asterisk users - Apr 2005 - How to prevent native bridging between SIP channels

Hello,

how can I prevent Asterisk from trying to create a native bridge between
an incoming call from a SIP provider and an extension attached to a
SIP ATA?

My Asterisk is behind a firewall, and the native bridge invariably fails.

Thanks in advance for any suggestion!

(I DID search the list archives for "native bridge" and found one
similar
query without any replies).

Regards,

Wolf Paul

add

canreinvite=no

to the sip user definition blocks for the SIP provider and for the SIP ATA.

Regards,

Marc

Wolf N. Paul wrote:
> Hello,
> 
> how can I prevent Asterisk from trying to create a native bridge between
> an incoming call from a SIP provider and an extension attached to a
> SIP ATA?
> 
> My Asterisk is behind a firewall, and the native bridge invariably fails.
> 
> Thanks in advance for any suggestion!
> 
> (I DID search the list archives for "native bridge" and found one
similar
> query without any replies).
> 
> Regards,
> 
> Wolf Paul
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
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Hi

Is there any help for me to register my quantium A800 (SIP) with my Asterisk
.

Please help me what should me my Sip.conf
now present i did

[1234567]
type=friend
context=sip
usernamesecretnat=yes
host=dynamic
canreinvite=no
defaultip=XXX.XXX.XXX.XXX
disallow=all
allow=g729
allow=gsm
allow=g723.1
allow=ulaw

and is there any special change need on quintum?


Bashir

Hi Basher,

Currently im using my A800 Quintum registered in my Asterisk SIP server. For
you to register your Quitum to Aterisk, define your asterisk in as proxy and
registrar IP at SIP config at quintum (I can give you the sample config at
private mail). Also setup a user account at sip_additionl.conf in your
asterisk defining the username and password used and defined in the quantum.

Here is the sample in the asterisk:

[199]
username=199
type=friend
secret=199
qualify=no
port=5060
pickupgroupnat=never
mailboxhost=xxx.xxx.xxx.xxx (IP add of your quintum)
dtmfmode=rfc2833
disallowcontext=from-internal
canreinvite=no
callgroupcallerid="Caller ID" <199>
allow
Shall you have more question, your free to ask.

In return I want to ask also if have you tried to managed to register a
D-Link DVG-1402S. I have here a demo unit but I can make it work... I am not
sure what is missing.. anyone here in asterisk users-list?

Regards,

Jessie

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Bashir Ullah -
www.Lamsre.Com
Sent: Monday, April 25, 2005 3:36 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Quantum A800 (SIP) <-> Asterisk Config 

Hi

Is there any help for me to register my quantium A800 (SIP) with my Asterisk
.

Please help me what should me my Sip.conf
now present i did

[1234567]
type=friend
context=sip
usernamesecretnat=yes
host=dynamic
canreinvite=no
defaultip=XXX.XXX.XXX.XXX
disallow=all
allow=g729
allow=gsm
allow=g723.1
allow=ulaw

and is there any special change need on quintum?


Bashir

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Marc Storck <marc.storck@msnetworks.lu> writes in reply to my question:
>add
>
>canreinvite=no
>
>to the sip user definition blocks for the SIP provider and for the SIP ATA.
>
>Regards,
>  
>
Unfortunately, I already have this parameter in the sip user definitons, 
as well as
a "t" option in the Dial command, both of which, according to the 
article on
SIP Media Path in the Asterisk-Wiki, should prevent Asterisk from trying 
to take
itself out of the loop. But it still does :-(

On the other hand, the same article says that Asterisk decides whether 
or not to
take itself out of the media path depends on "many variables" -- I was
hoping to
get some information on some of the _other_ variables, in addition to the
canreinvite=no and the "t"ransfer option to the Dial command.

Regards,

Wolf