[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #3373 - 14 msgs

Jain, Sonal Sonal.Jain at Sterlingbancorp.com
Thu Apr 8 12:42:46 MST 2004


Can anybody recommend a good web interface for asterisk that actually works.
I am looking for a web interface that can show how many callers are on the phone, should be able to transfer the calls and disconnect. I have tried using the flash operator but has been unsuccessful in making it work.
thanks

 -----Original Message-----
From: 	asterisk-users-admin at lists.digium.com [mailto:asterisk-users-admin at lists.digium.com]  On Behalf Of asterisk-users-request at lists.digium.com
Sent:	Thursday, April 08, 2004 3:30 PM
To:	asterisk-users at lists.digium.com
Subject:	Asterisk-Users digest, Vol 1 #3373 - 14 msgs

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Today's Topics:

   1. Re: can't hear vm audio (Mark Phillips)
   2. Re: Zapata required? (Mark Phillips)
   3. Re: TigerJet ISDN card (Michael Welter)
   4. Using Skinny with a 7905G phone (Chris Barnett)
   5. Re: Asterisk & 3com nbx 100 support (Eric Wieling)
   6. Re: Siemens EWSD 13 (CW_ASN)
   7. Auto Attendant?? (James Moran)
   8. Re: TigerJet ISDN card (Michael Welter)
   9. FreeBSD port of asterisk (David W. Chapman Jr.)
  10. TDM Stater kit all working - WOOHOO - wondering about Asterisk FAX Support (Kyle Hagan)
  11. Re: RE: RxFax/spandsp: not disconnecting (Derek)
  12. Local Calling Area database? (Scott Laird)
  13. Re: ADPCM 4-bit, 6 kHz (Steven Critchfield)
  14. RE: H.323 Seg faulting (Derek Samford)

--__--__--

Message: 1
Date: Thu, 8 Apr 2004 13:03:59 -0400 (EDT)
Subject: Re: [Asterisk-Users] can't hear vm audio
From: "Mark Phillips" <kc2eni at nyc-ares.org>
To: asterisk-users at lists.digium.com
Reply-To: asterisk-users at lists.digium.com

OK, this is too damn freaky!

Now it work but I didn't do anything.

Actually, that's not quite true. I found that if I do modprobe zaptel then
modprobe wcfxo then ztcfg the vm audio works. If I do service zaptel start
it doesn't.

What gives?


> So I've been fighting to get the X100P working. A battle which I've kinda
> won but not without a cost.
>
> Before I won the Zaptel battle I was able to hear all of the messages that
> asterisk plays. For example, when I'm accessing VoiceMail I would have
> been requested to input my password. This did work but now it doesn't.
> Asterisk does show that it is playing the file but no audio is heard.
>
> I have audio on regular SIP based calls as well as IAX based ones. I'me
> not getting and audio when I make a ZAP call.
>
> Ideas?
>
> Mark
>
>
>
> G7LTT/KC2ENI
> Mark Phillips
>
>
>
> G7LTT/KC2ENI
> Mark Phillips
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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
>


G7LTT/KC2ENI
Mark Phillips

--__--__--

Message: 2
Date: Thu, 8 Apr 2004 13:05:06 -0400 (EDT)
Subject: Re: [Asterisk-Users] Zapata required?
From: "Mark Phillips" <kc2eni at nyc-ares.org>
To: asterisk-users at lists.digium.com
Reply-To: asterisk-users at lists.digium.com

You did unhash the ztdummy in the Makefile before compiling it right?

> Steven Kokinos wrote:
>
>>Ho do I go about loading the ztdummy driver after unloading zap?
>>
>>
>>
> $ su -
> # modprobe ztdummy
>
>
>>Thanks,
>>
>>-Steve
>>
>>
>>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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
>


G7LTT/KC2ENI
Mark Phillips

--__--__--

Message: 3
Date: Thu, 08 Apr 2004 11:23:31 -0600
From: Michael Welter <mike at introspect.com>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] TigerJet ISDN card
Reply-To: asterisk-users at lists.digium.com

In Linux config, CAPI2.0 support is under the "Active ISDN Cards" 
category.  Does this mean CAPI will not work with static cards?

Thanks,

Mark Phillips wrote:

> Is it CAPI compliant? if so yes
> 
> 
> 
>>Is there any Linux/* support for the TigerJet ISDN card?
>>
>>-brian
>>
> 
> 
> 
> G7LTT/KC2ENI
> Mark Phillips
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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
> 
> 

-- 
Michael Welter
Introspect Telephony Corp.
Denver, Colorado
+1 303 674 2575
mike at introspect.com
www.introspect.com



--__--__--

Message: 4
Date: Thu, 8 Apr 2004 18:31:07 +0100
From: "Chris Barnett" <chris.barnett at insphire.com>
To: <asterisk-users at lists.digium.com>
Subject: [Asterisk-Users] Using Skinny with a 7905G phone
Reply-To: asterisk-users at lists.digium.com

Hi All,

I'm trying to get a Cisco 7905g IP Phone to work with our Asterisk
server but I'm having problems getting the phone to answer a call or
make a call. I'm using the stable branch of the asterisk CVS on a RH9
box.

I have got the phone to register with * and it retrieves it's extension
number, date & time etc., but when I pick the handset up it's just dead
silence (but I get tones when pressing the numbers on the keypad). There
are no debug messages or otherwise shown on the * console (even with
very verbose turned on and skinny debugging turned on).

If I ring the phone from a SIP PC softphone (X-Ten lite) the phone rings
but continues to ring even when the handset is picked up - until the 20
seconds timeout occurs and the call is transferred to the extensions
voicemail box. When the call is made I do see debug messages telling me
the call is being made and sent to the skinny extension.

I have experimented with allowing and disallowing various formats to no
avail. I'm sure I've missed something out but after many days and hours
searching every resource I can find I haven't been able to progress any
further - any ideas what I've done wrong or missed out - or is what I am
trying to achieve simply not possible at the moment.

Below is my skinny.conf;

[general]
port =3D 2000
Bindaddr =3D 0.0.0.0
dateFormat =3D D-M-Y
keepAlive =3D 120
; allow =3D all
; disallow =3D

[Ext104]
Device=3DSEP000E386DCF06
Version=3DP021114C	; version number from the phone
Nat=3D0
Callerid=3D"Cisco IP Phone" <104>
Mailbox=3D1004
Callwaiting=3D1
Transfer=3D1
Threewaycalling=3D1
Context=3Ddefault
Line =3D> 104
Host=3D192.168.1.55

And this is the section from extensions.conf;

<snip>

Exten =3D> 104,1,Dial(Skinny/104 at Ext104,20,tr)
Exten =3D> 104,2,VoiceMail,u104
Exten =3D> 104,102,VoiceMail,b104

<snip>


Many thanks,

Chris Barnett

--__--__--

Message: 5
Subject: Re: [Asterisk-Users] Asterisk & 3com nbx 100 support
From: Eric Wieling <eric at fnords.org>
To: asterisk-users at lists.digium.com
Organization: BTEL Consulting
Date: Thu, 08 Apr 2004 12:33:38 -0500
Reply-To: asterisk-users at lists.digium.com

Try www.google.com "site:lists.digium.com NBX" without the quotes, of
course.  There's also a google search box on the same page you use to
sign up for the mailing lists, but it never worked right for me.

On Thu, 2004-04-08 at 11:59, Jeremy Koski wrote:
> No. The only way I found to search the archives was month by month.
> 
> There's an option to download all of the mailing list archives, but its
> 129MB...
> 
> On Thu, 8 Apr 2004, Eric Wieling wrote:
> 
> > On Thu, 2004-04-08 at 10:30, Jeremy Koski wrote:
> > > Does anybody know if the 3com NBX 100 phones will work with
> > > Asterisk? The phones require a boot image to be sent either through
> > > layer2 or layer3 before they will function properly after being powered
> > > on each time.
> >
> > You didn't see the information when you searched the mailing list
> > archives?

-- 
          Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."


--__--__--

Message: 6
From: "CW_ASN" <cw_asn at fibertel.com.ar>
To: <asterisk-users at lists.digium.com>
Subject: Re: [Asterisk-Users] Siemens EWSD 13
Date: Thu, 8 Apr 2004 14:40:26 -0300
Reply-To: asterisk-users at lists.digium.com

In fact, with EWSD V13 you can't remove CRC4 in PRI mode.

Regards,

Gus

----- Original Message -----
From: "Storer, Darren" <starusers at comgate.tv>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, April 07, 2004 8:32 PM
Subject: RE: [Asterisk-Users] Siemens EWSD 13


> Hi,
>
> I had exactly the same symptoms today with a co-located * connected to a
> Public Switch here in the UK. The problem was solved by insisting that the
> Telco turned on CRC4 at their end and then, after an 'init 6', layer two
> settled down on both systems.
>
> I was taught that if you are connecting to a full specification Q.931
> circuit, CRC4 should be enabled by default; in the event that one end does
> not support CRC4 the other end should auto-negotiate back and the circuit
> should still align without problems. Having said all of this I have yet to
> see auto-negotiation of CRC4 on any equipment (Public Network or CPE) and
> suspect that I was not told the truth in the first place...
>
> Selection of CRC4 seems to be random from Telco to Telco even on an
install
> by install basis within the same Carrier. It's the first thing to check
when
> new kit appears to be unstable..
>
> HTH
>
> Darren
> --
> Comgate
> Telco>Internet<Broadcast
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of
> asterisk at geek.be
> Sent: 07 April 2004 14:59
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Siemens EWSD 13
>
>
> Hi all,
>
> Has anyone got any experience with hooking Asterisk up with a
> Siemens EWSD 13 switch over a E1/PRI ?
> We're located in Belgium (Europe) and one of our telecom partners
> uses this switch.
>
> We connected one of our TE410P ports with their switch, but the status
> light on the TE410P card keeps blinking red.
> On their side they are getting a DSA (distance service alarm) error, so
> this normally means the devices 'see' eachother.. but there are still
> problems with the signalling.
>
> Our config below is the same as we are using for MCI, one of our other
> telecom partners.
>
> We tried changing the LBO and timing, but no luck.
> As you see the signalling is carried over channel 16 (default).
>
> TX and RX have also been regularly switched, so no luck..
>
> Their switch is providing the timing.
>
> The telecom operator has double checked the asterisk config several
> times, and it's conform to their setup.
>
> The only thing they couldn't find in the Asterisk config is a
> 'multiframing' option. But I presume this is automatically detected or
> set by default ?
> They also tried normal/single(?) framing, but no difference.
>
> The card has also been tested with our MCI E1, and works flawlessly, so
> no hardware issue.
>
> Anyone got any further ideas ?
>
> Any info or help greatly appreciated!
>
> Our config,
>
> *** zaptel.conf ***
> span=1,1,6,ccs,hdb3,crc4,yellow
> bchan=1-15
> bchan=17-31
> dchan=16
>
> *** zapata.conf ***
> [channels]
> switchtype=euroisdn
> signalling=pri_cpe
> pridialplan=unknown
>
> group=1
> channel => 1-15,17-31
>
> <other zapata standard config>
>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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




--__--__--

Message: 7
From: James Moran <jmoran at potentialtech.com>
To: Asterisk <asterisk-users at lists.digium.com>
Organization: Potential Technologies
Date: Thu, 08 Apr 2004 13:48:24 -0400
Subject: [Asterisk-Users] Auto Attendant??
Reply-To: asterisk-users at lists.digium.com

I'm having trouble finding documentation for the auto attendant does
anyone have an idea where there might be some???


--__--__--

Message: 8
Date: Thu, 08 Apr 2004 11:51:56 -0600
From: Michael Welter <mike at introspect.com>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] TigerJet ISDN card
Reply-To: asterisk-users at lists.digium.com

In Linux config, CAPI2.0 support is under the "Active ISDN Cards" 
category.  Does this mean CAPI will not work with _passive_ cards?

Michael Welter wrote:

> In Linux config, CAPI2.0 support is under the "Active ISDN Cards" 
> category.  Does this mean CAPI will not work with static cards?
> 
> Thanks,
> 
> Mark Phillips wrote:
> 
>> Is it CAPI compliant? if so yes
>>
>>
>>
>>> Is there any Linux/* support for the TigerJet ISDN card?
>>>
>>> -brian
>>>
>>
>>
>>
>> G7LTT/KC2ENI
>> Mark Phillips
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at 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
>>
>>
> 

-- 
Michael Welter
Introspect Telephony Corp.
Denver, Colorado
+1 303 674 2575
mike at introspect.com
www.introspect.com



--__--__--

Message: 9
Date: Thu, 8 Apr 2004 12:55:11 -0500
From: "David W. Chapman Jr." <dwcjr at inethouston.net>
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] FreeBSD port of asterisk
Reply-To: asterisk-users at lists.digium.com

In our FreeBSD port of Asterisk, we have a lot of local patches and I 
was wondering if it were possible to get some of them merged into the 
Asterisk source base.

Thanks

-- 
David W. Chapman Jr.
dwcjr at inethouston.net	Raintree Network Services, Inc. <www.inethouston.net>

--__--__--

Message: 10
From: "Kyle Hagan" <khagan at nuvoinc.com>
To: <asterisk-users at lists.digium.com>
Date: Thu, 8 Apr 2004 10:54:22 -0700
Subject: [Asterisk-Users] TDM Stater kit all working - WOOHOO - wondering about Asterisk FAX Support
Reply-To: asterisk-users at lists.digium.com

 Finally got my Asterisk (Test) system up and working with a TDM Starter
kit. WOOHOO!!!
Goona be buying the T1 cards soon to fully implement the system and FINALLY
get rid of our old Fujitsu Starlog POS.

Was just wondering about the Fax Support. Features say its not complete. Any
idea when this could be available?


Kyle




--__--__--

Message: 11
From: Derek <derek at incrediblefresh.com>
Subject: Re: [Asterisk-Users] RE: RxFax/spandsp: not disconnecting
Date: Thu, 8 Apr 2004 14:02:27 -0400
To: asterisk-users at lists.digium.com
Reply-To: asterisk-users at lists.digium.com

I installed rxfax and spandsp. Everything looks good except when 
receiving a fax, only the first page is captured, even though the 
sending machine seems to be transmitting all pages. Am I doing 
something wrong?

-D

Derek Irwin
Information Technology
IncredibleFresh Inc.
Naples, FL

"I've been booting my Windows box daily for the last year. The 
computer's OK, but my shoe's starting to wear out."

On Mar 31, 2004, at 11:01 AM, Steve Underwood wrote:

> Reynaldo Simbulan wrote:
>
>> Hi Steve,
>>
>> I am having this problem in which RxFax is still holding the file 
>> after
>> receiving a complete fax. Somehow the zap channel is still active but 
>> on the
>> fax client it was sent successfully.
>> If you call the line it is still busy.
>>
>>
> spandsp-0.0.1k.tar.gz and updated app_rxfax.c and app_txfax.c files 
> are available for download. They address this disconnect issue, and 
> have a few other minor tweaks.
>
> Regards,
> Steve
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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
>


--__--__--

Message: 12
To: asterisk-users at lists.digium.com
From: Scott Laird <scott at sigkill.org>
Date: Thu, 8 Apr 2004 11:04:48 -0700
Subject: [Asterisk-Users] Local Calling Area database?
Reply-To: asterisk-users at lists.digium.com

Is there an easy way to get information about local calling areas out 
of telcos?  I'm trying to get a list of area codes and prefixes in my 
local calling area out of Verizon, and it looks like they've stopped 
providing the information online.  Is there an easy source that I'm 
missing, or do I need to call them and have them mail me a copy every 
few months?


Scott


--__--__--

Message: 13
Subject: Re: [Asterisk-Users] ADPCM 4-bit, 6 kHz
From: Steven Critchfield <critch at basesys.com>
To: asterisk-users at lists.digium.com
Date: Thu, 08 Apr 2004 13:20:15 -0500
Reply-To: asterisk-users at lists.digium.com

On Tue, 2004-04-06 at 10:49, Steve Underwood wrote:
> Why do people get this uncontrollable urge to post, when the don't know 
> the correct answer? :-)

Having the absolute correct answer isn't always important if it steers
the requester in the right direction of self enlightenment. 

Don't discourage new users too much from answering questions as we need
new blood to carry on the fight of educating the even newer users. I
know I for one have been pretty burnt out on it and skip all but a small
handful of questions a week.
-- 
Steven Critchfield  <critch at basesys.com>


--__--__--

Message: 14
Subject: RE: [Asterisk-Users] H.323 Seg faulting
Date: Thu, 8 Apr 2004 14:25:49 -0400
From: "Derek Samford" <dsamford at netphoneblue.com>
To: <asterisk-users at lists.digium.com>
Reply-To: asterisk-users at lists.digium.com

This is a multi-part message in MIME format.

------_=_NextPart_001_01C41D96.EAEAE4F8
Content-Type: text/plain;
	charset="us-ascii"
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I've placed a bounty on my bug. See
http://bugs.digium.com/bug_view_page.php?bug_id=3D0001334

=20

=20

=20

  _____ =20

From: Derek Samford=20
Sent: Wednesday, April 07, 2004 4:26 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] H.323 Seg faulting

=20

Can someone take a look, tell me if this is a bug, a possible resources
issue, or my own damn fault?

=20

http://bugs.digium.com/bug_view_page.php?bug_id=3D0001381

=20

=20

Thanks,

Derek


------_=_NextPart_001_01C41D96.EAEAE4F8
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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I&#8217;ve placed a bounty on my =
bug. See </span></font><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'><a
href=3D"http://bugs.digium.com/bug_view_page.php?bug_id=3D0001334">http:/=
/bugs.digium.com/bug_view_page.php?bug_id=3D0001334</a><o:p></o:p></span>=
</font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

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<div>

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<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Derek =
Samford <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, April =
07, 2004
4:26 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b>
asterisk-users at lists.digium.com<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Asterisk-Users] =
H.323
Seg faulting</span></font><o:p></o:p></p>

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12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Can someone take a look, tell me if this is a bug, a
possible resources issue, or my own damn =
fault?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><a
href=3D"http://bugs.digium.com/bug_view_page.php?bug_id=3D0001381">http:/=
/bugs.digium.com/bug_view_page.php?bug_id=3D0001381</a><o:p></o:p></span>=
</font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Thanks,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Derek<o:p></o:p></span></font></p>

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