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<div style="direction: ltr;font-family: Tahoma;color: #000000;font-size: 10pt;">hello.
<div>how can i block SIP messages by message content type. for exlamle : text/plain</div>
<div>tanks in advance!<br>
<div style="font-family: Times New Roman; color: #000000; font-size: 16px">
<hr tabindex="-1">
<div id="divRpF849399" style="direction: ltr;"><font face="Tahoma" size="2" color="#000000"><b>From:</b> sr-users [sr-users-bounces@lists.sip-router.org] on behalf of TEG AMJG [tegamjg@gmail.com]<br>
<b>Sent:</b> Tuesday, April 05, 2016 1:35 AM<br>
<b>To:</b> Kamailio (SER) - Users Mailing List<br>
<b>Subject:</b> Re: [SR-Users] dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (from asterisk call-id)<br>
</font><br>
</div>
<div></div>
<div>
<div dir="ltr">Hi Alberto, thanks for your reply
<div><br>
</div>
<div>yeah, i am using ds_select_dst("0","10") in this case because i use setid=0 in the dispatcher table as you can see in this route block</div>
<div><br>
</div>
<div>
<div>route[TOASTERISK] {</div>
<div>#!ifdef WITH_LOADBALANCE</div>
<div> xlog("L_INFO","UBICAR: Forwarding non-registrar requests to Asterisk \n");</div>
<div> if(!ds_select_dst("0", "10")) {</div>
<div> sl_send_reply("500", "Service Unavailable");</div>
<div> xlog("L_ERROR","UBICAR: No destinations available for $rd \n");</div>
<div> exit;</div>
<div> }</div>
<div> xlog("L_ERROR","UBICAR: AVP $avp(dsdstid)\n");</div>
<div> xlog("L_INFO","UBICAR: Dispatching to Asterisk Box $du\n");</div>
<div> t_set_fr(0,10000);</div>
<div> t_on_failure("MANAGE_FAILURE");</div>
<div> route(RELAY);</div>
<div> exit;</div>
<div>#!endif</div>
<div>}</div>
</div>
<div><br>
</div>
<div>Also both Asterisk are responding to ping probes done by Kamailio, they are actually responding 404 but are still kept alive thanks to this param:</div>
<div>
<div><br>
</div>
<div>modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404")</div>
</div>
<div><br>
</div>
<div>and the command kamctl dispatcher dump shows them with flags=AP</div>
<div><br>
</div>
<div>
<div>URI:: sip:<a href="http://192.168.10.191:5060" target="_blank">192.168.10.191:5060</a> flags=AP priority=0 attrs=duid=PBX2;my=pbx2;maxload=200</div>
<div>URI:: sip:<a href="http://192.168.10.190:5060" target="_blank">192.168.10.190:5060</a> flags=AP priority=0 attrs=duid=PBX1;my=pbx1;maxload=200</div>
</div>
<div><br>
</div>
<div>Also, i would like to know if there is a way to print some kind of table to see how kamailio are keeping these call loads recorded. I have been looking for it but i cannot find any way of doing it</div>
<div><br>
</div>
<div>Again, thank you very much for your reply </div>
<div><br>
</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">2016-04-04 14:23 GMT-03:00 Alberto Sagredo <span dir="ltr">
<<a href="mailto:alberto.sagredo@avanzada7.com" target="_blank">alberto.sagredo@avanzada7.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex; border-left:1px #ccc solid; padding-left:1ex">
<div dir="ltr">Hi
<div><br>
</div>
<div>Are you using <span style="color:rgb(0,0,0); background-color:rgb(254,254,238)">ds_select_dst("1", "10"); </span>somewhere ?</div>
<div><br>
</div>
<div>You have to check Asterisk is responding to OPTIONS as dispatcher considered UP when 200 OK from Asterisk or SIP Gateway is received.. and not if not received.</div>
<div><br>
</div>
<div>I have used dispatcher but not with 10 option on algo...</div>
<div><br>
</div>
<div>BR</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">
<div>
<div class="h5">2016-04-04 8:07 GMT+02:00 TEG AMJG <span dir="ltr"><<a href="mailto:tegamjg@gmail.com" target="_blank">tegamjg@gmail.com</a>></span>:<br>
</div>
</div>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex; border-left:1px #ccc solid; padding-left:1ex">
<div>
<div class="h5">
<div dir="ltr">
<div>Hi</div>
<div><br>
</div>
<div>I am new to Kamailio and i am just trying to setup a kamailio (v4.2.3) dispatcher to distribute calls to 2 asterisk using the call load distribution algorithm (alg 10). I started to test and see how all it is working and in the logs appears that the call
load record it cannot be found.</div>
<div><br>
</div>
<div>
<div>- ERROR: dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (call id generated by asterisk)</div>
</div>
<div><br>
</div>
<div>I am getting this error trying a 2 call leg conversation when asterisk is creating the BYE message destinated to the callee after receiving a BYE message from the caller, and the BYE message by Asterisk reach Kamailio to then apply the ds_load_update
function</div>
<div><br>
</div>
<div>From what i understand about ds_load_update it is that it sets the load to an specific destination if Kamailio receives a 2xx reply of an INVITE message, and removes it if Kamailio receives a BYE message based from the Call-id received in that message,
where it is actually failing in my implementation at the moment</div>
<div><br>
</div>
<div>Here are the blocks of code of the dispatcher parameters and where the function ds_load_update() is being used:</div>
<div>
<div>#!ifdef WITH_LOADBALANCE</div>
<div><br>
</div>
<div>#Parameters using Dispatcher from DB</div>
<div># ----------</div>
<div>modparam("dispatcher", "db_url", DBURL)</div>
<div>modparam("dispatcher", "table_name", "dispatcher")</div>
<div># ----------</div>
<div><br>
</div>
<div>#Enabling failover mechanism</div>
<div># -----------</div>
<div>modparam("dispatcher", "flags", 2)</div>
<div># -----------</div>
<div><br>
</div>
<div>#Setting dispatcher parameters in AVP</div>
<div># -----------</div>
<div>modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")</div>
<div>modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")</div>
<div>modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")</div>
<div>modparam("dispatcher", "dstid_avp", "$avp(dsdstid)")</div>
<div># -----------</div>
<div><br>
</div>
<div>#Setting dispatcher result in PV variables</div>
<div># -----------</div>
<div>modparam("dispatcher", "attrs_pvname", "$var(attrs)")</div>
<div># -----------</div>
<div><br>
</div>
<div><br>
</div>
<div>#Enabling dispatcher algorithm according to call loads</div>
<div># -----------</div>
<div>modparam("dispatcher", "ds_hash_size",8)</div>
<div>modparam("dispatcher", "ds_hash_expire", 3600)</div>
<div>modparam("dispatcher", "ds_hash_initexpire", 60)</div>
<div># -----------</div>
<div><br>
</div>
<div># Setting parameters for probing dispatcher destinations</div>
<div># -----------</div>
<div>modparam("dispatcher", "ds_ping_interval", 20)</div>
<div>modparam("dispatcher", "ds_ping_from", "sip:kamailio1@<Kamailio IP>")</div>
<div>modparam("dispatcher", "ds_probing_mode", 1)</div>
<div>modparam("dispatcher", "ds_probing_threshhold", 2)</div>
<div>modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404")</div>
<div># -----------</div>
</div>
<div><br>
</div>
<div>...</div>
<div>
<div>route[WITHINDLG] {</div>
<div> if (has_totag()) {</div>
<div> # sequential request withing a dialog should</div>
<div> # take the path determined by record-routing</div>
<div><br>
</div>
<div>#!ifdef WITH_LOADBALANCE</div>
<div> if(is_method("BYE|CANCEL")){</div>
<div> xlog("L_ERROR","UBICAR BYE ds_load_update requesturi=$ru from=$fu method=$rm callid=$ci to=$tu reply=$mt rrcode=$rr rcode=$rs\n");</div>
<div> ds_load_update();</div>
<div> }</div>
<div>#!endif</div>
</div>
<div>...</div>
<div>}</div>
<div><br>
</div>
<div>...</div>
<div><br>
</div>
<div>
<div>onreply_route[MANAGE_REPLY] {</div>
<div>#!ifdef WITH_LOADBALANCE</div>
<div>if(is_method("INVITE")) {</div>
<div> if(status=~"2[0-9][0-9]") {</div>
<div> xlog("L_ERROR","UBICAR setting ds_load_update requesturi=$ru from=$fu method=$rm callid=$ci to=$tu reply=$mt rrcode=$rr rcode=$rs");</div>
<div> ds_load_update();</div>
<div> }else if(status=~"[3-7][0-9][0-9]") {</div>
<div> ds_load_unset();</div>
<div> }</div>
<div>}</div>
<div>#!endif</div>
<div><br>
</div>
<div>if(status=~"[12][0-9][0-9]"){</div>
<div> route(NATMANAGE);</div>
</div>
<div> }</div>
<div>}</div>
<div><br>
</div>
<div><br>
</div>
<div>This is what i am getting from logs after the 2 call legs conversation completes:</div>
<div><br>
</div>
<div>
<div>kam1 /usr/sbin/kamailio[4082]: ERROR: <script>: UBICAR: FROMASTERISK <a href="mailto:70175143146f92975a5bff8876435911@192.168.10.194" target="_blank">
70175143146f92975a5bff8876435911@192.168.10.194</a><br>
</div>
<div><br>
</div>
<div>kam1 /usr/sbin/kamailio[4082]: ERROR: <script>: UBICAR setting ds_load_update requesturi=<null> from=<a href="mailto:sip%3A103@192.168.10.194" target="_blank">sip:103@192.168.10.194</a> method=INVITE callid=<a href="mailto:28d0f6e71625c9e87baec07f46167863@192.168.10.194" target="_blank">28d0f6e71625c9e87baec07f46167863@192.168.10.194</a>
to=<a href="mailto:sip%3A104@192.168.10.194" target="_blank">sip:104@192.168.10.194</a> reply=2 rrcode=OK rcode=200</div>
<div><br>
</div>
<div>kam1 /usr/sbin/kamailio[4083]: ERROR: <script>: UBICAR setting ds_load_update requesturi=<null> from=<a href="http://sip:103@192.168.10.194:5060" target="_blank">sip:103@192.168.10.194:5060</a> method=INVITE callid=<a href="mailto:1-3692@192.168.20.2" target="_blank">1-3692@192.168.20.2</a>
to=<a href="http://sip:104@192.168.10.194:5060" target="_blank">sip:104@192.168.10.194:5060</a> reply=2 rrcode=OK rcode=200</div>
<div><br>
</div>
<div>kam1 /usr/sbin/kamailio[4084]: ERROR: <script>: UBICAR BYE ds_load_update requesturi=<a href="http://sip:104@192.168.10.190:5060" target="_blank">sip:104@192.168.10.190:5060</a> from=<a href="http://sip:103@192.168.30.1:5060" target="_blank">sip:103@192.168.30.1:5060</a>
method=BYE callid=<a href="mailto:1-3692@192.168.20.2" target="_blank">1-3692@192.168.20.2</a> to=<a href="http://sip:104@192.168.30.1:5060" target="_blank">sip:104@192.168.30.1:5060</a> reply=1 rrcode=<null> rcode=<null></div>
<div><br>
</div>
<div>kam1 /usr/sbin/kamailio[4085]: ERROR: <script>: UBICAR BYE ds_load_update requesturi=sip:104@192.168.30.2:22523;transport=UDP from=<a href="mailto:sip%3A103@192.168.10.194" target="_blank">sip:103@192.168.10.194</a> method=BYE callid=<a href="mailto:28d0f6e71625c9e87baec07f46167863@192.168.10.194" target="_blank">28d0f6e71625c9e87baec07f46167863@192.168.10.194</a>
to=<a href="mailto:sip%3A104@192.168.10.194" target="_blank">sip:104@192.168.10.194</a> reply=1 rrcode=<null> rcode=<null></div>
<div><br>
</div>
<div>kam1 /usr/sbin/kamailio[4085]: ERROR: dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (<a href="mailto:28d0f6e71625c9e87baec07f46167863@192.168.10.194" target="_blank">28d0f6e71625c9e87baec07f46167863@192.168.10.194</a>)</div>
</div>
<div><br>
</div>
<div>Kamailio IP: 192.168.10.194</div>
<div><br>
</div>
<div>Asterisk IP:</div>
<div><br>
</div>
<div>
<div>+----+-------+-------------------------+-------+----------+-------------------------------+-------------+</div>
<div>| id | setid | destination | flags | priority | attrs | description |</div>
<div>+----+-------+-------------------------+-------+----------+-------------------------------+-------------+</div>
<div>| 1 | 0 | sip:<a href="http://192.168.10.190:5060" target="_blank">192.168.10.190:5060</a> | 2 | 0 | duid=PBX1;my=pbx1;maxload=200 | pbx1 |</div>
<div>| 2 | 0 | sip:<a href="http://192.168.10.191:5060" target="_blank">192.168.10.191:5060</a> | 2 | 0 | duid=PBX2;my=pbx2;maxload=200 | pbx2 |</div>
<div>+----+-------+-------------------------+-------+----------+-------------------------------+-------------+</div>
</div>
<div><br>
</div>
<div><br>
</div>
</div>
<br>
</div>
</div>
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</blockquote>
</div>
<br>
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<br>
_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
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</blockquote>
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</div>
-- <br>
<div class="gmail_signature">
<div dir="ltr">-<br>
Saludos a todos<br>
</div>
</div>
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