Am Thu, 20 Jun 2024 14:14:11 -
schrieb smartin114--- via sr-users :
> Invite to carrier is good, sent over tls and so the reply from the carrier,
> the 200 ok. The ACK is sent via UDP and so is the Bye, which I didn't
> include.
I fear I need more information and a complete example to see
Hi
> Originating invite towards carrier. Yes, the contact includes the
> transport=tls.
I think I'm missing something.
Which message is not correctly routed?
The 200 OK reply to an INVITE which was initiated via transport tls? Or
messages in a new transaction of that call, possibly from the
Hi
The INVITE would also be interesting.
Does the Contact: Header in the invite contain the transport=tls
attribute?
Souldn't the 200 OK reply contain at least one Record-Route (the lowest
one) stating transport=tls?
PS: I was facing a similar issue with a commercial SBC which, when
TLS/TCP is
Hi
To determine, if rtpengine was engaged during an invite or not, I make
use of rr params for all messages in following transactions and I set
an avp for all messages in the current transaction.
When the call is canceled, I would need to call rtpengine_delete to
make sure rtpengine stops
Hi Gang
To process spiraling calls without the dialog module. I'm trying to use
a rr param to identify in which iteration I am.
So when I have a call spiraling through the same instance 3 times,
every time I process an invite I would do something like:
add_rr_param(";sp-count=$avp(sp-count)");
Hi Christian
> this is interesting but when i install the package all start automatically.
> How can i disable this function (compilation for kernel module)?
> Maybe i need to install a different packege?
To disalbe kernel module usage:
rtpengine.conf
### for userspace forwarding only:
table =
Hi Alex
> I wouldn't worry too much about these. They're kind of an anachronism.
I just stumbled over your article:
https://www.cnblogs.com/shunzh/p/14360712.html
This helps a bit. After looking again at the issue I observe, I think I
have narrowed it down on the alias not being set on the
Hi Gang
Somehow I don't get my head around NAT Flags and the nathelper module
https://www.kamailio.org/docs/modules/5.7.x/modules/nathelper.html
In the examples I found, there is: FLT_NATS and FLB_NATB
If I got it right, FLB_NATB is a branch flag, which shall indicate that
the device is
Hi List
I have stumbled over this challenge:
I have a kamailio server without dialog module. Only tmx module, acting
as registrar / rtpengine host.
If the call is routed over that server once, this works fine.
If the call is routed more than once (call from one location to another
on same
Hi Bernd
I also use dual-stack.
When the destination has an ipv6 and ipv4 address in the DNS, then
kamailio should be able to use either of those protocols to reach it.
So:
CPE => ipv4 => Kamailio => ipv6 => CPE
should work, if the involved CPE play along!
Example: Cisco SPA112 don't know
Ciao Christian
PCAP file would be easier to read. But let's see..
1st Leg:
Linode Client1 IP: 93.189.137.22 => Kamailio Server: 44.3.44.40
(73 from HB9EUE :-) )
That Linode Client has RTP on: 192.168.1.21
2nd Leg:
Kamailio (44.3.44.40) => Client2 (88.116.10.134) Grandstream
RTP Side 1:
Ciao Christian
> I set up a Kamailio SIP server in a virtual machine on a private network and
> i connect to this with a WireGuard VPN.
Maybe, provide more insight
Your Kamailio-Server also acts as the WireGuard Server?
All your SIP clients connect using WireGuard?
Can you ping from one
Ciao Christian
Unless you use rtpengine or similar, Kamailio won't do anything with
RTP.
So I fear, you have to sniff the INVITE + DSP and 200 OK + SDP of one
of those calls without audio. (tcpdump -nvs0 port 5060 on the kamailio
host for example)
Look at the
c=IN and m=audio lines in the SDP
Hi all
I have found another solution:
Don't use: setflag(FLT_DLG);
call dlg_manage() on every message entering request_route before trying
to set dlg_vars.
On a 422 reply I now get two CDR, but that is OK as long as I have
correct dlg_vars on the second almost identical call (endpoint
Hi Omar
> Thanks for your replay. Lookup is depending on pseudo-variables and in my
> case I want to lookup for a static user "test" which is not included in the
> R-uri or not even the request SIP message.
>
> So is there any another way to do it?
Not sure what you exactly want to do. You
Hi List
I am just wondering...
When I am sending the initial INVITE to a customer CPE, this goes
throug the whole location lookup and through a branch route in which I
make some last adjustments to the headers, like removing header the
customer shall not get (like P-Asserted-Identity which would
Hi Calvin
> I'm trying to add something simple like the following:
>
> append_hf("X-testheader: True\r\n", "From");
>
> However, I don't see my X-testheader in a packet capture. Are there
> any common pitfalls that would prevent append_hf from working as
> expected?
Weird, I use this a lot and
Hi Matthew
> However in my scenario I wish to first try one endpoint and then if it gets
> a 4XX response forward it to another UAC, and update the TO accordingly.
>
> Currently trying to call uac_replace_to twice shows an error and corrupts
> the TO header. Is there a work around for this?
I
Hi
By setting: branch_expire to the same value as dialog_expire which is
12 hours, now my headers get correctly restored on UPDATE and BYE.
But in this UPDATE scenario:
CPE A => Registrar A (topos) => Core => Registrar B (topos) => CPE B
I now have the phenomena that the Contact Header is
Hi Karsten
> did you use some kind of db backend for topos like redis or mysql?
Well, you have to, to be able to store the route and via sets :-)
MySQL in use.
We might have figured out a possible cause.
This is the topology:
CPE <=> Kamailio Registrar <=> Kamailio Core <=> Interconnections.
Hi John
> Those 5 vias remind me an old firmware of Cisco equipment. I think I found
> that
> once and solved with sems or something else acting as b2bua.
It's not a Cisco. It's an ARRIS CPE which used to be operated as
PacketCable CPE but for which also a SIP Firmware is available.
I would
Hi
https://lists.kamailio.org/pipermail/sr-users/2020-July/109801.html
I have the exact same issue.
When the B side is starting a new transaction (UPDATE to refresh the
session in my case) without topos enabled, that transaction contains
one or multiple route header and a to_tag.
Therefore
Hi Alex
> There are lots of strategies for reducing message size, although the
> RFC-recommended (better said, mandated :-) approach is just to switch to TCP
> whenever your message sizes come within 200 bytes of the MTU, if I'm not
> mistaken.
Those CPE do not support TCP.
> Otherwise:
>
>
Hi Alex
> Not really a great answer for you, but I think you should reconsider using
> `topos` to reduce SIP message size.
>
> `topos` is complex and I'm not sure the added complexity pays off in this
> case, from a purely thermodynamic point of view.
So, any idea how to solve the issue?
At
Hi
I'm having another go at topos trying to solve an issue of sip messages
getting to big for certain clients to process.
Unfortunately some methods (ACK/PRACK/BYE probably CANCEL) do not
contain a contact header, therefore the topology is still revealed (and
potentially a lot of via header
Good morning Henning
> one really effective way is the topos module. If you can not use this, you
> could switch to compact sip header. This way you save some bytes per
> header-field. But this is something that needs to be supported from the user
> agent as well. You could remove not needed
Hi all
I had a mistake in the branch routes, which caused the AVP not to get
added to the stack in a case. That's why I had such a hard time with
the indexes being all over the place.
Mit freundlichen Grüssen
-Benoît Panizzon-
--
I m p r o W a r e A G-Leiter Commerce Kunden
Hi List
avp stacking is starting to drive me crazy... Somehow I don't
understand what is going on in this situation:
I have a main branch, and add two more branches with append_branch()
I use a on_branch_route trigger:
branch_route[BR_TO_CPE]
{
[Here, some code setting $var(needbh)]
Hi Bastian
> you can add a Record Route parameter to the initial INVITE [1] and for that
> parameter after loose_route() for in-dialog INVITE [2]
Thank you, that looks promising.
But now I face another issue with parallel branching.
On the branch route I do add_rr_param(";rtp=yes") if I need
Hi Alex
Only drawback I have with this is, when cycling through registered
contacts and adding branches. I can only have one trigger.
> request_route {
>...
>
>if(jittery)
>t_on_branch("BRANCH_DECAF");
>else
>t_on_branch("BRANCH_REGULAR");
Hi Alex
> As is often the case, there is a useful opportunity here to step back, switch
> to decaf, and ask why you want to call 'exit' from a branch_route in the
> first place.
:-)
Actually I intended to call 'return(1)' to stop processing that actual
route and continue in the calling route
Hi Unai
> It looks like now I have all pieces configured and I can see output when I
> run "kamctl dialog show”. Is there a way to verify that the load balancing is
> working as expected (aside from reproducing the production scenario)? Is
> there any log to enable and/or look at? Any stats or
Hi
I was wondering what exactly happens with parallel branching in this
situation:
branch_route[BR_T]
{
if (condition)
{
# Stuff below not required
exit or return
}
do other stuff before relaying
}
Will this break, stop
Hi Henning
> somehow the usrloc internal record id got duplicated. Are you seeing this
> error frequently or just on some occasions, like the restart you mentioned?
> You can try to activate this parameter:
> https://kamailio.org/docs/modules/5.7.x/modules/usrloc.html#usrloc.p.db_insert_update
Hi Gang
I would like to set a variable specific to the branch. Is there a way?
append_branch($var(aor1));
$(branch(dst_uri)[-1]) = $var($dst1)
$(branch(want_plus)[-1]) = 1;
append_branch($var(aor2));
$(branch(dst_uri)[-1]) = $var($dst2)
$(branch(want_plus)[-1]) = 0;
t_on_branch("BR_T");
Hi List
Via sql_xquery I get a stacked xavp aka array.
I would like to store that stacked xavp into a hash table for later usage in
other transactions. Is this
possible?
Via xarp_parameters explode or implode does not work as that would stringy the
same key multiple times.
Mit freundlichen
Hi gang
location database contains multiple contacts for an AOR
The invite therefore is parallel branched to those multiple contacts.
Each reply from each branch is handled by MANAGE_REPLY.
rtpengine_manage() is called in MANAGE_REPLY to stop rtpengine
processing a failed call.
Unfortunately,
> Is there a way to trigger a branch route for only some of the
> destinations added with append_branch? Or to trigger different
> branch routes for different destinations in one set added with
> append_branch?
Always the same, when the email is sent and I start thinking
Hi List
Imagine the following situation:
Two Kamailio registrar proxies handling the same domain. Location
Information synced via DMQ.
So if a device registers we don't know on which of the two registrar
proxies.
Core proxy is dispatching calls to either one of those registrars.
Core also
Hi
Just a guess:
https://www.kamailio.org/docs/modules/devel/modules/sqlops.html#idm85
If you define 'localhost' then the default socket (probably somewhere
in your MySQL config) ist taken.
Mit freundlichen Grüssen
-Benoît Panizzon-
--
I m p r o W a r e A G-Leiter Commerce Kunden
Hi Alex
> Perhaps it is helpful to ask a more basic question: why are you using
> server-side keepalives?
Because of a *stupid* cloud hosted PBX vendor solution.
That vendor is hosting his PBX behind a NAT without SIP ALG.
NAT UDP Timeout 90 seconds!
That vendor refused to send any kind of
Hi
We have reached 2 registered CPE and start facing issues.
Am I observing correctly that when ka_mode 4 is enabled, OPTIONS are
send simultaneously to all registered CPE and not spread over the
interval?
They could be the cause very high pps peaks on network equipment?
If so, is there a
Hi List
Kamailio 5.5
# kamcmd stats.get_statistics dialog:
dialog:active_dialogs = 17
dialog:early_dialogs = 18446744073709551615
dialog:expired_dialogs = 0
dialog:failed_dialogs = 31198
dialog:processed_dialogs = 107519
Where could this early_dialog value come from?
Mit freundlichen Grüssen
Hi
> But no matter how I try, I get a 500 error...
>
> # kamcmd ul.db_users location
Still curious why this does not work.
> Or would I need to query this directly from the database?
Solved my requirement:
snmpget -Oqv -v 2c -c NotActualCommunity localhost
Hi List
I want to find a way to check how many users are regsitered.
https://www.kamailio.org/docs/modules/devel/modules/usrloc.html#usrloc.r.db_users
But no matter how I try, I get a 500 error...
# kamcmd ul.db_users location
And yes, the table is called 'location'.
I did try to find out
Hi Daniel
> the B should not do record_route() when forwards back INVITE to A.
Right, moving record_route() from the beginning of request_route{} to
the branch routes requiring this, solved this for the RR header on my
registrar node.
> Via is for routing back replies, if you remove it, node B
Hi Alex
> If you're having to think about how to do things that break basic SIP
> semantics, it may be time to rethink your design.
:-) We went into production far down that rabbit hole now. It would be
quite hard to pull out from that far in.
> More particularly, passing requests from A to B
Hi List
Two Kamailio Nodes situation.
Node A: Routing Instance.
Node B: Registrar Instance.
An invite is sent from Node A to B.
Customer registered on B is 'busy' as example.
B initiates Call Forwarding by adding a Diversion Header and sending
the Invite back to A with a new R-URI towards the
Hi
I'm opening an issue on github as I consider this a bug.
This fixes the issue:
route[DMQ_CAPTURE]
{
if(is_method("KDMQ"))
{
if(has_body("application/json") && $fU == 'dialog')
{
if (jansson_get("lifetime", $rb,
> Is there a way to crank up debug output in the dialog and/or dmq module?
debug=3 is what I was looking for.
And that looks good:
DEBUG: dialog [dlg_dmq.c:142]: dlg_dmq_handle_msg(): body:
{"action":2,"h_entry":3217,"h_id":4523,"state":4,"start_ts":1700748833,"lifetime":43265,
...
That is
Hi
I'm trying to change the lifetime timer on a KDMQ dialog message before
handing it with dmq_handle_message();
if(has_body("application/json") && $fU == 'dialog')
{
if (jansson_get("lifetime", $rb, "$var(lifetime)"))
{
Hi Daniel
> there is no automatic reload, be sure you do not have some timer routine
> (inside kamailio.cfg or cron.d) that does it, or an external application
> (e.g., management application).
Thank you, that was the cause. We have a cronjob which downloads rules
from a spit call blocklist
Hi
I'm still looking for a better way to tell Kamailio that we want to
enter 'maintenance' and STAY in maintenance after a restart.
Maintenance is: Reject all messages without totag with 503 to prevent
creating new dialogues.
I could use a shared pv and the use kamcmd pv.shvSet to toggle it.
Hi
> I have a scenario on wich requires sequential fork (up to 30 possible
> destinations) with different
> Call-ID for each leg. I am using topoh module enabled, but it always puts the
> same Call-ID.
> Someone have a solution for this scenario?
AFAIK that is not possible as Kamailio, being a
Hi Anthony
You were spot on. Not a single crash occurred anymore after setting:
modparam( "db_mysql", "opt_ssl_mode", 1 )
Mit freundlichen Grüssen
-Benoît Panizzon-
--
I m p r o W a r e A G-Leiter Commerce Kunden
__
Hi Karsten
> how are you measure the traffic you don't want to your Homer? Print stuff
> in that Kamailio event route or something else?
Basically, I want to mirror sip messages to homer, only sip messages
relevant to calls, not anything else which might be processed by
kamailio.
Hi
Further trying to eliminate every possible cause of UDP drops...
We use Homer as HEP server in conjunction with the siptrace module.
So to prevent DQM traffic to be sent to Homer I added:
event_route[siptrace:msg] {
if(is_method("KDMQ")) {
drop();
}
}
And had a closer look
Hi Gang
While still hunting DMQ issues, I noticed that the OS is reporting UDP
drops. Maybe DMQ packets? Would DMQ re-send a lost packet?
I increased OS UDP RX buffers times 10 and monitoring counters.
I also found that DMQ can use tcp or tls as transport.
Could this help to prevent loosing
Hi Team
I'm still hunting down DMQ dialog issues.
https://www.kamailio.org/docs/modules/devel/modules/dialog.html#dialog.p.enable_dmq
Quote:
"Notably, it is not possible to send in-dialog requests on any but the
original proxy instance."
I make sure, that if a procied call (with same callID)
Hi List
At the moment, we challenge every invite (and re-invite) to make sure
the customer is authenticated.
Now we have one kind of PBX, which never does not authenticate when we
challenge a Re-Invite.
According to the vendor of that PBX's RFC interpretation, answering a
challenge to a
Am Mon, 18 Sep 2023 10:09:40 +0200
schrieb Benoit Panizzon :
> append_to_reply("X-RR: (append reply) Final Dispatch Failure
> $avp(dispgroup):$avp(trunkname)\r\n");
Found a mistake. append_to_reply works.
Mit freundlichen Grüssen
-Benoît Panizzon-
--
I m p r
Hi List
I'm attempting to add a header to a reply message on failure.
Situation:
Registrar => Core => (dispatcher list) => Various IC
I would like to know on the registrar, which of the dispatcher groups
failed.
What I attempted sofar:
failure_route[DISPATCH_FAILURE]
{
$(branch(dst_uri)[-1]) = $(ulc(aor=>received)[$var(i)]);
seems to be correct.
Mit freundlichen Grüssen
-Benoît Panizzon-
--
I m p r o W a r e A G-Leiter Commerce Kunden
__
Zurlindenstrasse 29 Tel +41 61 826 93 00
Hi Yuriy
I fear I still need some help
What I am doing now is:
if(reg_fetch_contacts("location", "$var(lookupuri)", "aor")) {
t_newtran();
xlog("L_INFO","$cfg(route): Existing registration contacts
count: $(ulc(aor=>count))\n");
$var(i) =
Hi
This is how I handle them on the registrar:
[...]
# Record original destination before performing call forwarding
$avp(pre_redir_destination) = $rU;
[...]
Somewhere in the config we start reacting to replies from the
customer like when busy...
if (t_check_status("(486)|(600)")) {
Hi David
If you change $tU outside a branch route, kamailio just appends the new
value to the existing one.
You have to use branch routes, the changes are only valid in the branch
so if the branch fails without the call being connected (and for
example a failure route is triggered) you can set
Hi
Kamailio 5.5 and 5.6
Server version: 8.0.33-0ubuntu0.20.04.2 (Ubuntu)
Using db_cluster
Only issue I occasionally observe is when a query times out or has some
other issue, then the query fails and is not retried on the next
cluster member. But I guess this is how db_cluster works and not
Hi Team
To offer some diagnose functionality I created an API that returns the
location list via JSONRPC
But this creates huge variables.
Jun 12 09:43:32 prod-cpereg02 kamailio[2174548]: ERROR:
[core/pvapi.c:1491]: pv_printf_mode(): no more space for spec value - printed:0
token:8599
Hi Sergey
> 3.1.1 WireLess
>
> How do you want to route such call types without a postal code?
We only offer Fixed Landline Services. No mobile. So this is not the
issue. We know the 'installation' location of every fixed line.
Of course, if a customer takes his phone somewhere else, this is
Hi Sergey
Swisscom is operating all the emergency call infrastructure in
Switzerland.
Most documentation is public:
https://www.swisscom.ch/en/business/enterprise/offer/alarming-solutions-ealarm-emergency/sos-database.html?campID=SC_emergencylocalization
Referring to this document:
Hi Sergey
Thank you for your reply.
> if(is_method("REGISTER")) {
>sl_send_reply("100", "Checking your credentials");
> } else {
>sl_send_reply("100", "Attempting to connect your call");
> }
I left auto_inv_100 to 1 but I am now experimenting with calling
t_reply before doing the
Hi gang
In 2024, Switzerland will start using NG112 (NG911) procedures to
transmit the caller location via Geolocation URL via LIS Server to a
PSAP.
So time to start testing this on my devel plattform. Kamailio ships
with the lost module which looks promising.
In short, the Procedure is as
Hi list
Since updating 5.5 to 5.6 i often see this message in the logs:
ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on
query: Duplicate entry 'uloc-645ca178-f29e-1' for key 'location.ruid_idx' (1062)
ERROR: [db_query.c:244]: db_do_insert_cmd(): error while
Hi Daniel
> But I wanted to check if you started with --atexit=no cli param, because
> your error messages seemed linked to start up time.
Yes, --atexit=no parameter is set in the systemd file. Shall I try
without?
Mit freundlichen Grüssen
-Benoît Panizzon-
--
I m p r o W a r e A G-
Hi Дилян
> in case Kamailio is linked with OpenSSL 3.0 it will fail. Linking with
> Openssl 1.1.1 works.
Version: 5.6.4+ubuntu22.04
Architecture: amd64
Maintainer: Kamailio Admin Group
Installed-Size: 1072
Depends: kamailio (= 5.6.4+ubuntu22.04), libc6 (>= 2.34), libcurl4 (>= 7.16.2),
Hi Alex
Thank you for your input. Determining if the call is for a locally
registered user seems pretty easy.
Now the next challenge...
There might be multiple registered contacts for an AOR using multiple
protocols and residing on different registrars.
I was considering using dispatcher mode
Hi Yuriy
> You don't need to do so in branch route. You can do this within
> request_route.
Bug accessing:
> > $var(socket) = $(ulc(aor=>socket)[$T_branch_idx]);
Is only possible in the branch route as I need that index, right?
Mit freundlichen Grüssen
-Benoît Panizzon-
--
I m p r o W a r e
Hello
Is there a possibility to send a reply from within a branch route?
Experimental case (yes, I know it's a non working set-up, but I just
wonder if that would be possible).
Two registrar nodes, location information shared between the two.
CPE registers via TLS, therefore the existing
Hi Team
I think, I might have found something that was overseen in the
rtpengine module.
A (ipv6) <=> rtpengine <=> B (ipv4)
Invite+SDP from B to A
It looks like this works as expected. A SDP c= line is created
containing an ipv6 address.
Invite, no SDP from A to B
Reply with SDP offer from B
Hi
Before I go too deep into try and error, I guess others have been there
too.
What are sensible value you use for the pike module to detect /
mitigate abusive behavior, especially dictionary attacks?
Are there better solutions that the pike module?
Mit freundlichen Grüssen
-Benoît Panizzon-
Hi Stefan
> I would like to do topos on the SIP-Proxy direction PSTN but only for
> specific hosts.
I ran in a quite similar issue with topos which made it unsuitable for
my needs.
Same situation:
CPE <=> Kamailio Registrar (TM Module) <=> Kamailio Routing Core
(Dialogue aware) <=> IC
I
Hi All
I would like to avoid having run away calls as good as possible.
For this, I would like to require timers and set the maximum expires
timer to say 1800
I had a look at the sst module, but I see no way to require timers or
to set a max timers value.
Do my intentions make sense? How did
Hi
We have two registrars.
Today I observed a CPE doing this:
CPE REGISTER (no auth) => Registrar 1 => 407 Challenge NONCE A
CPE REGISTER (auth to NONCE A => Registrar 2 => 407 Challenge NONCE B
CPE REGISTER (auth to NONCE B => Registrar 1 => 407 Challenge NONCE A
and so on.
Time on both
Hi Volker
An Example by which I query a profile counter:
curl -X POST http://VoiceSwitch:8080/RPC -H 'Content-Type: application/json'
-d '{"jsonrpc": "2.0", "method": "dlg.profile_get_size", "params" :
["channels-inuse", "Customer-"], "id": 1}'
I think, with jsonrpc 2.0 you could pass an
Hi Daniel, Alex and all
I'm having an argument with the Vendor of our SBC towards IC carriers.
According to $vendor tech, if the SIP Message was received via UDP, the
transport= attribute of the Contact Header has no meaning and therefore
the RURI of a following transaction (like BYE) will not
Hi James
> It should automatically re-use an existing TCP connection.
> Can you give an example?
After digging more in this issue, it looks like I found the solution:
Upon successful REGISTER authentication I do:
if ($proto != 'udp') {
Hi Alex and Daniel
Thank you for the quick reply.
I have now also set double_rr = 2 but still no joy, same two RR header
are added as before.
modparam("rr", "enable_full_lr", 0)
modparam("rr", "append_fromtag", 1)
modparam("rr", "enable_double_rr", 2)
Situation, IP's and Usernames a bit
Hi List
Digging further into my tcp/tls issues...
Upon successful authentication I call:
tcp_keepalive_enable("60", "5", "5");
tcp_set_connection_lifetime("120");
I tought, this would do to keep the connection alive.
I noticed, despite keepalive packets being exchanged, kamailio is
closing
Hi
CPE registering with TCP (or TLS).
2nd leg is UDP.
Within one transaction (INVITE to OK including all ACK and PRACK)
transport is kept as desired for both legs.
But when the B side disconnects (sending BYE, new reverse transaction),
this is sent via UDP to the A CPE which initially talked
Hi List
CPE behind Firewall, registering to Kamailio via TCP (or TLS).
When a call is sent to the CPE, kamailio attemts to open a new TCP
connection and is blocked by Firewall.
Is there an option to tell kamailio to use the existing registered TCP
connection?
Mit freundlichen Grüssen
-Benoît
Hi all
I'm trying to incorporate rtcp data from rtpengine into our CDRS. So
trying with the first interesting value, the average mos.
modparam("rtpengine", "mos_average_pv", "$avp(mos_average)")
If I understood right, to get this variable set, I need to call
rtpengine_manage() on the message
Some update after more testing...
not droping topos to the core and setting
modparam("topos", "rr_update", 1)
solves some situations but has caused at least a new one.
Situation:
Two CPE on same Registrar.
A CPE => +
+ REGISTRAR <=> CORE
B CPE <= +
A is calling B.
Call route: A
Hi Henning
> you've seems to have already found a workaround, as posted later.
>
> Generally, topos should work for PRACK, there were several enhancements/fixes
> committed in the last year.
Unfortunately, I didn't get it working as I anticipated.
CPE - REGISTRAR - CORE
I would like to hide
Hi Patrick
Agreed, this could become an issue when we do dual stack ipv4 and ipv6
unless we use a hostname which will probably cause other issues.
> But overall sip compliant component must Always follow Route ip before
> contact IP.
I will test how the know buggy client behaves.
While
Hi
I'm trying to work around Route-Header and Via Issues with the two
topology hiding modules topoh and topos and trying to figure out, which
one works better for our environment.
My conclusion so far:
topos creates very clean header, but needs a database or redis. I'm
always reluctant in
Hi
I am struggling with DMQ and dialog DB storage leading to orphan or
duplicate entries in database and not loading all dialog information
upon restart.
usrloc on the other side, also using local db storage and dmq, works
flawlessly even when restarting one node. All location information is
Hi Gang
I have this code snipplet:
if (has_credentials("$fd")) {
xlog("L_INFO", "$cfg(route): got $rm with credentials. Validate
them!\n");
if ($aU == $null) {
xlog("L_INFO", "$cfg(route): no auth user, send
challenge\n");
Hi out there!
While experimenting with the listen and adverize 'hostname' option I
came across a voice switch, which I suppose fails if the Via header is
not an IPv4 address.
There is a record_route_advertised_address(address) in the RR module,
to set a customer Record-Route Header.
Is there
Hi List
Testing failure situations, I discovered unset_dlg_profile can
not be used in request_route:
I count the channels per customer in a dlg_profile to know when they
are busy. Residential POTS customer have 1 channel.
Now this situation (Trying to mimik POTS behavior)
Kamailio <=> CPE of
Hello World and Rick
> A quick update. Now it works. I guess the newly added DNS hostname was
> not yet fully propagated to all DNS caches.
I start doubting, this was a clever idea...
I wonder what solutions other have come up to.
After some testing I noticed: SIP ALG on NAT devices do work by
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