RE: RE: CLI using XAP and Sipura SPA 3000 / Linksys3102

["Nigel Giddings" <nigel@xxxxxxxxxxxxxx>]

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Patrick,

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Attached the log for calling 160 (Echo test) over the hosted asterisk
server...

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The SPA 3000 is set to debug 2... Call progress seems to be in there...

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HTH

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Nigel

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<159>[0]Off Hook

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<151>Calling:160@xxxxxxx:0

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<151>[0:0]AUD ALLOC CALL (port=3D16408)

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<151>[0:0]RTP Rx Up

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<159>RSE_DEBUG: reference domain:nat.ipwire.co.uk

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<159>RSE_DEBUG: reference domain:nat.ipwire.co.uk

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<151>CC:CallProgress

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<151>[0:0]ENC INIT 18

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<151>[0:0]RTP Tx Up (pt=3D18->c16fc8cd:38698)

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<151>[0:0]RTCP Tx Up

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<159>RSE_DEBUG: unref domain, nat.ipwire.co.uk

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<159>RSE_DEBUG: reference domain:nat.ipwire.co.uk

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<151>[0:0]RTP Tx Dn

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<151>[0:0]ENC INIT 18

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<151>[0:0]RTP Tx Up (pt=3D18->c16fc8cd:38698)

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<151>CC:Remote Resume

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<151>CC:Connected

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<151>[0:0]RTP Rx 1st PKT @16408(2)

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<151>[0:0]DEC INIT 18

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<151>[0:0]LAT-- 7(2)

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<151>[0:0]LAT-- 5(2)

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<159>[0]On Hook

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<151>[0:0]AUD Rel Call

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<159>RSE_DEBUG: reference domain:nat.ipwire.co.uk

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<151>DLG Terminated

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<159>RSE_DEBUG: unref domain, nat.ipwire.co.uk

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<159>RSE_DEBUG: unref domain, nat.ipwire.co.uk

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<159>RSE_DEBUG: unref domain, nat.ipwire.co.uk

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<159>RSE_DEBUG: last unref for domain nat.ipwire.co.uk

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<151>Sess Terminated

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<151>CC:Clean Up

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<151>--- OBJ POOL STAT ---

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<151>OP:RTPRXB =3D  96 ( 96  192)=20=20=20

<151>OP:RTPREB =3D  40 ( 40   48)

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<151>OP:RTPTXB =3D  64 ( 64  108)=20=20=20

<151>OP:TIMEOU =3D  49 ( 60   40)

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<151>OP:SIPCOR =3D   0 (  1   28)=20=20=20

<151>OP:SIPCTS =3D  32 ( 32  680)

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<151>OP:SIPSTS =3D  30 ( 32 3524)=20=20=20

<151>OP:SIPAUS =3D   5 (  8  340)

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<151>OP:SIPDLG =3D  10 ( 10  140)=20=20=20

<151>OP:SIPSES =3D  12 ( 12 7748)

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<151>OP:SIPREG =3D   3 (  4  228)=20=20=20

<151>OP:SIPLIN =3D   0 (  2  908)

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<151>OP:STUNTS =3D  16 ( 16   68)=20=20

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-----Original Message-----
From: xap_automation@xxxxxxx
[mailto:xap_automation@xxxxxxx] On
Behalf Of Patrick Lidstone
Sent: 11 June 2009 17:16
To: xap_automation@xxxxxxx
Subject: Re: [xap_automation] RE: CLI using XAP and Sipura SPA 3000 /
Linksys3102

=20






At least on my setup, it would appear not... but the config to make
SPA's work with asterisk differs from the standalone - might be worth
downloading a syslogger and pointing your SPA at it & seeing whether
you
do get outgoing call data as well? All James would have to do is get xAP
switchboard to pick up UDP packets from the syslog port and extract the
text - it's all human readable in the form
"...caller<number>...". Happy
to look at doing a standalone conduit type thing too if there is the
interest...

P.

2009/6/11 Kevin Hawkins <yahoogroupskh@xxxxxxx
<mailto:yahoogroupskh@xxxxxxx>
>

Interested here too P ... :-)

Is there a way of picking up the dialled info + call progress as well -
or would that require * ?
It would be nice to have the Sipura standalone integrate with  xAP
switchboard

K


Patrick Lidstone wrote:
>
>
> OK, I've had a quick look, and incoming call information (but not
> outgoing) is sent to syslog (over the network, via UDP), if you have
> the SPA3k configured right. It shouldn't be too difficult to convert
> those into xAP messages. Which caller id schema do your existing
> solutions use, Nigel? (A sample message from a xAP viewer would be
> useful).
>
> Patrick
>
> 2009/6/11 Patrick Lidstone <patrick@xxxxxxx
<mailto:patrick@xxxxxxx>=20

> <mailto:patrick@xxxxxxx <mailto:patrick@xxxxxxx>
>>

>
>     Re. asterisk - I've been running asterisk for about 8 years now,
>     and it is nothing short of awesome - in the last couple of years,
>     it's got to the point that it's rock solid. I'm a big fan of the
>     astlinux build, which can run from compact flash, reducing power
>     consumption and improving reliability.
>
>     In terms of Nigel's original ask - it looks like incoming calls
>     are notified by SNMP. I'll have a look into whether that's
>     actually the case, and if so, what we can do to catch the traps. A
>     quick and easy hack would be to simply attach a meteor caller id
>     unit to the phone side of the SPA3k and use one of several
>     existing apps to catch the caller id data and retransmit is as a
>     xAP message. This will work because SPA delivers caller id on the
>     analogue side (albeit in the US format by default, but that makes
>     little/no difference).
>
>     HTH
>     Patrick
>
>     2009/6/11 Paul Gale <groups2@xxxxxxx
<mailto:groups2@xxxxxxx>=20

>     <mailto:groups2@xxxxxxx <mailto:groups2@xxxxxxx>
>>

>
>
>
>         Hi Nigel - can't help with the time aspect! But did you
>         consider running one of the easy-to-install asterisk packages
>         such as FreePBX or Trixbox in a VMWare server environment - no
>         extra hardware needed and works well (so far in my testing).
>
>
>
>         Of course, you still need a bit of time to install it all! And
>         still maybe overkill for just getting CLI from the VOIP call?
>
>
>
>         Paul.
>
>
>
>         *From:* xap_automation@xxxxxxx
<mailto:xap_automation@xxxxxxx>=20
>         <mailto:xap_automation@xxxxxxx
<mailto:xap_automation@xxxxxxx>
>
>         [mailto:xap_automation@xxxxxxx
<mailto:xap_automation@xxxxxxx>=20
>         <mailto:xap_automation@xxxxxxx
<mailto:xap_automation@xxxxxxx>
>] *On Behalf Of *Nigel
>         Giddings
>         *Sent:* 11 June 2009 08:14
>
>         *To:* xap_automation@xxxxxxx
<mailto:xap_automation@xxxxxxx>=20
>         <mailto:xap_automation@xxxxxxx
<mailto:xap_automation@xxxxxxx>
>

>         *Subject:* [xap_automation] CLI using XAP and Sipura SPA 3000

>         / Linksys3102
>
>
>
>
>
>
>         Hi Patrick,
>
>
>
>         Thanks for picking up on the thread...
>
>
>
>         The only information I could find is at this link
>
>
>
>         http://www.voipuser.org/forum_topic_1213.html
<http://www.voipuser.org/forum_topic_1213.html>=20
>         <http://www.voipuser.org/forum_topic_1213.html
<http://www.voipuser.org/forum_topic_1213.html>
>
>
>
>
>         There is mention of xAP but it looks as if it never came to
>         anything?
>
>
>
>         I'm not using Asterisk. I have played in the past and it is
>         very good but can't really justify the overhead (H/W &
time)
>         at the moment... I am using the SPA with a VoIP account on a

>         www.gradwell.com/voip <http://www.gradwell.com/voip>
<http://www.gradwell.com/voip <http://www.gradwell.com/voip>
> service

>         at the moment just while I do some work with a company, easy
>         way for them to pick up the bills and of course it is cheaper
>         to make calls 'on-net'.
>
>
>
>         I do have a small PBX and the ATA presents as an incoming
>         line.... SWMBO is now confused because if we receive an
incoming
>         VoIP call there is no announcement, where as there is on the
>         PSTN....
>
>
>
>         Have to note here that the voice announcements get a very high
>         SWMBO approval rating...
>
>
>
>         TIA
>
>
>
>         Nigel
>
>
>
>
>
>
>
>
>
>         PS I have sent this as a separate mail as I can't send a
>         response with this attached for some reason....
>
>
>
>
>
>
>
>
>




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