Hello,
I have a problem with the following configuration.
I want to make calls from Asterisk to a browser using RTPEngine as relay.
Everything works fine, if Kamailio is not natted (See
kamailio_without_nat.log).
If it's address is translated, then 200 OK responses from the browser don't
seem t
Just a question for the RPM maintainers...
Is there a reason for some of the modules (such as carrierroute) not being
included in the RPMS?
It's been bugging me for a while, when I do an installation, I have to go and
compile all the sources Not that it's an issue, just easier to do it fro
Hi Ovidiu,
On 18/10/14 00:37, Ovidiu Sas wrote:
Which is bad, it should be the IP of the FS server.
I investigated and I'm not sure this is the issue.
Unfortunately when I named the various addresses it obscured the fact
that the ip address kamailio.int is the IP address of the freeswitch
se
I just tested the same thing on a Rackspace VPS (Xen I think) and am
seeing the same climbing load average with 2 async workers. It seems to
top out at 1.05.
On 10/23/2014 02:13 PM, Alex Balashov wrote:
Another thing I have found is that having a certain amount of async
workers running, even
Hello,
As explained in my eirlier email. My topology is like below:
sipml5(behind NAT) -> (Public Interface)kamailio(Private Interface) <>
Freeswitch(172.16.0.150)
As Freeswitch is doing the ICE handling. My SIPml5 cliemt used to receive a SDP
like the below:
v=0
o=FreeSWITCH 1414067668
And what returns prtpengine at log when changing this packet.
Returning to SIP proxy: d3:sdp316:v=0#015#012o=root 1195474335 1195474335
IN IP4 2.10.39.16#015#012s=Asterisk PBX 12.6.1#015#012c=IN IP4
2.10.39.16#015#012t=0 0#015#012m=audio 30614 RTP/AVP 8 3 0
101#015#012a=rtpmap:8 PCMA/8000#015#012a
No SDP body only one. but packet like this
INVITE
sip:device-200@sip:1.21.10.2:45437;rinstance=07f88c423145358e;transport=UDP
SIP/2.0
Record-Route:
Via: SIP/2.0/UDP sip.myservice.com:5068
;branch=z9hG4bKca7d.2d16143316e23fac46bf686bb41780b3.2
Via: SIP/2.0/UDP 17.74.28.7:50600;branch=z9hG4bK22c67
On 10/23/14 15:06, Yuriy Gorlichenko wrote:
> Still have same error...
> Now rtpproxy_manage("co-sp") for classic call. At log I see that
> rtpproxy wirked gine. For each step it generate write body, but t_Relay
> still send strange "compinated" packet to UDP with SDP for WS...
Do you mean that t
Still have same error...
Now rtpproxy_manage("co-sp") for classic call. At log I see that rtpproxy
wirked gine. For each step it generate write body, but t_Relay still send
strange "compinated" packet to UDP with SDP for WS...
2014-10-23 20:42 GMT+04:00 Yuriy Gorlichenko :
> Oh. Ok. I will try. T
Hi Gonzalo,
Thanks a lot for your reply. I took a closer look at the issue and I guess we
figured out the reason why this is happening. But, still couldnt get the
solution unfortunately. Our topology is like this:
sipml5(behind NAT) -> (Public Interface)kamailio(Private Interface) <>
Fre
Hi,
Using method json_get_field i am able to extract json string. However, this
string comes with quotes which cause a hurdle in assigning to various
pseudo variables, for example,
--
json_get_field($redis(r=>value), "to_number", "$var(wim_to)");
json_get_field($redis(r=>val
Another thing I have found is that having a certain amount of async
workers running, even if they are not doing anything, appears to cause
unexplained CPU load, even if the Kamailio instance is completely idle
and not processing any calls.
Here is the baseline load with no async workers:
[roo
You can implement whatever routing logic you want in kamailio.
Also, you can use different sets of modules to implement same type of
routing logic.
The To header is irrelevant in SIP, no need for re-write.
When a call is received from an endpoint, based on it's IP address you
can choose what to do
Hi Fred,
Its more that the "user" on Kamailio is actually a PBX with extensions on it.
On asterisk I'd usually do Dial(SIP/peername/extension) but I obviously cant do
this as Kamailio is the peer that the call is being routed to initially.
What I need to figure out is how to on kamailo maybe us
Oh. Ok. I will try. Thanks for advice. I very hope it hepls.
2014-10-23 20:18 GMT+04:00 Richard Fuchs :
> On 10/23/14 12:17, Yuriy Gorlichenko wrote:
> > What you mean under "full set of flags"? At reply I use mirror (+/-)
> > flags off course. More, it work without branches fine ( i select only
On 10/23/14 12:17, Yuriy Gorlichenko wrote:
> What you mean under "full set of flags"? At reply I use mirror (+/-)
> flags off course. More, it work without branches fine ( i select only
> one endpoint). I have issue only with branches.
I mean that instead of using rtpproxy_manage("co") you should
What you mean under "full set of flags"? At reply I use mirror (+/-) flags
off course. More, it work without branches fine ( i select only one
endpoint). I have issue only with branches.
23.10.2014 18:56 пользователь "Richard Fuchs" написал:
>
> On 10/23/14 06:03, Yuriy Gorlichenko wrote:
> > Hell
Hello,
I have a problem with the following configuration.
I want to make calls from Asterisk to a browser using RTPEngine as relay.
Everything works fine, if Kamailio is not natted (See
kamailio_without_nat.log).
If it's address is translated, then 200 OK responses from the browser don't
seem t
On 10/23/14 11:32, Marino Mileti wrote:
> Version is 3.3.0.0 mr3.5.0.0
>
> I've seen that the problem is during the "ping-pong test" when the value of
> bencode_item is assigned
>
> The error is on __bencode_dictionary_init, and the istruction is :
> dict->value=0
>
> I've read ARM documentation
If you want to call a user on Kamailio from Asterisk...
example...
exten => s,1,Verbose(4,calling user on kamailio)
same => n,Dial(SIP/USERNAME@KAMAILIO,time,options)
same => n,--after dial logic --
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-95
Version is 3.3.0.0 mr3.5.0.0
I've seen that the problem is during the "ping-pong test" when the value of
bencode_item is assigned
The error is on __bencode_dictionary_init, and the istruction is :
dict->value=0
I've read ARM documentation and it says to add __attribute__((packed)) to
value defin
Hi Fred,
Thanks for the quick response. I already do use some Kamailio features on our
internal network for load balancing.
The use case that I'm interested in is to effectively replace an asterisk
server that I use for SIP trunking to remote phone systems with a Kamailio
registrar/proxy and
Hi Kenny,
This depends on the carriers and scenarios that you may use. I know
"depends" is a horrible answer, but one of the great aspects of Kamailio
is the flexibility of the modules.
Some deployments may have a group of Asterisk servers all configured
similarly for handling calls. With th
On 10/23/14 06:03, Yuriy Gorlichenko wrote:
> Hello all. I use rtpengine and rtpproxy-ng module at kamailio for
> proxying RTP and modifying SDP between endpoints. I use two types of
> clients - such as WSS based and UDP based clients.
>
> I have a trouble with append_branch and rtpengine handling
On 10/23/14 05:28, Marino Mileti wrote:
> Hi guys,
>
> I've cross compiled Kamailio (4.1.6) for ARM (Wandboard-IMX6). Everything
> seems works fine, but when I try to enable rtpproxy-ng module and Alignment
> trap error occurred just during rtpp_test (exchanging ping-ponge message
> betweek rtppro
Hi,
I have a few asterisk servers providing some basic SIP trunking and routing.
We have remote PBXs trunked onto asterisk which calls come into asterisk and
are routing down to extensions on the remote PBX via prefix routing.
I'm looking to have a central Kamailio Registrar/Proxy/Loadbalancer
Hello,
are you using branch 4.2 or some tgz/package from 4.2.0 release? There
was a fix related to such issue, so 4.2 branch must be used (or debian
nightly builds for 4.2).
I will have to check the code how flexible that is, but afaik, the limit
on reload interval was thought to avoid sending ag
Hello,
what should be happen, is the following:
- invite from controller to first parameter (caller of desired call)
- after 200ok comes from 'caller', kamailio sends REFER to it pointing
to the second parameter (callee of desired call) and then BYE, getting
out of the initial call
- after gettin
I seem to be going round in circles… I am trying to use dlg_bridge() from the
dialog module to establish a call between two SIP endpoints. I have tested
with Snom phones and linphone soft phone with the same result.
I get an outbound call to the first (from) end point, I answer the phone… and
Hello all. I use rtpengine and rtpproxy-ng module at kamailio for proxying
RTP and modifying SDP between endpoints. I use two types of clients - such
as WSS based and UDP based clients.
I have a trouble with append_branch and rtpengine handling for this
packets.
I try to implement this logic of m
Hi guys,
I've cross compiled Kamailio (4.1.6) for ARM (Wandboard-IMX6). Everything
seems works fine, but when I try to enable rtpproxy-ng module and Alignment
trap error occurred just during rtpp_test (exchanging ping-ponge message
betweek rtpproxy and rtpproxy-ng module of Kamailio)
rtpengine[22
Hey Guys,
I am in the process of testing the reloading of the uac registrations
(thanks Daniel for that), and hit some issue:
* when using uac_reg_info twice in a row, second command always hangs.
"""
root@Dev2:/etc/kamailio# kamcmd -s tcp:127.0.0.1:2046 uac.reg_info
l_uuid 301
{
l_uuid
The $TV(Sn) returns a string containing the seconds and milliseconds.
To get the current DateTime I have to add the seconds since standard epoch.
What about the milliseconds? Are those the current milliseconds?
I'm using the following C# code to create a DateTime object using the value
from $TV(S
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