asterisk users - Nov 2004 - direct asterisk to asterisk SIP calls without external SIP provider

Hi all,

I have a small system of two hardware boxes (residential gateways)
running Linux with Asterisk on them. Each RG has some FXS ports to which
analog telephones can be connected.

I already had a working system including an external SIP provider, where
both RGs would register to that provider with a telephone number and
they could call each other via that telephone number. Each RG had a line
register => <telephone number>:<password>@sip.myprovider.com
in sip.conf. I also included a section

[mysipprovider]
type=peer
context=fromINTERNET
host=sip.myprovider.com

and used Dial(SIP/<telephone number>@mysipprovider) in my dialplan.
Context fromINTERNET only consisted of
exten => s,1,Dial(FXSport/0,,tH)

This setup was working great, but now I want to have the two RGs
communicate directly to each other via SIP (such that an analog phone on
one RG can call the other RG), leaving out an external SIP provider
completely.

I tried to just remove the register => line from sip.conf and use
Dial(SIP/<other RG's IP address>) in my dialplan, but asterisk debug
on
the called RG would say that it was looking for an empty user name and
did not find it (I am not at my work any more now, I don't remember the
exact reply).

I also tried using SIPp (sipp.sourceforge.net) on my Windows XP laptop,
via the command
sipp <one RG's IP address> -s <telephone number> -m 1
but that only resulted in an ethereal trace showing two
Request: INVITE sip:<telephone number>@<RG's IP address>:5060,
with
session description
messages sent from my laptop, followed by a
Status: 404 Not Found
message from the RG to my laptop, and finally one
Request: ACK sip:<telephone number>@<RG's IP address>:5060
message.h

Leaving out the -s <telephone number> command line option would give
similar messages with 'service' instead of <telephone number> as
the
destination, again resulting in a 404 reply message.

I wonder if I would need to include any extra peer/user/friend sections
in sip.conf to make this setup work, or even still have a register =>
line to the other RG somehow.

Thanks in advance for any help you can offer me.

With kind regards,
Rene Kramer.

If you are trunking between two * servers, why not use IAX instead?
Besides, IAX will handle the NAT translations, you probably have at each RG,
better than SIP.

Lyle

----- Original Message ----- 
From: "Kramer, R.D.J." <R.D.J.Kramer@tue.nl>
To: <asterisk-users@lists.digium.com>
Sent: Friday, November 26, 2004 6:15 PM
Subject: [Asterisk-Users] direct asterisk to asterisk SIP calls
withoutexternal SIP provider


Hi all,

I have a small system of two hardware boxes (residential gateways)
running Linux with Asterisk on them. Each RG has some FXS ports to which
analog telephones can be connected.

I already had a working system including an external SIP provider, where
both RGs would register to that provider with a telephone number and
they could call each other via that telephone number. Each RG had a line
register => <telephone number>:<password>@sip.myprovider.com
in sip.conf. I also included a section

[mysipprovider]
type=peer
context=fromINTERNET
host=sip.myprovider.com

and used Dial(SIP/<telephone number>@mysipprovider) in my dialplan.
Context fromINTERNET only consisted of
exten => s,1,Dial(FXSport/0,,tH)

This setup was working great, but now I want to have the two RGs
communicate directly to each other via SIP (such that an analog phone on
one RG can call the other RG), leaving out an external SIP provider
completely.

I tried to just remove the register => line from sip.conf and use
Dial(SIP/<other RG's IP address>) in my dialplan, but asterisk debug
on
the called RG would say that it was looking for an empty user name and
did not find it (I am not at my work any more now, I don't remember the
exact reply).

I also tried using SIPp (sipp.sourceforge.net) on my Windows XP laptop,
via the command
sipp <one RG's IP address> -s <telephone number> -m 1
but that only resulted in an ethereal trace showing two
Request: INVITE sip:<telephone number>@<RG's IP address>:5060,
with
session description
messages sent from my laptop, followed by a
Status: 404 Not Found
message from the RG to my laptop, and finally one
Request: ACK sip:<telephone number>@<RG's IP address>:5060
message.h

Leaving out the -s <telephone number> command line option would give
similar messages with 'service' instead of <telephone number> as
the
destination, again resulting in a 404 reply message.

I wonder if I would need to include any extra peer/user/friend sections
in sip.conf to make this setup work, or even still have a register =>
line to the other RG somehow.

Thanks in advance for any help you can offer me.

With kind regards,
Rene Kramer.
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