Re: [SR-Users] dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (from asterisk call-id)

[TEG AMJG]

Hi again

There is something that i am starting to realize and i think is the source
of the problem. Since i am not sure, I think someone can clarify this to me.

As i said the problem rises when i do a 2 call leg conversation. In which,
since Asterisk is a B2BUA and creates another channel (which ends up being
another dialog), there is a point where Asterisk initiates the SIP flow.

Sip Client 1 ---> Kamailio --> Asterisk (dialog 1)
Asterisk -> Kamailio -> Sip Client 2 (dialog 2)

Now, since both Asterisk takes part of the destination addresses which the
dispatcher modules are managing at the moment, there is no way (from what i
know) that the ds_load_update() actually records that call load.

I am pretty sure that there is some way to manage this better, and since i
am new to Kamailio, I think you guys can help me out easily

Thank you very much for your time



2016-04-04 18:35 GMT-03:00 TEG AMJG :

> Hi Alberto, thanks for your reply
>
> 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
>
> route[TOASTERISK] {
> #!ifdef WITH_LOADBALANCE
> xlog("L_INFO","UBICAR: Forwarding non-registrar requests to
> Asterisk \n");
>  if(!ds_select_dst("0", "10")) {
> sl_send_reply("500", "Service Unavailable");
> xlog("L_ERROR","UBICAR: No destinations available for $rd
> \n");
> exit;
> }
> xlog("L_ERROR","UBICAR: AVP $avp(dsdstid)\n");
> xlog("L_INFO","UBICAR: Dispatching to Asterisk Box $du\n");
> t_set_fr(0,1);
> t_on_failure("MANAGE_FAILURE");
> route(RELAY);
> exit;
> #!endif
> }
>
> 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:
>
> modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404")
>
> and the command kamctl dispatcher dump shows them with flags=AP
>
> URI:: sip:192.168.10.191:5060 flags=AP priority=0
> attrs=duid=PBX2;my=pbx2;maxload=200
> URI:: sip:192.168.10.190:5060 flags=AP priority=0
> attrs=duid=PBX1;my=pbx1;maxload=200
>
> 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
>
> Again, thank you very much for your reply
>
>
> 2016-04-04 14:23 GMT-03:00 Alberto Sagredo 
> :
>
>> Hi
>>
>> Are you using ds_select_dst("1", "10"); somewhere ?
>>
>> 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.
>>
>> I have used dispatcher but not with 10 option on algo...
>>
>> BR
>>
>>
>>
>>
>> 2016-04-04 8:07 GMT+02:00 TEG AMJG :
>>
>>> Hi
>>>
>>> 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.
>>>
>>> - ERROR: dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find
>>> load for (call id generated by asterisk)
>>>
>>> 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
>>>
>>> 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
>>>
>>> Here are the blocks of code of the dispatcher parameters and where the
>>> function ds_load_update() is being used:
>>> #!ifdef WITH_LOADBALANCE
>>>
>>> #Parameters using Dispatcher from DB
>>> # --
>>> modparam("dispatcher", "db_url", DBURL)
>>> modparam("dispatcher", "table_name", "dispatcher")
>>> # --
>>>
>>> #Enabling failover mechanism
>>> # ---
>>> modparam("dispatcher", "flags", 2)
>>> # ---
>>>
>>> #Setting dispatcher parameters in AVP
>>> # ---
>>> modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
>>> modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
>>> modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
>>> modparam("dispatcher", "dstid_avp", "$avp(dsdstid)")
>>> # ---
>>>
>>> #Setting dispatcher result in PV variables
>>> # ---
>>> modparam("dispatcher", "attrs_pvname", "$var(attrs)")
>>> # ---
>>>
>>>
>>> #Enabling dispatcher algorithm according to call loads
>>> # 

Re: [SR-Users] dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (from asterisk call-id)

[Vladimer Gabunia]

hello.
how can i block SIP messages  by message content type. for exlamle : text/plain
tanks in advance!

From: sr-users [sr-users-boun...@lists.sip-router.org] on behalf of TEG AMJG 
[tega...@gmail.com]
Sent: Tuesday, April 05, 2016 1:35 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] dispatcher [dispatch.c:1416]: ds_load_remove(): cannot 
find load for (from asterisk call-id)

Hi Alberto, thanks for your reply

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

route[TOASTERISK] {
#!ifdef WITH_LOADBALANCE
xlog("L_INFO","UBICAR: Forwarding non-registrar requests to Asterisk 
\n");
 if(!ds_select_dst("0", "10")) {
sl_send_reply("500", "Service Unavailable");
xlog("L_ERROR","UBICAR: No destinations available for $rd \n");
exit;
}
xlog("L_ERROR","UBICAR: AVP $avp(dsdstid)\n");
xlog("L_INFO","UBICAR: Dispatching to Asterisk Box $du\n");
t_set_fr(0,1);
t_on_failure("MANAGE_FAILURE");
route(RELAY);
exit;
#!endif
}

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:

modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404")

and the command kamctl dispatcher dump shows them with flags=AP

URI:: sip:192.168.10.191:5060<http://192.168.10.191:5060> flags=AP priority=0 
attrs=duid=PBX2;my=pbx2;maxload=200
URI:: sip:192.168.10.190:5060<http://192.168.10.190:5060> flags=AP priority=0 
attrs=duid=PBX1;my=pbx1;maxload=200

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

Again, thank you very much for your reply


2016-04-04 14:23 GMT-03:00 Alberto Sagredo 
<alberto.sagr...@avanzada7.com<mailto:alberto.sagr...@avanzada7.com>>:
Hi

Are you using ds_select_dst("1", "10"); somewhere ?

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.

I have used dispatcher but not with 10 option on algo...

BR




2016-04-04 8:07 GMT+02:00 TEG AMJG 
<tega...@gmail.com<mailto:tega...@gmail.com>>:
Hi

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.

- ERROR: dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for 
(call id generated by asterisk)

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

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

Here are the blocks of code of the dispatcher parameters and where the function 
ds_load_update() is being used:
#!ifdef WITH_LOADBALANCE

#Parameters using Dispatcher from DB
# --
modparam("dispatcher", "db_url", DBURL)
modparam("dispatcher", "table_name", "dispatcher")
# --

#Enabling failover mechanism
# ---
modparam("dispatcher", "flags", 2)
# ---

#Setting dispatcher parameters in AVP
# ---
modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
modparam("dispatcher", "dstid_avp", "$avp(dsdstid)")
# ---

#Setting dispatcher result in PV variables
# ---
modparam("dispatcher", "attrs_pvname", "$var(attrs)")
# ---


#Enabling dispatcher algorithm according to call loads
# ---
modparam("dispatcher", "ds_hash_size",8)
modparam("dispatcher", "ds_hash_expire", 3600)
modparam("dispatcher", "ds_hash_initexpire", 60)
# ---

# Setting parameters for probing dispatc

Re: [SR-Users] dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (from asterisk call-id)

[TEG AMJG]

Hi Alberto, thanks for your reply

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

route[TOASTERISK] {
#!ifdef WITH_LOADBALANCE
xlog("L_INFO","UBICAR: Forwarding non-registrar requests to
Asterisk \n");
 if(!ds_select_dst("0", "10")) {
sl_send_reply("500", "Service Unavailable");
xlog("L_ERROR","UBICAR: No destinations available for $rd
\n");
exit;
}
xlog("L_ERROR","UBICAR: AVP $avp(dsdstid)\n");
xlog("L_INFO","UBICAR: Dispatching to Asterisk Box $du\n");
t_set_fr(0,1);
t_on_failure("MANAGE_FAILURE");
route(RELAY);
exit;
#!endif
}

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:

modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404")

and the command kamctl dispatcher dump shows them with flags=AP

URI:: sip:192.168.10.191:5060 flags=AP priority=0
attrs=duid=PBX2;my=pbx2;maxload=200
URI:: sip:192.168.10.190:5060 flags=AP priority=0
attrs=duid=PBX1;my=pbx1;maxload=200

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

Again, thank you very much for your reply


2016-04-04 14:23 GMT-03:00 Alberto Sagredo :

> Hi
>
> Are you using ds_select_dst("1", "10"); somewhere ?
>
> 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.
>
> I have used dispatcher but not with 10 option on algo...
>
> BR
>
>
>
>
> 2016-04-04 8:07 GMT+02:00 TEG AMJG :
>
>> Hi
>>
>> 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.
>>
>> - ERROR: dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load
>> for (call id generated by asterisk)
>>
>> 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
>>
>> 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
>>
>> Here are the blocks of code of the dispatcher parameters and where the
>> function ds_load_update() is being used:
>> #!ifdef WITH_LOADBALANCE
>>
>> #Parameters using Dispatcher from DB
>> # --
>> modparam("dispatcher", "db_url", DBURL)
>> modparam("dispatcher", "table_name", "dispatcher")
>> # --
>>
>> #Enabling failover mechanism
>> # ---
>> modparam("dispatcher", "flags", 2)
>> # ---
>>
>> #Setting dispatcher parameters in AVP
>> # ---
>> modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
>> modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
>> modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
>> modparam("dispatcher", "dstid_avp", "$avp(dsdstid)")
>> # ---
>>
>> #Setting dispatcher result in PV variables
>> # ---
>> modparam("dispatcher", "attrs_pvname", "$var(attrs)")
>> # ---
>>
>>
>> #Enabling dispatcher algorithm according to call loads
>> # ---
>> modparam("dispatcher", "ds_hash_size",8)
>> modparam("dispatcher", "ds_hash_expire", 3600)
>> modparam("dispatcher", "ds_hash_initexpire", 60)
>> # ---
>>
>> # Setting parameters for probing dispatcher destinations
>> # ---
>> modparam("dispatcher", "ds_ping_interval", 20)
>> modparam("dispatcher", "ds_ping_from", "sip:kamailio1@")
>> modparam("dispatcher", "ds_probing_mode", 1)
>> modparam("dispatcher", "ds_probing_threshhold", 2)
>> modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404")
>> # ---
>>
>> ...
>> route[WITHINDLG] {
>>  if (has_totag()) {
>>   # sequential request withing a dialog should
>>   # take the path determined by record-routing
>>
>> #!ifdef WITH_LOADBALANCE
>> if(is_method("BYE|CANCEL")){
>> xlog("L_ERROR","UBICAR BYE ds_load_update  requesturi=$ru
>> from=$fu method=$rm callid=$ci to=$tu reply=$mt rrcode=$rr rcode=$rs\n");
>> ds_load_update();
>> }
>> #!endif
>> ...
>> }
>>
>> ...
>>
>> onreply_route[MANAGE_REPLY] {
>> #!ifdef WITH_LOADBALANCE
>> if(is_method("INVITE")) {
>> 

Re: [SR-Users] dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (from asterisk call-id)

[Alberto Sagredo]

Hi

Are you using ds_select_dst("1", "10"); somewhere ?

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.

I have used dispatcher but not with 10 option on algo...

BR




2016-04-04 8:07 GMT+02:00 TEG AMJG :

> Hi
>
> 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.
>
> - ERROR: dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load
> for (call id generated by asterisk)
>
> 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
>
> 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
>
> Here are the blocks of code of the dispatcher parameters and where the
> function ds_load_update() is being used:
> #!ifdef WITH_LOADBALANCE
>
> #Parameters using Dispatcher from DB
> # --
> modparam("dispatcher", "db_url", DBURL)
> modparam("dispatcher", "table_name", "dispatcher")
> # --
>
> #Enabling failover mechanism
> # ---
> modparam("dispatcher", "flags", 2)
> # ---
>
> #Setting dispatcher parameters in AVP
> # ---
> modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
> modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
> modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
> modparam("dispatcher", "dstid_avp", "$avp(dsdstid)")
> # ---
>
> #Setting dispatcher result in PV variables
> # ---
> modparam("dispatcher", "attrs_pvname", "$var(attrs)")
> # ---
>
>
> #Enabling dispatcher algorithm according to call loads
> # ---
> modparam("dispatcher", "ds_hash_size",8)
> modparam("dispatcher", "ds_hash_expire", 3600)
> modparam("dispatcher", "ds_hash_initexpire", 60)
> # ---
>
> # Setting parameters for probing dispatcher destinations
> # ---
> modparam("dispatcher", "ds_ping_interval", 20)
> modparam("dispatcher", "ds_ping_from", "sip:kamailio1@")
> modparam("dispatcher", "ds_probing_mode", 1)
> modparam("dispatcher", "ds_probing_threshhold", 2)
> modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404")
> # ---
>
> ...
> route[WITHINDLG] {
>  if (has_totag()) {
>   # sequential request withing a dialog should
>   # take the path determined by record-routing
>
> #!ifdef WITH_LOADBALANCE
> if(is_method("BYE|CANCEL")){
> xlog("L_ERROR","UBICAR BYE ds_load_update  requesturi=$ru
> from=$fu method=$rm callid=$ci to=$tu reply=$mt rrcode=$rr rcode=$rs\n");
> ds_load_update();
> }
> #!endif
> ...
> }
>
> ...
>
> onreply_route[MANAGE_REPLY] {
> #!ifdef WITH_LOADBALANCE
> if(is_method("INVITE")) {
> if(status=~"2[0-9][0-9]") {
> xlog("L_ERROR","UBICAR setting ds_load_update
>  requesturi=$ru from=$fu method=$rm callid=$ci to=$tu reply=$mt rrcode=$rr
> rcode=$rs");
> ds_load_update();
> }else if(status=~"[3-7][0-9][0-9]") {
> ds_load_unset();
> }
> }
> #!endif
>
> if(status=~"[12][0-9][0-9]"){
> route(NATMANAGE);
>  }
> }
>
>
> This is what i am getting from logs after the 2 call legs conversation
> completes:
>
> kam1 /usr/sbin/kamailio[4082]: ERROR: 

[SR-Users] dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (from asterisk call-id)

[TEG AMJG]

Hi

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.

- ERROR: dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load
for (call id generated by asterisk)

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

>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

Here are the blocks of code of the dispatcher parameters and where the
function ds_load_update() is being used:
#!ifdef WITH_LOADBALANCE

#Parameters using Dispatcher from DB
# --
modparam("dispatcher", "db_url", DBURL)
modparam("dispatcher", "table_name", "dispatcher")
# --

#Enabling failover mechanism
# ---
modparam("dispatcher", "flags", 2)
# ---

#Setting dispatcher parameters in AVP
# ---
modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
modparam("dispatcher", "dstid_avp", "$avp(dsdstid)")
# ---

#Setting dispatcher result in PV variables
# ---
modparam("dispatcher", "attrs_pvname", "$var(attrs)")
# ---


#Enabling dispatcher algorithm according to call loads
# ---
modparam("dispatcher", "ds_hash_size",8)
modparam("dispatcher", "ds_hash_expire", 3600)
modparam("dispatcher", "ds_hash_initexpire", 60)
# ---

# Setting parameters for probing dispatcher destinations
# ---
modparam("dispatcher", "ds_ping_interval", 20)
modparam("dispatcher", "ds_ping_from", "sip:kamailio1@")
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_probing_threshhold", 2)
modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404")
# ---

...
route[WITHINDLG] {
 if (has_totag()) {
  # sequential request withing a dialog should
  # take the path determined by record-routing

#!ifdef WITH_LOADBALANCE
if(is_method("BYE|CANCEL")){
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");
ds_load_update();
}
#!endif
...
}

...

onreply_route[MANAGE_REPLY] {
#!ifdef WITH_LOADBALANCE
if(is_method("INVITE")) {
if(status=~"2[0-9][0-9]") {
xlog("L_ERROR","UBICAR setting ds_load_update
 requesturi=$ru from=$fu method=$rm callid=$ci to=$tu reply=$mt rrcode=$rr
rcode=$rs");
ds_load_update();
}else if(status=~"[3-7][0-9][0-9]") {
ds_load_unset();
}
}
#!endif

if(status=~"[12][0-9][0-9]"){
route(NATMANAGE);
 }
}


This is what i am getting from logs after the 2 call legs conversation
completes:

kam1 /usr/sbin/kamailio[4082]: ERROR: