Just to compare.
This call seems to be OK.
Another order of log entries.
«using iLBC/8000» 2 lines.
And an «Average MOS» results (no such like in previous log).
10.45.145.33 — asterisk
10.45.144.69, 195.209.XXX.YY — opensips/rtpengie (int, ext)
217.66.158.139 — cell phone provider nat box
11
Hi list,
I would like to discuss RTPEngine logs,
to know your opinion if such log is normal or not.
The client complained that haven’t heard anything during the call.
10.45.145.33 is Asterisk
195.209.XXX.YY is OpenSIPS + RTPEngine
217.66.157.207 is cell phone network from which an UAC connects
Hi Johan,
sure I can
maybe you were confused because the ‘ @ ‘ symbol
had been substituted in e-mail by ‘at’ word
I’ll try to post SQL insert once again, let’s have a look
what will happen
INSERT INTO registrant (registrar, aor, username, password, binding_URI,
expiry) VALUES (‘sip:11.11.1
Hi Johan ,
your SQL insert will look like this:
INSERT INTO registrant (registrar, aor, username, password, binding_URI,
expiry) VALUES (‘sip:11.11.11.11’, ’sip:79993332...@voip-isp.com’,
’79993332211’, ’PaSsWoRd’, ‘sip:79993332211@22.22.22.22’, 300);
where:
- 11.11.11.11 is VoIP ISP IP ad
Hi list,
is it normal at all — to authenticate BYE requests?
I’ve checked RFC 3261 now, but haven’t found anything corresponding to this.
Neither if it’s normal, nor if it’s not normal.
---
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Hi list.
I’m looking for a solution how to set caller id from a pre-defined list of
phone numbers _randlomly_ .
So, there is some number of caller ids allowed by our VoIP provider for
outgoing calls.
And I need to set it randomly, from the list.
I know the solution how it can be achieved i
Hi Ali,
overall, you need to configure auto-scaling [1][2] and the event interface [3]
usage.
I never tried fetching exactly this event, but I think you haven’t configured
event_route usage [4][5].
Have a try something like this:
event_route[E_CORE_PROC_AUTO_SCALE] {
xlog(" … some
Hi Mehdi,
onreply_route [1] is used not for generating replies by your OpenSIPS,
it it used for actions to be taken for SIP replies which are going towards
OpenSIPS
from other VoIP entities.
To drop a call with some code you may use these functions:
sl_send_reply(); [2]
send_repl
Hi list,
a small contribution to the community,
not as a yum-repository yet, but still.
https://alexeyka.zantsev.com/rpm/
Often creating rpm packages of RTPEngine
becomes a headache, so I decided to share.
Hope this will be useful!
---
BR, Alex
Hi Stefan,
as I understood, you use CentOS, so
this command should be exactly what you need:
yum --showduplicates list available opensips*
You may also browse repository:
https://yum.opensips.org/3.2/releases/el/7/x86_64/
https://yum.opensips.org/browse.php
-
Hello Stefan,
as we see, the packages are available from 2 repos in your system —
an old version from ‘epel’ and an up-to-date one from ‘opensips’.
It can be useful to run such command to list available packages
and repositories they belong to:
yum --showduplicates list available opensip
Hi,
what if to try to use some custom t_on_reply route ?
Something like this:
if (is__method("REGISTER")) {
xlog("L_INFO", "forwarding REGISTER to main registrar...\n");
$ru = "sip: 10.0.0.3:5070 ";
t_on_reply("main_reg_replies");
if (!t_relay()) {
send_reply(500, "
Hi list.
What do these errors mean?
This started after migrating the virtual machine and changing IP addresses.
Everything else remain unchanged.
Version 3.2.2
ERROR:mid_registrar:unregister_record: 'from' key not found, skipping
De-REGISTER
ERROR:mid_registrar:mid_reg_aor_event: failed to
Hi Sasmita,
if there’s not too much IP addresses (not hundreds or thousands) in your
dr_gatewas table,
you still can add them to the config file and re-read it with reload_routes [1]
function.
There should not be service interruption according to the documentation.
So, you may try to add tru
Hi Saurabh,
not clear from your message what routing table do you mean —
either Linux kernel routing table or drouting [1] module.
Anyway, you may use ratelimit [2] module or pike [3] modules
specifying the IP addresses as the source address to check/not to check
the requests coming from it.
I
Discovered that script_helper.so is incompatible with topology_hiding.
(Simply because there is no place to use topology_hiding_match
function in case of script_helper.so usage — because
when using script_helper.so, we don’t add to the script the section with
if(has_totag()) ).
So, I fixed my e
The reason was that i executed route[relay] in the wrong place
of the script.
Like this:
…
…
route(relay); # I added this, and it was a mistake (too early)
…
record_route; # default, but calls haven’t reached this line
...
route(relay);
---
BR, A
Time to try the script_helper module has come :D
Now works as expected https://ibb.co/6D3R0fJ
Thanks everybody for advice!
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Hi list,
feeling shamefully but still can not fix the problem, trying different
configurations.
I’m setting drouting (INVITEs are routed well).
But BYEs from UAC go directly to dr_gateway, not through OpenSIPS.
I understand that this all is about in-dialog request, having to_tag,
loose_rou
If I also enable profile replication, MI command shows that it’s synced
successfully,
with the same settings as for other types of replication.
But the dialogs and dr_gateways status is not synced, according to MI output,
and I can not understand what’s wrong with the configuration.
Why somethi
I changed a bit according to ‘active-backup’ scheme, but still the same.
### active node
modparam("clusterer", "sharing_tag", "vip/1=active")
### backup node
modparam("clusterer", "sharing_tag", "vip/1=backup")
And this for both nodes:
if (is_method("INVITE")) {
create_dialog(
Hi list,
can not find the reason why MI ‘clusterer_list_cap’ command
shows that the dialog replication is not active.
It’s necessary to mention that profiles synced successfully,
when I configured it with the ‘profile_replication_cluster’ parameter (later I
removed this due to documentation [1
Hi Shah Hussain,
maybe this will be useful to you:
https://www.opensips.org/Documentation/TipsFAQ#toc2
http://controlpanel.opensips.org/htmldoc_8_X_X/user_management.html —
‘Password mode’ section.
---
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Hello,
maybe the second parameter (‘contact’) of the ‘is_contact_registered’ function
[1] could be useful?
And speaking about Expires: 0 — according to SIP RFC [2] it’s absolutely normal,
it means tat a device disables its registration, this is done either by sending
Expires: 0 SIP header,
or
Nice article, but nothing about RTPEngine High Availability ;)
Storing RTPEngine data in 2 Redis instances (RW and RO) by each OpenSIPS node
is the way to keep active calls going on in case of any problems with some of
nodes.
---
BR, Alexey
http://a
Hello,
the regexp matching digits will look like \d+
(backslash, d letter, plus sign).
Maybe this can be also useful to you:
https://www.opensips.org/Documentation/Script-Tran-3-1#toc6
---
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__
Hello Julio,
there was an article [1] in OpenSIPS Blog which may be helpful.
[1]:
https://blog.opensips.org/2019/02/25/auto-process-scaling-a-cure-for-load-and-resources-concerns/
---
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Hi all,
I’m sure there are VoIP engineers from South America (Brazil preferably) among
us.
If anybody knows where it’s possible to find VoIP equipment of
Yeastar/Neogate/Leucotron
vendors, please send me an e-mail.
I’m located on the other continent and it’s not so easy to find what I need
Hi Bogdan,
thank you for the advice!
Moreover, I had a mistake: I need to use && rarther than ||
I fixed it, and now only valid INVITEs are allowed
(either from VoIP ISPs from address table, or from registered users):
# antiflood
if(!is_myself("$si") && $Rp == 5060) {
Hi list,
OpenSIPS 2.4.8
is it possible to use is_registered() finction
if the registered users are saved by mid_registrar module,
and forwarded further to the main registrar server?
mid_registrar saves users’ location well, to the «location» table,
‘opensipsctl fifo ul_dump’ shows the list o
- also added ‘t_newtran();’ for each incoming REGISTER.
We’ve also noticed that CPU utilization remained high when
restarting OpenSIPS under high load.
CPU utilization became normal when 1) dropping sip traffic with firewall,
2) then restarting OpenSIPS (with no load), and 3) accepting sip tra
The problem seems to be fixed after 2 changes:
- adding ‘use_children 8’ to both listeners — external and internal
(from which the $fs is done)
- doing $fs="udp:10.229.3.33:5070"; after ‘mid_registrar_save’,
(earlier that was done before it, in one of cases in script).
Now the CPU utilization
Hi list,
CentOS 7.7.1908
OpenSIPS 2.4.8 x86_64
2 OpenSIPS processes use almost 100% of CPU.
I see it in top.
This load is always generated by the same listener/interface:
Process:: ID=17 PID=44694 Type=SIP receiver udp:10.229.3.33:5070
Process:: ID=18 PID=44695 Type=SIP receiver u
Hi Sasmita,
isn’t the ‘n’ option of the function what you need?
https://opensips.org/docs/modules/2.4.x/drouting.html#func_is_from_gw
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Hi Sasmita,
I checked one of my deployments and see this:
loadmodule "auth_db.so"
modparam("auth_db", "calculate_ha1", 0 )
modparam("auth_db", "password_column", "ha1")
modparam("auth_db", "db_url",
"mysql://dbu_opensips:pAsS@10.10.10.10/db_opensips")
modparam("auth_db", "use_domain", 1)
Th
Start with creating the ‘version’ table [1],
as it is used when creating any other table.
[1]
https://github.com/OpenSIPS/opensips/blob/3.0/scripts/postgres/standard-create.sql
---
BR, Alexey
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Hi Ryan,
do you need to create the database exactly with Opensips cli?
Another way is to do it using a DB cli (SQL command).
This is for 3.0 and PGSQL:
https://github.com/OpenSIPS/opensips/tree/3.0/scripts/postgres
---
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Hi, Răzvan
Will it be a kind of alternative for RTPEngine?
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Yes, it’s my fault.
I re-checked the script.
mid_registrar is used in the other section, not in those one,
which processes the REGISTER requests I watched using sngrep.
Excuse me for the wrong information.
---
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http://alexeyka.zantsev.com
Hmm.. I checked our configs and saw that REGISTER messages go
through OpenSIPS (mode 2) and ip address in Contact header
remains unchanged...
>Hi, Social Boh!
>
>When using mid-registrar, the INVITE flow will be:
>
> INVITE INVITE INVITE INVITE
>Alice > MID-RE
>Well, let’s check out in which mode is mid_registrar configured?
---
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>
It is absolutely correct.
And that’s your job, as a VoIP adminitsrator, to forward this INVITE
to the appropriate server.
Any SIP client will send INVITE to the server defined in its settings,
at least if something additional is not configured («Outbound proxy»
or «Route» or maybe it’s called
Even more, speaking about SIP protocol and REGISTER requests in general,
it’s worth saying that you can specify some ‘third-party’ address with which
the Contact: header of your REGISTER will be filled, and the INVITES for that
endpoint will be sent to that address and not to the device which in
excuse me please, wrong thread...
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Hi, Social Boh !
It’s absolutely OK, because when mid-registrar
re-send a REGISTER, it does not change the IP address
in the Contact: header.
So, when the ‘main server’ sends an INVITE to the end-point,
it sends it not to the mid-registrar, but to the IP address
being pointed in the Contact: h
Hi, Social Boh !
It’s absolutely OK, because when mid-registrar
re-send a REGISTER, it does not change the IP address
in the Contact: header.
So, when the ‘main server’ sends an INVITE to the end-point,
it sends it not to the mid-registrar, but to the IP address
being pointed in the Contact: h
I also configured both OpenSIPS nodes to use the same MongoDB:
modparam("usrloc", "cachedb_url",
"mongodb://1.1.1.1:27017/opensipsDB.userlocation")
and the result was the same:
— REGISTERs were handled fine;
— calls were successful
— ‘ul_dump’ still showed nothing.
Hi list.
I have a 2.4 full-sharing-cluster and trying to
make it as redundant as possible.
So I configured
modparam("usrloc", "working_mode_preset", "full-sharing-cachedb-cluster")
and separate MongoDB to each OpenSIPS node:
# node 1
modparam("usrloc", "cachedb_url",
"mongodb://
Hi Steve,
thank you.
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Hi list,
I’m using 2.4.6 in a full-sharing cluster.
Everything is OK but the ‘seed_fallback_interval’ [1] parameter.
When I configure it as in the docs, OpenSIPS can not start and
shows an error:
ERROR:core:set_mod_param_regex: parameter not found in
module
CRITICAL:core:yyerror: parse er
Hi list,
I’m using 2.4.6 in a full-sharing cluster.
Everything is OK but the ‘seed_fallback_interval’ [1] parameter.
When I configure it as in the docs, OpenSIPS can not start and
shows an error:
ERROR:core:set_mod_param_regex: parameter not found in
module
CRITICAL:core:yyerror: parse er
Hi Mark,
have you tried to achieve this with rtpengine_offer/rtpengine_answer
functions?
It will rewrite the o= and c= SDP fields.
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Hi guys,
as we see, there are also 2 Content-length headers in the incoming to OpenSIPS
INVITE.
I haven’t found anything regarding the number of reoccurrences of this
header in the SIP message [1].
What about measuring the length with Wireshark, to determine the
position of this header with
Hey Artiom,
the main difference in call-flow between federated and full-sharing
architecture is as follows:
Federated cluster:
++++
Hey Liviu,
thank you for the informative answer!
By the way, I configured a full-sharing usrloc cluster
and it seems to be what I need.
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Hi, solarmon
no, there’s no need to sync dialogs. It’s another type of cluster.
As for now, the main purpose is just to have a common
user location dataset, when OpenSIPS nodes will know
the location of each registered device. If it’s connected locally,
it will serve the call locally. If the d
Hi list
I’m trying to find the best solution for setting
federated user location cluster.
Key notes:
- geographically distributed OpenSIPS nodes
- the same numbering plan for all devices (no strict
ranges for Europe/Asia and so on)
- ability to be in service (at least inside of each node)
in cas
Hi Răzvan,
thank you.
So, 2.4 will be the best choice as it it LTS.
>Hi, Alexey!
>
>Most likely clustering 2.4 and 3.0 versions would not work, due to
>different replication packets versions.
>
>Best regards,
>Răzvan
---
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http://alexeyka
Hi list,
is it a good/bad idea to create a cluster of 2.4 and 3.0 versions?
Or it’s of vital importance to have the same version of instances in the
cluster?
The idea is to create a federated user location cluster. [1]
[1]
https://opensips.org/Documentation/Tutorials-Distributed-User-Loca
Also tried to delete codecs in branch_route, but still no success.
Well, enough flooding:)
Maybe somebody has an idea how to remove codecs.
---
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Now I'm testing with this snippet:
$var(aline_1_before) = $(rb{sdp.line,a,0});
xlog("Route: $rT . 1 a line in the SDP body is $var(aline_1_before)\n");
$var(aline_2_before) = $(rb{sdp.line,a,1});
xlog("Route: $rT . 2 a line in the SDP body is $var(aline_2_before)
Maybe I need to use this function in some other
place of the script?
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Hi Bogdan,
in fact, I'm just playing with this and have no real need (at least right now).
But I'm curious why the function is not working properly for me.
Script debugging shows smth strange about SDP presence:
DBG:sipmsgops:create_codec_lumps: creating 1 streams
DBG:sipmsgops:get_associa
Dominic,
so, you use the same weight of gateways for random selection, right?
>i tried it it out and looks to be working (random).
>thanks for the help Ben
---
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Hi Ben,
does it mean that the gateways will be selected in random order if they have
the same weight?
>You just need to set the carrier flag to use weights for routing and define
>the list of gateways with their weights. It’s described in the Overview
>section of the dr_routing module documen
Hi Sasmita,
in spite of some python-related warnings, your commands are working (running
3.0).
Try like this:
root@vds1261:~# opensips-cli -x mi get_statistics tm:
/usr/local/lib/python3.5/dist-packages/sqlalchemy/sql/functions.py:68:
SAWarning: The GenericFunction 'array_agg' is al
Hi Mark,
I haven't understood well enough what data do you store in the table.
Something designed by OpenSIPS developers? E.g. 'subscribers' table.
Or some custom data which you use in your script say via AVPs?
If it's possible to store it as key-value pairs, I'd recommend you using
local cache
Hi list
Trying to delete all codecs except one.
No success.
route[relay] {
if (is_method("INVITE")) {
t_on_branch("per_branch_ops");
t_on_reply("handle_nat");
t_on_failure("1");
}
script_trace( 1, "$rb(application/sdp)", "t
Hi John!
>i am trying for some time now to integrate Opensips with Asterisk,
> but without success. I have seen the links to the Opensips blog for
> Asterisk integration, but it is outdated both for Opensips and Opensips.
Well, what confused you in this tutorial? It seems to be what you need:
htt
Hello,
isn't it better to consider something from these modules:
- DROUTING [1]
- LOAD_BALANCER [2]
- CARRIERROUTE [3]
1 https://opensips.org/html/docs/modules/3.0.x/drouting.html
2 https://opensips.org/html/docs/modules/3.0.x/load_balancer.html
3 https://opensips.org/html/docs/modules/3.0.x/car
Hi, Răzvan,
now clear, thank you.
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Accroding to logfile,
ban time is 5-6 seconds instead of configured 20.
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Hi list,
Server: OpenSIPS (3.0.0 (x86_64/linux)) nightly.
Why does the ban time (accroding to logs) differ from that from config
parameters?
Config:
modparam("pike", "remove_latency", 20)
Log:
Jul 24 12:38:03 vds1261 /usr/sbin/opensips[21658]: PIKE - BLOCKing ip
195.154.177.142, n
hmmm...
but the output seems a litle bit strange
well, anyway, checking data via SQL is also appropriately.
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Oh, sure. Knew about this some time ago and then forgot.
Thank you Liviu.
(opensips-cli): mi subnet_dump partition=default
part: default
Destinations:
grp: 0
ip: 255.255.0.0
port: 0
proto: any
pattern:
context_info:
grp: 0
ip: 255.255.0.0
port: 0
proto:
Hi Liviu,
i installed -nightly release, now I can see 'address_dump',
but is shows only one row from several:
root@vds1261:~# opensips-cli -x mi address_dump
Partitions:
name: default
Destinations:
grp: 0
ip: 195.209.116.4
mask: 32
port: 0
proto: a
Hi list,
the question regarding OpenSIPS-cli for 3.0 version.
I hope somebody can help me here, even as the project is a stand-alone one.
I can not execute 'address_dump' MI command - it needs
a 'partition' option. I'm doing everything accroding to the documentation [1]
but all in vain:
(ope
Being more accurate, it's worth mentioning the following:
* SIP-accounts:
le...@alexeyka.zantsev.com
lex...@alexeyka.zantsev.com
lex...@alexeyka.zantsev.com
* register.deny file contents:
ALL : ALL
* regi
The night brings counsel.
For "Deny all except ... " policy -
register.deny file contents:
ALL : ALL
register.allow file contents:
"^sip:user@alexeyka\.zantsev\.com$" : ALL
And the script:
if (is_method("REGISTER"))
{
if(!allow_register("register")) {
sl_send_re
Hello Pavel,
nice approach, never used regex module,
just read about it.
Thank you for an advice.
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Hi list,
is it possible to filter REGISTER requests with permissions.so [1] module,
based on username?
It's written " Main purpose of the function is to prevent registration of
"prohibited" IP addresses. " When speaking about IP filtering,
I'd rather use check_address or check_source_address func
> Values with timeout still existed.
I mean they exist the period of time defined as the timeout parameter.
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As for now, more than 24 hours passed, the local cache value is OK
(haven't been cleared). So, I don't know what is this - when it flushes.
And only permanent value have been flushed.
Values with timeout still existed. Strange...
---
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http:/
Checked the value via fifo command after 2h 20 min -
it's empty.
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Hello Anton,
you'd better show here the load_balancer part of your config,
as it has some parameters regarding to working in the cluster.
>Понедельник, 27 мая 2019, 17:17 +05:00 от Антон Ершов :
>
>Hello friends!
> I'm testing new features of opensips 3. And I discovered the following
> situati
Hi list,
I'm using 2.4.5 and noticed that local cache [1] value
expires suddenly, even if I don't set the timeout.
This is my startup_route config:
startup_route {
cache_store("local", "incoming:ratelimit", "15");
}
I checked this setting with "opensipsctl fifo cache_fetch local
It should work.
A conference consists of the same SIP requests and replies
as any other SIP call.
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Hi Konrad !
Will the conference be created on the Asterisk/FreeSWITCH ?
I'm sure that in this case all calls which go through OpenSIPS
will be absolutely identical like any other calls.
I have a HA cluster (active-backup scheme), where OpenSIPS
is a frontend and there is an Asterisk behind. I ca
Hi Bogdan,
that would be nice, thank you!
Tell me please where can I find some more info
about the output of 'fifo rl_list', besides
https://opensips.org/html/docs/modules/2.4.x/ratelimit.html#mi_rl_list ?
root@deb-osips1:# opensipsctl fifo rl_list
PIPE:: id=pipe_111 algorithm=SBT limi
Hi Ben!
Sure, that was the clue! Thank you.
NOT working:
cache_fetch("local", "incoming:ratelimit", $var(rl));
xlog("L_NOTICE", "cfg_line $cfg_line ; var(rl) is $var(rl)");
if (!rl_check("pipe_$rU", "$var(rl{s.int})", "SBT")) {
NO
Hi list !
The idea is to store limit value in local cache, to be able to change it without
OpenSIPS restart. But when I store it in local cache, it is not applied
to 'limit' parameter of rl_check [1] function. Though it is fetched
successfully.
[1] https://opensips.org/html/docs/modules/2.4.x/r
And warnings related to memory...
WARNING:core:init_reactor_size: shrinking reactor size from 262144
(autodetected via rlimit) to 13107 (limited by memory of 10% from 4Mb)
WARNING:core:init_reactor_size: use 'open_files_limit' to enforce other limit
or increase pkg memory
-
Guys helped me to find, but still any ideas are welcome
https://opensips.org/pipermail/users/2008-December/001872.html
---
BR, Alexey
http://alexeyka.zantsev.com/
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Users mailing list
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I noticed also that these errors appear in log suddenly, maybe this is the cause
of OpenSIPS restart? But I haven't examined SIP traffic yet. Just watch the
logs.
ERROR:core:forward_reply: no 2nd via found in reply
---
BR, Alexey
http://alexeyka.zan
I noticed also that these errors appear in log suddenly, maybe this is the cause
of OpenSIPS restart? But I haven't examined SIP traffic yet. Just watch the
logs.
ERROR:core:forward_reply: no 2nd via found in reply
---
BR, Alexey
http://alexeyka.za
Hello John,
have you read this tutorial?
https://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8
I hope it has a plenty of information you need.
---
BR, Alexey
http://alexeyka.zantsev.com/
__
Hi list
Today I noticed how OpenSIPS 2.4.5 stopped
responding to INVITE messages from upstream VoIP provider.
fifo commands were haven't worked either.
I restarted OpenSIPS and everything became OK.
Not sure if this is the cause of a problem, but found this in OpenSIPS log:
Apr 21 13:23:55 vo
Hello,
not sure, but maybe some of these:
EXEC - https://opensips.org/html/docs/modules/2.4.x/exec.html
JSON - https://opensips.org/html/docs/modules/2.4.x/json.html
---
BR, Alexey
http://alexeyka.zantsev.com/
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