Daniel,
Sorry, I had meant to send it to the list.
You are correct about missing messages. The ACK is being sent, but it is
sent to it's self (loop condition) over the 'lo' device.
So basically its a design flaw on my part of the kamailio.cfg file.
My current config works for initial message
Hi, I have confgured Kamailio with TLS in an Kamailio-Asterisk
implementation. The phones (Bria - Eyebeam - Aastra 57i) are registering in
Kamailio without problems, but the registration is not being forwarded to
Asterisk.
Kamailio is listening on port tls:5061 only and Asterisk on udp:5080. Should
Daniel-Constantin Mierla writes:
> I am thinking of packaging v3.1.4 next week on Thursday -- last minor
> release was more than 1.5month ago.
> Any comments against it?
the bug still exists, where AVP_CLASS_DOMAIN avps are not saved in the
transaction. i guess jan forgot about it.
-- juha
_
On 5/20/11 9:56 PM, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
if I understood right, it is about 200 ACK which are forwarded
stateless. If yes, then should be only in forward.c, the condition at
line 545, where the assignment can be changed in a memcpy.
for example like this? if
Daniel-Constantin Mierla writes:
> if I understood right, it is about 200 ACK which are forwarded
> stateless. If yes, then should be only in forward.c, the condition at
> line 545, where the assignment can be changed in a memcpy.
for example like this? if so, i can go and make the commit.
--
On 05/20/2011 12:54 PM, Daniel-Constantin Mierla wrote:
the support is in core since 2002, but not many entered the challenge
so far :-)
That's a fairly apt metaphor for the state of IPv6 adoption as a whole. :)
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2
On 5/20/11 6:33 PM, Jon Farmer wrote:
On 20 May 2011 17:25, Daniel-Constantin Mierla wrote:
the module is good, but not for this case -- the warning seems to be from
startup, saying that no reverse dns could be done to the ipv6 listen address
and it is harmless (done for the purpose of build
On 5/20/11 5:42 PM, Timo Reimann wrote:
Hey Jon,
On 20.05.2011 17:31, Jon Farmer wrote:
How do I represent the following if the address was IPv6?
if(src_ip ==127.0.0.1)
If I specify the IPv6 address with our without [] I get
WARNING:core:fix_socket_list: could not rev. resolve
Maybe Iñak
Hey Jon,
On 20.05.2011 17:31, Jon Farmer wrote:
> How do I represent the following if the address was IPv6?
>
> if(src_ip ==127.0.0.1)
>
> If I specify the IPv6 address with our without [] I get
>
> WARNING:core:fix_socket_list: could not rev. resolve
Maybe Iñaki's brand new ipops module is
Hi
How do I represent the following if the address was IPv6?
if(src_ip ==127.0.0.1)
If I specify the IPv6 address with our without [] I get
WARNING:core:fix_socket_list: could not rev. resolve
Regards
Jon
--
Jon Farmer
Tel: 07795 118140
Email: viperdud...@gmail.com
Twitter: @viperdudeuk
_
On Friday 20 May 2011, Carsten Bock wrote:
> here is the doodle link:
> http://doodle.com/sipgepeyp4us4yey
>
> If anyone suggests other dates, i can add them accordingly.
Hi Carsten,
great, i've added my availability as well.
Cheers,
Henning
___
SIP
Daniel;
Sorry for not being clear;
I understand the HTTP stuff;
I was asking if it is a simple as just calling the dialog bridge method
within the HTTP event route.
for example;
event_route[xhttp:request] {
dlg_bridge("sip:m...@myproxy.com", "sip:y...@yourproxy.com",
"sip:myproxy.com:5080"
Dear all,
I'm trying to change the From uri and Dsplay but without sucess.
My config is as follow:
if(dp_translate("", "$avp(s:frm_user_name)/$avp(s:test_frm_user_name)"))
--> i'm sending 2112202701 and I get back corectly 701
{
$avp(s:display) = $avp
Please do not write private emails, keep the mailing list cc-ed.
Can you get the output of:
ngrep -d any -qt -W byline port 5040
on kamailio server for such a call? Include all messages, since you may
miss something in your filtering (like you did below), of course you can
mask the ip addres
On 5/20/11 3:44 PM, David J. wrote:
Thanks Daniel;
What creates the Initial Dialog and REFER method
Just calling
dlg_bridge("sip:m...@myproxy.com", "sip:y...@yourproxy.com",
"sip:myproxy.com:5080");
When I get an HTTP event;
not sure you ask something or not, but if you don't know wher
Thanks Daniel;
What creates the Initial Dialog and REFER method
Just calling
dlg_bridge("sip:m...@myproxy.com", "sip:y...@yourproxy.com",
"sip:myproxy.com:5080");
When I get an HTTP event;
On 5/20/11 9:32 AM, Daniel-Constantin Mierla wrote:
Hello,
On 5/20/11 3:25 PM, David J. w
Hello,
On 5/20/11 3:25 PM, David J. wrote:
I was wondering if I can make an HTTP request to Kamailio;
and then have kamailio do a lookup based on passed parameters
to connect callers.
I am trying to make a click-to-dial type application;
I was looking at the HTTP server inside kamailio;
It was
I was wondering if I can make an HTTP request to Kamailio;
and then have kamailio do a lookup based on passed parameters
to connect callers.
I am trying to make a click-to-dial type application;
I was looking at the HTTP server inside kamailio;
It was interesting to me to try to use this as an a
Thanks Daniel...
Best Regards
2011/5/20 Daniel-Constantin Mierla
> Hello,
>
>
> On 5/20/11 2:26 PM, Bruno Bresciani wrote:
>
>> Hi,
>>
>> on kamailio 3.1.2 the one_time_nonce parameter of the auth module is
>> similar to the nonce_reuse parameter present on kamailio 1.5.0?
>>
>> I want to disa
Hello,
On 5/20/11 2:26 PM, Bruno Bresciani wrote:
Hi,
on kamailio 3.1.2 the one_time_nonce parameter of the auth module is
similar to the nonce_reuse parameter present on kamailio 1.5.0?
I want to disable the check of the nonce reuse in kamailio 3.1.2...
yes, you have to use that one_time_no
Hi,
on kamailio 3.1.2 the one_time_nonce parameter of the auth module is similar
to the nonce_reuse parameter present on kamailio 1.5.0?
I want to disable the check of the nonce reuse in kamailio 3.1.2...
regards
___
SIP Express Router (SER) and Kama
Hi all, I have it running using OPTIONS but not when using INFO. I was missing
an s exten in my asterisk config.
Does anyone have INFO working with Asterisk?
Regards
Lee
-Original Message-
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On
OK, I will see what I can do at my end but it's odd that when the ping first
kicks off a 481 is sent back but the server stays listed as active.
Thanks for the help.
Regards
Lee
-Original Message-
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.or
Hi,
i think i added the feature in 3.1 to configure the valid responses.
In prior versions, only the following reply codes are considered a
"valid" reply:
- 200 OK
- 501: Cisco-Gateways reply with a "501 Not supported" to the request.
- 403: Aastra-Gateways reply with a "403" to the request.
- 405
Hi Carsten, is this valid syntax for 3.0.4? I've been up and down the doc for
the dispatcher module and can't see this in the 3.0 docs. It is replying with
a 481.
Thanks
Lee
-Original Message-
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org
Hi,
what are the destination systems replying to the INFO-request?
It should be a "200 OK" in order to reactivate the destination. If it
replies something else, you need to configure the other replies:
modparam("dispatcher", "ds_ping_reply_codes",
"class=2;code=403;code=488;class=3")
See:
http:/
Hi Carsten, I see the INFO requests on the target server and the packets being
sent and received on the kamailio server. Logs aren't showing me anything,
even in debug mode.
Regards
Lee
-Original Message-
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-
Hi,
here is the doodle link:
http://doodle.com/sipgepeyp4us4yey
If anyone suggests other dates, i can add them accordingly.
Carsten
2011/5/20 Daniel-Constantin Mierla :
> Hi,
>
> On 5/20/11 10:24 AM, Carsten Bock wrote:
>>
>> Hi,
>>
>> June 4th would be a saturday?
>
> ... that means Saturday i
Hi,
do you see any outgoing "INFO" Requests?
Do you see anything in the logs?
Carsten
2011/5/20 Lee Archer :
> Hi Daniel, I can see the gateway flagged as P after so many failed attempts
> but when I bring the server back up I can see the pings being received by
> the server but Kamailio still h
Hi Daniel, I can see the gateway flagged as P after so many failed
attempts but when I bring the server back up I can see the pings being
received by the server but Kamailio still has a P against the dispatcher
dump. The server is definitely working and pings are getting through.
SET:: 4430125
Hi,
On 5/20/11 10:24 AM, Carsten Bock wrote:
Hi,
June 4th would be a saturday?
... that means Saturday is a free day in your company?!?! :-)
I wanted Thursday or Friday, so it should be June 2 or 3.
Maybe we could create a doodle (http://www.doodle.com/) for this,
where everyone can mark the
Hi,
June 4th would be a saturday?
Maybe we could create a doodle (http://www.doodle.com/) for this,
where everyone can mark the dates, when he is available
Carsten
2011/5/20 Daniel-Constantin Mierla :
> Hello,
>
> we should decide the roadmap to next major release v3.2.0, so probably an
> IR
Hello,
we should decide the roadmap to next major release v3.2.0, so probably
an IRC meeting for some real time chatting would be good. I am thinking
of June 3 or 4, 15:00GMT. Anyone interested in the meeting and available
in one of those dates? The date with most of devs available will be
se
Hi,
is the TM-Module loaded before the dispatcher module? Otherwise, the
automatic Pinging is deactivated.
Also, please check the Parameters set to the dispatcher module.
Kind regards,
Carsten
2011/5/20 Daniel-Constantin Mierla :
>
>
> On 5/19/11 4:05 PM, Lee Archer wrote:
>
> Hi all I wonder if
Hello,
I am thinking of packaging v3.1.4 next week on Thursday -- last minor
release was more than 1.5month ago.
Any comments against it?
Cheers,
Daniel
--
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda
___
On 5/19/11 4:05 PM, Lee Archer wrote:
Hiall I wonder if someone canhelp bypointingme in the right
direction. Ihave dispatcher set toping and set any failedgatewaysto
probing but how do you get them out ofthis state when they start
working again? I have run some tests and Imust bemissing s
On 5/19/11 1:49 PM, Andrew O. Zhukov wrote:
Right now I'm try to add the subj feature.
Linksys/SPA2102-3.3.6 (just from the box) respond
SIP/2.0 480 Temporarily not available
Landline over Asterisk with E1:
SIP/2.0 603 Limit exceeded
Ether I so unlucky, or just forget it and do not provide i
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