September 2008 - sr-users

[Kamailio-Users] failure route problem - duplicate invites
by Matthew McGuire 10 Sep '08

10 Sep '08

I'm having a problem with the route sections below. When my proxy receives an Invite to proxy to a PSTN gateway, the proxy is sending 2 invites very fast to the gateway in the first route block which ends up creating a mess with the gateway. Either I have something wrong in the routing code or I have an SER/OS/hardware timing problem. Any suggestions ? route[1] { xlog("L_INFO", "Sending to route 1\n%mb\n"); rewritehostport ("1.1.1.1:5060"); append_branch(); t_on_failure("1"); t_relay(); } failure_route[1] { if(t_check_status("487")) { break; } xlog( "L_INFO", "failure on route 1\n%mb\n"); append_branch(); route(2); break; } route[2] { xlog("L_INFO", "Sending to route 2\n%mb\n"); rewritehostport ("2.2.2.2:5060"); append_branch(); t_on_failure("2"); t_relay(); } failure_route[2] { if(t_check_status("487")) { break; } xlog( "L_INFO", "failure on route 2\n%mb\n"); append_branch(); route(3); break; } route[3] { xlog("L_INFO", "Sending to rout 3\n%mb\n"); rewritehostport ("3.3.3.3:5060"); t_relay(); } Thanks, Matt

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[Serusers] help required
by Engr. Muhammad Noman Saeed 10 Sep '08

10 Sep '08

Dear All, I need your help for establishing SER as a SIP server for my organization. I have downloaded it at my fedora 8.0 machine, but as i am new with linux and SER it is hard to find me how to install/configure. kindly help me as a new SER user. regards -- ========================= Noman =========================

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[Serusers] how can i config ser.cfg
by 邱磊 10 Sep '08

10 Sep '08

hi friends: i need some help! frist,my asterisk receive a sip message from sip proxy via SER.when my asterisk send 200 to SER and then receive ACK from proxy directly! when asterisk send BYE to the proxy and can hangup the call. but now i redirect the BYE to SER, i found the SER can not send the message to proxy! what can i do in the ser.cfg? thanks in advance 2008-09-10 邱磊

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[Kamailio-Users] add number to an avp ?
by BERGANZ François 09 Sep '08

09 Sep '08

I need to add 1 to an avp like : $avp(i:681)=$avp(i:681)+1; but don't work! how to? thank you Cordialement BERGANZ François http://www.acropolistelecom.net TEL/FAX : 33 (0) 1 70 72 50 15

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[Serusers] Dubt on RFC 4484 and SIP Proxy Server behavior
by Francesco la Torre 09 Sep '08

09 Sep '08

Hi all, I've a doubt regarding a statement at the end of page 5 "... Some important design decisions are associated with carrying assertions in a SIP request. If an assertion is carried by value, or uses a MIME-based content indirection system, then proxy servers will be unable to inspect the assertion themselves. If the assertion were referenced in a header, however, it might be possible for the proxy to acquire and inspect the assertion itself. There ..." Why a proxy server will be unable to inspect the assertion themself ? SER has functions by which it could be read message body, does this behavior not respect any rfc ? I'm missing something :-S Thanks in advance, Francesco la Torre

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[Kamailio-Users] [Fwd: Re: Unconditional call forward]
by Stefano Palleschi 09 Sep '08

09 Sep '08

Hi Gustavo, thanks for your reply. I've changed the *avp type* from "2" to "0" but I get always the same error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57> Now I'm going to reboot the server... If I'll have news I'll let you know. Regards, Stefano -------- Messaggio Originale -------- Oggetto: Re: [Kamailio-Users] Unconditional call forward Data: Wed, 3 Sep 2008 10:38:42 -0300 Da: Gustavo Mistrinelli <gmistrinelli(a)gmail.com> A: Stefano Palleschi <stefano.palleschi(a)okcom.it> CC: users(a)lists.kamailio.org Referenze: <48BE6299.5000805(a)okcom.it> Hi Stefano, Please check *avp type* that you are using, use type 0 instead of 2 See below the different types for avps: 0 - *AVP* with string name and string value 1 - *AVP* with string name and integer value 2 - *AVP* with integer name and string value 3 - *AVP* with integer name and integer value Regards, Gustavo On Wed, Sep 3, 2008 at 7:10 AM, Stefano Palleschi <stefano.palleschi(a)okcom.it <mailto:stefano.palleschi@okcom.it>> wrote: Hello all, i'm trying to do an unconditional call forward with openser. I've added the instructions below in the script: if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); xlog("$avp(s:callfwd)"); route(1); exit; } and i've created a record on the usr_preferences table of openser database: mysql> select * from usr_preferences; +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | 1 | 0666620996 | 81.29.176.57 <http://81.29.176.57> | callfwd | sip:0666653218@81.29.176.57 <mailto:sip%3A0666653218@81.29.176.57> | 2 | 2008-09-03 09:51:33 | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ 1 row in set (0.00 sec) like showed, i would like forward the call that arrive at the sip client 0666620996 to the pstn number 0666653218. At openser restarting I get always this error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57 <http://81.29.176.57>> without make any call. Why in the uri there isn't the first part (username@)? Thanks in advance. Stefano Palleschi _______________________________________________ Users mailing list Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users -- Gustavo Mistrinelli

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[Kamailio-Users] 302 redirect inside redirect
by Kent Pirlo 09 Sep '08

09 Sep '08

Ok, I am trying to get a redirect server working correctly, only cache is I need to hit another redirect server first, and grab info from the contact in the that 302 reply before processing and sending back a 300 to the original switch. call flow would be something like this: call comes into kamailio, 1) kamailio sends the invite to another proxy(1), 2) that proxy returns a 302,(if it times out it moves on to step 5) 3) kamailio sends an ACK back to that proxy(1), 4) kamailio looks at the contact header of the 302 reply, 5) depending on that info, I decide where to tell the originating switch where to go, 5) kamailio sends originating switch a 300 redirect with new contact header. 6) originating switch sends back ACK. I have tried doing this with a first step going to route(1) , which does a "t_on_reply" and "t_relay" to the second proxy. The onreply route, graps the contact saves it into variable and then sends to route(2), Route(2) looks at the variable, does some other lookups and sends t_reply("300","redirect"). The switch then sends the ACK, but kamailio says its not a local ACK and ignores it.. Then it keeps sending 300's back to the switch... Please help.. Thanks in Advance. kent

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[Kamailio-Users] Documentation Site Down
by Jon Farmer 09 Sep '08

09 Sep '08

Hi When I go to http://kamailio.org/docs/ i get a 404 error. Any ideas anyone? Regards Jon

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[Kamailio-Users] Forward call
by Thiago Rondon 09 Sep '08

09 Sep '08

Folks, My problem for forward call, that I need to change username/password and realm to do my forward. So, I try this: modparam("uac","from_restore_mode", "auto") modparam("uac","auth_realm_avp","$avp(s:uac_realm)") modparam("uac","auth_username_avp","$avp(s:uac_username)") modparam("uac","auth_password_avp","$avp(s:uac_password)" route { ... if(uri=~"^sip:0[0-9]{10}@") { xlog("L_INFO","uac_atual: $avp(s:uac_username) !!"); $avp(s:uac_username)="username"; $avp(s:uac_realm)="1.1.1.1"; $avp(s:uac_password)="password"; xlog("L_INFO","uac_trocado $avp(s:uac_username) !!"); route(3); exit; } .. } ... route[3] { t_on_failure("3"); # reset flag to mark no authentication yet performed resetflag(7); # forward to PSTN uac_replace_from("username","sip:username@1.1.1.1"); rewritehostport("1.1.1.1:5060"); xlog("L_INFO","rewritehost !!!!!!!!"); t_relay(); } .. So, when I do xlog, in the first entry, uac_username don't display, why ? And its strange, because in one supplier its ok, and in another its not run. But I don't undestand why uac_username is empty in the first xlog! Thanks, -Thiago Rondon

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[Kamailio-Users] populate_leg_info
by David Villasmil 08 Sep '08

08 Sep '08

Anyone lnow what this means? dialog:populate_leg_info: bad sip message or missing Contact hdr ERROR:dialog:dlg_onreq: could not add further info to the dialog openser 1.3.2 thnks

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[Kamailio-Users] New OpenXCAP release 1.0.0
by Adrian Georgescu 08 Sep '08

08 Sep '08

Hello, There is a new release of OpenXCAP server available. It contains many bug fixes and additions for important features like XML partial manipulation, xcap-caps, xcap-diff and RLS services that were not available at the initial release last year. With this features OpenXCAP fulfills now most if not all relevant requirements for an XCAP server implementation. A generic xcap client library and command line client is available in the clients directory, you can use it to manipulate documents on OpenXCAP or other XCAP server implementation. The server software is also operational at http://sip2sip.info so you can test against it by registering a SIP account. Changes in version 1.0.0 * Added RLS services (RFC 4662 and RFC 4826) * Added support for xcap-diff based on draft-ietf-simple-xcap-diff-09 * Added partial get/put/delete of elements in the XCAP documents * Added test suite for rls-services, resource-lists and partial updates * Many bug fixes from field experiences * Development status from beta to production * Switched to Python 2.5 * Improved documentation and testing suite * Changed license to GPL * Fixed MySQL operational error (2006) The software can be downloaded from: http://openxcap.org Many thanks to Denis Bilenko who did most of the works for this new release. Kind regards, Adrian Georgescu

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[Kamailio-Users] CDRTool and Prepaid
by Adrian Georgescu 08 Sep '08

08 Sep '08

Hello Thanks for the feedback from everyone. You can try cdrtool version 6.6.0, I have added parallel prepaid call support. The algorithm is described in PREPAID.txt cdrtool (6.6.0) unstable; urgency=low * Allow simultaneous calls for prepaid accounts, see the Simultaneous prepaid calls section in doc/PREPAID.txt * Added pseudo code examples for call control engine * Apply the changes from setup/mysql/alter_tables.mysql * Added interface to kill ongoing sessions in Prepaid page Testing and feedback is useful and welcome. Regards, Adrian

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[Kamailio-Users] call forward problem
by muhammad akl 06 Sep '08

06 Sep '08

hey all after making call forward feature using avpops module and its exported functions avp_db_load and avp_pushto i have a problem : assuming all calls to extension 12001 will be forwarded to extension 12008, in opensers's database i have added the following : username=12001 domain =mydomain attribute = callfwd value = sip:12008@mydomain type = 0 untill that every thing was fine and all calls comes to 12001 forward to 12008 without any problem , the problem comes from when i am using 12008 to call 12001 it calls itself ( mean 12008) so anyone can tell me what i have missed ? i have configured my openser.cfg like this in loaded modules section added loadmodules "avpops.so" > in module parameter section added : modparam("avpops", "avp_url", "mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") in Invite message handler added : if(avp_db_load("$ru/username","$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); route(1); exit; } any suggestions ?

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[Kamailio-Users] two proxies sharing the same database
by toly 05 Sep '08

05 Sep '08

Greetings, I have 2 proxies having the same database on backend. One is the primary, second is standby - OpenSER 1.2.1. That's how I implement DNS based failover. What I see, when all UA's registered on primary and I start standby, I's starting to cleanup location table according to it's own timer. So record is gone, when primary trying to updated the record on re-register it of course fails, as a result location table is wrong. What will be a solution? Thanks, Toly -- View this message in context: http://www.nabble.com/two-proxies-sharing-the-same-database-tp19335185p1933… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] How to solve early media during serial forking
by Iñaki Baz Castillo 05 Sep '08

05 Sep '08

Hi, AFAIK there is an issue in typical SIP/PSTN enviroments in the case of failover using different gateways serially. The main issue is that SIP doesn't define how a UAC should handle the case in which it receives various "183 Session Progress" from different early-dialogs, each of them containing early-media. It occurs to me that a SIP/PSTN gateway sometimes replies 183 with early media for 1 or 2 seconds, and inmediatelly it replies a 502. Well, this 502 could be caused by other node in the PSTN network, it's not necessarialy a problem fo the gateway [1]. In this case I do serial forking forwarding the request to other gateway. But the UAC already received early media from the first gateway, and most of the SIP phones just render the first early media they receive in case of multiple 183, and drop others. So basically the problem is that the UAC user doesn't listen the ringing coming from PSTN after the request has been forwarded to the second gateway. Is there something I could do at proxy level to avoid this issue? Sincerely I don't think so, since it makes no sense to disallow the first 183 (I cannot know that the gateway will later reply a 502). Thanks. [1] Anyway, *who* is creating the early media? Why the PSTN generates it if the request doesn't arrive to the called? "imagine there is no PSTN..." -- Iñaki Baz Castillo

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[Kamailio-Users] custom routing
by toly 05 Sep '08

05 Sep '08

Greetings, Suppose I need to do custom routing for outbound calls from UA - softphone. Each user account may have the following info attached to it: 1:provider_a:2:provider_b:3:provider_c which means: forward call to provider_a, if failed forward call to provider_b if failed forward call to provider_c if failed send response back to UA I do database lookup based on from, and get above string. Suppose I wrote my own dispatcher module which initially gets set to above string: 1:provider_a:2:provider_b:3:provider_c and forwards to first provider, then on failure_route I call next_provider and so on. Will it work across multiple messages? Will it work correctly for each child - I mean joe@domain and paul@domain should be handled according their own providers? Any other suggestions? Thanks, Toly -- View this message in context: http://www.nabble.com/custom-routing-tp19322478p19322478.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] connection to remote mysql database
by Asim Riaz 05 Sep '08

05 Sep '08

I am using Kamailio 1.2 , in t openser.cfg I defined the the url for db, which is not local ("auth_db|permissions|uri_db|usr_loc", "db_url", "mysql://openser:db_password@db_ip_address/openser") but still getting the error message below when start openser; /sbin/openser[13064]: permissions - initializing /sbin/openser[13064]: WARNING: File not found: //etc/openser/permissions.allow /sbin/openser[13064]: Default allow file (//etc/openser/permissions.allow) not found => empty rule set /sbin/openser[13064]: WARNING: File not found: //etc/openser/permissions.deny /sbin/openser[13064]: Default deny file (//etc/openser/permissions.deny) not found => empty rule set /sbin/openser[13064]: new_connection: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) /sbin/openser[13064]: ERROR: group_db_bind: unable to connect to the database /sbin/openser[13064]: ERROR:group:mod_init: unable to open database connection /sbin/openser[13064]: init_mod(): Error while initializing module group /sbin/openser[13064]: INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to kill any idea ? thanks in advance Asim

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[Kamailio-Users] Openser & Centile
by BERGANZ François 05 Sep '08

05 Sep '08

Hello, I have some problem with, openser & centile! When do a call from centile to openser, the call don't do "ringing", just "trying" and after, centile say "BYE" Have you an idea? Thank you

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Re: [Kamailio-Users] Openser & Centile
by BERGANZ François 05 Sep '08

05 Sep '08

I could have a look, but Asterisk-Openseràno problems Centile-openseràdont work ! Cordialement BERGANZ François <http://www.acropolistelecom.net> http://www.acropolistelecom.net TEL/FAX : 33 (0) 1 70 72 50 15 De : Darren Sessions [mailto:dmsessions@gmail.com] Envoyé : vendredi 5 septembre 2008 15:26 À : BERGANZ François Objet : Re: [Kamailio-Users] Openser & Centile Would be helpful to see some of the SIP signaling. _____________________________ Darren Sessions dmsessions(a)gmail.com http://www.darrensessions.com _____________________________ On Sep 5, 2008, at 7:24 AM, BERGANZ François wrote: Hello, I have some problem with, openser & centile! When do a call from centile to openser, the call don't do "ringing", just "trying" and after, centile say "BYE" Have you an idea? Thank you _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

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[Kamailio-Users] CDRTool and Prepaid
by Adrian Georgescu 05 Sep '08

05 Sep '08

The problem with concurrent prepaid calls and single balance is that you have to correlate between the call control and rating angine somehow so that all calls terminate when balnce becomes zero. The problem is a bit complex: Example: Balance = 10. A call starts to destination XXX, for the sake of example max session time = 2 minutes After one minute, you start second call to destination YYY which has a different price and your balance is not anymore 10 but depends on the duration of the first call which is in progress. What is the maximum session time for it given that the first call is already in progress? What should happen with the first call? I am looking for suggestions on implementing a proper algorithm to deal with this situation in the rating engine. If you have any I would be glad to hear it. Adrian

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[Serusers] /tmp/ser_fifo disaapears after rebooting
by Joseph Myalla 05 Sep '08

05 Sep '08

I have revmoved the ser and reinstalled again, just after the installation ser works, i can see the PID, calls are made and text messages between computers, lokking in the tmp/ser_fifo is availabe, but it desappears after rebooting, what could be the reason On Wed, Sep 3, 2008 at 11:00 AM, <serusers-request(a)lists.iptel.org> wrote: > Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.iptel.org/mailman/listinfo/serusers > or, via email, send a message with subject or body 'help' to > serusers-request(a)lists.iptel.org > > You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Serusers digest..." > > > Today's Topics: > > 1. Error opening ser's fifo /tmp/ser_fifo (Joseph Myalla) > 2. Fwd: Error opening ser's fifo /tmp/ser_fifo sorry its ubuntu > 8.04LTs (Joseph Myalla) > 3. Re: Error opening ser's fifo /tmp/ser_fifo (Atle Samuelsen) > 4. SER 0.9.7-pre3 and timers (Steven C. Blair) > 5. how can i get the ip ? ( ?? ) > 6. how can i get the ip? ( ?? ) > 7. how can i get the IP? ( ?? ) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 2 Sep 2008 12:12:39 +0100 > From: "Joseph Myalla" <chanangae(a)gmail.com> > Subject: [Serusers] Error opening ser's fifo /tmp/ser_fifo > To: serusers(a)lists.iptel.org > Message-ID: > <531a7e5d0809020412n27b87d6g8f7346a569d98286(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > I have downloades ser's debian packages and installed them in Ubuntu > Linux 6.0Ls Desktop edition > ser_0.9.7_i386.deb and the correspoding ser_mysql modules, > > trying commands serctrl ps or ul show i get an error "Error opening > ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I > get error SER/FIFO not accessible: 2 > > the are some instructions in the mailing list saying I should add > fifo_mode=0777 in the configuration file, I have done it but still the > errors exist. > > can any one help. > > > -- > Myalla J.C > Kibaha Education Centre > P.O Box 30054 Kibaha > Pwani > Tanzania > +255787680744 > > > ------------------------------ > > Message: 2 > Date: Tue, 2 Sep 2008 12:14:24 +0100 > From: "Joseph Myalla" <chanangae(a)gmail.com> > Subject: [Serusers] Fwd: Error opening ser's fifo /tmp/ser_fifo sorry > its ubuntu 8.04LTs > To: serusers(a)lists.iptel.org > Message-ID: > <531a7e5d0809020414q25b69c15ya029023b51ce367e(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > I have downloades ser's debian packages and installed them in Ubuntu > Linux 8.0Ls Desktop edition > ser_0.9.7_i386.deb and the correspoding ser_mysql modules, > > trying commands serctrl ps or ul show i get an error "Error opening > ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I > get error SER/FIFO not accessible: 2 > > the are some instructions in the mailing list saying I should add > fifo_mode=0777 in the configuration file, I have done it but still the > errors exist. > > can any one help. > > > -- > Myalla J.C > Kibaha Education Centre > P.O Box 30054 Kibaha > Pwani > Tanzania > +255787680744 > > > > -- > Myalla J.C > Kibaha Education Centre > P.O Box 30054 Kibaha > Pwani > Tanzania > +255787680744 > > > ------------------------------ > > Message: 3 > Date: Tue, 2 Sep 2008 13:18:02 +0200 > From: Atle Samuelsen <clona(a)cyberhouse.no> > Subject: Re: [Serusers] Error opening ser's fifo /tmp/ser_fifo > To: Joseph Myalla <chanangae(a)gmail.com> > Cc: serusers(a)lists.iptel.org > Message-ID: <20080902111802.GD63335(a)mystic.cyberhouse.no> > Content-Type: text/plain; charset=us-ascii > > Hi Joseph, > > have you started ser? because, if ser is not running you will not be > able to get access to /tmp/ser_fifo. > > You can also check if it's there by doing a ls /tmp/ser_fifo :) > > > - Atle > > > * Joseph Myalla <chanangae(a)gmail.com> [080902 13:13]: >> I have downloades ser's debian packages and installed them in Ubuntu >> Linux 6.0Ls Desktop edition >> ser_0.9.7_i386.deb and the correspoding ser_mysql modules, >> >> trying commands serctrl ps or ul show i get an error "Error opening >> ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I >> get error SER/FIFO not accessible: 2 >> >> the are some instructions in the mailing list saying I should add >> fifo_mode=0777 in the configuration file, I have done it but still the >> errors exist. >> >> can any one help. >> >> >> -- >> Myalla J.C >> Kibaha Education Centre >> P.O Box 30054 Kibaha >> Pwani >> Tanzania >> +255787680744 >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > > > ------------------------------ > > Message: 4 > Date: Tue, 2 Sep 2008 11:14:46 -0400 > From: "Steven C. Blair" <blairs(a)isc.upenn.edu> > Subject: [Serusers] SER 0.9.7-pre3 and timers > To: "serusers(a)iptel.org" <serusers(a)iptel.org> > Message-ID: > <20C52563FC353043A5A9B80D847590E102D6824900(a)MAILBOX.exchange.upenn.edu> > > Content-Type: text/plain; charset="us-ascii" > > > I'd like to ask if anyone if running a service using SER 0.9.7-pre3 software with greater than 2000 registrations? If so are any of you experiencing issues which seem to be related to timers not firing when expected. We set the inv_timeout to a number which approximates a certain number of rings. This worked well when our subscriber base was small however as the number of subscribers, and associated calling activity, grew the timers seem to take longer to fire. We are planning on upgrading to a newer version of SER but I'd like to hear about other real world experiences. > > Thanks, Steve > > > > >

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[Kamailio-Users] Mediaproxy vs. RTPproxy.
by Alex Balashov 05 Sep '08

05 Sep '08

This must be an incredibly dumb question, but why would anyone ever use mediaproxy over rtpproxy? If one is written in Python and one in C, it seems to me that the performance and scalability makes the choice of rtpproxy a no-brainer. What benefit is there to using mediaproxy? More features and capabilities, ease of configuration perhaps? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] FW: Call Hangup from Caller-End not working with TLS
by Ali Jawad 05 Sep '08

05 Sep '08

Hi All I am using using openser 1.3..if I make a call between two softphones on the same lan or a a pstn call to my mobile phone..and the called/receiver party does hang-up the call. It works fine in UDP mode and the call get's hang-up. However in TLS mode this does not work. Anything I might have missed here? Since both udp and tls use the same routes, and voice is fine and no one way audio ..etc. I notice this http://pastebin.com/m38c979f6 on rtp proxy. However in the logs of openser I can't see any errors. However on the wire shark I can see icmp destination unreachable...port unreachable. I would have said it is a NAT issue. However it works for simple UDP. However I did notice the following in the logs Aug 28 13:41:01 [8565] DBG:tm:set_timer: relative timeout is 4000000 Aug 28 13:41:01 [8565] DBG:tm:insert_timer_unsafe: [7]: 0xb6137794 (45900000) Aug 28 13:41:01 [8565] DBG:tm:retransmission_handler: retransmission_handler : done Aug 28 13:41:02 [8564] DBG:core:parse_msg: SIP Request: Aug 28 13:41:02 [8564] DBG:core:parse_msg: method: <BYE> Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri: <sip:michofr@193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B200 8F980980E97E42F8EC;nat=yes> Aug 28 13:41:02 [8564] DBG:core:parse_msg: version: <SIP/2.0> Aug 28 13:41:02 [8564] DBG:core:parse_headers: flags=2 Aug 28 13:41:02 [8564] DBG:core:parse_via_param: found param type 232, <branch> = Shouldn't the transport=TLS ? Regards

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[Serusers] how can i get the ip?
by 邱磊 05 Sep '08

05 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? regards 2008-09-05 邱磊

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[Kamailio-Users] Help with SerMyAdmin
by Tiago Rés 04 Sep '08

04 Sep '08

Hello all, I'm a new user in this list, but I have already seen some posts about serMyAdmin. I have openser 1.2.2 installed, running with mysql Version 14.12. I'm trying to use a web interface for openser, such as sermyadmin or openser administrator, but I'm unable to accomplish that. I have followed the instructions from sermyadmin website, but I had some errors: I couldn't add a user in mysql with that command, so I used something like: openserctl add admin openserrw admin@localhost The remaining steps went ok, but everytime I try to open http://(myserver):8080/serMyAdmin I get an error page from Tomcat. Any suggestion? Thanks in advance. Tiago

2 3

[Kamailio-Users] Fax Detection
by Abdul Hakeem 04 Sep '08

04 Sep '08

Hello, Is there a way to identify in incoming call as a fax (perhaps by detecting CNG tone ). Regards, Abdul Hakeem

2 1

[Kamailio-Users] ERROR:mysql:db_mysql_new_connection: driver error: Too many connections
by Samael - 04 Sep '08

04 Sep '08

Hi, We are deploying OpenSER in a large infrastructure together with Asterisk to support thousands of current transactions. We are using udp listener and mysql module in openser. We had to increase the children parameter of the openser.cfg to 200 because Openser couldn't handle the high amount of concurrent SIP REQUESTs from ASTERISK (Not only INVITES, but also a lot of SIP OPTIONs), that means that Openser couldn't handle more than 'children' REQUEST loosing the rest of the concurrent REQUEST. children=200 But then, the following error appeared in the openser: ERROR:mysql:db_mysql_new_connection: driver error: Too many connections This is because the Standard parameter of mysql "max_connections" is set to 100. In mysql CLI, we can see that every child process of openser has a mysql connection. (see in Mysql> SHOW PROCESSLIST) The recommended max_connections parameter is 500-1000 => we have a limit of 500-1000 concurrent SIP REQUEST to be processed by the openser. Why there is a process for every child? Why is neccesary a mysql connection for every child? Can the openser have a pool of mysql connection to be used by thousands of children? Regards

4 6

[Kamailio-Users] OpenSER reply 513 when register use FQDN
by Steven Wu 04 Sep '08

04 Sep '08

Hi, I just want use FQDN in my networking environment. My SJPhone use FQDN register to OpenSER 1.2.2 but was rejected by 513 Message to big. However it can be registed use the same profile just replace FQDN to IP address. The DNS query process worked fine and got the right IP address. My environment elments looks like: OpenSER 1.2.2 at host 192.168.1.120 / ipcsip.org DNS server at host 10.57.39.53 SJPhone at host 192.168.1.105 REGISTER package looks like: 192.168.1.105 -> 192.168.1.120 REGISTER sip:ipcsip.org:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.105;branch=z9hG4bKc0a801690000001f48bf8e4700006b9400000002;rport From: "unknown" <sip:1000@ipcsip.org:5062>;tag=328715c58a2 To: <sip:1000@ipcsip.org:5062> Contact: <sip:1000@192.168.1.105> Call-ID: 444CD1265B43407EBBF2B26BD4C01C6B0xc0a80169 CSeq: 1 REGISTER Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 192.168.1.120 -> 192.168.1.105 SIP/2.0 513 Message too big Via: SIP/2.0/UDP 192.168.1.105;branch=z9hG4bKc0a801690000001f48bf8e4700006b9400000002;rport=5060 From: "unknown" <sip:1000@ipcsip.org:5062>;tag=328715c58a2 To: <sip:1000@ipcsip.org:5062>;tag=2560d490c3265ff35995c6bbde62a7c3.b7d8 Call-ID: 444CD1265B43407EBBF2B26BD4C01C6B0xc0a80169 CSeq: 1 REGISTER Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 I got same result evenif enabled dns & rev_dns in openser.cfg (dns=yes rev_dns=yes). So, What wrong with me? Coudle some give me some instructions? Best regards, Steven Wu

4 7

[Kamailio-Users] Unconditional call forward
by Stefano Palleschi 04 Sep '08

04 Sep '08

Hello all, i'm trying to do an unconditional call forward with openser. I've added the instructions below in the script: if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); xlog("$avp(s:callfwd)"); route(1); exit; } and i've created a record on the usr_preferences table of openser database: mysql> select * from usr_preferences; +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | 1 | 0666620996 | 81.29.176.57 | callfwd | sip:0666653218@81.29.176.57 | 2 | 2008-09-03 09:51:33 | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ 1 row in set (0.00 sec) like showed, i would like forward the call that arrive at the sip client 0666620996 to the pstn number 0666653218. At openser restarting I get always this error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57> without make any call. Why in the uri there isn't the first part (username@)? Thanks in advance. Stefano Palleschi

2 1

[Serusers] Which kind of SIP message can I use to exchange Data out of session?
by Peter Cai 04 Sep '08

04 Sep '08

Hi All, If a SIP client has to exchange some data out of a Session with the proxy server, such as get his "buddy list", what SIP Message can I use? I didn't find any thing suit for this purpose. Should I use something beside SIP such as HTTP.

3 3

[Serusers] cannot load module acc.so
by 静于 04 Sep '08

04 Sep '08

hello, I cannot load the acc.so module.The config file is below: # ------------- version 0.8.11-0 # ------------- Initial global variables debug=3 fork=yes log_stderror=no listen=10.1.2.3 listen=127.0.0.1 # hostname matching an alias will satisfy the condition uri==myself". alias=bigu.edu alias=10.1.2.3 # dns - Uses dns to check if it is necessary to add a "received=" field # to a via. Default is no. # rev_dns - Same as dns but use reverse DNS. dns=no rev_dns=no port=5060 children=4 # check_via - Turn on or off Via host checking when forwarding replies. # Default is no. arcane. looks for discrepancy between name and # ip address when forwarding replies. check_via=yes # syn_branch - Shall the server use stateful synonym branches? It is # faster but not reboot-safe. Default is yes. syn_branch=yes # memlog - Debugging level for final memory statistics report. Default # is L_DBG -- memory statistics are dumped only if debug is set high. memlog=3 # sip_warning - Should replies include extensive warnings? By default # yes, it is good for trouble-shooting. sip_warning=yes # fifo - FIFO special file pathname fifo="/tmp/ser_fifo" # server_signature - Should locally-generated messages include server's # signature? By default yes, it is good for trouble-shooting. server_signature=yes # reply_to_via - A hint to reply modules whether they should send reply # to IP advertised in Via. Turned off by default, which means that # replies are sent to IP address from which requests came. reply_to_via=no # user | uid - uid to be used by the server. 99 = nobody. uid="nobody" # group | gid - gid to be used by the server. 99 = nobody. gid="nobody" # mhomed -- enable calculation of outbound interface; useful on # multihomed servers. mhomed=0 # ------------- external module loading loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # ------------- tm parameters modparam("tm", "fr_timer", 12) modparam("tm", "fr_inv_timer", 24) # ------------- rr parameters # set ";lr" tag to ?�?;lr=true?�? modparam("rr", "enable_full_lr", 1) # ------------- accounting parameters modparam("acc", "log_missed_flag", 3) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) # ------------- usrloc parameters # 2 enables write-back to persistent mysql storage for speed # disable=0, write-through=1 modparam("usrloc", "db_mode", 2) # minimize write back window - default is 60 seconds modparam("usrloc", "timer_interval", 10) # database location modparam("usrloc", "db_url", "sql://ser:>password>@localhost/ser") # ------------- auth parameters # database location modparam("auth_db", "db_url", "sql://ser:>password>@localhost/ser") # allows clear text passwords in the mysql database modparam("auth_db", "calculate_ha1", yes) # name of password column in mysql database modparam("auth_db", "password_column", "password") # ------------- routing logic route { # ------------- routine checks # stop forwarding at 10 hops to prevent infinite loops if (!mf_process_maxfwd_header("10")) { log(1, "LOG: Too many hops\n"); sl_send_reply("483", "Too many hops"); break; }; # prevents private ip space from being used if (search("^(Contact|m): .*(a)(192\.168\.|10\.|172\.16)")) { if (method=="REGISTER") { log(1, "LOG: Someone trying to register from private IP\n"); sl_send_reply("479", "Please don't use private IP addresses" ); break; }; }; # separate the destination r-uri from the set of proxies that must be traversed loose_route(); # if the host portion of the request uri is not local, send it directly # to route processing. if (!(uri==myself)) { route(2); break; }; # All REGISTER attempts are processed and must always be authenticated if (method=="REGISTER") { # make sure that users don't register infinite loops if (search("^(Contact|m): .*(a)(10\.1\.2\.3|(proxy\.)?bigu\.edu)")) { log(1, "LOG: alert: someone trying to set aor==contact\n"); sl_send_reply("476", "No Server Address in Contacts Allowed" ); break; }; # challenge/response if (!www_authorize("bigu.edu", "subscriber")) { www_challenge("bigu.edu", "0"); break; }; # only registered users are allowed if (!is_user("replicator") & !check_to()) { log(1, "LOG: unregistered user registration attempt\n"); sl_send_reply("403", "Only registered users are allowed"); break; }; # it is an authenticated request, update Contact database now if (!save("location")) { sl_reply_error(); }; break; }; # process traffic local to BigU and the PSTN # Find the canonical username lookup("aliases"); # check domain again, if it is not still local after the alias # table lookup, just send it on its way. We do not authenticate # traffic we forward if (!(uri=~"^sip:(.+@)?(10\.1\.2\.3|(proxy\.)?bigu\.edu)([:;\?].*)?$")) { route(5); break; }; # now check for destinations through the gateway. 911 and 9911 # are always sent to the gateway. The assumption is that other all # numeric usernames between 5 and 20 digits are really pstn numbers # and so they are routed to the gateway if ( (uri=~"^sip:911@.*") | (uri=~"^sip:9911@.*") | (uri=~"sip:[0-9]{5,20}@.*") ) { route(3); break; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail (ser->grp) group?) if (is_user_in("Request-URI", "voicemail")) { t_on_failure("4"); setflag(4); }; # handle local SIP destinations not found in usrloc db # mostly offline or non-existent users if (!lookup("location")) { route(4); break; }; # check whether some inventive user has uploaded gateway # contacts to usrloc to bypass authorization logic if (uri=~"@10\.1\.2\.5([;:].*)*" ) { log(1, "LOG: Gateway address in UsrLoc\n"); route(3); break; }; # this flag is used with the acc module to report missed calls # to syslog. setflag(3); # do it (words to live by) append_hf("P-hint: USRLOC\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } /* end of initial routing logic */ # ------------- process traffic leaving BigU for Internet route[2] { # outbound requests are allowed only for registered BigU users if (!(src_ip==10.1.2.3) & !(proxy_authorize("bigu.edu", "subscriber"))) { # ACK and CANCEL have no security mechanisms so they are just # noted if (method=="ACK" | method=="BYE") { log(1, "LOG: failed outbound authentication for ACK granted\n"); } else if (method=="CANCEL") { log(1, "LOG: failed outbound authentication for CANCEL granted\n"); } else { proxy_challenge("bigu.edu", "0"); break; }; }; # to maintain credibility of our proxy, we check >From in INVITEs if (!src_ip==10.1.2.3 & method=="INVITE" & !check_from()) { log(1, "LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; append_hf("P-hint: OUTBOUND ON INTERNET\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- process traffic leaving Internet for PSTN route[3] { # all calls through the gateway must be record routed to assure # acl acceptance on the gateway record_route(); # send out emergency calls to pstn gateway immediately if ( (uri=~"^sip:911@.*") | (uri=~"^sip:9911@.*") ) { rewritehostport("10.1.2.5:5060"); forward(uri:host, uri:port); break; }; # five digit numeric addresses are internal freebies sent to the pbx # without authentication if (uri=~"^sip:[0-9]{5}@(10.1.2.3|(proxy\,)?\.bigu\.edu)") { rewritehostport("10.1.2.5:5060"); forward(uri:host, uri:port); break; }; # all numeric addresses beginning with 9 go to the pbx on the way # to the PSTN # first the caller needs to be authenticated if (uri=~"^sip:9[0-9]*@(10.1.2.3|(proxy\.)?bigu\.edu)") { if (!(src_ip==10.1.2.3 | method==ACK | method=="CANCEL" | method=="BYE")) { if (!proxy_authorize("bigu.edu", "subscriber")) { proxy_challenge( "bigu.edu","0"); break; } else if (method=="INVITE" & !check_from()) { log(1, "LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; }; if (method=="INVITE") { # if the r-uri begins 91, does the authenticated user have # permission for long distance if (uri=~"sip:91[0-9]*@.*") { if (!is_user_in("credentials", "ld")) { sl_send_reply("403", "Local calls only"); break; }; }; }; # authenticated and authorized, now accounting is set setflag(1); }; rewritehostport("10.1.2.5:5060"); append_hf("P-hint: GATEWAY\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- process calls for users offline route[4] { if (!t_newtran()) { sl_reply_error(); }; if (!t_reply("404", "Not Found")) { sl_reply_error(); }; break; } # ------------- process aliased outbound traffic # inbound requests that have been aliased to a non-BigU domain # are not authenticated by BigU route[5] { append_hf("P-hint: ALIASED-OUTBOUND\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- CC-Diversion to voicemail failure_route[4] { append_branch("sip:80000@10.1.2.5"); append_urihf("CC-Diversion: ", "\r\n"); append_hf("P-hint: OFFLINE-VOICEMAIL\r\n"); t_relay(); } Can anyone help BR Ivy ___________________________________________________________ �Ż��䣬��䣡 http://cn.mail.yahoo.com/

1 0

Re: [Kamailio-Users] svn diff
by Vadim Berezniker 04 Sep '08

04 Sep '08

Oops hit Ctrl+Enter svn diff http://openser.svn.sourceforge.net/svnroot/openser/tags/1.2.2/ http://openser.svn.sourceforge.net/svnroot/openser/tags/1.2.3/ I've been trying to run this against the public SVN but it never completes.... I hate sourceforge. From: Vadim Berezniker Sent: Wednesday, September 03, 2008 1:57 PM To: 'users(a)lists.kamailio.org' Subject: svn diff Can someone with access to the dev svn run this for me?

3 3

[Kamailio-Users] Could you please help me how to integrate RLS AS and PS AS with SERIMS?
by mahesh.peddi＠info-spectrum.com 04 Sep '08

04 Sep '08

Hi, I am working on openser 1.3.2 as RLS application server and presence application server. I want to integrate this two application servers with SERIMS. Could you please help me how to integrate and make run this. To do this I have run SERIMS components on two servers? Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

1 0

[Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
by Kristian Kielhofner 03 Sep '08

03 Sep '08

Hello everyone, Feel free to set $OSSPROXY to whatever you like = Kamailio/OpenSIPS/OpenSER (for those of you who can't let go). $OSSPROXY is my favorite SIP toolkit. In addition to being an excellent proxy and registrar it serves as a platform to do SIP mangling and wrangling. With that in mind let me propose a crazy idea... How hard would it be to get $OSSPROXY to be able to parse what is commonly referred to as SIP-T (RFC 3372)? I understand $OSSPROXY has no business getting too involved in the SDP but I feel it makes sense in some instances (like this one). What if I want to make a routing decision based on an encapsulated ISUP parameter of some sort? Who knows what it could be!?!? Maybe there are some other uses but that makes the most sense and seems to be the most practical. It should be possible to develop an $OSSPROXY module to: - Use the SDP parser to find the ISUP SDP part: http://www.kamailio.net/dokuwiki/doku.php/development:sdp-parsing - Pass that part into a "real" ISUP parser (if necessary). I couldn't seem to find the perfect library but perhaps something like this could work: http://www.openss7.org/isup.html - Make the ISUP parameters available via AVPs, pseudo variables, etc for use in the $OSSPROXY script. I don't really have a need for this but it seems like it could be cool. Am I nuts? -- Kristian Kielhofner http://blog.krisk.org

3 3

[Serusers] Is setdsturi supported in SER
by Steven C. Blair 03 Sep '08

03 Sep '08

I am wondering if setdsturi is supported in SER 0.9.7-pre3 Thanks

1 0

[Kamailio-Users] svn diff
by Vadim Berezniker 03 Sep '08

03 Sep '08

Can someone with access to the dev svn run this for me?

1 0

[Kamailio-Users] Unconditional call forward
by Stefano Palleschi 03 Sep '08

03 Sep '08

Hello all, i'm trying to do an unconditional call forward with openser. I've added the instructions below in the script: if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); xlog("$avp(s:callfwd)"); route(1); exit; } and i've created a record on the usr_preferences table of openser database: mysql> select * from usr_preferences; +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | 1 | 0666620996 | 81.29.176.57 | callfwd | sip:0666653218@81.29.176.57 | 2 | 2008-09-03 09:51:33 | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ 1 row in set (0.00 sec) like showed, i would like forward the call that arrive at the sip client 0666620996 to the pstn number 0666653218. At openser restarting I get always this error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57> without make any call. Why in the uri there isn't the first part (username@)? Thanks in advance. Stefano Palleschi

2 1

[Kamailio-Users] VQ_MALLOC memory manager
by Henning Westerholt 03 Sep '08

03 Sep '08

Hi all, does somebody uses the VQ_MALLOC memory manager? I thought about removing this one, as it don't work correctly at the moment (it lacks a realloc implementation). This would be a good cleanup in the core. If somebody still depends on this memory manager, perhaps has also some fixes created for this, please reply to this thread. If nobody speaks up i'll remove the VQ_MALLOC implementation at the end of this month. Cheers, Henning

1 0

[Kamailio-Users] NAT transversal without mediaproxy or rtpproxy
by BERGANZ François 03 Sep '08

03 Sep '08

Hello, I am looking for to do nat transversal without mediaproxy or rtpproxy! I cant find any example without mediaproxy or rtpproxy. Have you an idea? I need to register my users, and contact them after Thank you

4 6

[Kamailio-Users] t_relay to two diffrent proxy
by Asim Riaz 03 Sep '08

03 Sep '08

Hi , I am trying to relay invite from Kamailio to one of two proxies, as long as it receive the reply from one it should not forward to second but incase if the proxy1 does not reply then relay to proxy2. here is my config route(1); if((src_ip != proxy1_ip_address) || (src_ip !=proxy2_ip_address) ) { if(!t_relay_to_udp("proxy1_ip_address", "5060")) || (!t_relay_to_udp("proxy2_ip_address","5060")) { end_media_session(); sl_reply_error(); } } I am also allowing only my proxies can see the location tables; f ((src_ip== Proxy1_ip_address) || (src_ip= Proxy2_ip_address)) { if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; for some reason incoming calls from my proxies are looping means Kamailio sending back to one of my proxyies, i dont know if there is somthing missing in my configuration or somthing else is causing this problem. Regards Asim Riaz

2 2

[Kamailio-Users] Kamailio dispatcher Module redirection
by Asim Riaz 03 Sep '08

03 Sep '08

Hi All I am trying to configure Openser as loadblancer for two more Kamailio. I wan loardbalancer not stay in the middle once routed to other kamilio servers. my first question is, dispathcer list always take IPs or I can configure proxy names as well. at the moment I am checking the to_uri and then redirecting the traffic to my one of Kamilio on my network, it seems to work fine with Linksys but not other phones e.g polycom xlite or even sjphone. if(to_uri=~"sip:.+@sip.mydomain.ie <sip%3A.%2B(a)sip.mydomain.ie>") { ds_select_domain("1", "4"); sl_send_reply("300","Redirect"); exit; } any suggestion. Thanks in advance Regards Asim Riaz

4 8

[Kamailio-Users] uac_redirect question
by Zahid Mehmood 03 Sep '08

03 Sep '08

Hi, Need a little help with get_redirects(), using openser 1.2. I am using the following in failure_route to capture contacts from a "multiple choices" reply from cisco media gateway. Following code sets the ruri to the last contact, however, the $ds show more information about other contacts. Is there a way to reset this so that only ruri information is used? Cisco response and xlog information is also given below. if(!get_redirects("*:*")) { xlog("L_ERROR", "Failed to fetch contact '$ct' from 301/302 - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci \n"); acc_db_request("480", "acc"); t_reply("480", "Temporarily Unavailable"); exit; } # get last URI from destination-set and set it as R-URI xlog("L_INFO", "Redirect from UAC intercepted 1 - M= $rm RURI=$ru D=$ds B=$bR \nF=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:tmp)/g"); $avp(s:tmp) = $ds; avp_subst("$avp(s:tmp)", "/.*(sip:.+@[^:;>]+).*$/\1/"); avp_pushto("$ru", "$avp(s:tmp)"); setflag(29); xlog("L_INFO", "Redirect from UAC intercepted 2 - M= $rm RURI=$ru D=$ds B=$bR \ntmp=$avp(s:tmp) \nF=$fu T=$tu IP=$si ID=$ci \n"); append_branch(); route(17); # process exit; } Thanks in advance for your advice. -- Zahid The response from gateway: U 2008/08/26 08:19:04.440411 10.10.0.32:5060 -> 10.10.0.98:5060 SIP/2.0 300 Multiple Choices. Via: SIP/2.0/UDP 10.10.0.98;branch=z9hG4bK901b.335cb695.0,SIP/2.0/UDP 10.10.12.140;branch=z9hG4bK612a4f20B9293665. From: "10521" <sip:10521@devproxy.myip.org>;tag=B395F9F2-4618AC3F. To: <sip:40001@devproxy.myip.org;user=phone>;tag=195535C4-11AC. Date: Tue, 26 Aug 2008 12:19:04 GMT. Call-ID: f7932e23-d89cd014-e3ea20a9(a)10.10.12.140. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 2 INVITE. Allow-Events: telephone-event. Diversion: <sip:40001@10.10.0.32>;reason=unconditional;counter=1. Contact: <sip:10512@10.10.0.98>,<sip:10512@10.10.0.32>. Content-Length: 0. xlog entries: Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted 1 - M=INVITE RURI=sip:40001@10.10.0.32:5060;transport=udp D=Contact: sip:40001@10.10.0.32:5060;transport=udp, <sip:10512@10.10.0.32 >;q=0.01, <sip:10512@10.10.0.98>;q=0.01 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140 Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted 2 - M=INVITE RURI=sip:10512@10.10.0.98 D=Contact: sip:10512@10.10.0.98, <sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98 >;q=0.01 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01 tmp=sip:10512@10.10.0.98 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140 Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted - M=INVITE RURI=sip:10512@10.10.0.98 D=Contact: sip:10512@10.10.0.98, <sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98 >;q=0.01, sip:10512@10.10.0.98 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01, sip:10512@10.10.0.98 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140

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[Kamailio-Users] t_relay
by sabino frisardi 03 Sep '08

03 Sep '08

Hi all, I'm using openser 1.4. Can anyone tell me how can i call t_relay() from another module. Can anyone help me? any suggestion? Thanks for your help. Regards -- Sabino Frisardi

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[Serusers] how can i get the IP?
by 邱磊 03 Sep '08

03 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? 2008-09-03 邱磊

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[Serusers] how can i get the ip?
by 邱磊 03 Sep '08

03 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? regard~ 2008-09-03 邱磊

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[Serusers] how can i get the ip ?
by 邱磊 03 Sep '08

03 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? regard~ 2008-09-03 邱磊

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Re: [Kamailio-Users] CDRTool and Prepaid
by Matteo D'Amato 03 Sep '08

03 Sep '08

Thanks David for your answer, but can you explain how radius can do that since sending BYE messages is more at the signaling level. I dont quite understand how radius can even communicate at the SIP signaling level. From: David Villasmil Sent: Sunday, August 31, 2008 4:33 PM To: Matteo D'Amato Subject: Re: [Kamailio-Users] CDRTool and Prepaid You could use radius. On Sun, Aug 31, 2008 at 10:09 PM, Matteo D'Amato > wrote: Hello, In "/doc/PREPAID.txt" it talks about "an external call control module (not provided by CDRTool) that maintains call status and terminates calls by sending BYEs to both SIP end-points" What is this 3-party module and where can I get it? --Matteo _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

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[Serusers] SER 0.9.7-pre3 and timers
by Steven C. Blair 02 Sep '08

02 Sep '08

I'd like to ask if anyone if running a service using SER 0.9.7-pre3 software with greater than 2000 registrations? If so are any of you experiencing issues which seem to be related to timers not firing when expected. We set the inv_timeout to a number which approximates a certain number of rings. This worked well when our subscriber base was small however as the number of subscribers, and associated calling activity, grew the timers seem to take longer to fire. We are planning on upgrading to a newer version of SER but I'd like to hear about other real world experiences. Thanks, Steve

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[Serusers] Error opening ser's fifo /tmp/ser_fifo
by Joseph Myalla 02 Sep '08

02 Sep '08

I have downloades ser's debian packages and installed them in Ubuntu Linux 6.0Ls Desktop edition ser_0.9.7_i386.deb and the correspoding ser_mysql modules, trying commands serctrl ps or ul show i get an error "Error opening ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I get error SER/FIFO not accessible: 2 the are some instructions in the mailing list saying I should add fifo_mode=0777 in the configuration file, I have done it but still the errors exist. can any one help. -- Myalla J.C Kibaha Education Centre P.O Box 30054 Kibaha Pwani Tanzania +255787680744

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[Serusers] Fwd: Error opening ser's fifo /tmp/ser_fifo sorry its ubuntu 8.04LTs
by Joseph Myalla 02 Sep '08

02 Sep '08

I have downloades ser's debian packages and installed them in Ubuntu Linux 8.0Ls Desktop edition ser_0.9.7_i386.deb and the correspoding ser_mysql modules, trying commands serctrl ps or ul show i get an error "Error opening ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I get error SER/FIFO not accessible: 2 the are some instructions in the mailing list saying I should add fifo_mode=0777 in the configuration file, I have done it but still the errors exist. can any one help. -- Myalla J.C Kibaha Education Centre P.O Box 30054 Kibaha Pwani Tanzania +255787680744 -- Myalla J.C Kibaha Education Centre P.O Box 30054 Kibaha Pwani Tanzania +255787680744

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[Serusers] Local number portability
by inge 02 Sep '08

02 Sep '08

Hi all, Is someone know if it's possible to use SER for prefix calls for LNP ? Then the call will be sent to AS5350 and the prefix must define the carrier for out ported number. I search information on this topic but I find nothing about SIP and LNP. Thanks for our informations. Regards

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[Kamailio-Users] route number
by BERGANZ François 02 Sep '08

02 Sep '08

Hello, I need to have a route number like route(1000) But after some tests, I cant have more than route(39) How can I have more? Thank you

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[Kamailio-Users] How to config OpenSER as IMS server
by Steven Wu 01 Sep '08

01 Sep '08

Hi everybody, Currently we are working on an IMS project. This project need a IMS server. Finally we choiced OpenSER as the IMS server. But have no experience how to use OpenSER as a IMS server. Could someone tell me how to config it? Best regards, Steven Wu

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[Kamailio-Users] Mediaproxy 2.0 - Dispatcher vs Relay
by Nuno Marques 01 Sep '08

01 Sep '08

Hi, I'm trying to understand better how mediaproxy works. Just read the mediaproxy-ng.org page... i understood the necessity of using mediaproxy for NAT traversal, but i have one doubt... - What's the difference between the dispatcher and the relay? Can someone explain me the concept or point me to some literature so i can clear my mind :) Thanks in advance, Nuno

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[Kamailio-Users] kamailio 1.5 and beyond
by Henning Westerholt 01 Sep '08

01 Sep '08

Hi all, i want to share some thoughts where i think the priorities for the next major release(s) of Kamailio should be. This is the continuation of the work i (and many others) have done for the previous releases, but i thought it makes sense to summarize it a bit on a higher (abstraction) level. 1. Refactoring and cleanup We still have quite a lot of old code in our repository, which is no longer working or obsolete. Some implementations uses way to much gotos, or macros that are several pages long. This makes important parts of the core and modules harder to understand and maintain then necessary. Especially in the area of utility functions there is still to much duplicated code. proposed actions: - identification and removal of no longer working code in the core and modules - identification and unification of redundant (duplicated) code in core and modules to one implementation - identification and removal of obselete code parts in core and modules. - refactoring of code to use less gotos and macros 2. User and developer documentation With regards to user documentation quite a lot of work has been done in the past releases. But with the developer documentation, especially in certain modules, there is still plenty of content missing. As the involvement of individuals and companies can change, documentation is necessary to ensure a continued maintenance of the code. proposed actions: - conversion of already existing documentation to doxygen format - add a short doxygen module definition to every module, explaining the purpose - add a short doxygen file definition to every file, explaining the purpose - extend doxygen docs in important areas like the core and modules like TM - autogeneration of database related module documentation from the XML scheme definitions 3. Consolidation in core and modules At the moment we've 85 modules in our repository. Some modules implement only one or two functions, which makes no real sense with regards to the maintainance overhead that comes with it. 24 of the modules are using database access and 7 implement database interfaces. There is no clear boundary in some areas of the core and modules, this hinders the further development of the server. proposed actions: - integrate several modules that share responsibilities and not implement that much functionality on their own into one - autogeneration of module interface and database infrastructure code from the XML database scheme definitions - identify obselete module functions that can be done easily with pseudo-variables now. Remove these functions. - move certain core parts that do not need to be there into modules, making the core smaller and more flexible - identify core APIs which are used from modules, evaluate eventual generalisations to make the modules less dependent from the actual core implementation 4. Quality assurance The complexity of several modules, and the special dependencies that some of them carry increase the work to test the server and reproduce bugs. Without proper test coverage maintenance work is slowed down and flexibility is lost. All "easy" bugs that get caught automatically will not reach the user, increasing the quality. proposed actions: - extend available test suite to cover more of core and modules - continue maintenance of available infrastructure for build tests (build bot, debian package autobuilder), evaluate extensions to increase coverage All this work will be done as in the past in an open and accountable way on the developer list. For bigger changes a request for comment will be posted in advance on this list. If existing functionality is deprecated, then this RFC will be also posted to the user list. If some functionality is moved or changed than this will be documented in the wiki (porting docs) as usual. I understand some changes will be probably add work on your side, in porting, configuration adjustments and the like. But i think this is necessary to ensure the further development of the Kamailio project, and to be able to deal with new challenges that will come up in the future. Any input and help with this regards to this individual tasks is of course much appreciated. In the end the success of the project depends in the future (like it did in the past) on your support. Without your involvement in contributions, discussions and bug reports i would be only to a much lesser extend involved in this project. Thank you, Henning Westerholt -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

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sr-users September 2008