HI!
I need help to configure RTPProxy.
I have working Kamailio on Centos 6.5, On LAN everything is working fine.
Now I want my other users to connect using 3G or any other network.
Please help me with that.
Statis Public IP (DSL Modem) - Proxy Box with 2 Lan cards ---
192.168.2.0 and 192.168.0
Hello,
I am trying to forward the register to multiple Asterisk using Kamailio.
The basic Idea is:
Softphone A>Kamailio-> Asterisk 1
--> Asterisk 2.
I follow Asipto howto http://lylix.net/kamailio, but the problem I see in
my case, is that registr
On 05/29/2015 11:58 AM, Ali Taher wrote:
Can I save current gateway in the avp list to a pseudo variable and add it
to append_to_reply, then use use_next_gw() to get next gateway and so on ,
before using t_relay() ?
Sure. But you won't want to call t_relay() if you're sending redirects
-- jus
Hi,
Can I save current gateway in the avp list to a pseudo variable and add it
to append_to_reply, then use use_next_gw() to get next gateway and so on ,
before using t_relay() ?
Regards,
Ali
-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
Al
I'm not sure it can be done automatically.
--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
Sent from
Hi,
How can all available contacts extracted from the database be appended
automatically?
Thanks for your fast help :)
Ali
-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
Alex Balashov
Sent: Friday, May 29, 2015 6:37 PM
To: sr-users@lists.si
On 05/29/2015 11:34 AM, Ali Taher wrote:
In this way , all gateways will be sent by Kamailio by one SIP response ?
If you like!
append_to_reply("Contact: ;q=0.1\r\n");
append_to_reply("Contact: ;q=0.2\r\n");
sl_send_reply("300", "Multiple Choices");
exit;
--
Alex Balashov | Principal | Evar
Hi,
In this way , all gateways will be sent by Kamailio by one SIP response ?
Regards,
Ali
-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
Alex Balashov
Sent: Friday, May 29, 2015 6:13 PM
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Use
On 29/05/15 11:16 AM, Vasiliy Ganchev wrote:
> On May 29, 2015; 3:19pm, Richard Fuchs wrote:
>> A good way to start debugging this is to run rtpengine with log-level 7
>> and post the full log for such a call.
> Hi Richard! Thanks for answer!
> Call log written on WS_Kamailio, rtpengine with log
Al S writes:
> Thank you for the reply:You're saying I ned to use "aliases" table to
> provision my TNs with each IP-PBX.What module I can use to add TNs to
> this table?
Check if ALIAS_DB module fits your needs.
-- Juha
___
SIP Express Router (SER) a
On May 29, 2015; 3:19pm, Richard Fuchs wrote:
> A good way to start debugging this is to run rtpengine with log-level 7
> and post the full log for such a call.
Hi Richard! Thanks for answer!
Call log written on WS_Kamailio, rtpengine with log-level 7
Call from UA_WS 272 calling to UA_SIP 271 a
Ali,
It's up to you where you pull the gateway list from. After that, you can
use append_to_reply() to construct the Contact header and send the
redirect message.
So, yes, it can be done.
-- Alex
On 05/29/2015 10:35 AM, Ali Taher wrote:
Hello,
We need to integrate Kamailio with an exchan
Thank you for the reply:You're saying I ned to use "aliases" table to provision
my TNs with each IP-PBX.What module I can use to add TNs to this table?
Thanks,Ali
> there are many ways you can achieve that. One is to define aliases for
> one number and then register that.
> Date: Fri, 29 May 20
Hello,
We need to integrate Kamailio with an exchange switch in order to have
routing decision.
Kamailio must send to the switch the list of available gateways in the
contact tag of SIP response as shown in the below example:
SIP/2.0 300 Multiple choices
Via: SIP/2.0/UDP 172.16.5.112:5060
Olli Attila writes:
> Could the "ICE bridge" solve this issue or is there another way around
> this?
Yes, see ICE flag in rtpengine README.
-- Juha
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.o
Al S writes:
> Does Kamilio have a predefined module to support "Multiple Phone
> Numbers" registration with a single Registration?I am trying to
> register multiple numbers on behalf of IP-PBXs.
there are many ways you can achieve that. One is to define aliases for
one number and then register
Hi,
Does Kamilio have a predefined module to support "Multiple Phone Numbers"
registration with a single Registration?I am trying to register multiple
numbers on behalf of IP-PBXs.
Thanks,Al
Please see in http://www.kamailio.org/wiki/cookbooks/4.2.x/pseudovariables
$_s(format) - Evaluate dynamic format
*$_s(format)* - returns the string after evaluating all pseudo-variables in
format
$var(x) = "sip:" + $rU + "@" + $fd;
# is equivalent of:$var(x) = $_s(sip:$rU@$fd);
BR
On 29/05/15 05:48 AM, Vasiliy Ganchev wrote:
Юрий, Thanks for your answer!
Sorry but I couldn't understand how to do "Reverse rtp_manage functions". If
it is possible, extend your idea with some more example/explanation?
Maybe someone else can help suggest something, or do I need to give more
in
Hello;
it works both ways. Could you explain what i did here?
$rU=*$_s*($avp(i:69)$rU);
$rU=*$_s*($avp(s:69)$rU);
Thanks.
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/Kamailio-V4-2-5-append-prefix-to-Request-Uri-problem-tp138258p138277.html
Sent from the Users m
Hi,
Try to use:
$rU=$_s($avp(69)$rU);
BR
Julia
On Fri, May 29, 2015 at 12:13 PM, Yasin CANER
wrote:
> Hello;
> i try to append prefix to $rU like below in Kamailio V4.2.5 . i get
> error even if i set string value. it works only when i append like that
> $rU="3002"+$rU;
>
> -
Somehow got improvement, I am able to start the kamailio service and able
to register two extensions. but again , am getting some error in Asterisk
console and in /var/log/messages
*Asterisk console *
chan_sip.c:16090 parse_register_contact: Not a valid SIP contact (missing
sip:/sips:) trying to us
Юрий, Thanks for your answer!
Sorry but I couldn't understand how to do "Reverse rtp_manage functions". If
it is possible, extend your idea with some more example/explanation?
Maybe someone else can help suggest something, or do I need to give more
information?
Thanks in advance!
--
View this
On Thursday 28 May 2015 19:48:20 Alexandru Covalschi wrote:
> First question: is it possible to create users not via kamctl command, but
> directly inserting new entries to this tables?
Yes.
> And the second one: if it is possible, which hash alghorythm Kamailio uses
> when creating a new user on
Hello;
i try to append prefix to $rU like below in Kamailio V4.2.5 . i
get error even if i set string value. it works only when i append like
that $rU="3002"+$rU;
- 1 test
$var(prefix)=$avp(s:69);
$rU=$var(prefix)+$rU;
-FAILED
-2 test
$rU=$avp(s:69
Hello Daniel,
I am implementing conference bridge using kamalio and RTP
porxy without media mixing. Since for now we are ok with this. My question
is as per the flow of RFC 4579, UA may send SUBSCRIBE to the conference URI
with event as conference. In kamalio modules presence
I think I posted the wrong log.
Please use this log instead
###
[May 28 21:54:51] VERBOSE[2641] chan_sip.c:
<--- SIP read from UDP:127.0.0.1:5060 --->
INVITE sip:*6...@voiptosave.com SIP/2.0
Record-Route:
Record-Route:
Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG
Hello.
I think I understand the problem, but I do not know how to fix it.
I have done the following when the problem accord:
-Download Elastix MT 64bit (downloaded from -
http://www.elastix.com/en/downloads/)
-Install it in VMware ESXi 5.1. Elastix works fine within the LAN
-Then I pushed the VM t
Hello,
I have a problem in call creation between websocket client (sipml5)
and traditional sip client (linphone ios). If i make a call from
ws-client to linphone (through kamailio with rtpengine installed), i get
error from ws-client console when linphone answers the call with 200ok/sdp:
"Fai
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