Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.11.0 major

[Kneeoh]

I'd like to request distributed dialogs via the cachedb interface for couchbase.

Thank you


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Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.11.0 major release

[Ryan Bullock]

List looks good.

I think the B2B could use some work for its DB interaction. Similar to what
was done for the dialog module (separate timer process, bulk deletes, etc).
We notice that a B2B instance running a similar load as a proxy w/ dialogs
hits the DB quite a bit harder.

Regards,

Ryan


On Mon, Nov 4, 2013 at 10:09 AM, Saúl Ibarra Corretgé
wrote:

> Hi Bogdan,
>
> Can we have "async TLS" added to the list?
>
> Nice list for Santa, btw :-P
>
> On Nov 4, 2013, at 6:58 PM, Bogdan-Andrei Iancu 
> wrote:
>
> > Hi all,
> >
> > I would like to start a discussion about the next OpenSIPS major release
> > - and in this discussion anyone is welcomed with options, ideas, critics
> > and other. Your feedback is important to drive the project into a
> > direction that reflects the user's needs!.
> >
> > So, I will list here the starting points, for both release planing and
> > release content.
> >
> >
> > Content
> > ---
> > What was done:
> >http://www.opensips.org/About/Version-1-11-0#toc2
> > What is planned:
> >http://www.opensips.org/About/Version-1-11-0#toc9
> > Planned items have priorities (for being addressed); it is a must to
> > have all items done for the next release, as we need to fit into a time
> > frame. Whatever is not done, will be left for the next release (1.12 ?)
> >
> > Additional thinks (not listed on web) we are considering are:
> >- new "call queuing" module
> >- new SMPP module
> >- async operations at script level (doing async db ops, exec, rest
> > queries)
> >- dropping avpops module (and replacing with dbops module)
> >- simplify scripting/logic by dropping the usage of AVPs (defined as
> > module params) in favor of explicit func. params
> >- better handling of UAC transactions (being able to set failure
> > routes for them, to fire new requests from script)
> >- Quality routing in Dynamic Routing
> > Also we target so work in the RTPProxy area (still under heavy planing)
> > like restart persistence, replication and statistics .
> >
> >
> > Planing
> > ---
> > Release candidate:
> >second half of January 2014, depending on the progress with the
> > items to be done.
> > Testing phase:
> >1 month allocated (it may be extended if critical problems show up)
> > Stable release:
> >second half of February (after the testing phase is done).
> >
> >
> > Once again, your feedback on these matters is important to us.
> >
> >
> > Best regards,
> >
> > --
> > Bogdan-Andrei Iancu
> > OpenSIPS Founder and Developer
> > http://www.opensips-solutions.com
> >
> >
> > ___
> > Devel mailing list
> > de...@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
>
>
>
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.11.0 major release

[Saúl Ibarra Corretgé]

Hi Bogdan,

Can we have "async TLS" added to the list?

Nice list for Santa, btw :-P

On Nov 4, 2013, at 6:58 PM, Bogdan-Andrei Iancu  wrote:

> Hi all,
> 
> I would like to start a discussion about the next OpenSIPS major release
> - and in this discussion anyone is welcomed with options, ideas, critics
> and other. Your feedback is important to drive the project into a
> direction that reflects the user's needs!.
> 
> So, I will list here the starting points, for both release planing and
> release content.
> 
> 
> Content
> ---
> What was done:
>http://www.opensips.org/About/Version-1-11-0#toc2
> What is planned:
>http://www.opensips.org/About/Version-1-11-0#toc9
> Planned items have priorities (for being addressed); it is a must to
> have all items done for the next release, as we need to fit into a time
> frame. Whatever is not done, will be left for the next release (1.12 ?)
> 
> Additional thinks (not listed on web) we are considering are:
>- new "call queuing" module
>- new SMPP module
>- async operations at script level (doing async db ops, exec, rest
> queries)
>- dropping avpops module (and replacing with dbops module)
>- simplify scripting/logic by dropping the usage of AVPs (defined as
> module params) in favor of explicit func. params
>- better handling of UAC transactions (being able to set failure
> routes for them, to fire new requests from script)
>- Quality routing in Dynamic Routing
> Also we target so work in the RTPProxy area (still under heavy planing)
> like restart persistence, replication and statistics .
> 
> 
> Planing
> ---
> Release candidate:
>second half of January 2014, depending on the progress with the
> items to be done.
> Testing phase:
>1 month allocated (it may be extended if critical problems show up)
> Stable release:
>second half of February (after the testing phase is done).
> 
> 
> Once again, your feedback on these matters is important to us.
> 
> 
> Best regards,
> 
> -- 
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> 
> 
> ___
> Devel mailing list
> de...@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/devel

--
Saúl Ibarra Corretgé
AG Projects




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[OpenSIPS-Users] [RELEASES] Planing OpenSIPS 1.11.0 major release

[Bogdan-Andrei Iancu]

Hi all,

I would like to start a discussion about the next OpenSIPS major release
- and in this discussion anyone is welcomed with options, ideas, critics
and other. Your feedback is important to drive the project into a
direction that reflects the user's needs!.

So, I will list here the starting points, for both release planing and
release content.


Content
---
What was done:
http://www.opensips.org/About/Version-1-11-0#toc2
What is planned:
http://www.opensips.org/About/Version-1-11-0#toc9
Planned items have priorities (for being addressed); it is a must to
have all items done for the next release, as we need to fit into a time
frame. Whatever is not done, will be left for the next release (1.12 ?)

Additional thinks (not listed on web) we are considering are:
- new "call queuing" module
- new SMPP module
- async operations at script level (doing async db ops, exec, rest
queries)
- dropping avpops module (and replacing with dbops module)
- simplify scripting/logic by dropping the usage of AVPs (defined as
module params) in favor of explicit func. params
- better handling of UAC transactions (being able to set failure
routes for them, to fire new requests from script)
- Quality routing in Dynamic Routing
Also we target so work in the RTPProxy area (still under heavy planing)
like restart persistence, replication and statistics .


Planing
---
Release candidate:
second half of January 2014, depending on the progress with the
items to be done.
Testing phase:
1 month allocated (it may be extended if critical problems show up)
Stable release:
second half of February (after the testing phase is done).


Once again, your feedback on these matters is important to us.


Best regards,

-- 
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


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Re: [OpenSIPS-Users] FIFO from scripts

[Saúl Ibarra Corretgé]

Hi!

On Nov 4, 2013, at 4:49 PM, Andreas Sikkema  wrote:

> Hi,
> 
> What are OpenSIPS users using to talk to OpenSIPS (FIFO probably)?
> 

We use FIFO.

> I want to run a cron job every now and then to check if there are calls 
> running for a very long time and close them.
> 
> I can do this by hand and than it's not difficult. I'd *love* to do this 
> using a python script.

> Does anyone have an example on how to talk to the FIFO from python?

Have a look at this: 
http://mediaproxy.ag-projects.com/projects/mediaproxy/repository/entry/mediaproxy/interfaces/opensips.py

that's how MediaProxy communicates with OpenSIPS for ending dialogs. Extending 
it to support other fifo methods should be pretty straightforward.


Cheers,

--
Saúl Ibarra Corretgé
AG Projects




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Re: [OpenSIPS-Users] FIFO from scripts

[Arnold Vriezekolk NETZOZEKER B . V .]

Hi Andreas,

First off, wouldn't you use the fr_timer and fr_inv_timer for this purpose? [1]
Or maybe the default_timeout from the dialog module? [2]

Anyway, the way i use the fifo from my scripts is via de mi_datagram module. [3]
You can just make an udp socket that you can connect to from your script and 
execute commands / receive replies from it. As explained in [4]

Best Regards,
Arnold Vriezekolk

[1] http://www.opensips.org/html/docs/modules/1.8.x/tm.html#id250196
[2] 
http://www.opensips.org/html/docs/modules/devel/dialog.html#default-timeout-id
[3] http://www.opensips.org/html/docs/modules/devel/mi_datagram.html
[4] http://www.opensips.org/html/docs/modules/devel/mi_datagram.html#id248952

On Mon, 4 Nov 2013, Andreas Sikkema wrote:


Hi,

What are OpenSIPS users using to talk to OpenSIPS (FIFO probably)?

I want to run a cron job every now and then to check if there are calls running 
for a very long time and close them.

I can do this by hand and than it's not difficult. I'd *love* to do this using 
a python script.

Does anyone have an example on how to talk to the FIFO from python?

--
Andreas




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[OpenSIPS-Users] FIFO from scripts

[Andreas Sikkema]

Hi,

What are OpenSIPS users using to talk to OpenSIPS (FIFO probably)?

I want to run a cron job every now and then to check if there are calls
running for a very long time and close them.

I can do this by hand and than it's not difficult. I'd *love* to do this
using a python script.

Does anyone have an example on how to talk to the FIFO from python?

-- 
Andreas
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Re: [OpenSIPS-Users] Invite with Replaces header

[Nick Altmann]

You may save callid in dialog variables for each dialog. When you receive
INVITE with Replaces you can check dialogs with saved callid using
get_dialog_info() function.

Or you may use cachedb.

--
Nick


2013/11/4 Dani Popa 

> Hi all,
>
> There is any way to check if Opensips instance have dialog in any state
> defined by  Replaces Header of new incoming  call ?
>
> --
> Dani Popa
>
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Re: [OpenSIPS-Users] 481 Call does not exists

[Bogdan-Andrei Iancu]

Hi Miha,

You mean the BYEs at lines 426 and 438 ? If so, note that they have some
Call-IDs different from whatever other Call-ID you have in that trace -
they do not match anything
Are they maybe related to some previous call ?? Considering both BYEs
have the same CallID it means they belong to the call, maybe different
legs of the same call ; but as both legs are to FS, it only match the
case when the call from FS was sent to OS, OD received a 3xx and
redirected call back to FS (like the second callid you have in the trace).

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 11/04/2013 11:56 AM, Miha wrote:
> Hi Bogdan,
>
> this is not the same thing as we were taling last time. I manage to
> get accouting working by your help and now all CDR works as it should:)
>
> But this is different thing as I do not know why Opensips send two
> more BYE request and FS replay with 481 code.
>
> Here is a sip trace: http://pastebin.com/zdPfuQUt
>
> Thank you!
>
> Miha
>
>
> Dne 10/30/2013 9:35 AM, piše Bogdan-Andrei Iancu:
>> Hi,
>>
>> Issue sorted out off list, by using a custom hdr to link the two SIP
>> calls (for accounting purposes).
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>>
>>
>> On 10/24/2013 11:03 AM, Miha wrote:
>>> Hi,
>>>
>>> I need a little help.
>>>
>>> When I am having 302 that is handled by Opensips I am gettinf 481 by
>>> FS when opensips sends BYE request.
>>>
>>> I am guessing that the problem is with initial invite due to port 5080
>>> invite but opensips send to 5060, please let me know if I am wrong:)
>>>
>>> How can I change this behaviour as BYE is not send by Opensips and
>>> call is not droped as it should on other side.
>>>
>>> Here is a sip trace:
>>> http://pastebin.com/pwwuY4Uu
>>>
>>> Tnx!
>>>
>>> Miha
>>>
>>> ___
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>>>
>

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[OpenSIPS-Users] Invite with Replaces header

[Dani Popa]

Hi all,

There is any way to check if Opensips instance have dialog in any state
defined by  Replaces Header of new incoming  call ?

-- 
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Re: [OpenSIPS-Users] opensips 1.10 Bug

[M.Khaled W Chehab]

Please can you update me if it's a bug or  it should has a special coding ?

 

Regards

 

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of M.Khaled W Chehab
Sent: Friday, November 01, 2013 3:55 PM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] opensips 1.10 Bug

 

Dears,

 

 

Kinldy note that the  modparam("acc", "db_extra_bye", "sender=
$DLG_end_reason ") is   working in route  script and  its value inserted in
acc table in mysql,but in 

 

But in   

local_route{ 

the $DLG_end_reason is printed correctly in xlog  but it has a  have a null
value in acc table 

 

 

please advise

 

 

Khaled Chehab

Senior Network Engineer

Description: icucall

Operations Office - Lebanon

Office: +961 1 515155 ext 300

Mobile  : +961 3 045212

E-mail: kche...@icucall.com

MSN ID :khalidche...@hotmail.com 

Skype: k_chehab 

Web Site: http://www.icucall.com  

 http://www.allohi.com  

 

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Re: [OpenSIPS-Users] OpenSIPS mhomed=yes and shared IP address using corosync

[Bogdan-Andrei Iancu]

Hi Remco,

OK, if your approach does not work, keep in mind you can still use
local_route to change the outbound socket for the probing OPTIONs.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 11/02/2013 02:50 PM, Remco . wrote:
> Hi Bogdan,
>
> Thanks for your reply. The feature of setting the probe interface for
> DR would be great. In the meantime, I studied the exact behavior of
> the IPaddr2 resource agent a bit more. It turns out it uses iproute2
> to bind the address. The VIP is added as a secondary IP address to the
> interface - no wonder the kernel picks the primary. I will see if I
> can modify the resource agent a bit so it will add the VIP as the
> primary IP (swap the IP addresses round). That way, the mhomed=yes
> option will work. I will report back my findings.
>
> Thanks,
>
> Remco.
>
>
> On Fri, Nov 1, 2013 at 12:27 PM, Bogdan-Andrei Iancu
> mailto:bog...@opensips.org>> wrote:
>
> Hello Remco,
>
> In mhomed, yes you let the kernel to pick the source IP based on
> the routing table - so this approach delegate the logic from
> OpenSIPS to the kernel. And it is up to ho well the network part
> is set.
>
> In the future I would like to add to the DR module the possibility
> to set the probing interface (as you have now in the dispatcher
> module). For now, what you can do is to use the local_route to
> catch the DR pings and use force_send_socket() to change the
> outgoing interface.
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
> On 11/01/2013 10:39 AM, Remco . wrote:
>> Hi all,
>>
>> I have a clustered OpenSIPS setup (using corosync with a virtual
>> IP address). This has been working great over the last couple of
>> years. I now want to add an extra IP address to the boxes, again
>> floating a VIP over these interfaces. These interfaces will be
>> used to communicate with PSTN gateways. I noticed however upon
>> enabling these interfaces, the drouting module starts to ping the
>> gateways using the wrong source address, i.e
>>
>> 1.1.1.1 = VIP on eth0
>> 2.2.2.2 = VIP on eth1
>>
>> OpenSIPS is configured to listen on the the two VIPs with a
>> listen directive.
>>
>> According to the kernel's routing table, it should use 2.2.2.2
>> but it uses 1.1.1.1 which results in failure. As I understood,
>> mhomed=yes should achieve just this behavior by asking the kernel
>> for the appropriate source address on sending out a packet.
>> However, when I enable the mhomed option, OpenSIPS starts to
>> complain about not having a socket to send out the packets. I
>> assume this is caused by the kernel returning the real IP from
>> the interfaces (first) instead of the VIPs.
>>
>> Just because of the dynamic nature of the interface selection, I
>> won't be able to use force_send_socket().
>>
>> I know this question has come up on the list on several
>> occasions, but nothing recent and I was still wondering if
>> someone has a workaround or solution for this. I can imagine when
>> using OpenSIPS as a load-balancer with two interfaces (in and
>> out) you might encounter this problem as well if you try to add
>> high availability (in which you often cannot avoid the Virtual IP
>> scenario).
>>
>> Thanks,
>> Remco.
>>
>>
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Re: [OpenSIPS-Users] uac_auth

[Bogdan-Andrei Iancu]

  
  
Hi Rik,
  
  The truth is in the middle. The second invite from opensips (the
  one with credentials) must not be considered a retransmission - it
  has a totally different VIA branch -> different transaction.
  Also, OpenSIPS should increase the CSeq when answering to the
  challenge, but not able to do so as OpenSIPS is mainly a SIP
  proxy, not a b2bua.
  
  Regards,
  

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 11/04/2013 12:22 PM, Rik Broers wrote:

  
  
  
  
  
Hello Bogdan,
 
Yes
I’m very sure that the proper credentials are used ;)
 
I’m
going to try and calculate the response according to the
RFC.
 
One
thing I found is that asterisk seems to ignore my second
invite with Authorization because of retransmit?
It
seems that I should increase my CSEQ on second invite.. How
can I do this neatly?
 
[Nov  4 11:08:25]
  DEBUG[22804]:
chan_sip.c:22448 handle_incoming:  Received INVITE (5) -
Command in SIP INVITE
[Nov  4 11:08:25]
  DEBUG[22804]:
chan_sip.c:22467 handle_incoming: Ignoring SIP message
because of retransmit (INVITE Seqno 12481, ours 12481)
Ignoring this INVITE request
 
 

  Met
  vriendelijke groet,
   
  

  

  Rik Broers
  Voice Engineer

  

  
   

 

  
From: Bogdan-Andrei
Iancu [mailto:bog...@opensips.org]

Sent: vrijdag 1 november 2013 12:34
To: Rik Broers
Cc: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] uac_auth
  

 
Hello Rik,

It may be silly , but are you sure you filled in the
  proper credentials (realm, auth user and password) ??

Also, based on how the response for digest is computed,
  you can double check the OpenSIPS auth response
  (calculating the HA and md5 sums as per RFC 2617).

Regards,

  
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 11/01/2013 01:09 PM, Rik Broers wrote: 

  Yes, thats correct. Opensips sends out an
  invite with Authorization header as response on the 401
  unauthorized.
  This authorization header contains the
  correct Nonce.
  Instead of being authorized I receive another
  401 unauthorized which opensips replies again with new
  nonce and so on until max branches is reached.
   
  
Met
  vriendelijke groet,
Regards,
 

  

  
Rik
  Broers
Voice Engineer
  

  

 
  
   
  

  From: Bogdan-Andrei
  Iancu [mailto:bog...@opensips.org]
  
  Sent: vrijdag 1 november 2013 11:49
  To: OpenSIPS users mailling list
  Cc: Rik Broers
  Subject: Re: [OpenSIPS-Users] uac_auth

  
   
  Hello Rik,
  
So
OpenSIPS generates a new INVITE with credentials (as a
result of the uac_auth() ), but this is also rejected ?
  
Regards,
  
  

  Bogdan-Andrei Iancu
  OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
  
  On 10/31/2013 11:46 AM, Rik Broers wrote: 
  
Hi,
 
I’m trying to use the uac_auth() function
to add Authorization to my invite after I received a 401
Unauthorized.

I call the function in the failure route
and according to Debug the authorization header is
inserted. I also see this in a trace.
Unfortunately I haven’t been able to
authorize successfully, double checked everything and
also tried with phones to ensure the credentials are
correct and my asterisk is working.
I’m filling the credential

Re: [OpenSIPS-Users] uac_auth

[Rik Broers]

Hello Bogdan,

Yes I'm very sure that the proper credentials are used ;)

I'm going to try and calculate the response according to the RFC.

One thing I found is that asterisk seems to ignore my second invite with 
Authorization because of retransmit?
It seems that I should increase my CSEQ on second invite.. How can I do this 
neatly?

[Nov  4 11:08:25] DEBUG[22804]: chan_sip.c:22448 handle_incoming:  Received 
INVITE (5) - Command in SIP INVITE
[Nov  4 11:08:25] DEBUG[22804]: chan_sip.c:22467 handle_incoming: Ignoring SIP 
message because of retransmit (INVITE Seqno 12481, ours 12481) Ignoring this 
INVITE request


Met vriendelijke groet,

Rik Broers
Voice Engineer



From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: vrijdag 1 november 2013 12:34
To: Rik Broers
Cc: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] uac_auth

Hello Rik,

It may be silly , but are you sure you filled in the proper credentials (realm, 
auth user and password) ??

Also, based on how the response for digest is computed, you can double check 
the OpenSIPS auth response (calculating the HA and md5 sums as per RFC 2617).

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com

On 11/01/2013 01:09 PM, Rik Broers wrote:
Yes, thats correct. Opensips sends out an invite with Authorization header as 
response on the 401 unauthorized.
This authorization header contains the correct Nonce.
Instead of being authorized I receive another 401 unauthorized which opensips 
replies again with new nonce and so on until max branches is reached.

Met vriendelijke groet,
Regards,

Rik Broers
Voice Engineer



From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: vrijdag 1 november 2013 11:49
To: OpenSIPS users mailling list
Cc: Rik Broers
Subject: Re: [OpenSIPS-Users] uac_auth

Hello Rik,

So OpenSIPS generates a new INVITE with credentials (as a result of the 
uac_auth() ), but this is also rejected ?

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com

On 10/31/2013 11:46 AM, Rik Broers wrote:
Hi,

I'm trying to use the uac_auth() function to add Authorization to my invite 
after I received a 401 Unauthorized.
I call the function in the failure route and according to Debug the 
authorization header is inserted. I also see this in a trace.
Unfortunately I haven't been able to authorize successfully, double checked 
everything and also tried with phones to ensure the credentials are correct and 
my asterisk is working.
I'm filling the credentials with a modparam not with AVP.

In DBG I see this: DBG:uac_auth:build_authorization_hdr: hdr is 
So it seems to match correctly.

I'm authenticating against Asterisk. And my failure route looks like this:
failure_route[FailPBX]{
xlog("Im in failpbx route");
uac_auth();
t_on_failure("FailPBX");
t_relay();
}

What happens is the following
-> Invite
<- 100 Giving a try
<- 401 Unauthorized (Unique nonce 1)
-> ACK
-> invite with authorization header (unique Nonce 1)
<- 100 Giving a try
<- 401 Unauthorized (Unique nonce 2)
-> invite with authorization header (unique Nonce 2)
. and so on until ERROR:tm:add_uac: maximum number of branches exceeded.


Only thing left for me now is to verify that the Digest calculated is correct. 
How can I do this? What functions should I use on linux..
Below my authorization challenge.
[cid:image005.png@01CED93A.521D1010]

Or are there any other things I'm missing?
Im using NOTICE:core:main: version: opensips 1.10.0-notls (x86_64/linux)


Met vriendelijke groet,
Regards,

Rik Broers
Voice Engineer







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Re: [OpenSIPS-Users] 481 Call does not exists

[Miha]

Hi Bogdan,

this is not the same thing as we were taling last time. I manage to get 
accouting working by your help and now all CDR works as it should:)


But this is different thing as I do not know why Opensips send two more 
BYE request and FS replay with 481 code.


Here is a sip trace: http://pastebin.com/zdPfuQUt

Thank you!

Miha


Dne 10/30/2013 9:35 AM, piše Bogdan-Andrei Iancu:

Hi,

Issue sorted out off list, by using a custom hdr to link the two SIP
calls (for accounting purposes).

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 10/24/2013 11:03 AM, Miha wrote:

Hi,

I need a little help.

When I am having 302 that is handled by Opensips I am gettinf 481 by
FS when opensips sends BYE request.

I am guessing that the problem is with initial invite due to port 5080
invite but opensips send to 5060, please let me know if I am wrong:)

How can I change this behaviour as BYE is not send by Opensips and
call is not droped as it should on other side.

Here is a sip trace:
http://pastebin.com/pwwuY4Uu

Tnx!

Miha

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