date:20090908

Re: [OpenSIPS-Users] questions about log?

2009-09-08 Thread Uwe Kastens

Hi,

you can define the syslog facility in opensips.cfg. After that you can
put the log to any location.

BR

Uwe



ASHWINI NAIDU schrieb:
> By default the logging of opensips will be done in */var/log/syslog* in
> debian systems and */var/log/messages* in redhat based systems
> 
> 2009/9/9 zhangchao1  >
> 
> 
> Hello everybody, dose anyone know where the log file is?
> 
> 
> "中国制造",讲述中国60年往事
> 
> ___
> Users mailing list
> Users@lists.opensips.org 
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 
> 
> 
> -- 
> Thanking You,
> Ashwini BR Naidu
> 
> 
> 
> 
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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Re: [OpenSIPS-Users] questions about log?

2009-09-08 Thread ASHWINI NAIDU

By default the logging of opensips will be done in */var/log/syslog* in
debian systems and */var/log/messages* in redhat based systems

2009/9/9 zhangchao1 

>
> Hello everybody, dose anyone know where the log file is?
>
>
> --
> "中国制造",讲述中国60年往事
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


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Re: [OpenSIPS-Users] Regarding Uac_replace_from

2009-09-08 Thread ASHWINI NAIDU

Hi Bogdan,

   That problem is solved by using the diverter_avp modparam in call control
module. thanks for replying

On Tue, Sep 8, 2009 at 8:37 PM, Bogdan-Andrei Iancu
wrote:

> Hi Ashwini,
>
> I do not know about the Call-Control module too much, but can you also
> push to it the new FROM value from the script ? or you cannot influence
> the info that is sent to CC module ?
>
> Regards,
> Bogdan
>
> ASHWINI NAIDU wrote:
> > Hi Bogdan,
> >
> > The whole scenario is i want the new from(after uac_replace_from)
> > header value to be used by the call controller. is there any way i can
> > do it.
> >
> > On Fri, Sep 4, 2009 at 6:14 PM, Bogdan-Andrei Iancu
> > mailto:bog...@voice-system.ro>> wrote:
> >
> > Hi Ashwini,
> >
> > The variables that provide information from the SIP message (like
> $fu)
> > get the values from the original received message, disregarding
> > whatever
> > changes you do .The only exception is $ru (and family) vars.
> >
> > Regards,
> > Bogdan
> >
> >
> > ASHWINI NAIDU wrote:
> > > Hi all,
> > >
> > >
> > >  I am trying to replace the from header using the
> > > *uac_replace_from* in uac module. the thing i noticed the
> > replacement
> > > is happening in the* sip headers*. After using the
> > *uac_replace_from*
> > > and try to print the *$fu and $fU *the previous values before
> > > replacement can be seen . can anyone tell me what is the problem?
> > >
> > >
> > > --
> > > Thanking You,
> > > Ashwini BR Naidu
> > >
> >
> 
> > >
> > > ___
> > > Users mailing list
> > > Users@lists.opensips.org 
> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > >
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org 
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> >
> >
> > --
> > Thanking You,
> > Ashwini BR Naidu
> > 
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



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[OpenSIPS-Users] questions about log?

2009-09-08 Thread zhangchao00001


Hello everybody, dose anyone know where the log file is?___
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[OpenSIPS-Users] Newbie To OpenSIPS

2009-09-08 Thread Kemp, Larry

First time compiling OpenSIPS; installing on CentOS. Downloaded source as well 
as RPM's from http://centos.leurent.eu. I was wondering if any of you that have 
already done this successfully and had any resources or notes from your 
deployment that you might be willing to share, detailed how-to info for 
database and making/installing gui, or could recommend which of the many books 
that exist to read on this. Info seems splintered since the OpenSER split. 
To date I have:
Compiled & started /usr/local/sbin/opensips
   Listening on 
 udp: 127.0.0.1 [127.0.0.1]:5060
 udp: 172.20.30.184 [172.20.30.184]:5060
 tcp: 127.0.0.1 [127.0.0.1]:5060
 tcp: 172.20.30.184 [172.20.30.184]:5060
   Aliases: 
 tcp: localhost:5060
 tcp: localhost.localdomain:5060
 tcp: hostname:5060
 tcp: hostname.domain.com:5060
 udp: localhost:5060
 udp: localhost.localdomain:5060
 udp: hostname:5060
 udp: hostname.domain.com:5060
Any guidance would be appreciated from the community that has done this 
already. I installed current revision.

Lars


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Re: [OpenSIPS-Users] Newbie To OpenSIPS

2009-09-08 Thread Alex Balashov

Kemp, Larry wrote:
> Can OpenSIPS be used as a Session Border controller sitting at my edge 
> passing and receiving SIP traffic to others I SIP peer with? If not, what 
> other open-source would anyone suggest to act as SBC's? I too would rather do 
> it via open-source and x86 or 64bit chip, less costly. Thanks.

This is mostly a semantic issue.

OpenSIPS is not an SBC in the way that commercial SBCs are SBCs, and 
lacks a number of key aspects, including ASIC-assisted processing in the 
higher-end ones.

But it can be used for subscriber or carrier-facing edge duty in the 
same way SBCs are often used (unnecessarily so).

-- 
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: [OpenSIPS-Users] Newbie To OpenSIPS

2009-09-08 Thread Brett Nemeroff

It can do some things like a SBC, but it isn't an SBC at heart.
What functionality are you looking for specifically?

-Brett


On Tue, Sep 8, 2009 at 1:05 PM, Kemp, Larry wrote:

> Can OpenSIPS be used as a Session Border controller sitting at my edge
> passing and receiving SIP traffic to others I SIP peer with? If not, what
> other open-source would anyone suggest to act as SBC's? I too would rather
> do it via open-source and x86 or 64bit chip, less costly. Thanks.
>
> Lars
>
>
> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
> Sent: Tuesday, September 08, 2009 1:50 PM
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
>
> Hi,
>
> Good question and not easy to answer. ACME is expensive AND you will
> need somebody to configure it in a way you will need it. So as an
> redundant option your talking about 100-150K.
>
> To buy a big name won't prevent you from implementing, bugsearching.
>
>
> My personal opinion: Take less money, look for good consultants and try
> it with opensource.
>
>
> BR
>
> Uwe
>
>
> Kemp, Larry schrieb:
> > Certainly. If I just wanted to pass my SIP to other carriers or have them
> connect to my SIP customers could I use OpenSIPS for that alone, or would I
> still need some other sort of session border controller?
> >
> > Larry Kemp
> >
> >
> > -Original Message-
> > From: users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
> > Sent: Tuesday, September 08, 2009 1:31 PM
> > To: OpenSIPS users mailling list
> > Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
> >
> > Hi,
> >
> > Yes, this could be an option. But a very expensive one :-)
> >
> > BR
> >
> > Uwe
> >
> >
> > Kemp, Larry schrieb:
> >> So I would use OpenSIPS behind say like an Acme Packet
> http://www.acmepacket.com/ Session Border Controller or a MetaSwitch
> http://www.metaswitch.com/ connecting to the PSTN, then use OpenSIPS to
> talk to those Gateways via SIP and route my customer's VOIP traffic from
> their Asterisk PBX's to those devices that speak SIP, right?
> >>
> >> Lars
> >>
> >>
> >> -Original Message-
> >> From: users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
> >> Sent: Tuesday, September 08, 2009 12:19 PM
> >> To: OpenSIPS users mailling list
> >> Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
> >>
> >> Hi,
> >>
> >> Opensips is a SIP router not a media gateway. So far you will need
> >> something that will take care of comvert TDM/PSTN to sip.
> >>
> >> There should be lot of examples for this kind of setup.
> >>
> >> BR
> >>
> >> Uwe
> >>
> >>
> >>
> >> Kemp, Larry schrieb:
> >>> Thank you for the response UK. We are looking at using OpenSIPS on
> Linux
> >>> to act as a SIP Router to handoff SIP calls to other carriers that are
> >>> also using SIP, and explore the use of the agent registration server
> >>> functionality. I have already previously installed and debugged
> Asterisk
> >>> on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP
> >>> phones. My understanding is that you can route your Asterisk systems to
> >>> OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or
> >>> SS7 carriers, or your own Class 4 or 5 environment. Maybe I am
> >>> misunderstanding as far as what OpenSIPS can actually do? Thanks.
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>> -Original Message-
> >>> From: users-boun...@lists.opensips.org
> >>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
> >>> Sent: Tuesday, September 08, 2009 11:32 AM
> >>> To: OpenSIPS users mailling list
> >>> Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
> >>>
> >>>
> >>>
> >>> Hi Lars,
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
>  Any guidance would be appreciated from the community that has done
>  this already. I installed current revision.
> >>>
> >>>
> >>> Could be helpfull to know what you want to do with opensips :-)
> >>>
> >>>
> >>>
> >>> BR
> >>>
> >>>
> >>>
> >>> Uwe
> >>>
> >>>
> >>>
> >>> --
> >>>
> >>>
> >>>
> >>> kiste lat: 54.322684, lon: 10.13586
> >>>
> >>>
> >>>
> >>> ___
> >>>
> >>> Users mailing list
> >>>
> >>> Users@lists.opensips.org
> >>>
> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>>
> >>>
> >>>
> 
> >>>
> >>> ___
> >>> Users mailing list
> >>> Users@lists.opensips.org
> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
> >
> >
>
>
> --
>
> kiste lat: 54.322684, lon: 10.13586
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> ___
> Users mailing list
>

Re: [OpenSIPS-Users] Newbie To OpenSIPS

2009-09-08 Thread Kemp, Larry

Can OpenSIPS be used as a Session Border controller sitting at my edge passing 
and receiving SIP traffic to others I SIP peer with? If not, what other 
open-source would anyone suggest to act as SBC's? I too would rather do it via 
open-source and x86 or 64bit chip, less costly. Thanks.

Lars


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
Sent: Tuesday, September 08, 2009 1:50 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS

Hi,

Good question and not easy to answer. ACME is expensive AND you will
need somebody to configure it in a way you will need it. So as an
redundant option your talking about 100-150K.

To buy a big name won't prevent you from implementing, bugsearching.


My personal opinion: Take less money, look for good consultants and try
it with opensource.


BR

Uwe


Kemp, Larry schrieb:
> Certainly. If I just wanted to pass my SIP to other carriers or have them 
> connect to my SIP customers could I use OpenSIPS for that alone, or would I 
> still need some other sort of session border controller?
> 
> Larry Kemp
> 
> 
> -Original Message-
> From: users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
> Sent: Tuesday, September 08, 2009 1:31 PM
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
> 
> Hi,
> 
> Yes, this could be an option. But a very expensive one :-)
> 
> BR
> 
> Uwe
> 
> 
> Kemp, Larry schrieb:
>> So I would use OpenSIPS behind say like an Acme Packet 
>> http://www.acmepacket.com/ Session Border Controller or a MetaSwitch 
>> http://www.metaswitch.com/ connecting to the PSTN, then use OpenSIPS to talk 
>> to those Gateways via SIP and route my customer's VOIP traffic from their 
>> Asterisk PBX's to those devices that speak SIP, right?
>>
>> Lars
>>
>>
>> -Original Message-
>> From: users-boun...@lists.opensips.org 
>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
>> Sent: Tuesday, September 08, 2009 12:19 PM
>> To: OpenSIPS users mailling list
>> Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
>>
>> Hi,
>>
>> Opensips is a SIP router not a media gateway. So far you will need
>> something that will take care of comvert TDM/PSTN to sip.
>>
>> There should be lot of examples for this kind of setup.
>>
>> BR
>>
>> Uwe
>>
>>
>>
>> Kemp, Larry schrieb:
>>> Thank you for the response UK. We are looking at using OpenSIPS on Linux
>>> to act as a SIP Router to handoff SIP calls to other carriers that are
>>> also using SIP, and explore the use of the agent registration server
>>> functionality. I have already previously installed and debugged Asterisk
>>> on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP
>>> phones. My understanding is that you can route your Asterisk systems to
>>> OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or
>>> SS7 carriers, or your own Class 4 or 5 environment. Maybe I am
>>> misunderstanding as far as what OpenSIPS can actually do? Thanks.
>>>
>>>  
>>>
>>>  
>>>
>>>  
>>>
>>> -Original Message-
>>> From: users-boun...@lists.opensips.org
>>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
>>> Sent: Tuesday, September 08, 2009 11:32 AM
>>> To: OpenSIPS users mailling list
>>> Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
>>>
>>>  
>>>
>>> Hi Lars,
>>>
>>>  
>>>
>>>  
>>>
>>>  
>>>
 Any guidance would be appreciated from the community that has done
 this already. I installed current revision.
>>>  
>>>
>>> Could be helpfull to know what you want to do with opensips :-)
>>>
>>>  
>>>
>>> BR
>>>
>>>  
>>>
>>> Uwe
>>>
>>>  
>>>
>>> -- 
>>>
>>>  
>>>
>>> kiste lat: 54.322684, lon: 10.13586
>>>
>>>  
>>>
>>> ___
>>>
>>> Users mailing list
>>>
>>> Users@lists.opensips.org
>>>
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>> 
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> 
> 


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Re: [OpenSIPS-Users] Newbie To OpenSIPS

2009-09-08 Thread Uwe Kastens

Hi,

Good question and not easy to answer. ACME is expensive AND you will
need somebody to configure it in a way you will need it. So as an
redundant option your talking about 100-150K.

To buy a big name won't prevent you from implementing, bugsearching.


My personal opinion: Take less money, look for good consultants and try
it with opensource.


BR

Uwe


Kemp, Larry schrieb:
> Certainly. If I just wanted to pass my SIP to other carriers or have them 
> connect to my SIP customers could I use OpenSIPS for that alone, or would I 
> still need some other sort of session border controller?
> 
> Larry Kemp
> 
> 
> -Original Message-
> From: users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
> Sent: Tuesday, September 08, 2009 1:31 PM
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
> 
> Hi,
> 
> Yes, this could be an option. But a very expensive one :-)
> 
> BR
> 
> Uwe
> 
> 
> Kemp, Larry schrieb:
>> So I would use OpenSIPS behind say like an Acme Packet 
>> http://www.acmepacket.com/ Session Border Controller or a MetaSwitch 
>> http://www.metaswitch.com/ connecting to the PSTN, then use OpenSIPS to talk 
>> to those Gateways via SIP and route my customer's VOIP traffic from their 
>> Asterisk PBX's to those devices that speak SIP, right?
>>
>> Lars
>>
>>
>> -Original Message-
>> From: users-boun...@lists.opensips.org 
>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
>> Sent: Tuesday, September 08, 2009 12:19 PM
>> To: OpenSIPS users mailling list
>> Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
>>
>> Hi,
>>
>> Opensips is a SIP router not a media gateway. So far you will need
>> something that will take care of comvert TDM/PSTN to sip.
>>
>> There should be lot of examples for this kind of setup.
>>
>> BR
>>
>> Uwe
>>
>>
>>
>> Kemp, Larry schrieb:
>>> Thank you for the response UK. We are looking at using OpenSIPS on Linux
>>> to act as a SIP Router to handoff SIP calls to other carriers that are
>>> also using SIP, and explore the use of the agent registration server
>>> functionality. I have already previously installed and debugged Asterisk
>>> on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP
>>> phones. My understanding is that you can route your Asterisk systems to
>>> OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or
>>> SS7 carriers, or your own Class 4 or 5 environment. Maybe I am
>>> misunderstanding as far as what OpenSIPS can actually do? Thanks.
>>>
>>>  
>>>
>>>  
>>>
>>>  
>>>
>>> -Original Message-
>>> From: users-boun...@lists.opensips.org
>>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
>>> Sent: Tuesday, September 08, 2009 11:32 AM
>>> To: OpenSIPS users mailling list
>>> Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
>>>
>>>  
>>>
>>> Hi Lars,
>>>
>>>  
>>>
>>>  
>>>
>>>  
>>>
 Any guidance would be appreciated from the community that has done
 this already. I installed current revision.
>>>  
>>>
>>> Could be helpfull to know what you want to do with opensips :-)
>>>
>>>  
>>>
>>> BR
>>>
>>>  
>>>
>>> Uwe
>>>
>>>  
>>>
>>> -- 
>>>
>>>  
>>>
>>> kiste lat: 54.322684, lon: 10.13586
>>>
>>>  
>>>
>>> ___
>>>
>>> Users mailing list
>>>
>>> Users@lists.opensips.org
>>>
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>> 
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> 
> 


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Re: [OpenSIPS-Users] Newbie To OpenSIPS

2009-09-08 Thread Kemp, Larry

Certainly. If I just wanted to pass my SIP to other carriers or have them 
connect to my SIP customers could I use OpenSIPS for that alone, or would I 
still need some other sort of session border controller?

Larry Kemp


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
Sent: Tuesday, September 08, 2009 1:31 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS

Hi,

Yes, this could be an option. But a very expensive one :-)

BR

Uwe


Kemp, Larry schrieb:
> So I would use OpenSIPS behind say like an Acme Packet 
> http://www.acmepacket.com/ Session Border Controller or a MetaSwitch 
> http://www.metaswitch.com/ connecting to the PSTN, then use OpenSIPS to talk 
> to those Gateways via SIP and route my customer's VOIP traffic from their 
> Asterisk PBX's to those devices that speak SIP, right?
> 
> Lars
> 
> 
> -Original Message-
> From: users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
> Sent: Tuesday, September 08, 2009 12:19 PM
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
> 
> Hi,
> 
> Opensips is a SIP router not a media gateway. So far you will need
> something that will take care of comvert TDM/PSTN to sip.
> 
> There should be lot of examples for this kind of setup.
> 
> BR
> 
> Uwe
> 
> 
> 
> Kemp, Larry schrieb:
>>
>> Thank you for the response UK. We are looking at using OpenSIPS on Linux
>> to act as a SIP Router to handoff SIP calls to other carriers that are
>> also using SIP, and explore the use of the agent registration server
>> functionality. I have already previously installed and debugged Asterisk
>> on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP
>> phones. My understanding is that you can route your Asterisk systems to
>> OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or
>> SS7 carriers, or your own Class 4 or 5 environment. Maybe I am
>> misunderstanding as far as what OpenSIPS can actually do? Thanks.
>>
>>  
>>
>>  
>>
>>  
>>
>> -Original Message-
>> From: users-boun...@lists.opensips.org
>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
>> Sent: Tuesday, September 08, 2009 11:32 AM
>> To: OpenSIPS users mailling list
>> Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
>>
>>  
>>
>> Hi Lars,
>>
>>  
>>
>>  
>>
>>  
>>
>>> Any guidance would be appreciated from the community that has done
>>> this already. I installed current revision.
>>  
>>
>> Could be helpfull to know what you want to do with opensips :-)
>>
>>  
>>
>> BR
>>
>>  
>>
>> Uwe
>>
>>  
>>
>> -- 
>>
>>  
>>
>> kiste lat: 54.322684, lon: 10.13586
>>
>>  
>>
>> ___
>>
>> Users mailing list
>>
>> Users@lists.opensips.org
>>
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>> 
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 


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Re: [OpenSIPS-Users] Newbie To OpenSIPS

2009-09-08 Thread Jeff Pyle

Personally I prefer the Sonus GSX9000 gear for a PSTN gateway.  I even put
one in my kid's room.


- Jeff



On 9/8/09 1:30 PM, "Uwe Kastens"  wrote:

> Hi,
> 
> Yes, this could be an option. But a very expensive one :-)
> 
> BR
> 
> Uwe
> 
> 
> Kemp, Larry schrieb:
>> So I would use OpenSIPS behind say like an Acme Packet
>> http://www.acmepacket.com/ Session Border Controller or a MetaSwitch
>> http://www.metaswitch.com/ connecting to the PSTN, then use OpenSIPS to talk
>> to those Gateways via SIP and route my customer's VOIP traffic from their
>> Asterisk PBX's to those devices that speak SIP, right?
>> 
>> Lars
>> 
>> 
>> -Original Message-
>> From: users-boun...@lists.opensips.org
>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
>> Sent: Tuesday, September 08, 2009 12:19 PM
>> To: OpenSIPS users mailling list
>> Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
>> 
>> Hi,
>> 
>> Opensips is a SIP router not a media gateway. So far you will need
>> something that will take care of comvert TDM/PSTN to sip.
>> 
>> There should be lot of examples for this kind of setup.
>> 
>> BR
>> 
>> Uwe
>> 
>> 
>> 
>> Kemp, Larry schrieb:
>>> 
>>> Thank you for the response UK. We are looking at using OpenSIPS on Linux
>>> to act as a SIP Router to handoff SIP calls to other carriers that are
>>> also using SIP, and explore the use of the agent registration server
>>> functionality. I have already previously installed and debugged Asterisk
>>> on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP
>>> phones. My understanding is that you can route your Asterisk systems to
>>> OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or
>>> SS7 carriers, or your own Class 4 or 5 environment. Maybe I am
>>> misunderstanding as far as what OpenSIPS can actually do? Thanks.
>>> 
>>>  
>>> 
>>>  
>>> 
>>>  
>>> 
>>> -Original Message-
>>> From: users-boun...@lists.opensips.org
>>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
>>> Sent: Tuesday, September 08, 2009 11:32 AM
>>> To: OpenSIPS users mailling list
>>> Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
>>> 
>>>  
>>> 
>>> Hi Lars,
>>> 
>>>  
>>> 
>>>  
>>> 
>>>  
>>> 
 Any guidance would be appreciated from the community that has done
 this already. I installed current revision.
>>>  
>>> 
>>> Could be helpfull to know what you want to do with opensips :-)
>>> 
>>>  
>>> 
>>> BR
>>> 
>>>  
>>> 
>>> Uwe
>>> 
>>>  
>>> 
>>> -- 
>>> 
>>>  
>>> 
>>> kiste lat: 54.322684, lon: 10.13586
>>> 
>>>  
>>> 
>>> ___
>>> 
>>> Users mailing list
>>> 
>>> Users@lists.opensips.org
>>> 
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>> 
>>> 
>>> 
>>> 
>>> ___
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>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>> 
> 


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Re: [OpenSIPS-Users] Newbie To OpenSIPS

2009-09-08 Thread Uwe Kastens

Hi,

Yes, this could be an option. But a very expensive one :-)

BR

Uwe


Kemp, Larry schrieb:
> So I would use OpenSIPS behind say like an Acme Packet 
> http://www.acmepacket.com/ Session Border Controller or a MetaSwitch 
> http://www.metaswitch.com/ connecting to the PSTN, then use OpenSIPS to talk 
> to those Gateways via SIP and route my customer's VOIP traffic from their 
> Asterisk PBX's to those devices that speak SIP, right?
> 
> Lars
> 
> 
> -Original Message-
> From: users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
> Sent: Tuesday, September 08, 2009 12:19 PM
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
> 
> Hi,
> 
> Opensips is a SIP router not a media gateway. So far you will need
> something that will take care of comvert TDM/PSTN to sip.
> 
> There should be lot of examples for this kind of setup.
> 
> BR
> 
> Uwe
> 
> 
> 
> Kemp, Larry schrieb:
>>
>> Thank you for the response UK. We are looking at using OpenSIPS on Linux
>> to act as a SIP Router to handoff SIP calls to other carriers that are
>> also using SIP, and explore the use of the agent registration server
>> functionality. I have already previously installed and debugged Asterisk
>> on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP
>> phones. My understanding is that you can route your Asterisk systems to
>> OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or
>> SS7 carriers, or your own Class 4 or 5 environment. Maybe I am
>> misunderstanding as far as what OpenSIPS can actually do? Thanks.
>>
>>  
>>
>>  
>>
>>  
>>
>> -Original Message-
>> From: users-boun...@lists.opensips.org
>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
>> Sent: Tuesday, September 08, 2009 11:32 AM
>> To: OpenSIPS users mailling list
>> Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
>>
>>  
>>
>> Hi Lars,
>>
>>  
>>
>>  
>>
>>  
>>
>>> Any guidance would be appreciated from the community that has done
>>> this already. I installed current revision.
>>  
>>
>> Could be helpfull to know what you want to do with opensips :-)
>>
>>  
>>
>> BR
>>
>>  
>>
>> Uwe
>>
>>  
>>
>> -- 
>>
>>  
>>
>> kiste lat: 54.322684, lon: 10.13586
>>
>>  
>>
>> ___
>>
>> Users mailing list
>>
>> Users@lists.opensips.org
>>
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>> 
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 


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Re: [OpenSIPS-Users] Newbie To OpenSIPS

2009-09-08 Thread Kemp, Larry

So I would use OpenSIPS behind say like an Acme Packet 
http://www.acmepacket.com/ Session Border Controller or a MetaSwitch 
http://www.metaswitch.com/ connecting to the PSTN, then use OpenSIPS to talk to 
those Gateways via SIP and route my customer's VOIP traffic from their Asterisk 
PBX's to those devices that speak SIP, right?

Lars


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
Sent: Tuesday, September 08, 2009 12:19 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS

Hi,

Opensips is a SIP router not a media gateway. So far you will need
something that will take care of comvert TDM/PSTN to sip.

There should be lot of examples for this kind of setup.

BR

Uwe



Kemp, Larry schrieb:
> 
> 
> Thank you for the response UK. We are looking at using OpenSIPS on Linux
> to act as a SIP Router to handoff SIP calls to other carriers that are
> also using SIP, and explore the use of the agent registration server
> functionality. I have already previously installed and debugged Asterisk
> on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP
> phones. My understanding is that you can route your Asterisk systems to
> OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or
> SS7 carriers, or your own Class 4 or 5 environment. Maybe I am
> misunderstanding as far as what OpenSIPS can actually do? Thanks.
> 
>  
> 
>  
> 
>  
> 
> -Original Message-
> From: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
> Sent: Tuesday, September 08, 2009 11:32 AM
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
> 
>  
> 
> Hi Lars,
> 
>  
> 
>  
> 
>  
> 
>> Any guidance would be appreciated from the community that has done
> 
>> this already. I installed current revision.
> 
>>
> 
>  
> 
> Could be helpfull to know what you want to do with opensips :-)
> 
>  
> 
> BR
> 
>  
> 
> Uwe
> 
>  
> 
> -- 
> 
>  
> 
> kiste lat: 54.322684, lon: 10.13586
> 
>  
> 
> ___
> 
> Users mailing list
> 
> Users@lists.opensips.org
> 
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 
> 
> 
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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Re: [OpenSIPS-Users] Newbie To OpenSIPS

2009-09-08 Thread Uwe Kastens

Hi,

Opensips is a SIP router not a media gateway. So far you will need
something that will take care of comvert TDM/PSTN to sip.

There should be lot of examples for this kind of setup.

BR

Uwe



Kemp, Larry schrieb:
> 
> 
> Thank you for the response UK. We are looking at using OpenSIPS on Linux
> to act as a SIP Router to handoff SIP calls to other carriers that are
> also using SIP, and explore the use of the agent registration server
> functionality. I have already previously installed and debugged Asterisk
> on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP
> phones. My understanding is that you can route your Asterisk systems to
> OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or
> SS7 carriers, or your own Class 4 or 5 environment. Maybe I am
> misunderstanding as far as what OpenSIPS can actually do? Thanks.
> 
>  
> 
>  
> 
>  
> 
> -Original Message-
> From: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
> Sent: Tuesday, September 08, 2009 11:32 AM
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS
> 
>  
> 
> Hi Lars,
> 
>  
> 
>  
> 
>  
> 
>> Any guidance would be appreciated from the community that has done
> 
>> this already. I installed current revision.
> 
>>
> 
>  
> 
> Could be helpfull to know what you want to do with opensips :-)
> 
>  
> 
> BR
> 
>  
> 
> Uwe
> 
>  
> 
> -- 
> 
>  
> 
> kiste lat: 54.322684, lon: 10.13586
> 
>  
> 
> ___
> 
> Users mailing list
> 
> Users@lists.opensips.org
> 
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 
> 
> 
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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Re: [OpenSIPS-Users] explizit handling auf replyto

2009-09-08 Thread Uwe Kastens

Hi,

>>   
> you do not need to do any routing in onreply route at all, in none of 
> the case (stateless or statefull)
>> I will make a trace and post it to the list. One with forward and the
>> other without.
>>   
> make a trace and opensips logs.
> 

I have attached opensips.log with debug=9.
w_forward_an.gz => with forward
wo_forward_an.gz => without forward

BR

Uwe

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w_forward_an.gz
Description: GNU Zip compressed data


wo_forward_an.gz
Description: GNU Zip compressed data
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Re: [OpenSIPS-Users] Newbie To OpenSIPS

2009-09-08 Thread Kemp, Larry

Thank you for the response UK. We are looking at using OpenSIPS on Linux to act 
as a SIP Router to handoff SIP calls to other carriers that are also using SIP, 
and explore the use of the agent registration server functionality. I have 
already previously installed and debugged Asterisk on CentOS to act as a 
branded appliance issuing DHCP and DNS to VOIP phones. My understanding is that 
you can route your Asterisk systems to OpenSIPS as a SIP-Proxy-where they can 
then get router to other SIP or SS7 carriers, or your own Class 4 or 5 
environment. Maybe I am misunderstanding as far as what OpenSIPS can actually 
do? Thanks.

[cid:image001.jpg@01CA307D.1602ADC0]





-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens
Sent: Tuesday, September 08, 2009 11:32 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS



Hi Lars,







> Any guidance would be appreciated from the community that has done

> this already. I installed current revision.

>



Could be helpfull to know what you want to do with opensips :-)



BR



Uwe



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Re: [OpenSIPS-Users] explizit handling auf replyto

2009-09-08 Thread Bogdan-Andrei Iancu

Uwe Kastens wrote:
> Hello,
>
> Ok. I need to have that forward() in my configuration the get answers
> like 404, 486, 487 back to my asterisk. Reading your statements this
> should not be possible since I use t_relay for the requests and the
> replys should be routed by default.
>   
you do not need to do any routing in onreply route at all, in none of 
the case (stateless or statefull)
> I will make a trace and post it to the list. One with forward and the
> other without.
>   
make a trace and opensips logs.

Regards,
Bogdan
> BR and Thanks in advance
>
> Uwe
>
>
>
> Bogdan-Andrei Iancu schrieb:
>   
>> Uwe,
>>
>> forward() is a function exclusivly used for REQUESTS - for replies, 
>> nothing needs to be done as OpenSIPS will do it automatically:
>>
>> 1) if the requests was statefully forwarded (via t_relay() ), the 
>> transaction will contain all the info to route back the reply
>>
>> 2) if the requests was statelessly forwarded (via forward() ), the VIA 
>> stack (in received reply) will contain all the info to route back the reply
>>
>> Regards,
>> Bogdan
>>
>>
>> Uwe Kastens wrote:
>> 
>>> Hi Bogdan,
>>>
>>>   
>>>   
>  I need to route this
> replys with an reply_route and forward them explicitly to the pstn 
> gateway.
>   
>   
>   
 and how do you do reply routing ?? replies are automatically routed 
 based on VIA stack and you cannot influence this from script.
 
 
>>> ...
>>> if (!t_relay()) {
>>>   sl_reply_error();
>>>  };
>>>
>>> t_on_reply("1");
>>> }
>>> onreply_route[1]  {
>>> xlog("L_DBG","== route [1] t_on_failure msg-len from $si:$sp:
>>> $ml \n$mb\n ==\n");
>>> forward("10.20.30.101:5100");
>>> }
>>>
>>> BR
>>>
>>> Uwe
>>>
>>>
>>>
>>>   
>>>   
 Regards,
 Bogdan
 
 
> This not as it should be?
>
>
>
> BR
>
> Uwe
>   
>   
>   
>> using forward() forces a stateless behaviour of OpenSIPS. To be able to 
>> catch replies for a requests (what on_reply route does) it requires a 
>> statefull approach. This is the reason it does not work for you.
>>
>> If you use t_relay() instead of forward(), you should be able to use 
>> failure and onreply routes.
>>
>> Regards,
>> Bogdan
>>
>> Uwe Kastens wrote:
>> 
>> 
>> 
>>> Hello,
>>>
>>> I am using opensips 1.5.1 and I have the problem, that busy messages are
>>> not passed to the mediagw. Anything else is working (ACK etc.pp.). If I
>>> route that message via forward("IP:port") to the media gw in on_replyto,
>>> busy is processed correctly. If not, the busy is not send to the 
>>> mediagw.
>>>
>>> So I was wondering if I had to handle some replyto messages?
>>>
>>> BR
>>>
>>> uwe
>>>
>>>
>>>   
>>>   
>>>   
>>>   
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>> 
>> 
>   
>   
>   
 ___
 Users mailing list
 Users@lists.opensips.org
 http://lists.opensips.org/cgi-bin/mailman/listinfo/users
 
 
>>>   
>>>   
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>
>
>   


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Re: [OpenSIPS-Users] explizit handling auf replyto

2009-09-08 Thread Uwe Kastens

Hello,

Ok. I need to have that forward() in my configuration the get answers
like 404, 486, 487 back to my asterisk. Reading your statements this
should not be possible since I use t_relay for the requests and the
replys should be routed by default.

I will make a trace and post it to the list. One with forward and the
other without.

BR and Thanks in advance

Uwe



Bogdan-Andrei Iancu schrieb:
> Uwe,
> 
> forward() is a function exclusivly used for REQUESTS - for replies, 
> nothing needs to be done as OpenSIPS will do it automatically:
> 
> 1) if the requests was statefully forwarded (via t_relay() ), the 
> transaction will contain all the info to route back the reply
> 
> 2) if the requests was statelessly forwarded (via forward() ), the VIA 
> stack (in received reply) will contain all the info to route back the reply
> 
> Regards,
> Bogdan
> 
> 
> Uwe Kastens wrote:
>> Hi Bogdan,
>>
>>   
  I need to route this
 replys with an reply_route and forward them explicitly to the pstn gateway.
   
   
>>> and how do you do reply routing ?? replies are automatically routed 
>>> based on VIA stack and you cannot influence this from script.
>>> 
>>
>> ...
>> if (!t_relay()) {
>>   sl_reply_error();
>>  };
>>
>> t_on_reply("1");
>> }
>> onreply_route[1]  {
>> xlog("L_DBG","== route [1] t_on_failure msg-len from $si:$sp:
>> $ml \n$mb\n ==\n");
>> forward("10.20.30.101:5100");
>> }
>>
>> BR
>>
>> Uwe
>>
>>
>>
>>   
>>> Regards,
>>> Bogdan
>>> 
 This not as it should be?



 BR

 Uwe
   
   
> using forward() forces a stateless behaviour of OpenSIPS. To be able to 
> catch replies for a requests (what on_reply route does) it requires a 
> statefull approach. This is the reason it does not work for you.
>
> If you use t_relay() instead of forward(), you should be able to use 
> failure and onreply routes.
>
> Regards,
> Bogdan
>
> Uwe Kastens wrote:
> 
> 
>> Hello,
>>
>> I am using opensips 1.5.1 and I have the problem, that busy messages are
>> not passed to the mediagw. Anything else is working (ACK etc.pp.). If I
>> route that message via forward("IP:port") to the media gw in on_replyto,
>> busy is processed correctly. If not, the busy is not send to the mediagw.
>>
>> So I was wondering if I had to handle some replyto messages?
>>
>> BR
>>
>> uwe
>>
>>
>>   
>>   
>>   
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 
   
   
>>> ___
>>> Users mailing list
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>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>> 
>>
>>   
> 
> 
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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Re: [OpenSIPS-Users] explizit handling auf replyto

2009-09-08 Thread Alex Balashov

Bogdan-Andrei Iancu wrote:

> 2) if the requests was statelessly forwarded (via forward() ), the VIA 
> stack (in received reply) will contain all the info to route back the reply

I think the question is whether stateless forwarding can be used to 
override default processing of Via and route the reply somewhere else.

You can, for example, do this (whether in stateless or stateful request 
forwarding mode):

onreply_route[1] {
drop;
}

... I think it's along that general train of thought.

-- 
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: [OpenSIPS-Users] explizit handling auf replyto

2009-09-08 Thread Bogdan-Andrei Iancu

Uwe,

forward() is a function exclusivly used for REQUESTS - for replies, 
nothing needs to be done as OpenSIPS will do it automatically:

1) if the requests was statefully forwarded (via t_relay() ), the 
transaction will contain all the info to route back the reply

2) if the requests was statelessly forwarded (via forward() ), the VIA 
stack (in received reply) will contain all the info to route back the reply

Regards,
Bogdan


Uwe Kastens wrote:
> Hi Bogdan,
>
>   
>>>  I need to route this
>>> replys with an reply_route and forward them explicitly to the pstn gateway.
>>>   
>>>   
>> and how do you do reply routing ?? replies are automatically routed 
>> based on VIA stack and you cannot influence this from script.
>> 
>
>
> ...
> if (!t_relay()) {
>   sl_reply_error();
>  };
>
> t_on_reply("1");
> }
> onreply_route[1]  {
> xlog("L_DBG","== route [1] t_on_failure msg-len from $si:$sp:
> $ml \n$mb\n ==\n");
> forward("10.20.30.101:5100");
> }
>
> BR
>
> Uwe
>
>
>
>   
>> Regards,
>> Bogdan
>> 
>>> This not as it should be?
>>>
>>>
>>>
>>> BR
>>>
>>> Uwe
>>>   
>>>   
 using forward() forces a stateless behaviour of OpenSIPS. To be able to 
 catch replies for a requests (what on_reply route does) it requires a 
 statefull approach. This is the reason it does not work for you.

 If you use t_relay() instead of forward(), you should be able to use 
 failure and onreply routes.

 Regards,
 Bogdan

 Uwe Kastens wrote:
 
 
> Hello,
>
> I am using opensips 1.5.1 and I have the problem, that busy messages are
> not passed to the mediagw. Anything else is working (ACK etc.pp.). If I
> route that message via forward("IP:port") to the media gw in on_replyto,
> busy is processed correctly. If not, the busy is not send to the mediagw.
>
> So I was wondering if I had to handle some replyto messages?
>
> BR
>
> uwe
>
>
>   
>   
>   
 ___
 Users mailing list
 Users@lists.opensips.org
 http://lists.opensips.org/cgi-bin/mailman/listinfo/users
 
 
>>>   
>>>   
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>
>
>   


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Re: [OpenSIPS-Users] explizit handling auf replyto

2009-09-08 Thread Alex Balashov

Uwe Kastens wrote:
> Hi,
> 
>> Replies are automatically routed only if they are statefully routed.
> 
> Statefull = t_relay() ?

Keep in mind, you can't really mix stateful and stateless handling for 
what are hopefully obvious reasons.  If you statelessly direct a reply, 
the stateful transaction layer used to open the transaction from the 
initial request will not know that you did so, and vice versa, etc etc.

-- 
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Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: [OpenSIPS-Users] explizit handling auf replyto

2009-09-08 Thread Alex Balashov

Uwe Kastens wrote:

>> Replies are automatically routed only if they are statefully routed.
> 
> Statefull = t_relay() ?

Yes.

-- 
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Re: [OpenSIPS-Users] explizit handling auf replyto

2009-09-08 Thread Uwe Kastens

Hi,

> Replies are automatically routed only if they are statefully routed.

Statefull = t_relay() ?

BR

Uwe
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Re: [OpenSIPS-Users] explizit handling auf replyto

2009-09-08 Thread Alex Balashov

Replies are automatically routed only if they are statefully routed.

Uwe Kastens wrote:

> Hi Bogdan,
> 
>>>  I need to route this
>>> replys with an reply_route and forward them explicitly to the pstn gateway.
>>>   
>> and how do you do reply routing ?? replies are automatically routed 
>> based on VIA stack and you cannot influence this from script.
> 
> 
> ...
> if (!t_relay()) {
>   sl_reply_error();
>  };
> 
> t_on_reply("1");
> }
> onreply_route[1]  {
> xlog("L_DBG","== route [1] t_on_failure msg-len from $si:$sp:
> $ml \n$mb\n ==\n");
> forward("10.20.30.101:5100");
> }
> 
> BR
> 
> Uwe
> 
> 
> 
>> Regards,
>> Bogdan
>>> This not as it should be?
>>>
>>>
>>>
>>> BR
>>>
>>> Uwe
>>>   
 using forward() forces a stateless behaviour of OpenSIPS. To be able to 
 catch replies for a requests (what on_reply route does) it requires a 
 statefull approach. This is the reason it does not work for you.

 If you use t_relay() instead of forward(), you should be able to use 
 failure and onreply routes.

 Regards,
 Bogdan

 Uwe Kastens wrote:
 
> Hello,
>
> I am using opensips 1.5.1 and I have the problem, that busy messages are
> not passed to the mediagw. Anything else is working (ACK etc.pp.). If I
> route that message via forward("IP:port") to the media gw in on_replyto,
> busy is processed correctly. If not, the busy is not send to the mediagw.
>
> So I was wondering if I had to handle some replyto messages?
>
> BR
>
> uwe
>
>
>   
>   
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>>>   
>>
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> 
> 


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Re: [OpenSIPS-Users] Newbie To OpenSIPS

2009-09-08 Thread Uwe Kastens

Hi Lars,



> Any guidance would be appreciated from the community that has done
> this already. I installed current revision.
> 

Could be helpfull to know what you want to do with opensips :-)

BR

Uwe

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Re: [OpenSIPS-Users] explizit handling auf replyto

2009-09-08 Thread Uwe Kastens

Hi Bogdan,

>>  I need to route this
>> replys with an reply_route and forward them explicitly to the pstn gateway.
>>   
> and how do you do reply routing ?? replies are automatically routed 
> based on VIA stack and you cannot influence this from script.


...
if (!t_relay()) {
  sl_reply_error();
 };

t_on_reply("1");
}
onreply_route[1]  {
xlog("L_DBG","== route [1] t_on_failure msg-len from $si:$sp:
$ml \n$mb\n ==\n");
forward("10.20.30.101:5100");
}

BR

Uwe



> 
> Regards,
> Bogdan
>> This not as it should be?
>>
>>
>>
>> BR
>>
>> Uwe
>>   
>>> using forward() forces a stateless behaviour of OpenSIPS. To be able to 
>>> catch replies for a requests (what on_reply route does) it requires a 
>>> statefull approach. This is the reason it does not work for you.
>>>
>>> If you use t_relay() instead of forward(), you should be able to use 
>>> failure and onreply routes.
>>>
>>> Regards,
>>> Bogdan
>>>
>>> Uwe Kastens wrote:
>>> 
 Hello,

 I am using opensips 1.5.1 and I have the problem, that busy messages are
 not passed to the mediagw. Anything else is working (ACK etc.pp.). If I
 route that message via forward("IP:port") to the media gw in on_replyto,
 busy is processed correctly. If not, the busy is not send to the mediagw.

 So I was wondering if I had to handle some replyto messages?

 BR

 uwe


   
   
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>>> 
>>
>>   
> 
> 
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Re: [OpenSIPS-Users] Access to Radius AVP's and "radius_extra" parameter

2009-09-08 Thread Irina Stanescu

Hello John,

On Tue, Sep 8, 2009 at 5:43 PM, John Quick wrote:

> I have been experimenting with Radius authentication on an older version of
> OpenSIPS (actually my test
> unit has OpenSER v1.3.2, but I don't think it will be much different to
> 1.5.x for this). I want to be
> able to read several values returned from the Radius server. Is this what
> the modparam "radius_extra"
> is for? Or is it to allow extra values to be passed to the Radius server
> for accounting?
>
>
The "radius_extra" parameter only allows you to send extra values to the
Radius server, to be logged for accounting. You cannot fetch any value from
an accounting reply.



> My Radius server is configured to return three AVP's like this:
> Reply-Message = "Opensips rules!",
> Sip-RPId = sip:123...@mysip.com ,
> SIP-AVP = "#14:012...@mynumber.biz <14%3a012...@mynumber.biz>"
>
> Using Wireshark, I can see that the values are all being sent back to the
> OpenSIPS server, but I have
> only had success in accessing the value in SIP-AVP. Is there a trick to
> this or are those other values
> unavailable in the OpenSIPS script?
>
> Is the functionality I'm looking for only available in the new AAA modules?
>
>
The functionality you are looking for is provided by the new AAA_RADIUS
module. Using the functions exported by this module you can yield directly
from the script any type of Radius query and also inspect replies for
certain attributes.
For more information, check out the
tutorialand the
documentation
 .
If you have any other uncertainties, please let me know. Any other
suggestions are welcome as well.



> John Quick
> Smartvox Limited
> Web: www.smartvox.co.uk
>
> Smartvox is a limited company, registered in England and Wales, number
> 5005263.
> Registered office: Spectrum House, Dunstable Road, Redbourn, St.Albans,
> Herts AL3 7PR
>
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>


Best regards,
Irina Stanescu
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[OpenSIPS-Users] b2bua top hiding + authentication

2009-09-08 Thread Jeff Pyle

Hello,

What impact does authentication have on the b2bua modules?  When I put
b2b_init_request("top hiding") into my otherwise functioning script, very
strange things happen.

In order to achieve topology hiding, is it as simple as inserting this init
line at some point, or is there more to it?


- Jeff


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Re: [OpenSIPS-Users] explizit handling auf replyto

2009-09-08 Thread Bogdan-Andrei Iancu

Hi Uwe,


Uwe Kastens wrote:
> Hi Bogdan,
>
> Seems that my question was not very clear.
>
> I would expect that reply messages would be handled automatically, if I
> use t_relay.
whatever forwarding function you are using (forward / t_relay), the 
replies are automatically routed back by opensips core without any help 
from script.
>  This seems not to happen in my setup.
could you post a sip trace of a failing reply ?

>  I need to route this
> replys with an reply_route and forward them explicitly to the pstn gateway.
>   
and how do you do reply routing ?? replies are automatically routed 
based on VIA stack and you cannot influence this from script.

Regards,
Bogdan
> This not as it should be?
>
>
>
> BR
>
> Uwe
>   
>> using forward() forces a stateless behaviour of OpenSIPS. To be able to 
>> catch replies for a requests (what on_reply route does) it requires a 
>> statefull approach. This is the reason it does not work for you.
>>
>> If you use t_relay() instead of forward(), you should be able to use 
>> failure and onreply routes.
>>
>> Regards,
>> Bogdan
>>
>> Uwe Kastens wrote:
>> 
>>> Hello,
>>>
>>> I am using opensips 1.5.1 and I have the problem, that busy messages are
>>> not passed to the mediagw. Anything else is working (ACK etc.pp.). If I
>>> route that message via forward("IP:port") to the media gw in on_replyto,
>>> busy is processed correctly. If not, the busy is not send to the mediagw.
>>>
>>> So I was wondering if I had to handle some replyto messages?
>>>
>>> BR
>>>
>>> uwe
>>>
>>>
>>>   
>>>   
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>
>
>   


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[OpenSIPS-Users] Call Control module flexibility (renamed)

2009-09-08 Thread Bogdan-Andrei Iancu

Hi Ashwini,

I see you point - the question is how flexible is the Call Control 
module in order to allow you to customize the "billing account" (and not 
to use all the time the FROM URI).

Maybe somebody involved in this module devel can answer you.

Regards,
Bogdan

ASHWINI NAIDU wrote:
>
> Hi Bogdan,
>
>I have a scenario where one user Ashwini can have 2 phnos as her 
> accounts. When Ashwini calls from any of there 2 phnos or Ashwini the 
> value shd be deducted from the ashw...@.com 
>  which is provided.
>
> *Example:
> Ashwini has 91992 and 919xxx19
> If i call from 91992 my debit balance should be retrieved from 
> ashwini
> *
> This is the whole scenario. The call-controller is picking the 
> original From header value.
>
> other than this is there any way to achieve the above scenario.
>
>
> On Fri, Sep 4, 2009 at 7:36 PM, Brett Nemeroff  > wrote:
>
> Why not just use whatever value you are using as a parameter for
> uac_replace_from ?
>
>
> On Fri, Sep 4, 2009 at 8:07 AM, ASHWINI NAIDU
> mailto:ashwini.na...@gmail.com>> wrote:
>
> Hi Bogdan,
>
> The whole scenario is i want the new from(after
> uac_replace_from)  header value to be used by the call
> controller. is there any way i can do it.
>
>
> On Fri, Sep 4, 2009 at 6:14 PM, Bogdan-Andrei Iancu
> mailto:bog...@voice-system.ro>> wrote:
>
> Hi Ashwini,
>
> The variables that provide information from the SIP
> message (like $fu)
> get the values from the original received message,
> disregarding whatever
> changes you do .The only exception is $ru (and family) vars.
>
> Regards,
> Bogdan
>
>
> ASHWINI NAIDU wrote:
> > Hi all,
> >
> >
> >  I am trying to replace the from header using the
> > *uac_replace_from* in uac module. the thing i noticed
> the replacement
> > is happening in the* sip headers*. After using the
> *uac_replace_from*
> > and try to print the *$fu and $fU *the previous values
> before
> > replacement can be seen . can anyone tell me what is the
> problem?
> >
> >
> > --
> > Thanking You,
> > Ashwini BR Naidu
> >
> 
> 
> >
> > ___
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> > Users@lists.opensips.org 
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
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>
>
>
> -- 
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> Ashwini BR Naidu
>
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>
>
>
> -- 
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> Ashwini BR Naidu
> 
>
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Re: [OpenSIPS-Users] Regarding Uac_replace_from

2009-09-08 Thread Bogdan-Andrei Iancu

Hi Ashwini,

I do not know about the Call-Control module too much, but can you also 
push to it the new FROM value from the script ? or you cannot influence 
the info that is sent to CC module ?

Regards,
Bogdan

ASHWINI NAIDU wrote:
> Hi Bogdan,
>
> The whole scenario is i want the new from(after uac_replace_from)  
> header value to be used by the call controller. is there any way i can 
> do it.
>
> On Fri, Sep 4, 2009 at 6:14 PM, Bogdan-Andrei Iancu 
> mailto:bog...@voice-system.ro>> wrote:
>
> Hi Ashwini,
>
> The variables that provide information from the SIP message (like $fu)
> get the values from the original received message, disregarding
> whatever
> changes you do .The only exception is $ru (and family) vars.
>
> Regards,
> Bogdan
>
>
> ASHWINI NAIDU wrote:
> > Hi all,
> >
> >
> >  I am trying to replace the from header using the
> > *uac_replace_from* in uac module. the thing i noticed the
> replacement
> > is happening in the* sip headers*. After using the
> *uac_replace_from*
> > and try to print the *$fu and $fU *the previous values before
> > replacement can be seen . can anyone tell me what is the problem?
> >
> >
> > --
> > Thanking You,
> > Ashwini BR Naidu
> >
> 
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org 
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
> ___
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>
>
>
>
> -- 
> Thanking You,
> Ashwini BR Naidu
> 
>
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Re: [OpenSIPS-Users] Nextone - Proxy-Authorization and AuthorizationHeader

2009-09-08 Thread Bogdan-Andrei Iancu

Hi Julien,

It most of the cases, your approach is correct , but to be 100% safe, 
use "remove_credentials()" instead of "remove_hf()" - remove_credentials 
will remove only the credentials/headers that were using on the local 
authentication, which is useful when a requests carries more than one 
auth sets (multi stage authentication)

Regards,
Bogdan

Julien Chavanton wrote:
> We need to deal with the fact that the remote Nextone UA does not 
> reply when Proxy-Authorization or Authorization is present in the body
> In this example, removing the Proxy-Authorization after authentication 
> as been done on Opensips seems to work.
>  
>  # local and authenticated user ? route the call, else challenge 
> authentication
>  if(is_from_local()){
> if (proxy_authorize("", "subscriber")){
> remove_hf("Proxy-Authorization");
> t_relay();
> exit;
> }
> else{
>proxy_challenge("", "1");
>exit;
> }
>  }
>
> 
> *From:* users-boun...@lists.opensips.org on behalf of Julien Chavanton
> *Sent:* Mon 07/09/2009 4:06 PM
> *To:* Users@lists.opensips.org
> *Subject:* [OpenSIPS-Users] Nextone - Proxy-Authorization and 
> AuthorizationHeader
>
> Hi, when forwarding call to Nextone SBC if there is
>  
> Proxy-Authorization or Authorization header it does not reply to INVITE
>  
> Can I configure Opensip not to forward the authentication information 
> used with his clients softphones ?
> 
>
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Re: [OpenSIPS-Users] New AAA module: Which release of Opensips?

2009-09-08 Thread Bogdan-Andrei Iancu

Hi John,

The new AAA/RADIUS support is present only in 1.6 (current devel 
version). Options are:

1) see if the old RADIUS support does not fit your needs somehow -> use 
opensips 1.5

2) go for the new AAA/RADIUS support and use 1.6 (even if devel, you can 
take the chance to use it - there are other people doing that)

3) go for the new AAA/RADIUS support, but wait for the official release 
of 1.6 in mid October.

Best regards,
Bogdan

John Quick wrote:
> Hi,
>
> I want to use the new AAA/Radius module, but it is for a business critical 
> service so am reluctant to
> move to release 1.6 if it is not quite bedded in yet. What choices are 
> available?
>
> John Quick
> Smartvox Limited
> Web: www.smartvox.co.uk
>
> Smartvox is a limited company, registered in England and Wales, number 
> 5005263.
> Registered office: Spectrum House, Dunstable Road, Redbourn, St.Albans, Herts 
> AL3 7PR
>  
>
>
>
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Re: [OpenSIPS-Users] Newbie To OpenSIPS

2009-09-08 Thread Brett Nemeroff

I hate to sound like a pain, but the information really isn't splintered at
all. The docs are all at www.opensips.org. Much of the older openser docs
are still very relevant, but you shouldn't even need those anymore. Check
the opensips web site. Go over the tutorials. Check the example
configuration and then ask us some specific questions and we'll be happy to
help you out.
Compiling and installing opensips is very straightforward. a make/make
install should cover 95% of most installations. The only modifications I
ever really find necessary is to allow compilation of db_mysql in the
Makefile, which I do by hand so I can recompile in the future without
remembering the switches.

-Brett

On Tue, Sep 8, 2009 at 9:36 AM, Kemp, Larry wrote:

> First time compiling OpenSIPS; installing on CentOS. Downloaded source as
> well as RPM's from http://centos.leurent.eu. I was wondering if any of you
> that have already done this successfully and had any resources or notes from
> your deployment that you might be willing to share, detailed how-to info for
> database and making/installing gui, or could recommend which of the many
> books that exist to read on this. Info seems splintered since the OpenSER
> split.
> To date I have:
> Compiled & started /usr/local/sbin/opensips
>   Listening on
> udp: 127.0.0.1 [127.0.0.1]:5060
> udp: 172.20.30.184 [172.20.30.184]:5060
> tcp: 127.0.0.1 [127.0.0.1]:5060
> tcp: 172.20.30.184 [172.20.30.184]:5060
>   Aliases:
> tcp: localhost:5060
> tcp: localhost.localdomain:5060
> tcp: hostname:5060
> tcp: hostname.domain.com:5060
> udp: localhost:5060
> udp: localhost.localdomain:5060
> udp: hostname:5060
> udp: hostname.domain.com:5060
> Any guidance would be appreciated from the community that has done this
> already. I installed current revision.
>
> Lars
>
>
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[OpenSIPS-Users] Access to Radius AVP's and "radius_extra" parameter

2009-09-08 Thread John Quick

I have been experimenting with Radius authentication on an older version of 
OpenSIPS (actually my test
unit has OpenSER v1.3.2, but I don't think it will be much different to 1.5.x 
for this). I want to be
able to read several values returned from the Radius server. Is this what the 
modparam "radius_extra"
is for? Or is it to allow extra values to be passed to the Radius server for 
accounting?

My Radius server is configured to return three AVP's like this:
Reply-Message = "Opensips rules!",
Sip-RPId = sip:123...@mysip.com,
SIP-AVP = "#14:012...@mynumber.biz"

Using Wireshark, I can see that the values are all being sent back to the 
OpenSIPS server, but I have
only had success in accessing the value in SIP-AVP. Is there a trick to this or 
are those other values
unavailable in the OpenSIPS script?

Is the functionality I'm looking for only available in the new AAA modules?

John Quick
Smartvox Limited
Web: www.smartvox.co.uk

Smartvox is a limited company, registered in England and Wales, number 5005263.
Registered office: Spectrum House, Dunstable Road, Redbourn, St.Albans, Herts 
AL3 7PR
 



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[OpenSIPS-Users] New AAA module: Which release of Opensips?

2009-09-08 Thread John Quick

Hi,

I want to use the new AAA/Radius module, but it is for a business critical 
service so am reluctant to
move to release 1.6 if it is not quite bedded in yet. What choices are 
available?

John Quick
Smartvox Limited
Web: www.smartvox.co.uk

Smartvox is a limited company, registered in England and Wales, number 5005263.
Registered office: Spectrum House, Dunstable Road, Redbourn, St.Albans, Herts 
AL3 7PR
 



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Re: [OpenSIPS-Users] rpid_avp and NULL values

2009-09-08 Thread Carlo Dimaggio


Il giorno 22/lug/09, alle ore 11:09, Bogdan-Andrei Iancu ha scritto:

> Hi Carlo,
>
> you must set the line :
>   modparam("auth_db", "load_credentials", "$avp(s:rpid)=rpid")
>
> This line instructs opensips to load at db auth time tht "rpid"  
> field into the $avp(s:rpid) variable.

Hi Bogdan,

Last july I have modified the opensips.cfg adding "modparam("auth_db",  
"load_credentials", "$avp(s:rpid)=rpid")" and all worked.
Now I have executed some test after upgrading to 1.5.3 and I'm  
experiencing with the same problem (if rpid field of user A is NULL,  
after some calls user A takes the rpid of another user B (that is not  
null))
In addition, user A has a different domain of user B.

What do you think about this behaviour?

Thanks and Regards,
Carlo Dimaggio

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Re: [OpenSIPS-Users] Checksum incorrect maybe caused by "UDP" checksum offload

2009-09-08 Thread Stanisław Pitucha

2009/9/8  :
> The mysterious point is that all packets orginating from the interface
> (192.168.20.20) towards UAC(192.168.20.1) have Bad checksum in the UDP
> level.

If you're capturing at the machine that has the 192.168.20.20
interface, then everything is ok. You're seeing wrong checksum on your
traces because it's calculated on your network card instead - so when
you capture the packet the checksum is probably still filled with
zeros.
Just ignore it, or disable the highlighting of bad checksums. If you
disable the offloading, you'll see the result you expect, but you'll
also give your CPU a bit more work on every packet, so for RTP that's
impacting both latency and throughput.

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

2009-09-08 Thread Jeff Pyle

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?


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 ³No
>> information available² when I click on the Media Information link.
>> 
>> In one particular example, the query looks for
>> ³SDjui5c99-324092c7=2B1=2B2b6b0008=2B96cdd538² but the database has
>> ³SDjui5c99-324092c7+1+2b6b0008+96cdd538².  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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Re: [OpenSIPS-Users] Checksum incorrect maybe caused by "UDP" checksum offload

2009-09-08 Thread Thomas Gelf

> The mysterious point is that all packets orginating from the interface  
> (192.168.20.20) towards UAC(192.168.20.1) have Bad checksum in the UDP
> level. This happened with/without RTPproxy, so i'm wondering whether I  
> should care about this problem or not because the sound is conveyed  
> and everything seems okay.

I guess everything is fine. Try to disable checksum offloading (using
ethtool), your capture should look fine then.

Cheers,
Thomas

-- 
 mail: tho...@gelf.net
  web: http://thomas.gelf.net/


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Re: [OpenSIPS-Users] Checksum incorrect maybe caused by "UDP" checksum offload

2009-09-08 Thread Ghaith . ALKAYYEM

I'm using Wireshark

Saúl Ibarra  a écrit :

> How are you making the packet captures? If tcpdump, did you use -s0?
>
>
> --
> /Saúl
> http://www.saghul.net | http://www.sipdoc.net
>
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Re: [OpenSIPS-Users] Checksum incorrect maybe caused by "UDP" checksum offload

2009-09-08 Thread Saúl Ibarra

How are you making the packet captures? If tcpdump, did you use -s0?


-- 
/Saúl
http://www.saghul.net | http://www.sipdoc.net

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[OpenSIPS-Users] Checksum incorrect maybe caused by "UDP" checksum offload

2009-09-08 Thread Ghaith . ALKAYYEM

Hello,

Could anybody tell me what is the problem with such a configuration:


--- 
   
(UAC)192.168.10.1 |===> |192.168.10.10 [OpenSIPS]  
192.168.20.20|===> |(UAC)192.168.20.1 |
--- 
   


The mysterious point is that all packets orginating from the interface  
(192.168.20.20) towards UAC(192.168.20.1) have Bad checksum in the UDP
level. This happened with/without RTPproxy, so i'm wondering whether I  
should care about this problem or not because the sound is conveyed  
and everything
seems okay.

Regards.


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Re: [OpenSIPS-Users] RURI domain on NAT'd endpoints

2009-09-08 Thread Jeff Pyle

All very good points.  I've largely been able to avoid NAT up until now, so
I'm afraid I'm a bit too green to offer any clarification.


- Jeff



On 9/8/09 1:08 AM, "Thomas Gelf"  wrote:

> Thomas Gelf wrote:
>> Jeff Pyle wrote:
>>> if (client_nat_test("3")) {
>>>   force_rport();
>>>   $avp(s:received_uri) = $source_uri;
>>>   if (!is_method("REGISTER")) fix_contact();
>>>   setbflag(7);
>>> }# nat_keepalive() further down after some pref checks
> 
> One more thing: I'm doing client_nat_test("1) in reply routes (should
> make sense), client_nat_test("7") for REGISTERs and client_nat_test("3")
> for other requests. While I'm pretty sure regarding the reply_route part
> (you cannot use test 2 and 4 as Via headers are nothing but copies from
> Request), I'm wondering whether the latter distinction between REGISTER
> and other request is making sense.
> 
> Googling and having a look to Flavios book shows that also others are
> doing so - why?? If test 4 matches, the request would for sure also
> trigger test 2 wouldn't it? So, may I completely skip test 4?
> 
> There is one special case that comes to my mind: does test 2 somehow
> respect the rport parameter? I mean, if topmost Via has a private IP
> and rport set - is test 2 then still triggered?
> 
> Best regards,
> Thomas Gelf


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