ot break the first
existing call-leg still known to the kamailio registrar.
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for a specific ip or for specific situations after it
was engaged.
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7) it does not know to some 4XX default.
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Hi Mustafa
> SIP/2.0 607.
I didn't check the RFC. But doesnt a reply need to be in the format
SIP/2.0 XXX AA
Is the text missing after the numerical error code?
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ith limited
memory which struggle with a long record-route list.
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k_number){s.substr,1,0});
}
}
And of course I have a corresponding TRANSLATE_TO_LOCAL to display numbers to
the customer in a format they are used to in Switzerland.
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ur SBC and
interconnection would be sent to the customer as the 2nd invocation of
rtpengine would be ignored.
I hope this makes sense.
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ot stuck when trying to nicely tell rtpengine to stop listening
for callID-extraID on a CANCEL as I have not yet found a way to get
hold of that extraID.
rtpengine will eventually time out, but I try to tidy up correctly.
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every reply
status >= 300 is considered a failure in unconfirmed dialog state. Not
sure if this is true for the 422 status, this does not terminate a call
but causes the caller to re-issue the invite.
=> More testing ongoing.
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ag causes a
race condition in handling the dialog variables.
I see there are two attempts by the dialog module to write a CDR with
the same callID. After processing the 422 and when the call ends while
processing BYE.
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_var after loose_route() it is still
null.
Any idea what the cause could be?
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ssible?
I would like to avoid using the topos module as I issues with this
module completely failing in certain situations.
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is already handling that
callid?
if(rtpengine_query_v("j", "$var(dummy)")) {
rtpengine_manage();
}
basically works, but generates a couple of ugly error messages in the
log as rtpengine_query_v does not find the call.
Is there a nicer way?
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irst one picks up the call. Is
rtpengine stuck with IPv6 or will it listen to both protocols because
it processed two offer for the same callid?
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'0',
`partition` int NOT NULL DEFAULT '0',
PRIMARY KEY (`id`),
UNIQUE KEY `ruid_idx` (`ruid`),
KEY `account_contact_idx` (`username`,`domain`,`contact`),
KEY `expires_idx` (`expires`),
KEY `connection_idx` (`server_id`,`connection_id`),
KEY `tcpcon_idx` (`c
append_branch works by adding at the end of the index and never
inserting a branch in between.
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Zur
7:27:45 dev-cpereg01 kamailio[2708456]: ERROR: usrloc [urecord.c:419]:
wb_timer(): inserting contact into database failed (aor: somedudetp0216)
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into a branch route which is specific to
a branch route like the user_agent associated with the registration to
which the call is sent?
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_branch_idx]) = $null;
}
}
I don't get this right! Call is still sent to that contact.
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Zurlin
t;dmq_usrloc", "enable", 1)
# Do a full sync on start
modparam("dmq_usrloc", "sync", 1)
I assume, when kamailio is being restartet there are some expired entries still
present in the database which kamailio tries to insert on a new registration.
This leads to th
> }
>
> etc.
Thank you. So there is no 'direct' way I missed. Ok, I'll find a way
around ;-)
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_
ith variable injections via SQL. Immagine
some use sets a password ";var=value" this would lead to this var being
overwritten I guess.
We are moving towards storing ha1 hashed passwords, so that would solve my
issue I guess.
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t;e are null
Am I right to assume the issue being caused by the comma? Is there a
way to allow a comma?
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terminated?
With 'return' or with 'exit'?
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wrong node.
So is this a bug? Shall I report an issue on github?
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) = 3
}
=> Still timing out after 5 seconds, same issue.
Ping Olle and Alex. Any idea how to fix this issue?
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on restarting kamailio, the dialog
suddenly is back and after timing out again, a CDR is then written with
a way too long duration.
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denly being back?
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CH-4133 Pratteln
ng if that also occurs on 5.6
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so. Usually the crashes occur
after restarting. At the moment all 4 dev registrars have restarted
without crashing in the first 2 or so minute. So this somehow looks
promising!
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09:45:11 dev-cpereg01 kamailio[11905]: ALERT: [main.c:779]:
handle_sigs(): core was generated
Could anyone point me the correct direction on how to fix this issue?
Am I the only one seeing this issue?
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rg/docs/modules/stable/modules/rtpengine.html
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C
b0 tl->next=0x7f9618e11198 tl->prev=0x7f9617e20548
Sep 22 07:32:29 dev-core02 kamailio[915066]: CRITICAL: dialog [dlg_dmq.c:314]:
dlg_dmq_handle_msg(): Unable to insert dlg timer 0x7f9617d96750 [1598:12168]
Are the numbers in brackets [hash_id:hash_entry] ?
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fix for opensips
to kamailio.
Indeed, I see this message occur occasionally after managing the 200 OK
reply and routing the ACK.
Does anyone know, if this is still a open bug with Kamailio 5.5? Or
what else could be the cause of this message?
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e1, but NOT on Node2
What I fear happens next: They dialog_vars remaining on the Database of
Node 2 accumulate more and more, making the database slower. When a
call happens to re use the same hash id and entry, things start going
really badly.
Could I be right?
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isting TCP connection.
Do I have to set $fs? But how do I set this only for a specific branch?
Assume I have multiple contacts, some using TCP other UDP or TLS.
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only add the ones having a socket set.
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d, all 4 get traffic.
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CH-4133 Pr
1?
And, what happens if I set:
$branch(uri) = $null
will $(branch(uri)[0]) become $branch(uri) and eveything shift one
index down?
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27; loop did not output the contact registered on the other
registrar.
So WHY is there still the INVITE being sent to the fourth, non local contact?
=> Confused!
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}
Result:
* Call is still branched to CPE
* $(branch(uri)[$T_branch_idx]) is null on all subsequent branches after the
first one.
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in case of UDP and NAT via correct know ip for ALG
port-forwarding).
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other TSP and Carriers, I would have to check, at
the moment, they are all connected via a commercial vendor SBC so if
that SBC can handle 305 on Invites (it can in register) that would work.
But one of our goals is to eventually also get rid of that SBC
ly have a stable config
which will last for several months, but at the moment this is the
situation.
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Hi Alex
> If profile membership is synchronised, does that mean it is possible
> to get a count of dialogs in a given profile on any node?
Correct. I'm using this to implement a channel limit per trunk or per
customer on two dialog DMQ synced nodes.
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CH-4133 PrattelnFax +41
x27;3891401256-2097511021'
acc [acc_cdr.c:229]: db_write_cdr(): fallback to dlg_only search
because of message doesn't exist.
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> Kamcmd complains the table is not found. What would be the correct
> syntax to provide the table name?
Sigh, once more, answering myself: db_mode=3 only!
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ul.db_contacts s:database.location
Kamcmd complains the table is not found. What would be the correct
syntax to provide the table name?
Attempted with Kamailio 5.5 and 5.6
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-cpereg01 kamailio[36143]: INFO: [main.c:799]:
handle_sigs(): terminating due to SIGCHLD
Has there been a fundamental change in TLS? Or did I miss something
important to do on a 5.5 => 5.6 upgrade?
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se.
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us code 488 to allow fall-back on a failed re-negotiation.
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ng.
But now I had a look on core2 and this is where the call is expiring.
So I guess the refreshing of the session timer via AVP is not DMQ
synced to the other node? Should I replace the AVP by a DLG var?
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turn is passed through
rtpengine_manage() the existing RTP stream is deleted from rtpengine
instead of being left running with the previously agreed codec?
Has anyone already figured out a list of what reply codes should be
exempt from calling rtpengine_manage()?
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orphan database
entry.
I would like to digg deeper into this. Is there a way to log more
details from the dialogue module like when it triggers dialogue expiry?
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n-Expires){s.int});
}
On an INVITE containing those header:
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: 100rel,timer
Expires: 330
Now Kamailio replies:
SIP/2.0 422 Session Timer Too Small
Min-SE: 300
What am I doing wrong?
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g", 6)
But when reading the docs to the dialog modules, I understood that the dlg_flag
is purely optional and does not need to be set when using dlg_manage as I do.
Maybe it's not optional when using dialog AND sst together?
Also I wonder why no BYE is being sent, when the dialogue times
MD htable.reload spithash
$KAMCMD mtree.reload rules
$KAMCMD mtree.reload prefix
$KAMCMD mtree.reload antispoof
Could this cause kamailio to forget about running dialogues? That could
also explain, why I keep seeing the dialogue tables filling up with
stale dialogues over time.
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ilio not to find the corresponding dialogue on
this re-invite?
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ng the call and if so, call
rtpengine_manage() to keep backhauling traffic on the Re-Invites
refrreshing the session.
Unfortunately, I found no way to check if rtpengine_query finds the
call. Did I miss something?
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-
I guess I figured it out...
don't use via-branch=auto, but via-branch=auto-next then it works by
magic :-)
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inbound via, not the one generated on the branched
call.
How can this be accomplished with kamailio 5.4?
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unicate
via SIP IPv6.
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var(i) = $var(i) + 1;
}
jansson_set("array","result","$var(aor_a)","$var(json_reply)");
sql_result_free("sql_res");
} else {
jansson_set
ost}));
But then uac complains:
replace_uri(): Already called uac_replace for this dialog
Why isn't just the most from version kept and used?
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getting full route set being passed on.
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CH
headers, but failed.
Is this a known limitation of using topos?
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# True
route(RELAY);
} else {
# discard ACK not matching transaction
exit;
}
}
Ack is routed.
Any explanation why topos causes t_check_trans() to return false on ACK
and PRACK?
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nfigured correctly for
> this expectation?
>
> https://kamailio.org/docs/modules/5.6.x/modules/dialog.html#dialog.p.update_period
>
> https://kamailio.org/docs/modules/5.6.x/modules/dialog.html#dialog.p.db_mode
I will give it another try :-)
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t
entries kept accumulating with each restart in the 'dialog' and
'dialog_vars' tables, never expiring even the next day, when the dialog
timeout we set to 12 hours should have made sure they got expired.
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th the authentication
username (equal to the AOR in our case) of the customer to which the
dialogue belongs to.
As I kept running into DMQ sync issues with storing dialogue information
in database, I switched off database support for dialogue.
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tcp write queue overflows up at shy over 1MB
reply no matter what I change.
Any ideas what else to toggle?
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Master-Master registrar scenario, the INVITE towards a CPE might
originate from the IP of the registrar that CPE did not register to
and therefore the SIP ALG not knowing about and never make it through
NAT.
PS: Still looking for ideas, how this could be solved.
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Hi
> Try quoting the pseudovariable?
>
>t_on_branch("$avp(broute_trigger)");
Does this make he variable more 'constant'? :-)
I already built a switch/case contruct around it on a
REARM_B_TRIGGER route which I now call in every failure route.
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not a situation which occurs often so
having the CANCEL to time out and only then removing the dialogue is
probably acceptable.
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t_on_branch($avp(broute_trigger));
route(RELAY);
exit;
}
But no...
function t_on_branch: parameter 1 is not constant
Any 'tricks'?
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is routed after the first path failed
(marked with RIGHT HERE!) then the appropriate branch route is not
triggered again and the headers are wrong. But I can not set t_on_branch
within route[DISPATCHCALL] or route[DISPATCH_FAILURE] because this is
called with a different pre-set branch trigger, d
> What else could cause this not to match?
Nevermind... being blind...
There was an inactivating #!ifdef around the part matching the
dispatcher in question.
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_is_from_list() returns nothing.
I have also sniffed the connection. The ip addresses in the packets are
the ones I expect.
modparam("dispatcher", "ds_ping_from",
"sip:survivabil...@sip.example.com")
pings the machines and is getting answers.
What else could ca
replicated via DMQ between two instances which I too
care never to restart at the same time.
Will completely stop and then restart the two affected instances and
clear the dialog and dialog_vars tables to see if this issue is
manifesting itself again.
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Hi
Is there a way to get register messages into accounting?
This could come in handy while helping figure out if clients were not
registered while calls attemted to reach them.
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ile calls-tests are running.
I just wanted to make sure there is not need to specifically clear
dialog variables at the end of a dialogue.
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rst facing
that requirement. So my hope was that I missed some module or way to
accomplish this.
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PRACK with t_on_reply. It looks like they somehow pass completely
unprocessed through Kamailio (not completely Via Header are removed
accordingly)
Am I missing something?
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ler way to achieve this)
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> $fU,$rU,$tU within MANAGE_REPLY
>
> I get: do_action(): bad uri
Ok, it took me some time to realize, that REPLIES to INVITE do not have
an $rU.
Still wondering, if there is a more elegant way to achieve this.
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rks fine it looks as it's only changing
the usernames that causes this issue.
What would be the proper was to change usernames in replies?
Is there any module which would apply the same user translations as
done with the invite to all replies and re-invites?
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t I may be do wrong?
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atter if I do this on a failure route
or a branch route.
Or did I miss a trick? :-)
Would I need to also use branch_route to send the initial call towards
the customer CPE and set the 'localized' $fU there? Is this message
being dropped and the original one used again when triggering
fa
p with the original destination and new
destination concatenated.
Is there a way to 'reset' the $tU and $fU after setting them once, so
that they can be set again?
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notice,
that $ct contains the value of the INVITE contact header sent to the
customer, not of the 302 reply contact header.
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Hi Tim
> set_body($rb+"\r\na=prime:20", $cT);
set_body($rb+"a=prime:20\r\n", $cT);
works fine! Thanks a lot!
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Hi Patrick
> You can check the uac_redirect module.
>
> Not sure it handle Diversion header but its a good starting point.
Merci! That sounds promising I'll try. The ability to add a 'reason'
let's me hope it might create a Diversion header.
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27;replace' a body line.
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C
on.
But I wonder, is there maybe a module which handles all of this out of
the box?
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nd when
ds_select_domain ?
Could ds_select_domain be safely used for any situation?
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_select_dst
I can not figure out, why I see this difference.
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s by reading header from an 'input buffer'
and writing to an 'output buffer'.
I guess in case of t_on_failure the complete output buffer with corrected
$ru and additional header is being discarded so I have to do all changes again?
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URI is reset to the
value it had on 'relaying'. Is this on 'inbound replaying' or on
'outbound relaying' when route(RELAY) was called to send the call TO
the first annoucement server?
Is there a bit more specific information on that behaviour?
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