- sr-users - kamailio.org

[Kamailio-Users] Dialog profile size not resetting in 1.4.4.
by Alex Balashov 21 May '09

21 May '09

Greetings, I've run into an issue with 1.4.4 in which the following appears to be happening, and I'm yet to determine the cause: The total amount of open inbound calls (dialogs) is tracked with the dialog module by putting them in a global profile. Because of a segfault bug with profiles with "no values," a workaround is used wherein a profile "with" values is created and the hash key is always the same value (i.e. "1"). The dialog tracking flag is 2: modparam("dialog", "dlg_flag", 2) And profile looks like this: modparam("dialog", "profiles_with_value", "globalinbound;specificinbound") Anyway, inbound INVITEs get handed off to a route (route[1]) that performs some logic like this: route[1] { # Check if DID is assigned to an account with database # and store value in AVP, along with other values. if($avp(S:did_assigned) == "0") { sl_send_reply("404", "Not Found"); exit; } get_profile_size("specificinbound", "$avp(S:account_id)", "$var(a_concurrent_calls)"); if($var(a_concurrent_calls) >= $avp(S:port_limit)) { sl_send_reply("486", "Busy"); exit; } setflag(2); # Track dialog. set_dlg_profile("globalinbound", "1"); set_dlg_profile("specificinbound", "$avp(S:account_id)"); } What appears to be happening is that calls coming into unassigned numbers ($avp(S:did_assigned) == "0") increase the size of the "globalinbound" profile, which is something of a mystery to me since I am neither setting flag 2 nor calling set_dlg_profile() until further down in the logic! And after the 404 Not Found reply is sent, the profile size is not decremented. Is the use of statefully tracked replies (t_reply()) required in order to propagate dialog state through to the dialog module? My impression is this was not the case. More pressing concern is that profile size is increased and not decreased; in my interpretation, such calls should not be added to the 'globalinbound' profile at all. I have not had the opportunity to try to reproduce this with 1.5.x. Thanks! -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

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[Kamailio-Users] "Detached" timer messages.
by Alex Balashov 21 May '09

21 May '09

Greetings, I was wondering if someone fluent in the anatomy of the TM module can tell me what this condition actually implies in a commonsensical sort of way, from a user perspective -- it is on line 861 in modules/tm/timer.c: /* check first if we are on the "detached" timer_routine list, * if so do nothing, the timer is not valid anymore * (sideffect: reset_timer ; set_timer is not safe, a reseted timer * might be lost, depending on this race condition ) */ if (new_tl->timer_list==DETACHED_LIST){ LM_CRIT("set_timer for %d list called on a \"detached\" " "timer -- ignoring: %p\n", list_id, new_tl); goto end; } Reason I am wondering is that I got a few of these error messages in my logs: May 20 18:23:50 ser1 /usr/local/sbin/kamailio[25232]: CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" timer -- ignoring: 0x2a96cc0d50 And was not sure what to make of them. Thank you! -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

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[Kamailio-Users] Mailing list questions
by Kareem Hamdy 21 May '09

21 May '09

Is there an easier way to search through the mailing list other than going to http://lists.kamailio.org/pipermail/users/, and searching in my Firefox based on the subject name? Thanks

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[Kamailio-Users] RLS - is rls-services supported ?
by Pascal Maugeri 21 May '09

21 May '09

Hi I'm doing an integration of the RLS module for Kamailio 1.5 and a third-party XCAP server. I see that when my client sends a SIP SUBSCRIBE to a list that is defined into the document rls-services, I only see the traffic from Kamailio/RLS to the XCAP server to retrieve a resource-lists document, nothing about the rls-services document ?! Does RLS support rls-services ? If not what lists should contain the resource-lists document ? Cheers Pascal

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Re: [Kamailio-Users] XCAP_CLIENT failing to fetch/parse resource-lists document
by Daniel-Constantin Mierla 21 May '09

21 May '09

On 05/20/2009 01:04 PM, Pascal Maugeri wrote: > > that is caused by CURLE_WRITE_ERROR. > > Do you get a pkg memory or so error before those pasted above? > > > Well I only get the following line before: > > May 20 13:59:52 sandbox /usr/sbin/kamailio[11424]: > WARNING:core:fm_free: free(0) called > > [...] > > Any idea of what could go wrong ? Do you get the error for all xcap queries? Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Serusers] SER fail authentication using RADIUS
by Leon Li 21 May '09

21 May '09

Hi, I got the following error when using FreeRADIUS to authenticate SER registration. May 21 04:46:57 ser[3352]: REGISTER May 21 04:46:57 ser[3352]: REGISTER: challenging user May 21 04:46:57 ser[3354]: REGISTER May 21 04:46:57 ser[3354]: pre_auth(): Credentials received are not filled properly What does this mean? Also, I don't see any RADIUS traffic generated by SER. Please assist. Thanks Leon

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[Kamailio-Users] DTMF and Kamailio
by Simon Muyal 20 May '09

20 May '09

Hello, I have the following topology: IPphones <-> IPBX-1 <-> <-> IPBX-A <-> IPphones IPphones <-> IPBX-2 <-> KAMAILIO <-> IPBX-B <-> IPphones ... IPphones <-> IPBX-N <-> IPBX-X <-> IPphones version: kamailio 1.4.1-notls (i386/linux) I experienced problems regarding DTMF digits transport. DTMF is transported through RTP data and it is not recognized at the end. Does anyone could recommend me how to force the negotiation between Kamailio and IPBXs to be sure that IPphones and IPBXs will use another method to transport DTMF (like OOB SIP INFO or RFC2833) Many thanks, Simon -- -------------------------------------------------------------- Simon Muyal Suivi des Services aux Usagers (SSU) Services operations & support Réseau National de Télécommunications pour la Technologie, l'Enseignement et la Recherche Tel: +33 (0) 1 53 94 20 40 | GIP RENATER Fax: +33 (0) 1 53 94 20 41 | c/o ENSAM E-mail: muyal(a)renater.fr | 151 Boulevard de l'Hôpital http://www.renater.fr | 75013 PARIS --------------------------------------------------------------

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[Kamailio-Users] Load Based Load Balancing
by Kareem Hamdy 20 May '09

20 May '09

Hello: I've been going through your lists and your documentation, and I'm unable to find how you set up load based load balancing. To further illustrate: For example, let's say we have the following live calls:1,2,3,4,5,6,7 Round Robin Works Well ---------------------- +-----+ | SER | +-----+ | | +--------------+ |-------| FreeSWITCH 1 | [Calls 1,3,5,7] | +--------------+ | | +--------------+ |-------| FreeSWITCH 2 | [Calls 2,4,6] +--------------+ But let's say some of the calls end significantly quicker than the others, how can I send calls by their load (current calls, cpu, etc) For example, let's say we have the following live calls:1,2,3,4,5,6,7 - Let's say calls 4 and 5 lasted about 30 seconds, while the others lasted about 5 minutes. How would I be able to route calls to the box that's least used - in this example, "FreeSWITCH 2" at any given time had the least number of calls, so calls 4,5, and 6 were dispatched to it. Load Based Load Balancing ------------------------- +-----+ | SER | +-----+ | | +--------------+ |-------| FreeSWITCH 1 | [Calls 1,3,7] | +--------------+ | | +--------------+ |-------| FreeSWITCH 2 | [Calls 2,6] +--------------+ Thanks, Kareem

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[Kamailio-Users] Fwd: Debian distributions
by David 20 May '09

20 May '09

To: users(a)listskamailio.org Debian used to pack with OpenSER, does Kamailio ?

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Re: [Kamailio-Users] [Kamailio-Devel] Bugs on "kamctl" related to CR??? - Moving from Devel to User List
by Edson - Lists 20 May '09

20 May '09

Hi, Ovidiu I'm replicating this trade on User list, as You suggested, but take look on what I had done: Kamailio-151:/etc/kamailio # kamdbctl reinit kamailio MySQL password for root: INFO: Database kamailio deleted INFO: test server charset INFO: creating database kamailio ... INFO: Core Kamailio tables succesfully created. Install presence related tables? (y/n): y INFO: creating presence tables into kamailio ... INFO: Presence tables succesfully created. Install tables for imc cpl siptrace domainpolicy carrierroute userblacklist htable purple? (y/n): y INFO: creating extra tables into kamailio ... INFO: Extra tables succesfully created. Kamailio-151:/etc/kamailio # kamctl cr addrt 0 'XXXX' ERROR 1146 (42S02) at line 1: Table 'kamailio.route_tree' doesn't exist ERROR: cr - SQL Error Kamailio-151:/etc/kamailio # and the error/bug persist. Even with a clean DB definition... Where could the problem be? I understand that it's on the "kamctl" script, but I can be wrong... Edson. Ovidiu Sas escreveu: > This is a question for users, not for devel. > Please install the extra tables via kamdbctl. > Take a look also at /etc/kamailio/kamctlrc where the extra tables are defined. > > > Regards, > Ovidiu Sas > > On Tue, May 19, 2009 at 4:25 PM, Edson - Lists <4lists(a)gmail.com> wrote: >> Hi, Guys... >> >> After some tests, I realize that CarrierRoute will feet better on we >> want to do, but when trying to include a carrier, I got: >> >> # kamctl cr addrt 1 XXXX >> ERROR 1146 (42S02) at line 1: Table 'kamailio.route_tree' doesn't exist >> ERROR: cr - SQL Error >> >> Then I try to include a route by hand with the commands: >> >> INSERT INTO carrier_name (carrier) VALUES ("default"); >> INSERT INTO domain_name (domain) VALUES ("default"); >> INSERT INTO carrierroute (carrier,domain,scan_prefix,flags,mask,prob,strip,rewrite_host,rewrite_prefix,rewrite_suffix,description) >> VALUES("1","1","1000","0","0","1","0","PEER-01","","","FRANCE"); >> >> And try to see the results, but I got: >> >> # kamctl cr show >> cr routing tree >> ERROR 1146 (42S02) at line 1: Table 'kamailio.route_tree' doesn't exist >> cr routes >> >> Now is this resulting from bug(s) in the "kamctl" script or from wrong >> definitions on the "kamctlrc" file? >> >> Edson. >> >> >> _______________________________________________ >> Kamailio (OpenSER) - Devel mailing list >> Devel(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/devel >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/devel >> >

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[Kamailio-Users] rtpproxy error
by Chandrakant Solanki 20 May '09

20 May '09

Hi I have found following errors... Anybody how it occurs... Is there any special way to install rtpproxy..?? May 20 05:01:33 asterisk4 /sbin/kamailio[24792]: ERROR:nathelper:send_rtpp_command: can't connect to RTP proxy May 20 05:01:33 asterisk4 /sbin/kamailio[24792]: ERROR:nathelper:send_rtpp_command: proxy <unix:/var/run/rtpproxy.sock> does not respond, disable it May 20 04:11:40 asterisk4 /sbin/kamailio[24525]: WARNING:nathelper:rtpp_test: can't get version of the RTP proxy May 20 04:11:40 asterisk4 /sbin/kamailio[24525]: WARNING:nathelper:rtpp_test: support for RTP proxy <unix:/var/run/rtpproxy.sock> has been disabled temporarily -- Regards, Chandrakant Solanki

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[Kamailio-Users] new feature: event_route
by Daniel-Constantin Mierla 20 May '09

20 May '09

Hello, I just introduced support for a new type of route in configuration file - event_route. The prototype for it is: event_route[groupid:eventid] { [actions] } The main purpose for it is to allow modules (and core) to be able to execute code written in configuration file when a specific event happens, without altering the config grammar. Also, in short term, the modules that fire events should be able to impose what type of actions can be used in the respective event_route - now functions allowed in request route can be used. First module that uses this feature is htable, executing an event route only once, when all modules were initialized. A typical use case is to initialize some items in a hash table. Example: modparam("htable", "htable", "a=>size=4;") event_route[htable:mod-init] { $sht(a=>calls-to::10.10.10.10) = 0; $sht(a=>max-calls-to::10.10.10.10) = 100; } route { if(is_method("INVITE") && !has_totag()) { switch($rd) { case "10.10.10.10": lock("calls-to::10.10.10.10"); $sht(a=>calls-to::10.10.10.10) = $sht(a=>calls-to::10.10.10.10) + 1; unlock("calls-to::10.10.10.10"); if($sht(a=>calls-to::10.10.10.10)>$sht(a=>max-calls-to::10.10.10.10)) { sl_send_reply("500", "To many calls to .10"); exit; } break; ... } } } This system will be used to get kamailio's error_route and local_route functionalities. Other cases I have in mind now are: - auto-expired dialog route introduced in kamailio 1.5.0 to become event_route[dialog:auto-expired] - call event route when a location record has expired on timer: event_route[usrloc:auto-expired] making available via PV details of expired contact - rtimer module routes to become event routes Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] XCAP_CLIENT failing to fetch/parse resource-lists document
by Pascal Maugeri 20 May '09

20 May '09

I see that xcap_client is failing to parse a document from the XCAP server: May 20 12:41:47 sandbox /usr/sbin/kamailio[11280]: ERROR:xcap_client:send_http_get: while performing curl option May 20 12:41:47 sandbox /usr/sbin/kamailio[11280]: ERROR:rls:get_resource_list: while fetching data from xcap server May 20 12:41:47 sandbox /usr/sbin/kamailio[11280]: ERROR:rls:rls_handle_subscribe: while attepmting to get a resource list May 20 12:41:47 sandbox /usr/sbin/kamailio[11280]: ERROR:rls:rls_handle_subscribe: occured in rls_handle_subscribe I know the document is retrieved by the XCAP client (I see the HTTP traffic). The document content is very simple: <?xml version="1.0" encoding="UTF-8"?> <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> </resource-lists> >From that point I'm a bit confused on how to troubleshoot this issue. How could I get more log from curl ? Regards, Pascal

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[Kamailio-Users] auth to a provider ?
by BERGANZ François 20 May '09

20 May '09

Hello all, I need that Kamailio auth to my provider how to auth kamailio to my provider? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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[Kamailio-Users] Kamailio Callgroup
by Andre Buchwald 19 May '09

19 May '09

Hi, is it possible to configure a callgroup by Kamailio? UA_1 calls UA_2 that is a callgroup-member. In this case all members receive "invite" from UA_1 and if one member accepts the "invite" all other invites should be cancelled. Thanks

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[Kamailio-Users] Realtime Integration Of Asterisk 1.4 With Kamailio 1.5.x
by Gavin Henry 19 May '09

19 May '09

This has just appeared on voip-info via my Google Alerts. It will answer a *lot* of questions that appear on the list: http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+Wit… (Written by the Daniel so you know it works!) Cheers.

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Re: [Kamailio-Users] VoiceMail MWI
by jh＠tutpro.com 19 May '09

19 May '09

Chandrakant Solanki writes: > But how can i use... can i have example or .cfg file for same... you send publish requests to kamailio using mi/xmlrpc commands. there is nothing to configure there. in addition, you of course need to process mwi subscribe requests from your users. -- juha

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[Serusers] VoiceMail MWI
by Chandrakant Solanki 19 May '09

19 May '09

Hi all, I am newbie to kamailio and asterisk... I am using ... *Asterisk 1.6.0.5 on CentOS 5* (172.18.100.65) *kamailio 1.5.0-notls on CentOS 5* (172.16.0.10) I am successfully registered my sip phone with kamailio server and call were flow from one number to another. I have multiple Asterisk Server and one Kamailio server... when user dial number, but unable to reach he/she can put voicemail for that user on asterisk server where he/she can called... After a few minutes (say 10 minutes) i will execute one script in cron (every asterisk server) that will find voice mail from and put it on centeralized voicemail server, which is same as above configuration... How can i use MWI ... to indicate user that he/she has voicemaili in his/her mailbox... What changes should be requiered at kamailio site as well as asterisk side... Plz help me out. Thanks in advance -- Regards, Chandrakant Solanki

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[Serusers] FreeRADIUS authentication via radiusclient-ng
by Leon Li 19 May '09

19 May '09

Hi, I installed SER and FreeRADIUS, radiusclient-ng. However, SER seems never talk with FreeRADIUS, as I found no RADIUS auth logs. After digging radiusclient.conf, I found it calls FreeRADIUS using a script (/usr/local/sbin/login.radius), but this script does nothing only output as below. $ sudo /usr/local/sbin/login.radius This is the dummy login.radius script. If you want that this script does something useful you'll have to replace it. The following RADIUS environment variables are set: Bye, bye. How can I configure radiusclient-ng to call FreeRADIUS when there is an authentication request? Thanks Leon

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[Kamailio-Users] RLS with XCAP server - document changes notification
by Pascal Maugeri 18 May '09

18 May '09

Hi I am planning to use the RLS module with an external XCAP server (integrated_xcap_server=0) to store users resource-lists documents. My question is related to the changes of a such resource-lists document and how the RLS server behaves. I've read carrefully the module documentation<http://kamailio.org/docs/modules/1.5.x/rls.html#id2469762>but this point remains unclear. Let's say that user U1 has subscribed to her resource-lists, stored in a XCAP server, that contains contacts U2 and U3. U1 receives notifications of status updates of U2 and U3. If U1 adds a new contact U4 to the resource-lists (eg. using XCAP), does the RLS detects this change and subscribes to U4 status updates ? Regards, Pascal

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[Serusers] Warning: run_failure_handlers - ser rel2_0_0
by Tomasz Zieleniewski 18 May '09

18 May '09

Hi I quite often receive such warning: ser: Warning: run_failure_handlers: no negative handler (0, 166) What does it mean? Should I bother at all? Thank You in advance. Best Regards, - Tomasz Zieleniewski

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[Kamailio-Users] error : CRITICAL **: pecan_stream_write_full: assertion `stream' failed in kamailio
by rachid 15 May '09

15 May '09

Hello, I use kamailio to communicate between a sip client and a MSN client but when a sip client send a message to MSN client, the kamailio print me this message:CRITICAL **: pecan_stream_write_full: assertion `stream' failed someone can tell me where is the probleme? thank you -- ------------------------------------------------------------ BAHA Rachid Elève ingénieur en 3ème année à l'INPT, Rabat GSM:(+33) 0619787609

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[Kamailio-Users] early media and first SIP/Kamailio Configuration
by bahanfar＠hotmail.de 15 May '09

15 May '09

Hi, I am newcomer in SIP and Kamailio Wold an have a question. I have two IP Telefones and would like to realize with Kamailio an IP Communication. The Problem: a. The UA 1 calls UA 2 b. UA 2 start recording the video form UA 1, before a SIP call is established (early media) How can I do that? Can anybody help me or can give me any Info about example application. Thank you very much. Bahanfar _________________________________________________________________ http://redirect.gimas.net/?n=M0905xIE8_MSN4 Internet Explorer 8 jetzt für MSN - kostenlos!

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Re: [Kamailio-Users] kamailio multi-domain
by karhu 15 May '09

15 May '09

>You mean you've added the users to the subscriber table? Yes i added User1 and User2 in the subscriber's mysql table. Xlite conf : usr : User1 psw : xxxx Domain : 10.10.10.2 usr: User2 psw : xxxx Domain : 192.168.0.2 In mysql table i added : User2 with xxx psw and 192.168.0.2 domain. (don't really know if it is the righ way...!?) It seems the registering request are fine, below are the wireshark captures for user1 and user2 : http://www.nabble.com/file/p23555745/User1_Register.jpg http://www.nabble.com/file/p23555745/User2_Register.jpg Etheir i try to call from User2 to User1 or User1 to User2 i got nothing. On XliteUser2 i tried for the call : User1(a)10.10.10.2 >You've enabled auth in the cfg given below, so this is the expected behaviour. >Your UA should reply with the proper credentials, then the call should be >established. Take a look to the log file of the kamailio, and do a sip trace >(e.g. with ngrep -W byline port 5060) to get more information about the >eventual problem. Below ngrep -W byline port 5060 capture, User1 call User2 INVITE sip:boby@10.10.10.2 SIP/2.0. Via: SIP/2.0/UDP 10.10.10.3:41394;branch=z9hG4bK-d87543-18094a3f30346a6c-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:1000@10.10.10.3:41394>. To: "boby"<sip:boby@10.10.10.2>. From: "toto"<sip:1000@10.10.10.2>;tag=89767174. Call-ID: MzE3NDc5YzMwNDI2YjdjMWE4Zjg2Y2RlYTQ3YTZiMzY.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username="1000",realm="10.10.10.2",nonce="4a0d276f000000535a5a47a45038d6cbfab9392dd8d576cf",uri="sip:boby@10.10.10.2",response="87afe587be968300f615006a989c9c7f",algorithm=MD5. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 534. . v=0. o=- 2 2 IN IP4 10.10.10.3. s=CounterPath X-Lite 3.0. c=IN IP4 10.10.10.3. t=0 0. m=audio 20562 RTP/AVP 107 119 0 98 8 3 101. a=alt:1 1 : wbrcVqeO gWxBJcbu 10.10.10.3 20562. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:119 BV32-FEC/16000. a=rtpmap:98 iLBC/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. m=video 31202 RTP/AVP 115 34. a=alt:1 1 : Ph3NZX4n 9x8Vo2nU 10.10.10.3 31202. a=fmtp:115 QCIF=1 I=1 J=1 K=1 MAXBR=1960. a=fmtp:34 QCIF=1 MAXBR=1960. a=rtpmap:115 H263-1998/90000. a=rtpmap:34 H263/90000. a=sendrecv. # U 10.10.10.2:5060 -> 10.10.10.3:41394 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 10.10.10.3:41394;branch=z9hG4bK-d87543-18094a3f30346a6c-1--d87543-;rport=41394. To: "boby"<sip:boby@10.10.10.2>;tag=0c6c077910af0e0627388e55f0ad5733-3061. From: "toto"<sip:1000@10.10.10.2>;tag=89767174. Call-ID: MzE3NDc5YzMwNDI2YjdjMWE4Zjg2Y2RlYTQ3YTZiMzY.. CSeq: 2 INVITE. Server: Kamailio (1.5.0-notls (x86_64/linux)). Content-Length: 0. . # U 10.10.10.3:41394 -> 10.10.10.2:5060 ACK sip:boby@10.10.10.2 SIP/2.0. Via: SIP/2.0/UDP 10.10.10.3:41394;branch=z9hG4bK-d87543-18094a3f30346a6c-1--d87543-;rport. To: "boby"<sip:boby@10.10.10.2>;tag=0c6c077910af0e0627388e55f0ad5733-3061. From: "toto"<sip:1000@10.10.10.2>;tag=89767174. Call-ID: MzE3NDc5YzMwNDI2YjdjMWE4Zjg2Y2RlYTQ3YTZiMzY.. CSeq: 2 ACK. Content-Length: 0. . Cheers, Karhu -- View this message in context: http://www.nabble.com/kamailio-multi-domain-tp23539467p23555745.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] Error when connecting to mysql
by Alex R.S.M 15 May '09

15 May '09

Hi, I am getting the following error when kamailio wants to connect to mysql within a simple perl program. "mysql" module is already loaded in the config file. Is there any modparam for mysql that has to be set? ERROR:core:XS_OpenSER__Message_log: perl error: Can't locate object method "connect" via package "DBI" at /etc/openser/test.pl line 17. simple perl program: #!/usr/bin/perl use OpenSER; use OpenSER::Constants; sub run { ??? } 1 Thanks, A. R. _________________________________________________________________ Insert movie times and more without leaving Hotmail®. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutor…

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[Kamailio-Users] xmpp:cmd_send_message: invalid content-type 0x100ff
by Joao Gomes Pereira 15 May '09

15 May '09

Hello to all Im trying to use Kamailio with xmpp module in Component Mode. So, I have Jabberd2 installed and running in the same machine, and this is my config: loadmodule "xmpp.so" modparam("xmpp", "domain_separator", "*") modparam("xmpp", "gateway_domain", "jabber.servebeer.com") modparam("xmpp", "xmpp_domain", "jabber.servebeer.com") modparam("xmpp", "xmpp_host", "my_server_IP") modparam("xmpp", "xmpp_port", 5347) if (method == "MESSAGE") { log("*** xmpp-handled MESSAGE message.\n"); if (xmpp_send_message()) { t_reply("200", "Accepted"); } else { t_reply("404", "Not found"); } return; } but I'm receiving this errors: kamailio[8218]: ERROR:xmpp:cmd_send_message: invalid content-type 0x300ff What could be the problem? Thanks regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira(a)startel.pt

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[Kamailio-Users] sr-users mailing list
by Daniel-Constantin Mierla 15 May '09

15 May '09

Hello, the users mailing list for sip router project is available as: sr-users(a)lists.sip-router.org. you can subscribe to it at: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users People starting testing sip router and willing to help with documentation are invited to join. The other mailing lists dedicated to sip router project are: - sr-dev - for development related discussions http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev - sr-users-es - sip router Spanish community mailing list http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-es Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] kamailio multi-domain
by karhu 14 May '09

14 May '09

Hello, First of all i would like to say i am a newbee in Kamailio :). Here is my network configuration : User 1 User2 10.10.10.3 10.10.10.2 192.168.0.2 192.168.0.10 -----------------*KAMAILIO*----------------- User1 and User2 are registered in mysql. When i installed Kamailio, i edited kmctlrc to have SIP_DOMAIN = 10.10.10.2. I am using XLite Siphone for both users. When User 1 is registering to Kamailio there is no problem, but when User2 try to do the same i get 407 Proxy Authentication Required.. What i would like to do is User1 call User2 trough kamailio. Obviously i activated nathelper module and started rtpproxy. Do i need to configure multi-domain module ? Here is my .conf file : ############################################# ####### Global Parameters ######### debug=9 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" #port=5060 /* 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:192.168.0.2 listen=udp:10.10.10.2 ####### Modules Section ######## #set module path mpath="/usr/local/lib64/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- 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) # ----- rr 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) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* 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", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("domain", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ modparam("presence|presence_xml", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:10.10.10.2:5060") # -- nathelper modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7890") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { 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 setflag(2); route(1); } route[1] { if (check_route_param("nat=yes")) { setbflag(6); } if (isflagset(5) || isbflagset(6)) { route(5); } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { 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; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($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 } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { unforce_rtp_proxy(); } if (t_was_cancelled()) { 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(); ##} } ############################################## Thanks -- View this message in context: http://www.nabble.com/kamailio-multi-domain-tp23539467p23539467.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] x-lite 3.0 on linux
by Daniel-Constantin Mierla 14 May '09

14 May '09

Hello, during last weeks while testing the kamailio (openser) modules with sip-router core I got x-lite 3.0 working on linux (ubuntu) with wine 1.1.20. I posted guidelines at: http://appdb.winehq.org/objectManager.php?sClass=version&iId=16413 Voice, IM&Presence seem to work just fine. Video was not tested as I couldn't make any wine application see the video cam, although it is working fine on linux with skype or ekiga. For me it is a great tool to test the presence, hope it will be useful for some of you as well. Apart of x-lite, I got other sip softphone working with wine: - pangolin - works fine, but the presence status changes are not sent to server (no PUBLISH) -- might be a bug in the software, maybe someone running it on windows can confirm it or not - zoiper communicator rc1 free edition for windows - linux version seems to be available now, but requires to register with their site Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] kamailio and/or ser 3.0 with sip router core
by Daniel-Constantin Mierla 14 May '09

14 May '09

Hello, long awaited period of SIP Router integration phase is more and more approaching the end. Apart of 4 modules that are not integrated yet from Kamailio - seas, siptrace, uac_redirect and nat_traversal (work being undertaken) -- the rest of 88 modules are ready for http://sip-router.org core. I have created a page to collect guidelines for the default Kamailio configuration file used with sip-router. I invite all of you to test it, not only this one, but your private configs as well. http://sip-router.org/wiki/migration/kamailio-3.0-config Feel free to add to the wiki page or post questions about it to sr-users(a)lists.sip-router.org. This is a new list created for users of sip-router project, you can subscribe to it at: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users At this stage, sip router can be used with modules from both Kamailio (OpenSER) and SER projects, mixed in same instance at the same time. Therefore you can choose the best ones that fit your needs. For Kamailio (OpenSER) users there is a page that tries to collect new features they got from SER side (still a lot to add there, hope ser developers will contribute what they find missing). Another page collects links for ser and kamailio users to find what the other project brings: http://sip-router.org/wiki/migration/new-features-for-k-users http://sip-router.org/wiki/features/inherited-features It was not only integration work in the past months, sip router has quite a bunch of brand new feature. The wiki page to collect the summary is available at: http://sip-router.org/wiki/features/new-in-devel Jan just set online documentation portal, available for web browsing: http://sip-router.org/docbook/sip-router/branch/master/ Common modules are located in directory 'modules' while kamailio and ser specific modules can be found under module_k and modules_s. Project's wiki portal got also good content and it is under heavy work -- note that everyone can contribute to wiki: http://sip-router.org/wiki Couple of core and tm features are not yet integrated or not exactly as they were in Kamailio. Please report the ones you are missing in your config at: http://sip-router.org/wiki/devel/kamailio-integration#tm_and_core_extensions It is the place that should set the priority rank for integration. I hope that in 1-2 months time frame to get into beta stage and prepare a stable release. Many thanks for those that allocated resources to make things evolve to this stage and many thanks in advance for those that will start testing and help improving the sip router project. Beware that is still very early stage and not recommended for production. Cheers, Daniel PS. Codename 3.0 refers to Kamailio (OpenSER) or SER using sip router core, there is not a stable release yet. -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] dialplan subst issue
by Andreas Granig 14 May '09

14 May '09

Hi all, I was playing around with dialplan module (kam 1.4) for the first time and it seems that the repl_exp doesn't handle back references correctly: mysql root@kamailio> select * from dialplan\G *************************** 1. row *************************** id: 1 dpid: 1 pr: 1 match_op: 1 match_exp: @voicebox\.local$ match_len: 0 subst_exp: ^sip:(.*)$ repl_exp: sip:xx_\1_yy attrs: voicebox 1 row in set (0.00 sec) In my config, I do: dp_translate("1", "$ru/$var(dptest)"); xlog("L_INFO", "DP result='$var(dptest)', attr='$avp(s:dialplan_attr)\n"); and the result is: DP result='sip:xx_user@voicebox.local', attr='voicebox' So the string after the back reference ("_yy" in this case) is not inserted. Any ideas? Andreas

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[Kamailio-Users] presence pres-rules and database watchers table
by Dubravko Caric 13 May '09

13 May '09

Hi all, is there any way to automatically change "reason" column from "rejected" back to "NULL" or "allow" in watchers table after UAC#1 moves the UAC#2 from its blacklist to whitelist? i see that kamailio (1.5.0) correctly reads index.xml file from database, but "rejected" reason is still in the watcher table until I delete it manually and restart UAC. ...and is this even suppose to work like that or should I use different approach? Thanks Dubravko

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[Kamailio-Users] kamailio
by Abou NDIAYE 13 May '09

13 May '09

Hello, In fact I started with kamailio. The calls go smoothly on my router (with NAT) kamailio that the proxy is started or not, the calls pass. I put what in the conf file for the proxy intercepts all calls sip and then forward. Thank you in advance. _________________________________________________________________ Découvrez Windows Live Spaces et créez votre site Web perso en quelques clics ! http://spaces.live.com/signup.aspx

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[Kamailio-Users] Get send_socket if I have no new branch
by catalina oancea 13 May '09

13 May '09

Hello all In my kamailio.cfg I don't use append_branch at all and I listen on multiple addresses. Before running t_relay I need to know on which socket will the request be sent. I don't want to change the socket, but I want to know which socket is chosen. I tried $(branch(send_socket)[0]) and $(branch(send_socket)) but I got error "ERROR:pv:pv_get_branchx: error accessing branch [0]" Does anybody know how I can get this info? Thanks Catalina

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Re: [Kamailio-Users] two UACs behind same NAT
by Dubravko Caric 13 May '09

13 May '09

Hi, ----- Original Message ---- From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: Iñaki Baz Castillo <ibc(a)aliax.net> Cc: users(a)lists.kamailio.org Sent: Monday, May 11, 2009 10:04:49 AM Subject: Re: [Kamailio-Users] two UACs behind same NAT >> >> Bad luck then... :( >> SIP ALG is the worst enemy for SIP. >> > workarounds could be: > - run sip server also on a different port than 5060 (say 5070) - > kamailio is just fine doing it. Point the users behind such ALGs to this > port. Still alg can detect it, but most of them do the detection by port > 5060 > - use TCP if the phone supports it - most of algs do not touch TCP > connections, but ... > - use TLS if the phone supports it - safest - the alg cannot touch it yes, we are planning to use TLS and I hope we won't have that kind of problems then > - recommended - send back the router and ask for money return well, I could do that but the problem is not just with this router. I saw a list of this kind of routers (small home routers) that are not recommended to use with SIP, and I think that TLS is only universal solution. Best regards Dubravko Cheers, Daniel

2 1

[Serusers] SER with RADIUS
by Leon Li 13 May '09

13 May '09

Hi, I am trying to make SER working with FreeRADIUS. However, I got an error using www_challenge, " 0(3184) parse error (119,25-26): unknown command, missing loadmodule?" The 119 line is www_challenge("", "0"); I did load auth module. Any idea? Here is my config: # # $Id: ser-basic.cfg,v 1.1.2.2 2008/06/12 13:15:06 alfredh Exp $ # # This a very basic config file w aliases and anamed route but # w/o authentication, accounting, database, multi-domain support etc. # Please refer to ser.cfg for a more complete example # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) #memdbg=10 # memory debug message level #memlog=10 # memory statistics log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=202.158.197.134 port=5060 children=4 #user=ser #group=ser #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) # # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/ctl.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ctl params # by default ctl listens on unixs:/tmp/ser_ctl if no other address is # specified in modparams; this is also the default for sercmd modparam("ctl", "binrpc", "unixs:/tmp/ser_ctl") # listen on the "standard" fifo for backward compatibility modparam("ctl", "fifo", "fifo:/tmp/ser_fifo") # listen on tcp, localhost #modparam("ctl", "binrpc", "tcp:localhost:2046") # -- auth_radius params -- modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_reply("483","Too Many Hops"); break; } if (msg:len >= max_len ) { sl_reply("513", "Message too big"); break; } # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(FORWARD); break; } if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(FORWARD); break; } # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # authentication if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save_contacts("location"); break; } # native SIP destinations are handled using our USRLOC DB if (!lookup_contacts("location")) { sl_reply("404", "Not Found"); break; } append_hf("P-hint: usrloc applied\r\n"); } route(FORWARD); } route[FORWARD] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); } } Thanks, Leon

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[Kamailio-Users] BAN a scanner
by BERGANZ François 12 May '09

12 May '09

Hello, Is it possible to ban an IP which scan my servers ? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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[Kamailio-Users] WARNING:core:fm_free: free(0) called in kamailio
by rachid 12 May '09

12 May '09

Hello, when i start a sip client (x-lite) i have in kamailio debug message this error: WARNING:core:fm_free: free(0) called someone can tell me where is the problem?? thank you ------------------------------------------------------------ BAHA Rachid Elève ingénieur en 3ème année à l'INPT, Rabat GSM:(+33) 0619787609

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[Kamailio-Users] RTP timeout and nathelper
by mayamatakeshi 11 May '09

11 May '09

Hello, RTPProxy 1.2.0 can be instructed to notify and external app when RTP timeout occurs. It seems this notification must be enabled on a per session basis. So, does module nathelper have the means to request this notification? regards, takeshi

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[Kamailio-Users] two UACs behind same NAT
by dubravko caric 11 May '09

11 May '09

Hi all, I have a problem when using two UACs behind same NAT on Linksys WRT54GL router. Registrations, Invites, ... are handled correctly, but after 30sec in the call UACs hangs up the call (RTP goes directly between clients). This happens because UAC#1 (which answerd the call with OK) didn't recive ACK from UAC#2. This isn't happening when I have this same clinets behind NAT on PIX525 and everything is working fine. Everithing is also fine when UAC#1 calls someone on PSTN (RTP is handled by RTPProxy) or someone behind other NAT. Strange thing is that UAC#1 sends Ringing response and OK message with this information in Contact header: Contact: <sip:bob@public_ip_address:31479;rinstance=e0f8548cf68b5b6e> but proxy forwards this Ringing response and OK message to UAC#2 with this information in Contact header: Contact: <sip:bob@public_ip_address:0;rinstance=b18efeaa8a0e3f0f> so port is "0" and with ngrep on proxy I saw that this port is replaced with port "5060" and ACK is sent to the UAC#1 with this parameters public_ip_address:5060 which isn't correct because UAC#1 is using 31478-31488 port range and expects traffic on these ports. Some of debug messages that i see on proxy are: 1. sl_filter_ACK: to late to be a local ACK!, 2. e2e proxy ACK found building branch for end2end ACK totag for e2e ACK found: 1 I'm using Kamailio 1.5.0 with standard kamailio.cfg and newest XLite (with STUN and ICE options turn on). I would be greatfull if someone can give me any guidelines what to look next in order to solve this issue. Best regards Dubravko

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[Kamailio-Users] creating openser db on an external mysql server
by Alex R.S.M 11 May '09

11 May '09

Hi, Creating openser database on the localhost works fine using "kamdbctl create". However, I need to create openser database on an external mysql server. I tried the following command but it's not working: kamdbctl create openser@<external_mysql_server_ip> What's the correct command? Thanks, A R _________________________________________________________________ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_BR_life_in_synch_052009

4 11

[Kamailio-Users] call routing problem
by bhrugu mehta 08 May '09

08 May '09

Hi all, I am new to openser. I have register two sip user in openser (as register server) and call handling in asterisk. when 1001 user do a call to 1002 nothing happen. call rejected. If posible give a sip.conf and extension.conf snap of this scenario. any suggestion? Bhrugu Mehta

3 2

[Kamailio-Users] H.323 / SIP signalling gateway for OpenSER
by Bilbo 07 May '09

07 May '09

Hi, Has anyone tried connecting the 'Cisco Unified Border Element' to OpenSER as a H.323 /SIP signalling gateway to allow H.323 video calls to be placed over OpenSER? Can you see any problems trying to connect these two systems together? Thanks -- View this message in context: http://www.nabble.com/H.323---SIP-signalling-gateway-for-OpenSER-tp23401501… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 3

[Kamailio-Users] sip router presentation at amoocon 2009
by Daniel-Constantin Mierla 06 May '09

06 May '09

Hello, those interested in seeing the slides I presented at Amoocon 2009, can get the link to them at: http://sip-router.org/2009/05/05/amoocon-2009-presentation/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

2 1

[Kamailio-Users] H.323 signalling gateway for OpenSER
by Bilbo 06 May '09

06 May '09

Hi, Has anyone tried connecting the 'Cisco Unified Border Element' to OpenSER as a H.323 /SIP signalling gateway to allow H.323 video calls to be placed over OpenSER? Can you see any problems trying to connect these two systems together? Thanks -- View this message in context: http://www.nabble.com/H.323-signalling-gateway-for-OpenSER-tp23401428p23401… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[Kamailio-Users] web interface to add new users
by Joao Gomes Pereira 04 May '09

04 May '09

Hello to all Does someone knows of a Web application to allow users to add an account in Kamailio? SIREMIS is only for administrator use. Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira(a)startel.pt

2 1

[Kamailio-Users] Arbitrary contact added in 302 generated responses
by Francisco Javier Lizaran Vilches 04 May '09

04 May '09

Hi all: Have running this script on Openser 1.2.3-notls version managing forwards this way: route[3] { ... t_on_failure("1"); if avp_db_load("$ru", "*") { if (is_avp_set("$avp(s:fwdbusy)/s")) { setflag(23); }; ... failure_route[1] { ... if ((isflagset(23)) && (t_check_status("486"))) { if (avp_pushto("$ru", "$avp(s:fwdbusy)")) { t_reply("302","Moved Temporarily"); return; }; }; ... User A calls user B and B has fwdbusy parameter set in user preferences; if B is busy, proxy sends 302 message back to A with contact set to $avp(s:fwdbusy) value. This setup usually works fine. However, sometimes the proxy shows a strange behaviour. It starts to append an arbitrary contact to the contact header in every forwarding it does. The contact appended has nothing to do with A or B or the uri set in the avp. The same uri is appended in all call forwardings performed in the system. If the proxy is restarted, the problem disappears. Example: SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.100.2.254:5060;branch=z9hG4bK51461DD5 From: <sip:30132@10.100.2.254 <sip%3A30132(a)10.100.2.254>>;tag=92DAB388-EE7 To: <sip:20050@domain.com <sip%3A20050(a)domain.com> >;tag=880a5593aeb097bc75600b31d6e17107-78ac Call-ID: C0C84A1C-33CE11DE-BEBEEAA9-C0D323DF(a)192.168.2.40 CSeq: 101 INVITE Contact: sip:30050@domain.com <sip%3A30050(a)domain.com>, <sip:030410@10.172.0.254:5060;transport=udp>;q=0 Server: OpenSER (1.2.3-notls (i386/linux)) Content-Length: 0 <sip:030410@10.172.0.254:5060;transport=udp>;q=0 is added in all forwardings done the system. Unfortunately I have no debug info cause it happens in production environment. I couldn't reproduce the problem in test environment. Have you got any idea on what could make it happen? Thanks a lot: Fran Lizaran

2 3

[Kamailio-Users] Http_query
by sasirekha jaganathan 04 May '09

04 May '09

Hi, Is it possible to get the XML response(i.e content other than text) in http_query function? Thanks Now surf faster and smarter ! Check out the new Firefox 3 - Yahoo! Edition http://downloads.yahoo.com/in/firefox/?fr=om_email_firefox

3 2

[Kamailio-Users] Callback functionality
by sasirekha jaganathan 04 May '09

04 May '09

Hi, Is it possible to implement a custom callback in kamailio modules? Thanks Now surf faster and smarter ! Check out the new Firefox 3 - Yahoo! Edition http://downloads.yahoo.com/in/firefox/?fr=om_email_firefox

3 2

[Serusers] ser problem
by Akhmad Dhany 04 May '09

04 May '09

Hi,, I send my problem in attach files. Thanks very much. Regards, Dhany

2 1

[Kamailio-Users] communication between SIP and MSN users without using purplemap table
by rachid 04 May '09

04 May '09

Hello , I use kamailio to have communication between sip client(bob) and MSN client (alice) and i added msn address in bob budy liste(in x-lite) so i want to have bidirectional communication without using purplemap !! someone can help me or tel me witch "file.c" to modifie thank you -- ------------------------------------------------------------ BAHA Rachid Elève ingénieur en 3ème année à l'INPT, Rabat GSM:(+33) 0619787609

3 2

[Kamailio-Users] changelog page not found
by ram 03 May '09

03 May '09

http://www.kamailio.org/mos/view/News/NewsItem/Kamailio-v1.5.1-Released//pu… The page you are trying to access does not exist. Please select a page from the main menu. This is from page404.php Ram

2 1

[Kamailio-Users] Multiple SIP Proxy Environment / Socket Information
by Brandon Armstead 02 May '09

02 May '09

Hello guys, Is there a method upon using lookup("location") to also pull out the "socket" information for the original location the UAC registered to, for scenarios of this example: P1 & P2 share same usrloc database. UA1 registers to P1 UA2 registers to P2 UA1 calls UA2 UA1 invites -> P1 -> INVITES -> UA2 (bypassing P2 -- where the actual nat binding is). Now upon P1 looking up usrloc for UA2, I would like to recognize that P1 is not the Proxy to deliver the call, and forward the request to P2 to send to UA2. So currently I have: UA1 INVITE -> P1 INVITE -> UA2 I wish to have: UA1 INVITE -> P1 INVITE -> P2 INVITE -> UA2 Is there an easy method to do this? I have been looking at the new nat traversal module it looks like it is doable with this (any further input as far as that?). Also is it possible with the classic Nat Helper module? Any input is appreciated, thanks!

4 23

[Kamailio-Users] CDR / EDR Accounting With Kamailio
by Brandon Armstead 02 May '09

02 May '09

Hello, First off going to thank you guys ahead of time, I'm currently running into the following issue: I'm trying to account for internal calls, i.e. example scenarios such as: UA1 and UA2 are both on P1 UA1 calls UA2 so, effectively: UA1 -> P1 -> UA2 Now the problem I'm running into is with the direction of the calls, I am having trouble accounting for both sides of the call, i.e. I store an avp such as: $avp(s:account-id) Obviously the AVP is going to be set to one account_id, and my problem is that the Callee is what ends up "being accounted" for not, the "Caller". Another scenario I would like some insight on and having problems with is: UA1 is on P1, UA2 is on P2 UA1 calls P1 -> P2 -> UA2 same type scenario, only one end of the call (Callee is being accounted for) -- how can I accurately account using kamailio acc module for both sides of the call for both users? Any thought / input on this is very much appreciated, thanks!

3 2

[Serusers] SER can't connect to mysql
by Paul Osier 01 May '09

01 May '09

Hi. When starting SER, I get the following line in the logs: new_connection(): Access denied for user 'ser'@'localhost' (using password: YES) I have the following line in ser.cfg: modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") along with the required modules loaded. I can log into mysql with mysql -h localhost -u ser -pheslo ser , so I think all the mysql stuff is set right and the parameter has the correct info. Any ideas why it's not connecting with the ser.cfg file setup? Thanks.

1 0

[Kamailio-Users] LCR Private Memory
by Stagg Shelton 01 May '09

01 May '09

I am working to get LCR implemented on my migration system, and when I try to start kamailio i find the below information in the log file, and kamailio fails to start. My lcr table has 30667 rows in it. I have read about changing the config.h file to increase the private memory, but my question is how much should I allocate. I forsee going up to more than 200,000 routes within the next 12 month period. I am currently working towards migrating up from openser 1.3.3 to kamailio-1.4.4-notls. I am currently not using LCR in my production openser 1.3.3 system. This is new functionality that we are implementing as part of the upgrade. Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:db_mysql:db_mysql_convert_row: no private memory left Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:db_mysql:db_mysql_convert_rows: error while converting row #2808 Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:db_mysql:db_mysql_convert_result: error while converting rows Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:db_mysql:db_mysql_store_result: error while converting result Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:core:db_do_query: error while storing result Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:lcr:reload_gws: Failed to query lcr data Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: CRITICAL:lcr:mod_init: Failed to reload gateways and routes Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:core:init_mod: failed to initialize module lcr Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:core:main: error while initializing modules Thanks Stagg

5 11

[Kamailio-Users] SIREMIS installation
by Nikolaos Markolefas 30 Apr '09

30 Apr '09

Hi all, just installed SIREMIS on a Ubuntu server, on top of kamailio 1.5. It seems to work alright, but for the charts sections. What I see is : """ Open Flash Chart JSON Parse Error [Syntax Error] Error at character 2, line 2: 0: 1: <div style='font-size: 12px; color: blue; font-family:Arial; font-weight:bold;'>[2009-04-09 12:40:1 """ Apache produces the following log (error.log): *[Thu Apr 09 15:39:58 2009] [notice] Apache/2.2.9 (Ubuntu) PHP/5.2.6-2ubuntu4 with Suhosin-Patch configured -- resuming normal operations [Thu Apr 09 15:40:07 2009] [error] [client 192.168.110.25] File does not exist: /var/www/siremis-0.9.2/siremis-web/js/sound.js, referer: http://v-srv-obama/siremis/bin/controller.php?view=main.CrtMainView [Thu Apr 09 15:40:09 2009] [error] [client 192.168.110.25] File does not exist: /var/www/siremis-0.9.2/siremis-web/js/sound.js, referer: http://v-srv-obama/siremis/bin/charts.php?cg=system * The file indeed does not exist. I'd also like to note that while i downloaded the 0.9.2 version, SIREMIS homepage shows v0.9.1 . Any ideas?

3 9

[Kamailio-Users] from replace
by Gentrice's kaiser 30 Apr '09

30 Apr '09

Dear Sir: I wrote a failure route for serial call mechanism, it help for no answer call ... but all these serial calls need to have a new CallerID, so I use replace_from for this purpose. modparam("uac","from_restore_mode","auto") failure_route[3] { log(1,"******* I'm failure_route[3] *******\n"); if (avp_pushto("$ru", "$avp(forward)")) { append_branch(); avp_delete("$avp(forward)"); uac_replace_from("sip:1234@$fd"); # change from header t_on_failure("3"); # try next , recursive t_relay(); } } If only one forward number in AVP array , this code working fine, the issue will happen if multiple forward number trying, ( forward 2nd,3rd ...number) the UAC module will restore from header in 200 OK of its following message when a UA answered, we will have a duplicated "from header" in 200 OK (sending out opensips). It means we will have 2 "from" headers (in 200 OK) to caller, if 2nd forwarded callee answer. Caller ---> Invite ---> opensips ----> UA 0, no answer ----> UA 1, no answer ---->UA 2, answer <----200 OK <---200OK<--- this 200ok will have 2 "from" header. Most of UA just ignore redundant from header, but some can not. I think it should be UAC modules issue, anyone can help me? best regards KK

3 7

[Serusers] SIP Diversion for SER 0.9.6
by inge 30 Apr '09

30 Apr '09

Hi all, I look for a way to use SIP DIVERSION. In case of a call transfer, I have to conserve the original called number and send it to SS7 gateway. But for a good interoperability between SIP and SS7 in this kind of application I just see the way of SIP DIVERSION support. Can someone help me ? Regards, A.L

3 5

[Kamailio-Users] Kamailio v1.5.1 Released
by Daniel-Constantin Mierla 30 Apr '09

30 Apr '09

Kamailio v1.5.1 is out - a minor release of the branch 1.5, including fixes since v1.5.0 - configuration file and database compatibility is preserved. Kamailio 1.5.1 is based on the latest version of branch 1.5, including many fixes in code and documentation, therefore those running 1.5.0 are advised to upgrade. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/1.5.1/src/ <http://www.kamailio.org/pub/kamailio/1.5.1/src> Detailed changelog: http://www.kamailio.org/pub/kamailio/1.5.1/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 kamailio Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.5.1/ Project site at SourceForge.net (still using old name): http://sourceforge.net/projects/openser/ Binaries and packages will be uploaded at: http://www.kamailio.org/pub/kamailio/1.5.1/ Modules' documentation: http://www.kamailio.org/docs/modules/1.5.x/ <http://www.kamailio.org/docs/modules/1.5.x> What is new in 1.5.x release series is summarized in the announcement of v1.5.0: http://www.kamailio.org/mos/view/Kamailio-OpenSER-v1.5.0-Release-Notes /Note: Kamailio is the new name of OpenSER project. First version under Kamailio name was 1.4.0. Older versions will continue to use OpenSER name. / -- Daniel-Constantin Mierla http://www.asipto.com/

3 2

[Kamailio-Users] insert_hf not persistant
by David Aldworth 30 Apr '09

30 Apr '09

I think I hijacked a thread when I posted this previously. Sorry 'bout that. The issue we are having is where Kamailio relays an INVITE with the correct To header field domain, however when it relays the subsequent CANCEL it seems like it does not correctly change the To header field domain. Here is what we use to try and put in the correct To header in the CANCEL: remove_hf("To"); if ($tt == null) insert_hf("To: <sip:$rU@1.2.3.4>\r\n", "From"); else insert_hf("To: <sip:$rU@1.2.3.4>;tag=$tt\r\n", "From"); xlog("DBG1: Trying to fix CANCEL: td=$td ci=$ci rm=$rm ru=$ru ou=$ou\n"); Changes to the left side of the at-symbol persist, however changes to the domain are reverted and changed back to what was received from adjacent proxy. Any ideas why it reverts?

2 1

[Kamailio-Users] openser CDR Diversion calls
by Ilya Pichugin 29 Apr '09

29 Apr '09

Hi all! I'm new in openSER and I'm trying to get correct CDRs when the call is forwarded on busy, noanswer and uncond., but i have some troubles :( I use accounting flags by acc module and radius - so, I can get cdrs from acc table or raddact table When I do any type of FWD call by AVP module, for example, A (pstn) -> B (ser user, fwd to GSM) -> C (GSM) i get cdrs for A -> C, but I want get cdrs for A - B and B - C Example: mysql> SELECT * FROM openser.usr_preferences u; +----+---------+----------+------------+------------+------+-------------+---------------------+ | id | uuid | username | domain | attribute | type | value | last_modified | +----+---------+----------+------------+------------+------+-------------+---------------------+ | 12 | 3097999 | 3097999 | xx.xxx.com | fwdall | 0 | 89112820419 | 0000-00-00 00:01:00 | Make call 8123364000 (pstn) - 3097999 (SERuser fwdall to GSM) - 89112820419 (GSM) and CDRs from raddact table: mysql> SELECT callingstationid,canonicalURI, acctstarttime,acctstoptime, acctsessiontime FROM radacct r where acctstarttime>'2009-04-23' and callingstationid like '8123364000'; +------------------+--------------+---------------------+---------------------+-----------------+ | callingstationid | canonicalURI | acctstarttime | acctstoptime | acctsessiontime | +------------------+--------------+---------------------+---------------------+-----------------+ | 8123364000 | 89112820419 | 2009-04-24 10:29:10 | 2009-04-24 10:29:12 | 2 | +------------------+--------------+---------------------+---------------------+-----------------+ from acc table: mysql> SELECT id,method,sip_code,sip_reason,time,src_user,dst_user FROM openser.acc a; +------+--------+----------+------------+---------------------+------------+-------------+ | id | method | sip_code | sip_reason | time | src_user | dst_user | +------+--------+----------+------------+---------------------+------------+-------------+ | 2420 | INVITE | 200 | OK | 2009-04-24 10:29:10 | 8123364000 | 89112820419 | | 2421 | ACK | 200 | OK | 2009-04-24 10:29:10 | 8123364000 | 89112820419 | | 2422 | BYE | 200 | OK | 2009-04-24 10:29:12 | 3097999 | 8123364000 | +------+--------+----------+------------+---------------------+------------+-------------+ So, how do you bill diversion calls? Help, pls. -- Best Regards, Ilya Pichugin i.pichugin(a)ptl.ru

3 2

[Kamailio-Users] \0 in SIP-Messsages - uac_replac_from - Lumps - etc...
by Carsten Bock 29 Apr '09

29 Apr '09

Hi, short question: I've noticed, when the UAC-Module (from 1.3) replaces the From-Headerin subsequential requests, the From-Header is filled up with "zeros". For most clients this is absolutely fine (tested with Snom's, AVM's, Cisco's and many more). I've noticed, that some Zyxel devices however do not like this and ignore the request in this case... Is this somehow defined in the RFC? Can somebody give me hint on a chapter or something? Or does it break the RFC? Thanks in advance, Carsten

2 8

[Kamailio-Users] insert_hf not persistant
by David Aldworth 29 Apr '09

29 Apr '09

The issue we are having is where Kamailio relays an INVITE with the correct To header field domain, however when it relays the subsequent CANCEL it seems like it does not correctly change the To header field domain. Here is what we use to try and put in the correct To header in the CANCEL: remove_hf("To"); if ($tt == null) insert_hf("To: <sip:$rU@1.2.3.4>\r\n", "From"); else insert_hf("To: <sip:$rU@1.2.3.4>;tag=$tt\r\n", "From"); xlog("DBG1: Trying to fix CANCEL: td=$td ci=$ci rm= $rm ru=$ru ou=$ou\n"); Changes to the left side of the at-symbol persist, however changes to the domain are reverted and changed back to what was received from adjacent proxy. Any ideas why it reverts?

1 0

[Kamailio-Users] Amoocon 2009
by Daniel-Constantin Mierla 29 Apr '09

29 Apr '09

Amoocon (former AsteriskTag) 2009 will take place in Rostock, Germany, May 04-05, 2009. Kamailio (OpenSER) and SIP-Router.org projects will be presented at the event by talks and case studies of: * Daniel-Constatin Mierla: http://www.amoocon.de/users/14 * Olle E. Johansson: http://www.amoocon.de/users/61 If you are around Berlin the days before the event or in Rostock during the event and want to meet, drop me an email. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

[Serusers] SER proxy compatibility with Red Hat Enterprise Linux 5
by inge 28 Apr '09

28 Apr '09

Hi, I want to buy a RHEL5 to install a SER Proxy 0.9.6. I want to know if this version is compatible whith the RHEL 5. Thanks.

2 1

[Kamailio-Users] sharing vedio between sip (x-lite) and MSN
by rachid 28 Apr '09

28 Apr '09

Hello all, I instaled kamailio and i want to have sharing video between a sip user (X-lite) and MSN user someone can help me please? -- ------------------------------------------------------------ BAHA Rachid Elève ingénieur en 3ème année à l'INPT, Rabat GSM:(+33) 0619787609

4 3

Re: [Kamailio-Users] Load Balancing Asterisk using Kamailio
by Daniel-Constantin Mierla 27 Apr '09

27 Apr '09

Hello, not cc-ing to mailing list could lead to no answer... private messages from mailing list threads are ignored. Other benefits with mailing lists is that others can answer quicker or better. Thanks, Daniel On 04/27/2009 04:52 PM, Kurt Weasel wrote: > Hi Daniel, thank you for the reply. > I have some more questions to ask. > > 1. Since the dial plan is configured exactly on Asterisk (at > etc/asterisk/sip.conf and /etc/asterisk/extension.conf), do we need to > enable MySQL for Kamailio installation? Or the default installation > configuration will do (without additional dialplan or MySQL setting)? > My idea is only using Kamailio as pure load balancer so we hope to do > this as simple as possible without degrading the function. > > 2. I assume using Kamailio dispatcher module will forward some of the > users to asterik server 1 and some of them to asterisk server 2. Then > if user A is registered with Asterisk server 1 and user B is > registered with Asterisk server 2, can user A call user B? Without > Kamailio, the answer is no since they are in different Asterisk > server. With Kamailio, I am expecting both user A and B can call each > other although they are registered with different servers. Is my > configuration enough to achieve this function? Or do we need some > additional setup for this? > > Thanks again. > > Best Regards, > Kurt > > On Mon, Apr 27, 2009 at 8:02 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > > On 04/27/2009 07:10 AM, Kurt Weasel wrote: > > Hi, > > I searched around the web to load balance asterisk servers and > found Kamailio for possible solution. Let's say I have two > identical asterisk servers with same dialplan and > configuration and I want both servers look like they have same > IP address from clients. Based on my understanding, we need 1 > Kamailio as load balancer and 2 Asterisk servers as the real > servers. Let's say the setup is : > > Kamailio load balancer 192.168.2.1 > Asterisk Server #1 192.168.2.2 > Asterisk Server #2 192.168.2.3 > > My question is, X-Lite softphone Configuration should be set > to domain 192.168.2.1, right? > > I also want to know the step by step configuration to set > kamailio as load balancer. I have not used Kamailio before. > However after searching the documentation, the step (based on > my understanding) is somewhat like this : > > 1. Install Kamailio. I will use the step by step here : > http://kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn > > 2. Then, using dipatcher module, I will configure it using > this guide here : > http://kamailio.org/dokuwiki/doku.php/asterisk:load-balancing-and-ha > Then modify the dispatcher.list file to match the IP address > of my asterisk servers : > > /1 sip:192.168.2.2:5060 <http://192.168.2.2:5060> > <http://192.168.2.2:5060> > 1 sip:192.168.2.3:5060 <http://192.168.2.3:5060> > <http://192.168.2.3:5060>/ > > > Am I missing some steps? > Do I also need to configure dialplan or any other file at > Kamailio load balancer? Or those two steps basically done it > all for simple load balancing configuration? Thanks for your > responses. > > these steps are ok if you do not deal with NAT. If yes, then > things get a bit more complex, you need PATH support on load > balancer, but AFAIK, Asterisk does not support it -- maybe I am wrong. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > > -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

[Kamailio-Users] LCR gateway failthrough..
by Nick Warr 27 Apr '09

27 Apr '09

I'm using the LCR module to send traffic based on just the prefix of the number at the moment. I have two gateway groups, one with a local provider to connect locally, one with a larger one to do international calls (they can also connect locally but with a higher price).. When I call next_gw() will LCR use all possible gateway groups, or just the gateway(s) in the first matching group? The second question is tangential to the first, if it does send the traffic to other gateway group after failing through the first, does it regenerate the RURI, because the local provider wants a numerical prefix on the calls I send, and the other one no (basically do I need to account for this in my config?)

4 6

[Serusers] SER with RADIUS
by Leon Li 27 Apr '09

27 Apr '09

Hi, Does anyone still have the instruction of SER RADIUS Howto? The link http://iptel.org/ser/ser_radius.html Found in an old thread doesn't work now. (see below for the old thread) [Serusers] ser RADIUS Howto Jan Janak jan at iptel.org <mailto:serusers%40lists.iptel.org?Subject=%5BSerusers%5D%20ser%20RADIUS %20Howto&In-Reply-To=> Fri Sep 12 02:11:33 UTC 2003 * Previous message: [Serusers] serweb main page display <http://lists.iptel.org/pipermail/serusers/2003-September/002595.html> * Next message: [Serusers] Implementing Voicemail <http://lists.iptel.org/pipermail/serusers/2003-September/002596.html> * Messages sorted by: [ date ] <http://lists.iptel.org/pipermail/serusers/2003-September/date.html#2591 > [ thread ] <http://lists.iptel.org/pipermail/serusers/2003-September/thread.html#25 91> [ subject ] <http://lists.iptel.org/pipermail/serusers/2003-September/subject.html#2 591> [ author ] <http://lists.iptel.org/pipermail/serusers/2003-September/author.html#25 91> ________________________________ Hello, I finished SER RADIUS Howto. The document thoroughly describes installation and configuration of RADIUS server and client libraries for authentication and accounting with ser. RADIUS related questions appear quite often on the mailing lists so I hope the document helps. I would appreciate if anyone using another RADIUS server than FreeRADIUS could describe the setup and send it to me (plaintext is OK, I will convert it to docbook). You can get the document from here: http://iptel.org/ser/ser_radius.html Any feedback is appreciated. Jan. Thanks Leon

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[Kamailio-Users] switch to TCP when UDP message is bigger than MTU
by Sandro Bordacchini 26 Apr '09

26 Apr '09

Hi all. I am doing some test about tcp/udp transport protocols in SIP and IP fragmentation. I would like to test this behaviour, as stated in par. 18.1.1 of rfc3261: If a request is within 200 bytes of the path MTU, or if it is larger than 1300 bytes and the path MTU is unknown, the request MUST be sent using an RFC 2914 [43] congestion controlled transport protocol, such as TCP. If this causes a change in the transport protocol from the one indicated in the top Via, the value in the top Via MUST be changed. This prevents fragmentation of messages over UDP and provides congestion control for larger messages. However, implementations MUST be able to handle messages up to the maximum datagram packet size. To do this, I am running a OpenSer 1.3.2 with a "default" configuration (and very simple: no accounting, no auth, no DB) and I have two SIP phones registered on that server (for example, ext. 100 and ext. 101). When 100 calls 101, the INVITE reaches the OpenSer that appends some "dummy" headers (I have added some append_hf in the route), just to let the message be bigger than the MTU. The INVITE is routed towards 101 with UDP protocol and I have IP fragmentation. Do you have some hint to get the OpenSer work as described in the RFC (i.e. switch automatically to TCP)? Or is this only possible with UACs and not with proxies? TIA -- Ing. Sandro Bordacchini Dexgate Sr. Tech. Engineer P.zza Fermi 1 - Fornacette IT-56012 Calcinaia (Pisa) Tel. +390587424145 ext 580

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[Serusers] Error in CANCEL generation while forking - ser 2.0
by Tomasz Zieleniewski 24 Apr '09

24 Apr '09

Hi, I found an error in the SER proxy behavior. I use SER v_2_0_0 from CVS. When SER forks requests because there is a number of registered users for particular AOR, and then receives first final response, it in the standard way cancels all other transaction. Problems is that it is not compliant with RFC3261 because it doesn't add Max-Forwards header to the CANCEL requests. RFC3261 says that Max-Forwards must be present in any SIP method. CANCEL sip:16@212.180.179.42:9360 SIP/2.0 Via: SIP/2.0/UDP 212.180.179.42:7060;branch=z9hG4bKb721.ceeb7aa5.1 From: <sip:pgr@sip.rd.touk.pl <sip%3Apgr(a)sip.rd.touk.pl>>;tag=374595 Call-ID: 0da793c0ff2e7f2fc81b23e4166bbab3(a)212.180.179.42 To: <sip:tzl@sip.rd.touk.pl <sip%3Atzl(a)sip.rd.touk.pl>> CSeq: 479638 CANCEL Route: <sip:sip.rd.touk.pl:7060;lr> User-Agent: Sip EXpress router(2.0.0 (x86_64/linux)) Content-Length: 0 Kind regards, Tomasz Zieleniewski

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[Kamailio-Users] textops usage
by Juan Perez 24 Apr '09

24 Apr '09

Hello I am receiving the SDP from a provider with this format: g729/8000/1, which I interpret as codec g729, sampling rate 8000 and the number of channels is 1. The calling party [linksys pa2p] apparently does not like the "1" and terminates the call. I want to strip it using textops module and pass back the SDP w/o the "1" and do this in the On_Reply and the Route sections: if (search_body("g729/8000/1")) { xlog("L_INFO","mylog: On Reply 1 section. Found the string.\n"); replace_body("g729/8000/1","g729/8000"); } but it can not even find it. Is leaving me with several questions: 1- is the "re" that I am using right? 2- Do I have to escape the "/"? 3- does the replace and search function go "deep" enough in the SDP? cheers jp

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[Kamailio-Users] Problem with LCR
by Antonio Reale 24 Apr '09

24 Apr '09

Hi all, I'm using kamailio 1.4.4. In kamailio.cfg I have call forward on busy and call forward no answer that works good when I redirect the call to a local user. If I redirect the call to a PSTN number using lcr module I receive the error: ERROR:lcr:next_gw: No ruri_user AVP If I call directly the PSTN number all is OK and I see in the debug: DBG:lcr:next_gw: Added ruri_user_avp <0255XXXXXX> Where I need to set the ruri_user_avp? I just have modparam("lcr", "ruri_user_avp", "$avp(i:500)") in the script. With a very similar configuration and openser 1.2.3 the problem doesn't occur. Thanks. Regards. Antonio.

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[Kamailio-Users] bidirectionnal chating and presence between sip and MSN
by rachid 23 Apr '09

23 Apr '09

Hello all , I want to have a bidirectionnal chating and presence between sip and MSN ,but i dont know how to configure kamailio thank you for your help -- ------------------------------------------------------------ BAHA Rachid Elève ingénieur en 3ème année à l'INPT, Rabat GSM:(+33) 0619787609

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Re: [Serusers] SER Support for SIP Diversion header
by Stefan Sayer 23 Apr '09

23 Apr '09

Hello, o Randy Jackson [04/22/09 19:49]: > Stefan, > > > > For a SER/SEMS configuration using the voicemail app, 4605 calls 4610. > 4610 doesn’t answer and call is forwarded to 9200. If the forwarded > call is routed to SER/SEMS (i.e. see example below) when the SER > (ser-2.0.0-rc1) receives the below INVITE which contains the Diversion > header in the message header portion of the request, what module and > methods do you use to retrieve the parameters from the Diversion header, > like the URI? 9200 does not have a vm account. The vm app will not > answer the call for account 4610 based on the below INVITE. Thanks in > advance. The question is: where do you get the email address from? If it's in DB I think you could get the user from the diversion header into avp with some clever textops and then do the DB query with it to get email address and user into P-App-Param header. If you are only using mode=box (only voicebox, no mail) though, it may be simpler to get the user in SEMS like with this patch to voicemail app: Index: AnswerMachine.cpp =================================================================== --- AnswerMachine.cpp (revision 1354) +++ AnswerMachine.cpp (working copy) @@ -531,10 +531,16 @@ throw AmSession::Exception(500, "voicemail: no email address"); } - user = get_header_keyvalue(iptel_app_param,"usr", "User"); + // user = get_header_keyvalue(iptel_app_param,"usr", "User"); + AmUriParser div_parser; + if (!div_parser.parse_contact(getHeader(req.hdrs, "Diversion"), 0, end)) { + ERROR("could not parse Diversion header"); + throw AmSession::Exception(500, "voicemail: no user found"); + } + user = div_parser.uri_user; if (!user.length()) user = req.user; - + sender = get_header_keyvalue(iptel_app_param, "snd", "Sender"); if (!sender.length()) sender = req.from; btw, isn't Diversion supersedet by History-Info? BR Stefan > > > > > > INVITE sip:9200@192.168.1.1;user=phone > <mailto:9200@192.168.1.1;user=phone> SIP/2.0 > > Via: SIP/2.0/TCP 192.168.1.2;branch=z9hG4bKac816850588;alias > > Max-Forwards: 70 > > From: <sip:4605@192.168.1.2>;tag=1c816838653 > > To: <sip:9200@_192.168.1.1_;user=phone> > > Call-ID: 816837947271200020365(a)192.168.1.2 > > CSeq: 1 INVITE > > Contact: <sip:4605@192.168.1.2;transport=tcp> > > Supported: em,100rel,timer,replaces,path,resource-priority > > Allow: > REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE > > *Diversion: <sip:4610@192.168.1.1>;reason=no-answer;screen=no;privacy=off* > > User-Agent: > > Content-Type: application/sdp > > Content-Length: 270 > > > -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.com www.iptego.com IPTEGO GmbH Wittenbergplatz 1 10789 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

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[Kamailio-Users] SR-Users-ES maillist
by Iñaki Baz Castillo 23 Apr '09

23 Apr '09

Hi, for the spanish speakers community there is a maillist about Sip-Router: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-es This maillist has been merged with the previous Kamailio Users-es maillist, so the maillist history remains. Best regards. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] New listen IP without restart
by catalina oancea 23 Apr '09

23 Apr '09

Hello everybody, Does anybody know if there is a way to add a new listen line in kamailio.cfg and make kamailio start listening on that IP without restarting the server? Thanks a lot Catalina

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[Serusers] is serusers@lists.iptel.org mailing list still in use?
by Leon Li 23 Apr '09

23 Apr '09

Hi, I am just wondering whether serusers(a)lists.iptel.org is still the one to request support? I didn't received any emails in last several days. Cheers Leon

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Re: [Kamailio-Users] Multiple SIP Proxy Environment / Socket Information
by Iñaki Baz Castillo 21 Apr '09

21 Apr '09

El Martes, 21 de Abril de 2009, Brandon Armstead escribió: > Inaki, Please, mantain the thread in the maillist. > Thanks for the input -- that does look similar to what I'm looking to > do, however not 100%. From looking at the RFC it looks as if there is > always a certain path being traversed, however in my case, its more of a > "this is not the server that holds this user location socket binding, lets > forward to the server that does". > > So in essence: > > UA1 can register to P1, P2, P3, and so on..... > > However it does not register to P3 via the means of UA1 REGISTER -> P1 -> > P2 -> P3 (REGISTRATION). > > It simply registers directly to P1, or P2, or P3, etc.... > > In which case upon user lookup: lookup("location"), I want to be able to > say "Hey I do not hold the socket for this registration, forward to server > that does to do the actual lookup(location) and t_relay() to the UA. Yes, it's a different scenario. You could do some trick: - When P1 receives a REGISTER it sets a bflag for it (bflag 1). - When P2 receives a REGISTER it sets a bflag for it (bflag 2). - When P3receives a REGISTER it sets a bflag for it (bflag 3). When P1 receives a request and performs location for that AoR, it will also extract the bflags, so P1 checks if bflag 1 is on. If not and bflag 2 is on, then it routes (by setting $du => without changing the resolved RURI) the request to P2. :) -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] Disable auto ACK
by kévin l'heveder 21 Apr '09

21 Apr '09

Hi, i want to generate my own ACK after a OK ( After ringing). but even if i drop the OK in my reply_route a ACK is auto generated by Kamailio and sent How could i prevent this behavior ?? I found some clues in the sources : *#define no_autoack(_t_) ((_t_)->flags&T_NO_AUTOACK_FLAG)* But i didn't see how i could flag a message to disable AUTOACK If someone have an idea or THE response, that would be great!! Thanks. Koon

2 1

[Kamailio-Users] configuration
by shiva prasad 21 Apr '09

21 Apr '09

Hi All, I've successfully installed kamailio 1.5x version on fedora 9 os, could you please let me know is it possible to configure Ekiga(client) to connect kamailio server,. if possible, could you please let me know how to configure the client? Thanks in advance. -- Regards, Shivaprasad BS

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[Kamailio-Users] help using rewrite prefix
by Carlos A. Alvarez 20 Apr '09

20 Apr '09

I need help setting up carrierroute using the rewrite prefix and the strip function. It seem like every time I use the Rewrite Prfix function the carrierroute module will rewrite the prefix, but then I don't get certain reply messages on kamailio. At least from what I can see on the log. INFO:carrierroute:cr_do_route: uri 5555550001 was rewritten to sip:0115555550001@aslab.commx.net, carrier 1, domain 10 INFO:carrierroute:rewrite_uri_recursor: URI or prefix tree nodes empty, empty rule list INFO:carrierroute:cr_do_route: rewrite_uri_recursor doesn't complete, uri 0115555550001, carrier 1, domain 10 Is this normal behavior? I also noticed that some messages coming back from the host are not being processed by kamailio. For example on this particular call I purposely get a 604 from the server. The thing is I can't process a failure route for this message, because kamailio doesn't see is, even though it is being sent back to the server. Also when I use the Strip function is looks like carrierroute does a second lookup on the route table. In the example below I am calling a number using the prefix 551... The route table is below. +----+---------+--------+--------------+-------+------+------+-------+-----------------+----------------+----------------+------------------------+ | id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description | +----+---------+--------+--------------+-------+------+------+-------+-----------------+----------------+----------------+------------------------+ | 30 | 1 | 1 | 551 | 0 | 0 | 1 | 2 | 10.10.10.1 | | | LD-1 | 31 | 1 | 1 | 1 | 0 | 0 | 1 | 0 | 10.10.10.1 | | | LD | Log for that call INFO:carrierroute:cr_do_route: uri 5515555550004 was rewritten to sip:15555550004@10.10.10.1, carrier 1, domain 1 INFO:carrierroute:cr_do_route: uri 15555550004 was rewritten to sip:15555550004@10.10.10.1, carrier 1, domain 1 Carrier found!! ("1", "131.226.0.144", "15555550004", "15555550004", "COMMX-LD"); Now please note that if I remove route id 31 then the call log looks like it is below: INFO:carrierroute:cr_do_route: uri 5515555550004 was rewritten to sip:15555550004@10.10.10.1, carrier 1, domain 1 INFO:carrierroute:rewrite_uri_recursor: URI or prefix tree nodes empty, empty rule list INFO:carrierroute:cr_do_route: rewrite_uri_recursor doesn't complete, uri 15555550004, carrier 1, domain 1 Please help!!! Carlos It will also help if there was complete routing example using carrierroute v 1.5.0, since the one in the wiki is outdated. Thanks.

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Re: [Kamailio-Users] bidirectionnal chat and presence sharing between a SIP and a jabber user
by Daniel-Constantin Mierla 20 Apr '09

20 Apr '09

Please cc all the time to mailing lists. Emailing privately will make the message dropped. Also, mailing list has the benefit of providing access to other people that might know better the answer. Thanks, Daniel On 04/20/2009 01:18 PM, rachid wrote: > *Hello, > I use the purplemap table to mapping sip uri and jabber client,so i > want to have a bidirectionnal chat and sharing between a dip and > jabber users > > I tried with purple modules* > > 2009/4/20 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > On 04/17/2009 11:22 AM, rachid wrote: > > Hello all, > > someone can help me to configure kamailio to i was able to have a > bidirectionnal chat and presence sharing between a SIP and a > jabber user ? > thank you > > what is the problem? Did you get or not any communication between > sip and xmpp? Have you tried with xmpp or purple modules? > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > > > > > -- > ------------------------------------------------------------ > BAHA Rachid > Elève ingénieur en 3ème année à l'INPT, Rabat > GSM:(+33) 0619787609 > > -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

[Kamailio-Users] Question about route[4] for NAT
by x-kamailio＠sidell.org 20 Apr '09

20 Apr '09

Sorry for the potentially stupid question... I'm trying to make sense of the standard kamailio.cfg file. At the top, it says to remove the #n# comments to enable nat processing. Looking down in the code, I see that one of the sections to be enabled is route[4]. But, no where else in the file do I see that route(4) is ever called. Is the cfg file missing something, or is route(4) called in some way that is not apparent from reading the file? Thanks! -- Mark Sidell Partner Forte, Inc. 919-942-7068 fax 919-969-2844 www.forteinc.com

2 1

[Kamailio-Users] NAT & Branching in Parallel Forking Scenarios
by Brandon Armstead 20 Apr '09

20 Apr '09

Hello Guys, Going to say thanks for any input ahead of time. My question is currently we may have a scenario where we have a client using several UAC's registered to the same username. Now what I've seen is that sometimes one endpoint of User A may be NATed, another may be public. This seems to be causing some call flow control issues in a sense for when we handle for NAT / vs NOT NAT, at the time of a request and all of this is in parallel being handled. Example Scenario. User A calls User B -- (both internal). User B has end points X (NAT), Y(PUBLIC), and Z(NAT). So a call is being branched to X, Y, and Z in parallel. Now we fix_nated_contact for endpoint X, next we go to Y, however we've already fixed_nated_contact, thus this call is deemed bad, and then Z continues. Any thoughts / examples / suggestions for these types of scenarios? This is just an example one, one particular case where issues we have seen is more like follows: PSTN Calls User B (external to internal) User B has the same previous end points (X,Y,Z) + (PSTN End Points that we are adding in parallel to fork to). Same type issue, same type scenario. The end result of the issues are lack of audio, mis routed contacts, etc. So any help and input is appreciated, thanks!

3 2

[Kamailio-Users] run openser and asterisks in one box
by Metshet Seleshi 20 Apr '09

20 Apr '09

hi all, is it possible to run openser and asterisks in one box, I have googled but all the result I got have rewritehostport("X.X.X.X:5060) with a diffrent IP than the server IP, I am assuming either they have 2 network card or refering to another host . but will it be possible to put them togather? if so how, thanks, please help

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[Kamailio-Users] Profile takes some time to be cleared
by mayamatakeshi 20 Apr '09

20 Apr '09

Hello, I'm using the set_dlg_profile function from the dialog module (kamailio 1.4). I use it to control limit of simultaneous calls to clients. I'm noticing that after this function is called, even if the call terminates immediately, it takes some 3 to 4 seconds for the profile to be cleared. Is there any reason for this? I can see even if I call unset_dlg_profile on failure_route, the profile will take that time span to be cleared. regards, mayama

2 9

[Kamailio-Users] shift/reduce conflict
by priz＠libero.it 20 Apr '09

20 Apr '09

Hi, I'm a newbie of kamailio. I'm an Asterisk expert, and I want to explore the "*SER" world, because I need a real SIP proxy. I planned kamailio project to begin. I want the radius auth. I have a question about a compile message. If I try to compile kamailio- 1.5.0-tls, I always receive the message "cfg.y: conflicts: 1 shift/reduce" Before build: # make distclean # make mantainer- clean and then: # make all bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce flex cfg.lex The rest of build is OK, without errors. Is this a problem or is normal? And if is not normal, how to avoid this problem? I have a Debian Lenny # uname -a Linux voicelab 2.6.26-1-686 #1 SMP Sat Jan 10 18:29: 31 UTC 2009 i686 GNU/Linux Uncomment: Makefile: TLS=1 FREERADIUS=1 Makefile. vars MODS_MYSQL=on and MODS_RADIUS=on /usr/local/src/kamailio/kamailio-1.5.0- tls/modules/acc/Makefile ENABLE_RADIUS_ACC=true Many thanks Priz

4 3

[Kamailio-Users] opening 2 and more channels on 1 SIP account
by Tamer Higazi 18 Apr '09

18 Apr '09

Hi! I am new to opensips, and I am looking for a powerfull sip gateway sollution, because asterisk isn't able to handle such thing. My case: I have a Grandstream VoIP Device, at which a DECT base with 2 cordless phones are connected at 1 FX1 Port. If a call is placed and made through one cordless phone the other cordless phone appears as busy. What I want: I want to place and receive as many calls at the same time through 1 1 FXS Port with one single SIP Account, and through this where the base station is connected through. Question: Does kamilio have the ability for such things?! For any help and advise I would thank you kindly Tamer

2 1

[Kamailio-Users] More Complex Example Config File
by Alberto Listas 17 Apr '09

17 Apr '09

Hi, I am having a difficult time setting up a config file that works properly for a simple dispatcher with failover, database authentication and persistence. Could someone send me a more extensive example than the one that comes by default? Thanks, Alberto Furtado

2 1

[Kamailio-Users] Lumps again
by Alex Hermann 17 Apr '09

17 Apr '09

Hello all, i got bitten by the lumps mess again. I can understand the lump concept makes a bit of sense if requests are just passing though a proxy, but nowadays Kamailio obtains more and more UA functionality where it is completely unappropriate. How do I store a modified PUBLISH body? I change it with subst_body() before calling handle_publish() but ofcourse the original body is stored by the presence UA. Is there currently a method to store a modified body? (apart from looping the message back to myself) Would implementing an apply_lumps() function or something like that be very difficult? -- Greetings, Alex Hermann

4 3

[Serusers] switch
by hanane zahidi 17 Apr '09

17 Apr '09

Hi All; I need to use switch in my ser.cfg file .But ser give me some errors .so i need the correct syntax to do it here is my ser.cfg if ( method=="REGISTER"){ i = %fu % 2; switch(i){ case"1": route(1); break; case"0": route(2); break; } } thank tou for your help best regards

2 1

[Kamailio-Users] get sdp audio port in a script variable
by antonroman 17 Apr '09

17 Apr '09

Hi all, I need to get the INVITE sdp audio port in a script variable to be used in an external command. Is this possible without modifying the source code? I think it can't be done with the TEXTOP module, can be? Thanks in advance Regards -- View this message in context: http://www.nabble.com/get-sdp-audio-port-in-a-script-variable-tp23061909p23… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 3

[Serusers] SER authentication with LDAP
by Leon Li 17 Apr '09

17 Apr '09

Hi all, I just would like to know if we can link SER to LDAP server (Microsoft AD) for user authentication, rather than using MySQL. If yes, how can we achieve it? Thanks Leon

1 0

[Kamailio-Users] prefix() cannot take pseudovariable argument?
by Alex Balashov 16 Apr '09

16 Apr '09

Why can't prefix() accept a pseudovariable argument, e.g. prefix("$avp(S:the_prefix)"); ? It is interpreted literally. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

2 3

[Serusers] Looking for commercial support
by Jean-Paul Smets 16 Apr '09

16 Apr '09

Hi, We are looking for commercial support for SER and SEMS. More precisely, we are looking for an engineer or a company who could help us to improve our existing SER based infrastructure. We would also be happy to get support from someone who has good experience with SER, SEMS and python. We do not need much in terms of features (registrar + proxy + voice2mail), but we want to be able to solve any little issue related to specific softphone or hardware, and to learn how to write new scripts for SER and SEMS. Do you have any advice? Regards, JPS. -- Jean-Paul Smets-Solanes, Nexedi CEO - Tel. +33(0)6 29 02 44 25 ERP5 Enterprise: Free / Open Source ERP for Critical Applications http://www.erp5.com ERP5 Express: Hosted Open Source ERP for small companies http://www.myerp5.com Nexedi: Consulting and Development of Free / Open Source Software http://www.nexedi.com

1 0

[Kamailio-Users] reducing T1 timer per route
by Juan Perez 16 Apr '09

16 Apr '09

hello I am using lcr and have 3 gws that I send the calls to, so far I am handling error responses using "on_failure_route" and calling the next_gw function and it is working fine. But in cases when 1 of those gws is disconnnected or super busy and a 100 Trying is not received in the first 3 - 5 seconds I would like to drop that call and use the next_gw. Right now I have to wait until all retries fail to try the next gw, but that happens after 64*T1 = 32 seconds. How can I accomplish that? cheers jp

3 4

[Kamailio-Users] starting kamailio using kamailio binary
by Juan Perez 16 Apr '09

16 Apr '09

hello when using kamctl to start/stop kamailio it also provides a lot more "functions" like "kamctl lcr show", "kamctl ul show", etc. If I start kamailio using kamailio -f "path to my config" etc how can I have the same functionalities? regards jp

5 7

mailing list description
by Daniel-Constantin Mierla 16 Apr '09

16 Apr '09

SIP Router Project - Users mailing list

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