RE: [Asterisk-Users] Sipura 3000 dialing noise

2005-05-31 Thread David Phelan
Have you updated with the lastest firmware..
It now does an on-hook forward to asterisk

Dave
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Bishop
Sent: Tuesday, 31 May 2005 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Sipura 3000 dialing noise

Hi all,

We have several sipura 3000's working well for outbound calls, however the
issue we have is that when calls are sent to the Sipura with
Dial(SIP/${EXTEN:[EMAIL PROTECTED]) the Sipura does a SIP answer immediately and
then proceeds with the call in band therefore sending dialing sounds back
to the caller. Other SIP gateways we have notably the Vegastream and others
do not do a SIP answer until the call is successfully connected to the
called party.

Any ideas?
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Re: [Asterisk-Users] Sipura 3000 dialing noise

2005-05-31 Thread Eric Bishop
Yeah tried it. 

Unfortunately I need this feature in reverse. I need the call to stay
on hook when going from Asterisk to Sipura. Staying onhook from Sipura
to Asterisk workd fine.



On 5/31/05, David Phelan [EMAIL PROTECTED] wrote:
 Have you updated with the lastest firmware..
 It now does an on-hook forward to asterisk
 
 Dave
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Eric Bishop
 Sent: Tuesday, 31 May 2005 3:45 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Sipura 3000 dialing noise
 
 Hi all,
 
 We have several sipura 3000's working well for outbound calls, however the
 issue we have is that when calls are sent to the Sipura with
 Dial(SIP/${EXTEN:[EMAIL PROTECTED]) the Sipura does a SIP answer immediately 
 and
 then proceeds with the call in band therefore sending dialing sounds back
 to the caller. Other SIP gateways we have notably the Vegastream and others
 do not do a SIP answer until the call is successfully connected to the
 called party.
 
 Any ideas?
 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 --
 No virus found in this incoming message.
 Checked by AVG Anti-Virus.
 Version: 7.0.322 / Virus Database: 267.3.0 - Release Date: 30/05/2005
 
 
 --
 No virus found in this outgoing message.
 Checked by AVG Anti-Virus.
 Version: 7.0.322 / Virus Database: 267.3.0 - Release Date: 30/05/2005
 
 

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Re: [Asterisk-Users] Sipura 3000 dialing noise

2005-05-31 Thread David John Walsh
Eric

A completly off topic response (and not even a response in that I'm
asking you a question - sorry)

you say that you have several 3000 devices and you show your dial string as :

Dial(SIP/${EXTEN:[EMAIL PROTECTED])

Is the sipura1 section referencing a single sipura or the group of
several.  The only reason that I ask if it is the latter - how are
you grouping them.

Thanks for you response if i figure the answer to your question I
will post back.

David

On 31/05/05, Eric Bishop [EMAIL PROTECTED] wrote:
 Hi all,
 
 We have several sipura 3000's working well for outbound calls, however
 the issue we have is that when calls are sent to the Sipura with
 Dial(SIP/${EXTEN:[EMAIL PROTECTED]) the Sipura does a SIP answer immediately
 and then proceeds with the call in band therefore sending dialing
 sounds back to the caller. Other SIP gateways we have notably the
 Vegastream and others do not do a SIP answer until the call is
 successfully connected to the called party.
 
 Any ideas?
 ___
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 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
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[Asterisk-Users] Sipura 3000 dialing noise

2005-05-30 Thread Eric Bishop
Hi all,

We have several sipura 3000's working well for outbound calls, however
the issue we have is that when calls are sent to the Sipura with
Dial(SIP/${EXTEN:[EMAIL PROTECTED]) the Sipura does a SIP answer immediately
and then proceeds with the call in band therefore sending dialing
sounds back to the caller. Other SIP gateways we have notably the
Vegastream and others do not do a SIP answer until the call is
successfully connected to the called party.

Any ideas?
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