Re: [asterisk-users] I used to use an Asterisk server, but now it is overkill, ...

[Alan Lord]

Ronald Wiplinger wrote:
 I had installed in the office an Asterisk server, but the company is
 gone and I could keep the server.
 
 However, for my family with three members and two phone lines this
 server is overkill. I am looking for a compact solution, which is more
 suitable for me.
 
 I want a small  silent box, which can connect two phone lines and 6
 internal VoIP phones and about 6 external VoIP phones.
 I would like to have:
 1. Announcements for callers (dial the extension number)
 2. voice mail with mail forwarding
 3. wakeup call
 4. pickup group
 5. call forwarding after 20 seconds, ...
 6. ISN support, Sipbroker support
 7. remote gateway support
 
 I guess that is all what I would need at home.

Hi Ronald,

I built my own small low-power server that runs Linux and provides a 
host of services for our home and our home businesses. Asterisk is just 
one of the functions and it runs very happily (well, the box has *never* 
stopped or needed rebooting apart from when I wanted to change something).

The VIA C7 board I bought runs at about 7W, has no fan and I have even 
downclocked it from 1.2Ghz to 1Ghz.

I have written some articles on my blog about it, here's the first article:

http://www.theopensourcerer.com/2007/09/08/untangle-asterisk-pbx-and-file-server-all-in-one/

For the other instalments use the tag cloud and Asterisk.

With the new Atom processor you might get even better power consumption 
although I have read somewhere that the associated chipsets for the Atom 
are very thirsty (+20W)...

Hope this helps.

Alan


-- 
The way out is open!
http://www.theopensourcerer.com


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Re: [asterisk-users] OT: Asterisk on fitPC

[Gordon Henderson]

On Tue, 12 Aug 2008, mail-lists wrote:


Hm. $300 in the US and the UK disty is selling them for just short of
£240, so they can go stuff themselves, low-power or not. (I buy 1GHz

 systems with 1GB of RAM, running at 15W for half that. No drive though)

Gordon,

If you don't mind my asking: What do you get for $150.00 ?


According to Google this morning:

150 U.S. dollars = 79.0305585 British pounds

But because I live in the UK and deal exclusively in GBP, I meant half the 
UK price. That might not have been clear in my email.


So, I can't get a lot for that ($150/£79), but for £129 I can get a 1GHz 
Via processor in a nice thin client box with an external brick PSU and 
1GB of RAM.


http://linitx.com/viewproduct.php?prodid=12008

For me, it's the perfect small-office PBX system and I've deployed quite a 
few of them now.


For a long time there's been a US - UK conversion for imported stuff 
where they basically replace the $ sign with a £ sign, and it seems this 
is almost what's going on here. (We call it rip-off Britain)


At the current exchange rate, $300 is just over £150 and for that, I can 
get my 1GHz systems complete with a 128MB flash IDE drive.


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Re: [asterisk-users] VICIDial error

[Brad]

Solved!

You have to get to the end of the scratch install directions to find the 
database setup.

This information SHOULD be in the standard vicidial install instructions.

Classic case of stupid flippn' administrator combined with poor documentation.

Install the database.
du!


--- On Fri, 8/8/08, Brad [EMAIL PROTECTED] wrote:

 From: Brad [EMAIL PROTECTED]
 Subject: [asterisk-users] VICIDial error
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Date: Friday, August 8, 2008, 6:02 PM
 Warning: Cannot modify header information - headers already
 sent by (output started at
 /home/telecom/public_html/vicidial/admin.php:1175) in
 /home/telecom/public_html/vicidial/admin.php on line 1187
 
 Warning: Cannot modify header information - headers already
 sent by (output started at
 /home/telecom/public_html/vicidial/admin.php:1175) in
 /home/telecom/public_html/vicidial/admin.php on line 1188
 
 Has anyone ever seen this?
 
 I am getting a double header sent with all aspects of the
 Astisk GUI including VICIDial
 
 
   
 
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Re: [asterisk-users] BLF functionality

[Johansson Olle E]

13 aug 2008 kl. 00.45 skrev Dan Peters:

We have had Asterisk up and running for a while now and it works  
very well.  Recently we tried to integrate a Linsys SPA962 with the  
associated SPA932 console.  We can get the BLF lights to blink when  
a phone is ringing and we can get the BLF lights to go solid when  
that call is picked up.  My question is about the BLF for the phone  
that placed the call.  Is the BLF supposed to light up when the  
handset is picked up and a dial tone is heard?  Right now that is  
not happening.  The BLF lights only seem to operate for phones that  
are RECEIVING calls and not MAKING them.


It all depends on the version of Asterisk you are using. In Asterisk  
1.4, there's an option called limitonpeers that will

make sure that both incoming and outgoing calls are accounted for.

The subscription only checks the peer part of a SIP type=friend. With  
limitonpeers set, Asterisk only uses the peer

call counter for both incoming and outgoing calls.

In Asterisk 1.6.0 beta, there's an additional setting for setting a  
busy level, allowing for a few more calls while still

busy - in order to allow call transfers while on the phone.

In Asterisk SVN trunk the user object is removed and the problem  
doesn't exist any more.


/O

smime.p7s
Description: S/MIME cryptographic signature
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[asterisk-users] cmdRecord issue related to iax2 received mini frame before first full voice frame?

[Novak Joe]

Hi,
  I tried sending this message a few months back but never lucked into
a response.  I thought I'd try one more time, juicing up the subject
heading a bit, as I am still seeing this behavior intermittently.

  I'm running several asterisk servers in combination with dundi.  The
servers are in different data centers, but other than that they are
running identical copies of the same os image, asterisk configuration,
etc.  One server acts as the trunk and is used to terminate pstn
calls, and pass them on to another server via dundi, which then
answers the call.
  I recently noticed that one of the call-answering servers was
responding and playing back voice prompts fine, but was failing to
record any user generated audio.  After opening up the CLI on this
server and running a test call through it, I noticed reams of the
following warning message any time audio was being played or recorded:
[May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process:
Received mini frame before first full voice frame
[May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process:
Received mini frame before first full voice frame
[May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process:
Received mini frame before first full voice frame
[May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process:
Received mini frame before first full voice frame
[May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process:
Received mini frame before first full voice frame
[May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process:
Received mini frame before first full voice frame
[May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process:
Received mini frame before first full voice frame

  I found the following related post:
http://lists.digium.com/pipermail/asterisk-users/2006-January/136982.html

  However this doesn't explain why I should be unable to record
anything.  The issue seems to be related to network activity, and I'm
not seeing it on any other servers.

  A more detailed explanation of what the above warning means/implies,
and how or why it might be preventing recordings would be greatly
appreciated.

  I'm running Asterisk 1.4.11 on debian Etch.

  Cheers

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[asterisk-users] Newbie: Queue and CDR Reporter and Analyser

[Lee, John (Sydney)]

I am trying to look for a software (open source or proprietory) that could do 
reporting on both queue and CDR in Asterisk 1.4.*

Could someone give me some suggestions?

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Re: [asterisk-users] FWD $30 membership-fee

[Alex Robar]

I stand corrected, I finally received a few of these yesterday. They're not
unclear about the process; The $30 yearly charge is mandatory. The messages
do state that you can link as many accounts to one payment as you'd like
though.
-- 
Alex Robar
[EMAIL PROTECTED]


On Thu, Aug 7, 2008 at 3:05 PM, Alex Robar [EMAIL PROTECTED] wrote:

 FWD has had paid membership options for years. The paid memberships help to
 improve the network and increase it's reach. As far as I've heard (and as
 far as the site mentions), paid membership is not a requirement. That would
 sort of go against the talk... for free... for good slogan.

 AR

 --
 Alex Robar
 [EMAIL PROTECTED]


 On Thu, Aug 7, 2008 at 2:48 PM, SIP [EMAIL PROTECTED] wrote:


  From what I can ascertain, this is a way to essentially fund Jeff
 Pulver's political agenda. I remember writing something a couple of
 years back (

 http://neil.ideasip.com/2006/03/08/von-coalition-and-the-ideals-of-the-little-guy/
 ) about how the VON Coalition, which is meant to be a political action
 committee to help foster new communications, has a somewhat high barrier
 to entry (minimum $10,000 per year).

 As far as I can tell, this FWD membership is a less expensive way for
 people to put their money behind a similar agenda (well... okay, Jeff's
 agenda, whatever that may be).

 The only real issue I see with it is that, a political action committee
 is a committee. The FWD membership seems a little less transparent. It
 could very well be a way to fund Jeff Pulver's personal vision. While
 he's done some great things in the community, I still feel awkward with
 the idea of funding the whole One man. One voice. One decision. No
 oversight idea.

 I'm eager to see how it pans out, though.


 N.


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Re: [asterisk-users] Off Topic: Disable Polycom Soundpoint DoNotDisturb Feature

[Kevin P. Fleming]

Roi Stork wrote:

 However, the problem is that there is still no ringing sound so the user
 can't hear it. Is there a way to make the ringing tone audible?

You can remap the DND key to do something else (or nothing). It may
still be possible for the user to set DND status via the menus, though.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk Experience (TM)

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Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser

[Faraz Khan]

queuemetrics

Lee, John (Sydney) wrote:
 I am trying to look for a software (open source or proprietory) that 
 could do reporting on both queue and CDR in Asterisk 1.4.*
 
 Could someone give me some suggestions?
 
 
 
 
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-- 
Faraz R Khan
Chief Architect
Emergen Consulting Pvt Ltd
+92.21.529.0381 x200
www.emergen.biz


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Re: [asterisk-users] CDR accuracy

[Klaus Darilion]

Steve Murphy wrote:
 On Tue, 2008-08-12 at 16:39 +0200, Klaus Darilion wrote:
 Hi!

 I wonder how Asterisk measures the call duration. The CDR files have a 
 accuracy of seconds. Thus, what happens if the call duration is 0.3 
 seconds. What will Asterisk report? 0 seconds? 1 second?

 What logic will be used by Asterisk: floor? ceil? round?

 thanks
 klaus
 
 Klaus--
 
 The duration/billsec fields are stored as simple integers.
 A simple integer subtraction is performed for both; duration
 is end time minus start time; billsec is end time minus answer time.
 
 Operations are done on system time, in seconds. If the .3 sec spans
 a system second increment, then the time will be 1, if not, then the
 time will be 0. It would seem to me the probability of .3 sec spanning
 a clock tick would be .3...
 
 CDR's do, internally, store finer increments than seconds. (struct
 timeval),
 but the interface yields plain seconds. I just checked the code, and 
 sure enough, just the seconds field is used. So, truncation seems to be
 the rounding method.
 
 In general, we never fussed much about the microseconds, because on
 most interfaces, the slop in how much time it took to make a connection
 made the precision laughable.

Hi Steve!

Thanks for the detailed information. What about the following scenario:
ANSWER and HANGUP happens in the same second. Thus, the call duration 
will be 0 seconds. How are such use cases usually handled in the billing 
system? Are you billing the user (e.g. 1 second or the minimum fee) if 
the call is ANSWERED even if Asterisk reports 0 seconds?

regards
klaus

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Re: [asterisk-users] CDR accuracy

[Gerald Begumisa]

I think that the most appropriate answer for this would be it depends
on your setup and requirements.

Some of our customers bill all answered calls for the entire minimum
duration/increment (even if duration is 0) while others have
configured a rule not to bill all calls whose duration is less than a
certain threshold.

www.yo.co.ug

On 8/13/08, Klaus Darilion [EMAIL PROTECTED] wrote:
 Steve Murphy wrote:
 On Tue, 2008-08-12 at 16:39 +0200, Klaus Darilion wrote:
 Hi!

 I wonder how Asterisk measures the call duration. The CDR files have a
 accuracy of seconds. Thus, what happens if the call duration is 0.3
 seconds. What will Asterisk report? 0 seconds? 1 second?

 What logic will be used by Asterisk: floor? ceil? round?

 thanks
 klaus

 Klaus--

 The duration/billsec fields are stored as simple integers.
 A simple integer subtraction is performed for both; duration
 is end time minus start time; billsec is end time minus answer time.

 Operations are done on system time, in seconds. If the .3 sec spans
 a system second increment, then the time will be 1, if not, then the
 time will be 0. It would seem to me the probability of .3 sec spanning
 a clock tick would be .3...

 CDR's do, internally, store finer increments than seconds. (struct
 timeval),
 but the interface yields plain seconds. I just checked the code, and
 sure enough, just the seconds field is used. So, truncation seems to be
 the rounding method.

 In general, we never fussed much about the microseconds, because on
 most interfaces, the slop in how much time it took to make a connection
 made the precision laughable.

 Hi Steve!

 Thanks for the detailed information. What about the following scenario:
 ANSWER and HANGUP happens in the same second. Thus, the call duration
 will be 0 seconds. How are such use cases usually handled in the billing
 system? Are you billing the user (e.g. 1 second or the minimum fee) if
 the call is ANSWERED even if Asterisk reports 0 seconds?

 regards
 klaus

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Re: [asterisk-users] BLF functionality

[Dave Fullerton]

Johansson Olle E wrote:
 
 13 aug 2008 kl. 00.45 skrev Dan Peters:
 
 We have had Asterisk up and running for a while now and it works very 
 well.  Recently we tried to integrate a Linsys SPA962 with the 
 associated SPA932 console.  We can get the BLF lights to blink when a 
 phone is ringing and we can get the BLF lights to go solid when that 
 call is picked up.  My question is about the BLF for the phone that 
 placed the call.  Is the BLF supposed to light up when the handset is 
 picked up and a dial tone is heard?  Right now that is not happening.  
 The BLF lights only seem to operate for phones that are RECEIVING 
 calls and not MAKING them.

 It all depends on the version of Asterisk you are using. In Asterisk 
 1.4, there's an option called limitonpeers that will
 make sure that both incoming and outgoing calls are accounted for.
 
 The subscription only checks the peer part of a SIP type=friend. With 
 limitonpeers set, Asterisk only uses the peer
 call counter for both incoming and outgoing calls.
 
 In Asterisk 1.6.0 beta, there's an additional setting for setting a busy 
 level, allowing for a few more calls while still
 busy - in order to allow call transfers while on the phone.
 
 In Asterisk SVN trunk the user object is removed and the problem doesn't 
 exist any more.
 

I find it amazing how often I find myself stuck on a problem and then 
someone else posts a question about it to the list. I am in the same 
boat with the OP (although I never thought to test incoming calls until 
I read his message). If I call a phone it will show busy, however if I 
make a call from that phone it still shows as idle. I've set call-limit 
and limitonpeers and restarted asterisk but still no joy. What am I 
missing? I'm running 1.4.21.2

Relevant sip.conf:

[lan-soundpointip](!)
type=friend
host=dynamic
disallow=all
allow=ulaw
dtmfmode=rfc2833
qualify=no
call-limit=10
limitonpeers=yes

[3900](lan-soundpointip)
username=3900
secret=sdjghdfkjhgdf
context=phone-operator
callerid=Operator 3900

[3917](lan-soundpointip)
username=3917
secret=dfkghdjfhdkfd
context=phone-isdept
callerid=Dave Fullerton 3917
mailbox=3117

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[asterisk-users] Sending Set Asynchronous Balanced Mode Extended

[Jon Weisman]

So we've got a TE410P configured as E-1. The PRI is showing up as normal, I 
have green lights, but d channel doesnt seem to come up and i keep getting this 
error if i do a pri intense debug

The carrier swears up and down that there are no issues on their end. Any 
thoughts?

localhost*CLI 
 Unnumbered frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode extended

TIA,

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Re: [asterisk-users] OT: Asterisk on fitPC

[Mark Hamilton]

I love the thin client stuff. It probably looks as big as the Samsung
SWA-4000. But in terms of hardware, don't I need a PCI card to get it
working? How would that work?

Sorry, I have no idea about Asterisk working for home, but just SIP related
stuff. :(

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: August 12, 2008 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT: Asterisk on fitPC

On Tue, 12 Aug 2008, mail-lists wrote:

 I can't see why not. You should easily have enough power for asterisk.

 You can probably also run it as your firewall in a home environment 
 thanks to the dual RJ45's

 I don't know whether or not you can use the built in RJ11 to interface 
 with your POTS line though - maybe someone else could speak to this?

Looks like it's rs232, so I suspect not :)

However if the AMD Geode 500MHz processor is any good, and I'd expect it to
be better than the old 500MHz Via processor I use as my test/development
system then you'll be able to run well over a dozen concurrent calls (not
transoding) without any issues..


Hm. $300 in the US and the UK disty is selling them for just short of £240,
so they can go stuff themselves, low-power or not. (I buy 1GHz systems with
1GB of RAM, running at 15W for half that. No drive though)

Gordon

  

 Hi,



 I?d like to install Asterisk at home. But don?t want to use a full 
 blown PC to host it. I was thinking of using fitPC www.fit-pc.com 
 http://www.fit-pc.com to do all the Asterisk work, interfacing with 
 the local Bell Canada line, and using a SIP VoIP line as well.



 What do you experts think of it?



 Thanks,

 Mark.


 -
 ---

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Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser

[Mark Hamilton]

Doesn't Queuemetrics run on a license basis?
Anything else that's probably open source and free?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faraz Khan
Sent: August 13, 2008 8:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser

queuemetrics

Lee, John (Sydney) wrote:
 I am trying to look for a software (open source or proprietory) that 
 could do reporting on both queue and CDR in Asterisk 1.4.*
 
 Could someone give me some suggestions?
 
 
 
 
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-- 
Faraz R Khan
Chief Architect
Emergen Consulting Pvt Ltd
+92.21.529.0381 x200
www.emergen.biz


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Re: [asterisk-users] cmdRecord issue related to iax2 received mini frame before first full voice frame?

[Tilghman Lesher]

On Wednesday 13 August 2008 04:50:24 Novak Joe wrote:
 [May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process:
 Received mini frame before first full voice frame

   A more detailed explanation of what the above warning means/implies,
 and how or why it might be preventing recordings would be greatly
 appreciated.

A mini frame is simply a frame containing minimal information about the
call itself (the meta data), and a full frame contains all of the meta-data
information.  Sending mini-frames is part of the IAX protocol, as a way of
saving significant bandwidth over the course of a call.  However, a mini frame
cannot be interpreted correctly independently of a full frame.  In every media
stream, a full frame is send approximately once every 60 seconds, to sync the
timestamps.

You could think about it in a different way, by considering a video encoding
method, whereby the full image is sent every once in a while, but only the
differences (which are much smaller) are sent most of the time.  If you have
only a frame containing the differences to an image that you never got in the
first place, then it's very difficult to know what to do with that.

   I'm running Asterisk 1.4.11 on debian Etch.

There have been many changes, bug fixes, and even security issues fixed since
1.4.11.  I'd really recommend that you try something more recent (and even the
latest, because we fixed 2 security issues in the latest release, 1.4.21.2).

-- 
Tilghman

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Re: [asterisk-users] Sending Set Asynchronous Balanced Mode Extended

[Steve Totaro]

On Wed, Aug 13, 2008 at 9:56 AM, Jon Weisman [EMAIL PROTECTED] wrote:
 So we've got a TE410P configured as E-1. The PRI is showing up as normal, I
 have green lights, but d channel doesnt seem to come up and i keep getting
 this error if i do a pri intense debug

 The carrier swears up and down that there are no issues on their end. Any
 thoughts?

 localhost*CLI
 Unnumbered frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
 extended

 TIA,
 Jon

Let's see your zap confs.  Have you tried looping your card back with
pri_net from another port?

As a side note, I usually keep calling the telco until I get someone
on the phone that is actually helpful and not in a rush to get you off
the phone.  Then I get their extension or DID and of course ask if
they don't mind me calling them directly.  I can usually tell by the
way they answer the phone whether they are really going to try to
help, you can hear it in their voice.

Anyways, you could also really get tough with them once you are sure
the issue is not on your side.  I escalate daily and CC everyone on
the telco side, the sales reps, the managers, everyone on the
escalation food chain.

First, double check everything on your side, configs, cables, punch
down, then insist that they send out a tech with a t-bird (not sure
what the equivelent for e1 is) and stand there and watch him/her do
the testing.  They usually call into the CO and have a series of 1s,
0s, and alternating patterns sent to the t-bird.  Sometimes I have had
this happen on the same day as the call.

Of course, when it is resolved, send glowing happy emails to everyone
you have been complaining to and if someone was really helpful in
particular, give credit where it is due.

That is the way I have learned to get things working more quickly than usual.

Thanks,
Steve Totaro

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Re: [asterisk-users] CDR accuracy

[Jay R. Ashworth]

On Wed, Aug 13, 2008 at 02:37:58PM +0200, Klaus Darilion wrote:
 Thanks for the detailed information. What about the following scenario:
 ANSWER and HANGUP happens in the same second. Thus, the call duration 
 will be 0 seconds. How are such use cases usually handled in the billing 
 system? Are you billing the user (e.g. 1 second or the minimum fee) if 
 the call is ANSWERED even if Asterisk reports 0 seconds?

Seems to me that's a policy issue; ie: why are you asking *us*?  :-)

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 30

[dimitri . osler]

Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a 
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al 
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], 
altrimenti vi risponderò al mio rientro.

Dimitri Osler

I will be on vacation until Tuesday 19th of August with limited access to voice 
and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 
0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on 
my return.

Dimitri Osler





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[asterisk-users] Decline issue

[michel freiha]

Dear Sir,
Please find attached the log file that I took from the asterik server during
a call...Please check the SIP packets exchanged between OpenSer that send an
Invite SIP packet to theasterisk server and the asterisk Server and let me
know wat this DECLINEd message means...
Regards


Asterisk Log.rtf
Description: RTF file
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Re: [asterisk-users] OT: Asterisk on fitPC

[Drew Gibson]

You can use an external ATA.

I have a Linksys SPA3102 which has 1 FXO (Bell line) port, 1 FXS (phone) 
port and one Ethernet port to connect via SIP to Asterisk.

Alternatively, there are USB connected adapters eg. Xorcom.com but I 
haven't used them.

regards,

Drew



Mark Hamilton wrote:
 I love the thin client stuff. It probably looks as big as the Samsung
 SWA-4000. But in terms of hardware, don't I need a PCI card to get it
 working? How would that work?

 Sorry, I have no idea about Asterisk working for home, but just SIP related
 stuff. :(

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Gordon
 Henderson
 Sent: August 12, 2008 4:41 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] OT: Asterisk on fitPC

 On Tue, 12 Aug 2008, mail-lists wrote:

   
 I can't see why not. You should easily have enough power for asterisk.

 You can probably also run it as your firewall in a home environment 
 thanks to the dual RJ45's

 I don't know whether or not you can use the built in RJ11 to interface 
 with your POTS line though - maybe someone else could speak to this?
 

 Looks like it's rs232, so I suspect not :)

 However if the AMD Geode 500MHz processor is any good, and I'd expect it to
 be better than the old 500MHz Via processor I use as my test/development
 system then you'll be able to run well over a dozen concurrent calls (not
 transoding) without any issues..


 Hm. $300 in the US and the UK disty is selling them for just short of £240,
 so they can go stuff themselves, low-power or not. (I buy 1GHz systems with
 1GB of RAM, running at 15W for half that. No drive though)

 Gordon

   
   
 Hi,



 I?d like to install Asterisk at home. But don?t want to use a full 
 blown PC to host it. I was thinking of using fitPC www.fit-pc.com 
 http://www.fit-pc.com to do all the Asterisk work, interfacing with 
 the local Bell Canada line, and using a SIP VoIP line as well.



 What do you experts think of it?



 Thanks,

 Mark.


 -
 ---

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-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


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[asterisk-users] Asterisk and Radius

[Salvatore Del Popolo]

Hello,
I would like to know, if in Asterisk version 1.6.0-beta9 is it possible 
to use Radius as an external authentication server.
Thank you for your help
Ciao
Salvo

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Re: [asterisk-users] LNP Problems

[Adam Moffett]

Thanks everyone for the input.

 A CSR is nothing more than a listing of the numbers by your current
 provider on some sort of letterhead to indicate you actually are the
 subscriber who these numbers belong to (ie, you pay the bill for
 them).

 Is it necessary for the actual LNP process - no, not technically but
 companies require it to make sure they are not porting some else's
 numbers.  Most CLEC's will just use a copy of your bill as the CSR.
 RBOC's have a more formal record which lists USOCs and other data that
 is completely unnecessary.  The company doing the LNP will also need
 an LOA from you to request the CSR from the current provider.

 Time Warner most likely does have to give you one if they operate as a
 CLEC in your state or residence but it wont be you they give it to.
 You should provide TWTelecom with an LOA and then they can request the
 CSR from Time Warner.  If the numbers in question are not numbers
 native to Time Warner - ie, Time Warner ported them from Bell or your
 regional LEC, then TWTelecom can force the issue and just port them by
 updating the LNP database with their service provider id and other
 appropriate information.  Time Warner does not have to release the
 number to them for this.

 Regardless, its useless for you to bother calling Time Warner and ask
 for a CSR because the only people who would know what you are talking
 about are in the LNP/Carrier division and unless you work for another
 carrier, you wont get to them.  Your new Telco will have to do this.
 If they cant accomodate this, I would find another provider.

 On Tue, Aug 12, 2008 at 3:42 PM, Adam Moffett [EMAIL PROTECTED] wrote:
   
 What is the deal with CSR's?

 TWTelecom is telling me that I can't port a number to their service
 without a Customer Service Record.  Apparently this is easy with
 Verizon, and not so easy with some other companies.

 Basically I'm at a brick wall with a couple of ports because TWTelecom
 is telling me I HAVE to get a CSR and certain other providers (Time
 Warner Cable for one) are telling me that's wrong, that I don't need one
 and they don't have one to give me.

 Does anybody know what to do at this point?



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Re: [asterisk-users] Decline issue

[Felippe Silvestre]

post the response of the command bellow
 
dialplan show [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 
 
att.
 
Felippe Silvestre
 




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of michel
freiha
Sent: Wednesday, August 13, 2008 11:30
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Decline issue


Dear Sir,
Please find attached the log file that I took from the asterik
server during a call...Please check the SIP packets exchanged between
OpenSer that send an Invite SIP packet to theasterisk server and the
asterisk Server and let me know wat this DECLINEd message means...
Regards

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[asterisk-users] ENUM lookup

[Adrian Marsh]

Hi All,

 

For a 1.4 version asterisk, whats the recommended mechanism for dialling
with ENUM lookup?  At the moment I user SIPbroker, but am getting tired
of it hanging on certain numbers, so I was thinking about implementing
it myself.

 

I've seen various vo-ip.info pages
(http://www.voip-info.org/wiki/view/Asterisk+cmd+EnumLookup) talking
about the func ENUMLOOKUP instead of EnumLookup Application, but then
I'll need to implement my own logic around this right??

 

Thanks,

 

Adrian

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Re: [asterisk-users] BLF functionality

[Sean Dennis]

 I find it amazing how often I find myself stuck on a problem and then 
 someone else posts a question about it to the list. I am in the same 
 boat with the OP (although I never thought to test incoming calls until 
 I read his message). If I call a phone it will show busy, however if I 
 make a call from that phone it still shows as idle. I've set call-limit 
 and limitonpeers and restarted asterisk but still no joy. What am I 
 missing? I'm running 1.4.21.2

 Relevant sip.conf:

 [lan-soundpointip](!)
 type=friend
 host=dynamic
 disallow=all
 allow=ulaw
 dtmfmode=rfc2833
 qualify=no
 call-limit=10
 limitonpeers=yes

 [3900](lan-soundpointip)
 username=3900
 secret=sdjghdfkjhgdf
 context=phone-operator
 callerid=Operator 3900

 [3917](lan-soundpointip)
 username=3917
 secret=dfkghdjfhdkfd
 context=phone-isdept
 callerid=Dave Fullerton 3917
 mailbox=3117

   
In my general section of my sip.conf I have:

allowsubscribe=yes
notifyringing=yes
limitonpeer=yes
notifyhold=yes

and it works both ways.


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Re: [asterisk-users] help...i cant do more...

[David Thomas]

On Fri, Apr 25, 2008 at 4:38 AM, Bruno Pereira
[EMAIL PROTECTED] wrote:
 Thanks for the answers.
 I need to say that this command is executed from another machine, with the
 command ssh
 because in ocalhost is all ok, with sudo or with root.

 I will try that trace to see if it helps me, but the bg probem is start the
 service from another machine with ssh .

Did anyone ever find a solution to this issue. I have the same problem
when trying to start asterisk from another computer via SSH. It starts
fine on the local box, but over SSH it just hangs forever. I am using
root as the user, and issuing the command: ssh 10.0.0.10
'/etc/init.d/asterisk start'.

Thanks!
Dave

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Re: [asterisk-users] FC2 and Zaptel

[Jay Ray]

1.4 from CVS..

 Last nite...I corrected the below error by change the check in Makefile for 
HOTPLUG=yes to a no and it does not try to download and install that tar.gz 
file anymore..Not sure if it will cause a prob..

Later all teh install steps worked fine until I ran ztcfg which errored with - 
unable to open devide.. I found that /dev/zap did not exist...now a little 
more searching revealed that my devices for zap wer einstalledin /udev..i justl 
inked those to .dev.zap and ztcfg workedlooks like my  card is up and 
running..

Can I Just connect my homephone line to the LINE port of the X100P card and use 
a softphone to outpulse calls over that Line??

--- On Wed, 8/13/08, Paul Hales [EMAIL PROTECTED] wrote:
From: Paul Hales [EMAIL PROTECTED]
Subject: Re: [asterisk-users] FC2 and Zaptel
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com
Date: Wednesday, August 13, 2008, 12:26 AM

Which versions of Zaptel have you tried to build?

PaulH


Jay Ray wrote:
 Any ideas, please they are highly appreciated

 --- On *Mon, 8/4/08, Jay Ray /[EMAIL PROTECTED]/* wrote:

 From: Jay Ray [EMAIL PROTECTED]
 Subject: [asterisk-users] FC2 and Zaptel
 To: asterisk-users@lists.digium.com
 Date: Monday, August 4, 2008, 12:02 AM

 Hi,

  I am using an older Fedora - FC2 and trying to install
 zaptel.(for X100P card I have - FXO with one line port and one
 Phone port)

 Fist I tried installin from RPM...as given here (also tried
 installing Zapata)
 http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora (FC2 is
 at the end)

 But looks like zaptel executable was not there...same for zapata...

 Then I started on downloading the source, I successfully completed
 MAKE for zaptel...but make install has following error...Full o/p
 follows:
 =
 [EMAIL PROTECTED] zaptel]# make install
 make[1]: Entering directory `/usr/src/zaptel'
 make -C /lib/modules/2.6.10-1.771_FC2/build ARCH=i386
 SUBDIRS=/usr/src/zaptel/kernel HOTPLUG_FIRMWARE=yes
 KBUILD_OBJ_M=pciradio.o tor2.o torisa.o wcfxo.o wct1xxp.o
wctdm.o
 wcte11xp.o wcusb.o zaptel.o ztd-eth.o ztd-loc.o ztdummy.o
 ztdynamic.o zttranscode.o wct4xxp/ wctc4xxp/ xpp/ wctdm24xxp/
 wcte12xp/ modules
 make[2]: Entering directory `/lib/modules/2.6.10-1.771_FC2/build'

   Building modules, stage 2.
   MODPOST
 *** Warning: class_device_destroy
 [/usr/src/zaptel/kernel/zaptel.ko] undefined!
 make[2]: Leaving directory `/lib/modules/2.6.10-1.771_FC2/build'
 make[2]: Entering directory `/usr/src/zaptel/kernel/xpp/utils'
 make[2]: Nothing to be done for `all'.
 make[2]: Leaving directory `/usr/src/zaptel/kernel/xpp/utils'
 make[1]: Leaving directory `/usr/src/zaptel'
 install -d /etc/udev/rules.d
 build_tools/genudevrules  /etc/udev/rules.d/zaptel.rules
 build_tools/uninstall-modules dahdi 2.6.10-1.771_FC2
 make -C /lib/modules/2.6.10-1.771_FC2/build ARCH=i386
 SUBDIRS=/usr/src/zaptel/kernel HOTPLUG_FIRMWARE=yes
 KBUILD_OBJ_M=pciradio.o tor2.o torisa.o wcfxo.o wct1xxp.o
wctdm.o
 wcte11xp.o wcusb.o zaptel.o ztd-eth.o ztd-loc.o ztdummy.o
 ztdynamic.o zttranscode.o wct4xxp/ wctc4xxp/ xpp/ wctdm24xxp/
 wcte12xp/ INSTALL_MOD_PATH= INSTALL_MOD_DIR=misc modules_install
 make[1]: Entering directory `/lib/modules/2.6.10-1.771_FC2/build'
   INSTALL /usr/src/zaptel/kernel/pciradio.ko
   INSTALL /usr/src/zaptel/kernel/tor2.ko
   INSTALL /usr/src/zaptel/kernel/torisa.ko
   INSTALL /usr/src/zaptel/kernel/wcfxo.ko
   INSTALL /usr/src/zaptel/kernel/wct1xxp.ko
   INSTALL /usr/src/zaptel/kernel/wct4xxp/wct4xxp.ko
   INSTALL /usr/src/zaptel/kernel/wctc4xxp/wctc4xxp.ko
   INSTALL /usr/src/zaptel/kernel/wctdm.ko
   INSTALL /usr/src/zaptel/kernel/wctdm24xxp/wctdm24xxp.ko
   INSTALL /usr/src/zaptel/kernel/wcte11xp.ko
   INSTALL /usr/src/zaptel/kernel/wcte12xp/wcte12xp.ko
   INSTALL /usr/src/zaptel/kernel/wcusb.ko
   INSTALL /usr/src/zaptel/kernel/xpp/xpd_fxo.ko
   INSTALL /usr/src/zaptel/kernel/xpp/xpd_fxs.ko
   INSTALL /usr/src/zaptel/kernel/xpp/xpd_pri.ko
   INSTALL /usr/src/zaptel/kernel/xpp/xpp.ko
   INSTALL /usr/src/zaptel/kernel/xpp/xpp_usb.ko
   INSTALL /usr/src/zaptel/kernel/zaptel.ko
   INSTALL /usr/src/zaptel/kernel/ztd-eth.ko
   INSTALL /usr/src/zaptel/kernel/ztd-loc.ko
   INSTALL /usr/src/zaptel/kernel/ztdummy.ko
   INSTALL /usr/src/zaptel/kernel/ztdynamic.ko
   INSTALL /usr/src/zaptel/kernel/zttranscode.ko
 make[1]: Leaving directory `/lib/modules/2.6.10-1.771_FC2/build'
 [ `id -u` = 0 ]  /sbin/depmod -a 2.6.10-1.771_FC2 || :
 make[1]: Entering directory `/usr/src/zaptel/kernel/xpp/utils'
 make[1]: Nothing to be done for `all'.
 make[1]: Leaving directory `/usr/src/zaptel/kernel/xpp/utils'
  

Re: [asterisk-users] help...i cant do more...

[Anthony Francis]

David Thomas wrote:
 On Fri, Apr 25, 2008 at 4:38 AM, Bruno Pereira
 [EMAIL PROTECTED] wrote:
   
 Thanks for the answers.
 I need to say that this command is executed from another machine, with the
 command ssh
 because in ocalhost is all ok, with sudo or with root.

 I will try that trace to see if it helps me, but the bg probem is start the
 service from another machine with ssh .
 

 Did anyone ever find a solution to this issue. I have the same problem
 when trying to start asterisk from another computer via SSH. It starts
 fine on the local box, but over SSH it just hangs forever. I am using
 root as the user, and issuing the command: ssh 10.0.0.10
 '/etc/init.d/asterisk start'.

 Thanks!
 Dave

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The real solution is to run Asterisk as a service, but if you must have 
it run from a console then I would suggest starting it in a screen.

That is, make sure you have screen installed, run it, and then start 
asterisk.
After that disconnect from the screen session by pressing ctrl+a and then d.
To reconnect to the screen session at anytime you simply do screen -r.

The issue with simply running the asterisk command from an ssh session 
is that its process is started as a child of your remote shell.

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Re: [asterisk-users] OT: Asterisk on fitPC

[Mark Hamilton]

Don't you think the USB ones would be amazing considering how the ATA is
another extra piece of equipment?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson
Sent: August 13, 2008 10:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT: Asterisk on fitPC

You can use an external ATA.

I have a Linksys SPA3102 which has 1 FXO (Bell line) port, 1 FXS (phone) 
port and one Ethernet port to connect via SIP to Asterisk.

Alternatively, there are USB connected adapters eg. Xorcom.com but I 
haven't used them.

regards,

Drew



Mark Hamilton wrote:
 I love the thin client stuff. It probably looks as big as the Samsung
 SWA-4000. But in terms of hardware, don't I need a PCI card to get it
 working? How would that work?

 Sorry, I have no idea about Asterisk working for home, but just SIP
related
 stuff. :(

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Gordon
 Henderson
 Sent: August 12, 2008 4:41 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] OT: Asterisk on fitPC

 On Tue, 12 Aug 2008, mail-lists wrote:

   
 I can't see why not. You should easily have enough power for asterisk.

 You can probably also run it as your firewall in a home environment 
 thanks to the dual RJ45's

 I don't know whether or not you can use the built in RJ11 to interface 
 with your POTS line though - maybe someone else could speak to this?
 

 Looks like it's rs232, so I suspect not :)

 However if the AMD Geode 500MHz processor is any good, and I'd expect it
to
 be better than the old 500MHz Via processor I use as my test/development
 system then you'll be able to run well over a dozen concurrent calls (not
 transoding) without any issues..


 Hm. $300 in the US and the UK disty is selling them for just short of
£240,
 so they can go stuff themselves, low-power or not. (I buy 1GHz systems
with
 1GB of RAM, running at 15W for half that. No drive though)

 Gordon

   
   
 Hi,



 I?d like to install Asterisk at home. But don?t want to use a full 
 blown PC to host it. I was thinking of using fitPC www.fit-pc.com 
 http://www.fit-pc.com to do all the Asterisk work, interfacing with 
 the local Bell Canada line, and using a SIP VoIP line as well.



 What do you experts think of it?



 Thanks,

 Mark.


 -
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-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


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Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 31

[dimitri . osler]

Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a 
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al 
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], 
altrimenti vi risponderò al mio rientro.

Dimitri Osler

I will be on vacation until Tuesday 19th of August with limited access to voice 
and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 
0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on 
my return.

Dimitri Osler





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Re: [asterisk-users] BLF functionality

[Dave Fullerton]

Sean Dennis wrote:
 I find it amazing how often I find myself stuck on a problem and then 
 someone else posts a question about it to the list. I am in the same 
 boat with the OP (although I never thought to test incoming calls until 
 I read his message). If I call a phone it will show busy, however if I 
 make a call from that phone it still shows as idle. I've set call-limit 
 and limitonpeers and restarted asterisk but still no joy. What am I 
 missing? I'm running 1.4.21.2

 Relevant sip.conf:

 [lan-soundpointip](!)
 type=friend
 host=dynamic
 disallow=all
 allow=ulaw
 dtmfmode=rfc2833
 qualify=no
 call-limit=10
 limitonpeers=yes

 [3900](lan-soundpointip)
 username=3900
 secret=sdjghdfkjhgdf
 context=phone-operator
 callerid=Operator 3900

 [3917](lan-soundpointip)
 username=3917
 secret=dfkghdjfhdkfd
 context=phone-isdept
 callerid=Dave Fullerton 3917
 mailbox=3117

   
 In my general section of my sip.conf I have:
 
 allowsubscribe=yes
 notifyringing=yes
 limitonpeer=yes
 notifyhold=yes
 
 and it works both ways.
 

The problem was I had limitonpeers in the wrong place. I didn't have it 
set in the [general] section, only on the peers/users. If I had bothered 
to search the sample sip.conf file before hand I would have seen that.

Thanks.

-Dave

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[asterisk-users] call forward spa 841 and asterisk 1.4.21

[Walter Willis]

i am install asterisk with asterisk-gui , the clients have spa841, but I am
call number outsite celular and I try to transfer the call to client sip,
not show XTRAN in the phone.

any idea??
configuration bad ???
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[asterisk-users] seeking hardware recommendation PCI versus PCI Express E1 card (te407p vs te420bf)

[Rajesh Kumar Mallah]

Dear List ,

I have to make a choice between TE407P  (5.0V PCI slot) and TE420BF (3.3V PCI 
express).

I have a IBM x220 with 2 x 1.2 GHz PIII CPU and OnBoard SCSI Ultra160 Drives
but it does not have PCI Express slot. So i cannot use TE420BF with it.

The system i am willing to build should support upto 60 voice channels
and there will be at least 10 channels under call recording.

My question is will the CPU power (2 * 1200 MHz PIII ) and the PCI bus (33 MHz 
bus)
be sufficient to handle the load ?

I can always buy the card  and test , and if the need be i can put the
card (TE407P) on the 5 V PCI slot of any modern motherboard also, the
question here is , Am i loosing anything by not buying a PCI-Express card
and instead going for a normal 33MHz , 5.0 V PCI card ?

to summarise is TE407P in anyway inferior to TE420BF Card ?
does TE407P saturate the 33mhz PCI bus when all voice channels are used ?

The most new mobos are having 5.0V PCI slots , PCI x1 and x16 slots, so
if i buy te407p i test it with the old ibm x220 server and if need be
i can use it with the newer mobos.

regds
Rajesh Kumar Mallah.


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[asterisk-users] rtc issue

[Jerry Geis]

Hi

I am using centos 4.6 on an ebox 4300. Everything seems to be working 
except the
/proc/interrupts rtc is always constant. On other machines the rtc 
(which ztdummy uses) is always incrementing.
the uhci_hcd and ehci_hcd are both running.

What dont I have right on the system so rtc increments?

Thanks,

Jerry
--

  0: 274276IO-APIC-edge  timer
  1:  9IO-APIC-edge  i8042
  2:  0  XT-PIC  cascade
  8:  2IO-APIC-edge  rtc
145:   5522   IO-APIC-level  eth0
153:  10279   IO-APIC-level  HDA Intel
161:  0   IO-APIC-level  uhci_hcd
169:  0   IO-APIC-level  uhci_hcd
177:  17491   IO-APIC-level  ehci_hcd
NMI:  0
LOC: 274158
ERR:  0
MIS:  0

lspci shows
lspci
00:00.0 Host bridge: VIA Technologies, Inc. CX700 Host Bridge (rev 10)
00:00.1 Host bridge: VIA Technologies, Inc. CX700 Host Bridge
00:00.2 Host bridge: VIA Technologies, Inc. CX700 Host Bridge
00:00.3 Host bridge: VIA Technologies, Inc. CX700 Host Bridge
00:00.4 Host bridge: VIA Technologies, Inc. CX700 Host Bridge
00:00.7 Host bridge: VIA Technologies, Inc. CX700 Host Bridge
00:01.0 PCI bridge: VIA Technologies, Inc. VT8237 PCI Bridge
00:08.0 Ethernet controller: Realtek Semiconductor Co., Ltd. 
RTL-8139/8139C/8139C+ (rev 10)
00:0f.0 IDE interface: VIA Technologies, Inc. CX700M2 IDE
00:10.0 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1 
Controller (rev 90)
00:10.1 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1 
Controller (rev 90)
00:10.4 USB Controller: VIA Technologies, Inc. USB 2.0 (rev 90)
00:11.0 ISA bridge: VIA Technologies, Inc. CX700 PCI to ISA Bridge
00:11.7 Host bridge: VIA Technologies, Inc. CX700 Internal Module Bus
00:13.0 PCI bridge: VIA Technologies, Inc. CX700 Host Bridge
01:00.0 VGA compatible controller: VIA Technologies, Inc. CX700M2 
UniChrome PRO II Graphics (rev 03)
02:01.0 Audio device: VIA Technologies, Inc. VIA High Definition Audio 
Controller (rev 10)

lsmod shows
 lsmod
Module  Size  Used by
md5 4161  1
ipv6  236929  10
autofs430405  0
sunrpc163237  1
ztdummy 4180  0
zaptel203268  3 ztdummy
crc_ccitt   2241  1 zaptel
ipt_REJECT  6721  1
ipt_state   1985  26
ip_conntrack   41077  1 ipt_state
iptable_filter  3009  1
ip_tables  17601  3 ipt_REJECT,ipt_state,iptable_filter
dm_mirror  31045  0
dm_mod 67577  1 dm_mirror
uhci_hcd   31705  0
snd_hda_intel 347644  4
snd_pcm_oss44456  0
snd_mixer_oss  19072  1 snd_pcm_oss
snd_pcm82824  4 snd_hda_intel,snd_pcm_oss
snd_timer  30724  3 snd_pcm
snd_page_alloc 11528  2 snd_hda_intel,snd_pcm
snd_hwdep  10884  1 snd_hda_intel
snd70276  10 
snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer,snd_hwdep
soundcore  10017  1 snd
8139too26177  0
mii 5313  1 8139too
ehci_hcd   31429  0
usb_storage60937  2
ext3  118217  1
jbd72665  1 ext3
ata_piix   15173  0
libata111645  1 ata_piix
sd_mod 17345  3
scsi_mod  125901  3 usb_storage,libata,sd_mod


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Re: [asterisk-users] seeking hardware recommendation PCI versus PCI Express E1 card (te407p vs te420bf)

[Kevin P. Fleming]

Rajesh Kumar Mallah wrote:

 to summarise is TE407P in anyway inferior to TE420BF Card ?
 does TE407P saturate the 33mhz PCI bus when all voice channels are used ?

There is no effective performance difference between using the PCI-X and
PCI-E versions of these cards; the cards are essentially identical, with
the addition of a PCI-Express bridge on the PCI-E card.

Keep in mind that even if you use 4 E1 circuits with the card, the total
bandwidth consumption of card is approximately 1 megabyte per second (4
times 2 megabits per second), which is drastically below the PCI bus
bandwidth of 132 megabytes per second (33 MHz bus with 32-bit
transfers). No quad-T1/E1 card will ever be able to saturate a PCI bus,
especially not PCI-X or PCI-E.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk Experience (TM)

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Re: [asterisk-users] TDM400P FXO not seeing ringing after software update

[Drew Gibson]

Kevin P. Fleming wrote:
 Tony Mountifield wrote:

   
 My guess is that 1.2.24 will work, which was at revision 3842 of the tree
 (rev 3741 of wctdm.c). The next change to wctdm.c (rev 4126) looks innocuous
 enough, but the follwing two (revs 4128 and 4132) look likely culprits,
 from looking at the areas of code that they affect.
 

 It is very likely 4128, based on the code it affects and the behavior
 that is being reported. Please let us know as soon as you can (anyone
 who has this problem), if reverting r4128 from current Zaptel branch 1.2
 SVN solves the problem.

   

Did anyone find a fix for this (besides down-grading to 1.2.24)?

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


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Re: [asterisk-users] TDM400P FXO not seeing ringing after software update

[Kevin P. Fleming]

Drew Gibson wrote:

 Did anyone find a fix for this (besides down-grading to 1.2.24)?

It has been fixed in Subversion already, but a new Zaptel release hasn't
been made. That will be done in the next few days.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk Experience (TM)

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Re: [asterisk-users] rtc issue

[Jerry Geis]

Jerry Geis wrote:
 Hi

 I am using centos 4.6 on an ebox 4300. Everything seems to be working 
 except the
 /proc/interrupts rtc is always constant. On other machines the rtc 
 (which ztdummy uses) is always incrementing.
 the uhci_hcd and ehci_hcd are both running.

 What dont I have right on the system so rtc increments?

 Thanks,

 Jerry
 --

  0: 274276IO-APIC-edge  timer
  1:  9IO-APIC-edge  i8042
  2:  0  XT-PIC  cascade
  8:  2IO-APIC-edge  rtc
 145:   5522   IO-APIC-level  eth0
 153:  10279   IO-APIC-level  HDA Intel
 161:  0   IO-APIC-level  uhci_hcd
 169:  0   IO-APIC-level  uhci_hcd
 177:  17491   IO-APIC-level  ehci_hcd
 NMI:  0
 LOC: 274158
 ERR:  0
 MIS:  0

 lspci shows
 lspci
 00:00.0 Host bridge: VIA Technologies, Inc. CX700 Host Bridge (rev 10)
 00:00.1 Host bridge: VIA Technologies, Inc. CX700 Host Bridge
 00:00.2 Host bridge: VIA Technologies, Inc. CX700 Host Bridge
 00:00.3 Host bridge: VIA Technologies, Inc. CX700 Host Bridge
 00:00.4 Host bridge: VIA Technologies, Inc. CX700 Host Bridge
 00:00.7 Host bridge: VIA Technologies, Inc. CX700 Host Bridge
 00:01.0 PCI bridge: VIA Technologies, Inc. VT8237 PCI Bridge
 00:08.0 Ethernet controller: Realtek Semiconductor Co., Ltd. 
 RTL-8139/8139C/8139C+ (rev 10)
 00:0f.0 IDE interface: VIA Technologies, Inc. CX700M2 IDE
 00:10.0 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1 
 Controller (rev 90)
 00:10.1 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1 
 Controller (rev 90)
 00:10.4 USB Controller: VIA Technologies, Inc. USB 2.0 (rev 90)
 00:11.0 ISA bridge: VIA Technologies, Inc. CX700 PCI to ISA Bridge
 00:11.7 Host bridge: VIA Technologies, Inc. CX700 Internal Module Bus
 00:13.0 PCI bridge: VIA Technologies, Inc. CX700 Host Bridge
 01:00.0 VGA compatible controller: VIA Technologies, Inc. CX700M2 
 UniChrome PRO II Graphics (rev 03)
 02:01.0 Audio device: VIA Technologies, Inc. VIA High Definition Audio 
 Controller (rev 10)

 lsmod shows
 lsmod
 Module  Size  Used by
 md5 4161  1
 ipv6  236929  10
 autofs430405  0
 sunrpc163237  1
 ztdummy 4180  0
 zaptel203268  3 ztdummy
 crc_ccitt   2241  1 zaptel
 ipt_REJECT  6721  1
 ipt_state   1985  26
 ip_conntrack   41077  1 ipt_state
 iptable_filter  3009  1
 ip_tables  17601  3 ipt_REJECT,ipt_state,iptable_filter
 dm_mirror  31045  0
 dm_mod 67577  1 dm_mirror
 uhci_hcd   31705  0
 snd_hda_intel 347644  4
 snd_pcm_oss44456  0
 snd_mixer_oss  19072  1 snd_pcm_oss
 snd_pcm82824  4 snd_hda_intel,snd_pcm_oss
 snd_timer  30724  3 snd_pcm
 snd_page_alloc 11528  2 snd_hda_intel,snd_pcm
 snd_hwdep  10884  1 snd_hda_intel
 snd70276  10 
 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer,snd_hwdep
 soundcore  10017  1 snd
 8139too26177  0
 mii 5313  1 8139too
 ehci_hcd   31429  0
 usb_storage60937  2
 ext3  118217  1
 jbd72665  1 ext3
 ata_piix   15173  0
 libata111645  1 ata_piix
 sd_mod 17345  3
 scsi_mod  125901  3 usb_storage,libata,sd_mod


I stopped everything and reloaded modprobe ztdummy debug=1
and all I have from dmesg is :
ztdummy: init() finished

it has been 10 minutes. I so nothing else logged.
It is supposed to log every 5 seconds.

How can I get ztdummy working with rtc correctly.

my cpuinfo is
 more /proc/cpuinfo
processor   : 0
vendor_id   : CentaurHauls
cpu family  : 6
model   : 13
model name  : VIA Eden Processor  500MHz
stepping: 0
cpu MHz : 499.027
cache size  : 128 KB
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: no
fpu : yes
fpu_exception   : yes
cpuid level : 1
wp  : yes
flags   : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge cmov 
pat clflush acpi mmx fxsr sse sse2 tm nx pni est tm2 xtpr rng rng_en ace 
ace_en
bogomips: 998.94

Thanks,

Jerry

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[asterisk-users] New GUI for Realtime Asterisk - RAGUI

[Mike Clark]

Our company, WebPoint IT Solutions has just released an open source (GPL 
V2 license), Ruby on Rails based gui manager for Realtime Asterisk 
called RAGUI.

RAGUI is definitely a work in progress and has rough edges, but we 
expect to polish it up in the upcoming weeks and months. All comments, 
contributions, and criticisms are welcomed!

Here are the links:
Sourceforge: http://sourceforge.net/projects/ragui/
Website: http://www.ragui.net

Enjoy!

Mike Clark
WebPoint IT Solutions


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[asterisk-users] Asterisk stops sending RTP packets to ethernet interface

[Jonathan Miller]

From what I can determine while troubleshooting a voice-dropping
issue, the Asterisk server in my organization has been dropping RTP
packets between the asterisk server process and the network interface.

I determined this from an RTP debug that showed packets sent to the
phone and packets received from the phone during the entire call. A
tcpdump done on the server for the interface that would deliver the
packet to the wire does not show the packets.

Is there somewhere I can look to resolve this? Something anybody has
come across? It is happening frequently and with great discomfort to
many users.

I had upgraded from 1.2.x to latest 1.4.x in attempts to resolve this.
I also disabled a lot of COM/LPT and USB devices in the BIOS to free
up some IRQ's. no devices are sharing IRQ's at this point, with I
thought might have been part of the issue, but has proved to at least
not be directly related.

These calls are from a PRI to a Cisco 7940 using SIP. There is a
Juniper EX switch between the two. Both sides negotiate at
100Mbps/Full Duplex.

I have ruled the switch out of the problem as it's not seeing the
packets on the wire when the issue is occuring.

Please help or point me to someone that can.

-Jonathan
[EMAIL PROTECTED]

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Re: [asterisk-users] TDM400P FXO not seeing ringing after software update

[Tony Mountifield]

In article [EMAIL PROTECTED], Drew Gibson [EMAIL PROTECTED] wrote:
 Kevin P. Fleming wrote:
  Tony Mountifield wrote:

  My guess is that 1.2.24 will work, which was at revision 3842 of the tree
  (rev 3741 of wctdm.c). The next change to wctdm.c (rev 4126) looks 
  innocuous
  enough, but the follwing two (revs 4128 and 4132) look likely culprits,
  from looking at the areas of code that they affect.
 
  It is very likely 4128, based on the code it affects and the behavior
  that is being reported. Please let us know as soon as you can (anyone
  who has this problem), if reverting r4128 from current Zaptel branch 1.2
  SVN solves the problem.
 
 Did anyone find a fix for this (besides down-grading to 1.2.24)?

I believe Russell did apply a fix to the 1.2 branch of zaptel (rev 4442),
but I have been too busy with other things to try it out.

There has not been a zaptel release since the fix was committed.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] New GUI for Realtime Asterisk - RAGUI

[bkruse]

That's pretty cool, I love ruby. What method does it use to
communicate to Asterisk? Does it manipulate raw config
files or use manager or something similar?

I am curious :)

-Brandon

Mike Clark wrote:
 Our company, WebPoint IT Solutions has just released an open source (GPL 
 V2 license), Ruby on Rails based gui manager for Realtime Asterisk 
 called RAGUI.

 RAGUI is definitely a work in progress and has rough edges, but we 
 expect to polish it up in the upcoming weeks and months. All comments, 
 contributions, and criticisms are welcomed!

 Here are the links:
 Sourceforge: http://sourceforge.net/projects/ragui/
 Website: http://www.ragui.net

 Enjoy!

 Mike Clark
 WebPoint IT Solutions


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[asterisk-users] ztdummy on centos 4.6 i386

[Jerry Geis]

I am running centos 4.6 i386 kernel 2.6.9-67
ztdummy compiles fine,   loads fine,  but does not work.

modprobe ztdummy debug=1

dmesg shows
ztdummy: init() finished

however the debug is supposed to print something every 5 seconds
it does not do this. Nor does /proc/interrupts rtc value increment.

What else do I need to do to get this to work correctly?

Jerry

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Re: [asterisk-users] Asterisk and Radius

[Philippe Sultan]

Ciao Salvo,

That is not directly possible. But, you can integrate a GPL PERL
RADIUS client with Asterisk :
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth

It works good, I use it to make Asterisk work as an IVR with a billing
system, that acts as a RADIUS server.

Cheers,

-- 
Philippe Sultan

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Re: [asterisk-users] New GUI for Realtime Asterisk - RAGUI

[Mike Clark]

Brandon:

It manages the MySQL database tables directly via RoR. It is 
specifically built for using Realtime Asterisk. It uses #exec statements 
in the extensions.conf file to set up the context declarations.

Thanks,

Mike

bkruse wrote:
 That's pretty cool, I love ruby. What method does it use to
 communicate to Asterisk? Does it manipulate raw config
 files or use manager or something similar?

 I am curious :)

 -Brandon

 Mike Clark wrote:
   
 Our company, WebPoint IT Solutions has just released an open source (GPL 
 V2 license), Ruby on Rails based gui manager for Realtime Asterisk 
 called RAGUI.

 RAGUI is definitely a work in progress and has rough edges, but we 
 expect to polish it up in the upcoming weeks and months. All comments, 
 contributions, and criticisms are welcomed!

 Here are the links:
 Sourceforge: http://sourceforge.net/projects/ragui/
 Website: http://www.ragui.net

 Enjoy!

 Mike Clark
 WebPoint IT Solutions


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[asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?

[Jonathan Miller]

From what I can determine while troubleshooting a voice-dropping
issue, the Asterisk server in my organization has been dropping RTP
packets between the asterisk server process and the network interface.

I determined this from an RTP debug that showed packets sent to the
phone and packets received from the phone during the entire call. A
tcpdump done on the server for the interface that would deliver the
packet to the wire does not show the packets.

Is there somewhere I can look to resolve this? Something anybody has
come across? It is happening frequently and with great discomfort to
many users.

I had upgraded from 1.2.x to latest 1.4.x in attempts to resolve this.
I also disabled a lot of COM/LPT and USB devices in the BIOS to free
up some IRQ's. no devices are sharing IRQ's at this point, with I
thought might have been part of the issue, but has proved to at least
not be directly related.

These calls are from a PRI to a Cisco 7940 using SIP. There is a
Juniper EX switch between the two. Both sides negotiate at
100Mbps/Full Duplex.

I have ruled the switch out of the problem as it's not seeing the
packets on the wire when the issue is occuring.

Please help or point me to someone that can.

-Jonathan
[EMAIL PROTECTED]

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[asterisk-users] Cisco 7960

[Shawn L]

This one is a little off-topic, it's more about the phone than asterisk
itself.

I have a cisco 7960 configured with 2 lines to 2 different sip providers
(cant get
asterisk to register with the 2nd provider, but that's another story).  Is
there a
way yo determine which direction speed-dial buttons will go out?  I'd like
to have
speed-dial buttons that will go out on line2 instead of line 1.  Anyone know
if this
is possible?

Thanks


Shawn
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[asterisk-users] Callwaiting and CallerID on ZAP PSTN Line

[Jim Duda]

I have a standard analog POTS service attached to a TDM401 card.

My zapata.conf for this line has:

; PSTN connected here
;immediate=no
;busydetect=yes
;busycount=8
;musiconhold=default
mailbox=100
mwimonitor=fsk
mwilevel=512
mwimonitornotify=/usr/local/sbin/zapnotify.sh
faxdetect=incoming
signalling=fxs_ks
context=incoming
callwaiting=yes
callwaitingcallerid=yes
channel = 4

When I'm on the phone and another call comes in, I hear the callwaiting 
tone.  However, Asterisk doesn't appear to give me any CallerID 
information for the other call.  Is there some way I can gain access to 
the CallerID info on the other call?  I assume there is some 
extensions.conf magic required?  An old CallerID box can do this, so I 
would expect Asterisk to be able to do this too.

Any advice appreciated.

Best,

Jim


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Re: [asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?

[Steve Totaro]

On Wed, Aug 13, 2008 at 7:40 PM, Jonathan Miller [EMAIL PROTECTED] wrote:
 From what I can determine while troubleshooting a voice-dropping
 issue, the Asterisk server in my organization has been dropping RTP
 packets between the asterisk server process and the network interface.

 I determined this from an RTP debug that showed packets sent to the
 phone and packets received from the phone during the entire call. A
 tcpdump done on the server for the interface that would deliver the
 packet to the wire does not show the packets.

 Is there somewhere I can look to resolve this? Something anybody has
 come across? It is happening frequently and with great discomfort to
 many users.

 I had upgraded from 1.2.x to latest 1.4.x in attempts to resolve this.
 I also disabled a lot of COM/LPT and USB devices in the BIOS to free
 up some IRQ's. no devices are sharing IRQ's at this point, with I
 thought might have been part of the issue, but has proved to at least
 not be directly related.

 These calls are from a PRI to a Cisco 7940 using SIP. There is a
 Juniper EX switch between the two. Both sides negotiate at
 100Mbps/Full Duplex.

 I have ruled the switch out of the problem as it's not seeing the
 packets on the wire when the issue is occuring.

 Please help or point me to someone that can.

 -Jonathan
 [EMAIL PROTECTED]

Jonathan,

It sounds hardware specific to me.  Is this a new install or a new problem?

If it is a new problem, then what has changed?  Is the NIC in question
onboard?  What hardware are you using?  Brands, MoBo, NIC, etc...

If I were you, I would remove or disable the NIC and stick a tried and
true old school 3Com NIC in the server and try that.

Thanks,
Steve Totaro

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[asterisk-users] Asterisk 1.4 T38 UDPTL Pass Through MAX TNT and Linksys 2102

[JR Richardson]

Hi All,

I finally got the time to test t38 pass through with a TNT, * 1.4.21.1 and
Linksys 2102:

PRI TNT SIP Asterisk 2102 SharpFax

Faxing either direction, the call sets up with ulaw rtp, when fax tones hit
the line, both the TNT and the 2102 switch to t38 and udptl packets fly
through Asterisk.  All looks good, but, once udptl sets up, every few
seconds, I get a warning: 'rtp Read too short' on the Asterisk CLI from the
TNT side of the session.  Faxes never complete, not even a half page,
nothing, transmission just ends.

There are only a few parameters on the TNT that effect t38 and I've adjusted
them all with no change in the results.

Pretty much the same results when testing t38 pass through to a Cisco pri
gateway as well.

So my question is: Does anyone else have this solution working and wouldn't
not mind sharing configs?

Thanks.

JR
-- 
-
JR Richardson
Engineering for the Masses
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Re: [asterisk-users] Very loud noise on TDM400

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Carlos Chavez wrote:
   I am having a problem with and Asterisk installation where two ports
 connected to a TDM400 card will have a very loud noise when you try to
 dial.  The server has an OpenVox D110P, a TDM04B and a Xorcom Astribank
 8 fxs.  It is running Zaptel 1.4.11 and Asterisk 1.4.18.

Hi,

I had a similar problem a while back with a 2N box.  For some reason
there was a very loud click which caused the echo canceller to lose the
plot.

Have you tried disabling the echo can on that port?

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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5uGaI5kuU92tGvoNLC887Ks=
=Wn8O
-END PGP SIGNATURE-

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Re: [asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?

[Eric ManxPower Wieling]

Steve Totaro wrote:
 On Wed, Aug 13, 2008 at 7:40 PM, Jonathan Miller [EMAIL PROTECTED] wrote:
 From what I can determine while troubleshooting a voice-dropping
 issue, the Asterisk server in my organization has been dropping RTP
 packets between the asterisk server process and the network interface.
[snip]
 I have ruled the switch out of the problem as it's not seeing the
 packets on the wire when the issue is occuring.

 
 It sounds hardware specific to me.  Is this a new install or a new problem?
 
 If it is a new problem, then what has changed?  Is the NIC in question
 onboard?  What hardware are you using?  Brands, MoBo, NIC, etc...
 
 If I were you, I would remove or disable the NIC and stick a tried and
 true old school 3Com NIC in the server and try that.

This is my advice as well.  Disable the on-board NIC, remove any other 
NIC card, replace it with a well known good card.

-- 
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, 
T-1, PRI, Frame Relay, Linux, and network design.  Based near 
Birmingham, AL.  Now accepting clients worldwide.

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Re: [asterisk-users] Slighly OT?.. headset for Linksys SPA922

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Paul Hales wrote:
 That's a good question - the plantronics are available with 
 interchangeable ends - which makes them easy to move between different 
 phones.

Problem is, the headset button only works for the minijack thingy so if
you use the plantronics (plugged into the handset thing) you still need
to take the phone off the hook.

Unlike the snom 360 where there is a separate socket.

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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H5a80ulb78zhH2Iz1t2pSms=
=mJvc
-END PGP SIGNATURE-

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Re: [asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?

[Steve Totaro]

On Wed, Aug 13, 2008 at 10:41 PM, Eric ManxPower Wieling
[EMAIL PROTECTED] wrote:


 Steve Totaro wrote:
 On Wed, Aug 13, 2008 at 7:40 PM, Jonathan Miller [EMAIL PROTECTED] wrote:
 From what I can determine while troubleshooting a voice-dropping
 issue, the Asterisk server in my organization has been dropping RTP
 packets between the asterisk server process and the network interface.
 [snip]
 I have ruled the switch out of the problem as it's not seeing the
 packets on the wire when the issue is occuring.


 It sounds hardware specific to me.  Is this a new install or a new problem?

 If it is a new problem, then what has changed?  Is the NIC in question
 onboard?  What hardware are you using?  Brands, MoBo, NIC, etc...

 If I were you, I would remove or disable the NIC and stick a tried and
 true old school 3Com NIC in the server and try that.

 This is my advice as well.  Disable the on-board NIC, remove any other
 NIC card, replace it with a well known good card.

 --
 Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
 T-1, PRI, Frame Relay, Linux, and network design.  Based near
 Birmingham, AL.  Now accepting clients worldwide.


NIC card is redundant ;-)

Thanks,
Steve T

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Re: [asterisk-users] New GUI for Realtime Asterisk - RAGUI

[Steve Totaro]

Thanks to Mike and your team.  I will be checking this out and will
report back to the list.  Good stuff.

Thanks,
Steve Totaro

On Wed, Aug 13, 2008 at 7:32 PM, Mike Clark [EMAIL PROTECTED] wrote:
 Brandon:

 It manages the MySQL database tables directly via RoR. It is
 specifically built for using Realtime Asterisk. It uses #exec statements
 in the extensions.conf file to set up the context declarations.

 Thanks,

 Mike

 bkruse wrote:
 That's pretty cool, I love ruby. What method does it use to
 communicate to Asterisk? Does it manipulate raw config
 files or use manager or something similar?

 I am curious :)

 -Brandon

 Mike Clark wrote:

 Our company, WebPoint IT Solutions has just released an open source (GPL
 V2 license), Ruby on Rails based gui manager for Realtime Asterisk
 called RAGUI.

 RAGUI is definitely a work in progress and has rough edges, but we
 expect to polish it up in the upcoming weeks and months. All comments,
 contributions, and criticisms are welcomed!

 Here are the links:
 Sourceforge: http://sourceforge.net/projects/ragui/
 Website: http://www.ragui.net

 Enjoy!

 Mike Clark
 WebPoint IT Solutions


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Re: [asterisk-users] Slighly OT?.. headset for Linksys SPA922

[Darrick Hartman]

Matt Riddell wrote:
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1
 
 Paul Hales wrote:
 That's a good question - the plantronics are available with 
 interchangeable ends - which makes them easy to move between different 
 phones.
 
 Problem is, the headset button only works for the minijack thingy so if
 you use the plantronics (plugged into the handset thing) you still need
 to take the phone off the hook.
 
 Unlike the snom 360 where there is a separate socket.
 
 - --
 Kind Regards,
 
 Matt Riddell
 Director

Matt,

You can get an adapter for the Plantronics that will plug into the 2.5mm 
jack on the phone.

Darrick


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Re: [asterisk-users] Slighly OT?.. headset for Linksys SPA922

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Darrick Hartman wrote:
 Matt Riddell wrote:
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 Paul Hales wrote:
 That's a good question - the plantronics are available with 
 interchangeable ends - which makes them easy to move between different 
 phones.
 Problem is, the headset button only works for the minijack thingy so if
 you use the plantronics (plugged into the handset thing) you still need
 to take the phone off the hook.

 Unlike the snom 360 where there is a separate socket.

 - --
 Kind Regards,

 Matt Riddell
 Director
 
 Matt,
 
 You can get an adapter for the Plantronics that will plug into the 2.5mm 
 jack on the phone.

Aha!  That might solve another issue I'm working on!

Cool, thanks for that Darrick!

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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pvkELSksMP6Zq6KOTxa7N3o=
=Kcji
-END PGP SIGNATURE-

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[asterisk-users] Unable to create ZAP Channel

[Jay Ray]

Her eis my call flow 

Sofphone -- Asterisk-- X100P card-- Home Phone line

Home phone line plugged in LINE port of the FXO ccard (2 port)

Here is the error I get 


Aug 13 22:35:31 VERBOSE[8625]: -- Executing 
1;36;40mDial0;37;40m(1;35;40mSIP/xlite1-a0fb0;37;40m, 
1;35;40mZap/g1/30350612330;37;40m) in new stack
Aug 13 22:35:31 NOTICE[8625]: Unable to create channel of type 'Zap'

===
COnfigurations are:

/etc/asterisk/zapata.conf  

[channels]
group=1
stripmsd=0
signalling=fxs_ks
channel = 1

/etc/asterisk/extensions.conf 

[default]
exten = _0X,1,Dial,Zap/g1/${EXTEN:1}






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Re: [asterisk-users] Unable to create ZAP Channel

[Tzafrir Cohen]

On Wed, Aug 13, 2008 at 09:51:07PM -0700, Jay Ray wrote:
 Her eis my call flow 
 
 Sofphone -- Asterisk-- X100P card-- Home Phone line
 
 Home phone line plugged in LINE port of the FXO ccard (2 port)
 
 Here is the error I get 
 
 
 Aug 13 22:35:31 VERBOSE[8625]: -- Executing 
 1;36;40mDial0;37;40m(1;35;40mSIP/xlite1-a0fb0;37;40m, 
 1;35;40mZap/g1/30350612330;37;40m) in new stack
 Aug 13 22:35:31 NOTICE[8625]: Unable to create channel of type 'Zap'
 
 ===
 COnfigurations are:
 
 /etc/asterisk/zapata.conf  
 
 [channels]
 group=1
 stripmsd=0
 signalling=fxs_ks
 channel = 1
 
 /etc/asterisk/extensions.conf 
 
 [default]
 exten = _0X,1,Dial,Zap/g1/${EXTEN:1}

What is the output of:

  asterisk -rx 'zap show channel 1'

?

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 32

[dimitri . osler]

Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a 
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al 
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], 
altrimenti vi risponderò al mio rientro.

Dimitri Osler

I will be on vacation until Tuesday 19th of August with limited access to voice 
and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 
0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on 
my return.

Dimitri Osler





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