Re: [OpenSIPS-Users] Users Digest, Vol 40, Issue 76

[auto-reply from antonio.spirande...@longwave.eu]

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Re: [OpenSIPS-Users] BYE on 180 causing dialog loop

[Vlad Paiu]

Hello,

We will try and fix the BYE in early stage problem with Topology Hiding 
as soon as possible.


Regards,

Vlad Paiu
OpenSIPS Developer


On 11/20/2011 02:28 PM, ddgiants wrote:

Any other thoughts on this guys?
Tx
D

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[OpenSIPS-Users] Problem to relay sip message recieved port 9090 to asterisk port 7160

[arif tuhin]

I'm Trying to relay sip messages from sip client to asterisk. Kamailio should 
accept the msg on 9090 udp port and it will forward the sip message to 7160 udp 
port of Asterisk.
I thought the default cfg file will be enough to accomplish this simple task. 
But kamailio is receiving the msg ok but its not forwarding the msg. Another 
interesting thing is if i enable tls and give a tls port of kamailio , it 
easily forwards and the client registers ok.
With Best Regards
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[OpenSIPS-Users] delay for first invite

[Dani Popa]

Hi,

I know it's a weird question, but still, it is possible to add a delay 
(let's say 5 seconds) for the first invite(somehow to increase post dial 
delay with 5 seconds).


Thanks,
Dani

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Re: [OpenSIPS-Users] delay for first invite

[Sammy Govind]

We can add delay for a particular host, add error, packet drop and packet
reordering in network layer but for just first invite !! ummm...yes in
configuration where you detect _first_ INVITE put a sleep in there but then
it won't be true network latency simulation.

On Tue, Nov 22, 2011 at 6:26 PM, Dani Popa  wrote:

> Hi,
>
> I know it's a weird question, but still, it is possible to add a delay
> (let's say 5 seconds) for the first invite(somehow to increase post dial
> delay with 5 seconds).
>
> Thanks,
> Dani
>
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Re: [OpenSIPS-Users] delay for first invite

[Bogdan-Andrei Iancu]

Hi all,

Using sleep() functions in script is really dangerous as actually you 
block an opensips process in doing the sleep. So if you have 8 processes 
and you have 8 calls in sleep for 5 secs, you will end up blocking your 
entire opensips for all SIP traffic.


A not simple approach, but more efficient is to first set fr_timer to 5 
and send the invite to a destination that does not exists / answer -> in 
5 seconds you will end up in failure route and you can resume the 
processing there.and there is no blocking in opensips.


Regards,
Bogdan

On 11/22/2011 03:46 PM, Sammy Govind wrote:
We can add delay for a particular host, add error, packet drop and 
packet reordering in network layer but for just first invite !! 
ummm...yes in configuration where you detect _first_ INVITE put a 
sleep in there but then it won't be true network latency simulation.


On Tue, Nov 22, 2011 at 6:26 PM, Dani Popa > wrote:


Hi,

I know it's a weird question, but still, it is possible to add a
delay (let's say 5 seconds) for the first invite(somehow to
increase post dial delay with 5 seconds).

Thanks,
Dani

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Re: [OpenSIPS-Users] Problem to relay sip message recieved port 9090 to asterisk port 7160

[Bogdan-Andrei Iancu]

Hi,

1) if you have kamailio questions, why do you post on the opensips 
mailing list ???


2) cross pointing (posting in the same time on multiple lists) is a 
really bad idea !!!


3) maybe you should consider using OpenSIPS and you things will start 
working ;)


Regards,
Bogdan

On 11/22/2011 02:14 PM, arif tuhin wrote:
I'm Trying to relay sip messages from sip client to asterisk. Kamailio 
should accept the msg on 9090 udp port and it will forward the sip 
message to 7160 udp port of Asterisk.
I thought the default cfg file will be enough to accomplish this 
simple task. But kamailio is receiving the msg ok but its not 
forwarding the msg. Another interesting thing is if i enable tls and 
give a tls port of kamailio , it easily forwards and the client 
registers ok.

With Best Regards


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Re: [OpenSIPS-Users] BYE on 180 causing dialog loop

[Bogdan-Andrei Iancu]

Hi Saul,

Just to clarify - while the call is still in early stage, the control is 
done at transaction level (the INVITE transaction) - if transaction is 
successful (200OK) -> call established; if transaction fails (negative 
reply) -> call fails.


So, the dialog module is not interested in the CANCEL -> it will wait to 
see the feedback on the INVITE level, like the 487 reply (as a result of 
the CANCEL being accepted).


The BYE (instead of CANCEL) works in a similar way - the dialog module 
will simply wait to see what will happen with the INVITE.


So, from standard dialog state, the dialog module does not care about 
the CANCELs or BYEs in early state.


Of course, things are a bit different when using topology hiding with 
dialog module - there you have the "topo hide" the BYE also ;).and 
this needs to be fixed


Regards,
Bogdan

On 11/17/2011 02:24 PM, Saúl Ibarra Corretgé wrote:

Hi,

On Nov 17, 2011, at 12:53 PM, Vlad Paiu wrote:


Hello,

The problem lies in the fact that the device send BYE while the dialog was not 
established yet, when in fact a Cancel should have been used.


The caller MAY use a BYE instead of a CANCEL to terminate a dialog in the early 
stage, that is, if it received a 101-199 response with a to-tag. (RFC 3261, sec 
15)

Since no SIP trace was provided I don't know if that was the case. Anyway, 
would OpenSIPS handle that situation correctly?


Regards,

--
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Re: [OpenSIPS-Users] BYE on 180 causing dialog loop

[Saul Ibarra Corretge]

Hi Bogdan,

On Nov 22, 2011, at 3:17 PM, Bogdan-Andrei Iancu wrote:

> Hi Saul,
> 
> Just to clarify - while the call is still in early stage, the control is done 
> at transaction level (the INVITE transaction) - if transaction is successful 
> (200OK) -> call established; if transaction fails (negative reply) -> call 
> fails.
> 
> So, the dialog module is not interested in the CANCEL -> it will wait to see 
> the feedback on the INVITE level, like the 487 reply (as a result of the 
> CANCEL being accepted).
> 
> The BYE (instead of CANCEL) works in a similar way - the dialog module will 
> simply wait to see what will happen with the INVITE.
> 
> So, from standard dialog state, the dialog module does not care about the 
> CANCELs or BYEs in early state.
> 
> Of course, things are a bit different when using topology hiding with dialog 
> module - there you have the "topo hide" the BYE also ;).and this needs to 
> be fixed
> 

Thanks for the detailed explanation!

Regards,

-- 
Saúl Ibarra Corretgé
AG Projects






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Re: [OpenSIPS-Users] Users Digest, Vol 40, Issue 77

[auto-reply from antonio.spirande...@longwave.eu]

Sarò assente fino al 25 Novembre compreso. Per urgenze rivolgersi direttamente 
ad assiste...@longwave.eu o chiamare lo 0522375500. Saluti

I will be out of office till  November 25th 2011.

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Re: [OpenSIPS-Users] LRN dips with Dynamic Routing

[Bogdan-Andrei Iancu]

Hi Kpirlo,

When sending the call to the dip provider, use a failure route in order 
to catch the 3xx reply you get back. In the failure route, use the 
uac_redirect module with the get_redirects() function 
(http://www.opensips.org/html/docs/modules/1.7.x/uac_redirect.html#id250367) 
in order to extract the redirect contacts from the reply and push them 
as new destinations.


Regards,
Bogdan

On 11/20/2011 08:04 PM, Kpirlo wrote:
We are currently using the Dynamic routing module for our least cost 
routing.


Now we are looking at implementing an LRN dipping service, where we 
will send the call to a dip provider first and receive a 302 redirect 
back which will have the LRN returned in the contact header as "rn=" 
if the number has been ported or will include ";npdi"  in the contact 
header if it has not been ported.


Im asking for any advice anyone has on how to implement this and how 
it could work with dymanic routing to choose the route based on rn if 
available, but actually send the call using the original "to" number.


Thank you in advance for any help.

Kent


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Re: [OpenSIPS-Users] SRV lookups with b2bua

[Bogdan-Andrei Iancu]

Hi Ryan,

Could you post the domain and the SRV records you have populated?  
OpenSIPS DNS engine is able to handle priorities and weights between 
records, so .either records are not right, either there is a bug 
somewhere...


Regards,
Bogdan

On 11/21/2011 03:39 PM, Ryan Revels wrote:

Ovidiu,

The DNS server replies with both SRV records & indicates they are of 
equal priority and weight.
My question is why OpenSIPS would choose the same A record each and 
every time?


Thanks,
Ryan

On Fri, Nov 18, 2011 at 3:44 PM, Ovidiu Sas > wrote:


Probably your DNS server doesn't do round robin on your records.

On Fri, Nov 18, 2011 at 4:32 PM, Ryan Revels mailto:r...@revelous.net>> wrote:
> Ovidiu,
>
> I am seeing the same behavior when in proxy mode. Can you help
me understand
> where the problem may be occurring?
>
> Thanks,
> Ryan
>
> On Fri, Nov 18, 2011 at 12:37 PM, Ovidiu Sas
mailto:o...@voipembedded.com>> wrote:
>>
>> The B2B modules are using the tm API to construct requests and
>> therefore the handling of SRV records should not be influenced
by how
>> INVITE requests are handled: proxy or b2b.
>> Try to handle the INVITE in proxy mode and check if you have
the same
>> behaviour.
>>
>> Regards,
>> Ovidiu Sas
>>
>> On Fri, Nov 18, 2011 at 1:12 PM, Ryan Revels mailto:r...@revelous.net>> wrote:
>> > Bump
>> >
>> > On Nov 11, 2011 2:34 PM, "Ryan Revels" mailto:r...@revelous.net>> wrote:
>> >>
>> >> Do the b2bua modules fully support resolving SRV records?
>> >> I have 2 records with equal priority and weight, but the
b2bua always
>> >> chooses the same one:
>> >>
>> >> 10.10.10.1 -> 8.8.8.8 DNS Standard query NAPTR
sip.notreal.com 
>> >> 8.8.8.8 -> 10.10.10.1 DNS Standard query response NAPTR 10 10 S
>> >> 10.10.10.1 -> 8.8.8.8 DNS Standard query SRV
_sip._udp.sip.notreal.com 
>> >> 8.8.8.8 -> 10.10.10.1 DNS Standard query response SRV 0 0 5060
>> >> test2.notreal.net  SRV 0 0 5060
test1.notreal.net 
>> >> 10.10.10.1 -> 8.8.8.8 DNS Standard query A test1.notreal.net

>> >> 8.8.8.8 -> 10.10.10.1 DNS Standard query response A 1.2.3.4
>> >> 10.10.10.1 -> 8.8.8.8 DNS Standard query NAPTR
sip.notreal.com 
>> >> 8.8.8.8 -> 10.10.10.1 DNS Standard query response NAPTR 10 10 S
>> >> 10.10.10.1 -> 8.8.8.8 DNS Standard query SRV
_sip._udp.sip.notreal.com 
>> >> 8.8.8.8 -> 10.10.10.1 DNS Standard query response SRV 0 0 5060
>> >> test2.notreal.net  SRV 0 0 5060
test1.notreal.net 
>> >> 10.10.10.1 -> 8.8.8.8 DNS Standard query A test1.notreal.net

>> >> 8.8.8.8 -> 10.10.10.1 DNS Standard query response A 1.2.3.4
>> >> 10.10.10.1 -> 8.8.8.8 DNS Standard query NAPTR
sip.notreal.com 
>> >> 8.8.8.8 -> 10.10.10.1 DNS Standard query response NAPTR 10 10 S
>> >> 10.10.10.1 -> 8.8.8.8 DNS Standard query SRV
_sip._udp.sip.notreal.com 
>> >> 8.8.8.8 -> 10.10.10.1 DNS Standard query response SRV 0 0 5060
>> >> test2.notreal.net  SRV 0 0 5060
test1.notreal.net 
>> >> 10.10.10.1 -> 8.8.8.8 DNS Standard query A test1.notreal.net

>> >> 8.8.8.8 -> 10.10.10.1 DNS Standard query response A 1.2.3.4
>> >> 10.10.10.1 -> 8.8.8.8 DNS Standard query NAPTR
sip.notreal.com 
>> >> 8.8.8.8 -> 10.10.10.1 DNS Standard query response NAPTR 10 10 S
>> >> 10.10.10.1 -> 8.8.8.8 DNS Standard query SRV
_sip._udp.sip.notreal.com 
>> >> 8.8.8.8 -> 10.10.10.1 DNS Standard query response SRV 0 0 5060
>> >> test2.notreal.net  SRV 0 0 5060
test1.notreal.net 
>> >> 10.10.10.1 -> 8.8.8.8 DNS Standard query A test1.notreal.net

>> >> 8.8.8.8 -> 10.10.10.1 DNS Standard query response A 1.2.3.4
>> >>
>> >> Any help would be much appreciated.
>> >> Thanks.
>> >
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Re: [OpenSIPS-Users] Accounting 200 ACKS with Radius

[Bogdan-Andrei Iancu]

Maybe switching:
modparam("acc", "report_ack", 0)

to 1 will help you.

Regards,
Bogdan

On 11/20/2011 12:50 AM, discodo...@aol.com wrote:

Here is the entire config.

http://pastebin.com/ZJRNJupn


-Original Message-
From: Nick Khamis 
To: OpenSIPS users mailling list 
Sent: Sat, Nov 19, 2011 1:34 am
Subject: Re: [OpenSIPS-Users] Accounting 200 ACKS with Radius

Hard to say with just a route. Is it possible to post the rest of the config?

Nick.

On Fri, Nov 18, 2011 at 8:02 PM,mailto:discodo...@aol.com>>  wrote:
>  Hello all,
>  I have opensips 1.6 and I am using the radius accounting.  Everything is
>  working great.  What I am trying to do is also account the 200 ACK on a call
>  connect.  When I was using the database to log CDR's it was working.  Now
>  that I moved to Radius for accounting my additional setflag does not seem to
>  create a request to my radius server.
>  I only get a start and a stop in my radius log.  I am looking for my 200
>  ACK.
>  Here is my on reply route.
>  onreply_route[2] {
>  xlog("ACK with rs= $rs/ rb= $rb / fd= $fd / fu=  $fu / rr= $rr / rs= $rs /
>  Ri= $Ri / si= $si");
>  if ($rs == "200"&&  $si == "xx.x.x.xx") {
>   xlog("==ACC==  $si | $rs ");
>  setflag(1); # do accounting
>  }
>  if (has_body("application/sdp")) {
>   xlog("Onreply 2 turn on rtp");
>  rtpproxy_answer();
>  }
>  }
>
>  Is there a flag that I am missing?  Or syntax that I am not aware of?
>  ___
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>
>

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Re: [OpenSIPS-Users] Users Digest, Vol 40, Issue 78

[auto-reply from antonio.spirande...@longwave.eu]

Sarò assente fino al 25 Novembre compreso. Per urgenze rivolgersi direttamente 
ad assiste...@longwave.eu o chiamare lo 0522375500. Saluti

I will be out of office till  November 25th 2011.

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Re: [OpenSIPS-Users] Auth_DB's load_credentials isn't grabbing RPID info

[Bogdan-Andrei Iancu]

Hi Duane,

You need to fix your script logic and when you cache the fetched 
password  also cache (in a separate key) the fetched RPID.  So load , 
cache and fetch the passwd and rpid all the time together.


Regards,
Bogdan

On 11/19/2011 03:15 AM, osiris123d wrote:

I think I know the issue with this.  For INVITES I am using the
cachedb_local.  So if a user dials out the first time then credentials are
pulled from the database and the RPID info is gotten.  The second time the
user calls someone the cached credentials are used and the RPID is never
pulled.  So that is why my RPID is not getting attached to my INVITES.

Is there a way to fix this or should I just not use cachedb_local setup when
INVITES are being authenticated???

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Re: [OpenSIPS-Users] delay for first invite

[Dani Popa]

thanks,

Dani

On 11/22/11 16:09, Bogdan-Andrei Iancu wrote:

Hi all,

Using sleep() functions in script is really dangerous as actually you 
block an opensips process in doing the sleep. So if you have 8 
processes and you have 8 calls in sleep for 5 secs, you will end up 
blocking your entire opensips for all SIP traffic.


A not simple approach, but more efficient is to first set fr_timer to 
5 and send the invite to a destination that does not exists / answer 
-> in 5 seconds you will end up in failure route and you can resume 
the processing there.and there is no blocking in opensips.


Regards,
Bogdan

On 11/22/2011 03:46 PM, Sammy Govind wrote:
We can add delay for a particular host, add error, packet drop and 
packet reordering in network layer but for just first invite !! 
ummm...yes in configuration where you detect _first_ INVITE put a 
sleep in there but then it won't be true network latency simulation.


On Tue, Nov 22, 2011 at 6:26 PM, Dani Popa > wrote:


Hi,

I know it's a weird question, but still, it is possible to add a
delay (let's say 5 seconds) for the first invite(somehow to
increase post dial delay with 5 seconds).

Thanks,
Dani

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[OpenSIPS-Users] filter 503 messages

[Robert R]

Hi,

How can I filter the receiving 5xx messages, i.e. how can I parse 503
messages received by the proxy?

I have tried the following and none works:

t_check_status("503")
is_method("503")

Thanks,
R
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Re: [OpenSIPS-Users] filter 503 messages

[Ovidiu Sas]

You need to use t_check_status inside a failure_route.
Take a look at the default config to see how a failure_route is enabled.

Regards,
Ovidiu Sas


On Tue, Nov 22, 2011 at 3:27 PM, Robert R  wrote:
> Hi,
>
> How can I filter the receiving 5xx messages, i.e. how can I parse 503
> messages received by the proxy?
>
> I have tried the following and none works:
>
> t_check_status("503")
> is_method("503")
>
> Thanks,
> R
>
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>

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Re: [OpenSIPS-Users] About prepaid question

[Adrian Georgescu]

Is very difficult to help somebody see the forest because of the trees. That 
document is the most detailed explanation in existence about your question.

Please improve the document if is not good enough.

On Nov 22, 2011, at 4:30 AM, Nick wrote:

> Hi
> 
> Thanks for your support.
> 
> I saw this document http://download.ag-projects.com/CDRTool/doc/RATING.txt
> About Pricing formula, I don't understand.
> 
> In billing_customer, increment and min_duration set 0.
> In global.inc, 
> "priceDenominator"   => 100, 
> "priceDecimalDigits" => 2, 
> "durationPeriodRated"=> 60,  
> "trafficSizeRated"   => 1024,  
> "reportMissingRates" => 0, 
> "rate_longer_than"   => 0,  
> "MaxSessionTime" => 36000  
> 
> In billing_rates, durationRate set 100.
> 
> It's a Pricing formula.
>Pricing formula
>---
> 
>if min_duration then
>   minimumDurationCharged = min_duration
>else if minimumDurationCharged set in global inc
>   use minimumDurationCharged from global.inc
>else
>   minimumDurationCharged = call duration
> 
>if increment then
>   durationForRating = round to the next increment
>else
> durationForRating = call duration
> 
>if durationForRating >= minimumDurationCharged then
>   Price = connectCost/priceDenominator+
>   durationRate*durationForRating/durationPeriodRated/priceDenominator
>else 
> Price = 0
> 
> durationForRating = ?? What is durationForRating?? 
> minimumDurationCharged = 0
> connectCost = 0 
> How much is price??
> 
> I don't understand to calculate this price.
> Can you give me a detail explanation?? 
> Thanks
> 
> Nick 
> 
> 
> 
> On 2011年11月21日 21:34, Adrian Georgescu wrote:
>> 
>> http://download.ag-projects.com/CDRTool/doc/PREPAID.txt
>> 
>> 
>> On Nov 21, 2011, at 8:38 AM, Nick wrote:
>> 
>>> Hi
>>> 
>>> I have a question for callcontrol and cdrtool.
>>> 
>>> What command can i real-time found enough money to online user??
>>> 
>>> If I have account 09, It have 5 dollars.
>>> Now, It call to 09. How much money for 09 a real-time???
>>> 
>>> In MySQL. Only saw total money.  When 09 don't enough money,  Can I 
>>> notice to user??
>>> 
>>> Thanks for your support.
>>> 
>>> Nick
>>> 
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Re: [OpenSIPS-Users] Auth_DB's load_credentials isn't grabbing RPID info

[duane . larson]

Will do. Thanks.

On , Bogdan-Andrei Iancu  wrote:

Hi Duane,




You need to fix your script logic and when you cache the fetched password  
also cache (in a separate key) the fetched RPID. So load , cache and  
fetch the passwd and rpid all the time together.





Regards,



Bogdan





On 11/19/2011 03:15 AM, osiris123d wrote:




I think I know the issue with this. For INVITES I am using the



cachedb_local. So if a user dials out the first time then credentials are



pulled from the database and the RPID info is gotten. The second time the



user calls someone the cached credentials are used and the RPID is never



pulled. So that is why my RPID is not getting attached to my INVITES.




Is there a way to fix this or should I just not use cachedb_local setup  
when



INVITES are being authenticated???





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View this message in context:  
http://opensips-open-sip-server.1449251.n2.nabble.com/Auth-DB-s-load-credentials-isn-t-grabbing-RPID-info-tp6988909p7010370.html



Sent from the OpenSIPS - Users mailing list archive at Nabble.com.





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Bogdan-Andrei Iancu



OpenSIPS Founder and Developer



OpenSIPS solutions and "know-how"





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Re: [OpenSIPS-Users] Users Digest, Vol 40, Issue 79

[auto-reply from antonio.spirande...@longwave.eu]

Sarò assente fino al 25 Novembre compreso. Per urgenze rivolgersi direttamente 
ad assiste...@longwave.eu o chiamare lo 0522375500. Saluti

I will be out of office till  November 25th 2011.

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[OpenSIPS-Users] using dialog based topology_hiding in failover scenario

[Jayesh Nambiar]

Hi All,
I tested the topology_hiding function in dialog module and it works well.
Now my scenario requires that I use topology_hiding function based on the
carrier where the call is supposed to go. And I use failure_route to
failover between multiple carriers. So the condition is, if 1st carrier
requires topology_hiding I enable it and route the call and if that call
fails, the next carrier might not need topology_hiding enabled so I need to
somehow undo the topology_hiding that I called while routing to the first
carrier.
Moreover, if I call the topology_hiding again for the second carrier, the
contact header gets malformed since the contact header is appended again
and the call fails because of invalid contact header.
One possible solution I thought of was calling the topology_hiding function
in the branch_route, but opensips would not start and logs an error saying
"Command cannot be used in the block".
Anyone with any possible workaround for this? Any help is appreciated !!

Thanks,

--- Jayesh
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