Hello Bogdan,
Now we changed the behaviour of the UAC. One of them will send a BYE and
this is relayed to the PSTN GW which drops the call, since opensips will
not handle the BYE locally. So loose_route is done and the BYE is
relayed to the PSTN GW.
The following is happening:
1) INVITE from
Hi
After re-subscribing to a user presence, we receive immediately a NOTIFY
with no content:
NOTIFY sip:us...@xx.xx.xx.xx:5060 SIP/2.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6667;branch=z9hG4bK34ab.7700bc14.0
To: sip:us...@mycompany.com sip%3aus...@mycompany.com;tag=14257163
From: sip:us...@mycompany.com
Hi,
I am facing the issue posted previously.
It is very clear what happens. Your rls-services document looks like this:
rls-services xmlns:rl=urn:ietf:params:xml:ns:resource-lists
xmlns=urn:ietf:params:xml:ns:rls-services
service uri=sip:alice at net1.test
This is the mail system at host posta.websolutions.it.
I'm sorry to have to inform you that your message could not
be delivered to one or more recipients. It's attached below.
For further assistance, please send mail to postmaster
If you do so, please include this problem report. You can
delete
Hi all.Anybody know what is the schema tel-uri? OpenSIPS works with the schema tel-uri?Thanks.Airton Kuada
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Hi Sanjeev,
Unfortunately this feature is not implemented in 1.6 either.
Regards,
Anca
Sanjeev BA wrote:
Hi,
I am facing the issue posted previously.
It is very clear what happens. Your rls-services document looks like this:
rls-services xmlns:rl=urn:ietf:params:xml:ns:resource-lists
There are two major issues related to the implementing of such feature:
1. You expose your server to information provisioned in a remote system
2. OpenSIPS child may block waiting for the response from the remote
XCAP server
So though this feature is well defined in the standards, it can
Hello,I dont see the parameter append_branches in the registrar module of
1.6 version. I was using it so that Openser does not route calls to all the
AORs. My requirement was that the last registered client should get the
call.
So i used append_branches as 0 in the registrar module and
I'm not sure if it's a real good idea to rerwrite the contact header.
That may break loose routing? Do you have a good reason to do this?
On Tue, Oct 20, 2009 at 8:22 PM, Alex Balashov
abalas...@evaristesys.com wrote:
Try escaping the : as well. They have special meaning in regex.
Ariadne
Hi,
I have the following requirement:
If a from tm generated cancel is answered with a 200 OK I want to send a
BYE to the UAC.
Is this possible?
BR
Uwe
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Thanks Khan. I tried this before and it did not work, but your documentation
is the most comprehensive one, great job. May be i need some more tune up
for my setup. Basically I can make a call from UA behind NAT to PSTN through
opesips/rtpproxy outbound. I have installed opensips and rtpproxy on
I have a server located on the internet running opensips and asterisk. When
registering directly to asterisk I can perform echo tests and make calls.
If I register to Opensips and use the load_balance there is one way audio. I
can hear sounds coming from the asterisk server but sound from
NHi Duane,
There are is a firewall on the server end however all ports are open, no NAT at
the server end however there is NATing on the end of the soft phone. Though
when registering with asterisk directly there is no issue.
Regards,
Ross
Date: Wed, 21 Oct 2009 15:23:04 +
So in the wireshark trace you see RTP traffic coming from the Asterisk
servers IP address, but what about the traffic coming from the softphone?
What IP address is that going towards?
On Oct 21, 2009 10:35am, Ross Beer ross_b...@hotmail.com wrote:
NHi Duane,
There are is a
Hi,
I am trying to get an RLS server setup working and find the following issue.
Client sends SUBSCRIBE with list URI. RLS expands the list URI to individual
URIs.
Sends individual subscribe to Presence Server (co-located)
Presence server responds with 202 or 200 response code.
I have
Yep, traffic comes from the asterisk server and can be heard on the softphone,
but when the echo test starts no audo can be heard.
Therfore the flow goes like this:
Asterisk --- Opensips Softphone
But NOT:
Softphone --- Opensips Asterisk
Which is strange, if opensips is not
I have installed rtpproxy on the same host as that of opensips, which as
only one external public IP. My rtpproxy is not running bridge
mode. Rtprpoxy NAT traversal is not working for users from 2 different
network. To make this case work, should I use bridge mode ? with 2 IP
interfaces ? or is
No.
--
Sent from mobile device
On Oct 21, 2009, at 9:34 AM, Uwe Kastens ki...@kiste.org wrote:
Hi,
I have the following requirement:
If a from tm generated cancel is answered with a 200 OK I want to
send a
BYE to the UAC.
Is this possible?
BR
Uwe
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kiste lat: 54.322684,
Hi Alex,
Any other option to solve this 200 OK for INVITE relayed after CANCEL
issue with opensips and asterisk?
http://lists.kamailio.org/pipermail/devel/2008-August/015209.html
BR
Uwe
Alex Balashov schrieb:
No.
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Sent from mobile device
On Oct 21, 2009, at 9:34 AM, Uwe Kastens
I am also setting up FindMe/FollowMe type conditional based routing on a test
box and have got it working with avp_db_query. I am inserting the mobile
and home numbers for the called user into the location database table with
specific Q values depending on how the user wants the phones to ring
It looks like it is sending in to the server's IP address and back to it's self
which is strange.
I think this has something to do with the SDP and possibly my router. I am
doing an echo test so audio should come back, however Asterisk should stay in
the media path as it does when directly
I need some advice:
I have a test case that looks like this:
outside customer calls a phone number, number is busy.
opensips looks up the customer preference and forwards the busy call to another
phone.
the first (busy) number is behind a nat.
the second is not.
I am using rtpproxy for my
Do any one know if there is a function like fixed_nated_subscribe() , just
like fix_nated_register() ? I'm facing problem where opensips is not fixing
the natted subscribes ? I'm using rtpproxy for INVITE/Rtp and it works fine.
Please help
Mani
On Wed, Oct 21, 2009 at 3:33 PM, presc...@wcoil.com
El Miércoles, 21 de Octubre de 2009, Airton Kuada escribió:
Hi all.
Anybody know what is the schema tel-uri?
http://www.google.com/search?q=tel+URI+RFCie=UTF-8oe=UTF-8
OpenSIPS works with the schema tel-uri?
Not very well but it accepts it and parses it as follows:
- username = TEL
El Miércoles, 21 de Octubre de 2009, Manivasagam Sivaraman escribió:
Do any one know if there is a function like fixed_nated_subscribe() , just
like fix_nated_register() ? I'm facing problem where opensips is not fixing
the natted subscribes ? I'm using rtpproxy for INVITE/Rtp and it works
Hi,
I have a question related to my load balancing configuration of opensips.
I have an X-Lite softphone that connects to Opensips server, which transfers
the INVITE request to one of the asterisk boxes.
All of them are behind firewall on the same network. Then asterisk calls to
my cell phone
On Wednesday 21 October 2009 23:13:36 Justin L wrote:
Hi,
I have a question related to my load balancing configuration of opensips.
I have an X-Lite softphone that connects to Opensips server, which
transfers the INVITE request to one of the asterisk boxes.
All of them are behind firewall
Here is the INVITE:
INVITE sip:13101234...@ask00-rvn SIP/2.0
Record-Route: sip:10.1.3.130;lr;ftag=c020195b;did=d08.3a8259b2
Via: SIP/2.0/UDP 10.1.3.130;branch=z9hG4bK88c5.4ae45bf5.0
Via: SIP/2.0/UDP 172.16.100.159:21874
Hi,
you can do the Follow me / call hunting by placing the additional
destinations in the usrloc (and use serialize), or you can let usrloc to
work in the normal way (no permanent contacts) and in failure route,
when you got the negative reply from the registered phone, you can load
the new
Hi Uwe,
as I understand from you, from end devices (GW, as1 and as2) everything
work ok, but the dialog state on opensips is not properly kept??
Regards,
Bogdan
Uwe Kastens wrote:
Hello Bogdan,
Now we changed the behaviour of the UAC. One of them will send a BYE and
this is relayed to the
Hi Jayesh,
The append_branch() module param was converted to a function param for
the lookup(). See:
http://www.opensips.org/html/docs/modules/devel/registrar.html#id271025
the b flag.
Regards,
Bogdan
Jayesh Nambiar wrote:
Hello,
I dont see the parameter append_branches in the
Hi,
Running 1.6 non SVN we are getting random crashes, it appears this is
from the uac_replace_from and uac_replace_to, we did not have this
problem until we started using these function. Below is the bt from gdb.
The only errors we get in the logs are from memcache, which I think are
Hi Uwe,
I'm trying to follow why you actually need this ? sorting out multiple
200 OKs of a call is not typically a just for a proxy, but rather
something that needs to handled between end points.
There are ways to do the BYEing on OpeSIPS (for the second 200 OK), but
I have the feeling that
Yeah. I did some more testing with manually placing temporary records in
usrloc and I am thinking thats not a good method because if one of the
internal users gets called by multiple people at the same time then my
current config would place duplicate Home and Mobile records in usrloc and
that
Hi Ross,
Actually you do not need any media relay (mediaproxy or rtpproxy) here.
As time as Asterisk is on the public side, it should directly work even
with a natted client.
What you have to check is the SDP received by the nated client in the
200 OK - check what IP it is instructed to send
Hi Mani,
so opensips and rtpproxy are situated in the public internet and you
have clients behind different NATs ? is this correct? If so, you do not
bridging mode for RTPproxy.
Regards,
Bogdan
Manivasagam Sivaraman wrote:
I have installed rtpproxy on the same host as that of opensips, which
Hi Kelly,
There are 2 approaches:
1) if you enabled rtpproxy in request route for the INVITE, then,
whatever branches you keep forking, take care and force rtpproxy in all
200 OK (whatever branch - nated or not).
2) enable rtproxy individually, per branch - instead of using the
request route
Hi Justin,
a trace means all SIP messages from that call (not only the INVITE) :).
Also, audio problem means there is not audio at all or means you have
one way audio ?
Regards,
Bogdan
Justin L wrote:
Here is the INVITE:
INVITE sip:13101234...@ask00-rvn SIP/2.0
Record-Route:
Hi Brad,
Do you use both replace_to and replace_from in the same time ?
Do you still have the core file ? could you check in frame 0 for
request and request-dst_uri ?
Thanks and regards,
Bogdan
Brad Bendy wrote:
Hi,
Running 1.6 non SVN we are getting random crashes, it appears this is
Hi Bogdan,
Well, sometimes we do, sometimes we only call or the other, 90% of the
time we use both on the same call.
I do still have the core, inside of gdb do I need to run a command?
When I run "bt", frame 0 shows: #0 0x000801517b39 in
pre_print_uac_request (t=0x802976678, branch=1,
#0 0x000801517b39 in pre_print_uac_request (t=0x802976678,
branch=1, request=0x801647020) at t_fwd.c:132
132 memcpy( p, request-dst_uri.s,
request-dst_uri.len);
$2 = (struct sip_msg *) 0x801647020
(gdb) print request-dst_uri
$3 = {s = 0x0, len = -1}
I see len = -1, so basically a value
Brad,
Please apply the attached patch and see if it's fixing the problem for you.
Regards,
Bogdan
Brad Bendy wrote:
#0 0x000801517b39 in pre_print_uac_request (t=0x802976678,
branch=1, request=0x801647020) at t_fwd.c:132
132 memcpy( p, request-dst_uri.s,
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