Daniel-Constantin Mierla writes:
> > Also, there is no "=". So instead of ";sn=", the code should look for
> > "sn".
>
> Indeed, I will push a fix for it.
I also did one, but yours is cleaner than mine. Thanks. Tested and it
works, no more warnings to syslog.
Looks like no-one had ever
Juha Heinanen writes:
> I think I found the bug:
>
> ./src/core/config.h:#define SOCKNAME_PARAM ";sn="
>
> but in parse_params() result params are without ";".
Also, there is no "=". So instead of &q
I think I found the bug:
./src/core/config.h:#define SOCKNAME_PARAM ";sn="
but in parse_params() result params are without ";".
-- Juha
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Daniel-Constantin Mierla writes:
> Do you have the rr modparam related socket name mode set?
Yes, sorry, forgot to mention it:
modparam("rr", "sockname_mode", 1)
I have added some debug to loose.c and some converted DBGs to INFOs in
rr_do_force_send_socket() function:
I did more tests and found that loose_route() does not find socket by
name given in ;sn param:
Feb 12 10:40:59 lab /usr/bin/sip-proxy[16975]: WARNING: rr [loose.c:799]:
rr_do_force_send_socket(): no socket found to match second RR
Seems to be OK not to include ;sn param in IP address R-R URIs. At
least I didn't get any errors or warning to syslog when I called
loose_route() on in-dialog requests that had such Route headers.
-- Juha
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Daniel-Constantin Mierla writes:
> Set names to the sockets
> (https://www.kamailio.org/wiki/cookbooks/devel/core#listen):
>
> listen= . name "s1"
>
> And then add parameter "sn=s1" to the specific Record-Route header.
If sockname_mode param has value 1:
modparam("rr", "sockname_mode",
Alex Balashov writes:
> That is the main reason I had previously thought it wasn’t possible!
Have you changed your mind?
The document
https://skalatan.de/de/blog/kamailio-sbc-teams
tells
Change in your configuration the existing record_route() function call
to this one:
Daniel-Constantin Mierla writes:
> The record_route_preset() is supposed to offer the flexibility of
> setting the address part in RR headers via variables. I do not see how
> the record_route() can be better variant, because it will end up in the
> same kind functionality: for which to use the
Daniel-Constantin Mierla writes:
> If not, then that has to be added because the purpose of
> record_route_preset() is to be able to set rr header as one needs based
> on variables that can be used in parameters. Note that the parameters
> are only "host:port;transport=", without protocol, for
Daniel-Constantin Mierla writes:
> The code of add_rr_param() used after the record-route header is added
> indicates that the values should be added. I didn't have the time to
> check what is with the cached params if add_rr_param() used beore the
> record-route header is added. I will try to
Henning Westerholt writes:
> not 100% sure what the other blog post is suggesting - but in my
> setups there is no need for a special record_route(..) in reply
> handling (like reply route).
Its is not question about reply handling, but direction of INVITE. See
my earlier message.
-- Juha
Carsten Bock writes:
> I don't understand the trouble:
> Wouldn't the following lead to the exact example:
>
> listen=tls:1.2.3.4:8007 advertise teams.tutpro.com:8007
> listen=tcp:1.2.3.4:5070
That would not work when K is a multi-tenant carrier Teams SBC proxy
that requires tenant specific
Daniel-Constantin Mierla writes:
> What exactly do you need to do for R-R in replies?
>
> The R-R headers are mirrored in replies, so if you add R-R for requests
> using the right order of parameters based on the direction, there is
> nothing to be done for replies, or at least I haven't had to
Sergey Safarov writes:
> Maybe need to move into more global changes...
>
> How about DNS hostname usage for all headers?
> "Via", "Record-Route" maybe "Contact" when Kamailio with extra modules
> play media?
>
> how about global directive like
> header_with_hostname=tue
I'm not sure if that
Pepelux writes:
> Try to put SBC-FQDN instead of SBC-IP-ADDR in the Record-Route:
>
> Record-Route: :5060;ftag=c3da9477e05d45fca31c24a155af3318;lr=on>
Also, it better to follow OpenSIPS tips on the matter
https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/
since it also mentions
to keep on using record_route() as before, but tell it
to use a FQDN instead of IP address of the proxy in the first or second
R-R header that it is adding.
-- Juha
-
Daniel-Constantin Mierla writes:
>
> On 10.02.21 21:01
When request is sent from Kamailio to MS Teams SIP Proxy, the top R-R
URI needs to contain FQDN of Kamailio SIP proxy instead of its IP
address. Document
https://skalatan.de/de/blog/kamailio-sbc-teams
suggest to replace record_route(); call with
Daniel-Constantin Mierla writes:
> Get also the output for gdb commands in frame 0:
>
> list
>
> p *cbp
Here, but not much:
Program terminated with signal SIGSEGV, Segmentation fault.
#0 0x7f601d4b2f32 in run_trans_callbacks_internal
(cb_lst=0x7f5ffb53e318, type=512,
We got the crash below with 5.4.2. Before the crash there was
db_mysql_async_exec_task(): failed to execute query
related error messages in syslog.
-- Juha
Program terminated with signal SIGSEGV, Segmentation fault.
#0 0x7f601d4b2f32 in run_trans_callbacks_internal
I read kamailio-sbc-teams tutorial
https://skalatan.de/blog/kamailio-sbc-teams
and it does not mention, which CA root certificates need to be included
tls config ca_list so that Kamalio is able to verify
sipX.pstnhub.microsoft.com server certificates.
-- Juha
Daniel-Constantin Mierla writes:
> You can add the new file there and eventually a few notes in the
> README.md how to install it.
Done, Juha
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Daniel-Constantin Mierla writes:
> this is syntax highlighting for kamailio.cfg, right?
Yes, plus indentation, which it inherits from c-mode.
> We can create a repo on kamailio github oraganization, like
> emacs-kamailio-syntax, similar to what is now for vim:
>
> *
After noticing that emacs mode is missing, below is my first cut of
kamailio-mode.el.
-- Juha
---
;; kamailio mode for emacs (function list is not complete)
;; activate by starting your kamailio.cfg with this kind of line:
;;
beer Ll writes:
> Has anybody use a android SIP client with TLS client authentication enabled
> with kamailio?
One possibility is baresip and baresip+ available from Play Store and
F-Droid.
-- Juha
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Daniel-Constantin Mierla writes:
> So, to summarize, there should be syntax highlighting (maintained at
> different levels) for the editors:
>
> * vim (I guess also neovim)
> * vscode
> * atom
> * mcedit
Long list, but unfortunately the oldest editor is missing: emacs.
-- Juha
But these could be used:
detailed_ip_type (ip, result)
detailed_ipv4_type (ip, result)
detailed_ipv6_type (ip, result)
-- Juha
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Daniel-Constantin Mierla writes:
> this one:
> https://www.kamailio.org/docs/modules/stable/modules/ipops.html#ipops.f.is_ip_rfc1918
If it is really is based on RFC1918, then I doubt that it would handle
IPv6 private addresses, since when that RFC was written, IPv6 didn't
exist.
-- Juha
I looked in the wiki for a utility function that would check if given ip
address is private, but didn't find one. Does it exist?
-- Juha
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Daniel-Constantin Mierla writes:
> New function added, but no testing done -- if there are issues, open a
> bug report.
Works as expected, thanks, Juha
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Daniel-Constantin Mierla writes:
> New function added, but no testing done -- if there are issues, open a
> bug report.
Thanks, will test later today,
-- Juha
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Kjeld Flarup writes:
> My question is still, which port is the BYE from the server supposed to be
> sent to?
If client is behind NAT, the server must use the connection that the
client opened to it to send the request to client. There is no way the
server opens the connection and uses it.
--
Kjeld Flarup writes:
> How is TCP SIP actually supposed to handle a BYE, when the client is
> behind NAT.
Client behind NAT is supposed to keep its TCP connection to SIP Proxy
alive and use it for all requests of the call. If the connection breaks
for some reason, the client sets up a new one
Daniel-Constantin Mierla writes:
> I wanted to say that the new function to be added should be a bit more
> generic, not targeting only set_reply_close()+send_reply(), but have a
> "mode" parameter to control what other operations should be done before
> sending the reply out.
That is how I
Daniel-Constantin Mierla writes:
> set_reply_close() is very minimal wrapper action from the core setting
> an internal flag. It can be merged with a reply function, but maybe it
> should be a new function a little bit more generic, like
> send_reply_mode(code, reason, mode) where mode is a
Many times (especially when there is a hacking attempt) I want to close
TCP or TLS session after sending reply. So, for example, I write:
set_reply_close();
send_reply("403", "Forbidden");
It would be more convenient if these two calls could be combined into
one, for example:
Added note about branch flags in registrar readme:
https://github.com/kamailio/kamailio/commit/f2996bf733a0b5e00fe124440353b9a69a3532a2
-- Juha
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I didn't find any mention in registrar module README that save()
function saves also branch flags and that lookup() retrieves them.
Is that documented somewhere?
-- Juha
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What I wrote in below, is not correct.
(1) Pointing Chrome to
https://:5061
does result in successful handshake:
Oct 17 17:53:06 lohi /usr/bin/sip-proxy[13274]: INFO: tls [tls_domain.c:751]:
sr_ssl_ctx_info_callback(): SSL handshake started
Oct 17 17:53:06 lohi /usr/bin/sip-proxy[13274]:
Jeff Bilyk writes:
> https://lists.kamailio.org/pipermail/sr-users/2013-March/077235.html may
> contain a helpful workaround,
Jeff,
Thanks for your reply. I do have
tcp_accept_aliases=no
and this problems appears before event_route [xhttp:request], that is,
already during TLS handshake.
--
Henning Westerholt writes:
> Interesting. Does e.g. 5.4 still works for you? Might be then a
> regression in trunk.
Same problem with 5.4.
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Simple way to show this problem without any WebRTC SIP client is to
point Chrome browser to K's TLS listening port:
https://:5061
and look with wireshark or tshark how the handshake gets terminated by
Chrome right after Server Hello.
The same with Firefox works fine.
-- Juha
Daniel-Constantin Mierla writes:
> In the code is possible to lookup the hash table at runtime, being also
> done by some of the htable cfg functions. It will require some new code
> to evaluate the name expression every time at at runtime, then parse and
> lookup the hash table structure...
A
Is there some good reason why htable in $sht(htable=>key) cannot be a
dynamic string containing pseudo variables?
-- Juha
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Daniel-Constantin Mierla writes:
> If nobody else here can add more, this aspect can be clarified by asking
> on sip-implementors mailing list or maybe ietf sipcore wg mailing
> list.
Perhaps Inaki or Jose as authors of the RFC could comment. I agree that
secure transport is the only one that
Daniel-Constantin Mierla writes:
> For SIP URI the parameter value is always ws:
>
> * https://tools.ietf.org/html/rfc7118#section-5.2
Yes, according to that RFC, but does it make any sense, since how you
tell based on usrloc received value if ws or wss was used for the
registration?
-- Juha
It may be related that also usrloc received uri has ;transport=ws:
sip:192.168.43.83:51176;transport=ws
instead of ;transport=wss.
-- Juha
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No matter if UA uses ws or wss to connect to Kamailio, when K routes
INVITE from this UA and record_route() is called, K always adds
;transport=ws parameter to Record-Route URI, for example,
Record-Route:
In this test, Kamailio is listening on that port like this:
# SIP over WebSocket
sagar malam writes:
> I have tried ul_rm_contact but it is not working :
> curl POST --data
> '{"jsonrpc":"2.0","method":"ul.rm_contact","params":{"0":"location","1":"
> 30...@x.com
>
sagar malam writes:
> I am using the usrloc module to store registrations. I could not find any
> way to remove stale registrations before expires time. Most of the
> registrations are mobile app so they wont run in background to save battery
> of phone hence we end up having lots of
Daniel-Constantin Mierla writes:
> Should be fixed now.
Yes, thanks, Juha
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tls_locking.c: In function 'tls_init_locks':
tls_locking.c:191:1: warning: label 'error' defined but not used
[-Wunused-label]
error:
^
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Daniel-Constantin Mierla writes:
> The DNS NAPTR is also turned off by default, which should be on, based
> on RFC, iirc, but that adds extra DNS query and slows down everything as
> most of services I saw so far do no relay on NAPTR.
It is OK to have NAPTR lookup off by default, but if it is
What is the point of dns_naptr_ignore_rfc default value:
dns_naptr_ignore_rfc
If the DNS lookup should ignore the remote side's protocol preferences,
as indicated by the Order field in the NAPTR records and mandated by RFC
2915.
dns_naptr_ignore_rfc = yes | no (default yes)
In my (and RFC
Henning Westerholt writes:
> * 5 new modules have been added (pv_headers, kafka, secsipid,
> * systemdops, dlgs)
Could not find many of these on page:
https://kamailio.org/docs/modules/devel/
-- Juha
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Daniel-Constantin Mierla writes:
> Should be fixed now. What OS/compiler you use that reports such warning?
> My clang doesn't do it ...
I'm using Debian 10 with C compiler gcc 8.3.0.
-- Juha
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Got this when building 5.4;
CC (gcc) [sip-proxy]core/dset.o
core/dset.c: In function 'uri_trim_rcv_alias':
core/dset.c:1042:6: warning: variable 'proto' set but not used
[-Wunused-but-set-variable]
int proto;
^
-- Juha
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Duarte Rocha writes:
> Does Kamailio have any method to transform an History-Info with redirecting
> info into a Diversion or the other way around?
I don't know of any builtin function for that, but you can handle it
with normal config script statements.
-- Juha
This 488 think reminds me that SIP over webrtc is broken.
Webrtc UAS (at least JsSIP) cannot issue 488 before the UAS has started
to ring. It is very frustrating for the callee to get such a spam ring.
RFC3261 section "8.2 UAS Behavior" tells:
Note that request processing is atomic. If a
José Lopes writes:
> Hello Daniel,
>
> Thanks for your reply.
>
> On the next link, I put the original SDP from webrtc client and the SDP
> after kamailio that exposes the issue.
> I am using Kamailio version 5.3.3 with sdpops and rtpengine module.
>
> https://pastebin.com/bYr0AcVT
Perhaps it
Prakash Ganesan writes:
> We are planning to use kamilio/RTP engine for our WebRTC calls. I would
> like to know how the H264 codec licences are handled as i know there is a
> royalty fee associated with this video codec from the server perspective.
> As of now there is plan to use H264 pass
Daniel-Constantin Mierla writes:
> These commits were pushed to branch 5.3 already.
Yes, I noticed, thanks, Juha
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Daniel-Constantin Mierla writes:
> I pushed two commits on master and branch 5.3 to skip reusing tcp
> connection id. See if it solves your case.
I tried with master and now setting of $du in branch route worked OK.
-- Juha
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Daniel-Constantin Mierla writes:
> > Mar 29 10:19:28 char /usr/bin/sip-proxy[26162]: INFO: * du =
> > sip:127.0.0.1:5090;transport=tcp
>
> Does this happen after a usrloc location lookup? Trying to figure out
> what destination-related processing is done before to see what fields
> were
Daniel-Constantin Mierla writes:
> Have you tested with tcp/tls or with udp? There was a group of commits
> trying to propagate the tcp connection id as much as possible to speed
> up connection search, maybe that needs to be reset -- I can look into
> it, if you tested for tcp/tls.
This is what
Daniel-Constantin Mierla writes:
> The branch route is executed inside t_fwd.c (like 354 in master branch).
> You can try to do a diff of tm module between the branches to see if you
> can spot something related. I remember I grouped some variables in a
> structure at some point, but it should
Juha Heinanen writes:
> I noticed that in 5.3 and master setting $du in branch route has no
> effect. It works fine in 5.2 using exactly the same config.
This is pretty serious. If someone can point to possible source code
files, where this bug might have been originated from, I can
I noticed that in 5.3 and master setting $du in branch route has no
effect. It works fine in 5.2 using exactly the same config.
-- Juha
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Daniel-Constantin Mierla writes:
> Would it be possible to combine like playing a file first, then do echo
> mode?
As you noticed, play and echo use different audio_source and thus I
don't think it is possible to combine the two.
-- Juha
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Regarding echo, one can use same kind of baresip accounts file as
with play, such as
;auth_user=echo;auth_pass=xx;outbound="sip:192.168.43.82:5060;transport=tcp";ptime=20;audio_codecs=OPUS/48000/2,PCMU,PCMA;regint=600;pubint=0;regq=0.5;sipnat=outbound;answermode=auto
The difference is that
Daniel-Constantin Mierla writes:
> > Here is an example configuration of a SIP UA that plays a file when it
> > receives a call:
> >
> > .baresip/config:
> > ...
> > audio_sourcegst,file:///tmp/file_to_play.wav
> > ...
> > module gst1.so
gst1 module was recently
Daniel-Constantin Mierla writes:
> >> wondering if anyone here is aware of a lightweight sip app that can
> >> answer a call, play some file and/or do echo mode, mainly targeted at
> >> using it for basic sip routing and call testing.
> > baresip cli app can do all that.
>
> OK, thanks, I will
Daniel-Constantin Mierla writes:
> wondering if anyone here is aware of a lightweight sip app that can
> answer a call, play some file and/or do echo mode, mainly targeted at
> using it for basic sip routing and call testing.
baresip cli app can do all that.
-- Juha
In kamailio/etc/kamailio.cfg NAT test is based on nat_uac_test("19").
19 includes test 16:
16 - Test if the source port is different from the port in the “Via”
header. If the “Via” header contains no port, it uses the default SIP
port 5060
Based on a couple of tests using baresip, looks
Daniel-Constantin Mierla writes:
> do you have an external application that keeps persistent connection for
> rpc/jsonrpc commands?
That was it.
-- Juha
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It turned out that the unknown error messages were produced by process
32058 that was not in ctl ps list at all. So it must have survived
alive from a previous restart of K.
Question: how it is possible that a tcp worker that does not belong to
the current running K instance gets to service a
Daniel-Constantin Mierla writes:
> Try to install debugging symbols for libmariadb and maybe there will be
> more hints in the backtace of what it does internally, which can improve
> troubleshooting by searching on the web for similar cases.
I did that already earlier when I reported about
Henning Westerholt writes:
> If you only saw it once it is probably not worth to dig that deep into
> it, might be also caused from some external factors.
It has appeared several times and may be related to json request.
Perhaps tcp connection is broken before K has delivered the result.
--
Any idea where this kind of error message could come from:
Mar 12 07:16:17 rox2 /usr/bin/sip-proxy[32058]: CRITICAL:
[core/pass_fd.c:277]: receive_fd(): EOF on 61
Mar 12 07:16:17 rox2 /usr/bin/sip-proxy[32058]: ERROR:
[core/io_wait.h:607]: io_watch_del(): trying to delete already erased
Daniel-Constantin Mierla writes:
> > DB access works fine from K worker processes during the time when
> > insert is stuck in timer process.
>
> Are new records inserted in acc table? I think there are some
> tools/commands for mysql to inspect the state a database table, check
> acc table and
Daniel-Constantin Mierla writes:
> If it happens periodically, maybe you can track why: try to identify
> apps accessing the database for back up, cdr generation, etc ... as well
> as infrastructure maintenance operations (vm backup snapshot).
DB access works fine from K worker processes during
Daniel-Constantin Mierla writes:
> It seems to be the case of a retransmission timeout:
>
> #17 0x7f7dc04d4aca in acc_onreply (t=0x7f7d9e3b0650, req=0x7f7d9e357650,
> reply=0x, code=408) at acc_logic.c:604
>
> Code is 408 and the reply is faked value. This case is happening
Regarding db_mysql timeout_interval, it has its default value 2, which
means 6 seconds. The insert was hanging in the timer process much
longer and no error messages related to abort appear in syslog. How is
that possible?
-- Juha
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Daniel-Constantin Mierla writes:
> There is no async-insert done by acc with db_cluster, it is always
> standard (sync) insert. You would have to track why the mysql server (or
> the client library) is blocking from time to time, I don't think it is
> something that kamailio can do.
OK, thanks
Daniel-Constantin Mierla writes:
> The process is stuck in mysql client library.
>
> The async insert is not implemented in the db_cluster module, so at this
> moment, if you want it, you have to use acc directly with db_mysql
> module.
Then how is it possible that most of the time accounting
Daniel-Constantin Mierla writes:
> If you can reproduce it, watch what the timer processes do during that
> time frame. Get the list of processes with 'kamctl ps', then when the
> issue is exposed, grab the backtraces of all processes with:
>
> kamctl trap
>
> A file is created with the
Daniel-Constantin Mierla writes:
> It is ok, just add the documentation for the new function as well.
Done, Juha
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Daniel-Constantin Mierla writes:
> The function can be added, should be easy to extract the code from the
> rpc command function.
How about the tm.c patch below? Or should the function be placed in
some other tm/.c file and if so, which one?
-- Juha
---
***
Daniel-Constantin Mierla writes:
> There was an unsafe list iteration - can you try with:
>
> -
> https://github.com/kamailio/kamailio/commit/325a45e846faae3e5dfa333727d5fab294e44dac
>
> If all ok, you can backport.
Thanks for the patch. The crash has only happened once, so it is not
easy
Daniel-Constantin Mierla writes:
> If you can reproduce it, watch what the timer processes do during that
> time frame. Get the list of processes with 'kamctl ps', then when the
> issue is exposed, grab the backtraces of all processes with:
>
> kamctl trap
>
> A file is created with the
Daniel-Constantin Mierla writes:
> The rpc command tm.clean is like a last resort option when dealing with
> an unexpected situation that messed up the timer process, otherwise the
> transactions should be cleaned as they are expired or terminated because
> of final response sent upstream, with a
Here is some more details about the shm usage. Usage was steady at
about 10 MB until time 17:05. Then it usage started to steadily grow
during 15 min period and 17:21 hit 80 MB at which point tm.clean was
executed. After that usage dropped straight back to 10 MB level.
How is that possible,
Daniel-Constantin Mierla writes:
> The $stat(name) can be used for any internal statistic. I think it takes
> only the name of statistic, not the group, so something like
> $stat(free_size).
Thanks, free_size and the others in
core.shmmem
{
total: 67108864
free: 58274976
What kind of delay there is on cleaning of an expired transaction? I'm
asking, since tm.clean released lots of shm.
-- Juha
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John Petrini writes:
> The times we've seen this is when transactions are waiting on something so
> they pile up consuming shared memory. Do you have any database lookups or
> calls out to external services or scripts?
John,
Thanks for your reply. Yes, there are MySQL operations both during
K reported during about 90 sec period that it is out of shared memory:
Feb 28 09:47:28 rox1 /usr/bin/sip-proxy[19725]: ERROR: tm
[t_hooks.c:136]: insert_tmcb(): out of shm. mem
Feb 28 09:47:28 rox1 /usr/bin/sip-proxy[19725]: ERROR: acc
[acc_logic.c:394]: acc_onreq(): cannot register additional
Daniel-Constantin Mierla writes:
> Hello,
>
> I didn't know about limitation, so far I needed only set_rtpengine_set()
> with one parameter. Now I am wondering why won't work for
> rtpengine_manage() because internally it calls the
> rtpengine_offer()/_answer().
I have never used
So, what is the conclusion on this? It is not good if information about
failed branch route gets lost. Would it be possible to return some
internal error reply and make t_relay() always succeed?
-- Juha
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;Sent from mobile, with due apologies for brevity and errors.
>
>> On Dec 16, 2019, at 3:00 AM, Juha Heinanen wrote:
>>
>> Daniel-Constantin Mierla writes:
>>
>>> t_relay() should return negative (false) in such case, but I am not
>sure
>>> it ret
Daniel-Constantin Mierla writes:
> t_relay() should return negative (false) in such case, but I am not sure
> it returns a specific value for it -- this can be a variant to add if
> needed.
Yes, t_relay() returns false, but the branch flags I set in the branch
route are lost and I don't know
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