What you are asking to do breaks the protocol rules of SIP. Neither Opensips
nor any other SIP proxy will do it.
I am not sure which you actually wanted as the result as CANCEL and 480 are not
the same thing, but you cannot convert a BYE into either one.
A CANCEL has to refer to an in-progress
It doesn’t sound like it has anything to do with the registration. It sounds
like your router has some sort of SIP Helper application that is trying to
assist by re-writing the ports in the INVITE. Many modern routers come with
this functionality enabled by default, even though in my experience
g example just so I have something else to stare at?
And if anyone else has any bright ideas, I'm definitely all ears (or eyes);
Thanks very much;
- Jock
On Fri, Jan 27, 2017 at 12:17 PM, Newlin, Ben
<ben.new...@inin.com<mailto:ben.new...@inin.com>> wrote:
That is not correct. OpenSIP
That is not correct. OpenSIPS 1.11 definitely contains the index transformation
[1]. I’m using it in my code!
I think the problem is the quoting. For the example that uses a hard-coded
search string it should work if you leave off the quotes. This is how I’m using
it.
For the variable
You said you are doing load balancing as well. Are you doing load balancing on
the ACK? What module are you using (dispatcher, loadbalancer, etc.)?
Load balancing functions can change the R-URI.
Ben Newlin
From: on behalf of Denis
I would recommend just using $du. [1]
$du = “sip:” + $var(Fqdn) + “:5060”;
[1] http://www.opensips.org/Documentation/Script-CoreVar-2-2#toc35
Ben Newlin
From: on behalf of "Ramachandran, Agalya
(Contractor)"
Reply-To:
3eb34e5355e54575f8ec391740d3@192.168.36.68:5060<mailto:7f643eb34e5355e54575f8ec391740d3@192.168.36.68:5060>
CSeq: 103 ACK
User-Agent: FPBX-12.0.76.4(11.5.1)
Content-Length: 0
Regards,
Richard
On 10/10/2016 17:52, Newlin, Ben wrote:
It’s impossible to say whether the ACK or the 4XX re
s!
On Fri, Oct 21, 2016 at 11:10 PM, Alex Balashov
<abalas...@evaristesys.com<mailto:abalas...@evaristesys.com>> wrote:
On 10/21/2016 06:36 PM, Newlin, Ben wrote:
Not only that, but provisional responses (except 100 Trying) are
required to have a To tag [1]. So you would likely r
Not only that, but provisional responses (except 100 Trying) are required to
have a To tag [1]. So you would likely run into issues with UAs if you start
returning messages without them.
You could just drop the provisional replies, but you may run into issues with
timeouts in that case as user
4XX after ACK
The ack is from the cp. opensips is sending the 4xx.
I'll try the same on another provider. Thanks for the info
On 7 Oct 2016 18:27, "Newlin, Ben"
<ben.new...@inin.com<mailto:ben.new...@inin.com>> wrote:
The most likely cause is that there is something wrong in t
t four and it still
sends subsequebt ones
Regards
Richard
On 07/10/2016 19:46, Richard Robson wrote:
The ack is from the cp. opensips is sending the 4xx.
I'll try the same on another provider. Thanks for the info
On 7 Oct 2016 18:27, "Newlin, Ben"
<ben.new...@inin.com<mailt
The most likely cause is that there is something wrong in the format of the ACK
which is causing the far end to not recognize it as being the ACK for that 4XX
response. So the far end will continue to retransmit the 4XX response. On the
OpenSIPS side, these are recognized as retransmissions so
No problem.
Thanks, but I am not using 2.2 and not using the topology_hiding module. I am
using the Dialog module with the topology_hiding function in 1.11.
Ben Newlin
From: Bogdan-Andrei Iancu <bog...@opensips.org>
Date: Friday, September 30, 2016 at 4:39 AM
To: "Newlin, Be
in the logs.
Ben Newlin
From: Bogdan-Andrei Iancu <bog...@opensips.org>
Date: Tuesday, October 4, 2016 at 5:28 AM
To: "Newlin, Ben" <ben.new...@inin.com>, OpenSIPS users mailling list
<users@lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topo
The priority of the codecs is given by their ordering on the “m=audio” line.
PCMU is “0” and you can see in the after SDP that “0” has been moved to the
front of the list.
The ordering of the “a=rtpmap” lines is not important and does not indicate the
preferred codec order.
Ben Newlin
From:
not sure I would recommend it.
Ben Newlin
From: H Yavari <hyav...@rocketmail.com>
Reply-To: H Yavari <hyav...@rocketmail.com>
Date: Wednesday, September 14, 2016 at 2:10 AM
To: "Newlin, Ben" <ben.new...@inin.com>, OpenSIPS users mailling list
<users@lists.opens
<hyav...@rocketmail.com>
Date: Wednesday, September 7, 2016 at 9:26 AM
To: "Newlin, Ben" <ben.new...@inin.com>, OpenSIPS users mailling list
<users@lists.opensips.org>
Subject: Re: [OpenSIPS-Users] mulitple instance for different rules
Hi,
I want to have physical
I would recommend using Docker for this to ensure the 2 instances of OpenSIPS
are isolated from each other. We are running OpenSIPS in Docker with multiple
containers on a single machine with great success. It also makes local testing
very simple, especially with a SIPp container as well.
I see now that status is a global parameter that should also be the reply
status.
I would still recommend using t_check_status().
Ben Newlin
From: <users-boun...@lists.opensips.org> on behalf of "Newlin, Ben"
<ben.new...@inin.com>
Reply-To: OpenSIPS user
The comma in your regex is unnecessary; it allows a match for “30,”. It should
just be “30[12]”.
Also, where is the status variable being set from? There are two places to get
the return code: $rs and $T_reply_code.
Or you could use t_check_status() like so:
if (t_check_status(“30[12]”)) {
take a while for me to
revive an older configuration that allows me to enable the memory debugger.
Ben Newlin
From: Bogdan-Andrei Iancu <bog...@opensips.org>
Date: Tuesday, August 2, 2016 at 3:47 AM
To: "Newlin, Ben" <ben.new...@inin.com>, OpenSIPS users mailling list
<
is not applied and $du is set
to the IP from the incoming message’s Route header, which is my server.
Ben Newlin
From: Bogdan-Andrei Iancu <bog...@opensips.org>
Date: Monday, August 1, 2016 at 7:13 AM
To: "Newlin, Ben" <ben.new...@inin.com>, OpenSIPS users mailling list
<
.
Ben Newlin
From: Bogdan-Andrei Iancu <bog...@opensips.org>
Date: Monday, August 1, 2016 at 10:57 AM
To: OpenSIPS users mailling list <users@lists.opensips.org>, "Newlin, Ben"
<ben.new...@inin.com>
Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
-Andrei Iancu <bog...@opensips.org>
Date: Monday, August 1, 2016 at 10:01 AM
To: "Newlin, Ben" <ben.new...@inin.com>, OpenSIPS users mailling list
<users@lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()
Hi Ben,
1) I'm
of the incompatibility between these
modules would be beneficial as well, as it would have saved me many hours
attempting to make this work and may save others that time as well.
Ben Newlin
From: Bogdan-Andrei Iancu <bog...@opensips.org>
Date: Monday, August 1, 2016 at 7:13 AM
To: "
topology.
Ben Newlin
From: Bogdan-Andrei Iancu <bog...@opensips.org>
Date: Friday, July 29, 2016 at 8:40 AM
To: "Newlin, Ben" <ben.new...@inin.com>, OpenSIPS users mailling list
<users@lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()
This is 1.11.6, running on CentOS 7.
Ben Newlin
From: <users-boun...@lists.opensips.org> on behalf of Bogdan-Andrei Iancu
<bog...@opensips.org>
Reply-To: OpenSIPS users mailling list <users@lists.opensips.org>
Date: Friday, July 29, 2016 at 8:50 AM
To: "Newlin, Be
g list <users@lists.opensips.org>, "Newlin, Ben"
<ben.new...@inin.com>
Subject: Re: [OpenSIPS-Users] How to tell if uac_auth() is successful
Hi Ben
The uac_auth() should return true if the authentication was successfully
performed. Are you sure you correctly perform the test
their behavior.
Ideally the Record-Route headers from previous replies could be used in this
case to allow the call to succeed, but I don’t know if that is possible.
Thanks,
Ben Newlin
From: "Newlin, Ben" <ben.new...@inin.com>
Date: Wednesday, July 27, 2016 at 9:44 AM
To: Bogdan-A
:57 AM
To: OpenSIPS users mailling list <users@lists.opensips.org>, "Newlin, Ben"
<ben.new...@inin.com>
Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
Hi Ben,
First, if you use TH, makes no sense to do Record-Routing - there are 2 SIP
concepts that overlaps. Yo
so there will be no Route/RR headers at all. So no need to do loose_route or
so. You just do TH matching for the sequential requests and nothing more.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 22.07.2016 16:48, Newlin, Ben wrote:
Hi,
I have had 3 OpenSIPS server crashes in the last week. All were due to
segmentation faults. I was not able to capture core dumps; I am configuring
that now to catch the next crash.
My logs leading up to the crash are full of errors from fix_route_dialog()
complaining about invalid URIs for
I am using the UAC and UAC_AUTH modules to perform trunk authentication. The
uac_auth() function will only use credentials when the realm matches one in the
401/407 challenge. Is there any way to tell whether there was a successful
authentication match?
The function doesn’t provide any return
Failure routes, reply routes, and branch routes are all transaction specific
and will only be triggered for the specific transaction for which they have
been armed; in this case, the initial INVITE. If you want replies to sequential
requests to also hit the reply route, you must add
Hi,
I am using the Dialog module with topology_hiding() in my server and I have a
need to Record-Route the call on my server as I am advertising a different
address than I am listening on. I have found what I believe is an inconsistency
in the handling of Record-Route within the Dialog
AVPs are tied to a transaction, so the transaction must be matched before they
will be available. You should use t_check_trans() to do this.
However, I think this will not work for you because ACKs are their own
transactions and I don’t believe they will have access to the AVPs from the
INVITE
;
Date: Tuesday, June 28, 2016 at 5:45 AM
To: "Newlin, Ben" <ben.new...@inin.com>, "users@lists.opensips.org"
<users@lists.opensips.org>
Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong
address in VIA header
Hi Ben,
You mean you used the re
to perform any tests or get tracing/logs. Thanks!
[1] https://github.com/OpenSIPS/opensips/issues/917
Ben Newlin
From: "Newlin, Ben" <ben.new...@inin.com>
Date: Monday, June 27, 2016 at 11:06 AM
To: Bogdan-Andrei Iancu <bog...@opensips.org>, "users@lists.opensips.org"
&l
: Bogdan-Andrei Iancu <bog...@opensips.org>
Date: Monday, June 27, 2016 at 10:41 AM
To: "Newlin, Ben" <ben.new...@inin.com>, "users@lists.opensips.org"
<users@lists.opensips.org>
Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong
addr
appears to be working, but the local generated ACK is not using that address.
Ben Newlin
From: Bogdan-Andrei Iancu <bog...@opensips.org>
Date: Monday, June 27, 2016 at 5:37 AM
To: "users@lists.opensips.org" <users@lists.opensips.org>, "Newlin, Ben"
<ben.new...@ini
I have run into the same problem that was described in this previous post [1],
however it doesn’t appear it was ever solved at the time.
I am using the dispatcher module to route calls to external carriers and I am
using set_advertised_address to set the outgoing public address prior to
It also seems like AVPOPS module [1] may be a good solution here as it has
functions to pull data from a database into AVPs based by user.
[1] http://www.opensips.org/html/docs/modules/2.2.x/avpops.html
Ben Newlin
From: on behalf of sevpal
Bogdan,
Thanks. I assume the parameter will be available in 1.11.9?
Ben Newlin
From: Bogdan-Andrei Iancu <bog...@opensips.org>
Date: Tuesday, June 21, 2016 at 9:42 AM
To: OpenSIPS users mailling list <users@lists.opensips.org>, "Newlin, Ben"
<ben.new...@inin.com
In my script I send all “100 Trying” responses manually* and use the 0x01 flag
when calling t_relay so that it will not send its own “100 Giving a try”
response, as per the documentation [1].
Unfortunately, in 1.11.7 and 1.11.8 this flag seems to have stopped working
which results in multiple
It looks like a new tarball for OpenSIPS 1.11.7 was uploaded earlier today, but
it appears to have a bad format. I cannot extract it with any of my usual
archive utilities. Can anyone else confirm?
It was working fine yesterday with the previous tarball.
You could also have wrapped the DB_PASS in an extra set of m4-style quotes:
define( `DB_PASS',``some#password'')
Ben Newlin
From:
> on
behalf of John Nash
Stefan,
$var type variable have process-level scope [1]. This means they are only valid
in the same script process that initialized them. It is very likely in your
case that the startup_route is running in a different child process than the
message processing routes and that is causing the
In on_reply_route all pseudo-variables are evaluated by default in the context
of the reply, but in failure_route they are evaluated by default in the context
of the request. I think you are probably getting a different value for $si than
you are expecting in the failure_route case and there is
In the case where you perform the SQL query I don’t see where you are actually
setting the result into the Request-URI. You seem to only be storing it in the
cache. I think you need to add:
$rd = $avp(PBX);
to the else leg.
Ben Newlin
From:
I know I have used presence extensively with Polycom phones using pidf+xml. I
know they support it. Maybe there is some setting in your model specifying the
remote server type? If that is set to Microsoft Lync the Polycom may be sending
xpidf for compatibility.
But Polycom phones absolutely
If you use uac_replace_from in a branch route then the changes are specific to
that branch. If the call fails the header will be reverted and can be modified
again in another branch.
Ben Newlin
From:
> on
behalf of
users mailling list
Date: Wednesday, August 12, 2015 at 2:40 AM
To: OpenSIPS users mailling list, Newlin, Ben
Subject: Re: [OpenSIPS-Users] CARRIERROUTE module
Hi Ben,
The carrierroute module is not maintained (for like 6 years).
Still, if you want to do a routing based on both caller and caller
Hi all,
We have some interest in possibly using the CARRIERROUTE module instead of
DROUTING as it allows prefix matching of the originating number as well as the
destination, whereas DROUTING must match the originating user exactly. However,
we noticed that the CARRIERROUTE module is labeled
/expanded headers.
Ben Newlin
From: Bogdan-Andrei Iancu
Date: Tuesday, August 4, 2015 at 12:25 PM
To: OpenSIPS users mailling list, Newlin, Ben
Subject: Re: [OpenSIPS-Users] Compact/Full Headers
Hi Ben,
The OpenSIPS 2.1 has the compression module that is able to do that for you
automatically
that for any
given header?
Ben Newlin
From: Newlin, Ben
Date: Friday, July 10, 2015 at 5:54 PM
To: users@lists.opensips.orgmailto:users@lists.opensips.org
Subject: Compact/Full Headers
Is there a simple way to convert Compact headers to their full form and vice
versa while processing a message? I
We are using the topology_hiding functionality of the Dialog module in OpenSIPS
1.11. We have 2 OpenSIPs servers set up in tandem, with one acting as the
load-balancer and the other as the proxy. Only the load-balancer has a public
IP address, so on the proxy we use
Is there a simple way to convert Compact headers to their full form and vice
versa while processing a message? I haven’t found much mention of compact
headers except that they are supported. I also saw the new compression module
in 2.1, but we are using 1.11.
Ben Newlin
to the first index
of avp
On Jul 1, 2015 3:44 PM, Newlin, Ben
ben.new...@inin.commailto:ben.new...@inin.com wrote:
Sorry, actually the documentation confused me.
You are correct in the way that it should work if the do_routing() and
route_to_gw() are really pushing the functions on the end
access the avp, we can either get explicitly the specific index, or
we get the last index which is `gw2`.
So if I'm not mistaken, then I don't understand what the problem is here?
Correct me if this is wrong.
On Wed, Jul 1, 2015 at 11:46 AM, Newlin, Ben
ben.new...@inin.commailto:ben.new
is `gw2`.
So if I'm not mistaken, then I don't understand what the problem is here?
Correct me if this is wrong.
On Wed, Jul 1, 2015 at 11:46 AM, Newlin, Ben
ben.new...@inin.commailto:ben.new...@inin.com wrote:
I found similar behavior in my implementation, except that I use dr_routing()
instead
I found similar behavior in my implementation, except that I use dr_routing()
instead of route_to_gw(). When the AVP is already populated, these functions
simply push on the end. This may be an implementation decision, but it is
certainly not intuitive or what I expected.
I am currently using
NOTIFY to the phone.
The only thing is that the packet has 2 Via headers with the same IP and port
of OpenSIPS server.
As I said, it works, but looks weird.
On Wed, Jun 3, 2015 at 7:00 PM, Newlin, Ben
ben.new...@inin.commailto:ben.new...@inin.com wrote:
It sounds like you may be sending
It sounds like you may be sending the NOTIFY to yourself when you use the
domain name instead of the IP. Have you verified the address that the domain
resolves to? Is it the same as the OpenSIPS instance?
Ben Newlin
From: Stas Kobzar
Reply-To: OpenSIPS users mailling list
Date: Wednesday, June
James,
Only SIP Requests are processed by the route function. That is why it makes
sense for the INVITEs, REGISTERs, and OPTIONs, BYEs, ACKs, etc. The 486 is a
reply and is handled by OpenSIPS according to normal SIP rules. It will not
cause the script to run at all unless you are using the
The usage of $dlg_val is described in the module documentation [1]. You set and
read them just like any other pseudo-variable:
$dlg_val(foo) = “bar”;
The only thing I don’t know is whether these variables can be accessed from the
E_DLG_STATE_CHANGED event route, as it is not clear whether the
You can add variables to the Dialog using $dlg_var.
Ben Newlin
From: DanB
Reply-To: OpenSIPS users mailling list
Date: Monday, May 11, 2015 at 8:11 AM
To: Bogdan-Andrei Iancu, OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Dialog start event over EVI
For now no emergency since we
There is also a function in the URI module for checking if the user portion of
a URI is an E.164 number.
http://www.opensips.org/html/docs/modules/1.11.x/uri.html#id294513
From: Podrigal, Aron
Reply-To: OpenSIPS users mailling list
Date: Thursday, April 30, 2015 at 1:20 PM
To: OpenSIPS users
Something in between is manipulating the addressing in the SIP message. You
said that ALG was disabled in the router, but either that is incorrect or there
is some other piece of equipment changing the message.
This is clear because the contact is not the only header that is changed. The
do ALG?
On Thu, Apr 9, 2015 at 10:39 AM, Newlin, Ben
ben.new...@inin.commailto:ben.new...@inin.com wrote:
Something in between is manipulating the addressing in the SIP message. You
said that ALG was disabled in the router, but either that is incorrect or there
is some other piece of equipment
“$rU@$rd:$rp”
$rU – R-URI Username
$rd – R-URI Domain
$rp – R-URI Port
http://www.opensips.org/Documentation/Script-CoreVar-1-11
Ben Newlin
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Satish Patel
Sent: Tuesday, April 07, 2015 10:23 AM
To:
Terrance,
Yes, I am sure it works. I have changed the values in onreply_route and see
them change in the message. I believe Razvan’s comment applies only when
operating in normal proxy mode. The usage of the Dialog module and
topology_hiding() is what makes it work.
Ben Newlin
From:
Yes, topology_hiding(). I am actually doing exactly what you have described
wanting to do. I am using set_advertised_address() in request route and then in
onreply_route to change the address in each direction. When topology_hiding()
is being used, the module is already going to replace the IP
Terrance,
I am doing something similar and have found the “topology hiding” feature of
the dialog module will do this. The feature’s purpose is to re-write all IP
information in SIP headers to hide network topology, but in the process it will
also perform the type of translation you are
-Andrei Iancu [mailto:bog...@opensips.org]
Sent: Thursday, March 26, 2015 10:02 AM
To: OpenSIPS users mailling list; Newlin, Ben
Subject: Re: [OpenSIPS-Users] DROUTING module changing Carrier IDs
Hi Ben,
If you run the dr_carrier_status MI command, do you see the same #012 in the
ID of the carrier
places in the logs, which also do not appear when logging to stderr.
Sorry for the false alarm, but I appreciate the help to point me in the right
direction.
Ben Newlin
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Newlin, Ben
Sent: Thursday, March 26
I am using OpenSIPS 1.11 and I am recently getting some weird behavior from the
DROUTING module. The module is corrupting the Carrier ID that is being returned
via AVP. This behavior is occurring only with the Carrier ID, not the gateway
or rule IDs, and only when it is returned from the
| ben.new...@inin.com
Interactive Intelligence
Deliberately Innovative
www.inin.comhttp://www.inin.com/
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: Friday, December 26, 2014 2:59 AM
To: OpenSIPS users mailling list; Newlin, Ben
Subject: Re: [OpenSIPS-Users] B2BUA Proxy Interaction
Hi
I have been reading through the documentation for the B2BUA functionality. I
want to use a B2BUA while also having many of the proxy functionalities like
drouting, accounting, etc. However, it is indicated that they do not work
easily together.
The B2BUA Tutorial says this:
A solution is to
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