Hello Everyone,
How to correct message 484
Is need use txt module to fill string with correct information ?
<--- SIP read from UDP:192.168.100.145:5060 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6
From: "asterisk" ;tag=as0a530a8d
To: ;tag=
humm,
When datagram server is initialized in method "mi_init_datagram_server", it
set tx_sock equal to rx_sock, instead of creating a new reply socket. So
that when an async command is received in method "mi_datagram_server" with
tx_sock reference, i pass it to "build_async_handler" method which i
Hello Daniel,
After looking on debug I have question, about this debug
9(2046) DEBUG: [parser/parse_addr_spec.c:893]: parse_addr_spec(): end of
header reached, state=29
9(2046) DEBUG: [parser/msg_parser.c:190]: get_hdr_field(): DEBUG:
get_hdr_field: [82]; uri=[sip:1...@networklab.loc;tran
Daniel,
Following up.
Thanks
ve
On 03/27/2014 04:44 PM, i...@vintageelectronics.ca wrote:
Daniel,
I found your writeup at
http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour
and tried to follow it.
Jitsi would not connect even though it is running in the same box as
Kama
Hello Daniel,
Please see screen shot,
https://www.dropbox.com/s/zot54rs9d13dthy/loop1.png
Slava.
- Original Message -
From: "Daniel-Constantin Mierla"
To: "Slava Bendersky" , "Kamailio (SER) - Users Mailing
List"
Sent: Friday, March 28, 2014 6:54:20 PM
Subject: Re: [SR-Users]
Sweet!
On 28 March 2014 15:24:10 GMT-04:00, Kelvin Chua wrote:
>It worked! perfect
>
>Kelvin Chua
>
>
>On Fri, Mar 28, 2014 at 1:34 AM, Alex Balashov
>wrote:
>
>> On 03/28/2014 04:32 AM, Kelvin Chua wrote:
>>
>> I tried, AVPs, they are not accessible on the
>>> event_route[tm:local-request]
>>>
Hello,
there is no loop there -- the log is incomplete, not showing what
happened with the INVITE request, only with its replies -- but anyhow,
the replies 180 and 486 are coming from callee, the user-agen: z ...
So all seems ok with routing of the invite, just the callee is busy, not
answer
Hello Daniel,
I was looking where is problem, and I see call is looping meaning coming back
to caller.
I putted couple xlog to see where is breaks.
$avp(oexten) = $rU;
xlog("L_INFO", "This is $avp(oexten)");
if (!lookup("location")) {
xlog("L_INFO", "This $rU and this $du");
this is log
Have you tested with async mi commands? The function
build_async_handler() is cloning the information in shared memory,
expecting that the response to the mi command to be done from another
process. In that case, the socket is no longer valid, because it was
created only in the mi process.
I
Please find attached updated patch as requested.
On Fri, Mar 28, 2014 at 8:49 PM, Daniel-Constantin Mierla wrote:
> Thanks for the patch. Can you resend it without commented code? Just
> remove the lines that are no longer used, to have clean code.
>
> Cheers,
> Daniel
>
>
> On 28/03/14 01:29,
Okay, good news. After reverting back to using avp, I'm handling 1000+
ports without any issues. Not sure what was wrong during the previous test,
but it doesn't appear to be here now.
Thanks for your help Daniel
Ryan
Ryan Brindley
Software Development Officer
Stratics Networks, Inc.
1.866.635.6
I found a part of the problem.
The kam cfg reg ex had a typo, which i've removed.
1. route[FSVBOX] {
2. if(!($rU=~"^1[0-9][0-9]+$"))
3. return;
4. prefix("vb-");
5. route(FSRELAY);
6. }
So now when I dial 4
Thanks for the patch. Can you resend it without commented code? Just
remove the lines that are no longer used, to have clean code.
Cheers,
Daniel
On 28/03/14 01:29, Muhammad Shahzad wrote:
Hi,
After wasting most of the day trying to make mi_datagram over UDP
socket work. I eventually realize
I wasn't using msg_apply_changes and I could've been wrong that it showed
up. That was yesterday when I was testing that. One of my tests though did
get xlog to show the variable, but it could've been with avp.
I'll convert my configs back to using avp and begin testing again with
multiple channel
Hi there.
I'm still trying to integrate Kamailio and freeswitch... where kamailio acts as
a proxy and registrar ... and freeswitch provides conference calls and
voicemail.
I have calls between two polycoms working and conference calls work.
But when I try to leave a voice message for a user by
Should not crash, can you send the backtrace from the core dump? What
version are you using?
Also, setting debug=3 in config file and sending the log messages will help.
Saving $avps() is something very common in my configs and haven't
encountered any issues in the past years.
Cheers,
Daniel
Hello,
On 28/03/14 19:58, Ryan Brindley wrote:
Hey community,
I'm trying to get a 2nd leg custom header value stored in my acc table.
Right now I'm appending the header with append_hf in a failure route
and trying to use:
modparam("acc","db_extra","custom=$hdr(Custom)")
A sip trace shows t
It worked! perfect
Kelvin Chua
On Fri, Mar 28, 2014 at 1:34 AM, Alex Balashov wrote:
> On 03/28/2014 04:32 AM, Kelvin Chua wrote:
>
> I tried, AVPs, they are not accessible on the
>> event_route[tm:local-request]
>>
>
> That's because AVPs are only transaction-persistent, not dialog-persistent
It's also worth noting that I tried to use avp and it successfully added
the custom value, but was causing Kamailio to crash once it tried to handle
multiple calls at once.
I'm pretty green to Kamailio, so you'll have to pardon my inexperience on
this probably pretty easy question.
On Fri, Mar 28
Hey community,
I'm trying to get a 2nd leg custom header value stored in my acc table.
Right now I'm appending the header with append_hf in a failure route and
trying to use:
modparam("acc","db_extra","custom=$hdr(Custom)")
A sip trace shows the header was properly added and xlog output shows t
I'm rewriting the Contact header in the REGISTER to be Kamailio on port
5060. For some reason Asterisk is sending INVITEs on the source port of
the message and not 5060. I thought rport would correct that, but it
sounds like rport is only intended to for subsequent messages in a given
transaction
Hi Alexandr
On 28 March 2014 15:36, Alexandr Usov wrote:
> I am already have some practice to integrate Kamailio with Asterisk, when
> all users creates and registers in Kamailio, and calls go to/from Asterisk
> with static "host=kamailio_ip" settings for each user on Asterisk side.
>
> I can't
Hello,
rport and Via headers are used for routing the SIP responses back to the
device sending the request. They are not used at all for routing INVITE
requests.
The tutorial you refer to is generating a new REGISTER from Kamailio to
Asterisk, putting in it the Contact header with the addres
Basically, I'm trying to get Asterisk to send all future INVITEs to
Kamailio on port 5060 and not the source port of the REGISTER. My REGISTER
looks like this:
U 1.1.1.1:59738 -> 2.2.2.2:5060
REGISTER sip:2.2.2.2:5060 SIP/2.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK404.05c2e5a7.0.
Via: SIP/2
I have a Kamailio server forwarding REGISTERs to an Asterisk box, similar
to the way Daniel shows here:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
Can I force Kamailio to append rport=5060 to the topmost Via header, prior
to forwarding the REGISTER to Asterisk? I
Hello,
if you want caching, look at htable or mtree modules, if you want with
database real time query, see auth_db is_subscriber() or just use slqops.
Cheers,
Daniel
On 28/03/14 17:53, James Cloos wrote:
Given a proxy where the only registrations will be next-hop pbxes, where
those names wi
Hello,
it is recommended to use the latest 4.0.x, which is 4.0.6 at this
moment. There was a fix to registrar module that might be the reason of
the issue you faced.
On the other hand, the backtrace is important, send the output of:
bt full
Cheers,
Daniel
On 28/03/14 19:10, Igor Potjevlesc
Hello,
I experienced an issue yesterday.
I move a customer to Kamailio 4.0.4 (previously he was on an old SER
instance without issue).
He uses the same SIP account for connecting two IPBX (same platform, same
firmware).
When the second REGISTER came on Kamailio and, may be 500ms/1s after (I
On Mar 28, 2014, at 11:36 AM, Alexandr Usov wrote:
> I am already have some practice to integrate Kamailio with Asterisk, when all
> users creates and registers in Kamailio, and calls go to/from Asterisk with
> static "host=kamailio_ip" settings for each user on Asterisk side.
>
> I can't (do
Given a proxy where the only registrations will be next-hop pbxes, where
those names will not be INVITEable, what is the ideal way to tell kama
the list of accepted usernames and authentication info for REGISTER?
It is important to ensure that any INVITE to any of the REGISTERable
usernames 404s.
Thanks Alex.
What are the components that I should take into account. Based on answer in
another thread I will be using SIP trunk with an NGN to route the outside
traffic (other than SIP-SIP).
I'll have NATed clients so I'll need the media proxy or rtp proxy as well.
What about their dimensionin
I am already have some practice to integrate Kamailio with Asterisk, when
all users creates and registers in Kamailio, and calls go to/from Asterisk
with static "host=kamailio_ip" settings for each user on Asterisk side.
I can't (don't know - how to) use in same way integration with FreeSWITCH.
Ca
Here's a copy of my cfg file: http://pastebin.com/GtDwWKWr
I tried to run with the switches you mentioned... but i still don't get
anything logged to syslog.
?
sorry... and thanks for your patience.
From: Olle E. Johansson
To: mark li
Cc: Olle E Johansso
1. Kamailio does not handle RTP, so audio is not a scaling factor at all;
2. The only scaling factors are transactional memory (TM) and any dialog
state you are keeping;
3. Without I/O wait from external sources, Kamailio can handle a
practically infinite amount of concurrent calls and CPS. B
HI Guys,
can you refer me to some resources which help me dimension the hardware for
a setup.
Total users will be 20,000. What percentage should i assume for concurrent
audio and video calls? What is the standard practice? How does this all map
to the cpu, ram and storage etc. given that I will b
On 03/28/2014 04:32 AM, Kelvin Chua wrote:
I tried, AVPs, they are not accessible on the event_route[tm:local-request]
That's because AVPs are only transaction-persistent, not dialog-persistent.
however, your idea on $dlg_vars was spot on, why didn't i think of that.
Thanks a bunch!
Well..
Hi Alex,
yes, locally generated BYEs
I was trying to avoid the unique constraint approach as it is not very
elegant.
I tried, AVPs, they are not accessible on the event_route[tm:local-request]
however, your idea on $dlg_vars was spot on, why didn't i think of that.
Thanks a bunch!
Kelvin Chua
Oh, you were talking about the BYEs that are internally generated by
'dialog'. In that case, ignore my advice; it's not applicable, because
locally generated BYEs will not have the RR header and this won't work.
In those situations, in my personal opinion, your best bet is to use a
database-si
On 03/28/2014 03:53 AM, Kelvin Chua wrote:
I have a situation here,
when a dialog expires, it sends a BYE to both call legs and this does
not generate an entry on acc table.
however, when i use acc_db_request() on event_route[tm:local-request], i
get 2 entries on acc, 1 for the downstream, and
I have a situation here,
when a dialog expires, it sends a BYE to both call legs and this does not
generate an entry on acc table.
however, when i use acc_db_request() on event_route[tm:local-request], i
get 2 entries on acc, 1 for the downstream, and 1 for the upstream.
is there a way of handli
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