Re: [OpenSIPS-Users] [OpenSIPS-Business] OpenSIPS 2.0 code release available

[Norman Brandinger]

Congratulations !

Norm
On 2/17/11 7:30 AM, Bogdan-Andrei Iancu wrote:
> Hi all,
> 
> After like one year of heavy development, I'm proud to announce the
> first code release of OpenSIPS 2.0 .
> 
> This is a version still under heavy development, with limited capability
> and functionality.
> 
> For all details (what can do, how to use, where to get), visit:
> http://www.opensips.org/Development/NewDesignStage1
> 
> This is not a production-suitable version, but I want to encourage
> people to download, compile, test and analyse it as much as possible.
> Any feedback or comments on the code, approached with used, etc, are
> more than welcome.
> 
> Also, I would like to publicly give all the credits for this work to :
>* Andrei DRAGUS - and...@opensips.org
>* Bogdan IANCU - bog...@opensips.org
>* Vlad PAIU - vladp...@opensips.org
> 
> 
> Waiting for your feedback :)
> 
> Best regards,
> Bogdan
> 


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Re: [OpenSIPS-Users] Getting Error When Configuring OpenSIPS + FreeRadius

[Norman Brandinger]

Please post the current error that you're receiving so that someone on 
the list might be able to point you in the correct direction.

Regards,
Norm

Ahmed Munir wrote:
> Hi, 
>
> Thanks for replying, Norman I've added the line in dictionary.opensips 
> i.e. ATTRIBUTE   Source-IP  214 string, and start 
> freeradius service and make a call and still getting same error as I 
> mentioned on my previous mail.
>
> Kindly assist me to resolve this problem because its quite too long 
> that I'm this problem.
>
>
> Date: Thu, 04 Mar 2010 12:52:49 -0500
> From: Norman Brandinger mailto:n...@goes.com>>
> Subject: Re: [OpenSIPS-Users] Getting Error When Configuring  
>  OpenSIPS
>+   FreeRadius
> To: OpenSIPS users mailling list  <mailto:users@lists.opensips.org>>
> Message-ID: <4b8ff371.6000...@goes.com
> <mailto:4b8ff371.6000...@goes.com>>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> I've got the following (among other attributes) defined in a
> local.dictionary file:
>
> ATTRIBUTE   Source-IP  214 string
>
>
> If it's added to opensips.dictionary, then it no longer needs to be in
> my local.dictionary.
>
> Regards,
> Norm
>
> Bogdan-Andrei Iancu wrote:
> > Hi Ahmed,
> >
> > Do you see in any RADIUS dictionary the Source-IP AVP ? if yes,
> please
> > post here its definition.
> >
> > Regards,
> > Bogdan
> >
> > Ahmed Munir wrote:
> >
> >> Hi,
> >>
> >> Thanks for your reply, I've copied dictionary.opensips to
> >> /etc/freeradius directory and added few lines in it after that
> when I
> >> make the call I'm getting this same error as mentioned in my
> previous
> >> mail earlier, I even copied the dictionary.freeradius to
> >> /etc/clientradius_ng directory as well.
> >>
> >> Lines for dictionary.opensips are listed below;
> >>
> >>  Attributes ###
> >>
> >> ATTRIBUTE Sip-Uri-User 208  string # Proprietary,
> auth_radius
> >> ATTRIBUTE Sip-Group211  string # Proprietary,
> group_radius
> >> ATTRIBUTE Sip-Rpid 213  string # Proprietary,
> auth_radius
> >> ATTRIBUTE SIP-AVP  225  string # Proprietary,
> avp_radius
> >> ATTRIBUTE Sip-Method   101  integer
> >>
> >> ### Service-Type Values ###
> >> VALUE Service-Type   Group-Check  12   # Proprietary,
> group_radius
> >> VALUE Service-Type   Sip-Session  15
> >> VALUE Service-Type   SIP-Caller-AVPs  30   # Proprietary,
> avp_radius
> >> VALUE Service-Type   SIP-Callee-AVPs  31   # Proprietary,
> avp_radius
> >>
> >> ### Sip-Method Values ###
> >> VALUE Sip-Method Undefined  0
> >> VALUE Sip-Method Invite 1
> >> VALUE Sip-Method Cancel 2
> >> VALUE Sip-Method Ack4
> >> VALUE Sip-Method Bye8
> >> VALUE Sip-Method Info   16
> >> VALUE Sip-Method Options32
> >> VALUE Sip-Method Update 64
> >> VALUE Sip-Method Register   128
> >> VALUE Sip-Method Message256
> >> VALUE Sip-Method Subscribe  512
> >> VALUE Sip-Method Notify 1024
> >> VALUE Sip-Method Prack  2048
> >> VALUE Sip-Method Refer  4096
> >> VALUE Sip-Method Other  8192
> >>
> >>
> >> And my dictionary settings are listed below;
> >>
> >> dictionary(freeradius)
> >>
> >> $INCLUDE/opt/freeradius/share/freeradius/dictionary
> >> $INCLUDE/opt/freeradius/etc/raddb/dictionary.opensips
> >>
> >> dictionary(radiusclient)
> >>
> >> ##At the end of the line
> >> $INCLUDE  /opt/freeradius/etc/raddb/dictionary.opensips
> >>
> >>
> >> Kindly tell me from which section I'm facing this problem.
> Waiting for
> >> your reply.
> >>
> >> Date: Fri, 26 Feb 2

Re: [OpenSIPS-Users] Getting Error When Configuring OpenSIPS + FreeRadius

[Norman Brandinger]

I've got the following (among other attributes) defined in a 
local.dictionary file:

ATTRIBUTE   Source-IP  214 string


If it's added to opensips.dictionary, then it no longer needs to be in 
my local.dictionary.

Regards,
Norm

Bogdan-Andrei Iancu wrote:
> Hi Ahmed,
>
> Do you see in any RADIUS dictionary the Source-IP AVP ? if yes, please 
> post here its definition.
>
> Regards,
> Bogdan
>
> Ahmed Munir wrote:
>   
>> Hi,
>>
>> Thanks for your reply, I've copied dictionary.opensips to 
>> /etc/freeradius directory and added few lines in it after that when I 
>> make the call I'm getting this same error as mentioned in my previous 
>> mail earlier, I even copied the dictionary.freeradius to 
>> /etc/clientradius_ng directory as well.
>>
>> Lines for dictionary.opensips are listed below;
>>
>>  Attributes ###
>>
>> ATTRIBUTE Sip-Uri-User 208  string # Proprietary, auth_radius
>> ATTRIBUTE Sip-Group211  string # Proprietary, group_radius
>> ATTRIBUTE Sip-Rpid 213  string # Proprietary, auth_radius
>> ATTRIBUTE SIP-AVP  225  string # Proprietary, avp_radius
>> ATTRIBUTE Sip-Method   101  integer
>>
>> ### Service-Type Values ###
>> VALUE Service-Type   Group-Check  12   # Proprietary, group_radius
>> VALUE Service-Type   Sip-Session  15
>> VALUE Service-Type   SIP-Caller-AVPs  30   # Proprietary, avp_radius
>> VALUE Service-Type   SIP-Callee-AVPs  31   # Proprietary, avp_radius
>>
>> ### Sip-Method Values ###
>> VALUE Sip-Method Undefined  0
>> VALUE Sip-Method Invite 1
>> VALUE Sip-Method Cancel 2
>> VALUE Sip-Method Ack4
>> VALUE Sip-Method Bye8
>> VALUE Sip-Method Info   16
>> VALUE Sip-Method Options32
>> VALUE Sip-Method Update 64
>> VALUE Sip-Method Register   128
>> VALUE Sip-Method Message256
>> VALUE Sip-Method Subscribe  512
>> VALUE Sip-Method Notify 1024
>> VALUE Sip-Method Prack  2048
>> VALUE Sip-Method Refer  4096
>> VALUE Sip-Method Other  8192
>>
>>
>> And my dictionary settings are listed below;
>>
>> dictionary(freeradius)
>>
>> $INCLUDE/opt/freeradius/share/freeradius/dictionary
>> $INCLUDE/opt/freeradius/etc/raddb/dictionary.opensips
>>
>> dictionary(radiusclient)
>>
>> ##At the end of the line
>> $INCLUDE  /opt/freeradius/etc/raddb/dictionary.opensips
>>
>>
>> Kindly tell me from which section I'm facing this problem. Waiting for 
>> your reply.
>>
>> Date: Fri, 26 Feb 2010 15:21:03 +0200
>> From: Andrew Pogrebennyk > >
>> Subject: Re: [OpenSIPS-Users] Getting Error When Configuring OpenSIPS
>>+   FreeRadius`
>> To: OpenSIPS users mailling list > >
>> Message-ID: <4b87cabf.9000...@portaone.com 
>> >
>> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>>
>> On 26.02.2010 14:33, Ahmed Munir wrote:
>> 
>>> Feb 26 14:47:47 rose /usr/local/sbin/opensips[25988]:
>>> ERROR:aaa_radius:rad_avp_add: failure
>>> Feb 26 14:47:47 rose /usr/local/sbin/opensips[25988]:
>>> ERROR:acc:acc_aaa_request: failed to add Source-IP, 13
>>>
>>>
>>> And I also check table radacct in mysql database, no records are 
>>>   
>> inserted
>> 
>>> into it.
>>>   
>> I think this means an incorrect RADIUS dictionary. You should verify
>> that the extra attributes you have defined are present there.
>>
>> --
>> Sincerely,
>> Andrew Pogrebennyk
>>
>>  
>> --
>> Regards,
>>
>> Ahmed Munir
>>
>>
>>
>>
>>
>>
>> 
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>   
>> 
>
>
>   


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Re: [OpenSIPS-Users] [RFC] What repo to use for 2.0 ?

[Norman Brandinger]

I have found that the speed of mercurial seems to be an issue with 
larger sized project.

For example, try downloading adium: hg clone http://hg.adium.im/adium

My vote is for git.

Regards,
Norm

Bogdan-Andrei Iancu wrote:
> Hi everyone,
>
> As we are getting closed to first working code for OpenSIPS 2.0 (a proto 
> core including config, transport, threading, reactor and dispatcher), we 
> will upload the code the public SF repo.
>
> I would like to get some feedback from you about what repo type to use - 
> as we have the opportunity to start a fresh new piece of code, we have 
> the option to use a different repo than SVN.
>
> SF offers the following options
> - SVN
> - git
> - mercurial
> - bazaar
>
> Should we keep SVN ? pros ? minuses ? something much better ?
>
>
> Regards,
> Bogdan
>
> PS: should we create a poll on the project web site ?
>
>   


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Re: [OpenSIPS-Users] Iñaki Baz Castillo all wisecrac k but no knowledge

[Norman Brandinger]

Rather than have this thread continue its downward spiral, I would like to 
suggest the following book:


Thanks to Flavio E. Goncalves, a new edition of "Building Telephony 
Systems with OpenSIPS" is now available.

It covers the latest stable release, the OpenSIPS 1.6, updating existing 
topics (NAT traversal, accounting, etc), but also approaching new 1.6 
specific technologies (data caching, dialog usage, etc):

The book will teach you how to build scalable and robust telephony 
systems using SIP:

- Build a VoIP Provider based on the SIP Protocol
- Cater to scores of subscribers efficiently with a robust telephony 
system based in pure SIP
- Gain a competitive edge using the most scalable VoIP technology
- Learn how to avoid pitfalls using precise billing
- Packed with rich practical examples and case studies on the latest 
OpenSIPS version 1.6

More on : 
http://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book

Regards,
Bogdan


Regards,
Norm 


Alex Balashov wrote:
> I think Iñaki was trying to tell you politely that your question, was 
> you formulated it, is somewhat naive.  This is mainly so because you 
> are looking for a quick answer (as though it could be devised from 
> reading the cookbook) to a potentially rather complicated, larger 
> architectural issues.
>
> Certain questions just can't be tackled that way, and certain things 
> can't be learned nor solved in one day, or with a few hours of effort 
> to find a "recipe."
>
> On 01/21/2010 07:18 AM, Slot Zero wrote:
>
>   
>> Well at least he shouldn't have been the one to start it. as far as 
>> personality trait well you got two of us; the only difference is that I need 
>> loud mouthed wisecracking trigger. So I am still better as far as 
>> personality goes. If he is such an expert he could have at least sent a 
>> decent reply to my request. I think that is pretty much what I am asking you 
>> experts.
>>
>>
>>
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>
>
>   


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Re: [OpenSIPS-Users] RES: Billing System

[Norman Brandinger]

Congratulations,

Norm

Steven E. Ames wrote:
> Congratulations! Just ordered a copy.
>
> -Original Message-
> From: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org]on Behalf Of AsteriskGuide
> Sent: Wednesday, January 20, 2010 12:41 PM
> To: 'OpenSIPS users mailling list'
> Subject: [OpenSIPS-Users] RES: Billing System
>
>
> A new book about OpenSIPS was released today
>
> http://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book
>
> Sample chapter: http://www.packtpub.com/article/installation-of-opensips-1.6
>
> Regards,
>
> Flavio E. Goncalves
>
> -Mensagem original-
> De: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] Em nome de sergio
> Enviada em: Wednesday, January 20, 2010 3:33 PM
> Para: users@lists.opensips.org
> Assunto: [OpenSIPS-Users] Billing System
>
> Need a Billing System for Commercial OpenSIPS.
>
> Someone could show me some? I need some good book about OpenSIPS.
>
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>
>
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>
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>
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Re: [OpenSIPS-Users] Polycom 650 can't Register behind NAT

[Norman Brandinger]

Try changing the port that the Polycom is listening on from 5060 to 
something else.

Regards,
Norm

osiris123d wrote:
> I am sure this is not a OpenSIPs issue, but I was just wondering if people
> have run into this and what they have done to resolve this.  I currently
> have been testing my OpenSIPs config with Counterpaths Bria client and also
> the SJphone.  They don't have any issues registering behind NAT.  I just
> purchases a Polycom 650 phone and I cannot get it to register.  When I do an
> ngrep I see that the phone is wanting OpenSIPS to talk back to it via port
> 5060.  When I do an ngrep with Bria or SJphone I see that when OpenSIPS
> replies back to the softphone it is not on port 5060.  
>
> Here is the ngrep showing a polycom trying to register
>
> U 75.65.x.x:5060 -> 66.80.x.x:5060
> REGISTER sip:sip..com SIP/2.0.
> Via: SIP/2.0/UDP 192.168.0.100;branch=z9hG4bK484090eaC6F4185B.
> From: "DL" ;tag=2AED749F-64CCEB50.
> To: .
> CSeq: 1 REGISTER.
> Call-ID: 86e5160c-a73b99fd-e4c3b...@192.168.0.100.
> Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS,
> INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPD
> ATE, REFER".
> User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.1.2.0392.
> Accept-Language: en.
> Max-Forwards: 70.
> Expires: 3600.
> Content-Length: 0.
> .
>
>
> U 66.80.x.x:5060 -> 75.65.x.x:5060
> SIP/2.0 401 Unauthorized.
> Via: SIP/2.0/UDP
> 192.168.0.100;branch=z9hG4bK484090eaC6F4185B;received=75.65.x.x.
> From: "DL" ;tag=2AED749F-64CCEB50.
> To: ;tag=eb4f595d3de5f0bd1aa7f6687dc18bd8.e526.
> CSeq: 1 REGISTER.
> Call-ID: 86e5160c-a73b99fd-e4c3b...@192.168.0.100.
> Server:  SIP Proxy.
> Content-Length: 0.
> .
>
>
> U 75.65.x.x:5060 -> 66.80.x.x:5060
> REGISTER sip:sip..com SIP/2.0.
> Via: SIP/2.0/UDP 192.168.0.100;branch=z9hG4bK8e1a28414313B72.
> From: "DL" ;tag=2AED749F-64CCEB50.
> To: .
> CSeq: 2 REGISTER.
> Call-ID: 86e5160c-a73b99fd-e4c3b...@192.168.0.100.
> Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS,
> INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPD
> ATE, REFER".
> User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.1.2.0392.
> Accept-Language: en.
> Max-Forwards: 70.
> Expires: 3600.
> Content-Length: 0.
>
>   


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Re: [OpenSIPS-Users] CDRTool query to media_sessions table

[Norman Brandinger]

#  $Id: mysql-dialup.conf,v 1.1 2006/08/31 22:10:41 pnixon Exp $
#
# FreeRADIUS "dialup" SQL Queries for the MySQL Dialect
# Safe characters list for sql queries. Everything else is replaced
# with their mime-encoded equivalents.
# The default list should be ok
#safe-characters = 
"@abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ0123456789.-_: /"

Regards,
Norm

Dan Pascu wrote:
> On 8 Sep 2009, at 15:01, Jeff Pyle wrote:
>
>   
>> Dan,
>>
>> Unfortunately I'm not finding any references to escapaing characters  
>> in
>> Freeradius.  I realize this isn't your home turf, but having said  
>> that, do
>> you have any pointers?
>> 
>
> There is some option in sql.conf, but I don't recall the option name.  
> Look in the freeradius provided sample sql.conf, I'm sure the comments  
> will help you find it.
>
>   
>> Thanks,
>> Jeff
>>
>>
>>
>> On 9/7/09 8:02 AM, "Dan Pascu"  wrote:
>>
>> 
>>> On 5 Sep 2009, at 23:01, Jeff Pyle wrote:
>>>
>>>   
 Hello,

 On CDRTool v6.8.0 and Mediaproxy 2.3.6 I notice sometimes I get “No
 information available” when I click on the Media Information link.

 In one particular example, the query looks for
 “SDjui5c99-324092c7=2B1=2B2b6b0008=2B96cdd538” but the database has
 “SDjui5c99-324092c7+1+2b6b0008+96cdd538”.  Is there a way to
 encourage CDRTool to query with the non-modified version of the
 characters?  Or, is another solution preferable?
 
>>> CDRTool queries with the information available from the radius
>>> accounting table. You should configure your radius server (I presume
>>> freeradius) not to escape those characters.
>>>
>>> The difference appears from the fact that the information is inserted
>>> unmodified in the media sessions table, but the radius server escapes
>>> certain characters when inserting it in the accounting table.
>>>
>>>   
 - Jeff
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>>> --
>>> Dan
>>>
>>>
>>>
>>>
>>> ___
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>>>   
>> ___
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>> 
>
>
> --
> Dan
>
>
>
>
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Re: [OpenSIPS-Users] stop accounting

[Norman Brandinger]

One solution is to take a look at mediaproxy from AG Projects: 
http://mediaproxy.ag-projects.com

Regards,
Norm

kaiser wrote:
> Dear sir,
>
> The opensips script coded with radius accouting,
> if the call is terminated with BYE and 200 OK, radius stop accouinging
> will be sent out.
>
> I try a call without 200 OK for BYE, I made an UA discoonected with LAN.
>
> acc module seems fail in radius accouting, no radius packert send out.
> How do I fix it?
>
>
> kk
>
>
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Re: [OpenSIPS-Users] error in config file - invalid routing table number

[Norman Brandinger]

I have some (very) large config files.  At some point increasing the 
following variable helped (also in config.h):

#define PKG_MEM_POOL_SIZE 4*1024*1024

To answer your previous question, no, the route table numbers do not 
have to be sequential.

Regards,
Norm

Patrick wrote:
> Would having more than 100 route tables mean I might run into issues 
> with private memory?  The is no command argument to change that 
> (unlike shared mem)...
>
>
> On May 19, 2009, at 4:57 PM, Norman Brandinger wrote:
>
> Take a look at config.h, specifically the following setting:
>
> #define RT_NO 100
>
> You'll have to recompile after changing the value.
>
> Regards,
> Norm
>
> Patrick wrote:
>> Hello,
>> I've run into a limitation in the config file.   I'm defining  
>> routes tables as
>>
>> route[1] {
>>...
>> }
>>
>> 
>>
>> route[100] {
>>...
>> }
>>
>>
>> opensips: CRITICAL:core:yyerror: parse error in config file, line  
>> 2000, column 1-2: invalid routing table number
>>
>>
>> Is there a limit of only 1-99 for routing tables?   Also, do these  
>> numbers have to be sequential?
>>
>>
>>
>> Thanks,
>>
>> Patrick
>>
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>>
>
>
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Re: [OpenSIPS-Users] error in config file - invalid routing table number

[Norman Brandinger]

Take a look at config.h, specifically the following setting:

#define RT_NO 100

You'll have to recompile after changing the value.

Regards,
Norm

Patrick wrote:
> Hello,
>  I've run into a limitation in the config file.   I'm defining  
> routes tables as
>
> route[1] {
> ...
> }
>
> 
>
> route[100] {
> ...
> }
>
>
> opensips: CRITICAL:core:yyerror: parse error in config file, line  
> 2000, column 1-2: invalid routing table number
>
>
> Is there a limit of only 1-99 for routing tables?   Also, do these  
> numbers have to be sequential?
>
>
>
> Thanks,
>
> Patrick
>
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Re: [OpenSIPS-Users] Paid Consultation Request

[Norman Brandinger]

There have been a couple of people (myself included) that have taken 
advantage of this 'bug' / 'feature'.  Probably should get some input 
from Bogdan as to whether it's something to be used or avoided.

Regards,
Norm

Geoffrey Mina wrote:
> Either a 'bug' or a 'feature'... whatever you want to call it :)  It
> worked as I _expected_, but not as intended.
>
> thanks,
> Geoff
>
> On Wed, Feb 11, 2009 at 6:18 PM, Iñaki Baz Castillo  wrote:
>   
>> El Jueves, 12 de Febrero de 2009, escribió:
>> 
>>> I have addressed the ds_select_domain() usage issue after re-reading
>>> the documentation upon your guidance.  The last config I sent over is
>>> using ds_next_dst() in the failure_route.
>>>
>>> I am not sure why OpenSIPS didn't have a startup error, but it
>>> appeared to have started cleanly...
>>>   
>> As you can see in module "dispatcher.c":
>>
>>  {"ds_select_domain", (cmd_function)w_ds_select_domain, 2, fixup_igp_igp, 0,
>> REQUEST_ROUTE},
>>
>> So "ds_select_domain" can be invoked just in REQUEST_ROUTE. I'm suspecting
>> that since the funcion is called into a ROUTE block (called from a
>> FAILURE_ROUTE block) the check is "bypassed". I'd consider it a bug...
>>
>> --
>> Iñaki Baz Castillo
>>
>> 
>
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Re: [OpenSIPS-Users] Paid Consultation Request

[Norman Brandinger]

Geoff,

The old OpenSER/Kamailio site has a "business" mailing list that is 
dedicated to this type of topic.  Seems like a good idea to create a 
similar mailing list for the OpenSIPS folks.

I have to agree with Adrian's recent comments in regard to narrowing 
down the scope of your problem/request.

There are no doubt people out there that will take your money and not 
provide the value you're looking for.  You may not be happy with some of 
the responses to your request, but those that made the responses are 
really trying to help you look out for your own interests.

Since you're a network engineer, you should know that; 1) trying to load 
balance SIP traffic, and 2) using Asterisk to process the media of the 
load balanced traffic; isn't necessarily trivial.

If you've got something up and running in a lab environment error free, 
I congratulate you as you probably don't need any outside help.  
However, if you're throwing errors in your lab environment, as Adrian 
suggested, you might want to consider detailing the problem and asking 
for some help.

Regards,
Norm


Geoffrey Mina wrote:
> There is no NAT
> There is no Firewall (running local IPTables)
> There are no 'users'
> There are no 'phones'
> There is a single upstream Carrier
> There ARE a plurality of asterisk based IVR systems which are
> processing the media.  All configured identically, all on the same
> network.
>
> I have been doing SIP for years, just not using OpenSIPS as my
> Loadbalancer/Gateway of choice.
> I have been writing networked applications for 5 times the length of
> my SIP experience... an OpenSIPS N00B perhaps, but a network/SIP N00B
> I think not.
>
> I know enough to know that my configuration is NOT complex.  I am
> shocked to hear that so many people on this list think this type of
> OpenSIPS configuration is complicated to the point that an experienced
> user couldn't take a look at a simple 150 line configuration file and
> point out any glaring issues and identify some best practice
> scenarios.
>
> Hopefully there are some experienced users out there who are
> interested in contributing something useful instead of this dribble
> about how smart you are and how complex your OpenSIPS configuration
> is.
>
> -Geoff
>
>
>
> On Wed, Feb 11, 2009 at 2:25 PM, John Rose  wrote:
>   
>> Often people and companies underestimate the complexities of a SIP proxy
>> installation. There are many variables particularly ones with carriers,
>> phones, NAT's, firewalls...  Many assume it is just configuration. A lot of
>> it is but unless you have worked it in depth you won't know.
>>
>> Here is a free "blessing" - Seek and you shall find my dear OpenSIPS N00B.
>> Let the light guide you through the valley of SIP and networking
>> complexities :)
>>
>> John
>>
>> 
>>> -Original Message-
>>> From: users-boun...@lists.opensips.org [mailto:users-
>>> boun...@lists.opensips.org] On Behalf Of Geoffrey Mina
>>> Sent: Wednesday, February 11, 2009 9:47 AM
>>> To: users@lists.opensips.org
>>> Subject: [OpenSIPS-Users] Paid Consultation Request
>>>
>>> Hello,
>>> I am looking for anyone who would consider themselves an 'expert' in
>>> the field of OpenSIPS.  My company is launching an OpenSIPS deployment
>>> to front-end all the SIP traffic entering our network.  I would like
>>> to have someone experienced look over my config to give it the
>>> proverbial 'blessing'.
>>>
>>> We run a very high profile network and I can't afford to have any
>>> minor misconfigurations or problems cause issues down the road.
>>>
>>> I would be willing to pay $150/hr (USD) via PayPal.  I am guessing I
>>> will need 2(ish) hours, and IM/MSN/Skype chat would be the best.
>>>
>>> If anyone is interested, please let me know.
>>> Thanks,
>>> Geoff
>>>
>>> p.s. we are primarily using the dispatcher module, so any interested
>>> party should know that one inside and out!
>>>
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Re: [OpenSIPS-Users] Call recording

[Norman Brandinger]

Keep a close eye on the number of file descriptors in use.  Have seen 
cases where the os ran out of descriptors using this tool.

Norm

Brett Nemeroff wrote:
> I know this isn't really "call recording" but I've used a really cool 
> tool called "pcapsipdump" that dumps each call, rtp and SIP into it's 
> own pcap file. 
>
> I made some modifcations so it doesn't log OPTIONS dialogs. If you're 
> very resourceful, you can make it dump a recording of sorts. And it'd 
> be pretty passive, if that's desirable.
>
> -Brett
>
>
> On Sun, Jan 25, 2009 at 5:12 AM, Jimmy Svensson  > wrote:
>
> Hi!
>  
> I'm looking for a pure call recording solution that generates
> pcap-files of the data from one specific call. I've been looking
> at using the a merged OpenSIPS+rtpproxy solution to achieve this.
> The rtpproxy would not be proxying any RTP traffic in this case,
> just terminating and recording. Does anyone have experience with
> this kind of solution?
>  
> Now I found that rtpproxy seems to be replaced by mediaproxy but I
> can't find any call recording features on the mediaproxy. Does
> anyone know if there are any plans to implement call recording in
> mediaproxy or do I have to go for the rtpproxy?
>  
> Thanks!
> /Jimmy
>
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Re: [OpenSIPS-Users] client.conf file for CDRTool

[Norman Brandinger]

The reference in the opensips.cfg file should generally be to 
radiusclient-ng/radiusclient.conf

Regards,
Norm


Gabriel Bermudez wrote:
> Hi everyone,
>
> I'm following CDRTool's install guide from 
> http://cdrtool.ag-projects.com/wiki/Install.  But I don't know what the 
> contents for the
>
> /etc/opensips/radius/client.conf
>
> file referenced on the guide show be.  It's used in the configurations 
> files of opensips.cfg and mediaproxy's config.ini, but I can't find an 
> example of it.  Can some one point me in the right direction or the 
> basic information that this file has to have.  I'm putting all the 
> software (opensips, cdrtool, mediaproxy, freeradius and mysql) on the 
> same box for testing, but for production I will split this software on 2 
> or more servers.  I hope some one knows this info so thanks for your 
> help in advance.
>
> Regards,
>
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Re: [OpenSIPS-Users] FREERADIUS problem

[Norman Brandinger]

Take a look at radiusclient-ng.  The two files of interest are: 
radiusclient.conf and servers.

Regards,
Norm


ASHWINI NAIDU wrote:
> hi all,
>
>
>I have freeradius-1.1.3 running on debain machine. i have 
> integrated it with openser-1.3.4. when i try to make a call  i get the 
> following errors
>
> Jan 23 13:27:44 localhost /sbin/openser[9910]: 
> ERROR:acc:acc_rad_request: radius-ing failed
> Jan 23 13:28:17 localhost /sbin/openser[9906]:* rc_send_server: no 
> reply from RADIUS server jp.local:1813*
> Jan 23 13:28:17 localhost /sbin/openser[9906]:* 
> ERROR:acc:acc_rad_request: radius-ing failed*
> Jan 23 13:29:29 localhost /sbin/openser[9913]: ERROR:tm:w_t_relay: 
> t_forward_nonack failed
>
>
> myhostname is jp.
>
> I wanted to know where i can set the ip address of the radius server 
> so that it can access the accounting server of freeradius.
>
> -- 
> Thankin You,
> Ashwini BR Naidu
> 
>
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Re: [OpenSIPS-Users] Problem When Loading auth_radius

[Norman Brandinger]

Hi,

radiusclient.conf should have an entry that points to a dictionary. It's 
this dictionary that should be able to locate the attribute in question 
(either directly or though another included dictionary).

Regards,
Norm

ASHWINI NAIDU wrote:
> But i have made radiusclient.conf file available to it. what else has 
> to be added to remove this error.
>
> On Tue, Jan 13, 2009 at 6:14 PM, Norman Brandinger  <mailto:n...@goes.com>> wrote:
>
> The attribute you're looking for can be found in radiusclient-ng.
>
> Regards,
> Norm
>
> ASHWINI NAIDU wrote:
>
> hi all,
>
> When i try to load the module auth_radius and start openser i
> am getting the following errors
>
>
> Jan 13 23:35:32 localhost openser: INFO:core:init_tcp: using
> epoll_lt as the TCP io watch method (auto detected)
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> NOTICE:core:main: version: openser 1.3.4-notls (i386/linux)
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> INFO:core:main: using 128 Mb shared memory
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> INFO:core:main: using 1 Mb private memory per process
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> INFO:sl:mod_init: Initializing StateLess engine
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> INFO:tm:mod_init: TM - initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> INFO:maxfwd:mod_init: initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> INFO:usrloc:ul_init_locks: locks array size 512
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> INFO:registrar:mod_init: initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> INFO:textops:mod_init: initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> WARNING:permissions:parse_config_file: file not found:
> //etc/openser/permissions.allow
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> WARNING:permissions:mod_init: default allow file
> (//etc/openser/permissions.allow) not found => empty rule set
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> WARNING:permissions:parse_config_file: file not found:
> //etc/openser/permissions.deny
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> WARNING:permissions:mod_init: default deny file
> (//etc/openser/permissions.deny) not found => empty rule set
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> INFO:permissions:init_trusted: db_url parameter of permissions
> module not set, disabling allow_trusted
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> INFO:permissions:init_addresses: db_url parameter of
> permissions module not set, disabling allow_addr
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> INFO:avpops:avpops_init: initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> INFO:xlog:mod_init: initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> INFO:acc:mod_init: initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> INFO:auth:mod_init: initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> INFO:auth_db:mod_init: initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> INFO:auth_radius:mod_init: initializing...
> *Jan 13 23:35:32 localhost /sbin/openser[18155]:
> ERROR:auth_radius:mod_init: auth_radius: can't get code for
> the Digest-Response attribute
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> ERROR:core:init_mod: failed to initialize module auth_radius
> Jan 13 23:35:32 localhost /sbin/openser[18155]:
> ERROR:core:main: error while initializing modules
> *
> What does it mean by "*can't get code for the Digest-Response
> attribute"
>
> how can i make it available for it.
> *
> -- 
> Thankin You,
> Ashwini BR Naidu
> 
> 
>
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>
>
>
>
>
> -- 
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Re: [OpenSIPS-Users] Problem When Loading auth_radius

[Norman Brandinger]

The attribute you're looking for can be found in radiusclient-ng.

Regards,
Norm

ASHWINI NAIDU wrote:
> hi all,
>
> When i try to load the module auth_radius and start openser i am 
> getting the following errors
>
>
> Jan 13 23:35:32 localhost openser: INFO:core:init_tcp: using epoll_lt 
> as the TCP io watch method (auto detected)
> Jan 13 23:35:32 localhost /sbin/openser[18155]: NOTICE:core:main: 
> version: openser 1.3.4-notls (i386/linux)
> Jan 13 23:35:32 localhost /sbin/openser[18155]: INFO:core:main: using 
> 128 Mb shared memory
> Jan 13 23:35:32 localhost /sbin/openser[18155]: INFO:core:main: using 
> 1 Mb private memory per process
> Jan 13 23:35:32 localhost /sbin/openser[18155]: INFO:sl:mod_init: 
> Initializing StateLess engine
> Jan 13 23:35:32 localhost /sbin/openser[18155]: INFO:tm:mod_init: TM - 
> initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]: INFO:maxfwd:mod_init: 
> initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]: 
> INFO:usrloc:ul_init_locks: locks array size 512
> Jan 13 23:35:32 localhost /sbin/openser[18155]: 
> INFO:registrar:mod_init: initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]: INFO:textops:mod_init: 
> initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]: 
> WARNING:permissions:parse_config_file: file not found: 
> //etc/openser/permissions.allow
> Jan 13 23:35:32 localhost /sbin/openser[18155]: 
> WARNING:permissions:mod_init: default allow file 
> (//etc/openser/permissions.allow) not found => empty rule set
> Jan 13 23:35:32 localhost /sbin/openser[18155]: 
> WARNING:permissions:parse_config_file: file not found: 
> //etc/openser/permissions.deny
> Jan 13 23:35:32 localhost /sbin/openser[18155]: 
> WARNING:permissions:mod_init: default deny file 
> (//etc/openser/permissions.deny) not found => empty rule set
> Jan 13 23:35:32 localhost /sbin/openser[18155]: 
> INFO:permissions:init_trusted: db_url parameter of permissions module 
> not set, disabling allow_trusted
> Jan 13 23:35:32 localhost /sbin/openser[18155]: 
> INFO:permissions:init_addresses: db_url parameter of permissions 
> module not set, disabling allow_addr
> Jan 13 23:35:32 localhost /sbin/openser[18155]: 
> INFO:avpops:avpops_init: initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]: INFO:xlog:mod_init: 
> initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]: INFO:acc:mod_init: 
> initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]: INFO:auth:mod_init: 
> initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]: INFO:auth_db:mod_init: 
> initializing...
> Jan 13 23:35:32 localhost /sbin/openser[18155]: 
> INFO:auth_radius:mod_init: initializing...
> *Jan 13 23:35:32 localhost /sbin/openser[18155]: 
> ERROR:auth_radius:mod_init: auth_radius: can't get code for the 
> Digest-Response attribute
> Jan 13 23:35:32 localhost /sbin/openser[18155]: ERROR:core:init_mod: 
> failed to initialize module auth_radius
> Jan 13 23:35:32 localhost /sbin/openser[18155]: ERROR:core:main: error 
> while initializing modules
> *
> What does it mean by "*can't get code for the Digest-Response attribute"
>
> how can i make it available for it.
> *
> -- 
> Thankin You,
> Ashwini BR Naidu
> 
>
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Re: [OpenSIPS-Users] presence not starting

[Norman Brandinger]

Hi David,

Try loading the new "signaling" modules.

loadmodule "signaling.so";

Regards,
Norm

David Villasmil wrote:
> Hello all,
>
>  I've compiled opensips 1.5 with presence support, this is what my 
> MAkefile looks like:
>
> # if not set on the cmd. line or the env, exclude this modules:
> exclude_modules?= jabber cpl-c xmpp rls mi_xmlrpc \
> db_postgres db_unixodbc db_oracle db_berkeley \
> avp_radius auth_radius group_radius uri_radius \
> osp perl snmpstats perlvdb seas peering carrierroute mmgeoip \
> pua pua_bla pua_mi pua_usrloc pua_xmpp \
> ldap h350
>
> I've taken out presence, presence_mwi, presence_xml and xcap_client 
> (besides others like db_mysql)
>
> I compile and everything looks good, no errors, but when I start 
> OpenSIPS I get:
>
> Nov  9 09:46:12 chef-os-01 /sbin/opensips[13068]: 
> NOTICE:presence:mod_init: initializing module ...
> Nov  9 09:46:12 chef-os-01 /sbin/opensips[13068]: 
> DBG:core:find_cmd_export_t:  not found 
> Nov  9 09:46:12 chef-os-01 /sbin/opensips[13068]: 
> ERROR:presence:load_sig_api: can't import load_sig
> Nov  9 09:46:12 chef-os-01 /sbin/opensips[13068]: 
> ERROR:presence:mod_init: can't load signaling functions
> Nov  9 09:46:12 chef-os-01 /sbin/opensips[13068]: ERROR:core:init_mod: 
> failed to initialize module presence
> Nov  9 09:46:12 chef-os-01 /sbin/opensips[13068]: ERROR:core:main: 
> error while initializing modules
> Nov  9 09:46:12 chef-os-01 /sbin/opensips[13068]: 
> DBG:presence_xml:destroy: start
> Nov  9 09:46:12 chef-os-01 /sbin/opensips[13068]: 
> NOTICE:presence:destroy: destroy module ...
>
>
> Is there any other module I need?
>
> Thanks in advance!
>
>
> cheers
>
> David
> 
>
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