- sr-users - kamailio.org

Limit call duration
by Hans Goossens 07 Mar '11

07 Mar '11

Hi, I'm wondering if there is a easy way to limit the call duration in Kamailio ? I would like to automaticaly end the call when the max. duration (for example 12 hours) is reached. In the feature overview of Kamailio 1.5 I found: "option to send BYE when dialog expires" but i'm can't figure out how to get this working (if it's still implemented at all in the current version). Thanks in advance, Kind regards Hans

2 2

about syn_branch
by Juha Heinanen 07 Mar '11

07 Mar '11

we noticed that if sr 3.1 config does not contain syn_branch=0 then acks to 200 oks that are t_relayed by sr, contain branch=0 param in topmost via. in kamailio 1.5, via branch param has proper value even when tm param syn_branch is not set. a couple of questions: 1) according to core cookbook, syn_branch core param only affects stateless forwarding. in my config, all acks are t_relayed, i.e., syn_branch param should not have any effect, but is does. how is that possible? is there a bug in core cookbook? 2) should t_relaying of acks to 200 oks use proper (i.e.. non 0) via branch param even when syn_branch is not set to any value in config? if not why? if yes, will branch=0 be used only when ack to 200 ok does not match any invite transaction? -- juha

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How to access additional xavp's
by Alex Hermann 07 Mar '11

07 Mar '11

Hello, I'm currently toying with xavp's and have some trouble accessing the values. I want to have access to the xavp that isn't the last added one. From the wiki page on http://sip-router.org/wiki/devel/xavp I got the impression that indices are supported, but that doesn't seem to work. In the following fragment i want access to the values 1A & 1B, how to do that? $xavp(test=>a) = "1A"; $xavp(test[0]=>b) = "1B"; $xavp(test=>a) = "2A"; $xavp(test[0]=>b) = "2B"; xlog("$xavp(test[0]=>a)"); xlog("$xavp(test[0]=>b)"); xlog("$xavp(test[1]=>a)"); xlog("$xavp(test[1]=>b)"); pv_xavp_print(); Which gives the following result: Dec 21 14:30:35 veyron wsproxy1[10398]: ERROR: <script>: 2A Dec 21 14:30:35 veyron wsproxy1[10398]: ERROR: <script>: 2B Dec 21 14:30:35 veyron wsproxy1[10398]: ERROR: <script>: <null> Dec 21 14:30:35 veyron wsproxy1[10398]: ERROR: <script>: <null> Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:470]: +++++ XAVP list: 0xb3a38770 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: test Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 2063405720 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 6 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:496]: XAVP value: <xavp:0xb3a387c4> Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:470]: +++++ XAVP list: 0xb3a387c4 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: b Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 110 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 2 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:484]: XAVP value: 2B Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: a Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 109 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 2 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:484]: XAVP value: 2A Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:505]: ----- XAVP list Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: test Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 2063405720 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 6 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:496]: XAVP value: <xavp:0xb3a386c8> Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:470]: +++++ XAVP list: 0xb3a386c8 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: b Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 110 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 2 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:484]: XAVP value: 1B Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:473]: *** XAVP name: a Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:474]: XAVP id: 109 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:475]: XAVP value type: 2 Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:484]: XAVP value: 1A Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:505]: ----- XAVP list Dec 21 14:30:35 veyron wsproxy1[10398]: INFO: <core> [xavp.c:505]: ----- XAVP list -- Greetings, Alex Hermann

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Balancing Asterisk
by Andy Lippitt 07 Mar '11

07 Mar '11

Hello all, I've read as many of the asterisk balancing threads as I can find. Either my situation is unusual or I simply haven't understood anything I've read. In short, I'm building an web/phone mashup which uses Asterisk's AGI to get its work done. My only users are on the PSTN connected to Asterisk through a SIP trunk provider. So presently, in and out through the same trunk, apps live on the single Asterisk box. My goal is scaling and failover. I don't have any need for cross talk or transfers between the asterisk instances, and the algo's in dispatcher seem fine. It seems to me that I should be setting the sip-router up a replacement for the existing peer in Asterisk. What leaves me scratching my head is how I then register the sip-router with the upstream provider. Alternatively, if I use the sip-router as an outboundproxy from asterisk (which seems like it's going to take some hacking to make this work in 1.4), doesn't this now mean I have multiple UAC's trying to register for the same name? Can someone set me on the right track? Thanks, Andy Lippitt

2 1

Newly acquired SIP fails authorization from softphone
by Larry Baumbach 06 Mar '11

06 Mar '11

I am a newbie to the world of VIOP. I am attempting to set up an ATA with SIP. I created a SIP at account iptel.com and received an email confirmation stating "We are reserving the following SIP address for you: sip:larry.baumbach@iptel.org". I tried to testing this address in Xlite but got messages saying: "Account failed to enable. Account Iptel could not be enabled. Verify your user ID, password and authorization name. When I set up the SIP account in Xlite I used: UserID: larry.baumbach(a)iptel.org ( I also tried "larry.baumbach" & "sip:larry.baumbach@iptel.org") Domain: sip.iptel.org Password: my password Authorization name: (I left blank as I did not receive any) I can log into SERweb with the same UserID and Password and access my account. What am I doing wrong? Or is there some kind of wait time before my SIP address is activated? I have spent too much time trying to get this to work on my own. Thanks very much for any help you can provide. -- Larry Baumbach 920.358.8695 larry.baumbach(a)gmail.com

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Re: [SR-Users] [sr-dev] planning release of v3.1.2
by Bernhard Suttner 06 Mar '11

06 Mar '11

Hi, whats about: http://sip-router.org/tracker/index.php?do=details&task_id=108 I think that some users are trying to register to an asterisk system. Or does somebody have a solution for that? A new release would be nice!!! Thanks in advance. Best regards, Bernhard ----- Original Message ----- From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] To: kamailio [mailto:sr-users@lists.sip-router.org], sr-dev [mailto:sr-dev@lists.sip-router.org] Sent: Fri, 28 Jan 2011 20:15:50 +0100 Subject: [sr-dev] planning release of v3.1.2 > Hello, > > I think it is time to release v3.1.2, first date that comes in my mind > is next Thursday if everyone feels it is enough time to take care of > backporting any fix he/she did and it is not yet there. That will > provide us a fresh release for the FOSDEM event. If not, then maybe the > other week, Tuesday, so the participants at the Kamailio Devel training > in Barcelona can practice on it. > > Soon after we should plan also a release for previous stable, branch 3.0. > > Anyone having other options? > > Thanks, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > > _______________________________________________ > sr-dev mailing list > sr-dev(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >

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Kamailio auth_radius: duplicate User-Name attribute
by Kosilov Fedor 06 Mar '11

06 Mar '11

Hello List! I'm trying to set up authorization with our billing proprietary radius server, using Freeradius as a proxy. Currently I'm experiencing the following problem: The Access-Request packet, sent by Kamailio, contains two User-Name attribute records Here is a log from the Freeradius server: rad_recv: Access-Request packet from host 127.0.0.1 port 59294, id=112, length=298 User-Name = "2219001(a)example.com" Digest-Attributes = 0x0a0932323139303031 Digest-Attributes = 0x01106c696e6b2d726567696f6e2e7275 Digest-Attributes = 0x022254584452634531773045524b7368796f30684a70544f4f6a69424d386b32534a Digest-Attributes = 0x04147369703a6c696e6b2d726567696f6e2e7275 Digest-Attributes = 0x030a5245474953544552 Digest-Attributes = 0x050661757468 Digest-Attributes = 0x090a3030303030303031 Digest-Attributes = 0x080c32383034636535373032 Digest-Response = "e79b47955c02401fe52d05f7956609aa" Service-Type = Sip-Session Sip-Uri-User = "2219001" * User-Name = "call-id=domcmqmnychbwlp@koffe-work"* NAS-Identifier = "kamserv.example.com" NAS-Port = 5060 NAS-IP-Address = 127.0.0.1 # Executing section authorize from file /etc/freeradius/sites-enabled/default +- entering group authorize {...} ++[preprocess] returns ok ++[chap] returns noop ++[mschap] returns noop [digest] Checking for correctly formatted Digest-Attributes [digest] Digest-Attributes look OK. Converting them to something more usful. Digest-User-Name = "2219001" Digest-Realm = "example.com" Digest-Nonce = "TXDRcE1w0ERKshyo0hJpTOOjiBM8k2SJ" Digest-URI = "sip:example.com" Digest-Method = "REGISTER" Digest-QOP = "auth" Digest-Nonce-Count = "00000001" Digest-CNonce = "2804ce5702" [digest] Adding Auth-Type = DIGEST ++[digest] returns ok [suffix] Looking up realm "example.com" for User-Name = "2219001(a)example.com " [suffix] Found realm "example.com" [suffix] Adding Realm = "example.com" [suffix] Proxying request from user 2219001 to realm example.com [suffix] Preparing to proxy authentication request to realm "example.com" ++[suffix] returns updated [eap] No EAP-Message, not doing EAP ++[eap] returns noop ++[files] returns noop ++[expiration] returns noop ++[logintime] returns noop ++[pap] returns noop Sending Access-Request of id 250 to 127.0.0.1 port 1822 User-Name = "2219001(a)example.com" Digest-Attributes = 0x0a0932323139303031 Digest-Attributes = 0x01106c696e6b2d726567696f6e2e7275 Digest-Attributes = 0x022254584452634531773045524b7368796f30684a70544f4f6a69424d386b32534a Digest-Attributes = 0x04147369703a6c696e6b2d726567696f6e2e7275 Digest-Attributes = 0x030a5245474953544552 Digest-Attributes = 0x050661757468 Digest-Attributes = 0x090a3030303030303031 Digest-Attributes = 0x080c32383034636535373032 Digest-Response = "e79b47955c02401fe52d05f7956609aa" Service-Type = Sip-Session Sip-Uri-User = "2219001" * User-Name = "call-id=domcmqmnychbwlp@koffe-work"* NAS-Identifier = "kamserv.example.com" NAS-Port = 5060 NAS-IP-Address = 127.0.0.1 Proxy-State = 0x313132 Proxying request 1 to home server 127.0.0.1 port 1822 As I understand, this second User-Name attribute has to be a call-id attribute.

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Calls are killed during kamailio restart
by Efelin Novak 04 Mar '11

04 Mar '11

Hi, I'd like to ask whether my situation is normal. During kamailio restart calls are dropped from mediaproxy. Those two programs are connected using kamailio.sock. Why RTP strem is dropped when SIP proxy is restarted? I would expect just undelivered BYE or something. Thanks for answer, Efelin

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WARNING: <core> [db_id.c:281]: identical DB URLs, but different DB connection pid [30123/30107]
by Klaus Darilion 04 Mar '11

04 Mar '11

Any idea what does it mean? Where does it come from? Do I have to change my config? How? WARNING: <core> [db_id.c:281]: identical DB URLs, but different DB connection pid [30123/30107] Thanks Klaus

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re-SUBSCRIBE and Record-Route
by marius zbihlei 04 Mar '11

04 Mar '11

Hello list, An UAC sends a resubscribe in the form SUBSCRIBE sip:Alice@example.com SIP/2.0 Route: <sip:sip.example.com;lr=on> Route:<sip:sip.next-hop.example.com;lr=on> .... The request has a to tag, from tag... Basic resubscribe(R-URI is the same as original SUBSCRIBE) The proxy that handles it is a Kamailio (1.5 but this refers to 3.1 as well). The problem is that loose_route() treats this request as coming from a strict routing router, not from the UAC, so it applies the strict-to-loose rules and rewrites the request as: SUBSCRIBE sip:sip.next-hop.example.com;lr=on SIP/2.0 Route: <sip:sip.example.com;lr=on> .... The segment that handles this is pretty standard: if (loose_route()) { # sequential request withing a dialog should take the path # determined by record-routing if(has_totag()) { forward(); exit; } It sees that message has Route set, it finds the to tag (so the routes were not a preset), and then , the important thing, sees that the domain from the RURI matches itself (kamailio has aliases configured for example.com), thus considering the request as coming from a strict router. Then it takes the bottom Route header and overwrites the RURI. What is the best way to ensure that loose routing is applied to the resubscribe as described above? Cheers, Marius

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Re: [SR-Users] set_dlg_profile() in failure_route
by Timo Reimann 04 Mar '11

04 Mar '11

Hey 侯旭光, please keep SR-Users in CC so others can share in some assistance and benefit from a solution. On 04.03.2011 08:29, 侯旭光 wrote: > I did trace the dialog in the initial REQUEST > > my script: > ... > modparam("dialog","dlg_flag","4") > ... > route{ > > if (is_method("INVITE")){ > dlg_setflag("2"); > dlg_manage(); > ... > if(avp_db_load("$ruri/username","$avp(s:busyforward)")) > { > setflag(26); > t_on_failure(FAIL_ONE); > } > ... > } > failure_route[FAIL_ONE]{ > ... > if(isflagset(26)&&t_check_status("486")) > { > $ru=$avp(s:busyforward); > append_branch(); > dlg_setflag("2"); > avp_delete("$avp(s:busyforward)"); > resetflag("26"); > lookup("location"); > route(RELAY); > } > ... > } > route[RELAY]{ > if(!t_relay()){ > sl_reply_error(); > } > } > #----------------------------------------------------------------------# > > Is anything wrong? You set flag 2 but showed flag 4 to be modparam'ed to the dialog flag. Is that intentional? Anyways, since you are doing dlg_manage() in the INVITE transaction, you should still be fine. Just for the sake of debugging, can you set flag 4 in your 'is_method("INVITE")' block and see if it makes a difference? I have no in-depth experience with on demand dialog tracking as we use the dialog flag exclusively. By the way, I see no calls to set_dlg_profile() in your route logic. Is this really an excerpt of the configuration you are having trouble with? Cheers, --Timo > > 2011/3/3 Timo Reimann <timo.reimann(a)1und1.de>: >> Hi, >> >> >> On 03.03.2011 14:47, 侯旭光 wrote: >>> set_dlg_profile() function carshed in failure_route when I need do the >>> call-forwarding. >>> >>> When in busy-call-forward or no-answer-forward mode ,it'll run into >>> the failure_route.But at the time , the route_type of the dialog is >>> FAILURE_ROUTE and the get_current_dialog() return NULL, >>> >>> then the set_dlg_profile() function won't work. >>> >>> How can I set dialog profile in the busy or no-answer call-forwarding ? >>> >>> CRITIAL dialog [dlg_profile.c]: BUG - dialog not found in non > REQUEST route >>> ERROR dialog [dialog.c] : failed to set profile >> >> I think you did not track the dialog in the first place, i.e., when the >> initial request associated to the dialog you seek to profile was > processed. >> >> Are you using dlg_flag, and if so, have you enabled it in REQUEST_ROUTE? >> If you want to track dialogs on demand instead, try calling dlg_manage() >> prior to calling set_dlg_profile(). I'm not sure if this is the most >> elegant approach for your use case but it might at least work. >> >> >> Cheers, >> >> --Timo

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set_dlg_profile() in failure_route
by 侯旭光 04 Mar '11

04 Mar '11

Hello set_dlg_profile() function carshed in failure_route when I need do the call-forwarding. When in busy-call-forward or no-answer-forward mode ,it'll run into the failure_route.But at the time , the route_type of the dialog is FAILURE_ROUTE and the get_current_dialog() return NULL, then the set_dlg_profile() function won't work. How can I set dialog profile in the busy or no-answer call-forwarding ? LOG file CRITIAL dialog [dlg_profile.c]: BUG - dialog not found in non REQUEST route ERROR dialog [dialog.c] : failed to set profile thanks

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Re: [SR-Users] About RPM of Kamailio
by Ovidiu Sas 02 Mar '11

02 Mar '11

Please post your questions to the mailing list. If you want to build an rpm, the location of the spec file is available in the source tree: ./pkg/kamailio/rpm/kamailio.spec.CenOS If you prefer a link: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=pkg/kama… Regards, Ovidiu Sas On Wed, Mar 2, 2011 at 5:56 AM, Suresh Bhandari <bring2sb(a)yahoo.com> wrote: > Hello Ovidiu, > > I searched through the kamailio.org/pub but can't find what you said. > > Can you verify the exact location of package that you've downloaded. > > Thanks. > Suresh > >

1 0

how to combine alias_db_lookup() with lookup()
by x-kamailio＠sidell.org 02 Mar '11

02 Mar '11

I'm trying to use the db_alias module as a way to define "generic" addresses that map to a set of actual phones. For example, I'd like the alias "home(a)foo.bar" to map to "kitchen(a)foo.bar" and "office(a)foo.bar", so that both phones ring when a call comes in to "home". I have set the append_branches param to 1: modparam("alias_db", "append_branches", 1) I also modified the dbaliases database table so that key "alias_idx" isn't unique, thereby allow me to add multiple rows for the same alias. The relevant script section is taken verbatim from 3.1 kamailio.cfg: # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: xlog( "L_WARN", "XXX $ru $fu\n"); t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } When I place a call to an alias, the kamailio debug log shows that alias_db_lookup() is correctly setting the ruri to the first entry found in the table, and using append_branch() to add the others. But only the first matching phone gets an INVITE, not the others. I suspect that the lookup() call is blowing away the branches set up by alias_db_lookup() and replacing them with the single phone that matches the ruri for the first alias entry. Is there a way to get alias_db_lookup() and lookup() to play together, so that the first function can set up a list of branches, and the second function can resolve all of the branches to the actual device locations? -- Mark Sidell Partner Forte, Inc. 919-942-7068 fax 919-969-2844 www.forteinc.com

2 1

Radius authentication
by Pablo Ros 02 Mar '11

02 Mar '11

I am trying to authenticate through radius (info in LDAP database). I am using kamailio 3.1 First of all I would like to clear up an issue: As shown the way to do the authentication is done with: *Code:* if (! radius_www_authorize("uu.net")) { www_challenge("uu.net", "0"); return; } Ok, when I installed openser I did it with kamailio "flavour" so it was using the auth_radius module belonging to it. Does anyone know which would be the correct way to do the challenge to the user cause is was not working at all. The radius client does not even send it. I got stuck some time till I just decided to load the module from ser modules folder and then freeradius server started to get correctly my requests. Actually I can even see it authenticates them correctly. However Openser/Kamailio doesn't seem to see the same and it doesn't saves location. *Code:* route[AUTH] { #!ifdef WITH_AUTH # Primeramente comprobamos si pertenece a nuestro dominio; si no ya no hace falta comprobar credenciales. if (uri==myself) { if (is_method("REGISTER")) { xlog("L_NOTICE","KAM-INFO: r[AUTH] - REGISTER - User info: ($fu):($si)>\n"); if (!radius_www_authorize("i2cat.net")){ route(RADIUS); #www_challenge("i2cat.net","0"); exit; } } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } Before doing the challenge then it just goes throught: *Code:* route[RADIUS] { sl_send_reply("100", "Trying"); append_to_reply("Expires: 600\r\n"); append_to_reply("Min-Expires: 240\r\n"); xlog("L_NOTICE","KAM-INFO: RADIUS AUTHENTICATION - AUTHORIZING USER $fU - <$fu>:<$si>\n"); xlog("L_NOTICE","KAM-INFO: CHALLENGING. - RETCODE-> $rc \n"); www_challenge("i2cat.net", "0"); switch($rc){ case -5: xlog("L_INFO", "-> 500: internal server error"); sl_send_reply("500", "Internal Server Error"); case -4: xlog("L_INFO", "-> 404: credentials not found"); sl_send_reply("404", "Credentials Not Found"); case -3: xlog("L_INFO", "-> 400: bad request - stale nonce"); sl_send_reply("400", "Bad Request"); case -2: xlog("L_INFO", "-> 401: invalid password"); sl_send_reply("401", "Invalid Password"); case -1: xlog("L_INFO", "-> 401: invalid user"); sl_send_reply("401", "Invalid User"); default: xlog("L_INFO", "-> 401: unauthorized"); sl_send_reply("401", "Unauthorized"); } } Buuuuuuuuuuuuut... I got that in the debug of Kamailio: *Code:* 4(31099) DEBUG: auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1 4(31099) DEBUG: auth_radius [sterman.c:271]: radius_authorize_sterman(): Success 4(31099) WARNING: auth_radius [authorize.c:89]: RADIUS server did not send SER-UID attribute in digest authentication reply 4(31099) DEBUG: auth [challenge.c:102]: build_challenge_hf: realm=' i2cat.net' 4(31099) DEBUG: auth [challenge.c:113]: build_challenge_hf: qop='auth' 4(31099) DEBUG: auth [challenge.c:236]: auth: 'WWW-Authenticate: Digest realm="i2cat.net", nonce="TWZJLk1mSAKFVzL0b+dVPzkuyyAnZHQs", qop="auth" I guess it has something to do with this SER-UID attribute and thus something about the dictonary? It is weird seeing that the radius server says 'ok' but then openser is not authenticating it. I need some clues! Thank you!. -- Pablo Ros

2 1

rtpproxy and connection information field
by Dominguez Jover, Ricardo 01 Mar '11

01 Mar '11

Hi all, I'm using rtpproxy 1.2.1 and kamailio. RTPproxy is working fine in almost every case. However I have a problem with some call. When softphone A using sip2sip.info account calls softphone B using my Kamailio server account, the Kamilio receives SIP packets from IP1 (81.23.228.129). The invite packet has IP2 (81.23.228.150) in the conecction information field (c=IN IP4 81.23.228.150): INVITE sip:12345@xxxxxxxx.xxx;transport=udp SIP/2.0 Record-Route: <sip:81.23.228.129;lr;ftag=d9e99adf;did=e81.3538c317> Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bKd14b.12802794.0 Via: SIP/2.0/UDP 192.168.xx.xx:7964;received=88.xx.xx.xx;branch=z9hG4bK-d8754z-8ba1b4b47e37bb89-1---d8754z-;rport=7964 Max-Forwards: 69 Contact: <sip:54321@88.xxx.xxx.xxx:7964;transport=udp> To: "2205"<sip:12345@xxxxxxxx.xxx> From: "R sip2sip"<sip:54321@sip2sip.info>;tag=d9e99adf Call-ID: OGE5MTFiMzljZjE0NTYxN2M0N2VkZjgzNDRhMzE2ZTA. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent:xxxxxxxxxxxxxxxxxxxxxxxxx Content-Length: 279 v=0 o=- 12943454020854250 1 IN IP4 192.168.xxx.xx s=xxxxxxxxxxxxxxxxxxxx c=IN IP4 81.23.228.150 t=0 0 m=audio 52854 RTP/AVP 107 0 8 18 101 a=rtpmap:107 BV32/16000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Then Kamilio translates "c=" IP address to my RTPproxy IPaddress and sends the INVITE to softphone B. After the OK softphone B sends RTP packets to the RTPproxy as specified in the "c=" field, however the RTPproxy sends RTP packets to IP1 instead of send packets to IP2. How can I tell the RTPproxy to send packets to IP2 (the one in the c= field?) Thanks, Ricardo Dominguez

1 1

Segfault in Kamailio 1-3-4 to 1.5.5
by Andrew O. Zhukov 01 Mar '11

01 Mar '11

In continue of letters: Kamailio 1.5.5 No TLS Segmentation Fault After upgrade from openser 1.3.4 to kamailio 1.5.5 the same crash set Can someone from developers provide me commercial support to fix this bug in malloc module. If so, contact me directly.

6 12

modulo operator
by Klaus Darilion 01 Mar '11

01 Mar '11

Hi! Using kamailio 3.1.1, I failed to use '%' as described in the core cookbook. Using 'mod' instead seems to work. regards Klaus

2 3

Kamailio Installation
by Suresh Bhandari 01 Mar '11

01 Mar '11

Hello Community, I am new to Kamailio, and this list as well. I am trying to install Kamailio 3.1.2, but I am getting too many errors. I have fixed some but still not getting the way. I am using Red Hat Enterprise Linux (RHEL) 5, and /usr/local directory. When I ran the following command: make group_include="standard standard-dep mysql" include_modules="carrierroute peering" install it prompted not found error for the file docbookx.dtd, I found it (modules/auth/auth.xml, and modules_s/acc_syslog/acc_syslog.xml) and fixed it as it was errorenous URL location. For reference earlier it was http://www.oasis-open.org/docbookid/id/g/4.5/docbookx.dtd, which I changed to http://www.oasis-open.org/docbook/xml/4.5/docbookx.dtd. Now when I run the previous command again, I am getting the follwing errors: *nsgmls:<URL>http://www.oasis-open.org/docbook/xml/4.5/docbookx.dtd:116:17:E: "X20AC" is not a function name* If I ignore this, and continue, I am not able to find the "sip-router" service in /etc/init.d. The entire errors is attached here. Please help me solve the issue. TIA Suresh

2 1

no private memory left after mi mt_reload command
by Javier Gallart 28 Feb '11

28 Feb '11

Hi all we are running kamailio 3.1.1. Basically for every call it has to query a mtree that is reloaded every 15 minutes. The mtree table is about 4M records and is stored in a Pgostgres database. For the reload we use the command mi mt_reload embedded in a xmlrpc request. The code for this is: route[XMLRPC]{ if (method == "POST" || method == "GET") { if (search("^User-Agent:")) { xlog("xmlrpc request\n"); set_reply_close(); set_reply_no_connect(); # optional dispatch_rpc(); } exit; } } This server has been running with no issued for some weeks now, but at some point during the weekend the mi command started to fail. This is what the logs show: ERROR: <core> [db_res.c:181]: no private memory left ERROR: db_postgres [km_res.c:225]: could not allocate rows ERROR: db_postgres [km_dbase.c:280]: failed to convert rows ERROR: mtree [mtree_mod.c:627]: Error while fetching result ERROR: db_postgres [km_dbase.c:318]: invalid parameter value ERROR: mtree [mtree_mod.c:738]: cannot re-load info from database I've restarted kamailio and now it's working fine. Should we increase the private memory? On the other hand, we'd like to detect this failure condition on the script and act accordingly (for instance sending an email with an alarm message), is there anyway to do this? Thanks in advance Regards Javier

2 7

Re: [SR-Users] Can only tranfer (refer) a call one time
by Youngjin Park 28 Feb '11

28 Feb '11

> Hi, > > Can you tell me what advantage on INVITE has against REFER? > > REFER is a kind of blind transfer and INVITE in sipXbridge is 3PCC? > > Thanks in advance. > > Youngjin > > > On Sun, Feb 27, 2011 at 6:40 AM, Grzegorz Stanislawski < > stangrze(a)netitel.pl> wrote: > >> Hi. >> We have proverb for this, i don't know english version but it goes like >> this: >> "When it isn't known what it's all about, it's about money" >> >> Your ITSP had troubles with proper handling second transfer for billing >> purposes so decided to disable it. >> Proxy doesnt participate in call transfer, but ITSP it must charge users >> properly: Alice should pay just for call to Bob, Bob for "his" call to >> Charlie and so on. >> If You are using sipX You should use sipXbridge, it replaces REFER with >> INVITE and bridges calls. >> >> Grzegorz Stanislawski >> >> >> >> W dniu 2011-02-25 11:40, niklas rehnberg pisze: >> >>> Hi, >>> Thank for the quick response. >>> The issue occur only when the Alice is a PSTN client. >>> My ITSP says that they only supporting one call transfer (very strange). >>> They can not explain why etc... >>> PSTN client: Alice >>> MGW/MGC(ITSP): Cisco/SER >>> Our sip server: SIPX >>> >>> BR Niklas >>> 2011/2/25 Iñaki Baz Castillo <ibc(a)aliax.net <mailto:ibc@aliax.net>> >>> >>> 2011/2/25 niklas rehnberg <niklas.rehnberg(a)gmail.com >>> <mailto:niklas.rehnberg@gmail.com>>: >>> > Hi, >>> > Have following issue: >>> > >>> > Alice calling Bob. >>> > Bob make call transfer to Charlie (works fine) >>> > Charlie transfer Alice to David. (the call break) >>> > >>> > Why is not possible to transfer the call more than one time? >>> > Is it any parameters? >>> > >>> > My ITSP use SER together with Cisco MGW. >>> >>> Niklas, nothing in SIP protocol neither in SER/Kamailio makes your >>> scenario to fail. It must be a problem in your custom setup. Try >>> identifying the problem capturing SIP traces. >>> Also take into account that a proxy doesn't participate at all in the >>> process of a "call transfer". It's just a transparent mechanism for a >>> proxy. >>> >>> -- >>> Iñaki Baz Castillo >>> <ibc(a)aliax.net <mailto:ibc@aliax.net>> >>> >>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >

2 1

Re: [SR-Users] kamctl db exec
by Henning Westerholt 28 Feb '11

28 Feb '11

On Saturday 26 February 2011, 侯旭光 wrote: > But I need to use the feature, is it aviliable or not? Hi 侯旭光, ok.. Have you tried if its work with an simple example? kamctl exec "show databases" Cheers, Henning

1 0

UC Expo 2011, London, next week
by Daniel-Constantin Mierla 28 Feb '11

28 Feb '11

Hello, I will be visiting the first day of UC Expo in London (http://www.ucexpo.co.uk/), Tuesday, March 8. If you are going to be there and want to meet for a chat about Kamailio & VoIP, drop me an email. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Re: [SR-Users] Unable to start Siremis
by jack＠4voice.net 28 Feb '11

28 Feb '11

1 0

Re: [SR-Users] Unable to start Siremis
by hb 28 Feb '11

28 Feb '11

Hi jack： I got the same problem with you. I know little about apache and HTML, would you tell me how to configuration apache specific. Thanks a lot. haibo huang

1 0

how to route to new URI on failure
by x-kamailio＠sidell.org 27 Feb '11

27 Feb '11

I'm trying to route failed calls to a voicemail URI. The failure route couldn't be simpler: failure_route[FAIL_ONE] { if (t_is_canceled()) { exit; } if( t_check_status("486|408") { append_branch( "sip:foo@2.2.2.2"); t_relay(); } } But, it doesn't work. For example, let's say the initial INVITE resolves to a local device "me(a)1.1.1.1". This works fine, and the phone rings. After a timeout, the failure_route executes. The branch "foo(a)2.2.2.2" gets appended, and kamailio sends a new INVITE, but instead of determining the correct proxy for the new address, it sends the INVITE, with the new URI, to the device that original received the INVITE, "me(a)1.1.1.1". Obviously, this doesn't work. I've been able to force kamailio to route the call correctly by modifying failure_route[ to use t_relay_to_udp() as follows: failure_route[FAIL_ONE] { if (t_is_canceled()) { exit; } if( t_check_status("486|408") { append_branch( "sip:foo@2.2.2.2"); t_relay_to_udp( "2.2.2.2", "5060"); } } But, it seems like kamailio should figure what where to send the new INVITE itself. What am I doing wrong? -- Mark Sidell Partner Forte, Inc. 919-942-7068 fax 919-969-2844 www.forteinc.com

2 2

Can only tranfer (refer) a call one time
by niklas rehnberg 27 Feb '11

27 Feb '11

Hi, Have following issue: Alice calling Bob. Bob make call transfer to Charlie (works fine) Charlie transfer Alice to David. (the call break) Why is not possible to transfer the call more than one time? Is it any parameters? My ITSP use SER together with Cisco MGW. BR Niklas

4 4

Lcr Command Disabled
by Amit Nepal 25 Feb '11

25 Feb '11

Lcr is working fine but Why would i get LCR commands disabled ? kamctl lcr reload ERROR: command disabled-- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

3 8

Rtpproxy newbie question
by Marc 25 Feb '11

25 Feb '11

Hi, I installed a kamailio from the debian repository, i use the default configuration. The only enable the tcp and tls transport modes, and configured mysql login. Everyting seems to be working fine. I would need now to configure rtprpoxy, but i just cant make it work. My goal is to force rtpproxy for all calls. Any help is highly appreciated!! Thank you Marc Rtpproxy is installed and seems to be running: This is my rtpproxy configuration: ------------ # Defaults for rtpproxy # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: CONTROL_SOCK=udp:127.0.0.1:22222 LISTEN_ADDR=[external_ip] # Additional options that are passed to the daemon. EXTRA_OPTS="-l ${LISTEN_ADDR}" /usr/sbin# ------------- It seems to be running: ----------- /etc/init.d# ./rtpproxy start Starting RTP relay: rtpproxy. /etc/init.d# netstat -an|grep 22222 udp 0 0 127.0.0.1:22222 0.0.0.0:* /etc/init.d# ----------------- Here is my kamailio configuration, default as is installed: ------------------------------------------- #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.1 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define ""mysql:[my_mysql_db_config]"" "mysql://openser:openserrw@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=2 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_DAEMON fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ disable_tcp=no /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:[external_ip]:5060 listen=tcp:[external_ip]:5062 listen=tls:[external_ip]:5061 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 port=5062 tls_port_no=5061 enable_tls=yes #!ifdef WITH_TLS enable_tls=yes #!endif ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting #pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## # set paths to location of modules #!ifdef LOCAL_TEST_RUN mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "tls.so" modparam("tls", "config", "/etc/kamailio/tls.cfg") #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", ""[my_mysql_db_config]"") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", "[my_mysql_db_config]") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", ""[my_mysql_db_config]"") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", "[my_mysql_db_config]") modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", "[my_mysql_db_config]") modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", "[my_mysql_db_config]") # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", "[my_mysql_db_config]") # ----- presence_xml params ----- modparam("presence_xml", "db_url", "[my_mysql_db_config]") modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@[mydomain]") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") modparam("tls", "tls_debug", 10) #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); } # Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} }

2 2

postgres in Kamailio 3.1.2
by Bruno Bresciani 24 Feb '11

24 Feb '11

Hi, I configure the Kamailio 3.1.2 with postgres but I cann't start. In the log file is generated the following error: ** ERROR: load_module: could not open module </home2/local/kamailio/lib/kamailio/modules/db_ postgres.so>: /home2/local/kamailio/lib/kamailio/modules/db_postgres.so: undefined symbol: PQdescribePrepared Why this error is generated? Best Regards

2 1

tcp_persistent_flag
by Bruno Bresciani 24 Feb '11

24 Feb '11

Hi, there is in kamailio 3.1.2 some exported parameter that work as tcp_persistent_flag paramenter? I want to keep alive the connections TCP/TLS as long as the register contact is valid. Best Regards

2 1

dialplan regex confusion
by thrillerbee 24 Feb '11

24 Feb '11

I'm using the following in dialplan: id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs ----+------+----+----------+----------------------------------------+-----------+----------------------------------------------+----------+---------------------------------- 3 | 3 | 0 | 1 | ^30\d{3}#\+?1?[2-9]\d{2}[2-9]\d{6}$ | 0 | ^(30\d{3}#)(\+?1?)([2-9]\d{2}[2-9]\d{6}$) | \11\3 | normalize prefixed domestic A R-URI goes in as 30001#12345678910 It comes out as 30001#*\1*12345678910 Can someone help me understand why? I'd expect to be exactly what it was originally but for some reason '\1' is thrown in the middle... Thanks, Ryan

2 2

pipelimit db schema
by thrillerbee 24 Feb '11

24 Feb '11

Can anyone point me to the db schema for the new pipelimit module? Thanks, Ryan

3 6

kamctl db exec
by 侯旭光 24 Feb '11

24 Feb '11

Hello How to use "kamctl db exec <query> " function? My SQL_ENGINE is MySQL How to write the command line when I want to do the query "select * from acc" with kamctl db tool? thanks

2 1

NAT Traversal
by Spinov Evgeniy 24 Feb '11

24 Feb '11

Hello, all. Using nathelper + rtpproxy for subj. Kamailio has public and private network interfaces. Asterisk is only private. RTP Proxy is working in bridge mode and relaying traffic from UAC to Asterisks. Everything is working fine, except one configuration. When the client is behind router ( a specific one, I do not have an access there to check ), and this UAC is making a call to other public extension, which is behind router, then RTP Proxy is relaying traffic to the caller, using another UDP port, then the packets arrive. For instance: UAC 1 -> UAC 2 PUBLIC_IP:10 > KAMAILIO_IP:5555 KAMAILIO_IP:5678 > PUBLIC_IP:12 While for the UAC 2 it looks like: PUBLIC_IP:20 > KAMAILIO_IP:6767 KAMAILIO_IP:4564 > PUBLIC_IP:20 The source and destination UDP ports are the same. As result, I can hear UAC 1 and he cannot hear me. In case of we have UAC 3, which is behind other router, call is working fine with same configuration. "It's routers fault" you can say, but in the same configuration ( I mean network, not kamailio ) it worked, but when RTPProxy was not in bridge mode and Kamailio and Asterisks were in public network. Reinvites are not allowed in both cases. The question is, why the source and destination UDP ports are different? Using STUN in first case, cause without it, private IP written in contacts and as result, traffic relayed from Kamailio is incorrect, cause heading to private network which is unreachable. Any ideas where to dig?

1 0

Rtpproxy newbie question
by Marc 24 Feb '11

24 Feb '11

Hi, I installed a kamailio from the debian repository, i use the default configuration. The only enable the tcp and tls transport modes, and configured mysql login. Everyting seems to be working fine. I would need now to configure rtprpoxy, but i just cant make it work. My goal is to force rtpproxy for all calls. Any help is highly appreciated!! Thank you Marc Rtpproxy is installed and seems to be running: This is my rtpproxy configuration: ------------ # Defaults for rtpproxy # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: CONTROL_SOCK=udp:127.0.0.1:22222 LISTEN_ADDR=[external_ip] # Additional options that are passed to the daemon. EXTRA_OPTS="-l ${LISTEN_ADDR}" /usr/sbin# ------------- It seems to be running: ----------- /etc/init.d# ./rtpproxy start Starting RTP relay: rtpproxy. /etc/init.d# netstat -an|grep 22222 udp 0 0 127.0.0.1:22222 0.0.0.0:* /etc/init.d# ----------------- Here is my kamailio configuration, default as is installed: ------------------------------------------- #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.1 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define ""mysql:[my_mysql_db_config]"" "mysql://openser:openserrw@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=2 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_DAEMON fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ disable_tcp=no /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:[external_ip]:5060 listen=tcp:[external_ip]:5062 listen=tls:[external_ip]:5061 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 port=5062 tls_port_no=5061 enable_tls=yes #!ifdef WITH_TLS enable_tls=yes #!endif ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting #pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## # set paths to location of modules #!ifdef LOCAL_TEST_RUN mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "tls.so" modparam("tls", "config", "/etc/kamailio/tls.cfg") #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", ""[my_mysql_db_config]"") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", "[my_mysql_db_config]") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", ""[my_mysql_db_config]"") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", "[my_mysql_db_config]") modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", "[my_mysql_db_config]") modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", "[my_mysql_db_config]") # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", "[my_mysql_db_config]") # ----- presence_xml params ----- modparam("presence_xml", "db_url", "[my_mysql_db_config]") modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@[mydomain]") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") modparam("tls", "tls_debug", 10) #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); } # Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} }

1 0

403 "Not Relaying" problem
by yan wang 24 Feb '11

24 Feb '11

Dear Friends, With Kamailio 3.1.2, after successfully registering, the caller sends INVITE to the proxy, and is always responded with 403 not relaying. I found that in the route[AUTH] subroutine, like below. Regarding the red comparing, it failed. I also check the code, and found Kamailio will use the host name in the From header to compare all the socket's IP address, such as host = mysipserver.com, but socket's ip = 127.0.0.1 or 10.1.1.5. As you can see, this comparing will never be equal.Could anyone show me how to handle this problem? Thanks a lot! # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } Best Regards,Spencer

1 0

how to forward on busy
by x-kamailio＠sidell.org 23 Feb '11

23 Feb '11

Is there an example somewhere that shows how to do forward-on-busy, starting with the kamailio.cfg from version 3.1? Ultimately, I want to use a per-user AVP to obtain the forwarding URI, but just an example that shows how to forward to a fixed URI from the failure route would be great. -- Mark Sidell Partner Forte, Inc. 919-942-7068 fax 919-969-2844 www.forteinc.com

1 0

Re: [SR-Users] Visiting CeBIT 2011 in Hanover (Carsten Bock)
by Dan-Cristian Bogos 23 Feb '11

23 Feb '11

Daniel, I will be there on Thursday/Friday. Cheers. DanB

1 0

422 Session timer too small
by Mino Haluz 23 Feb '11

23 Feb '11

Hi, I see in my cdrtool rarely 422 Session Timer too small - the reason why the call was not initiated. Do you have some suggestion how to fix this ? Maybe SST module ? What is the default value of this timer, so that I could increase it ? kamailio 3.1.0 (i386/linux) 1e204f Thank you, Mino

3 2

kamailio 3.1: forward("10.0.0.10:5060") not working?
by Min Wang 23 Feb '11

23 Feb '11

HI It seems the forward("host:port") not working in the kamailio 3.1? the simple route like: route { forward("67.154.xx.xx:5080"); } the error is: <core> [proxy.c:278]: ERROR: mk_proxy: could not resolve hostname: "67.154.xx.xx:5080" Anything missing? the detailed log is: 0(8500) DEBUG: <core> [sr_module.c:625]: find_export_record: found <load_tm> in module tm [/usr/lib/kamailio/modules/tm.so] 0(8500) DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_newtran> in module tm [/usr/lib/kamailio/modules/tm.so] 0(8500) DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_relay_to_tcp> in module tm [/usr/lib/kamailio/modules/tm.so] 0(8500) DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_relay_to_udp> in module tm [/usr/lib/kamailio/modules/tm.so] 0(8500) DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_relay> in module tm [/usr/lib/kamailio/modules/tm.so] 0(8500) DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_forward_nonack> in module tm [/usr/lib/kamailio/modules/tm.so] 0(8500) DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_release> in module tm [/usr/lib/kamailio/modules/tm.so] 0(8500) DEBUG: <core> [main.c:2371]: Expect (at least) 18 SER processes in your process list 0(8500) DEBUG: <core> [proxy.c:278]: DEBUG: mk_proxy: doing DNS lookup... 0(8500) DEBUG: <core> [dns_cache.c:567]: dns_hash_find(_sip._udp.67.154.xx.xx:5080(28), 33), h=780 0(8500) DEBUG: <core> [resolve.c:727]: get_record: lookup(_sip._udp.67.154.xx.xx.:5080, 33) failed 0(8500) DEBUG: <core> [dns_cache.c:895]: dns_cache_mk_bad_entry(_sip._udp.67.154.xx.xx:5080, 33, 60, 1) 0(8500) DEBUG: <core> [dns_cache.c:828]: dns_cache_add: adding _sip._udp.67.154.xx.xx:5080(28) 33 (flags=1) at 780 0(8500) DEBUG: <core> [dns_cache.c:567]: dns_hash_find(67.154.xx.xx:5080(18), 1), h=16 0(8500) DEBUG: <core> [resolve.c:727]: get_record: lookup(67.154.xx.xx:5080, 1) failed 0(8500) DEBUG: <core> [dns_cache.c:895]: dns_cache_mk_bad_entry(67.154.xx.xx:5080, 1, 60, 1) 0(8500) DEBUG: <core> [dns_cache.c:828]: dns_cache_add: adding 67.154.xx.xx:5080(18) 1 (flags=1) at 16 0(8500) : <core> [proxy.c:278]: ERROR: mk_proxy: could not resolve hostname: "67.154.xx.xx:5080" 0(8500) ERROR: <core> [route.c:1161]: fixing failed (code=-478) at cfg:/etc/kamailio/kamailio.cfg:394 ERROR: error -478 while trying to fix configuration 0(8500) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start 0(8500) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table 0(8500) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing semaphores 0(8500) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb lists 0(8500) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done 0(8500) DEBUG: <core> [mem/shm_mem.c:236]: shm_mem_destroy 0(8500) DEBUG: <core> [mem/shm_mem.c:239]: destroying the shared memory lock thx. -- Kind regards / Mit freundlichen Grüßen Min Wang

2 2

Re: [SR-Users] How to get the dialog information via dlg_list command?
by yan wang 23 Feb '11

23 Feb '11

Hi Daniel, Thanks a lot for your quick reply. I recompile and install the Kamailio as the following page described.http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-… I also successfully created two SIP users: "User1" and "User2". Now the registeration is passed for the two users. But when "User1" sends INVITE to "User2" through the Kamailio proxy, it always response with 403 "Not relaying" and not forward the INVITE to "User2". Could you please give me some help on this case? Thanks. BTW, I have export the environment variable SIP_DOMAIN to "mysipserver.com". And please check the attached enclosed config file and trace. Best Regards,Spencer --- On Wed, 2/23/11, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] How to get the dialog information via dlg_list command? To: "yan wang" <rtosvip(a)yahoo.com.cn> Cc: sr-users(a)lists.sip-router.org Date: Wednesday, February 23, 2011, 2:42 PM Hello, the problem is you are using the dialog module coming from SER, not the one coming from Kamailio. The loadpath (or mpath) has to include directorins ending in modules_k and modules. I recommend you use kamailio.cfg as a start to build your config and then add the dialog module. Before compiling, when you are using the sources, you have to do: make FLAVOUR=kamailio cfg You can see more at: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git If you prefer to install deb files (on Debian/Ubuntu), see: http://www.kamailio.org/dokuwiki/doku.php/packages:debs Cheers, Daniel On 2/23/11 7:32 AM, yan wang wrote: Dear Friends, I need your help on the following question: I am using Kamailio 3.1.2. I want to get the dialog information dynamically via the FIFO command "dlg_list" by "kamctl fifo dlg_list". But unfortunately, I could get NOTHING in the output. And the fact is that I have setup 5 pairs SIP calls with the Kamailio SIP proxy. Could anyone show me the clues and how to get the correct dialog information? Thanks. BTW, I attached the config file. Best Regards, Spencer _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Re: [SR-Users] get_profile_size() function help (Alex Balashov)
by 侯旭光 23 Feb '11

23 Feb '11

On 02/21/2011 04:50 AM, ??? wrote: > ALTER:<core> [main.c 722] : child process 29651 exited by sinal 11 > ALTER:<core> [main.c 725] : core was generated > INFO :<core> [main.c 737] : terminating due to SIGCHILD This is a crash. It's a bug. Then how to get the number of dialogs belonging to a profile?

3 3

How to get the dialog information via dlg_list command?
by yan wang 23 Feb '11

23 Feb '11

Dear Friends, I need your help on the following question: I am using Kamailio 3.1.2. I want to get the dialog information dynamically via the FIFO command "dlg_list" by "kamctl fifo dlg_list". But unfortunately, I could get NOTHING in the output. And the fact is that I have setup 5 pairs SIP calls with the Kamailio SIP proxy. Could anyone show me the clues and how to get the correct dialog information? Thanks. BTW, I attached the config file. Best Regards,Spencer

2 1

Re: [SR-Users] Unable to start Siremis
by jack＠4voice.net 22 Feb '11

22 Feb '11

1 0

Unable to start Siremis
by jack＠4voice.net 22 Feb '11

22 Feb '11

2 1

UAC Module: test registration to a Voip provider
by Matteo Campana 22 Feb '11

22 Feb '11

Hi all, we are using the UAC module ( http://www.kamailio.org/docs/modules/stable/modules_k/uac.html#id2910015) to register the proxy to an external DID provider. I know that the module takes care of sending REGISTER on the basis of credentials stored in uacreg table, but my question is: if I add a new row in the uacreg table, kamailio will register the new username after the database update or I need a restart of kamailio (or some kamailio module)? If I edit the row in uacreg table and I call the rpc command *sercmd uac.reg_dump *I see the old values in the output, but if I restart kamailio I see the new values. Tanks in advance, Matteo

2 1

how to forward on busy
by x-kamailio＠sidell.org 22 Feb '11

22 Feb '11

I'm trying to do something very basic, which is to forward a call to a different number on busy. But, the various things I've tried from the examples don't work. Starting with the kamailio.cfg that is included with version 3.1, what code would I add to forward all busy calls to "sip:foo@bar.com", a target that is not necessarily local to the server? Many thanks! -- Mark Sidell Partner Forte, Inc. 919-942-7068 fax 919-969-2844 www.forteinc.com

2 1

Kamailio/SIP Router in RHEL 5
by Suresh Bhandari 22 Feb '11

22 Feb '11

Hi, I wanted to run Kamailio/SIP Server in RHEL 5 (Being an RHCE, I prefer this :)), but the INSTALL document you provided has no description based on that OS :(. I found that there are no RPMs for this as well. Are you preparing them? or Do you have any plans for that? or you prefer me to change the OS. Thanks and Regards, Suresh.

3 2

Visiting CeBIT 2011 in Hanover
by Daniel-Constantin Mierla 22 Feb '11

22 Feb '11

Hello, I am going to visit CeBIT show this year (first week of March in Hanover, Germany), if anyone else here is there and want to meet for a chat about latest developments in the project and VoIP world, drop me an email, maybe we have some overlapping days and can sit together for a bit. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

3 2

HTABLE problem
by Ricardo Martinez 21 Feb '11

21 Feb '11

Hello I’m trying to use “htable” module with the example to block a user with three time s failed password. When a REGISTER arrives I’m getting this error : Feb 18 17:26:34 /usr/local/sbin/kamailio[5840]: ERROR: <core> [lvalue.c:358]: setting pvar failed Feb 18 17:26:34 /usr/local/sbin/kamailio[5840]: ERROR: <core> [lvalue.c:411]: assignment failed at pos: (322,54-322,81) Feb 18 17:26:34 /usr/local/sbin/kamailio[5840]: ERROR: <core> [lvalue.c:358]: setting pvar failed Feb 18 17:26:34 /usr/local/sbin/kamailio[5840]: ERROR: <core> [lvalue.c:411]: assignment failed at pos: (328,47-328,49) The lines pointing the error are these : 312 if ( !radius_www_authorize("10.0.0.208") ) 313 { 314 switch ($retcode) { 315 case -1: 316 xlog("L_INFO","authentication failed from radius\n"); 317 if($sht(a=>$au::auth_count) == "null") 318 { 319 $sht(a=>$au::auth_count) = 0; 320 } 321 $sht(a=>$au::auth_count) = $sht(a=>$au::auth_count) + 1; 322 if($sht(a=>$au::auth_count) == 3) 323 { 324 xlog("L_INFO","auth failed 3rd time - src ip: $si\n"); 325 } 326 $sht(a=>$au::last_auth) = $Ts; 327 break; 328 case -5: 329 sl_send_reply("403", "Forbidden"); 330 exit; 331 } 332 www_challenge("10.0.0.208", "1"); 333 exit; 334 } What could be the problem? Thanks in advance. Regards, Ricardo.-

2 3

Kamailio Awards 2010
by Daniel-Constantin Mierla 21 Feb '11

21 Feb '11

Hello, almost a tradition by now, being the 4th edition, I published Kamailio Awards for 2010 - a blog post that tries to summarize the top of the activities in the public space related to Kamailio project that happened during the previous year. You can browse it at: http://asipto.com/u/ka10 Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

AVP Error in version 3.1.0
by Jijo 21 Feb '11

21 Feb '11

Hi All, On register we store the contact in an avp variable and do a t_relay(). After t_relay() the $avp variable becomes null. I printed the value before after t_relay() to determine this behavior. This happens only on registration load test around 2000 subcribers with ( 4 REGISTER/sec). This happens only for one subscriber out of 2000 subscribers. I did the similar test with $var and its working fine. Anybody observed similar behavior with avp? This was working in kamailio 1.4 version. We did the upgrade recently to 3.1.0 and started observing this issue. How do we debug this issue.? Thanks Jijo

2 5

get_profile_size() function help
by 侯旭光 21 Feb '11

21 Feb '11

Hello bother again what's the problem with the function get_profile_size() My script modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "dlg_mode", 1) modparam("dialog", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("dialog", "profiles_no_value", "inbound ; outbound") ... if(is_method("INVITE")) { ... dlg_manage(); set_dlg_profile("inbound"); ... get_profile_size("inbound","$avp(size)"); ... } ============================================ when program runs to the function get_profile_size(), the proxy will be terminate. log file: ALTER: <core> [main.c 722] : child process 29651 exited by sinal 11 ALTER: <core> [main.c 725] : core was generated INFO : <core> [main.c 737] : terminating due to SIGCHILD ============================================ what' s the problem? Did I use the func wrong? thanks

2 1

pcre mystery or bug in dialplan module?
by Juha Heinanen 21 Feb '11

21 Feb '11

i have in dialplan table record 287 14 10 1 ^(.+)\+358 0 ^(.+):\+358(.+) \1:0\2 and when it is applied to uri Feb 21 08:05:15 sip /usr/sbin/sip-proxy[24071]: INFO: uri <sip:+35892345670@test.fi;user=phone> the resulting uri looks like this: Feb 21 08:05:15 sip /usr/sbin/sip-proxy[24071]: INFO: uri <sip\1:092345670@test.fi;user=phone> i can't figure out, where \1 comes to the result, because \1 is has already produced 'sip' to it. i have added debug to dialplan module and looks like it compiles correct subst and repl expressions: Feb 21 08:05:05 sip /usr/sbin/sip-proxy[24067]: INFO: dialplan [dp_db.c:392]: parsing repl_exp <\1:0\2> Feb 21 08:05:05 sip /usr/sbin/sip-proxy[24067]: INFO: dialplan [dp_db.c:404]: compilig subst_exp <^(.+):\+358(.+)> is there some stupid mistake in the expressions or could it be a bug somewhere? -- juha

1 1

decimal fraction problem
by 侯旭光 20 Feb '11

20 Feb '11

Hello I need to add q value while using function append_branch(),but the function only takes decimal fraction as the parameter. What if I want to use pv to add q value? The $var and $avp just have string and integer type. Thanks a lot!

1 0

My new blog http://cciev.ciscovoicetech.com/
by Arun Kumar 20 Feb '11

20 Feb '11

Hi Guys, Soon, I'll be starting a new section related to Asterisk (around 4 years of full time experience with Asterisk, Trixbox, SER, OpenSer, MediaProxy, AGI*) so let me know if you like to see some topic coming. Cheers Arun

1 0

Failure Route and Reply Routes
by Brandon Armstead 19 Feb '11

19 Feb '11

Hello, Is there any specific information / documentation as to what kind of response causes the failure_route to trigger over the reply_route. For example, it seems that 486 response triggers the failure route while as a 408 triggers the reply_route. Perhaps I am over looking something? I would imagine that 5xx and 6xx (final / non provisional responses) would trigger the failure route. While as 4xx would trigger the reply route -- however this does not always seem to be the case in the example of 486 vs 408. Any information / help in advance is greatly appreciated, thanks! Sincerely, Brandon Armstead

3 3

Kamailio NOTIFY
by Spinov Evgeniy 19 Feb '11

19 Feb '11

Hello all. I'm having problem with sending NOTIFY packet for event dialog. The problem is, that all NOTIFY packets sent with this event is empty. ( Content-length: 0 ) and in fact, not passing actual status to the phone. In my case this is SPA962. NOTIFY packets for event presence are sent fine and status is changing normally. The same phone, works fine with Asterisk. Asterisk is sending NOTIFY with non-zero content and changing status for the subscriber as well. I've turned on pua module to send PUBLISH requests at Kamailio, it's filling DB table with subscription, everything runs fine and NOTIFY packets are coming to subscribed phone, but they are empty. I also have presence_dialoginfo module on and pua_dialoginfo module on. I've tried on Kamailio 3.0.3 and 3.1.1. Content always empty. There is no sense of pasting here packet dumps, cause everything is like should be: (UAC) SUBSCRIBE, event: presence ---> K K ---> 202 OK (SPA) SUBSCRIBE, event: dialog ---> K K ---> 202 OK PUBLISH, event: presence ---> K K ---> 200 OK NOTIFY, event: presence, Content-Length: XXX ---> UAC NOTIFY, event: dialog, Content-Length: 0 ---> SPA962 As result, softphone changes status and hardphone ( SPA962 ) - no. May be it's cause of PUBLISH event value? Please advise.

2 10

why function result -1 cannot be assigned to variable?
by Juha Heinanen 19 Feb '11

19 Feb '11

i have a local function x that returns a positive integer or -1. when it returns -1, assignment $var(tmp) = function_x("$rd"); produces to syslog an error: Feb 19 07:32:44 sip /usr/sbin/sip-proxy[27284]: ERROR: <core> [lvalue.c:411]: assignment failed at pos: (1415,40-1415,40) i have had that function for years, but have not noticed these errors in earlier versions than 3.1. any ideas why the error is produced and how to get rid of it? -- juha

1 0

Interfaces guestures
by Spinov Evgeniy 18 Feb '11

18 Feb '11

Hello, all. I'm having Kamailio 3.0.4. Server have 2 physical interfaces, one is private, second is public. INVITE request is coming to Kamailio from public interface and being routed by Kamailio to one of ther asterisk boxes, which is in private network. Packet is routed to correct interface, but SRC address of packet, containing public Kamailio IP. As of that, asterisk is unable to communicate properly with Kamailio. I'm not applying any packet changes in Kamailio config and cannot get why Kamailio is doing that? Kamailio is listening both networks and able to send packets with correct IP, but somewhy sending an invalid one. Am I doing something wrong? Thank you.

2 1

Re: [SR-Users] pdb module timeouts
by Thomas Baumann 18 Feb '11

18 Feb '11

Hello Marius, thanks a lot for submitting the patch. It works like a charm now. But to be honest I had troubles to find this patch, I just checked the daily tarball [http://sip-router.org/tarballs/sr/] and found this change. How I can get patch e914f97b4 in a better way ? regards, Thomas >Hello >I have made a patch to 3.1(e914f97b4) . The poll() event list was not >cleared when a response was received so when 2 servers responded one >after the other it might block in an incorrect recv(). Can you please >test with this? >Marius ___________________________________________________________ Schon gehört? WEB.DE hat einen genialen Phishing-Filter in die Toolbar eingebaut! http://produkte.web.de/go/toolbar

4 4

Re: [SR-Users] ERROR: failover functions used, but AVPs paraamters required are NULL
by Gary Chen 18 Feb '11

18 Feb '11

Never mind. I used wrong kamailio.cfg file. From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Thursday, February 17, 2011 8:51 AM To: Gary Chen Cc: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] ERROR: failover functions used, but AVPs paraamters required are NULL Hello, can you send all log messages printed by: kamailio -E -ddd Thanks, Daniel On 2/17/11 2:42 PM, Gary Chen wrote: kamailio version 3.1.2 I am trying to setup dispatcher to use its failover feature. Here is dispatcher part of configure file: loadmodule "dispatcher.so" modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshhold", 10) modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=4") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_from", "sip:lb1 <at> m-lab-ca805-sig.kd-lab.de") modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "flags", 2) But it is still give the following error: 0(1917) ERROR: dispatcher [dispatcher.c:624]: failover functions used, but AVPs paraamters required are NULL -- feature disabled Does anybody know why? Gary _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com

2 1

Re: [SR-Users] Bad File Descriptor
by Daniel-Constantin Mierla 17 Feb '11

17 Feb '11

Please keep the mailing list cc-ed. My previous email had other comments/questions inline, can you answer them as well? They are relevant in troubleshooting. Thanks, Daniel On 2/17/11 7:57 PM, David J. wrote: > version: kamailio 3.1.1 (x86_64/linux) 88bda8 > flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, > USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, > SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, > FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, > USE_DST_BLACKLIST, HAVE_RESOLV_RES > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4MB > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > id: 88bda8 > compiled on 04:39:27 Dec 9 2010 with gcc 4.3.2 > > > > On 2/17/11 1:46 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> what is the version you are using? >> >> On 2/17/11 7:35 PM, David J. wrote: >>> 5(20390) ERROR: <core> [udp_server.c:586]: ERROR: udp_send: >>> sendto(sock,0x7f67f7f94d14,4,0,69.117.34.101:2048,16): Bad file >>> descriptor(9) >>> 5(20390) ERROR: nathelper [nathelper.c:1722]: udp_send failed >>> >>> >>> I see this error when I try to restart kamailio after crash; >> >> What is the cause for the crash? Do you have a log or core for that? >> >>> I see the cause of this problem is 'stale' entries in the location >>> table; >>> >>> If I delete this entries kamailio starts fine; any suggestions to >>> prevent this from continuously happening. >> Is any change in the IP address of the server upon restart? >> >> Do the errors persist or they stop after a while? They are related to >> NAT keepalives sent by nathelper module, so they don't affect the sip >> traffic and if they are for stale contacts then they are fully >> harmless. The stale contacts should be automatically removed after a >> while by usrloc module. >> >> Cheers, >> Daniel >> > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Bad File Descriptor
by David J. 17 Feb '11

17 Feb '11

5(20390) ERROR: <core> [udp_server.c:586]: ERROR: udp_send: sendto(sock,0x7f67f7f94d14,4,0,69.117.34.101:2048,16): Bad file descriptor(9) 5(20390) ERROR: nathelper [nathelper.c:1722]: udp_send failed I see this error when I try to restart kamailio after crash; I see the cause of this problem is 'stale' entries in the location table; If I delete this entries kamailio starts fine; any suggestions to prevent this from continuously happening. Thanks.

2 1

Kamailio 1.5.5 No TLS Segmentation Fault
by Stagg Shelton 17 Feb '11

17 Feb '11

Hello, We have been having a problem with Kamilio faulting and dumping core files on occasion. I have not been able to reproduce the failure at will, but notice the back trace seems to point toward actions with the dialogue. Below is from a backtrace of a core file from just a few minutes ago. Can anyone determine what may have caused the system to error and stop processing? Thanks Stagg Core was generated by `/sbin/kamailio -m 512'. Program terminated with signal 11, Segmentation fault. #0 0x00007f8a11d55fa7 in unref_dlg (dlg=0x7f89f7e07470, cnt=1) at dlg_hash.c:474 474 d_entry = &(d_table->entries[dlg->h_entry]); Missing separate debuginfos, use: debuginfo-install bzip2-libs-1.0.5-5.fc11.x86_64 db4-4.7.25-11.fc11.x86_64 e2fsprogs-libs-1.41.9-2.fc11.x86_64 elfutils-libelf-0.147-1.fc11.x86_64 glibc-2.10.2-1.x86_64 keyutils-libs-1.2-5.fc11.x86_64 krb5-libs-1.6.3-31.fc11.x86_64 libacl-2.2.49-3.fc11.x86_64 libattr-2.4.43-3.fc11.x86_64 libcap-2.16-4.fc11.1.x86_64 libconfuse-2.6-2.fc11.x86_64 libgcc-4.4.1-2.fc11.x86_64 libselinux-2.0.80-1.fc11.x86_64 lm_sensors-3.1.0-1.fc11.x86_64 lua-5.1.4-3.fc11.x86_64 mysql-libs-5.1.46-1.fc11.x86_64 net-snmp-libs-5.4.2.1-13.fc11.x86_64 nspr-devel-4.8.4-1.3.fc11.x86_64 nss-devel-3.12.6-1.2.fc11.x86_64 nss-softokn-freebl-3.12.6-1.2.fc11.x86_64 openssl-0.9.8n-1.fc11.x86_64 pcre-7.8-2.fc11.x86_64 perl-libs-5.10.0-82.fc11.x86_64 popt-1.13-5.fc11.x86_64 radiusclient-ng-0.5.6-4.fc11.x86_64 rpm-libs-4.7.2-1.fc11.x86_64 tcp_wrappers-libs-7.6-55.fc11.x86_64 xz-libs-4.999.9-0.1.beta.20091007git.fc11.x86_64 zlib-1.2.3-22.fc11.x86_64 (gdb) bt full #0 0x00007f8a11d55fa7 in unref_dlg (dlg=0x7f89f7e07470, cnt=1) at dlg_hash.c:474 d_entry = 0x0 __FUNCTION__ = "unref_dlg" #1 0x00007f8a11d5180f in unref_dlg_from_cb (t=0x7f89f7d9c660, type=4096, param=0x7f8a1836a6e0) at dlg_handlers.c:622 dlg = 0x7f89f7e07470 #2 0x00007f8a18138ea3 in run_trans_callbacks (type=4096, trans=0x7f89f7d9c660, req=0x0, rpl=0x0, code=0) at t_hooks.c:240 cbp = 0x7f89f7dc30e8 backup = 0x71a9d0 trans_backup = 0xffffffffffffffff __FUNCTION__ = "run_trans_callbacks" #3 0x00007f8a181273cc in free_cell (dead_cell=0x7f89f7d9c660) at h_table.c:132 b = 0x0 i = 1 rpl = 0x0 tt = 0x0 foo = 0x7fff4282f190 p = 0x7f89f7d3b068 #4 0x00007f8a18127bb6 in free_hash_table () at h_table.c:345 p_cell = 0x7f89f7d9c660 tmp_cell = 0x0 i = 4075 #5 0x00007f8a181342a4 in tm_shutdown () at t_funcs.c:109 __FUNCTION__ = "tm_shutdown" #6 0x00000000004529f6 in destroy_modules () at sr_module.c:321 t = 0x7349d0 foo = 0x734910 __FUNCTION__ = "destroy_modules" #7 0x000000000041f6b4 in cleanup (show_status=1) at main.c:331 No locals. #8 0x0000000000420597 in handle_sigs () at main.c:517 chld = 0 chld_status = 134 i = 12 do_exit = 1 ---Type <return> to continue, or q <return> to quit--- shutdown_time = 60 __FUNCTION__ = "handle_sigs" #9 0x00000000004217b5 in main_loop () at main.c:859 chd_rank = 12 i = 4 pid = 21442 si = 0x0 __FUNCTION__ = "main_loop" #10 0x0000000000423410 in main (argc=3, argv=0x7fff4282f498) at main.c:1321 cfg_log_stderr = 0 cfg_stream = 0x1fe1010 c = -1 r = 0 tmp_len = 0 port = 0 proto = 4910128 ret = -1 rfd = 4 tmp = 0x7fff4282ff8a "" options = 0x4b77e0 "f:cCm:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:" rand_source = 0x4b7d9c "/dev/urandom" seed = 3628387751 __FUNCTION__ = "main" (gdb) (gdb) quit

3 7

Failover with UCARP and Monit
by Hugh Waite 17 Feb '11

17 Feb '11

Hello, I am setting up a high-availablilty kamailio system using UCARP to failover between active and standby instances. To detect failure, we intend to use Monit. Monit can monitor the kamailio PID and start the process when needed (Example on the wiki http://www.kamailio.org/dokuwiki/doku.php/install:configure-initd-script) and it can also do OPTIONS pings to verify it is working. If the pings fail we will initiate a ucarp swap. However if the server is currently in standby, it does not have the V-IP address, so I don't want to run the OPTIONS pings (I think). Does anyone use a similar system and can provide an example of how ucarp, monit and kamailio can work together? Many thanks, Hugh Waite

4 5

Kamailio and Asterisk Realtime Integration using Asterisk Database
by Pavel Miskov 17 Feb '11

17 Feb '11

Hi all, I am trying to integrate Kamailio and Asterisk as explained in http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb document but I have some problems. 1. When Asterisk sends the call back to Kamailio, Asterisk generates new INVITE but with wrong From URI. In From header, "Display name" is OK (Phone A) but "From URI" is wrong (it is URI of Phone B). 2. When Phone B answers the call, Asterisk generates another INVITE to Phone B. When Phone B sends OK to this second INVITE Asterisk generates another INVITE to Phone A. 3. When Phone B sends BYE, Asterisk generates INVITE to Phone A and when it receives OK from Phone A it sends BYE to Phone A. I can fix the first problem with transformation in Kamailio but is this the way how it is supposed to work or I misconfigured something? I have tried this with kamailio 3.1 and asterisk 1.6.2.16.1 (and 1.8.2.3). Thanks Pavel

2 1

Mediaproxy 2.4.4 sessions
by Ricardo Martinez 17 Feb '11

17 Feb '11

Hello. I’m using kamailio 3.1.2 and mediaproxy 2.4.4 . Is there a way to see the active media sessions like in the old mediaproxy? I was using the command ./sessions Thanks Ricardo-.

2 1

Kamailio 3.0.4: Segfault at 4 [..] error 4 in dialog.so
by Klaus Feichtinger 17 Feb '11

17 Feb '11

Hello, does anybody have an idea, what this error messages could mean? Feb 16 11:23:27 SipSrvLstI1 kernel: [429767.707165] kamailio[10758]: segfault at 4 ip b78ba6e5 sp bfacece0 error 4 in dialog.so[b78a0000+32000] Feb 16 13:40:42 SipSrvLstI1 kernel: [438003.071797] kamailio[24880]: segfault at 4 ip b79416e5 sp bff7d460 error 4 in dialog.so[b7927000+32000] Feb 16 18:47:25 SipSrvLstI1 kernel: [456406.166885] kamailio[20939]: segfault at 4 ip b78d56e5 sp bfd821c0 error 4 in dialog.so[b78bb000+32000] I am setting the dialog flag for any incoming call..... Whenever a load test is running, this error occurs. I am not sure if it is reproduceable, or not. But I think it should be, because in worst case (during a load test over night) this error was indicated in a period of 1-2 minutes. But I've found this error only yesterday, because the server was down (as a result of this error....). The daemon (kamailio_3.0.4+lenny1_i386.deb) is running on a virtual machine (ESX) with OS Debian Lenny. How can I find / deliver additional info to this error / fault for analyzation? Thanks for any hints. Klaus

2 1

ACC og ACC_RADIUS module
by Morten Isaksen 17 Feb '11

17 Feb '11

Hi, We have a OpenSER 1.1 platform running with radius accounting and I am in the progress of updating it to Kamailio 3.1. I am trying to decide if I should do accounting via Radius or directly to MySQL on the new platform. The only benefits a can see with Radius is that you can build some redundancy into your radius client. If one Radius server is failing then try the next and you can configure radius to log to a file if the DB is down. But i think you can get the same level of redundancy with a replicated DB setup with heartbeat/pacemaker. If I choose to do the accounting direct to MySQL I will skip the Radius layer (and one error source). Are there any other pros and cons? -- Morten Isaksen

2 1

ERROR: failover functions used, but AVPs paraamters required are NULL
by Gary Chen 17 Feb '11

17 Feb '11

kamailio version 3.1.2 I am trying to setup dispatcher to use its failover feature. Here is dispatcher part of configure file: loadmodule "dispatcher.so" modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshhold", 10) modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=4") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_from", "sip:lb1 <at> m-lab-ca805-sig.kd-lab.de") modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "flags", 2) But it is still give the following error: 0(1917) ERROR: dispatcher [dispatcher.c:624]: failover functions used, but AVPs paraamters required are NULL -- feature disabled Does anybody know why? Gary

2 1

Re: [SR-Users] pdb module timeouts
by Thomas Baumann 16 Feb '11

16 Feb '11

2 1

Re: [SR-Users] Kamailio 1.5.4 and beyond - mhomed issue under Linux
by Sean O'Donnell 16 Feb '11

16 Feb '11

Hi Marius, Thanks for the reply. I found the patch in the 1.5 CVS branch. By the way, the 1.5.5 source tarball in /pub/kamailio/1.5.5/src was created before this patch was written, so the patch is not in there. The tarball I created from the 1.5 branch did have it. Thanks again, Sean O'Donnell Senior Engineer uReach Technologies, Inc. ---- On Mon, 14 Feb 2011, marius zbihlei (marius.zbihlei(a)1and1.ro) wrote: On 02/11/2011 09:14 PM, Sean O'Donnell wrote: > Hi all, > Hello Sean, > I checked the GIT repository and noticed that there was a patch in forward.c for > this issue. Looks like it was done 11/4/2010. Two questions: > > 1) Is there any reason that patch didn't make it into Kamailio 3.1.2 ? > Indeed, there is no reason. Actually I have cherry-picked the patches into the 3.1 branch (commit_notice 3.1 3ec1e9 and 9ef0a1e). Thanks for noticing. > 2) Any reason I shouldn't apply that patck to my 1.5.4 release? > The patch was already applied to the svn 1.5 branch (don't know if the latest 1.5.5 tar ball has it) but you can install it from svn and you can find it there (commit 6050) Cheers Marius > Thanks, > Sean O'Donnell > Senior Engineer > uReach Technologies, Inc. > > > > > > ---- On Thu, 10 Feb 2011, Sean O'Donnell (skodonnell(a)ureach.com) wrote: > > Hi all, > > I just started deploying Kamailio release 1.5.4, and I think there's an issue > with how Kamailio identifies an outgoing interface when mhomed is enabled under > Linux. > > I use Kamailio as a call distributor/proxy between a soft-switch/SBC and a > voicemail platform. It > runs on a CentOS 5.3 (Linux 2.6 kernel) host with two network interfaces and is > configured such that it listens on both interfaces. One interface (public > interface) handles traffic with the SBC, the other (private interface) handles > with the VM platform. The 'mhomed' option is enabled. > > After upgrading from 1.5.3 to 1.5.4, I started noticing problems with UDP > packets coming out of the public interface. After looking at some ngrep > captures on that interface, I noticed that some packets had the source IP > address of the private interface and also had Record-Route and Via headers for > the private interface only - no headers for the public interface were there. > > Usually when I see the wrong source IP in a UDP packet, it's an issue with how > routes are set up on the host. However, I had our network engineer double check > them, and they seem fine (no ambiguous routes). The fact that I captured these > messages on the public interface also indicates to me that the kernel is routing > the message correctly. The missing Record-Route and Via for the public > interface, however, lead me to believe that the proxy didn't correctly identify > the outgoing interface in the first place. > > After looking at the ChangeLog for 1.5.4, I noticed that the some new logic was > put in to improve performance when mhomed is enabled (r5971) in forward.c, and I > think this is the issue. > > As I understand it, prior to 1.5.4, when mhomed was enabled, Kamailio determined > the outgoing interface by creating a temporary UDP socket, invoking connect() on > the socket with the packet destination, then checking the source IP of the > socket that the kernel assigned using getsockname(). After the source address > was determined, the temp socket was closed closed. As of 1.5.4, this was > modified to reuse the temporary socket and just re-invoking connect() with a new > destination address. > > The problem with the enhancement is that Linux (again, at least in the 2.6 > kernels I'm using) > doesn't seem to rebind a new source address to the socket when connect() is > called more than once on > a UDP socket. Instead, it keeps the original one, and thus the wrong interface > is assumed. > > I wrote a small program to confirm this - basically creates a UDP socket, calls > connect()/getsockname() > multiple times using different destination addresses. I ran it on several 2.6 > kernels, including > Centos4.x and Centos5. The result was always that the source address of the > socket wasn't changed after the first connect(), regardless of the destination > address. The only way I could get it work as > required was to first do a connect() using a zero'd out AF_UNSPEC address before > doing the > connect() to the remote address. I also ran it on Solaris and it worked. Go > figure. > > I've downloaded the latest stable release (3.1.2) but I think the issue is still > there, and I don't see > anything in the user groups that addresses this. > > Any help would be appreciated. > > Thanks, > > Sean > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Error when tring to load lcr module.
by Gary Chen 16 Feb '11

16 Feb '11

I am using Kamailio 3.1.2. I added lcr module in my kamailio.cfg file. After run this: ./kamctl -E I got this error: 0(23744) ERROR: lcr [lcr_mod.c:1235]: lcr_gw params at row <0> does not start with ';' 0 I have set up some gateway information in lcr_gw , lcr_rule and lcr_rule_target. From log it seems retrieve all the information but then got above error. Here is my lcr part of config: loadmodule "lcr.so" modparam("lcr", "db_url", DBURL) modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") Gary

2 1

Re: [SR-Users] pdb module timeouts
by Thomas Baumann 15 Feb '11

15 Feb '11

Hello, I did a fallback to the original source, without sucess. I still have massive timeouts. So far I have no idea, how to go on. ps. what lenght of the carrier codes would be ok ? 3...10 digits, additional characters needed like A,B,C,D,E,F? regards, Thomas Hi Henning, this change was done at pdbt.c: $ diff pdbt.c pdbt_changed.c 245c245 < bufsize = slen + 1 + 1 + 3 + 1 + 1; // line buff er (telephone number + colon + white space + carrier ID + newline + \0) --- > bufsize = slen + 1 + 1 + 4 + 1 + 1; // line buff er (telephone number + colon + white space + carrier ID + newline + \0) 257,258c257,258 < ret = snprintf(p, 5, "%d\n", node->carrier); < if (ret < 1 || ret > 4) { --- > ret = snprintf(p, 6, "%d\n", node->carrier); > if (ret < 1 || ret > 5) { If I run querys in filemode against the server, where is no timeout. pdbt query -f /tmp/numbers3 -r 10.12.18.21:10001 -q 1 | grep answer regards, Thomas -----Ursprüngliche Nachricht----- Von: "Henning Westerholt" <henning.westerholt(a)1und1.de> Gesendet: 11.02.2011 09:33:05 An: sr-users(a)lists.sip-router.org Betreff: Re: [SR-Users] pdb module timeouts On Wednesday 09 February 2011, Thomas Baumann wrote: > I am using the PDB module and server components for number portability. 2 > instances of PDB Server runs on (10.12.19.51/10001/10002), Kamailio on > (10.12.19.21). With a small amount of traffic (-cmax 150 -cps 10 > -callduration 3), where are timeouts: WARNING: pdb [pdb.c:260]: exceeded > timeout while waiting for response. > > One requested number was 307111094, where the module prints out a timeout. > > The funny part is, that I can see the responses at least arriving at the > 10.12.19.21 interface on time. > > Request send: 0,200855 s > Answer received: 0,201027 s > > That are 0,172 ms and far away from a timeout. > > What could be the reason ? Hi Thomas, a bit difficult to say on a first sight.. Maybe some scheduling or load issues on the local machine? I've just checked two production server here, i don't see it so far in the available logs. How much load do you've on the machine? And what kind of timeout value do you specified? > ps. A small change on the server part was done: handle 4 digit carrier > codes. Ah, i see. I guess its not a big change? Regards, Henning ___________________________________________________________ Schon gehört? WEB.DE hat einen genialen Phishing-Filter in die Toolbar eingebaut! http://produkte.web.de/go/toolbar

2 1

CORE postgres module
by Bruno Bresciani 15 Feb '11

15 Feb '11

Hi, During my tests with kamailio 1.5.0 a core is generated when postgres is disconnected and I try register a user or make a call in kamailio. Analising the module postgres source code I notice that core is generated by the PQescapeStringConn located in db_postgres_val2str function. Someone know why this core is generated? Best Regards

2 2

Issue with avp
by Amit Nepal 14 Feb '11

14 Feb '11

Hi, I have been trying to load rpid while loading credentials. modparam("auth_db", "load_credentials", "$avp(i:123)=rpid") Now, I am trying to do a check in my routing logic. xlog("L_NOTICE","The avp is :$avp(i:123)"); (I dont get the value here either, i can't see the value when i do avp_print() if($avp(i:123)<5) { sl_send_reply("408","Message Here"); } I dont get the value of avp at that place. Any guidance please. -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

2 4

kamailio 1.5.5 crashes continue
by Andrew O. Zhukov 14 Feb '11

14 Feb '11

Core was generated by `/usr/local/sbin/kamailio -P /var/run/openser/openser.pid -m 32 -u openser -g op'. Program terminated with signal 6, Aborted. #0 0x00000039d8c30265 in raise () from /lib64/libc.so.6 (gdb) backtrace #0 0x00000039d8c30265 in raise () from /lib64/libc.so.6 #1 0x00000039d8c31d10 in abort () from /lib64/libc.so.6 #2 0x000000000046c397 in qm_debug_frag (qm=0x733c00, f=0x841420) at mem/q_malloc.c:137 #3 0x000000000046d99a in qm_free (qm=0x733c00, p=0x841450, file=0x4e4d30 "parser/digest/digest.c", func=0x4e4da0 "free_credentials", line=95) at mem/q_malloc.c:439 #4 0x0000000000495fac in free_credentials (_b=0x2b875404a130) at parser/digest/digest.c:95 #5 0x0000000000471a36 in clean_hdr_field (hf=0x2b875404a100) at parser/hf.c:116 #6 0x00002b8750ab78de in clean_msg_clone (msg=0x2b8754049130, min=0x2b8754049130, max=0x2b875404a748) at sip_msg.h:54 #7 0x00002b8750ab77b7 in run_trans_callbacks (type=2, trans=0x2b8754057ec0, req=0x2b8754049130, rpl=0x7c0eb8, code=200) at t_hooks.c:245 #8 0x00002b8750abe39d in t_reply_matching (p_msg=0x7c0eb8, p_branch=0x7fffbe5d11e8) at t_lookup.c:888 #9 0x00002b8750abe997 in t_check (p_msg=0x7c0eb8, param_branch=0x7fffbe5d11e8) at t_lookup.c:964 #10 0x00002b8750acd79b in reply_received (p_msg=0x7c0eb8) at t_reply.c:1395 #11 0x000000000041c6db in forward_reply (msg=0x7c0eb8) at forward.c:576 #12 0x000000000043ccf0 in receive_msg ( buf=0x712980 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP XXX.XX.XXX.13;branch=z9hG4bK8aa9.93e271f1.0;received=XXX.XX.XXX.13\r\nVia: SIP/2.0/UDP XXX.XX.XXX.236:5060;received=XXX.XX.XXX.236;branch=z9hG4bK37dacff3;rport=5060\r\nRec"..., len=576, rcv_info=0x7fffbe5d1340) at receive.c:212 #13 0x00000000004692e3 in udp_rcv_loop () at udp_server.c:449 #14 0x0000000000420ecb in main_loop () at main.c:774 #15 0x0000000000422e0f in main (argc=11, argv=0x7fffbe5d15c8) at main.c:1321 (gdb) q [root@ tmp]# openser -V version: kamailio 1.5.5-notls (x86_64/linux) flags: STATISTICS, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ main.c compiled on 09:42:37 Feb 10 2011 with gcc 4.1.2 Andrew.

2 3

kamailio and MySQL licensing question
by Siddhardha Garige 14 Feb '11

14 Feb '11

hello all, I have a question regarding MySQL licensing. We configured kamailio to work in edge proxy mode with drive-by registration to support our network architecture. One of our clients is interested in conducting some trails with us and we are setting up a test network at client's facility. As a part of this process we will be installing our custom kamailio on their servers with MySQL. We are not charging our client for kamailio and mysql implementation since they both are open source implementations. But we will be charging them for our time at clients facility. Are we violating any MySQL license policies by installing our custom kamailio with MySQL for our client? thanks Sid "May the light be with you." ______________________________________________ Siddhardha Garige www.luminepixels.com

2 1

Re: [SR-Users] Kamailio 1.5.4 and beyond - mhomed issue under Linux
by Sean O'Donnell 14 Feb '11

14 Feb '11

Hi all, I checked the GIT repository and noticed that there was a patch in forward.c for this issue. Looks like it was done 11/4/2010. Two questions: 1) Is there any reason that patch didn't make it into Kamailio 3.1.2 ? 2) Any reason I shouldn't apply that patck to my 1.5.4 release? Thanks, Sean O'Donnell Senior Engineer uReach Technologies, Inc. ---- On Thu, 10 Feb 2011, Sean O'Donnell (skodonnell(a)ureach.com) wrote: Hi all, I just started deploying Kamailio release 1.5.4, and I think there's an issue with how Kamailio identifies an outgoing interface when mhomed is enabled under Linux. I use Kamailio as a call distributor/proxy between a soft-switch/SBC and a voicemail platform. It runs on a CentOS 5.3 (Linux 2.6 kernel) host with two network interfaces and is configured such that it listens on both interfaces. One interface (public interface) handles traffic with the SBC, the other (private interface) handles with the VM platform. The 'mhomed' option is enabled. After upgrading from 1.5.3 to 1.5.4, I started noticing problems with UDP packets coming out of the public interface. After looking at some ngrep captures on that interface, I noticed that some packets had the source IP address of the private interface and also had Record-Route and Via headers for the private interface only - no headers for the public interface were there. Usually when I see the wrong source IP in a UDP packet, it's an issue with how routes are set up on the host. However, I had our network engineer double check them, and they seem fine (no ambiguous routes). The fact that I captured these messages on the public interface also indicates to me that the kernel is routing the message correctly. The missing Record-Route and Via for the public interface, however, lead me to believe that the proxy didn't correctly identify the outgoing interface in the first place. After looking at the ChangeLog for 1.5.4, I noticed that the some new logic was put in to improve performance when mhomed is enabled (r5971) in forward.c, and I think this is the issue. As I understand it, prior to 1.5.4, when mhomed was enabled, Kamailio determined the outgoing interface by creating a temporary UDP socket, invoking connect() on the socket with the packet destination, then checking the source IP of the socket that the kernel assigned using getsockname(). After the source address was determined, the temp socket was closed closed. As of 1.5.4, this was modified to reuse the temporary socket and just re-invoking connect() with a new destination address. The problem with the enhancement is that Linux (again, at least in the 2.6 kernels I'm using) doesn't seem to rebind a new source address to the socket when connect() is called more than once on a UDP socket. Instead, it keeps the original one, and thus the wrong interface is assumed. I wrote a small program to confirm this - basically creates a UDP socket, calls connect()/getsockname() multiple times using different destination addresses. I ran it on several 2.6 kernels, including Centos4.x and Centos5. The result was always that the source address of the socket wasn't changed after the first connect(), regardless of the destination address. The only way I could get it work as required was to first do a connect() using a zero'd out AF_UNSPEC address before doing the connect() to the remote address. I also ran it on Solaris and it worked. Go figure. I've downloaded the latest stable release (3.1.2) but I think the issue is still there, and I don't see anything in the user groups that addresses this. Any help would be appreciated. Thanks, Sean _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Error running kamailio 3.1.0
by Lucas Alvarez 12 Feb '11

12 Feb '11

Hi, I have compiled kamailio 3.1.0 without any error and I having this error when running kamailio: ERROR: <core> [sr_module.c:523]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/sl.so>: /usr/local/lib/kamailio/modules_k/sl.so: undefined symbol: fm_malloc Any will be appreciated. Regards, Lucas

2 1

After upgrade from openser 1.3.4 to kamailio 1.5.5 the same crash set
by Andrew O. Zhukov 12 Feb '11

12 Feb '11

[root@ tmp]# /usr/local/sbin/kamailio -V version: kamailio 1.5.5-notls (x86_64/linux) flags: STATISTICS, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ main.c compiled on 12:38:36 Feb 2 2011 with gcc 4.1.2 ----------------------------- Core was generated by `/usr/local/sbin/kamailio -P /var/run/openser/openser.pid -m 32 -u openser -g op'. Program terminated with signal 11, Segmentation fault. #0 0x000000000046b0e3 in fm_malloc (qm=0x72dc00, size=32) at mem/f_malloc.c:354 354 if ((*f)->size>=size) goto found; (gdb) backtrace #0 0x000000000046b0e3 in fm_malloc (qm=0x72dc00, size=32) at mem/f_malloc.c:354 #1 0x00002b30f2803087 in build_rr (_l=0x76f110, _l2=0x76fe80, user=0x7fffe9c5a500, tag=0x777a58, params=0x0, _inbound=0) at record.c:176 #2 0x00002b30f2802b7a in record_route (_m=0x76e0e0, params=0x0) at record.c:322 #3 0x00002b30f28047db in w_record_route (msg=0x76e0e0, key=0x0, bar=0x0) at rr_mod.c:212 #4 0x000000000040ed9b in do_action (a=0x73f5a0, msg=0x76e0e0) at action.c:874 #5 0x000000000040c03a in run_action_list (a=0x73f5a0, msg=0x76e0e0) at action.c:145 #6 0x000000000040e6a7 in do_action (a=0x73f810, msg=0x76e0e0) at action.c:746 #7 0x000000000040c03a in run_action_list (a=0x73e418, msg=0x76e0e0) at action.c:145 #8 0x000000000040c2a9 in run_actions (a=0x73e418, msg=0x76e0e0) at action.c:120 #9 0x000000000040c357 in run_top_route (a=0x73e418, msg=0x76e0e0) at action.c:195 #10 0x000000000043bda4 in receive_msg ( buf=0x70c980 "NOTIFY sip:XXXXXX.com SIP/2.0\r\nVia: SIP/2.0/UDP XX.XXX.101.68:5060;branch=z9hG4bK-6ee3865\r\nFrom: VTHome <sip:101650@XXXXXX.com>;tag=129d73a13db8ec7fo0\r\nTo: <sip:XXXXX.com>\r\nCall-ID: e3fd1da9-142a0a17"..., len=373, rcv_info=0x7fffe9c5ae90) at receive.c:175 #11 0x0000000000467eeb in udp_rcv_loop () at udp_server.c:449 #12 0x000000000042097b in main_loop () at main.c:774 #13 0x00000000004228b0 in main (argc=11, argv=0x7fffe9c5b118) at main.c:1321 (gdb) print size $1 = 32 (gdb) quit -------------------------------------------- Core was generated by `/usr/local/sbin/kamailio -P /var/run/openser/openser.pid -m 32 -u openser -g op'. Program terminated with signal 11, Segmentation fault. #0 0x000000000046bf7b in fm_status (qm=0x72dc00) at mem/f_malloc.c:609 609 size+=f->size,f=f->u.nxt_free,i++,j++){ (gdb) backtrace #0 0x000000000046bf7b in fm_status (qm=0x72dc00) at mem/f_malloc.c:609 #1 0x000000000041feb3 in sig_usr (signo=15) at main.c:563 #2 <signal handler called> #3 0x00000039d8cd4a51 in __recvfrom_nocancel () from /lib64/libc.so.6 #4 0x0000000000467bf4 in udp_rcv_loop () at udp_server.c:408 #5 0x000000000042097b in main_loop () at main.c:774 #6 0x00000000004228b0 in main (argc=11, argv=0x7fffe9c5b118) at main.c:1321 (gdb) print i $1 = 402 (gdb) print j $2 = 1 (gdb) print size $3 = 7234295468789601279 (gdb) print f $4 = (struct fm_frag *) 0x3738656435393838 (gdb) print f->size Cannot access memory at address 0x3738656435393838 ------------------------------------------------------------------- Andrew O. Zhukov

2 6

pdb module timeouts
by Thomas Baumann 11 Feb '11

11 Feb '11

Hello, I am using the PDB module and server components for number portability. 2 instances of PDB Server runs on (10.12.19.51/10001/10002), Kamailio on (10.12.19.21). With a small amount of traffic (-cmax 150 -cps 10 -callduration 3), where are timeouts: WARNING: pdb [pdb.c:260]: exceeded timeout while waiting for response. One requested number was 307111094, where the module prints out a timeout. The funny part is, that I can see the responses at least arriving at the 10.12.19.21 interface on time. Request send: 0,200855 s Answer received: 0,201027 s That are 0,172 ms and far away from a timeout. What could be the reason ? regards, Thomas ps. A small change on the server part was done: handle 4 digit carrier codes. level3_logs: Feb 9 17:19:42 node2 /openser/sbin/kamailio[21200]: DEBUG: pdb [pdb.c:201]: querying '307111094'... Feb 9 17:19:42 node2 /openser/sbin/kamailio[21200]: WARNING: pdb [pdb.c:260]: exceeded timeout while waiting for response. ___________________________________________________________ Schon gehört? WEB.DE hat einen genialen Phishing-Filter in die Toolbar eingebaut! http://produkte.web.de/go/toolbar

3 4

Change transport=tls within Contact Header
by Bernhard Suttner 11 Feb '11

11 Feb '11

Hi, is there a function to change the transport field within the Contact Header? t_relay does relay it to the UDP port of the PBX but the transport within the Contact Header is still set to TLS. Later, the PBX does try to establish a TLS connection. The idea is now, that kamailio does set the transport of the Contact Header to UDP. Another possible way would to be, that I use the re.subst method but maybe there is a special function for that. Thanks in advance. Best regards, Bernhard

2 1

NAT, TLS and location table
by Bernhard Suttner 11 Feb '11

11 Feb '11

Hi, I am using TLS and recognize the following problem: The TLS connection are build up successfully but the natping (natping_interval = 10) does not send small dummy packets to the phones. The phones are behind a firewall with NAT. Registered phones with NAT but UDP do work correctly. They are getting the natping every 10 seconds. After 120 seconds (should be the tcp_connection_timeout) kamailio does send a FIN to the TLS phone to close the TLS connection. Should I increase the tcp_connection_timeout to a value bigger than the registration timeout? I thought I do not need that, because of the natping_interval. Is it maybe better to use a SIP-Options Ping instead of the small dummy packets? I would prefer the dummy packets because they are much smaller. BTW: Kamailio 3.1.1 and snom360 with latest v8 firmware. Thanks in advance! Best regards, Bernhard

2 3

handle_new_connect ... Too many open files
by Efelin Novak 11 Feb '11

11 Feb '11

Hi, today during two hours kamailio made 2G of a syslog messages. This message was repeated for 10 000 000 times. /sbin/kamailio[24272]: ERROR: <core> [tcp_main.c:3959]: WARNING: handle_new_connect: error while accepting connection(24): Too many open files It was preceded by following error. /sbin/kamailio[24265]: ERROR: <core> [tcp_read.c:269]: error reading: Connection reset by peer (104) /sbin/kamailio[24265]: ERROR: <core> [tcp_read.c:882]: ERROR: tcp_read_req: error reading Finally it stopped without any given reason. I'm using kamailio 3.1.0 (i386/linux) 1e204f, callcontrol 2.0.11 and clustered mysql. Was this some kind of attack, bug, error, misconfiguration or what? Thank for answer, Efelin

2 1

TLS Support
by Bernhard Suttner 11 Feb '11

11 Feb '11

Hi, how could I compile the TLS module and only the TLS module - not the other sources. BTW: The documentation within the http://www.kamailio.org/docs/tls.html is not up-to-date. Thanks in advance. Best regards, Bernhard

2 2

Kamailio as a sip client
by Amit Nepal 11 Feb '11

11 Feb '11

Hi, Is it possible to have kamailio register to a different proxy and route calls through that proxy ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

3 3

Kamailio 1.5.4 and beyond - mhomed issue under Linux
by Sean O'Donnell 10 Feb '11

10 Feb '11

Hi all, I just started deploying Kamailio release 1.5.4, and I think there's an issue with how Kamailio identifies an outgoing interface when mhomed is enabled under Linux. I use Kamailio as a call distributor/proxy between a soft-switch/SBC and a voicemail platform. It runs on a CentOS 5.3 (Linux 2.6 kernel) host with two network interfaces and is configured such that it listens on both interfaces. One interface (public interface) handles traffic with the SBC, the other (private interface) handles with the VM platform. The 'mhomed' option is enabled. After upgrading from 1.5.3 to 1.5.4, I started noticing problems with UDP packets coming out of the public interface. After looking at some ngrep captures on that interface, I noticed that some packets had the source IP address of the private interface and also had Record-Route and Via headers for the private interface only - no headers for the public interface were there. Usually when I see the wrong source IP in a UDP packet, it's an issue with how routes are set up on the host. However, I had our network engineer double check them, and they seem fine (no ambiguous routes). The fact that I captured these messages on the public interface also indicates to me that the kernel is routing the message correctly. The missing Record-Route and Via for the public interface, however, lead me to believe that the proxy didn't correctly identify the outgoing interface in the first place. After looking at the ChangeLog for 1.5.4, I noticed that the some new logic was put in to improve performance when mhomed is enabled (r5971) in forward.c, and I think this is the issue. As I understand it, prior to 1.5.4, when mhomed was enabled, Kamailio determined the outgoing interface by creating a temporary UDP socket, invoking connect() on the socket with the packet destination, then checking the source IP of the socket that the kernel assigned using getsockname(). After the source address was determined, the temp socket was closed closed. As of 1.5.4, this was modified to reuse the temporary socket and just re-invoking connect() with a new destination address. The problem with the enhancement is that Linux (again, at least in the 2.6 kernels I'm using) doesn't seem to rebind a new source address to the socket when connect() is called more than once on a UDP socket. Instead, it keeps the original one, and thus the wrong interface is assumed. I wrote a small program to confirm this - basically creates a UDP socket, calls connect()/getsockname() multiple times using different destination addresses. I ran it on several 2.6 kernels, including Centos4.x and Centos5. The result was always that the source address of the socket wasn't changed after the first connect(), regardless of the destination address. The only way I could get it work as required was to first do a connect() using a zero'd out AF_UNSPEC address before doing the connect() to the remote address. I also ran it on Solaris and it worked. Go figure. I've downloaded the latest stable release (3.1.2) but I think the issue is still there, and I don't see anything in the user groups that addresses this. Any help would be appreciated. Thanks, Sean

1 0

Merging nathelper modules
by Ovidiu Sas 10 Feb '11

10 Feb '11

Hello all, The nathelper module in modules_k was split in two: - nathelper (dealing with signaling); - rtpproxy (dealing with rtpproxy protocol). I would like to move the rtpproxy module from "modules_k" into "modules" and remove rtpproxy functionality from nathelper (s). This will give to (s) and (k) users: - rtpproxy: a single module for dealing with rtpproxy servers; - nathelper: two variants for dealing with NAT signaling. Next step, will be to merge the two nathelper modules into a single one. Thoughts? Regards, Ovidiu Sas

2 2

nathelper-module and sdp offer in 200 OK
by Emil Kroymann 10 Feb '11

10 Feb '11

Hi, We recently had a problem with the nathelper module and rtpproxy in a scenario where the SDP offer is sent only in the 200 OK. We use sip-router 3.1 and rtp-proxy from git master. The sip-router configuration uses the rtpproxy_offer() and rtpproxy_answer() functions in appropriate places. The problem is, that the arguments sent to the rtpproxy, when the ACK with the sdp answer arrives, seems to be not in the order, that rtpproxy expects. On the 200 OK, the nathelper module sends callid, to-tag, from-tag to rtpproxy. On the ACK, the nathelper module sends callid, from-tag, to-tag (with different command prefixes, of course, but I cannot remember them atm). The version of rtpproxy that we are using seems to expect, that the order of arguments sent on the ACK request is the same as on the 200 OK. My question: are there any module parameters, to correct this behaviour? Regards, Emil -- Emil Kroymann VoIP Services Engineer Email: emil.kroymann(a)isaco.de Tel: +49-30-203899885 ISACO GmbH Kurfürstenstraße 79 10787 Berlin Germany Amtsgericht Charlottenburg, HRB 112464B Geschäftsführer: Daniel Frommherz

3 12

Use different certificate for different client with TLS
by Daniel GROTTI 09 Feb '11

09 Feb '11

Hi all, I would like to use kamailio 3.1 with TLS and verified also a client certificate. My tls.cfg file is as follow: --- tls.cfg ---- ..... ..... [server:MY_IP:5061] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = default_key.pem certificate = default_cert.pem ca_list = default_ca.pem [client:default] verify_certificate = no require_certificate = no .... .... So I want to verify the client certificate, to do that I use "is_peer_verified()" function in kamailio.cfg. As tls.cfg shows, I have to send to my clients the CA certificate and the client certificate (default_cert.pem + default_key.pem - signing by the CAcert). This client certificate is unique for all clients. Everything works fine. But suppose I wanted to create a client certificate for client 1 (cert_1.crt), and a different client certificate for client 2 (cert_2.crt) and I want to configure kamailio to be able to verified this different certificates. Does it possible ? How can I configure the tls.cfg file to do that ? I try to do something like this: [server:MY_IP:5061] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = default_key_1.pem certificate = default_cert_1.pem private_key = default_key_2.pem certificate = default_cert_2.pem ca_list = default_ca.pem But when kamailio restart it seems that it read only the last couple of row certifcate/private_key. Regards, Daniel G

2 1

[OpenSER-Users] uaCSTA support in openser
by Hugo Koblmueller 09 Feb '11

09 Feb '11

high.all! i'm wondering if there is any support of uaCSTA in openser (planned)? i'm just working on the integration of asterisk (*) environment to OCS 2007 environment, having openSER in the middle (mainly for TCP/UDP translation and smoothing out the protocol deficienes on both sides). in this setup the * having the openSER in front is talking to the OCS (and vice versa) via the OCS mediation server, which is moreorless sending standard SIP messages, which enables normal softphone (integration to *) of the office communicator. this configuration is already working... now i'm planning to go for the CTI integration, where there is no OCS mediation server in between OCS and openSER, doing the translation of SIP/CSTA to SIP. i'm thinking about using openSER for this task, that's why i'm looking for a CSTA module or perl programm, which is capable of this functionality. afaik for the CTI communication there isn't the full complexity of CSTA needed, just a subset mainly for call setup and call clearing. anyone having experience on this topic? thx & cheers -hugo Great Ideas for Small Devices Hugo Koblmueller Senior Staff Engineer Software Development COMNEON electronic technology GmbH & Co. OHG Freistaedter Strasse 400 4040 Linz Austria hugo.koblmueller(a)comneon.com tel: fax: mobile: Skype ID: +43 (5) 1777 - 15730 <http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%285%2 9+1777+%2D+15730&Email=hugo(a)koblmueller.com> +43 (5) 1777 - 15810 +43 (676) 82051280 <http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%28676 %29+82051280&Email=hugo(a)koblmueller.com> drhookson Want to always have my latest info? <https://www.plaxo.com/add_me?u=21475050628&src=client_sig_212_1_banner_join &invite=1&lang=en> Want a signature like <http://www.plaxo.com/signature?src=client_sig_212_1_banner_sig&lang=en> this?

7 13

Re: [SR-Users] module postgres SER
by Daniel-Constantin Mierla 09 Feb '11

09 Feb '11

please keep the mailing list cc-ed, sending private messages is not in the spirit of public mailing lists. Others may want to follow up the discussion now or later. Thanks, Daniel On 2/9/11 5:01 PM, Bruno Bresciani wrote: > Daniel, > > thanks for your reply, really ser-0.8.1.4 is too old but i need to > solve this problem on that version. My great doubt is Why the aug_free > function corrupt the url of database after some attemps to reconnect. > Well, I'll try to understand this question... > > Best Regards > > > 2011/2/9 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > On 2/9/11 2:19 PM, Bruno Bresciani wrote: > > Hi, > > I've seen the problem in a postgres module (SER-0.8.1.4), if > the connection fails and module tries to reparse url it fails > as CON_SQLURL(_h) is corrupted by the function aug_free() > after some reconnect attempts . When the postgres database > back to work, some modules doesn't get reconnect because the > db_url is corrupted. Why this is happening? There are some > solution for this problem? > > ser 0.8.1.4 is soooo old and I cannot fully remember, but I think > postgres module had no reconnect functionality at all by that time. > > However, version 3.1.x of SER (as well as Kamailio flavour) has db > reconnect functionality for postgres. You can try it and report if > something is not working, it will be fixed in 3.x, but I think > nobody is still developing on 0.8.x to backport anything there. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

problem with bye using rtpproxy
by Amit Nepal 09 Feb '11

09 Feb '11

I have been trying to figure this out While using kamailio and rtpproxy, the caller is not receiving the bye when callee hangs up but audio is two way and everything seems to be working fine, any one had this issue ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

2 1

Multiple call accounting from an IP
by Amit Nepal 09 Feb '11

09 Feb '11

Hi everyone, I am sure someone have been working with this scenario, how about accounting multiple calls from same account or ip address while using ip auth ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

2 1

fosdem 2011 presentation about p_usrloc
by Henning Westerholt 09 Feb '11

09 Feb '11

Hello, this year the FOSDEM developer conference was a again a really nice event, the first time with an own room dedicated completely to open source telephony solutions! If you're interested in our presentation about the new p_usrloc module and how to scale location services with Kamailio, you can find it at the usual place on our webserver: http://kamailio.org/events/2011-fosdem/p_usrloc.pdf Best regards, Henning

3 3

module postgres SER
by Bruno Bresciani 09 Feb '11

09 Feb '11

Hi, I've seen the problem in a postgres module (SER-0.8.1.4), if the connection fails and module tries to reparse url it fails as CON_SQLURL(_h) is corrupted by the function aug_free() after some reconnect attempts . When the postgres database back to work, some modules doesn't get reconnect because the db_url is corrupted. Why this is happening? There are some solution for this problem? Best Regard

2 1

Fosdem2011 presentation: SIP Web20 Lua
by Daniel-Constantin Mierla 09 Feb '11

09 Feb '11

Hello, the presentation I did during Fosdem in Brussels last weekend is available at: http://www.kamailio.org/events/2011-fosdem/dcm-sip-web-lua.pdf The focus was to show how to interact with Kamailio via HTTP and how Kamailio can interact with Web services via HTTP, using Lua to make it easier. There are slides for a demo config of sending asynchronous notifications to Twitter on missed calls (using modules app_lua, mqueue, sqlops and rtimer). Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Serusers] problem with SER not proxying messages
by Asuncion Merino Rodríguez 09 Feb '11

09 Feb '11

Hola, no he visto respuesta a este foro, por favor alguien podría dar alguna respuesta?

1 0

Kamailio v3.1.2 Released
by Daniel-Constantin Mierla 08 Feb '11

08 Feb '11

Hello, Kamailio SIP Server v3.1.2 stable release is out. This is a maintenance release of latest stable branch, 3.1, that includes fixes since release of v3.1.1. There is no change to database or configuration file required to upgrade to 3.1.2 from 3.1.0 or 3.1.1 versions, therefore it is strongly recommended to upgrade to v3.1.2. For more details about version 3.1.2, visit: http://www.kamailio.org/w/2011/02/kamailio-v3-1-2-released/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Problem with LCR!
by Ricardo Martinez 08 Feb '11

08 Feb '11

Hello. We’re having critical issues with our LCR, we have the Kamailio version 1.5.2, several calls are arriving to our LCR and from some reason the next message appears in the log : Feb 6 12:01:55 /usr/local/sbin/kamailio[14185]: NOTICE:lcr:gw_set_state: trunk "xx.xx.xx.135:5060" from group: <2099> is OFFLINE! We can see this message repeated for every single gateway in the “gw” table from the LCR, after this happened, the LCR rounting process is blocked and no more calls are processed. Can someone help us with this problem? Thanks in advance, *Regards,* *Ricardo Martinez.-* * *

3 3

planning release of v3.1.2
by Daniel-Constantin Mierla 08 Feb '11

08 Feb '11

Hello, I think it is time to release v3.1.2, first date that comes in my mind is next Thursday if everyone feels it is enough time to take care of backporting any fix he/she did and it is not yet there. That will provide us a fresh release for the FOSDEM event. If not, then maybe the other week, Tuesday, so the participants at the Kamailio Devel training in Barcelona can practice on it. Soon after we should plan also a release for previous stable, branch 3.0. Anyone having other options? Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 1

Problem with load_gws()
by Amit Nepal 08 Feb '11

08 Feb '11

Hi, Below is my config and i am getting error , missing load module modparam("lcr","db_url",DBURL) modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") route[LCR] { if(!load_gws()){ xlog("yes"); }; This gives me error : loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio 0(1265) : <core> [cfg.y:3412]: parse error in config file routes/route-lcr.cfg unknown command, missing loadmodule? for load_gws() ERROR: bad config file (1 errors) Any guidance please ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

3 4

Handling timers from an Avaya PBX (throwing 404s)
by Graham Wooden 08 Feb '11

08 Feb '11

Hi there, I turned up a few new sip trunk customers this past week, from which one of them was an Avaya system. The customer wasn¹t able to do a username/password auth, so I set them up as a permanent entry in the location table. I am running Kamailio 1.5.5. The issue is that I am throwing back a 404 to them when they do a timer event, as they are passing sip:Unknown@my.ip.addr. I don¹t have a user name Unknown, hence the 404. I noticed that with the timer events from other customer¹s PBXs, that they are passing sip:my.addr, from which I am giving back 200 OK. What is the best way to handle this? Just let it go or add a ³Unknown² user (I don¹t like the latter, really)? Calls to and from their system is fine, so right now it¹s just entries in my logs but I hate spending CPU cycles on stuff that isn¹t going to benefit anyone. I sent an email back to the customer to see if maybe he can resolve it on his end, but I am doubtful on getting it resolved on his end. Thanks, -graham

3 5

Proposal to extend acc/dialog modules in order to log CDRs
by Timo Reimann 07 Feb '11

07 Feb '11

Hi all, I'd like to bring forward a proposal for creating CDRs (Call Data Records) from Kamailio by means of extending both acc and dialog modules. Feedback is very welcome. So far, CDR generation requires post-processing of log entries created by the (transaction-based) acc module or some other means. Looking at OpenSIPS (which is already capable of producing CDRs), my idea is to copy the basic notion of tracking dialogs for to-be-accounted calls and log a proper CDR on termination. Creating basic CDRs including start time, end time, and call duration can be accomplished quite easily this way. Apart from this very minimum CDR content, however, one can think of a number of CDR fields that cannot be filled easily. For instance, caller identity isn't always contained in the From header (think of CLIR calls) but need to be determined from other headers depending on the type of call, possibly even involving a database lookup. That's why I'd like to provide a way to pass additional, CDR-specific data from the configuration file to the dialog structure associated with the call. When the dialog is about to terminate, the extended acc module would make sure that the added CDR data associated to the dialog is included in the CDR. This way, users may define CDR fields to their personal needs from the configuration file in a flexible manner. In order to accomplish this, the dialog module would need to be extended such that dialog-specific data may be stored for the duration of a dialog (which is not possible to this day AFAICS). A new setter function attached to the dialog interface (struct dlg_binds) could accept a simple key/value pair for storage purposes. Exporting this function to the configuration file and accompanying it with a corresponding getter function will ensure that dialog-specific data may be written and read generically. Thus, improvements to the dialog module may be leveraged not just for CDR-accounting purposes. Regarding the acc module, a couple of new features need to be implemented: First, the introduction of another module parameter called something like "cdr_fields" that comprises the set of key names designated for CDR inclusion. Hence, a line like modparam("acc", "cdr_fields", "caller, callee, foo, bar") would require Kamailio configurators to store dialog-specific values for the keys "caller", "callee", "foo", and "bar", respectively, during call processing. The values will be fed to the dialog structure employing the new dialog store function explained above. Second, when the acc module is in CDR mode, dialogs shall be tracked from creation to termination using dialog callbacks. On termination of a dialog, values from keys referenced in the "cdr_fields" parameter will be fetched from the terminating dialog structure and associated with the new, fresh CDR. Third, the CDR is persisted to either log file, database, or both. Note that my approach would not touch the existing acc module functionality: CDR mode is supposed to be working in addition to transaction-based accounting as people will likely want to keep transaction logs for debugging or similar needs. A flag would enable and disable CDR generation at will. I have not fully investigated re-usability possibilities but believe that acc's persistence functions may be taken advantage of by the new CDR-producing parts as well. That's it with regards to the basic mode of operation. I am also thinking of having a few other features, such as using a switch to set the call start time to the moment the ACK or OK is seen instead of the INVITE (because users shouldn't be billed for call setup times). Looking forward to seeing your feedback on this proposal. Cheers, --Timo

3 6

topoh, angle brackets and Contact URI params interpretation
by Andrew Pogrebennyk 07 Feb '11

07 Feb '11

I'm having an issue with topoh module in Kamailio 3.1.0. When Contact header is formed like this: Contact: 0991 <sip:192.168.0.107;line=sr-N6IAzBysz.tyz.D4M.VLOBMfOBFuWxvfMxV4> The other party responds properly. But when there is no angle brackets in Contact: Contact: sip:192.168.0.107;line=sr-N6IAzB3AWxyfz.stM.quOBFZMJZfWxj7W.y-MljAWBy* Really, many parsers implement "prefer shift to reduce" principle, which means if something can be interpreted in more enclosed expression, it will be interpreted this way and no as part of less enclosed expression so ;line is interpreted as header parameter but not URI parameter. RFC 3261 section 20 suggests that any URI parameters be contained within angle brackets: === The Contact, From, and To header fields contain a URI. If the URI contains a comma, question mark or semicolon, the URI MUST be enclosed in angle brackets (< and >). Any URI parameters are contained within these brackets. If the URI is not enclosed in angle brackets, any semicolon-delimited parameters are header-parameters, not URI parameters. === I think the topoh module should force the angle brackets. BTW it seems that parameter needs to be urlencoded, see rule 'other-param' in RFC 3261 section 25.1: other-param = pname [ "=" pvalue ] pname = 1*paramchar pvalue = 1*paramchar paramchar = param-unreserved / unreserved / escaped param-unreserved = "[" / "]" / "/" / ":" / "&" / "+" / "$" No .-* characters are allowed in the paramchar. But at least that's not causing me any problems. -- Sincerely, Andrew Pogrebennyk

3 6

Compiling Just One Module
by Amit Nepal 07 Feb '11

07 Feb '11

Could someone guide me how to compile and install just one module with flavor set to kamailio ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

3 4

authentication is not working
by Danny Dias 06 Feb '11

06 Feb '11

Hello my friends, I'm trying to configure authentication on my Kamailio and is not working at all :( I've added the following to the script to make it work: (but it doesn't) ... loadmodule "auth.so" loadmodule "auth_db.so" ... modparam("usrloc", "db_url", "mysql://kamailio:kamailiorw@localhost/kamailio") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://kamailio:kamailiorw@localhost/kamailio") modparam("auth_db", "load_credentials", "") ... if (!(method=="REGISTER") && from_uri==myself) /*no multidomain version*/ { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!db_check_from()) { sl_send_reply("403","Forbidden auth ID"); exit; } consume_credentials(); } ... if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } ## if (!db_check_to()) { sl_send_reply("403","Forbidden auth ID"); exit; } if (!save("location")) sl_reply_error(); exit; } But is not working at all...take a look: # U 2011/02/03 09:31:04.402891 172.30.140.22:48752 -> 172.30.140.8:5060 REGISTER sip:172.30.140.8 SIP/2.0 Via: SIP/2.0/UDP 172.30.140.22:48752 ;branch=z9hG4bK-d87543-9a54af22967ae417-1--d87543-;rport Max-Forwards: 70 Contact: <sip:1000@172.30.140.22:48752;rinstance=fcade2df86ce0ab8> To: "1000"<sip:1000@172.30.140.8 <sip%3A1000(a)172.30.140.8>> From: "1000"<sip:1000@172.30.140.8 <sip%3A1000(a)172.30.140.8>>;tag=cd3e2323 Call-ID: MmU0YjM1NThiNTg0ZjhiNGM4ODA4ZmU1YWFiYjBmNTc. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 # U 2011/02/03 09:31:04.404039 172.30.140.8:5060 -> 172.30.140.22:48752 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.140.22:48752 ;branch=z9hG4bK-d87543-9a54af22967ae417-1--d87543-;rport=48752 To: "1000"<sip:1000@172.30.140.8 <sip%3A1000(a)172.30.140.8> >;tag=c97b4d1cb1f3d0da549e06a8d482ef63.34dc From: "1000"<sip:1000@172.30.140.8 <sip%3A1000(a)172.30.140.8>>;tag=cd3e2323 Call-ID: MmU0YjM1NThiNTg0ZjhiNGM4ODA4ZmU1YWFiYjBmNTc. CSeq: 1 REGISTER Contact: <sip:1000@172.30.140.22:48752 ;rinstance=fcade2df86ce0ab8>;expires=3600 Content-Length: 0 Am i missing something in my configuration? Thanks in advance!!!

5 12

503 Server error response for REGISTER request (Asterisk v1.6.0.5)
by Vikram Ragukumar 04 Feb '11

04 Feb '11

Hello, I had posted this question earlier on asterisk-users, but didn't receive any reply, so I'm posting on sr-users in the hope that someone can provide guidance on how to debug this problem. Our setup involves a Sip softphone registering with a fresh install of Asterisk v1.6.0.5 through an outbound proxy (Kamailio v3.1.1). We observe that about 80% of the time, immediately after startup of the softphone, when the softphone sends a REGISTER message, Asterisk replies with a "503 Server error". However, subsequent REGISTER messages are replied to appropriately (with a 401 Unauthorized). There are very few devices (under 5) registered with Asterisk, so this is not an overload issue. Please find attached a SIP trace for device registration. - What might cause this behavior in Asterisk ? - Are there any config file parameters that require to be changed ? - Would it help to see any other specific logs, if so which ones ? Thanks and Regards, Vikram.

2 3

SIP Router 3.03 topoh
by dotnetdub 04 Feb '11

04 Feb '11

Hi All, Kamailio had just started to get SIP attacked and it looked like they were sending a bad header and Kamailio died.. Nov 20 19:48:51 sipproxy sip[3383]: INFO: <script>: [R-MAIN =>] Request method OPTIONS is to our domain (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3378]: INFO: <script>: [ROUTE-1] Received registration from sip:1368250498@sipproxy (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3378]: INFO: <script>: [ROUTE-1 !] Credentials invalid; issuing challenge Nov 20 19:48:51 sipproxy sip[3381]: INFO: <script>: [ROUTE-1] Received registration from sip:100@sipproxy (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3381]: INFO: <script>: [ROUTE-1 !] Credentials invalid; issuing challenge Nov 20 19:48:51 sipproxy sip[3374]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.401446] kamailio[3374]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3373]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.636774] kamailio[3373]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3388]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.680305] kamailio[3388]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3390]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.710444] kamailio[3390]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3383]: INFO: <script>: [ROUTE-1] Received registration from sip:2478223614@sipproxy (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3383]: INFO: <script>: [ROUTE-1 !] Credentials invalid; issuing challenge Nov 20 19:48:51 sipproxy sip[3378]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.753334] kamailio[3378]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3379]: INFO: <script>: [ROUTE-1] Received registration from sip:noauth@sipproxy (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3379]: INFO: <script>: [ROUTE-1 !] Credentials invalid; issuing challenge Nov 20 19:48:51 sipproxy sip[3375]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.768137] kamailio[3375]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3383]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.775628] kamailio[3383]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3386]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.790925] kamailio[3386]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3379]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.798926] kamailio[3379]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3381]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.815493] kamailio[3381]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:54 sipproxy sip[3344]: INFO: <core> [main.c:737]: INFO: terminating due to SIGCHLD Nov 20 19:48:54 sipproxy sip[3399]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3396]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3400]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3392]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3401]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3394]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3402]: INFO: <core> [main.c:788]: INFO: signal 15 received Thanks Brian

4 20

dispatcher - ds_mark_dst("i"); is inactive only few seconds
by Thomas Baumann 04 Feb '11

04 Feb '11

Dear sr-users, I use this straight forward dispatcher config. In the failure route I am calling this function: ds_mark_dst("i"); Also this action is logged: Feb 2 17:20:31 node5 /openser/sbin/kamailio[15945]: INFO: : FAIL_ONE:time [Wed Feb 2 17:20:31 2011] destination sip:10.12.19.31:5060 marked as inactive The gateway is inactive: [root@node5 ~]# kamctl 'fifo' ds_list URI:: sip:10.12.19.31:5060 flag=I priority=0 attrs=test1=123;test2=456 URI:: sip:10.12.19.21:5060 flag=A priority=0 attrs=test1=123;test2=456 Few seconds later the gateway is active again (there is still now response to SIP-OPTIONS) [root@node5 ~]# kamctl 'fifo' ds_list URI:: sip:10.12.19.31:5060 flag=A priority=0 attrs=test1=123;test2=456 URI:: sip:10.12.19.21:5060 flag=A priority=0 attrs=test1=123;test2=456 Does anybody have a idea why ? regards, Thomas DB: mysql> select * from dispatcher -> ; +----+-------+----------------------+-------+----------+---------------------+-------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+----------------------+-------+----------+---------------------+-------------+ | 1 | 1 | sip:10.12.19.21:5060 | 0 | 0 | test1=123;test2=456 | AS 1 | | 2 | 1 | sip:10.12.19.31:5060 | 0 | 0 | test1=123;test2=456 | AS 2 | +----+-------+----------------------+-------+----------+---------------------+-------------+ CONFIG: #!KAMAILIO ####### Defined Values ######### #!define WITH_MYSQL #!define WITH_IPAUTH # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@10.12.18.51/openser" #!endif debug=2 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ alias="m-lab-ca805-sig.kd-lab.de:5060" alias="m-lab-ca805-sig.kd-lab.de" alias="10.12.19.51" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:10.12.19.51:5060 listen=tcp:10.12.19.51:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 ####### Modules Section ######## # set paths to location of modules mpath="/usr/local/kamailio-3.1/lib/kamailio/modules_k/:/usr/local/kamailio-3.1/lib/kamailio/modules/" # ------------------ module loading ---------------------------------- loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "dialplan.so" loadmodule "avpops.so" loadmodule "dispatcher.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- dispatcher params ----- #modparam("dispatcher", "list_file", "/openser/etc/kamailio/dispatcher.list") modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshhold", 10) modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=4") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_from", "sip:lb1@m-lab-ca805-sig.kd-lab.de") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; #!ifdef WITH_IPAUTH if (!allow_source_address()) { sl_send_reply("403", "Forbidden"); exit; } #!endif # Send to application servers for number evaluation if ($rU=~"^0[1-9][0-9]{5,20}$") { ds_select_dst("1","0"); if (method=="INVITE") { t_on_failure("FAIL_ONE"); } t_relay(); exit(); } sl_send_reply("403", "Forbidden"); exit; } # Sample failure route failure_route[FAIL_ONE] { if (t_check_status("408")) { xlog("L_INFO", "FAIL_ONE:time [$Tf] destination $avp(dsdst) marked as inactive \n"); ds_mark_dst("i"); } exit; } ___________________________________________________________ NEU: FreePhone - kostenlos mobil telefonieren und surfen! Jetzt informieren: http://produkte.web.de/go/webdefreephone

2 5

Reloading a module without restart
by Stagg Shelton 04 Feb '11

04 Feb '11

Is it possible to reload a module such as siptrace without requiring a full restart of kamailio. I have changed my db_url modparam for the siptrace module and would like to avoid restarting kamailio if possible. Thanks Stagg

3 3

Problem with ser
by Asuncion Merino Rodriguez 02 Feb '11

02 Feb '11

Hello, i have insalled a ser server sucessfully. To try it, we was registered by a softphone, but when i am calling to the another registered user, he hang the phone but we cant ear anything. Can somebody help me? Thanks

1 0

Redundancy between 2 Kamailio servers
by Danny Dias 02 Feb '11

02 Feb '11

Hello my friends, I wonder if someone has done this before? i would like to implement 2 Kamailio servers with redundancy, something like the following: SERVER_A is working as the primary sip proxy (virtual IP as the sip signaling), if it fails, the other server (sleeping) should UP the virtual IP and takes all the traffic....i guess there is many Linux implementations thah can do this, but i would like to know if there is someone that has done this before and hear some recomendations... I've read some difficulty in the synchronisation of registrations because Kamailio works best when it stores registrations in memory and registrations are constantly changing - they expire and are renewed, as well as new ones joining and old ones leaving. To make the failover solution function seamlessly, it is necessary to synchronise the in-memory registrations between the primary and the backup server . This can be done by forking a copy of the registration request to the backup server, but there are some practical problems in doing this, has anyone do something with this? Thanks in advance!

6 21

Weird IPv6 Registration Issue
by Jon Farmer 02 Feb '11

02 Feb '11

Hi I am experimenting with a dual homed (IPv4 and IPv6) Openser 1.1.0-tls server. If I register using a IPv6 address then the initial registration works and the details are saved in the location table. However when the UA trys to re-register on expiry the SIP returns a 200 OK but the location data is not updated in the table. Looking at the syslog it is reporting that it didn't save the data as there was an existing entry on index with key 1. This is referring I presume to the primary index on "username, domain, contact) on the location table. So what happens is the first location entry expires and gets deleted from the table and thus there is no record for the subsequent register and thus openser thinks the UA has dropped off until the next re-register when because there is no longer a location record the data gets saved correctly. This doesn't happen if I register the same UA on IPv4 on the same openser server. Anyone got any ideas why this is happening? Regards Jon -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

3 2

ACC error | failed to insert into database
by alex pappas 02 Feb '11

02 Feb '11

Hi all, Hve anyone seen before the following error? *Kamailio acc [acc.c:398]: failed to insert into database* After a Kamailio restart it is ok but it start again afetr x time. Thank you Alex

2 2

Re: [SR-Users] t_local_replied() ?
by Bernhard Suttner 01 Feb '11

01 Feb '11

argh. I just mixed up the true/false return values in the eMail but you still got it :-) Thanks a lot! ----- Original Message ----- From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org Sent: Tue, 01 Feb 2011 20:31:17 +0100 Subject: Re: [SR-Users] t_local_replied() ? > Hello, > > not sure if you ask about the options, or you tried them and don't give > you the needed feature, since there are some improper true/false return > codes in your email. t_branch_replied() will return false if the 408 is > generated locally. > > Cheers, > Daniel > > On 2/1/11 8:25 PM, Bernhard Suttner wrote: > > Hi, > > > > just to be sure: > > > > - If the gateway does send back a 100 Trying and then a 408 is detected > within failure_route the method t_branch_replied does return false (means: > gateway is up and running) - dont go to next gateway (dispatcher) > > > > - If the gateway does not respond and a 408 is detected within > failure_route (= 408 was generated from kamailio) t_branch_replied does > return true (means: gateway is down) - go to the next gateway (dispatcher) > > > > Is that correct or am I wrong? > > > > Best regards, > > Bernhard > > > > > > > > ----- Original Message ----- > > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > > To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] > > Cc: sr-users(a)lists.sip-router.org > > Sent: Tue, 01 Feb 2011 13:30:54 +0100 > > Subject: Re: [SR-Users] t_local_replied() ? > > > > > >> Hello, > >> > >> On 2/1/11 12:08 PM, Bernhard Suttner wrote: > >>> Hi, > >>> > >>> I am using the dispatcher module and want to check within the > >> failure_route if the 408 was internally generated from kamailio or it was > >> received from the dispatcher gateway. > >>> There was previously a function called t_local_replied() in the > TM-module > >> but I could not find this function in the current documentation. Was it > >> removed? > >>> Is there a alternative to check if the 408 was local generated or if it > >> was received from the peer (= from the dispatcher gateway)? > >> see the example of: > >> http://kamailio.org/docs/modules/stable/modules/tm.html#t_branch_replied > >> > >> Cheers, > >> Daniel > >> > >> -- > >> Daniel-Constantin Mierla > >> http://www.asipto.com > >> > >> > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla > http://www.asipto.com > >

1 0

Re: [SR-Users] t_local_replied() ?
by Bernhard Suttner 01 Feb '11

01 Feb '11

Hi, just to be sure: - If the gateway does send back a 100 Trying and then a 408 is detected within failure_route the method t_branch_replied does return false (means: gateway is up and running) - dont go to next gateway (dispatcher) - If the gateway does not respond and a 408 is detected within failure_route (= 408 was generated from kamailio) t_branch_replied does return true (means: gateway is down) - go to the next gateway (dispatcher) Is that correct or am I wrong? Best regards, Bernhard ----- Original Message ----- From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org Sent: Tue, 01 Feb 2011 13:30:54 +0100 Subject: Re: [SR-Users] t_local_replied() ? > Hello, > > On 2/1/11 12:08 PM, Bernhard Suttner wrote: > > Hi, > > > > I am using the dispatcher module and want to check within the > failure_route if the 408 was internally generated from kamailio or it was > received from the dispatcher gateway. > > > > There was previously a function called t_local_replied() in the TM-module > but I could not find this function in the current documentation. Was it > removed? > > > > Is there a alternative to check if the 408 was local generated or if it > was received from the peer (= from the dispatcher gateway)? > see the example of: > http://kamailio.org/docs/modules/stable/modules/tm.html#t_branch_replied > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > >

2 1

Radius and kamailio: avp unknown attribute
by Dominguez Jover, Ricardo 01 Feb '11

01 Feb '11

Hi everybody, I have a problem with Kamalio and Radius, when a try to authenticate a user I get the following: Feb 1 11:50:48 kamailiodes /usr/local/kamailio-3.1/sbin/kamailio[5941]: rc_avpair_new: unknown attribute 5 Feb 1 11:50:48 kamailiodes /usr/local/kamailio-3.1/sbin/kamailio[5941]: ERROR: auth_radius [sterman.c:412]: authorization failed In fact I can't see any packet sended to the Radius Server from Kamailio. I've read several posts and I've chmod of /var/run/radius.seq, include dictionary.sip in dictionary.radius and so on. Changing "if (!radius_www_authorize("$td"))" with " if (!radius_www_authorize(""))" I don't get any error but and can't neither see any packet sended to the Radius Server. I've checked the connection between radius client and radius server and it's OK, I get: Received response ID 91, code 2, length = 211 Any help would be appreciated. Kind regards, Ricardo Dominguez

2 2

Trunk connection asterisk-kamailio
by Mino Haluz 01 Feb '11

01 Feb '11

Hi, I have a question about kamailio-asterik interconnection. I'd like to connect 1000 numbers with a trunk, but it would be painful to add 1000 trunks on asterisk. Do you have some idea how could I group those numbers into one trunk connected to kamailio? asterisk and kamailio configuration settings. Thank you very much.

2 1

t_local_replied() ?
by Bernhard Suttner 01 Feb '11

01 Feb '11

Hi, I am using the dispatcher module and want to check within the failure_route if the 408 was internally generated from kamailio or it was received from the dispatcher gateway. There was previously a function called t_local_replied() in the TM-module but I could not find this function in the current documentation. Was it removed? Is there a alternative to check if the 408 was local generated or if it was received from the peer (= from the dispatcher gateway)? Thanks in advance! Best regards, Bernhard

2 1

kamailio and sip-router at FOSDEM 2011 - social event
by Henning Westerholt 31 Jan '11

31 Jan '11

Hello, in about two and a half week the annual open source developer conference FOSDEM (http://fosdem.org/2011/) takes place in Brussels. As in the last two years we would like to meet here for a social event, probably a dinner on Saturday evening, 6th February. This developers already confirmed their participance: - Daniel-Constantin Mierla - Marius Zbihlei - Henning Westerholt This year we'll also be present in the Open Source Telephony room on sunday with two talks, "Scaling location services in large SIP networks with Kamailio from Marius and myself, and "Unifying SIP and Web worlds with Lua (Kamailio)" from Daniel. Olivier Taylor agreed helping us again in choosing a nice place for the evening. Participation at the event is free of charge, everybody pays for its own expenses. In order to organize a restaurant and do a reservation please send me a short mail if you would like to attend. Looking forward to see you in Brussels! Best regards, Henning P.S.: Photos of the 2009 event can be found on the web site: http://www.kamailio.org/events/2009-FOSDEM/FOSDEM2009-Album/ -- Henning Westerholt - Head of IT Operations Internet Access & Communications 1&1 Internet AG, Karlsruhe, Germany

2 3

xlogl and pseudo variables
by Bernhard Suttner 29 Jan '11

29 Jan '11

Hi, I am using xlogl for debugging and logging the most important actions within the kamailio script. Does it take much resources if I use the pseudo variables with the xlogl statements? Example: xlogl("L_NOTICE", "Route $rm to destination! F=$fu T=$tu IP=$si \n"); I´m planning to do this in all the xlogl statements to be able to debug / "see whats going on" on the system. Thanks in advance. Best regard, Bernhard

2 1

Re: [SR-Users] onreply_route and failure_route
by Bernhard Suttner 29 Jan '11

29 Jan '11

Hi, that was _really_ fast. Thanks. :-) One more question: Could I somehow test with the route[NAT] for example, if the request was from the main route { ... } or from the onreply_route[] ? Best regards, Bernhard ----- Original Message ----- From: Alex Balashov [mailto:abalashov@evaristesys.com] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org [mailto:sr-users@lists.sip-router.org] Sent: Sat, 29 Jan 2011 21:37:34 +0100 Subject: Re: [SR-Users] onreply_route and failure_route > Yes. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > On Jan 29, 2011, at 3:35 PM, "Bernhard Suttner" <bernhard.suttner(a)winet.ch > > wrote: > > > Hi, > > > > is it possible to call a route within onreply_ and failure_route? I > > mean, within the main route { ... } its possible to call the route > > [NAT]. Is this also possible for this "special" routes? If this is > > possible, is it also allowed to call a route which will normally > > called from the main route { ... } > > > > Thanks in advance. > > > > Best regards, > > Bernhard > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > > list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

1 0

onreply_route and failure_route
by Bernhard Suttner 29 Jan '11

29 Jan '11

Hi, is it possible to call a route within onreply_ and failure_route? I mean, within the main route { ... } its possible to call the route[NAT]. Is this also possible for this "special" routes? If this is possible, is it also allowed to call a route which will normally called from the main route { ... } Thanks in advance. Best regards, Bernhard

2 1

Joining strings
by Lee Archer 29 Jan '11

29 Jan '11

Hi, is it possible to join strings? I'd like to prepend a number to a string prior to processing it to an int? Thanks Lee

2 2

Can you use a bigint with Kamailio?
by Lee Archer 29 Jan '11

29 Jan '11

Hi, I am trying to run dispatcher based on the inbound number called. This works fine for some numbers but not all. Is there a bigint string function as this would solve my problems? The numbers I am receiving are 14 digits long which s.int is returning as something completely different to what it should be. Thanks Lee

2 2

LCR next_gw() - certain GW, source from other IP address
by Graham Wooden 29 Jan '11

29 Jan '11

Hi there, I have a requirement now that for one of my GW¹s in my LCR, needs to be sourced from another IP address (two different rate-decks with the carrier). This ratedeck will always be at least 2nd inline ... I was thinking something along the sorts ... Some pseudo code: failure_route [2] { . . if (next_gw()) { if ( name = 'gw_name') { force_send_socket(udp:ip.address.of.certain.ip:5060); } else { # just dummy, as we will send out our default IP for everyone else. } t_on_failure("2"); route(19); } . . . } What variable will have the value that's in the gw_name column? I can't go by the IP address in $ru because the destination IP is the same between the two ratedecks. Overall, Is that the best way to handle this requirement? Thanks all, -graham

3 2

Re: [SR-Users] LCR next_gw() - certain GW, source from other IP address
by GP Wooden 29 Jan '11

29 Jan '11

Same destination IP for both rate decks, so I need to go by something else, like the gw_name. ----- Reply message ----- From: "Stagg Shelton" <stagg(a)vocalcloud.com> Date: Sat, Jan 29, 2011 8:27 am Subject: [SR-Users] LCR next_gw() - certain GW, source from other IP address To: <sr-users(a)lists.sip-router.org> Do the provider gateways have the same IP address for the different rate plans? If the gateways are different then the following may be relevant. if (next_gw()) { # prepare for lcr failover t_on_failure("2"); if($rd=="aaa.aaa.aaa.aaa") { xlog ("L_INFO", "This is a request for Some Rate Plan"); force_send_socket(udp:xxx.xxx.xxx.xxx:5060); } else { xlog ("L_INFO", "This is a request for another rate plan"); force_send_socket(udp:yyy.yyy.yyy.yyy:5060); }; route(2); } else { xlog("No gateways found!"); Stagg On 1/29/11 9:10 AM, Graham Wooden wrote: > Hi there, > > I have a requirement now that for one of my GW¹s in my LCR, needs to be > sourced from another IP address (two different rate-decks with the carrier). > This ratedeck will always be at least 2nd inline ... > > I was thinking something along the sorts ... Some pseudo code: > > failure_route [2] { > . > . > if (next_gw()) { > if ( name = 'gw_name') { > force_send_socket(udp:ip.address.of.certain.ip:5060); > } else { > # just dummy, as we will send out our default IP for everyone > else. > } > t_on_failure("2"); > route(19); > } > . > . > . > } > > What variable will have the value that's in the gw_name column? I can't go > by the IP address in $ru because the destination IP is the same between the > two ratedecks. > > Overall, Is that the best way to handle this requirement? > > Thanks all, > > -graham > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

3 4

Kamailio Devel Training, Barcelona - update
by Daniel-Constantin Mierla 28 Jan '11

28 Jan '11

Hello, a quick update about the next training in Barcelona, Feb 10-11, 2011, focusing on Kamailio development. We got a bigger room (for second time :-) ), so we increased the capacity to 28 seats. We are already 23 people in the class and several more joining the dinner Thursday evening, so there is going to be a great time. If you plan to come, register asap, there is no time to get a bigger room, so once we are full booked this time, we close the participants list. You can see more details at: http://www.kamailio.org/w/2011/01/kamailio-development-training-barcelona-f… Also, do not forget Fosdem next weekend, it is a bunch of us going there, two talks about Kamailio, come and say hi: http://www.kamailio.org/w/2011/01/social-networking-event-brussels-belgium/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Kamailio as a routing engine
by Gang Liu 28 Jan '11

28 Jan '11

Hello, I wrote a H.323/SIP IWF program before, it was based on B2BUA framework. Currently I am planning to use Kamailio as internal routing server and let IWF become a class 4 soft switch.That is Inbound ---- H.323/ SIP -- > ---- SIP routing request-----> Kamailio IWF/SBC | Outbound <--- H.323/SIP -- <--- SIP ----------------------------- Is it ok to store all voip provider's information at database of Kamailio and pass all to IWF at private SIP headers and let IWF as a outbound proxy? Is Kamailio as a SIP redirect server or it is better to stay at call signaling path until call session ended? It is great to share any information about this. regards, Gang

2 1

Load_gw failed
by Thiago Maluf 27 Jan '11

27 Jan '11

Hi Kamailio Users, I'm trying to use LCR module in Kamailio 3.1.1 and I', having some problems with this function (load_gws). I'm my kamailio.cfg , I have this route: # SEARCH DEST route[NUMBER] { xlog("L_ALERT", "NUMBER IS $rU \n"); if ($rU=~"^55211020[0-9]{4}$") { # IS MY USERS route(LDAP_NUM2USER); } else { # NOT IS MY USER if (!load_gws(1,$rU)) { xlog("L_ALERT", "LOAD LOCAL GW FAILED $rU \n"); route(DSER); return; } else { xlog("L_ALERT", "LOAD GW DEBUG $avp(i:709)\n"); } ; if (!next_gw()) { xlog("L_ALERT", "NEXT LOCAL GW FAILED\n"); sl_send_reply("503", "Service not available - No gateways"); return; }; setflag(29); route(RELAY); return; }; }; When my call arrive to load_gws, I have this problem: (2198) ERROR: <core> [action.c:1158]: runtime fixup failed for load_gws param 2 (2198) ERROR: <core> [action.c:1511]: run action error at: /etc/kamailio/kamailio.cfg:472 I already saw the Kamailio Code and this error doesn't make sense. Would somebody Help me with it? Very thanks, Thiago Maluf. -- ---------------------------------------------------------------- Thiago Maluf Resende Tel: +55 21 9700-9113 e-mail: malufrj(a)gmail.com

2 1

SIP Recorder
by Danny Dias 27 Jan '11

27 Jan '11

Hello my friends, I have a requeriment, which indicates that i have to record every SIP conversation between peers (also for callings to the PSTN); the recording server will be built for our company following this requeriments (also requested for the client): My doubt is: How can i handle sip conversations recording when all the calls are passing through a Proxy Server? I do understand that the media is always peer to peer and the signaling goes through the Proxy, but in this case the media not only has to pass between the peers because it must be recorded. How should i handle this?

7 19

Kamailio CFG file size ??
by Jijo 27 Jan '11

27 Jan '11

Hi All, is there any limitation on the size of kamailo.cfg file. When i tried to add few avp_subst, then the kamailio starts showing up the error out of package memory on receiving SIP message. If i remove those lines then everything works fine.. avp_subst is simple one. Our kamailio cfg file is quite big.. Please let me know how to debug this issue? Here is the error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [db_row.c:116]: no private memory left 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_res.c:293]: could not allocate row 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_res.c:258]: failed to convert row #1 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_res.c:75]: failed to convert rows 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_dbase.c:416]: error while converting result 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [db_query.c:137]: error while storing result 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: avpops [avpops_db.c:333]: cannot do the query 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: maxfwd [mf_funcs.c:62]: parsing MAX_FORWARD header failed! 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: rr [loose.c:98]: failed to parse headers 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: siputils [checks.c:64]: To parsing failed 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [action.c:831]: ERROR: do_action: memory allocation failure 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [action.c:1511]: run action error at: :0 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: pv [pv_core.c:1761]: do action failed 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [route.c:1846]: ERROR: eval_elem: bad or missing From: header 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [msg_translator.c:2229]: ERROR: build_res_buf_from_sip_req: alas, parse_headers failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: rr [loose.c:70]: failed to parse To header field 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: siputils [checks.c:64]: To parsing failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [action.c:831]: ERROR: do_action: memory allocation failure 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [action.c:1511]: run action error at: :0 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1740]: do action failed) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1381]: error parsing headers 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [data_lump.c:318]: ERROR: del_lump: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: textops [textops.c:1081]: no memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [route.c:1846]: ERROR: eval_elem: bad or missing From: header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [backLinkMode] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:418]: cannot parse From header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:400]: cannot parse To header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [temp] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [node] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [binding] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_shv.c:367]: no more pkg mem 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:418]: cannot parse From header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:400]: cannot parse To header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [temp] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [node] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [binding] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:418]: cannot parse From header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:1257]: osb_get_osv_current_binding(): no binding available 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <script>: R1 - binding IP=0 not available

2 3

NAT Problem in Asterisk Realtime + Kamailio config
by andrzej.ciupek＠asterisk.edu.pl 27 Jan '11

27 Jan '11

Hello I have started to use configuration like: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb I works fine, but I have problem with dealing with NAT. kamctl ul show gives me: AOR:: andrzej_104 Contact:: sip:andrzej_104@10.5.0.94:14368;rinstance=9c2b256a13210aa7 Q= Expires:: 46 Callid:: ZDUwYzIzMDNiZTI3MjkyNmFmOWEyOTI2ODcyNjVmNDg. Cseq:: 60 User-agent:: eyeBeam release 1102q stamp 51814 State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp:77.xx.yyy.zzz:5060 Methods:: 5087 AOR:: andrzej_103 Contact:: sip:andrzej_103@10.5.0.165:5060 Q= Expires:: 926 Callid:: 11631436-5060-1(a)10.5.0.165 Cseq:: 2257 User-agent:: Grandstream HT-502 V1.2A 1.0.3.10 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:77.xx.yyy.zzz:5060 Methods:: 6879 So when I have incomming call from Asterisk to Kamailio, Kamailio start to send call to LAN IP. I have tried to use help from: http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-… But I have never get: Received:: and Cflag:: 192 it is always Cflag:: 0. field in my kamctl ul show. In my configuration Asterisk is Callweaver and is on the other server. I use rtpproxy-1.2.1 and kamailio-3.0.1 and mofified config from link: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Please help.

2 2

RLS question
by "Andrés S. García Ruiz" 27 Jan '11

27 Jan '11

Hi, I'm trying to create a RLS "rls-services" document with a contacts-list inside it. Kamailio states that "No xcap document found for uri:x". When creating a subscribe I must set the uri to the service uri. What I want to know is the sql query Kamailio does in order to get the contacts list for that uri. For example: XCAP document: "<?xml version=\"1.0\" encoding=\"UTF-8\"?>" + "<rls-services xmlns=\"urn:ietf:params:xml:ns:rls-services\"" + " xmlns:rl=\"urn:ietf:params:xml:ns:resource-lists\">" + "<service uri=\"sip:restricted_areas_presentities@open-ims.test\">" + "<list name=\"friends\">" + "<rl:entry uri=\"sip:testuser01@open-ims.test\"/>" + "<rl:entry uri=\"sip:testuser02@open-ims.test\"/>" + "</list>" + "<packages>" + "<package>presence</package>" + "</packages>" + "</service>" + "</rls-services>"; Is stored in /xcap-root/rls-services/users/sip:restricted_areas@open-ims.test/index The subscribe is from sip:restricted_areas@open-ims.test to sip:restricted_areas_presentities@open-ims.test What am I doing wrong? Thanks, Andrés.

3 5

[kamailio-users] kamailio on amazon
by Chandrakant Solanki 26 Jan '11

26 Jan '11

Hi I have installed Kamailio with MediaProxy and asterisk on Amazon Server.. While kamailio/MediaProxy and Asterisk both running on different amazon's instance. Kamailio : 1.5.0-notls MediaProxy : 2.3.8 Asterisk : 1.6.2.6 Firewall port is opened for mediaproxy from 10000-20000 (UDP), 5060 (UDP) etc on both amazon machine.. while I tried to play an audio... it plays sound file but unable to found audio on device. when I put tcpdump on asterisk machine it gives following error... # tcpdump -i eth0 udp portrange 10000-20000 -w test1.pcap 01:39:52.850279 IP (tos 0xb8, ttl 64, id 59, offset 0, flags [DF], proto: UDP (17), length: 60) ip-W.X.Y.Z.compute.internal.15246 > xyz.com.10010: [bad udp cksum 3cfd!] UDP, length 32 01:39:52.870279 IP (tos 0xb8, ttl 64, id 60, offset 0, flags [DF], proto: UDP (17), length: 60) ip-W.X.Y.Z.compute.internal.15246 > xyz.com.10010: [bad udp cksum 97fe!] UDP, length 32 01:39:52.890281 IP (tos 0xb8, ttl 64, id 61, offset 0, flags [DF], proto: UDP (17), length: 60) ip-W.X.Y.Z.compute.internal.15246 > xyz.com.10010: [bad udp cksum 75c0!] UDP, length 32 Any Idea..!! -- Regards, Chandrakant Solanki

3 2

Re: [SR-Users] fix_nated_sdp()
by Bernhard Suttner 25 Jan '11

25 Jan '11

Hi, thanks for your response. I think I understand the scenario very good - but im not so familiar with the fix_nated_sdp() functionality. Currently the SDP will be modified with the source IP of the message. This does work good but currently I only do that for some special devices (matching the User-Agent / Server string). I want to do that fix_nated_sdp() in all cases and not only for some special devices. The FreeSWITCH RTP-Auto Adjustment does work in nearly every case but in one (currently known) scenario it does not. Maybe I should just give it a try to do fix_nated_sdp() in combination with nat_uac_test (RFC1918 private address in SDP). Thanks a lot for this discussion. Best Regards, Bernhard ----- Original Message ----- From: Ovidiu Sas [mailto:osas@voipembedded.com] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: Carsten Bock [mailto:lists@bock.info], sr-users(a)lists.sip-router.org Sent: Tue, 25 Jan 2011 20:38:14 +0100 Subject: Re: [SR-Users] fix_nated_sdp() > You need to experiment with nathelper module and while doing that, you > need to inspect the SIP messages in order to understand how it works. > Only when you fully understand your scenarios, you will be able to > properly configure, debg and setup kamailio. > > Best thing to do here is to use rtpproxy and relay media if needed. > > Another option would be to let the SDP untouched and let freeswitch do > it's magic (if it can). Not sure if this will work in all scenarios. > > Calling fix_nated_sdp() on the SBC will definitely not work (you are > just messing with the IP on the SDP and you don't know if the real RTP > port on the NATed side will match the advertised port in your "fixed" > SDP). > > > Regards, > Ovidiu Sas > > On Tue, Jan 25, 2011 at 1:23 PM, Bernhard Suttner > <bernhard.suttner(a)winet.ch> wrote: > > Hi, > > > > thanks for your answer. > > > > The question was about, if the fix_nated_sdp() (in the example below) > could maybe break something. I am sure, that the fix_nated_sdp() would work > in "some" cases but I am not sure, if the function could break something: > > > > > > if (method==INVITE && has_sdp() && nat_uac_test(8)) fix_nated_sdp(3) > > > > and within the onreply route: > > > > if (status=~(180|183|200) && has_sdp() && nat_uac_test(8)) > fix_nated_sdp(3) > > > > > > Or is there a better solution to fix these damend NAT scenarios? > > > > Best regards, > > Bernhard > > > > > > -----Ursprüngliche Nachricht----- > > Von: kaiserbock2(a)googlemail.com [mailto:kaiserbock2@googlemail.com] Im > Auftrag von Carsten Bock > > Gesendet: Dienstag, 25. Januar 2011 18:56 > > An: Bernhard Suttner > > Cc: sr-users(a)lists.sip-router.org > > Betreff: Re: [SR-Users] fix_nated_sdp() > > > > Hi Bernhard, > > > > have you checked the SDP which is going to and from the > > Freeswitch-Server? Is it modified? > > > > If yes: You may want to check your NAT-Firewall. A working (but not > > nice) solution might be to relay the RTP through an proxy in this > > case... > > If No: Verify the result of "nat_uac_test": Why is it returning "false"? > > > > Carsten > > > > 2011/1/25 Bernhard Suttner <bernhard.suttner(a)winet.ch>: > >> Hi, > >> > >> someone has an idea? > >> > >> Thanks in advance! > >> > >> Best regards, > >> Bernhard > >> > >> -----Ursprüngliche Nachricht----- > >> Von: sr-users-bounces(a)lists.sip-router.org > [mailto:sr-users-bounces@lists.sip-router.org] Im Auftrag von Bernhard > Suttner > >> Gesendet: Montag, 24. Januar 2011 20:38 > >> An: sr-users(a)lists.sip-router.org > >> Betreff: [SR-Users] fix_nated_sdp() > >> > >> Hi, > >> > >> I have a question about the fix_nated_sdp() function. Different devices > (phones, pbxes, gateways) are using Kamailio as SBC. The SBC does t_relay() > the message to a pool of FreeSWITCH Servers. > >> > >> The scenario does work quite well but: > >> > >> if a call is initiated from one of the freeswitch server, pass it through > kamailio to the PBX and the PBX does have a forwarding back to kamailio and > in the end to freeswitch I have no audio because the PBX is behind a NAT and > uses private ip addresses in all the SDP of 200 OK and 183 Session Progress. > Therefore FreeSWITCH could not do the (very cool) RTP Auto-Adjustment (check > if IP in incoming packet is different to the IP in the SDP) to detect the > real IP address. > >> > >> Therefore I have to use fix_nated_sdp(). I am not sure, what this > function could break therefore I want to be very sure and use it only, if > really necessary. Test which User-Agent or Server the peer (in this example > the PBX) has, is not really possible because some devices does not send this > header. > >> > >> What do you think about the follwoing pseudo code: > >> > >> if (method==INVITE && has_sdp() && nat_uac_test(8)) fix_nated_sdp(3) > >> > >> and within the onreply route: > >> > >> if (status=~(180|183|200) && has_sdp() && nat_uac_test(8)) > fix_nated_sdp(3) > >> > >> Somebody has a better idea to handle this? What could go wrong? > >> > >> Thanks in advance. > >> > >> Best regards, > >> Bernhard Suttner > >> > >> _______________________________________________ > >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > >> sr-users(a)lists.sip-router.org > >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >> > >> > >> _______________________________________________ > >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > >> sr-users(a)lists.sip-router.org > >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >> > > > > > > > > -- > > Carsten Bock > > Schomburgstr. 80 > > 22767 Hamburg > > Germany > > > > Mobile +49 179 2021244 > > Home +49 40 34927217 > > Büro (Verl) +49 5246 801427 > > Fax +49 40 34927218 > > mailto:carsten@bock.info > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > >

1 0

fix_nated_sdp()
by Bernhard Suttner 25 Jan '11

25 Jan '11

Hi, I have a question about the fix_nated_sdp() function. Different devices (phones, pbxes, gateways) are using Kamailio as SBC. The SBC does t_relay() the message to a pool of FreeSWITCH Servers. The scenario does work quite well but: if a call is initiated from one of the freeswitch server, pass it through kamailio to the PBX and the PBX does have a forwarding back to kamailio and in the end to freeswitch I have no audio because the PBX is behind a NAT and uses private ip addresses in all the SDP of 200 OK and 183 Session Progress. Therefore FreeSWITCH could not do the (very cool) RTP Auto-Adjustment (check if IP in incoming packet is different to the IP in the SDP) to detect the real IP address. Therefore I have to use fix_nated_sdp(). I am not sure, what this function could break therefore I want to be very sure and use it only, if really necessary. Test which User-Agent or Server the peer (in this example the PBX) has, is not really possible because some devices does not send this header. What do you think about the follwoing pseudo code: if (method==INVITE && has_sdp() && nat_uac_test(8)) fix_nated_sdp(3) and within the onreply route: if (status=~(180|183|200) && has_sdp() && nat_uac_test(8)) fix_nated_sdp(3) Somebody has a better idea to handle this? What could go wrong? Thanks in advance. Best regards, Bernhard Suttner

3 4

accounting serial forked transactions with 302 from LCR
by thrillerbee 25 Jan '11

25 Jan '11

I'm converting my OpenSIPS routers to Kamailio & have run into a small complication. The proxy pushes all INVITEs to a least-cost router. This LCR responds with a list of routes as contact instances in a 302 Redirect. Calls are routing a serially forking normally. Connected & failed calls account normally. However, if the caller cancels the call, the acc module includes the 302 in the transaction record as the final response as opposed to the actual final response - the 487 Request Canceled. Is there something I could be missing that would cause this? Thanks, Ryan

2 4

Re: [SR-Users] $ua in on_reply route not set?
by Bernhard Suttner 25 Jan '11

25 Jan '11

Hi, found the problem. The device does sometimes use User-Agent and sometimes Server. Is it better to use $hdr() or the Search() function? I use this to check for the User-Agent and then to do a "fix_nated_sdp()" (in route[] and onreply[]) because I am not really sure, if the fix_nated_sdp() could break something. Or should kamailio break nothing here? Sometimes the User-Agent/Server is missing in Session-Progress 183. Therefore a "global" fix_nated_sdp() would be nice to have. Best regards, Bernhard ----- Original Message ----- From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org Sent: Mon, 24 Jan 2011 17:09:12 +0100 Subject: Re: [SR-Users] $ua in on_reply route not set? > Hello, > > On 1/24/11 5:03 PM, Bernhard Suttner wrote: > > Hi, > > > > could it be, that the $ua pseudo variable is not set within in a onreply > route? (Version 3.1). > no, should be set, there was no change in this regard for quite long > time. Can you sent the sip reply plus log with debug=3? > > > > What is the best alternative for that? Search()? > > The alternative is $hdr(User-Agent) which is practically returning the > same value as $ua, using generic header search mechanism. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > >

2 1

Re: [SR-Users] pdb module best practice
by Henning Westerholt 25 Jan '11

25 Jan '11

On Monday 24 January 2011, Thomas Baumann wrote: > the downtime is not a problem, I already have 2 deamons running. > I want to reduce the time , how changes become active. > > 1. update xxx entry > 2. export table to *.csv (5s) > 3. generate *.mmap file (25 seconds) > > So changes will take at lease 30 seconds to become active. Only 20% of my > numbers are updated very often, I want to export them and the generation > of the *.mmap file takes only 5 seconds. > > But in this case I need to specify the daemons per request and not in the > general parameters. Because later on it might be necessary to ask the > daemons hosting the rest of my numbers. Or maybe 30 seconds are ok ? Hi Thomas, i understand. You could of course change/ extend the interface of the pdb, but at the moment this functionality is not available. During module startup the sockets and server data structures are initialized, so the structure needs probably a bit changed. But let me ask, what kind of changes are done to your routing? Is the routing data then completely invalid, or just a more expensive route is taken for this particular target? In the latter case a update period of 30 seconds sounds not like a big deal to me, after all you probably have also some manual steps involved before you do an update. Cheers, Henning

1 0

Re: [SR-Users] SIP between IPv4 and IPv6 agents
by Komáromi Péter 25 Jan '11

25 Jan '11

Hello! I went through your letters and start as following: ------------------------------------------------------- root@ubuntu:/home/yeti/rtpproxy# /etc/init.d/kamailio restart Restarting kamailio: kamailioloading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ Listening on udp: 192.168.124.158:5060 udp: FEC0:0:0:0:0:0:0:2:5060 tcp: 192.168.124.158:5060 tcp: FEC0:0:0:0:0:0:0:2:5060 Aliases: tcp: ubuntu:5060 udp: ubuntu:5060 root@ubuntu:/home/yeti/rtpproxy# kamctl restart INFO: Stopping Kamailio : INFO: stopped INFO: Starting Kamailio : INFO: started (pid: 7597) root@ubuntu:/home/yeti/rtpproxy# rtpproxy restart ------------------------------------------------------- So I guess everything is starting fine. The users can register and call each other. Their data are in the mysql database`s location table. I put this into the kamailio.cfg file: -------------------------------------------------------- # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")) { if (af == inet) { if ( $(rU{s.substr,0,1}) == "6") { force_rtp_proxy("ie"); } else { force_rtp_proxy("ee"); } } else { if ( $(rU{s.substr,0,1}) == "4") { force_rtp_proxy("ei"); } else { force_rtp_proxy("ii"); } } } else if (is_method("200OK")) { if (af == inet) { if ( $(rU{s.substr,0,1}) == "6") { force_rtp_proxy("ei"); } else { force_rtp_proxy("ee"); } } else { if ( $(rU{s.substr,0,1}) == "4") { force_rtp_proxy("ie"); } else { force_rtp_proxy("ii"); } } } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } ---------------------------------------------------------- But, unfortunately there is no RTP traffic yet. :( What do you thing guys, what can be the problem? Regards, Peter Am 02.12.2010 13:06, schrieb Komáromi Péter: > Hi! > > So if you say it is possible to solve the problem with the only > location table, the location_inet4 and location_inet6 is not > certainly necessary... do I _have to_ use the 4to6.cfg file from the > source of kamailio, or not? When you call force_rtp_proxy() then you have to provide the i/e flags to tell rtpproxy how it should bridge the call. Therefore you need to know for every call: - is the caller IPv4 or IPv6? - is the callee IPv4 or IPv6? Finding out the protocol version of the caller is rather easy: if (af == inet) { # caller ipv4 ... } else { # caller ipv6 ... } But finding out if the callee is v4 or v6 is difficult. Actually it may even be that the callee is registered with 2 clients at the same time - one using v4 and v6. As far as I know this multi-registration scenario can not be handled currently - only single-registrations or multi-registrations which use the same protocol version are supported. Anyway - there are several approaches how to detect the IP version of the callee. - fixed mapping: for example if you know that all usernames starting with 1 are IPv4 clients, and all usernames starting with 2 are IPv6 clients, then you can detect the callee's protocol version manually, e.g: if ( $(rU{s.substr,0,1}) == "1") { # callee ipv4 ... } else { # callee ipv6 ... } - check destination URI if it contains a IPv4 or IPv6 address, e.g. use http://www.kamailio.org/dokuwiki/doku.php/transformations:devel#resubst_exp… to test if $dd contains '.' which would indicate a ipv4 address. - use dedicated location tables > (Because it uses them, for instance here: if (method == "REGISTER") > { if (af == inet) { save("location-inet4"); } else if (af == inet6) > { save("location-inet6");...) > > If I do not have to use it, I do not need to merge it with my > kamailio.cfg file, the steps I have to do are: install rtpproxy, > start it with the appropriate parameters and set the forcing in the > kamailio.cfg file?! The more I think about it I prefer the approach with separated location tables. It should be rather easy to integrate the 2 location tables into your existing config. regards Klaus

3 4

Problem reloading carrierroute
by Morten Isaksen 25 Jan '11

25 Jan '11

Hi, Kamailio 3.0.3. A few days ago we got a problem when "kamctl cr reload" was executed. Jan 24 08:50:27 sip-core-2 /usr/local/sbin/kamailio[1639]: ERROR: db_mysql [km_dbase.c:346]: no memory left Jan 24 08:50:27 sip-core-2 /usr/local/sbin/kamailio[1639]: ERROR: carrierroute [cr_db.c:331]: Fetching rows failed Jan 24 08:50:27 sip-core-2 /usr/local/sbin/kamailio[1639]: ERROR: carrierroute [cr_data.c:181]: could not load routing data I tried to increase the shared memory from 32M to 64M but that did not seem to help. It failed again after a few hours. The last thing I tried was to insert modparam("carrierroute", "fetch_rows", 2000) into kamailio.cfg and is waiting to see if that helped. The carrierroute table is about 16K rows. Is it private og shared memory that is missing? Any other ideas how to fix this? -- Morten Isaksen

2 3

Building specific commit
by mayamatakeshi 25 Jan '11

25 Jan '11

Hello, I am using these commands to build and install kamailio: git clone http://git.sip-router.org/sip-router kamailio cd kamailio git checkout -b 3.1 origin/3.1 make FLAVOUR=kamailio cfg make group_include="mysql" all make group_include="mysql" install Now, If I want to build a specific commit like a0d376d85aae2f6e46e381a01fb85c5830250d4d, what should I do? regards, takeshi

2 2

kamctl fifo <command> not responding
by Anton Roman 24 Jan '11

24 Jan '11

Hi all, I'm having trouble trying to execute fifo commands with "kamctl fifo <command>". Just after restarting Kamailio it works fine, however, sometimes after some days running it doesn't respond. kamailio1:~#* kamctl fifo which* It doesn't respond so I input *Crtl+c* and I get: /usr/local/lib/kamailio//kamctl/kamctl.fifo: line 89: /tmp/kamailio_fifo: Interrupted system call If I delete and create the fifo file again (with "rm /tmp/kamailio_fifo" and "mkfifo /tmp/kamailio_fifo" and "chmod 660 /tmp/kamailio_fifo") it keeps not responding. Any help is welcome, what can be happening? Below you can find info about the pipe and the running kamailio. Thanks in advance, Best regards Antón kamailio1:~# *ls -hall /tmp/kamailio_fifo * prw-rw---- 1 root root 0 ene 17 12:01 /tmp/kamailio_fifo After deleting and creating the fifo file again: kamailio1:~# *ls -hall /tmp/kamailio_fifo * prw-rw-r-- 1 root root 0 ene 18 10:28 /tmp/kamailio_fifo kamailio1:~#* ps -ef | grep kama* root 17369 17245 0 10:12 pts/0 00:00:00 grep kama kamailio 23277 1 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23289 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23291 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23293 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23294 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23295 23277 0 Jan15 ? 00:01:14 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23299 23277 0 Jan15 ? 00:01:13 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23300 23277 0 Jan15 ? 00:01:13 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23303 23277 0 Jan15 ? 00:01:13 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23305 23277 0 Jan15 ? 00:05:29 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23306 23277 0 Jan15 ? 00:05:33 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23309 23277 0 Jan15 ? 00:05:30 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23311 23277 0 Jan15 ? 00:05:31 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23312 23277 0 Jan15 ? 00:00:02 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23313 23277 0 Jan15 ? 00:00:39 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23315 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23320 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23321 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23322 23277 0 Jan15 ? 00:00:05 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23323 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio

2 5

$ua in on_reply route not set?
by Bernhard Suttner 24 Jan '11

24 Jan '11

Hi, could it be, that the $ua pseudo variable is not set within in a onreply route? (Version 3.1). What is the best alternative for that? Search()? Thanks in advance! Best regards, Bernhard

2 1

Multiple Sip
by aram amin 24 Jan '11

24 Jan '11

Dear Sir / Madam I registered for your services today and the main purpose was to register my other voip/sip providers with your services so that I can have more options in making out bound calls with my sip phone, however could you please explain to me that if for instance, I have three accounts with three different voip providers (i.e JustVoip, Smartvoip,and Voipcheap) ,then how do i register them with my account that you have opened for me today. could you also tell me how is it possible to make specific calls from one of the voip providers above . Thank you Regards Aram

2 1

pdb module best practice
by Thomas Baumann 24 Jan '11

24 Jan '11

Hello, I tested the pdb module and the pdb_server. It is a fast solution even with a huge amount of numbers. But how I can update the pdb_server in a fast and effective way ? I tried it in the following way: - number;carrier are stored in a mysql table (20 million entries) - 2 pdb_servers are running - the table is exported into a csv file (time: 5 seconds) - pdbt will create a new mmappable file (time: 25 seconds) - the pdb_servers will be restarted to activate the changes The time of 30 seconds is good but not fast enought. Some of my numbers are updated very often, to export them and create a new mmappable file will take only 3 seconds. I had the idea to split my numbers in a more or less static area and a dynamic area. What I want to do later is to have the pdb_server not static configured, I want to define the server within the request: 1. query in dynamic range: pdb_query("$rU", "$avp(i:1000)", "localhost:10001,localhost:10002"); if not found look into the static area: pdb_query("$rU", "$avp(i:1000)", "localhost:10003,localhost:10004"); Does it make sence, or has somebody a better idea how changes activated within 10s or less ? regards, Thomas ___________________________________________________________ Neu: WEB.DE De-Mail - Einfach wie E-Mail, sicher wie ein Brief! Jetzt De-Mail-Adresse reservieren: https://produkte.web.de/go/demail02

2 1

pdb module best practice
by Thomas Baumann 22 Jan '11

22 Jan '11

1 0

Problem with parallel forking
by Daniel Grotti 21 Jan '11

21 Jan '11

Hi all, I'm using kamailio 3.1 and I have some problems with parallel forking. I need to implement parallel forking to different users registered on kamailio. So, when call arrives with R-URI= sip:003912345678@<IP_server>, I need to fork te call to (for example) 2 users: 1001 and 1001. To do that, I've created my usr_preferences table like this: +----+------+------------------+--------+-----------+------+-------+---------------------+ |/ id | uuid | username | domain | attribute | type | value | /last_modified | +----+------+------------------+--------+-----------+------+-------+---------------------+ |/ 1 | | /003912345678/ | | fork | 0 | 1001 | | /|/ 2 | | /003912345678/ | | fork | 0 | 1002 | | /+----+------+------------------+--------+-----------+------+-------+---------------------+ and I've added a code to my kam.cfg like this: if (is_method("INVITE")) { xlog("L_INFO", "REQUEST Invite - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(avp_db_load("$ru/username","$avp(s:fork)")) { avp_pushto("$ru/username","$avp(s:fork)/g"); } if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); exit; } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); t_relay(); } } If 1001 and 1002 are registered everything works fine (1001 Ringing and 1002 ringing). Ifonly 1001 registered everything works fine (1001 ringing and 1002 is offline.). But when 1002 is registered and 1001 in offline, kamailio try to call 1001, find that it's offline ( and I get 404 User Offline") but no call to 1002 is attempted. What's wrong ? Regards, Daniel

2 2

Too many false positives?
by Jeremya 21 Jan '11

21 Jan '11

A large bunch of recent email on this list got consigned to the spam bin by my Thunderbird. This hasn't happened before. Is there some list change that triggered this? Regards Jeremy A

2 1

Proxying INVITE message without port information in R-URI
by Vikram Ragukumar 20 Jan '11

20 Jan '11

Hello, The diagram below shows the SIP message flow in our setup when Softphone A places a call to Softphone B. ----------- ----- ---------- ----- ----------- |Softphone A|-<->-|Proxy|-<->-|Sip Server|-<->-|Proxy|-<->-|Softphone B| ----------- ----- ---------- ----- ----------- IP: a.b.c.d IP: a.b.c.d Proxies shown in the diagram above are physically the same entity. For convenience of diagrammatic representation the figure above has been unwrapped. Both Softphone A and Softphone B (hereafter referred to as A,B) are behind a NAT and are registered to the Sip Server through the same proxy (IP: a.b.c.d). Proxy runs Kamailio v3.0.1. When A places a call to B, an INVITE is proxied to the Sip Server from A. The Sip Server now sends another INVITE to the proxy, however the R-URI of this INVITE does not contain the port number that the INVITE is supposed to be sent to. Hence the proxy sends the INVITE to the default SIP port 5060. This port is not open in the firewall hence Softphone B never receives the INVITE. My questions are 1) Is it mandatory for R-URI of the INVITE to contain the port number ? 2) If not, is there something that can be added to the config script in Kamailio so that the proxy forwards the INVITE to the appropriate port? (maybe by keeping track of the rcv port of the REGISTER message that Softphone B sends...) ? Thanks and Regards, Vikram. PS : The Sip Server in the above discussion is Voipswitch.

3 10

Re: [SR-Users] Refer Using UAC.
by Daniel-Constantin Mierla 20 Jan '11

20 Jan '11

private mails are simply ignored after first advise in this regard, please CC the mailing list always. If you read the config from the tutorial, you see how the invite is relayed to asterisk. Refer should go to asterisk in the same way if it is an out of dialog request, or follow record route/contact address for within dialog requests. Cheers, Daniel On 1/20/11 11:37 AM, David J. wrote: > could you point me to the docs? > > just use forward() or rewritehostport()? > > > > On 1/20/11 5:15 AM, Daniel-Constantin Mierla wrote: >> If asterisk is in the path of the call, then just forward the REFER >> to it, there is no need to generate a new one. >> >> Also, note that REFER is many times part of a dialog, uac_req_send() >> creates requests out of the dialog. >> >> Cheers, >> Daniel >> >> On 1/16/11 11:37 PM, David J. wrote: >>> I realize that kamailio is not a b2bua; >>> But because we are using Asterisk in the path; >>> >>> To extend the Asterisk Realtime Tutorial; >>> >>> I was wondering if I could do something like this... >>> >>> Kind of like how we use UAC to send a register to Asterisk; >>> Could we do the same and modify the method to use REFER instead? >>> >>> I know it is more complex; but I am not sure where to handle this case; >>> >>> Thanks for any pointers. >>> >>> if(is_method("REFER")){ >>> $var(rip) = $sel(cfg_get.asterisk.bindip); >>> $uac_req(method)="REFER"; >>> $uac_req(ruri)="sip:" + $var(rip) + ":" + >>> $sel(cfg_get.asterisk.bindport); >>> $uac_req(furi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $ft; >>> $uac_req(turi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $tt; >>> $uac_req(hdrs)="Contact: <sip:" + $au + "@" >>> + $sel(cfg_get.kamailio.bindip) >>> + ":" + >>> $sel(cfg_get.kamailio.bindport) + ">\r\n"; >>> if($sel(contact.expires) != $null) >>> $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + >>> $sel(contact.expires) + "\r\n"; >>> else >>> $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + >>> $hdr(Expires) + "\r\n"; >>> >>> >>> uac_req_send(); >>> >>> >>> } >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

1 0

SL local ACK found -> dropping it! error ...
by Sébastien Cramatte 20 Jan '11

20 Jan '11

Not sure how todo this SIP capture. Can you give me an hand and explain how can I get this trace. I can post my configuration if it's necesary. But I'use the default NAT setup that comes with Debian package. 2011/1/19 Alex Balashov <abalashov(a)evaristesys.com> You'd need to post the SIP capture, but it sounds like Kamailio is not > associating the ACK with any known transaction, perhaps because the ACK has > an unexpected RURI, Via branch parameter, or CSeq number. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

2 5

Handling call transfer in the Asterisk Realtime setup.
by David J. 20 Jan '11

20 Jan '11

I am trying to add support for call transfer in the Asterisk realtime tutorial on Asipto; I am not sure what I would have to do to get this feature working; Perhaps I have to handle "refer" messages; but I am not sure how I send that to Asterisk; Any advice would be greatly appreciated.

2 1

Refer Using UAC.
by David J. 20 Jan '11

20 Jan '11

I realize that kamailio is not a b2bua; But because we are using Asterisk in the path; To extend the Asterisk Realtime Tutorial; I was wondering if I could do something like this... Kind of like how we use UAC to send a register to Asterisk; Could we do the same and modify the method to use REFER instead? I know it is more complex; but I am not sure where to handle this case; Thanks for any pointers. if(is_method("REFER")){ $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REFER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $ft; $uac_req(turi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $tt; $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); }

2 1

Reconnection to db
by Javier Gallart 20 Jan '11

20 Jan '11

Hi list we have been testing the mtree module. We do periodical reloads of the mtree table with the "mi mt_reload" command through xmlrpc. It works fine except when the db is restarted. If we manually restart the db and then try the mt_reload commands we get this error: Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: ERROR: db_postgres [km_dbase.c:152]: 0xa6a490 PQsendQuery Error: server closed the connection unexpectedly#012#011This probably means the server terminated abnormally#012#011before or while processing the request.#012 Query: select partition,prefix,route from trees order by partition Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: ERROR: <core> [db_query.c:101]: error while submitting query Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: ERROR: mtree [mtree_mod.c:622]: Error while querying db Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: ERROR: mtree [mtree_mod.c:738]: cannot re-load info from database Is there a way to reconnect to the db before sending that command? Thanks in advance Regards Javi

3 15

avp_db_query() question
by 侯旭光 19 Jan '11

19 Jan '11

Hello avp_db_query(query[,dest]) can get a database query and store the results in the avps. But what if the results returns many rows,and how can I get all the results? How to set the [dest] parameter ? I've tried the method describered inhttp://www.kamailio.org/docs/avp_db_query.html,but it doesn't work. like below mysql<select mem_user from tgroup where grp_name='1234'; +----------+ | mem_user | +----------+ | 1013 | | 2013 | +----------+ 2 rows in set (0.00 sec) kamailio.cfg if(avp_db_query("select mem_user from tgroup where grp_name='1234'","$avp(name)")) { xlog("L_INFO","query results[1] :$avp(name[1])\n"); xlog("L_INFO","query results[2] :$avp(name[2])\n"); } syslog <INFO> query results[1] :<null> <INFO> query results[2] :<null> version: kamailio 3.0.2 MySQL 5.0 thank you very much!

2 1

SL local ACK found -> dropping it! error ...
by Sébastien Cramatte 19 Jan '11

19 Jan '11

Hello, I'm new with Kamailio. I've just setup on Debian Squeeze using mostly the example configuration. I've been able to route inbound / outbound call on local networks W/O NAT and over mobile 3G+NAT ... Unfortunateley I've tried over 3 wifi / local + NAT and Kamailio drop call and return me this message "SL local ACK found -> dropping it! error ..." My Kamailio server has public fixed IP and listen on 5060:udp It's look and issue with NAT setup ... How can we solve it ? Best regards

3 3

Return code after fr_inv_timer hit
by Mino Haluz 18 Jan '11

18 Jan '11

Hi, I would like to force kamailio to send another code as Request timeout when fr invite timeout is hit. Is there some nice way how to achieve it, or I have to edit the code ? :( Thank you!

3 7

ACK problem with FreeSwitch-Kamailio SBC implementation.
by Ricardo Martinez 18 Jan '11

18 Jan '11

Hello. I was wondering if someone can help me with some problem that I’m having with the combination of FreeSwitch and Kamailio as SBC. I’m having and ACK not routed by kamailio and dropped. Here is the debug. 100.100.100.211:5060 : Kamailio 100.100.100.211:5090 : FreeSwitch 100.100.100.108 : Caller 100.100.100.212 : Calle Request-Line: INVITE sip:02131234@100.100.100.211<sip%3A02131234(a)100.100.100.211>SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@64.76.148.108:42772> To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: Bria release 2.2 stamp 45414 Content-Length: 388 Message Body Session Description Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Status-Code: 100 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Server: kamailio (3.1.1 (i386/linux)) Content-Length: 0 Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=8422566b To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Contact: <sip:02131234@100.100.100.211:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Accept: application/sdp Message Body Session Description Protocol Request-Line: ACK sip:02131234@100.100.100.211:5090;transport=udp SIP/2.0 Method: ACK [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-182d537bcb534165-1---d8754z-;rport Max-Forwards: 70 Route: <sip:100.100.100.211;lr> Contact: <sip:1234@64.76.148.108:42772> To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 ACK User-Agent: Bria release 2.2 stamp 45414 Content-Length: 0 Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: True] [Suspected resend of frame: 7] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=8422566b To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Contact: <sip:02131234@100.100.100.211:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Accept: application/sdp Message Body Session Description Protocol When I removed Freeswitch from the path the ACK is routed ok. The ACK is routed by loose_route, but then it fails because the if ( t_check_trans() ) { What could be happening? Regards, Ricardo.- *Ricardo Martínez **|* Ingeniero de Desarrollo RedVoiss *|* rmartinez(a)redvoiss.net Santiago - Chile *|* +56 2 2408196 www.redvoiss.net

2 1

ACK problem with FreeSwitch-Kamailio SBC implementation.
by Ricardo Martinez 17 Jan '11

17 Jan '11

Hello. I was wondering if someone can help me with some problem that I’m having with the combination of FreeSwitch and Kamailio as SBC. I’m having and ACK not routed by kamailio and dropped. Here is the debug. 100.100.100.211:5060 : Kamailio 100.100.100.211:5090 : FreeSwitch 100.100.100.108 : Caller 100.100.100.212 : Calle User Datagram Protocol, Src Port: 42772 (42772), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:02131234@100.100.100.211<sip%3A02131234(a)100.100.100.211>SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@64.76.148.108:42772> To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: Bria release 2.2 stamp 45414 Content-Length: 388 Message Body Session Description Protocol User Datagram Protocol, Src Port: sip (5060), Dst Port: 42772 (42772) Session Initiation Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Status-Code: 100 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Server: kamailio (3.1.1 (i386/linux)) Content-Length: 0 User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:3031234@100.100.100.212:5060 SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Record-Route: <sip:100.100.100.211;lr=on> Via: SIP/2.0/UDP 100.100.100.211;branch=z9hG4bK3916.41fc6fb3.0 Via: SIP/2.0/UDP 100.100.100.211:5090 ;rport=5090;branch=z9hG4bK1QH1XDH609cUm Max-Forwards: 66 From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=QmrB15eKrQ3Bj To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> Call-ID: 54a74b6a-9cf6-122e-05a0-0050bad00883 CSeq: 7304076 INVITE Contact: <sip:mod_sofia@100.100.100.211:5090> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 200 X-FS-Support: update_display Remote-Party-ID: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;party=calling;screen=yes;privacy=off Message Body Session Description Protocol User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Status-Code: 100 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 100.100.100.211;branch=z9hG4bK3916.41fc6fb3.0;received=100.100.100.211 Via: SIP/2.0/UDP 100.100.100.211:5090 ;rport=5090;branch=z9hG4bK1QH1XDH609cUm Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=QmrB15eKrQ3Bj To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> Call-ID: 54a74b6a-9cf6-122e-05a0-0050bad00883 CSeq: 7304076 INVITE Server: Thirdlane Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: <sip:3031234@100.100.100.212<sip%3A3031234(a)100.100.100.212> > Content-Length: 0 User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 100.100.100.211;branch=z9hG4bK3916.41fc6fb3.0;received=100.100.100.211 Via: SIP/2.0/UDP 100.100.100.211:5090 ;rport=5090;branch=z9hG4bK1QH1XDH609cUm Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=QmrB15eKrQ3Bj To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=as6727124a Call-ID: 54a74b6a-9cf6-122e-05a0-0050bad00883 CSeq: 7304076 INVITE Server: Thirdlane Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: <sip:3031234@100.100.100.212<sip%3A3031234(a)100.100.100.212> > Content-Type: application/sdp Content-Length: 239 Message Body Session Description Protocol User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: ACK sip:3031234@100.100.100.212<sip%3A3031234(a)100.100.100.212>SIP/2.0 Method: ACK [Resent Packet: False] Message Header Record-Route: <sip:100.100.100.211;lr=on> Via: SIP/2.0/UDP 100.100.100.211;branch=0 Via: SIP/2.0/UDP 100.100.100.211:5090 ;rport=5090;branch=z9hG4bK20atZ819Xj3Dg Max-Forwards: 69 From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=QmrB15eKrQ3Bj To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=as6727124a Call-ID: 54a74b6a-9cf6-122e-05a0-0050bad00883 CSeq: 7304076 ACK Contact: <sip:mod_sofia@100.100.100.211:5090> Content-Length: 0 User Datagram Protocol, Src Port: sip (5060), Dst Port: 42772 (42772) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=8422566b To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Contact: <sip:02131234@100.100.100.211:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "Outbound Call" <sip:02131234@100.100.100.211<sip%3A02131234(a)100.100.100.211> >;party=calling;privacy=off;screen=no Message Body Session Description Protocol User Datagram Protocol, Src Port: 42772 (42772), Dst Port: sip (5060) Session Initiation Protocol Request-Line: ACK sip:02131234@100.100.100.211:5090;transport=udp SIP/2.0 Method: ACK [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-182d537bcb534165-1---d8754z-;rport Max-Forwards: 70 Route: <sip:100.100.100.211;lr> Contact: <sip:1234@64.76.148.108:42772> To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 ACK User-Agent: Bria release 2.2 stamp 45414 Content-Length: 0 User Datagram Protocol, Src Port: sip (5060), Dst Port: 42772 (42772) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: True] [Suspected resend of frame: 7] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=8422566b To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Contact: <sip:02131234@100.100.100.211:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "Outbound Call" <sip:02131234@100.100.100.211<sip%3A02131234(a)100.100.100.211> >;party=calling;privacy=off;screen=no Message Body Session Description Protocol When I removed Freeswitch from the path the ACK is routed ok. The ACK is routed by loose_route, but then it fails because the if ( t_check_trans() ) { What could be happening? * * *Regards,* *Ricardo.-*

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Re: [SR-Users] OpenIMSCore and Kamalio Integration
by Carsten Bock 17 Jan '11

17 Jan '11

Hi Andres, no problem ;-) See my answers inline. 2011/1/17 "Andrés S. García Ruiz" <asgarcia(a)um.es>: > > Hi Carsten, > > Thanks a lot for all the information and the access to your developments. > > I haven't been able to answer last week cause I got the flu. I'm fine right > now. > > In a previous mail you wrote: > > "The goal of the project is to create a stable, fast, open-source, > feature-rich IMS implementation based on Kamailio and the > OpenIMS-Core." > > > > I suppose that you've chosen Kamailio as a Sip Server for XCAP, Presence, > RLS, presence-rules, etc. and OpenIMSCore as the heart of the IMS. So your > work aims -I think- to make an easy integration of Kamailio with the > OpenIMSCore. Is that true? No, not exactly. I've replaced the SER-Core from the OpenIMS-Core with the Kamailio-Core and we are in the progress of creating a new OpenIMS-Core (based on the "old" one). We do not continue to use the existing OpenIMS-Core as such. The OpenIMS-Core implementation has some limitations and has some aspects, which i personally do not like and which i am trying to change (don't misunderstand me here: For a research project, there was really great work done). However, there are a few parts, which i want to change, in order to make the OpenIMS-Core more flexible (and in some issues more stable): - The current OpenIMS-Core borrowed some code from other modules, e.g. there was some code taken from the dialog module, from the Nathelper module and maybe others, in order to implement the CSCF-Components. As the intented show-case and prototype, that is totally acceptable. However, this results in improvements and fixes done to those modules not necessarily leading to fixes in the OpenIMS-Modules - The used Core is SER 2.1-dev1. I think, that was the version of SER when the OpenIMS-Core was released. I wanted to replace the core with a more current core (in my case Kamailio 3.1/3.2), in order to bring the OpenIMS-Core forward. - many aspects of the implementation can be replaced in the current version with Core-Functions (e.g. header manipulation), that will make the code easier to maintain - i want to add the missing documentation for the modules - i wanted to include the modules in the Kamailio/sip-router.org project, in order to keep them maintained My work has nothing to do with integrating a Kamailio-Server (e.g. Presence) into a system consisting of the OpenIMS-Core, but it has something to do with improving the OpenIMS-Core. First i want to improve the OpenIMS-Core, and then we will see, what's next. All i did so far, in order to make a Kamailio-based-Presence-Server work with the IMS-Core, was to modify the Trigger-Point on my HSS a little bit... (but more a dirty hack and nothing, were i am sure if it is the currect or even safe to do; i made it work rather than thinking of, if it was correct...). You need to reconfigure your trigger-points (check the Session Case) in order to make this work. I have attached my trigger-point for presence, may this will help you... > I'm trying to design and implement a similar architecture, but I've had some > issues, mainly with how the OpenIMSCore communicates with the external SIP > presence servers. For example, when I send a RLS SUBSCRIBE to the IMS core, > it does the following: > - If the subscribed URI is an user of the domain, it translates its SIP URI > to the sip contact uri. > - If the subscribed URI is not an user, it answers with "HSS User unknown". > > Could you tell me what kind of problems have you solved till now? > > Kind regards, > Andrés. > > PS: I'm using Ubuntu. Is Ubuntu compatible with your distribution? Since Ubuntu is based on Debian, i think that should work out of the box (but no warranty for that). Kind regards, Carsten P.S: Please always use the reply to all button, in order to keep the lists in CC.

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Wrong ACK packet sent from Cisco gw
by Nawfel Oujdi 17 Jan '11

17 Jan '11

Hello! I m facing the same strange behaviour with my AS5300 voice gateway. When the gw is connected directly to PBX everythings works well but when i put a sip proxy forwarding calls between gw and PBX all the calls hangs up after 5 sec (+or -). Looking into the trace sip i realize that gw send a wrong ACK in reply of INVITE , then sip proxy discard it and PBX hangs the call cause he never receive the ACK. ACK sip:79.125.120.12:5060;lr=on;did=ce.3716ea02 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 Route: <sip:911111500@PBX:5060> To: <sip:911111500@sip_proxy>;tag=as7f388e3f Date: Mon, 17 Jan 2011 09:26:36 GMT Call-ID: B6F61A2E-215211E0-802BD462-C4432B89@cisco_gw To work fine , the content of Route header should be in ACK header and viceversa. I tried to compare between the sip trace of a wrong call and a good one (using other cisco gw AS5350 who works well with sip proxy in the same escenario) and i realize that the only difference is the INVITE of wrong case doesn' t send branch number in the via header. INVITE sip:911111500@sip_proxy:5060 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 To: <sip:911111500@sip_proxy> i m using c5300-is-mz.123-26.bin ios version. Anybody understand what is happening in there?? is there any solution?? i ll send more information if it s requested. Thanks in advance. Nawfel Oujdi here is the result of ngrep: U 2011/01/13 15:14:43.791514 cisco_gw:51703 -> sip_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. 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:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.791893 sip_server:5060 -> cisco_gw:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: OpenSIPS (1.6.3-notls (i386/linux)). Content-Length: 0. . U 2011/01/13 15:14:43.791957 sip_server:5060 -> asterisk_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 5. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.792775 asterisk_server:5060 -> sip_server:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Length: 0. . U 2011/01/13 15:14:43.793770 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.794688 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:43.856520 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:43.993417 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.993613 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:44.038774 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:44.193431 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. -- ------------------------------ Aviso legal: Este mensaje electrónico está dirigido únicamente a la(s) dirección(es) indicadas anteriormente; el carácter confidencial, personal e intransferible del mismo está protegido legalmente. Cualquier revelación, uso o reenvío no autorizado, completo o en parte, está prohibido. Si ha recibido este mensaje por equivocación, notifíquelo inmediatamente a la persona que lo ha enviado y borre el mensaje original junto con sus ficheros anexos sin leerlo ni grabarlo, total o parcialmente.

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[OpenSER-Users] Strange ACK format gives OpenSER error?
by Nawfel Oujdi 17 Jan '11

17 Jan '11

Hello! I m facing the same strange behaviour with my AS5300 voice gateway. When the gw is connected directly to PBX everythings works well but when i put a sip proxy forwarding calls between gw and PBX all the calls hangs up after 5 sec (+or -). Looking into the trace sip i realize that gw send a wrong ACK in reply of INVITE , then sip proxy discard it and PBX hangs the call cause he never receive the ACK. ACK sip:79.125.120.12:5060;lr=on;did=ce.3716ea02 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 Route: <sip:911111500@PBX:5060> To: <sip:911111500@sip_proxy>;tag=as7f388e3f Date: Mon, 17 Jan 2011 09:26:36 GMT Call-ID: B6F61A2E-215211E0-802BD462-C4432B89@cisco_gw To work fine , the content of Route header should be in ACK header and viceversa. I tried to compare between the sip trace of a wrong call and a good one (using other cisco gw AS5350 who works well with sip proxy in the same escenario) and i realize that the only difference is the INVITE of wrong case doesn' t send branch number in the via header.i m not really sure that cause teh problem. INVITE sip:911111500@sip_proxy:5060 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 To: <sip:911111500@sip_proxy> i m using c5300-is-mz.123-26.bin ios version. Anybody understand what is happening in there?? is there any solution?? i ll send more information if it s requested. Thanks in advance. Nawfel Oujdi here is the result of ngrep: U 2011/01/13 15:14:43.791514 cisco_gw:51703 -> sip_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. 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:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.791893 sip_server:5060 -> cisco_gw:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: OpenSIPS (1.6.3-notls (i386/linux)). Content-Length: 0. . U 2011/01/13 15:14:43.791957 sip_server:5060 -> asterisk_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 5. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.792775 asterisk_server:5060 -> sip_server:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Length: 0. . U 2011/01/13 15:14:43.793770 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.794688 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:43.856520 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:43.993417 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.993613 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:44.038774 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:44.193431 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. -- ------------------------------ Aviso legal: Este mensaje electrónico está dirigido únicamente a la(s) dirección(es) indicadas anteriormente; el carácter confidencial, personal e intransferible del mismo está protegido legalmente. Cualquier revelación, uso o reenvío no autorizado, completo o en parte, está prohibido. Si ha recibido este mensaje por equivocación, notifíquelo inmediatamente a la persona que lo ha enviado y borre el mensaje original junto con sus ficheros anexos sin leerlo ni grabarlo, total o parcialmente.

1 0

tm module question
by Prelle, Stefan 14 Jan '11

14 Jan '11

Hi folks, I have a problem with a scenario where an incoming INVITE is forwarded to a redirect server which responds with 302 and a new target in the Contact header (B). The call is being forwarded to the new contact correctly. All responses from B that are either provisional or a 200 OK are being routed back to the caller. If B responds with any code >=400 the 302 from the redirect server is being returned to the caller instead of the received code. A B C |--INVITE->| | | |--INVITE->| | |<---302---| D | |----INVITE--->| | |<-----404-----| |<---302---| Is this intended? And if so, is there any way to change this behaviour so the latest received response is returned? I tested using the version from repository from 13.01.2011 My reduced config looks like this: route{ if (@to.tag != "") { if (loose_route()) { record_route(); t_relay(); } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } record_route(); } exit; } else { record_route(); } t_on_failure("REDIRECT_REPLY"); if (!t_relay_to_udp("192.168.0.10","5060")) { sl_reply_error(); }; exit; } failure_route[REDIRECT_REPLY] { if (t_check_status("3[0-9][0-9]")) { get_redirects("*:1"); t_relay(); exit; } } Regards, Stefan

2 2

IP Address
by Bernhard Suttner 14 Jan '11

14 Jan '11

Hi, I have a very short question: Kamailio does listen to many IP Addresses and does forward messages to the peer with t_relay. Which IP address will kamailio use for the outgoing traffic as source-ip, if there are many IP addresses on the interface? Is it possible to specify this somehow? Thanks in advance. Best regards, Bernhard

2 2

manual dispatching
by Klaus Darilion 13 Jan '11

13 Jan '11

Hi! I would like to use probing implementation in dispatcher module to keep track of the state of the gateways, but I want to choosing a gateway manually in the config. For example I defined a dispatcher set with 6 entries. Then in .cfg I want to retrieve a certain entry from the set (by index) and its state (active, inactive, probing). AFAIS there is currently now way to do this. Does anybody knows a workaround or maybe another way for manual dispatching but automatic probing? Thanks Klaus

1 2

Looking for the RTP Proxy
by Mallik 13 Jan '11

13 Jan '11

Dear sir, we re looking for RTP proxy training or solution, can you please give us more details on this thanking you with regards Mallik

1 0

How to test xhttp with wget or curl
by mayamatakeshi 13 Jan '11

13 Jan '11

Hello, can someone give me a sample of calling to xhttp from curl or wget? I have this in my kamailio.cfg: listen=tcp:192.168.2.105:80 loadmodule "xhttp.so" event_route[xhttp:request] { xhttp_reply("200", "OK", "text/html", "<html><body>OK - [$si:$sp]</body></html>"); } But when I try wget or curl, kamailio closes the connection without sending a response: [root@centos55-02105 tmp]# curl http://192.168.2.105/test curl: (52) Empty reply from server [root@centos55-02105 tmp]# wget http://192.168.2.105/test --2011-01-14 00:11:49-- http://192.168.2.105/test Connecting to 192.168.2.105:80... connected. HTTP request sent, awaiting response... No data received. I see this in the log file: Jan 14 00:04:47 centos55-02105 /usr/local/sbin/kamailio[10479]: ERROR: <core> [tcp_read.c:904]: ERROR: tcp_read_req: bad request, state=7, error=4 buf: GET /test HTTP/1.0^M User-Agent: Wget/1.11.4 Red Hat modified^M Accept: */*^M Host: 192.168.2.105^M Connection: Keep-Alive^M ^M parsed: GET /test HTTP/1.0^M User-Agent: Wget/1.11.4 Red Hat modified^M Accept: */*^M Host: 192.168.2.105^M Connection: Keep-Alive^M ^M br, takeshi

2 3

Dispatcher Setting
by Bernhard Suttner 13 Jan '11

13 Jan '11

Hi, I am using the Dispatcher module to load-balance the calls to several freeswitch servers. On which responses within the failure_route should be a fallback to another server (dst_next_dst())? I am sure, that on a 408 response there should be a fallback but whats about other 4XX responses and 5XX or 6XX? Thanks in advance. Best regards, Bernhard Suttner

2 2

OpenIMSCore and Kamalio Integration
by "Andrés S. García Ruiz" 13 Jan '11

13 Jan '11

Hi everybody, I'm trying to deploy an IMS network with OpenIMSCore and Kamailio. Since OpenIMSCore has been already tested along with Mobicents, now I want substitute Mobicents for Kamailio. I've also successfully installed Kamailio. I can run it without any problem, but when SIP Publish messages arrive at Kamalio, it answers with "407 Proxy Authentication Required". The IMS presentity is already registered against the OpenIMSCore. How can I solve that problem? The publish message sent: PUBLISH sip:testuser01@open-ims.test SIP/2.0 Route: <sip:ciervo.inf.um.es:5060;lr>, <sip:iscmark@scscf.open-ims.test:6060;lr;s=1;h=0;d=0;a=7369703a74657374757365723031406f70656e2d696d732e74657374> Call-ID: K28QO9PYAE5vn3RNdYOkIEddMEd6yDiqeAW7DwSt6XV. CSeq: 1 PUBLISH From: "testuser01" <sip:testuser01@open-ims.test>;tag=ff123bda To: "testuser01" <sip:testuser01@open-ims.test> Via: SIP/2.0/UDP 155.54.210.135:6060;branch=z9hG4bKa31a.6cba1cd2.0 Via: SIP/2.0/UDP 155.54.210.134:4060;branch=z9hG4bKa31a.71481d13.0;i=1 Via: SIP/2.0/TCP 155.54.190.166:8060;rport=41624;branch=z9hG4bK-d8754z-ce7487ff34be55f9-1---d8754z- Max-Forwards: 15 Content-Type: application/pidf+xml Expires: 30000 Event: presence Contact: <sip:testuser01@155.54.190.166:8060> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite IMS-OSGi-Client 0.1 CVS-Mon_Nov_29_10-14-33_CET_2010 Content-Length: 451 P-Asserted-Identity: <sip:testuser01@open-ims.test> P-Charging-Vector: icid-value="P-CSCFabcd000000004cf3708400000002";icid-generated-at=155.54.210.134;orig-ioi="open-ims.test" <?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' entity='sip:testuser01@open-ims.test'><tuple id='t6b9a6ab3'><status><basic>open</basic></status></tuple><dm:person id='p34b126e5'><rpid:activities><rpid:Online/></rpid:activities><dm:note>Online</dm:note></dm:person></presence> Kamailio answer: SIP/2.0 407 Proxy Authentication Required Call-ID: K28QO9PYAE5vn3RNdYOkIEddMEd6yDiqeAW7DwSt6XV. CSeq: 1 PUBLISH From: "testuser01" <sip:testuser01@open-ims.test>;tag=ff123bda To: "testuser01" <sip:testuser01@open-ims.test>;tag=b27e1a1d33761e85846fc98f5f3a7e58.3d3a Via: SIP/2.0/UDP 155.54.210.135:6060;branch=z9hG4bKa31a.6cba1cd2.0 Via: SIP/2.0/UDP 155.54.210.134:4060;branch=z9hG4bKa31a.71481d13.0;i=1 Via: SIP/2.0/TCP 155.54.190.166:8060;rport=41624;branch=z9hG4bK-d8754z-ce7487ff34be55f9-1---d8754z- Proxy-Authenticate: Digest realm="open-ims.test", nonce="TPNwMkzzbwZRE1piAM8yHaXGsXvwS16v" Server: kamailio (3.1.0 (i386/linux)) Content-Length: 0 Kamailio log: 5(15391) DEBUG: <core> [parser/msg_parser.c:628]: SIP Request: 5(15391) DEBUG: <core> [parser/msg_parser.c:630]: method: <PUBLISH> 5(15391) DEBUG: <core> [parser/msg_parser.c:632]: uri: <sip:testuser01@open-ims.test> 5(15391) DEBUG: <core> [parser/msg_parser.c:634]: version: <SIP/2.0> 5(15391) DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <PUBLISH> 5(15391) DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=10 5(15391) DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [45]; uri=[sip:testuser01@open-ims.test] 5(15391) DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: to body ["testuser01" <sip:testuser01@open-ims.test> ] 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bKa31a.6cba1cd2.0>; state=16 5(15391) DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 5(15391) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 5(15391) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via 5(15391) DEBUG: <core> [receive.c:145]: After parse_msg... 5(15391) DEBUG: <core> [receive.c:186]: preparing to run routing scripts... 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bKa31a.71481d13.0>; state=6 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 236, <i> = <1>; state=16 5(15391) DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 5(15391) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=100 5(15391) DEBUG: <core> [parser/msg_parser.c:526]: parse_headers: this is the second via 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 235, <rport> = <41624>; state=6 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bK-d8754z-ce7487ff34be55f9-1---d8754z->; state=16 5(15391) DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 5(15391) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=100 5(15391) DEBUG: maxfwd [mf_funcs.c:85]: value = 15 5(15391) DEBUG: <core> [parser/msg_parser.c:199]: DEBUG: get_hdr_body : content_length=451 5(15391) DEBUG: <core> [parser/msg_parser.c:101]: found end of header 5(15391) DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=ff123bda 5(15391) DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 5(15391) DEBUG: sanity [mod_sanity.c:217]: all sanity checks passed 5(15391) DEBUG: siputils [checks.c:73]: no totag 5(15391) DEBUG: tm [t_lookup.c:1081]: DEBUG: t_check_msg: msg id=1 global id=0 T start=0xffffffff 5(15391) DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=41274, isACK=0 5(15391) DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed 5(15391) DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found 5(15391) DEBUG: tm [t_lookup.c:1150]: DEBUG: t_check_msg: msg id=1 global id=1 T end=(nil) 5(15391) DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==9 && [open-ims.test] == [127.0.0.1] 5(15391) DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 5(15391) DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==14 && [open-ims.test] == [155.54.190.245] 5(15391) DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 5(15391) DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==9 && [open-ims.test] == [127.0.0.1] 5(15391) DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 5(15391) DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==14 && [open-ims.test] == [155.54.190.245] 5(15391) DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 5(15391) DEBUG: auth_db [authorize.c:239]: realm value [open-ims.test] 5(15391) DEBUG: auth [api.c:85]: auth:pre_auth: Credentials with realm 'open-ims.test' not found 5(15391) DEBUG: auth_db [authorize.c:257]: not authenticated 5(15391) DEBUG: auth [challenge.c:102]: build_challenge_hf: realm='open-ims.test' 5(15391) DEBUG: auth [challenge.c:236]: auth: 'Proxy-Authenticate: Digest realm="open-ims.test", nonce="TPNwMkzzbwZRE1piAM8yHaXGsXvwS16v" ' 5(15391) DEBUG: sl [sl.c:278]: reply in stateless mode (sl) 5(15391) DEBUG: <core> [msg_translator.c:207]: check_via_address(155.54.210.135, 155.54.210.135, 0) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [receive.c:289]: receive_msg: cleaning up Thanks in advance, Andrés. -- ----------------------- Andrés S. García Ruiz ----------------------- e-mail: asgarcia(a)um.es Teléfono: (+34) 868 888258 Dpto. de Ingeniería de la Información y las Comunicaciones Facultad de Informática Universidad de Murcia 30100 Campus de Espinardo

5 13

set_dlg_profile not working?
by Ricardo Martinez 12 Jan '11

12 Jan '11

Hello. I’m trying to count the calls to a specific gateway. So I’m using the dialog module. This route is done before the final relay. route[CHANNELS] { if( is_method("INVITE") ) { if(!isflagset(31) ) { # user has max channel limit set as preference if(is_avp_set("$avp(s:channels)") && avp_check("$avp(s:channels)", "gt/s:0")) { # get current calls for uuid xlog("L_INFO","[$ci] : avp_channels existe y es mayor que 0 ?"); if (get_profile_size("gws","$rd","$var(calls)")) { xlog("L_INFO","Actualmente el gateway $rd tiene $var(calls) llamadas simultaneas\n"); } # check within limit if($avp(s:channels) > $var(calls)) { xlog("L_INFO", "CHANNELS : terminador '$rd' tiene '$var(calls)' de '$avp(s:channels)' active calls before this one\n"); $var(setprofile) = 1; } else { xlog("L_INFO", "Call crtl: user channel limit exceeded [$var(calls)/$avp(s:channels)]\n"); send_reply("487", "Request Terminated: Channel limit exceeded"); exit; } } else { xlog("L_INFO","[$ci] : the value if channels is $avp(s:channels)"); if( !is_avp_set("$avp(s:channels)") ) { xlog("L_INFO","[$ci] : avp_channel not set"); }; if(!avp_check("$avp(s:channels)", "gt/s:0") ) { xlog("L_INFO","[$ci] : avp_channel menor que 0?"); }; $var(setprofile) = 0; } } if($var(setprofile) > 0) { * if ( set_dlg_profile("gws","$rd") ) {* * xlog("L_INFO","set_dlg_profile exitoso");* } } When i made the call and try to see the profile with this command : [root@sipvoiss kamailio]# kamctl fifo profile_get_size gws 10.0.0.220 profile:: name=gws value=10.0.0.220 count=0 I’m not seeing the call added to the profile, despite the fact I using the * set_dlg_profile("gws","$rd")* What could be happening? This is part of the debug : Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] INVITE dominio:10.0.0.208 LOCAL DOMAIN Jan 11 18:07:03 .: DEBUG: misc_radius [functions.c:212]: success Jan 11 18:07:03 .: DEBUG: misc_radius [functions.c:133]: AVP 'tranum'/0=' sip:02444444444@10.0.0.208 <sip%3A02444444444(a)10.0.0.208>'/0 has been added Jan 11 18:07:03 .: DEBUG: misc_radius [functions.c:133]: AVP 'channels'/0='2'/0 has been added Jan 11 18:07:03 .: DEBUG: avpops [avpops_impl.c:1047]: 1 avps were processed Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] INVITE con dominio:10.0.0.208 checks de usuario via avp_radius OK Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] CHECKS SECTION :: INVITE:sip:02444444444@10.0.0.208 <INVITE%3Asip%3A02444444444(a)10.0.0.208>checks ok... Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:1748]: added matched_gws[0]=[1, 4, 1, 8207736] Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:1645]: added gw_uri_avp <1|1|4|550|3700771904||5060||1|0> with weight <8207736> Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:1911]: r_uri < sip:5504444444@10.0.0.220:5060>, dst_uri <> Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:2015]: added ruri_user_avp <02444444444> Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:2050]: added flags_avp <0> Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:2057]: added defunct_gw_avp <3700771904> Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] Ruta LCR : cambiando sip:02444444444@10.0.0.208 <sip%3A02444444444(a)10.0.0.208> por sip:5504444444@10.0.0.220:5060 Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] : entro en ruta 39 de call control Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] : no esta seteado el flag 31 Jan 11 18:07:03 .: DEBUG: avpops [avpops_impl.c:1180]: check <2> against <0> as str /16 Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] : avp_channels existe y es mayor que 0 ? Jan 11 18:07:03 .: INFO: <script>: Actualmente el gateway 10.0.0.220 tiene 0 llamadas simultaneas Jan 11 18:07:03 .: INFO: <script>: CHANNELS : terminador '10.0.0.220' tiene '0' de '2' active calls before this one Jan 11 18:07:03 .: INFO: <script>: set_dlg_profile exitoso Thanks in advance, *Regards,* *Ricardo.-*

2 5

AFTER MAINTENANCE DIALOG MODULE NOT WORKING
by Phillman25 Kyriacou 12 Jan '11

12 Jan '11

Hi I performed the following maintenance actions to my server that is currently installed with kamailio v3.1 cd /usr/local/src/kamailio-3.1.0/kamailio git pull origin make all make install After this action the dialog module is not working, after investigation, i found out that the following entry is causing this issue: modparam("dialog", "db_mode", 1) In my syslog i find the following errors: Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7397]: CRITICAL: dialog [dlg_db_handler.c:117]: BUG - db connection found already open Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 33 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 34 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7397]: ERROR: dialog [dialog.c:620]: failed to connect to database (rank=24) Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7397]: ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module dialog (/usr/local/lib/kamailio/modules_k/dialog.so) Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7397]: ERROR: <core> [pt.c:481]: ERROR: fork_tcp_process(): init_child failed for process 30, pid 7397, "tcp receiver child=7" Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7397]: ERROR: <core> [tcp_main.c:4811]: ERROR: tcp_main: fork failed: Success Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 37 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 43 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 30 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 41 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 5 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: WARNING: <core> [db_id.c:281]: identical DB URLs, but different DB connection pid [7396/7349] Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7391]: INFO: <core> [main.c:807]: INFO: signal 15 received Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 47 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: CRITICAL: dialog [dlg_db_handler.c:117]: BUG - db connection found already open Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: ERROR: dialog [dialog.c:620]: failed to connect to database (rank=23) Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module dialog (/usr/local/lib/kamailio/modules_k/dialog.so) Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: ERROR: <core> [pt.c:481]: ERROR: fork_tcp_process(): init_child failed for process 29, pid 7396, "tcp receiver child=6" Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 39 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: ERROR: <core> [tcp_main.c:4811]: ERROR: tcp_main: fork failed: Success Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 45 Only the dialog module is facing this problem Has anyone faced this before? Thanks Phillip

2 2

mi_fifo question
by Ricardo Martinez 11 Jan '11

11 Jan '11

Hello. I’m trying to use the kamctl command to get information about dialogs. I’m using the next line : kamctl fifo profile_get_size:kamailio_fifo and I’m getting the next output in the log file : Jan 11 12:34:38 /usr/local/sbin/kamailio[15051]: ERROR: mi_fifo [fifo_fnc.c:223]: open error (/tmp/kamailio_fifo:openser_receiver_15073): No such file or directory Jan 11 12:34:38 /usr/local/sbin/kamailio[15051]: ERROR: mi_fifo [fifo_fnc.c:484]: cannot open reply pipe /tmp/kamailio_fifo:openser_receiver_15073 What I’m doing wrong? This is part of the configuration file : # ----- mi_fifo params ------ modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("mi_fifo", "fifo_group", 0) modparam("mi_fifo", "fifo_group", "root") modparam("mi_fifo", "fifo_user", 0) modparam("mi_fifo", "fifo_user", "root") and this is the file kamailio_fifo in the /tmp directory : [root@kamailio]# ls -la /tmp total 52 drwxrwxrwt. 4 root root 4096 Jan 11 12:38 . dr-xr-xr-x. 25 root root 4096 Jan 7 09:25 .. drwxrwxrwt 2 root root 4096 Jan 7 09:25 .ICE-unix *prw-rw---- 1 root root 0 Jan 11 12:34 kamailio_fifo* drwx------ 2 root root 4096 Jan 11 09:54 ssh-gpTCQ12700 Hope someone could help me. Regards, Ricardo.-

2 3

Re: [SR-Users] Avp_write missing modules ?
by Alex Balashov 10 Jan '11

10 Jan '11

Please keep the discussion on list - cc: the list. I see no indication in the documentation that avp_write() is a function that exists, even going back to 1.4.x. On 01/10/2011 04:54 PM, Amit Nepal wrote: > Kamailio 3.1 > > Thank You > Amit Nepal > Systems Administrator > Phoenix Internet > Phone: 602-385-0731 > 602-234-0917#112 > http://www.phoenixinternet.net > > > On 1/10/2011 2:50 PM, Alex Balashov wrote: >> On 01/10/2011 04:48 PM, Amit Nepal wrote: >>> I am getting missing loadmodule issue with avp_write. I have the >>> following statement on line 11 and i am getting this error. Could you >>> please help me on this. But i can load from db using avp_db_load and i >>> can delete avp values but only avp_write is not working. >>> >>> >>> avp_write("$ruri", "s:fwd_blind"); >>> >>> loading modules under >>> /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ >>> 0(2361) : <core> [cfg.y:3412]: parse error in config file >>> /usr/local/etc/kamailio/cfgs/pstn.cfg, line 11, column 39: unknown >>> command, missing loadmodule? >>> >>> ERROR: bad config file (1 errors) >> >> Which version of Kamailio? >> -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

3 5

Avp_write missing modules ?
by Amit Nepal 10 Jan '11

10 Jan '11

I am getting missing loadmodule issue with avp_write. I have the following statement on line 11 and i am getting this error. Could you please help me on this. But i can load from db using avp_db_load and i can delete avp values but only avp_write is not working. avp_write("$ruri", "s:fwd_blind"); loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ 0(2361) : <core> [cfg.y:3412]: parse error in config file /usr/local/etc/kamailio/cfgs/pstn.cfg, line 11, column 39: unknown command, missing loadmodule? ERROR: bad config file (1 errors) Thank you Amit

2 1

Problem with contact header parsing and get_redirects
by Stefan Prelle 10 Jan '11

10 Jan '11

Hi folks, Since upgrading vom Kamailio 1.5 to 3.1 I have a problem using get_redirects from the uac_redirect module. It seems that there is a problem with parsing the Contact header from the received 302. This is my script excerpt: if (t_check_status("3[0-9][0-9]")) { get_redirects("*"); if (!t_relay()) { xlog("L_ERR","t_relay failed\n"); } } Some 302 messages lead (as expected) to INVITEs send to targets in the Contact header, for others t_relay fails. A working example would be Contact:<sip:+49406686100@192.168.0.1:5060;user=phone>;q=0.5,<sip: +49406686100@192.168.0.2:5060;user=phone>;q=0.25 It fails on Contact: sip:+49406686100@192.168.0.1:5060 The following log messages appear when calling get_redirects (and failing t_relay): kamailio[7372]: DEBUG: uac_redirect [rd_funcs.c:74]: resume branch=0 kamailio[7372]: DEBUG: uac_redirect [rd_funcs.c:81]: checking branch=0 (added=0) kamailio[7372]: DEBUG: uac_redirect [rd_funcs.c:85]: branch=0 is a redirect (added=0) kamailio[7372]: WARNING: <core> [mem/q_malloc.c:366]: qm_malloc(0x82ba320, 32) called from <core>: parser/msg_parser.c: parse_headers(339) kamailio[7372]: WARNING: <core> [mem/q_malloc.c:406]: qm_malloc(0x82ba320, 32) returns address 0x8311d94 frag. 0x8311d7c (size=32) on 1 -th hit kamailio[7372]: DEBUG: uac_redirect [rd_funcs.c:233]: contact hdr not found in sh_rpl The Contact header that fails obviously has no parameters for the URI and contact itself. As far as I understand the RFC 3261 BNF this is valid. The 302 response is by the way created by kamailio when using the LCR module (like this) if (!load_gws("1")) {... exit;} if (!next_gw()) {... exit; } sl_send_reply("302", "Not here"); Did I miss anything when upgrading or is this a problem with the parser? Cheers, Stefan

2 5

Re: [SR-Users] ENUM changes?
by Juha Heinanen 09 Jan '11

09 Jan '11

klaus, could you try with master version if it fixes the problem. -- juha

2 2

Kamailio 3.1.1 + Sipp
by Rochdi ABID 09 Jan '11

09 Jan '11

Hello community, I want to test Kamailio 3.1.1 with SIPP. I followed many ideas that I found them on the internet but without no success. In particular, I followed this link http://www.troubleshootingwiki.org/OpenSER#Stress_Test_The_SIP_Signaling I can see traffic from the uac --> Kamailio --> uas ans in the other sens but no successful call. Can you give me any ideas please. Regards, Rochdi

2 1

Kamailio 3.1.1 + Sipp
by Rochdi ABID 08 Jan '11

08 Jan '11

Hello community, I want to test Kamailio 3.1.1 with SIPP. I followed many ideas that I found them on the internet but without no success. In particular, I followed this link http://www.troubleshootingwiki.org/OpenSER#Stress_Test_The_SIP_Signaling I can see traffic from the uac --> Kamailio --> uas ans in the other sens but no successful call. Can you any ideas please. Regards, Rochdi

1 0

Re: [SR-Users] Change the SIP-To-User
by Bernhard Suttner 07 Jan '11

07 Jan '11

Hi, thanks to all responses. I wonder if there is a function to modify the To/From Header stateful. I mean, change the TO header in the INVITE to A and then the 200 OK (for example) reverse the change back to the previous value to B? but the "magic" $tU is amazing... Best regards, Bernhard ----- Original Message ----- From: Olle E. Johansson [mailto:oej@edvina.net] To: Alex Balashov [mailto:abalashov@evaristesys.com] Cc: sr-users(a)lists.sip-router.org Sent: Fri, 07 Jan 2011 11:09:41 +0100 Subject: Re: [SR-Users] Change the SIP-To-User > > 7 jan 2011 kl. 10.55 skrev Alex Balashov: > > > On 01/07/2011 04:52 AM, Carsten Bock wrote: > > > >> the trick, when using "uac_replace_from" is, that it will > >> "automagically" change subsequest requests. This is required in order > >> to work with most SIP-Endpoints. > >> If you just want to change the "From" for the current request, that is > fine. > > > > I understand. Changing To/From by a proxy has long been a thorny issue, > in the sense that on the one hand those fields are purely cosmetic in > strict-3261 mode, but on the other hand, changing them breaks 2543 > compatibility, and it's just not something a proxy is supposed to do. > Well, that changed with a new rfc (put number here) about caller ID updates. > I do not remember the proxy role, but a UA can now change From/To identities > mid call, like after a call transfer in a pbx. > > /O > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

4 3

RLS server as SUBSCRIBE proxy?
by Jeremya 07 Jan '11

07 Jan '11

Hi, My situation is that I have a number of SIP end points (server) that support SUBSCRIBE method on various resources they hold. Primarily these are dialog resources, but include arbitrary other resources with custom MIME types. I also have a large number of SIP end-points (clients) that use SUBSCRIBE to resources held by these end-points. Some of these end-points are standard phones (SPA-962) monitoring dialog state. Other end points are custom clients that monitor a variety of custom resources and MIME types. I have got to the point where the number of SUBSCRIBE clients is making it impractical for the SUBSCRIBE sources to support them. I want to make a SUBSCRIBE proxy server that makes arbitrary SUBSCRIBE requests to my SIP end points (server) and in turn accepts SUBSCRIBE requests from my consumer SIP clients (Phones etc). In effect the server would aggregate SUBSCRIBE requests into a single SUBSCRIBE to a target resource. Ideally the SUBSCRIPTION would be dynamic, generated by the SUBSCRIBE requests from the SIP phones and other clients, but this can be static. I've read the documentation for the RLS module in Kamailio but I can't actually figure out what it does. It may do what I want (if only in some database mode) but I'd appreciate advice as to whether to pursue this as an option or go to plan-B Regards Jeremy

2 2

Re: [SR-Users] Change the SIP-To-User
by Bernhard Suttner 07 Jan '11

07 Jan '11

Hi, ah, thanks. Would be good to have a uac_replace_to. Sounds like a "30min" task to someone familiar with kamailio programming :-) BR, Bernhard ----- Original Message ----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org Sent: Fri, 07 Jan 2011 21:59:55 +0100 Subject: Re: [SR-Users] Change the SIP-To-User > Bernhard Suttner wrote: > > Hi, > > > > thanks to all responses. I wonder if there is a function to modify the > To/From Header stateful. I mean, change the TO header in the INVITE to A and > then the 200 OK (for example) reverse the change back to the previous value > to B? > > > > but the "magic" $tU is amazing... > > If you use uac_replace_from(), the changes will stateful and fixed for > every in-dialog requests and responses. > > But there is no similar feature for To header manipulation. > > regards > klaus > > >

1 0

Using Dispatcher Module From Mysql
by Amit Nepal 07 Jan '11

07 Jan '11

Hi everyone, Again stuck at the dispatcher module.Do anyone happen to have a working config for kamailio with dispatcher list from mysql database ? if not can someone please guide me on this ? Do i need to pass module parameters for the column and the table names ? I have the following config : loadmodule "dispatcher.so" if(uri=~"^sip:1[2-9][0-9]{9}@") { if(is_user_in("credentials","longdistance")) { #route(PSTN); #exit; ds_select_dst("1","8"); forward(); } else{ sl_send_reply("403","No Permissions for Long Distance calls"); exit; }; }; -- Thank You Amit Nepal

3 5

Dispatcher Attributes field
by alex pappas 07 Jan '11

07 Jan '11

Dear all, I'm trying with the Dispatcher module to terminate a call to three different gateways serially. For each gw I use the attribute field to get the attribute and add it as a prefix to the RU. The problem is that the $avp(dsattrs) is returning always the first value of the first gw that is using. example: destination sip:100.12.12.12:5060 attrs 111 destination sip:100.12.12.11:5060 attrs 222 destination sip:100.12.12.10:5060 attrs 333 In this example the rU will always be 111+number. I tried also to initiate the avp in the Failure route by giving null value but still does not change the result. In Request route: if(ds_select_domain("$avp(s:disp_dstgrp)", "4")) { $avp(s:term_prefix) = $avp(dsattrs); # avp for CDR purpose $rU = $avp(dsattrs) + $rU; t_on_failure("FAIL_NATIONAL"); t_relay(); exit; } In Failure route: if(!ds_next_domain()) { t_reply("503", "Service Unavailable"); exit; } else { $rU = $avp(dsattrs) + $rU; t_on_failure("FAIL_THIS"); append_branch(); if(!t_relay()) { t_reply("503","Service Unavailable"); exit; }

2 5

Change the SIP-To-User
by Bernhard Suttner 07 Jan '11

07 Jan '11

Hi, what is the best way to replace the userpart of the SIP-To URI with the content of a pseudo variable? Thanks for every hint. Best regards, Bernhard

6 8

problem in MAKE command during installation Sip Express Router
by city spider 07 Jan '11

07 Jan '11

hello i want to install Sip Express Router on CENT OS 5.5. the problem is that after untar the ser-0.9.6.src.tar.gz package my system doesn't operate MAKE command ,and returns 127 errors containing message "the system doesn't find the folder...." is there any solution for this problem... waiting for your response.

2 1

Call Routing Not working
by Tim King 07 Jan '11

07 Jan '11

Below is my current code and my calls are just getting 100 trying -- your call is important to us even between to locally registered extension. Any guidance as to how to simplify troubleshooting this routing? ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE") ) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } #!ifdef WITH_FREESWITCH # save callee ID $avp(callee) = $rU; route(FSDISPATCH); #!endif # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif #!ifdef WITH_CFGSAMPLES /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } #!endif if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_FREESWITCH if(route(FSINBOUND)) return; #!endif #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } #!ifdef WITH_FREESWITCH # FreeSWITCH routing blocks route[FSINBOUND] { if($si== $sel(cfg_get.freeswitch.bindip) && $sp==$sel(cfg_get.freeswitch.bindport)) return 1; return -1; } route[FSDISPATCH] { if(!is_method("INVITE")) return; if(route(FSINBOUND)) return; # dial number selection switch($rU) { case /"^41$": # 41 - voicebox menu # allow only authenticated users if($au==$null) { sl_send_reply("403", "Not allowed"); exit; } $rU = "vm-" + $au; break; case /"^441[0-9][0-9]$": # starting with 44 folowed by 1XY - direct call to voice box strip(2); route(FSVBOX); break; case /"^433[01][0-9][0-9]$": # starting with 433 folowed by (0|1)XY - conference strip(2); break; case /"^45[0-9]+$": strip(2); break; default: # offline - send to voicebox if (!registered("location")) { route(FSVBOX); exit; } # online - do bridging prefix("kb-"); if(is_method("INVITE")) { # in case of failure - re-route to FreeSWITCH VoiceMail t_on_failure("FAIL_FSVBOX"); } } route(FSRELAY); exit; } route[FSVBOX] { if(!($rU=~"^1[0-9][0-9]+$")) return; prefix("vb-"); route(FSRELAY); } # Send to FreeSWITCH route[FSRELAY] { $du = "sip:" + $sel(cfg_get.freeswitch.bindip) + ":" + $sel(cfg_get.freeswitch.bindport); if($var(newbranch)==1) { append_branch(); $var(newbranch) = 0; } route(RELAY); exit; } #!endif #!ifdef WITH_FREESWITCH failure_route[FAIL_FSVBOX] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } if (t_check_status("486|408")) { # re-route to FreeSWITCH VoiceMail $rU = $avp(callee); $var(newbranch) = 1; route(FSVBOX); } } #!endif # sample config file for dispatcher module #!ifdef WITH_DISPATCHER modparam("dispatcher", "db_url", "mysql://openser:password@localhost /openser") route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); forward(); # t_relay(); } #!endif

2 1

Routing error
by Tim King 07 Jan '11

07 Jan '11

My calls are not making it to the dispatcher and I am not sure what I am doing wrong. Can anyone see what I need to modify so that my calls can be distributed to my free switch boxes via the dispatcher module? cat /usr/local/etc/kamailio/kamailio.cfg #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_FREESWITCH #!define WITH_ANTIFLOOD #!define WITH_DISPATCHER #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:password@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif #!ifdef WITH_FREESWITCH freeswitch.bindip = "127.0.0.1" desc "FreeSWITCH IP Address" freeswitch.bindport = "5090" desc "FreeSWITCH Port" #!endif ####### Modules Section ######## # set paths to location of modules #!ifdef LOCAL_TEST_RUN mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "dispatcher.so" # loadmodule "modules/tm/tm.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org<sip%3Apinger(a)kamailio.org> ") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } #!ifdef WITH_FREESWITCH # save callee ID $avp(callee) = $rU; route(FSDISPATCH); #!endif # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif #!ifdef WITH_CFGSAMPLES /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } #!endif if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_FREESWITCH if(route(FSINBOUND)) return; #!endif #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } #!ifdef WITH_FREESWITCH # FreeSWITCH routing blocks route[FSINBOUND] { if($si== $sel(cfg_get.freeswitch.bindip) && $sp==$sel(cfg_get.freeswitch.bindport)) return 1; return -1; } route[FSDISPATCH] { if(!is_method("INVITE")) return; if(route(FSINBOUND)) return; # dial number selection switch($rU) { case /"^41$": # 41 - voicebox menu # allow only authenticated users if($au==$null) { sl_send_reply("403", "Not allowed"); exit; } $rU = "vm-" + $au; break; case /"^441[0-9][0-9]$": # starting with 44 folowed by 1XY - direct call to voice box strip(2); route(FSVBOX); break; case /"^433[01][0-9][0-9]$": # starting with 433 folowed by (0|1)XY - conference strip(2); break; case /"^45[0-9]+$": strip(2); break; default: # offline - send to voicebox if (!registered("location")) { route(FSVBOX); exit; } # online - do bridging prefix("kb-"); if(is_method("INVITE")) { # in case of failure - re-route to FreeSWITCH VoiceMail t_on_failure("FAIL_FSVBOX"); } } route(FSRELAY); exit; } route[FSVBOX] { if(!($rU=~"^1[0-9][0-9]+$")) return; prefix("vb-"); route(FSRELAY); } # Send to FreeSWITCH route[FSRELAY] { $du = "sip:" + $sel(cfg_get.freeswitch.bindip) + ":" + $sel(cfg_get.freeswitch.bindport); if($var(newbranch)==1) { append_branch(); $var(newbranch) = 0; } route(RELAY); exit; } #!endif #!ifdef WITH_FREESWITCH failure_route[FAIL_FSVBOX] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } if (t_check_status("486|408")) { # re-route to FreeSWITCH VoiceMail $rU = $avp(callee); $var(newbranch) = 1; route(FSVBOX); } } #!endif # sample config file for dispatcher module #!ifdef WITH_DISPATCHER modparam("dispatcher", "db_url", "mysql://openser:password@localhost /openser") route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); forward(); # t_relay(); } #!endif

3 2

Re: [SR-Users] avp_db_query returning wrong data
by Alex Balashov 07 Jan '11

07 Jan '11

On 11/23/2010 06:41 PM, Jijo wrote: > Yes it work with many UDP children. There were some concurrency issues with the Postgres module a while back, discovered after the 3.1.0 release. I remember something about it because we are extremely heavy users of the Postgres module as well. Unfortunately, I can't find the thread. Try the latest git master HEAD: git clone git://git.sip-router.org/sip-router cd sip-router make FLAVOUR=kamailio cfg make all And see if it changes. -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street NE 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

3 4

SIREMIS logging problem.
by Ricardo Martinez 06 Jan '11

06 Jan '11

Hello. I have Siremis 2.0 installed from scratch, but I’m unable to logging. This is the error that I’m getting from the logs : '01/05/2011','12:41:36','ERR','DATAOBJ','Query Error: SQLSTATE[42S02]: Base table or view not found: 1146 Table 'siremis.contact' doesn't exist','n/a' I was checking the SIREMIS db and I can’t found the table “contact”. Is this a bug?.. can someone help me? Thanks Regards, * * *Ricardo Martinez.-*

2 3

Kamailio Development Training, Barcelona, Feb 10-11, 2011
by Daniel-Constantin Mierla 05 Jan '11

05 Jan '11

Hello, for those interested in learning how to write C code for Kamailio & SER, the next development training class takes place in Barcelona, Spain, during February 10-11, 2011. You can find more details at: http://asipto.com/u/r Number of seats is limited, the acceptance policy is first come first served. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

1 0

ENUM changes?
by Klaus Darilion 05 Jan '11

05 Jan '11

Hi! I just found out that my ENUM config didn't worked any more - and it worked for sure some time ago. Were there any recent changes, e.g. from 3.0 to 3.1? I used if enum_query("e164.arpa.","+sip+voice:sip") { which should accept normal E2U+sip and E2U+voice:sip NAPTRs. But it doesn't work with kamailio 3.1.0. But enum_query("e164.arpa.") works. thanks klaus

4 3

Re: [SR-Users] RLS issue
by Klaus Darilion 05 Jan '11

05 Jan '11

1. Kamailio will send the request to the address of the list member (normal DNS lookups). 2. If there is a problem with OpenIMS then you should ask the OpenIMS guys, we are not familiar with OpenIMS. regards klaus Am 22.12.2010 12:57, schrieb "Andrés S. García Ruiz": > > Thanks a lot for your response! > > I've been thinking about it and there's something abnormal. Firs, the > RLS is suppose to send the initial SUBSCRIBE to the S-CSCF instead of > sending it to the I-CSCF. Then, I don't know why the S-CSCF translates > the SUBSCRIBE URI (sip:testuser02@open-ims.test) to > "sip:testuser01@155.54.190.245:8060;rinstance=9b7761b4bcaa4bd0". All the > routing is the default configuration of both cscfs and rls. > > How can I change the routing of the RLS in order it to send the > SUBSCRIBE messages to the S-CSCF? > > Here's the configuration file: > / > #!KAMAILIO > > #!ifdef ACCDB_COMMENT > ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL > DEFAULT ''; > #!endif > > ####### Defined Values ######### > > #!define WITH_DEBUG > #!define WITH_AUTH > #!define WITH_MYSQL > #!define WITH_USRLOCDB > #!define WITH_PRESENCE > #!define WITH_XCAPSRV > > > # *** Value defines - IDs used later in config > #!ifdef WITH_MYSQL > # - database URL - used to connect to database server by modules such > # as: auth_db, acc, usrloc, a.s.o. > #!define DBURL "mysql://openser:openserrw@localhost/openser" > #!endif > #!ifdef WITH_MULTIDOMAIN > # - the value for 'use_domain' parameters > #!define MULTIDOMAIN 1 > #!else > #!define MULTIDOMAIN 0 > #!endif > > # - flags > # FLT_ - per transaction (message) flags > # FLB_ - per branch flags > #!define FLT_ACC 1 > #!define FLT_ACCMISSED 2 > #!define FLT_ACCFAILED 3 > #!define FLT_NATS 5 > > #!define FLB_NATB 6 > #!define FLB_NATSIPPING 7 > > ####### Global Parameters ######### > > #!ifdef WITH_DEBUG > debug=4 > log_stderror=yes > #!else > debug=2 > log_stderror=no > #!endif > > memdbg=5 > memlog=5 > > log_facility=LOG_LOCAL0 > > fork=yes > children=4 > > /* uncomment the next line to disable TCP (default on) */ > #disable_tcp=yes > > > /* uncomment the next line to disable the auto discovery of local aliases > based on reverse DNS on IPs (default on) */ > #auto_aliases=no > > /* add local domain aliases */ > alias="open-ims.test" > > /* uncomment and configure the following line if you want Kamailio to > bind on a specific interface/port/proto (default bind on all available) */ > #listen=udp:10.0.0.10:5060 > > /* port to listen to > * - can be specified more than once if needed to listen on many ports */ > port=5060 > > #!ifdef WITH_TLS > enable_tls=yes > #!endif > > #!ifdef WITH_XCAPSRV > tcp_accept_no_cl=yes > #!endif > > ####### Custom Parameters ######### > > # These parameters can be modified runtime via RPC interface > # - see the documentation of 'cfg_rpc' module. > # > # Format: group.id = value 'desc' description > # Access: $sel(cfg_get.group.id) or @cfg_get.group.id > # > > #!ifdef WITH_PSTN > # PSTN GW Routing > # > # - pstn.gw_ip: valid IP or hostname as string value, example: > # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" > # > # - by default is empty to avoid misrouting > pstn.gw_ip = "" desc "PSTN GW Address" > #!endif > > > ####### Modules Section ######## > > # set paths to location of modules > #!ifdef LOCAL_TEST_RUN > mpath="modules_k:modules" > #!else > mpath="/usr/local/kamailio-3.1-xcap-rls/lib/kamailio/modules_k/:/usr/local/kamailio-3.1-xcap-rls/lib/kamailio/modules/" > #!endif > > #!ifdef WITH_MYSQL > loadmodule "db_mysql.so" > #!endif > > loadmodule "mi_fifo.so" > loadmodule "kex.so" > loadmodule "tm.so" > loadmodule "tmx.so" > loadmodule "sl.so" > loadmodule "rr.so" > loadmodule "pv.so" > loadmodule "maxfwd.so" > loadmodule "usrloc.so" > loadmodule "registrar.so" > loadmodule "textops.so" > loadmodule "siputils.so" > loadmodule "xlog.so" > loadmodule "sanity.so" > loadmodule "ctl.so" > loadmodule "mi_rpc.so" > loadmodule "acc.so" > # loadmodule "pua_mi.so" > # loadmodule "pua_usrloc.so" > > > #!ifdef WITH_AUTH > loadmodule "auth.so" > loadmodule "auth_db.so" > #!ifdef WITH_IPAUTH > loadmodule "permissions.so" > #!endif > #!endif > > #!ifdef WITH_ALIASDB > loadmodule "alias_db.so" > #!endif > > #!ifdef WITH_MULTIDOMAIN > loadmodule "domain.so" > #!endif > > #!ifdef WITH_PRESENCE > loadmodule "presence.so" > loadmodule "presence_xml.so" > loadmodule "pua.so" > loadmodule "rls.so" > #!endif > > #!ifdef WITH_NAT > loadmodule "nathelper.so" > loadmodule "rtpproxy.so" > #!endif > > #!ifdef WITH_TLS > loadmodule "tls.so" > #!endif > > #!ifdef WITH_ANTIFLOOD > loadmodule "htable.so" > loadmodule "pike.so" > #!endif > > #!ifdef WITH_XMLRPC > loadmodule "xmlrpc.so" > #!endif > > #!ifdef WITH_XCAPSRV > loadmodule "xhttp.so" > loadmodule "xcap_server.so" > #!endif > > # ----------------- setting module-specific parameters --------------- > > > # ----- mi_fifo params ----- > modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") > > > # ----- tm params ----- > # auto-discard branches from previous serial forking leg > modparam("tm", "failure_reply_mode", 3) > # default retransmission timeout: 30sec > modparam("tm", "fr_timer", 30000) > # default invite retransmission timeout after 1xx: 120sec > modparam("tm", "fr_inv_timer", 120000) > > > # ----- rr params ----- > # add value to ;lr param to cope with most of the UAs > modparam("rr", "enable_full_lr", 1) > # do not append from tag to the RR (no need for this script) > modparam("rr", "append_fromtag", 0) > > > # ----- registrar params ----- > modparam("registrar", "method_filtering", 1) > /* uncomment the next line to disable parallel forking via location */ > # modparam("registrar", "append_branches", 0) > /* uncomment the next line not to allow more than 10 contacts per AOR */ > #modparam("registrar", "max_contacts", 10) > > > # ----- acc params ----- > /* what special events should be accounted ? */ > modparam("acc", "early_media", 0) > modparam("acc", "report_ack", 0) > modparam("acc", "report_cancels", 0) > /* by default ww do not adjust the direct of the sequential requests. > if you enable this parameter, be sure the enable "append_fromtag" > in "rr" module */ > modparam("acc", "detect_direction", 0) > /* account triggers (flags) */ > modparam("acc", "log_flag", FLT_ACC) > modparam("acc", "log_missed_flag", FLT_ACCMISSED) > modparam("acc", "log_extra", > "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") > modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) > /* enhanced DB accounting */ > #!ifdef WITH_ACCDB > modparam("acc", "db_flag", FLT_ACC) > modparam("acc", "db_missed_flag", FLT_ACCMISSED) > modparam("acc", "db_url", DBURL) > modparam("acc", "db_extra", > "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") > #!endif > > > # ----- usrloc params ----- > /* enable DB persistency for location entries */ > #!ifdef WITH_USRLOCDB > modparam("usrloc", "db_url", DBURL) > modparam("usrloc", "db_mode", 2) > modparam("usrloc", "use_domain", MULTIDOMAIN) > #!endif > > > # ----- auth_db params ----- > #!ifdef WITH_AUTH > modparam("auth_db", "db_url", DBURL) > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("auth_db", "load_credentials", "") > modparam("auth_db", "use_domain", MULTIDOMAIN) > > # ----- permissions params ----- > #!ifdef WITH_IPAUTH > modparam("permissions", "db_url", DBURL) > modparam("permissions", "db_mode", 1) > #!endif > > #!endif > > > # ----- alias_db params ----- > #!ifdef WITH_ALIASDB > modparam("alias_db", "db_url", DBURL) > modparam("alias_db", "use_domain", MULTIDOMAIN) > #!endif > > > # ----- domain params ----- > #!ifdef WITH_MULTIDOMAIN > modparam("domain", "db_url", DBURL) > # use caching > modparam("domain", "db_mode", 1) > # register callback to match myself condition with domains list > modparam("domain", "register_myself", 1) > #!endif > > > #!ifdef WITH_PRESENCE > # ----- presence params ----- > modparam("presence", "db_url", DBURL) > > # ----- presence_xml params ----- > modparam("presence_xml", "db_url", DBURL) > modparam("presence_xml", "force_active", 1) > modparam("presence_xml", "integrated_xcap_server", 1) > modparam("rls", "db_url", DBURL) > modparam("rls", "xcap_table", "xcap") > modparam("rls", "rlsubs_table", "rls_watchers") > modparam("rls", "rlpres_table", "rls_presentity") > modparam("rls", "clean_period", 100) > modparam("rls", "waitn_time", 50) > modparam("rls", "max_expires", 10800) > modparam("rls", "hash_size", 9) > modparam("rls", "xcap_root", "http://ciervo.inf.um.es/xcap-root:5060") > modparam("rls", "integrated_xcap_server", 1) > modparam("rls", "to_presence_code", 10) > modparam("rls", "rls_event", "presence") > #!endif > > > #!ifdef WITH_NAT > # ----- rtpproxy params ----- > modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") > > # ----- nathelper params ----- > modparam("nathelper", "natping_interval", 30) > modparam("nathelper", "ping_nated_only", 1) > modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) > modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") > > # params needed for NAT traversal in other modules > modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") > modparam("usrloc", "nat_bflag", FLB_NATB) > #!endif > > > #!ifdef WITH_TLS > # ----- tls params ----- > modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") > #!endif > > #!ifdef WITH_ANTIFLOOD > # ----- pike params ----- > modparam("pike", "sampling_time_unit", 2) > modparam("pike", "reqs_density_per_unit", 16) > modparam("pike", "remove_latency", 4) > > # ----- htable params ----- > # ip ban htable with autoexpire after 5 minutes > modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") > #!endif > > #!ifdef WITH_XMLRPC > # ----- xmlrpc params ----- > modparam("xmlrpc", "route", "XMLRPC"); > modparam("xmlrpc", "url_match", "^/RPC") > #!endif > > #!ifdef WITH_XCAPSRV > # ----- xcap_server params ----- > modparam("xcap_server", "db_url", DBURL) > # modparam("rls", "integrated_xcap_server", 1) > #!endif > > ####### Routing Logic ######## > > > # Main SIP request routing logic > # - processing of any incoming SIP request starts with this route > route { > > # per request initial checks > route(REQINIT); > > # NAT detection > route(NAT); > > # handle requests within SIP dialogs > route(WITHINDLG); > > ### only initial requests (no To tag) > > # CANCEL processing > if (is_method("CANCEL")) > { > if (t_check_trans()) > t_relay(); > exit; > } > > t_check_trans(); > > # authentication > route(AUTH); > > # record routing for dialog forming requests (in case they are routed) > # - remove preloaded route headers > remove_hf("Route"); > if (is_method("INVITE|SUBSCRIBE")) > record_route(); > > # account only INVITEs > if (is_method("INVITE")) > { > setflag(FLT_ACC); # do accounting > } > > # dispatch requests to foreign domains > route(SIPOUT); > > ### requests for my local domains > > # handle presence related requests > route(PRESENCE); > > # handle registrations > route(REGISTRAR); > > if ($rU==$null) > { > # request with no Username in RURI > sl_send_reply("484","Address Incomplete"); > exit; > } > > # dispatch destinations to PSTN > route(PSTN); > > # user location service > route(LOCATION); > > route(RELAY); > } > > > route[RELAY] { > #!ifdef WITH_NAT > if (check_route_param("nat=yes")) { > setbflag(FLB_NATB); > } > if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { > route(RTPPROXY); > } > #!endif > > /* example how to enable some additional event routes */ > if (is_method("INVITE")) { > #t_on_branch("BRANCH_ONE"); > t_on_reply("REPLY_ONE"); > t_on_failure("FAIL_ONE"); > } > > if (!t_relay()) { > sl_reply_error(); > } > exit; > } > > # Per SIP request initial checks > route[REQINIT] { > #!ifdef WITH_ANTIFLOOD > # flood dection from same IP and traffic ban for a while > # be sure you exclude checking trusted peers, such as pstn gateways > # - local host excluded (e.g., loop to self) > if(src_ip!=myself) > { > if($sht(ipban=>$si)!=$null) > { > # ip is already blocked > xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); > exit; > } > if (!pike_check_req()) > { > xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); > $sht(ipban=>$si) = 1; > exit; > } > } > #!endif > > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > exit; > } > > if(!sanity_check("1511", "7")) > { > xlog("Malformed SIP message from $si:$sp\n"); > exit; > } > } > > # Handle requests within SIP dialogs > route[WITHINDLG] { > if (has_totag()) { > # sequential request withing a dialog should > # take the path determined by record-routing > if (loose_route()) { > if (is_method("BYE")) { > setflag(FLT_ACC); # do accounting ... > setflag(FLT_ACCFAILED); # ... even if the transaction fails > } > route(RELAY); > } else { > if (is_method("SUBSCRIBE") && uri == myself) { > # in-dialog subscribe requests > route(PRESENCE); > exit; > } > if ( is_method("ACK") ) { > if ( t_check_trans() ) { > # no loose-route, but stateful ACK; > # must be an ACK after a 487 > # or e.g. 404 from upstream server > t_relay(); > exit; > } else { > # ACK without matching transaction ... ignore and discard > exit; > } > } > sl_send_reply("404","Not here"); > } > exit; > } > } > > # Handle SIP registrations > route[REGISTRAR] { > if (is_method("REGISTER")) > { > if(isflagset(FLT_NATS)) > { > setbflag(FLB_NATB); > # uncomment next line to do SIP NAT pinging > ## setbflag(FLB_NATSIPPING); > } > if (!save("location")) > sl_reply_error(); > > exit; > } > } > > # USER location service > route[LOCATION] { > > #!ifdef WITH_ALIASDB > # search in DB-based aliases > alias_db_lookup("dbaliases"); > #!endif > > if (!lookup("location")) { > switch ($rc) { > case -1: > case -3: > t_newtran(); > t_reply("404", "Not Found"); > exit; > case -2: > sl_send_reply("405", "Method Not Allowed"); > exit; > } > } > > # when routing via usrloc, log the missed calls also > if (is_method("INVITE")) > { > setflag(FLT_ACCMISSED); > } > } > > # Presence server route > route[PRESENCE] { > > if(!is_method("PUBLISH|SUBSCRIBE")) > return; > > #!ifdef WITH_PRESENCE > > if (!t_newtran()) > { > sl_reply_error(); > exit; > }; > > if(is_method("PUBLISH")) > { > handle_publish(); > t_release(); > } > else if( is_method("SUBSCRIBE")) > { > $var(ret_code)= rls_handle_subscribe(); > > if($var(ret_code)== 10) { > handle_subscribe(); > } > t_release(); > > }else if(method=="NOTIFY") > { > rls_handle_notify(); > } > > exit; > #!endif > # if presence enabled, this part will not be executed > if (is_method("PUBLISH") || $rU==$null) > { > sl_send_reply("404", "Not here"); > exit; > } > return; > } > > # Authentication route > route[AUTH] { > return; > #!ifdef WITH_AUTH > if (is_method("REGISTER")) > { > # authenticate the REGISTER requests (uncomment to enable auth) > if (!www_authorize("$td", "subscriber")) > { > www_challenge("$td", "0"); > exit; > } > > if ($au!=$tU) > { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } else { > > #!ifdef WITH_IPAUTH > if(allow_source_address()) > { > # source IP allowed > return; > } > #!endif > # authenticate if from local subscriber > if (from_uri==myself) > { > if (!proxy_authorize("$fd", "subscriber")) { > proxy_challenge("$fd", "0"); > exit; > } > if (is_method("PUBLISH")) > { > if ($au!=$tU) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } else { > if ($au!=$fU) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } > > consume_credentials(); > # caller authenticated > } else { > # caller is not local subscriber, then check if it calls > # a local destination, otherwise deny, not an open relay here > if (!uri==myself) > { > sl_send_reply("403","Not relaying"); > exit; > } > } > } > #!endif > return; > } > > # Caller NAT detection route > route[NAT] { > #!ifdef WITH_NAT > force_rport(); > if (nat_uac_test("19")) { > if (method=="REGISTER") { > fix_nated_register(); > } else { > fix_nated_contact(); > } > setflag(FLT_NATS); > } > #!endif > return; > } > > # RTPProxy control > route[RTPPROXY] { > #!ifdef WITH_NAT > if (is_method("BYE")) { > unforce_rtp_proxy(); > } else if (is_method("INVITE")){ > force_rtp_proxy(); > } > if (!has_totag()) add_rr_param(";nat=yes"); > #!endif > return; > } > > # Routing to foreign domains > route[SIPOUT] { > if (!uri==myself) > { > append_hf("P-hint: outbound\r\n"); > route(RELAY); > } > } > > # PSTN GW routing > route[PSTN] { > #!ifdef WITH_PSTN > # check if PSTN GW IP is defined > if (strempty($sel(cfg_get.pstn.gw_ip))) { > xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); > return; > } > > # route to PSTN dialed numbers starting with '+' or '00' > # (international format) > # - update the condition to match your dialing rules for PSTN routing > if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) > return; > > # only local users allowed to call > if(from_uri!=myself) { > sl_send_reply("403", "Not Allowed"); > exit; > } > > $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); > > route(RELAY); > exit; > #!endif > > return; > } > > # XMLRPC routing > #!ifdef WITH_XMLRPC > route[XMLRPC] > { > # allow XMLRPC from localhost > if ((method=="POST" || method=="GET") > && (src_ip==127.0.0.1)) { > # close connection only for xmlrpclib user agents (there is a bug in > # xmlrpclib: it waits for EOF before interpreting the response). > if ($hdr(User-Agent) =~ "xmlrpclib") > set_reply_close(); > set_reply_no_connect(); > dispatch_rpc(); > exit; > } > send_reply("403", "Forbidden"); > exit; > } > #!endif > > # Sample branch router > branch_route[BRANCH_ONE] { > xdbg("new branch at $ru\n"); > } > > # Sample onreply route > onreply_route[REPLY_ONE] { > xdbg("incoming reply\n"); > #!ifdef WITH_NAT > if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) > && status=~"(183)|(2[0-9][0-9])") { > force_rtp_proxy(); > } > if (isbflagset("6")) { > fix_nated_contact(); > } > #!endif > } > > # Sample failure route > failure_route[FAIL_ONE] { > #!ifdef WITH_NAT > if (is_method("INVITE") > && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { > unforce_rtp_proxy(); > } > #!endif > > if (t_is_canceled()) { > exit; > } > > # uncomment the following lines if you want to block client > # redirect based on 3xx replies. > ##if (t_check_status("3[0-9][0-9]")) { > ##t_reply("404","Not found"); > ## exit; > ##} > > # uncomment the following lines if you want to redirect the failed > # calls to a different new destination > ##if (t_check_status("486|408")) { > ## sethostport("192.168.2.100:5060"); > ## append_branch(); > ## # do not set the missed call flag again > ## t_relay(); > ##} > } > > #!ifdef WITH_XCAPSRV > # #!define WITH_XHTTPAUTH > event_route[xhttp:request] { > xdbg("===== xhttp: request [$rv] $rm => $hu\n"); > > #!ifdef WITH_XHTTPAUTH > if (!www_authorize("xcap", "subscriber")) > { > www_challenge("xcap", "0"); > exit; > } > #!endif > xdbg("===== xhttp: Antes del if de $hu\n"); > if($hu=~"^/xcap-root/") > { > xdbg("===== xhttp: Dentro del if de $hu\n"); > set_reply_close(); > set_reply_no_connect(); > # xcap ops > $xcapuri(u=>data) = $hu; > if($xcapuri(u=>xuid)=~"^sip:.+@.+") > $var(uri) = $xcapuri(u=>xuid); > else if($xcapuri(u=>xuid)=~".+@.+") > $var(uri) = "sip:" + $xcapuri(u=>xuid); > else > $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; > xlog("===== xhttp: $xcapuri(u=>auid) : $xcapuri(u=>xuid)\n"); > if($xcapuri(u=>auid)=="xcap-caps") > { > $var(xbody) = > "<?xml version='1.0' encoding='UTF-8'?> > <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> > <auids> > <auid>rls-services</auid> > <auid>pidf-manipulation</auid> > <auid>xcap-caps</auid> > <auid>resource-lists</auid> > <auid>pres-rules</auid> > <auid>org.openmobilealliance.pres-rules</auid> > </auids> > <extensions> > </extensions> > <namespaces> > <namespace>urn:ietf:params:xml:ns:rls-services</namespace> > <namespace>urn:ietf:params:xml:ns:pidf</namespace> > <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> > <namespace>urn:ietf:params:xml:ns:resource-lists</namespace> > <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> > </namespaces> > </xcap-caps>"; > xhttp_reply("200", "ok", "application/xcap-caps+xml", > "$var(xbody)"); > exit; > } > > #!ifdef WITH_XHTTPAUTH > # be sure auth user access only its documents > if ($au!=$(var(uri){uri.user})) { > xhttp_reply("403", "Forbidden", "text/html", > "<html><body>$si:$sp</body></html>"); > exit; > } > > #!endif > xdbg("===== xhttp: Antes del switch $rm => $hu\n"); > switch($rm) { > case "PUT": > xcaps_put("$var(uri)", "$hu", "$rb"); > if($xcapuri(u=>auid)=~"pres-rules") > { > xlog("===== xhttp put: refreshing watchers for $var(uri)\n"); > pres_update_watchers("$var(uri)", "presence"); > pres_refresh_watchers("$var(uri)", "presence", 1); > } > exit; > break; > case "GET": > xlog("===== xhttp: get $var(uri) => $hu\n"); > xcaps_get("$var(uri)", "$hu"); > exit; > break; > case "DELETE": > xcaps_del("$var(uri)", "$hu"); > if($xcapuri(u=>auid)=~"pres-rules") > { > xlog("===== xhttp del: refreshing watchers for $var(uri)\n"); > pres_update_watchers("$var(uri)", "presence"); > pres_refresh_watchers("$var(uri)", "presence", 1); > } > exit; > break; > } > } > > # http ops > xhttp_reply("200", "ok", "text/html", > "<html><body>OK: $si:$sp</body></html>"); > exit; > } > #!endif/ > > > Regards, > Andrés. > > El 21/12/10 10:22, Klaus Darilion escribió: >> Am 16.12.2010 13:36, schrieb "Andrés S. García Ruiz": >>> >>> B.5) P-CSCF ---- SUBSCRIBE ----> Presentity??? >>> >>> SUBSCRIBE sip:testuser01@155.54.190.245:8060;rinstance=9b7761b4bcaa4bd0 >>> SIP/2.0 >>> Record-Route: <sip:mt@pcscf.open-ims.test:4060;lr> >>> Record-Route: <sip:mt@scscf.open-ims.test:6060;lr> >>> Via: SIP/2.0/TCP 155.54.210.134:4060;branch=z9hG4bK07df.baa7cf24.0 >>> Via: SIP/2.0/UDP >>> 155.54.210.135:6060;received=155.54.210.135;rport=6060;branch=z9hG4bK07df.8a35e4f3.0 >>> >>> >>> Via: SIP/2.0/UDP 155.54.210.136;branch=z9hG4bK07df.3688b985.0 >>> Via: SIP/2.0/UDP 155.54.190.245;branch=z9hG4bK07df.964e0ba7.0 >>> To: sip:testuser01@open-ims.test >>> From: >>> sip:restricted_areas@open-ims.test;tag=533cb9e91f4b999cf76861cbb9ed54ed-32d5 >>> >>> >>> CSeq: 10 SUBSCRIBE >>> Call-ID: 7fd8dfdd-21694(a)127.0.0.1 >>> Content-Length: 0 >>> User-Agent: kamailio (3.2.0-dev1 (i386/linux)) >>> Max-Forwards: 14 >>> Event: presence >>> Contact: <sip:155.54.190.245:5060;transport=udp> >>> Expires: 10810 >>> Max-Forwards: 70 >>> Support: eventlist >>> P-Called-Party-ID: <sip:testuser01@open-ims.test> >>> >>> >>> The step B.5 is sent directly to the presentity testuser01. Instead of >>> that, I thought that message was suppose to be sent to the presence >>> server in the URI that is shown in Contact header. >> >> >> If the P-CSCF is not forwarding the request as supposed, I guess you >> have to fix the routing on the P-CSCF. >> >> klaus >

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Re: [SR-Users] sr-users Digest, Vol 68, Issue 7
by Javier Gallart 05 Jan '11

05 Jan '11

Thanks Alex, it looks like we'll have to move to mysql in that case (we are much more experienced in postgres). Is this feature expected to be added for db_postgres? Regards Javi > > Message: 6 > Date: Tue, 04 Jan 2011 11:54:20 -0500 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: Re: [SR-Users] Reconnection to db > To: sr-users(a)lists.sip-router.org > Message-ID: <4D2350BC.8080207(a)evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > On 01/04/2011 09:46 AM, Javier Gallart wrote: > > Hi list > > > > we have been testing the mtree module. We do periodical reloads of the > > mtree table with the "mi mt_reload" command through xmlrpc. It works > > fine except when the db is restarted. If we manually restart the db and > > then try the mt_reload commands we get this error: > > Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: > > ERROR: db_postgres [km_dbase.c:152]: 0xa6a490 PQsendQuery Error: server > > closed the connection unexpectedly#012#011This probably means the server > > terminated abnormally#012#011before or while processing the request.#012 > > Query: select partition,prefix,route from trees order by partition > > Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: > > ERROR: <core> [db_query.c:101]: error while submitting query > > Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: > > ERROR: mtree [mtree_mod.c:622]: Error while querying db > > Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: > > ERROR: mtree [mtree_mod.c:738]: cannot re-load info from database > > > > Is there a way to reconnect to the db before sending that command? > > Failure to automatically reconnect to Postgres is one of the > distinguishing features of db_postgres in contrast to db_mysql for the > moment. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > > > ------------------------------ > > _______________________________________________ > sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > End of sr-users Digest, Vol 68, Issue 7 > *************************************** >

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Use of SIPP along with kamailio - REGISTER followed by INVITE not working
by Stephen McVarnock 05 Jan '11

05 Jan '11

Guys, I am not sure in which world to expect to get an answer to this so I am sending this to both SIPP and SR-User mailing lists. I am trying to get an SIPP end-point to register with a kamailio proxy. I do this by sending the REGISTER from the SIPP end-point, which is successfully responded to by kamailio with a 200 OK. No problems there. The problem is then with sending an INVITE from kamailio to the SIPP instance. This is where the problems arise. 1) If I include the expected INVITE in the same xml scenario file as the REGISTER/200, then SIPP will complain about the INVITE as it will have a different Call-ID from the REGISTER method. I have a hunt around the forums and this was acknowledged as an issue but as far as I can see this has not been fixed/implemented. The following link will take you to a proposed fix for this issue. http://sipp.sourceforge.net/wiki/index.php/Patches#Pre.2FPost_scenarios Does anyone know what the current state of play is for this proposed patch or if there is another way to get around this issue? 2) I tried to REGISTER the SIPP endpoint in a single xml scenario file with kamailio. This works as per usual. I then killed the SIPP instance and ran a new SIPP script listening on the same port before trying to send the INVITE to it. I expected this to work as the SIPP scenarios (both sending REGISTER and expecting INVITE) listened on the same port. However, the INVITE was not received by the SIPP endpoint. Can anyone think of a reason for this? If any further info is required then please let me know. I have spent quite a bit of time on this and its an itch that needs scratched. Thanks in advance, Steve. -- ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Stephen McVarnock, AePONA Ltd, Interpoint Building, 20-24 York Street, Belfast BT15 1AQ Tel: +44 (0)28 9026 9114 Fax: +44 (0)28 9026 9111 http://www.aepona.com ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

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