HI Moacir
1.- Take a look to keepalived.org
2.- LCR Module or dispatcher could do what you look for
Best Regards
2016-06-05 16:41 GMT+02:00 Moacir Ferreira :
> Hi,
>
> I got two questions regarding high availability:
>
> 1 - Should my Kamailio server fail, I would like another Kamailio
> "box/
ing auth
Provider 2 -> Requesting auth
Thanks
2016-05-30 20:17 GMT+02:00 Juha Heinanen :
> Alberto Sagredo writes:
>
> > Actually im using
> ...
> > if(!next_gw()) {
> >
> > xlog("No hay GW de Backup");
> >
> > sl_send_reply(&
Hi
Im using LCR module and trying to handle 503/408 errors on provider to
route to a second one.
Actually im using
if(!load_gws(1, $rU, $var(caller_uri))) {
xlog("Gateways no disponibles");
sl_send_reply("500", "Server Internal Error - No gateway");
exit;
} else {
xlo
Hi Mikko
Thanks !
I will test and let you know
2016-05-09 9:21 GMT+02:00 Mikko Lehto :
> Alberto Sagredo :
>
> > Im trying to remove my Asterisk s= line on SDP but when doing:
> >
> > sdp_remove_line_by_prefix("s=Asterisk");
> >
> > Nothing happen
Thanks Both Fred/Juha
System is in production when i have a "free slot" i will check repeating
each rule with lower priority. I think that would work as i expect
Thanks for your tips guys!
Have a nice week
Alberto
2016-05-06 13:44 GMT+02:00 Juha Heinanen :
> Alberto Sagredo writ
HI cao
You would need to take a look to SDPs on INVITE and ACK to check , external
IP address are there in order to work properly. If that is the case, you
would need to check that packets are forwarded internally behind two NATs.
What RTP Proxy are you using? I highly recommend to use rtpengine
Hi
I have it working but i have re-read documentation and do not see how to do
what i need.
I explain it :)
Now i have only one LCR provider and i need to add a backup one.
I do not know if its enough to add under same lcr_id or its better to add
with different one and add several lcr_rule and
t; Ali
>
>
>
> *From:* sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On
> Behalf Of *Alberto Sagredo
> *Sent:* Thursday, May 05, 2016 2:24 PM
> *To:* Kamailio (SER) - Users Mailing List
> *Subject:* Re: [SR-Users] Kamailio - Sip_capture Module
>
>
>
Someone answered your partner Zainab about a question quite similar to
yours.
Take a look
Thanks
2016-05-05 11:35 GMT+02:00 Ali Taher :
> Hello,
>
>
>
> We are currently using kamailio sip_capture module, and then we need to
> create a Wireshark file (pcap format)from fields in the database.
>
Do you mean this?
http://www.kamailio.org/events/2014-KamailioWorld/day2/26-Daniel-Constantin.Mierla-Kamailio.cfg-Async.pdf
BR
2016-04-29 14:45 GMT+02:00 Safdar Khan :
> Hello,
> How i enable push notification in kamailio.How i proceed to configure the
> kamailio configuration file and which ka
;ra"); i need ip address like below
>
> mysql> select ip from ext where ext=202
> -> ;
> ++
> | ip |
> ++
> | 192.168.55.157 |
> ++
> 1 row in set (0.00 sec)
>
>
>
> On Thu, Apr 28,
Just a point . Its not needed to resend same email every day. Will try to
do the best with mailing list, but sometimes we are ill, occupied.. or just
do not know what to answer..
BR
2016-04-27 16:18 GMT+02:00 Shahid Mehmood :
> Dear All,
> i am getting error when i trying to fetch da
uot;,"valora");
sql_result_free("valora");
}
With no issues..
Any error on mysql¿
BR
2016-04-28 13:33 GMT+02:00 Alberto Sagredo :
> Hi
>
> Im my case
>
>
> modparam("sqlops","sqlcon","ca=>mysql://root:pass@IP/kamailio")
&g
Hi
Im my case
modparam("sqlops","sqlcon","ca=>mysql://root:pass@IP/kamailio")
2016-04-28 12:13 GMT+02:00 Shahid Mehmood :
> Dear All,
> i am getting error when i trying to fetch data from mysql data
> base using kamailio.
>
> i have following code into kamailio.cfg file
>
> loa
Hi Ankitt
Kamailio does not care about RTP. Just SIP Signaling
If you want to handle RTP between SIP Clients, depending of they are under
NAT or not, you would need RTPENGINE to proxy RTP packets between them.
Take a look how to install and configure rtpengine inside kamailio
(rtpengine module)
Hi Humberto
Take a look to RTPENGINE and rtpengine module in Kamailio.
Thats for RTP handling.
WITH_FREESWITCH allows in code to enable some parts to handle SIP correctly
agains Freeswitch. You should configure some values as FreeSwitch IP and
port
BR
2016-04-19 21:46 GMT+02:00 Humberto H :
>
Maybe it would be useful to use WITH_DEBUG and check /var/log/syslog for
detecting what is falling
Usually default script allows to call between extensions and register
without any issues.
Are you behind nat or locally?
BR
2016-04-14 7:11 GMT+02:00 Shiv Patidar :
> In kamailio server register
;>> Is Siremis supposed to work with Postgres and/or Nginx?
>>>
>>> On 13/04/16 17:24, Serge S. Yuriev wrote:
>>>
>>>> Hello,
>>>>
>>>> It's step 2 - Database configuration.
>>>> Latest GIT as mentioned before.
>>>
Have you taken a look to Homer project? http://sipcapture.org
Maybe you would find there what you are talking about.
BR
2016-04-14 17:48 GMT+02:00 Ali Taher :
> Hello,
>
>
>
> I’m using sipcapture module in Kamailio to capture sip packets and save
> them into mysql database.
>
> Everything i
HI Serge
When did you get that error?
I have recently installed on Debian 8 without any use. Use GIT version
solves some problems.
BR
2016-04-13 12:56 GMT+02:00 Serge S. Yuriev :
> Hello,
>
> I'm trying to setup latest Siremis from GIT and stuck on DB stage.
>
> ERROR: SQLSTATE[08006] [7] FA
o drop database location and install it using the sql statement from
> the file:
>
> utils/kamctl/mysql/usrloc-create.sql
>
> Cheers,
> Daniel
>
>
> On 07/04/16 22:20, Alberto Sagredo wrote:
>
> Im getting this error using USERLOCDB
>
> Apr 7 22:11:56 kamailio-
Im getting this error using USERLOCDB
Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR:
db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query:
Duplicate entry '0--1' for key 'connection_idx' (1062)
Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]
Hi Nelson
Im using rpid column to set Name as follows
if(avp_db_query("select rpid from subscriber where username = '$rU'",
"$avp(s:callerid)"))
{
uac_replace_from("\"$avp(s:callerid)\"","");
xlog(" Valor de RPID es : $avp(s:callerid)");
}
Hope this helps
Put tha
Just a point about using 302 redirect. It is expected to be handled by any
SIP device.. but maybe you could find some (I found them on Spectralink
DECT ) that could not handle 302 redirect and had to change way to use
dispatching for that devices..
BR
2016-03-28 22:16 GMT+02:00 Marrold :
> Well
Hi
Are you using ds_select_dst("1", "10"); somewhere ?
You have to check Asterisk is responding to OPTIONS as dispatcher
considered UP when 200 OK from Asterisk or SIP Gateway is received.. and
not if not received.
I have used dispatcher but not with 10 option on algo...
BR
2016-04-04 8:07
Take a look to LCR module and DIALPLAN module
BR
2016-04-04 6:43 GMT+02:00 li...@neulinx.com :
> hi all;
>
> I want to use kamailio achieve:
>
> First phase:
>
> 1) inbound caller number translation;
> 2) outbound caller number translation;
> 3) inbound callee number tran
Great. You can share with us how did you did it :)
Others would thank you also
BR
2016-04-04 15:22 GMT+02:00 NITESH BANSAL :
> Thanks guys for your ideas, I finally think that I have an idea on how to
> do it.
>
> Nitesh
>
> --
> Date: Fri, 1 Apr 2016 16:28:02 +0200
Hi!
I plan to use rpid field but when trying to get from database, subscriber
table has rpid field filled, i get always NULL.
Is there any more needed than:
modparam("auth_db", "load_credentials", "rpid")
?
Just a check of variable
xlog(" Valor de RPID es : $avp(s:rpid)");
Shows
Hi Federico. In mi case i have to force some RURIs to a determinate
Asterisk . There was some logic to do that instead letting kamailio to
handle that situations
I haven't tested option 3 , but it must do what you expect i think. Maybe
you would need to adjust
ds_hash_expire
Regards
2016-04-01 1
I have done something similar as follows
if($rU=~"^[0-3][0-9][0-9]$")
{
$var(valor)=1;
}
And later..
if(!ds_select_domain("$var(valor)", "4")) {
sl_send_reply("500", "Service Unavailable");
xlog("L_INFO","[$fU@$si:$sp]{$rm} Sin destinos disponibles
para $rd \n");
r
> requests, such as INVITEs/SUBSCRIBEs/etc.
>
> Cheers,
> Daniel
>
>
> On 31/03/16 10:57, Alberto Sagredo wrote:
>
> I have it working, authenticating in Kamailio. And just works.
>
> But there is a farm with lots of Asterisk and i was looking for an easy
> w
$fu or $au it may cause auth issues.
>
> On Thu, Mar 31, 2016 at 9:49 AM, Alberto Sagredo <
> alberto.sagr...@avanzada7.com> wrote:
>
>> Hi Marrold. I want to authenticate user on Kamailio but want Asterisk to
>> see them registered with kamailio ip as contact.
>&
ust relay the registration messages to Asterisk itself? What's
> the use case?
>
> Cheers
>
> On Thu, Mar 31, 2016 at 9:34 AM, Alberto Sagredo <
> alberto.sagr...@avanzada7.com> wrote:
>
>> Hi
>>
>> Im trying to modify example for Asterisk/Kamailio
Hi
Im trying to modify example for Asterisk/Kamailio integration that miconda
did, and trying to handle Forwarded register to Asterisk in an
authenticated way instead removing secret in asterisk.
I have made this changes
route[REGFWD] {
if(!is_method("REGISTER"))
{
Umm it was my fault :(
Sorry to bother you!
2016-03-22 16:36 GMT+01:00 Alberto Sagredo :
> I have installed Kamailio 4.4.x and im getting this error when trying to
> reload dispatcher?
>
> Any idea where to look?
>
> kamctl dispatcher reload
>
> 500 ERROR Reloadin
I have installed Kamailio 4.4.x and im getting this error when trying to
reload dispatcher?
Any idea where to look?
kamctl dispatcher reload
500 ERROR Reloading data
Also kamcmd dispatcher.reload gave me same result.
___
SIP Express Router (SER) and
You could find something related also on this link
Its in spanish
https://blog.irontec.com/integracion-kamailio-y-asterisk-con-path/
2016-03-01 11:25 GMT+01:00 Jurijs Ivolga :
> Hi,
>
> I would recommend you to take a look on path module:
>
> http://kamailio.org/docs/modules/1.4.x/path.html
>
I have just edited kamctl in order to make lcr commands work. Hope that
helps someone.
Using Kamailio 4.3.0
Please find kamctl attached also
### LCR management
#
lcr() *{*
*case* $1 *in*
show_gws)
# merr "command disabled"
#
rtised_address variable.
>> Removing advertised_address solved the issue.
>>
>> 2016-02-25 17:49 GMT+02:00 Alberto Sagredo > >:
>>
>>> :) Great
>>>
>>> So you will have maybe now something as this
>>>
>>> Record-Route:
>>>
25 17:02 GMT+02:00 Alberto Sagredo
> :
>
>> HI Alexandru i talk about something like this maybe in your RELAY route
>> or similar.
>>
>> I think you would have issues with ACKs until you would have
>> Record-Route: doubled
>>
>> if (dst
P_ADDR
>> alias = MY_INT_IP_ADDR
>> alias = MY_DOMAIN
>>
>> #!ifdef WITH_TLS
>> listen=MY_WSS_ADDR
>> #!endif
>>
>> port=5060
>>
>> ...
>>
>> # - rr params -
>> modparam("rr", "enable_full_lr", 1)
&g
Hi Alexandru
How is your configuration about Public IP and Private IP?
Do you use advertise?
Maybe you need to force Outbound traffic to Public IP Socket and inside
traffic to Private IP .
Do you have double record routing?
BR
2016-02-25 1:24 GMT+01:00 Alexandru Covalschi <568...@gmail.com>:
As Daniel mention, try to ngrep or tcpdump if messages are received or not.
Some Phones need to activate an special option to use DNS SRV instead A
registers
BR
Alberto
2016-02-22 10:10 GMT+01:00 Daniel Tryba :
> On Sat, Feb 20, 2016 at 03:08:00AM +, Derek Bolichowski wrote:
> > I am not a
relax their configuration
problems to your side.. Somes not, and SIP traces talks most of the times..
I just wonder if its possible, but seems with s= inside body not. Just
being curious while playing..
2016-02-24 15:52 GMT+01:00 Daniel Tryba :
> On Wed, Feb 24, 2016 at 03:22:46PM +0100, Albe
.org] *On
> Behalf Of *Alberto Sagredo
> *Sent:* 24 February 2016 14:03
> *To:* Kamailio (SER) - Users Mailing List
> *Subject:* Re: [SR-Users] sdp_remove_line_by_prefix
>
>
>
> s= MUST exist but thought remove.. will force to "s= "
>
>
>
> Any idea to
providers, sometimes
help you more :P
2016-02-24 15:11 GMT+01:00 Daniel Tryba :
> On Wed, Feb 24, 2016 at 03:02:32PM +0100, Alberto Sagredo wrote:
> > s= MUST exist but thought remove.. will force to "s= "
> >
> > Any idea to remove "Asterisk" traces
s= MUST exist but thought remove.. will force to "s= "
Any idea to remove "Asterisk" traces from SDP?
2016-02-24 14:49 GMT+01:00 Alberto Sagredo :
> Hi
>
> Im trying to remove my Asterisk s= line on SDP but when doing:
>
> sdp_remove_line_by_prefix("s=Ast
Hi
Im trying to remove my Asterisk s= line on SDP but when doing:
sdp_remove_line_by_prefix("s=Asterisk");
Nothing happens.
Loaded sdpops.so module as well.
Is there any additional needed?
BR
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-
l port:
>
> listen=udp:localip:5060
> listen=udp:localip:5080 advertise externip:5080
>
> Then you just need to force the socket to be used before sending out to
> use the right port and record_route() will do the rest.
>
> Cheers,
> Daniel
>
>
> On 24/02/16 09:
i plan to change in order to do not lose ACks .
Any idea to do it better ?
BR
2016-02-24 8:56 GMT+01:00 Daniel-Constantin Mierla :
> Hello,
>
> can you check in the config and see if there is any other record_route*()
> function used?
>
> Cheers,
> Daniel
>
>
>
Im fancing some issues due to kamailio behind nat and thought double record
route could help me .
But enabling that :
modparam("rr", "enable_double_rr", 1)
And later using
record_route_advertised_address("X.X.X.X:5060");
is giving me this error:
w_record_route_advertised_address(): Double at
modparam("dialog", "hash_size", 4096 )
Hope this would help to anyone.
2016-02-19 9:47 GMT+01:00 Alberto Sagredo :
> Looking into code is related to this previously error
>
> ERROR: acc [acc_cdr.c:852]: init_cdr_generation(): can't load dialog API
Looking into code is related to this previously error
ERROR: acc [acc_cdr.c:852]: init_cdr_generation(): can't load dialog API
2016-02-19 9:28 GMT+01:00 Alberto Sagredo :
> I have a strange behaviour when configuring ACC module to put CDRs
>
> It must be as simple as this:
>
I have a strange behaviour when configuring ACC module to put CDRs
It must be as simple as this:
modparam("acc", "cdr_enable", 1)
But got this error:
feb 19 09:26:14 kamailio /usr/local/sbin/kamailio[23712]: ERROR: acc
[acc_mod.c:604]: mod_init(): failed to init cdr generation
feb 19 09:26:14 k
Thank you!
That resolved my issue :)
2016-02-18 17:23 GMT+01:00 ycaner :
> There is parameter for lcr_id. you need to set max lcr_count in .cfg.
>
> lcr_count
> lcr_gw_count
>
>
>
> --
> View this message in context:
> http://sip-router.1086192.n5.nabble.com/LCR-Module-issue-tp145861p145863.ht
Hi
I have LCR working but not able to add more gateways?
It seems only first one is accepted
What could be wrong?
I show you database data and kamcmd commands output.
mysql> select * from lcr_rule_target;
+++-+---+--++
| id | lcr_id | rule_id | gw_id
x.
>
> This 4.2.6 was released for those running 4.2.x and are not intending for
> the moment to upgrade to 4.3.x, which may require config/db adjustments.
>
> Cheers,
> Daniel
>
>
> On 30/07/15 16:07, Alberto Sagredo wrote:
>
> Thanks. Upgraded to 4.2.6
>
> Is
Thanks Alex. Finally i made two kamailio.cfg depending on which case,
behind NAT or not, and advertise address to local when no NAT.
Will look for a clean solution later
Thanks for your attention
2015-07-29 10:25 GMT+02:00 Alex Balashov :
> Is it possible you have a default OS 'iptables' con
Thanks. Upgraded to 4.2.6
Is there any option to upgrade database and avoid problems due to database
versions?
i have been using reinit.. but did not see if there is any option for
upgrading
BR
2015-07-30 12:27 GMT+02:00 Daniel-Constantin Mierla :
> Hello,
>
> Kamailio SIP Server v4.2.6 stable
There is a dialplan module, maybe it Could help you in some topics
Best regards
El martes, 28 de julio de 2015, Nelson Migliaro
escribió:
> Hello everybody,
>
> I am currently using Kamailio in order to separate traffic based on dialed
> number.
>
> Some traffic goes to several Asterisk in a l
ov :
> On 07/29/2015 04:19 AM, Alberto Sagredo wrote:
>
> No. ACK seems to do not arrive to Kamailio
>>
>
> The 'tcpdump' filter I gave you won't show you that. It was designed
> specifically to capture packet flow between Kamailio and the phone.
>
No. ACK seems to do not arrive to Kamailio
I will review some things and let you know back.
2015-07-29 9:52 GMT+02:00 Alex Balashov :
> If you take a raw capture on the Kamailio box:
>
>tcpdump -i any -A -s 0 -n udp port 8002 and host PUBLIC_IP
>
> Does it not show Kamailio attempting to rel
cs without beeing answered
insted of answered call (No ACK for 200 OK issue) as expected
2015-07-29 9:44 GMT+02:00 Alex Balashov :
> On 07/29/2015 03:44 AM, Alberto Sagredo wrote:
>
> Could be Kamailio discarding this ACK In anyway?
>>
>
> It's possible, but have
Umm Alex
I see ACK sent to Kamailio from Asterisk, but not seen it send again.
Will take some more traces. and let you know back
Could be Kamailio discarding this ACK In anyway?
2015-07-29 9:40 GMT+02:00 Alex Balashov :
> Alberto,
>
> Based on the 200 OK you included, the end-to-end ACK from
Hi Alex
1.- Kamailio -> 172.26.101.50:8002 (Floating IP)
Asterisk -> 172.26.101.10:5080
2.- Transmitting (no NAT) to 192.168.0.170:8002:
ACK sip:110@IP_PUBLIC_IP:5066 SIP/2.0
Via: SIP/2.0/UDP 172.26.101.10:5080;branch=z9hG4bK0a330ae6
Route:
Max-Forwards: 70
From: "asterisk" ;tag=as14d7523
}
Any other recommendation is well appreciated
BR
2015-07-29 9:10 GMT+02:00 Alex Balashov :
> Alberto,
>
> On 07/28/2015 06:22 AM, Alberto Sagredo wrote:
>
> I have seen an issue with ACK as show when Asterisk sends ACK to
>> Kamailio, it sends to advertised Address i
I have Kamailio in a local network with RTPPROXY and NAT MANAGE
I have seen an issue with ACK as show when Asterisk sends ACK to Kamailio,
it sends to advertised Address instead to Kamailio IP address.
What could i add to solve this?
route[WITHINDLG] {
if (has_totag()) {
Hi
You would need to get a sip trace in order to see what happens.
Today i have forgotten my crystal ball at home ;)
BR
2015-07-28 14:38 GMT+02:00 goReeves Admin :
> Hi -
>
> I just installed Kamailio 4.3.0 on Ubuntu 14.04.2. I can connect using
> jitsi on windows and csipcimple on the andro
ribution)?
>
> http://www.kamailio.org/docs/modules/4.3.x/modules/dispatcher.html
>
> You will need to set a few parameters such as duid, hash_size, etc. and
> then utilize the special attribute of maxload.
>
> Then, using ds_load_update and ds_load_unset dispatcher will upda
same state in two different servers is looking for trouble.
>
> /O
>
> On 23 Jul 2015, at 11:16, Alberto Sagredo
> wrote:
>
> Sorry :)
>
> I found two scripts, the one on other mail worked fine
>
> per user limit is this one
>
> https://www.mail-archive.com/sr-u
Sorry :)
I found two scripts, the one on other mail worked fine
per user limit is this one
https://www.mail-archive.com/sr-users@lists.sip-router.org/msg07072.html
Best Regards :)
Any comment to improve it is welcome
2015-07-23 9:48 GMT+02:00 Alberto Sagredo :
> Hi
>
> Hi
Hi
Hi have read documentation but it seems dispatcher does not keep how many
calls has been dispatched or currently are in any of dispatcher destinations
I have take a look to code on:
http://lists.sip-router.org/pipermail/sr-users/2012-July/073919.html
But it seems to use calls limit per user,
+1 also here :)
2015-07-21 12:25 GMT+02:00 Alekzander Spiridonov :
> Agree
>
> 2015-07-20 22:18 GMT+03:00 Victor Seva >:
>
>> On 07/20/2015 08:58 PM, Daniel-Constantin Mierla wrote:
>> > My proposal is to move generation of self signed certificates to kamctl.
>> > There can be a kamctl.tls file
Using RTPBreak and sox im able to convert rtpproxy rtp files to wav. Later
you could convert to mp3 if liked
Just to share with list
https://github.com/albersag/rtpproxy-utils
Any comment is well appreciated
Best Regards
___
SIP Express Router (SER) a
quot;, nonce="5b30f8aa"
>> Content-Length: 0
>>
>>
>> <>
>> Scheduling destruction of SIP dialog '2ee5ec48557bba33-31464@ KamailioIP'
>> in 32000 ms (Method: REGISTER)
>> Scheduling destruction of SIP dialog '2ee5ec48
Take a look to :
http://kamailio.org/docs/modules/4.1.x/modules/sdpops.html
BR
2015-07-16 20:58 GMT+02:00 Alberto Sagredo :
> Hi Djamel
>
> As Daniel i do not understand what do you need.
>
> Kamailio does not care about codecs.. if you are not using RTP PROXY .
>
> You c
Hi Djamel
As Daniel i do not understand what do you need.
Kamailio does not care about codecs.. if you are not using RTP PROXY .
You could use sdops module to check things on SDP if it that helps
BR
2015-07-09 11:30 GMT+02:00 Djamel Bahamid :
> Hi,
>
> I have terminals of video conference wh
You could put several kamailio on cascade.. but you have to be careful with
via routes .
You could add to request_route { a route to diver all calls to other
kamailio
Having two location databases is not a good idea, maybe you could use
LOCATION on Database and have same user database for both
gt; confidential and exempt from disclosure under applicable law. If the reader
> of this message is not the intended recipient, you are hereby notified that
> any dissemination, distribution or copying of this communication is
> strictly prohibited. If you have received this message in error,
You could remove secret= on extensiones to check if its related to
authentication or not
You must not request authentication to kamailio in order to work properly
in front of Asterisk
As Daniel mention check if Kamailio peer is created and extensiones have no
secret.. you would need to add altern
t();
}
}
#!endif
return;
}
2015-07-15 16:59 GMT+02:00 Roberto Fichera :
> On 07/15/2015 08:44 AM, Alberto Sagredo wrote:
>
> Hi Alberto,
>
> can you also share part of the relevant place where you are calling that
> route?
>
> Cheers,
> Roberto Fichera.
>
>
You are welcome!
2015-07-15 15:42 GMT+02:00 Daniel Tryba :
> On Wednesday 15 July 2015 08:44:13 Alberto Sagredo wrote:
> > Kamailio is for hard people and fun :)
>
> No comment on this one
>
> > Thanks Visily i finnaly got it working with your tip. You were ri
#!endif
#!ifdef WITH_RTPENGINE
rtpengine_manage("external internal
replace-origin replace-session-connection ICE=remove RTP AVP");
#!endif
}
}
}
}
Hi daniel
In my tests rtpproxy recording waste less resources than asterisk
That was one of the reasons
BR
ALBERTO
El martes, 14 de julio de 2015, Daniel Tryba escribió:
> On Tuesday 14 July 2015 17:01:38 Alberto Sagredo wrote:
> > What i try is to detect if i have SAVP from end
Ganchev :
> Alberto Sagredo-2 wrote
> > Thanks Vasily i have changed a little today using a RTPPROXY route.
> >
> > Thats what i have right now
> >
> > But its not working as expected
> >
> > What i try is to detect if i have SAVP from endpoint and
set_rtp_proxy_set("1");
rtpproxy_manage("fwie");
start_recording();
#!endif
#!ifdef WITH_RTPENGINE
set_r
rovide the sdp bodies for both Grandstream (that matched) and
> Yealink (that didn't match). We have to compare how the SAVP is advertised
> and how the function is making the check.
>
> Cheers,
> Daniel
>
>
> On 08/07/15 16:45, Alberto Sagredo wrote:
>
> Im using
Thanks Daniel!
2015-07-14 13:11 GMT+02:00 Daniel Tryba :
> >If i turn STUN off as Olle mentioned, all worked fine.. but if i have a
> >customer that for error it turns ON it does not work in my case..
>
> > Any way to force RTP in that cases?
>
> Just always start the rtpproxy, especially when
customer that for error it turns ON it does not work in my case..
Any way to force RTP in that cases?
BR
2015-07-14 12:07 GMT+02:00 Daniel Tryba :
> On Tuesday 14 July 2015 11:56:38 Alberto Sagredo wrote:
> > I have found an issue detecting NAT of GS 1.4.23 Phones when using Stun
>
I have found an issue detecting NAT of GS 1.4.23 Phones when using Stun on
this phones
Usual nat_uac_test with numbers 19 and 3 does not seems to detect is behind
nat so NATMANAGE is not called
Here its some trace. ANy clue how to handle this phones if they activate
STUN for example
U 80.26.x.x
Hi
Im using an Asterisk cluster behind Kamailio and some services are running
on Asterisk. Traffic always must go to Asterisk so RTPPROXY and RTPNGINE do
the work and send traffic to Asterisk and traffic from Asterisk go to RTP
Proxy
I have been able to get rtpproxy working and later tried RTPeng
Im using if(sdp_with_transport("RTP/SAVP")) to detect with endpoint is send
SAVP or not to divert to and rtp proxy or rtpengine, as you know rtpproxy
supports recording and rtpengine does not yet.
So when using if(sdp_with_transport("RTP/SAVP")) with Grandstream Phones
all worked fine, but when c
Kamailio is not a Radius server.. Can use them.
Kamailio is other "THING"
2015-07-08 1:20 GMT+02:00 Ali Taher :
> Hello,
>
> I'm wondering if kamailio can act as radius server who's responsible of
> authentication,authorization and accounting, where kamailio receive
> accounting record types (st
7-07 12:02 GMT+02:00 Alberto Sagredo :
> Hi
>
> I have a Kamailio in front of an Asterisk Cluster and moved to rtpengine
> to give support for SRTP from rtpproxy
>
> What i want is to have SRTP on Kamailio-rtpengine leg but RTP inside
> network for Asterisk.
>
> I have found
Hi
I have a Kamailio in front of an Asterisk Cluster and moved to rtpengine to
give support for SRTP from rtpproxy
What i want is to have SRTP on Kamailio-rtpengine leg but RTP inside
network for Asterisk.
I have found that config
https://loadmultiplier.com/sites/default/files/kamailio/pcscf/kam
Hi Fadi
As Daniel and Loic commented .
Create users using kamctl or adding with a script to database mysql (if you
are using it) and you could use SIPP
Take a look to :
http://www.sipfish.com/blog/generating-voip-traffic-with-sipp/
BR
2015-07-06 9:28 GMT+02:00 Loic Chabert :
> Hello,
>
> Yes
elect * from user_role;
Empty set (0.00 sec)
I get a splash screen saying and error but it changes to next screen on
wizard
I have this version
Using Debian 8.1
Any log where i could see what is going with wizard?
BR
2015-07-03 13:35 GMT+02:00 Alberto Sagredo :
> Thanks Daniel
>
> I d
n wizard for creating the siremis database content. You can redo
> the installation wizard, remove the install.lock file -- see the
> installation docs for more details.
>
> Cheers,
> Daniel
>
>
> On 03/07/15 12:09, Alberto Sagredo wrote:
>
> Umm I have checked mysql table
Umm I have checked mysql tables and are empty for siremis
mysql> use siremis;
Database changed
mysql> show tables;
Empty set (0.00 sec)
mysql>
So maybe wizard is not installing sql on it? Any way to make it manually?
2015-07-03 12:04 GMT+02:00 Alberto Sagredo :
> Hi
>
Hi
Im installing Siremis 4.2 and getting this error when login. INstallation
Wizard was fine and all dependencies are installed following Asipto Guide
for Siremis.
Any idea what could be wrong?
Popup i get is..
{"target":"ERROR","content":"
\n[2015-07-03 10:04:05 (GMT)] An exception occurred wh
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