Hello,Various GUIDs and other signatures in SIP exist to link branches, transactions, and dialogs, but none that are uniquely generated per packet. The best you're going to be able to do is to compute a checksum,
thank you very much Daniel, GOD Bless you
i have done it successful by calling route msilo only in case method is
message.
one more thing is it possible that server send delivery message to the
sender, when it (kamailio server) sends offline message to the offline user.
--
View this message
Juha Heinanen writes:
> i have not been able to figure out, why i can't hear any audio when i
> call from webrtc client (using UDP/TLS/RTP/SAVPF profile) to sems echo
> server (using RTP/AVP profile):
>
> 1) webrtc client - UDP/TLS/RTP/SAVPF - rtpengine - RTP/AVP - sems echo server
i was able to
So im logging lots of information to Syslog to assist our helpdesk in
diagnosing customer issues.
Im seeing LOTS of user agents that re-use the sip Call-ID ( especially in
registers ).
What im wanting to be able to do is link a log message ( that I log at some
point in my dialplan ) to a specific
Hello,
I am forwarding an incoming call to a destination and on time-out forward
to another and them another.
I need to send a '181 Call Is Being Forwarded' to my SBC to reset the early
media stream when forwarding to the next destiation which I can do
with sl_send_reply("181","Call
Is Being Forwa
I started agian using
cd /usr/local/sbin/
./kamdbctl create
then
cd /usr/local/sbin/
./kamctrl
add user1 user1
1(11480) DEBUG: db_text [dbt_lib.c:289]: dbt_db_get_table(): cache or mtime
succeeded for [subscriber]
1(11480) DEBUG: db_text [dbt_base.c:232]: dbt_query(): new res with 1 cols
Hello Daniel,
I tried and it works! Thank you.
So, as suggested, I write $du=$null before call RELAY route.
Regards,
Igor.
-Message d'origine-
De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com]
Envoyé : lundi 18 mai 2015 16:42
À : 'Kamailio (SER) - Users Mailing List'
Objet : R
Hi Daniel, thanks for your prompt response. I read the module's GitHub readme
and there it says that it can be installed with 4.2:
Fetch kamailio code:
git clone --depth 1 --no-single-branch git://git.sip-router.org/kamailio
kamailio
cd kamailio
git checkout -b 4.2 origin/4.2
Add the modu
Hello Daniel,
I can try this. But there are cases where lookup is called and the
redirection to VoiceMail is working fine.
Could it be an issue with a missing "append_branch()" instruction?
Regards,
Igor.
-Message d'origine-
De : sr-users [mailto:sr-users-boun...@lists.sip-router.org]
Hello Dmitri,
Yes, the $ru is ok and contains the right domain name.
Regards,
Igor.
De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de
Dmitri Savolainen
Envoyé : vendredi 15 mai 2015 10:51
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] Issue
On 18.05.2015 15:41, Daniel-Constantin Mierla wrote:
> Hello,
>
> On 18/05/15 14:57, Klaus Darilion wrote:
>> Hi!
>>
>> Should log_prefix also work for xlog in reply routes? In my case it
>> doesn't (Kamailio 4.2.2)
> it should work for received sip replies.
Not here, neither in the default rep
cat > /usr/local/etc/kamailio/dbtext/subscriber
username(str) password(str) ha1(str) domain(str) ha1b(str)
user1:user1:xxx:10.1.1.1:xxx
9350) DEBUG: auth_db [authorize.c:498]: auth_check(): realm [10.31.8.1] table
[subscriber] flags [1]
2(9350) DEBUG: auth [api.c:96]: pre_auth(): auth: dige
Hi all,
I have successfully setup a Kamailio IMS environment using just a single
machine running three kamailio instances. I am using kamailio 4.2.4
compiling it myself. I followed this guide:
http://nil.uniza.sk/instant-messaging/simple/configuring-im-and-presence-kamailio-31-howto
to setup
Hello,
On 18/05/15 14:57, Klaus Darilion wrote:
> Hi!
>
> Should log_prefix also work for xlog in reply routes? In my case it
> doesn't (Kamailio 4.2.2)
it should work for received sip replies.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in
Hello,
you can find the db_text database schema in the source tree, at:
utils/kamctl/dbtext/kamailio/
I think kamdbctl will create it if you set the appropriate engine in
kamctlrc.
Apparently, the readme needs some updates.
Cheers,
Daniel
On 18/05/15 10:33, jaflong jaflong wrote:
>
> /usr/loc
Hi!
Should log_prefix also work for xlog in reply routes? In my case it
doesn't (Kamailio 4.2.2)
Thanks
Klaus
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/
Hello,
can you run with debug=3 and see if the function is actually executed?
Cheers,
Daniel
On 18/05/15 12:31, José Seabra wrote:
> Hello,
>
> I'm using the function sdp_remove_codecs_by_id from sdpops module in
> order to remove some codecs in INVITE request before send it to
> freeswitch, bu
Hi
Did you use msg_apply_changes() before relaying the INVITE
?http://kamailio.org/docs/modules/4.1.x/modules/textopsx.html#textopsx.f.msg_apply_changes
Regards,Dragos
From: José Seabra
To: Kamailio (SER) - Users Mailing List
Sent: Monday, May 18, 2015 12:31 PM
Subject: [SR-Users] Fun
Hello,
I'm using the function sdp_remove_codecs_by_id from sdpops module in order
to remove some codecs in INVITE request before send it to freeswitch, but
the function doesn't remove the codec, and it doesn't give any error
message.
I'm using this function in request route.
Kamailio version i
/usr/local/sbin/kamailio -V
version: kamailio 4.2.4 (x86_64/linux) be62bd
flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE,
USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC,
USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAIL
i have not been able to figure out, why i can't hear any audio when i
call from webrtc client (using UDP/TLS/RTP/SAVPF profile) to sems echo
server (using RTP/AVP profile):
1) webrtc client - UDP/TLS/RTP/SAVPF - rtpengine - RTP/AVP - sems echo server
if i call from the same webrtc client to bares
Hi Alex,
On Thu, May 14, 2015 at 5:47 AM, Alex Balashov
wrote:
> According to the rtpengine module documentation for rtpproxy_manage(),
> that's exactly what rtpproxy_manage() does:
>
>
>
> http://kamailio.org/docs/modules/4.2.x/modules/rtpengine.html#rtpengine.f.rtpengine_manage
>
> i.e.
>
> -
Hello,
the INVITE is sent out and the acc record is written, indicating that
the 200ok was received.
But logs show that you don't do anymore the RTPPROXY handling,
therefore, when clients are behind the nat, no voice.
You should call route(MSILO) only if the method is MESSAGE and let the
INVITE
Yes, that's the one, actually it was fix_nated_contact() that was commented out.
Thanks for the pointer to set_contact_alias() and handle_ruri_alias(). I see
them already in the cfg here so hopefully they've been used correctly. Tests
will tell.
From: Daniel-Cons
Hello,
if the password is correct set on the client, check also the realm
parameter and be sure that server and clients are using the same.
Cheers,
Daniel
On 17/05/15 13:45, Sanaii, Maziar wrote:
>
> Hello there,
>
>
>
> I’m trying to configure Kamailio 4.2.4 to authenticate client
> REGISTERs
Hello,
what is the command you start kamailio with?
Are those log messages you sent everything you see? Because it doesn't
get to complete start.
Cheers,
Daniel
On 17/05/15 13:44, Nahum Nir wrote:
> Hi All,
>
> I am using Kamailio 4.0.3 and can't logs.
> .cfg:
> #!define WITH_DEBUG
> ### LOG Le
Hello,
but fix_nated_sdp() is changing the SDP (body) not Contact header. Maybe
you had fix_nated_contact(). If yes, you should use set_contact_alias()
instead, along with handle_ruri_alias() -- see default kamailio.cfg for
version 4.2 for a sample of how to use these two functions.
Cheers,
Danie
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