sr-users March 2007

sr-users@lists.kamailio.org

192 participants
380 discussions

by H Quintana

Hi everybody, I'm using Xlite 3.0 and postgresql 8.2.3, the information about the presentity is properly stored in the table. The query brings the value for the field "body" but after the "conversion" to string the value is empty. Is Someone running the presence module with postgres? The log is at: http://cut.and.paste.org/index.php?id=952 I added this to notify.c in the line 514: DBG("PRESENCE:NOTIFY:Body array n=%d, content:<%.*s>\n", i, body_array[i]->len, body_array[i]->s); and in the log you will find: 0(14253) PRESENCE:NOTIFY:Body array n=0, content:<> before the line: 0(14253) PRESENCE:agregate_xmls: ERROR while parsing xml body message Thanks for your help, Humberto Quintana ____________________________________________________________________________________ TV dinner still cooling? Check out "Tonight's Picks" on Yahoo! TV. http://tv.yahoo.com/

17 years, 3 months

[Users] lcr, branch route and setdsturi

by Andreas Granig

Hi, Is it possible to set the dst-uri in a branch-route (in openser 1.1)? It would be useful if you could for example do least-cost-routing with lcr, where some of the gateways need an extra hop via an outbound-proxy, which could be marked with a prefix in the gw-table, like: route[1] { # let's assume uri = 'sip:12345@something' load_gws(); next_gw(); # lcr module selects a gw according to lcr patterns, and some of # them needs an outbound-proxy, which are marked by prefix 'out' t_on_failure("1"); t_on_branch("1"); # do something else and t_relay() } failure_route[1] { if( /* error code which triggers failover */ ) { if(next_gw()) { t_on_branch("1"); t_on_failure("1"); t_relay(); } else { /* error */ } } } branch_route[1] { # check if it's marked for using an outbound proxy: if(uri =~ "^sip:out[0-9]+@") setdsturi("sip:my.outbound.ip:5060"); } Would this work? Comments? Regards, Andreas This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

17 years, 3 months

[Serusers] NAT Handling - SER behind a router

by TZieleniewski

Hi!! Right now I have SER based voip system but inside the local network. I would like to enable the external internet connection. Is it possible to realize the NAT handling but with SER-RTPProxy or SER-MediaProxy not with the public IP so still in the local net but with the strict forwarding rules on the router. where router ofcaurse has the public IP So that for instance having SER and media proxy inside local network but with whole SIP traffic routed to SER and RTP to media proxy?? Is one of the above methods more recommened at the moment with ser2?? Thank you in advance Best Tomasz

17 years, 3 months

[Serusers] Call forwarding

by Jai Rangi

Hello, I have a ser and asterisk setup. Users register on ser and all the national and international calls go through asterisk. Asterisk handles the billing part for long distance calls. Now I got an issue with call forwarding. Here is the situation. Useragent A calls Usergent B. User B has setup the call the call forwarding to his cell phone. So when user A calls user B, ser check that user B has set the forwarding and send the call to Asterisk box. Asterisk box sees user A in the callerID and bills user A for that call. We want user B to be billed for that call, cause he is forwarding the calls to his cell phone (long distance number). How can I achieve this. Thank you, -Jai

17 years, 3 months

Re: 答复: [Users] ERROR 1062 (23000): Duplicateentry

by sukerry

郭军,您好！ please modify openserctrlrc, set ?WEB?(some what) to be yes ======= 2007-03-21 16:49:35 您在来信中写道：======= >Why don't you use openserctl add ....to add new user? > >-----邮件原件----- >发件人: alper ozbilen [mailto:alperbilen@hotmail.com] >发送时间: 2007年3月21日 16:45 >收件人: users(a)openser.org >主题: [Users] ERROR 1062 (23000): Duplicate entry > >Dear All, > >I can not insert a new subscriber from mysql console. >When I look at subscireber table, ha1 & ha1b & and phplib_id fields looks >null. So, i can only insert one new subscriber. When i attempt to insert >second one, i get error written below: > >ERROR 1062 (23000): Duplicate entry '' for key 3 > >Note : My insert sentence like that: >insert into subscriber (username, domain, password, email_address) values >("testuser","myfon.com","testpasswd","emailoftestuser(a)myfon.com"); > >I am confused about what i should to do to fix null statement of ha1 & ha2 & >and phplib_id fields. > >on the other side, there is no probem to add new user through openserctl > >What was the problem? I will wait your kind suggestion. > >Regards, >Alper Ozbilen > >_________________________________________________________________ >Exercise your brain! Try Flexicon. >http://games.msn.com/en/flexicon/default.htm?icid=flexicon_hmemailtaglinema… > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　sukerry 　　　　　　　　sukerry(a)126.com 　　　　　　　　　　2007-03-21

17 years, 3 months

[Users] ERROR 1062 (23000): Duplicate entry

by alper ozbilen

Dear All, I can not insert a new subscriber from mysql console. When I look at subscireber table, ha1 & ha1b & and phplib_id fields looks null. So, i can only insert one new subscriber. When i attempt to insert second one, i get error written below: ERROR 1062 (23000): Duplicate entry '' for key 3 Note : My insert sentence like that: insert into subscriber (username, domain, password, email_address) values ("testuser","myfon.com","testpasswd","emailoftestuser(a)myfon.com"); I am confused about what i should to do to fix null statement of ha1 & ha2 & and phplib_id fields. on the other side, there is no probem to add new user through openserctl What was the problem? I will wait your kind suggestion. Regards, Alper Ozbilen _________________________________________________________________ Exercise your brain! Try Flexicon. http://games.msn.com/en/flexicon/default.htm?icid=flexicon_hmemailtaglinema…

17 years, 3 months

[Users] OpenSER and FWD

by Michel Bensoussan

Hello. How can I configure OpenSER to outgoing all call to FWD and more important to ensure that all INCOMING call will go through OpenSER? Thanks. Regards, Michel.

17 years, 3 months

[Users] RPID single quotes problem

by Daryl Sanders

Is there an easy way to strip the single quotes from RPID in DBTEXT? When I use sc.dbtext it automatically adds single quotes to the entry like so... '17145551212' As a result when I add RPID to my header with the folowing line, I get the single quotes in the header. append_hf("P-Asserted-Identity: \"User\" <sip:+$avp(s:rpid)@x.x.x.x>\r\n"); Regards, Daryl

17 years, 3 months

Re: [Users] loose_route problem with 1.2

by jh＠tutpro.com

it turned out that if i have in openser.cfg listen=x.x.x.x alias="sip.foo.bar:5060" port=5060 auto_aliases=no then openser advertises at startup: Aliases: *: sip.foo.bar:5060:* and loose_route DOES NOT recognize that Route header <sip:sip.foo.bar;lr;transport=TCP> is local. on the other hand, if i comment line #auto_aliases=no openser advertises at startup: Aliases: tcp: sip.foo.bar:5060 udp: sip.foo.bar:5060 *: sip.foo.bar:5060:* and it DOES recognize that above Route header is local. if i uncomment auto_aliases=no and change alias="sip.foo.bar:5060" to alias="udp:sip.foo.bar:5060" alias="tcp:sip.foo.bar:5060" then loose_route DOES NOT recognize that the route header is local. my question is, how should i write the alias statements so that the route header is recognized local even when i have auto_aliases=no? -- juha

17 years, 3 months

Re: R: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem

by Kostas Marneris

Hello, What I understand (from Frame12) Session Initiation Protocol Request-Line: BYE sip:000003@1.255.ua_priv__IP SIP/2.0 is that your GW does not comply with the fixed SIP UA Contact address Contact: <sip:000003@213.156.ua_pub_IP:1176> which SER seems to correctly fix it (from Frame6). GW must send the BYE request back to the fixed addr : 213.156.ua_pub_IP:1176. Try a different PSTN-GW or try to isolate the RTP-Proxy solution. I am really confused of your ser.cfg. You use force_rtp_proxy("l") on Loose Route Section / INVITE block, but you don't use force_rtp_proxy() on INVITE handler (Route[3]) and you don't also use rtp_proxy on BYE block at Loose Route Section. I tried this senario without Mediaproxy or RTPproxy with success. regards, Kostas --- K.Marneris(a)otenet.gr ----- Original Message ----- From: "Fabio Macchi" <f.macchi(a)keeptelecom.com> To: <greger(a)teigre.com> Cc: "'Kostas Marneris'" <K.Marneris(a)otenet.gr>; <serusers(a)lists.iptel.org> Sent: 20 March 2007 19:56 Subject: R: R: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem Hi Greger, I attached an ethereal SIP call trace of a test call ( summary and detailed, I simple maskerade final ip numbers ): below only the INVITE relayed from proxy to gateway: Session Initiation Protocol Request-Line: INVITE sip:9999001234@194.244.gatewayIP:5060 SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Record-Route: <sip:194.244.Proxy__IP;ftag=12e1e2e19d527792;lr=on> Via: SIP/2.0/UDP 194.244.Proxy__IP;branch=z9hG4bKb24f.5133a2c4.0 Transport: UDP Sent-by Address: 194.244.Proxy__IP Branch: z9hG4bKb24f.5133a2c4.0 Via: SIP/2.0/UDP 1.255.ua_priv__IP;rport=1176;received=213.156.ua_pub_IP;branch=z9hG4bK35ca7a df63e3094f Transport: UDP Sent-by Address: 1.255.ua_priv__IP RPort: 1176 Received: 213.156.ua_pub_IP Branch: z9hG4bK35ca7adf63e3094f From: "000003" <sip:000003@194.244.Proxy__IP>;tag=12e1e2e19d527792 SIP Display info: "000003" SIP from address: sip:000003@194.244.Proxy__IP SIP tag: 12e1e2e19d527792 To: <sip:9999001234@194.244.Proxy__IP> SIP to address: sip:9999001234@194.244.Proxy__IP Contact: <sip:000003@213.156.ua_pub_IP:1176> Contact Binding: <sip:000003@213.156.ua_pub_IP:1176> URI: <sip:000003@213.156.ua_pub_IP:1176> SIP contact address: sip:000003@213.156.ua_pub_IP:1176 Supported: replaces Call-ID: 953e8996cfcc4ccc(a)1.255.ua_priv__IP CSeq: 60577 INVITE Sequence Number: 60577 Method: INVITE User-Agent: Grandstream HT386 1.0.3.64 FXS0 Max-Forwards: 16 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 327 Message body As you can see, contact informations are correctly fixed with the pubblic UA address, but when the callee hungs up, the BYE is relayed to the private IP address: am I missing something ? In this invite I see UA private address only in VIA: does BYE look to this parameter ? Later caller hangs up too, and the OK is relayed to the correct IP/port. Any help would be high appreciate, thanks in advance. Fabio _____ Da: Greger V. Teigre [mailto:greger@teigre.com] Inviato: martedμ 20 marzo 2007 9.59 A: Fabio Macchi Cc: 'Kostas Marneris'; serusers(a)lists.iptel.org Oggetto: Re: R: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem Normally this happens because you haven't fixed the Contact if the original INVITE or OK. g-) Fabio Macchi wrote: First, thanks for answer. I've tryed your trik and in effect this solve the problem of the '200 ok' forwarded to the UA, but my problem still remain alive: when BYE is sent from Gateway, it reaches correctly SER, but it still forward it to the private UA address. I was wondering about the nat_uac_test in this case, as the source of the BYE message is the gateway ( not natted ) and not the UA. Have any idea about this ? Fabio -----Messaggio originale----- Da: Kostas Marneris [mailto:K.Marneris@otenet.gr] Inviato: giovedμ 15 marzo 2007 20.39 A: Fabio Macchi Cc: serusers(a)lists.iptel.org Oggetto: Re: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem Hello, I was working on about the same problem today either with 'Mediaproxy solution' or with 'SER's Nathelper only solution' . The NAT issue is a nightmare, not because of SER but because of different implementations on NAT boxes. Actually my problem was : if the NATed UA send a BYE to SER, SER forward it to PSTN-GW, then the '200 Ok' Response from PSTN-GW is forwarded by SER to UA to the wrong port (Contact or Via header port). I used the following block on Loose Route section, (because BYE is loose_routed if you use Record-Route), and it seems to work. # --------------------------------------- # Loose Route Section # --------------------------------------- if (loose_route()) { # mark routing logic in request if (method == "BYE") { if (nat_uac_test("22")) { xlog("L_NOTICE", "*** LR -> NATed BYE - Use force_rport()"); force_rport(); }; }; route(1); break; }; I faced up your second problem too. The solution was to move the NAT handling block before proxy_authorize block. I think that the different behaviour does not come with the 'standard RFC1918 addresses', but with the different NAT type. I realize that the provisional mesgs '100 Trying' and '407 Proxy Authentication Required' are relayed back to the real IP addr of NATed UA (this is correct), but to the WRONG port (that of Contact/Via header and not the signalling received port). It seems that these mesgs use the IP address part of 'Received' field of Location DB but not the port. It happens to work if NAT box use the SAME port (eg. 5060) on NAT translation (10.10.10.1:5060 --> Real_IP:5060) (eg. with a SAGEM1500 Router) But it does not work if NAT box doesn't use the same port (10.10.10.1:5060 --> Real_IP:38181) I think that this has to be verified by SER developers or SER experts. Kostas --- K.Marneris(a)otenet.gr ----- Original Message ----- From: "Fabio Macchi" <mailto:f.macchi@keeptelecom.com> <f.macchi(a)keeptelecom.com> To: <mailto:serusers@lists.iptel.org> <serusers(a)lists.iptel.org> Sent: 15 March 2007 19:34 Subject: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem Hi all, I'm running the following schema: UA ( possibly natted ) -> SER -> PSTN Gateway I have a problem with UA belonging to a particular network with private address not RFC1918 compliant ( class 1.x.x.x ), SER and PSTN Gateway have pubblic address. The problem is that, after a succesfull call, if the PSTN gateway send a BYE to SER, then SER forward BYE to the private address of UA instead of pubblic one. I don't understand which is the section that handle BYE messages and how can I solve this problem: anyone may help ? Second, another question: with this particular network I had problem with INVITE too, because SER was sending "proxy authorization request" to the wrong TCP port. To solve this, I've moved the nat handling ( with force_rport ) before the proxy_authorize block and it's working, but why this is not necessary on standard RFC1918 compliant natted address ? Many thanks for any explanation Fabio _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------------------- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.15/728 - Release Date: 20/03/2007 08:07

17 years, 3 months

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sr-users March 2007