[OpenSIPS-Users] dispatcher not doing fail over itself

Bogdan-Andrei Iancu bogdan at opensips.org
Fri Sep 19 10:28:17 CEST 2014


The output show that only one destination was used by dispatcher (which 
was directly pushed into the message, so no pending destination in case 
of failover) - see the "273" AVP with value "1".

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 17.09.2014 18:06, Satish Patel wrote:
> Here you go
>
>
>
>
> /sbin/opensips[28000]: Sending call to ===> Freeswitch
> /sbin/opensips[28000]: INFO:avpops:ops_print_avp: p=0x7f9ed01dd420, 
> flags=0x0000
> /sbin/opensips[28000]: INFO:avpops:ops_print_avp: #011#011#011name=<274>
> /sbin/opensips[28000]: INFO:avpops:ops_print_avp: #011#011#011id=<10>
> /sbin/opensips[28000]: INFO:avpops:ops_print_avp: #011#011#011val_int=<0>
> /sbin/opensips[28000]: INFO:avpops:ops_print_avp: p=0x7f9ed01df208, 
> flags=0x0000
> /sbin/opensips[28000]: INFO:avpops:ops_print_avp: #011#011#011name=<273>
> /sbin/opensips[28000]: INFO:avpops:ops_print_avp: #011#011#011id=<9>
> /sbin/opensips[28000]: INFO:avpops:ops_print_avp: #011#011#011val_int=<1>
> /sbin/opensips[28000]: INFO:avpops:ops_print_avp: p=0x7f9ed01e6518, 
> flags=0x0002
> /sbin/opensips[28000]: INFO:avpops:ops_print_avp: #011#011#011name=<272>
> /sbin/opensips[28000]: INFO:avpops:ops_print_avp: #011#011#011id=<12>
> /sbin/opensips[28000]: INFO:avpops:ops_print_avp: 
> #011#011#011val_str=< / 0>
> /sbin/opensips[28000]: dispatcher: Attempting to dispatch call to 
> sip:182.xx.xx.xxx:5071
> /sbin/opensips[28005]: Inside dispatcher failure route
> /sbin/opensips[28005]: ds_dispatcher <null> <null> > <null>
> /sbin/opensips[28005]: 
> R-DISPATCHER-ROLLOVER:fU4SNHMvcNmnjW19gsSj0g..-S No more gateways in 
> route set
>
>
> On Wed, Sep 17, 2014 at 10:25 AM, Bogdan-Andrei Iancu 
> <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>     Hi,
>
>     As mentioned, place the avp_print() just after you did the
>     ds_select_dst(), before relaying to the first destination.
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>     OpenSIPS Founder and Developer
>     http://www.opensips-solutions.com
>
>     On 17.09.2014 14:09, Satish Patel wrote:
>
>         Confirmed  probing/inactive thing is working,
>
>         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.
>
>         Please advice if my script has any issue.
>
>         Sent from my iPhone
>
>         On Sep 17, 2014, at 3:30 AM, Bogdan-Andrei Iancu
>         <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>             Hi Satish,
>
>             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.
>
>             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.
>
>             Regards,
>
>             Bogdan-Andrei Iancu
>             OpenSIPS Founder and Developer
>             http://www.opensips-solutions.com
>
>             On 16.09.2014 21:02, Satish Patel wrote:
>
>                 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..
>
>                 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.
>
>
>

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