Re: [OpenSIPS-Users] openSIPS with an AS that would handle the routing logic

[Vlad Paiu]

Hello,

Since OpenSIPS is a SIP server, I would try to leverage it's 
capabilities and communicate over SIP with the AS - that means scripting 
your way through.



On 20.04.2024 11:50, julien.royann...@orange.com wrote:


Hello everyone,

I'm reaching out to get your opinion on using openSIPS with a business 
Application Server (AS) that would handle the routing logic.


At first glance, the SEAS module seems designed for this purpose, but 
it doesn't appear to be a good fit as it seems highly coupled with a 
Sip Servlet implementation using a specific protocol only supported 
for WeSIP.


It might be better to use REST interfaces, a 302 redirect-based 
mechanism, or possibly another module.


Thank you for your insights & advice!

JR

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Re: [OpenSIPS-Users] How to access to a column of a location database

[Vlad Paiu]

Hello,

You can use 
https://opensips.org/html/docs/modules/3.3.x/registrar.html#param_attr_avp 
in order to populate any custom info at save() time and the info will 
automatically be populated for you at lookup() time.



Regards,
Vlad


On 29.03.2024 15:08, guillaume.desgeo...@orange.com wrote:


Hi Bogdan,

Thank you for your answer.

And is it possible to use that lookup("location") function to put the 
“registered socket” field in a variable in order to use it for my script ?


Regards,

  Guillaume

*De :*Bogdan-Andrei Iancu 
*Envoyé :* jeudi 28 mars 2024 11:15
*À :* OpenSIPS users mailling list ; 
DESGEORGE Guillaume INNOV/IT-S 
*Objet :* Re: [OpenSIPS-Users] How to access to a column of a location 
database


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Hi Guillaume,

The registered contact (and the additional info) is fetched via the 
lookup("location") function. It also sets the registered socket for 
the routing the current request.


Regards,

Bogdan-Andrei Iancu
  
OpenSIPS Founder and Developer

   https://www.opensips-solutions.com
   https://www.siphub.com

On 22.03.2024 10:57, guillaume.desgeo...@orange.com wrote:

Hi everybody,

In my routing logic of my “opensips.cfg” file, I’m trying to
access the “socket” field of the location table of registered
contacts.

I have the registering informations correctly written in MySQL
location table but can’t find a function to access it.

I’d like to have a function which I give the registered username
and can give me back the associated socket.

Is the lookup () function the good one ? I didn’t understand how
to use it that way.

Thanks for your help,

   Guillaume

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Re: [OpenSIPS-Users] Load a DB Table to hash with MI control

[Vlad Paiu]

Hello,

Check out the sql_cacher [1] module.

[1] https://opensips.org/html/docs/modules/2.4.x/sql_cacher.html


Regards,
Vlad
On 17.10.2022 10:36, Antonis Psaras wrote:


Dear all

I am looking to implement a functionality similar to htable module 
were I will have a db table with some key values which will be loaded 
during service start and will expire every x sec or forced to be 
reloaded by MI. Using OpenSIPs 2.4


Any suggestion?

Best regards

*Antonis Psaras*


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Re: [OpenSIPS-Users] Check the User-Agent for registered user

[Vlad Paiu]

Hello,

At registration time you could save the User Agent in the attributes per 
registration [1] and at INVITE time you can check that in branch_route 
and make your decision there.


[1] 
https://opensips.org/html/docs/modules/3.2.x/registrar.html#param_attr_avp



Regards,
Vlad
On 14.10.2022 17:02, Callum Guy wrote:

Hi All,

I'm working on a project that requires me to evaluate the user agent 
of a registered contact before making a decision on a current 
registration attempt.


Is there a method to do this natively? If not, what is the best approach?

I haven't found anything yet, the current options i'm evaluating are to:

1. Use mi_script module to allow me to pull the contact record 
via ul_show_contact

2. Use rest_get to pull this data via a web service

Neither is ideal for my requirements but both would do the job, I'm 
hoping someone in the community has a cleaner solution before I start!


Best regards,

Callum



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Re: [OpenSIPS-Users] Generate CANCEL on 180

[Vlad Paiu]

Hello,

In 3.1, create_dialog [1] supports passing it the 'E' flag which will 
lead to the call termination if a dialog race condition occurs.


The dialog will be ended after race_condition_timeout [2] seconds.

Some of the most frequent race conditions are documented in RFC 5407 
[3], and the dialog module currently supports terminating the call in 
case races 3.1.2 and 3.1.3 from the RFC occur.


[1] 
https://opensips.org/html/docs/modules/3.1.x/dialog.html#func_create_dialog 
<https://opensips.org/html/docs/modules/3.1.x/dialog.html#func_create_dialog>


[2] 
https://opensips.org/html/docs/modules/3.1.x/dialog.html#race_condition_timeout 
<https://opensips.org/html/docs/modules/3.1.x/dialog.html#race_condition_timeout>


[3] https://tools.ietf.org/html/rfc5407 
<https://tools.ietf.org/html/rfc5407>


On 21.04.2021 15:52, Antonis Psaras wrote:

Hello Vlad

I am using 2.4 so race_condition_timeout is not available but I can upgrade if 
required.

Actually is not very clear to me what that parameter does. Can you explain a 
bit more?

Regards


Antonis Psaras / Managing Director
   


-Original Message-
From: Users  On Behalf Of Vlad Paiu
Sent: Τετάρτη, 21 Απριλίου 2021 15:45
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180

Hello,

Try using the race_condition_timeout dialog param along with the 'E'
flag when creating the dialog.

https://opensips.org/html/docs/modules/3.1.x/dialog.html#race_condition_timeout
<https://opensips.org/html/docs/modules/3.1.x/dialog.html#race_condition_timeout>

Best Regards,
Vlad
On 21.04.2021 12:46, Antonis Psaras wrote:

Hello Bogdan

The flow is the following

INVITE
Trying (instantly)
(after 2sec)
183
(after 3sec)
180
(here I do the process but during that I receive)
200 (instantly)
(here the CANCEL is sent)
CANCEL

And the call is keep going, hence the CANCEL was ignored by the carrier by the 
carrier because the call was answered.

Regards


Antonis Psaras

-Original Message-
From: Bogdan-Andrei Iancu 
Sent: Τετάρτη, 21 Απριλίου 2021 12:32
To: apsa...@microbase.gr; OpenSIPS users mailling list

Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180

Hi Antonis,

What exactly does not work ? sending the CANCEL out? or the callee "refuses" to 
cancel and sends a 200 OK ?

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
 https://www.opensips-solutions.com
OpenSIPS Bootcamp 2021 online
 https://opensips.org/training/OpenSIPS_eBootcamp_2021/

On 4/20/21 11:13 PM, Antonis Psaras wrote:

I did the following

if (t_check_status("180"))
{
t_cancel_branch();
  drop;
}

But there is an issue.

When 180 is followed by 200 instantly, the CANCEL is not working as expected.

When I add a delay on Answer ie 1sec then CANCEL works.

Any suggestion?

Antonis Psaras

-Original Message-
From: Users  On Behalf Of Kingsley
Tart
Sent: Τρίτη, 20 Απριλίου 2021 20:10
To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180

Firstly, I'm new to OpenSIPS so treat my comments accordingly.

But, can you do something in an onreply route?

eg, in a test setup I have, when I get an INVITE I do this:

create_dialog("pPB");
t_on_reply("doodle");

(I can't remember whether the dialog is needed for this)

and then I have this:

onreply_route[doodle] {
# expect $T_reply_code to likely first be 100
# then 180 or 183 for a progressing call
# 200 when call is answered
# or failure code (eg 4xx) or whatever
if (t_check_status("^1[0-9][0-9]$")) {
switch ($T_reply_code) {
case 180: $acc_extra(t_ringing) = $Ts; break;
case 183: $acc_extra(t_progress) = $Ts; break;
}
} else if (t_check_status("^2[0-9][0-9]$")) {
$acc_extra(t_answer) = $Ts;
} else {
xlog("Something else\n");
}
}

so when a 180 is received, it calls the above route function. Could you send a 
CANCEL from there?

Cheers,
Kingsley.

On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote:

Dear all

I am trying to create a service which will generate missed calls. In
order to be more accurate, I want to CANCEL the request when 180 is
received.

The scenario is the following

Asterisk Invite -> OpenSIPs -> Carrier

Carrier 183 -> OpenSIPs -> Asterisk

Carrier 180 -> OpenSIPs

OpenSIPs Cancel -> Carrier


Is that possible to be done from script without external app?

Regards

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Re: [OpenSIPS-Users] Generate CANCEL on 180

[Vlad Paiu]

Hello,

Try using the race_condition_timeout dialog param along with the 'E' 
flag when creating the dialog.


https://opensips.org/html/docs/modules/3.1.x/dialog.html#race_condition_timeout 



Best Regards,
Vlad
On 21.04.2021 12:46, Antonis Psaras wrote:

Hello Bogdan

The flow is the following

INVITE
Trying (instantly)
(after 2sec)
183
(after 3sec)
180
(here I do the process but during that I receive)
200 (instantly)
(here the CANCEL is sent)
CANCEL

And the call is keep going, hence the CANCEL was ignored by the carrier by the 
carrier because the call was answered.

Regards


Antonis Psaras

-Original Message-
From: Bogdan-Andrei Iancu 
Sent: Τετάρτη, 21 Απριλίου 2021 12:32
To: apsa...@microbase.gr; OpenSIPS users mailling list 

Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180

Hi Antonis,

What exactly does not work ? sending the CANCEL out? or the callee "refuses" to 
cancel and sends a 200 OK ?

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Bootcamp 2021 online
https://opensips.org/training/OpenSIPS_eBootcamp_2021/

On 4/20/21 11:13 PM, Antonis Psaras wrote:

I did the following

if (t_check_status("180"))
{
t_cancel_branch();
 drop;
}

But there is an issue.

When 180 is followed by 200 instantly, the CANCEL is not working as expected.

When I add a delay on Answer ie 1sec then CANCEL works.

Any suggestion?

Antonis Psaras

-Original Message-
From: Users  On Behalf Of Kingsley
Tart
Sent: Τρίτη, 20 Απριλίου 2021 20:10
To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180

Firstly, I'm new to OpenSIPS so treat my comments accordingly.

But, can you do something in an onreply route?

eg, in a test setup I have, when I get an INVITE I do this:

create_dialog("pPB");
t_on_reply("doodle");

(I can't remember whether the dialog is needed for this)

and then I have this:

onreply_route[doodle] {
# expect $T_reply_code to likely first be 100
# then 180 or 183 for a progressing call
# 200 when call is answered
# or failure code (eg 4xx) or whatever
if (t_check_status("^1[0-9][0-9]$")) {
switch ($T_reply_code) {
case 180: $acc_extra(t_ringing) = $Ts; break;
case 183: $acc_extra(t_progress) = $Ts; break;
}
} else if (t_check_status("^2[0-9][0-9]$")) {
$acc_extra(t_answer) = $Ts;
} else {
xlog("Something else\n");
}
}

so when a 180 is received, it calls the above route function. Could you send a 
CANCEL from there?

Cheers,
Kingsley.

On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote:

Dear all

I am trying to create a service which will generate missed calls. In
order to be more accurate, I want to CANCEL the request when 180 is
received.

The scenario is the following

Asterisk Invite -> OpenSIPs -> Carrier

Carrier 183 -> OpenSIPs -> Asterisk

Carrier 180 -> OpenSIPs

OpenSIPs Cancel -> Carrier

   
Is that possible to be done from script without external app?
   
Regards
   
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Re: [OpenSIPS-Users] [BLOG] Real-Time Rating and Cost Based Routing in OpenSIPS 3.1

[Vlad Paiu]

Hello,

Currently the module can only be used for a basic postpaid type of rating.

Regards, Vlad

On 09.04.2020 19:47, Mehdi Shirazi wrote:

Hi
Thanks for new features.
Is it possible(or planed) to use this module for basic prepaid billing ?

Regards
M.Shirazi

>While there are numerous external rating and billing engines available 
>in the wild, having a quick and easy way of putting a price for a call, 
>without relying on external applications, can be a valuable asset to have.


>https://blog.opensips.org/2020/04/07/real-time-rating-and-cost-based-routing-in-opensips-3-1/
  
<https://blog.opensips.org/2020/04/07/real-time-rating-and-cost-based-routing-in-opensips-3-1/>

>Thank you Vlad Paiu for the valuable contribution and post !

>Best Regards,

>Bogdan-Andrei Iancu



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Re: [OpenSIPS-Users] Opensips change the order of requests

[Vlad Paiu]

Hello,

I would say this an UA issue - OpenSIPS does not do any enforcing on the 
order of outgoing SIP packets, so if the UA sends an INVITE immediately 
followed by an UPDATE, there exists the risk of the UPDATE reaching the 
endpoint first ( even if the proxy would maintain the the packet order, 
since UDP is used as transport, there would still be no guarantee that 
the endpoint receives them in the same order ).


Can you configure the UA to only send the UPDATE after the INVITE 
transaction has finished ? This would be the only way to ensure there 
are no races.


Best Regards,

Vlad Paiu
OpenSIPS Developer

On 22.10.2015 20:05, Patrick Wakano wrote:

Hello Opensips list!

I already explained my scenario in this e-mail 
(http://lists.opensips.org/pipermail/users/2015-October/032859.html) 
where the Ack was propagated with wrong CSeq number when using the 
in-dialog ping options. Vlad did a great job and already correct this 
bug! (https://github.com/OpenSIPS/opensips/issues/680)


Now in a scenario where I disabled the in-dialog ping, I faced another 
issue. I mentioned that sometimes, due to unknown reasons, when I 
received a Re-Invite immediately followed by an Update, the Update 
gets forwarded first. The problem is that, since CSeq numbers are not 
altered, when this inversion happens, my peer rejects the Re-Invite 
because its CSeq number is lower than the Update one.
Probably Opensips has to respect the ordering of in-dialog requests 
and forward them accordingly, instead of handling them asynchronously 
and forwarding right away...


Best regards,
Patrick



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Re: [OpenSIPS-Users] Exported Asyncronous functions

[Vlad Paiu]

Hello,

There is no way to do that.
Just move your 'line after exec;' to the resume route, or put it in a 
new route that will get called from within the resume route.


Regards,

Vlad Paiu
OpenSIPS Developer

On 22.10.2015 17:39, Dragomir Haralambiev wrote:

Hello,
I try to test Exported Asyncronous functions in OpenSips 2.1.

What I do to return back in stript (at line after exec) when finished 
route[resume] ?

{

async( exec("test.sh.","$ru","$avp(return)"), resume );
line after exec;
}

route[resume] {
xlog("Exec return$avp(return)");
}

Best regards,
PlayMen


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Re: [OpenSIPS-Users] Opensips and CSeq number handling

[Vlad Paiu]

Hello,

Yes, if you are using in-dialog OPTIONS pings, OpenSIPS will mangle the 
CSEQs of all in-dialog requests, after it has sent the first pings.


Just making sure I understand the scenario... so the client is sending a 
Re-INVITE and immediately after that it sends an UPDATE, and the ACK for 
the RE-INVITE gets propagated with the CSEQ of the UPDATE ?


Best Regards,

Vlad Paiu
OpenSIPS Developer

On 20.10.2015 21:28, Patrick Wakano wrote:

Hi list,
An update about this issue.
The behavior I mentioned about Opensips incrementing the CSeq values 
only after second Re-Invite is incorrect. More tests showed me that 
this happens after the in-dialog Options the dialog module sends (I am 
creating it with "pP" options). This also explains why Opensips is 
changing the CSeq number, because it has to couple the CSeq numbering 
of the local generated Options with the original requests of the dialog!
The problem regarding the Ack with wrong CSeq number still occurs 
anyways. It seems that Opensips is not matching the Ack with the 
correct Invite transaction...


Patrick


On Tue, Oct 20, 2015 at 12:31 PM, Patrick Wakano <pwak...@gmail.com 
<mailto:pwak...@gmail.com>> wrote:


Hello Opensips list!

Hope you all doing fine!

The purpose of this e-mail is to explain a problem I am facing and
to understand a little bit more about the handling done by
Opensips over the CSeq number when forwarding messages to the
destination. I couldn't find any real good explanation over this
subject so I wrote this huge e-mail, sorry for that...

I am trying to use the remote ID feature of Asterisk, but in some
transfer scenarios the call gets dropped and after digging the
problem I think it is related to the CSeq handling done by Opensips.
This remote ID feature is configured to use the
P-Asserted-Identity header to transmit the callee ID to the caller
and it causes the exchange of Re-Invites and/or Updates during the
call. The transfer scenario I mentioned is entirely handled by
Asterisk, and as a result of the transfer it sends to Opensips the
identity of the new peer, using a Re-Invite and an Update.
First I would like to know how Opensips handles the CSeq number
when proxying the Invite from one side to the other? My tests
showed me that Opensips does not change the CSeq for the first
Invite and first Re-invite, however for the second Re-invite and
for requests after that it is always incrementing the value by one
when forwarding it. Although it haven't caused any errors so far,
I am not sure if this is correct. Why is Opensips incrementing it?
My understanding is that the proxy was not supposed to change this
field...

Now the problem I am facing: In a blind transfer scenario, the
remote ID feature causes Asterisk to send a Re-Invite and right
after an Update. Opensips increments the CSeq of both(because this
happens to be the second Re-Invite of the dialog) and forward them
to the destination. Both messages are answered with 200 Ok. This
follows by Asterisk sending an Ack with the same CSeq number used
in the Re-Invite. This is the point where Opensips fails, it gets
this Ack and forward it using the CSeq number of the Update and
not the one of the Re-Invite. Because of this the destination
discards this Ack and keeps retransmitting the 200 Ok for the
Re-Invite, eventually the call is dropped by timeout or because
some other Re-Invite happens without the prior one being properly
handled.

Useful information:
- If the Re-Invite followed by the Update is the first of the
dialog, then the problem does not happen. The CSeq numbers are not
incremented and the CSeq for the Ack is correct.
- If due to unknown timing reasons, the Update gets forwarded
before the Re-Invite (even though the Re-Invite is received first)
the problem also does not happen. The CSeq numbers are incremented
but the CSeq for the Ack gets the correct value. So it seems to me
that the Ack is getting the last CSeq used to forward, and not the
one of the corresponding Invite.
- When I enable more traces(debug=4), I always fall in the case
where the Update is forwarded before the Re-Invite and then the
problem doesn't happen.
- In an attended transfer, Asterisk does not send the Update so
the problem does not happen.
- Not sure why Asterisk is sending the Re-Invite immediately
followed by an Update, nevertheless technically I couldn't see a
problem with it.
- I am using Opensips 1.11.3


Best regards for all and sorry again for such a huge e-mail.

Patrick




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Re: [OpenSIPS-Users] Opensips and CSeq number handling

[Vlad Paiu]

Hello,

Think a SIP trace would help a lot in debugging this situation - my 
guess is that there is sort of a "race" between the RE-INVITE being sent 
out, and in-dialog options pings ( eg. Re-INVITE goes out, then OpenSIPS 
sends in-dialog OPTION pings, then the ACK comes in and it's CSEQ is 
wrongly increased ).


Can you please provide a SIP trace for your scenario ?

Best Regards,

Vlad Paiu
OpenSIPS Developer

On 21.10.2015 12:32, Vlad Paiu wrote:

Hello,

Yes, if you are using in-dialog OPTIONS pings, OpenSIPS will mangle 
the CSEQs of all in-dialog requests, after it has sent the first pings.


Just making sure I understand the scenario... so the client is sending 
a Re-INVITE and immediately after that it sends an UPDATE, and the ACK 
for the RE-INVITE gets propagated with the CSEQ of the UPDATE ?


Best Regards,
Vlad Paiu
OpenSIPS Developer
On 20.10.2015 21:28, Patrick Wakano wrote:

Hi list,
An update about this issue.
The behavior I mentioned about Opensips incrementing the CSeq values 
only after second Re-Invite is incorrect. More tests showed me that 
this happens after the in-dialog Options the dialog module sends (I 
am creating it with "pP" options). This also explains why Opensips is 
changing the CSeq number, because it has to couple the CSeq numbering 
of the local generated Options with the original requests of the dialog!
The problem regarding the Ack with wrong CSeq number still occurs 
anyways. It seems that Opensips is not matching the Ack with the 
correct Invite transaction...


Patrick


On Tue, Oct 20, 2015 at 12:31 PM, Patrick Wakano <pwak...@gmail.com 
<mailto:pwak...@gmail.com>> wrote:


Hello Opensips list!

Hope you all doing fine!

The purpose of this e-mail is to explain a problem I am facing
and to understand a little bit more about the handling done by
Opensips over the CSeq number when forwarding messages to the
destination. I couldn't find any real good explanation over this
subject so I wrote this huge e-mail, sorry for that...

I am trying to use the remote ID feature of Asterisk, but in some
transfer scenarios the call gets dropped and after digging the
problem I think it is related to the CSeq handling done by Opensips.
This remote ID feature is configured to use the
P-Asserted-Identity header to transmit the callee ID to the
caller and it causes the exchange of Re-Invites and/or Updates
during the call. The transfer scenario I mentioned is entirely
handled by Asterisk, and as a result of the transfer it sends to
Opensips the identity of the new peer, using a Re-Invite and an
Update.
First I would like to know how Opensips handles the CSeq number
when proxying the Invite from one side to the other? My tests
showed me that Opensips does not change the CSeq for the first
Invite and first Re-invite, however for the second Re-invite and
for requests after that it is always incrementing the value by
one when forwarding it. Although it haven't caused any errors so
far, I am not sure if this is correct. Why is Opensips
incrementing it? My understanding is that the proxy was not
supposed to change this field...

Now the problem I am facing: In a blind transfer scenario, the
remote ID feature causes Asterisk to send a Re-Invite and right
after an Update. Opensips increments the CSeq of both(because
this happens to be the second Re-Invite of the dialog) and
forward them to the destination. Both messages are answered with
200 Ok. This follows by Asterisk sending an Ack with the same
CSeq number used in the Re-Invite. This is the point where
Opensips fails, it gets this Ack and forward it using the CSeq
number of the Update and not the one of the Re-Invite. Because of
this the destination discards this Ack and keeps retransmitting
the 200 Ok for the Re-Invite, eventually the call is dropped by
timeout or because some other Re-Invite happens without the prior
one being properly handled.

Useful information:
- If the Re-Invite followed by the Update is the first of the
dialog, then the problem does not happen. The CSeq numbers are
not incremented and the CSeq for the Ack is correct.
- If due to unknown timing reasons, the Update gets forwarded
before the Re-Invite (even though the Re-Invite is received
first) the problem also does not happen. The CSeq numbers are
incremented but the CSeq for the Ack gets the correct value. So
it seems to me that the Ack is getting the last CSeq used to
forward, and not the one of the corresponding Invite.
- When I enable more traces(debug=4), I always fall in the case
where the Update is forwarded before the Re-Invite and then the
problem doesn't happen.
- In an attended transfer, Asterisk does not send the Update so
the problem does not happen.
- Not sure why Asterisk is sending the Re

Re: [OpenSIPS-Users] Regarding chachedb_mongdb use .

[Vlad Paiu]

Hello,

Looks like there is an error with your MongoDB sharding - the errors you 
are receiving from MongoDB look something like this : 
https://jira.mongodb.org/browse/SERVER-12899


Can you manually login to a MongoDB node and run the version query by hand ?

Regards,

Vlad Paiu
OpenSIPS Developer

On 05.10.2015 14:58, Sasmita Panda wrote:

Hi All ,

   I an using the NoSql module cachedb_mongodb as database with 
db_cachedb .


I have tested it running mongodb and opensips in a single instance . 
Now I am trying to run mongodb and opensips in different instances . I 
am facing some error in this case .


 What i have done :
I have created a db with name opensips read/write permission. I am 
also created the version table with name "my_version_table" inside 
that db .Now I am  trying to connect to that DB from my opensips 
server . But its giving following error :


 DBG:core:db_bind_mod: using db bind api for db_cachedb
 DBG:db_cachedb:db_cachedb_bind_api: BINDING API for : 
cachedb://mongodb:instance1
 DBG:db_cachedb:db_cachedb_init: Found matching URL : 
[mongodb:instance1://mongodb:mongodbserver@x.x.x.x:27017/db.opensips]

 DBG:core:cachedb_bind_mod: Binded to mod mongodb
 DBG:core:parse_cachedb_url: parsing 
[mongodb:instance1://mongodb:mongodbserver@x.x.x.x:27017/db.opensips]

 DBG:core:parse_cachedb_url: in host - :
 DBG:core:cachedb_do_init: opening new connection
 DBG:cachedb_mongodb:mongo_new_connection: Set timeout to 3000 millis
 INFO:cachedb_mongodb:mongo_new_connection: Connected at server 
x.x.x.x with version 3.0.6 , to db db.opensips
 DBG:db_cachedb:db_cachedb_init: Succesfully initiated connection to 
[mongodb:instance1]
 DBG:cachedb_mongodb:mongo_db_query_trans: Running raw mongo query on 
table db.my_version_table
 ERROR:cachedb_mongodb:mongo_db_query_trans: Failed to run query. Err 
= 0, 0 , 0

 DBG:cachedb_mongodb:mongo_db_query_trans: Fetched key connectionId
 DBG:cachedb_mongodb:mongo_db_query_trans: (int) 12
 DBG:cachedb_mongodb:mongo_db_query_trans: Fetched key n
 DBG:cachedb_mongodb:mongo_db_query_trans: (int) 0
 DBG:cachedb_mongodb:mongo_db_query_trans: Fetched key syncMillis
 DBG:cachedb_mongodb:mongo_db_query_trans: (int) 0
 DBG:cachedb_mongodb:mongo_db_query_trans: Fetched key writtenTo
 DBG:cachedb_mongodb:mongo_db_query_trans: (unknown type 10)
 DBG:cachedb_mongodb:mongo_db_query_trans: Fetched key err
 DBG:cachedb_mongodb:mongo_db_query_trans: (unknown type 10)
 DBG:cachedb_mongodb:mongo_db_query_trans: Fetched key ok
 DBG:cachedb_mongodb:mongo_db_query_trans: (double) 1.00e+00
 ERROR:core:db_table_version: error in db_query
 ERROR:core:db_check_table_version: querying version for table silo
 ERROR:msilo:mod_init: error during table version check.
 ERROR:core:init_mod: failed to initialize module msilo
 ERROR:core:main: error while initializing modules


   Please help me to solve this error .

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/


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Re: [OpenSIPS-Users] topology hiding not accepting BYE before 200 OK

[Vlad Paiu]

Hello Stuard,

What is the full SIP scenario for this ? The callee cannot send a BYE 
before the 200OK is sent ( from SIP point of view ). Can you please post 
a SIP trace for this ? A full debug OpenSIPS log would also help.


Regards,

Vlad Paiu
OpenSIPS Developer

On 05.10.2015 11:46, Stuart Marsden wrote:

Hi

we are experimenting with topology hiding on 2.1

I think we see the same issue once a call is set up  if UAC and UAS both send 
BYE  at  “the same time”

we cannot reproduce at will because of the small timing window required to 
receive the 2 BYEs

Stuart
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Re: [OpenSIPS-Users] Query regarding db_cachedb and cachedb_mongodb.

[Vlad Paiu]

Hello,

See the instructions at 
http://www.opensips.org/html/docs/modules/1.11.x/db_cachedb#id249688


Basically, in your script you have to set

 db_version_table="my_version_table"

and then in your MongoDB database go ahead and manually insert the 
record for the ACC table version , like :


db.my_version_table.insert({table_version : NumberInt(6), 
table_name : "acc"})


Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 19.09.2015 08:26, Sasmita Panda wrote:


HI All ,

My mongodb problem is solved . Now I can store and fetch data 
through mongodb module through raw query .


Now I need to understand something related to db_cachedb and 
cachedb_mongodb combined integration . .


  I have gone through the bellow link
http://www.opensips.org/html/docs/modules/1.11.x/db_cachedb.html

  In the end of this its saying I need to create a schema for version 
table in mongodb so that opensips modules can use that table .


What should I do for this ? I am new to mongodo and also opensips-1.11 
. So please can anybody explain me how to create same version table 
schema in mongodb ?


I have tried to put the version table in mongodb as a collection 
but it wont works for me .


I am trying to use mongodb for acc module and its giving bellow error :


 ERROR:core:db_check_table_version: invalid version 0 for table acc 
found, expected 6

 ERROR:acc:acc_db_init: error during table version check
 ERROR:acc:mod_init: failed...did you load a database module?
 ERROR:core:init_mod: failed to initialize module acc
 ERROR:core:main: error while initializing modules

I think this error will come for all the modules too . So what is the 
common solution for this ? Waiting for a reply .




*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/


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Re: [OpenSIPS-Users] core generated by opensips

[Vlad Paiu]

Hello,

As per our meeting today, please enable memory debugging on your server, 
and let me know when another crash happens, so I can take a look at the 
newly generated core file.


Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 03.09.2015 17:02, Vlad Paiu wrote:

Hello,

Can you replicate the crash on demand, or does it happen randomly ?
Would you be able to provide access to a server containing the core 
file and the OpenSIPS binary, in order to speed-up the debugging 
process ( avoid the ping pong on the mailing list ) ?


Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  
On 02.09.2015 18:47, Gupta, Rahul wrote:


Hi Bogdan, any update on this ?

*From:*Gupta, Rahul
*Sent:* Monday, August 31, 2015 4:42 PM
*To:* 'Bogdan-Andrei Iancu'; OpenSIPS users mailling list
*Subject:* RE: [OpenSIPS-Users] core generated by opensips

Hi Bogdan, following is the info you requested.

#0 0x7f2095572e2c in vfprintf () from /lib64/libc.so.6

#1 0x7f209560fed0 in __vsyslog_chk () from /lib64/libc.so.6

#2 0x7f2095610100 in syslog () from /lib64/libc.so.6

#3 0x004c4202 in tcp_read_req (con=0x7f2081a933c0, 
bytes_read=0x7ffc6bc97f0c) at tcp_read.c:597


#4 0x004c5908 in handle_io (fm=, idx=-1, 
event_type=) at tcp_read.c:1033


#5 0x004c8083 in io_wait_loop_epoll (unix_sock=optimized out>) at io_wait.h:845


#6 tcp_receive_loop (unix_sock=) at tcp_read.c:1141

#7 0x004b12e9 in tcp_init_children (chd_rank=out>, startup_done=0x0) at tcp_main.c:2389


#8 0x0043aebf in main_loop (argc=, 
argv=) at main.c:1011


#9 main (argc=, argv=) at 
main.c:1612


(gdb) f 3

#3 0x004c4202 in tcp_read_req (con=0x7f2081a933c0, 
bytes_read=0x7ffc6bc97f0c) at tcp_read.c:597


597 tcp_read.c: No such file or directory.

in tcp_read.c

(gdb) p con->con_req

$1 = (struct tcp_req *) 0x7f20931a97e0

(gdb) p _req

$2 = (struct tcp_req *) 0x82d720

(gdb) p req

$3 = (struct tcp_req *) 0x7f20931a97e0

(gdb) p *con

$4 = {s = 30, fd = 21, write_lock = 0, id = 19, rcv = {src_ip = {af = 
2, len = 4, u = {addrl = {2621885450, 0}, addr32 = {2621885450, 0, 0, 
0}, addr16 = {52234, 40006, 0, 0, 0, 0, 0, 0},


addr = "\n\314F\234", '\000' }}, dst_ip = 
{af = 2, len = 4, u = {addrl = {2588331018, 0}, addr32 = {2588331018, 
0, 0, 0}, addr16 = {52234, 39494, 0, 0, 0, 0, 0, 0},


addr = "\n\314F\232", '\000' }}, src_port = 
5060, dst_port = 5070, proto = 2, proto_reserved1 = 19, 
proto_reserved2 = 0, src_su = {s = {sa_family = 2,


sa_data = "\023\304\n\314F\234\000\000\000\000\000\000\000"}, 
sin = {sin_family = 2, sin_port = 50195, sin_addr = {s_addr = 
2621885450}, sin_zero = "\000\000\000\000\000\000\000"}},


bind_address = 0x7f2093171398}, refcnt = 2, type = PROTO_TCP, 
flags = 2, state = S_CONN_CONNECT, extra_data = 0x0, timeout = 50039, 
lifetime = 0, id_hash = 19, id_next = 0x0, id_prev = 0x0, c_next = 0x0,


  c_prev = 0x0, con_aliases = {{parent = 0x7f2081a933c0, next = 0x0, 
prev = 0x0, port = 5060, hash = 974}, {parent = 0x0, next = 0x0, prev 
= 0x0, port = 0, hash = 0}, {parent = 0x0, next = 0x0, prev = 0x0,


  port = 0, hash = 0}, {parent = 0x0, next = 0x0, prev = 0x0, 
port = 0, hash = 0}}, aliases = 1, con_req = 0x7f20931a97e0, 
msg_attempts = 1, async_chunks = 0x7f2081a93530, async_chunks_no = 0,


  oldest_chunk = 0}

(gdb)

*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* Monday, August 31, 2015 4:39 PM
*To:* Gupta, Rahul; OpenSIPS users mailling list
*Subject:* Re: [OpenSIPS-Users] core generated by opensips

And printing :
*con

Thanks,

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

On 31.08.2015 22:50, Gupta, Rahul wrote:

Hi Bogdan, following is the info you requested. This is not the
first tcp read, this server is been running for a while and
taking calls.

(gdb) bt

#0 0x7f2095572e2c in vfprintf () from /lib64/libc.so.6

#1 0x7f209560fed0 in __vsyslog_chk () from /lib64/libc.so.6

#2 0x7f2095610100 in syslog () from /lib64/libc.so.6

#3 0x004c4202 in tcp_read_req (con=0x7f2081a933c0,
bytes_read=0x7ffc6bc97f0c) at tcp_read.c:597

#4 0x004c5908 in handle_io (fm=,
idx=-1, event_type=) at tcp_read.c:1033

#5 0x004c8083 in io_wait_loop_epoll (unix_sock=) at io_wait.h:845

#6 tcp_receive_loop (unix_sock=) at
tcp_read.c:1141

#7 0x004b12e9 in tcp_init_children (chd_rank=, startup_done=0x0) at tcp_main.c:2389

#8 0x0043aebf in main_loop (argc=,
argv=) at main.c:1011

#9 main (argc=, argv=)
at main.c:1612

(gdb) f 3

#3 0x004c4202 in tcp_read_req (con=0x7f2081a933c0,
bytes_read=0x7ffc6bc97f0c) at tcp_read.c:597

597 tcp_read.c: No such file or directory.

in tcp_read.c

(gdb) p con->con_req

$1 = (struct tcp_req *) 0x7f20

Re: [OpenSIPS-Users] core generated by opensips

[Vlad Paiu]

Hello,

Also, on a second note on this, do you happen to have full debug logs 
from the time of the crash ? Those could be very useful as well.


Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 03.09.2015 17:02, Vlad Paiu wrote:

Hello,

Can you replicate the crash on demand, or does it happen randomly ?
Would you be able to provide access to a server containing the core 
file and the OpenSIPS binary, in order to speed-up the debugging 
process ( avoid the ping pong on the mailing list ) ?


Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  
On 02.09.2015 18:47, Gupta, Rahul wrote:


Hi Bogdan, any update on this ?

*From:*Gupta, Rahul
*Sent:* Monday, August 31, 2015 4:42 PM
*To:* 'Bogdan-Andrei Iancu'; OpenSIPS users mailling list
*Subject:* RE: [OpenSIPS-Users] core generated by opensips

Hi Bogdan, following is the info you requested.

#0 0x7f2095572e2c in vfprintf () from /lib64/libc.so.6

#1 0x7f209560fed0 in __vsyslog_chk () from /lib64/libc.so.6

#2 0x7f2095610100 in syslog () from /lib64/libc.so.6

#3 0x004c4202 in tcp_read_req (con=0x7f2081a933c0, 
bytes_read=0x7ffc6bc97f0c) at tcp_read.c:597


#4 0x004c5908 in handle_io (fm=, idx=-1, 
event_type=) at tcp_read.c:1033


#5 0x004c8083 in io_wait_loop_epoll (unix_sock=optimized out>) at io_wait.h:845


#6 tcp_receive_loop (unix_sock=) at tcp_read.c:1141

#7 0x004b12e9 in tcp_init_children (chd_rank=out>, startup_done=0x0) at tcp_main.c:2389


#8 0x0043aebf in main_loop (argc=, 
argv=) at main.c:1011


#9 main (argc=, argv=) at 
main.c:1612


(gdb) f 3

#3 0x004c4202 in tcp_read_req (con=0x7f2081a933c0, 
bytes_read=0x7ffc6bc97f0c) at tcp_read.c:597


597 tcp_read.c: No such file or directory.

in tcp_read.c

(gdb) p con->con_req

$1 = (struct tcp_req *) 0x7f20931a97e0

(gdb) p _req

$2 = (struct tcp_req *) 0x82d720

(gdb) p req

$3 = (struct tcp_req *) 0x7f20931a97e0

(gdb) p *con

$4 = {s = 30, fd = 21, write_lock = 0, id = 19, rcv = {src_ip = {af = 
2, len = 4, u = {addrl = {2621885450, 0}, addr32 = {2621885450, 0, 0, 
0}, addr16 = {52234, 40006, 0, 0, 0, 0, 0, 0},


addr = "\n\314F\234", '\000' }}, dst_ip = 
{af = 2, len = 4, u = {addrl = {2588331018, 0}, addr32 = {2588331018, 
0, 0, 0}, addr16 = {52234, 39494, 0, 0, 0, 0, 0, 0},


addr = "\n\314F\232", '\000' }}, src_port = 
5060, dst_port = 5070, proto = 2, proto_reserved1 = 19, 
proto_reserved2 = 0, src_su = {s = {sa_family = 2,


sa_data = "\023\304\n\314F\234\000\000\000\000\000\000\000"}, 
sin = {sin_family = 2, sin_port = 50195, sin_addr = {s_addr = 
2621885450}, sin_zero = "\000\000\000\000\000\000\000"}},


bind_address = 0x7f2093171398}, refcnt = 2, type = PROTO_TCP, 
flags = 2, state = S_CONN_CONNECT, extra_data = 0x0, timeout = 50039, 
lifetime = 0, id_hash = 19, id_next = 0x0, id_prev = 0x0, c_next = 0x0,


  c_prev = 0x0, con_aliases = {{parent = 0x7f2081a933c0, next = 0x0, 
prev = 0x0, port = 5060, hash = 974}, {parent = 0x0, next = 0x0, prev 
= 0x0, port = 0, hash = 0}, {parent = 0x0, next = 0x0, prev = 0x0,


  port = 0, hash = 0}, {parent = 0x0, next = 0x0, prev = 0x0, 
port = 0, hash = 0}}, aliases = 1, con_req = 0x7f20931a97e0, 
msg_attempts = 1, async_chunks = 0x7f2081a93530, async_chunks_no = 0,


  oldest_chunk = 0}

(gdb)

*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* Monday, August 31, 2015 4:39 PM
*To:* Gupta, Rahul; OpenSIPS users mailling list
*Subject:* Re: [OpenSIPS-Users] core generated by opensips

And printing :
*con

Thanks,

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

On 31.08.2015 22:50, Gupta, Rahul wrote:

Hi Bogdan, following is the info you requested. This is not the
first tcp read, this server is been running for a while and
taking calls.

(gdb) bt

#0 0x7f2095572e2c in vfprintf () from /lib64/libc.so.6

#1 0x7f209560fed0 in __vsyslog_chk () from /lib64/libc.so.6

#2 0x7f2095610100 in syslog () from /lib64/libc.so.6

#3 0x004c4202 in tcp_read_req (con=0x7f2081a933c0,
bytes_read=0x7ffc6bc97f0c) at tcp_read.c:597

#4 0x004c5908 in handle_io (fm=,
idx=-1, event_type=) at tcp_read.c:1033

#5 0x004c8083 in io_wait_loop_epoll (unix_sock=) at io_wait.h:845

#6 tcp_receive_loop (unix_sock=) at
tcp_read.c:1141

#7 0x004b12e9 in tcp_init_children (chd_rank=, startup_done=0x0) at tcp_main.c:2389

#8 0x0043aebf in main_loop (argc=,
argv=) at main.c:1011

#9 main (argc=, argv=)
at main.c:1612

(gdb) f 3

#3 0x004c4202 in tcp_read_req (con=0x7f2081a933c0,
bytes_read=0x7ffc6bc97f0c) at tcp_read.c:597

597 tcp_read.c: No such file or directory.

in tcp_read.c

(gdb) p con->con_req

$1 = (struct tcp_req *) 0x7f20931a97e0

(gdb) p _req

$2 = 

Re: [OpenSIPS-Users] core generated by opensips

[Vlad Paiu]

Hello,

Can you replicate the crash on demand, or does it happen randomly ?
Would you be able to provide access to a server containing the core file 
and the OpenSIPS binary, in order to speed-up the debugging process ( 
avoid the ping pong on the mailing list ) ?


Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 02.09.2015 18:47, Gupta, Rahul wrote:


Hi Bogdan, any update on this ?

*From:*Gupta, Rahul
*Sent:* Monday, August 31, 2015 4:42 PM
*To:* 'Bogdan-Andrei Iancu'; OpenSIPS users mailling list
*Subject:* RE: [OpenSIPS-Users] core generated by opensips

Hi Bogdan, following is the info you requested.

#0 0x7f2095572e2c in vfprintf () from /lib64/libc.so.6

#1 0x7f209560fed0 in __vsyslog_chk () from /lib64/libc.so.6

#2 0x7f2095610100 in syslog () from /lib64/libc.so.6

#3 0x004c4202 in tcp_read_req (con=0x7f2081a933c0, 
bytes_read=0x7ffc6bc97f0c) at tcp_read.c:597


#4 0x004c5908 in handle_io (fm=, idx=-1, 
event_type=) at tcp_read.c:1033


#5 0x004c8083 in io_wait_loop_epoll (unix_sock=optimized out>) at io_wait.h:845


#6 tcp_receive_loop (unix_sock=) at tcp_read.c:1141

#7 0x004b12e9 in tcp_init_children (chd_rank=out>, startup_done=0x0) at tcp_main.c:2389


#8 0x0043aebf in main_loop (argc=, 
argv=) at main.c:1011


#9 main (argc=, argv=) at 
main.c:1612


(gdb) f 3

#3 0x004c4202 in tcp_read_req (con=0x7f2081a933c0, 
bytes_read=0x7ffc6bc97f0c) at tcp_read.c:597


597 tcp_read.c: No such file or directory.

in tcp_read.c

(gdb) p con->con_req

$1 = (struct tcp_req *) 0x7f20931a97e0

(gdb) p _req

$2 = (struct tcp_req *) 0x82d720

(gdb) p req

$3 = (struct tcp_req *) 0x7f20931a97e0

(gdb) p *con

$4 = {s = 30, fd = 21, write_lock = 0, id = 19, rcv = {src_ip = {af = 
2, len = 4, u = {addrl = {2621885450, 0}, addr32 = {2621885450, 0, 0, 
0}, addr16 = {52234, 40006, 0, 0, 0, 0, 0, 0},


addr = "\n\314F\234", '\000' }}, dst_ip = 
{af = 2, len = 4, u = {addrl = {2588331018, 0}, addr32 = {2588331018, 
0, 0, 0}, addr16 = {52234, 39494, 0, 0, 0, 0, 0, 0},


addr = "\n\314F\232", '\000' }}, src_port = 
5060, dst_port = 5070, proto = 2, proto_reserved1 = 19, 
proto_reserved2 = 0, src_su = {s = {sa_family = 2,


sa_data = "\023\304\n\314F\234\000\000\000\000\000\000\000"}, 
sin = {sin_family = 2, sin_port = 50195, sin_addr = {s_addr = 
2621885450}, sin_zero = "\000\000\000\000\000\000\000"}},


bind_address = 0x7f2093171398}, refcnt = 2, type = PROTO_TCP, 
flags = 2, state = S_CONN_CONNECT, extra_data = 0x0, timeout = 50039, 
lifetime = 0, id_hash = 19, id_next = 0x0, id_prev = 0x0, c_next = 0x0,


  c_prev = 0x0, con_aliases = {{parent = 0x7f2081a933c0, next = 0x0, 
prev = 0x0, port = 5060, hash = 974}, {parent = 0x0, next = 0x0, prev 
= 0x0, port = 0, hash = 0}, {parent = 0x0, next = 0x0, prev = 0x0,


  port = 0, hash = 0}, {parent = 0x0, next = 0x0, prev = 0x0, port 
= 0, hash = 0}}, aliases = 1, con_req = 0x7f20931a97e0, msg_attempts = 
1, async_chunks = 0x7f2081a93530, async_chunks_no = 0,


  oldest_chunk = 0}

(gdb)

*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* Monday, August 31, 2015 4:39 PM
*To:* Gupta, Rahul; OpenSIPS users mailling list
*Subject:* Re: [OpenSIPS-Users] core generated by opensips

And printing :
*con

Thanks,

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

On 31.08.2015 22:50, Gupta, Rahul wrote:

Hi Bogdan, following is the info you requested. This is not the
first tcp read, this server is been running for a while and taking
calls.

(gdb) bt

#0 0x7f2095572e2c in vfprintf () from /lib64/libc.so.6

#1 0x7f209560fed0 in __vsyslog_chk () from /lib64/libc.so.6

#2 0x7f2095610100 in syslog () from /lib64/libc.so.6

#3 0x004c4202 in tcp_read_req (con=0x7f2081a933c0,
bytes_read=0x7ffc6bc97f0c) at tcp_read.c:597

#4 0x004c5908 in handle_io (fm=,
idx=-1, event_type=) at tcp_read.c:1033

#5 0x004c8083 in io_wait_loop_epoll (unix_sock=) at io_wait.h:845

#6 tcp_receive_loop (unix_sock=) at
tcp_read.c:1141

#7 0x004b12e9 in tcp_init_children (chd_rank=, startup_done=0x0) at tcp_main.c:2389

#8 0x0043aebf in main_loop (argc=,
argv=) at main.c:1011

#9 main (argc=, argv=)
at main.c:1612

(gdb) f 3

#3 0x004c4202 in tcp_read_req (con=0x7f2081a933c0,
bytes_read=0x7ffc6bc97f0c) at tcp_read.c:597

597 tcp_read.c: No such file or directory.

in tcp_read.c

(gdb) p con->con_req

$1 = (struct tcp_req *) 0x7f20931a97e0

(gdb) p _req

$2 = (struct tcp_req *) 0x82d720

(gdb) p req

$3 = (struct tcp_req *) 0x7f20931a97e0

(gdb)

*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* Monday, August 31, 2015 3:45 PM
*To:* Gupta, Rahul; OpenSIPS users mailling lis

[OpenSIPS-Users] OpenSIPS Workshop 2015 in Chicago - Post Facts

[Vlad Paiu]

Hi everyone !

Following the requests of people who were not able to attend, we put 
together a nice collection of presentations and videos - all are shared 
here:


http://www.opensips.org/Community/Workshop-2015Chicago

Enjoy !
Thanks again to our sponsors, to the people who helped organize the 
event and, of course, to our beautiful audience !


Best regards,

--
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com


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Re: [OpenSIPS-Users] Ack without To tag

[Vlad Paiu]

Hello,

Please post a trace of this.

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 25.06.2015 06:25, John Nash wrote:

Let me add some more details which I noticed.

As i explained in previous post in the same subject, we receive ACK 
without to tag from one UA, since it will not match any dialog, I 
tried to route it to destination with topology_hiding but topology 
hiding used different branch tag than the 200 OK received (It only 
replaced .0 with .2). I think this is happening because I am also 
using drouting module and UAC which use branches.


I can post trace also if someone wants to have a look.


On Wed, Jun 24, 2015 at 10:06 PM, John Nash john.nash...@gmail.com 
mailto:john.nash...@gmail.com wrote:


I am using opensips 2.1 with topology_hiding module. I have an
issue only with one SIP endpoint. This endpoint sends Ack message
(after 200 OK to Invite) without any to tag because of that it is
not matching with In dialog request section.

Can a UA send ACK without to tag?...If yes any way I can match it
with ongoing dialog?

John




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Re: [OpenSIPS-Users] redis HMGET nil value problem

[Vlad Paiu]

Hello,

Currently, only STRING and INTEGER data types are supported - no NULL 
support in the cachedb raw interface.


Please open a GITHUB issue for this and will fix it as soon as possible.

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 28.05.2015 16:21, Jayesh Nambiar wrote:

Hi,
I'm using cache_raw queries to get data from redis and I have a 
problem with accessing nil values that gets returned from it.

For eg I do the following:
cache_raw_query(redis:myRedis HMGET one two three, $avp(result));

Here the value of one is abc, two is not present in redis and three is 
xyz. Now if I access the values, I ideally expect it to be the following:
$(avp(result)[0]) = xyz, $(avp(result)[1]) = null and 
$(avp(result)[2]) = abc.
But since key two is not present in redis for whatever reasons, the 
result I get back is:

$(avp(result)[0]) = xyz
$(avp(result)[1]) = abc
$(avp(result)[2]) = null

This actually makes my value matching go for a toss. Is there a better 
solution to access the array that is returned back from redis or at 
least store the value as NULL in that array element of avp. Any help 
is greatly appreciated.


Thanks,

- Jayesh


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Re: [OpenSIPS-Users] Websocket connection closes after tcp_connection_lifetime expires

[Vlad Paiu]

Hello,

I've just fixed on GIT the issue where the tcp_persistent_flag was 
ignored for websocket TCP connections ( as of commit 320c825 ) - please 
update your sources and the issue should be fixed, without having to 
change the tcp_connection_lifetime.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 19.05.2015 23:07, Patrick Wakano wrote:

Hello,

More tests with WebRTC in Opensips 2.1 and I noticed that the 
websocket connection of my browser was being disconnected some time 
after it registered, this caused the javascript lib to reconnect and 
re-register. This is not good because each time it happens, the 
browser uses a new port and I get one more contact to the same AOR in 
my location table...
Investigating the problem I noticed the websocket connection was being 
dropped because the Opensips' TCP connection lifetime was being 
reached and so it closed the connection.
First thing I did was to use the tcp_persistent_flag from the 
Registrar module. Unfortunately it didn't work So next test was to 
increased the tcp_connection_lifetime parameter to something greater 
than the Register expire time and luckly it worked! The websocket 
connection wasn't dropped anymore!

Even though now things are working, I think we have some bug in here...
I know the websocket runs over TCP but should it be ruled by the 
tcp_connection_lifetime parameter? I was expecting some independence, 
although I would completely understand if it is not possible from the 
implementation point of view.
Despite that, considering websockets fall under the same rules as TCP, 
then the tcp_persistent_flag should apply for it too, right?


Regards,
Patrick


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Re: [OpenSIPS-Users] Can't register outside local network

[Vlad Paiu]

Hello,

If your clients shows 408, then probably there was a timeout reaching 
the OpenSIPS IP.
What IP is OpenSIPS listening on - public vs private ? Also, is it 
reachable from the outside ? Check for issues like firewall / iptables 
rules blocking the 5060 port.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 20.05.2015 12:20, Manar Boukhari wrote:

Hi,

I've successfully installed OpenSIPs in Ubuntu and can register 
locally, but if i try to register from different network (from other 
post).  Got  : Account :xx could not be anabled, Problem at Server 
(SIP error 408). Try again later (I'm using X-Lite to register)


Is there a configuration that i need to add to make my opensips server 
reachable from outside ?


I'm a complete beginner in SIP and opensips, so I can't figure out 
what i'm missing ...


Any help would be appreciated,

Best regards,

Manar






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Re: [OpenSIPS-Users] Can't register outside local network

[Vlad Paiu]

Hello,

Just change the listen= directive at the very beginning of the script.

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 20.05.2015 16:19, Manar Boukhari wrote:


Hi,

Thanks for your reply, OpenSIPS is listening on a private IP . How can 
i use a public IP ?


And i have already opended the 5060 port.



2015-05-20 14:45 GMT+02:00 Vlad Paiu vladp...@opensips.org 
mailto:vladp...@opensips.org:


Hello,

If your clients shows 408, then probably there was a timeout
reaching the OpenSIPS IP.
What IP is OpenSIPS listening on - public vs private ? Also, is it
reachable from the outside ? Check for issues like firewall /
iptables rules blocking the 5060 port.

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  


On 20.05.2015 12:20, Manar Boukhari wrote:

Hi,

I've successfully installed OpenSIPs in Ubuntu and can register
locally, but if i try to register from different network (from
other post).  Got  : Account :xx could not be anabled,
Problem at Server (SIP error 408). Try again later (I'm using
X-Lite to register)

Is there a configuration that i need to add to make my opensips
server reachable from outside ?

I'm a complete beginner in SIP and opensips, so I can't figure
out what i'm missing ...

Any help would be appreciated,

Best regards,

Manar






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--
Cordialement
--
BOUKHARI Manar
Ingénieur d'études et de développement, Or-Robotics




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Re: [OpenSIPS-Users] Dynamic routing based on a shell variable with Opensips

[Vlad Paiu]

Hello,

You can fetch a shell variable in OpenSIPS by using the $env pvar in the 
cfgutils module ( see 
http://www.opensips.org/html/docs/modules/2.1.x/cfgutils#id294802 )


After that, routing to the actual nodes can be done by using the module 
of your choice ( dispatcher, stateless in terms of dialog counting, or 
load_balancer which takes into consideration the number of calls in each 
node )


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 01.05.2015 00:38, Kaan Dandin wrote:

Dear all,


I would like to consult to you if it is possible to make SIP routing( 
for my case to different IMS nodes) according to a shell variable . 
For example is the variable is 1 make the routing to IMS node1 , if 2 
make the routing to IMS node2 


What is the correct way to implement this scenario? Which module need 
to be used? Dispatcher module? Load Balancer Module or Dynamic Routing 
Module?


Thanks for your kind support

BR,
Kaan


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Re: [OpenSIPS-Users] not trigger on_failure_route[0] when use mi interface

[Vlad Paiu]

Hello,

Not sure I understand what is taking more than 10 minutes ?
Why are you trying to change the maximum number of milliseconds of 
OpenSIPS trying to retransmit a message ?


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 16.04.2015 07:04, chow wrote:

Hi：
 It work for me with use $du that redirect request to Main route.
 have another problem about  retransmission
 
 eg：

 I followed the order sent 50 messages.   40 of  message will  be  fast
store into db.
 the remaining message (40-50) store into  db  that  Spend  more than 10
minute.
 I  modify:
 modparam(tm, fr_timer, 5)
 modparam(tm, fr_inv_timer, 30)
 modparam(tm, T2_timer, 1000)
 
but it  always   spend long time to do it.

why  T2_timer no  work with a loss  value?
have some timer like TIMER B  or TIME F  for non-invite message?



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Re: [OpenSIPS-Users] SIP/2.0 477 Send failed (477/TM) - Route

[Vlad Paiu]

Hello,

Try to have an OpenSIPS localhost UDP listener, and do 
force_send_socket(udp:127.0.0.1:5060) before attempting to relay to 
127.0.0.1:9
In your case it fails since you're trying to relay via TCP, which fails 
since nobody listens over there - you won't have those issues with UDP.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 15.04.2015 23:04, leo wrote:

Enabling more detailed logs i  think the error would be $du =
sip:127.0.0.1:9;
Wouldn't be $ru instead of $du? And i receive
ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR
[server=127.0.0.1:9] (111) Connection refused

Thanks one more time,

Leo



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Re: [OpenSIPS-Users] command get_statistics is not available

[Vlad Paiu]

Hello,

The accepted syntax’s are either

 opensipsctl fifo get_statistics all

in order to fetch all statistics, or

 opensipsctl fifo get_statistics core:

in order to fetch all statistics form a certain module, or

 opensipsctl fifo get_statistics core rcv_requests

in order to fetch a single statistic.

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 15.04.2015 17:42, Dale Harris wrote:

Hi,

I have an OpenSIPS system that isn't allowing me to run opensipctl
fifo get_statistics.  Log show an error of:

opensips[51324]: ERROR:mi_fifo:mi_fifo_server: command get_statistics
is not available

Any suggestions on how to fix this?




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Re: [OpenSIPS-Users] Memory leak issue?

[Vlad Paiu]

Hello,

Is the leak you are reporting happening in shared memory, or pkg ? Your 
logs contain just PKG memory dump ?


OpenSIPS 1.9 is no longer supported ( see [1] ), so I would advise you 
to update to the latest OpenSIPS 1.11 ( latest minor release is 1.11.4 )


[1] www.opensips.org/About/AvailableVersions

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 09.04.2015 09:09, microx wrote:

Hi all,

I'm using OpenSIPS version 1.9.2 to set up a SIP outbound proxy and two
internal SIP servers in my testing environment.
All SIP requests and responses follow the same following routes.
SIP client -- SIP outbound proxy -- SIP server -- SIP outbound proxy --
SIP clients

With some stress tests in several days (many SIP client continue
registration), I find that OpenSIPS eats much more memory than what I
configuration. The eaten memory is not released after more than several hour
after the stress test is done. Following the official troubleshooting guide
about memory issue, I try to identify whether this is a memory leak issue.
Please kindly help to take a look at the attached log files. Besides, is it
possible that my incorrect configuration leads to memory leak? I am afraid
that I do not properly write the script to produce the memory leak issue.
Many thanks for any comment.

opensips_memory_leak_sipserver.gz
http://opensips-open-sip-server.1449251.n2.nabble.com/file/n7596394/opensips_memory_leak_sipserver.gz

opensips_memory_leak_sipoutbound.gz
http://opensips-open-sip-server.1449251.n2.nabble.com/file/n7596394/opensips_memory_leak_sipoutbound.gz


Best regards,
Chen-Che



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Re: [OpenSIPS-Users] SIP/2.0 477 Send failed (477/TM) - Route

[Vlad Paiu]

Hello,

What you can do is send the call to a destination which is not available 
at all, control the amount of time you want to give the client to 
register via the fr_timer, and when that timeout is exceeded try to 
route the call to the client.


Short snippet of code would be

if (!t_relay(3))
{
if((is_method(INVITE)))
{
exec_avp(some script to wake-up the client);
# relay to localhost, discart port
$du = sip:127.0.0.1:9;
# wait two seconds for the client to register
$T_fr_timeout = 2;
t_on_failure(route_to_client);
t_relay();
exit;
}
}

failure_route[route_to_client] {
if (t_was_cancelled())
exit;

# after two seconds, this will get called
# see if the client is registered now

lookup(location);
t_relay();
exit;
}

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 13.04.2015 18:20, leo wrote:

Hello Guys,

could you give the last clue on this? The point is once the UA is
re-registered, how to forward the call to it?
Thanks,

Leo



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Re: [OpenSIPS-Users] lookup(location) extract received field value

[Vlad Paiu]

Hello,

From you ul show output, you don't have to do anything special for this 
to work - OpenSIPS will automatically relay the call to the Received:: 
value that's displayed in the ul show output, setting it as $du, while 
the actual Request-URI of the message will contain the private Contact 
that the client registered.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 14.04.2015 18:09, Satish Patel wrote:


Hi,

I have following User registred over public IP but that client doesn't 
support STUN so contact info showing private IP 192.168.1.6


lookup function default extract Contact:: sip:1001@192.168.1.6:27098 
http://sip:1001@192.168.1.6:27098


Is there a way i can extract  Received:: sip:173.XX.XX.215:27098  so 
i can create new URI and send call to that?


if (lookup(location)) {
..
..
}


[root@sip ~]# opensipsctl ul show
Domain:: location table=512 records=1
AOR:: 1...@sip.example.com mailto:1...@sip.example.com
Contact:: 
sip:1001@192.168.1.6:27098;rinstance=e223da1c59d774db Q=

Expires:: 3585
Callid:: 
NjIyYzg5NzU0NGNlYjFhZTEyMDZlNDk2NTgzMDUzYjY

Cseq:: 2
User-agent:: X-Lite 4.7.1 74247-44615bc7-W6.1
Received:: sip:173.XX.XX.215:27098
State:: CS_SYNC
Flags:: 0
Cflags:: NAT
Socket:: udp:182.XX.XX.164:5060
Methods:: 5951


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Re: [OpenSIPS-Users] how do I check destination user is offline

[Vlad Paiu]

Hello,

If the destination is TCP, then you can detect the failure to relay in 
your OpenSIPS script. See the flags that t_relay() takes at 
www.opensips.org/html/docs/modules/1.11.x/tm#id294528


if (!t_relay(0x02)) {
# failure to relay , treat error here
}

Make sure to also set low tcp_send_timeout and tcp_connect_timeout 
values , in order to detect the failures faster ( 
http://www.opensips.org/Documentation/Script-CoreParameters-1-11#toc92 )


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 14.04.2015 15:02, chow wrote:

Hi ALL:
User A  sent register to opensips. then User A client core-dump,  so
User A will  never sent unregister message to opensips.
in this case , how do I  Immediately know User A unreachable .
registered function can not know this,
by the way,  transport protocol is TCP.
 thanks a lot.



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Re: [OpenSIPS-Users] loose_route() sending ACK itself

[Vlad Paiu]

Hello,

Looking in your SIP trace, I see that in the 200OK Contact, you have 
Contact: sip:72.XX.XX.140;did=7de.9accc6f5. , and when OpenSIPS is 
routing the ACK out, it is routing it to


 U 182.XX.XX.164:5060 - 72.XX.XX.140:5060
ACK sip:72.XX.XX.140;did=7de.9accc6f5 SIP/2.0.

so not sure where exactly is the loop.

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 07.04.2015 19:32, Satish Patel wrote:

It is URGENT!!

can some one help?  This is very strange issue and i am stuck here :(

loose_route()  sending ACK/BYE itself instead of next hope :(

I have removed all entries from domain table but no luck :(

On Thu, Mar 26, 2015 at 12:09 AM, Satish Patel satish@gmail.com 
mailto:satish@gmail.com wrote:


Hi,

senario:

[UA]-[Opensips]-[Freeswitch]


UA sending correct ACK to freeswitch but Opensips loose_route()
sending it to itself and it break dialog, If use
fix_dialog_route() then it works, I don't have any IP address
added in domain table also.

How do i check whether Freeswitch using loose_route for strict route?


I have following script:

if (has_totag()) {

if (loose_route()) {

   if (is_method(BYE)) {
#setflag(ACC_DO); # do accounting ...
#setflag(ACC_FAILED); # ... even if the transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is
useless, do RR for
# re-INVITEs alos, as some buggy
clients do change route set
# during the dialog.
record_route();
}

if (check_route_param(nat=yes))
setflag(NAT);

# route it out to whatever destination was
set by loose_route()
# in $du (destination URI).
route(relay);
 }  else {

if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but
stateful ACK; must be an ACK after
# a 487 or e.g. 404 from
upstream server
xlog(non loose-route section\n);
#t_relay();
exit;
} else {
# ACK without matching
transaction -
# ignore and discard
xlog(ACK without matching transaction\n);
exit;
}
}






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Re: [OpenSIPS-Users] Opensips is listenning port 5060, but NMap shows 5060 is closed

[Vlad Paiu]

Hello,

It seems you have a firewall issue, since indeed OpenSIPS is listening 
on TCP port 5060.

Fix that, and OpenSIPS should start receiving traffic.

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 27.03.2015 09:17, jacky wrote:

I got Opensips on an Ubuntu Cloud Server, it is listenning the port 5060.
http://opensips-open-sip-server.1449251.n2.nabble.com/file/n7596174/2015-03-27_150231.bmp

But I test on the remote pc client with tools, and shows that the 5060 was
closed:
http://opensips-open-sip-server.1449251.n2.nabble.com/file/n7596174/QQ%E5%9B%BE%E7%89%8720150327150551.png
http://opensips-open-sip-server.1449251.n2.nabble.com/file/n7596174/2015-03-27_150700.bmp

I am wondering that will it results to nothing received from sip clients
Jitsi on my pc sended REGISTER, meanwhile, the remote Opensips got nothing.

Thanks for your attention, I really appreciate your help!


Best regards!



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[OpenSIPS-Users] [Releases] Minor releases for 1.11, 1.10 and 1.8 branches

[Vlad Paiu]

Hello all,

New minor releases for the currently maintained versions will be done 
Thursday  2nd of April, as follows:

- 1.11.4
- 1.10.4
- 1.8.7

(see http://www.opensips.org/About/AvailableVersions).

Please let us know if you have any pending bug reports or suggestions.

Best Regards,

--
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OpenSIPS Developer
http://www.opensips-solutions.com


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Re: [OpenSIPS-Users] 'opensipsdbctl create' returns 'ERROR: relation call_center_id_seq does not exist'

[Vlad Paiu]

Hello,

This has been fixed on the 2.1 GIT branch.
Please pull the latest changes and try again.

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 20.03.2015 23:05, Emrah wrote:

Hi all,
A Google search on this pointed me to here, the mailing list.
I am using OpenSIPS2.1.0 and am trying to populate my Postgres 
database with the 'opensipsdbctl' script.

I am getting the following error:
ERROR:  relation call_center does not exist
ERROR:  relation call_center_id_seq does not exist
*ERROR: Grant privileges to extra tables failed!*

OpenSIPS is also my gateway to learning about Postgres, and I 
apologize for the limited info in this post.
A Google search returned a recent post about the same issue: 
http://opensips-open-sip-server.1449251.n2.nabble.com/Table-creation-errors-with-Postgres-td7595323.html
I didn't find anything more on the subject and resorted on posting 
here to ask for help.


Any pointers would be much appreciated!

Best,
Emrah


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Re: [OpenSIPS-Users] Intermittent call dropping issue

[Vlad Paiu]

Hello,

OpenSIPS shouldn't respond with the ACk - that should originate from the 
caller side and OpenSIPS should relay it.

Please post the full SIP trace for your call.

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 20.03.2015 16:13, Stuart Mills, Invosys wrote:


Hi,

I’m currently experiencing an issue when using OpenSIPS and FreeSWITCH 
within the same network.


Calls arrive at the OpenSIPS border, then I use the dispatcher module 
to send to one of 3 FreeSWITCH instances.


On a bad call it seems like 200OK is being sent back from FreeSWITCH 
on calls that will drop, it tries time and time again until giving up 
after 30 seconds, but OpenSIPS isn’t responding with the ACK. The IP 
addresses in the trace I’ve got look good, all internal NAT addresses 
so I’m not sure how it could be missing this packet.


Have any one experienced this using a similar setup? I can post the 
opensips.cfg file if necessary.


Regards,

*Stuart Mills*

Senior Network Engineer

*Invosys Ltd.*

Direct. 0161 444 7403

Web. www.invosys.com http://www.invosys.com/


cid:862D2963-B682-4FBA-A237-D1F118B1E624

*DISCLAIMER:* This e-mail and files transmitted with it are 
confidential and intended solely for the individual to whom they are 
addressed, and upon the basis that the recipient will conduct the 
appropriate virus checks. If you are not the addressee, you are not 
authorised to use the information or to place any reliance upon it, 
nor should you copy it or show it to anyone. Internet communications 
are not secure and Invosys Ltd are not responsible for the abuse by 
third parties, nor for any alteration or corruption in transmission, 
nor for any damage or loss caused by any virus or other defect. 
Registered and correspondence address: Invosys Ltd (5799390), New 
Bridgewater House, Mayfield Avenue, Worsley, Manchester, M28 3JF




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Re: [OpenSIPS-Users] ERROR:dialog:dlg_validate_dialog:

[Vlad Paiu]

Hello,

Just to recap, you are saying that the Contact the user agent is sending 
is broken and you are happy that OpenSIPS is properly fixing the 
message, but you want to get rid of the ERRORs in the log ? If this is 
the case, you can use setdebug [1] for this.


Try something like

setdebug(-3)
if ($DLG_status!=NULL  !validate_dialog() ) {
xlog( in-dialog bogus request \n);
fix_route_dialog();
}
setdebug()

http://www.opensips.org/Documentation/Script-CoreFunctions-2-1#toc48

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 18.03.2015 22:47, Satish Patel wrote:
I know you guys are super busy in OpenSIPS 2.1 release, but any 
suggestion on above issue?


On Wed, Mar 18, 2015 at 12:17 AM, Satish Patel satish@gmail.com 
mailto:satish@gmail.com wrote:


I am getting following error in log, I can understand my contact:
and Route: values mismatching here. why it is happening? is there
a way to get rid on this error?

Following is scenario. Only getting error in BYE message.

[UA][OpenSIP]---[Freeswitch]-[Opensip]-[SIP
Provide]


ERROR:dialog:dlg_validate_dialog: failed to validate remote
contact: dlg=[sip:16463737221
tel:16463737221@188.178.235.222:5061;transport=udp] ,
req=[sip:188.178.235.222;lr;ftag=840e2e35;did=1f4.ca6a6956]

I am using fix_route_dialog() in loose_route()

if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route() || match_dialog())  {
if ($DLG_status!=NULL 
!validate_dialog() ) {
xlog( in-dialog bogus request \n);
fix_route_dialog();
 }

xlog(L_INFO, Loose route failed on
$hdr(route)\n);
if (is_method(BYE)) {
#setflag(ACC_DO); # do accounting ...
#setflag(ACC_FAILED); # ... even
if the transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is
useless, do RR for
# re-INVITEs alos, as some buggy
clients do change route set
# during the dialog.
record_route();
}

if (check_route_param(nat=yes))
setflag(NAT);

# route it out to whatever destination was
set by loose_route()
# in $du (destination URI).
route(relay);
 }  else {

if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but
stateful ACK; must be an ACK after
# a 487 or e.g. 404 from
upstream server
xlog(non loose-route
section\n);
t_relay();
exit;
} else {
# ACK without matching
transaction -
# ignore and discard
xlog(ACK without matching
transaction\n);
exit;
}
}
xlog(L_INFO, destination uri after
loose_route: $du\n);
sl_send_reply(404,Not here);
}
exit;
}









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Re: [OpenSIPS-Users] error 483

[Vlad Paiu]

Hello,

483 usually means 'Too Many Hops'. If you do a SIP trace on the server, 
do you see OpenSIPS looping the request to itself ? Maybe the SIP phone 
sends the IP of the server instead of the domain that you have 
configured, and your script is configured to route out such requests.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 19.03.2015 04:15, Carlos Cruz wrote:


Hi;

Can someone tell me why  or where I may find the  info I need; I'm 
able to register external remote phones (hard phones), but the 
internal phone (soft phones) within the same network as the OpenSIPS 
test server report error 483.


Thanks!!

Carlos



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Re: [OpenSIPS-Users] Dispatcher user specific route question - 2.1

[Vlad Paiu]

Hello,

If you want to do dispatching between multiple setids, ds_select_dst() 
allows that. See the docs at [1] , you can provide a comma separated 
list of setids - so your $avp(zone) can contain '1,2' and OpenSIPS will 
try to first send to the servers in setid 1, and then, if those fail, to 
the servers in setid 2.


[1] http://www.opensips.org/html/docs/modules/1.11.x/dispatcher#id294368

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 19.03.2015 06:17, Satish Patel wrote:

I have add extra zone column in subscriber table,

+--+-+
| username |  zone |
+--+-+
|1001 |1|
|1002 |2|
+--+-+


In dispatcher table I have following two Freeswitch in two groups.

+---+-++
| setid | destination  | description|
+---+--+---+
| 1 | sip:fs1.example.com http://fs1.example.com | Freeswitch-1 |
| 2 | sip:fs2.example.com http://fs2.example.com | Freeswitch-2 |
+---+--+---+


in opensips.cfg script i am query subscriber table base on incoming 
username and storing zone in avp(zone) variable, and calling same 
variable in following code


if ( !ds_select_dst($avp(zone), 4, FM10))

Question: now either user belongs to zone 1 or 2, so it is *NOT* going 
to do load-balancing between two. But if I want to allow some user to 
do load-balancing then how it will be possible in above scenario?


Can i set setid on fly so i can pass request along with user request 
and set same group for both switch and user call load-balance on both 
switch?


Any other idea?


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Re: [OpenSIPS-Users] Jitsi sent REGISTER, Opensips received nothing

[Vlad Paiu]

Hello,

Well, if you did a tcpdump on the OpenSIPS box and saw nothing, then it 
means the packages aren't actually reaching the box. Please check that 
there are no firewalls in between the client and OpenSIPS that are 
blocking the traffic.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 19.03.2015 08:23, jacky wrote:

I have a test with one Jitsi using Opensips on the Internet
Wireshark showed me that Jitsi sent several REGISTER packages, in the same
time I used the command tcpdump to listen on the Opensips Server , but got
nothing.
Anyway, on the server, opensips, rtpproxy, media-dispatcher, media-relay are
running!
what happened to opensips server? why it won't response to distanced
request?

I appreciate your opinion, thanks a lot!

Best regards




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Re: [OpenSIPS-Users] Dispatcher user specific route question - 2.1

[Vlad Paiu]

Hello,

It will do fail-over.

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 19.03.2015 18:39, Satish Patel wrote:

Thanks Vlad,

Superb! so it will do round-robin? or fail-over?

On Thu, Mar 19, 2015 at 12:30 PM, Vlad Paiu vladp...@opensips.org 
mailto:vladp...@opensips.org wrote:


Hello,

If you want to do dispatching between multiple setids,
ds_select_dst() allows that. See the docs at [1] , you can provide
a comma separated list of setids - so your $avp(zone) can contain
'1,2' and OpenSIPS will try to first send to the servers in setid
1, and then, if those fail, to the servers in setid 2.

[1]
http://www.opensips.org/html/docs/modules/1.11.x/dispatcher#id294368

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  


On 19.03.2015 06:17, Satish Patel wrote:

I have add extra zone column in subscriber table,

+--+-+
| username |  zone |
+--+-+
|1001 |1|
|1002 |2|
+--+-+


In dispatcher table I have following two Freeswitch in two groups.

+---+-++
| setid | destination  | description|
+---+--+---+
| 1 | sip:fs1.example.com http://fs1.example.com |
Freeswitch-1 |
| 2 | sip:fs2.example.com http://fs2.example.com |
Freeswitch-2 |
+---+--+---+


in opensips.cfg script i am query subscriber table base on
incoming username and storing zone in avp(zone) variable, and
calling same variable in following code

if ( !ds_select_dst($avp(zone), 4, FM10))

Question: now either user belongs to zone 1 or 2, so it is *NOT*
going to do load-balancing between two. But if I want to allow
some user to do load-balancing then how it will be possible in
above scenario?

Can i set setid on fly so i can pass request along with user
request and set same group for both switch and user call
load-balance on both switch?

Any other idea?


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[OpenSIPS-Users] [NEW] Topology Hiding Module

[Vlad Paiu]

Hello,

I am happy to announce the new topology_hiding module.

Topology hiding is usually utilized as an approach to enhance SIP 
network security. Since, in regular SIP traffic, critical IP address 
data is forwarded to other networks, the concern is that third parties 
can use that information in order to direct attacks at your internal SIP 
network.


The new topology_hiding module strips and restores the headers that 
contain topology information (Via, Record-Route, Route and Contact 
headers) , and optionally it can also change the Call-ID of the requests.


Compared to the topology hiding solution offered by the B2B modules, it 
has the great advantage that it can be used together with other script 
functionalities that were previously incompatible ( eg. dialog vars  
profiles, accounting, etc.).
Compared to the previous solution of topology hiding which was 
integrated into the dialog module ( which only supported topology hiding 
for INVITE based dialogs ), the new module can work in stand-alone mode 
as well ( without dialog support ), thus allowing to do topology hiding 
for all types of dialogs ( eg. Presence dialogs ) and also for 
standalone initial requests ( eg. SIP MESSAGE ).


Find the module readme at [1] and a tutorial with a script example at [2].

Testing and feedback are very much welcome.

[1] http://www.opensips.org/html/docs/modules/2.1.x/topology_hiding.html
[2] http://www.opensips.org/Documentation/Tutorials-Topology-Hiding

Best Regards,

--
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com


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Re: [OpenSIPS-Users] TLS handling

[Vlad Paiu]

Hello,

OpenSIPS complains that there is an error when connecting via TCP to 
that endpoint.
Now, are you sure you do not have multple branches when relaying that 
SIP MESSAGE,out of which only one fails ? In t_relay(), if you have 
multiple branches and at least one succceeds, you will get a 1 return code.


Please post the complete debug=4 logs for the processing. In the 
previous email, you've truncated the logs to the moment OpenSIPS gets 
blocked in trying to connect to the endpoint - what happens afterwards ( 
after connet timeout ) would also be helpfull.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 06.03.2015 11:06, Чалков Артём wrote:

do anyone have any idea about how to handle that?

05.03.2015, 16:22, Чалков Артём achal...@ya.ru:

debug=4

Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:tcp_read_req: We're 
releasing the connection in state 3
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:io_watch_del: 
io_watch_del op on index -1 36 (0x77dee0, 36, -1, 0x10,0x1) fd_no=2 called
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:release_tcpconn:  
releasing con 0x7f2be91663a8, state 0, fd=36, id=1
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:release_tcpconn:  
extra_data (nil)
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_msg: SIP 
Request:
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_msg:  method:  
MESSAGE
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19028]: DBG:core:handle_tcp_child: 
reader response= 7f2be91663a8, 0 from 0
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_msg:  uri: 
sip:achalkov1@x.x.x.x:3631;transport=TCP
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19028]: DBG:core:io_watch_add: 
io_watch_add op on 52 (0x77dd80, 52, 2, 0x7f2be91663a8,1), fd_no=38
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_msg:  version: 
SIP/2.0
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_headers: 
flags=2
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_via_param: found param 
type 232, branch = z9hG4bK-d8754z-668ef50b1a4c0a31-1---d8754z-; state=6
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_via_param: found param 
type 235, rport = n/a; state=17
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_via: end of 
header reached, state=5
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_headers: via 
found, flags=2
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_headers: this 
is the first via
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:receive_msg: After 
parse_msg...
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:receive_msg: 
preparing to run routing scripts...
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_headers: 
flags=800
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_to: end of 
header reached, state=10
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_to: 
display={}, ruri={sip:achalkov1@x.x.x.x:3631;transport=TCP}
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:get_hdr_field: To 
[51]; uri=[sip:achalkov1@x.x.x.x:3631;transport=TCP]
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:get_hdr_field: to body 
[sip:achalkov1@x.x.x.x:3631;transport=TCP#015#012]
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:get_hdr_field: cseq CSeq: 
3 MESSAGE
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: 
DBG:maxfwd:is_maxfwd_present: value = 70
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_headers: 
flags=
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:get_hdr_field: 
content_length=3
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:get_hdr_field: 
found end of header
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_headers: 
flags=
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:decode_mime_type: 
Decoding MIME type for:[text/plain]
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_to_param: 
tag=b2b91161
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_to: end of 
header reached, state=29
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_to: 
display={achalkov}, ruri={sip:achalkov@x.x.x.x:3631;transport=TCP}
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_headers: 
flags=
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_methods: 
methods 0x173F
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:uri:has_totag: no totag
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_headers: 
flags=78
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:tm:t_lookup_request: 
start searching: hash=32018, isACK=0
Mar  5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:tm:matching_3261: 
RFC3261 transaction matching

Re: [OpenSIPS-Users] TLS handling

[Vlad Paiu]

Hello,

Since TLS doesn't support async in 1.11, you should get an error 
straight out of t_relay()

Can you please post the full debug logs here ?

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 05.03.2015 13:44, Чалков Артём wrote:

Hi all!

Can someone help us?. I cannot understand how TLS in opensips 1.11 works.
The problem is when we use TLS, i cannot handle connection problems.

When i use TCP in async mode, i have 408 in failure route when outgoing TCP 
connection fails, when i use TCP in sync mode, i have negative status after 
t_relay(), however, after TLS, i cannot catch neither 408, or negative 
t_relay() status. So, how to handle TLS connection error?

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Re: [OpenSIPS-Users] Boolean values from DB and avp

[Vlad Paiu]

Hello,

From the OpenSIPS lexer :

YES yes|true|on|enable
NO  no|false|off|disable

All of them are equivalent, and are correspondent to integer 1 and 
respectively integer 0.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 23.02.2015 17:09, Patrick Wakano wrote:

Thank you very much for the answer Vlad!

This raises me another question. I've been using boolean values with 
avp and found no problem so far. For instance, I set $avp(test) = 
true, and then use it in an if clause and everything works fine! But 
I'm really not sure if this is correct, and the docs are a little 
fuzzy regarding avp and boolean values. So can you clarify me what is 
the real support of boolean types in Opensips?


Regards,
Patrick


On Mon, Feb 23, 2015 at 11:58 AM, Vlad Paiu vladp...@opensips.org 
mailto:vladp...@opensips.org wrote:


Hello,

In OpenSIPS, AVPs can have either integer or string values.
For boolean values in Postgres, the module internally stores them
as strings, so you will either get 't' or 'f' in your script - so
yes, it's an OpenSIPS restriction in some ways.

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  


On 23.02.2015 16:36, Patrick Wakano wrote:

Hello list,

I've been using Opensips 11.3.1 with a Postgres DB 9.1 and
noticed that if I perform an avp_db_query call to retrieve any
boolean value from DB, the result gives me an avp with a string
and not the boolean value. For instance if I have any false value
in my DB, my avp_db_query gives me an avp with the string value
f and not the boolean value false.
Is this behaviour due to some misconfiguration in my DB or
Opensips, or is it an Opensips restriction? The docs shows The
value type of the AVP (string or integer) will be derived from
the type of the columns. but have no mention to boolean...

Thanks,

Patrick


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Re: [OpenSIPS-Users] Boolean values from DB and avp

[Vlad Paiu]

Hello,

In OpenSIPS, AVPs can have either integer or string values.
For boolean values in Postgres, the module internally stores them as 
strings, so you will either get 't' or 'f' in your script - so yes, it's 
an OpenSIPS restriction in some ways.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 23.02.2015 16:36, Patrick Wakano wrote:

Hello list,

I've been using Opensips 11.3.1 with a Postgres DB 9.1 and noticed 
that if I perform an avp_db_query call to retrieve any boolean value 
from DB, the result gives me an avp with a string and not the boolean 
value. For instance if I have any false value in my DB, my 
avp_db_query gives me an avp with the string value f and not the 
boolean value false.
Is this behaviour due to some misconfiguration in my DB or Opensips, 
or is it an Opensips restriction? The docs shows The value type of 
the AVP (string or integer) will be derived from the type of the 
columns. but have no mention to boolean...


Thanks,

Patrick


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Re: [OpenSIPS-Users] remove_hf does not remove header

[Vlad Paiu]

Hello,

Due to OpenSIPS internals, the changes to the message are stored and 
only applied when the message goes out - that's why removing a header 
and then checking for it's existence will not appear consistent. But 
nonetheless, the change you make in the script to the SIP message will 
get propagated.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 12.02.2015 11:22, Podrigal, Aron wrote:

Hi,


I'm want to remove the Remote-Party_ID header, but it does not work.

if (is_present_hf(Remote-Party-ID)) {
xlog(removing Remote-Party-ID);
 remove_hf(Remote-Party-ID);
}

if (is_present_hf(Remote-Party-ID))
xlog(Remote-Party-ID is still present);


I tried this in several  places, within a branch_route, main route. 
but it does not work.

Any help? what am I doing wrong?

Thanks


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Re: [OpenSIPS-Users] Testing Event_Route with USRLOC

[Vlad Paiu]

Hello Duane,

The usrloc module triggers a generic event - to which you subscribe via 
various ways ( run an OpenSIPS route, simple UDP packet, xmlrpc, rabbitmq ).


If you want a route to be run , you simple need to load the 
event_route.so module - then, by simply having an event route with the 
name of the exported events, you will  be auto-registered to them.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 10.02.2015 06:31, Duane Larson wrote:
I'm trying out event_route with USRLOC but I am not seeing any xlogs.  
I'm not 100% clear how to set up event_route's but it seems like it is 
basically just creating the following


modparam(usrloc, db_mode,   2)

event_route[E_UL_AOR_INSERT] {
xlog(The E_UL_AOR_INSERT event was raised\n);
  }

event_route[E_UL_AOR_DELETE] {
xlog(The E_UL_AOR_DELETE event was raised\n);
  }



Is there anything else that needs to be done in the script?  If I 
register or register a client I am not seeing any xlog messages.  And 
if I delete a client using opensipsctl fifo ul_rm I don't see 
anything.  Running OpenSIPS 1.11.2-notls



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Re: [OpenSIPS-Users] Last Modified AOR for User

[Vlad Paiu]

Hello,

Indeed, date fields where not properly supported in the cachedb_mongodb 
module.
I've just committed a fix for this - so please update your GIT sources 
to the latest ones.


I've just tested this with a small script like :

if (!save(location))
sl_reply_error();

sleep(1);

cache_raw_query(mongodb:instance1,{ \op\ : 
\find\, \ns\ : \location.location\, \query\: {\username\ : 
\vlad\} },$avp(mongo_result));

$json(json_res) := $avp(mongo_result);
$avp(expiration) = $json(json_res/expires)-$Ts;
$avp(last_edited) = $Ts-$json(json_res/last_modified);
xlog(Username vlad will expired in $avp(expiration) 
seconds and was edited $avp(last_edited) seconds ago \n);


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 10.02.2015 04:26, sevpal wrote:
The data in the DB are stored correctly, there aren't any problems 
there, I can view all the fields properly set with 
db.location.find() . On a raw Query though, all the other fields 
return with Json data except the date/time fields. It seems mongo 
needs a different kind of directive than what is currently in the 
module to retrieve the data.

*From:* Bogdan-Andrei Iancu mailto:bog...@opensips.org
*Sent:* Monday, February 09, 2015 4:24 AM
*To:* sevpal mailto:sev...@aol.com ; OpenSIPS users mailling list 
mailto:users@lists.opensips.org

*Subject:* Re: [OpenSIPS-Users] Last Modified AOR for User
Hi Jalung,

If you list the records in mongoDB collection, do you see the 
last_modified field properly set ? (I'm trying to understand if you 
have a problem with the data in DB or with the query itself)


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 07.02.2015 01:47, sevpal wrote:
If I'm storing the location table in a mongodb collection, how to 
query the most current AOR for a user? I can query all the fields 
except the date/time fields eg; last_modified, they return empty. 
I'm doing this using the mongo raw  query in Opensips.

Jalung


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Re: [OpenSIPS-Users] Validate dialog confusion

[Vlad Paiu]

Hello,

In order to be able to route such broken sequential requests with no 
Route headers you must first catch them in your script with match_dialog 
, eg :


if (loose_route() || match_dialog())

and then you need to fix them

if ($DLG_status!=NULL  !validate_dialog() ) {
if (fix_route_dialog())
xlog( received bogus in-dialog request, but properly fixed \n);
else
xlog( in-dialog bogus request \n);
}

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 31.12.2014 14:27, John Nash wrote:

Or may be should try to match_dialog and then relay

On Wed, Dec 31, 2014 at 5:57 PM, John Nash john.nash...@gmail.com 
mailto:john.nash...@gmail.com wrote:


OK got it. I also have a situation where I receive BYE from callee
which has To tag but no route header. As per default script this
request should not be allowed to go further. Would
fix_route_dialog(); work or I should just relay it?

On Wed, Dec 31, 2014 at 1:37 AM, Bogdan-Andrei Iancu
bog...@opensips.org mailto:bog...@opensips.org wrote:

Hi John,

1) if the request does not match any dialog it will go on the
else branch where it simply logs the valid request - this
is a simple example, but the idea is that without dialog state
you cannot check if valid or not; and you maybe do not create
dialogs for all your calls and maybe not all your sequential
request are part of a INVITE dialog (you may have re-SUBSCRIBE
or NOTIFYs)

2) the function just fixes, you need to send it out as usual
via the t_relay(),

Regards,

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

On 30.12.2014 12:16, John Nash wrote:

I was looking at usage of validate_dialog() and saw the
following snippet from module documentation page. If you
check the else part... as per my understanding if a dialog is
not found ($DLG_status) will be NULL) it will go to else part
which indicates a message has To tag but dialog not found.
How can it be a valid in-dialog request.

 if (has_totag()) {
 loose_route();
 if ($DLG_status!=NULL  !validate_dialog() ) {
 xlog( in-dialog bogus request \n);
 } else {
 xlog( in-dialog valid request - $DLG_dir !\n);
 }
 }
Also there is one more code
if (has_totag()) {
 loose_route();
 if ($DLG_status!=NULL)
 if (!validate_dialog())
 fix_route_dialog();
 }

After calling fix_route_dialog() do I need to just relay the
message? I mean this function will fix the message and
send?I was trying to look for some security check where
someone can inject a route header and a to tag (Bogus of
course) to reach to my gateways through my proxy.



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Re: [OpenSIPS-Users] INVITE/FROM/TO Header change possible?

[Vlad Paiu]

Hello,

The R-URI domain is a R/W variable, $rd ( 
http://www.opensips.org/Documentation/Script-CoreVar#toc65 )


For the FROM and TO headers, you need to use the uac_replace_from / 
uac_replace_to functions from the UAC module ( see 
http://www.opensips.org/html/docs/modules/1.11.x/uac.html#id293640 )


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 11.12.2014 21:34, bluerain wrote:

I have Asterisk as media gateway and Opensip in front.  So the issue I am
running into is that, in reference to Asterisk, it send calls to Opensips,
it is not aware of the carrier which Opensips is sending to.

Thus:
1. INVITE header would be the IP address of the Opensips
2. From header would be IP of asterisk itself.
3. To header is also the IP of the Opensips.

Now, when Opensips send the call out to the far end carrier, it doesn't
manipulate any of these information.  But then it cause the far end
carrier not liking it.  The far end carrier want to see:

1. INVITE header would be the IP address of the Carrier's SBC (which
Opensips knows, but not asterisk)
2. From header would be IP of Opensips itself
3. To header is also the IP of the Caririer.

I checked the core variable doc, It does not have R/W on these herader.
So how can I change them?

Thx in advance.



--
View this message in context: 
http://opensips-open-sip-server.1449251.n2.nabble.com/INVITE-FROM-TO-Header-change-possible-tp7594681.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

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Re: [OpenSIPS-Users] Parse initial INVITE Request-URI

[Vlad Paiu]

Hello,

Who is doing the actual forward ? Is it a SIP proxy / entity located 
before OpenSIPS ?


Just by looking at the INVITE message you've posted, the originally 
dialled number ( 858555 ) is located in the TO username ( readable 
via the $tU variable in the script ).


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 12.12.2014 03:57, discodo...@aol.com wrote:

Good day to all.
Here is what I am trying to do and I am hoping someone can set me in 
the right direction.  I am using Opensips 1.11.2


I have an inbound SIP call.  This call is actually a forward.  When I 
dial the number 858555 it is forwarded to 858555.


Dialed number: 858555 forwards to 1855222

When the call comes in Opensips this is my initial invite...


INVITE sip:+1855222@111.222.333.444:5060 SIP/2.0
Via: SIP/2.0/UDP 11.11.11.11:5060;branch=z9hG4bK08B4ca619b00ce6405c
From: sip:+1858666@11.11.11.11:5060;isup-oli=0;tag=gK084a85fd
To: sip:858555@111.222.333.444:5060
Call-ID: 537432242_53736858@11.11.11.11
CSeq: 17719 INVITE
Max-Forwards: 93
Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE
Accept: application/sdp, application/isup, application/dtmf, 
application/dtmf-relay,  multipart/mixed

Contact: sip:+1858666@11.11.11.11:5060
P-Asserted-Identity: sip:+1858666@11.11.11.11:5060
Diversion: sip:+1858555@11.11.11.11:5060;privacy=off;screen=no; 
reason=no-answer; counter=1

Supported: 100rel
Content-Length:  200
Content-Disposition: session; handling=required
Content-Type: application/sdp

I see that the initial INVITE shows the toll free number that the call 
was forwarded to but the TO: shows the original dialed number.
My question is can I grab the phone number from the initial INVITE?  I 
have goggled around and tried using the search but it just returns a 1 
or 0 for a match.  I know once I am able to grab the number from the 
INVITE can then update the TO:field.  I have dumped out almost all of 
the vars from Opensips and none of them show me the phone number from 
the initial INVITE.


Any help is appreciated.

Thanks,


James



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Re: [OpenSIPS-Users] NAT configuration residential script

[Vlad Paiu]

Hello,

Always using rtpproxy_engage() on the initlal INVITE is not an ideal 
solution.
At the script level, in OpenSIPS you can know if either the source or 
destination is NATed.


The residential script can be configured to properly support NAT - do 
'make menuconfig' , then go to 'Generate OpenSIPS Script' - Residenti 
Script - Configure Residential Script, and Check USE_NAT option ( 
spacebar key ), then go back ( q key ) and hit Generate Residential Script .


That should give you a good starting point on how to properly handle 
both the source and destination NAT.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 03.12.2014 06:50, campusvtv wrote:

Hello,

I'm tryng residential script in a basic scenario.

I'm using Jitsi and XLite for my tests. Both are behind a NAT and 
OpenSIPs listen on a Public IP.


If I call from Jitsi to X-LIte the audio work on both ways because 
OpenSIPs detect JITSI SDP INVITE contain a private address:


c=IN IP4 192.168.1.4.

and change it to rtpproxy public address before send INVITE to X-Lite 
because I'm using nat_uac_test(31) where 8 - SDP is searched for 
occurrence of RFC1918 / RFC6598 addresses


If I call from X-Lite to Jitsi, audio only on X-Lite.

The problem is on Jitsi SDP 200 ok there is this line:

c=IN IP4 192.168.1.4.

So X-Lite try to send RPT stream to this address.

If I use rtpproxy_engage on the initial INVITE, audio work always but 
I think not is the best solution.


Any hint?

Regards

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Re: [OpenSIPS-Users] Some phones can call, others not...

[Vlad Paiu]

Hello,

If the UDP packet is fragmented and due to whatever reasons the UA 
doesn't receive it properly, there are a couple of things that you can 
do to reduce the SIP message size :


1 - enable topology hiding in OpenSIPS ( see 
http://www.opensips.org/html/docs/modules/1.11.x/dialog.html#id296413 )
2 - manually remove optional headers from the OpenSIPS script ( 
http://www.opensips.org/html/docs/modules/1.11.x/sipmsgops#id293936 )
3 - remove some of the codecs from the SDP ( 
http://www.opensips.org/html/docs/modules/1.11.x/sipmsgops#id294821 )


Also, another alternative would be to switch to TCP in order to avoid 
these fragmentation issues.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 01.12.2014 16:19, Stefano Pisani wrote:
In the first case the last UDP packet (INVITE with authentication 
header) is fragmented.

It's very common to lost fragmented UDP packet.

s

Il 28/11/2014 09.42, Michele Pinassi ha scritto:

Hi all,

i'm experiencing a strange issue. Some VoIP phones, like mine (5002),
cannot call other phones (like 5023) but i receive calls from 5023. Same
config, same context.

I did some sipgrep:

Call from 5002 to 5023 (FAILED):http://pastebin.com/BF6YyWHr
Call from 5023 to 5002 (SUCCESS):http://pastebin.com/rW3AKr22

My config:http://pastebin.com/9gP9xncd

Thanks for all your help.

Michele



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Re: [OpenSIPS-Users] Not forwarding 200 ok sporatically

[Vlad Paiu]

Hello,

Looking at your OpenSIPS logs, it seems that the 200 OK was succesfully 
relayed ( 2302 bytes written ) :


Nov 13 20:05:44 mediaproxy2 /sbin/opensips[311]: DBG:core:tcp_send: 
after write: c= 0x7f80dc6230d0 n=2302 fd=41
Nov 13 20:05:44 mediaproxy2 /sbin/opensips[311]: DBG:core:tcp_send: 
buf=#012SIP/2.0 200 Ok#015#012Via: SIP/2.0/TCP 
192.168.2.93:45442;received=24.186.16.85;branch=z9hG4bK.gLQAkbdo-;rport=45442#015#012From: 

Nov 13 20:05:44 mediaproxy2 /sbin/opensips[311]: DBG:tm:relay_reply: 
sent buf=0x7f80e70df3a8: SIP/2.0 2..., shmem=0x7f80dc616370: SIP/2.0 2


You do a ngrep trace on the OpenSIPS box, you do not see that 200 OK 
going out ?

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 02.12.2014 13:41, Tito Cumpen wrote:
Any idea what would cause this issue?? Not sure Michele and I are 
facing the same issue.


On Wed, Nov 26, 2014 at 3:53 AM, Michele Pinassi 
michele.pina...@unisi.it mailto:michele.pina...@unisi.it wrote:


A similar issue happens to me, when from some voip phones i try to
call other voip phones.

For example, when from 5002 i try to call 5023.

Here's my config: http://pastebin.com/9gP9xncd

This is what happens on server 172.20.1.2 http://172.20.1.2:
http://pastebin.com/gZNSwyHB

This is the log on 5002: http://pastebin.com/RXtqwMtz

But on 5023 no INVITE arrive... :-( this happens only from some
phones while on others calling 5023 works as expected.

Michele


Il 25/11/2014 17:24, Vlad Paiu ha scritto:

Hello,

Please provide the SIP trace and the OpenSIPS full debug log.

Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  
On 25.11.2014 11:50, Tito Cumpen wrote:

group


I am having an issue in which opensips will ignore the the 200
ok response to an invite . I am using Opensips 1.11. This will
cause the the b leg to assume a call and the A leg to be treated
by the failure route. I cant find the initial invite in the
debug logs but see the 200 ok response. I have captures of a
working call vs a non working and cannot find a difference that
would be the reason behind the failure. Is there anything I can
provide to help identify the cause of this issue?


Thanks,
Tito


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-- 
Michele Pinassi

Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di 
Siena
tel: 0577.(23)2169 - fax: 0577.(23)2053

Per trovare una soluzione rapida ai tuoi problemi tecnici
consulta le FAQ di Ateneo,http://www.faq.unisi.it  



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Re: [OpenSIPS-Users] Some phones can call, others not...

[Vlad Paiu]

Hello,

For the  call from 5002 to 5023, OpenSIPS is properly delivering the 
call, but gets no reply from the UAS.


Is the 5023 account actually available at 172.20.1.27:32768 ? Can you 
take a trace on the actual phone and see if the packets are reaching it ?


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 28.11.2014 10:42, Michele Pinassi wrote:

Hi all,

i'm experiencing a strange issue. Some VoIP phones, like mine (5002),
cannot call other phones (like 5023) but i receive calls from 5023. Same
config, same context.

I did some sipgrep:

Call from 5002 to 5023 (FAILED): http://pastebin.com/BF6YyWHr
Call from 5023 to 5002 (SUCCESS): http://pastebin.com/rW3AKr22

My config: http://pastebin.com/9gP9xncd

Thanks for all your help.

Michele



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Re: [OpenSIPS-Users] Not forwarding 200 ok sporatically

[Vlad Paiu]

Hello,

Please provide the SIP trace and the OpenSIPS full debug log.

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 25.11.2014 11:50, Tito Cumpen wrote:

group


I am having an issue in which opensips will ignore the the 200 ok 
response to an invite . I am using Opensips 1.11. This will cause the 
the b leg to assume a call and the A leg to be treated by the failure 
route. I cant find the initial invite in the debug logs but see the 
200 ok response. I have captures of a working call vs a non working 
and cannot find a difference that would be the reason behind the 
failure. Is there anything I can provide to help identify the cause of 
this issue?



Thanks,
Tito


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[OpenSIPS-Users] OpenSIPS Summit 2014 in Las Vegas - Post Facts

[Vlad Paiu]

Hi everyone !

Following the requests of people who were not able to attend, we put 
together a nice collection of presentations, videos and photos - all are 
shared here:


http://www.opensips.org/Community/Summit-2014LasVegas

Enjoy !
Thanks again to our sponsors, to the people who helped organize the 
event and, of course, to our beautiful audience !


Best regards,

--
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com


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Re: [OpenSIPS-Users] Dialog Replication Behaviour

[Vlad Paiu]

Hello,

This bug was fixed as of 
https://github.com/OpenSIPS/opensips/commit/59a0360a9dbaf0c13b95ce7d038b3225ce0e4caa


Please update your sources to the latest 1.11 branch, there have been 
several other dialog BIN replication fixes.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 09.09.2014 22:44, Federico Edorna wrote:
Hello, I'm working on a redundant solution with Opensips version 
1.11.2 using dialog replication via Binary Interface.


I've noticed that the active server sends the binary create command to 
the backup server only when the dialog is confirmed.
The problem I've found is than when a call is canceled by the caller 
(the dialog in active server is unconfirmed or in early state), active 
server destroy the local dialog and also send the delete command to 
the backup server. As the backup server does not have the dialog 
created, I get an error:


ERROR:dialog:dlg_replicated_delete: dialog not found (callid: 
|64f64dce-96d9658a@192.168.

20.108| ftag: |f2fcd0fe6f793c1ao1|


Is this behaviour correct? Unconfirmed/Early State Dialog are not 
replicated to backup server? If this is the case, should I ignore this 
errors? Or maybe replication for deletion should only be sent if 
dialog was confirmed?


Many Thanks!

Federico


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Re: [OpenSIPS-Users] Blocking Premium and Special Routes

[Vlad Paiu]

Hello,

One alternative would be to use the drouting module to store the prefix 
list for the premium / special services routes, and use do_routing() 
just for matching those prefixes - if you have a match, simply reject 
the call.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 31.08.2014 03:48, Nick Cameo wrote:

Hello Everyone,

We would like to stop users from trying to terminate calls to premium 
and special
services routes. Most vendors already block this however, just in case 
we come

across one that does not, we would like to address the issue on our side.

What we have come up with is storing the codes in a MySQL db, and use
memchached to dip into the table on interval and cache the current 
info from

the persistent storage. Finally, use the cachedb module to pull the key
value pair from the memcached on every call.

Any other approaches or suggestions would be greatly appreciated. Just
weighing in our options.

Thanks in Advance,

Nick.


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Re: [OpenSIPS-Users] Blocking Premium and Special Routes

[Vlad Paiu]

Hello,

This will most likely perform better than any other solution - since 
every prefix is fully cached and also drouting stores all the prefixes 
in a trie, so the search doesn't depend on the overall number of blocked 
prefixes.


By default do_routing() will do longest prefix matching. For your case, 
since you're not doing routing, you should use the 'C' flag, just for 
checking the prefix.

If you want to do exact number lookup, look at the 'L' flag.

http://www.opensips.org/html/docs/modules/1.11.x/drouting.html#id294716

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 01.09.2014 15:44, Nick Cameo wrote:

One more thing. If I do use the drouting module, which I would like to.
 how will the matching work (ie, longest, shortest, exact?).

N.



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Re: [OpenSIPS-Users] topology_hiding() not executing

[Vlad Paiu]

Hello,

That's why I was asking for the call's SIP trace - seems to be an issue 
with the newly added CSEQ increase features.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 29.08.2014 08:11, Satish Patel wrote:
Very interesting thing happened, If i am authentication trunk using 
uac_auth() function then it is not handling BYE from callee, but if i 
use IP base authentication on trunk between opensips and asterisk it 
works! and also handling BYE properly. Don't you think it is wired?


Here is my asterisk sip.conf  if i comment out permit ip and 
insecure=invite its works!  with BYE message also. Why it is not 
working properly with UAC authentication?


[4545]
type=friend
host=182.72.242.164
context=office
;insecure=invite
;deny=0.0.0.0/0.0.0.0 http://0.0.0.0/0.0.0.0
;permit=182.xx.xx.xx/255.255.255.255 http://255.255.255.255
secret=password
allow=all
nat=yes



On Thu, Aug 28, 2014 at 8:40 AM, Vlad Paiu vladp...@opensips.org 
mailto:vladp...@opensips.org wrote:


Hello,

Please privately send me again the SIP trace for the call.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  


On 28.08.2014 15:23, Satish Patel wrote:

I have tried to remove RR but it didn't help :(  I have put top
hide after RR no luck.

Sent from my iPhone

On Aug 28, 2014, at 3:16 AM, Vlad Paiu vladp...@opensips.org
mailto:vladp...@opensips.org wrote:


Hello,

When doing topology hiding, there's no need to call
record_route() at all, so please remove the call to
record_route() from your script, or move topology_hiding() after
the record_route() function call.

Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  
On 27.08.2014 20:25, Satish Patel wrote:

Hi Vlad,

I have created pastebin for Asterisk, Opensips and opensips.cf
http://opensips.cf file, I am going to send you in your
private email address because of security reason.

Let me know if you see any issue in my configuration, or any
kind of suggestion.

Appreciate your help.


On Mon, Aug 25, 2014 at 11:48 PM, Satish Patel
satish@gmail.com mailto:satish@gmail.com wrote:

I have put topology_hiding() function at following place in
script but its not hiding VIA header following is my senerio

[UA][Opensips]---[Asterisk/SIP gateway]

I want to hind my UA IP address so Asterisk doesn't see
them, currently my asterisk can see what IP address UA
coming from, where should i put them generally


if (is_method(INVITE)) {
...
...
if  ( uri=~^sip:[0-9]*@.* mailto:%5Esip:[0-9]*@.*) {
uac_replace_from(sip:4545@65.111.170.127
mailto:sip%3A4545@65.111.170.127);
t_on_failure(3);
resetflag(7);
t_relay( udp:65.111.170.127:5065
http://65.111.170.127:5065 );
 topology_hiding();
exit;
};









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Re: [OpenSIPS-Users] topology_hiding() not executing

[Vlad Paiu]

Hello,

When doing topology hiding, there's no need to call record_route() at 
all, so please remove the call to record_route() from your script, or 
move topology_hiding() after the record_route() function call.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 27.08.2014 20:25, Satish Patel wrote:

Hi Vlad,

I have created pastebin for Asterisk, Opensips and opensips.cf 
http://opensips.cf file, I am going to send you in your private 
email address because of security reason.


Let me know if you see any issue in my configuration, or any kind of 
suggestion.


Appreciate your help.


On Mon, Aug 25, 2014 at 11:48 PM, Satish Patel satish@gmail.com 
mailto:satish@gmail.com wrote:


I have put topology_hiding() function at following place in script
but its not hiding VIA header following is my senerio

[UA][Opensips]---[Asterisk/SIP gateway]

I want to hind my UA IP address so Asterisk doesn't see them,
currently my asterisk can see what IP address UA coming from,
where should i put them generally


if (is_method(INVITE)) {
...
...
if  ( uri=~^sip:[0-9]*@.*) {
uac_replace_from(sip:4545@65.111.170.127
mailto:sip%3A4545@65.111.170.127);
t_on_failure(3);
resetflag(7);
t_relay( udp:65.111.170.127:5065
http://65.111.170.127:5065 );
 topology_hiding();
exit;
};




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Re: [OpenSIPS-Users] topology_hiding() not executing

[Vlad Paiu]

Hello,

Please post the relevant part of your script when calling 
topology_hiding() and when routing sequential requests, and also please 
pastebin a full SIP trace showing the traffic for such a dialog.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 26.08.2014 15:38, Satish Patel wrote:
I have tried your logic and it works but it is not handling BYE 
message, after caller hang up phone, caller not receiving BYE and 
caller phone is still in connected state not getting hung up.


Sent from my iPhone

On Aug 26, 2014, at 5:09 AM, Vlad Paiu vladp...@opensips.org 
mailto:vladp...@opensips.org wrote:



Hello,

You must call topology_hiding() before t_relay() - please try that.
Also, make sure to change your sequential request handling from

 if (loose_route())

to

 if (loose_route() || match_dialog())


Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  
On 26.08.2014 06:48, Satish Patel wrote:
I have put topology_hiding() function at following place in script 
but its not hiding VIA header following is my senerio


[UA][Opensips]---[Asterisk/SIP gateway]

I want to hind my UA IP address so Asterisk doesn't see them, 
currently my asterisk can see what IP address UA coming from, where 
should i put them generally



if (is_method(INVITE)) {
...
...
if  ( uri=~^sip:[0-9]*@.*) {
uac_replace_from(sip:4545@65.111.170.127 
mailto:sip%3A4545@65.111.170.127);

t_on_failure(3);
resetflag(7);
t_relay( udp:65.111.170.127:5065 
http://65.111.170.127:5065 );

 topology_hiding();
exit;
};



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Re: [OpenSIPS-Users] need help - Insert_hf when Route: missing

[Vlad Paiu]

Hello,

Please be a little bit more patient - last reply for your questions was 
yesterday :)


About your problem, you do not need to handle Route headers at all - 
this is the entire point of topology hiding, OpenSIPS adds no Route: 
header to it, since it acts like the endpoint, and the callee must 
contact OpenSIPS by the Contact headers only.


As requested in the previous users list email, please post SIP traces of 
your calls to see exactly what's going on ( leave plain topology hiding 
in place, please remove your hacks with the Route headers ).


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 27.08.2014 14:08, Satish Patel wrote:

I have post many question on topology hiding any not get any reply back from 
people and developers now I don't have any option except some goofy hack

When I use topology hiding it removes Route: and because of that callee doesn't 
able to send BYE back to opensips.

I want use insert_hf to inject Route: so I get clean BYE message. Do you guys 
have any suggestion?

Sent from my iPhone
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Re: [OpenSIPS-Users] topology_hiding() not executing

[Vlad Paiu]

Hello,

You must call topology_hiding() before t_relay() - please try that.
Also, make sure to change your sequential request handling from

 if (loose_route())

to

 if (loose_route() || match_dialog())


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 26.08.2014 06:48, Satish Patel wrote:
I have put topology_hiding() function at following place in script but 
its not hiding VIA header following is my senerio


[UA][Opensips]---[Asterisk/SIP gateway]

I want to hind my UA IP address so Asterisk doesn't see them, 
currently my asterisk can see what IP address UA coming from, where 
should i put them generally



if (is_method(INVITE)) {
...
...
if  ( uri=~^sip:[0-9]*@.*) {
uac_replace_from(sip:4545@65.111.170.127 
mailto:sip%3A4545@65.111.170.127);

t_on_failure(3);
resetflag(7);
t_relay( udp:65.111.170.127:5065 http://65.111.170.127:5065 );
 topology_hiding();
exit;
};



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Re: [OpenSIPS-Users] URGENT! uac_auth PSTN gateway authentication issue

[Vlad Paiu]

Hello,

What OpenSIPS version are you currently using ?
I've just committed a fix that implements a preliminary version of this, 
see commits :


https://github.com/OpenSIPS/opensips/commit/3316a2a518a2ac27401408369e4bd3adc70b4e48
and
https://github.com/OpenSIPS/opensips/commit/7989c4fcf1825afccb8102b65d94d66105dbdf33

Please apply them to your sources and let me know how it oges
Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 25.08.2014 13:31, Satish Patel wrote:
Great!! I can see light in tunnel now because last 1 week I tried 
everything and now I was planing to go for B2B but I guess as you said 
you guys working on so I'm holding my breath.


This is must needed solution because SIP service provide most of time 
provide password to make outbound trunk call.


Sent from my iPhone

On Aug 24, 2014, at 11:13 PM, Bogdan-Andrei Iancu bog...@opensips.org 
mailto:bog...@opensips.org wrote:



Hi Satish,

It is an known issue that OpenSIPS does not increases the cseq number 
when performing UAC auth against another party. Asterisk does not 
like that and consider the new branch INVITE with credentials a 
simple retransmission (even if it has a different VIA-branch :P) and 
discards them - this is why you get that timeout from asterisk.


We have ongoing work (hopefully to be ready in 1-2 weeks) for 
increasing the cseq number is a sip-wise manner. Just keep an eye on 
the mailing list.


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 25.08.2014 04:47, Satish Patel wrote:
I am seeing following and all transaction has CSeq: 2 INVITE, I have 
notice one thing asterisk asking for 407 but opensips never send any 
challenge response


Opensips --- INVITE --- Asterisk
Asterisk - 407 -- Opensips (Proxy-Authenticate: Digest 
algorithm=MD5, realm=asterisk, nonce=3a710e79.)

Opensips  ACK - Asterisk

Here opensips challenging SIP client  and saying giving try to 
asterisk and then following


Opensips  INVITE --- Asterisk
Opensips  INVITE  Asterisk
Opensips  INVITE  Asterisk

After 3 tries opensips send SIP client 408 Request timeout..


On Sun, Aug 24, 2014 at 4:26 PM, Stefano Pisani 
stefano.pis...@omnianet.it mailto:stefano.pis...@omnianet.it wrote:


Check if the cseq was incremented by one in the second try.
Use ngrep.



Il 24/08/2014 22.24, Satish Patel ha scritto:


Hi,

my Opensips (UAC) registered to PSTN gateway and now i am
trying to call using my SIPphone which is register to opensip
but no success. I am getting 407 Proxy authentication issue.. 
I am using following method but it didn't work. I need solution

badly..

PSTN gateway sending 407 Proxy auth and then my Opensip sending
407 proxy auth to SIP phone.

Does anyone has any working example or some kind of document? I
haven;t see any single doc anywhere in Internet about uac_auth
issue



modparam(uac,credential,username:domain:password)

route {

t_on_failure(2);
t_relay( udp:ip_addr:5060 );
...
}

failure_route[2] {
  uac_auth();
  t_relay(udp:ip_addr:5060);
}


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Re: [OpenSIPS-Users] URGENT! uac_auth PSTN gateway authentication issue

[Vlad Paiu]

Hello,

Ok then, the patches should apply just fine -
https://github.com/OpenSIPS/opensips/commit/3316a2a518a2ac27401408369e4bd3adc70b4e48

was already backported, so just apply
https://github.com/OpenSIPS/opensips/commit/7989c4fcf1825afccb8102b65d94d66105dbdf33

and test it out.

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 25.08.2014 13:36, Satish Patel wrote:

I'm using 1.11 last week installed.

Sent from my iPhone

On Aug 25, 2014, at 6:34 AM, Vlad Paiu vladp...@opensips.org 
mailto:vladp...@opensips.org wrote:



Hello,

What OpenSIPS version are you currently using ?
I've just committed a fix that implements a preliminary version of 
this, see commits :


https://github.com/OpenSIPS/opensips/commit/3316a2a518a2ac27401408369e4bd3adc70b4e48
and
https://github.com/OpenSIPS/opensips/commit/7989c4fcf1825afccb8102b65d94d66105dbdf33

Please apply them to your sources and let me know how it oges
Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  
On 25.08.2014 13:31, Satish Patel wrote:
Great!! I can see light in tunnel now because last 1 week I tried 
everything and now I was planing to go for B2B but I guess as you 
said you guys working on so I'm holding my breath.


This is must needed solution because SIP service provide most of 
time provide password to make outbound trunk call.


Sent from my iPhone

On Aug 24, 2014, at 11:13 PM, Bogdan-Andrei Iancu 
bog...@opensips.org mailto:bog...@opensips.org wrote:



Hi Satish,

It is an known issue that OpenSIPS does not increases the cseq 
number when performing UAC auth against another party. Asterisk 
does not like that and consider the new branch INVITE with 
credentials a simple retransmission (even if it has a different 
VIA-branch :P) and discards them - this is why you get that timeout 
from asterisk.


We have ongoing work (hopefully to be ready in 1-2 weeks) for 
increasing the cseq number is a sip-wise manner. Just keep an eye 
on the mailing list.


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 25.08.2014 04:47, Satish Patel wrote:
I am seeing following and all transaction has CSeq: 2 INVITE, I 
have notice one thing asterisk asking for 407 but opensips never 
send any challenge response


Opensips --- INVITE --- Asterisk
Asterisk - 407 -- Opensips   (Proxy-Authenticate: Digest 
algorithm=MD5, realm=asterisk, nonce=3a710e79.)

Opensips  ACK - Asterisk

Here opensips challenging SIP client  and saying giving try to 
asterisk and then following


Opensips  INVITE --- Asterisk
Opensips  INVITE  Asterisk
Opensips  INVITE  Asterisk

After 3 tries opensips send SIP client 408 Request timeout..


On Sun, Aug 24, 2014 at 4:26 PM, Stefano Pisani 
stefano.pis...@omnianet.it mailto:stefano.pis...@omnianet.it 
wrote:


Check if the cseq was incremented by one in the second try.
Use ngrep.



Il 24/08/2014 22.24, Satish Patel ha scritto:


Hi,

my Opensips (UAC) registered to PSTN gateway and now i am
trying to call using my SIPphone which is register to opensip
but no success. I am getting 407 Proxy authentication
issue..  I am using following method but it didn't work. I
need solution badly..

PSTN gateway sending 407 Proxy auth and then my Opensip
sending 407 proxy auth to SIP phone.

Does anyone has any working example or some kind of document?
I haven;t see any single doc anywhere in Internet about
uac_auth  issue



modparam(uac,credential,username:domain:password)

route {

t_on_failure(2);
t_relay( udp:ip_addr:5060 );
...
}

failure_route[2] {
  uac_auth();
  t_relay(udp:ip_addr:5060);
}


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Re: [OpenSIPS-Users] URGENT! uac_auth PSTN gateway authentication issue

[Vlad Paiu]

Hello,

Sorry about that - failed to commit auth.h , also apply

https://github.com/OpenSIPS/opensips/commit/fc2ed8ace7040734f5c1b11fe235478014817de5

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 25.08.2014 15:26, Satish Patel wrote:
I downloaded latest repo from git whatever you mentioned. At compile 
time I got following error when it was compiling uac.c


Fatal error: auth.h file and directory not found.

Does it require any dependencies?

Sent from my iPhone

On Aug 25, 2014, at 8:10 AM, Vlad Paiu vladp...@opensips.org 
mailto:vladp...@opensips.org wrote:



Hello,

Ok then, the patches should apply just fine -
https://github.com/OpenSIPS/opensips/commit/3316a2a518a2ac27401408369e4bd3adc70b4e48

was already backported, so just apply
https://github.com/OpenSIPS/opensips/commit/7989c4fcf1825afccb8102b65d94d66105dbdf33

and test it out.

Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  
On 25.08.2014 13:36, Satish Patel wrote:

I'm using 1.11 last week installed.

Sent from my iPhone

On Aug 25, 2014, at 6:34 AM, Vlad Paiu vladp...@opensips.org 
mailto:vladp...@opensips.org wrote:



Hello,

What OpenSIPS version are you currently using ?
I've just committed a fix that implements a preliminary version of 
this, see commits :


https://github.com/OpenSIPS/opensips/commit/3316a2a518a2ac27401408369e4bd3adc70b4e48
and
https://github.com/OpenSIPS/opensips/commit/7989c4fcf1825afccb8102b65d94d66105dbdf33

Please apply them to your sources and let me know how it oges
Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  
On 25.08.2014 13:31, Satish Patel wrote:
Great!! I can see light in tunnel now because last 1 week I tried 
everything and now I was planing to go for B2B but I guess as you 
said you guys working on so I'm holding my breath.


This is must needed solution because SIP service provide most of 
time provide password to make outbound trunk call.


Sent from my iPhone

On Aug 24, 2014, at 11:13 PM, Bogdan-Andrei Iancu 
bog...@opensips.org mailto:bog...@opensips.org wrote:



Hi Satish,

It is an known issue that OpenSIPS does not increases the cseq 
number when performing UAC auth against another party. Asterisk 
does not like that and consider the new branch INVITE with 
credentials a simple retransmission (even if it has a different 
VIA-branch :P) and discards them - this is why you get that 
timeout from asterisk.


We have ongoing work (hopefully to be ready in 1-2 weeks) for 
increasing the cseq number is a sip-wise manner. Just keep an eye 
on the mailing list.


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 25.08.2014 04:47, Satish Patel wrote:
I am seeing following and all transaction has CSeq: 2 INVITE, I 
have notice one thing asterisk asking for 407 but opensips never 
send any challenge response


Opensips --- INVITE --- Asterisk
Asterisk - 407 -- Opensips (Proxy-Authenticate: Digest 
algorithm=MD5, realm=asterisk, nonce=3a710e79.)

Opensips  ACK - Asterisk

Here opensips challenging SIP client and saying giving try to 
asterisk and then following


Opensips  INVITE --- Asterisk
Opensips  INVITE  Asterisk
Opensips  INVITE  Asterisk

After 3 tries opensips send SIP client 408 Request timeout..


On Sun, Aug 24, 2014 at 4:26 PM, Stefano Pisani 
stefano.pis...@omnianet.it mailto:stefano.pis...@omnianet.it 
wrote:


Check if the cseq was incremented by one in the second try.
Use ngrep.



Il 24/08/2014 22.24, Satish Patel ha scritto:


Hi,

my Opensips (UAC) registered to PSTN gateway and now i am
trying to call using my SIPphone which is register to
opensip but no success. I am getting 407 Proxy
authentication issue..  I am using following method but it
didn't work. I need solution badly..

PSTN gateway sending 407 Proxy auth and then my Opensip
sending 407 proxy auth to SIP phone.

Does anyone has any working example or some kind of
document? I haven;t see any single doc anywhere in Internet
about uac_auth  issue



modparam(uac,credential,username:domain:password)

route {

t_on_failure(2);
t_relay( udp:ip_addr:5060 );
...
}

failure_route[2] {
  uac_auth();
  t_relay(udp:ip_addr:5060);
}


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Re: [OpenSIPS-Users] tcp_no_new_conn_bflag usage

[Vlad Paiu]

Hello,

The tcp_no_new_conn_bflag only instructs OpenSIPS to not attempt opening 
a new TCP connection, and only re-use existing connections for relaying 
the request ( if no connection available, a negative reply will 
automatically be generated ) - it is currently not linked to the status 
of an AOR ( registered or not ).



Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 20.08.2014 14:04, Alex Massover wrote:


Hi,

I have a question regarding tcp_no_new_conn_bflag usage.

When tcp_no_new_conn_bflag removes AOR, upon TCP disconnect or as a 
result of t_relay? I'm trying to implement the scenario where AORs are 
removed upon TCP disconnect without waiting for the next INVITE to 
this client.


--

*Alex Massover | Telefónica Digital*

Architect

M +972-54-2279512

a...@jajah.com mailto:a...@jajah.com



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Re: [OpenSIPS-Users] OpenSIPS Summit, almost there

[Vlad Paiu]

Hello,

There will be no live streaming for the event, but all the presentations 
will be recorded and uploaded after the event ( there will be a followup 
with the download / streaming links ).


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 01.08.2014 15:11, Anshuman S Rawat wrote:

Is a live or recorded webcast planned for those of us who cannot be there?

Regards,
Anshuman


-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Thursday, July 31, 2014 11:53 PM
To: users@lists.opensips.org; developensips; n...@lists.opensips.org;
busin...@lists.opensips.org
Subject: [OpenSIPS-Users] OpenSIPS Summit, almost there

There are less then 4 days to the OpenSIPS Summit in Chicago !

We are proud and excited to have such a valuable team of speakers:
   http://www.opensips.org/Community/Summit-2014Chicago-Schedule


To be honest, I'm personally looking forward to be there and listen on
those presentations, to see what other mad things people did with
OpenSIPS :).

For our participants, see you in Chicago next Monday !!

Regards,




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Re: [OpenSIPS-Users] Proposed change to cachedb

[Vlad Paiu]

Hello Brett,

What OpenSIPS version are you using ?

In the latest GIT for all maintained branches, the code was fixed to 
propagate the exact return from the module implementing the module 
connectivity, except for the 0 return code, which is converted to 1 ( 
success ) - done in order not to break the script execution.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 23.07.2014 17:18, Brett Nemeroff wrote:

Hey All,

Using memcached..

So I've noticed that while performing a cache_fetch I can't tell the 
difference between a cache failure and a NOT_FOUND. It seems the 
problem is actually in cachedb.c because we do this at the end 
of cachedb_fetch:


return cde-cdb_func.get(con,attr,val)0?-1:1;

So I read this to basically say to return -1 on any failure regardless 
of the failure code. This is probably because of the generalized 
nature of the cache interface and since each cache backend has it's 
own return codes. I get that, but that being said, I can't tell what 
the failure is and respond properly.


So I changed that one return line to look more like this:

res = cde-cdb_func.get(con,attr,val);
if (res  0) {
  return res;
} else {
  return 1;
}

Is this acceptable? Will I run into problems I'm not thinking about? 
The only real problem I can see is that a specific error number on one 
cache backend might mean something different on another. Obviously the 
only way to really fix this would be to have each cache backend to 
match up it's own backend's reply codes to a set of generic opensips 
cache engine reply codes separately enumerated.


Thoughts?

Thanks,
Brett


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Re: [OpenSIPS-Users] json module issues

[Vlad Paiu]

Hello,

There have been some recent changes to the JSON module in terms of 
parsing the error codes ( some old code got deprecated on versions  0.9 
) and we had to do some compile-time detection of the version of libjson 
used ( since the JSON library only exports a version macro starting with 
0.10 ), so this might be the source of your issues.


What is the version of the libjson do you currently have installed on 
your system ? So far tested with 0.9, 0.10 and 0.12 and does not seem to 
replicate.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 08.07.2014 19:57, Kurtis Heimerl wrote:

I bumped up the debug, seeing this in the logs:

Jul  8 16:51:47 SERVERNAME /usr/sbin/opensips[30188]: 
ERROR:json:pv_set_json: Error parsing json: success
Jul  8 16:51:47 SERVERNAME /usr/sbin/opensips[30188]: 
ERROR:core:do_assign: setting PV failed
Jul  8 16:51:47 SERVERNAME /usr/sbin/opensips[30188]: 
ERROR:core:do_assign: error at line: 151


Looks like it thinks the json isn't parsing right, but the config is 
still dead simple...


$json(k) := [1,2];
xlog(L_ERR,Kurtis2 $json(k));



On Tue, Jul 8, 2014 at 9:48 AM, Kurtis Heimerl 
kheim...@cs.berkeley.edu mailto:kheim...@cs.berkeley.edu wrote:


Huh. I tried the default json code too
(http://lists.opensips.org/pipermail/devel/2009-September/004177.html)


   $json(obj1) := {};   # initialize an empty JSON object
   $json(obj1/key) = value; #replace or insert the (key,value)
  #pair into the json object;
   xlog($json(obj1));   # print the serialized version of
the object

Which still doesn't work (printing null), so something deeply
screwed up.

As far as versions, I'm running 1.11.2-notls (x86_64/linux),
according to opensips. This is out of the default opensips debian
repo, as is the json module itself.

Anyone have any idea why the json module would be failing like
this? It doesn't seem like an issue with my config.


On Mon, Jul 7, 2014 at 11:28 PM, Ra(zvan Crainea
raz...@opensips.org mailto:raz...@opensips.org wrote:

Hi, Kurtis!

I've just run your test and the output seems ok:
Kurtis [ 1, 2 ]
Are you sure there is no other function between those two
lines that could delete the json? Also, what version of
OpenSIPS are you using?

Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com


On 07/08/2014 03:01 AM, Kurtis Heimerl wrote:

Hey All,

I seem to be having some very simple issues with the json
module. The
following code in my config:

$json(k) := [1,2];
xlog(L_ERR,Kurtis $json(k));

Is producing the following output:

Jul  7 23:59:00 NAME /usr/sbin/opensips[6379]: Kurtis2 null

This looks to be as simple as I can get a json command,
but I can't
figure out why it's null. Changing the  := to a = and
the [1,2] to
a 4 works, but that's just a normal variable. Anyone
know what's up?

Thanks!


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Re: [OpenSIPS-Users] Failed to generate script

[Vlad Paiu]

Hello,

It seems that menuconfig fails when running m4 to build the config file, 
so it seems that there might be something wrong with the deb or how it 
is installed.
Can you please let me know where the 'menuconfig_templates' folder is 
located on your system ? You can you use


 find / -name menuconfig_templates

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 30.06.2014 18:36, Ludmann, Dieter wrote:

Hello,

I use Version 1.10.1 (i386).
I did not compile from sources, but used the package on 
http://apt.opensips.org/pool/main/o/opensips/
Please find attached the curses.out file.
Thanks for support.

Mit freundlichen Grüßen,
Kind regards,

Dieter Ludmann

CTDI GmbH
Headquarters Europe
Test Design
Stephanstr. 4-8
76316 Malsch
Germany
Fon: +49 7246 80-3678
Mobile: +49 160 97851055
Fax: +49 7246 80-3259
Email:  dieter.ludm...@ctdi.eu
http://www.ctdi.eu/
_

CTDI GmbH, Supervisory Board: Dr. Bruno Jacobfeuerborn (Chairman)
Management Board: Dieter Hollenbach, Monika Rüth, David Burt
Commercial Registration: Amtsgericht Mannheim HRB 362406
Registered Office: Malsch, VAT ID # DE813160881
_

-Ursprüngliche Nachricht-
Von: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] 
Im Auftrag von Vlad Paiu
Gesendet: Montag, 30. Juni 2014 12:24
An: users@lists.opensips.org
Cc: gr...@ucs.cam.ac.uk
Betreff: Re: [OpenSIPS-Users] Failed to generate script

Hello,

What OpenSIPS version are you currently using ?
Can you please pastebin the file 'curses.out' from the OpenSIPS main sources 
folder ?

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 27.06.2014 17:17, TDDiLu wrote:

Gordon Ross wrote

I'm trying to use osipsconfig to generate an OpenSIPs script. But
when I do Generate Residential Script,I get the error:

Config generated :
/etc/opensips/opensips_residential_2013-12-24_13:20:33.cfg =  FAILED.
Press any key to continue

If I exit, I can see the program has created a file - but it's empty.

I have the same issue, with all three scripts (residential, trunking
and load-balancer).
Any solution or hint available meanwhile?
The directory /etc/opensips exists, and every time I try Generate it
produces a new file with actual timestamp, but this file is empty (0 bytes).

Any help appreciated.
Dieter



--
View this message in context:
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te-script-tp7589129p7592105.html Sent from the OpenSIPS - Users
mailing list archive at Nabble.com.

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Re: [OpenSIPS-Users] OpenSIPS 1.11 : Sending OPTIONS during ongoing session

[Vlad Paiu]

Hello,

How are you creating the dialog within your script ?
The within dialog SIP Option Pings are only sent if you create the 
dialog using the 'Pp' flags, so please remove those if you're using them.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 30.06.2014 19:29, Gary Nyquist wrote:


Hi,

I am trying to migrate to OpenSIPS 1.11.

Observing that, it is sending out OPTIONS packets, when the session is 
going on (after the ACK); using the same Call-ID as that of the 
ongoing Call.


Is there a way to stop these OPTIONS packets to be originated (during 
ongoing sessions) from the OpenSIPS?


Thanks in adavance.

--Gary



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Re: [OpenSIPS-Users] TCP Errors

[Vlad Paiu]

Hello,

First of all, are you using the async_tcp option in your OpenSIPS script ?
Also, what architecture / OS are you running on ?
What steps are you taking in reproducing this ?

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 27.06.2014 19:59, Gary Nyquist wrote:

Thanks Bogdan for looking into it.
Here is the version:
opensips -V
version: opensips 1.11.1-tls (x86_64/linux)
flags: STATS: On, USE_TCP, USE_TLS, DISABLE_NAGLE, SHM_MEM, SHM_MMAP, 
PKG_MALLOC, F_MALLOC, USE_SHM_MEM, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
MAX_URI_SIZE 1024, BUF_SIZE 65535

poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
git revision: dbc8653
main.c compiled on 16:17:43 Jun 26 2014 with gcc 4.4.7
Yes, this bug is reproducable.
When it appears, the log file instantly gets filled with hundreds of 
repeating lines like this:
Jun 26 20:30:31 ip-10-0-0-30 /usr/sbin/opensips[15229]: 
DBG:core:handle_tcpconn_ev: data available on 0x7f40bcec7398 6
Jun 26 20:30:31 ip-10-0-0-30 /usr/sbin/opensips[15229]: 
DBG:core:io_watch_del: io_watch_del op on index -1 6 (0x7dff20, 6, -1, 
0x0,0x1) fd_no=110 called
Jun 26 20:30:31 ip-10-0-0-30 /usr/sbin/opensips[15229]: 
ERROR:core:io_watch_del: BUG - trying to del fd 6 with flags 2 1
Jun 26 20:30:31 ip-10-0-0-30 /usr/sbin/opensips[15229]: 
DBG:core:handle_tcpconn_ev: data available on 0x7f40bcec7398 6
Jun 26 20:30:31 ip-10-0-0-30 /usr/sbin/opensips[15229]: 
DBG:core:io_watch_del: io_watch_del op on index -1 6 (0x7dff20, 6, -1, 
0x0,0x1) fd_no=110 called
Jun 26 20:30:31 ip-10-0-0-30 /usr/sbin/opensips[15229]: 
ERROR:core:io_watch_del: BUG - trying to del fd 6 with flags 2 1

Thanks again for your help.
BR
-Gary
*Sent:* Thursday, June 26, 2014 at 10:56 AM
*From:* Bogdan-Andrei Iancu bog...@opensips.org
*To:* Gary Nyquist g...@gmx.us
*Cc:* OpenSIPS users mailling list users@lists.opensips.org
*Subject:* Re: [OpenSIPS-Users] TCP Errors
Hi,

That bug log actually says OpenSIPS tries to remove a connection 
marked as READ from a list for WRITEs :)..


I will look into that. What exact version do you use (opensips -V) ? 
also, can you reproduce this  bug ?


Thanks and regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 25.06.2014 20:27, Gary Nyquist wrote:

Thanks Bogdan for the detailed reply.
I implemented your advice.
Those error messages are now gone.
But seeing a new error now; not sure if it is related to that...
ERROR:core:io_watch_del: BUG - trying to del fd 36 with flags 2 1
Any advice?
BR
-Gary
*Sent:* Wednesday, June 25, 2014 at 5:54 AM
*From:* Bogdan-Andrei Iancu bog...@opensips.org
*To:* OpenSIPS users mailling list users@lists.opensips.org,
g...@gmx.us
*Subject:* Re: [OpenSIPS-Users] TCP Errors
Hi,

Those messages say that OpenSIPS tried to open a TCP connection to
a party which does not respond - there was a timeout for connect
in 10 seconds. Because of this blocking in connects, there were
not more opensips workers available to handle other traffic.

So what you need to do is :
- minimize the impact of the blocking connect - see my
previous email on reducing the connect timeout
- you may configure OpenSIPS not to open new TCP connect (but
to reuse the existing ones, open by clients). See
tcp_no_new_conn_bflag
http://www.opensips.org/Documentation/Script-CoreParameters-1-11#toc96
- try to understand the SIP patterns where such TCP connect
fails so you can avoid them at script level.

Regards,

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

On 24.06.2014 21:34, Gary Nyquist wrote:

Hi,

The following lines are repeating in the log.

ERROR:core:tcp_blocking_connect: timeout 10 s elapsed from 10 s
ERROR:core:tcpconn_connect: tcp_blocking_connect failed
ERROR:core:tcp_send: connect failed
ERROR:tm:msg_send: tcp_send failed

INFO:core:send2child: no free tcp receiver, connection passed
to the least busy one

Any guess, what could be the reason?

Best

--Gary

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Re: [OpenSIPS-Users] Failed to generate script

[Vlad Paiu]

Hello,

What OpenSIPS version are you currently using ?
Can you please pastebin the file 'curses.out' from the OpenSIPS main 
sources folder ?


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 27.06.2014 17:17, TDDiLu wrote:

Gordon Ross wrote

I’m trying to use osipsconfig to generate an OpenSIPs script. But when I
do “Generate Residential Script”,I get the error:

Config generated :
/etc/opensips/opensips_residential_2013-12-24_13:20:33.cfg =  FAILED.
Press any key to continue

If I exit, I can see the program has created a file - but it’s empty.

I have the same issue, with all three scripts (residential, trunking and
load-balancer).
Any solution or hint available meanwhile?
The directory /etc/opensips exists, and every time I try Generate it
produces a new file with actual timestamp, but this file is empty (0 bytes).

Any help appreciated.
Dieter



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Re: [OpenSIPS-Users] Using external MySQL server

[Vlad Paiu]

Hello,

To make things simpler, you can also do

modparam(module1|module2|module3,db_url,YOUR_URL_HERE)

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 26.06.2014 11:13, Alain BIEUZENT wrote:


Hi gary,

You can use m4 « compilator » for this

Have look to : http://www.opensips.org/Documentation/Tools

Regards

*De :*users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] *De la part de* Gary Patton

*Envoyé :* jeudi 26 juin 2014 01:29
*À :* users@lists.opensips.org
*Objet :* [OpenSIPS-Users] Using external MySQL server

Hi again!  Another newbie question.

If I want a module to use an external MySQL db server, I simply modify 
the modparam for each module in my residential config file by adding 
the external server's IP address to the DB_URL parameter for each module.


But isn't there a simpler way?  I guess what I'm asking is whether I 
can enter the external db server's IP address on a single line in a 
config file somewhere in OpenSIPS and OpenSIPS will use the external 
MySQL server for all the modules included in my residential config file?


Hope this question isn't too silly.

Regards

Gary



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[OpenSIPS-Users] OpenSIPS 1.11 Stable Release Scheduled

[Vlad Paiu]

Hello,

OpenSIPS version 1.11 is scheduled to be released as stable on 
Wednesday, 7th of May.


If you still have unresolved or unreported issues, please visit the 
officlal tracker at 
https://github.com/OpenSIPS/opensips/issues?page=1state=open


Many thanks to all the people who got involved in testing and fixing 
work on 1.11We are almost there !


Best Regards,

--
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com


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Re: [OpenSIPS-Users] Miliseconds precision for accounting module

[Vlad Paiu]

Hello,

Which OpenSIPS version are you using ?
You could use get_timestamp [1] from the Core to get the current second 
and microsecond,

and set the two variables at INVITE time, and set them as db_extra [2] .

Then, at BYE time call again the get_timestamp function, store them in 
some AVPs and set those AVPs in [3]. This way you should get both the 
INVITE and BYE timestamps with microseconds precision in the CDR record.


[1] http://www.opensips.org/Documentation/Script-CoreFunctions-1-10#toc18
[2] http://www.opensips.org/html/docs/modules/1.10.x/acc.html#id295028
[3] http://www.opensips.org/html/docs/modules/1.10.x/acc.html#id295056

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 12.04.2014 23:44, Maciej Bylica wrote:

Hello Ryan,

I am using dialog accounting, so each row is fully qualified cdr 
record, not only single transaction of a call.
Couldn't i just use two extra db variables which will gather the $time 
inside INVITE {} and BYE {}?


Thanks,
Mac


2014-04-12 6:39 GMT+02:00 Ryan Mitchell r...@tcl.net 
mailto:r...@tcl.net:


Hello Mac,

Each row in the acc table is for a transaction.  To make a proper
CDR out of the data, you have to combine rows to find the start
and end of the call.  That can be harder than it sounds,
especially with forking (parallel, or the more common case of
serial forking when you are LCR routing or simply sending calls to
alt destinations after a timeout).  I wrote scripts that implement
a simple dialog state machine to make sense of all the distinct
legs of a call, though there should be an easier way with the
auto-cdr / multi call-legs accounting feature of the acc module
(anyone comment on this please?).

The time field in the acc table will be the timestamp of the
response for the given transaction.  If you assign an extra field
for another timestamp, it will depend on where you assign that var
in your script.  In my case I assign it in the main routing
section so the timestamp indicates the start of the transaction.

best regards,
Ryan



On Fri, Apr 11, 2014 at 10:06 AM, Maciej Bylica mb...@gazeta.pl
mailto:mb...@gazeta.pl wrote:

Ryan,

One more question.
Currently i have some db extra attrs setup. My acc table looks
like following:


++--+--+-+-++

| Field  | Type | Null | Key | Default |
Extra  |


++--+--+-+-++

| id | int(10) unsigned | NO   | PRI | NULL|
auto_increment |

| method | char(16) | NO   | | | 
  |


| from_tag   | char(64) | NO   | | | 
  |


| to_tag | char(64) | NO   | | | 
  |


| callid | char(64) | NO   | MUL | | 
  |


| sip_code   | char(3)  | NO   | | | 
  |


| sip_reason | char(32) | NO   | | | 
  |


| time   | datetime | NO   | | NULL| 
  |


| duration   | int(11) unsigned | NO   | | 0   | 
  |


| setuptime  | int(11) unsigned | NO   | | 0   | 
  |


| SourceAddr | char(30) | NO   | | NULL| 
  |


| DestAddr   | char(30) | NO   | | NULL| 
  |


| Anum   | char(30) | NO   | | NULL| 
  |


| Bnum_rU| char(30) | NO   | | NULL| 
  |


| Bnum_tU| char(30) | NO   | | NULL| 
  |


| created| datetime | YES  | | NULL| 
  |



++--+--+-+-++


modparam(acc, db_extra, SourceAddr=$si; DestAddr=$rd;
Anum=$fU; Bnum_rU=$rU; Bnum_tU=$tU)


Now using additional data like $time will give me the exact
moment the call is ended, nothing more, am i right?

To have detailed call duration i need to know exact answer and
disconnect timestamps.


Btw: i am using OpenSIPS (1.9.1-notls (x86_64/linux))


Thanks,

Mac



2014-04-10 22:03 GMT+02:00 Ryan Mitchell r...@tcl.net
mailto:r...@tcl.net:

Using db_extra to stuff custom data into your acc table,
use the $time var with a format such as %s.%N or similar.

Or, as you suggested, do it on the database level with a
trigger or auto-update column.



On Thu, Apr 10, 2014 at 10:01 AM, Maciej Bylica
mb...@gazeta.pl

Re: [OpenSIPS-Users] OpenSIPS fails on urn:service:sos RURI

[Vlad Paiu]

Hello Anders,

I've just committed a fix for this on OpenSIPS trunk, see 
https://github.com/OpenSIPS/opensips/commit/edef0fd73cc03fbb9cd31191529cd9b988bb6394


The patch accepts urn:service type URIs and also allows the OpenSIPS 
script writer to extract the actual service that the request targets. 
Here a short script example that I've used to test


if (is_method(INVITE)  $ru =~ ^urn:service) {
xlog(Received INVITE going to service $(ru{uri.host}) 
\n);

record_route();
t_relay(udp:192.168.2.134:5070);
exit;
}

Patch was only comitted on trunk, so please apply the patch to your 
sources and re-test.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 01.04.2014 21:48, Bogdan-Andrei Iancu wrote:

Hello Anders,

I was not aware of this RFC (to be honest its approach seems odd to me 
as it is over-complicating such a simple problem - generic naming for 
services) . OpenSIPS is not supporting URN schema. I haven;t read the 
RFC in all details to understand if a SIP proxy is required to 
actually support and parse such schema or it should simply act as a 
blind forwarder for it (based on OBP info). The RFC keeps mentioning:


quote

Since service URNs are not routable, a SIP proxy or user agent has to
   translate the service URN into a routable URI for a location-
   appropriate service provider, such as a SIP URL

/quote

I will do more digging into this.

Best regards,

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

On 19.03.2014 03:08, Anders Kristensen wrote:

I'm trying to route calls to emergency services through an OpenSIPS
proxy. It seems that the tm module fails because it (or OpenSIPS
itself) is unable to parse urn:service:sos RURI (defined in RFC 5031).

Mar 18 17:27:31 [8807] ERROR:core:parse_uri: bad uri,  state 0 parsed:
urn: (4) / urn:service:sos (15)
Mar 18 17:27:31 [8807] ERROR:core:parse_sip_msg_uri: bad uri 
urn:service:sos

Mar 18 17:27:31 [8807] DBG:core:set_err_info: ec: 1, el: 3, ei: 'error
parsing r-uri'
Mar 18 17:27:31 [8807] ERROR:tm:new_t: uri invalid
Mar 18 17:27:31 [8807] ERROR:tm:t_newtran: new_t failed

I'm routing the call using t_relay(udp:127.0.0.1:5070) so there
should be no real need to parse the RURI. (What I'd really like to do
instead is have the client insert a preloaded route in the INVITE and
have OpenSIPS route based on that and leave the RURI untouched.)

A quick search shows that this exact issue came up back in 2008:
http://lists.sip-router.org/pipermail/users/2008-March/016680.html
Also, RFC 5031 was published in 2008 so I'm surprised if this is still
an issue. Can you experts think of *any* way to get OpenSIPS to send
an INVITE with RURI urn:service:sos? Preferably one that doesn't
involve writing a new module just for this purpose.

I tried this with the latest OpenSIPS 1.10.

Thanks,
Anders

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[OpenSIPS-Users] Supported OpenSIPS Versions update

[Vlad Paiu]

Hello,

A short reminder about the supported version of OpenSIPS.

According to the OpenSIPS version management [1], this is the lifetime 
of the currently supported versions:


* 1.8 will be supported until the end of 2014 (LTS)
* 1.9 becomes unmaintained (STS)
* 1.10 will be supported until the end of 2014 (STS)
* 1.11 will be supported for two years after it's stable release (LTS)

The full overview of the currently available versions is also listed at [2]

[1] http://opensips.org/Development/Development#toc2
[2] http://www.opensips.org/About/AvailableVersions

Best Regards

--
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com


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Re: [OpenSIPS-Users] Timer shift on dr-pinger

[Vlad Paiu]

Hello,

First of all, what kind of probing mode do you have defined ? ( [1] )

Possible reasons are that either you are talking to your gateways via 
TCP, and OpenSIPS is losing some time establishing new connections or 
actually sending out SIP messages via TCP. You can set some thresholds 
where OpenSIPS will warn you in case TCP gets slow, see [2].


Another possibility would be that you have your gateways defined as 
FQDNs, and OpenSIPS is wasting lots of time doing DNS queries.You can 
set some thresholds where OpenSIPS will warn you in case DNS gets slow, 
see [3].


[1] http://www.opensips.org/html/docs/modules/1.11.x/drouting.html#id250145
[2] http://www.opensips.org/Documentation/Script-CoreParameters-1-11#toc87
[3] http://www.opensips.org/Documentation/Script-CoreParameters-1-11#toc53

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 02.03.2014 17:38, Remco wrote:

Hi all,

I'm experiencing timer shifts running OpenSIPS 1.9.
The following message is in the logfile:

Mar  2 06:49:15 [hostname] /usr/sbin/opensips[18796]: 
CRITICAL:core:timer_ticker: timer handler dr-pinger lasted (1991 
us) for more than timer tick (100 us) - potential timer shifting


It seems that dr-pinger is experiencing some sort of delay which 
shifts the internal timer in the opensips process. Although the delays 
vary, they are not significantly increasing. After some time issues 
with processing traffic start to arise. Eventually some gateways are 
disabled by Drouting, although that takes a couple of hours.


To me it seems that we have some sort of networking issue, what i 
cannot understand why drouting is causing the internal timers to 
shift. As the main function of the dr-pinger is to detect unreachable 
gateways and disable them. Perhaps someone has an idea what might be 
causing this behavior?


Thanks,
Remco.


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Re: [OpenSIPS-Users] Adding Proxy-Authorization header

[Vlad Paiu]

Hello,

The registrant module is to be used only for generating REGISTER 
requests ( with auth included ).
For proxied calls, you need to use the uac and uac_auth modules ( [1] ) 
for adding the auth headers - call uac_auth() ( [2] ) function within 
failure route when receiving a challenge.


[1] http://www.opensips.org/html/docs/modules/1.11.x/uac_auth.html
[2] http://www.opensips.org/html/docs/modules/1.11.x/uac.html#id250288

Best Regards

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 24.02.2014 17:33, Stefano Pisani wrote:

You can use module UAC_AUTH

Il 24/02/2014 16.18, Diego Barberio ha scritto:

Hi all,

I have opensips registered to an IP-PBX using registrant module and I 
want to make an outbound call to that PBX through the proxy.


I'm sending and INVITE from my application to the proxy with a From 
that is actually registered by the proxy, however OpenSIPs is not 
adding the Proxy-Authorization header so the INVITE is rejected with 
a 401 Unauthorized and that response is forwarded to my application.


I just want opensips to add the Proxy-Authorization header so the 
call is not rejected by the IP-PBX. Is it possible to achieve this?


Thanks
Diego


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Re: [OpenSIPS-Users] fifo profile_get_values

[Vlad Paiu]

Hello,

Any updates on this ?

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 16.01.2014 18:13, Vlad Paiu wrote:

Hello Jeff,

Well then it seems like a bug - if you can, please provide more info 
on how you can replicate this and open a ticket on github. Thanks for 
the help in debugging this.


Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  
On 15.01.2014 20:47, Jeff Pyle wrote:

Hi Vlad,

That's encouraging!  I had tried a few months ago on 1.10.  It wasn't 
shared - in fact, very similar to your example.  In my simple case I 
created another profile without values and queried that when I needed 
a total across all values.



- Jeff


On Wed, Jan 15, 2014 at 1:19 PM, Vlad Paiu vladp...@opensips.org 
mailto:vladp...@opensips.org wrote:


Hello,

I've just tested this on the latest master, with a simple scenario

create_dialog();
set_dlg_profile(caller,$fU);

and it seems to be working for me :

scripts/opensipsctl fifo profile_get_values caller
value:: vlad count=1

Is that profile, by any chance, a shared dialog profile ?
Because, as the docs states (
http://www.opensips.org/html/docs/modules/1.11.x/dialog.html#id296552
) the function will currently not work with shared profiles.

if it's not the case of a shared profile, please let me know what
OpenSIPS version are you currently using. Also, please try to run
'opensipsctl fifo list_all_profiles' and see if you profile is
listed there ( assuming you're running an OpenSIPS version = 1.9
http://www.opensips.org/html/docs/modules/1.9.x/dialog.html#id296393
)

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  


On 15.01.2014 20:05, Jeff Pyle wrote:

Ok, good.

Bogdan, Rasvan, others on the dev team... is this a bug, or a
change in design?  Should we open a bug report?


- Jeff



On Wed, Jan 15, 2014 at 9:43 AM, Dragomir Haralambiev
goup2...@gmail.com mailto:goup2...@gmail.com wrote:

All worked fine.


2014/1/15 Jeff Pyle jp...@fidelityvoice.com
mailto:jp...@fidelityvoice.com

Since you are defining the profile with a value, your
query needs to contain a value as well.  This wasn't the
case in, say, 1.6.  This may be a bug in more current
versions.

Try defining the profile without values.  See if it
changes your result.


- Jeff


On Wed, Jan 15, 2014 at 9:20 AM, Dragomir Haralambiev
goup2...@gmail.com mailto:goup2...@gmail.com wrote:

Hello,

This is setings in opensips.cfg:
modparam(dialog, profiles_with_value, caller)

When run
opensipsctl fifo profile_get_values caller

I receive:
404 Profile not found

Where is my mistake?

Regards,
PlayMen



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Re: [OpenSIPS-Users] Error on make menuconfig

[Vlad Paiu]

Hello,

Seems the Solaris sed does not have the -i option ( which is used to 
edit some files in place to change the installation path ) - 
http://stackoverflow.com/questions/4121711/sed-i-option-is-not-working-on-solaris 
, seems that this is just a GNU sed feature.


Will try to do a work around for this.

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 21.01.2014 10:19, Nathaniel L Keeling III wrote:

Hello,

I currently have versions 5.9 rev=2011.11.21 of libncurses development 
installed. I also, just for experimentation, changed the Makefile for 
menuconfig to -lcurses and tried to compiled. This changed allowed 
menuconfig to compile and display the menu but when I tried to compile 
opensips this is the error that I get:


gcc -g -Wall 
-DMENUCONFIG_CFG_PATH=\/opt/opensips.1.10//opensips//menuconfig_templates/\ 
-DMENUCONFIG_GEN_PATH=\/opt/opensips.1.10//etc/opensips/\ 
-DMENUCONFIG_HAVE_SOURCES=0-c -o items.o items.c
gcc -g -Wall 
-DMENUCONFIG_CFG_PATH=\/opt/opensips.1.10//opensips//menuconfig_templates/\ 
-DMENUCONFIG_GEN_PATH=\/opt/opensips.1.10//etc/opensips/\ 
-DMENUCONFIG_HAVE_SOURCES=0-c -o commands.o commands.c
gcc -g -Wall 
-DMENUCONFIG_CFG_PATH=\/opt/opensips.1.10//opensips//menuconfig_templates/\ 
-DMENUCONFIG_GEN_PATH=\/opt/opensips.1.10//etc/opensips/\ 
-DMENUCONFIG_HAVE_SOURCES=0-c -o menus.o menus.c
gcc -g -Wall 
-DMENUCONFIG_CFG_PATH=\/opt/opensips.1.10//opensips//menuconfig_templates/\ 
-DMENUCONFIG_GEN_PATH=\/opt/opensips.1.10//etc/opensips/\ 
-DMENUCONFIG_HAVE_SOURCES=0-c -o parser.o parser.c
gcc -g -Wall 
-DMENUCONFIG_CFG_PATH=\/opt/opensips.1.10//opensips//menuconfig_templates/\ 
-DMENUCONFIG_GEN_PATH=\/opt/opensips.1.10//etc/opensips/\ 
-DMENUCONFIG_HAVE_SOURCES=0-c -o main.o main.c
gcc -o configure -g -Wall 
-DMENUCONFIG_CFG_PATH=\/opt/opensips.1.10//opensips//menuconfig_templates/\ 
-DMENUCONFIG_GEN_PATH=\/opt/opensips.1.10//etc/opensips/\ 
-DMENUCONFIG_HAVE_SOURCES=0  cfg.o curses.o items.o commands.o menus.o 
parser.o main.o -lcurses
make[2]: Leaving directory 
`/usr/local/src/opensips/opensips_1_10/menuconfig'

mkdir -p /opt/opensips.1.10//opensips//menuconfig_templates/
touch   menuconfig/configs/* 
/opt/opensips.1.10//opensips//menuconfig_templates/
ginstall -m 644 menuconfig/configs/* 
/opt/opensips.1.10//opensips//menuconfig_templates/

sed -i -e s#/usr/local/lib/opensips#lib64/opensips# \
/opt/opensips.1.10//opensips//menuconfig_templates/*
sed: illegal option -- i
make[1]: *** [opensipsmc] Error 2
make[1]: Leaving directory `/usr/local/src/opensips/opensips_1_10'


Thanks

Nathaniel

On 1/16/14, 4:59 AM, Vlad Paiu wrote:

Hello,

Seems to be a linking error.

Do you have libncurses dev library installed on that Solaris machine, 
or just libcurses dev library ?
1.10 has the old libcurses replaced with libncurses - previously 
libcurses was just a sym link to libncurses but some newer OSs 
started to remove the libcurses link and just present the libncurses so.


Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  
On 16.01.2014 07:54, Nathaniel L Keeling III wrote:

Hello,

I am trying to upgrade to Opensips 1.10 on Solaris 10. I download 
the git source and ran the make menuconfig command and got this 
error. I had no problems with Opensips 1.8 when I ran make menuconfig.


make[1]: Entering directory 
`/usr/local/src/opensips/opensips_1_10/menuconfig'
gcc -o configure -g -Wall 
-DMENUCONFIG_CFG_PATH=\menuconfig/configs/\ 
-DMENUCONFIG_GEN_PATH=\etc/\ -DMENUCONFIG_HAVE_SOURCES=1  cfg.o 
curses.o items.o commands.o menus.o parser.o main.o -lncurses

Undefined   first referenced
 symbol in file
initscr32   main.o
w32attron   curses.o
w32attroff  curses.o
ld: fatal: Symbol referencing errors. No output written to configure
make[1]: *** [all] Error 1
make[1]: Leaving directory 
`/usr/local/src/opensips/opensips_1_10/menuconfig'

./menuconfig/configure --local
make: ./menuconfig/configure: Command not found
make: *** [menuconfig] Error 127


Thanks

Nathaniel


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Re: [OpenSIPS-Users] Adjusting Headers

[Vlad Paiu]

Hello,

Can you please send us a SIP trace for this, maybe we can figure out 
what the device doesn't like ?


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 06.02.2014 08:46, Alec Doran-Twyford wrote:

Hi All,

Solved the problem with manipulating the Headers but this has not 
fixed the problem of getting a 401 or 407 authentication request from 
the other device when i send them an invite.
It just hangs, by sending me back a trying response. I have compared 
the Wireshark to the other device I have been testing and nothing seem 
to be different now other than thing which are different throughout 
all the device.


anyone got some helpful hint which could cause this problem? my 
thoughts are that it doesn't like some one the following head which 
had been put into the SIP INIVITE

RecordRoute or the Via's

as I couldn't see them in any of my other test on diffrent device.

Thanks

Alec



Alec Doran-Twyford

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| E-mail: a.dorantwyf...@ivstel.com mailto:a.dorantwyf...@ivstel.com 
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On 6 February 2014 11:38, Alectronic a.dorantwyf...@ivstel.com 
mailto:a.dorantwyf...@ivstel.com wrote:


Hi All,
Thanks my device is now able to register with other device.

However I now have another problem which is the other end does not
trying to
authenticate when I send an Invite (It just hang there) I have
compared a
trace call of a freepbx and a SIP phone over wireshark connecting
to the
other endpoint and it would seem I need to manipulate the header
so that
the To header is that of the register IP/domain.

What would it take to be able to do this manipulation?

Thanks

Alec



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Re: [OpenSIPS-Users] Setting do_routing

[Vlad Paiu]

Hello,

Exactly as Nick suggested, you can create a carrier which contains the 
gateway list that you are interested to specifically route to - further 
more, you get the added advantage that you can change that gateway list 
without having to change your config and restart OpenSIPS.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 06.02.2014 05:54, Nick Altmann wrote:

Why not to create carrier with gateways list and then route to carrier?

--
Nick

2014-02-06 Nick Cameo sym...@gmail.com:

Hello Bogdan,

Thank you so much for your response. Looking at the documentation it seems
that both `route_to_carrier`, and `route_to_gw`
accept a single parameter that is the ID of the carrier or gateway. Is there
anything that will accept a gateway list by any chance?

It would be perfect if we can pass a gwlist to a method and have it iterate
through the gateways given a failed relay.

Thanks in Advance,

N.

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Re: [OpenSIPS-Users] Dialog | Topology_hiding

[Vlad Paiu]

Hello,

Please post your entire sequential message processing here - you 
shouldn't get to that preloaded route part for sequential requests, 
since sequentials are actually expected to have Route headers.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 04.02.2014 22:21, Wilmar Campos wrote:

Hi All,

I am using Opensips 1.8.3 and I am trying to setup topology hiding.

The problem I have is with bye messages:
15.068600 2.x.x.1 - 2.x.x.2 SIP Request: BYE
sip:2.x.x.2;did=1e4.0a44
15.074164 2x.x.x.2 - 2.x.x.1 SIP Status: 403 Preload Route
denied

I think is because of this line:
   if(loose_route()) {
xlog(L_ERR,Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]);

if (!is_method(ACK))
  send_reply(403,Preload Route denied);
  exit;
}
   }


Can you please guide me here?

Thanks,

Wilmar


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Re: [OpenSIPS-Users] Setting do_routing

[Vlad Paiu]

Hello,

You could use route_to_carrier ( see [1] ) for that, which accepts a 
carrier instead of a single gateway.


[1] http://www.opensips.org/html/docs/modules/1.11.x/drouting.html#id294558

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 04.02.2014 22:50, Nick Cameo wrote:

Hello Bogdan,

Sorry to re-live this old email however, from what I understand 
route_to_gw() only takes a single gw?

Which function allows for explicit ordered gwlist?

Kind Regards,

Nick.


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Re: [OpenSIPS-Users] topology hiding

[Vlad Paiu]

Hello,

The sequential processing part is a little bit wrong - you should have

if (loose_route() || match_dialog()) {
if ($DLG_status==NULL) {
xlog( cannot match request to a dialog \n);
# something wrong - might want to drop such 
requests
}


Can you please also post a trace of the traffic flow when the Route 
header gets that bogus \304 header ? Trying to replicate this on my side 
and see what's wrong.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 03.02.2014 20:36, BJ Quinn wrote:

Oh and the only manual manipulation of the route headers was an attempt to get 
rid of that \304 in the header.

I think the \304 thing may be a red herring for now.  I still can't get the 
topology hiding to work.  Below is my config file.  It's literally the default 
config file with nothing changed but I've put in my IP address on the listen 
line, added a couple of aliases, added UAC module to try to change the from 
header (that works) and the dialog module and a couple of modifications to the 
route to make topology hiding work (not working for me).

Am I putting this in the wrong part of the route?

Thx

-BJ Quinn

---
debug=3
log_stderror=no
log_facility=LOG_LOCAL0

fork=yes
children=4

auto_aliases=no

listen=udp:xx.xx.xx.9:5060

disable_tcp=yes

disable_tls=yes

alias=xx.xx.xx.76:5060
alias=xx.xx.xx.77:5060

mpath=/usr/lib64/opensips/modules

loadmodule signaling.so

loadmodule sl.so

loadmodule tm.so
modparam(tm, fr_timer, 5)
modparam(tm, fr_inv_timer, 30)
modparam(tm, restart_fr_on_each_reply, 0)
modparam(tm, onreply_avp_mode, 1)

loadmodule rr.so
modparam(rr, append_fromtag, 0)

loadmodule maxfwd.so

loadmodule sipmsgops.so

loadmodule mi_fifo.so
modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)
modparam(mi_fifo, fifo_mode, 0666)

loadmodule uri.so
modparam(uri, use_uri_table, 0)

loadmodule usrloc.so
modparam(usrloc, nat_bflag, NAT)
modparam(usrloc, db_mode,   0)

loadmodule registrar.so
modparam(registrar, tcp_persistent_flag, TCP_PERSISTENT)

loadmodule acc.so
modparam(acc, early_media, 0)
modparam(acc, report_cancels, 0)
modparam(acc, detect_direction, 0)
modparam(acc, failed_transaction_flag, ACC_FAILED)
modparam(acc, log_flag, ACC_DO)
modparam(acc, log_missed_flag, ACC_MISSED)

# added to rewrite from header
loadmodule uac.so
loadmodule uac_auth.so
modparam(uac,restore_mode,manual)

#added for topology hiding
loadmodule dialog.so

route{
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}

if (has_totag()) {
if (loose_route()) {
# added for topology hiding 
if ($DLG_status==NULL  !match_dialog() ) {
xlog( cannot match request to a dialog \n);
}
#/added for topology hiding

if (is_method(BYE)) {
setflag(ACC_DO);
setflag(ACC_FAILED);
} else if (is_method(INVITE)) {
record_route();
}

route(relay);
} else {

if ( is_method(ACK) ) {
if ( t_check_trans() ) {
t_relay();
exit;
} else {
exit;
}
}
sl_send_reply(404,Not here);
}
exit;
}

if (is_method(CANCEL))
{
if (t_check_trans())
t_relay();
exit;
}

t_check_trans();

if ( !(is_method(REGISTER)  ) ) {
if (from_uri==myself)
{
} else {
if (!uri==myself) {
send_reply(403,Rely forbidden);
exit;
}
}
}

if (loose_route()) {
xlog(L_ERR,
Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]);
if (!is_method(ACK))
sl_send_reply(403,Preload Route denied);
exit;
}

if (!is_method(REGISTER|MESSAGE))
record_route();

if (is_method(INVITE)) {

setflag(ACC_DO); # do accounting
}

 if (is_method(INVITE)) {
# rewrite from header
uac_replace_from(sip:$f...@xx.xx.xx.9);
# trying to fix that /304 problem

Re: [OpenSIPS-Users] check if ip address belongs to ip and subnet subscriber

[Vlad Paiu]

Hello,

Internally, when using ip.pton, OpenSIPS stores the binary 
representation of the IP as a character array ( due to the different 
size of IPv4 vs IPv6 ), so using the  will not work properly - that's 
why you're code snippet is not working as expected.  Will try to look 
and see how to fix this - if possible at all.


In the mean time, I'd strongly suggest using the permissions module - 
there is no performance penalty when reloading the address table ( while 
the new table info is loading, OpenSIPS will hold the old table info in 
memory , and once the loading is done there new and old IP lists will 
just be swapped ).


I'd suggest having a address groupid integer stored in the subscriber 
table which needs be added in the load_credentials param ( to be loaded 
at auth time ), and then run

check_source_address($avp(subscriber_grp))

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 04.02.2014 09:24, Edwin wrote:

Vlad,

The $avp(sourceip_net) in the test was 255.255.255.0. I want to use a
netmask so clients can use any ip from the ip block we have assigned them.

I'm also testing the permission module (as Stefano suggested) which is of
course perfect in this case. The only thing I 'worry about' is that
everytime a ip is changed in the address table and we hit the 'address
reload' is this has an impact on a live system with many registrations per
second... (so will there be a little timeout or does the process seamless
continue). This because clients can change there own ip / subnet in a web
based management system.



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Re: [OpenSIPS-Users] OpenSIPS At FOSDEM '14

[Vlad Paiu]

Hello,

The presentation has been uploaded on the OpenSIPS website, at
http://www.opensips.org/Community/Event-Fosdem2014

Also, I would like to thank all the people participating to this event 
and hope to see you all again at FOSDEM 15 !


Best,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 16.01.2014 20:13, Vlad Paiu wrote:

Hello all,

At the beggining of February, the OpenSIPS team will be attending 
FOSDEM '14 conference [1], 1-2 February 2014, where we will hold a 
lightning talk focusing on building fully redundant, distributed 
platforms usiing OpenSIPS [2] .


If you are at the event and are looking to discover, understand and 
evaluate OpenSIPS, or just looking to meet the OpenSIPS devs and have 
an informal talk, come and meet us!


Looking forward to seeing you there!

[1] https://fosdem.org/2014/
[1] https://fosdem.org/2014/schedule/event/opensips/

Best Regards,




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Re: [OpenSIPS-Users] topology hiding

[Vlad Paiu]

Hello,

No, you should not regex out those bogus characters, this seems like a 
bug - could you please send us to SIP trace for your scenario so I can 
understand how and when it's happening ? Are you currently doing any 
manual manipulation on the Route headers in your script ?


Also, if possible, Please open an issue on 
https://github.com/OpenSIPS/opensips/issue for this so we can better 
keep track of it.



Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 01.02.2014 02:26, BJ Quinn wrote:

Hi,

I'd like to use topology_hiding(), but I can't quite understand how to 
integrate it into the routing part of the configuration file. I have my 
opensips box on a public IP and some machines initiating calls through the 
opensips box that are also on public IPs, so no NAT going on or anything like 
that. However, a couple of the carriers we're trying to use don't like seeing 
the IP address of the machines initiating the call (in Route and Contact 
headers, etc.) and that's causing problems including some carriers don't think 
the call has set up properly (even though it goes through), which leads to 
missing BYEs. Anyway, seems like topology_hiding() is a great idea anyway, 
regardless of the fact that I've had a carrier specifically request it.

I'm using 1.10.  So I've started with the basic Residential scenario made from 
osipsconfig. I didn't check any of the options (like ENABLE_TCP, USE_ALIASES, 
etc.) and modified only my IP address and added a couple of aliases for the 
machines making the calls.  I added the following outside of the routing logic 
to load the dialog module to make topology_hiding() available.

   loadmodule dialog.so

Then, under if(has_totag()) { if (loose_route()) { I added --

   if ($DLG_status==NULL  !match_dialog() ) {
 xlog( cannot match request to a dialog \n);
   }

And outside of the if(has_totag()) section I added --

   if (is_method(INVITE)) {
 create_dialog();
 topology_hiding();
   }

Without these added sections, things are fine on some carriers and with other 
carriers I have the problems described above which causes me to want to enable 
topology hiding.  With these added sections, I get 408 timeouts since it 
appears that the opensips box is responding NOT HERE to the carrier's 200 OKs.

Also, possibly unrelated, in either case I'm getting a weird \304 added to my 
Route header.  Should I just replace the Route header and regex that out?

Route: sip:xx.xx.xx.xx:\304;lr

Thanks!

-BJ Quinn

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Re: [OpenSIPS-Users] check if ip address belongs to ip and subnet subscriber

[Vlad Paiu]

Hello,

First of all, you can't do $(avp(sourceip_mask){ip.pton}) since ip 
transformations only work against on IP addresses, and from what I see 
in your previous script examples, $avp(sourceip_mask) was something like 
24, when instead you should be trying 255.255.255.0 .


Also, do you really need to also check netmasks, or wouldn't
 if ($si == $avp(sourceip_net))

work for you ?

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 02.02.2014 00:03, Edwin wrote:

Stefano,

In fact you have a point. And probably I will put the network address in the
database.

But still, I hate it when I don't understand why a logical comparison
doesn't work like I expect it to do.

In this case the output of $(avp(sourceip_mask){ip.pton})  $(si{ip.pton})
should be the same as $avp(sourceip_net) and it gives a error.

So, or it is a 'bug' or I do it wrong (sometimes the docs are a little bit
to summier...)

Is it possible to put == between to ip.xxx statements?



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Re: [OpenSIPS-Users] Opensips 1.9 stop working after upgrade

[Vlad Paiu]

Hello,

I guess you were still using your old database after migrating to 1.9 ?
If yes, you should check out the tips at
http://www.opensips.org/Documentation/Migration-1-8-0-to-1-9-0

for migrating both DB and script, since 1.9 brings some new columns for 
the subscriber table, which is exactly what OpenSIPS was complaining 
about with those ERROR messages.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 03.02.2014 15:35, Chandra Prakash wrote:

Hi guys,

Problem is solved, I need to add the extra column 'attr' in aliases table
and also need to change the version table.

I think it is is a bug, Pls correct me If I'm wrong.

Thanks

-Original Message-
From: Chandra Prakash [mailto:chandraprak...@virtualemployee.com]
Sent: Monday, February 3, 2014 5:50 PM
To: 'users@lists.opensips.org'
Subject: Opensips 1.9 stop working after upgrade

Hi,

I've aupdated the opensips 1.9 after upgrade it stopped working and giving
this error.

CRITICAL:db_mysql:wrapper_single_mysql_real_query: driver error (1054):
Unknown column 'attr'  in 'field list'
Jan 27 04:56:55 debiansip01 /sbin/opensips[3728]: ERROR:core:db_do_query:
error while submitting query - [select username,contact,expires,q,cal
lid,cseq,flags,cflags,user_agent,received,path,socket,methods,last_modified,
sip_instance,attr from aliases ]
Jan 27 04:56:55 debiansip01 /sbin/opensips[3728]:
ERROR:usrloc:preload_udomain: db_query (1) failed Jan 27 04:56:55
debiansip01 /sbin/opensips[3728]: ERROR:usrloc:child_init: child(1): failed
to preload domain 'aliases'
Jan 27 04:56:55 debiansip01 /sbin/opensips[3728]: ERROR:core:init_mod_child:
failed to initializing module usrloc, rank 1 Jan 27 04:56:55 debiansip01
/sbin/opensips[3728]: ERROR:core:main_loop: init_child failed for UDP
listener

Pls help


-Original Message-
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Today's Topics:

1. Re: check if ip address belongs to ip and subnet subscriber
   (Edwin)
2. Re: check if ip address belongs to ip and subnet subscriber
   (Stefano Pisani)
3. Re: check if ip address belongs to ip and subnet subscriber
   (Edwin)
4. Re: check if ip address belongs to ip and subnet subscriber
   (Stefano Pisani)
5. Re: check if ip address belongs to ip and subnet subscriber
   (Edwin)


--

Message: 1
Date: Sat, 1 Feb 2014 08:44:26 -0800 (PST)
From: Edwin eahaselh...@gmail.com
Subject: Re: [OpenSIPS-Users] check if ip address belongs to ip and
subnet  subscriber
To: users@lists.opensips.org
Message-ID: 1391273066233-7589398.p...@n2.nabble.com
Content-Type: text/plain; charset=us-ascii

This helped a bit, so I came up with:

$var(sourceip_net) = $(avp(sourceip_mask){ip.pton}) 
$(avp(sourceip){ip.pton});
$var(si_net) = $(avp(sourceip_mask){ip.pton})  $(si{ip.pton});

if($var(sourceip_net) == $var(si_net))
{
 xlog(L_INFO,  ip $si belongs to $au\n); } else {
 xlog(L_INFO,  ip $si does not belong to $au\n);
 sl_send_reply(403, Forbidden);
 exit;
}

But I like to write i like this:

if( [ $(avp(sourceip_mask){ip.pton})  $(avp(sourceip){ip.pton}) ] == [
$(avp(sourceip_mask){ip.pton})  $(si{ip.pton}) ] ) {
 xlog(L_INFO,  ip $si belongs to $au\n); } else {
 xlog(L_INFO,  ip $si does not belong to $au\n);
 sl_send_reply(403, Forbidden);
 exit;
}

But this gives an error (column 121-123: syntax error, column 121-123: bad
command!)



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Message: 2
Date: Sat, 01 Feb 2014 17:50:40 +0100
From: Stefano Pisani stefano.pis...@omnianet.it
Subject: Re: [OpenSIPS-Users] check if ip address belongs to ip and
subnet subscriber
To: OpenSIPS users mailling list users@lists.opensips.org
Message-ID: 52ed25e0.5060...@omnianet.it
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

why you are using []? use () instead.

Il 01/02/2014 17.44, Edwin ha scritto:

This helped a bit, so I came up with:

$var(sourceip_net) = $(avp(sourceip_mask){ip.pton}) 
$(avp

Re: [OpenSIPS-Users] loadbalancer module and SUBSCRIBE

[Vlad Paiu]

Hello,

The load_balancer module works on top of the dialog module, which only 
supports INVITE based dialogs - thus you'll have to use the dispatcher 
module for handling the distributing of all methods other than INVITE.



Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 27.01.2014 17:33, Bobby Smith wrote:
Any ideas?  What are some ways people load balance presence to their 
application servers? (when opensips is just a proxy, and without using 
any of the presence module support)


Thanks much,



On Wed, Jan 22, 2014 at 6:06 PM, Bobby Smith bobby.sm...@gmail.com 
mailto:bobby.sm...@gmail.com wrote:


Greetings list,

I'd like to be able to load balance subscribes across the same set
of servers I'm using to handle calls with the loadbalancer module.
 I think I've read somewhere (I haven't looked at the code yet)
that the loadbalancer module only handles dialog creating INVITEs,
and not dialog creating SUBSCRIBES (for sip presence).

How do others achieve this?  Through the dispatcher module?  It
would be really nice to reuse the current mechanism we have,
because I want dialog load balancing behavior to be consistent
over the pool of servers we have servicing both functionalies for.

I think a lot of the other modules have problems with SUBSCRIBE as
well.  An initial dialog-creating subscribe (for long running
subscriptions) doesn't create a dialog when create_dialog() is
called from the routing script, which seems to cause some
re-SUBSCRIBE messages later on to have issues processing 200 OK's
back (not getting to the right place).  So are SUBSCRIBES not
supported at all from within the dialog module infrastructure?

Thanks,

Bobby




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Re: [OpenSIPS-Users] cannot parse the Contact URI with colon in display part

[Vlad Paiu]

Hello,

Going through the SIP RFC, I see that the display name is described as

display-name = *(token LWS)/ quoted-string

and token is

token = 1*(alphanum / - / . / ! / % / * / _ / + / ` 
/ ' / ~ )


Thus if your display name contains other symbols than the above, it 
should be a quoted string.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 21.01.2014 22:57, Martin Stock wrote:

Hi Guys,

IMHO the current OpenSIPS creates a strange error message if the 
display part of contact header contains a colon :.

E.g. the error messages looks like this:
--- snip ---
opensips[24791]: ERROR:core:parse_uri: bad uri,  state 0 parsed: Fax: 
(4) / Fax: Name sip:1001@192.168.16.158:5060 (62)
opensips[24791]: ERROR:nat_traversal:get_contact_uri: cannot parse the 
Contact URI

--- snap ---

The contact header in the of the UA:
Contact: Fax: Name sip:1001@192.168.16.158:5060

If I remove the : inside the display part the error message disappears.

In diplay parts of other URIs this is not a problem.

Is that a bug?

Regards
Martin

P.S.:
I'm using OpenSIPS version 1.10.

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Re: [OpenSIPS-Users] MSILO malfunction when using with MongoDB

[Vlad Paiu]

Hello,

As the module docs says ( [1] ) the module was tested with a small 
fraction of all the modules, and needs extensive testing to make sure 
it's compatible with each and every module that currently uses the 
direct DB interface.
Thanks for the testing and bug report - I will replicate this on my end 
and see why the message dumping is not working properly.


[1] 
http://www.opensips.org/html/docs/modules/1.11.x/db_cachedb.html#id249052


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 17.01.2014 05:11, Hoa Nguyen wrote:


It seems like because m_dump function try to query id column but the 
exact name is _id in mongo. Maybe it's not much deal to fix it J




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Re: [OpenSIPS-Users] Error on make menuconfig

[Vlad Paiu]

Hello,

Seems to be a linking error.

Do you have libncurses dev library installed on that Solaris machine, or 
just libcurses dev library ?
1.10 has the old libcurses replaced with libncurses - previously 
libcurses was just a sym link to libncurses but some newer OSs started 
to remove the libcurses link and just present the libncurses so.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 16.01.2014 07:54, Nathaniel L Keeling III wrote:

Hello,

I am trying to upgrade to Opensips 1.10 on Solaris 10. I download the 
git source and ran the make menuconfig command and got this error. I 
had no problems with Opensips 1.8 when I ran make menuconfig.


make[1]: Entering directory 
`/usr/local/src/opensips/opensips_1_10/menuconfig'
gcc -o configure -g -Wall 
-DMENUCONFIG_CFG_PATH=\menuconfig/configs/\ 
-DMENUCONFIG_GEN_PATH=\etc/\ -DMENUCONFIG_HAVE_SOURCES=1 cfg.o 
curses.o items.o commands.o menus.o parser.o main.o -lncurses

Undefined   first referenced
 symbol in file
initscr32   main.o
w32attron   curses.o
w32attroff  curses.o
ld: fatal: Symbol referencing errors. No output written to configure
make[1]: *** [all] Error 1
make[1]: Leaving directory 
`/usr/local/src/opensips/opensips_1_10/menuconfig'

./menuconfig/configure --local
make: ./menuconfig/configure: Command not found
make: *** [menuconfig] Error 127


Thanks

Nathaniel


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