Re: [asterisk-users] Can't compile on NSLU2 LE

[Drew Gibson]

Gunner wrote:
> >> Hi,
> >> I keep getting the following on my slug no matter what version I 
> try (1.2, 1.4, 1.6, 1.6svn):
> >>
>
> >Do you build this on the NSLU2 machine?
> >
> >Left-over files from an x86 build? Try 'make clean' .
>
> This is native on the NSLU2.  It was right from svn, but I just tried 
> a make clean and it still has the same error.

Which distro are you using on the slug?
Have you tried installing from Optware?

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call Stealing

[Drew Gibson]

Geoff Lane wrote:
> On Thursday, January 15, 2009, Danny Nicholas wrote:
>
>   
>> Why not use call-conferencing?  If you transferred your call into a
>> conference room, you could join the conference from any extension on
>> your *. When the caller hangs up, just end the conference.
>> 
>
> Thanks for the reply.
>
> AIUI, you need to set up the conference before leaving the extension
> on which you took the call. If so, call parking would probably be
> better since that leaves the original extension free for further
> calls.
>
> Thanks again,
>
>   

Would SLA (Shared Line Appearance) work for this?

Put call on hold, press button beside flashing light on second handset?

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] top posting again [was: Re: CDR Design]

[Drew Gibson]

David fire wrote:
> you are soamming my mail box whit this useless discution
> the solution is doble posting (top and bottom)
>
> 2008/12/17 Andrew Kohlsmith (lists)  >
>
> On December 17, 2008 05:03:00 pm Eric "ManxPower" Wieling wrote:
> > To me top posting is like people talking about "SIP Trunks".
>  There is
>



I submit the above as corollary to RE Kushner's point.

Is it the spelling that is difficult or the typing?

Really, there's nothing like a good, old-fashioned pre-Christmas flame 
to get you in the Holiday mood, eh? :-)

regards,

Drew


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Some Good News for VoIP

[Drew Gibson]

http://www.theregister.co.uk/2008/12/16/infonetics_enterprise_telephony_numbers_q3_2008/

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] get first month of trixbox free

[Drew Gibson]

Hi Don,

Don Kelly wrote:


Caution---top posting. It works for me--ignore it if you like.



Without forgiving Michael's commercial message to *-users, perhaps we 
can punish your top posting by highlighting your gross insensitivity to 
the "physically challenged"! :-)


And (flame follows) we would do it using careful use of English 
grammar and spelling, especially if we were




The only major error I found were "threw" instead of "through". Perhaps 
that is forgivable for someone using a handle of "creepyBLINDy"? N'est pas?


regards,

Drew

* From: * asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of 
*Babcock, Michael Alex*

*

hi;

threw the end of the year we are running a promo, when ordering any 
package on


http://gwhosting.net

including our vps servers and trixbox servers, you can get your first 
month off. Yes, that's right, enter "30free" with out the quote signs 
into the coupon code field during checkout to get your first month 
free. Give us a try, you won't be sorry. Your security is our number 
one priority. GW Hosting, your dedicated home on the web:


http://gwhosting.net

"30free" does truly get you your free month. Stop at any time during 
your first month and you won't be charged any more, no strings 
attached. Well, wait there is one string, you have to go to


http://gwhosting.net

and sign up using "30free" to get the free month.

thanks

Michael Babcock

Michael Babcock 


Owner of GW Hosting, http://www.gwhosting.net <http://www.gwhosting.net/>

For information on what I may be doing at the moment, please feel free 
to visit my blog, twitter or brightkite at the following links: 

Twitter: http://www.twitter.com/creepyblindy 


Blog: http://www.gwfans.net <http://www.gwfans.net/>

Brightkite: http://brightkite.com/people/creepyblindy




--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] 'dialer' application to trigger call betweenhardphone and number

[Drew Gibson]

Have you looked at ADM <http://adm.hamnett.org/> ?

I don't know how it works on Windows but on Linux you can highlight any 
number on the screen then leftclick on dialer icon, middleclick into the 
dialer to make a call.

regards,

Drew


Danny Nicholas wrote:
> It doesn't matter where the program resides as long as it can interact with
> your asterisk (FTP, HTTP, etc).  You just have to have the right accesses
> and security.
>
>   
>>
>> Does anyone know of a small lightweight windows 'dialer' application I
>> can use to trigger a call (via call file or AMI) from any
>> application?  (The call would be placed between the target number, and
>> the preconfigured DN of the hardphone at the user's desk)
>>
>>  
>>
>> Ideally a phone number would be 'selected' from within any windows
>> application and the call would be triggered via hotkey, or a
>> right-click menu or by clicking a system tray icon.
>>
>> 

>> Ideal would be something very 'efficient' with at most two or three
>> discreet actions needed to dial-- (i.e. 1:Select, 2:Hotkey--done!)
>>
>>  
>>
>> Any ideas?  Any Happy customers?
>> 


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] OT: What do you guys think of this?

[Drew Gibson]

Alex Balashov wrote:
> http://www.theregister.co.uk/2008/12/01/richard_bennett_utorrent_udp/
>
> FUD?  Interesting?  Boring?  New news?  Old news?
>
>   

Seems the sky isn't falling (yet). The original article didn't have the 
full story, here's an update...

http://www.theregister.co.uk/2008/12/05/richard_bennett_bittorrent_udp/

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] OT: What do you guys think of this?

[Drew Gibson]

Ira wrote:
> At 12:44 PM 12/2/2008, you wrote:
>   
>> At 04:03 12/2/2008, Benny Amorsen wrote:
>>  >Doug <[EMAIL PROTECTED]> writes:
>>  >
>>  >> "Net Neutrality" is great in principle.  But ISP's need to
>>  >> somehow control those few percentage of users who suck down
>>  >> a huge majority of the bandwidth.  It's dollars and cents.
>>  >
>>  >Yes, just like the airlines need to somehow control those users who
>>  >keep showing up to the flight they booked, every single time! It's
>>  >impossible to do overbooking with customers like that, so we need to
>>  >find ways of punishing them.
>>
>> What happens if everyone who owns a car drives
>> it at the same time?  Owns a telephone and
>> uses it at the same time?
>> 
>
>
> If I could get the same plan for my internet as I get for my phones, 
> a few dollars a month plus a bit per minute(megabyte), I'd be all 
> over it, but even better, then the provider wouldn't have to care as 
> they'd be making a fair profit no matter what the user did.
>   

You make that sound almost reasonable. I'm sure initial pricing would 
only be slightly higher for the majority of customers with only the 
"bad" users being "punished".

Six months later, when all the fuss has died down, the price goes up by, 
say, 0.5c/MB. That's not much is it? Half a penny? After all, they used 
to raise it by $2/month and customers only grumbled a bit before paying. 
So 0.5c isn't much, is it?

...and BTW, I'm not a p2p or torrent user but I do enjoy having 
bandwidth available when the latest Fedora or Ubuntu comes out.

regards,

Drew


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] RTCP too short

[Drew Gibson]

Do you have Grandstream phones?

I noticed a similar issue last year with Grandstream GXP2000 phones. The 
phone was sending an empty RTP packet for the "keepalive" whilst on 
mute. I reported a bug to Grandstream but nothing happened.

regards,

Drew



Jon Weisman wrote:
> I get this all the time. Still haven't found a solution but it doesnt 
> seem to affect call quality or server performance. I think there's a 
> way to disable the message, but I lost the link. :(
>  
> -Jon
>
> - Original Message -
> *From:* michel freiha <mailto:[EMAIL PROTECTED]>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> <mailto:asterisk-users@lists.digium.com> ;
> [EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>
> *Sent:* Friday, November 28, 2008 10:07 AM
> *Subject:* [asterisk-users] RTCP too short
>
> Dear Sir,
>
> I'm running Asterisk 1.4.21.2 <http://1.4.21.2> on a CentOS
> machineWhen running asterisk -rv I can see a lot of
> messages about RTCP too short...
>
> -- Remote UNIX connection disconnected
> [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP
> Read too short
> [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP
> Read too short
> [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP
> Read too short
> [Nov 28 13:33:02] WARNING[29804]: rtp.c:891 ast_rtcp_read: RTCP
> Read too short
>
> Can you let me know how to fix this issue?
>
> Regards
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP to IAX2 with delayed echo

[Drew Gibson]

c james wrote:
> A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
> having a conversation.  Call quality is reported as good except for an
> echo with a 3 second delay.
>
> Most of my searches are saying echo happens only on the PSTN piece, but
> there isn't one here.
>
>   

Which end hears the echo?

If it is the Polycom end, try a better quality headset with the softphone.
Echo comes from analogue portions of the "circuit" and is usually caused 
at the end that doesn't hear it.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Recommend Wireless IP Phone

[Drew Gibson]

Wilton Helm wrote:
> Good points.  I got an access point instead of a router specifically 
> so I could locate it in the best position.  IMO Wi-Fi routers are dumb 
> by definition because where you want a router is probably NOT anywhere 
> close to the best point for the Wi-Fi part.  This unit has a 
> particularly sensitive receiver to compliment the higher power.  It 
> would have been nice it it had MIMO, too, as that always helps.  
> Repeaters would be a challenge in this case because most of the 
> property is natural wooded (so no power or protection) and I'm trying 
> to cover a road by only own property at one end.

"naturally wooded" does not bode well for WiFi. Trees are much better 
than walls at absorbing 2.4GHz signals due to their high water content. 
Mountains block 2.4GHz even better. If the woods are deciduous, it may 
work well in the winter but fade away come spring.

If the road is fairly straight, a directional antenna like a Yagi at one 
end might give you coverage there. As for the rest of your property, you 
will have to get an omni antenna up high, say one of your mountains.

You may be better off with something that uses lower frequencies. The 
old analogue cordless phones have much better range than 2.4GHz digital 
stuff.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dundi Issues

[Drew Gibson]

Jeremy Mann wrote:
> I'm not aware of any offloading done on this particular box, it's an HP ML110 
> G5 using the onboard NIC.
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson
> Sent: Wednesday, November 05, 2008 9:30 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Dundi Issues
>
> Jeremy Mann wrote:
>   
>> I don't know if it's related, but when doing a packet sniff with
>> wireshark, I see UDP checksum incorrect messages:
>>
>> 0.058230 source -> destination UDP Source port: 4520 Destination port:
>> 4520 [UDP CHECKSUM INCORRECT]
>>
>> 
>
> Be careful with this error, some network cards that can do "IP Offload"
> processing will show up with bad checksums in Wireshark. Check the specs
> for your NIC, this may be a Red Herring (or it might not! ).
>
>   

The HP docs indicate a setting for enable/disable checksum offload, try 
looking at the packets "on-the-wire" rather than on the server itself.

http://h2.www2.hp.com/bc/docs/support/SupportManual/c00846707/c00846707.pdf

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dundi Issues

[Drew Gibson]

Jeremy Mann wrote:
>
> I don’t know if it’s related, but when doing a packet sniff with 
> wireshark, I see UDP checksum incorrect messages:
>
> 0.058230 source -> destination UDP Source port: 4520 Destination port: 
> 4520 [UDP CHECKSUM INCORRECT]
>

Be careful with this error, some network cards that can do "IP Offload" 
processing will show up with bad checksums in Wireshark. Check the specs 
for your NIC, this may be a Red Herring (or it might not! ).

regards,

Drew


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] VoIP traffic shaping

[Drew Gibson]

Kristian Kielhofner wrote:
> On 11/1/08, OCG Technical Support <[EMAIL PROTECTED]> wrote:
>   
>>
>> This was so interesting I had to move it to its own thread!
>>
>>
>> Is anyone using this script?  How does it perform compared to the older
>> WonderShaper script?
>>
>> 
> It was based off Wondershaper originally, enhanced for VoIP traffic
> and gives the option to use HFSC or HTB.  Not only do I use it myself
> for AstLinux and Star2Star, most of the reports I've (we've) had have
> been favorable.
>
> Try it out!
>
>   

Has anyone tried this on OpenWRT?

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] network design philosophy and practice

[Drew Gibson]

Kristian Kielhofner wrote:
> On Wed, Oct 29, 2008 at 10:48 AM, Drew Gibson <[EMAIL PROTECTED]> wrote:
>   
>> Worst part is the few Cisco phones we have insist on "searching for
>> VLAN" (which doesn't exist) for 5 minutes on startup. Hopefully they
>>
>> 
>
> Drew,
>
>   Disable CDP on the phone and that will go away.  I know you said
> you're not using VLANs but...
>
>   You can use CDP and set your voice-vlan on Cisco switches.  Or...
> you can install cdp-tools on a Linux box and have it advertise a voice
> vlan for you!
>
> http://gpl.internetconnection.net/
>
>   I added the voice vlan support to cdp-tools. ;)
>
>   

I tried out the cdp-tools some time ago (it may have been on your 
recommendation, Kristian) but with no success.
Is it possible to disable CDP on the 7940 (image_version : "P0S3-08-2-00")?

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] network design philosophy and practice

[Drew Gibson]

David Gibbons wrote:
> Two separate networks? Did I miss something? I feel like I'm taking crazy 
> pills! Two separate physical networks means twice the hassle, twice the 
> maintenance, twice the cost, twice the headache. Not to mention the fact that 
> the whole idea of VOIP is to simplify IT and focus on converging data and 
> voice networks.
>
> This is what VLANs and QOS do best. I dare say it's what they were designed 
> foe. I can't think of any reason that I would ever recommend two ports per 
> desk to support telephony -- ever. It's ludicrous to think that two ports 
> will be better than one if we're setting up our VLANs and QOS properly. A 
> phone takes very, very little bandwidth away from the desktop and a decent 
> one will support tagging its frames for the alternate voice VLAN.
>
> --snip--
> In almost all cases it is much better to have two seperate networks.
> This may be impractical in some smaller installs, but in any office
> setting we always do this. The only reason I can think of not to is to
> eliminate the cost of the second cable.
> --snip--
>   


That's two _logically_ separate networks. The key point is that the 
"last yard" cable to the phone is not shared with the computer.
The issue is not a lack of bandwidth but that the phone has to try and 
get its little packets inserted between the massive packets of a 
database lookup or file transfer in a timely manner (latency and jitter).

You might get away with a single logical network on a smaller site or a 
larger one with very light traffic.

QoS is not required on lightly loaded links and will do nothing for you 
on over loaded ones. I only use it on WAN links where bandwidth is more 
expensive.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] network design philosophy and practice

[Drew Gibson]

Bill Michaelson wrote:
> I'm wondering how prevalent the practice of physically segregating 
> voice and data networks is in the Real World.
>
> What are the factors that typically lead to such a decision?  
> DIscussions of pros and cons are most welcome by me.
>
> Experiences, anybody?
>

We chose to go with a segregated network and certainly don't regret the 
choice. Voice and data are on separate ports at the desk, avoiding QoS 
issues completely and reducing confision amongst users who still expect 
separate Phone and Computer plugs on the wall.
The traffic does run through the same switches and inter-switch trunks 
but always on distinct VLANs.

My experience with connecting the desktop computer through the phone has 
been very poor. Audio breaks up when the computer does large data transfers.

"Yes, Sir. I'll just look that up in our 
datab...baba.bas.ss..ss..se"

In addition our users require gigabit to the desktop. The phones are 100Mb.

Worst part is the few Cisco phones we have insist on "searching for 
VLAN" (which doesn't exist) for 5 minutes on startup. Hopefully they 
will be replaced through attrition but despite being over-priced, 
over-featured and proprietary, Cisco do build robust kit. Sigh.

regards,

Drew


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Emerging dilema? DID forwarding meets SMS

[Drew Gibson]

Gordon Henderson wrote:
> On Thu, 23 Oct 2008, Karl Fife wrote:
>   
>> We have a number of DID's that do the standard VoIP tricks: ringing
>> multiple locations, findme-followme etc.  What is happening more and
>> more is that customers call those DID numbers, and draw the reasonable
>> conclusion that they are calling mobile numbers because they literally
>> can HEAR that the called party is on a mobile.  Consequently many of
>> those customers draw the conclusion that they can safely send SMS's to
>> those DID numbers.  Naturally the SMS messages disappear into the ether.
>> 
>
> Er, they don't dissapear for me. I send a TXT to a landline, the phone 
> rings and there is a text to speech robot which reads it out to you, or, 
>
>   

> Don't you have that facility?
>
> Maybe it depends on country and telco.
>
>   

Err, Gordon, you must be in a country from the 21st century.

North America is just beginning to emerge from the mobile Stone Age. 
Some people have heard of text messaging but most think you have to pay 
Blackberry to send emails.

I ran into the issues Karl mentions when trying to txt our ISP contact 
during our office move.
Can anyone clarify how SMS to non-mobile numbers are generally handled 
in North America?
Is it possible to have SMS delivered direct to your landline DIDs? Then 
have Asterisk relay it to the actual mobile DID.


regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] anoyingly answers already in use pstn line

[Drew Gibson]

Tzafrir Cohen wrote:
> On Fri, Oct 17, 2008 at 05:04:32PM -0400, Gleim, Jason wrote:
>>>
>>> I am using Asterisk and an X101P card as a glorified answering
>>>   
>> machine.
>> 
>>> We have a residential PSTN line with about six phones connected to it.
>>> Like an answering machine, I want Asterisk answer the line *only* when
>>> an incoming call is not answered after four rings.
>>>
>>> This mostly works. My extensions.conf is at the end of this message.
>>>
>>> The problem is that Asterisk will sometimes answer the line when
>>> someone
>>> is already talking on one of the six phones connected to it. Sometimes
>>> Asterisk will answer the line and start playing the greeting in the
>>> middle of a conversation! This is especially a problem when I am
>>> talking
>>>
>>>   
>> Others may wish to chime in and confirm or deny this but the card is
>> probably getting confused by you loading the line with the other phones.
>> I know most of the analog cards I've worked with (which does not include
>> the X101P) really get cranky if there is anything else hanging off that
>> line. The only solution I've seen to the problem is to change things
>> around so that the card is the only thing on the line.
>> 
>
> The "cranky" card here is not the issue. It would be the same with any
> other card.
>
>   
>> In know you said you haven't switched to IP or FXS but is there a reason
>> why? 
>> 
>
> That would require rewiring.
>
>   

I swapped my X101P for a Linsys SPA3102 some time back.

Calls come in to the SPA on its FXO port and get forwarded to Asterisk 
which then rings the "legacy" phones on the SPA's FXS port, as the other 
members of the household (read "wife") are used to. All of Asterisk's 
features are available.

If any issues arise, you just pull the power to the SPA and calls just 
pass through directly to the legacy phones.
In addition, the SPA is supposed to pass calls through to the FXS if it 
loses its registration with Asterisk (eg Asterisk crashes) but I never 
got this working.

One day, I would like to teach Asterisk to behave like an old-fashioned 
answering machine to allow a "second chance" to catch calls after the 
fourth ring but I haven't had the time yet.

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Text messaging and Asterisk

[Drew Gibson]

C. Savinovich wrote:
>   Can somebody please give a pointer to a complete neophyte (like me) on
> text messaging, what product can I use to send and automatic text message to
> a cell phone from within the asterisk dialplan? (the part of the dialplan I
> have down, the part of the text message no)
>
> Thanks
> C. Savinovich
>
>   

I don't use it but on my Asterisk 1.4 slug there was a file 
/etc/asterisk/fastsms.conf which had info about connecting to SMS 
services for about 4c per txt.

regards,

Drew


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] is there a way

[Drew Gibson]

Steve Totaro wrote:
>
> My only wish is that Linux had a facility like XP to bridge NICs 
> without running all sorts of commands for brctl.  Just a GUI like XP.  
> Last time I setup a bridge in Linux, I had to change many kernel 
> options and rebuild the entire kernel to get bridging working 
> properly.  With XP, you just select the NICS, right click and select 
> add to bridge.
>

I've always seen this as (YA) flaw in Windows. Bridging seems to be the 
lazy man's way of avoiding configuring Windows networking properly 
(admittedly, this is not a simple task), especially with OpenVPN. 
Bridging should be left to switches, they're the professionals!

> For linux, I find that running firestarter, ICS/Firewall is fine, my 
> end game is to get all of my traffic to go over an OpenVPN tunnel at 
> my colo which is the default gateway over OpenVPN.  Windows seems to 
> have the easiest method of getting this done.

OpenVPN has an option to do just this, "redirect-gateway", that will 
push (almost) all the traffic out through the tunnel. Works the same 
with any supported OS

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] PoE switch recommendations?

[Drew Gibson]

Ken D'Ambrosio wrote:
> Hey, all.  We're rolling out VoIP, and I'm wondering about PoE
> recommendations, as we're going to have to replace our current network
> equipment.  My first inclination would be to just plunk down the cash and
> do a Cisco system, but I'm relatively certain that would get shot down by
> finance.  Any recommendations for a couple-hundred-port solution with
> VLANs, PoE, and QoS?  Don't care much if it's in a single chassis or not,
> so long as it has Gbit uplinks.
>   

We tried a Linksys SRW208P about 2 years ago but couldn't use it due to 
the noise from the fans (think "jet engines")
We settled on the Netgear 16/8 port (8 PoE + 8 non-PoE) switches for the 
call centre "pods", each supporting 5 phones. None have caused problems 
to date.

In the server room, we have Dell PowerConnect 3548P switches which are 
fully managed driving a mix of phones, access points cameras. They are 
excellent value, Cisco are way overpriced.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] OT: Do You Know What the Problem With CDMA is?

[Drew Gibson]

Andrew Kohlsmith (lists) wrote:

On September 25, 2008 10:41:45 am Drew Gibson wrote:
  

Once CDMA has gone the way of the dodo in North America, I really will
miss one of my favourite scenes:-

Visiting Brit steps off plane and checks phone for messages...

Puzzled look appears as they ask "Why doesn't my phone work? It worked
fine in France/Italy/Germany/Timbuktu."

You start to explain about CDMA and their eyes open wide as they realize
they have just stepped back into the cellular stone age...



You don't have AT&T towers near airports?

-A


Nope, no AT&T north of Buffalo.

To be honest, it happened a few years ago (~2002).

We now have Rogers' towers near airports (and 3G iPhones in stores).

Bell Canada and Telus are moving to GSM 3G (side-stepping standard GSM 
so they don't have to admit their mistakes)



regards,

Drew

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] OT: Do You Know What the Problem With CDMA is?

[Drew Gibson]

Andrew Kohlsmith (lists) wrote:

On September 25, 2008 09:01:52 am Dean Collins wrote:
  

Yep you got it world coverage includes all the countries of the
world like USA, Canada and Mexico, and not something like USA and 212
other countries globally.

BTW I hear that Iraq also now uses CDMA (some senator shoe-horned it
into a funding bill for the war that they had to use CDMA to 'support
usa businesses'), of course that means that they now use a different
handset type to all of their neighbours... though I hear Iran will
also be forced to implement CDMA once they are 'liberated' which should
be any day now :)



That doesn't mean that GSM towers won't be built, it just means that the CDMA 
towers will be there first.


I dunno; GSM 3G is all CDMA tech anyway.
  


Yes, but it is a standard agreed upon by a large number of carriers 
around the world.


Once CDMA has gone the way of the dodo in North America, I really will 
miss one of my favourite scenes:-


Visiting Brit steps off plane and checks phone for messages...

Puzzled look appears as they ask "Why doesn't my phone work? It worked 
fine in France/Italy/Germany/Timbuktu."


You start to explain about CDMA and their eyes open wide as they realize 
they have just stepped back into the cellular stone age...


regards,

Drew

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] sip conversations overlapping!!!!

[Drew Gibson]

RoLaNd RoLaNd wrote:
> i appologize for not making myself clear..
>
>
> i have my asterisk box, connexted to 4 sipura3102..
>  these sipuras has 4 PSTN lines connected to them through one cable, 
> which has 8 lines inside of it (2 connected to an RJ11 and plugged 
> into its respecitve fxs port in the sipura)

Lines plugged into the fxS ports? I hope you have them in the "LINE" 
ports (fxO)
Are there any telephones plugged directly into the Sipuras? Into the 
"PHONE" ports (FXS)

> my prob is as such:
>
> when i call from softphone#1 to sipura #1, sound is pretty good and 
> everything is working perfectly.. though if asterisk recieves a call 
> from another sipura.. lets say its sipura #2, then! i could hear the 
> attendnat answering the incoming phone in my current conversation, and 
> i could hear some1 picking up and answerinfg the call..!
> if i ask them to hang up! my line breaks as well..

I would double check the wiring of the "8 line" cable. 4 POTS lines = 4 
pairs of tip & ring. Are there some of the tip&ring pairs mixed up? eg 
tip from line 1 mixed with ring from line 2, etc.
This is the most likely scenario since I can't imagine Asterisk bridging 
the calls without being asked to.

Otherwise, are we still missing something in the topology here?

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] sip conversations overlapping!!!!

[Drew Gibson]

RoLaNd RoLaNd wrote:
>
> Hi all,
>
> i'm facing this weird prob...my topology is as such:
>
>   
>  
> -
> -
>
> when am on a call, sometimes when some1 else tries to call out.. i 
> hear the actual tones which ends up preventing the other end from 
> talking to me..
> moroever, when some1 calls me through one sipura, while im talking on 
> another... i can hear the attendant welcoming message, then i hear the 
> voice of whoever have picked tht line up..! and if i ask that person 
> to hang up... my line breaks as well..!
> can any1 help me with this issue!
> below is my config:
>

How are the analogue phones wired? One phone plugged directly to one 
3102 FXS port? or is there common wiring ?
Are all the FXO ports connected to telco lines?


regards,

Drew

NOTE: Holding the  key down whilst typing the first person, 
singular, pronoun will produce stunningly readable results. Either 
 key will do, you can even use the  key if both of 
those are broken/can't locate them. You can also use this procedure for 
the first letter of each sentence, it makes everything much easier to read.

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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

[Drew Gibson]

We use

http://www.areski.net/asterisk-stat-v2/about.php
http://www.micpc.com/qloganalyzer/

on Asterisk 1.2, don't know how well they work with later versions

regards,

Drew


Mark Hamilton wrote:
> 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?
>>
>>
>> 
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>> 
>
>   


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Limit to the length of string ?

[Drew Gibson]

Gordon Henderson wrote:
> On Tue, 26 Aug 2008, Tilghman Lesher wrote:
>
>   
>> On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote:
>> 
>>> A-Ha... That string is 256 characters long... Now there's a fishy number
>>> if ever there was one.
>>>
>>> So, if this a real limitation? This is 1.2.30 if that makes a
>>> difference...
>>>
>>> Did this limit go away in 1.4 ?
>>>   
>> Yes, it did.
>> 
>
> OK. Thanks.
>
> Now I guess I have to play the 1.4 lottery :)
>
> (or will it get "fixed" in 1.2.31 ;-)
>
>   

Is there a maximum string length for use with the legacy interface 
chan_string?
Does it depend on the type of cup used? Does styrofoam give better range 
than paper?

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] implementing an intercom with asterisk

[Drew Gibson]

Jonathan Disher wrote:
> I am looking to replace the phone system at my father's shop with an  
> Asterisk box and some Cisco phones, but one piece of the  
> implementation is tripping me up.  He has two buildings (the office,  
> and the shop proper), separated by about 3-400 yards.  Currently with  
> the ancient Meridian system installed, there is a paging intercom (to  
> page employees, etc) on a dedicated extension - play a loud tone, then  
> set up a 2 way channel.  Anyone got any ideas, hardware wise, on how I  
> might implement this with an Asterisk system?
>
> Thanks, and if this isn't appropriate for this list, if anyone has a  
> better destination for the question, I"d be quite appreciative.
>
>   

Hi Jon,

how is the existing intercom implemented? Is it on the phones or 
separate speakers?

If it is a paging system wired into the Nortel, you may be able to reuse 
it via an ATA,

If it's done through the phones, try 
http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom 
for ideas

That is, once you've bridged the ethernet across the 3-400 yards. You 
can use fibre, wireless or Thicknet. Fibre is the most robust, wireless 
is the cheapest and Thicknet, well, good luck getting the parts! :-)

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Siemens Gigaset IP in USA (S685 IP in particular)

[Drew Gibson]

Paul Chambers wrote:
> For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP 
> range in the U.S. I'm particularly interested in the Gigaset S685 IP. 
> Since it's DECT 6.0, and there's an English (UK) version, I'm thinking 
> it should work just fine, after dealing with the walwart issue (and 
> maybe caller ID signalling).
>
> Anyone imported one from the UK and using it in the US? for how long? 
> impressions? anything not working?
>
> Have you purchased additional US-spec handsets and used them with the UK 
> basestation?
>
> Thanks in advance,
>
> Paul
>   

The original DECT standard uses 1880-1900MHz, as implemented in Europe.

The US FCC designated 1920-1930MHz. This is marketed as DECT 6.0.

The FCC might get angry at you for using regular DECT phones in the US.

And your neighbours with iPhones (GSM) might also get angry...

regards,

Drew


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Two peers, same IP and port

[Drew Gibson]

Chris Hastie wrote:
> Is it possible to have two peers register to Asterisk from the same
> IP/port combination?
>
> I have a Zoom 5821 two port ATA that can support up to 4 VOIP accounts.
> I want to use it to provide two different extensions on an Asterisk
> system. In the past I have configured two port ATAs to use a different
> SIP local port for each account, but the Zoom unit does not appear to
> allow the SIP local port to be specified on an account by account basis.
> Can I get the unit to register two separate accounts on Asterisk from
> the same port and IP?
>
>   
Hi Chris,

from testing I did a  year ago with 1.2, I would say that his is not 
possible. Asterisk was tracking the registration by IP and could not 
differentiate the accounts by port number alone.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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

[Drew Gibson]

Do you have a firewall enabled? If so, disable it.

Else run "/etc/init.d/iptables stop" (or equivalent) and try again.

regards,

Drew



Jonathan Miller wrote:
> We put a 3c509 in, just for posterity, and it did not help the issue.
> I verified that the NIC is not sharing any interrupts, that there is
> no excessive disk wait, and that asterisk thinks it is sending the
> packets. They are simply not making it to the physical interface.
>
> Is there somewhere else I can look or something else I can do? How
> would I go about prioritizing the asterisk process? There's not a lot
> of processes running besides asterisk itself, dhcpd, tftpd and
> postfix. Really just kernel stuff after that...
>
> This is killing me. Voice drops out for various periods (between 1/2
> and 5 seconds) and lost packets do not show up with a rtp rtcp
> stats...
>
> This is weird. Any help you can offer would be appreciated. We spent 6
> hours on phone with Digium support yesterday and could not locate an
> issue within asterisk itself.
>
> -Jonathan
>
>
> On Wed, Aug 13, 2008 at 9:08 PM, Steve Totaro
> <[EMAIL PROTECTED]> 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.
>>>
>>> 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
>>
>> 


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Problem with Aastra 480ci and qualify=yes

[Drew Gibson]

James Lamanna wrote:
> Hi,
> We have a few Aastra 480ci phones and we've noticed that in order to
> get the phone to receive a call, qualify must be = no.
> Apparently the Aastras do not respond to the qualify message (or
> respond in a way Asterisk doesn't understand) and Asterisk thinks the
> phone is unreachable.
> However, this now prevents MWI from working properly on the phones.
>
> Does anyone know how to get MWI working without qualify? Or how to get
> qualify working again with the Aastras?
>
>   

We have a number of 480i and one 480ct all setup with "qualify=yes" 
(Asterisk 1.2.24)
Our inbound call centre seems to be pretty busy and my own MWI lights up 
far too often. Never had a problem with either.

Which version of firmware?
Which version of Asterisk?
What's in your sip.conf?
What error messages show on the console?
Anything relevant in the logs?

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Multi-homed Asterisk

[Drew Gibson]

Jon Weisman wrote:
> Hey guys its me again! So I need to setup our Asterisk server with multiple 
> IP providers. The server has two NICs, we have two providers, now we want 
> redundancy! Any guides on how to set this up? We're running Fedora Core 5 w/ 
> Asterisk 1.4.
>
> What I would like is that incase one link goes down, or even if the link is 
> showing up, but for some reason its not routing traffic the other link 
> should take over. This is for both inbound and outbound SIP. Now the hard 
> part... I need this to be reliable.
>
> Your help is greatly appreciated,
>
> Jon 
>
>   

I assume it is the Internet connection that you want to make redundant.

If you have any inbound connections, you will have to have a router 
running BGP and your two providers will have to co-operate to get this 
setup and working properly. We have two connections from the same 
provider and it did not cost us anything to add BGP support.

The router can be a dedicated device such as from Cisco, et al or can be 
a Linux-based machine running Zebra.

There are other "budget" methods to create a form of redundancy but they 
have drawbacks (see post from GH)

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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

[Drew Gibson]

Kevin P. Fleming wrote:
> 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.
>
>   
Thanks Kevin,

sorry I am unable to test the svn version because the site is 
(literally) on the other side of the world. On-site visits for screw-ups 
are rather pricey! :-)

No rush for the update, I will keep my eyes open for the new release.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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.
>>>
>>>
>>> -
>>> ---
>>>
>>> ___
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: 
>>> http://www.astricon.net
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>   
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: 
>> http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> 
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] AA50 Failover

[Drew Gibson]

Brent Torrenga wrote:
>> Set the primary unit to pick up on, say, 2 rings and the secondary unit 
>> to pick up on 4.
>> If the primary fails, it won't pick up the line and the secondary takes 
>> over.
>>
>> You could try having your phones register to both but I'm not sure how 
>> that would work
>>
>> regards,
>>
>> Drew
>> 
>
> That's exactly how I have things setup.  Phones register to both servers.
> The backup server doesn't do much - only answers incoming calls if the ring
> goes on for too long.  Keep in mind that you may want to configure the
> backup server to not use the lines for placing outbound calls - maybe setup
> a SIP trunk for that.  Otherwise, when the primary server comes back up, the
> two will be talking over each other trying to use the same lines.
>
> --Brent
>   

That's where the fax/modem sharing box comes in, if one device port has 
the line, all other device ports are blocked. Downside is that this will 
also cause dead air on an outbound call or two until things settle down 
again.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] AA50 Failover

[Drew Gibson]

Tzafrir Cohen wrote:
> On Thu, Jul 31, 2008 at 02:10:34PM -0400, Dave Welsh wrote:
>   
>> If I buy two AA50s can I set them up so that everything runs through the 
>> first one, but the second one will take over if the first one goes down? 
>>I can see the extensions recovering, because they use ethernet, but 
>> what about the FSO lines? Is there a way they can be spliced to both 
>> AA50s so that no one need to do any emergency rewiring?
>> 
>
> Thinking aloud:
>
> what happens if you just connect th two units on the same phone line?
>
>   

Use one of those fax/modem/phone sharing boxes that blocks the other 
devices when one is off hook.

> You basically need a way to prevent the slave unit from answering calls
> if the master is alive.
>
>   

Set the primary unit to pick up on, say, 2 rings and the secondary unit 
to pick up on 4.
If the primary fails, it won't pick up the line and the secondary takes 
over.

You could try having your phones register to both but I'm not sure how 
that would work

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] comparing pots solutions for asterisk

[Drew Gibson]

Eric Fort wrote:
> I've been looking at various solutions for getting FXS and FXO lines 
> in and out of asterisk. one solution is using TDM-400 cards.  Another 
> solution is using the grandstream GXW400x and GXW410x gateways.  Cost 
> per port seems lower on the gateways and no pci slot is required.  Why 
> would one choose to use the TDM-400 cards?  what would be the 
> advantages and disadvantages of each approach?
>

The internal card should give you higher reliability as there are fewer 
parts and cables although the external gateways could allow you to have 
redundant servers.

External gateways would also be easier to scale when you need more lines.

Does anyone have experience with the Grandstream gateways? Are they more 
reliable than their GXP2000 phones? Can they actually provide support yet?

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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

[Drew Gibson]

Drew Gibson wrote:
> Tony Mountifield wrote:
>   
>> A customer has an Asterisk box with two TDM400P cards, running [EMAIL 
>> PROTECTED] 2.8,
>> which has Asterisk 1.2.7.1 and Zaptel 1.2.5. This has been running ok
>> for a while, with only some small issues. They have FXO ports going
>> to analogue POTS (UK standard) lines, and SIP phones for extensions.
>>
>> I just tried updating Zaptel to 1.2.26 and Asterisk to 1.2.30.1, and
>> although the update appeared to go ok, the system would no longer
>> detect incoming calls. It appeared not to see the ringing signal.
>> Outgoing calls over the PSTN lines still worked fine.
>>
>> Does anyone know if there have been changes in the 1.2 series that
>> affects ring detection on the TDM400P FXO ports? Any critical settings
>> in zaptel.conf or zapata.conf?
>>
>> For now, I've had to revert to the original versions, in which ringing
>> once again works fine.
>>
>> Cheers
>> Tony
>>   
>> 
>
> YES!
>
> I have a 4xFXO board in Singapore. Having tried many various settings, 
> etc., I had assumed it was the I18N of the ringtones and signalling.
>
> Does anyone know at which rev the issue appeared?
>   
Tzafrir,

Tony tried Zaptel 1.2.26, and I was at 1.2.25, both were failing to 
pickup. I just downgraded to 1.2.24 and the system is now picking up 
incoming calls. Asterisk remained at 1.2.28.

 From the Changelog file I found a possibly relevant change...

2008-04-04 04:29 + [r4126-4132]  sruffell <[EMAIL PROTECTED]>:

 ... involved wctdm.c and fxo_modes.h. Merges with 1.4 code?

Any use?

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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

[Drew Gibson]

Tony Mountifield wrote:
> A customer has an Asterisk box with two TDM400P cards, running [EMAIL 
> PROTECTED] 2.8,
> which has Asterisk 1.2.7.1 and Zaptel 1.2.5. This has been running ok
> for a while, with only some small issues. They have FXO ports going
> to analogue POTS (UK standard) lines, and SIP phones for extensions.
>
> I just tried updating Zaptel to 1.2.26 and Asterisk to 1.2.30.1, and
> although the update appeared to go ok, the system would no longer
> detect incoming calls. It appeared not to see the ringing signal.
> Outgoing calls over the PSTN lines still worked fine.
>
> Does anyone know if there have been changes in the 1.2 series that
> affects ring detection on the TDM400P FXO ports? Any critical settings
> in zaptel.conf or zapata.conf?
>
> For now, I've had to revert to the original versions, in which ringing
> once again works fine.
>
> Cheers
> Tony
>   

YES!

I have a 4xFXO board in Singapore. Having tried many various settings, 
etc., I had assumed it was the I18N of the ringtones and signalling.

Does anyone know at which rev the issue appeared?

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] OT - How to test tftp for phones provisioning

[Drew Gibson]

Olivier wrote:
>
>
>  tftp is the client, do you have it installed ?... example:
>
>  # tftp hostname
>  tftp> get /srv/tftp/foo.txt
>  tftp> ^D
>  # cat foo.txt
>  ...
>
>
> That was exactly what I was after : I installed tftp on my Ubuntu 
> system and checked Debian tftp server
>
>
>
>  Things to check: is /srv/tftp the "tftp" directory or is it the
> os filesystem
>  directory where the tftp root resides ?
>
>  Also, the tftp daemon in CentOS is started by xinetd and can be
>  invoked with extra -v flags so as to increase logging verbosity.
>  Check your dist. This may help...
>
> Yes, that's the next step.
> I could see a tftp service is running ok on my server and I need to 
> increase its logs to pinpoint root cause.
>

Olivier,

also check that you don't have any firewalls in the way, especially if 
there is nothing in the logs. Turn them off for testing, on both server 
and CLIENT. I found out about the client f/w blocking tftp the hard way! :-)

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] need ata suggestion

[Drew Gibson]

randulo wrote:
> On Wed, Jun 18, 2008 at 5:05 PM, Tilghman Lesher
> <[EMAIL PROTECTED]> wrote:
>   
>> those devices, a tinfoil hat would block the interference (as long as it
>> contains no holes).
>> 
>
> How can you put holes in interference?
>
>   
... and how many would it take to fill the Albert Hall?

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] fxotune question

[Drew Gibson]

Drew Gibson wrote:
>>>> Tilghman Lesher wrote:
>>>> 
>>>> My fxotune.conf:-
>>>>
>>>> 13=7,255,251,251,2,255,255,1,255
>>>> 14=9,254,251,255,2,0,1,0,0
>>>> 15=9,254,251,255,2,0,1,0,0
>>>> 16=5,0,0,0,0,0,0,0,0
>>>> 17=9,254,251,255,2,0,1,0,0
>>>> 18=9,254,251,255,2,0,1,0,0
>>>> 19=9,254,251,255,2,0,1,0,0
>>>> 20=9,254,251,255,2,0,1,0,0
>>>> 21=9,254,251,255,2,0,1,0,0
>>>> 22=9,254,251,255,2,0,1,0,0
>>>> 23=9,254,251,255,2,0,1,0,0
>>>> 24=9,254,251,255,2,0,1,0,0
>>>>
>>>> regards,
>>>>
>>>> Drew
>>>>   
>>>> 
>>   
>> 
>
> Thanks Tilghman,
>
> I will try flipping one of the lines at the weekend, can't touch it 
> during the week.
>
> Our rxgain was raised from 0.0 to 2.0 at the end of February after 
> complaints that CC staff couldn't hear customers but the 'static' issues 
> pre-date that change.
>
> regards,
>
> Drew
>
>   

Hi Tilghman,

I thought I had found something, all of the lines were patched in with 
Cat 5 patch cords except Port 16 which had a telephone cable (which 
would flip the polarity). After changing all the patch cables to 
telephone type, I re-ran fxotune but the results were almost identical 
to the original settings.

new fxotune.conf:-

13=7,255,251,251,2,255,255,1,255
14=9,254,251,255,2,0,1,0,0
15=6,1,254,253,0,255,0,0,0
16=5,0,0,0,0,0,0,0,0
17=9,254,251,255,2,0,1,0,0
18=9,254,251,255,2,0,1,0,0
19=9,254,251,255,2,0,1,0,0
20=7,255,251,251,2,255,255,1,255
21=9,254,251,255,2,0,1,0,0
22=9,254,251,255,2,0,1,0,0
23=6,1,254,253,0,255,0,0,0
24=9,254,251,255,2,0,1,0,0

These results are very similar to John Morey's, eg. not much changed 
after reversing polarity.

The one line that shows zero's all the way across is the line with our 
DSL backup line on it. Is it the filter giving these results or does the 
telco take more care balancing lines for DSL?

Shouldn't I have seen a major change of settings after changing polarity 
for tip&ring?

regards,

Drew


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Anyone using zaptel analogue hardware in Singapore?

[Drew Gibson]

If anyone is using or have experience of Asterisk with zaptel hardware 
on a POTS line in Singapore? If so, would you mind sharing your zaptel 
and zapata configs?

I'm having a little trouble getting my new server to answer calls 
(outbound is working, see thread "Asterisk not picking up incoming 
calls...")

regards,

Drew


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk not picking up incoming calls from TDM400P

[Drew Gibson]

It's the thought that counts! :-)


Tim Nelson wrote:
> You are correct... my mistake. :-/
>
> Tim Nelson
> Systems/Network Support
> Rockbochs Inc.
> (218)727-4332 x105
>
> - Original Message -
> From: "Drew Gibson" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Friday, June 6, 2008 1:36:13 PM GMT -06:00 US/Canada Central
> Subject: Re: [asterisk-users] Asterisk not picking up incoming calls from 
> TDM400P
>
> Nope, didn't help.
>
> Doesn't the context declaration come *before* the channel assignment in 
> zapata.conf?
> It's working that way in our main Asterisk server.
>
> regards,
>
> Drew
>
>
>
> Tim Nelson wrote:
>   
>> It looks like you may be missing a context declaration right after your 
>> "channel => 1" line. Try adding "context=incoming" right after that. 
>>
>> Tim Nelson
>> Systems/Network Support
>> Rockbochs Inc.
>> (218)727-4332 x105
>>
>> - Original Message -
>> From: "Drew Gibson" <[EMAIL PROTECTED]>
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>> 
>> Sent: Friday, June 6, 2008 12:37:47 PM GMT -06:00 US/Canada Central
>> Subject: [asterisk-users] Asterisk not picking up incoming calls from TDM400P
>>
>> Hi,
>>
>> I am having some issues with a new server install in Singapore.
>>
>> Outbound calls work fine.
>>
>> Inbound calls are not picked up by Asterisk.
>>
>> Zaptel 1.2.25 and Asterisk 1.2.28 both built from source.
>> libpri installed
>>
>> wctdm and zaptel load without error
>>
>> Jun  6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface 
>> Registered on major 196
>> Jun  6 23:34:03 fs01 kernel: [211138.372937] Zaptel Version: 1.2.25
>> Jun  6 23:34:03 fs01 kernel: [211138.372943] Zaptel Echo Canceller: KB1
>> Jun  6 23:34:03 fs01 kernel: [211138.383639] Freshmaker version: 73
>> Jun  6 23:34:03 fs01 kernel: [211138.384053] Freshmaker passed register test
>> Jun  6 23:34:04 fs01 kernel: [211139.076180] Module 0: Installed -- AUTO 
>> FXO (FCC mode)
>> Jun  6 23:34:04 fs01 kernel: [211139.275847] Module 1: Installed -- AUTO 
>> FXO (FCC mode)
>> Jun  6 23:34:04 fs01 kernel: [211139.475514] Module 2: Installed -- AUTO 
>> FXO (FCC mode)
>> Jun  6 23:34:05 fs01 kernel: [211139.675182] Module 3: Installed -- AUTO 
>> FXO (FCC mode)
>> Jun  6 23:34:05 fs01 kernel: [211139.682518] Found a Wildcard TDM: 
>> Wildcard TDM400P REV I (4 modules)
>> Jun  6 23:34:08 fs01 kernel: [211142.686305] Registered tone zone 18 
>> (Singapore)
>> Jun  6 23:34:14 fs01 kernel: [211149.412565] -- Setting echo registers:
>> Jun  6 23:34:14 fs01 kernel: [211149.412990] -- Set echo registers 
>> successfully
>> Jun  6 23:34:14 fs01 kernel: [211149.413005] -- Setting echo registers:
>> Jun  6 23:34:14 fs01 kernel: [211149.413426] -- Set echo registers 
>> successfully
>> Jun  6 23:34:14 fs01 kernel: [211149.413435] -- Setting echo registers:
>> Jun  6 23:34:14 fs01 kernel: [211149.413848] -- Set echo registers 
>> successfully
>> Jun  6 23:34:14 fs01 kernel: [211149.413861] -- Setting echo registers:
>> Jun  6 23:34:14 fs01 kernel: [211149.414276] -- Set echo registers 
>> successfully
>> Jun  6 23:34:28 fs01 kernel: [211163.107241] Zaptel Transcoder support 
>> loaded
>>
>>
>> The one and only POTS line has been tuned with "fxotune and "fxotune -s" 
>> has been run.
>> ztmonitor shows incoming ring (volume peaks at 3-4000)
>>
>> No, nothing, nadda, zero response from Asterisk.
>>
>> Can anyone suggest a tool to help find the gap between zaptel and Asterisk?
>>
>> regards,
>>
>> Drew
>>
>>
>> zaptel.conf:-
>>
>> fxsks=1-4
>> ;(have tried fxsls=1-4 but no difference)
>> loadzone=sg
>> defaultzone=sg
>>
>>
>> zapata.conf:-
>>
>> [channels]
>> group=1
>> context=incoming
>> callprogress=no
>> rxgain=2.0
>> txgain=0.0
>> immediate=no
>> usecallerid=yes
>> callerid=asreceived
>> signalling=fxs_ks
>> relaxdtmf=yes
>> pickupgroup=1
>> faxdetect=incoming
>>
>> channel => 1
>>
>>
>> indications.conf:-
>>
>> [general]
>> country=sg
>>
>> < Other countries snipped --->
>>
>> [sg]
>>
>> ; Singapore section borrowed from 
>> http://csusap.csu.edu.au/

Re: [asterisk-users] Asterisk not picking up incoming calls from TDM400P

[Drew Gibson]

Nope, didn't help.

Doesn't the context declaration come *before* the channel assignment in 
zapata.conf?
It's working that way in our main Asterisk server.

regards,

Drew



Tim Nelson wrote:
> It looks like you may be missing a context declaration right after your 
> "channel => 1" line. Try adding "context=incoming" right after that. 
>
> Tim Nelson
> Systems/Network Support
> Rockbochs Inc.
> (218)727-4332 x105
>
> - Original Message -
> From: "Drew Gibson" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Friday, June 6, 2008 12:37:47 PM GMT -06:00 US/Canada Central
> Subject: [asterisk-users] Asterisk not picking up incoming calls from TDM400P
>
> Hi,
>
> I am having some issues with a new server install in Singapore.
>
> Outbound calls work fine.
>
> Inbound calls are not picked up by Asterisk.
>
> Zaptel 1.2.25 and Asterisk 1.2.28 both built from source.
> libpri installed
>
> wctdm and zaptel load without error
>
> Jun  6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface 
> Registered on major 196
> Jun  6 23:34:03 fs01 kernel: [211138.372937] Zaptel Version: 1.2.25
> Jun  6 23:34:03 fs01 kernel: [211138.372943] Zaptel Echo Canceller: KB1
> Jun  6 23:34:03 fs01 kernel: [211138.383639] Freshmaker version: 73
> Jun  6 23:34:03 fs01 kernel: [211138.384053] Freshmaker passed register test
> Jun  6 23:34:04 fs01 kernel: [211139.076180] Module 0: Installed -- AUTO 
> FXO (FCC mode)
> Jun  6 23:34:04 fs01 kernel: [211139.275847] Module 1: Installed -- AUTO 
> FXO (FCC mode)
> Jun  6 23:34:04 fs01 kernel: [211139.475514] Module 2: Installed -- AUTO 
> FXO (FCC mode)
> Jun  6 23:34:05 fs01 kernel: [211139.675182] Module 3: Installed -- AUTO 
> FXO (FCC mode)
> Jun  6 23:34:05 fs01 kernel: [211139.682518] Found a Wildcard TDM: 
> Wildcard TDM400P REV I (4 modules)
> Jun  6 23:34:08 fs01 kernel: [211142.686305] Registered tone zone 18 
> (Singapore)
> Jun  6 23:34:14 fs01 kernel: [211149.412565] -- Setting echo registers:
> Jun  6 23:34:14 fs01 kernel: [211149.412990] -- Set echo registers 
> successfully
> Jun  6 23:34:14 fs01 kernel: [211149.413005] -- Setting echo registers:
> Jun  6 23:34:14 fs01 kernel: [211149.413426] -- Set echo registers 
> successfully
> Jun  6 23:34:14 fs01 kernel: [211149.413435] -- Setting echo registers:
> Jun  6 23:34:14 fs01 kernel: [211149.413848] -- Set echo registers 
> successfully
> Jun  6 23:34:14 fs01 kernel: [211149.413861] -- Setting echo registers:
> Jun  6 23:34:14 fs01 kernel: [211149.414276] -- Set echo registers 
> successfully
> Jun  6 23:34:28 fs01 kernel: [211163.107241] Zaptel Transcoder support 
> loaded
>
>
> The one and only POTS line has been tuned with "fxotune and "fxotune -s" 
> has been run.
> ztmonitor shows incoming ring (volume peaks at 3-4000)
>
> No, nothing, nadda, zero response from Asterisk.
>
> Can anyone suggest a tool to help find the gap between zaptel and Asterisk?
>
> regards,
>
> Drew
>
>
> zaptel.conf:-
>
> fxsks=1-4
> ;(have tried fxsls=1-4 but no difference)
> loadzone=sg
> defaultzone=sg
>
>
> zapata.conf:-
>
> [channels]
> group=1
> context=incoming
> callprogress=no
> rxgain=2.0
> txgain=0.0
> immediate=no
> usecallerid=yes
> callerid=asreceived
> signalling=fxs_ks
> relaxdtmf=yes
> pickupgroup=1
> faxdetect=incoming
>
> channel => 1
>
>
> indications.conf:-
>
> [general]
> country=sg
>
> < Other countries snipped --->
>
> [sg]
>
> ; Singapore section borrowed from 
> http://csusap.csu.edu.au/~whaase01/itc308/asterisk/indications.conf
> description = Singapore
> ; Singapore
> ; Reference: 
> http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf
> ; Frequency specs are:   425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation 
> depth 100%; SIT +/- 50Hz
> ringcadence = 400,200,400,2000
> dial= 425
> ring= 425*24/400,0/200,425*24/400,0/2000 ; modulation should 
> be 100%, not 90%
> busy= 425/750,0/750
> congestion  = 425/250,0/250
> callwaiting = 425*24/300,0/200,425*24/300,0/3200
> stutter = 
> !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425
> info= 950/330,1400/330,1800/330,0/1000   ; not currently in 
> use acc. to reference
> dialrecall  = 425*24/500,0/500,425/500,0/2500; unspecified in 
> IDA reference, use repeating Holding Tone A,B
> record  = 1400/500,0/15000   ; unspecified in 
> IDA reference, use 0.5s tone e

[asterisk-users] Asterisk not picking up incoming calls from TDM400P

[Drew Gibson]

Hi,

I am having some issues with a new server install in Singapore.

Outbound calls work fine.

Inbound calls are not picked up by Asterisk.

Zaptel 1.2.25 and Asterisk 1.2.28 both built from source.
libpri installed

wctdm and zaptel load without error

Jun  6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface 
Registered on major 196
Jun  6 23:34:03 fs01 kernel: [211138.372937] Zaptel Version: 1.2.25
Jun  6 23:34:03 fs01 kernel: [211138.372943] Zaptel Echo Canceller: KB1
Jun  6 23:34:03 fs01 kernel: [211138.383639] Freshmaker version: 73
Jun  6 23:34:03 fs01 kernel: [211138.384053] Freshmaker passed register test
Jun  6 23:34:04 fs01 kernel: [211139.076180] Module 0: Installed -- AUTO 
FXO (FCC mode)
Jun  6 23:34:04 fs01 kernel: [211139.275847] Module 1: Installed -- AUTO 
FXO (FCC mode)
Jun  6 23:34:04 fs01 kernel: [211139.475514] Module 2: Installed -- AUTO 
FXO (FCC mode)
Jun  6 23:34:05 fs01 kernel: [211139.675182] Module 3: Installed -- AUTO 
FXO (FCC mode)
Jun  6 23:34:05 fs01 kernel: [211139.682518] Found a Wildcard TDM: 
Wildcard TDM400P REV I (4 modules)
Jun  6 23:34:08 fs01 kernel: [211142.686305] Registered tone zone 18 
(Singapore)
Jun  6 23:34:14 fs01 kernel: [211149.412565] -- Setting echo registers:
Jun  6 23:34:14 fs01 kernel: [211149.412990] -- Set echo registers 
successfully
Jun  6 23:34:14 fs01 kernel: [211149.413005] -- Setting echo registers:
Jun  6 23:34:14 fs01 kernel: [211149.413426] -- Set echo registers 
successfully
Jun  6 23:34:14 fs01 kernel: [211149.413435] -- Setting echo registers:
Jun  6 23:34:14 fs01 kernel: [211149.413848] -- Set echo registers 
successfully
Jun  6 23:34:14 fs01 kernel: [211149.413861] -- Setting echo registers:
Jun  6 23:34:14 fs01 kernel: [211149.414276] -- Set echo registers 
successfully
Jun  6 23:34:28 fs01 kernel: [211163.107241] Zaptel Transcoder support 
loaded


The one and only POTS line has been tuned with "fxotune and "fxotune -s" 
has been run.
ztmonitor shows incoming ring (volume peaks at 3-4000)

No, nothing, nadda, zero response from Asterisk.

Can anyone suggest a tool to help find the gap between zaptel and Asterisk?

regards,

Drew


zaptel.conf:-

fxsks=1-4
;(have tried fxsls=1-4 but no difference)
loadzone=sg
defaultzone=sg


zapata.conf:-

[channels]
group=1
context=incoming
callprogress=no
rxgain=2.0
txgain=0.0
immediate=no
usecallerid=yes
callerid=asreceived
signalling=fxs_ks
relaxdtmf=yes
pickupgroup=1
faxdetect=incoming

channel => 1


indications.conf:-

[general]
country=sg

< Other countries snipped --->

[sg]

; Singapore section borrowed from 
http://csusap.csu.edu.au/~whaase01/itc308/asterisk/indications.conf
description = Singapore
; Singapore
; Reference: 
http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf
; Frequency specs are:   425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation 
depth 100%; SIT +/- 50Hz
ringcadence = 400,200,400,2000
dial= 425
ring= 425*24/400,0/200,425*24/400,0/2000 ; modulation should 
be 100%, not 90%
busy= 425/750,0/750
congestion  = 425/250,0/250
callwaiting = 425*24/300,0/200,425*24/300,0/3200
stutter = 
!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425
info= 950/330,1400/330,1800/330,0/1000   ; not currently in 
use acc. to reference
dialrecall  = 425*24/500,0/500,425/500,0/2500; unspecified in 
IDA reference, use repeating Holding Tone A,B
record  = 1400/500,0/15000   ; unspecified in 
IDA reference, use 0.5s tone every 15s
; additionally defined in reference
nutone  = 425/2500,0/500
intrusion   = 425/250,0/2000
warning = 425/624,0/4376 ; end of period 
tone, warning
acceptance  = 425/125,0/125
holdinga= !425*24/500,!0/500 ; followed by holdingb
holdingb= !425/500,!0/2500


[incoming]
; Added Answer statement for troubleshooting
exten => s,1,Answer()

include => office-incoming
include => internal

[office-incoming]
; OANDA Office incoming calls
ignorepat => 9
exten => s,1,Wait,1 ; Waiting a little longer for CID
exten => s,n,Answer ; Answer the line
exten => s,n,Set(TIMEOUT(digit)=2)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Set(TIMEOUT(absolute)=14400)
;exten => s,n,Goto(ivr_menu)

<- rest of extensions.conf snipped  ->

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] fxotune question

[Drew Gibson]

Tilghman Lesher wrote:
> On Thursday 05 June 2008 09:50:05 Eric "ManxPower" Wieling wrote:
>   
>> Echo Canceler Freak Out, this happens when the rxgain is too high and
>> the echo canceler freaks out.  Some users describe it as "screeching",
>> "feedback", "static", or other useless terms.  If users report "static"
>> on a system where there cannot be static (all digital, PRI, SIP, etc),
>> you might be experiencing ECFO.
>>     
>
> Considering he's using an analog card, that last statement does not apply.
>
>   
>> Drew Gibson wrote:
>> 
>>> Tilghman Lesher wrote:
>>>   
>>>> On Wednesday 04 June 2008 22:02:19 John Morey wrote:
>>>> 
>>>>> Hello,
>>>>>
>>>>> I've run fxotune at different times but continue to get what seem to be
>>>>> strange numbers in /etc/fxotune.conf.  It ends up with:
>>>>>
>>>>> 5=7,255,251,251,2,255,255,1,255
>>>>> 6=7,255,251,251,2,255,255,1,255
>>>>> 7=7,255,251,251,2,255,255,1,255
>>>>> 8=9,2,250,253,4,252,0,255,255
>>>>> 9=4,0,0,0,0,0,0,0,0
>>>>> 10=5,0,0,0,0,0,0,0,0
>>>>> 11=0,0,0,0,0,0,0,0,0
>>>>> 12=0,0,0,0,0,0,0,0,0
>>>>> ports 5-10 have lines hooked up to them.  The first four lines seem
>>>>> strange when compaired to what others have posted and what ports 9 and
>>>>> 10 have.
>>>>>
>>>>> Also if I'm reading things right my echo ratios seem to be very
>>>>> high.  Running "fxotune -d -b 5 -w 1004" gives the following:
>>>>> Dumping module /dev/zap/5
>>>>> echo ratio = 0.1759 (1960.0 / 11145.0)
>>>>> Which I read to be over 17%.  This seems crazy.  Am I reading this
>>>>> right? Where should I start to look for problems?
>>>>>   
>>>> You might check to see if the tip and ring are reversed in your wiring. 
>>>> That can frequently cause weird echo problems.
>>>> 
>>> Which ports would you expect to be reversed? 5-8 or 9-10?
>>>
>>> I have similar settings in my fxotune.conf for a TDM2400P and I'm
>>> getting complaints of "static" on the line that I suspect are related to
>>> an overtaxed h/w echo canceller
>>>
>>> My fxotune.conf:-
>>>
>>> 13=7,255,251,251,2,255,255,1,255
>>> 14=9,254,251,255,2,0,1,0,0
>>> 15=9,254,251,255,2,0,1,0,0
>>> 16=5,0,0,0,0,0,0,0,0
>>> 17=9,254,251,255,2,0,1,0,0
>>> 18=9,254,251,255,2,0,1,0,0
>>> 19=9,254,251,255,2,0,1,0,0
>>> 20=9,254,251,255,2,0,1,0,0
>>> 21=9,254,251,255,2,0,1,0,0
>>> 22=9,254,251,255,2,0,1,0,0
>>> 23=9,254,251,255,2,0,1,0,0
>>> 24=9,254,251,255,2,0,1,0,0
>>>
>>> regards,
>>>
>>> Drew
>>>   
>
>   

Thanks Tilghman,

I will try flipping one of the lines at the weekend, can't touch it 
during the week.

Our rxgain was raised from 0.0 to 2.0 at the end of February after 
complaints that CC staff couldn't hear customers but the 'static' issues 
pre-date that change.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] fxotune question

[Drew Gibson]

Tilghman Lesher wrote:
> On Wednesday 04 June 2008 22:02:19 John Morey wrote:
>   
>> Hello,
>>
>> I've run fxotune at different times but continue to get what seem to be
>> strange numbers in /etc/fxotune.conf.  It ends up with:
>>
>> 5=7,255,251,251,2,255,255,1,255
>> 6=7,255,251,251,2,255,255,1,255
>> 7=7,255,251,251,2,255,255,1,255
>> 8=9,2,250,253,4,252,0,255,255
>> 9=4,0,0,0,0,0,0,0,0
>> 10=5,0,0,0,0,0,0,0,0
>> 11=0,0,0,0,0,0,0,0,0
>> 12=0,0,0,0,0,0,0,0,0
>> ports 5-10 have lines hooked up to them.  The first four lines seem strange
>> when compaired to what others have posted and what ports 9 and 10 have.
>>
>> Also if I'm reading things right my echo ratios seem to be very
>> high.  Running "fxotune -d -b 5 -w 1004" gives the following:
>> Dumping module /dev/zap/5
>> echo ratio = 0.1759 (1960.0 / 11145.0)
>> Which I read to be over 17%.  This seems crazy.  Am I reading this right?
>> Where should I start to look for problems?
>> 
>
> You might check to see if the tip and ring are reversed in your wiring.  That
> can frequently cause weird echo problems.
>
>   

Which ports would you expect to be reversed? 5-8 or 9-10?

I have similar settings in my fxotune.conf for a TDM2400P and I'm 
getting complaints of "static" on the line that I suspect are related to 
an overtaxed h/w echo canceller

My fxotune.conf:-

13=7,255,251,251,2,255,255,1,255
14=9,254,251,255,2,0,1,0,0
15=9,254,251,255,2,0,1,0,0
16=5,0,0,0,0,0,0,0,0
17=9,254,251,255,2,0,1,0,0
18=9,254,251,255,2,0,1,0,0
19=9,254,251,255,2,0,1,0,0
20=9,254,251,255,2,0,1,0,0
21=9,254,251,255,2,0,1,0,0
22=9,254,251,255,2,0,1,0,0
23=9,254,251,255,2,0,1,0,0
24=9,254,251,255,2,0,1,0,0

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Best Linux distribution to use in Asterisk server

[Drew Gibson]

Tzafrir Cohen wrote:
> On Fri, May 09, 2008 at 11:50:59AM -0400, Drew Gibson wrote:
>   
>> equis software wrote:
>> 
>>> Hi, I allways use Gentoo y my Asterisk servers and work well, but what 
>>> do you think about to use Ubuntu or another distibution??
>>>
>>> Thanks
>>>   
>> I have run Asterisk on several Fedora versions, Debian, Unslung (on the 
>> NSLU2 or "Slug") and recently Ubuntu. My most critical servers are on 
>> Debian and new ones will be on Ubuntu LTS.
>>
>> I have had very few OS specific issues. I have always built from source 
>> except on the Slug but I noticed that Ubuntu has it in the apt 
>> repository which would be a great convenience if you are new to Linux or 
>> manage a lot of servers.
>> 
>
> Now that you mention it, are packages of that distribution really
> maintained?
>
> The recent volnurability of AST-2008-006 is a good test case for that.
> If affects both 1.2 and 1.4 .
>
> The annoncement by Digium:
>
>   http://downloads.digium.com/pub/asa/AST-2008-006.html
>
> As with the previous ones, the text is quite clear about the fix.
> "backporting" that patch to a slightly older version is not that tricky
> (and it is something that a distribution package maintainer is used to
> doing anyway).
>
> So what about updates?
>
>   

The optware feed for Unslung (on the NSLU2) is up-to-date on Asterisk 
1.4 with 1.4.19.1 but a rev or two behind on 1.2 with 1.2.24

> The LWN page for this advisory only lists Fedora and Debian:
>
>   http://lwn.net/Articles/280318/
>
> Response ime in both was quite reasonable.
>
> LWN also tracks adsisories from various other distributions. You can see
> the list in http://lwn.net/Alerts/ . The following other distributions
> have 'asterisk' packages:
>
>  * Gentoo
>  * Mandriva (??? - probably only in contrib and is unsupported)
>  * rPath (Not sure. See below about AstriskNOW)
>  * SUSE
>  * Ubuntu (the package is in 'universe', and not officially supported)
>
> The issue is listed as corrected in AsteriskNOW 1.0.3, but the latest
> version available for download is 1.0.2 .
> If I read rpath's repository page correctly, then the most recently
> released version of Asterisk is 1.4.17-2 , from Feb-2008 and thus does
> not contain this fix.
>
> To see the versions of packages i Ubuntu:
>
>   http://packages.ubuntu.com/asterisk
>
> As you can see, both Hardy and the development distribution (Interpid)
> include the same version of the package. As you can see from following
> the changelog link:
> http://changelogs.ubuntu.com/changelogs/pool/universe/a/asterisk/asterisk_1.4.17~dfsg-2ubuntu1/changelog
>
> The security issues of 1.4.18.1 were backported to that 1.4.17 package.
> But nothing about the recent advisory.
>
>
> The Gentoo port is basically where the Ubuntu package is: missing only
> the last one:
>
>   http://packages.gentoo.org/package/asterisk
>
>
> The FreeBSD port has not been updated yet. It is still at 1.4.18, and no
> sign of backported fixes:
>
>   http://www.freebsd.org/cgi/cvsweb.cgi/ports/net/asterisk/
>
>
> OpenBSD port was updated pretty fast (by upgrading to asterisk 1.4.19.1)
>
>   http://www.openbsd.org/cgi-bin/cvsweb/ports/telephony/asterisk/
>
>
> I don't know where to look for in other distributions.
>
>   


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Best Linux distribution to use in Asterisk server

[Drew Gibson]

equis software wrote:
> Hi, I allways use Gentoo y my Asterisk servers and work well, but what 
> do you think about to use Ubuntu or another distibution??
>
> Thanks

I have run Asterisk on several Fedora versions, Debian, Unslung (on the 
NSLU2 or "Slug") and recently Ubuntu. My most critical servers are on 
Debian and new ones will be on Ubuntu LTS.

I have had very few OS specific issues. I have always built from source 
except on the Slug but I noticed that Ubuntu has it in the apt 
repository which would be a great convenience if you are new to Linux or 
manage a lot of servers.

For a newbie, I would recommend an Ubuntu LTS release.

Pick the distro you are most comfortable with.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Zap Channels Collide (Incoming & Outgoing)

[Drew Gibson]

Eric Wieling wrote:
> Drew Gibson wrote:
>   
>> Eric Wieling wrote:
>>     
>>> Drew Gibson wrote:
>>>   
>>>   
>>>> I think the scary thing is that, for most people, basic knowledge of 
>>>> telephony was almost impossible to come by outside the opaque and 
>>>> secretive world of telco.
>>>>
>>>> That is until Asterisk came along!
>>>>
>>>> Perhaps there should be a regulatory requirement to read The Future of 
>>>> Telephony, cover to cover, before installing any Asterisk system! :-)
>>>>
>>>> http://www.asteriskdocs.org/
>>>> 
>>>> 
>>> People that try to "wing it" and install Asterisk when they don't know 
>>> telecom just gives people a bad impression of Asterisk and VoIP in 
>>> general.  This helps nobody except the pocketbook of the consultant.
>>>
>>>   
>>>   
>> but how else do they learn?
>>
>> 
>
> Books are one of the best resources, the Wiki is not *too* bad when it 
> comes to general telecom stuff.  You can also build prototype systems.
>   

That's why http://www.asteriskdocs.org/
Is there a better place to start?

> No, Asterisk did not suddenly unleash the gates of knowledge in telecom. 
>   All that information was available before Asterisk.  What was not 
> available was info on the specific inner workings of traditional PBXs.
>
> Asterisk and Digium did reduce the hardware cost of building a PBX.
>   

The knowledge may have been there but there was no "index". It was like 
saying to a Unix newbie "just use the man pages, it's all in there". But 
newbies don't know which command to look up!! (there is no index)

Asterisk reduced the cost of entry to the market by a several orders of 
magnitude.
This, in turn, allowed more folks to become technically "telephony 
literate".
Which, in turn again, made the knowledge exponentially easier to come by.

My start in telephony was thanks to 3Com and their efforts to promote 
VoIP in the mid to late 90's but it is from the Asterisk community that 
I've learned the most.

> Traidional telecom is actually fairly simple if you compare it with IP 
> PSTN/IP PBXs.  With an IP PBX like Asterisk you need to understand 
> telecom, IP networking (including routing, NAT, ports), Linux, as well 
> as Asterisk itself.
>
>   
If it helps, I feel the same contempt for the telephony guys trying to 
learn IP networking. But I try to remember that I have as much to learn 
in their field as they do in mine. (Afraid I'm not much good at humility 
either)  

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Zap Channels Collide (Incoming & Outgoing)

[Drew Gibson]

Eric Wieling wrote:
> Drew Gibson wrote:
>   
>> I think the scary thing is that, for most people, basic knowledge of 
>> telephony was almost impossible to come by outside the opaque and 
>> secretive world of telco.
>>
>> That is until Asterisk came along!
>>
>> Perhaps there should be a regulatory requirement to read The Future of 
>> Telephony, cover to cover, before installing any Asterisk system! :-)
>>
>> http://www.asteriskdocs.org/
>> 
>
> People that try to "wing it" and install Asterisk when they don't know 
> telecom just gives people a bad impression of Asterisk and VoIP in 
> general.  This helps nobody except the pocketbook of the consultant.
>
>   

but how else do they learn?

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Zap Channels Collide (Incoming & Outgoing)

[Drew Gibson]

I think the scary thing is that, for most people, basic knowledge of 
telephony was almost impossible to come by outside the opaque and 
secretive world of telco.

That is until Asterisk came along!

Perhaps there should be a regulatory requirement to read The Future of 
Telephony, cover to cover, before installing any Asterisk system! :-)

http://www.asteriskdocs.org/

regards,

Drew


Al Baker wrote:
> I know that everyone has gaps in their knowledge, but I am just 
> staggered that
> systems are being sold/deployed with such fundamental TELCO workings not 
> being
> understood.  Frightening.
>
> C. Chad Wallace wrote:
>   
>> At 5:22 PM on 08 May 2008, Forrest Beck wrote:
>>
>>   
>> 
>>> I have a client that is using the Sangoma A200DE with two phone
>>> lines attached.
>>>
>>> The problem is:
>>>
>>> They use their phone (Grandstream GXP2020) to dial out of the system.
>>> Instead of getting ringing, there is someone on the other end of the  
>>> line that happened to dial in at the exact same moment.
>>>
>>> So now they are stuck talking with this person, instead of the one
>>> the originally called.
>>>
>>> The ZAP channels are in a dial plan context that instructs it to
>>> just dial the office phones.
>>>
>>> [zap1]
>>> exten => s,1,Dial(SIP/1001&SIP/1002&SIP/1003)
>>> exten => s,n,Voicemail([EMAIL PROTECTED])
>>>
>>> Anyone know how to get around this?
>>> 
>>>   
>> This is known in the telephony world as "glare", and there's not much
>> you can do about it, especially if you only have one line.
>>
>> If you have multiple lines on an over-ring (or hunt group or whatever
>> you call it), the best thing to do is find out which way the telco
>> assigns calls to those lines wrt how they are assigned to the Asterisk
>> box.  And then allocate outgoing calls in the other direction.  
>>
>> On our installation, the calls are allocated from the first FXO port
>> (Zap/25) up.  So we set Asterisk to dial out starting from the last FXO
>> port in the group by calling Dial(Zap/G2) (capital G means dial down
>> from last, lowercase g means dial up from first).  That minimizes glare.
>>
>> But, as I said before, if you only have one line, you can't do that...
>>
>>   
>> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] VOICEMAIL OPTIONS help needed

[Drew Gibson]

Steve Johnson wrote:
> Hi everyone,
>
> We have a particular user on our Asterisk 1.4.x system who always
> listens to his voicemail messages via email.
>
> - Is there some way to send the voicemail ONLY to email and not retain
> them on the phone?
>
> - Alternatively, can the voicemail system only keep, say, just the
> last 10 messages (as backup in case of email delivery failure or a
> message getting deleted in email accidentally before it is heard),
> purging out the oldest when a new one is received?
> (If we set the option maxmsg=10 on his mailbox in voicemail.conf, I
> think it will stop accepting voicemails after 10 messages, not turf
> the oldest one and accept a new one in its place).
>
> Everyone else uses the normal voicemail options on their phones, so
> the solution should be just for this single user.
>
>
> Thanks for any suggestions.
>
> S.
>   

from voicemail.conf (1.2.24):-

; delete=yes; After notification, the voicemail is deleted 
from the server. [per-mailbox only]

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] New generic sounds

[Drew Gibson]

Tilghman Lesher wrote:
> We're about to do another batch of sounds, and I see by my word count that we
> have some extra time left over.  So, suggestions will be entertained for
> additional prompts in English, Spanish, or French.  The only rules are: 1) the
> prompts have to be generic to Asterisk.  No "Welcome to so-and-so's business"
> unless the business is fake and the prompt is funny.  2) The prompt may not be
> profane.  Our professional speakers do have a sense of humor, but there are
> some things they just will not say.
>
> I'll open it to the floor now, with the caveat that since Digium is paying for
> the recording session, it maintains final editorial approval over which sounds
> are selected.
>
>   

For those providers that want to show they are a SERIOUS phone company.

"We don't care. We don't have to. (snort) We're the Phone Company!"

"No, no, no, you're dealing with the telephone company."
"We are not subject to city, state, or federal legislation."
"We are omnipotent."

Sorry?... What was that Ernestine?

Oh, it seems they've already been done!

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Simple queue announcements

[Drew Gibson]

Doug Lytle wrote:
> Chris Bagnall wrote:
>   
>> Greetings list,
>>
>> I've been playing around with queues on an old asterisk 1.2 box at a 
>> customer's site. They want to be able to add really simple queue 
>> announcements every minute, along the following lines:
>> "sorry for the delay, someone will be with you shortly".
>>   
>> 
>
>
> I believe you'll need to migrate them to 1.4.
>
> Doug
>
>   

Works fine for me on 1.2.24 ...

from queues.conf:-

periodic-announce = Custom/periodic-fxqueue01
periodic-announce-frequency=90

plays a brief "sorry,etc, press any key to leave a message or continue 
to hold for next available etc, etc"

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] question about queue

[Drew Gibson]

BJ Weschke wrote:
> Rilawich Ango wrote:
>   
>> Thanks.  I have checked that the queue.conf.  I keep the default
>> setting as autofill=yes in my tests.  That's mean even autofill=yes,
>> the 1st caller will still stick the whole queue.
>> asterisk version : 1.4.18
>>
>> --queue.conf--
>> ; AutoFill Behavior
>> ;The old/current behavior of the queue has a serial type behavior
>> ;in that the queue will make all waiting callers wait in the queue
>> ;even if there is more than one available member ready to take
>> ;calls until the head caller is connected with the member they
>> ;were trying to get to. The next waiting caller in line then
>> ;becomes the head caller, and they are then connected with the
>> ;next available member and all available members and waiting callers
>> ;waits while this happens. The new behavior, enabled by setting
>> ;autofill=yes makes sure that when the waiting callers are connecting
>> ;with available members in a parallel fashion until there are
>> ;no more available members or no more waiting callers. This is
>> ;probably more along the lines of how a queue should work and
>> ;in most cases, you will want to enable this behavior. If you
>> ;do not specify or comment out this option, it will default to no
>> ;to keep backward compatibility with the old behavior.
>> ;
>> autofill = yes
>>
>>   
>> 
>  This was something I put in a long while back on 1.2 branch because we 
> really needed it for 1.2 to "bug fix" the behavior, but also needed to 
> prevent the change in behavior for those that didn't want it to change. 
>
>   

Is this option active in 1.2.24? I thought it was only in 1.4
It's not mentioned in the queues.conf.sample.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] queue logging

[Drew Gibson]

Flash Operator Panel http://www.asternic.org/

regards,

Drew



Arjan Kroon | Mobillion wrote:
>
> Hi,
>
> I’m not looking for a programma that show the queue logging.
>
> But is there a way to check during a call, which member is connected 
> to the caller.
>
> Kind Regard,
>
> Arjan Kroon
>
> * From: * [EMAIL PROTECTED] [mailto: 
> [EMAIL PROTECTED] ] *On Behalf Of *Scott Wolfe
> *Sent:* woensdag 9 april 2008 17:19
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] queue logging
>
> You could ASTassistant to see this. Its Freeware.
>
> www.astassistant.com <http://www.astassistant.com>
>
> - Original Message -
>
> * From: * Arjan Kroon | Mobillion <mailto:[EMAIL PROTECTED]>
>
> * To: * Asterisk Users Mailing List - Non-Commercial Discussion
> <mailto:asterisk-users@lists.digium.com>
>
> * Sent: * Wednesday, April 09, 2008 1:01 AM
>
> * Subject: * [asterisk-users] queue logging
>
> Hi,
>
> I’ using with asterisk a queue with tree members and round robin.
>
> When a caller enters this queue and it is connecting to one of the
> members, is there a possibility to see which member the caller is
> connected to?
>
> And is there a way to see in de application to see if the
> connection from the caller to one of the members was successful of
> not successful?
>
> I know you can see it in de queue. log.
>
> But I want to know if I can see it also in the hangup (h) in de
> application?
>
> Kind Regards
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] RTCP not being sent when on hold

[Drew Gibson]

Adrian A wrote:
> Hello,
>
> When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 
> <http://1.4.18.1> and I place the call on hold, the call is dropped 
> after 30 seconds.
> It looks like there is no RTCP/RTP sent to the client from Asterisk 
> while on hold (music on hold playing to caller) thus client 
> disconnects the call. During this time, I get the following messages 
> in the CLI:
>
> NOTICE[24194] rtp.c: Unknown RTP codec 126 received from '0.0.0.0 
> <http://0.0.0.0>'
>
> In sip.conf I have rtpkeepalive=15 but that does not seem to help.
>
> Does anyone know what I can do to fix this, other than increase the 
> timeout on Bria?
>
> Thanks,
> Adrian


Is it not up to the phone to send the keep-alive packets?

Sounds like Asterisk does not understand the keep-alive packets coming 
from the phone. Try setting "rtptimeout=300" in sip.conf to test this. 
It should now hangup after 5 minutes.

regards,

Drew

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] FW: [newtech-1] Skype 24 Hour Rolling Analytics

[Drew Gibson]

Yes Dean,

we do use Asterisk at OANDA. We've been running our office on it for 2 
years now and the call centre's 1st Asterisk anniversary was April 1!

Glad you like our site, we've just launched FXGame Mobile 
(http://fxlabs.oanda.com) for the gadget lover and FXGlobal Transfer for 
ex-pat Aussies to send their money home. :-)

Perhaps I should have added a smiley for my fellow North Americans but I 
think my analysis is valid. Perhaps the interesting points is that % of 
Skype users seems to be fairly uniform across cultures around the world  
although perhaps a little higher in Europe (a.m. usage while North 
Americans are still sleeping). I suspect that this is due to the call 
billing structure in Europe. They make the North American telcos look 
positively philanthropic.

regards,

Drew


Dean Collins wrote:
> Wow Drew, I had no idea someone from Oanda was a subscriber to the
> Asterisk list (and therefore an asterisk user company?).
>
> Just wanted to say you guys run a fantastic sight and I've been a long
> time user for at least the last 2 years.
>
>
>
>
> Now for the irony part of your email. I found it interesting with
> regards to global penetration and world population.
>
> I'd also be interested in seeing some comparisons where this goes
> against the trend.
>
> Eg would love to see a transaction per hour chart for Oanda on a global
> basis (though I don't know what your customer spread is).
>
> I've already sent this email to the creator of twitter vision to see if
> he can do something similar against tweets per hour on a rolling basis.
>
> I'd also love to see something truly global like SMS use which is pretty
> much a global application. Something like number of SMS's sent per hour
> across the various countries over a 24 hour period would be hugely
> interesting (well to me anyway).
>
>  
>
> Regards,
>
> Dean Collins
> Cognation Pty Ltd
> [EMAIL PROTECTED] 
> +1-212-203-4357
> +61-2-9016-5642 (Sydney in-dial). 
>
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Drew Gibson
>> Sent: Thursday, 3 April 2008 9:38 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] FW: [newtech-1] Skype 24 Hour Rolling
>> 
> Analytics
>   
>> Dean Collins wrote:
>> 
>>>   
> http://deancollinsblog.blogspot.com/2008/04/skype-24-hour-rolling-analyt
> ics.html
>   
>>> <http://bp3.blogger.com/_jmYevHrBr6M/R_Pq-
>>>   
>> vgIjvI/Af0/PgE_8gFqrY8/s1600-
>> h/World%2Bpopulation%2Bawake.png>
>> 
>>> Totally stumbled across this really interesting post
>>> http://skypejournal.com/blog/2008/04/world_online_or_asleep.html
>>>
>>>   
>> So, it seems that people make more calls on Skype when they are not
>> sleeping and that people who don't have computers don't make so many
>> Skype calls, even when they are awake.
>>
>> Fascinating!
>>
>> regards,
>>
>> Drew
>> 

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] FW: [newtech-1] Skype 24 Hour Rolling Analytics

[Drew Gibson]

Dean Collins wrote:
>
>  
> http://deancollinsblog.blogspot.com/2008/04/skype-24-hour-rolling-analytics.html
>  
>
>
> <http://bp3.blogger.com/_jmYevHrBr6M/R_Pq-vgIjvI/Af0/PgE_8gFqrY8/s1600-h/World%2Bpopulation%2Bawake.png>
>  
>
>
> Totally stumbled across this really interesting post 
> http://skypejournal.com/blog/2008/04/world_online_or_asleep.html
>

So, it seems that people make more calls on Skype when they are not 
sleeping and that people who don't have computers don't make so many 
Skype calls, even when they are awake.

Fascinating!

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Had it with Dell Garbage - HP Question

[Drew Gibson]

My own "fear from hell" is having to call tech support!

My personal experience is that I get better support, faster, from 
mailing lists than from a "paid for" tech support. If it's a common 
failure, Google will get you there faster than you can dial 1-800. If 
it's more unusual, the "paid for " techie won't have it in his script 
and you will have to wait to be "escalated". If it actually requires 
some real troubleshooting skills and/or brain power, your mailing list 
buddies will be far more interested in helping you than an underpaid, 
under-challenged help line dweeb who just wishes it was Friday already.

The idea that "paid for" or "at least there's a neck to choke" support 
is better is just FUD.

BTW the original IBM phrase was Fear, Uncertainty and DEATH (not the 
P.C. "Doubt", which is redundant).

regards,

Drew


Al Baker wrote:
> Helps a bunch !!!
> One follow up question - out of all of your possible choices for the OS 
> how did you pick *Debian*.
> I 'm not saying is bad, I just know nothing about the particular disto. 
> and and very curious what
> it brought to the table that made you pick over say *RedHat* - where you 
> can *buy support *or *SUSE* - where you can *buy support*. My fear from 
> hell is that I' get 50 or 60 of these boxes in, start having kernel 
> panics, and have no damn body to help except the folks on mailing lists. 
> Mind you these are often really smart people, very generously giving of 
> their time, but not quite the say as a manned/paid support organization.
>
> Thx for sharing !!!
>
>
>
> Michiel van Baak wrote:
>   
>> On 08:02, Thu 27 Mar 08, Al Baker wrote:
>>   
>> 
>>> How do you get notifications ?
>>> Is this thru one of the add on packages HP sells for the box ?  Which One ?
>>> Could you be more specific about what you mean by a "recovery CD"
>>> and hod do you get console access below multi used to do "recovery ??
>>>
>>> What is "integrated ILO BIOS Access" sounds cool.
>>>
>>> What O/S you usin and what made you pick it ?
>>>
>>> What kind and how many RAIDS are you using. The HP site gave like 8 
>>> different RAID controllers and like 20 CPUs to chose from.  How did you 
>>> chose ?
>>>
>>> Thx for sharing !!!
>>> 
>>>   
>> I'm not the op, but sending a reply anyways.
>>
>> The notifications come from the HP tools you can download
>> for free from their website.
>>
>> The recovery cd is probably a selfmade installer for their
>> setup. At least that's what we have.
>>
>> the ILO stuff is to give you access to the box like you were
>> sitting right in front of it with a physical keyboard and
>> monitor, but over IP.
>> You can boot the machine, access the cd in your local
>> machine etc, even if the box is on the other side of the
>> moon.
>>
>> We use Debian. HP even supports it on their DL380 boxen.
>>
>> We use the P400 raid controller. Setup RAID5 with 3 disks.
>> CPU we use right now is the Intel E5405
>>
>> Hope this helps a bit.
>>
>>   
>> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] FYI about my Mona Vie business venture

[Drew Gibson]

Does it work with Asterisk 1.2?


BerkHolz, Steven wrote:
> Asterisk work does not pay all of my bills, so I have joined up with a 
> company that has a very good payment plan.
>
> I have recently become a Mona Vie Independent Distributor.
>
> I am not going to go into a sales pitch.
> This is just an FYI to this opportunity.
>
> The company has grown into a Billion dollar company in just 2 years.
> This company's compensation plan is the best and quickest that I have seen.
> My brother-in-law has only been in the business for a month and is already 
> making a profit.
>
> The first thing that I noticed when researching the opportunity, was that I 
> could find no negative statements about it.
>
> The product itself has many health benefits.
>
> So far:
> My knees no longer ache.
> We are both sleeping better. I literally do not stir once asleep.
> My restless leg syndrome has not been noticed.
> I seem to have more energy.
>
> The main ingredient is the acai berry.
>
> Here is a list of what it is supposed to do:
> Boosts energy levels
> Improves digestive function
> Improves mental clarity/focus
> Promotes sound sleep
> Provides all vital vitamins
> Contains several important minerals
> Is an extremely powerful free radical fighter
> Acai has very high levels of fibers
> Cleanses and Detoxifies the body of infectious toxins
> Strengthens your immune system
> Enhances sexual desire and performance
> Fights cancerous cells
> Slows down the aging process
> Promotes healthier and younger-looking skin
> Alleviates diabetes
> Normalizes and regulates cholesterol levels
> Helps maintain healthy heart function
> Minimizes inflammation
> Improves circulation
> Prevents artherosclerosis
> Enhances visual acuity
>
>
> The income can be made in two ways (actually more, but two primary ways)
> 1.  Reselling the product at a marked up price. This is something that I 
> have no interest in, and do not personally know anyone doing this.
> 2.  Team Commissions.
> a.  You make back 5 percent of the sales that occur below you in your 
> tree.
> b.  You only have to personally sign 2 people.  Other people above you 
> will be adding to your tree.
> c.  They call it a binary system, where you only have 2 people directly 
> under you, and any other people that you add go down to the bottom and 
> benefit others as well as yourself.
> d.  I already have two people underneath me and have not personally 
> signed anyone yet, so it is a quick growing tree, even for people that may 
> not be as motivated.
> e.  After a month, My brother-in-law has NO more out of pocket expenses 
> to stay in this system.  The money he is earning is paying for his Minimum 
> requirements. The rest is profit.
>
> To sign up to be a distributor , which is required to make money, is $54
> A case of Mona Vie is $120.
> A case will last 2 people a month. (you only take 2 ounces a day)
>
> This may seem like a lot, but:
> 1.  You will not need to buy any vitamins.
> 2.  My brother-in-law is already making $200 a month, after being in the 
> system for a month, So his cost for the Mona Vie is covered and he is making 
> $80 a month.
> 3.  As more people sign up, the amount he gets back will increase.
>
>
> Anyway, I am not intending this to be pushy or salesy, I just wanted to let 
> my associates, that may be looking for additional income,  know about this.
>
> Here is the Website, if you are interested in researching this:
> http://teamvie.blogspot.com/
> http://www.monavie.com
> Also, feel free to Google it.
>
> I am very excited with this, both in the health benefits I am already seeing, 
> and the income potential.
>
> Please feel free to let me know if this is something that you may be 
> interested in, and I can get you more information.
>
> Thank You,
> Steven B
> [EMAIL PROTECTED]
>
> Please visit us on the web at www.hirotecamerica.com
> HIROTEC AMERICA  Ph. 248-836-5100 Fx. 248-836-5101
>
> Please only print this email if it is necessary. Help spread environmental 
> awareness.
>
> This e-mail and any files transmitted with it are intended only for the 
> person or entity to which it is addressed and may contain confidential 
> material and/or material protected by law.  Any  retransmission or use of 
> this information may be a violation of that law.  If you received this in 
> error, please contact the sender and delete the material.
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.co

Re: [asterisk-users] Is Asterisk ready for Prime-Time?

[Drew Gibson]

The box has been up since we upgraded the UPS, time before was for the 
disk failure in Feb 2007.

Asterisk has now been up for 5 hours, 44 minutes (yes, by Murphy's Law, 
I'm troubleshooting a problem but"restart when convenient" does not 
impact real uptime) but yesterday it had been up for 63+ days (last 
restart was for queue config changes)

This is stock code on stock OS on stock hardware. We don't tweak it, 
poke at it, fiddle with it, update it unless necessary. We do OS and 
Asterisk updates on planned maintenance days infrequently)

KISS and don't fsck with it!

regards,

Drew

 

Mojo with Horan & Company, LLC wrote:
> An off-the-shelf 5+ year old MSI MS-6378X-L motherboard, 1.6GHz AMD, 512 
> RAM, 10 extensions, no more than three concurrent calls:
>
> [EMAIL PROTECTED] ~]$ uptime
>  11:31:45 up 103 days,  1:00,  2 users,  load average: 0.00, 0.00, 0.00
>
> But:
> [EMAIL PROTECTED] ~]$ sudo asterisk -rx 'core show uptime'
> System uptime: 9 hours, 32 minutes, 25 seconds
>
> I reboot every evening :)  Drew, what's the uptime on your asterisk 
> process on that box that's been up for 193 days?
>
> Drew Gibson wrote:
>   
>> Bill Andersen wrote:
>>   
>> 
>>> This is not a troll.  I've used my real email because I want this
>>> taken seriously.  I'm not trying to make anyone mad, I just want
>>> some real discussion on this issue.  Please bare with me...
>>>
>>> I'm a USER of Asterisk.  We purchased 3 commercially available
>>> "Asterisk Based" PBXs a little over a year ago. (I won't mention
>>> which one at this point - I don't want to bad mouth them - yet!)
>>> Two of the systems are very small (5 SIP lines/6 Polycom phones).
>>> The third is on a PRI with 30 Polycom phones.
>>>
>>> My smaller sites work pretty good.  I've only had to restart
>>> Asterisk every month or so.  However, my 30 station system
>>> is a continuous headache.  I average a restart at least once a
>>> week.  Sometimes a couple of times in the week.  I'm always being
>>> called to "fix" something that just stopped working.
>>>
>>> I DON'T WANT TO GET INTO A "Well, don't just complain, tell us
>>> your setup and we can help you get it working".  This list HAS
>>> helped me figure out some of the issues.  THANK YOU!  But the
>>> purpose of this post is more of a fact finding mission.
>>>
>>> 1) Was choosing Asterisk for our company the wrong decision...
>>>
>>>a) IF... I expect a phone system to just work.  Once it is
>>>   configured, a phone system should just work with
>>>   very little attention.  My previous system was a
>>>   Comdial with external voice mail on a DOS based PC.
>>>   I LITERALLY WENT OVER 4 YEARS WITHOUT HAVING TO REMOVE
>>>   POWER TO THE COMDIAL CONTROL OR RE-BOOT THE VOICE MAIL PC.
>>>  
>>>b) IF... I really only need a phone system that allows an operator
>>>   to answer each call and transfer them to the appropriate
>>>   person.  I need voice mail, but very little auto attendant
>>>   features (mostly after hours).  All the bells and whistles
>>>   that Asterisk offers are cool, but don't bring that much to
>>>   the table for our purpose.
>>>
>>>c) IF... Stability is more of an issue than high end features?
>>>
>>> 2) Are there any users out there that really DO have an Asterisk
>>>system that just works like clockwork?  I'm saying, once setup,
>>>run for a year (or more) without any issues?
>>>
>>> 3) If SO, Should I simply consider a different vendor?
>>>
>>> 4) If NOT, and if my expectations are that a system SHOULD just
>>>run and run without any problems.  Is Asterisk simply not my
>>>solution.  Is Asterisk not REALLY ready for production.  Because
>>>in my mind (as a user of phone services), "dealing" with the
>>>phone system, even on a MONTHLY basis, means that the system
>>>is NOT really production ready...  Before we installed an
>>>Asterisk based PBX, I spent maybe 4 hours per YEAR with phone
>>>issues (setting up a new station?).  Since we moved to an
>>>Asterisk based PBX, I spend 4 hours (or more) every WEEK!
>>>
>>>Am I expecting too much?
>>>
>>> Bill
>>>
>>>   
>>> 
>>>   
>> I don't think you are expecting t

Re: [asterisk-users] Is Asterisk ready for Prime-Time?

[Drew Gibson]

Bill Andersen wrote:
> This is not a troll.  I've used my real email because I want this
> taken seriously.  I'm not trying to make anyone mad, I just want
> some real discussion on this issue.  Please bare with me...
>
> I'm a USER of Asterisk.  We purchased 3 commercially available
> "Asterisk Based" PBXs a little over a year ago. (I won't mention
> which one at this point - I don't want to bad mouth them - yet!)
> Two of the systems are very small (5 SIP lines/6 Polycom phones).
> The third is on a PRI with 30 Polycom phones.
>
> My smaller sites work pretty good.  I've only had to restart
> Asterisk every month or so.  However, my 30 station system
> is a continuous headache.  I average a restart at least once a
> week.  Sometimes a couple of times in the week.  I'm always being
> called to "fix" something that just stopped working.
>
> I DON'T WANT TO GET INTO A "Well, don't just complain, tell us
> your setup and we can help you get it working".  This list HAS
> helped me figure out some of the issues.  THANK YOU!  But the
> purpose of this post is more of a fact finding mission.
>
> 1) Was choosing Asterisk for our company the wrong decision...
>
>a) IF... I expect a phone system to just work.  Once it is
>   configured, a phone system should just work with
>   very little attention.  My previous system was a
>   Comdial with external voice mail on a DOS based PC.
>   I LITERALLY WENT OVER 4 YEARS WITHOUT HAVING TO REMOVE
>   POWER TO THE COMDIAL CONTROL OR RE-BOOT THE VOICE MAIL PC.
>  
>b) IF... I really only need a phone system that allows an operator
>   to answer each call and transfer them to the appropriate
>   person.  I need voice mail, but very little auto attendant
>   features (mostly after hours).  All the bells and whistles
>   that Asterisk offers are cool, but don't bring that much to
>   the table for our purpose.
>
>c) IF... Stability is more of an issue than high end features?
>
> 2) Are there any users out there that really DO have an Asterisk
>system that just works like clockwork?  I'm saying, once setup,
>run for a year (or more) without any issues?
>
> 3) If SO, Should I simply consider a different vendor?
>
> 4) If NOT, and if my expectations are that a system SHOULD just
>run and run without any problems.  Is Asterisk simply not my
>solution.  Is Asterisk not REALLY ready for production.  Because
>in my mind (as a user of phone services), "dealing" with the
>phone system, even on a MONTHLY basis, means that the system
>is NOT really production ready...  Before we installed an
>Asterisk based PBX, I spent maybe 4 hours per YEAR with phone
>issues (setting up a new station?).  Since we moved to an
>Asterisk based PBX, I spend 4 hours (or more) every WEEK!
>
>Am I expecting too much?
>
> Bill
>
>   

I don't think you are expecting too much.

We have:-

130 physical extensions including 24x7 inbound call centre

Debian on Dell server

[EMAIL PROTECTED]:~# uptime
 13:15:31 up 192 days, 23:49,  2 users,  load average: 0.00, 0.01, 0.00

(Power was removed to switch to new UPS)

asterisk*CLI> show version
Asterisk 1.2.24 built by root @ asterisk on a i686 running Linux on 
2007-09-08 17:17:07 UTC
asterisk*CLI> show uptime
System uptime: 63 days, 4 hours, 26 minutes, 40 seconds

(Asterisk was restarted after queue config changes)


We had a single power supply and single drive fail in one incident in 
Feb 2007 (one drive of RAID 1). System stayed up but was taken down for 
15 minutes to swap the drive. PS was hot-swapped when it arrived later.


regards,

Drew






-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] capacity

[Drew Gibson]

Our office averages around 1.5MB / mailbox, call it 10MB for rounding.

6,000 x 10MB = 60GB (n'est pas?)

2 x 250GB drives, mirrored, should cover that and the system quite nicely.

regards,

Drew

Disclaimer: Most of our employees are programmers so probably don't have 
any friends to call and leave messages! :-)



Steve Totaro wrote:
> RAID arguments (preference really) aside, 4k - 6k worth of student
> voicemails is going to require quite a bit of storage space.
>
> Thanks,
> Steve Totaro
>
> On Wed, Mar 19, 2008 at 12:01 PM, Drew Gibson <[EMAIL PROTECTED]> wrote:
>   
>> Having ventured high enough and far enough to view the curvature of the
>>  Earth and having stayed up late enough long enough (why do disks only
>>  fail at the weekend?) to rebuild and restore RAID 5 sets, I proffer the
>>  following (not so) Humble Opinion .
>>
>>  Dual power supplies, two thumbs up
>>
>>  but RAID 5 is only good for reducing storage costs on large volumes of
>>  data. It reduces performance and reliability over RAID 1. Don't put the
>>  OS on RAID 5 unless you like rebuilding servers from bare metal. It's
>>  much easier to rebuild and restore the data on RAID 5 sets if the OS is
>>  already up and running.
>>
>>  Your OS and other system critical files (Asterisk) should be on RAID 1
>>  for performance, redundancy and cost reasons.
>>
>>  More disks = higher cost and higher chance of failure.
>>
>>  Asterisk in general does not need much disk storage. The minimum drive
>>  size available in a new server tends to be overkill. Two drives as RAID
>>  1 gives you redundancy and performance. Adding a third drive for RAID 5
>>  adds cost, increases complexity and reduces reliability just to add
>>  storage capacity that you don't really need. (but the reseller WILL make
>>  more money and impress you with their command of the big words and
>>  acronyms on the spec sheet.)
>>
>>  If and only if you need to store many hundreds of gigs of data (eg.
>>  recording a very large volume of calls) then RAID 5 becomes useful (or
>>  RAID 10 or RAID n). You should add this "bulk storage" IN ADDITION TO
>>  the mirrored pair holding the OS.
>>
>>  regards,
>>
>>  Drew
>>
>>
>>
>>
>>  Steve Totaro wrote:
>>  > And I can post a link that shows a bunch of guys think the earth is
>>  > flat with a 5/10 google ranking also (like the barf guys).
>>  > http://www.alaska.net/~clund/e_djublonskopf/Flatearthsociety.htm
>>  >
>>  > I usually just call my guy at CDW and give him my needs, he is a
>>  > former techie gone sales.  He puts together a quote and emails it to
>>  > me for approval.
>>  >
>>  > I find HP server are very robust and rock solid at a decent price
>>  > point (IBM as well).  I like the 380 because you get six hot swap scsi
>>  > bays and redundant power supplies in a 2u profile, also, Digium and
>>  > Sangoma T1 cards have never given me an issue.
>>  >
>>  > Many on this list love Supermicro, I have yet to try them but I will
>>  > in the near future.  I have not heard a single complaint, only rave
>>  > reviews.
>>  >
>>  > I guess my original point was going for redundancy as far as storage
>>  > and power supplies with your dollar, not the fastest proc or maxed out
>>  > RAM that will not be needed.  Regardless of the actual hardware or
>>  > RAID setup, that is the angle I suggest you take.  4k - 6k students
>>  > will require quite a bit of storage.
>>  >
>>  > Thanks,
>>  > Steve Totaro
>>  >
>>  > On Wed, Mar 19, 2008 at 9:38 AM, Ron Joffe <[EMAIL PROTECTED]> wrote:
>>  >
>>  >> On Tuesday 18 March 2008 22:12, Steve Totaro wrote:
>>  >>  > For your use, I would go for a RAID 5
>>  >>
>>  >>  I would highly recommend against a raid 5 set. I can give you more 
>> details if
>>  >>  you are interested, but these guys have most if it down : www.baarf.com 
>> see
>>  >>  the link on the left on "why should I not use Raid 5"
>>  >>
>>  >>  Ron
>>  >>
>>  >>
>>  >>
>>  >>
>>  >>
>>  >>  ___
>>  >>  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>  >>
>>  >>  asterisk-users mailing list
>>  >>  To UNSUBSCRIBE or update options visit:
>>  >>http://list

Re: [asterisk-users] capacity

[Drew Gibson]

Having ventured high enough and far enough to view the curvature of the 
Earth and having stayed up late enough long enough (why do disks only 
fail at the weekend?) to rebuild and restore RAID 5 sets, I proffer the 
following (not so) Humble Opinion .

Dual power supplies, two thumbs up

but RAID 5 is only good for reducing storage costs on large volumes of 
data. It reduces performance and reliability over RAID 1. Don't put the 
OS on RAID 5 unless you like rebuilding servers from bare metal. It's 
much easier to rebuild and restore the data on RAID 5 sets if the OS is 
already up and running.

Your OS and other system critical files (Asterisk) should be on RAID 1 
for performance, redundancy and cost reasons.

More disks = higher cost and higher chance of failure.

Asterisk in general does not need much disk storage. The minimum drive 
size available in a new server tends to be overkill. Two drives as RAID 
1 gives you redundancy and performance. Adding a third drive for RAID 5 
adds cost, increases complexity and reduces reliability just to add 
storage capacity that you don't really need. (but the reseller WILL make 
more money and impress you with their command of the big words and 
acronyms on the spec sheet.)

If and only if you need to store many hundreds of gigs of data (eg. 
recording a very large volume of calls) then RAID 5 becomes useful (or 
RAID 10 or RAID n). You should add this "bulk storage" IN ADDITION TO 
the mirrored pair holding the OS.

regards,

Drew


Steve Totaro wrote:
> And I can post a link that shows a bunch of guys think the earth is
> flat with a 5/10 google ranking also (like the barf guys).
> http://www.alaska.net/~clund/e_djublonskopf/Flatearthsociety.htm
>
> I usually just call my guy at CDW and give him my needs, he is a
> former techie gone sales.  He puts together a quote and emails it to
> me for approval.
>
> I find HP server are very robust and rock solid at a decent price
> point (IBM as well).  I like the 380 because you get six hot swap scsi
> bays and redundant power supplies in a 2u profile, also, Digium and
> Sangoma T1 cards have never given me an issue.
>
> Many on this list love Supermicro, I have yet to try them but I will
> in the near future.  I have not heard a single complaint, only rave
> reviews.
>
> I guess my original point was going for redundancy as far as storage
> and power supplies with your dollar, not the fastest proc or maxed out
> RAM that will not be needed.  Regardless of the actual hardware or
> RAID setup, that is the angle I suggest you take.  4k - 6k students
> will require quite a bit of storage.
>
> Thanks,
> Steve Totaro
>
> On Wed, Mar 19, 2008 at 9:38 AM, Ron Joffe <[EMAIL PROTECTED]> wrote:
>   
>> On Tuesday 18 March 2008 22:12, Steve Totaro wrote:
>>  > For your use, I would go for a RAID 5
>>
>>  I would highly recommend against a raid 5 set. I can give you more details 
>> if
>>  you are interested, but these guys have most if it down : www.baarf.com see
>>  the link on the left on "why should I not use Raid 5"
>>
>>  Ron
>>
>>
>>
>>
>>
>>  ___
>>  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>>  asterisk-users mailing list
>>  To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] British Summertime & Grandstream Phones

[Drew Gibson]

Gordon Henderson wrote:
> On Sun, 25 Mar 2007, Gordon Henderson wrote:
>
>   
>> and change the Optional Rule to:
>>  3,3,7,1,0;10,4,7,2,0;60
>> someone correct me if I've goofed!
>> 
>
> Well, it's been a year since I wrote this, and of-course I goofed and 
> no-one corrected me ;-)
>
> Bother.
>
> So I've looked again.
>
> In the UK, We "Spring Forward" on the last Sunday of March, and we "Autumn 
> Back" on the last Sunday in October.
>
> Not the 3rd as I put there, which worked last year.
>
> So there's still 1.5 weeks to go...
>
> So it needs to be:
>
>3,-1,7,1,0,10,-1,7,1,0;60
>
> I think...
>
> Well, I'll give this a go and see what happens...
>
> Gordon
>
>   

Numbers look OK but I think you have a typo, try

3,-1,7,1,0;10,-1,7,1,0;60

(with 2 semicolons)

Check the f/w rev too.

Grandstream fscked up the time change parsing in the f/w last year while 
trying to deal with the new US/Canada changes. It worked in the Spring 
2007 and didn't work in the Autumn. We upgraded to 1.1.4.18 and it did, 
to our amazement, change at the right time (for Ontario) this Spring!

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hardphone SIP phone costs

[Drew Gibson]

Anciso, Roy wrote:
>
> I ’ m trying to understand something that just doesn’t seem to 
> compute. How can companies like Cisco justify selling their hard 
> phones for as much as they do? I know there is a matter of recouping 
> R&D costs but when you look at the iPhone with all its amazing 
> features for less than $500.00 it just doesn ’ t make sense. Am I the 
> only one that thinks this?
>

Hi Roy,

Although Cisco generally make good gear, it does tend to be 
over-featured and over-priced. It's due to two main factors, branding 
and lock-in.

Branding:- NoLogo by Naomi Klein is a good primer (regardless of which 
side of the fence you sit), see also H. C. Andersen's "The Emperor's New 
Clothes" http://en.wikipedia.org/wiki/The_Emperor's_New_Clothes

Lock-in:- If you've already bought into The Brand (eg Cisco's Call 
Manager software) you don't have a choice if you want all the features. 
You _could_ try telling the boss that the $100k+ that was spent on The 
Brand was a waste of money, that it could have been done for $20k with a 
mix of Brands, with more relevant features and that you're _sure_ the VP 
would understand  but then, no one got fired for buying IBM.


regards,

Drew


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Telemarketer Torture....

[Drew Gibson]

James Finstrom wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Anyone have the telemarketer torture prompts? I would seriously like
> to revive this.
>
> - --
> James Finstrom
>
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.6 (GNU/Linux)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>
> iD8DBQFH3I8qdloC7YyaIOoRAlAjAJ9Hp+3SS2Z8179HecWIETp4RVDzWQCeMizp
> fW2JPZdYl/uxG1ziUwYnHGo=
> =QPbv
> -END PGP SIGNATURE-
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   

Hi James,

I have a copy of the prompts. Will the list accept attachments?

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] SNOM on "Do Not Call" list????

[Drew Gibson]

Some light relief 

SNOM say "Please note that you will not be able to reach us by phone."


http://www.theregister.co.uk/2008/03/13/dont_call_us/


regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] FXS channel banks

[Drew Gibson]

Chris Bagnall wrote:
> Greetings list,
>
> I've been asked to provide a system for 200 extensions, most of which will be 
> existing analogue POTS handsets, not IP handsets. I've not really had any 
> experience with large channel banks in the past (since most of our 
> deployments are strictly IP-only to the desk), so I'm at a loss as to which 
> ones are worth looking at.
>
> If anyone's had experience using channel banks on reasonably sizeable 
> installs I'd be interested to hear what device(s) you used, how simple or 
> complex they were to configure, and whether there'd be any issues attaching 
> multiple units to a single server.
>
> This install would be in the UK, so we do need to factor in the different 
> conditions expected by UK POTS handsets (line impedance, etc.). Are most 
> channel banks country-neutral, or do specific models need to be purchased for 
> different line conditions in each country?
>
> Thanks in advance.
>
> Regards,
>
> Chris
>   

www.citel.com

I used them a few years back in a pilot install with legacy Nortel 
phones and it worked well. I gather they have grown tremendously from 
there. I'm in North America, don't know how well they support UK stuff.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Automon reliability issue

[Drew Gibson]

Jaap Winius wrote:
> Hi list,
>
> Can someone please explain how to get one touch recording (automon) to  
> work reliably? I'm using Asterisk 1.4.14 on a Debian etch system. My  
>   

We made this function reliable by including the word "quickly" in our 
instructions for pressing the keycode to start the recording. (Asterisk 
1.2.24 on Debian Etch)

Although a private confirmation beep to the initiator of the recording 
would be handy, this is the way things have to be in order to use the 
features.conf codes and still allow the use of * and # when calling 
outside IVR and voicemail systems. eg. "Enter your password followed by 
the pound key..."

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to hookup to cell phone for outbound calls?

[Drew Gibson]

Erik Anderson wrote:

On Feb 5, 2008 2:37 PM, Drew Gibson <[EMAIL PROTECTED]> wrote:
  

How about http://www.mgamble.ca/oss/iphone_asterisk/ ?



Hah!  Cool, but quite ridiculous. :-)

  


I have a Linksys NSLU2 (Slug) at home running Asterisk (see 
http://www.nslu2-linux.org/ )


It's small, relatively cheap and runs Asterisk very well. You could slip 
it into a pocket.


I haven't tried yet but I have done a little reading and hope to connect 
the Slug to a mobile network when I get the time to play.


I know you said no bluetooth, Ed but if you're in North America and your 
cellular network is CDMA, AFAIK option 1 is the only one possible.  
These carriers generally won't allow devices on their networks unless 
they are purchased from the carrier.


If your cellular network is GSM then there are two approaches to try,

1. Slug, 4GB USB stick, USB Bluetooth dongle, dedicated bluetooth mobile 
phone + Asterisk 1.4 with a chan_mobile. Unfortunately, I doubt that 
chan_mobile is packaged for the slug (it's in 1.4 trunk) and you would 
have to build it. Cost ~$150 + phone + your time


2. Slug, 4GB USB stick + SIP-GSM gateway. Much easier to configure but 
it's a second box so less portable and more expensive than a BT dongle 
and an old phone. Probably more robust. Cost $250-$400


You could also substitute a Linksys WRT54GL ( http://openwrt.org/ ) for 
the Slug which would give you ethernet ports and wireless too.


Hope this gives you some ideas

regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to hookup to cell phone for outbound calls?

[Drew Gibson]

Ed W wrote:
> Hi
>
> I need a small PBX for use on the move.  This means that outbound calls 
> will need to be made over the cell phone network.
>
> Assuming a small hardware PBX with a spare mini-PCI slot or a USB slot 
> then what hardware options do I have to get an outbound cellular 
> channel?  Options need to be rock solid, so no bluetooth to a cell phone 
> kind of solutions need apply. 
>
> Can any of the 3G usb devices out there offer outbound analogue calls 
> (ie other than via voip)?
>
> Cheers
>
> Ed W
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   

How about http://www.mgamble.ca/oss/iphone_asterisk/ ?

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] pulling my hair out over voicemail

[Drew Gibson]

John Von Essen wrote:
> Any ideas what could be going on? I tried tweaking the extension 1000 
> so it looks like:
>
> exten => 1000,3,VoicemailMain,s6000
>
>   
It may be your syntax, try :-

exten => 1000,3,VoicemailMain(6000|s)


regards,

Drew


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] POE draw on Aastra 480i

[Drew Gibson]

Octavio Ruiz wrote:

Allen Casteran wrote:
Anyone know what the POE draw is for the Aastra 480i phones?
We have switches that will do 15 watts on 12 ports but only do 7.7 watts on all 
24 ports.
A Cisco 3560 switch will do 15.6 watts on all 24 ports.
Just trying to find out if we need that much power.
  


  

Drew wrote:
According to Aastra tech support, 5 watts (peak) per 480i.
We are testing five phones running on a Linksys SRW208P that will only support 
full 15W on up to
4 of 8 ports. I can power up the switch while all phones are connected without 
any issues.
I would expect your lower power switch will provide ample power.



But, PoE class does not matter? Did you plug five Aastra phones?

I'm suspicious about how that scenario worked, I mean, as far as i
know Aastra phones should register as a zero PoE class, that means it
would reserve up to 12.94 watts no matter how many watts uses.  So, my
guess here is even if the phone use only 5 watts, the switch already
reserved 12.94 watts for it. I would love to see what happens if you
plug a sixth phone or figure out if you used an Aastra phone. Can you
tell us what model/brand you used?

Dimensioning PoE devices over capable switches has been a new issue
which involves many factors like those described before.

Regards,


PD. Sorry about  the original thread break off, I've been unable to
find the original one.
  


Hi Octavio,

We are using Aastra 480i phones powered by PoE.  During the testing we 
decided to drop the Linksys SRW208P as it is extremely noisy due to the 
fans, not suitable for an office environment. We continued testing with 
Netgear FS116P switches. These have 8 PoE ports plus 8 non-powered 
ports. Completely silent, no fans. Our testing was done with 5 phones 
per switch as our call centre is laid out in "pods" with 5 seats per pod 
and will not require more devices powered by PoE.


I did not look into the issues of PoE class as there were no problems 
meeting our requirements with the equipment we had. Perhaps the Cisco 
switches are more particular about PoE class, they are much more complex 
devices. I like Cisco gear, they make good products but they tend to be 
over-featured and over-priced for many applications.


We have had 5 of the Netgear switches in production for almost a year 
now, each powering 5 Aastra 480i phones without any issues whatsoever.


regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Peak number of calls?

[Drew Gibson]

Gordon Henderson wrote:
> Is there any way to find-out the peak number of calls that an asterisk 
> system has had? Not the total number of calls, but the maximum number of 
> simultaneous calls.
>
> I know I can porobably go through the CDR logs and look for calls which 
> have overlapped in time, but I'm wondering if there's some counter 
> somewhere I could access...
>
> (I'm looking for evidence for an ISDN client who wants to know if he's 
> spent too much on the number of ISDN lines he has installed!)
>
> Cheers,
>
> Gordon
>   

We use Asterisk-stat from Areski (GPL). It will show peak number of 
calls by the hour. Select "Daily Load", scroll down and choose the hour 
you want and "Fluctuation Graph". Lots of other goodies too.

http://areski.net/areski/index.php?option=com_content&task=view&id=22&Itemid=54

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [Fwd: Re: Large issue - having trouble diagnosing.]

[Drew Gibson]

Hi Cameron,

I think this paragraph goes to the heart of the matter...

Cameron Hissey wrote:
 are connected to the standard switch, however the cabling was a bit 
of a rush job and consequently the PoE has proven unstable on many of 
the points, with some of them not even supplying data packets. 
Benny Amorsen  suggested trying a store-bought cable direct to a phone, 
how did that go?


I would have a very close look at the cabling. Were the "R45" plugs put 
straight onto the cable or terminated in a wall jack? There are two 
types of plug. The one that is most commonly available is not intended 
for the solid copper core cable that is most commonly available in bulk 
boxes and WILL cause flakey connections.


Check the cables for correct pinout (see link below), not just 
continuity. The price of a reasonable cable tester will save you a 
fortune here. Nothing fancy but it must be able to check for split 
pairs. Split pairs could cause the symptoms you describe.


http://www.ertyu.org/steven_nikkel/ethernetcables.html

regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] OT: Traffic Shaping

[Drew Gibson]

Dovid B wrote:
- Original Message - 
From: "Matt Riddell" <[EMAIL PROTECTED]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Erik Anderson wrote:


On Jan 9, 2008 9:40 PM, Matt Riddell <[EMAIL PROTECTED]> wrote:
  

Heh yeah that's what I was thinking of doing.  What's the traffic
shaping like?  Can I specify max bandwidth etc or use hfsc shaping?


DD-WRT will do both HTB and HFSC shaping, though I've only ever used HTB.
  

Sweet. We had been using HTB but upgraded all our CPE to use HFSC when
AstLinux did and found it great.



Here's the dd-wrt wiki page on its QoS implementation:

http://www.dd-wrt.com/wiki/index.php/Quality_of_Service
  

Cool, looks good, although I would have thought Default Bandwidth Level
would have been pretty self explanatory if it works as expected.

I guess I'll have to try it out and update the wiki if it does.



Looks like they don't recommend HFSC currently due to some lag issues.
That might have been fixed, though, in the more recent firmware
builds.
  

Will try both out and see how they go.

Thanks for the pointers.

- --
Kind Regards,

Matt Riddell
Director
___



Matt,
The WRT54GL is not dual port. 

  


Chorus from the audience:- "Oh, yes it is!"

It has 5 ports! Although the ports are labeled as 1 Internet port and 4 
LAN ports, each can be assigned to a VLAN of your choosing and you can 
use them as you please (at least you can under openWRT). So you could 
allocate 2 WAN ports on distinct networks and leave 3 for the LAN.


I haven't done it myself but a far more clever gentleman at 
http://garycourt.com/blog/post/openwrt-advanced-firewall/ seems to know how.


regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [OT] black dogs

[Drew Gibson]

Philipp Kempgen wrote:

Drew Gibson wrote:

  
A well-written application should attempt to minimize the amount of 
'conversion' the user/programmer has to do. Therefore the command 
structure SHOULD be in a form that is natural for the user/programmer, 
NOT to the machine.


Personally, I would vote for "show dogs colour black" but maybe I've 
spent too much time with Cisco's IOS! :-)



"show me all the colored dogs now. hurry up! don't spent
any time in those other threads."
  



"and while you're at it would you please fix that stupid
mistake i made in the config file"

  


Is this in the schedule for 1.6?

regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Drew Gibson]

Since we're WAY OT anyway

Tony Plack wrote:
> That being said, ordering in a command structure should make sense to the 
> application (less intelligent entity), not to the "programmer" (hopefully 
> more intelligent).
>   

If that were true then we really should be writing our dialplans in 
binary machine code, that is what that dumb computers REALLY understand.

Fortunately, it's not true. We can take advantage of a GOOD programmer's 
skill to have the computer do the grunt work of converting something 
real people understand into machine code. We  call the product of this 
process a High-Level Programming Language.

A well-written application should attempt to minimize the amount of 
'conversion' the user/programmer has to do. Therefore the command 
structure SHOULD be in a form that is natural for the user/programmer, 
NOT to the machine.

Personally, I would vote for "show dogs colour black" but maybe I've 
spent too much time with Cisco's IOS! :-)

regards,

Drew

PS. There does seem to be an assumption that programmers are 
intelligent, I'm not sure that this is a defensible position. ;-)

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] X100P Fxo card headaches

[Drew Gibson]

Chris Boczko wrote:
> Hello List,
>
> Im just dipping my feet into the asterisk world, and im having major
> fxo problems
>
> Im running Asterisk (from svn) + libpri (from svn) + asterisk-addons
> (from svn) + asterisk gui (svn 1.4 branch) + zaptel (svn 1.4) on a
> Debian Etch box, with 1gb ram, running all of the services for my home
> server (web / db / music server etc), and i would like to run my PSTN
> line from Kingston Comms, but i can't get this box the recoginsie this
> line!
>
> The X100p is a cheap clone i got off ebay for a tenner, so im not
> expecting much, i know they have echo issues, but im going to upgrade
> to a SPA3012 / TDM400B when i have the cash.
>
> Ztcfg -vv reports
>
> orange:~# ztcfg -vv
>
> Zaptel Version: SVN-branch-1.4-r3374
> Echo Canceller: MG2
> Configuration
> ==
>
>
> Channel map:
>
> Channel 01: FXS Kewlstart (Default) (Slaves: 01)
>
> 1 channels to configure.
>
> orange:~#
>
>
> and my zaptel.conf file contains
>
> orange:~# more /etc/zaptel.conf
> fxsks=1
> loadzone=uk
> defaultzone=uk
> orange:~#
>
> but zap show status in the command line shows
>
> orange*CLI> zap show status
> Description  Alarms  IRQbpviol CRC4
> Fra Codi Options  LBO
> Wildcard X100P Board 1   OK  0  0  0
> CAS Unk  YEL  0 db (CSU)/0-133 feet (DSX-1)
> orange*CLI>
>
> and i think the Unk means unknown...?
>
> Does anyone have any ideas on how to get this line to work, ive
> followed every howto i can find, and google seems to be comming up
> short, as far as i can see, ztcfg should report the card as
> configured, but it isn't, and ive no idea why.
>
> Hope you can help
>
> Chris
>
>   

Try adding channels=1 to the end of your zaptel.conf to assign the 
settings you have made to a channel.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [asterisk-dev] trunk working under windows!

[Drew Gibson]

t; rm -f gentone
>>> $(MAKE) -C misdn clean
>>>
>>> Modified: trunk/pbx/Makefile
>>> URL: 
>>> http://svn.digium.com/view/asterisk/trunk/pbx/Makefile?view=diff&rev=89454&r1=89453&r2=89454
>>> ==
>>> --- trunk/pbx/Makefile (original)
>>> +++ trunk/pbx/Makefile Tue Nov 20 10:12:10 2007
>>> @@ -25,6 +25,10 @@
>>>  
>>>  include $(ASTTOPDIR)/Makefile.moddir_rules
>>>  
>>> +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),)
>>> +  LIBS+= -lres_ael_share.so -lres_monitor.so
>>> +endif
>>> +
>>>  clean::
>>> rm -f ael/*.o
>>>  
>>>
>>> Modified: trunk/res/Makefile
>>> URL: 
>>> http://svn.digium.com/view/asterisk/trunk/res/Makefile?view=diff&rev=89454&r1=89453&r2=89454
>>> ==
>>> --- trunk/res/Makefile (original)
>>> +++ trunk/res/Makefile Tue Nov 20 10:12:10 2007
>>> @@ -25,6 +25,13 @@
>>>  
>>>  include $(ASTTOPDIR)/Makefile.moddir_rules
>>>  
>>> +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),)
>>> +  # order-only dependency: build res_monitor before res_features
>>> +  res_features.so: | res_monitor.so
>>> +  # res_features uses some functions from res_monitor
>>> +  res_features.so_LIBS:= -lres_monitor.so
>>> +endif
>>> +
>>>  ael/ael_lex.o: ael/ael_lex.c ../include/asterisk/ael_structs.h 
>>> ael/ael.tab.h
>>>  ael/ael_lex.o: ASTCFLAGS+=-I. -Iael 
>>>  
>>>   

-- 

Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] RTP Read too short

[Drew Gibson]

I saw this with Grandstream GXP2000. When the phone is on a call and on 
mute, the phone sends SIP "keepalive" packets that are, indeed, too 
short. So asterisk is correct in this case. Grandstream said that they 
were just warnings and to ignore them. We have chosen to ignore 
Grandstream and move to a different phone vendor.

regards,

Drew

PS. I missed your question earlier because it was a reply to an existing 
thread. If you want to be seen, start a new thread, don't hijack an old one.


John Faubion wrote:
> Am I the *ONLY* one that has this issue? 
>
> John Faubion
>  
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] Behalf Of John
>> Faubion
>> Sent: Thursday, November 01, 2007 11:01 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] RTP Read too short
>>
>>
>> Hello,
>> I'm getting the following logs:
>>
>> [Nov  1 10:54:37] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too
>> short
>> [Nov  1 10:54:39] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too
>> short
>> [Nov  1 10:54:40] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too
>> short
>>
>> Anyone know how to correct this? I'm using SIPConnect from 
>> CBeyond and this
>> appears on incoming calls. I haven't had any complaints about 
>> voice quality
>> and I haven't seen any dropped calls. Should I be concerned?
>>
>> Thanks,
>> John Faubion
>> 


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] (no subject)

[Drew Gibson]

[EMAIL PROTECTED] wrote:
> Hi all,
>
> We have a client that needs to setup about 80 desk phones (about 50  
> in one location and about another 30 in 5 different locations). Which  
> brand/model would you recommend. We were personally thinking in  
> recommending either Cisco, Aastra, Polycom, or Snom, for we've heard  
> great things about them. However, having no real experience with them  
> makes it hard in recommending one to our customer. The only  
> experience we've had is a very frustrating one trying to load the IP  
> software on a Cisco 7970G and so we assume that if we have to go  
> through that for all 80 phones, we'll probably commit suicide :)
>
> Thanks
>   

We have used Cisco and Aastra, can't comment on Polycom or Snom.

I cannot recommend Cisco, good sound quality but that's it. Ridiculously 
overpriced, too few usable features, incredibly awkward to manage.
Aastra have good sound quality, reasonable price, configs are plain text 
and not to hard to work with. We have the 9133i as our basic phone and 
480i in the Call Centre for the soft buttons. Both can be fed from the 
same config templates.
We used to use Grandstream but quality and support issues have driven us 
away.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Stuck Voicemails?

[Drew Gibson]

Matt wrote:
This question is about 1.2.x asterisk.  Please no flames, or "you 
should upgrade to 1.4".


Does anyone know what might be the cause for 'stuck voicemail's in 
1.2.6 asterisk?  By stuck, I mean the phones show a voicemail, and if 
you log in you get "you have 1 new voicemail", and if you delete it it 
says 'deleted', however it remains.   Going into the mail directory 
reveals that there is either a msg0001.txt.tmp or a msg0001.txt file, 
but no associated wav file.


It happens very randomly, not often, and so far has eluded me being 
able to figure out what causes it.


Why does this happen?
I don't know why it happens but we have run a very early 1.2 svn before 
jumping to 1.2.17 (fixed several issues) and incrementally through to 
1.2.24 and we have not seen this issue.


Any related errors in asterisk logs or system logs?
Has fsck been run recently on the relevant filesystem?
Is your mail stored on a local filesystem?
Would upgrading to 1.2.24 be an acceptable upgrade?

regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] FXO ATA Options?

[Drew Gibson]

Conall O'Brien wrote:
> Hello,
>
>
> I'm currently looking at FXO options to provide a POTS line to Asterisk to 
> trunk calls with.
>
>
> Does anyone have any experience using the Linksys Sipura 3201 as an FXO 
> device for Asterisk?
>
>   

I use one at home and can recommend it as functional and reliable. It 
has an unbelievable number of configuration options. Linksys docs are a 
bit sparse, try the Sipura site under SPA3000.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Grandstream GXP-2000's and Asterisk.

[Drew Gibson]

Thomas Kenyon wrote:

Thomas Kenyon wrote:
  

I have 5 GXP 2000's with firmware 1.1.4.25 running with Asterisk 1.4.13.

Is anyone else getting the following error in the asterisk console:

[Oct 22 11:39:01] WARNING[7100]: rtp.c:1142 ast_rtp_read: RTP Read too short

every couple of seconds when a handset is in a call?

I didn't notice this happening when I was using an older GXP2000 with 
the same firmware (doesn't mean that it didn't happen).


The Call in question is using G.729.

TIA for any help with this.

I will hopefully get a bit more time to play with this today. (When I'm 
in the office in question).



Changing codec doesn't appear to matter. I gather that the cause is that 
the GXP-2000 sends empty udp packets as keep-alives. (which is all well 
and good, but even with a handful of handsets with light call volume the 
logs fill up with notices, at the moment there is only 1 call going 
through the server and this is generating 2 notices/second.


Is there any way to make asterisk ignore the empty packets from certain 
peers?


  

Hi Thomas,

I have tried to work through these (and other) issues with Grandstream 
but they seem to have a short attention span. We now buy Aastra phones.


regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] BBC on Atserix

[Drew Gibson]

SIP wrote:

Drew Gibson wrote:
  

Tzafrir Cohen wrote:


On Thu, Oct 18, 2007 at 01:04:06PM +0100, Cartwright, Dave wrote:
  
  

Just for fun.

http://news.bbc.co.uk/1/hi/magazine/7049642.stm



It's "Asterix" != "Asterisk". Though named after "*".

  
  

In Britain, it's called "humour" :-)

regards,

Drew




Only if it's actually humourous. ;)

N.

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
  

It's looking good ...

http://news.bbc.co.uk/2/hi/entertainment/4343264.stm

regards,

Drew


--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] BBC on Atserix

[Drew Gibson]

Tzafrir Cohen wrote:

On Thu, Oct 18, 2007 at 01:04:06PM +0100, Cartwright, Dave wrote:
  

Just for fun.

http://news.bbc.co.uk/1/hi/magazine/7049642.stm



It's "Asterix" != "Asterisk". Though named after "*".

  


In Britain, it's called "humour" :-)

regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Refrigerator Alarms

[Drew Gibson]

Hi Balu,

http://www.digitemp.com/ has all the info you need. Cost of hardware 
around US$70. Will give you pretty fridge temperature graphs too!
You can easily hack some of Brian's scripts to do different levels of 
temp alarm and trigger calls in Asterisk.


regards,

Drew


Balu Raman wrote:

Omar,
I am hoping that there may be some temp sensor interface that can be
routed to a pc and if the temp falls out of a range, I can have this
event call someone. I know what to do in asterisk to make a call. I
have to do some research. may be,  someone has already done a similar
thing. Has to be event driven.
Thanks,
balu raman

On 10/17/07, Omar A. Sabek <[EMAIL PROTECTED]> wrote:
  

Balu,

Do you want events passed to Asterisk from the refrigerator? Or does a
reminder type phone call need to be placed on an interval? Please be
more specific, since this sounds like a special purpose refrigerator,
does it have any way of passing events to an external device?

Omar A. Sabek

On 10/17/07, Balu Raman <[EMAIL PROTECTED]> wrote:


Hi,
I want asterisk to call a person on the phone for monitoring the
refrigerator storing vaccines.
I am clueless where to look. Can someone clue me in ?
Thanks,
balu raman

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users

  

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
  



--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?

[Drew Gibson]

Hi Raul,

I think that Bacula is going to cause you many headaches with Asterisk. 
Backups are (surprise!) I/O intensive and when streaming data from the 
network to disk or tape it will saturate the available bandwidth. This 
will cause the I/O wait time on the CPU to run high and effectively 
block other processing (eg. voice traffic) for seconds at a time. Try 
running a full backup during office hours and see what your users have 
to say!


My recommendation is to move either Asterisk or Bacula to a separate 
machine.


regards,

Drew


Raúl Gómez C. wrote:

Well, this has become a hot topic! :p

Thinking about my original post, I was reluctant of installing my PBX 
on a shared system, is a Dell PowerEdge 2950 with 2 Intel Xeon Dual 
Core CPUs @2GHz (4 totals cores) and 4GB RAM which serves as Domain 
Controller and File Server (Samba), central backup server (Bacula with 
a LTO2 external tape drive), it has dual NIC in a bonding alb mode and 
redundant PSU (each one connected to a different UPS). It has a PCI 
slots in which I can install my Sangoma Remora A400D card.


But now I think the PBX will work just fine in this system, maybe 
breaking the channel bonding and dedicating a NIC for the PBX and the 
other NIC for the remaining task, what do you think? Or its better to 
install the PBX on a dedicated system? Let me know your opinions!


Regards...

Raul


On 10/12/07, *Mojo with Horan & Company, LLC* 
<[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:



You're correct, it IS 100 -- but 100%.  Expressed in decimal format,
this is of course 1.0 -- and as each cpu has this average, 4.0
indicates
that no threads regularly wait for execution.  This worldcommunitygrid
you mentioned binds your cpu by design it sounds like.




___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Grandstream GXP2020 / 2000

[Drew Gibson]

Erik Wartusch wrote:
> Hi,
>
> Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a 
> business graded installation (with really traffic on  not 3 calls a 
> day ;-) )
> Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall)
>
>   
Hi Erik,

we have about 75 Grandstream GXP2000 phones running with Asterisk 1.2. 
We also have about 25 Aastra 480i phones in our call centre and 10 Cisco 
phones in meeting rooms.

The Grandstreams work fine on the whole, they are probably good value 
for money in smaller installations. We use some simple bash scripts to 
manage the configurations so the need to convert the config files from 
text to a binary format is not an issue. Sound quality is OK, the 
speaker phone is not great. Most users are happy with the Grandstreams 
but we give Aastras to anyone who spends a lot of time on the phone.

The reason I cannot recommend these devices for larger installations is 
the mediocre response from Grandstream's  technical support. Tech 
support will acknowledge your initial problem report but then ignore you 
if they don't have an immediate fix. This is a pattern repeated over 
several reported incidents.
Our latest issue is with the GXP2000's running f/w <= 1.1.1.14. The 
phone does not send a "keep-alive" packet when the mute function is 
used, despite this bug being documented as fixed in a much earlier 
release. This results in a disconnect after 5 minutes of being on 
"mute". Very annoying when on a conference call or on hold to tech 
support. This is fixed in 1.1.4.18 but this release introduces an issue 
with very loud (for our environment) ring tones rendering the GXP2000 
unusable in our office.

We no longer purchase Grandstream phones but will consider them again in 
the future should the support issues be resolved.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] sip.conf best practices?

[Drew Gibson]

Erik Anderson wrote:
> All - I've been wrestling with how to best structure the sip device
> accounts on a new asterisk server I'm deploying.  All of the sip
> devices (currently only Linksys SPA941s) will reside on the same
> subnet as the server, and I have already set up a decent automatic
> provisioning system for the phones.  When the rollout is complete,
> there will be about 100 SIP devices authenticating and routing calls
> through this server.  The question is what to use for the username
> portion of the SIP account.
>
> Part of me says that I should standardize on using each phone's MAC
> address as the sip account UID, like so:
>
> ; Joe Smith, x123
> [000E08DA0409]
> secret = blahblah
> ... and so on and so forth
>
> Doing it that way is nice for standardization's sake, but it makes the
> dialplan quite a bit more complex.
>
> The obvious alternative is to use the extension as the sip UID:
>
> ; Joe Smith, x123
> [123]
> secret = blahblah
> ...
>
> This makes the dialplan *much* more simple, but when looking through
> sip.conf, it's not as immediately obvious what device should be
> authenticating with that account.
>
> Since this is my first large-ish asterisk deployment, I'm seeking the
> advice of those who have gone before me.  What tactic (one of the
> above options or otherwise) is best to keep your sip.conf sane?
>
> Thanks!
> -Erik
>
>   
Hi Erik,

We have around 100 devices and most of our changes are adds for new 
users/devices with occasional re-assignment of devices. We manage our  
users and devices with some simple scripts and good old vi for exceptions.

Our extensions.conf has a list of global vars that tie an extension to a 
sip (or iax, or whatever) device. (I think this is straight from TFOT v1)

eg EXT_100=SIP/100

This allows us to redirect extensions to different devices or make 
extensions ring on multiple devices by changing that var alone (no need 
to alter macros or other dialplan elements)

eg EXT_100=SIP/101&SIP/102

The device-specific hardware and the SIP configurations are generated 
from a master "map" that contains a line per device including technology 
and extension, MAC, user display name and email address.

Scripts create the phone hardware configs with device type determined by 
MAC address (eg. Aastra, Grandstream or Cisco) from the "map" file and 
add the user to sip.conf and voicemail.conf.

sip.conf ...

[grandstream]
; Aastra 480i phones for general office
... (general SIP settings)
context=office-dial

[aastra-cc]
; Aastra 480i phones for Call Centre only
... (general SIP settings)
context=cc-dial

[100](grandstream)
username=100
secret=***
mailbox=100
callerid="Joe Bloggs" <100>

[101](aastra-cc)
username=101
secret=***
mailbox=101
callerid="Agent 99" <101>


Initially we only had one "class" of user, general office types using 
Grandstreams. When we migrated the Call Centre to Asterisk, they were 
the only users with Aastra phones, apart from one or two in the general 
office. So each "class" of user had a different hardware type and was 
easy to automate, the exceptions are currently handled with manual edits.

This system is working well and is stable but not sufficiently flexible 
for the future. Our company is growing rapidly and since we will no 
longer be buying Grandstream devices, more Aastras are appearing in the 
general office environment. This means we now have two "classes" of 
users that require different configs for the same device type, general 
office users and Call Centre agents.  This means a choice will have to 
be made between updating the scripts to cope with the two user classes 
or moving to realtime. Where's my Magic 8-ball... :-)

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] how to route outgoing calls on IP-level

[Drew Gibson]

Kate Kretz wrote:

Dear Sirs,

out asterisk server has multiple network cards.

I want some outgoing calls (from several extensions) to use one IP 
address, and others to go through

another address.

is there a way to achive that using asterisk ?

Cheers,
Kate

This is the job of your network, not Asterisk. Policy-based routing is 
not much fun (unless you think the Cisco CLI is "really cool") but it 
can be done.


regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Cisco UC 500

[Drew Gibson]

Jeremy Mann wrote:


Is the Cisco UC 500 able to integrate with Asterisk?  Specifically 
does it work via SIP?  Just curious, as the Cold Call Cisco sales rep 
had no clue what SIP even was, and this device looks interesting.



Google "cisco UC500", hit #2 = 
http://www.cisco.com/en/US/products/ps7293/products_data_sheet0900aecd8061fb06.html


Quotes: 

"Core components of the Cisco Unified Communications 500 Series 
include:Cisco Unified IP phones, including wireless handsets and 
Session Initiation Protocol (SIP) phones"


"PSTN interfaces and features:  SIP trunks and RFC 2833 support"

Does that help?

I'll bet Asterisk is cheaper though. :-)

regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users