<div dir="ltr">Here you go<br><br><div><br><br><br>/sbin/opensips[28000]: Sending call to ===> Freeswitch<br>/sbin/opensips[28000]: INFO:avpops:ops_print_avp: p=0x7f9ed01dd420, flags=0x0000<br>/sbin/opensips[28000]: INFO:avpops:ops_print_avp: #011#011#011name=<274><br>/sbin/opensips[28000]: INFO:avpops:ops_print_avp: #011#011#011id=<10><br>/sbin/opensips[28000]: INFO:avpops:ops_print_avp: #011#011#011val_int=<0><br>/sbin/opensips[28000]: INFO:avpops:ops_print_avp: p=0x7f9ed01df208, flags=0x0000<br>/sbin/opensips[28000]: INFO:avpops:ops_print_avp: #011#011#011name=<273><br>/sbin/opensips[28000]: INFO:avpops:ops_print_avp: #011#011#011id=<9><br>/sbin/opensips[28000]: INFO:avpops:ops_print_avp: #011#011#011val_int=<1><br>/sbin/opensips[28000]: INFO:avpops:ops_print_avp: p=0x7f9ed01e6518, flags=0x0002<br>/sbin/opensips[28000]: INFO:avpops:ops_print_avp: #011#011#011name=<272><br>/sbin/opensips[28000]: INFO:avpops:ops_print_avp: #011#011#011id=<12><br>/sbin/opensips[28000]: INFO:avpops:ops_print_avp: #011#011#011val_str=< / 0><br>/sbin/opensips[28000]: dispatcher: Attempting to dispatch call to sip:182.xx.xx.xxx:5071<br>/sbin/opensips[28005]: Inside dispatcher failure route<br>/sbin/opensips[28005]: ds_dispatcher <null> <null> > <null><br>/sbin/opensips[28005]: R-DISPATCHER-ROLLOVER:fU4SNHMvcNmnjW19gsSj0g..-S No more gateways in route set<br><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Sep 17, 2014 at 10:25 AM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
As mentioned, place the avp_print() just after you did the ds_select_dst(), before relaying to the first destination.<span class="im HOEnZb"><br>
<br>
Regards,<br>
<br>
Bogdan-Andrei Iancu<br>
OpenSIPS Founder and Developer<br>
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.<u></u>com</a><br>
<br></span><div class="HOEnZb"><div class="h5">
On 17.09.2014 14:09, Satish Patel wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
ConfirmedĀ probing/inactive thing is working,<br>
<br>
Now problem is failover not working, ds_next_dst() not able to find next gateway and it says no more gateway available. If you check my script, i am calling failure route and inside it calling ds_next_dst() but you are saying do avp_print() on ds_select_dst() it won't print anything because after failure you are in failure route block.<br>
<br>
Please advice if my script has any issue.<br>
<br>
Sent from my iPhone<br>
<br>
On Sep 17, 2014, at 3:30 AM, Bogdan-Andrei Iancu <<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>> wrote:<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi Satish,<br>
<br>
Use the "opensipsctl fifo ds_list" to see the status of the gws in realtime . Be sure that once a gw is in non-active state (probing or inactive), it will not be used again for routing. Just take care of doing the ds_next_dst() in order to jump to the next available gw.<br>
<br>
If you do the avp_print() after ds_select_dst(), you can see how many other gw are prepared to used in case of failover.<br>
<br>
Regards,<br>
<br>
Bogdan-Andrei Iancu<br>
OpenSIPS Founder and Developer<br>
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.<u></u>com</a><br>
<br>
On 16.09.2014 21:02, Satish Patel wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
After doing couple of TEST look like its marking "Probing" for failed gateway but not auto failover to next gateway, i meant call get disconnect and i need to re-initiate call then all call goes to second active gateway..<br>
<br>
I believe it should first mark gateway "Probing" and then fall-back to second gateway automatically instead of call disconnect and i am getting 503 error code on my SIP Phone.<br>
</blockquote></blockquote></blockquote>
<br>
</div></div></blockquote></div><br></div>