;user=phone>;tag=500808430. To: <sip:06090003@SERVERIP;user=phone>. Call-ID: 704382462@PHONEIP. CSeq: 1 REGISTER. Contact: <sip:06090003@PHONEIP:5060;user=phone;transport=udp>;expires=3600. User-Agent: Cisco ATA 186 v2.16.2 ata18x (030829a). Content-Length: 0. # U SERVERIP:5060 -> PHONEIP:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP PHONEIP:5060. From: <sip:06090003@SERVERIP;user=phone>;tag=500808430. To: <sip:06090003@SERVERIP;user=phone>. Call-ID: 704382462@PHONEIP. CSeq: 1 REGISTER. Server: OpenSer (0.9.5 (i386/linux)). Content-Length: 0. Warning: 392 SERVERIP:5060 "Noisy feedback tells: pid=4490 req_src_ip=PHONEIP req_src_port=5060 in_uri=sip:SERVERIP out_uri=sip:SERVERIP via_cnt==1". # U SERVERIP:5060 -> PHONEIP:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP PHONEIP:5060. From: <sip:06090003@SERVERIP;user=phone>;tag=500808430. To: <sip:06090003@SERVERIP;user=phone>;tag=329cfeaa6ded039da25ff8cbb8668bd2.8af0 . Call-ID: 704382462@PHONEIP. CSeq: 1 REGISTER. WWW-Authenticate: Digest realm="talk.artel.rw", nonce="42edb29e1dbcc6fa814dd3396634ed7be68eea56". Server: OpenSer (0.9.5 (i386/linux)). Content-Length: 0. Warning: 392 SERVERIP:5060 "Noisy feedback tells: pid=4490 req_src_ip=PHONEIP req_src_port=5060 in_uri=sip:SERVERIP out_uri=sip:SERVERIP via_cnt==1". Any idea? Aimable

2 1

[Serusers] No Audio
by Ashutosh Kumar 02 Aug '05

02 Aug '05

Hi, I am using the following ser.cfg with ser and mediaproxy, but I am getting the audio only in one direction. This does not seem a problem of NAT, as the nathelper is correclt rewriting the sdp messages, and the aor shows all clients bound with public IP. Earlier t_on_reply call in route[1] was causing an error sl_reply_error used: I'm terribly sorry, server error occured (1/SL) when one calls from one phone to another. Earlier I was getting proper bi-directional audio when I was using STUN instead of mediaproxy. Is there something wrong with my ser.cfg #-------------------------------------------------- # # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri 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 #debug=7 #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 ---------------------------------- loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/acc.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/domain.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/msilo.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) modparam("uri", "db_url", "sql://ser:heslo@localhost/ser") modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") modparam("auth_radius", "service_type", 15) modparam("acc", "db_url", "sql://ser:heslo@localhost/ser") modparam("acc", "db_missed_flag", 3) modparam("acc", "db_flag", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") modparam("acc", "service_type", 15) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 1) modparam("acc", "log_level", 2) #modparam("acc", "radius_missed_flag", 3) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") # -- rr params -- modparam("rr", "enable_full_lr", 1) modparam("msilo", "registrar", "sip:registrar@mydomainxyz123.com") modparam("msilo", "db_url", "sql://ser:heslo@mydomainxyz123.com/ser") modparam("auth_db", "db_url", "sql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("uri", "subscriber_table", "subscriber") modparam("uri", "uri_table", "uri") modparam("uri", "uri_user_column", "username") modparam("uri", "uri_domain_column", "domain") modparam("uri", "uri_uriuser_column", "uri_user") modparam("uri", "subscriber_user_column", "username") modparam("uri", "subscriber_domain_column", "domain") # !! Nathelper 1 #modparam("registrar", "nat_flag", 6) #modparam("nathelper", "natping_interval", 30) # Ping interval 30 s #modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # !! NAthelper 1 ends # ------------------------- request routing logic ------------------- # main routing logic 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 too big");break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- # 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"){ # && client_nat_test("3")) { record_route_preset("mydomainxyz123.com:5060;nat=yes"); route(3); } else if (method!="REGISTER") {record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- # if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- lookup("aliases"); if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) {setflag(6); #use_media_proxy(); }; ; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- # 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 (!method=="BYE") { force_rport(); fix_nated_contact(); }; if (uri!=myself) {route(1);break;}; if (uri==myself) { if (method=="CANCEL") {route(3); break;}; if (method=="REGISTER") {route(2); break;}; if (method=="MESSAGE") {route(4); break;}; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { if ((method=="INVITE" || method=="ACK") && t_newtran() ){ route(3);break; }; sl_send_reply("404", "Not Foundbusip1"); break; } else {setflag(1);t_relay();break;}; # if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- # t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { #end_media_session(); ; }; # sl_send_reply("404", "balleballeUser Not Found"); # sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); #if (!search("^Contact:\ +\*") && client_nat_test("7")) { # setflag(6); #fix_nated_register(); # force_rport(); #}; #if (!radius_www_authorize("")) { #if (!www_authorize("","subscriber")) { # www_challenge("","0"); # break; #}; #if (!check_to()) { # sl_send_reply("401", "Unauthorized"); # break; #}; consume_credentials(); save("location"); m_dump(); acc_rad_request("Some comment"); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- #if (client_nat_test("3")) { # setflag(7); # force_rport(); # fix_nated_contact(); #}; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); acc_db_request("404 missed call", "missed_calls"); acc_rad_request("404 missed call"); setflag(3); break; }; if (method=="CANCEL") { route(1); 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) || isflagset(7)) { # use_media_proxy(); ; }; route(1); } route[4] { if (!t_newtran()) { sl_reply_error(); break; }; if (m_store("0")) { t_reply("202", "Accepted for Later Delivery"); break; }; t_reply("503", "Service Unavailable"); break; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { #if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; ; }; if (client_nat_test("1")) { fix_nated_contact(); }; ; } Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

2 1

[Serusers] Non eixsting File in mediaproxy modparam
by Ashutosh Kumar 02 Aug '05

02 Aug '05

Hi, In my ser.cfg being used with mediaproxy for NAT traversal, there are two lines reuiqred by this package. modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") But these two files do not exist (*anywhere*) on my machine. I suppose that they are a part of the mediaproxy package.Am I missing them or is the package outdated. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

2 1

[Serusers] Xlite not working with SER+SEMS
by yu wang 02 Aug '05

02 Aug '05

Hi, I've been working on conference with Xlite but it is not work. The problem is SEMS starts sending media (default wav) back to Xlite once SEMS receives the INVITE/SDP from Xlite. It seems a bug on Xlite not listening on the media port after sending the INVITE thus it replys with ICMP port unreachable. But I don't see anyone complain this on this list. Is something wrong with my configuration? Please please help me.... Thanks, Pat

1 0

[Serusers] serctll add error
by chris 02 Aug '05

02 Aug '05

hello i tried running serctl add 1001 1001 chris_cleofe2yahoo.com but i got this error: awk: syntax error near line 1 awk: bailing out near line 1 awk: syntax error near line 1 awk: bailing out near line 1 HA1 calculation failed pls advise. thnks.

4 10

[Serusers] ser and database
by Sebastian Kühner 01 Aug '05

01 Aug '05

Hi! Has anyone of you used a database to make routing? For example: If the caller marks 001 -> Rewritehost("xxx.xxx.xxx.111"); If the caller marks 0049 -> Rewritehost("xxx.xxx.xxx.222"); The two host names are in my postgres database... Once I used a command like this to stop calls of users that don't have credit: if (!exec_msg(' POS1=`expr index "$SIP_HF_FROM" sip: + 4`; LENGTH=`expr index "$SIP_HF_FROM" @ - $POS1`; USER=`expr substr "$SIP_HF_FROM" $POS1 $LENGTH`; QUERY="select credit from subscriber where username=$USER AND credit > 0"; RESULT=`psql -U postgres AAA -t -c "$QUERY"`; if [ -z "$RESULT" ] ; then exit 1; fi ;' )) { sl_send_reply("403", "Sorry, not enough credit to make this call."); break; }; ... but I have problems to get the result of a psql query... is there any way to do it?? Thanks!! Sebastian

1 0

[Serusers] gw-pstn.3.07.cfg with Call-Transfer support
by Bastian Schern 01 Aug '05

01 Aug '05

Hello, I tried out the SER config "gw-pstn.3.07.cfg" from www.onsip.org. Nearly everything works fine but "Call-Transfer" is not working. Is it possible to integrate "Call-Transfer" support into the "gw-pstn.3.07.cfg"? Does anybody integrates this already? Regards Bastian debug=3 fork=yes log_stderror=no listen=192.0.2.13 # INSERT YOUR IP ADDRESS HERE 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/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") 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=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.0.2.13:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else 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(5); 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" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); 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 (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (uri=~"^sip:011[0-9]*@") { # International PSTN route(4); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { # Domestic PSTN route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; route(5); route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("192.0.2.245"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); route(5); route(1); } route[5] { # ----------------------------------------------------------------- # RTP Proxy Enabler # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } route[6] { # ------------------------------------------------------------------------ # ACK Handler # ------------------------------------------------------------------------ # ------------------------------------------------------------------------ # Aliases Section # ------------------------------------------------------------------------ lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; lookup("location"); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

2 2

[Serusers] codec question
by Innocent Evil 01 Aug '05

01 Aug '05

Scenario: IP Phone --> SIP Server --> Asterisk --> (Voice Mailbox) (G723.1, G729) -> (GSM) -> (GSM, G729) GSM coded file Is ipphone with codec g723.1 and g729 cabable to retrived voice mail from the above picture? If not, please explain a little more. Thanks a lot.

2 3

[Serusers] UAC module status of CVS HEAD version
by Innocent Evil 01 Aug '05

01 Aug '05

Hello, Anybody know about the status of UAC module of CVS HEAD version. Does it work? Thanks,

4 4

[Serusers] NAT-flag and multiple branches
by Klaus Darilion 01 Aug '05

01 Aug '05

Hi! I wonder how the NAT flag will be set during lookup("location") in case of multiple branches and some of the contacts are behind NAT and some not. regards, klaus

2 1

AW: [Serusers] dbtext: table not loaded. example desired
by Müller Thorsten 01 Aug '05

01 Aug '05

Hi all, I'm still at dbtext: I have a line in ser.cfg which is now: modparam("auth_db", "db_url","dbtext:///usr/local/etc/ser/dbtext") where I have a file produced with the script from cesc called "subscriber". Given by DEBUG-mode from dbt_load_file the module "dbtext" searches this file in a directory called "/usr/local/etc/ser" and NOT in the dir given by the "modparam"-line Can somebody give me a hint, why this occurs? Another problem I had with the strncpy/strncat of database and table-name. I had to change a line in db_file.c at line 82 from strncpy(path+dbn->len+1,tbn->s,tbn->len) to strncat(path,tbn->s,tbn->len) Is this a bug or is something wrong with my ser.cfg? Thanks Thorsten ser-version: 0.8.14 mt> -----Ursprüngliche Nachricht----- mt> Von: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] mt> Gesendet: Freitag, 29. Juli 2005 10:55 mt> An: Müller Thorsten mt> Cc: serusers(a)lists.iptel.org mt> Betreff: Re: [Serusers] dbtext: table not loaded. example desired mt> mt> mt> Hello, mt> mt> On 07/29/05 11:16, Müller Thorsten wrote: mt> mt> > Hi all, mt> > mt> > I try to set up ser with dbtext as authentication basis. mt> I run in some mt> > problems because the database is not found: mt> > DBT:db_query: table not loaded. mt> > mt> > After some googling a found that Cesc mt> [cesc.santa(a)gmail.com] had also mt> > done some work with dbtext and mailed that to "serusers" mt> or "serdev" mt> > mailinglists mt> > mt> > In a mail from May and June 05 he told that he has mt> written a "serctl" mt> > for dbtext and a script to produce the necessary files. mt> > mt> > Has somebody got these things and can mail me these mt> files! I could not mt> > found them in the mailing history. mt> > mt> > Has somebody also dealed with dbtext and can give me some example mt> > files for ser.cfg and dbtext files for mt> "authentication/subscriber" and mt> > "location". mt> > mt> rhe readme file of dbtext module has some examples of table mt> structures mt> as well as a simple config file. mt> mt> http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/mod mt> ules/dbtext/README?rev=HEAD&content-type=text/vnd.viewcvs-markup mt> mt> Daniel mt> mt> > Thanks in advance mt> > mt> > Thorsten Mueller mt> > mt> > ser 0.8.14 cross-compiled for arm mt> > My ser.cfg: mt> > #################################################### mt> > ... mt> > # Added by mt for authentication with dbtext mt> > loadmodule "/usr/local/lib/ser/modules/auth.so" mt> > loadmodule "/usr/local/lib/ser/modules/auth_db.so" mt> > loadmodule "/usr/local/lib/ser/modules/dbtext.so" mt> > mt> > modparam("auth_db", "db_url","dbtext:///var/dbtext/ser") mt> > mt> > # -- auth params -- mt> > modparam("auth_db", "calculate_ha1", 1) mt> > modparam("auth_db", "password_column", "password") mt> > modparam("auth_db", "user_column", "username") mt> > modparam("auth_db", "domain_column", "domain") mt> > ... mt> > mt> > if (uri==myself) { mt> > mt> > if (method=="REGISTER") { mt> > mt> > if (!www_authorize("domain.com", "subscriber")) { mt> > www_challenge("domain.com", "0"); mt> > break; mt> > }; mt> > save("location"); mt> > break; mt> > } mt> > ... mt> > ##################################################### mt> > mt> > I have a file in /var/dbtext/ser which is called "subscriber": mt> > ##################################################### mt> > username(str) password(str) ha1(str) domain(str) ha1b(str) mt> > suser:supasswd:xxx:domain.com:xxx mt> > ##################################################### mt> > mt> >----------------------------------------------------------- mt> ------------- mt> > mt> >_______________________________________________ mt> >Serusers mailing list mt> >serusers(a)lists.iptel.org mt> >http://lists.iptel.org/mailman/listinfo/serusers mt> > mt> > mt>

2 1

[Serusers] Problem with digest authentication
by aimable 01 Aug '05

01 Aug '05

Hi serusers, I have been using various versions of SER from last year without any problem but recently I made a new installation of OpenSER 0.9.5. Since then I am having problems with digest authentication from some of the phones. I have a bunch of 186 ATAs and Cisco 7940 phones but they cannot register to the server, while all the soft phones can register successfully. The server says Credentials with given realm not found. I tried to change the realm to localhost, the IP address, with no luck. And below is the result of ngrep I tried to grep the messages from the phones and here below is one message from a Cisco 186 ATA which has failed to register ########### Beginning of the capture ################## U PHONEIP:5060 -> SERVERIP:5060 REGISTER sip:SERVERIP SIP/2.0. Via: SIP/2.0/UDP PHONEIP:5060. From: <sip:06090003@SERVERIP;user=phone>;tag=500808430. To: <sip:06090003@SERVERIP;user=phone>. Call-ID: 704382462@PHONEIP. CSeq: 1 REGISTER. Contact: <sip:06090003@PHONEIP:5060;user=phone;transport=udp>;expires=3600. User-Agent: Cisco ATA 186 v2.16.2 ata18x (030829a). Content-Length: 0. # U SERVERIP:5060 -> PHONEIP:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP PHONEIP:5060. From: <sip:06090003@SERVERIP;user=phone>;tag=500808430. To: <sip:06090003@SERVERIP;user=phone>. Call-ID: 704382462@PHONEIP. CSeq: 1 REGISTER. Server: OpenSer (0.9.5 (i386/linux)). Content-Length: 0. Warning: 392 SERVERIP:5060 "Noisy feedback tells: pid=4490 req_src_ip=PHONEIP req_src_port=5060 in_uri=sip:SERVERIP out_uri=sip:SERVERIP via_cnt==1". # U SERVERIP:5060 -> PHONEIP:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP PHONEIP:5060. From: <sip:06090003@SERVERIP;user=phone>;tag=500808430. To: <sip:06090003@SERVERIP;user=phone>;tag=329cfeaa6ded039da25ff8cbb8668bd2.8af0 . Call-ID: 704382462@PHONEIP. CSeq: 1 REGISTER. WWW-Authenticate: Digest realm="talk.artel.rw", nonce="42edb29e1dbcc6fa814dd3396634ed7be68eea56". Server: OpenSer (0.9.5 (i386/linux)). Content-Length: 0. Warning: 392 SERVERIP:5060 "Noisy feedback tells: pid=4490 req_src_ip=PHONEIP req_src_port=5060 in_uri=sip:SERVERIP out_uri=sip:SERVERIP via_cnt==1". Any idea? Aimable

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[Serusers] Adding Asterisk Support for Voice Messages
by ntandoh＠termilink.net 01 Aug '05

01 Aug '05

Does anyone have info on adding Asterisk Support for Voice Messages, that page is missing from the SIP - Getting Started - Issue 04a.doc www.ONsip.org ?

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[Serusers] sip_method
by Szabolcs Szasz 01 Aug '05

01 Aug '05

Hi my friends! I have been tested my SER config for a few weeks (ser-0.8.14). It's OK, but in the ser's MySQL database in the acc table appears a few row with sip_method=PRACK and sip_method=INFO (there are > 1100 rows with INVITE,ACK and BYE). What is this meaning? I cannot regenerate this row. can anybody help me? Thanks Szabolcs

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[Serusers] Existing Radius Authentication
by Ryan Pagquil 01 Aug '05

01 Aug '05

Hi, Can I use my existing radius server as my login authentication for ser? The existing radius uses the system to read the user accounts, but explained on the radius howto i must create the user accounts on users file of the freeradius. Please help. 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

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RE: [Serusers] 1 way voice issue
by Sam Lee 01 Aug '05

01 Aug '05

Could the 183 session progress have anything to do with the one way voice ? ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, August 01, 2005 4:54 PM To: Sam Lee; serusers(a)lists.iptel.org Subject: Re: [Serusers] 1 way voice issue Very hard to say without a complete ngrep trace. If both parties are on public IPs, there is probably something with the signalling, not the SDP payload. The 183 Session Progress is a so-called provisional reponse. It can be used for early media (to set up the rtp stream before connection). A double 183 sounds strange if it's the proxy that does it. g-) ---- Original Message ---- From: Sam Lee To: serusers(a)lists.iptel.org Sent: Monday, July 25, 2005 12:27 PM Subject: [Serusers] 1 way voice issue > Hi, > > I've recently encountered some problem with my SIP service whereby i > call out to a specific number and i encounter a one way voice. If i'm > the initiator, i cannot hear the other party but he can hear me. At > first i thought it was a return route issue (as i'm going thru NAT) , > so i switch my SIP to a public IP but i still face the same problem. > Its really only that specific PSTN number that i have dialed facing > this problem. The only difference that i can think of is that PSTN > number is on a different route. I did a NGREP from my SIP server for > the PSTN number that works (2-way voice) and the Number that doesn't > work (1-way voice) . The only difference is there is an extra : > > NGW --> Proxy > SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP > Proxy --> SIP Device > SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP > > > for the PSTN number that works (the one with 2-way voice). > > Anyone has idea what does the Session Progress is for ? Or what > problem am i facing ? > > Thanks a mILLION ! > > Regards, > Sam > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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[Users] Using permissions
by jayesh nambiar 01 Aug '05

01 Aug '05

hi there, Can you please tell me if I can use the permissions module to trust the requests coming towards my SIP server using source IP. But my problem is I want to use flat files to store the source IPs I trust. Is it possible And if the IPs are trusted I need to forward them without any authentication. I need to store the trusted IPs in flat files since I dont need the overhead of querying DB. Please help me out. Thanx --------------------------------- Start your day with Yahoo! - make it your home page

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[Serusers] Any known problems with running SER behind NAT?
by Alan 01 Aug '05

01 Aug '05

Hi List, I wonder if you could share your experience with me of running SER behind NAT. Are there any known problems with doing it this way? - Server side NAT and have SER fix NAT'ed clients and (Media)Proxying them where needed. My specific setup is as follows:- SER(0.9.13) with a static private IP Firewall performing static on-to-one NAT Users will be on public IP's/(behind NAT) Does this cause any problems with the SIP/SDP packets? Also it is the intention to run Asterisk in the same way to provide voicemail, will this setup produce any foreseeable problems? Many Thanks, Alan

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[Serusers] 1 way voice issue
by Sam Lee 01 Aug '05

01 Aug '05

Hi, I've recently encountered some problem with my SIP service whereby i call out to a specific number and i encounter a one way voice. If i'm the initiator, i cannot hear the other party but he can hear me. At first i thought it was a return route issue (as i'm going thru NAT) , so i switch my SIP to a public IP but i still face the same problem. Its really only that specific PSTN number that i have dialed facing this problem. The only difference that i can think of is that PSTN number is on a different route. I did a NGREP from my SIP server for the PSTN number that works (2-way voice) and the Number that doesn't work (1-way voice) . The only difference is there is an extra : NGW --> Proxy SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP Proxy --> SIP Device SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP for the PSTN number that works (the one with 2-way voice). Anyone has idea what does the Session Progress is for ? Or what problem am i facing ? Thanks a mILLION ! Regards, Sam

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[Serusers] STUN for NAT traversal...
by Roger Lewau 01 Aug '05

01 Aug '05

I'm trying to use STUN for NAT traversal for all clients not using 1 to 1 NATing, and mediaproxy for the rare few who does.But it seems that all traffic is proxied anyway, dont really understand why. I do call the routeblock for mediaproxy before relaying the call, but since STUN is used it should not detect that NAT is in place. Or am I missing some logic? Any Idea on how to accomplish this? Kind regards Roger - Getting one step closer to a really god service everyday ;)

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[Serusers] Mediaproxy Load Balacing
by smy＠addm.com 01 Aug '05

01 Aug '05

Hi all, I've tried playing with Mediaproxy for sometime now, and thou with some clarification from Greger, I'm still having some problems with getting it up and running. I'm trying to have 1 SER with multiple Mediaproxy, and here's my configuration. I'm desperate since I've experienced no improvement with the system after over a week playing with it! :< Any help is really appreciated. --------- ser.cfg --------- loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") ------ mediaproxy.ini (on SER server) ------ [Dispatcher] socket = /var/run/proxydispatcher.sock group = ser defaultProxy = server's own IP [MediaProxy] listen = server's own IP allow = proxy servers' ip proxyIP = server's own ?IP portRange = 30000:60000 ;TOS = 0xb8 idleTimeout = 60 holdTimeout = 1800 ;forceClose = 0 accounting = off -------- mediaproxy.ini (on remote proxy server) -------- [Mediaproxy] start = yes socket = /var/run/proxydispatcher.sock group = ser listen = server's own IP:5080 allow = SER server's IP proxyIP = server's own IP portRange = 35000:65000 TOS = 0xb8 idleTimeout = 60 holdTimeout = 3600 forceClose = 0 accounting = off

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[Serusers] NAT Problem (Mediaproxy)... Pls help
by Yan Yu Lim 01 Aug '05

01 Aug '05

Hi Guys, I have a NAT problem, whereby I cannot get the RTP packets from a NATed user. Hence, I am trying to implement the Mediaproxy solution from ONSIP. 1. Mediaproxy and Ser are installed on the same PC, say 202.123.45.85. I've set SER to listen at 202.123.45.85, port 5060. How should I configure the ini file for Mediaproxy? Can someone please provide me with a simple example of how the mediaproxy ini file should be configured? 2. Assuming that I set mediaproxy to listen at 202.123.45.85, port 5061, & SER at 202.123.45.85, port 5060, how should my client's sip proxy setting be set? Is it 202.123.45.85:5061? 3. Mediaproxy has settings for proxy dispatcher & mediaproxy. What should be the settings for them in my case? 4. Do I need to configure additional files other than ser.cfg & mediaproxy.ini? 5. Lastly, for SER I am using : modparam("usrloc", "db_mode", 0) to check for all registered users online, do I use the command : serctl ul show Many thanks any suggestions. Regards, Zan

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[Serusers] Mediaproxy -- pls help!
by yy lim 01 Aug '05

01 Aug '05

Hi guys, I'm trying to get my mediaproxy working without much success. Both mediaproxy and ser are installed on the same machine, i.e. 201.23.13.145 for example. SER is listening on port 5060, and mediaproxy is set up as below. Please kindly advise. Rgrds, Zan ----- Mediaproxy.ini ------ [MediaProxy] start = yes socket = /var/run/mediaproxy.sock group = ser listen = 201.23.13.145:5062 allow = any proxyIP = 201.23.13.145 portRange = 5062:5064 TOS = 0xb8 idleTimeout = 60 holdTimeout = 3600 forceClose = 0 accounting = off ------- ser.cfg ------- debug=3 fork=yes listen=201.23.13.145 log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:XXXXX@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/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/exec.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" # -- mediaproxy NAT modules --- loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- # -- auth_radius params -- modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:XXXXXX@localhost/ser") modparam("acc", "db_url", "mysql://ser:XXXXXX@localhost/ser") # -- usrloc params -- # store user location in memory, not using database modparam("usrloc", "db_mode", 0) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", 1) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # mysql accounting # -- acc params -- modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 1) modparam("acc", "failed_transactions", 1) # -- tm params -- # set time for which ser will be waiting for a final response; # fr_inv_timer sets value for INVITE transactions, # fr_timer for all others modparam("tm","fr_inv_timer",15) modparam("tm","fr_timer",10) modparam("tm","fr_inv_timer_avp", "inv_timeout") #modparam("permissions", "db_mode", 1) #modparam("permissions", "trusted_table", "trusted") # -- Mediaproxy params -- modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("registrar", "nat_flag", 6) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # main routing logic route{ # ----------------------------- # Sanity check # ----------------------------- # 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 > max_len ) { sl_send_reply("513", "Message too big"); break; }; # ----------------------------- # Record Route Section # ----------------------------- # perform NAT check if (method=="INVITE" && client_nat_test("3")) { record_route_preset("201.23.13.145:5060;nat=yes"); log(1,"Invite + NAT test- Record Route\n"); } # 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 else if(method!="REGISTER") { record_route(); }; # ------------------------------- # Call Tear Down Section # ------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); log(1,"Tear Down Section\n"); }; # ------------------------- # loose-route processing # ------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; log(1,"Loose route processing\n"); }; route(1); break; }; setflag(1); # ----------------------------- # Call Type Processing Section # ----------------------------- if(uri != myself) { route(5); route(1); break; }; if (uri==myself) { if(method == "ACK") { route(6); break; } else if (method =="CANCEL") { route(3); break; } else if(method=="INVITE") { route(3); t_on_failure("1"); break; } else if(method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; route(1); } # ------------------------------- # Route Processing Section # ------------------------------- # ------------------------------ # Default Message Handler # ------------------------------ route[1] { t_on_reply("1"); if(!t_relay()) { if(method=="INVITE" || method=="ACK") { end_media_session(); log(1,"Route 1 default message handler\n"); }; sl_reply_error(); }; } # REGISTER Message Handler # ---------------------------------------------------------------- route[2] { sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); log(1,"Nated register\n"); }; 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(); }; } # ----------------------------------------------------------------- # CANCEL & INVITE Message Handler # ----------------------------------------------------------------- route[3] { if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if(method=="INVITE" && !(src_ip==10.10.10.4)) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { # attempt handoff to PSTN if (uri=~"^sip:8[0-9]*@") { log(1, "Forwarding to PSTN\n"); route(4); break; }; # retrieve voicemail if (uri=~"^sip:2[0-9]*@") { log(1, "Retrieving voicemail\n"); route(7); break; }; sl_send_reply("404", "User Not Found!"); break; }; if(method=="CANCEL") { route(1); break; }; route(5); route(1); } # ----------------------------- # PSTN Handler # ----------------------------- route[4] { rewritehostport("10.10.10.4:5060"); avp_write("i:45", "inv_timeout"); # ------------------------------- # RTP Proxy Enabler # ------------------------------- route[5] { if(isflagset(6)||isflagset(7)) { use_media_proxy(); log(1,"Using rtp proxy\n"); }; } # ------------------------------- # ACK # ------------------------------- route[6] { if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found!"); break; }; route(5); route(1); } route(5); route(1); } # ------------------------------ # Asterisk voicemail # ------------------------------ route[7] { #redirect now! rewritehostport("201.23.13.145:5061"); append_branch(); t_relay_to_udp("201.23.13.145","5061"); } failure_route[1] { if(!(method=="BYE") && !(method=="CANCEL")) { log(1,"Activating voicemail!!\n"); }; #revert_uri(); #redirect now! rewritehostport("201.23.13.145:5061"); append_branch(); t_relay_to_udp("201.23.13.145","5061"); } onreply_route[1] { log(1,"On reply route/n"); if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: [Serusers] serctll add error
by chris 01 Aug '05

01 Aug '05

hi kofi, were you able to run and test my config file? any developments? :) much thanks. chris ----- Original Message ----- From: chris To: Kofi Obiri-Yeboah ; serusers(a)lists.iptel.org Sent: Wednesday, July 27, 2005 11:46 AM Subject: Re: [Serusers] serctll add error hi kofi, i did commented out the listen=IP line. but i gives me the same error. ERROR: bad config file (1 errors) by the way, when i tried the hello_world.cfg from onsip i did not encountered any problem and ser was able to run. thnks kofi, i will wait for you reply next week. chris. ----- Original Message ----- From: Kofi Obiri-Yeboah To: chris ; serusers(a)lists.iptel.org Sent: Wednesday, July 27, 2005 11:19 AM Subject: RE: [Serusers] serctll add error Chris, I will run your config file over the weekend and will get back to you. Clearly that fact that you are receiving a bad config file error is an indication that there is a problem. Could you please comment out the listen=IP address line and rerun ?????? KOFI -----Original Message----- From: chris [mailto:chris@fivestartel.com] Sent: Tuesday, July 26, 2005 10:19 AM To: Kofi Obiri-Yeboah; serusers(a)lists.iptel.org Subject: Re: [Serusers] serctll add error hi kofi, attached is my ser.cfg i got this sample script from onsip.org getting started doc. this is their auth-mysql.3.05.cfg sample script, i just change the IP address on the listen variable. running using this cfg file leads me to an error. ERROR: bad config file (1 errors) maybe u can help me fix this. i have no idea how i can give you more details on the error cause that is the only line the system gives me. thnks. debug=3 fork=yes log_stderror=no listen=203.115.128.41 # put your server IP address here 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/uri_db.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("usrloc", "db_mode", 2) 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(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (uri==myself) { 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 # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- 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(); lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } ----- Original Message ----- From: Kofi Obiri-Yeboah To: chris ; serusers(a)lists.iptel.org Sent: Tuesday, July 26, 2005 11:39 PM Subject: RE: [Serusers] serctll add error please send your config file -----Original Message----- From: chris [mailto:chris@fivestartel.com] Sent: Tuesday, July 26, 2005 6:34 AM To: Kofi Obiri-Yeboah; serusers(a)lists.iptel.org Subject: Re: [Serusers] serctll add error hi, i still got the same error bash-2.05# serctl add 1001 1001 chris(a)yahoo.com awk: syntax error near line 1 awk: bailing out near line 1 awk: syntax error near line 1 awk: bailing out near line 1 HA1 calculation failed bash-2.05# pls. advise on how to fix this. thnks. ----- Original Message ----- From: Kofi Obiri-Yeboah To: chris ; serusers(a)lists.iptel.org Sent: Tuesday, July 26, 2005 8:06 PM Subject: RE: [Serusers] serctll add error syntax is <username> <password> <email address> you have a typo in the email address use "@" instead of "2" use shift-2 before yahoo.com HTH -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org]On Behalf Of chris Sent: Tuesday, July 26, 2005 12:58 AM To: serusers(a)lists.iptel.org Subject: [Serusers] serctll add error hello i tried running serctl add 1001 1001 chris_cleofe2yahoo.com but i got this error: awk: syntax error near line 1 awk: bailing out near line 1 awk: syntax error near line 1 awk: bailing out near line 1 HA1 calculation failed pls advise. thnks.

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Re: [Serusers] Problem : Can SER process the reINVITE messages properly?
by Greger V. Teigre 01 Aug '05

01 Aug '05

BTW, I sketched out a few other solutions, where I believe at least a couple should be valid attempts. Did you try those and what were the results? g-) Ricardo Martinez wrote: > Hello Greger and SER users. > About this issue here are some test that i made. I added the > lookup("location") as you recommend me : > > # > > > > ----------------------------------------------------------------- # > Loose Route Section # > ----------------------------------------------------------------- if > (loose_route()) { > > if (has_totag() && (method=="INVITE" || > method=="ACK")) { lookup("location"); > if (isflagset(6) || client_nat_test("3") || > search("^Route:.*;nat=yes")){ > setflag(6); > use_media_proxy(); > }; > }; > > route(1); > break; > }; > > > Unfortunately, i'm still unable to have voice in both ways, but this > modification seems to handle better the reINVITE message. And this > is why. For the normal approach (i mean the initial onsip_pstn > configuration) the second INVITE was not using the mediaproxym as > Greger pointed. Now, as you can see in the debugs that i'm > attaching, the second invite is using the mediaproxy, and therefore > the "c" parameter in the SDP message is being correctly modified by > SER. So, i don't understand why this time the call is failing. Could > it be that mediaproxy is not capable to handle the reINVITEs?. For > example in the debug from mediaproxy (i'm also attaching this one), i > see a warning message like this : warning: Received packet from a > third party: 200.0.0.7:16432. It seems like mediaproxy is not > accepting "voice packets" from the second IP (the one indicated by > the reINVITE). Why? > Also, when the call is established this is the output from the > "session" command : > > Caller Via Called > Status Duration Codec Type Traffic > ---------------------------------------------------------------------------- > -------------------------------------- > 200.0.0.6:17480 - 200.0.0.5:35094 - 200.0.0.4:46115 inactive > 0'24" G729 Audio 0/16.75k/16.75k > > The call is established but between the first IP (Asterisk box) and > the endpoint public IP. > So, could this be a problem with mediaproxy?. I read the changelog > from the version 1.3.1 and this is what i found : > > Changes from version 1.2.1 to 1.3.0 > ----------------------------------- > > - Only create a new session if a request belongs to the first INVITE > (no longer creates new sessions on re-INVITEs) > > could it be related?. > > I also have a general question about the reINVITE. As you can see > the first INVITE has in his URI the domain name for the number : > 5555848114(a)sipvoiss.desarrollo.redvoiss.net (Frame 1 from the debug). > The reINVITE sent by Asterisk has the URI : 5555848114(a)200.0.0.4. Is > this ok?, despite of this the reINVITE is correctly redirected to the > endpoint. (You can check this in the frame 13 from the debug). > > Thanks again. > > Regards, > > Ricardo Martinez.- > > > >> -----Mensaje original----- >> De: Greger V. Teigre [mailto:greger@teigre.com] >> Enviado el: Miércoles, 20 de Julio de 2005 3:06 >> Para: Ricardo Martinez; serusers(a)lists.iptel.org >> Asunto: Re: [Serusers] Problem : Can SER process the reINVITE >> messages properly? >> >> >> Hm. This was a tricky one. I would like to ask a second >> opinion from you >> NAT gurus on my musings below. What is the most appropriate >> response to >> Ricardo's scenario? WARNING to others: This is getting fairly >> technical on >> NAT/SDP... ;-) >> >> The reINVITE from Asterisk has a valid IP address and a valid >> (but new) SDP >> media ip (C=IN IP4 200.0.0.7). nat=yes is correctly not >> present in the >> reINVITE from Asterisk (it is not NATed). Thus >> use_media_proxy is correctly >> not used in the reINVITE, i.e. client_nat_test("3") and test >> on nat=yes will >> not trigger under the has_totag if statement (loose route >> processing): >> >> if (loose_route()) { >> >> if (has_totag() && (method=="INVITE" || method=="ACK")) { >> >> if (client_nat_test("3") || search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> }; >> >> route(1); >> break; >> }; >> >> In the OK from the Linksys: >> U 200.0.0.5:5060 -> 200.0.0.6:5060 >> SIP/2.0 200 OK. >> To: >> <sip:5555848114@sipvoiss.desarrollo.redvoiss.net>;tag=64d7492d >> b018b13ai0. From: "5565625858196" >> <sip:5565625858196@sipvoiss.desarrollo.redvoiss.net>;tag=as5d352c9a. >> Call-ID: >> 0848ef78297116742562204e2c04cbcb(a)sipvoiss.desarrollo.redvoiss.net. >> CSeq: 104 INVITE. >> Via: SIP/2.0/UDP 200.0.0.6:5060;branch=z9hG4bK518d870e. >> Record-Route: <sip:200.0.0.5;ftag=as5d352c9a;lr=on>. >> Contact: 5555848114 <sip:5555848114@200.0.0.4:46081>. >> Server: Linksys/PAP2-2.0.10(LSb). >> Content-Length: 237. >> Content-Type: application/sdp. >> . >> v=0. >> o=- 33068 33068 IN IP4 192.168.15.200. >> s=-. >> c=IN IP4 192.168.15.200. >> t=0 0. >> m=audio 16424 RTP/AVP 18 100 101. >> a=rtpmap:18 G729/8000. >> a=rtpmap:100 NSE/8000. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-15. >> a=ptime:30. >> a=sendrecv. >> >> ...you have 192.168.15.200, a private c= address in SDP, but >> the Contact has >> correctly been fixed (200.0.0.4:46081). So it seems the OK is >> processed >> correctly in onreply even though it does not show up in the >> debug. Flags 6 >> and 7 are not set, so use_media_proxy is correctly not run. >> The problem is >> that flag 6 should have been set (the Linksys is NATed), but >> as the reINVITE >> is not processed using lookup("location") the flag will not >> be picked up >> from usrloc. >> ---------------------------------------- >> >> However, now that Asterisk in the reINVITE is requesting >> another (valid) IP >> as media destination, the question is: what is propor behavior? >> >> The most natural would be to add a lookup("location") after: >> if (has_totag() && (method=="INVITE" || method=="ACK")) >> and then add isflagset(6) to the or test: >> if (isflagset(6) || client_nat_test("3") || >> search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> >> However, I'm not sure the lookup will work, because the R-URI >> is likely to >> be the direct address (5555848114@200.0.0.4:46081) and not >> the registered >> URI (@domain.com). It would be tempting to try to see if we >> somehow could >> make nat=yes set in the early INVITE. We would have to split the >> record_route section in two. One to handle !="REGISTER && >> !="INVITE" with a >> regular record_route() and wait until AFTER >> lookup("location") (maybe in >> route[1], the default handler) to add the following: >> if(method=="INVITE")){ >> if(isflagset(6)) { >> # INSERT YOUR IP ADDRESS HERE >> record_route_preset("192.0.2.13:5060;nat=yes"); >> } else { >> record_route(); >> } >> } >> >> This would make sure that not only dialogs where caller is >> nated, but also >> where callee is nated, will be tagged as nated. This again >> would cause >> nat=yes to be set and rtp proxy forced, further flag 6 is >> set, and thus the >> OK also processed with use_media_proxy(). >> >> BUT: Would this nat=yes cause any problems in other >> scenarios??? I'm not >> sure, but I don't think so. If that is correct, this is >> maybe the right >> solution. >> --------------------------------------- >> OTHER (SIMPLER?) SOLUTIONS: >> ---------- >> Instead you could add ||client_nat_test("1") together with >> isflagset 6 and 7 >> in your reply route: >> if ((isflagset(6) || isflagset(7) || client_nat_test("1")) && >> (status=~"(180)|(183)|2[0-9][0-9]")) >> >> but it will not work because the reINVITE did not get >> use_media_proxy(). >> >> Or, you could instead add fix_nated_sdp("2"); in your onreply route: >> if (client_nat_test("1")) { >> fix_nated_contact(); >> fix_nated_sdp("2"); >> }; >> WARNING! You will have to add else in between this test and >> the previous to >> avoid messing up use_media_proxy(). >> >> This will fix the SDP (c=IN IP4), but it will probably not >> help, because the >> port is still unknown. You could use "3" instead in >> fix_nated_sdp, this >> will add direction=active to the OK. However, another thread >> stated that >> Asterisk does not support direction=active. If it does, it >> will probably >> work and you get a non-proxied stream between Asterisk and >> the Linksys. >> >> Or finally, the third option would be to add the (hopefully) >> fixed IP in the >> loose_route test: >> if (src_ip==200.0.0.6 || client_nat_test("3") || >> search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> >> This would cause the OK automatically to get use_media_proxy. >> >> Ricardo, could you please test these various solutions and >> see if they >> really work or if (where...) I go wrong :-) >> g-) >> >> Ricardo Martinez wrote: >>> Hello Greger. >>> Thanks for your answer on this topic. Now i'm attaching more debug >>> information (the /var/log/messages from mediaproxy, the ngrep output >>> and some xlog statements in the ser.cfg file) beside some >> comments in >>> the file reINVITE_debug_problem.txt. >>> >>>> - Is your session really set up initially (before the reINVITE)? >>>> (mediaproxy reports 0/0/0 bytes) >>> >>> At least i have ringback tone. Then, when the call is answered i >>> have an OK and a ACK message coming to my SER box, then inmediatly >>> the reINVITE message arrives Asterisk box. >>> >>> >>> For what i can see from the debug the "nat=yes" is never reached >>> because the caller has a "valid ip" and therefore the first "if" in >>> the statament >>> >>> if (method=="INVITE" && client_nat_test("3")) { >>> # INSERT YOUR IP ADDRESS HERE >>> record_route_preset("64.76.148.246:5060;nat=yes"); >>> xlog("L_INFO", "time [%Tf] RECORD ROUTE SECTION : >>> invite & client_nat_test(3) TRUE ,record_route_preset [%rm]\n"); >>> } else if (method!="REGISTER") { >>> xlog("L_INFO", "time [%Tf] RECORD ROUTE SECTION : >>> record_route [%rm]\n"); >>> record_route(); >>> }; >>> >>> from the RECORD ROUTE SECTION is FALSE. >>> >>> Also, i don't understand why the second OK (the one from the >>> reINVITE) is not procesed in the ONREPLY ROUTE, or at least i don't >>> see any statement from the "xlog" in the debug. Is this normal? >>> >>> Thanks.! >>> Regards, >>> >>> Ricardo Martinez.- >>> >>> >>>> -----Mensaje original----- >>>> De: Greger V. Teigre [mailto:greger@teigre.com] >>>> Enviado el: Lunes, 18 de Julio de 2005 2:45 >>>> Para: Ricardo Martinez; serusers(a)lists.iptel.org >>>> Asunto: Re: [Serusers] Problem : Can SER process the reINVITE >>>> messages properly? >>>> >>>> >>>> Hi Ricardo, >>>> Thanks for a detailed analysis. Some questions: >>>> - Is your session really set up initially (before the reINVITE)? >>>> (mediaproxy reports 0/0/0 bytes) >>>> - You didn't show the ngrep trace. The script uses nat=yes in >>>> the Route >>>> header of the INVITE to detect a nat'ed client. Can you >>>> verify that the >>>> reINVITE has the nat=yes? >>>> - You haven't showed the mediaproxy log (it will show the >>>> callers reporting >>>> in etc). That could help (default /var/log/messages) >>>> - You can put a log statement in the loose_route section >>>> after the test for >>>> nat=yes to see if use_media_proxy was called >>>> >>>> g-) >>>> >>>> Ricardo Martinez wrote: >>>>> Hello. >>>>> I'm having problems trying to make SER, NAT'd endpoints >>>> and reINVITE >>>>> work together. >>>>> I was using the "gw-pstn3.07.cfg" file from onsip.org to do some >>>>> tests, and this is what i have. In one side i have an >> Asterisk with >>>>> an endpoint registered in it (let's call it A). In the >> other side i >>>>> have a PAP2 under NAT (let's call it B). >>>>> >>>>> >>>>> A ---------- Asterisk ----------- SER ----------- B (NAT'd) >>>>> 200.0.0.7 200.0.0.6 200.0.0.5 >>>>> 10.0.0.4 >>>>> >>>>> When i make a call from "A" to "B" this is what i see (in terms of >>>>> SDP). Looking from SER. >>>>> >>>>> A --------- Asterisk ------------ SER ------------ B (NAT'd) >>>>> Public: >>>>> 200.0.0.4 >>>>> 200.0.0.7 200.0.0.6 200.0.0.5 >>>> Inside: >>>>> 10.0.0.1 >>>>> >>>>> INVITE >>>>> c:200.0.0.6:19996 >>>>> -------------------> >>>>> INVITE >>>>> c:200.0.0.5:35010 >>>>> ----------------> >>>>> >>>>> >>>>> Caller Via Called Status >>>>> Duration Codec Type Traffic >>>>> >>>> -------------------------------------------------------------- >>>> ------------ >>>>> 200.0.0.6:19996 - 200.0.0.5:35010 - ?.?.?.?:? inactive 0'04" >>>>> Unknown Audio 0/0/0 >>>>> >>>>> Total traffic: 0bps/0bps/0bps (in1/in2/out) >>>>> Session count: 1 >>>>> >>>>> So far is ok..........and the phone is answered >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <---------------- (the phone is >>>>> answered) >>>>> OK >>>>> c:200.0.0.5:35010 >>>>> <--------------------- >>>>> >>>>> reINVITE >>>>> c:200.0.0.7:19996 >>>>> ---------------------> >>>>> reINVITE >>>>> c:200.0.0.7:19996 >>>>> ----------------> >>>>> >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <---------------- >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <--------------------- >>>>> >>>>> Finally according to the "session" information : >>>>> >>>>> Caller Via Called >>>>> Status Duration Codec Type Traffic >>>>> >>>> -------------------------------------------------------------- >>>> -------------- >>>>> ---------- >>>>> 200.0.0.6:19996 - 200.0.0.5:35010 - 200.0.0.7:16420 inactive >>>>> 0'26" G729 Audio 0/11.48k/11.48k >>>>> >>>>> Total traffic: 0bps/0bps/0bps (in1/in2/out) >>>>> Session count: 1 >>>>> And the audio is only in one way. :( >>>>> >>>>> So. you can see the reINVITE message apparently is not being >>>>> processed as a call to a NAT'd endpoint and therefore is not using >>>>> the mediaproxy, you can see the second "OK" messsage has >> the invalid >>>>> IP from the NAT'd user is in his sdp information. >>>>> As i said it before i am using the gw-pstn configuration >>>> file from the >>>>> onsip.org and as far as i can remember this configuration >> can handle >>>>> the reINVITE? isn't >>>>> I'm also using the last version of the mediaproxy (1.3.1). >>>>> Can someone tell me what i'm doing wrong? >>>>> >>>>> Hope someone could help me here. >>>>> Thanks in advance. >>>>> Regards... >>>>> >>>>> Ricardo Martinez.- >>>>> >>>>> _______________________________________________ >>>>> Serusers mailing list >>>>> serusers(a)lists.iptel.org >>>>> http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] Problem : Can SER process the reINVITE messages properly?
by Greger V. Teigre 01 Aug '05

01 Aug '05

Ricardo, I think you are correct in your analysis of mediaproxy not accepting a third party. Also, I saw the post from somebody on the newest mediaproxy version. Have you tried it? Version 1.2.1 created a new session for reINVITEs, that's why the nat=yes is used in the ONsip.org configs. Version 1.3.0 added support for detecting that a reINVITE came in and would reuse the session (which I btw don't think is good for your scenario as you change the IP). 1.4.2 seems to address your problem, so it would be interesting to hear if you have any success. As for the reINVITE with the IP address instead of domain, that's ok. g-) Ricardo Martinez wrote: > Hello Greger and SER users. > About this issue here are some test that i made. I added the > lookup("location") as you recommend me : > > # > > > > ----------------------------------------------------------------- # > Loose Route Section # > ----------------------------------------------------------------- if > (loose_route()) { > > if (has_totag() && (method=="INVITE" || > method=="ACK")) { lookup("location"); > if (isflagset(6) || client_nat_test("3") || > search("^Route:.*;nat=yes")){ > setflag(6); > use_media_proxy(); > }; > }; > > route(1); > break; > }; > > > Unfortunately, i'm still unable to have voice in both ways, but this > modification seems to handle better the reINVITE message. And this > is why. For the normal approach (i mean the initial onsip_pstn > configuration) the second INVITE was not using the mediaproxym as > Greger pointed. Now, as you can see in the debugs that i'm > attaching, the second invite is using the mediaproxy, and therefore > the "c" parameter in the SDP message is being correctly modified by > SER. So, i don't understand why this time the call is failing. Could > it be that mediaproxy is not capable to handle the reINVITEs?. For > example in the debug from mediaproxy (i'm also attaching this one), i > see a warning message like this : warning: Received packet from a > third party: 200.0.0.7:16432. It seems like mediaproxy is not > accepting "voice packets" from the second IP (the one indicated by > the reINVITE). Why? > Also, when the call is established this is the output from the > "session" command : > > Caller Via Called > Status Duration Codec Type Traffic > ---------------------------------------------------------------------------- > -------------------------------------- > 200.0.0.6:17480 - 200.0.0.5:35094 - 200.0.0.4:46115 inactive > 0'24" G729 Audio 0/16.75k/16.75k > > The call is established but between the first IP (Asterisk box) and > the endpoint public IP. > So, could this be a problem with mediaproxy?. I read the changelog > from the version 1.3.1 and this is what i found : > > Changes from version 1.2.1 to 1.3.0 > ----------------------------------- > > - Only create a new session if a request belongs to the first INVITE > (no longer creates new sessions on re-INVITEs) > > could it be related?. > > I also have a general question about the reINVITE. As you can see > the first INVITE has in his URI the domain name for the number : > 5555848114(a)sipvoiss.desarrollo.redvoiss.net (Frame 1 from the debug). > The reINVITE sent by Asterisk has the URI : 5555848114(a)200.0.0.4. Is > this ok?, despite of this the reINVITE is correctly redirected to the > endpoint. (You can check this in the frame 13 from the debug). > > Thanks again. > > Regards, > > Ricardo Martinez.- > > > >> -----Mensaje original----- >> De: Greger V. Teigre [mailto:greger@teigre.com] >> Enviado el: Miércoles, 20 de Julio de 2005 3:06 >> Para: Ricardo Martinez; serusers(a)lists.iptel.org >> Asunto: Re: [Serusers] Problem : Can SER process the reINVITE >> messages properly? >> >> >> Hm. This was a tricky one. I would like to ask a second >> opinion from you >> NAT gurus on my musings below. What is the most appropriate >> response to >> Ricardo's scenario? WARNING to others: This is getting fairly >> technical on >> NAT/SDP... ;-) >> >> The reINVITE from Asterisk has a valid IP address and a valid >> (but new) SDP >> media ip (C=IN IP4 200.0.0.7). nat=yes is correctly not >> present in the >> reINVITE from Asterisk (it is not NATed). Thus >> use_media_proxy is correctly >> not used in the reINVITE, i.e. client_nat_test("3") and test >> on nat=yes will >> not trigger under the has_totag if statement (loose route >> processing): >> >> if (loose_route()) { >> >> if (has_totag() && (method=="INVITE" || method=="ACK")) { >> >> if (client_nat_test("3") || search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> }; >> >> route(1); >> break; >> }; >> >> In the OK from the Linksys: >> U 200.0.0.5:5060 -> 200.0.0.6:5060 >> SIP/2.0 200 OK. >> To: >> <sip:5555848114@sipvoiss.desarrollo.redvoiss.net>;tag=64d7492d >> b018b13ai0. From: "5565625858196" >> <sip:5565625858196@sipvoiss.desarrollo.redvoiss.net>;tag=as5d352c9a. >> Call-ID: >> 0848ef78297116742562204e2c04cbcb(a)sipvoiss.desarrollo.redvoiss.net. >> CSeq: 104 INVITE. >> Via: SIP/2.0/UDP 200.0.0.6:5060;branch=z9hG4bK518d870e. >> Record-Route: <sip:200.0.0.5;ftag=as5d352c9a;lr=on>. >> Contact: 5555848114 <sip:5555848114@200.0.0.4:46081>. >> Server: Linksys/PAP2-2.0.10(LSb). >> Content-Length: 237. >> Content-Type: application/sdp. >> . >> v=0. >> o=- 33068 33068 IN IP4 192.168.15.200. >> s=-. >> c=IN IP4 192.168.15.200. >> t=0 0. >> m=audio 16424 RTP/AVP 18 100 101. >> a=rtpmap:18 G729/8000. >> a=rtpmap:100 NSE/8000. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-15. >> a=ptime:30. >> a=sendrecv. >> >> ...you have 192.168.15.200, a private c= address in SDP, but >> the Contact has >> correctly been fixed (200.0.0.4:46081). So it seems the OK is >> processed >> correctly in onreply even though it does not show up in the >> debug. Flags 6 >> and 7 are not set, so use_media_proxy is correctly not run. >> The problem is >> that flag 6 should have been set (the Linksys is NATed), but >> as the reINVITE >> is not processed using lookup("location") the flag will not >> be picked up >> from usrloc. >> ---------------------------------------- >> >> However, now that Asterisk in the reINVITE is requesting >> another (valid) IP >> as media destination, the question is: what is propor behavior? >> >> The most natural would be to add a lookup("location") after: >> if (has_totag() && (method=="INVITE" || method=="ACK")) >> and then add isflagset(6) to the or test: >> if (isflagset(6) || client_nat_test("3") || >> search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> >> However, I'm not sure the lookup will work, because the R-URI >> is likely to >> be the direct address (5555848114@200.0.0.4:46081) and not >> the registered >> URI (@domain.com). It would be tempting to try to see if we >> somehow could >> make nat=yes set in the early INVITE. We would have to split the >> record_route section in two. One to handle !="REGISTER && >> !="INVITE" with a >> regular record_route() and wait until AFTER >> lookup("location") (maybe in >> route[1], the default handler) to add the following: >> if(method=="INVITE")){ >> if(isflagset(6)) { >> # INSERT YOUR IP ADDRESS HERE >> record_route_preset("192.0.2.13:5060;nat=yes"); >> } else { >> record_route(); >> } >> } >> >> This would make sure that not only dialogs where caller is >> nated, but also >> where callee is nated, will be tagged as nated. This again >> would cause >> nat=yes to be set and rtp proxy forced, further flag 6 is >> set, and thus the >> OK also processed with use_media_proxy(). >> >> BUT: Would this nat=yes cause any problems in other >> scenarios??? I'm not >> sure, but I don't think so. If that is correct, this is >> maybe the right >> solution. >> --------------------------------------- >> OTHER (SIMPLER?) SOLUTIONS: >> ---------- >> Instead you could add ||client_nat_test("1") together with >> isflagset 6 and 7 >> in your reply route: >> if ((isflagset(6) || isflagset(7) || client_nat_test("1")) && >> (status=~"(180)|(183)|2[0-9][0-9]")) >> >> but it will not work because the reINVITE did not get >> use_media_proxy(). >> >> Or, you could instead add fix_nated_sdp("2"); in your onreply route: >> if (client_nat_test("1")) { >> fix_nated_contact(); >> fix_nated_sdp("2"); >> }; >> WARNING! You will have to add else in between this test and >> the previous to >> avoid messing up use_media_proxy(). >> >> This will fix the SDP (c=IN IP4), but it will probably not >> help, because the >> port is still unknown. You could use "3" instead in >> fix_nated_sdp, this >> will add direction=active to the OK. However, another thread >> stated that >> Asterisk does not support direction=active. If it does, it >> will probably >> work and you get a non-proxied stream between Asterisk and >> the Linksys. >> >> Or finally, the third option would be to add the (hopefully) >> fixed IP in the >> loose_route test: >> if (src_ip==200.0.0.6 || client_nat_test("3") || >> search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> >> This would cause the OK automatically to get use_media_proxy. >> >> Ricardo, could you please test these various solutions and >> see if they >> really work or if (where...) I go wrong :-) >> g-) >> >> Ricardo Martinez wrote: >>> Hello Greger. >>> Thanks for your answer on this topic. Now i'm attaching more debug >>> information (the /var/log/messages from mediaproxy, the ngrep output >>> and some xlog statements in the ser.cfg file) beside some >> comments in >>> the file reINVITE_debug_problem.txt. >>> >>>> - Is your session really set up initially (before the reINVITE)? >>>> (mediaproxy reports 0/0/0 bytes) >>> >>> At least i have ringback tone. Then, when the call is answered i >>> have an OK and a ACK message coming to my SER box, then inmediatly >>> the reINVITE message arrives Asterisk box. >>> >>> >>> For what i can see from the debug the "nat=yes" is never reached >>> because the caller has a "valid ip" and therefore the first "if" in >>> the statament >>> >>> if (method=="INVITE" && client_nat_test("3")) { >>> # INSERT YOUR IP ADDRESS HERE >>> record_route_preset("64.76.148.246:5060;nat=yes"); >>> xlog("L_INFO", "time [%Tf] RECORD ROUTE SECTION : >>> invite & client_nat_test(3) TRUE ,record_route_preset [%rm]\n"); >>> } else if (method!="REGISTER") { >>> xlog("L_INFO", "time [%Tf] RECORD ROUTE SECTION : >>> record_route [%rm]\n"); >>> record_route(); >>> }; >>> >>> from the RECORD ROUTE SECTION is FALSE. >>> >>> Also, i don't understand why the second OK (the one from the >>> reINVITE) is not procesed in the ONREPLY ROUTE, or at least i don't >>> see any statement from the "xlog" in the debug. Is this normal? >>> >>> Thanks.! >>> Regards, >>> >>> Ricardo Martinez.- >>> >>> >>>> -----Mensaje original----- >>>> De: Greger V. Teigre [mailto:greger@teigre.com] >>>> Enviado el: Lunes, 18 de Julio de 2005 2:45 >>>> Para: Ricardo Martinez; serusers(a)lists.iptel.org >>>> Asunto: Re: [Serusers] Problem : Can SER process the reINVITE >>>> messages properly? >>>> >>>> >>>> Hi Ricardo, >>>> Thanks for a detailed analysis. Some questions: >>>> - Is your session really set up initially (before the reINVITE)? >>>> (mediaproxy reports 0/0/0 bytes) >>>> - You didn't show the ngrep trace. The script uses nat=yes in >>>> the Route >>>> header of the INVITE to detect a nat'ed client. Can you >>>> verify that the >>>> reINVITE has the nat=yes? >>>> - You haven't showed the mediaproxy log (it will show the >>>> callers reporting >>>> in etc). That could help (default /var/log/messages) >>>> - You can put a log statement in the loose_route section >>>> after the test for >>>> nat=yes to see if use_media_proxy was called >>>> >>>> g-) >>>> >>>> Ricardo Martinez wrote: >>>>> Hello. >>>>> I'm having problems trying to make SER, NAT'd endpoints >>>> and reINVITE >>>>> work together. >>>>> I was using the "gw-pstn3.07.cfg" file from onsip.org to do some >>>>> tests, and this is what i have. In one side i have an >> Asterisk with >>>>> an endpoint registered in it (let's call it A). In the >> other side i >>>>> have a PAP2 under NAT (let's call it B). >>>>> >>>>> >>>>> A ---------- Asterisk ----------- SER ----------- B (NAT'd) >>>>> 200.0.0.7 200.0.0.6 200.0.0.5 >>>>> 10.0.0.4 >>>>> >>>>> When i make a call from "A" to "B" this is what i see (in terms of >>>>> SDP). Looking from SER. >>>>> >>>>> A --------- Asterisk ------------ SER ------------ B (NAT'd) >>>>> Public: >>>>> 200.0.0.4 >>>>> 200.0.0.7 200.0.0.6 200.0.0.5 >>>> Inside: >>>>> 10.0.0.1 >>>>> >>>>> INVITE >>>>> c:200.0.0.6:19996 >>>>> -------------------> >>>>> INVITE >>>>> c:200.0.0.5:35010 >>>>> ----------------> >>>>> >>>>> >>>>> Caller Via Called Status >>>>> Duration Codec Type Traffic >>>>> >>>> -------------------------------------------------------------- >>>> ------------ >>>>> 200.0.0.6:19996 - 200.0.0.5:35010 - ?.?.?.?:? inactive 0'04" >>>>> Unknown Audio 0/0/0 >>>>> >>>>> Total traffic: 0bps/0bps/0bps (in1/in2/out) >>>>> Session count: 1 >>>>> >>>>> So far is ok..........and the phone is answered >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <---------------- (the phone is >>>>> answered) >>>>> OK >>>>> c:200.0.0.5:35010 >>>>> <--------------------- >>>>> >>>>> reINVITE >>>>> c:200.0.0.7:19996 >>>>> ---------------------> >>>>> reINVITE >>>>> c:200.0.0.7:19996 >>>>> ----------------> >>>>> >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <---------------- >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <--------------------- >>>>> >>>>> Finally according to the "session" information : >>>>> >>>>> Caller Via Called >>>>> Status Duration Codec Type Traffic >>>>> >>>> -------------------------------------------------------------- >>>> -------------- >>>>> ---------- >>>>> 200.0.0.6:19996 - 200.0.0.5:35010 - 200.0.0.7:16420 inactive >>>>> 0'26" G729 Audio 0/11.48k/11.48k >>>>> >>>>> Total traffic: 0bps/0bps/0bps (in1/in2/out) >>>>> Session count: 1 >>>>> And the audio is only in one way. :( >>>>> >>>>> So. you can see the reINVITE message apparently is not being >>>>> processed as a call to a NAT'd endpoint and therefore is not using >>>>> the mediaproxy, you can see the second "OK" messsage has >> the invalid >>>>> IP from the NAT'd user is in his sdp information. >>>>> As i said it before i am using the gw-pstn configuration >>>> file from the >>>>> onsip.org and as far as i can remember this configuration >> can handle >>>>> the reINVITE? isn't >>>>> I'm also using the last version of the mediaproxy (1.3.1). >>>>> Can someone tell me what i'm doing wrong? >>>>> >>>>> Hope someone could help me here. >>>>> Thanks in advance. >>>>> Regards... >>>>> >>>>> Ricardo Martinez.- >>>>> >>>>> _______________________________________________ >>>>> Serusers mailing list >>>>> serusers(a)lists.iptel.org >>>>> http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] LCR on onsip.org - No DB tables created?
by Dave 30 Jul '05

30 Jul '05

I used exactly that script. But It did not create any LCR related tables.!!! --- Ricardo Martinez <rmartinez(a)redvoiss.net> wrote: > Hello Dave. > If i'm not wrong, there is a shell script included > in the last SER > version to create all the DB tables related to SER. > I think there is in : > /sip_router/scripts/ and is called "ser_mysql.sh" > You can try that!. > > Regards, > Ricardo Martinez > > > > > -----Mensaje original----- > > De: Dave [mailto:ddx66@yahoo.com] > > Enviado el: Viernes, 29 de Julio de 2005 11:16 > > Para: serusers(a)lists.iptel.org > > Asunto: [Serusers] LCR on onsip.org - No DB tables > created? > > > > > > Hi, I downloaded and installed the ser-0.9.3 from > > onsip.org with the intention of using LCR. > Although, > > the LCR module was complied, I do not see the > tables > > lcr, gw in the DB? How do I create them? > > Thanks > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam > protection around > > http://mail.yahoo.com > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

3 3

RE: [Serusers] LCR on onsip.org - No DB tables created?
by Dave 30 Jul '05

30 Jul '05

Thanks Ricardo - Do I need to add these to "versions" table in ser database? If yes, what is the version number for these tables? Dave --- Ricardo Martinez <rmartinez(a)redvoiss.net> wrote: > Hello Dave. > I was almost sure that the tables for LCR were > defined there!, my mystake. > But, i have the table definitions (i can't rememeber > where i found it). You > can check with Juha if there is a mistake in the > tables. > > # > # Table structure for table 'gw' > # > > CREATE TABLE gw ( > gw_name VARCHAR(128) NOT NULL, > ip_addr INT UNSIGNED NOT NULL, > port SMALLINT UNSIGNED, > grp_id INT UNSIGNED NOT NULL, > PRIMARY KEY (gw_name), > KEY (grp_id) > ); > > # > # Table structure for table 'gw_grp' > # > > CREATE TABLE gw_grp ( > grp_id INT UNSIGNED NOT NULL AUTO_INCREMENT > PRIMARY KEY, > grp_name VARCHAR(64) NOT NULL > ); > > # > # Table structure for table 'lcr' > # > > CREATE TABLE lcr ( > prefix varchar(16) NOT NULL, > from_uri varchar(128) NOT NULL DEFAULT '%', > grp_id INT UNSIGNED NOT NULL, > priority TINYINT UNSIGNED NOT NULL, > KEY (prefix), > KEY (from_uri), > KEY (grp_id) > ); > > > Regards > Ricardo.- > > > -----Mensaje original----- > > De: Dave [mailto:ddx66@yahoo.com] > > Enviado el: Viernes, 29 de Julio de 2005 11:30 > > Para: Ricardo Martinez; serusers(a)lists.iptel.org > > Asunto: RE: [Serusers] LCR on onsip.org - No DB > tables created? > > > > > > I used exactly that script. But It did not create > any > > LCR related tables.!!! > > > > > > --- Ricardo Martinez <rmartinez(a)redvoiss.net> > wrote: > > > > > Hello Dave. > > > If i'm not wrong, there is a shell script > included > > > in the last SER > > > version to create all the DB tables related to > SER. > > > I think there is in : > > > /sip_router/scripts/ and is called > "ser_mysql.sh" > > > You can try that!. > > > > > > Regards, > > > Ricardo Martinez > > > > > > > > > > > > > -----Mensaje original----- > > > > De: Dave [mailto:ddx66@yahoo.com] > > > > Enviado el: Viernes, 29 de Julio de 2005 11:16 > > > > Para: serusers(a)lists.iptel.org > > > > Asunto: [Serusers] LCR on onsip.org - No DB > tables > > > created? > > > > > > > > > > > > Hi, I downloaded and installed the ser-0.9.3 > from > > > > onsip.org with the intention of using LCR. > > > Although, > > > > the LCR module was complied, I do not see the > > > tables > > > > lcr, gw in the DB? How do I create them? > > > > Thanks > > > > > > > > > __________________________________________________ > > > > Do You Yahoo!? > > > > Tired of spam? Yahoo! Mail has the best spam > > > protection around > > > > http://mail.yahoo.com > > > > > > > > > _______________________________________________ > > > > Serusers mailing list > > > > serusers(a)lists.iptel.org > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam > protection around > > http://mail.yahoo.com > > > ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

2 2

[Serusers] Parrallel forking using SER and Mediaproxy
by Ramu Yadav 30 Jul '05

30 Jul '05

Hi all, I am experiencing some problems while configuring SER in order to fork between different contacts (a SIP UA and a SIP Gateway) and passing through a mediaproxy: The scenario is the following: (3 calls) UA1 -> SER : INVITE sip:ramu at xyz.com SER forks: SER -> UA2 : INVITE sip:ramu@[local-ip] {SDP is changed by mediaproxy module} SER -> GW: INVITE sip:PSTN-NUMBER@[gateway-ip] {SDP is changed by mediaproxy module} SER -> GW: INVITE sip:PSTN-NUMBER@[gateway-ip] {SDP is changed by mediaproxy module GW -> SER: 183 Session Progress SER -> UA1: 183 Session Progress {SDP is changed by mediaproxy module} Audio session with the mediaproxy is Active GW -> SER: 183 Session Progress SER -> UA1: 183 Session Progress {SDP is changed by mediaproxy module} Audio session with the mediaproxy is Active UA2 -> SER: 180 Ringing SER -> UA1: 180 Ringing UA2 -> SER: 200 OK SER -> UA1: 200 OK {SDP is changed by mediaproxy module} Audio session with the mediaproxy is Active. SER -> GW: CANCEL .... The call gets connected, but the problem is that the audio session is mute: onreply_route[1] { if (status=~"(183)|2[0-9][0-9]") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; }; Can you help me on this ? Thank you a lot in advance, -- Ramu Yadav

2 2

[Users] feature request
by Klaus Darilion 30 Jul '05

30 Jul '05

Hi! In many situations it would be useful to verify the destination IP address of a message before sending the message. E.g. I want to make sure that accounting and authentication was applied before sending a request to a gateway. loose_route and DNS resolving make it impossible to handle all scenarios in the config script. Thinking about the problem I got the following idea: The tm module should, after all DNS lookups, verify the dest. IP against a blacklist (e.g. the GW table from lcr module). If the dest. IP is on the blacklist, the message will only be sent if a certain flag is set. This flag will be set in the config script after all checks (acc, auth...) has been applied. What do you think about this idea? Any other ideas for this problem? regards, klaus

3 11

[Serusers] jabber to simple transport
by David Everly 30 Jul '05

30 Jul '05

I may not have the subject right, but I'm looking for how to do the following: I have a jabber client and a jabber server on localhost. Everyone else in the company uses sip/simple by connecting Messenger to olcs.company.com. I would like to make it so that my jabber client can browse to or discover a transport that will behave as if it is a Messenger client to olcs.company.com. Unfortunately, I don't know the correct vocabulary to even use to express this idea what I have read doesn't seem to map (and I have made some attempt at this on my Debian system, but it seems that I have nothing working). Is all this even possible, and if so, how can it be achieved? Thanks in advance, Dave.

2 2

[Users] debian package conflict with ser
by Klaus Darilion 29 Jul '05

29 Jul '05

Hi! ser and openser debian packages can't be installed on the same machine as the gen_ha1 utility conflicts. What about renaming it to gen_ha1_openser? regards, klaus

2 2

[Serusers] Getting Started Workshop - Phone Help
by Mer lot 29 Jul '05

29 Jul '05

Were in the early stage of building up an initial SER configuration and find that even with the abundant information on onsip.orgs site and posts, a lot of things that can go wrong. We want to build up the full authenticated/NATed/ system specified in the onsip.orgs excellent 0.9.3 documentation and wed like to cooperate with several other people wanting to do the same in a sort of small virtual workshop. The goal is to help each other and accelerate our mutual progress and the emphasis is on direct communication (VOIP of course) or even SSH or VNC into a remote server for assistance. The config wed like to replicate in steps is everything in the SER Getting Started document up to and including the config on page 36 (5.1.1) Mediaproxy Transparent NAT Traversal. Were open to the Rtpproxy version also but the guide provides more info on Mediaproxy. In a subsequent step wed like to implement the PSTN gateway similar to that on page 52 though maybe not exactly as described there. Anyway that's a downstream step. For greatest benefit we think it would be useful to implement with a common Linux distro/build and UAs, which should all be freely available. Can discuss it more with interested participants. Once completed will be glad to contribute the notes and experiences back to onsip.org to the extent it might help others. Email us if you're interested. Will share VOIP numbers and more info. Regards, MC --------------------------------- Start your day with Yahoo! - make it your home page

1 0

RE: [Serusers] ser.pif file not found
by Sam Lee 29 Jul '05

29 Jul '05

What did u run it as ? Root ? If not , then probably when u start ser , it cant write to /var/run/ser.pid , which is the default path and requires root permission. Sam ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Colin Jordan Sent: Friday, July 29, 2005 5:28 PM To: serusers(a)lists.iptel.org Subject: [Serusers] ser.pif file not found I cannot control SER via serctl stop/start due to the fact that when it runs via this command it is not finding a ser.pid. I have searched for ser.pid on my system and it is not there (was just making sure that serctl was not looking in the wrong spot for the PID file). Any suggestions?? 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 2

RE: [Serusers] LCR on onsip.org - No DB tables created?
by Ricardo Martinez 29 Jul '05

29 Jul '05

Hello Dave. I was almost sure that the tables for LCR were defined there!, my mystake. But, i have the table definitions (i can't rememeber where i found it). You can check with Juha if there is a mistake in the tables. # # Table structure for table 'gw' # CREATE TABLE gw ( gw_name VARCHAR(128) NOT NULL, ip_addr INT UNSIGNED NOT NULL, port SMALLINT UNSIGNED, grp_id INT UNSIGNED NOT NULL, PRIMARY KEY (gw_name), KEY (grp_id) ); # # Table structure for table 'gw_grp' # CREATE TABLE gw_grp ( grp_id INT UNSIGNED NOT NULL AUTO_INCREMENT PRIMARY KEY, grp_name VARCHAR(64) NOT NULL ); # # Table structure for table 'lcr' # CREATE TABLE lcr ( prefix varchar(16) NOT NULL, from_uri varchar(128) NOT NULL DEFAULT '%', grp_id INT UNSIGNED NOT NULL, priority TINYINT UNSIGNED NOT NULL, KEY (prefix), KEY (from_uri), KEY (grp_id) ); Regards Ricardo.- > -----Mensaje original----- > De: Dave [mailto:ddx66@yahoo.com] > Enviado el: Viernes, 29 de Julio de 2005 11:30 > Para: Ricardo Martinez; serusers(a)lists.iptel.org > Asunto: RE: [Serusers] LCR on onsip.org - No DB tables created? > > > I used exactly that script. But It did not create any > LCR related tables.!!! > > > --- Ricardo Martinez <rmartinez(a)redvoiss.net> wrote: > > > Hello Dave. > > If i'm not wrong, there is a shell script included > > in the last SER > > version to create all the DB tables related to SER. > > I think there is in : > > /sip_router/scripts/ and is called "ser_mysql.sh" > > You can try that!. > > > > Regards, > > Ricardo Martinez > > > > > > > > > -----Mensaje original----- > > > De: Dave [mailto:ddx66@yahoo.com] > > > Enviado el: Viernes, 29 de Julio de 2005 11:16 > > > Para: serusers(a)lists.iptel.org > > > Asunto: [Serusers] LCR on onsip.org - No DB tables > > created? > > > > > > > > > Hi, I downloaded and installed the ser-0.9.3 from > > > onsip.org with the intention of using LCR. > > Although, > > > the LCR module was complied, I do not see the > > tables > > > lcr, gw in the DB? How do I create them? > > > Thanks > > > > > > __________________________________________________ > > > Do You Yahoo!? > > > Tired of spam? Yahoo! Mail has the best spam > > protection around > > > http://mail.yahoo.com > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com >

1 0

[Serusers] SER + PSTN problem
by Kuti Joe 29 Jul '05

29 Jul '05

Hi to all! I'm a new seruser, and I have a problem with SER and a PSTN gw what I can't solve. I have got 2 UAs, 1 SER server, 1 Cisco PSTN gw (by the side of the voip provider). All of them have public IP. Phone1 -------+ | SER --- PSTN gw | Phone2 -------+ When I call from Phone1 to Phone2 directly, it is working properly. But: If I call from Phone1 to Phone2 via the PSTN gw, then it's ringing out, and when I pick up Phone2 I hear the voice of Phone1, but in Phone1 I can't hear the voice of Phone2. If I call from Phone1 to other telephone numbers, I have the same problem. I would like to proxy the media on the server. I'm useing SER 0.8.14. ser.cfg attached Thank you very much for your effort, and sorry for my poor English... Regards, KutiJoe

2 1

RE: [Serusers] ser.pif file not found
by Sam Lee 29 Jul '05

29 Jul '05

Alternatively u could do a 'touch /var/run/ser.pid ' before doing serctl start -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Gary Reuter Sent: Friday, July 29, 2005 10:21 PM To: Colin Jordan Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] ser.pif file not found If you're using serctl to start ser, it should be setting the pid location properly... but the init.d ser script does not make ser create the pid file. The solution I used was to add OPTIONS="-P /var/run/ser.pid" after the RETVAL line in the init.d/ser script. On 7/29/05, Colin Jordan <cjordan(a)econetwireless.com> wrote: > > I cannot control SER via serctl stop/start due to the fact that when > it runs via this command it is not finding a ser.pid. I have searched > for ser.pid on my system and it is not there (was just making sure > that serctl was not looking in the wrong spot for the PID file). Any > suggestions?? > > _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] exec_msg problem
by Sam Lee 29 Jul '05

29 Jul '05

The problem is solved. Already post a reply earlier on with the solution. It was becos the function don't seem to like my syntax. Anyway, thanks for the response! -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Thomas Britis Sent: Friday, July 29, 2005 8:10 PM To: serusers(a)lists.iptel.org Subject: Re: [Serusers] exec_msg problem Do you run ser as root ? $ l /var/log/syslog -rw-r----- 1 root root 115568 2005-07-28 18:50 /var/log/syslog If you're not root, so you can't write to /var/log/syslog. Sam Lee wrote: > Tried. It didn't work. :< > Still getting the same error. The result still came out, its just that it is followed by the error message. > > 101 INVITE : 3902 <sip:3902@x.x.x.x>;tag=5086b56866af3edo0 to > <sip:967258002@x.x.x.x> Jul 29 16:26:16 abc > /usr/local/sbin/ser[10577]: ERROR: exec_msg: cmd echo "$SIP_HF_CSEQ : > $SIP_HF_FROM to $SIP_HF_ TO" >> /var/log/syslog failed. > exit_status=-1, errno=32: Broken pipe > > > > -----Original Message----- > From: Xavier TRENTIN [mailto:xtrentin@softiphone.net] > Sent: Friday, July 29, 2005 4:11 PM > To: Sam Lee > Subject: Re: [Serusers] exec_msg problem > > Try to quote the expression you want to echo like this: > > exec_msg('echo "$SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO">> > /var/log/syslog'); > > > Xavier. > > Sam Lee a écrit: > > >>*********************** >>Pas de virus detecté dans l'attachement no filename Pas de virus >>detecté dans l'attachement no filename Pas de virus detecté dans >>l'attachement no filename >> >>Votre Mail a été vérifié par InterScan. >>***********-*********** >> >> >> >> >>---------------------------------------------------------------------- >>-- >> >>Sujet: >>[Serusers] exec_msg problem >>Expéditeur: >>"Sam Lee" <sam.lee(a)super.net.sg> >>Date: >>Fri, 29 Jul 2005 11:35:53 +0800 >>Destinataire: >><serusers(a)lists.iptel.org> >> >> >>Hi, >> >>Have anyone seen this kind of error message before : >> >>ERROR: exec_msg: cmd echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO >> >>/var/log/syslog failed. exit_status=-1, errno=32: Broken pipe >> >>This command is actually in my ser.cfg where i capture this few >>environmental variables and push them into my /var/log/syslog. It >>effectively went in, but i still see this error message. >>Here is the original code in ser.cfg : >>exec_msg("echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO >> >>/var/log/syslog"); >> >>Please help! >> >>Regards, >>Sam >> >>---------------------------------------------------------------------- >>-- >> >>_______________________________________________ >>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 > > -- Thomas Storino Britis TCNet Inf e Telecom LTDA _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

RE: [Serusers] LCR on onsip.org - No DB tables created?
by Ricardo Martinez 29 Jul '05

29 Jul '05

Hello Dave. If i'm not wrong, there is a shell script included in the last SER version to create all the DB tables related to SER. I think there is in : /sip_router/scripts/ and is called "ser_mysql.sh" You can try that!. Regards, Ricardo Martinez > -----Mensaje original----- > De: Dave [mailto:ddx66@yahoo.com] > Enviado el: Viernes, 29 de Julio de 2005 11:16 > Para: serusers(a)lists.iptel.org > Asunto: [Serusers] LCR on onsip.org - No DB tables created? > > > Hi, I downloaded and installed the ser-0.9.3 from > onsip.org with the intention of using LCR. Although, > the LCR module was complied, I do not see the tables > lcr, gw in the DB? How do I create them? > Thanks > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[Serusers] LCR on onsip.org - No DB tables created?
by Dave 29 Jul '05

29 Jul '05

Hi, I downloaded and installed the ser-0.9.3 from onsip.org with the intention of using LCR. Although, the LCR module was complied, I do not see the tables lcr, gw in the DB? How do I create them? Thanks __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] ser.pif file not found
by Colin Jordan 29 Jul '05

29 Jul '05

I cannot control SER via serctl stop/start due to the fact that when it runs via this command it is not finding a ser.pid. I have searched for ser.pid on my system and it is not there (was just making sure that serctl was not looking in the wrong spot for the PID file). Any suggestions?? 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

AW: [Serusers] dbtext: table not loaded. example desired
by Müller Thorsten 29 Jul '05

29 Jul '05

Hi, Thanks for your fast response! I read this readme file, but I'm stucked with it. Which file or files do I need for dbtext authentication? Is there a special name necessary? When I'm right I give the name in the ser.cfg with the following line if (!www_authorize("domain.com", "subscriber")) { In my case "subscriber". And the directory where to find the file with the following line modparam("auth_db", "db_url","dbtext:///var/dbtext/ser") in my case /var/dbtext/ser How can I use not plain password but encrypted ones and finally are three "x" enough for ha1 and ha1b or do I need to hash some values and write them down here? Why does ser give me the reply "table not loaded", when these entries are correct meaning written after the README. How can I fill the "subscriber" file with users and passwords? Texteditor I think, because serctl is only for mysql. Thanks Thorsten Mueller mt> -----Ursprüngliche Nachricht----- mt> Von: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] mt> Gesendet: Freitag, 29. Juli 2005 10:55 mt> An: Müller Thorsten mt> Cc: serusers(a)lists.iptel.org mt> Betreff: Re: [Serusers] dbtext: table not loaded. example desired mt> mt> mt> Hello, mt> mt> On 07/29/05 11:16, Müller Thorsten wrote: mt> mt> > Hi all, mt> > mt> > I try to set up ser with dbtext as authentication basis. mt> I run in some mt> > problems because the database is not found: mt> > DBT:db_query: table not loaded. mt> > mt> > After some googling a found that Cesc mt> [cesc.santa(a)gmail.com] had also mt> > done some work with dbtext and mailed that to "serusers" mt> or "serdev" mt> > mailinglists mt> > mt> > In a mail from May and June 05 he told that he has mt> written a "serctl" mt> > for dbtext and a script to produce the necessary files. mt> > mt> > Has somebody got these things and can mail me these mt> files! I could not mt> > found them in the mailing history. mt> > mt> > Has somebody also dealed with dbtext and can give me some example mt> > files for ser.cfg and dbtext files for mt> "authentication/subscriber" and mt> > "location". mt> > mt> rhe readme file of dbtext module has some examples of table mt> structures mt> as well as a simple config file. mt> mt> http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/mod mt> ules/dbtext/README?rev=HEAD&content-type=text/vnd.viewcvs-markup mt> mt> Daniel mt> mt> > Thanks in advance mt> > mt> > Thorsten Mueller mt> > mt> > ser 0.8.14 cross-compiled for arm mt> > My ser.cfg: mt> > #################################################### mt> > ... mt> > # Added by mt for authentication with dbtext mt> > loadmodule "/usr/local/lib/ser/modules/auth.so" mt> > loadmodule "/usr/local/lib/ser/modules/auth_db.so" mt> > loadmodule "/usr/local/lib/ser/modules/dbtext.so" mt> > mt> > modparam("auth_db", "db_url","dbtext:///var/dbtext/ser") mt> > mt> > # -- auth params -- mt> > modparam("auth_db", "calculate_ha1", 1) mt> > modparam("auth_db", "password_column", "password") mt> > modparam("auth_db", "user_column", "username") mt> > modparam("auth_db", "domain_column", "domain") mt> > ... mt> > mt> > if (uri==myself) { mt> > mt> > if (method=="REGISTER") { mt> > mt> > if (!www_authorize("domain.com", "subscriber")) { mt> > www_challenge("domain.com", "0"); mt> > break; mt> > }; mt> > save("location"); mt> > break; mt> > } mt> > ... mt> > ##################################################### mt> > mt> > I have a file in /var/dbtext/ser which is called "subscriber": mt> > ##################################################### mt> > username(str) password(str) ha1(str) domain(str) ha1b(str) mt> > suser:supasswd:xxx:domain.com:xxx mt> > ##################################################### mt> > mt> >----------------------------------------------------------- mt> ------------- mt> > mt> >_______________________________________________ mt> >Serusers mailing list mt> >serusers(a)lists.iptel.org mt> >http://lists.iptel.org/mailman/listinfo/serusers mt> > mt> > mt>

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RE: [Serusers] exec_msg problem
by Sam Lee 29 Jul '05

29 Jul '05

Tried. It didn't work. :< Still getting the same error. The result still came out, its just that it is followed by the error message. 101 INVITE : 3902 <sip:3902@x.x.x.x>;tag=5086b56866af3edo0 to <sip:967258002@x.x.x.x> Jul 29 16:26:16 abc /usr/local/sbin/ser[10577]: ERROR: exec_msg: cmd echo "$SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_ TO" >> /var/log/syslog failed. exit_status=-1, errno=32: Broken pipe -----Original Message----- From: Xavier TRENTIN [mailto:xtrentin@softiphone.net] Sent: Friday, July 29, 2005 4:11 PM To: Sam Lee Subject: Re: [Serusers] exec_msg problem Try to quote the expression you want to echo like this: exec_msg('echo "$SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO">> /var/log/syslog'); Xavier. Sam Lee a écrit: >*********************** >Pas de virus detecté dans l'attachement no filename >Pas de virus detecté dans l'attachement no filename >Pas de virus detecté dans l'attachement no filename > >Votre Mail a été vérifié par InterScan. >***********-*********** > > > > > ------------------------------------------------------------------------ > > Sujet: > [Serusers] exec_msg problem > Expéditeur: > "Sam Lee" <sam.lee(a)super.net.sg> > Date: > Fri, 29 Jul 2005 11:35:53 +0800 > Destinataire: > <serusers(a)lists.iptel.org> > > > Hi, > > Have anyone seen this kind of error message before : > > ERROR: exec_msg: cmd echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO >> > /var/log/syslog failed. exit_status=-1, errno=32: Broken pipe > > This command is actually in my ser.cfg where i capture this few > environmental variables and push them into my /var/log/syslog. It > effectively went in, but i still see this error message. > Here is the original code in ser.cfg : > exec_msg("echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO >> > /var/log/syslog"); > > Please help! > > Regards, > Sam > >------------------------------------------------------------------------ > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > >

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[Serusers] dbtext: table not loaded. example desired
by Müller Thorsten 29 Jul '05

29 Jul '05

Hi all, I try to set up ser with dbtext as authentication basis. I run in some problems because the database is not found: DBT:db_query: table not loaded. After some googling a found that Cesc [cesc.santa(a)gmail.com] had also done some work with dbtext and mailed that to "serusers" or "serdev" mailinglists In a mail from May and June 05 he told that he has written a "serctl" for dbtext and a script to produce the necessary files. Has somebody got these things and can mail me these files! I could not found them in the mailing history. Has somebody also dealed with dbtext and can give me some example files for ser.cfg and dbtext files for "authentication/subscriber" and "location". Thanks in advance Thorsten Mueller ser 0.8.14 cross-compiled for arm My ser.cfg: #################################################### ... # Added by mt for authentication with dbtext loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/dbtext.so" modparam("auth_db", "db_url","dbtext:///var/dbtext/ser") # -- auth params -- modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") ... if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("domain.com", "subscriber")) { www_challenge("domain.com", "0"); break; }; save("location"); break; } ... ##################################################### I have a file in /var/dbtext/ser which is called "subscriber": ##################################################### username(str) password(str) ha1(str) domain(str) ha1b(str) suser:supasswd:xxx:domain.com:xxx #####################################################

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RE: [Serusers] exec_msg problem
by Sam Lee 29 Jul '05

29 Jul '05

Solved IT! Stupid error. Old : exec_msg("echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO>> /var/log/syslog"); New : exec_msg("(echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO)>> /var/log/syslog); That solves it! It seems like the exec_msg is not happy about the syntax that I gave it. :> Thank for all the help! Regards, Sam -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Sam Lee Sent: Friday, July 29, 2005 4:30 PM To: Xavier TRENTIN; serusers(a)lists.iptel.org Subject: RE: [Serusers] exec_msg problem Tried. It didn't work. :< Still getting the same error. The result still came out, its just that it is followed by the error message. 101 INVITE : 3902 <sip:3902@x.x.x.x>;tag=5086b56866af3edo0 to <sip:967258002@x.x.x.x> Jul 29 16:26:16 abc /usr/local/sbin/ser[10577]: ERROR: exec_msg: cmd echo "$SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_ TO" >> /var/log/syslog failed. exit_status=-1, errno=32: Broken pipe -----Original Message----- From: Xavier TRENTIN [mailto:xtrentin@softiphone.net] Sent: Friday, July 29, 2005 4:11 PM To: Sam Lee Subject: Re: [Serusers] exec_msg problem Try to quote the expression you want to echo like this: exec_msg('echo "$SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO">> /var/log/syslog'); Xavier. Sam Lee a écrit: >*********************** >Pas de virus detecté dans l'attachement no filename Pas de virus >detecté dans l'attachement no filename Pas de virus detecté dans >l'attachement no filename > >Votre Mail a été vérifié par InterScan. >***********-*********** > > > > > ---------------------------------------------------------------------- > -- > > Sujet: > [Serusers] exec_msg problem > Expéditeur: > "Sam Lee" <sam.lee(a)super.net.sg> > Date: > Fri, 29 Jul 2005 11:35:53 +0800 > Destinataire: > <serusers(a)lists.iptel.org> > > > Hi, > > Have anyone seen this kind of error message before : > > ERROR: exec_msg: cmd echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO >> > /var/log/syslog failed. exit_status=-1, errno=32: Broken pipe > > This command is actually in my ser.cfg where i capture this few > environmental variables and push them into my /var/log/syslog. It > effectively went in, but i still see this error message. > Here is the original code in ser.cfg : > exec_msg("echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO >> > /var/log/syslog"); > > Please help! > > Regards, > Sam > >----------------------------------------------------------------------- >- > >_______________________________________________ >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] BYE
by Sebastian Kühner 29 Jul '05

29 Jul '05

Hi! I'm connecting to a Cisco PSTN Gateway for international calls. I'm using a Port Restricted Cone NAT. I can make calls and make a hangup with MY phone, but if the callee at the other side of the gateway hangs up, I do not get the BYE-Message. That could be the problem of the Restricted Cone NAT... or someone has another idea? Thanks! Sebastian

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[Serusers] exec_msg problem
by Sam Lee 29 Jul '05

29 Jul '05

Hi, Have anyone seen this kind of error message before : ERROR: exec_msg: cmd echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO >> /var/log/syslog failed. exit_status=-1, errno=32: Broken pipe This command is actually in my ser.cfg where i capture this few environmental variables and push them into my /var/log/syslog. It effectively went in, but i still see this error message. Here is the original code in ser.cfg : exec_msg("echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO >> /var/log/syslog"); Please help! Regards, Sam

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RE: [Serusers] Problem : Can SER process the reINVITE messages pr operly?
by Ricardo Martinez 29 Jul '05

29 Jul '05

Thanks again Alberto. Are you using the ser.cfg from onsip.org? I'm using it, and i can't make it work with reinvite to a nat'd endpoints. Can you give me some hints here?. For example what version of mediaproxy are you using?. If you're using the onsip.org configuration, Did you modified this configuration to make it work for reINVITE's? Hope you can help me, Thanks again Regards, Ricardo Martinez.- -----Mensaje original----- De: Alberto Cruz [mailto:acruz@tekbrain.com] Enviado el: Miércoles, 27 de Julio de 2005 16:50 Para: Ricardo Martinez CC: 'serusers(a)lists.iptel.org' Asunto: Re: [Serusers] Problem : Can SER process the reINVITE messages pr operly? Importancia: Alta Yes it is working fine. Regards Ricardo Martinez wrote: Alberto. Thanks for your answer. You mean that your platform works well with the reINVITE's to a NAT'd endpoint?. Regards, Ricardo.- -----Mensaje original----- De: Alberto Cruz [ mailto:acruz@tekbrain.com <mailto:acruz@tekbrain.com> ] Enviado el: Miércoles, 27 de Julio de 2005 16:12 Para: Ricardo Martinez CC: 'Greger V. Teigre'; ' serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> ' Asunto: Re: [Serusers] Problem : Can SER process the reINVITE messages pr operly? Importancia: Alta Ricardo this block is working fine to me: #------------------------------------------------- # subsequent messages withing a dialog should take the # path determined by record-routing #------------------------------------------------- if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); # The following lines are added due media proxy if (has_totag()&&(method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { # Mark as NAT'ed setflag(6); use_media_proxy(); }; }; if ((method=="BYE") && ((client_nat_test("3") || search("^Route:.*;nat=yes")))) { fix_nated_contact(); force_rport(); }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here route(1); break; }; Maybe you have something wrong at your mediaproxy.ini file. Regards Alberto Cruz

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RE: [Serusers] i can't start ser could somebody help~~
by Sam Lee 29 Jul '05

29 Jul '05

It would help if you check either /var/log/messages or /var/log/syslog for error message pertaining to SER. Regards, Sam ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of ? ? Sent: Friday, July 29, 2005 2:25 PM To: serusers(a)lists.iptel.org Subject: [Serusers] i can't start ser could somebody help~~ hi the list! i have installed ser 0.8.12 and mysql,mysql is fine,but when restart ser then it appears 4 errors without saying why.i just did only a few modifies to ser.cfg ,this is the processing statements.could somebody tell me there the problem is.thanks very much and forgive my terrible english:) ot@localhost root]# ps -A PID TTY TIME CMD 1 ? 00:00:04 init 2 ? 00:00:00 ksoftirqd/0 3 ? 00:00:00 events/0 4 ? 00:00:00 kblockd/0 6 ? 00:00:00 khelper 5 ? 00:00:00 khubd 7 ? 00:00:00 pdflush 8 ? 00:00:00 pdflush 10 ? 00:00:00 aio/0 9 ? 00:00:00 kswapd0 117 ? 00:00:00 kseriod 156 ? 00:00:00 kjournald 1205 ? 00:00:00 kjournald 1206 ? 00:00:00 kjournald 1578 ? 00:00:00 syslogd 1582 ? 00:00:00 klogd 1603 ? 00:00:00 portmap 1623 ? 00:00:00 rpc.statd 1650 ? 00:00:00 rpc.idmapd 1744 ? 00:00:00 smartd 1754 ? 00:00:00 acpid 1766 ? 00:00:00 cupsd 1954 ? 00:00:00 sshd &n bsp;1969 ? 00:00:00 xinetd 1988 ? 00:00:00 sendmail 1997 ? 00:00:00 sendmail 2008 ? 00:00:00 gpm 2030 ? 00:00:00 crond 2079 ? 00:00:00 xfs 2098 ? 00:00:00 atd 2117 ? 00:00:00 dbus-daemon-1 2133 ? 00:00:00 mdadm 2159 tty1 00:00:00 mingetty 2165 tty2 00:00:00 mingetty 2171 tty3 00:00:00 mingetty 2177 tty4 00:00:00 mingetty 2183 tty5 00:00:00 mingetty 2189 tty6 00:00:00 mingetty 2195 ? 00:00:00 gdm-binary 2382 ? 00:00:00 gdm-binary 2393 ? 00:03:22 X 2423 ? 00:00:00 gnome-session 2485 ? 00:00:00 ssh-agent 2495 ? 00:00:01 gconfd-2 2498 ? 00:00:00 gnome-keyring-d 2500 ? 00:00:00 bonobo-activati 2502 ? 00:00:00 gnome-settings- 2507 ? 00:00:00 fam 2599 ? 00:00:01 metacity 2603 ? 00:00:01 gnome-panel ;2605 ? 00:00:02 nautilus 2607 ? 00:00:01 magicdev 2609 ? 00:00:00 eggcups 2611 ? 00:00:00 pam-panel-icon 2613 ? 00:00:11 rhn-applet-gui 2614 ? 00:00:00 pam_timestamp_c 2617 ? 00:00:00 gnome-vfs-daemo 2620 ? 00:00:00 clock-applet 2648 ? 00:00:00 mapping-daemon 2650 ? 00:00:00 notification-ar 2652 ? 00:00:00 mixer_applet2 2659 ? 00:00:00 wnck-applet 2661 ? 00:00:59 gnome-terminal 2662 ? 00:00:00 gnome-pty-helpe 2663 pts/1 00:00:01 bash 4506 pts/1 00:00:00 vsftpd 8355 pts/1 00:00:01 find 8871 pts/1 00:00:00 safe_mysqld 8895 pts/1 00:00:00 mysqld 9017 pts/2 00:00:00 bash 12474 pts/1 00:00:00 ps __________________________________________________ 赶快注册雅虎超大容量免费邮箱? http://cn.mail.yahoo.com

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[Serusers] i can't start ser could somebody help~~
by 皮面 29 Jul '05

29 Jul '05

hi the list! i have installed ser 0.8.12 and mysql,mysql is fine,but when restart ser then it appears 4 errors without saying why.i just did only a few modifies to ser.cfg ,this is the processing statements.could somebody tell me there the problem is.thanks very much and forgive my terrible english:) ot@localhost root]# ps -A PID TTY TIME CMD 1 ? 00:00:04 init 2 ? 00:00:00 ksoftirqd/0 3 ? 00:00:00 events/0 4 ? 00:00:00 kblockd/0 6 ? 00:00:00 khelper 5 ? 00:00:00 khubd 7 ? 00:00:00 pdflush 8 ? 00:00:00 pdflush 10 ? 00:00:00 aio/0 9 ? 00:00:00 kswapd0 117 ? 00:00:00 kseriod 156 ? 00:00:00 kjournald 1205 ? 00:00:00 kjournald 1206 ? 00:00:00 kjournald 1578 ? 00:00:00 syslogd 1582 ? 00:00:00 klogd 1603 ? 00:00:00 portmap 1623 ? 00:00:00 rpc.statd 1650 ? 00:00:00 rpc.idmapd 1744 ? 00:00:00 smartd 1754 ? 00:00:00 acpid 1766 ? 00:00:00 cupsd 1954 ? 00:00:00 sshd 1969 ? 00:00:00 xinetd 1988 ? 00:00:00 sendmail 1997 ? 00:00:00 sendmail 2008 ? 00:00:00 gpm 2030 ? 00:00:00 crond 2079 ? 00:00:00 xfs 2098 ? 00:00:00 atd 2117 ? 00:00:00 dbus-daemon-1 2133 ? 00:00:00 mdadm 2159 tty1 00:00:00 mingetty 2165 tty2 00:00:00 mingetty 2171 tty3 00:00:00 mingetty 2177 tty4 00:00:00 mingetty 2183 tty5 00:00:00 mingetty 2189 tty6 00:00:00 mingetty 2195 ? 00:00:00 gdm-binary 2382 ? 00:00:00 gdm-binary 2393 ? 00:03:22 X 2423 ? 00:00:00 gnome-session 2485 ? 00:00:00 ssh-agent 2495 ? 00:00:01 gconfd-2 2498 ? 00:00:00 gnome-keyring-d 2500 ? 00:00:00 bonobo-activati 2502 ? 00:00:00 gnome-settings- 2507 ? 00:00:00 fam 2599 ? 00:00:01 metacity 2603 ? 00:00:01 gnome-panel 2605 ? 00:00:02 nautilus 2607 ? 00:00:01 magicdev 2609 ? 00:00:00 eggcups 2611 ? 00:00:00 pam-panel-icon 2613 ? 00:00:11 rhn-applet-gui 2614 ? 00:00:00 pam_timestamp_c 2617 ? 00:00:00 gnome-vfs-daemo 2620 ? 00:00:00 clock-applet 2648 ? 00:00:00 mapping-daemon 2650 ? 00:00:00 notification-ar 2652 ? 00:00:00 mixer_applet2 2659 ? 00:00:00 wnck-applet 2661 ? 00:00:59 gnome-terminal 2662 ? 00:00:00 gnome-pty-helpe 2663 pts/1 00:00:01 bash 4506 pts/1 00:00:00 vsftpd 8355 pts/1 00:00:01 find 8871 pts/1 00:00:00 safe_mysqld 8895 pts/1 00:00:00 mysqld 9017 pts/2 00:00:00 bash 12474 pts/1 00:00:00 ps __________________________________________________ 赶快注册雅虎超大容量免费邮箱? http://cn.mail.yahoo.com

1 0

[Serusers] SER + SERNAT + MySQL
by Truong Chi Thanh 29 Jul '05

29 Jul '05

Hi all, My name is chithanh. I'm working at Vietnam. Now I'm using with SER. I have a case with SER. | ser database | | | ============================ || || | SER (5060) | | SERNAT + nathelper + rtpproxy (5082) | || || | user1 | | user2 | Explain : user1 registered SER ===> SER save("location") for user1 into table "location" of "ser" database. user2 registered SERNAT ===> SER save("location") for user2 into table "location" of "ser" database. My purpose : user1 can call for user2. user can call for user1. Real when I run with this case. User1 can't call for User2. (although i had setted db_mode=1 for both) because : + User1 INVITE User2 ==> SER lookup("location") into SER's memory. Not lookup in ser database. + Use2 INVITE User1 ==> SERNAT lookup("location") into SERNAT's memory. Not lookup in ser database. I want SER and SENAT when lookup("location"), they must lookup in table "location" of ser database. How do u to solve for this problem ? Please help me. =================================== My config of SERNAT : # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 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 debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5082 listen=abc.com alias=abc.com children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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/mysql.so" loadmodule "/usr/local//lib/ser/modules/nathelper.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" #loadmodule "/usr/local//lib/ser/modules/auth_radius.so" loadmodule "/usr/local//lib/ser/modules/acc.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) modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser") # -- 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) # Accounting modparam("acc", "log_level", 2) #modparam("acc", "log_flag", 3) #modparam("acc", "log_missed_flag", 3) modparam("acc", "db_flag", 3) modparam("acc", "db_missed_flag", 3) # ------------------------- 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 > 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("chinhnam.com","subscriber")) { www_challenge("chinhnam.com", "0"); break; }; fix_nated_contact(); save("location"); break; }; #End if(method=="REGISTER") if (method=="INVITE") { if (proxy_authorize("chinhnam.com","subscriber")) { if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }else { proxy_challenge("chinhnam.com", "0"); break; }; #End if (proxy_authorize("chinhnam.com", "subcriber")) }; #End if(method=="INVITE") fix_nated_contact(); force_rtp_proxy(); t_on_reply("1"); #setflag(3); }; #End if(uri=="myself") # 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(); }; } #End main routing logic onreply_route[1] { fix_nated_contact(); if (status=~"[12][0-9][0-9]") { force_rtp_proxy(); } } ============================================================================ My config of SER : # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 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 debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 listen=chinhnam.com alias=chinhnam.com children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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/mysql.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" #loadmodule "/usr/local//lib/ser/modules/auth_radius.so" loadmodule "/usr/local//lib/ser/modules/acc.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) modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser") # -- 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) # Accounting modparam("acc", "log_level", 2) #modparam("acc", "log_flag", 3) #modparam("acc", "log_missed_flag", 3) modparam("acc", "db_flag", 3) modparam("acc", "db_missed_flag", 3) # ------------------------- 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 > 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("chinhnam.com","subscriber")) { www_challenge("chinhnam.com", "0"); break; }; save("location"); break; }; #End if(method=="REGISTER") if (method=="INVITE") { if (proxy_authorize("chinhnam.com","subscriber")) { if(uri=~"sip:63[0-9]*@") { rewritehostport("192.168.200.248:5060"); }else if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }else { proxy_challenge("chinhnam.com", "0"); break; }; #End if (proxy_authorize("chinhnam.com", "subcriber")) }; #End if(method=="INVITE") #setflag(3); }; #End if(uri=="myself") # 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(); }; } #End main routing logic ============================================================================= Note: SER (5060) only signaling SIP for my user. Not use media proxy. SER (5082) : signaling SIP + RTP path for my user. Use nathelper + rtpproxy Thanks very much everything you had done for me. Truong Chi Thanh --------------------------------- Start your day with Yahoo! - make it your home page

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[Serusers] SIP Service Providers supporting Windows Messenger 4.6
by Michael Kendall 28 Jul '05

28 Jul '05

Hi, Reading the SERusers digest for a few months has allowed me to read about a lot of very complex and specialized problems. It's a bit intimidating. Although I might some day want to learn to administer a Linux system and run my own SER server, at this point I would rather just become a paying customer on someone else's SER server. Many of the list participants are clearly professionals operating large SIP service providers. But none of these professionals have identified their organizations clearly enough to let me become their customer. Perhaps because the traditions of the list prohibit users from advertising their commercial services. Can someone please give me a recommendation of a SIP service provider which has good service and reasonable pricing. I need a service which will allow me to call between two computers, each running Windows Messenger 4.6/4.7 . I am aware that the latest version, Windows Messenger 5.1 does not allow connecting to third party service providers, and that is the reason that I am using the older versions. The Microsoft site does not tell me how to configure version 5.1 for STUN or for symmetric NAT. Every link that I have found for downloading version 4.6 points me back to the download page at microsoft.com, and microsoft has actually removed the file so all I get from them is a polite apology. Fair enough. So here is an interesting legal question that I suspect a lot of you SER wizards will know the answer to: Is it illegal for third parties sites to publicly offer the download of version 4.6 after microsoft has chosen to stop offering it? It was after all being given away free by Microsoft a short time ago. I am aware that freeworlddialup.com (FWD) is a good no-cost SIP service provider. Does FWD use SER as their server software? The latest version of FWD offers several options for User Agent software, but none of these UAs include a whiteboard feature, which I need for my tutoring service. Therefore, I need a SIP Service Provider which supports Windows Messenger 4.6/4.7 and has some configuration instructions for this particular UA. Hopefully the config instructions will tell me how to set Messenger up for STUN when I have a standard NAT, and for mediaproxy when I encounter a symmetric NAT. Googling for "SIP Service Provider" and "Windows Messenger 4.6" has led me to a lot of dead ends and out-of-date pages, such as the "old FWD site" at pulver.com which has detailed instructions for configuring Messenger 4.6, but the instructions no longer work. "Service provider temporarily unavailable". (permanently) Can anybody offer a Service Provider recommendation or volunteer to take me on as a customer? thanks Michael Kendall

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[Serusers] Dest IP
by Sebastian Kühner 28 Jul '05

28 Jul '05

Hello, I need the destination IP of the user of the first INVITE message (user -> ser)... but dst_ip == xxx.xxx.xxx.xxx gives me the IP of ser. The problem is that I only want to force the rtp_proxy if the users are behind a Port restricted cone NAT (the scr-ips of them are known) Here is my code: # Force rtpproxy for users that are behind a Port Restricted Cone NAT if ((src_ip == xxx.xxx.xxx.xxx | dst_ip == xxx.xxx.xxx.xxx) && !(uri=~"sip:[0-9]{9,30}@.*")) { fix_nated_contact(); # Rewrite contact with source IP of signalling force_rport(); # Add rport parameter to topmost Via setflag(6); } Has anyone an idea how I can get the IP of the user? For example in the location database? Thanks! Sebastian

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[Users] new features in development version
by Daniel-Constantin Mierla 28 Jul '05

28 Jul '05

Hello, the new features of OpenSER development version are detailed a bit in: http://openser.org/features-0.10.x.php If you find something missing or you want to add more text, please send a mail. Hope it will convince some new users to start testing it. Daniel

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[Users] typos in documentation and use_domain question
by Klaus Darilion 28 Jul '05

28 Jul '05

Hi! In the uri_db documentation (all parameters): modparam("db_url", ".... ^^^^^^ shouldn't this be uri_db? In the group_radius documentation: modparam("group", ... ^^^^^^ shouldn't this be group_radius? I also wonder about the use_domain paramter in the radius modules. group_radius has this parameter, whereas all other radius modules (uri_radius, avp_radius, auth_radius) do not have this parameter? Is there a reason for this? regards, klaus

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[Serusers] Over-long fields can crash SER with PostgreSQL
by Karl A. Krueger 28 Jul '05

28 Jul '05

We are using SER with the PostgreSQL module, so that SER and our Asterisk PSTN gateway and voice mail system can use the same back-end database for user records. (Anyone who's interested in how we did this, let me know -- it's still a little shaky, but it pretty much works.) Recently we have encountered a problem: it seems that if any database error occurs, SER hangs or crashes. We have so far been able to mostly work around these conditions by being Really Careful with the database, but it is a little unsettling. The database schema we're working from is practically identical to the MySQL schema that ships with SER. Notably for this problem the lengths of all CHARACTER VARYING fields are the same. The User-Agent field is CHARACTER VARYING(50), for instance. So ... today boss got a copy of Xten's "eyeBeam" software, set it up to register to our SER ... and SER promptly locked up. I looked in the PostgreSQL database log and found this: Feb 10 13:39:32 mabell postgres[13058]: [267] ERROR: value too long for type character varying(50) Whoops. SER tried to insert eyeBeam's User-Agent data, PostgreSQL returned a data integrity error, and SER went bye-bye. So I did a little ALTER TABLE and adjusted the field to length 128 ... then put this in ser.cfg: # Disallow long user-agent if (search("^User-Agent: .{100,}") ) { log(1, "LOG: User agent too long\n"); sl_send_reply("479", "User agent too long!"); break; }; But this is obviously a kludge ... and doesn't fix the problem of one of the *other* fields being over-long. Anyone have any thoughts on this? Robustness patches for the PostgreSQL module or usrloc? Better ser.cfg workarounds to protect the world from malformed or over-long SIP header fields? Should I just change all the database fields to TEXT instead of CHARACTER VARYING(whatever) ? -- Karl A. Krueger <kkrueger(a)whoi.edu> Network Security -- Linux/Unix Systems Support -- Etc. Woods Hole Oceanographic Institution

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Re: [Serusers] asterisk and ser combine, 407 proxy authentucation required !
by Iqbal 28 Jul '05

28 Jul '05

tell ser that all packets from asterisk IP dont need to be authenticated, use trusted_ip setup for it, that should do the trick Iqbal On 7/28/2005, "zhaomin" <zhaomin(a)kingdream.com> wrote: >hello! > I want to realized asterisk+ser combine working,ser is working fine and asterisk working well I think,I configure sip.conf and extensions.conf voicemail.conf as the asterisk handbook ,but when I registered 2000 to dial 2001 but I get error :407 proxy authentucation required, > where is problem in asterisk or ser ? >blow is the log of x-lite diagnostic,Very appreciation anybody help! >SEND TIME: 201758109 >SEND >> 221.202.129.167:5060 >INVITE sip:2001@netvoip.cn SIP/2.0 >Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK1DD16D11800A4C6686A65DD7C5662FFF >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn> >Contact: <sip:2000@10.1.16.34:5060> >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53903 INVITE >Max-Forwards: 70 >Content-Type: application/sdp >User-Agent: X-Lite release 1103m >Content-Length: 244 > >v=0 >o=2000 201757640 201758109 IN IP4 10.1.16.34 >s=X-Lite >c=IN IP4 10.1.16.34 >t=0 0 >m=audio 8000 RTP/AVP 0 98 97 101 >a=rtpmap:0 pcmu/8000 >a=rtpmap:98 iLBC/8000 >a=rtpmap:97 speex/8000 >a=rtpmap:101 telephone-event/8000 >a=fmtp:101 0-15 > >RECEIVE TIME: 201758859 >RECEIVE << 221.202.129.167:5060 >SIP/2.0 407 Proxy Authentication Required >Via: SIP/2.0/UDP 10.1.16.34:5060;rport=5339;branch=z9hG4bK1DD16D11800A4C6686A65DD7C5662FFF >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn>;tag=b83deba332858fe2da009af570355a70.3a3b >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53903 INVITE >Proxy-Authenticate: Digest realm="netvoip.cn", nonce="42e896c750edabc9057e78d345379a4991b02378" >Server: Sip EXpress router (0.10.99-dev7 (i386/linux)) >Content-Length: 0 >Warning: 392 221.202.129.167:5060 "Noisy feedback tells: pid=28404 req_src_ip=61.136.210.7 req_src_port=5339 in_uri=sip:2001@netvoip.cn out_uri=sip:2001@netvoip.cn via_cnt==1" > > >SEND TIME: 201758859 >SEND >> 221.202.129.167:5060 >ACK sip:2001@netvoip.cn SIP/2.0 >Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK1DD16D11800A4C6686A65DD7C5662FFF >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn>;tag=b83deba332858fe2da009af570355a70.3a3b >Contact: <sip:2000@10.1.16.34:5060> >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53903 ACK >Max-Forwards: 70 >Content-Length: 0 > > >SEND TIME: 201758906 >SEND >> 221.202.129.167:5060 >INVITE sip:2001@netvoip.cn SIP/2.0 >Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn> >Contact: <sip:2000@10.1.16.34:5060> >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53904 INVITE >Proxy-Authorization: Digest username="2000",realm="netvoip.cn",nonce="42e896c750edabc9057e78d345379a4991b02378",response="337cb9deeceb1bf01e9768b6a5452568",uri="sip:2001@netvoip.cn" >Max-Forwards: 70 >Content-Type: application/sdp >User-Agent: X-Lite release 1103m >Content-Length: 244 > >v=0 >o=2000 201757640 201758109 IN IP4 10.1.16.34 >s=X-Lite >c=IN IP4 10.1.16.34 >t=0 0 >m=audio 8000 RTP/AVP 0 98 97 101 >a=rtpmap:0 pcmu/8000 >a=rtpmap:98 iLBC/8000 >a=rtpmap:97 speex/8000 >a=rtpmap:101 telephone-event/8000 >a=fmtp:101 0-15 > >RECEIVE TIME: 201759640 >RECEIVE << 221.202.129.167:5060 >SIP/2.0 100 trying -- your call is important to us >Via: SIP/2.0/UDP 10.1.16.34:5060;rport=5339;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn> >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53904 INVITE >Server: Sip EXpress router (0.10.99-dev7 (i386/linux)) >Content-Length: 0 >Warning: 392 221.202.129.167:5060 "Noisy feedback tells: pid=28398 req_src_ip=61.136.210.7 req_src_port=5339 in_uri=sip:2001@netvoip.cn out_uri=sip:2001@221.202.129.167:5061 via_cnt==1" > > >RECEIVE TIME: 201759671 >RECEIVE << 221.202.129.167:5060 >SIP/2.0 407 Proxy Authentication Required >Via: SIP/2.0/UDP 10.1.16.34:5060;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn>;tag=as6e0c8cea >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53904 INVITE >User-Agent: Asterisk PBX >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER >Contact: <sip:2001@221.202.129.167:5061> >Proxy-Authenticate: Digest realm="asterisk", nonce="43fb4632" >Content-Length: 0 > > >SEND TIME: 201759671 >SEND >> 221.202.129.167:5060 >ACK sip:2001@netvoip.cn SIP/2.0 >Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn>;tag=as6e0c8cea >Contact: <sip:2000@10.1.16.34:5060> >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53904 ACK >Max-Forwards: 70 >Content-Length: 0 > >

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[Serusers] SER Asterisk Voicemail - Message Playback
by Aisling 28 Jul '05

28 Jul '05

Hi Everyone, Unfortunately when I was upgrading my version of SER to 0.9.0 I deleted my config file which contained code for forwarding to asterisk voicemail.Therefore I am attempting to set up all over again (Doh! I know!) :-( Anyhow SER seems to be correctly forwarding to Asterisk, the user can leave a message if the phone isn't picked up after 3 rings and then the message is sent to the users email account. I want the user to also be able to listen to their voicemails from the end user device. It is my understanding from what I have coded, that user should be able to dial 9999 and when SER receives this it sends it to Asterisk' VoiceMailMain where the user will be asked for an account name and password. However there are problems when the user dials 9999. The user can hear nothing on the phone. Errors appear on both the SER screen and Asterisk screen so I am not sure what the problem is: On SER it says: Warning: sl_send_reply: I won't send a reply for ACK!! Forwarding to asterisk14(27355) contact_parser(): Empty body Parse_contact(): Error while parsing Get_contact_uri: Error while parsing Contact body On Asterisk it says: -- Executing VoiceMailMain("SIP/2092-dd89", "2092") in new stack -- Playing 'vm-login' (language 'en') WARNING: app_voicemail.c:3333 vm_execmain: Couldn't read username Spawn extension (test, 9999, 1) exited non-zero on 'SIP/2092-c660' The relevant parts of my ser.cfg and asterisk config files are shown below. Many thanks, Aisling. ------------------------------------------------------------------------ ----------------------------------- ser.cfg ------------------------------------------------------------------------ ---------------------------------- #Call Type Processing Section if(uri==myself){ if(uri=~"^sip:9[0-9]*@serveraddress"){ log(1, "forwarding to asterisk"); rewritehostport("serveraddress:port"); append_branch(); t_relay_to_udp("serveraddress", "port"); break; } if (method == "INVITE"){ t_on_failure("1"); route(3); break; } failure_route[1]{ revert_uri(); rewritehostport("serveraddress:port"); append_branch(); t_relay_to_udp("serveraddress", "port"); break(); } ------------------------------------------------------------------------ -------------------------------------- sip.conf ------------------------------------------------------------------------ ------------------------------------- [general] context=test port=5062 bindaddr=0.0.0.0 srvlookup=yes [2092] type=friend username=2092 canreinvite=no context=test mailbox=2092 host=dynamic nat=no dtmfmode=info disallow=all allow=alaw allow=ulaw [314] type=friend username=314 canreinvite=no context=test mailbox=314 host=dynamic nat=no dtmfmode=info disallow=all allow=alaw allow=ulaw ------------------------------------------------------------------------ ----------------------------------- extensions.conf ------------------------------------------------------------------------ ---------------------------------- [test] ;leave voice messages exten => 2092, 1, Voicemail(u2092) exten => 2092, 2, Hangup exten => 314, 1, Voicemail(u314) exten => 314, 2, Hangup ;play voice messages exten => 9999, 1, VoiceMailMain, 2092 exten => 9999, 2, Hangup exten => 9999, 1, VoiceMailMain, 314 exten => 9999, 2, Hangup ------------------------------------------------------------------------ ------------------------------------- voicemail.conf ------------------------------------------------------------------------ -------------------------------------- [default] 2092 => 1234, 2092, emailaddress 314 => 1234, 314, emailaddress -------------------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 : RE : [Serusers] forwarding and accounting
by Olivier Taylor 28 Jul '05

28 Jul '05

Sorry guy, u think I don't contribute, that's ok, but I try helping as much as I can. Respect for the job done by the Ser team, but sorry, I and many of us don't understand why we have now two Ser...(an open one and an open one...) If u really want to contribute, don't fork, just contribute, otherwise it will be another software. If you don't want people use your work, go and join the Microsoft team and forget free software. That's only the poor opinion of a guy using unfairly free software. Olivier Ps: make ur opinion available for all and put serusers in copy for the next mail. -----Message d'origine----- De : Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Envoyé : jeudi 28 juillet 2005 18:28 À : Olivier Taylor Objet : Re: RE : [Serusers] forwarding and accounting hmmm... I really do find annoying and unfair this kind of judgement from somebody who just enjoyed for free others people work without any contribution.....I will kindly ask you consider this before expressing such opinions. regards, bogdan Olivier Taylor wrote: >I can't accept the fork with Openser, sorry, I will find another >solution. Openser just have to rejoin Ser, that forking is very >annoying and probably unfair. > >Olivier > >-----Message d'origine----- >De : Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Envoyé : jeudi 28 juillet 2005 17:57 >À : Olivier Taylor >Objet : Re: [Serusers] forwarding and accounting > > >Hi Olivier, > >what you are looking for is multi-leg call accounting - SER has no >support for it, but OpenSER has. >see http://openser.org/pipermail/users/2005-June/000140.html > >regards, >bogdan > >Olivier Taylor wrote: > > > >>hi all, >> >>what about the accounting when forwarding a call. >> >>A-->B--ForwardTo-->C >> >>Forwarding is OK, even to Pstn, but... >>B as to pay the forwarding, but i don't see any record on acc neither >>radacct concerning this call for B. >> >>Any Idea? >> >>Olivier >> >>---------------------------------------------------------------------- >>- >>- >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> > > >

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[Serusers] forwarding and accounting
by Olivier Taylor 28 Jul '05

28 Jul '05

hi all, what about the accounting when forwarding a call. A-->B--ForwardTo-->C Forwarding is OK, even to Pstn, but... B as to pay the forwarding, but i don't see any record on acc neither radacct concerning this call for B. Any Idea? Olivier

2 1

[Users] timer question
by Klaus Darilion 28 Jul '05

28 Jul '05

Hi! What should happen if the invite_timer expires and there is no failure route defined? In my case happens nothing, the phone is still ringing. Shouldn't openser (CVS) cancel the call and responds with 408? If I define a failure_route, everything works fine. regards, klaus debuglog: 9(21752) DEBUG: reply relayed. buf=0x81148d0: SIP/2.0 1..., shmem=0x40606568: SIP/2.0 1 9(21752) reply_received: FR_INV_TIMER = 10 9(21752) DEBUG: add_to_tail_of_timer[1]: 0x40604d7c 9(21752) DEBUG:destroy_avp_list: destroying list (nil) 9(21752) receive_msg: cleaning up after 10 seconds: 10(21756) DEBUG: timer routine:1,tl=0x40604d7c next=(nil) 10(21756) DEBUG: final_response_handler: transaction silently dropped (0x40604c50) 10(21756) DEBUG: add_to_tail_of_timer[2]: 0x40604c98 phone is still ringing, some seconds later: 10(21756) DEBUG: timer routine:2,tl=0x40604c98 next=(nil) 10(21756) DEBUG: wait_handler : removing 0x40604c50 from table 10(21756) DEBUG: delete transaction 0x40604c50 10(21756) DEBUG: wait_handler : done phone is still ringing :-(

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[Serusers] Presence and messaging: UA with SER
by Cameron Beattie 28 Jul '05

28 Jul '05

I want to be able to offer the following functionality (in addition to calling) to my users namely: notification when a user is on/offline, directory search, and chat. These simple requirements are met with ease by Skype. And yet there doesn't seem to be any combination of SER and a freely-available Windows user agent that offers this. I have been a little bit out of touch with this recently so things may have moved on since I last checked. The main problem seems to be on the UA side. I've tried (amongst others) X-Ten, SJPhone, PhoneGaim, Firefly, and Windows Messenger - none of them come close to Skype's ease of configuration and use. Am I missing something? Can anyone offer some suggestions about how to offer this functionality? Regards Cameron

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[Serusers] asterisk and ser combine, 407 proxy authentucation required !
by zhaomin 28 Jul '05

28 Jul '05

hello! I want to realized asterisk+ser combine working,ser is working fine and asterisk working well I think,I configure sip.conf and extensions.conf voicemail.conf as the asterisk handbook ,but when I registered 2000 to dial 2001 but I get error :407 proxy authentucation required, where is problem in asterisk or ser ? blow is the log of x-lite diagnostic,Very appreciation anybody help! SEND TIME: 201758109 SEND >> 221.202.129.167:5060 INVITE sip:2001@netvoip.cn SIP/2.0 Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK1DD16D11800A4C6686A65DD7C5662FFF From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 To: <sip:2001@netvoip.cn> Contact: <sip:2000@10.1.16.34:5060> Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 CSeq: 53903 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 244 v=0 o=2000 201757640 201758109 IN IP4 10.1.16.34 s=X-Lite c=IN IP4 10.1.16.34 t=0 0 m=audio 8000 RTP/AVP 0 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 RECEIVE TIME: 201758859 RECEIVE << 221.202.129.167:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.16.34:5060;rport=5339;branch=z9hG4bK1DD16D11800A4C6686A65DD7C5662FFF From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 To: <sip:2001@netvoip.cn>;tag=b83deba332858fe2da009af570355a70.3a3b Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 CSeq: 53903 INVITE Proxy-Authenticate: Digest realm="netvoip.cn", nonce="42e896c750edabc9057e78d345379a4991b02378" Server: Sip EXpress router (0.10.99-dev7 (i386/linux)) Content-Length: 0 Warning: 392 221.202.129.167:5060 "Noisy feedback tells: pid=28404 req_src_ip=61.136.210.7 req_src_port=5339 in_uri=sip:2001@netvoip.cn out_uri=sip:2001@netvoip.cn via_cnt==1" SEND TIME: 201758859 SEND >> 221.202.129.167:5060 ACK sip:2001@netvoip.cn SIP/2.0 Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK1DD16D11800A4C6686A65DD7C5662FFF From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 To: <sip:2001@netvoip.cn>;tag=b83deba332858fe2da009af570355a70.3a3b Contact: <sip:2000@10.1.16.34:5060> Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 CSeq: 53903 ACK Max-Forwards: 70 Content-Length: 0 SEND TIME: 201758906 SEND >> 221.202.129.167:5060 INVITE sip:2001@netvoip.cn SIP/2.0 Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 To: <sip:2001@netvoip.cn> Contact: <sip:2000@10.1.16.34:5060> Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 CSeq: 53904 INVITE Proxy-Authorization: Digest username="2000",realm="netvoip.cn",nonce="42e896c750edabc9057e78d345379a4991b02378",response="337cb9deeceb1bf01e9768b6a5452568",uri="sip:2001@netvoip.cn" Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 244 v=0 o=2000 201757640 201758109 IN IP4 10.1.16.34 s=X-Lite c=IN IP4 10.1.16.34 t=0 0 m=audio 8000 RTP/AVP 0 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 RECEIVE TIME: 201759640 RECEIVE << 221.202.129.167:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.1.16.34:5060;rport=5339;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 To: <sip:2001@netvoip.cn> Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 CSeq: 53904 INVITE Server: Sip EXpress router (0.10.99-dev7 (i386/linux)) Content-Length: 0 Warning: 392 221.202.129.167:5060 "Noisy feedback tells: pid=28398 req_src_ip=61.136.210.7 req_src_port=5339 in_uri=sip:2001@netvoip.cn out_uri=sip:2001@221.202.129.167:5061 via_cnt==1" RECEIVE TIME: 201759671 RECEIVE << 221.202.129.167:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.16.34:5060;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 To: <sip:2001@netvoip.cn>;tag=as6e0c8cea Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 CSeq: 53904 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:2001@221.202.129.167:5061> Proxy-Authenticate: Digest realm="asterisk", nonce="43fb4632" Content-Length: 0 SEND TIME: 201759671 SEND >> 221.202.129.167:5060 ACK sip:2001@netvoip.cn SIP/2.0 Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 To: <sip:2001@netvoip.cn>;tag=as6e0c8cea Contact: <sip:2000@10.1.16.34:5060> Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 CSeq: 53904 ACK Max-Forwards: 70 Content-Length: 0

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[Users] creating postgres tables
by Klaus Darilion 28 Jul '05

28 Jul '05

Hi! I recently checked in scripts/postgresqldb.sh - a rewrite of the mysql version for postgresql. Currently tested are only the "create" and "drop" commands, thus the other commands are disabled. I had to modify the tables defintions and I'm not sure if this broke something. Thus, please report problems. regards, klaus

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[Serusers] Predictive Dialer
by Kapil Dhawan 28 Jul '05

28 Jul '05

Dear List Any clues on implementing some opensource Predictive Dialer with SER. Regards _________________________________________________________________ Aamir Khan is back! http://server1.msn.co.in/SP05/Mangalpandey/index.html See him in the mustached avatar in Mangal Pandey.

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[Serusers] my acc table works !!!
by zhu lizhong 28 Jul '05

28 Jul '05

Hello, guys: I sent a help message to you before since I got problems in acc tables. Now i made some changes, and it finally works. thanks you for guys. now I want to share my tips to you who are need acc records, because many people have this problem. first, make sure that you must confiugre Makefile in acc module and recompile it properly. second load acc module into ser.cfg. third, put all necessary into ser and route section. please follow the sequences. maybe some parts are not really necessary. but there is data in acc. Now i show you my ser.cfg. loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" modparam("auth_db|uri_db|usrloc|acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 1) modparam("rr", "enable_full_lr", 1) modparam("acc", "db_flag", 1) # Record Route Section # testing from zhu----------------------------------------------------------------- if (method=="INVITE" || method=="ACK") { log (1, "%method - INVITE or ACK MESSAGE RECEIVED - START ACCOUNTING\n"); setflag(1); # record_route(); }; if (method=="BYE" || method=="CANCEL") { log (1, "%method - STOP ACCOUNTING\n"); setflag(1); # record_route(); }; # end of testing -------------------------------------------- if (method!="REGISTER") { record_route(); setflag(2); }; this is data section received from acc table of mysql: | 333 <sip:333@192.168.1.26:5061>;tag=249979601 | <sip:800@192.168.1.26>;tag=1273583692 | 200 | INVITE | sip:800@192.168.1.188:5061 | sip:800@192.168.1.188:5061 | sip:333@192.168.1.26:5061 | sip:800@192.168.1.26 | 4E916DD5-2141-3B23-08CC-6F2CEA3B405B(a)192.168.1.26 | 333 | 192.168.1.26 | 249979601 | 1273583692 | 2005-07-28 01:50:01 | 20050728095001 | 0 | 0 | | 333 <sip:333@192.168.1.26:5061>;tag=249979601 | <sip:800@192.168.1.26>;tag=1273583692 | 200 | ACK | sip:800@192.168.1.188:5061 | sip:800@192.168.1.188:5061 | sip:333@192.168.1.26:5061 | sip:800@192.168.1.26 | 4E916DD5-2141-3B23-08CC-6F2CEA3B405B(a)192.168.1.26 | 333 | 192.168.1.26 | 249979601 | 1273583692 | 2005-07-28 01:50:01 | 20050728095001 | 0 | 0 | | 333 <sip:333@192.168.1.26:5061>;tag=249979601 | <sip:800@192.168.1.26>;tag=1273583692 | 200 | BYE | sip:800@192.168.1.188:5061 | sip:800@192.168.1.188:5061 | sip:333@192.168.1.26:5061 | sip:800@192.168.1.26 | 4E916DD5-2141-3B23-08CC-6F2CEA3B405B(a)192.168.1.26 | 333 | 192.168.1.26 | 249979601 | 1273583692 | 2005-07-28 01:50:03 | 20050728095003 | 0 | 0 | try it!

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[Serusers] No record inserted into the acc table of mysql!
by zhu lizhong 28 Jul '05

28 Jul '05

Hello, guys: thanks for your replies. I run SER and make sip call. I configured ser.cfg and some message are displayed as fowwlow: 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_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" modparam("acc","log_level",1) modparam("acc","log_flag",1) # modparam("acc", "db_url","flatstore:/var/log/acc") modparam("acc", "db_flag", 1) # modparam("acc", "db_missed_flag", 1) # modparam("acc", "failed_transactions", 2) modparam("auth_db|uri_db|usrloc|acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 1) if (method=="INVITE" || method=="ACK") { log (1, "%method - INVITE or ACK MESSAGE RECEIVED - START ACCOUNTING\n"); setflag(1); # record_route(); }; if (method=="BYE" || method=="CANCEL") { log (1, "%method - STOP ACCOUNTING\n"); setflag(1); # record_route(); }; # end of testing -------------------------------------------- if (method!="REGISTER") { record_route(); setflag(2); }; some messages are capured : ACK sip:800@192.168.1.188:5061 SIP/2.0 Record-Route: <sip:192.168.1.26;ftag=78086448;lr=on> Via: SIP/2.0/UDP 192.168.1.26;branch=0 Via: SIP/2.0/UDP 211.25.2.30:5061;received=192.168.1.26;rport=5061;branch=z9hG4bK255A602DB910243AC238B003FB36CE58 From: 333 <sip:333@192.168.1.26:5061>;tag=78086448 To: <sip:800@192.168.1.26>;tag=542148979 Contact: <sip:333@211.25.2.30:5061> Call-ID: 0CF994AF-C18C-4447-A6D3-758FF4EB99A8(a)192.168.1.26 CSeq: 9213 ACK Max-Forwards: 16 Content-Length: 0 SIP/2.0 200 Ok Via: SIP/2.0/UDP 211.25.2.30:5061;received=192.168.1.26;rport=5061;branch=z9hG4bK7D8A9CF509DF4CAC2EB9DF19BEB26EE3 From: 333 <sip:333@192.168.1.26:5061>;tag=78086448 To: <sip:800@192.168.1.26>;tag=542148979 Contact: <sip:800@192.168.1.188:5061> Call-ID: 0CF994AF-C18C-4447-A6D3-758FF4EB99A8(a)192.168.1.26 CSeq: 9214 BYE Server: X-Lite release 1105d Content-Length: 0 2(2160) WARNING:vqm_resize: resize(0) called 2(2160) DEBUG: add_to_tail_of_timer[2]: 0x422c2440 2(2160) DEBUG: reply relayed. buf=0x8103438: SIP/2.0 2..., shmem=0x422c3e68: SIP/2.0 2 2(2160) DBG: trans=0x422c23f8, callback type 128, id 0 entered 2(2160) ACC: transaction answered: method=BYE, i-uri=sip:800@192.168.1.188:5061, o-uri=sip:800@192.168.1.188:5061, call_id=0CF994AF-C18C-4447-A6D3-758FF4EB99A8(a)192.168.1.26, from=333 I know there is something happened, maybe, somewhere is worng in route section. Please give me some suggestions! thanks in advance! lizhong

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[Serusers] new replication module
by Jev 27 Jul '05

27 Jul '05

Hi, On line 136 of modules/tm/doc/tm.xml there is reference made to a new replication module. Is this a public module? Where can I find out more about this module? Cheers, -Jev

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Re: [Serusers] Problem : Can SER process the reINVITE messages pr operly?
by Alberto Cruz 27 Jul '05

27 Jul '05

Yes it is working fine. Regards Ricardo Martinez wrote: > Alberto. > Thanks for your answer. You mean that your platform works well > with the reINVITE's to a NAT'd endpoint?. > > Regards, > Ricardo.- > > -----Mensaje original----- > De: Alberto Cruz [mailto:acruz@tekbrain.com] > Enviado el: Miércoles, 27 de Julio de 2005 16:12 > Para: Ricardo Martinez > CC: 'Greger V. Teigre'; 'serusers(a)lists.iptel.org' > Asunto: Re: [Serusers] Problem : Can SER process the reINVITE > messages pr operly? > Importancia: Alta > > Ricardo this block is working fine to me: > > #------------------------------------------------- > # subsequent messages withing a dialog should take the > # path determined by record-routing > #------------------------------------------------- > if (loose_route()) { > append_hf("P-hint: rr-enforced\r\n"); > # The following lines are added due media proxy > if (has_totag()&&(method=="INVITE" || > method=="ACK")) { > if (client_nat_test("3") || > search("^Route:.*;nat=yes")) { > # Mark as NAT'ed > setflag(6); > use_media_proxy(); > }; > }; > if ((method=="BYE") && ((client_nat_test("3") || > search("^Route:.*;nat=yes")))) { > fix_nated_contact(); > force_rport(); > }; > # end media session for BYE and CANCEL is done above > # before entering the loose route. no need to call > it here > route(1); > break; > }; > > Maybe you have something wrong at your mediaproxy.ini file. > > Regards > > Alberto Cruz > >> >>

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[Serusers] SDP of payload settings
by Jason Mohler 27 Jul '05

27 Jul '05

Hello All, I am using SER and ran into a stumper. I am trying to set my SDP payload settings to the following below but I do not know how to config it in my invite message. Does anyone have a solution to my problem? Thank you for any information thqt you can provide.. a=rtpmap:0 PCMU/8000. a=rtpmap:100 X-NSE/8000. a=fmtp:100 192-194. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.

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Re: [Users] ACK processing (was: CANCEL processing)
by Michael Ulitskiy 27 Jul '05

27 Jul '05

Thanks, Bogdan. So it's still better to process CANCEL the same as INVITE to avoid this race condition. My next question is about ACK processing. In my observations if one uses statefull processing ACKs are either loose_routed or absorbed by t_relay(). In my test scenarios I've never seen otherwise. So, again, my question is if the following is OK: if (loose_route()) { #in-dialog ACKs are forwarded here do something t_relay(); exit; } if (is_method("ACK")) { #out-of-dialog ACKs are absorbed here t_relay(); exit; } if (uri==myself) { do lookups that rewrite RURI t_relay(); } Thank you, Michael > On Monday 25 July 2005 07:39 am, you wrote: > > Hi Michael, > > > > that is correct. just note you may encounter some races between INVITEs > > and CANCELs in the case of a fast CANCEL (CANCEL follows the INVITE > > almost instantly) - when one process is still handling the INVITE (the > > transaction may not be build yet), another process receives the CANCEL > > and it will not be able to match it to a transaction in this case, the > > CANCEL will be forward based on its RURI without any info from INVITE..... > > > > regards, > > bogdan > > > > Michael Ulitskiy wrote: > > > > >Hello, > > > > > >I have a question about CANCEL processing. > > >I read on the mailing list that CANCEL will be automatically matched by t_relay > > >to transaction it's cancelling, if needed transformation to RURI will be automatically > > >applied and then it will be automatically send to correct destination. > > >I'm experimenting with openser 0.10.x and it seems to be true, but I'd like to confirm > > >that the following is OK: > > > > > >if (loose_route()) { > > > do something > > > t_relay(); > > > break; > > >} > > >if (is_method("CANCEL")) { > > > t_relay(); > > > break; > > >} > > >if (uri==myself) { > > > do lookups that rewrite RURI > > > t_relay(); > > >} > > > > > >Thank you, > > > > > > > > > > > > > >

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RE: [Serusers] Problem : Can SER process the reINVITE messages pr operly?
by Ricardo Martinez 27 Jul '05

27 Jul '05

Alberto. Thanks for your answer. You mean that your platform works well with the reINVITE's to a NAT'd endpoint?. Regards, Ricardo.- -----Mensaje original----- De: Alberto Cruz [mailto:acruz@tekbrain.com] Enviado el: Miércoles, 27 de Julio de 2005 16:12 Para: Ricardo Martinez CC: 'Greger V. Teigre'; 'serusers(a)lists.iptel.org' Asunto: Re: [Serusers] Problem : Can SER process the reINVITE messages pr operly? Importancia: Alta Ricardo this block is working fine to me: #------------------------------------------------- # subsequent messages withing a dialog should take the # path determined by record-routing #------------------------------------------------- if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); # The following lines are added due media proxy if (has_totag()&&(method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { # Mark as NAT'ed setflag(6); use_media_proxy(); }; }; if ((method=="BYE") && ((client_nat_test("3") || search("^Route:.*;nat=yes")))) { fix_nated_contact(); force_rport(); }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here route(1); break; }; Maybe you have something wrong at your mediaproxy.ini file. Regards Alberto Cruz

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[Serusers] How to avoid sending messages more than once at the same destination?
by Alberto Cruz 27 Jul '05

27 Jul '05

Hi I have an issue. We are using B2BUA in the middle of SER and the UA and the PSTN-GW. Sometimes when a call is canceled the B2BUA finished the call transaction and clear the call id from the memory before it receives the 408 message from SER so SER continues sending the same message many times until it completes the max forwards time. Is there a way to tell SER if it don't receive a replay for that message it stop sending the message again? Regards Alberto Cruz

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RE: [Serusers] Problem : Can SER process the reINVITE messages pr operly?
by Ricardo Martinez 27 Jul '05

27 Jul '05

Hello Greger and SER users. About this issue here are some test that i made. I added the lookup("location") as you recommend me : # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { lookup("location"); if (isflagset(6) || client_nat_test("3") || search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; Unfortunately, i'm still unable to have voice in both ways, but this modification seems to handle better the reINVITE message. And this is why. For the normal approach (i mean the initial onsip_pstn configuration) the second INVITE was not using the mediaproxym as Greger pointed. Now, as you can see in the debugs that i'm attaching, the second invite is using the mediaproxy, and therefore the "c" parameter in the SDP message is being correctly modified by SER. So, i don't understand why this time the call is failing. Could it be that mediaproxy is not capable to handle the reINVITEs?. For example in the debug from mediaproxy (i'm also attaching this one), i see a warning message like this : warning: Received packet from a third party: 200.0.0.7:16432. It seems like mediaproxy is not accepting "voice packets" from the second IP (the one indicated by the reINVITE). Why? Also, when the call is established this is the output from the "session" command : Caller Via Called Status Duration Codec Type Traffic ---------------------------------------------------------------------------- -------------------------------------- 200.0.0.6:17480 - 200.0.0.5:35094 - 200.0.0.4:46115 inactive 0'24" G729 Audio 0/16.75k/16.75k The call is established but between the first IP (Asterisk box) and the endpoint public IP. So, could this be a problem with mediaproxy?. I read the changelog from the version 1.3.1 and this is what i found : Changes from version 1.2.1 to 1.3.0 ----------------------------------- - Only create a new session if a request belongs to the first INVITE (no longer creates new sessions on re-INVITEs) could it be related?. I also have a general question about the reINVITE. As you can see the first INVITE has in his URI the domain name for the number : 5555848114(a)sipvoiss.desarrollo.redvoiss.net (Frame 1 from the debug). The reINVITE sent by Asterisk has the URI : 5555848114(a)200.0.0.4. Is this ok?, despite of this the reINVITE is correctly redirected to the endpoint. (You can check this in the frame 13 from the debug). Thanks again. Regards, Ricardo Martinez.-

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[Users] rpid problem
by Klaus Darilion 27 Jul '05

27 Jul '05

Hi! I'm using openser CVS and try to deliver the rpid using auth_radius. But the rpid never appears in the AVPs. Also other AVP-SIP attributes (e.g. SIP-AVP = #101:TEST) do not appear in the AVP. Filling AVPs in the .cfg using avp_write("$ruri/username","$ruser"); works fine and are printed with avp_print. Any ideas how I can track down this problem? thanks klaus The radius server responds with a SIP-AVP (225) attribute (verfied with tcpdump): rpid:<sip:+4359966366102@1013cbc.com> my cfg snippet (full config attached): if (!radius_proxy_authorize("")) { xlog("L_WARN","wrong or no credentials - challenging client ..."); proxy_challenge("", "0"); exit; }; xlog("L_WARN","start avp_print()-"); avp_print(); xlog("L_WARN","stop avp_print()--"); the debug log: 8(5807) checking REGISTER authentication ... 8(5807) check_nonce(): comparing [42e78dca8153d605c63042be302f64af00e1abfc] and [42e78dca8153d605c63042be302f64af00e1abfc] 8(5807) DEBUG:auth_radius:radius_authorize_sterman: Success 8(5807) xl_printf: final buffer length 37 8(5807) start avp_print() - 8(5807) xl_printf: final buffer length 37 8(5807) stop avp_print() -- 8(5807) xl_printf: final buffer length 58 # # $Id: openser.cfg,v 1.1.1.1 2005/06/13 16:47:30 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=7 # 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/openser_fifo" listen=1.2.3.83:5060 log_facility=LOG_LOCAL4 # /var/log/openser.log # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" mpath="/usr/lib/openser/modules" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "postgres.so" loadmodule "domain.so" loadmodule "alias_db.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" loadmodule "auth_radius.so" loadmodule "avpops.so" # ----------------- setting module-specific parameters --------------- # database configuration modparam("usrloc", "db_url", "postgres://openser:xxxxxx@localhost/openser") modparam("domain|uri_db|alias_db", "db_url", "postgres://openserro:xxxxxxx@localhost/openser") # radius configuration modparam("auth_radius", "radius_config", "/etc/openser/radiusclient.conf") # multi domain configuration modparam("domain", "db_mode", 1) # Use caching in domain module modparam("alias_db|usrloc|registrar", "use_domain", 1) # group, group_radius, speeddial, uri_db, avpops # -- usrloc params -- modparam("usrloc", "db_mode", 1) # 0=no DB, 1 = write through, 2=caching # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # AVPs # I:101 = cbcprefix # call by call provider prefix # I:102 = ruser # userpart in request uri # I:103 = rdomain # domainpart in request uri # I:104 = dummy1 # I:105 = dummy2 # modparam("avpops","avp_aliases","cbcprefix=I:101;ruser=I:102;rdomain=I:103;dummy1=I:104;dummy2=I:105") # ------------------------- request routing logic ------------------- # main routing logic route{ xlog("L_WARN","[$Tf] $rm $ru ($fu --> $tu)"); #xlog("L_WARN","reference to message buffer: $mb"); xlog("L_WARN","playing around with avps..."); avp_write("$ruri/username","$ruser"); avp_write("$ruri/domain","$rdomain"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { xlog("L_WARN","too many hops, reply with 483 ..."); sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { xlog("L_WARN","message too big, reply with 513 ..."); sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!is_method("REGISTER")) { record_route(); } # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { log(1,"loose_route processing ..."); # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); exit; }; if (!is_method("REGISTER|CANCEL|ACK|BYE|OPTIONS|INFO")) { if (is_from_local()) { xlog("L_WARN","checking authentication ..."); # digest authentication if (!radius_proxy_authorize("")) { xlog("L_WARN","wrong or no credentials - challenging client ..."); proxy_challenge("", "0"); exit; }; xlog("L_WARN","start avp_print()--------"); avp_print(); xlog("L_WARN","stop avp_print()---------"); consume_credentials(); xlog("L_WARN","username/password correct ..."); log(1,"adding rpid header ..."); append_rpid_hf(); } } if (!is_uri_host_local()) { log(1,"outbound request ..."); # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); exit; }; # 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 (is_uri_host_local()) { if (is_method("REGISTER")) { if (is_from_local()) { log(1,"processing REGISTER ..."); xlog("L_WARN","checking REGISTER authentication ..."); # digest authentication if (!radius_www_authorize("")) { xlog("L_WARN","wrong or no credentials in REGISTER - challenging client ..."); www_challenge("", "0"); exit; }; xlog("L_WARN","start avp_print() of REGISTER--------"); avp_print(); xlog("L_WARN","stop avp_print() of REGISTER---------"); consume_credentials(); xlog("L_WARN","username/password correct in REGISTER, saving location ..."); save("location"); exit; } else { xlog("L_WARN","REGISTER for unknown domain received, reply with 403 ..."); sl_send_reply("403","Use your own proxy"); exit; } }; lookup("aliases"); if (!is_uri_host_local()) { append_hf("P-hint: outbound alias\r\n"); route(1); exit; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # send to at43 main proxy xlog("L_WARN","user not found, forwarding to main proxy ..."); t_relay_to_udp("1.2.3.160", "5060"); #sl_send_reply("404", "Not Found"); exit; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { xlog("L_WARN","route[1] entered ..."); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP xlog("L_WARN","t_relay the request ..."); if (!t_relay()) { sl_reply_error(); }; xlog("L_WARN","... leaving route[1]"); }

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[Users] "SIP-AVP" RADIUS AVP
by Bogdan-Andrei Iancu 27 Jul '05

27 Jul '05

Hi folks, I extended the syntax of "SIP-AVP" RADIUS AVP to allow definition of any types of OpenSER AVP - ID / STRING AVP name, INTEGER / STRING AVP value . The new syntax is: * value = SIP_AVP_NAME SIP_AVP_VALUE * SIP_AVP_NAME = STRING_NAME | '#'ID_NUMBER * SIP_AVP_VALUE = ':'STRING_VALUE | '#'NUMBER_VALUE . This syntax is backward compatible with the old one. Examples: "email:joe@yahoo.com" -> STRING NAME AVP (email) with STRING VALUE (joe(a)yahoo.com) "#14:joe@yahoo.com" -> ID AVP (14) with STRING VALUE (joe(a)yahoo.com) "age#28" -> STRING NAME AVP (age) with INTEGER VALUE (28) "#14#28" -> ID AVP (14) with INTEGER VALUE (28) RADIUS fans, please give it a try (if needed) and let me know if there are any issues. regards, bogdan PS: it will take some hours to become available on the public CVS server (sourceforge policy).

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[Serusers] fifo flat_rotate
by Steve Blair 27 Jul '05

27 Jul '05

Hello: I'm unclear on exactly what the command serctl fifo flat_rotate is suppose to do? If I merge all of the individual acc_N.log files into one report then delete these same files ser automatically creates new ones as expected. I don't see where this command has any affect. Do I really need to execute this command after removing (or moving) the accounting files? Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@upenn.edu

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[Serusers] Thank you for using voice.dyndns.tv's Voicemail.
by fedora＠voice.dyndns.tv 27 Jul '05

27 Jul '05

Hello 8000(a)voice.dyndns.tv, "8400(a)voice.dyndns.tv" <sip:8400@voice.dyndns.tv> left a voice message for you. Thank you for using voice.dyndns.tv's Voicemail. Your voicemail system. --------------------------------------------------------------- This message was created by voice.dyndns.tv's voicemail system. Please review www.voice.dyndns.tv for more information. ---------------------------------------------------------------

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[Serusers] Missed calls - no entry in database
by Aisling 27 Jul '05

27 Jul '05

Hi, Sorry for re-posting but I am still stuck with the following: 1) I have modified my ser.cfg (based on ser 0.9.0 onsip getting started document) for missed calls based on example from the Terena IP Telephony Cookbook. I get an error with the line: acc_request("404 Not Found"); parse error - unknown command, missing loadmodule? Even though I have loaded the acc.so and tm.so modules and used modparam("acc", "log_missed_flag", 3); 2) Besides that when I comment the above out, I am under the impression that if a user 314 rings user 2092 and hangs up before user 2092 answers i.e. sending a CANCEL message - an entry for this should now be in the "missed_calls" table fo the ser database....Is this correct?...I tested this scenario and the table was still empty. Could someone explain what I am missing? Many thanks (relevant part of ser.cfg included below) Aisling. //ser.cfg # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { #sl_send_reply("404", "Not Found"); #break; #new entered 20/07/05 #call invitations to off-line users are reported using #the acc_request action; to avoid #duplicate reports on request retransmissions, request # is processed statefully (t_newtran, t_reply) if ((method=="INVITE" || method=="ACK") && t_newtran()){ t_reply("404", "Not Found"); #acc_request("404 Not Found"); break; }; #all other requests to off-line users are simply #replied statelessly and no reports are issued sl_send_reply("404", "Not Found"); break; } else { #user on-line; report on failed transactions; #mark the transaction for reporting using the same #number as configured above; if the call is really missed, # a report will be issued setflag(3); #forward to the users current destination t_relay(); break; }; }; route(1); } -------------------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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[Users] debian postgresql package
by Klaus Darilion 27 Jul '05

27 Jul '05

Hi! HEAD version requires "libpq-dev" (libpq4), which is not in debian stable. I'm on stable and use "postgresql-dev" (libpq3) without any problems. Is there a reason why libpg4 is required? If not, I suggest stay with debian stable and require "postgresql-dev" (libpq3). regards, klaus

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[Serusers] Jabber module on latest stable (0.9.x) versions
by Pietro Ravasio 27 Jul '05

27 Jul '05

Hi! Is there anybody who's succesfully running jabber module on top of a recent "stable" ser/openser version? I can't get the server running, it hangs and I have then to kill it with "kill -9". This is the error log (running on openser 0.9.5, not forking, debug level = 8): ... 0(21394) DEBUG: init_mod_child (1): avpops 0(21394) get_connection(): Connection found in the pool 0(21394) XJAB:xjab_check_workers: worker[0][pid=21403] has exited - status=0 err=21403 errno=0 0(21394) XJAB:21402:xjab_check_workers: create a new worker[0] 0(21394) XJAB:xj_worker[0]:21405: exiting - wrong parameters 0(21394) XJAB:xjab_check_workers: worker[1][pid=21404] has exited - status=0 err=21404 errno=0 0(21394) XJAB:21402:xjab_check_workers: create a new worker[1] 0(21394) XJAB:xj_worker[1]:21406: exiting - wrong parameters 0(21394) SIGCHLD received: we do not worry about grand-children I don't really understand why I get different errors on different modules lowering the debug level (3): 2(0) get_connection(): Inherited open database connections, this is not a good idea 2(0) db_init(): Could not create a connection 2(0) ERROR: child_init(-2): Error while connecting database 2(0) init_mod_child(): Error while initializing module usrloc Please note that everything is working fine if I disable jabber module I have seen jabber module hasn't been modified recently, is it possible it is now somehow not compatibile with *sql modules? -- Ing. Pietro Ravasio - pietro.ravasio(a)abbeynet.it Enterprise Market Manager Abbeynet S.p.A. - http://www.abbeynet.it TEL: +390702109044 GSM: +393282172055 FAX: +390702109055

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[Users] Malformated packets, how to handle???
by Kanakatti Mahesh Subramanya 27 Jul '05

27 Jul '05

Every now and then, some of our users take it into their heads to muck around with their fones, and they (usually) do not do anything evil. However, I occassionally end up with sitiations where the fones generate 'fun' SUBSCRIBE packets that look like SUBSCRIBE <foo.bar>@aptela.com yeah, thats '<foo.bar>', not 'foo.bar' Which, by definition, causes openser to complain about parse_header failing, etc., etc. Short of scanning logs, doing ngreps, etc., is there any way to deal with this? I'd think the ability to deal with malformatted packets with a '400' or '416' would be useful? cheers

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[Serusers] Do i need Call id while sending SIP IM?
by Abhijit A.Mahajani 27 Jul '05

27 Jul '05

Do I need Call id field while sending any SIP Instant Message? Is it Mandatory ? Can i Send Instant Message just by specifying , from & to fields and the Msg? Regards. Abhijit

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[Users] gw-pstn.3.07.cfg with Call-Transfer support
by Bastian Schern 27 Jul '05

27 Jul '05

Hello, I tried out the SER config "gw-pstn.3.07.cfg" from www.onsip.org. Nearly everything works fine but "Call-Transfer" is not working. Is it possible to integrate "Call-Transfer" support into the "gw-pstn.3.07.cfg"? Does anybody integrates this already? Regards Bastian debug=3 fork=yes log_stderror=no listen=192.0.2.13 # INSERT YOUR IP ADDRESS HERE 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/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") 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=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.0.2.13:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else 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(5); 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" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); 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 (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (uri=~"^sip:011[0-9]*@") { # International PSTN route(4); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { # Domestic PSTN route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; route(5); route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("192.0.2.245"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); route(5); route(1); } route[5] { # ----------------------------------------------------------------- # RTP Proxy Enabler # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } route[6] { # ------------------------------------------------------------------------ # ACK Handler # ------------------------------------------------------------------------ # ------------------------------------------------------------------------ # Aliases Section # ------------------------------------------------------------------------ lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; lookup("location"); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

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Re:[Serusers] Missed calls - no entry in database
by Greger V. Teigre 27 Jul '05

27 Jul '05

Aisling, The CANCEL will have a to_tag and be caught by the loose_route test and t_relayed. You will have to have the CANELtest and set the flag somwhere there as well. g-) ---------------------------------------------------------------------- Hi, Sorry for re-posting but I am still stuck with the following: 1) I have modified my ser.cfg (based on ser 0.9.0 onsip getting started document) for missed calls based on example from the Terena IP Telephony Cookbook. I get an error with the line: acc_request("404 Not Found"); parse error - unknown command, missing loadmodule? Even though I have loaded the acc.so and tm.so modules and used modparam("acc", "log_missed_flag", 3); 2) Besides that when I comment the above out, I am under the impression that if a user 314 rings user 2092 and hangs up before user 2092 answers i.e. sending a CANCEL message - an entry for this should now be in the "missed_calls" table fo the ser database....Is this correct?...I tested this scenario and the table was still empty. Could someone explain what I am missing? Many thanks (relevant part of ser.cfg included below) Aisling. //ser.cfg # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { #sl_send_reply("404", "Not Found"); #break; #new entered 20/07/05 #call invitations to off-line users are reported using #the acc_request action; to avoid #duplicate reports on request retransmissions, request # is processed statefully (t_newtran, t_reply) if ((method=="INVITE" || method=="ACK") && t_newtran()){ t_reply("404", "Not Found"); #acc_request("404 Not Found"); break; }; #all other requests to off-line users are simply #replied statelessly and no reports are issued sl_send_reply("404", "Not Found"); break; } else { #user on-line; report on failed transactions; #mark the transaction for reporting using the same #number as configured above; if the call is really missed, # a report will be issued setflag(3); #forward to the users current destination t_relay(); break; }; }; route(1); } -------------------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.

1 0

RE: [Serusers] help!---how to see the debug information in detail
by Sam Lee 27 Jul '05

27 Jul '05

Are you talking about the amount of information u can see during a normal SIP Operation ? I haven't been able to make them show up using the normal method , so opted for a exec_msg way where i echo a message with a global variable into my /var/log/syslog . The steps are detailed here : http://www.iptel.org/ser/doc/seruser/seruser.html#AEN1140 Regards, Sam ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of chenhui Sent: Tuesday, July 26, 2005 5:03 PM To: serusers(a)lists.iptel.org Subject: [Serusers] help!---how to see the debug information in detail hi all, The /var/log/messages only shows limited information. Could anybody tell me how to see the debug information in detail? Thanks!

1 0

Re: [Serusers] help!---how to see the debug information in detail
by chenhui 27 Jul '05

27 Jul '05

I check the directory /var/log ,but have not found the file "syslog" and "debug". How to record the bebug messages in these files? Thanks a lot! ----- Original Message ----- From: Alberto Cruz To: chenhui Cc: serusers(a)iptel.org Sent: Tuesday, July 26, 2005 9:39 PM Subject: Re: [Serusers] help!---how to see the debug information in detail Try /var/log/syslog or /var/log/debug Regards Alberto Cruz chenhui wrote: hi all, The /var/log/messages only shows limited information. Could anybody tell me how to see the debug information in detail? Thanks! ---------------------------------------------------------------------------- _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers

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