Ross,
On 09/30/2010 10:14 AM, Ross Beer wrote:
> The phone sets up another call using INVITE and then uses REFER when
> the transfer takes place.
Please try to keep the list copied, as a matter of good practice.
If I understood your scenario correctly, you don't have a choice but
to send the REFER to the same Asterisk server as the one hosting the
original call leg, because that is the only server that is aware of
that call leg. Perhaps I have not adequately understood.
However, if your question is about whether Kamailio can initiate
transfers, it can, sort of. You can initiate REFERs and
INVITE-hold-REFER-BYE bridges from route script a la:
http://www.kamailio.org/docs/modules/3.0.x/modules_k/dialog.html#id2700697http://www.kamailio.org/docs/modules/3.0.x/modules_k/dialog.html#id2700815
You can also use the management interface to stimulate Kamailio to
initiate a 'bridge' of this nature from an outside script:
http://www.kamailio.org/docs/modules/3.0.x/modules_k/dialog.html#id2701383
If there is anything that might serve your needs, I think it is that.
-- Alex
--
Alex Balashov - Principal
Evariste Systems LLC
1170 Peachtree Street
12th Floor, Suite 1200
Atlanta, GA 30309
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
Hi,
I would like to know if the following is possible in Kamailio, I've
tried with OpenSIPs but I don't think it is ideal for my needs.
I would like to load balance multiple asterisk boxes which terminate
and originate calls. To transfer calls by attended transfer any new
calls originating from a phone need to be sent to the same server as
the held call. With the dialogue module I can add the call originating
from asterisk to a profile and the new call from the phone can check
if the user already belongs to a profile and then send the call to the
same gateway.
What I would like to know is if there is a better way to do this or if
Kamailio can perform the transfer without the need to send the call
back to the same asterisk box. I've noticed that the Kamailio dialogue
module has a few more features than its OpenSips counterpart.
I would be very grateful for any feedback.
Regards,
Ross
there is a broken grandstream sip ua that sometimes sends two initial
invites back-to-back. according to wireshark, sr received them about 10
microseconds apart. sr then forwards the request to pstn gw, but i
don't have wireshark dump of that side of the traffic. anyway, from the
ua side i see that about 10 milliseconds after receiving the invites, sr
replies to the ua "503 no more gateways available".
in my config that reply comes only from GW_FAILURE failure route,
which is set up like this:
t_on_failure("GW_FAILURE");
t_set_fr("120000", "5000");
t_relay()
so no way should failure route get executed before 5 seconds, but in
this back-to-back invite case it somehow gets executed almost
immediately. i have not seen this behavior in other situation.
this looks like a bug to me. perhaps there is some race condition
causing it? version of sr where this appears is 3.0.3.
-- juha
Hi,
I currently have 1 OpenSer server in front of a number of Asterisk
boxes. I want to move to a scenario where I have multiple OpenSer
boxes serving the same domain name in front of the Asterisk boxes. The
UA could be registered on either of the OpenSer via SRV. When Asterisk
has a call for the UA is it better to get Asterisk to dial both
OpenSER or get one OpenSER to fork the call to the other box?
Any opinions please?
Regards
Jon
--
Jon Farmer
Tel: 07795 118140
Email: viperdudeuk(a)gmail.com
Twitter: @viperdudeuk
Hello.
I’m wondering how the “Weight” column in the LCR ‘gw’ table Works.
I have 5 gateways defined with different weights.
gw_name grp_id ip_addr hostname port uri_scheme
transport strip tag weight ping flags
GW02 58 10.0.0.71 5060 1
1 4 0258 10 0 0
GW03 58 10.0.0.72 5060 1
1 4 0258 10 0 0
GW05 58 10.0.0.132 5060 1
1 4 0258 20 0 0
GW06 58 10.0.0.133 5060 1
1 4 0258 20 0 0
GW09 58 10.0.0.70 5060 1 1
4 0258 20 0 0
GW10 58 10.0.0.69 5060 1
1 4 0258 20 0 0
If I have 100 calls, I was expecting was to have 20 to GW05, 20 to GW06, 20
to GW09 and 20 to GW10
Finalle 10 to GW02 and 10 to GW02.
Is this the correct interpretation of this parameter?
Thanks in advance.
Regards,
Ricardo Martinez.-
Hello to all!
I need a little help with our ser installation (ser-2.0.0-rc1).
The continuous groving up of our infrastructure and using even more codecs, causes the INVITE (udp) to be over 1500bytes. An
external->incoming call to our proxy sip comes in with a size of ~1300 and will be forwarded to the end device (sua) over 1500.
Ser is adding more than 200bytes in the udp packet. I can see 'INVITE' from ser to customer, but the INVITE is not answerred due
the size. I'm sure about that, because in certain situations, deleting codec list the packet-size goes under the 1500bytes and all
works.
To main issue is from external calls to ser server to our customers.
Calls started from customers to ser and then out are all ok.
Now the question: Could the passage from UDP to TCP solve the issue?
Could someone give me an example and how implement it into our ser configuration?
The atteched configuration, ser on a mysql basis, is working well in expect the issue described.
Thank's a lot for your help!
Simon
Greetings,
Since the move to name-based routes in Kamailio v3.0, what
implications does this have for the 'timeout_route' dialog context
attribute? Has this been updated to reflect that change, or do I
still have to use numerical routes for it? Can I now set this as a
named constant, e.g.
$dlg_ctx(timeout_route) = DIALOG_TIMEOUT_ROUTE;
...
route[DIALOG_TIMEOUT_ROUTE] {
...
}
or is it a string?
$dlg_ctx(timeout_route) = "DIALOG_TIMEOUT_ROUTE";
Thanks!
--
Alex Balashov - Principal
Evariste Systems LLC
1170 Peachtree Street
12th Floor, Suite 1200
Atlanta, GA 30309
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
Hello,
Could somebody check if default ser.cfg file is correct in the following section:
route[AUTHENTICATION]
{
....
# check if the UID from the authentication meets the From header
$authuid = $uid;
if (!lookup_user("$fu.uid", "@from.uri")) {
xlog("L_INFO","fu.uid lookup failed\n");
del_attr("$uid");
}
if ($fu.uid != $fr.authuid) {
sl_reply("403", "Fake Identity");
drop;
}
..
Because it didn't work for me till I've replaced:
if (!lookup_user("$fu.uid", "@from.uri")) {
with:
if (!lookup_user("$fr.uid", "@from.uri")) {
Best regards,
Andrey
Hi,
Siremis version 2.0 is planned to work with upcoming Kamailio or SER 3.1
and by now is alpha version.
If you want to have a preview, then you can access it at:
http://demo.asipto.com/siremis20/
username: guest
password: guest
v2.0 brings a completely new face and many features, reusing the latest
openbiz development framework, among them:
- easy 4 step web-based installer
- siremis account management (including roles per users)
- customizable menu from web interface
- customizable views from web interface
- rework of many kamailio/ser related components such as subscriber,
siptrace, generic and usrloc charts
Siremis administration module, user profile management and web based
customization elements are disabled in the demo, to avoid breaking the
content.
Packaging is still under work, but if someone is good at php/xml, likes
to dig in by itself to sort out issues and wants to help with testing, I
can ship a raw tarball, just drop me an email. Feedback about this new
version is also appreciated, in order to improve it.
Regards,
Ramona
Hi all!
I really don't know why "Mitel" rejects my calls. I'm using Kamailio to
forward calls to Mitel.
A little more graphic:
Please see the picture:
http://s3.subirimagenes.com:81/otros/5226539form.jpg
SIP PHONE (Linksys) ---> Kamailio (1.5.4) ----> Mitel ----> Mitel Phone
Mitel rejects my calls with "404 Not Found". Ok, you may think: "the
extension that you are calling doesn't exists".. please dont think that.
(One more thing: If I try to make the same scene using Asterisk instead
Kamailio everything works fine.)
So, I made a sip capture to see what happens:
Sip Phone -> 100
192.168.10.140 -> Sip Phone
192.168.10.150 -> Kamailio
192.168.10.160 -> Mitel
Mitel Phone -> 200
Kamailio
U 192.168.10.140:5060 -> 192.168.10.150:5060
INVITE sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.140:5060;branch=z9hG4bK-d063d53a.
From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150>
>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>.
Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a
@).
CSeq: 101 INVITE.
Max-Forwards: 70.
Contact: "Sip Phone" <sip:100@192.168.10.140:5060>.
Expires: 240.
User-Agent: Linksys/SPA941-5.1.8.
Content-Length: 395.
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.
Supported: replaces.
Content-Type: application/sdp.
U 192.168.10.150:5060 -> 192.168.10.140:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 192.168.10.140:5060
;branch=z9hG4bK-d063d53a;rport=5060;received=192.168.10.140.
From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150>
>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>.
Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a
@).
CSeq: 101 INVITE.
Server: Kamailio (1.5.4-notls (i386/linux)).
Content-Length: 0.
U 192.168.10.150:5060 -> 192.168.10.160:5060
INVITE sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0.
Via: SIP/2.0/UDP 192.168.10.140:5060
;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a.
From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150>
>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>.
Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a
@).
CSeq: 101 INVITE.
Max-Forwards: 69.
Contact: "Sip Phone" <sip:100@192.168.10.140:5060>.
Expires: 240.
User-Agent: Linksys/SPA941-5.1.8.
Content-Length: 395.
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.
Supported: replaces.
Content-Type: application/sdp.
U 192.168.10.160:5060 -> 192.168.10.150:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0,SIP/2.0/UDP
192.168.10.140:5060
;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a.
From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150>
>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>
>;tag=0_4044193584-65506210.
Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a
@).
CSeq: 101 INVITE.
Content-Length: 0.
U 192.168.10.160:5060 -> 192.168.10.150:5060
SIP/2.0 404 Not Found.
Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0,SIP/2.0/UDP
192.168.10.140:5060
;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a.
From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150>
>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>
>;tag=0_4044193584-65506210.
Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a
@).
CSeq: 101 INVITE.
Contact: <sip:192.168.10.160>.
Content-Length: 0.
This is my Kamailio code from reenvites..
route[4] {
t_relay("udp:192.168.10.160:5060");
t_on_reply("1");
exit;
}
If you pay attention to INVITES (Kamailio SIP messages) you will see:
From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150>
>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>.
I think that should be:
From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150>
>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.160 <sip%3A200(a)192.168.10.160>>.
It could be the reason for Mitel rejects? Can I fix it? I can use TEXTOPS
but I cant understand why Mitel rejects the Kamailio INVITES.
I will thanks any help!
