asterisk users - Jul 2006 - Sipura ATA's Forwarding PSTN Calls to Asterisk

I'm working with a Sipura 3000 ATA here. I'm trying to get incoming PSTN
calls on the FXO port to go automatically to Asterisk. I have it working, but I
had to configure the ATA to register with Asterisk, which means that all calls
are being sent to Asterisk with a caller id of the username used to register
with Asterisk.

I want the real caller ID to be sent to Asterisk, which means I don't want
the ATA to register. The badly written Sipura docs aren't clear about how to
do this. Anyone set this up?

Doug.

> I want the real caller ID to be sent to Asterisk, which means I don't
want the ATA to register. The badly written Sipura docs aren't clear about
how to do this. Anyone set this up?
>
I am having the same problem...

Cheers,
Jorge Mauricio
-- 
<blog>
http://djmaucom.blogspot.com
http://jmauricio.blogspot.com
</blog>

On Fri, 2006-07-21 at 09:23 -0600, Douglas Garstang
wrote:
> I'm working with a Sipura 3000 ATA here. I'm trying to get incoming
PSTN calls on the FXO port to go automatically to Asterisk. I have it working,
but I had to configure the ATA to register with Asterisk, which means that all
calls are being sent to Asterisk with a caller id of the username used to
register with Asterisk.
> 
> I want the real caller ID to be sent to Asterisk, which means I don't
want the ATA to register. The badly written Sipura docs aren't clear about
how to do this. Anyone set this up?
> 
Yes, have you set up the Sipura as show in the document I sent you
privately 2 weeks ago? Because it all works on my systems.

 
-- 
Dave Cotton <dcotton@linuxautrement.com>

> -----Original Message-----
> From: Dave Cotton [mailto:dcotton@linuxautrement.com]
> Sent: Friday, July 21, 2006 9:51 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to
> Asterisk
> 
> 
> On Fri, 2006-07-21 at 09:23 -0600, Douglas Garstang wrote:
> > I'm working with a Sipura 3000 ATA here. I'm trying to get 
> incoming PSTN calls on the FXO port to go automatically to 
> Asterisk. I have it working, but I had to configure the ATA 
> to register with Asterisk, which means that all calls are 
> being sent to Asterisk with a caller id of the username used 
> to register with Asterisk.
> > 
> > I want the real caller ID to be sent to Asterisk, which 
> means I don't want the ATA to register. The badly written 
> Sipura docs aren't clear about how to do this. Anyone set this up?
> > 
> 
> Yes, have you set up the Sipura as show in the document I sent you
> privately 2 weeks ago? Because it all works on my systems.
Thanks, but no, because I have absolutely no idea what to do with that file, or
how to read it.

Doug.

Douglas Garstang wrote:
> I'm working with a Sipura 3000 ATA here. I'm trying to get incoming
PSTN calls on the FXO port to go automatically to Asterisk. I have it working,
but I had to configure the ATA to register with Asterisk, which means that all
calls are being sent to Asterisk with a caller id of the username used to
register with Asterisk.
> 
> I want the real caller ID to be sent to Asterisk, which means I don't
want the ATA to register. The badly written Sipura docs aren't clear about
how to do this. Anyone set this up?
> 
That's not correct.

My SPA-3000 FXO port registers with my Asterisk server, and when the 
PSTN calls come in, it uses the incoming caller's CallerID for the call.

Sounds like you have something misconfigured.

B.

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This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

> -----Original Message-----
> From: Brian Capouch [mailto:brianc@palaver.net]
> Sent: Friday, July 21, 2006 11:20 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to
> Asterisk
> 
> 
> Douglas Garstang wrote:
> > I'm working with a Sipura 3000 ATA here. I'm trying to get 
> incoming PSTN calls on the FXO port to go automatically to 
> Asterisk. I have it working, but I had to configure the ATA 
> to register with Asterisk, which means that all calls are 
> being sent to Asterisk with a caller id of the username used 
> to register with Asterisk.
> > 
> > I want the real caller ID to be sent to Asterisk, which 
> means I don't want the ATA to register. The badly written 
> Sipura docs aren't clear about how to do this. Anyone set this up?
> > 
> 
> That's not correct.
> 
> My SPA-3000 FXO port registers with my Asterisk server, and when the 
> PSTN calls come in, it uses the incoming caller's CallerID 
> for the call.
> 
> Sounds like you have something misconfigured.
Here's my invite Brian. The From: is always going to contain the auth id the
ATA used to register with Asterisk.

INVITE sip:2944009@xxx.187.130.42 SIP/2.0
Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK208dd2be;rport
From: "Cody XXX-527-7107"
<sip:atacody1@xxx.187.142.203>;tag=as3a94778b
To: <sip:2944009@xxx.187.130.42>
Contact: <sip:atacody1@xxx.187.142.203>
Call-ID: 6946cb0d3fc1b6d6763e1dea7e5c1d8c@xxx.187.142.203
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Cody XXX-527-7107"
<sip:atacody1@xxx.187.142.203>;privacy=off;screen=no
Date: Fri, 21 Jul 2006 17:44:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 28771 28771 IN IP4 xxx.187.142.203
s=session
c=IN IP4 xxx.187.142.203
t=0 0
m=audio 21652 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

> -----Original Message-----
> From: Dave Cotton [mailto:dcotton@linuxautrement.com]
> Sent: Friday, July 21, 2006 12:25 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to
> Asterisk
> 
> 
> On Fri, 2006-07-21 at 14:15 -0400, Brian Capouch wrote:
> 
> > I suspect there is an option somewhere on one of the "PSTN" 
> tabs on the 
> > SPA-3000 that has to be set correctly to enable the 
> pass-through.  I 
> > don't have time right now to play around with it--my system 
> is working 
> > just fine :-)  192.168.1.1 is my Asterisk server, and the ATA is at 
> > 192.168.1.113.
> 
> Yes, just set 
> 
> PSTN CID For VoIP CID:
> 
> to guess what:- YES.
Guess what: I already have it set to YES.

> -----Original Message-----
> From: Brian Capouch [mailto:brianc@palaver.net]
> Sent: Friday, July 21, 2006 12:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to
> Asterisk
> 
> 
> Douglas Garstang wrote:
> 
> > 
> > Here's my invite Brian. The From: is always going to 
> contain the auth id the ATA used to register with Asterisk.
> > 
> > INVITE sip:2944009@xxx.187.130.42 SIP/2.0
> > Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK208dd2be;rport
> > From: "Cody XXX-527-7107" 
> <sip:atacody1@xxx.187.142.203>;tag=as3a94778b
> > To: <sip:2944009@xxx.187.130.42>
> > Contact: <sip:atacody1@xxx.187.142.203>
> 
> And here's one from a call I just placed.  Note the dissimilarities 
> between the From: and Contact: fields on mine and the snippet 
> of yours 
> shown above.
> 
> I suspect there is an option somewhere on one of the "PSTN" 
> tabs on the 
> SPA-3000 that has to be set correctly to enable the pass-through.  I 
> don't have time right now to play around with it--my system 
> is working 
> just fine :-)  192.168.1.1 is my Asterisk server, and the ATA is at 
> 192.168.1.113.
> 
> "AstIn" is the display name I chose for the registration, btw.
> 
> B.
> 
> INVITE sip:s@192.168.1.1 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.113:5061;branch=z9hG4bK-3c04a2ec
> From: "Capouch B"
<sip:12192538181@192.168.1.1>;tag=5e2ab9e072a1a2cco1
> To: <sip:s@192.168.1.1>
> Call-ID: 1db84670-5816ff7c@192.168.1.113
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: AstIn <sip:12192538181@192.168.1.113:5061>
Well in that case, what's the point in having the ATA register with
Asterisk? You just direct all PSTN->VOIP calls to Asterisk with their PSTN
CID and destination, and VOIP->PSTN calls you just route with
Dial(${EXTEN}@ata).

I tried removing the registration info, thinking maybe that would make the ATA
not register, and send all PSTN->VOIP calls to Asterisk, but then the ATA
didn't even answer the incoming PSTN call.

Doug.

> -----Original Message-----
> From: Jonathan Attwood [mailto:jmattwood@gmail.com]
> Sent: Saturday, July 22, 2006 12:45 PM
> To: radamson@routers.com; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to
> Asterisk
> 
> 
> For the OP, do you have an entry against "Display Name" on the
PSTN
> tab, whilst logged in as admin/advanced? If I have an entry in this,
> what you describe happens for me. If the field is empty, CLID is sent
> correctly to my Asterisk box.
Thanks. Already tried that. Didn't work for me.