<table cellspacing="0" cellpadding="0" border="0" ><tr><td valign="top" style="font: inherit;">remove<br><br>--- On <b>Mon, 12/13/10, sr-users-request@lists.sip-router.org <i><sr-users-request@lists.sip-router.org></i></b> wrote:<br><blockquote style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"><br>From: sr-users-request@lists.sip-router.org <sr-users-request@lists.sip-router.org><br>Subject: sr-users Digest, Vol 67, Issue 33<br>To: sr-users@lists.sip-router.org<br>Date: Monday, December 13, 2010, 6:48 AM<br><br><div class="plainMail">Send sr-users mailing list submissions to<br> <a ymailto="mailto:sr-users@lists.sip-router.org" href="/mc/compose?to=sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br><br>To subscribe or unsubscribe via the World Wide Web, visit<br> <a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users"
target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>or, via email, send a message with subject or body 'help' to<br> <a ymailto="mailto:sr-users-request@lists.sip-router.org" href="/mc/compose?to=sr-users-request@lists.sip-router.org">sr-users-request@lists.sip-router.org</a><br><br>You can reach the person managing the list at<br> <a ymailto="mailto:sr-users-owner@lists.sip-router.org" href="/mc/compose?to=sr-users-owner@lists.sip-router.org">sr-users-owner@lists.sip-router.org</a><br><br>When replying, please edit your Subject line so it is more specific<br>than "Re: Contents of sr-users digest..."<br><br><br>Today's Topics:<br><br> 1. Re: Dialog module with 2 servers and 2 separate databases.<br> (Daniel-Constantin Mierla)<br> 2. First public release of sip:provider Community Edition<br>
(Andreas Granig)<br> 3. First public release of sip:provider Community Edition<br> (Andreas Granig)<br> 4. Re: Crash (michel freiha)<br><br><br>----------------------------------------------------------------------<br><br>Message: 1<br>Date: Mon, 13 Dec 2010 15:12:13 +0100<br>From: Daniel-Constantin Mierla <<a ymailto="mailto:miconda@gmail.com" href="/mc/compose?to=miconda@gmail.com">miconda@gmail.com</a>><br>Subject: Re: [SR-Users] Dialog module with 2 servers and 2 separate<br> databases.<br>To: "Pan B. Christensen" <<a ymailto="mailto:pan@ibidium.no" href="/mc/compose?to=pan@ibidium.no">pan@ibidium.no</a>><br>Cc: <a ymailto="mailto:sr-users@lists.sip-router.org" href="/mc/compose?to=sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>Message-ID: <<a ymailto="mailto:4D0629BD.8050501@gmail.com"
href="/mc/compose?to=4D0629BD.8050501@gmail.com">4D0629BD.8050501@gmail.com</a>><br>Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"<br><br>Hello,<br><br>one option you can do is to write to db immediately when a call is <br>active (see dialog module parameters) and do a query to the other server <br>database in addition to counting the local instance active calls.<br><br>Another one, different, is to use memcache for a communication system <br>between two or more instances.<br><br>Cheers,<br>Daniel<br><br>On 12/13/10 2:37 PM, Pan B. Christensen wrote:<br>> Thanks for your reply, Daniel.<br>> The purpose is to do busy forwarding without querying the client when <br>> the user already has >= X active calls. X will normally be 1 (call <br>> waiting inactive) or 2 (call waiting active). Advanced users may <br>> possibly set a higher value. Counting the number of calls on <br>> the server and doing busy forwarding
based on that rather than waiting <br>> for a "486 Busy here" from the client has several advantages.<br>> Currently, I've written code to do this with the dispatcher module, <br>> and it's working great with only one server. Here's a code snippet:<br>> $var(dlg_busy) = 0;<br>> get_profile_size("busy", "$avp(s:uid)", "$var(dlg_busy)");<br>> if ( $var(dlg_busy) >= $avp(s:busy_level) ) {<br>> if ($avp(s:cfb_status) == "on") {<br>> $rU = $avp(s:cfb_number);<br>> xlog("L_INFO", "-------------------- $avp(s:uid) has <br>> $var(dlg_busy) active calls. Treshold $avp(s:busy_level). Forwarding <br>> on busy to $rU --------------------\n");<br>>
route(10);<br>> }<br>> ...<br>> }<br>> Based on your reply, I guess one way to solve this would be to write <br>> the get_profile_size function in sqlops, query the two dialog <br>> databases and add the numbers. This would still require the customer <br>> to change their database design. Is there an easier or better way to <br>> do this?<br>> I also wote code to do busy forwarding if the client replies with 486 <br>> (do not disturb activated), 603 (call rejected) etc.<br>> This code works for normal busy forwarding if Polycom is set to 1 call <br>> per line key (default is 8). We'll then have to provision the <br>> $avp(s:busy_level) variable to the clients instead of handling it <br>> server-side. If a user now wants to change the setting, he'll have to <br>> reboot his phone after doing
so. Changing the <br>> reg.x.callsPerLineKey setting in the phone also limits the number of <br>> outgoing calls the user can make. We'll also have to make code for all <br>> the other hardphones the customer is planning to use plus make guides <br>> on how to change the setting for all kinds of softphones. We want to <br>> avoid all this.<br>> With kind regards,<br>> Pan<br>><br>> ----- Original Message -----<br>> *From:* Daniel-Constantin Mierla <mailto:<a ymailto="mailto:miconda@gmail.com" href="/mc/compose?to=miconda@gmail.com">miconda@gmail.com</a>><br>> *To:* Pan B. Christensen <mailto:<a ymailto="mailto:pan@ibidium.no" href="/mc/compose?to=pan@ibidium.no">pan@ibidium.no</a>><br>> *Cc:* <a ymailto="mailto:sr-users@lists.sip-router.org"
href="/mc/compose?to=sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>> <mailto:<a ymailto="mailto:sr-users@lists.sip-router.org" href="/mc/compose?to=sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a>><br>> *Sent:* Monday, December 13, 2010 12:26 PM<br>> *Subject:* Re: [SR-Users] Dialog module with 2 servers and 2<br>> separate databases.<br>><br>><br>><br>> On 12/10/10 2:17 PM, Pan B. Christensen wrote:<br>>> Hello,<br>>> My customer has the following database design.<br>>> Voip server 1 talks to SQL server 1.<br>>> Voip server 2 talks to SQL server 2.<br>>> Voip 1 and Voip 2 are load-balanced.<br>>>
Each SQL server has two databases. Database 1 contains<br>>> semi-static data like call forwarding properties for users and is<br>>> read-only. This is replicated from a third SQL server which the<br>>> web interface writes to. Database 2 is read/write, is not<br>>> replicated and contains data that is updated frequently like user<br>>> location and now dialog info.<br>>> Voip server 1 is not allowed to talk to SQL server 2 and vice versa.<br>>> I'm using forward() to send authenticated REGISTERs to the other<br>>> server so that it'll write this to RAM and its own SQL server.<br>>> Thus, both servers are aware of clients authenticated and<br>>>
registered by the other server.<br>>> How can I make both servers be aware of active calls on the other<br>>> server?<br>> what is the purpose?<br>><br>> Practically, it is not possible to track a call in two instances,<br>> because, unlike registration where is just a storage of mappings<br>> between contact and aor, call states of dialog module involve more<br>> processing logic, including timeouts and sending BYEs.<br>><br>> Cheers,<br>> Daniel<br>><br>> -- <br>> Daniel-Constantin Mierla<br>> Kamailio (OpenSER) Advanced Training<br>> Jan 24-26, 2011, Irvine, CA,
USA<br>> <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a><br>><br>> ------------------------------------------------------------------------<br>><br>> No virus found in this incoming message.<br>> Checked by AVG - www.avg.com<br>> Version: 8.5.449 / Virus Database: 271.1.1/3312 - Release Date:<br>> 12/12/10 19:34:00<br>><br>><br>> _______________________________________________<br>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>> <a ymailto="mailto:sr-users@lists.sip-router.org" href="/mc/compose?to=sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>> <a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users"
target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br><br>-- <br>Daniel-Constantin Mierla<br>Kamailio (OpenSER) Advanced Training<br>Jan 24-26, 2011, Irvine, CA, USA<br><a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a><br><br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL: <<a href="http://lists.sip-router.org/pipermail/sr-users/attachments/20101213/02fb715a/attachment-0001.htm" target="_blank">http://lists.sip-router.org/pipermail/sr-users/attachments/20101213/02fb715a/attachment-0001.htm</a>><br><br>------------------------------<br><br>Message: 2<br>Date: Mon, 13 Dec 2010 15:42:43 +0100<br>From: Andreas Granig <<a ymailto="mailto:agranig@sipwise.com" href="/mc/compose?to=agranig@sipwise.com">agranig@sipwise.com</a>><br>Subject: [SR-Users] First public release of sip:provider Community<br> Edition<br>To: <a
ymailto="mailto:sr-users@lists.sip-router.org" href="/mc/compose?to=sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>Message-ID: <<a ymailto="mailto:4D0630E3.7000204@sipwise.com" href="/mc/compose?to=4D0630E3.7000204@sipwise.com">4D0630E3.7000204@sipwise.com</a>><br>Content-Type: text/plain; charset="iso-8859-1"<br><br>Hi all,<br><br>We at Sipwise are excited to announce the first public release of the<br>sip:provider Community Edition (<a href="http://www.sipwise.com/products/spce/" target="_blank">http://www.sipwise.com/products/spce/</a>).<br>It is a fully open-source SIP based Class5 VoIP soft-switch, providing<br>every component an operator needs to offer VoIP services. It comes as a<br>communication platform leveraging the capabilities of Kamailio, SEMS and<br>Asterisk, complemented by our own open-sourced building blocks to<br>provide consistent and easy-to-use provisioning, billing and<br>configuration maintenance. The
different parts are carefully integrated<br>with each other to form a fully featured VoIP soft-switch.<br><br>The platform will make it much easier for new Kamailio users to get<br>started with VoIP, and will provide missing parts in the open-source<br>VoIP eco-system for more experienced users.<br><br>Please check<br><a href="http://www.sipwise.com/news/announcements/spce-first-release/" target="_blank">http://www.sipwise.com/news/announcements/spce-first-release/</a> for more<br>information on this release.<br><br>Have fun playing with it, we hope it's as useful to you as it is for us.<br>Andreas<br><br>-------------- next part --------------<br>A non-text attachment was scrubbed...<br>Name: signature.asc<br>Type: application/pgp-signature<br>Size: 900 bytes<br>Desc: OpenPGP digital signature<br>URL: <<a href="http://lists.sip-router.org/pipermail/sr-users/attachments/20101213/a31cdc10/attachment-0001.pgp"
target="_blank">http://lists.sip-router.org/pipermail/sr-users/attachments/20101213/a31cdc10/attachment-0001.pgp</a>><br><br>------------------------------<br><br>Message: 3<br>Date: Mon, 13 Dec 2010 15:45:36 +0100<br>From: Andreas Granig <<a ymailto="mailto:agranig@sipwise.com" href="/mc/compose?to=agranig@sipwise.com">agranig@sipwise.com</a>><br>Subject: [SR-Users] First public release of sip:provider Community<br> Edition<br>To: kamailio <<a ymailto="mailto:sr-users@lists.sip-router.org" href="/mc/compose?to=sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a>><br>Message-ID: <<a ymailto="mailto:4D063190.4000902@sipwise.com" href="/mc/compose?to=4D063190.4000902@sipwise.com">4D063190.4000902@sipwise.com</a>><br>Content-Type: text/plain; charset="iso-8859-1"<br><br>Hi all,<br><br>We at Sipwise are excited to announce the first public release of the<br>sip:provider Community Edition (<a
href="http://www.sipwise.com/products/spce/" target="_blank">http://www.sipwise.com/products/spce/</a>).<br>It is a fully open-source SIP based Class5 VoIP soft-switch, providing<br>every component an operator needs to offer VoIP services. It comes as a<br>communication platform leveraging the capabilities of Kamailio, SEMS and<br>Asterisk, complemented by our own open-sourced building blocks to<br>provide consistent and easy-to-use provisioning, billing and<br>configuration maintenance. The different parts are carefully integrated<br>with each other to form a fully featured VoIP soft-switch.<br><br>Please check<br><a href="http://www.sipwise.com/news/announcements/spce-first-release/" target="_blank">http://www.sipwise.com/news/announcements/spce-first-release/</a> for more<br>information on this release.<br><br>Have fun playing with it, we hope it's as useful to you as it is for us.<br>Andreas<br><br><br>-------------- next part --------------<br>A
non-text attachment was scrubbed...<br>Name: signature.asc<br>Type: application/pgp-signature<br>Size: 900 bytes<br>Desc: OpenPGP digital signature<br>URL: <<a href="http://lists.sip-router.org/pipermail/sr-users/attachments/20101213/e9873570/attachment-0001.pgp" target="_blank">http://lists.sip-router.org/pipermail/sr-users/attachments/20101213/e9873570/attachment-0001.pgp</a>><br><br>------------------------------<br><br>Message: 4<br>Date: Mon, 13 Dec 2010 16:48:02 +0200<br>From: michel freiha <<a ymailto="mailto:michofr@gmail.com" href="/mc/compose?to=michofr@gmail.com">michofr@gmail.com</a>><br>Subject: Re: [SR-Users] Crash<br>To: Daniel-Constantin Mierla <<a ymailto="mailto:miconda@gmail.com" href="/mc/compose?to=miconda@gmail.com">miconda@gmail.com</a>><br>Cc: <a ymailto="mailto:users@lists.kamailio.org" href="/mc/compose?to=users@lists.kamailio.org">users@lists.kamailio.org</a><br>Message-ID:<br>
<AANLkTi=5-JpFG3q=KQM-LK-dpKGYn03w1yYwLJpPfY=<a ymailto="mailto:N@mail.gmail.com" href="/mc/compose?to=N@mail.gmail.com">N@mail.gmail.com</a>><br>Content-Type: text/plain; charset="iso-8859-1"<br><br>Hello Daniel,<br><br>We are not facing any database problem even we are not using DNS anywhere in<br>Kamailio config...The only thing is that the debug level is 2 and syslog was<br>not configured asynchronous...Do you think if we configure it as<br>asynchronous, our issue will be solved?<br><br>Regards<br><br>On Mon, Dec 13, 2010 at 1:08 PM, Daniel-Constantin Mierla <<a ymailto="mailto:miconda@gmail.com" href="/mc/compose?to=miconda@gmail.com">miconda@gmail.com</a><br>> wrote:<br><br>> Hello,<br>><br>><br>> On 12/10/10 4:15 PM, michel freiha wrote:<br>><br>>> Hello Sir,<br>>><br>>> The crash issue has been successfully fixed after using GIT for the<br>>> version 3.1.0.<br>>><br>> thanks for
reporting back.<br>><br>><br>> Now we have another problem..>When the number of registered users exceeded<br>>> 2500 concurrent registered users, the kamailio stuck and each call will take<br>>> up to 1 minute to be established<br>>><br>>> Any comment on that?<br>>><br>> Do you have high debug level? If yes, is your syslog configured<br>> asynchronously?<br>><br>> Other than that, you can use benchmark module to spot which of your config<br>> actions takes so long to execute. Normally, such cases can happen when you<br>> have queries to slow database or dns servers.<br>><br>> Cheers,<br>> Daniel<br>><br>><br>><br>> --<br>> Daniel-Constantin Mierla<br>> Kamailio (OpenSER) Advanced Training<br>> Jan 24-26, 2011, Irvine, CA, USA<br>> <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a><br>><br>><br>-------------- next part
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