Re: [Asterisk-Users] Disconnection From IAXTel

[Dave Cotton]

On Sat, 2004-08-28 at 23:36 -0700, Muiz Motani wrote:
> I'm using the firefly third-party softphone. However, the same thing happened 
> when I used IAXphone 2.0.
> 

I can't offer any real solution because I was only testing the
connection with Firefly, but I got exactly the same symptoms whenever
the Firefly softphones tried to communicate, with or without the
notransfer= setting. I blamed it on the Firefly but you say the same
thing happens with IAXphone. Once hardphones where in place that problem
went away.






-- 
Dave Cotton <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Disconnection From IAXTel

[Muiz Motani]

I'm using the firefly third-party softphone. However, the same thing happened 
when I used IAXphone 2.0.



On 29 Aug 2004 at 7:13, you wrote:

> On Sat, 2004-08-28 at 14:01 -0700, Muiz Motani wrote:
> > How do I go about disallowing transfers when I am running an IAX soft 
> > phone. Is that setting not at the * server? Obviously, I don't have control over 
> > the configuration of the IAXTel * server.
> > 
> Which softphone are you using?
> 
> 
> -- 
> Dave Cotton <[EMAIL PROTECTED]>
> 
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-- 

Muiz Motani
Intelligent Distribution
72-6800 Lynas Lane, Richmond, B.C.  V7C 5E2
email: [EMAIL PROTECTED]
phone: +1 604 448 9293 fax: +1 604 448 9296

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Re: [Asterisk-Users] iaxtel and jitterbuffer

[steve]

On Sat, 28 Aug 2004, Andrew Kohlsmith wrote:

> Please note that it seems impossible to disable jitter buffer between 20040806 
> CVS HEAD endpoints.  The jitterbuffer numbers in iax2 show channels look 
> "live".  The numbers look right (jitbuf 0ms) between 20040806 and RC1 
> (Nufone).   I haven't upgraded since then.

The numbers get reported still in the older version, but the buffer IS 
turned off.

Steve

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Re: [Asterisk-Users] iaxtel and jitterbuffer

[steve]

On Sat, 28 Aug 2004, Michael George wrote:

> So even with X11 eliminated the sound is still bad to Digium.  I tried
> another's 1700 number, and it sounded the same, so it's not something unique
> to digium and me.
> 
> Would IAX/GSM be so sensitive to half-duplex that I cannot expect it to work
> with my ISP only giving me 1/2 duplex service?

If you think that the jitter buffer isn't working right and should fix 
this, then please capture debug from the buffer and send over to me.

To do that, in /etc/asterisk/logger.conf edit the debug line to be:

debug => notice,warning,error,debug,verbose

Then run asterisk like so:

/usr/sbin/asterisk -vv -g  -dd -c 

Then go "iax2 debug" at the CLI prompt.

Do a test call, then send me the resulting /var/log/asterisk/debug file.

THanks,
Steve

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[Asterisk-Users] where can I find spandsp?

[Rich Adamson]

Seems the opencall.org site has basically been unavailable for days/weeks.

Is there another location to obtain the current code?

Also, will spandsp install against the current * cvs?

Rich


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Re: [Asterisk-Users] Disconnection From IAXTel

[Dave Cotton]

On Sat, 2004-08-28 at 14:01 -0700, Muiz Motani wrote:
> How do I go about disallowing transfers when I am running an IAX soft 
> phone. Is that setting not at the * server? Obviously, I don't have control over 
> the configuration of the IAXTel * server.
> 
Which softphone are you using?


-- 
Dave Cotton <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] IAX dialing indication tone (PI = 8)

[spkao]

I played around with extensions.conf and found that if you 
just use the 'r'
option in Dial app and leave out the other options t, T, m, 
... then you get
dial indication. Haven't got around to try out the working 
option combinations
but there it is.
 
PK

  - Original Message - 
  From: 
  Matthew 
  Oulton 
  To: [EMAIL PROTECTED] 
  
  Sent: Saturday, August 28, 2004 8:13 
  AM
  Subject: [Asterisk-Users] IAX dialing 
  indication tone (PI = 8) 
  
  
  Hi,
   
  I also have the same problem, 
  from memory is this not progress indicator=8 that deals with the dialing indicator 
  ?
   
  Anyway I am also stuck with 
  not having any dial indication, has anybody got an idea.
   
  MO
  
  
  

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Re: [Asterisk-Users] iaxtel and jitterbuffer

[Andrew Kohlsmith]

On Saturday 28 August 2004 23:01, Michael George wrote:
> It's a PII 266 (okay, not the fatest system) with 192MB RAM.  X is not
> running and the Framebuffer has been turned off in /boot/grum/menu.lst.  I
> have disabled all the servers except for sshd.  I have the latest source
> from CVS HEAD as of about 30min ago.

Should be fine.  I ran * on a P90 for a while; it did everything I needed 
except iLBC.  :-)

> There is no Zap card in this sytem.  The only phone on it is a SIP phone.
> With it I dial in to digium's 1-700 number.  The audio is better, but still
> choppy and unacceptible.

Is your SIP phone doing any kind of silence suppression?  It must be turned 
off because asterisk takes its timing from the RTP stream and if the phone's 
not transmitting frames continuously you'll get shitty audio.

Note that latest CVS HEAD looks like they're making provisions for self-timing 
but without a stable clock source it's unlikely to help you.  There are 
ztdummy modules which use the RTC or certain brands of USB controller to 
provide adequate timing but ideally you want some kind of Zaptel hardware in 
there providing a 1000Hz interrupt.

Also -- make sure your uplink is acceptable.  First test: make sure there is 
nothing plugged into your upstream except for your asterisk box and the 
phone.  Some routers are known to play silly bugger with your packets which 
naturally wreaks havoc with asterisk.  :-)

> So even with X11 eliminated the sound is still bad to Digium.  I tried
> another's 1700 number, and it sounded the same, so it's not something
> unique to digium and me.

Perhaps something to do with your upstream or connection to IAXtel.  That's 
why I'm recommending having nothing but asterisk and the phone on the 
connection, at least until we nail down what the poor audio's being caused 
by.

> Would IAX/GSM be so sensitive to half-duplex that I cannot expect it to
> work with my ISP only giving me 1/2 duplex service?

It has nothing to do with IAX or GSM. Stop blaming them.  My upstream is half 
duplex as well (pretty much anyone on DSL or cable is on a half duplex 
connection whether they realize it or not).  

There are many, many people using asterisk every day for long distance and in 
environments where audio quality is crucial.  Let's stop blaming asterisk and 
take a good hard look at what's happenning, shall we?

-A.
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Re: [Asterisk-Users] POE

[Kevin P. Fleming]

Steven Pritchard wrote:
The dumb, cheap ($30 retail) 3com PoE injectors (3CNJPSE) are working
fine for me with my snom 200 phones.
And there are plenty of places to buy them cheaper as well. Provantage 
has them for around $22. www.soekris.com (a maker of embedded PC 
systems) has the same unit without the 3Com sticker on it (it's actually 
made by Ault) for $20, and as low as $15 in quantity.

And to echo everyone's else's comments, it works fine. We've used one 
with a Snom 200, an Avaya 4602 and a Cisco 7940G (with custom cable). 
Not all at the same time, of course .
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Re: [Asterisk-Users] iaxtel and jitterbuffer

[Michael George]

On Sat, Aug 28, 2004 at 05:08:30PM -0400, Andrew Kohlsmith wrote:
> On Saturday 28 August 2004 15:24, Michael George wrote:
> > I just saw a page on the wiki that mentions that running X11 or a VESA
> > frame buffer can cause jittery sound.  I only have this problem with IAX2,
> > but that might be cause when I use Zap <--> Zap or Zap <--> SIP there is no
> > en/decoding involved.
> 
> Asterisk is an application requiring hard realtime performance.  Pretty much 
> any telephony application is.  Running *anything* in addition to asterisk is 
> just asking for trouble.

Since X11 and other daemons  might be a problem on my main * server, I pulled
out my little testbed and fired it up.

It's a PII 266 (okay, not the fatest system) with 192MB RAM.  X is not running
and the Framebuffer has been turned off in /boot/grum/menu.lst.  I have
disabled all the servers except for sshd.  I have the latest source from CVS
HEAD as of about 30min ago.

There is no Zap card in this sytem.  The only phone on it is a SIP phone.
With it I dial in to digium's 1-700 number.  The audio is better, but still
choppy and unacceptible.

Looking at the * hardware recommendations page, this is by no means near the
smallest recorded setup, so teh system shouldn't be underpowered.

So even with X11 eliminated the sound is still bad to Digium.  I tried
another's 1700 number, and it sounded the same, so it's not something unique
to digium and me.

Would IAX/GSM be so sensitive to half-duplex that I cannot expect it to work
with my ISP only giving me 1/2 duplex service?


-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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[Asterisk-Users] Distinctive ring detection problem

[Paul Budden]

 
I am trying to get distinctive ring to work on my PSTN with no luck. I can get 2 different ring codes but it skips the context assigned... 
here is my complete zapata.conf: [channels] signalling=fxs_ks usecallerid=yes rxgain=1.0 txgain=1.0 language=en context=default usedistinctiveringdetection=yes dring1=134,0,0 dring2=137,0,0 dring1context=internal2 dring2context=default channel => 1 here is the debug output: Aug 18 10:36:20 NOTICE[1112767280]: chan_zap.c:5053 ss_thread: Got event 2 (Ring/Answered)... -- Detected ring pattern: 137,0,0 -- Distinctive Ring matched context internal2 -- Executing Wait("Zap/1-1", "1") in new stack -- Executing Answer("Zap/1-1", "") in new stack -- Executing VoiceMailMain("Zap/1-1", "") in new stack -- Playing 'vm-login' (language 'en') this should have passed on to the default context... 134,0,0 also hits the internal2 context. any ideas?
 
Paul Budden
[EMAIL PROTECTED]
EarthLink Revolves Around You.
 
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Re: [Asterisk-Users] POE

[Steven Pritchard]

On Sat, Aug 28, 2004 at 12:57:48PM -0500, Steve Maroney wrote:
> I was wondering what POE solutions are being used ?

The dumb, cheap ($30 retail) 3com PoE injectors (3CNJPSE) are working
fine for me with my snom 200 phones.

Steve
-- 
Steven Pritchard - K&S Pritchard Enterprises, Inc.
Email: [EMAIL PROTECTED] http://www.kspei.com/
Phone: (618)398-7360   Mobile: (618)567-7320
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RE: [Asterisk-Users] Are there any graphic designers on this list?

[Paterson, Mark]

 Is there an Asterisk Assistant for linux or windows?


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Shaw
Sent: Friday, August 27, 2004 10:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Are there any graphic designers on this
list?

ooohhh I'll take a crack at it! sounds like fun! :)
- Original Message -
From: "Sunrise Ltd" <[EMAIL PROTECTED]>
To: "astusr" <[EMAIL PROTECTED]>
Sent: Friday, August 27, 2004 8:47 AM
Subject: [Asterisk-Users] Are there any graphic designers on this list?


> Hi
>
> I had asked for some help with the Asterisk Assistants
>
>
http://www.voip-info.org/tiki-index.php?page=Asterisk+Assistants+for+Mac
OSX
>
> and many have offered assistance with translations which I am grateful

> for and like to say thank you again.
>
> However, there hasn't been a single response from a graphic designer 
> to offer help with a custom icon. Are there any graphic designers on 
> this list at all? If so, please take a look at the Wiki above and see 
> if you can help.
>
> thanks
> rgds
> benjk
>
> --
> Sunrise Telephone Systems Ltd
> 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan
>
> __
> GANBARE! NIPPON!
> Yahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE 
> http://mail.ganbare-nippon.yahoo.co.jp/
>
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Re: [Asterisk-Users] POE

[Kevin Day]

On Aug 28, 2004, at 1:13 PM, Michael Welter wrote:

Since the Cisco 79XX phones preceded the PoE standard, they are 
different--polarity is reversed.

IANAE, but as I understand the PoE devices, there are two types--one 
always applies -48VDC to the brown pair while the other senses (as per 
the PoE spec.) whether the device at the other end requires power.

I'm not willing to risk a $300 Cisco set, so I'm still using the wall 
wart.  Is anyone providing LAN power to 79XX phones at a reasonable 
cost?

Mike
I'm using one of these:
http://www.linksys.com/products/product.asp?grid=36&scid=47&prid=582  
(under $40)

This is a "dumb" device that's always powering the ethernet line, but 
it works. The only catch with 7940 and 7960 phones is that they expect 
the polarity to be reversed, compared to the rest of the world's POE 
specs.

So, you either have to make a funky ethernet cable (detailed in the 
Wiki) or cut the cable from the included 48V power supply and reconnect 
it backwards.

Works wonderfully.
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RE: [Asterisk-Users] asterisks and vonage

[Paterson, Mark]

Well, so I'm unable to get any inbound calls from Vonage but can call
outbound all day long. If you are able to get inbound calls working I
would appreciate it if you could share your configs via personal email. 


Rgs,
[EMAIL PROTECTED] 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Di
Martino
Sent: Saturday, August 28, 2004 10:24 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] asterisks and vonage

to start with i am new to asterisks and i am also a telcom idiot.
 
with that said i have one vonage line i would like to hook up in my soon
to be built Asterisk ippbx server.
Now with the one Vonage (with call waiting) line can i receive more one
call using an auto attendant route the call the approiate extention?
 
thanks
mike

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Re: [Asterisk-Users] POE

[Craig Guy]

We are using a Netgear FSM7326P to PoE a 7960 (with 7914 attached).

Craig

- Original Message - 
From: "Michael Welter" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Sunday, August 29, 2004 2:13 AM
Subject: Re: [Asterisk-Users] POE


> Steve Maroney wrote:
>
> > Hey guys,
> >
> > I was wondering what POE solutions are being used ? Ive done some
> > searching on google and found that PowerDsine seems to be good brand.
> >
> > Any comments,suggestions, and experiences on POE hubs other POE products
> > would be greatly appreciated.
> >
> > Thank you,
> > Steve Maroney
> >
> > ___
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> >
> >
>
> Since the Cisco 79XX phones preceded the PoE standard, they are
> different--polarity is reversed.
>
> IANAE, but as I understand the PoE devices, there are two types--one
> always applies -48VDC to the brown pair while the other senses (as per
> the PoE spec.) whether the device at the other end requires power.
>
> I'm not willing to risk a $300 Cisco set, so I'm still using the wall
> wart.  Is anyone providing LAN power to 79XX phones at a reasonable cost?
>
> Mike
>
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RE: [Asterisk-Users] SIP Provider for Reseller

[Chad Brown]

I've been in discussions with broadvoxdirect.

[EMAIL PROTECTED] has been encouraging our company to sign up. I have
yet to do so.

Chad

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, August 28, 2004 3:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Provider for Reseller

VoiceValley will do soon in NZ.
Quoting Beierlein Moritz <[EMAIL PROTECTED]>:

> Hi List,
> does somebody know a SIP Provider which offers reseller possibilities?
>
> Moritz



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Re: [Asterisk-Users] Newbie

[Michael Graves]

On Sat, 28 Aug 2004 17:44:53 -0700, Greg Broiles wrote:

>There are at least two ways to configure Vonage and Asterisk to
>connect to each other: you can connect the FXS port on your Vonaga ATA
>to an FXO port on an Asterisk box, or you can make a SIP connection to
>Vonage's server that Vonage thinks is coming from a "soft phone".
>
>If you connect the ports between the Vonage ATA and your Asterisk box,
>you will be limited to one call at a time.
>
>If you use the "soft phone" approach, another user has reported that
>multiple simultaneous calls appear to be possible.
>
>I don't know if I would want to gamble on that being true forever - my
>best guess is that it's an oversight on Vonage's part, not a choice.
>They don't seem to be very flexible or interested in the
>hobbyist/prosumer market.
>
>(Besides, if they let you start 2 or 3 or 4 simultaneous conversations
>with a single account, what's your motivation to sign up for multiple
>Vonage lines? very little, I suppose ..)
>
>Long-term, my impression is that if you want multiple simultaneous
>calls, you'll want to switch to another VoIP provider. I am in the
>same situation that you are (single Vonage line, want multiple inbound
>calls, don't want to change my phone number) and I'm slowly reaching
>the conclusion that I need to bite the bullet and dump Vonage, and my
>phone number along with it.
>
>-- 
>Greg Broiles, JD, EA
>[EMAIL PROTECTED] (Lists only. Not for confidential communications.)
>Law Office of Gregory A. Broiles
>San Jose, CA

Here, here! This is about the most sensible thing
that I've read all day. I too used Vonage for a year. I was very
pleased with all aspects of their service. However, two events
conspired to direct my migration elsewhere; I built and Asterisk server
and I realized that there were many weeks when I was entirely out of my
home office.

To the first point, well it simply didn't make too much sense to
convert a SIP line (Vonage) back to analog just to run it into an FXO
and make fresh digits. This was in Jan '04 when the soft phone account
was yet to be announced. Given the relative weakness in the small FXO
offerings from various vendors ATA>FXO seems less than ideal even
today.

On to my second observation, I travel a lot and was not in office to be
taking advantage of the great deal that "flat rate plans" posed. Which
in turn, means that they are not such a great deal at all.

Presently my Asterisk server registers with three separate providers
for outbound calling (VoipJet, NuFone, VoicePulse Connect) which each
support multiple simultaneous calls. All three are prepaid plans with
low or very low per-minute rates. I am now paying less for calls per
month than a single $40 Vonage account.

I also have an 800 number from Clearpath and a NYC DID from VoicePulse
Connect. On the road I use a Firefly soft phone on a laptop. It
registers with my own server but can also use one of my ITSP accounts
directly. 

I feel that I've just started to lever * for my personal home office
application. And apart from an unending desire to try new IP phones
(Polycom, SNOM, Pingtel & Zultys thus far) it's costing me less than
SBC or Vonage ever did.

My wife says I have a curious new hobby ;-)

Michael Graves

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262

"People said that there is no economic model for it, but there is:
the economic model of flower boxes...I put out flower boxes to raise 
my self esteem and make my house more attractive. If almost everyone 
does then the whole town becomes beautiful. The same thing can happen
with communications." - Nicolas Negroponte
 
** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704


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Re: [Asterisk-Users] Newbie

[Greg Broiles]

There are at least two ways to configure Vonage and Asterisk to
connect to each other: you can connect the FXS port on your Vonaga ATA
to an FXO port on an Asterisk box, or you can make a SIP connection to
Vonage's server that Vonage thinks is coming from a "soft phone".

If you connect the ports between the Vonage ATA and your Asterisk box,
you will be limited to one call at a time.

If you use the "soft phone" approach, another user has reported that
multiple simultaneous calls appear to be possible.

I don't know if I would want to gamble on that being true forever - my
best guess is that it's an oversight on Vonage's part, not a choice.
They don't seem to be very flexible or interested in the
hobbyist/prosumer market.

(Besides, if they let you start 2 or 3 or 4 simultaneous conversations
with a single account, what's your motivation to sign up for multiple
Vonage lines? very little, I suppose ..)

Long-term, my impression is that if you want multiple simultaneous
calls, you'll want to switch to another VoIP provider. I am in the
same situation that you are (single Vonage line, want multiple inbound
calls, don't want to change my phone number) and I'm slowly reaching
the conclusion that I need to bite the bullet and dump Vonage, and my
phone number along with it.

-- 
Greg Broiles, JD, EA
[EMAIL PROTECTED] (Lists only. Not for confidential communications.)
Law Office of Gregory A. Broiles
San Jose, CA
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Re: [Asterisk-Users] SIP Provider for Reseller

[sgup015]

VoiceValley will do soon in NZ.
Quoting Beierlein Moritz <[EMAIL PROTECTED]>:

> Hi List,
> does somebody know a SIP Provider which offers reseller possibilities?
>
> Moritz



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[Asterisk-Users] Newbie

[Michael Di Martino]

I am interested in setting up an Asterisk server as my  home phone system.

I ultimately want one 10 digit phone number, three  extensions, and an auto attendant  
My current phone service provider is Vonage, I have one line with call waiting.

My concern is will I need to add additional lines if I want the auto attendant  handle 
multilple calls.
For example a call comes in and the auto attendant sends the call to ext 1. Now while 
the person on ext 1 
Is still conversating can another call be handled by the auto attendant?
Regards,
Michael Di Martino
Director of MIS
The Telx Group
Office: 212 480 3300 X2022
Cell: 646 207 6603
[EMAIL PROTECTED]
--
Sent from my BlackBerry Wireless Handheld

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Re: [Asterisk-Users] SIP Provider for Reseller

[Roger Schreiter]

Beierlein Moritz schrieb:
Hi List,
does somebody know a SIP Provider which offers reseller possibilities?
...

Hi,
you are from Germany?
Sorry for advertising:
http://voip.planinternet.net
We are VoIP carrier and are supporting several
German VoIP providers.
Please contact us directly, if you think, our service is what you
are looking for, of if you have just further questions!
Roger.
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Re: [Asterisk-Users] POE

[Greg Boehnlein]

On Sat, 28 Aug 2004, Michael Welter wrote:

> Greg Boehnlein wrote:
> > On Sat, 28 Aug 2004, Michael Welter wrote:
> > 
> > 
> >>Since the Cisco 79XX phones preceded the PoE standard, they are 
> >>different--polarity is reversed.
> >>
> >>IANAE, but as I understand the PoE devices, there are two types--one 
> >>always applies -48VDC to the brown pair while the other senses (as per 
> >>the PoE spec.) whether the device at the other end requires power.
> >>
> >>I'm not willing to risk a $300 Cisco set, so I'm still using the wall 
> >>wart.  Is anyone providing LAN power to 79XX phones at a reasonable cost?
> > 
> > 
> > Please see:
> > 
> > http://www.voip-info.org/wiki-Cisco+POE
> > 
> > I am using 3Com POE injectors (the Dumb kind) with Cisco 7960s without a 
> > problem. You need a special cable that you can build or, you can buy a 
> > PowerDSine POE injector and get the following (part PD-PS-401-5/CSCS) 
> > which will do the same thing as the cable.
> > 
> 
> To add to my confusion, the wiki page you referenced says the 7960g 
> phones are different from the original 7960.

Not in terms of POE. No 7960 supports the 802.3af POE protocol and instead 
uses a proprietary Cisco protocol.

The cable that is referenced on those pages reverses the polarity of the 
wires, which with a dumb power injector like the 3com will allow you to 
use the Cisco. However, if you get a regular 802.3af injector, that 
actually speaks and respects the protocol, it STILL won't work with a 
Cisco.

I use the following dumb injector with my Cisco cable;
http://www.3com.com/prod/en_UK_EMEA/detail.jsp?tab=features&sku=3CNJPSE

It works.

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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Re: [Asterisk-Users] iaxtel and jitterbuffer

[Andrew Kohlsmith]

On Saturday 28 August 2004 15:24, Michael George wrote:
> I just saw a page on the wiki that mentions that running X11 or a VESA
> frame buffer can cause jittery sound.  I only have this problem with IAX2,
> but that might be cause when I use Zap <--> Zap or Zap <--> SIP there is no
> en/decoding involved.

Asterisk is an application requiring hard realtime performance.  Pretty much 
any telephony application is.  Running *anything* in addition to asterisk is 
just asking for trouble.

Actually I would be curious to see if asterisk performs better in a 
soft-realtime environment (i.e. what's actually easily possible with 
commodity PC hardware).

-A.
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Re: [Asterisk-Users] IAXy Power in Australia?

[Jeremy Bogan]

Have you tried feeding it less amps at all?
Not yet, but i'll see if I can find a power supply with less amps.
--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
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Re: [Asterisk-Users] iaxtel and jitterbuffer

[Andrew Kohlsmith]

On Saturday 28 August 2004 15:00, Michael George wrote:
> The difference between that and what I'm getting from IAX/GSM is profound,
> with GSM being intolerably poor quality.

That's odd; every single voice call coming in and out of the company I work 
for is using the GSM codec with asterisk and IAX2...  even the music on hold 
is passable.

> I have in my [general] section:

>   bandwidth=low
get rid of it; you're giving codecs below.

>   disallow=all
>   allow=gsm

>   jitterbuffer=yes
>   dropcount=10
>   maxjitterbuffer=500
>   maxexcessbuffer=100
>   minexcessbuffer=10
>   jittershrinkrate=1

My jitter settings are similar.  max 500, maxexcess 100, minexcess 50, 
dropcount=2 (10, are you *insane*?!), jittershrink of 1.  I'd slow down the 
shrink even more if I could, as even at 1 it's still noticeable.

Please note that it seems impossible to disable jitter buffer between 20040806 
CVS HEAD endpoints.  The jitterbuffer numbers in iax2 show channels look 
"live".  The numbers look right (jitbuf 0ms) between 20040806 and RC1 
(Nufone).   I haven't upgraded since then.

>   trunk=no

I found 20040806 CVS HEAD to have odd little problems with trunking too.  
Trunking between 20040806 and RC1 (again, with nufone) work fine.  I can't 
trunk to VPC at all or they can't hear me (I can hear them).

Just to make clear: I have completely disabled the jitter buffer between 
myself and Nufone and the call quality has gone up slightly.  I wasn't 
expecting this.

Regards,
Andrew
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Re: [Asterisk-Users] Disconnection From IAXTel

[Muiz Motani]

How do I go about disallowing transfers when I am running an IAX soft 
phone. Is that setting not at the * server? Obviously, I don't have control over 
the configuration of the IAXTel * server.


On 28 Aug 2004 at 19:03, you wrote:

> Disallowing an IAX
> transfer has always solved these problems for me. So you
> may want to give "notransfer=yes" a try.
> 
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Re: [Asterisk-Users] POE

[Michael Welter]

Greg Boehnlein wrote:
On Sat, 28 Aug 2004, Michael Welter wrote:

Since the Cisco 79XX phones preceded the PoE standard, they are 
different--polarity is reversed.

IANAE, but as I understand the PoE devices, there are two types--one 
always applies -48VDC to the brown pair while the other senses (as per 
the PoE spec.) whether the device at the other end requires power.

I'm not willing to risk a $300 Cisco set, so I'm still using the wall 
wart.  Is anyone providing LAN power to 79XX phones at a reasonable cost?

Please see:
http://www.voip-info.org/wiki-Cisco+POE
I am using 3Com POE injectors (the Dumb kind) with Cisco 7960s without a 
problem. You need a special cable that you can build or, you can buy a 
PowerDSine POE injector and get the following (part PD-PS-401-5/CSCS) 
which will do the same thing as the cable.

To add to my confusion, the wiki page you referenced says the 7960g 
phones are different from the original 7960.

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[Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting

[Ben Wern]

Can anyone who is using Asterisk with Broadvoice tell of their experiences with 3-way calling and call waiting? I can't get Broadvoice to respond to my question, but I understand that there is a per minute fee (3.9 c/minute?) if you go over your use allowances. 

My question is, how are 3 way and call waiting calls handled? Because Asterisk would just handle them as two different channels/calls -- does Broadvoice allow BYOD customers to have two active lines and then start charging for a third?

If so, does anyone have any configuration examples of limiting the number of sessions to a single provider?

Ben Wern





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[Asterisk-Users] Broadvoice BYOD Plans

[Ben Wern]

Can anyone who is 




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Re: [Asterisk-Users] POE

[Greg Boehnlein]

On Sat, 28 Aug 2004, Michael Welter wrote:

> Since the Cisco 79XX phones preceded the PoE standard, they are 
> different--polarity is reversed.
> 
> IANAE, but as I understand the PoE devices, there are two types--one 
> always applies -48VDC to the brown pair while the other senses (as per 
> the PoE spec.) whether the device at the other end requires power.
> 
> I'm not willing to risk a $300 Cisco set, so I'm still using the wall 
> wart.  Is anyone providing LAN power to 79XX phones at a reasonable cost?

Please see:

http://www.voip-info.org/wiki-Cisco+POE

I am using 3Com POE injectors (the Dumb kind) with Cisco 7960s without a 
problem. You need a special cable that you can build or, you can buy a 
PowerDSine POE injector and get the following (part PD-PS-401-5/CSCS) 
which will do the same thing as the cable.

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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Re: [Asterisk-Users] UK Disconnect supervision with TDM400P

[Richard Scobie]

Edward Eastman wrote:
I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN line,
loading wcfxs with OPERMODE=UK.  All's working well, except if I get an
incoming call through my bt line, and the remote party hangs up, I get
approx 20secs of the bt line hungup tone before asterisk hangs up, which
leads (if nothing else) to the well documented 20secs of beep on vm problem
:)
I have tried: busydetect=yes / busycount=7 / other busycounts /
callprogess=yes but none of these make any difference.  I have
loadzone/defaultzone=uk and country=uk in indications.conf and fxs_ks
signalling.
Try increasing your RX gain in 1db steps, until it reliably hangs up.
I had a box with X100Ps which busydetected perfectly with default gain 
settings. When they were replaced with TDM FXOs, busydetect stopped 
working and I needed 3db of RX gain added to get it working again.

Regards,
Richard
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Re: [Asterisk-Users] iaxtel and jitterbuffer

[Michael Graves]

On Sat, 28 Aug 2004 15:24:01 -0400, Michael George wrote:

>On Sat, Aug 28, 2004 at 03:00:26PM -0400, Michael George wrote:
>> On Sat, Aug 28, 2004 at 08:39:50AM -0600, Rich Adamson wrote:
>> >  
>> > I do a lot of work with companies throughout the US on network performance
>> > and we _frequently_ run into routers, switches, servers, etc, that are
>> > allowed to auto-negotiate their half vs full duplex nic interfaces. About
>> > 50% of the time, systems will get it wrong as there are no standards as
>> > to how the negotiation should be done.
>> > 
>> > A recent case this past week indicated that data flow between two servers
>> > on the same layer-2 network was around 400 kbps when it should have been
>> > able to sustain at least 80 meg.
>> > 
>> > You might double check each of your ethernet interfaces to ensure duplex
>> > settings are correct. If not at full duplex all the way through, you'll
>> > run into the strangeness you're seeing under varying traffic loads.
>
>I just saw a page on the wiki that mentions that running X11 or a VESA frame
>buffer can cause jittery sound.  I only have this problem with IAX2, but that
>might be cause when I use Zap <--> Zap or Zap <--> SIP there is no en/decoding
>involved.
>
>I am running * on my main home server, which does run X and other software.
>Perhaps that's the problem?  Maybe if I juiced it up with more RAM, might that
>help?  It's at .5GB now, but I can easily take it to 1GB.  Or, maybe a 900MHz
>Athlon still can't handle the coding with X11 running?

My understand, admittedly limited, is that the windowing system (X or
other) generates a lot of interupts. This can burden the system that is
also engaged in real-time tasks such as rpt for voip.

That said, my Asterisk server is is an AMD2500+ with 512 MB ram. I did
install the Gnome desktop with Fedora Core 1, but I don't do anything
else on the server. It runs headless. I just ssh in to tweak * as
needed.

Michael


Michael
--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262

"An ounce of pretention is worth a pound of manure."
 
** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704


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Re: [Asterisk-Users] iaxtel and jitterbuffer

[Michael George]

On Sat, Aug 28, 2004 at 03:00:26PM -0400, Michael George wrote:
> On Sat, Aug 28, 2004 at 08:39:50AM -0600, Rich Adamson wrote:
> >  
> > I do a lot of work with companies throughout the US on network performance
> > and we _frequently_ run into routers, switches, servers, etc, that are
> > allowed to auto-negotiate their half vs full duplex nic interfaces. About
> > 50% of the time, systems will get it wrong as there are no standards as
> > to how the negotiation should be done.
> > 
> > A recent case this past week indicated that data flow between two servers
> > on the same layer-2 network was around 400 kbps when it should have been
> > able to sustain at least 80 meg.
> > 
> > You might double check each of your ethernet interfaces to ensure duplex
> > settings are correct. If not at full duplex all the way through, you'll
> > run into the strangeness you're seeing under varying traffic loads.

I just saw a page on the wiki that mentions that running X11 or a VESA frame
buffer can cause jittery sound.  I only have this problem with IAX2, but that
might be cause when I use Zap <--> Zap or Zap <--> SIP there is no en/decoding
involved.

I am running * on my main home server, which does run X and other software.
Perhaps that's the problem?  Maybe if I juiced it up with more RAM, might that
help?  It's at .5GB now, but I can easily take it to 1GB.  Or, maybe a 900MHz
Athlon still can't handle the coding with X11 running?

-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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Re: [Asterisk-Users] iaxtel and jitterbuffer

[Michael George]

On Sat, Aug 28, 2004 at 08:39:50AM -0600, Rich Adamson wrote:
>  
> I do a lot of work with companies throughout the US on network performance
> and we _frequently_ run into routers, switches, servers, etc, that are
> allowed to auto-negotiate their half vs full duplex nic interfaces. About
> 50% of the time, systems will get it wrong as there are no standards as
> to how the negotiation should be done.
> 
> A recent case this past week indicated that data flow between two servers
> on the same layer-2 network was around 400 kbps when it should have been
> able to sustain at least 80 meg.
> 
> You might double check each of your ethernet interfaces to ensure duplex
> settings are correct. If not at full duplex all the way through, you'll
> run into the strangeness you're seeing under varying traffic loads.

My ISP has a half-duplex connection between me and the world-at-large.  It
doesn't seem like that should be a problem, though, because we've been running
VOIP with Multitech proprietary hardware for over two years now with little
trouble and excellent voice quality.  That was using a 9.6KBps codec.

The difference between that and what I'm getting from IAX/GSM is profound,
with GSM being intolerably poor quality.

As a test, I ran two internal * machines with IAX/GSM between them.  A
conversation would consume from 7-10KBps, varying.  Then I would call Digium's
iaxtel number and I could see traffic from 4.5-8KBps and the voice was all
choppy.  I called another person's system (knowing they had IVR, of course)
and the audio was also choppy, but when it got through the message and sent
ring tones, they sounded fine.  Then another voice message and it was choppy
again.

So I tried digium again.  This time I could see the bandwidth being consumed,
but I heard nothing on the line at all.

So I tried calling my own iaxtel number.  I could see my badwidth usage jump
to about 10KBps, as I would expect and * told me that it was playing out the
appropriate audo to the "incoming caller".  I heard nothing, however.

Does this perhaps give any further indication of what might be wrong?

I have in my [general] section:
bandwidth=low
disallow=all
allow=gsm
jitterbuffer=yes
dropcount=10
maxjitterbuffer=500
maxexcessbuffer=100 
minexcessbuffer=10
jittershrinkrate=1
trunk=no
register => me:[EMAIL PROTECTED]
tos=lowdelay

I'm working towards a client install of IAX which will be used for
inter-office VOIP, but I need to get this issue worked out or it's not
deployable.

-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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Re: [Asterisk-Users] POE

[Michael Welter]

Steve Maroney wrote:
Hey guys,
I was wondering what POE solutions are being used ? Ive done some
searching on google and found that PowerDsine seems to be good brand.
Any comments,suggestions, and experiences on POE hubs other POE products
would be greatly appreciated.
Thank you,
Steve Maroney
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Since the Cisco 79XX phones preceded the PoE standard, they are 
different--polarity is reversed.

IANAE, but as I understand the PoE devices, there are two types--one 
always applies -48VDC to the brown pair while the other senses (as per 
the PoE spec.) whether the device at the other end requires power.

I'm not willing to risk a $300 Cisco set, so I'm still using the wall 
wart.  Is anyone providing LAN power to 79XX phones at a reasonable cost?

Mike
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[Asterisk-Users] POE

[Steve Maroney]

Hey guys,

I was wondering what POE solutions are being used ? Ive done some
searching on google and found that PowerDsine seems to be good brand.

Any comments,suggestions, and experiences on POE hubs other POE products
would be greatly appreciated.

Thank you,
Steve Maroney

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Re: [Asterisk-Users] Broadvoice problem

[Ed Brady]

Marty Mastera wrote:

  
I had the same problem.  To fix it, I had to do two things

First:   I had to update to CVS head, this was as per broadvoice

  
  support.
  
  
Second:  After  updating, I had to change my sip.conf.  Originally my
sip.conf used hard coded ip addresses for broadvoice's IP servers, so

  
  I
  
  
had to change the following lines as such:
register => [mynumber]:[[EMAIL PROTECTED]
to read
register => [mynumber]:[[EMAIL PROTECTED]


  
  

Ed,

Weird things...I took your advice but executed it in stages...just like
you, I was registering with 147.135.8.129, hardcoded ip. My CVS-HEAD is
7/14/04.

The only thing I changed so far is to replace the 147.135.8.129 with
sip.broadvoice.com.  I didn't update from CVS, I also don't have SRV
lookups enabled (yet anyway).  It now registers and I can receive
inbound calls.

Does it make sense that BV may have implemented a change that would
allow registrations from a FQDN but not from a hardcoded ip? Just a
thought

Marty

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Marty,

Yeah,  I agree  it is pretty weird that Broadvoice would have made this
change.   When I called support they said that they had made some
changes to coverup up some kind of security "loop hole", however I am
not clear how this would relate to this FQDN change.  If nothing else,
it caused me to (finally) update my system.    

BTW, does the latest CVS code have better support for SRV lookups?

Ed


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Re: [Asterisk-Users] Broadvoice problem

[Ed Brady]

Marty Mastera wrote:

  
I had the same problem.  To fix it, I had to do two things

First:   I had to update to CVS head, this was as per broadvoice

  
  support.
  
  
Second:  After  updating, I had to change my sip.conf.  Originally my
sip.conf used hard coded ip addresses for broadvoice's IP servers, so

  
  I
  
  
had to change the following lines as such:
register => [mynumber]:[[EMAIL PROTECTED]
to read
register => [mynumber]:[[EMAIL PROTECTED]


  
  

Hey Ed,

When you replaced the hardcoded ip's with sip.broadvoice.com, did you
have to enabled SRV lookups to get it to work?

Thanks,

Marty

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Yes,    However I already it enabled so I didn't make any change on my
end.

Ed


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RE: [Asterisk-Users] Broadvoice problem

[Marty Mastera]

> I had the same problem.  To fix it, I had to do two things
> 
> First:   I had to update to CVS head, this was as per broadvoice
support.
> 
> Second:  After  updating, I had to change my sip.conf.  Originally my
> sip.conf used hard coded ip addresses for broadvoice's IP servers, so
I
> had to change the following lines as such:
> register => [mynumber]:[EMAIL PROTECTED]
> to read
> register => [mynumber]:[EMAIL PROTECTED]
> 


Ed,

Weird things...I took your advice but executed it in stages...just like
you, I was registering with 147.135.8.129, hardcoded ip. My CVS-HEAD is
7/14/04.

The only thing I changed so far is to replace the 147.135.8.129 with
sip.broadvoice.com.  I didn't update from CVS, I also don't have SRV
lookups enabled (yet anyway).  It now registers and I can receive
inbound calls.

Does it make sense that BV may have implemented a change that would
allow registrations from a FQDN but not from a hardcoded ip? Just a
thought

Marty

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RE: [Asterisk-Users] Broadvoice problem

[Marty Mastera]

> I had the same problem.  To fix it, I had to do two things
> 
> First:   I had to update to CVS head, this was as per broadvoice
support.
> 
> Second:  After  updating, I had to change my sip.conf.  Originally my
> sip.conf used hard coded ip addresses for broadvoice's IP servers, so
I
> had to change the following lines as such:
> register => [mynumber]:[EMAIL PROTECTED]
> to read
> register => [mynumber]:[EMAIL PROTECTED]
> 


Hey Ed,

When you replaced the hardcoded ip's with sip.broadvoice.com, did you
have to enabled SRV lookups to get it to work?

Thanks,

Marty

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Re: [Asterisk-Users] Broadvoice problem

[Ed Brady]

One other thing that I forgot to mention,  while I was hunting this 
problem down.  Broadvoice also reprovisioned my account, which caused me 
to have to get a new password.  I do not know if you will need to do 
this or not.

Ed
Ed Brady wrote:
I had the same problem.  To fix it, I had to do two things
First:   I had to update to CVS head, this was as per broadvoice support.
Second:  After  updating, I had to change my sip.conf.  Originally my 
sip.conf used hard coded ip addresses for broadvoice's IP servers, so 
I had to change the following lines as such:
register => [mynumber]:[EMAIL PROTECTED]
to read
register => [mynumber]:[EMAIL PROTECTED]

I also had to update my general settings for broadvoice
[Broadvoice]
type=peer
username=[mynumber]
fromuser=[mynumber]
secret=[secret]
context=incoming
host=147.135.8.129
fromdomain=147.135.8.129
nat=yes
canreinvite=no
dtmfmode=inband
[Broadvoice]
type=peer
username=[mynumber]
fromuser=[mynumber]
secret=[secret]
context=incoming
host=sip.broadvoice.comfromdomain=sip.broadvoice.com
nat=yes
canreinvite=no
dtmfmode=inband
After doing this, it worked like before
Hope this helps.
Ed

Russell Horn wrote:
Since Thursday evening my asterisk box has been failing to register with
broadvoice. I haven't changed any of my config files in the last week.
Can anyone suggest anything?
Asterisk is reporting:
*CLI> Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout:
Registration for '[EMAIL PROTECTED]' timed out, trying again
   -- Got SIP response 404 "Not found" back from 147.135.8.129
Urgent handler
My broadvoice config in sip.conf looks like:
[general]
context=incoming; Default context for incoming calls
externalip=82.41.201.XXX
register => 703XXX:[EMAIL PROTECTED]
port=5060   ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=0.0.0.0; address to bind to (0.0.0.0 binds 
to all)
srvlookup=no; Enable DNS SRV lookups on outbound 
calls

[Broadvoice]
type=peer
username=703XXX
fromuser=703XXX
secret=PASSWORD
host=147.135.8.129
context=flat
fromdomain=147.135.8.129
nat=no
disallow=all
allow=ulaw
canreinvite=no
dtmfmode=inband
qualify=yes


tethereal -V port 5060 reports:
Frame 11 (416 on wire, 416 captured)
   Arrival Time: Aug 28, 2004 16:17:05.72973
   Time delta from previous packet: 4.093142000 seconds
   Time relative to first packet: 20.001957000 seconds
   Frame Number: 11
   Packet Length: 416 bytes
   Capture Length: 416 bytes
Ethernet II
   Destination: 00:0d:66:23:84:54 (00:0d:66:23:84:54)
   Source: 00:40:95:35:d0:b8 (R.P.T._35:d0:b8)
   Type: IP (0x0800)
Internet Protocol, Src Addr: 
82-41-201-.cable.ubr11.edin.blueyonder.co.uk
(82.41.201.160), Dst Addr: 147.135.8.129 (147.135.8.129)
   Version: 4
   Header length: 20 bytes
   Differentiated Services Field: 0x10 (DSCP 0x04: Unknown DSCP; ECN: 
0x00)
   0001 00.. = Differentiated Services Codepoint: Unknown (0x04)
    ..0. = ECN-Capable Transport (ECT): 0
    ...0 = ECN-CE: 0
   Total Length: 402
   Identification: 0x000d
   Flags: 0x04
   .1.. = Don't fragment: Set
   ..0. = More fragments: Not set
   Fragment offset: 0
   Time to live: 64
   Protocol: UDP (0x11)
   Header checksum: 0x816c (correct)
   Source: 82-41-201.cable.ubr11.edin.blueyonder.co.uk
   Destination: 147.135.8.129 (147.135.8.129)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
   Source port: sip (5060)
   Destination port: sip (5060)
   Length: 382
   Checksum: 0x6327 (correct)
Session Initiation Protocol
   Request line: REGISTER sip:147.135.8.129 SIP/2.0
   Message Header
   Via: SIP/2.0/UDP 82.41.201.160:5060;branch=z9hG4bK30718407
   From: ;tag=as38aec91c
   To: 
   Call-ID: [EMAIL PROTECTED]
   CSeq: 106 REGISTER
   User-Agent: Asterisk PBX
   Expires: 120
   Contact: 
   Event: registration
   Content-Length: 0

Frame 12 (348 on wire, 348 captured)
   Arrival Time: Aug 28, 2004 16:17:05.995393000
   Time delta from previous packet: 0.265663000 seconds
   Time relative to first packet: 20.26762 seconds
   Frame Number: 12
   Packet Length: 348 bytes
   Capture Length: 348 bytes
Ethernet II
   Destination: 00:40:95:35:d0:b8 (R.P.T._35:d0:b8)
   Source: 00:0d:66:23:84:70 (00:0d:66:23:84:70)
   Type: IP (0x0800)
Internet Protocol, Src Addr: 147.135.8.128 (147.135.8.128), Dst Addr:
82-41-201-XXX.cable.ubr11.edin.blueyonder.co.uk (82.41.201.XXX)
   Version: 4
   Header length: 20 bytes
   Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00)
    00.. = Differentiated Services Codepoint: Default (0x00)
    ..0. = ECN-Capable Transport (ECT): 0
    ...0 = ECN-CE: 0
   Total Length: 334
   Identification: 0xf020
   Flags: 0x00
   .0.. = Don't fragment: Not set
   ..0. = More fragments: Not set
   Fragment offset: 0
   Time to live: 49
   Protocol: UDP (0x11)
   Header checksum: 0xe0ad (correct)
   Sour

Re: [Asterisk-Users] Broadvoice problem

[Ed Brady]

I had the same problem.  To fix it, I had to do two things
First:   I had to update to CVS head, this was as per broadvoice support.
Second:  After  updating, I had to change my sip.conf.  Originally my 
sip.conf used hard coded ip addresses for broadvoice's IP servers, so I 
had to change the following lines as such:
register => [mynumber]:[EMAIL PROTECTED]
to read
register => [mynumber]:[EMAIL PROTECTED]

I also had to update my general settings for broadvoice
[Broadvoice]
type=peer
username=[mynumber]
fromuser=[mynumber]
secret=[secret]
context=incoming
host=147.135.8.129
fromdomain=147.135.8.129
nat=yes
canreinvite=no
dtmfmode=inband
[Broadvoice]
type=peer
username=[mynumber]
fromuser=[mynumber]
secret=[secret]
context=incoming
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
nat=yes
canreinvite=no
dtmfmode=inband

After doing this, it worked like before
Hope this helps.
Ed

Russell Horn wrote:
Since Thursday evening my asterisk box has been failing to register with
broadvoice. I haven't changed any of my config files in the last week.
Can anyone suggest anything?
Asterisk is reporting:
*CLI> Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout:
Registration for '[EMAIL PROTECTED]' timed out, trying again
   -- Got SIP response 404 "Not found" back from 147.135.8.129
Urgent handler
My broadvoice config in sip.conf looks like:
[general]
context=incoming; Default context for incoming calls
externalip=82.41.201.XXX
register => 703XXX:[EMAIL PROTECTED]
port=5060   ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=0.0.0.0; address to bind to (0.0.0.0 binds to all)
srvlookup=no; Enable DNS SRV lookups on outbound calls
[Broadvoice]
type=peer
username=703XXX
fromuser=703XXX
secret=PASSWORD
host=147.135.8.129
context=flat
fromdomain=147.135.8.129
nat=no
disallow=all
allow=ulaw
canreinvite=no
dtmfmode=inband
qualify=yes


tethereal -V port 5060 reports:
Frame 11 (416 on wire, 416 captured)
   Arrival Time: Aug 28, 2004 16:17:05.72973
   Time delta from previous packet: 4.093142000 seconds
   Time relative to first packet: 20.001957000 seconds
   Frame Number: 11
   Packet Length: 416 bytes
   Capture Length: 416 bytes
Ethernet II
   Destination: 00:0d:66:23:84:54 (00:0d:66:23:84:54)
   Source: 00:40:95:35:d0:b8 (R.P.T._35:d0:b8)
   Type: IP (0x0800)
Internet Protocol, Src Addr: 82-41-201-.cable.ubr11.edin.blueyonder.co.uk
(82.41.201.160), Dst Addr: 147.135.8.129 (147.135.8.129)
   Version: 4
   Header length: 20 bytes
   Differentiated Services Field: 0x10 (DSCP 0x04: Unknown DSCP; ECN: 0x00)
   0001 00.. = Differentiated Services Codepoint: Unknown (0x04)
    ..0. = ECN-Capable Transport (ECT): 0
    ...0 = ECN-CE: 0
   Total Length: 402
   Identification: 0x000d
   Flags: 0x04
   .1.. = Don't fragment: Set
   ..0. = More fragments: Not set
   Fragment offset: 0
   Time to live: 64
   Protocol: UDP (0x11)
   Header checksum: 0x816c (correct)
   Source: 82-41-201.cable.ubr11.edin.blueyonder.co.uk
   Destination: 147.135.8.129 (147.135.8.129)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
   Source port: sip (5060)
   Destination port: sip (5060)
   Length: 382
   Checksum: 0x6327 (correct)
Session Initiation Protocol
   Request line: REGISTER sip:147.135.8.129 SIP/2.0
   Message Header
   Via: SIP/2.0/UDP 82.41.201.160:5060;branch=z9hG4bK30718407
   From: ;tag=as38aec91c
   To: 
   Call-ID: [EMAIL PROTECTED]
   CSeq: 106 REGISTER
   User-Agent: Asterisk PBX
   Expires: 120
   Contact: 
   Event: registration
   Content-Length: 0
Frame 12 (348 on wire, 348 captured)
   Arrival Time: Aug 28, 2004 16:17:05.995393000
   Time delta from previous packet: 0.265663000 seconds
   Time relative to first packet: 20.26762 seconds
   Frame Number: 12
   Packet Length: 348 bytes
   Capture Length: 348 bytes
Ethernet II
   Destination: 00:40:95:35:d0:b8 (R.P.T._35:d0:b8)
   Source: 00:0d:66:23:84:70 (00:0d:66:23:84:70)
   Type: IP (0x0800)
Internet Protocol, Src Addr: 147.135.8.128 (147.135.8.128), Dst Addr:
82-41-201-XXX.cable.ubr11.edin.blueyonder.co.uk (82.41.201.XXX)
   Version: 4
   Header length: 20 bytes
   Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00)
    00.. = Differentiated Services Codepoint: Default (0x00)
    ..0. = ECN-Capable Transport (ECT): 0
    ...0 = ECN-CE: 0
   Total Length: 334
   Identification: 0xf020
   Flags: 0x00
   .0.. = Don't fragment: Not set
   ..0. = More fragments: Not set
   Fragment offset: 0
   Time to live: 49
   Protocol: UDP (0x11)
   Header checksum: 0xe0ad (correct)
   Source: 147.135.8.128 (147.135.8.128)
   Destination: 82-41-201-XXX.cable.ubr11.edin.blueyonder.co.uk
(82.41.201.XXX)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
   Source port: sip (5060)
   Destination port: sip (5060)
   Length:

RE: [Asterisk-Users] Broadvoice problem

[Marty Mastera]

> Since Thursday evening my asterisk box has been failing to register
with
> broadvoice. I haven't changed any of my config files in the last week.
> 
> Can anyone suggest anything?
> 
> Asterisk is reporting:
> 
> *CLI> Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout:
> Registration for '[EMAIL PROTECTED]' timed out, trying again
> -- Got SIP response 404 "Not found" back from 147.135.8.129
> Urgent handler


I am having the exact same problem.  I have two incoming numbers with
BroadVoice and both started having this problem at the same time. I
spoke with BV support who acknowledged the 404 message and
"reprovisioned" my accounts, giving me a new password for each account.
None of this has helped.  I also tried registering with 147.135.0.129 in
case something had flip-flopped but that didn't help either.

I can still make outbound calls, and I haven't made any configuration
changes on my end.  When I called support the person spoke to alluded to
a known problem (he may have said something like "...why is this
happening?...there's been a bunch of these..."

Anyway, I was told to expect a call back once they looked into things
further.  When I hear more I will update.

Marty

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[Asterisk-Users] Broadvoice problem

[Russell Horn]

Since Thursday evening my asterisk box has been failing to register with
broadvoice. I haven't changed any of my config files in the last week.

Can anyone suggest anything?

Asterisk is reporting:

*CLI> Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout:
Registration for '[EMAIL PROTECTED]' timed out, trying again
-- Got SIP response 404 "Not found" back from 147.135.8.129
Urgent handler


My broadvoice config in sip.conf looks like:

[general]
context=incoming; Default context for incoming calls
externalip=82.41.201.XXX
register => 703XXX:[EMAIL PROTECTED]
port=5060   ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=0.0.0.0; address to bind to (0.0.0.0 binds to all)
srvlookup=no; Enable DNS SRV lookups on outbound calls

[Broadvoice]
type=peer
username=703XXX
fromuser=703XXX
secret=PASSWORD
host=147.135.8.129
context=flat
fromdomain=147.135.8.129
nat=no
disallow=all
allow=ulaw
canreinvite=no
dtmfmode=inband
qualify=yes






tethereal -V port 5060 reports:

Frame 11 (416 on wire, 416 captured)
Arrival Time: Aug 28, 2004 16:17:05.72973
Time delta from previous packet: 4.093142000 seconds
Time relative to first packet: 20.001957000 seconds
Frame Number: 11
Packet Length: 416 bytes
Capture Length: 416 bytes
Ethernet II
Destination: 00:0d:66:23:84:54 (00:0d:66:23:84:54)
Source: 00:40:95:35:d0:b8 (R.P.T._35:d0:b8)
Type: IP (0x0800)
Internet Protocol, Src Addr: 82-41-201-.cable.ubr11.edin.blueyonder.co.uk
(82.41.201.160), Dst Addr: 147.135.8.129 (147.135.8.129)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0x10 (DSCP 0x04: Unknown DSCP; ECN: 0x00)
0001 00.. = Differentiated Services Codepoint: Unknown (0x04)
 ..0. = ECN-Capable Transport (ECT): 0
 ...0 = ECN-CE: 0
Total Length: 402
Identification: 0x000d
Flags: 0x04
.1.. = Don't fragment: Set
..0. = More fragments: Not set
Fragment offset: 0
Time to live: 64
Protocol: UDP (0x11)
Header checksum: 0x816c (correct)
Source: 82-41-201.cable.ubr11.edin.blueyonder.co.uk
Destination: 147.135.8.129 (147.135.8.129)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Source port: sip (5060)
Destination port: sip (5060)
Length: 382
Checksum: 0x6327 (correct)
Session Initiation Protocol
Request line: REGISTER sip:147.135.8.129 SIP/2.0
Message Header
Via: SIP/2.0/UDP 82.41.201.160:5060;branch=z9hG4bK30718407
From: ;tag=as38aec91c
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 106 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: 
Event: registration
Content-Length: 0

Frame 12 (348 on wire, 348 captured)
Arrival Time: Aug 28, 2004 16:17:05.995393000
Time delta from previous packet: 0.265663000 seconds
Time relative to first packet: 20.26762 seconds
Frame Number: 12
Packet Length: 348 bytes
Capture Length: 348 bytes
Ethernet II
Destination: 00:40:95:35:d0:b8 (R.P.T._35:d0:b8)
Source: 00:0d:66:23:84:70 (00:0d:66:23:84:70)
Type: IP (0x0800)
Internet Protocol, Src Addr: 147.135.8.128 (147.135.8.128), Dst Addr:
82-41-201-XXX.cable.ubr11.edin.blueyonder.co.uk (82.41.201.XXX)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00)
 00.. = Differentiated Services Codepoint: Default (0x00)
 ..0. = ECN-Capable Transport (ECT): 0
 ...0 = ECN-CE: 0
Total Length: 334
Identification: 0xf020
Flags: 0x00
.0.. = Don't fragment: Not set
..0. = More fragments: Not set
Fragment offset: 0
Time to live: 49
Protocol: UDP (0x11)
Header checksum: 0xe0ad (correct)
Source: 147.135.8.128 (147.135.8.128)
Destination: 82-41-201-XXX.cable.ubr11.edin.blueyonder.co.uk
(82.41.201.XXX)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Source port: sip (5060)
Destination port: sip (5060)
Length: 314
Checksum: 0x15e1 (correct)
Session Initiation Protocol
Status line: SIP/2.0 404 Not found
Message Header
Via: SIP/2.0/UDP 82.41.201.XXX:5060;branch=z9hG4bK30718407
From: ;tag=as38aec91c
To:
;tag=SD30va299-239804385-1093709825857
Call-ID: [EMAIL PROTECTED]
CSeq: 106 REGISTER
Content-Length: 0



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Re: [Asterisk-Users] Queue Announcement not until after # accept call pressed

[Greg Boehnlein]

On Fri, 27 Aug 2004, Andrew Brown wrote:

> 
> When using the callback feature on agents I notice that when the queue calls
> one of the agents and the agent picks up the call they hear nothing until
> pressing the # to accept the call.
> 
> Only then does my announcement play back to the agent after which the call
> is immediately connected.
> 
> Is there a way to have the announcement played to the agent before they
> press # to accept. I have ackcall=yes in agent.conf
> 
> Can't find anything on the wiki.
> 
> Thanks
> 
> Andrew

Andrew,
You need my agent-preack-announce patch. This allows you to define 
a "preack=gsmfile" in your agents.conf, which will be played BEFORE the 
agent hits the "#" key to accept the call.

The patch and associated sound files etc.. are up on the bugtracker at:
http://bugs.digium.com/bug_view_page.php?bug_id=0001082

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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[Asterisk-Users] incomming call rejected using IAX2 with FWD

[Sunrise Ltd]

Storm D. J. Petersen wrote:
(B
(B>I cannot seem to accept incoming calls
(B>from FWD using IAX2. I followed the
(B>directions posted at
(B>www.fwd.pulver.com/advanced/iax
(B>I can make outgoing calls fine using
(B>IAX via FWD.  When someone calls me
(B>from FWD I get the following message:
(B>Chan_iax2.c:5251 socket_read: Reject
(B>connect attempt from 65.39.205.121
(B
(BFirst, the instructions on the FWD site are no good.
(B
(BI have just written a tool that configures FWD via IAX and
(BI, too, ran into some issues with the way the folks at FWD
(Bthink IAX should be configured.
(B
(BAnyway, for inbound you will need this in iax.conf ...
(B
(Bin section [general]
(B
(Bregister => 70707:[EMAIL PROTECTED]
(B
(Bassuming your FWD number was 70707 and your password was
(B"blah".
(B
(Bfurther ...
(B
(B[iaxfwd] ; inbound connections from FWD
(B; it's got to be 'iaxfwd' or it won't work
(Btype=user
(Bauth=rsa
(Binkeys=freeworlddialup
(Bdisallow=all
(Ballow=ulaw
(Bcontext=fwd-inbound
(B
(BYou need to make sure that you actually have FWD's public
(Bkey.
(B
(Bls -l /var/lib/asterisk/keys | grep freeworlddialup
(B
(Bshoud return something like ...
(B
(B-rw-r--r--  1 root  wheel  273 17 Jul 00:36
(Bfreeworlddialup.pub
(B
(BThe next thing to watch out for is the context. Make sure
(Byou have a matching context in your extensions.conf. For
(Bthe above example it would be something like this ...
(B
(B[fwd-inbound] ; context for incoming calls from FWD
(B;
(Bexten => 70707,1,NoOp(Incoming call for FWD #70707)
(Bexten => 70707,2,Dial(SIP/6001,60,r)
(Bexten => 70707,3,Hangup
(B
(Bagain assuming your FWD number was 70707 and you wanted
(Bthe call to come to a SIP phone with login 6001.
(B
(BIn general, you can see what's going on when you make a
(Btest call by switching on IAX debugging on the Asterisk
(Bconsole ...
(B
(B*CLI> iax2 debug
(B
(Bor if you run a server whose default is IAX2 then it would
(Bbe iax (without the 2).
(B
(Bthis will tell you when the call comes in from FWD, which
(Busername is being used and what context it is being sent
(Bto. Verify this with your configuration and make sure it
(Ball matches up.
(B
(BYou can turn the debugging off again using the command
(B"iax2 no debug" (or "iax no debug" depending on what you
(Bused to turn it on).
(B
(B
(BFinally, although you say your outbound FWD is working, I
(Brecommend you change the settings and not use FWD's
(Brecommended settings.
(B
(BIn particular, it is extremely bad style to use a dial
(Bcommand that contains the password, like so ...
(B
(BDial(IAX2/${FWDNUMBER}:[EMAIL PROTECTED] ...
(B
(BThis should only be used for trouble shooting but never
(Bever in production. It is very unfortunate that FWD is
(Bshowing this on their website leading many newbies to
(Blearn things the improper way.
(B
(BWhat you should be doing instead is put the password into
(Ban outbound peer definition and then use a reference to
(Bthat in your dial command. Here is how this looks like ...
(B
(Bin iax.conf
(B
(B[fwd-70707] ; outbound connections to FWD
(Btype=peer
(Bauth=md5
(Bsecret=blah
(Busername=70707
(Bqualify=yes
(Bhost=iax2.fwdnet.net
(Bdisallow=all
(Ballow=ulaw
(Bcontext=fwd-outbound
(Bcallerid="John Doe"<70707>
(B
(Band in extensions.conf
(B
(B[fwd-70707] ; context for outcoming calls via FWD #70707
(B;
(Bexten => _**393[1-9]X.,1,NoOp(Outgoing call to FWD
(B#${EXTEN:5})
(Bexten => _**393[1-9]X.,2,SetCIDNum(70707)
(Bexten => _**393[1-9]X.,3,SetCIDName(John Doe)
(Bexten =>
(B_**393[1-9]X.,4,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:5},60,r)
(Bexten => _**393[1-9]X.,5,Hangup
(B
(Bagain, assuming your FWD number was 70707 and you password
(Bwas "blah".
(B
(BThis is the proper way to configure outbound connections.
(B
(BOf course you can stick various bits into global
(Bvariables, but the password *doesn't* belong into the dial
(Bcommand nor into a global variable.
(B
(Bhope this helps
(Brgds
(Bbenjk
(B
(B
(B--
(BSunrise Telephone Systems Ltd
(B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan
(B
(B__
(BGANBARE! NIPPON!
(BYahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE
(Bhttp://mail.ganbare-nippon.yahoo.co.jp/
(B
(B___
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(B   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] SIP Provider for Reseller

[Beierlein Moritz]

Hi List,
does somebody know a SIP Provider which offers 
reseller possibilities?
 
Moritz
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[Asterisk-Users] UK Disconnect supervision with TDM400P

[Edward Eastman]

Hi

I know this gets covered fairly regularly, but I've had a search through the
archives and can't find anything dealing with this specifically - apologies
if I've missed it though.

I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN line,
loading wcfxs with OPERMODE=UK.  All's working well, except if I get an
incoming call through my bt line, and the remote party hangs up, I get
approx 20secs of the bt line hungup tone before asterisk hangs up, which
leads (if nothing else) to the well documented 20secs of beep on vm problem
:)

I have tried: busydetect=yes / busycount=7 / other busycounts /
callprogess=yes but none of these make any difference.  I have
loadzone/defaultzone=uk and country=uk in indications.conf and fxs_ks
signalling.

I've seen mention of the different BUSYDETECT flags in the * Makefile, but I
can't seem to find exactly what they do, or whether they're likely to
improve anything.

Does anyone have disconnect supervision with a TDM400P working well in the
UK? Can anyone provide some pointers to getting this working?

Thanks

Ed

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RE: [Asterisk-Users] Queue Announcement not until after # accept callpressed

[Edward Eastman]

This is something I'm after as well, what I have found is the following:
http://bugs.digium.com/bug_view_page.php?bug_id=0001082

http://lists.digium.com/pipermail/asterisk-dev/2004-February/003201.html

which pretty much does what I(you) want, the one problem with it is that
while the agent is listening to the pre # announcement, MOH for the queued
party stops.  Other than this I can confirm the patch works well with CVS
Head 08/03/04.

Does anyone else have anything better, or any status on the above patch?

Thanks

Ed

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew Brown
Sent: 27 August 2004 15:46
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Queue Announcement not until after # accept
callpressed


When using the callback feature on agents I notice that when the queue calls
one of the agents and the agent picks up the call they hear nothing until
pressing the # to accept the call.

Only then does my announcement play back to the agent after which the call
is immediately connected.

Is there a way to have the announcement played to the agent before they
press # to accept. I have ackcall=yes in agent.conf

Can't find anything on the wiki.

Thanks

Andrew


[exten.conf]

exten => s,1,Answer
exten => s,2,background(custom/100)

; Sales
exten => 1,1,ringing(2)
exten => 1,2,playback(custom/101)
exten => 1,3,queue(sales)

[queue.conf]

[default]
;
; Default settings for queues (currently unused)
;

[sales]


music = default

announce = sales_queue; This not played until after # pressed .. How can
i get announce to play as soon as call answered?

announce-frequency = 20

strategy = roundrobin

timeout = 15

retry = 5

maxlen = 0

member => Agent/7001
member => Agent/7005

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[Asterisk-Users] asterisks and vonage

[Michael Di Martino]

to start with i am new to asterisks and i am also a telcom idiot.
 
with that said i have one vonage line i would like to hook up in my soon to be built 
Asterisk ippbx server.
Now with the one Vonage (with call waiting) line can i receive more one call using an 
auto attendant route the call the approiate extention?
 
thanks
mike
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RE: [Asterisk-Users] incomming call rejected using IAX2 with FWD

[Bill Seddon]

Just by way of giving you some encouragement, I too run a version downloaded
on about the same day and use Telappliant successfully.  

I did get a "rejected" error until I converted one of my accounts to IAX
from the default SIP.  I don't think there is anything special about my
setup and the * PBX is behind a NAT firewall.  Also use FWD via their IAX2
service.

Bill Seddon

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: August 28, 2004 4:02 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] incomming call rejected using IAX2 with FWD

All,

I am experiencing this problem with an IAX link to the UK provider
TelAppliant.

chan_iax2.c:5251 socket_read: Rejected connect attempt from 217.14.132.162

Not sure what is causing this, however, it seems to have started sine I
downloaded Asterisk CVS-HEAD-08/19/04-19:55:53. Not sure if this is a red
herring., but, seems coincidental.

Again any idea's ?

Best regards

Steve Beaumont



-Original Message-
From:   [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Storm D. J.
Petersen
Sent:   09 January 2100 15:13
To: Asterisk-Users
Subject:[Asterisk-Users] incomming call rejected using IAX2 with FWD

Hi,
I cannot seem to accept incoming calls from FWD using IAX2.  I followed the
directions posted at www.fwd.pulver.com/advanced/iax   I can make outgoing
calls fine using IAX via FWD.  When someone calls me from FWD I get the
following message:
Chan_iax2.c:5251 socket_read: Reject connect attempt from
65.39.205.121
Any ideas?

Thanks,

S.

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[Asterisk-Users] IAX dialing indication tone (PI = 8)

[Matthew Oulton]

Hi,
 
I also have the same problem, 
from memory is this not progress indicator=8 that deals with the dialing indicator 
?
 
Anyway I am also stuck with not 
having any dial indication, has anybody got an idea.
 
MO

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Re: [Asterisk-Users] incomming call rejected using IAX2 with FWD

[Lyle Giese]

What do you have in extensions.conf for inbound from FWD?

In my default section, I have:

exten => ${FWDNUMBER},1,Goto(housemenu,s,1)

Lyle
- Original Message - 
From: "Storm D. J. Petersen" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, January 09, 2100 10:09 AM
Subject: [Asterisk-Users] incomming call rejected using IAX2 with FWD


> Hi,
> I cannot seem to accept incoming calls from FWD using IAX2.  I followed
the
> directions posted at www.fwd.pulver.com/advanced/iax   I can make outgoing
> calls fine using IAX via FWD.  When someone calls me from FWD I get the
> following message:
> Chan_iax2.c:5251 socket_read: Reject connect attempt from
> 65.39.205.121
> Any ideas?
>
> Thanks,
>
> S.
>
> ___
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RE: [Asterisk-Users] incomming call rejected using IAX2 with FWD

[steveb]

All,

I am experiencing this problem with an IAX link to the UK provider
TelAppliant.

chan_iax2.c:5251 socket_read: Rejected connect attempt from 217.14.132.162

Not sure what is causing this, however, it seems to have started sine I
downloaded Asterisk CVS-HEAD-08/19/04-19:55:53. Not sure if this is a red
herring., but, seems coincidental.

Again any idea's ?

Best regards

Steve Beaumont



-Original Message-
From:   [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Storm D. J.
Petersen
Sent:   09 January 2100 15:13
To: Asterisk-Users
Subject:[Asterisk-Users] incomming call rejected using IAX2 with FWD

Hi,
I cannot seem to accept incoming calls from FWD using IAX2.  I followed the
directions posted at www.fwd.pulver.com/advanced/iax   I can make outgoing
calls fine using IAX via FWD.  When someone calls me from FWD I get the
following message:
Chan_iax2.c:5251 socket_read: Reject connect attempt from
65.39.205.121
Any ideas?

Thanks,

S.

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Re: [Asterisk-Users] iaxtel and jitterbuffer

[Scott Laird]

On Aug 28, 2004, at 7:39 AM, Rich Adamson wrote:
I do a lot of work with companies throughout the US on network 
performance
and we _frequently_ run into routers, switches, servers, etc, that are
allowed to auto-negotiate their half vs full duplex nic interfaces. 
About
50% of the time, systems will get it wrong as there are no standards as
to how the negotiation should be done.
No standard?  Huh?  You mean besides 'NWay', which is part of 802.3?  
http://www.scyld.com/NWay.html

I've certainly seen problems, particularly with older Cisco switches 
and routers, but newer hardware seems to be pretty good.  In fact, 
autonegotiation is *required* with GigE; you aren't even allowed to 
disable it according to the specs.  Of course, that's sort of moot, 
because 1000/half isn't even slightly useful due to its 640-byte 
minimum packet size.

At my previous employer, we were having tons of duplex problems.  They 
mostly boiled down to forced duplex problems, where someone would force 
one end of a link, but leave the other end to autonegotiate.  With most 
of Cisco's hardware, forcing 100/full *completely* disables 
autonegotiation.  IMHO, it should still participate in autonegotiation, 
but only advertise the 100/full ability.  Instead, Cisco tells the 
other end "I don't negotiate."  So, if you set one end to 100/full and 
fail to force the other end, then it will try to negotiate, fail, and 
fall back to 100/half, because that's the only reasonable thing to when 
negotiation fails.  At this point, one end is 100/full and the other is 
100/half, and you're about to have trouble.  The really fun thing with 
this sort of link is that it works just fine with low traffic levels--a 
normal ping won't show problems, but it'll break when you actually try 
to use it for anything non-trivial.  Using larger ping packets helps: 
ping -s 1 totally fails if the duplex is broken anywhere along the 
link.

With newer IOS and CatOS builds, you can get around this by leaving CDP 
enabled; CDP v2 shares duplex information, and it'll log duplex 
mis-matches when both ends of the link use Cisco hardware.  I wrote a 
small CDP listener for Linux boxes and did the same thing, logging 
duplex mis-matches.  With 700 servers over 2 years, the only mismatches 
we ever found were caused by forced 100/full on the switches.

One easy fix that we found, at least for IOS switches, was to set the 
speed to auto but force the duplex.  That apparently leaves NWay 
negotiation running but only advertises full duplex as an option.  
Since nothing *ever* uses NWay to negotiate the speed of the link, this 
has the same result as forcing 100/full, but it fails in the right 
direction if you only force one end of the link.  Of course, knowing 
Cisco, this only applies for every third model of switch running 
even-numbered IOS builds.

Scott
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RE: [Asterisk-Users] G729 licenses

[Kevin Walsh]

Sergey Lapin [EMAIL PROTECTED] wrote:
> What will Asterisk do in the following case:
> 
> For example, we have 4 licenses, and have 4
> simultaneous calls, using G729.
> 
> Will asterisk allow incoming calls from peer,
> that can talk G729 and ulaw, and will it
> force it somehow to use ulaw in this case?
> 
> All phones there in LAN behind Asterisk
> prefer GSM codec, so it does transcoding.
> 
> So, what I mean is will Asterisk fall back
> to use other codecs, when it will be out of
> G729 licenses?
> 
Asterisk will not fall back to another codec if G.729 is requested
and not is available.  The general idea is that you work out how many
licenses you will need on your busiest day and then buy a couple of
extras.  Most of the time you will not be using what you have paid for.
The G.729 monopolists have made enough money out of their week's work,
so why give them more?

A better idea is to use a different codec, such as GSM, iLBC or even
ulaw (if you have the bandwidth), and ignore G.729 completely.  You
can add several choices in a list and allow the link to negotiate.
The SIP channel will try your codec choices in order of preference
but IAX2 will just pick one (usually ulaw).  Adding a few lines of code
will allow the IAX2 channel to respect the code preference order you
select.

If your supplier will only talk G.729 then switch to one that will
allow you to use a different codec.  There are plenty to choose from
these days.

-- 
   _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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Re: [Asterisk-Users] FXO interfaces used in UK?

[Rich Adamson]

> Impedance setting in the UK!?  OK, I've clearly missed something along the way.  A 
> search 
on the Wiki says something about Zcomplex impedance but I have absolutely no idea 
where or 
what this is.  If someone could point me in the right direction I'd be extremely 
grateful.
> 

That essentially means that not all telephone companies throughout 
the world are the same. The UK has its standards, the US has theirs, 
etc, etc. Your * fxo interface has to connect to the telephone 
company's pstn and use matching parameters "if" you want your fxo 
interface to work correctly. If they don't match you're likely to
incur echo, weird signaling problems, etc.

The standards often times include such things as the type of signaling
used on the pstn line, the impedence (600 ohms is the US standard) used
in designing the central office equipment and pstn cabling, etc.

The x100p fxo card is limited by the chip design on the card to US
standards, although some are using the card in the UK where different
standards exist.

The tdm04b (fxo) card uses a different set of chips, and has to be told
which international standard to use. Since * seems to default to US
standards, you will need to change that to uk standards. Might start by
reviewing the contents of /usr/src/zaptel/zaptel.conf.sample looking
for #loadzone=uk in that file. That file would have been copied to
/etc/zaptel.conf on your system when you installed asterisk.

Rich


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Re: [Asterisk-Users] iaxtel and jitterbuffer

[Rich Adamson]

> > Had this problem earlier this week - ensure 'trunk=no' in iax.conf if you're
> > using fairly current CVS code. There is something not right w/the trunking
> > that causes choppy sound. See the wiki for more info.
> 
> I am using current CVS code and I have trunk=no.  Still sounds crappy.  I need
> to check with my ISP and make sure they aren't throttling in that range.  I'm
> only getting about 4.5Kbps of throughput...  Any available codecs that can use
> that level of bandwidth?
 
I do a lot of work with companies throughout the US on network performance
and we _frequently_ run into routers, switches, servers, etc, that are
allowed to auto-negotiate their half vs full duplex nic interfaces. About
50% of the time, systems will get it wrong as there are no standards as
to how the negotiation should be done.

A recent case this past week indicated that data flow between two servers
on the same layer-2 network was around 400 kbps when it should have been
able to sustain at least 80 meg.

You might double check each of your ethernet interfaces to ensure duplex
settings are correct. If not at full duplex all the way through, you'll
run into the strangeness you're seeing under varying traffic loads.


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[Asterisk-Users] Re: Disconnection from IAXTel

[Brad Ediger]

Try an "iax2 debug" at the CLI and watch the logs as you get 
disconnected. See if it's something at the IAX protocol level. If you 
don't get any insight from that, use Ethereal on the segment to see 
what's going on.

Brad
I am using IAX soft clients (firefly, IAXComm, IAXPhone) from a Win2K 
machine on a NATed private LAN to connect to my IAXTel account. All of my 
calls to outside numbers(1-800, 1-877, etc) seem to be disconnected after 9 
seconds. Has this limit been placed on IAXTel? My call to the echo test 
number at 1-700-999-9613 seemed to run for as long as I wanted.
 

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Re: [Asterisk-Users] IAXy Power in Australia?

[Duane]

Jeremy Bogan wrote:
Maybe I have a busted IAXy, the power supply i've got is regulated and 
supports multiple voltages in different mA ratings, with the 9V DC at 
1500mA.
Have you tried feeding it less amps at all?
--
Best regards,
 Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
"I do not try to dance better than anyone else.
I only try to dance better than myself."
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[Asterisk-Users] switch statement in extensions.conf

[Michael George]

On the extensions.conf explanation page is a mention of the "switch" statement
and it refers one to the "connecting two * servers" page.  The only mention of
the switch statement there is brief and in the example.

However, the example seems to have some errors in it.  It shows a sample of
what's in the extensions.conf file, but it clearly has sections which would be
in the iax.conf file.  And there is no explanation of how the switch statement
works.

I've tried searching in the wiki but all references to switch seem to be for
phone switch banks.  Is there a more clear explanation of how that switch in
extensions.conf?  Or is the switch rarely used and a direct Dial() to another
IAX server the preferred way to deal with inter-* communications?

-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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Re: [Asterisk-Users] Queue Announcement not until after # accept call pressed

[Tim Robinson]

Tim Robinson wrote:
Andrew, I am looking for exactly the same thing, except using the 
normal Dial command via Zaptel with the c option to defer answering 
til you press the # key.  I am not a programmer but have briefly 
looked at the code in app_dial.c and I think it is doable - just need 
to find someone to help.

I deally what we need is something like the option A(x) where it plays 
file x after you answer - in this case after pressing the # key.  So 
the spec is along the lines of

B(x) where the file starts playing duing 'waiting for answer' phase to 
the called party, and terminates immediately on answer.  This should 
take another option L to loop the file.  'i.e. x is 'please press the 
# key to accept this incoming callplease press the # key to 
accepte...etc'

Any volunteers?
Rgds
Andrew Brown wrote:
When using the callback feature on agents I notice that when the 
queue calls
one of the agents and the agent picks up the call they hear nothing 
until
pressing the # to accept the call.

Only then does my announcement play back to the agent after which the 
call
is immediately connected.

Is there a way to have the announcement played to the agent before they
press # to accept. I have ackcall=yes in agent.conf
Can't find anything on the wiki.
Thanks
Andrew
[exten.conf]
exten => s,1,Answer
exten => s,2,background(custom/100)
; Sales
exten => 1,1,ringing(2)
exten => 1,2,playback(custom/101)
exten => 1,3,queue(sales)
[queue.conf]
[default]
;
; Default settings for queues (currently unused)
;
[sales]
music = default
announce = sales_queue; This not played until after # pressed .. 
How can
i get announce to play as soon as call answered?

announce-frequency = 20
strategy = roundrobin
timeout = 15
retry = 5
maxlen = 0
member => Agent/7001
member => Agent/7005
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Re: [Asterisk-Users] Queue Announcement not until after # accept call pressed

[Tim Robinson]

Tim Robinson wrote:
Andrew, I am looking for exactly the same thing, except using the 
normal Dial command via Zaptel with the c option to defer answering 
til you press the # key.  I am not a programmer but have briefly 
looked at the code in app_dial.c and I think it is doable - just need 
to find someone to help.

I deally what we need is something like the option A(x) where it plays 
file x after you answer - in this case after pressing the # key.  So 
the spec is along the lines of

B(x) where the file starts playing duing 'waiting for answer' phase to 
the called party, and terminates immediately on answer.  This should 
take another option L to loop the file.  'i.e. x is 'please press the 
# key to accept this incoming callplease press the # key to 
accepte...etc'

Any volunteers?
Rgds
Andrew Brown wrote:
When using the callback feature on agents I notice that when the 
queue calls
one of the agents and the agent picks up the call they hear nothing 
until
pressing the # to accept the call.

Only then does my announcement play back to the agent after which the 
call
is immediately connected.

Is there a way to have the announcement played to the agent before they
press # to accept. I have ackcall=yes in agent.conf
Can't find anything on the wiki.
Thanks
Andrew
[exten.conf]
exten => s,1,Answer
exten => s,2,background(custom/100)
; Sales
exten => 1,1,ringing(2)
exten => 1,2,playback(custom/101)
exten => 1,3,queue(sales)
[queue.conf]
[default]
;
; Default settings for queues (currently unused)
;
[sales]
music = default
announce = sales_queue; This not played until after # pressed .. 
How can
i get announce to play as soon as call answered?

announce-frequency = 20
strategy = roundrobin
timeout = 15
retry = 5
maxlen = 0
member => Agent/7001
member => Agent/7005
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Re: [Asterisk-Users] BT Communicator (SIP???) and Asterisk

[gARetH baBB]

On Mon, 23 Aug 2004, Robert Boardman wrote:

> Heartened by your that you have got x-lite working, I have been trying, 
> but failing to now get x-lite working, don suppose you could send me a 
> quick screen shot of you x-lite settings?

Not really, but it's not hard to get going.

Presuming you have an account [EMAIL PROTECTED], in System 
Settings->SIP Proxy->Default you put:

Display name: username
Username: username
Authorizarion User: username
Password: [password]
Domain/Realm: btinternet.com
SIP Proxy: sip.btcommunicator.bt.net
Out Bound Proxy: sip.btcommunicator.bt.net

I must get round to playing with the simple Outbound Proxy stuff in 
Asterisk CVS - though I think it's a global Outbound Proxy so not really 
useful to use in earnest.
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[Asterisk-Users] incomming call rejected using IAX2 with FWD

[Storm D. J. Petersen]

Hi,
I cannot seem to accept incoming calls from FWD using IAX2.  I followed the
directions posted at www.fwd.pulver.com/advanced/iax   I can make outgoing
calls fine using IAX via FWD.  When someone calls me from FWD I get the
following message:
Chan_iax2.c:5251 socket_read: Reject connect attempt from
65.39.205.121
Any ideas?

Thanks,

S.

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[Asterisk-Users] incomming call rejected using IAX2 with FWD

[Storm D. J. Petersen]

Hi,
I cannot seem to accept incoming calls from FWD using IAX2.  I followed the
directions posted at www.fwd.pulver.com/advanced/iax   I can make outgoing
calls fine using IAX via FWD.  When someone calls me from FWD I get the
following message:
Chan_iax2.c:5251 socket_read: Reject connect attempt from
65.39.205.121
Any ideas?

Thanks,

S.

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Re: [Asterisk-Users] IAXy Power in Australia?

[Jeremy Bogan]

Apart from voltage and current the main thing to watch out for is that
the PSU MUST be fully regulated or switched, an unregulated PSU will
give you all sorts of weird problems.
Maybe I have a busted IAXy, the power supply i've got is regulated and 
supports multiple voltages in different mA ratings, with the 9V DC at 
1500mA.

--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
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RE: [Asterisk-Users] AGI dtmf problems (with x-lite) (solved)

[Raul Elizondo (wizardteam)]

Hi Steven,

>IT is interesting you are even that far along with your AGI application
>when you haven't even figured out your mail client.

Do you mean my email client? or my voicemail?  Voicemail is working fine on
digim extensions, i even changed the language to spanish (btw, there are lot
of mistakes in the manner to say numbers).  The problem comes when i try to
use an agi with a x-lite client (softphone), it works fine with dtmf over
voicemail and over menu selection, but it does not recognice dtmf for agi's.


>What was the connection to the message about voice recognition that you
>replied to?

I am not using voice recognition, i m trying to use dtmf recognition with
agi's, but somehow, asterisk does not listen the x-lite only with agi's.
The rest of the digium fxs extensions works fine with my agi.

>If you wish to be lazy, go ALL OUT, and learn that you can start a new
>message with a new thread by clicking on the address of the mailing list
>in the headers section. It will bring up a new message with no content
>in it. It is a lot easier than deleting all the text in the old message.

huh?

>BTW, did you ANSWER() the call before going to AGI?

Actually no, and that was my problem.  When i begun to work with agi in tcl,
i was figuring out how it works from a perl script and i got some problems,
then, i used Answer() first times, i thought it was the error, when i
debuged my script, it worked fine with out the Answer() for digium
extensions.

Now is working after i added Answer() on the extensions.conf for that.
Thanks for the hint! ;)

-=Raul=-

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Re: [Asterisk-Users] chan_sccp2 & 7960 -- documentation and examplerequest.

[Craig Guy]

I have now today also configured a 7960 to work with asterisk via chan_sccp.
I have only used SCCP firmware 5.0 (5) with the 7960 and I gotta say that I
much prefer the SIP 7.2 firmware.  The real reason at this stage for going
SCCP is for support of the 7914 expansion module.  This phone will be the
reception phone and that means that we're after support for the 'progress'
lights on the expansion board.

I can't comment on the current support for the 7914 because I upgraded the
phone to firmware 5.0 (5) before I discovered that the 7914 firmware has to
be upgraded first.  And a version 5 firmware can't be downgraded :(

For anyone else trying to configure a 7960 for SCCP and Asterisk you will
need on the TFTP server the OS79XX.txt, XMLDefault.cnf.xml and
SEP.cnf.xml.  Documentation on the contents of SEP.cnf.xml was
very hard to track down.  I have added a sample copy of this file to the
wiki - http://www.voip-info.org/tiki-index.php?page=SCCP-HOWTO2

One other thing - I am running Fedora Core 2 and had to modify chan_sccp.c
and sccp_device.c, changing 'pthread_create' to 'ast_pthread_create' in
order for the compiled module to load in Asterisk.

Craig
- Original Message - 
From: "Matthew Boehm" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Thursday, August 26, 2004 9:56 PM
Subject: Re: [Asterisk-Users] chan_sccp2 & 7960 -- documentation and
examplerequest.


> I'm guessing that chan_sccp2 is the same one I am using which was
downloaded
> from http://chan-sccp.sourceforge.net ?
>
> If it is, we are in luck. I have 2 Cisco 7960G's all running just fine
with
> this new module.
>
> 1 phone has 1 line on it and the other has 2 lines on it. I am able to
dial
> all 3 lines from both phones and can call POTS numbers as well as all our
> SIP phones.
>
> I've got a custom services page running (hosted by someone else) and are
in
> the process of getting a custom directory working. Speeddials also work
> great displaying my custom name on the LCD screen.
>
> Have not yet figured out how to put call on hold but can transfer calls to
> another extension and can park calls.
>
> I also applied a patch to the module allowing a multi-line phone to answer
> an incomming call on any line. The non-patched version of CVS does not do
> this.
>
> Certain softkey menus (the 4 buttons along the bottom of the LCD) do not
> seem to be visible in the correct mode. For example, I can see a 'Hold'
> option visible right now even though there is no active call. And the
'Hold'
> button dissapears when a call is active; so I can't press 'Hold' or
> 'Transfer'. But I can use *'s internal #EXT to transfer.
>
> Haven't tested intercom abilities yet.
>
> Make sure that in your modules.conf you have a "noload => chan_skinny.so"
> otherwise the 7960's will continue to use that module instead of the new
> chan_sccp.
>
> sccp.conf
> -
> [general]
> keepalive = 60  ; How often the SCCP device does a keepalive ping
> context = default   ; default context that will be used if nothing
else
> is specified for
> dateFormat = D-M-Y  ; M-D-Y in any order (5 chars max)
> bindaddr = 1.2.3.4 ; replace 1.2.3.4 with the ip address of the asterisk
> box.
> port = 2000 ; listen on port 2000 (Skinny, default)
>
> [SEP000F3442E4A7]
> description = Matthew's 7960; A description, may be up to 16
charecters
> long. Used by * in 'sccp show'
> type = 7960 ; The model type needs to be defined so we
> know how to set it up.
> context   = matthew ; default context for outgoing calls.
> tzoffset  = -6  ; Timezone offset from GMT
> autologin = 1001
> speeddial = 4,John Doe
> speeddial = 7,Jack Trades
> speeddial = 8,Neverwinter Nights
>
>
> [SEP000F3442E199]
> description = Jack's 7960
> type = 7960
> context = matthew
> tzoffset = -6
> autologin = 1002,1003
> speeddial = 4,John Doe
> speeddial = 7,Jack Trades
> speeddial = 8,Neverwinter Nights
>
> [1001]
> id  = 1001  ; Id is a number that is dialed to login to the
line
> with.
> pin = 1234  ; The pin number needed to log into the device.
If
> pin is missing, anyone can log into it
> label   = 1001  ; The text to display on the display (on 7960)
> description = 1001  ; The text to display on the screen (on the 7910)
> context = matthew   ; Context outgoign calls are in.
> callwaiting = 1 ; If set to 1, call waiting will work.
> ;mailbox = 4; Check if this mailbox has any mail, and
if
> so, show the Message Waiting Indicator.
> callerid= "Theo" <1001> ; CallerId to use on outgoing calls from
> this line.
>
> [1002]
> id  = 1002
> pin = 1234
> label   = 1002
> description = 1002
> context = matthew
> callwaiting = 1
> ;mailbox = 5
> callerid= "Richard" <1002>
>
> [1003]
> id  = 1003
> pin = 1234
> label   = 1003
> description = 1

[Asterisk-Users] Disconnection From IAXTel

[Sunrise Ltd]

Muiz Motani wrote:
(B
(B>I am using IAX soft clients [snip]
(B>on a NATed private LAN [snip]
(B>[snip] disconnected after 9 seconds.
(B
(B8-10 seconds is roughly the time it takes for an IAX
(Btransfer to kick in. I have seen similar cases not with
(BIAXtel but other IAX connections where end points were
(Bunable to address each other directly and the transfer
(Bwould then cut off the connection. Disallowing an IAX
(Btransfer has always solved these problems for me. So you
(Bmay want to give "notransfer=yes" a try.
(B
(Brgds
(Bbenjk
(B
(B
(B--
(BSunrise Telephone Systems Ltd
(B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan
(B
(B__
(BGANBARE! NIPPON!
(BYahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE
(Bhttp://mail.ganbare-nippon.yahoo.co.jp/
(B
(B___
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[Asterisk-Users] G729 licenses

[Sergey Lapin]

Hi, all!!!
What will Asterisk do in the following case:
For example, we have 4 licenses, and have 4
simultaneous calls, using G729.
Will asterisk allow incoming calls from peer,
that can talk G729 and ulaw, and will it
force it somehow to use ulaw in this case?
All phones there in LAN behind Asterisk
prefer GSM codec, so it does transcoding.
So, what I mean is will Asterisk fall back
to use other codecs, when it will be out of
G729 licenses?
S.
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RE: [Asterisk-Users] IAXy Power in Australia?

[Roy Eddleston]

Jeremy

Apart from voltage and current the main thing to watch out for is that
the PSU MUST be fully regulated or switched, an unregulated PSU will
give you all sorts of weird problems.

Before I opted for 240V to 120V mains converters (they were on offer) to
use with the supplied US PSU's, I too had problems finding a regulated
1500mA PSU in the UK. The largest I could find was 900-1000mA but I did
find switched PSU's at 1500mA were easy enough to find and no more
expensive, around £20 in the UK.

Anyway, as I say the 800mA seems to be fine so far.

Cheers!

Roy...

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jeremy Bogan
> Sent: 28 August 2004 09:53
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] IAXy Power in Australia?
> 
> Hi Roy,
> 
> > I am in the UK but the IAXy's (x5) I just received from the states
have
> > 120VAC/9VDC 800mA supplies, I questioned this, but the supplier
stated
> > that this was what Digium are shipping with the IAXy.
> 
> Thanks for the info. I bought a power supply that does 1500mA at 9V
DC,
> but the IAXy doesn't power on. The end connector that goes into the
> IAXy sticks out a bit, so i'm wondering if that's the problem (but it
> does make a connection ok), one of the reasons i've been trying to
find
> a power supply within their specs.
> 
> --
> jeremy bogan[ [EMAIL PROTECTED] ]
> segment publishing - design.develop.host
> 
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Re: [Asterisk-Users] IAXy Power in Australia?

[Jeremy Bogan]

Hi Roy,
I am in the UK but the IAXy's (x5) I just received from the states have
120VAC/9VDC 800mA supplies, I questioned this, but the supplier stated
that this was what Digium are shipping with the IAXy.
Thanks for the info. I bought a power supply that does 1500mA at 9V DC, 
but the IAXy doesn't power on. The end connector that goes into the 
IAXy sticks out a bit, so i'm wondering if that's the problem (but it 
does make a connection ok), one of the reasons i've been trying to find 
a power supply within their specs.

--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
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RE: [Asterisk-Users] IAXy Power in Australia?

[Roy Eddleston]

Jeremy

I am in the UK but the IAXy's (x5) I just received from the states have
120VAC/9VDC 800mA supplies, I questioned this, but the supplier stated
that this was what Digium are shipping with the IAXy.

Digium state 1500mA on their docs but DigitNetworks state 1200mA on
theirs, but ship the 800mA

So far they are operating fine with UK spec phones, but time will tell
if the 800mA is enough, I am going to measure the current drain when I
get a few minutes.

Cheers!

Roy...

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jeremy Bogan
> Sent: 28 August 2004 08:08
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] IAXy Power in Australia?
> 
> > http://www.dse.com.au/cgi-bin/dse.storefront/
> > 412ff6210573f994273fc0a87f9
> > c0726/Product/View/M9917
> 
> I emailed Dick Smith with the requirements but none of the power
> supplies they have can do the 1500mA.
> 
> --
> jeremy bogan[ [EMAIL PROTECTED] ]
> segment publishing - design.develop.host
> 
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RE: [Asterisk-Users] IAXy Power in Australia?

[Florian Overkamp]

BTW, 

> -Original Message-
> I emailed Dick Smith with the requirements but none of the 
> power supplies they have can do the 1500mA.

I've got an IAXy running on a 1100mA powersup I found packaged with my
Lucent WLAN basestation. Be a little flexible and just try it.

Only thing: If the power is unsufficient, you'll have heavily distorted
dialtones and won't be able to make calls, so you'll know pretty soon when
the power is no good :)

Florian

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[Asterisk-Users] ISDN BRI card exepriences in UK

[David Gurr]

Looking for folks experiences with ISDN BRI cards in the UK ... what's good
and what's bad and any gotchas.

Thx

--
David Gurr
Congruity Ltd.
Hemel Hempstead
UK

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Re: [Asterisk-Users] IAXy Power in Australia?

[Gary]

the the real power requires and read the pages...

like when can you belive any sales gimp who works for dse ??

its in their catalogue, you asked, goto a store and find it !!

also check jaycar and altronics they are good bets as well.

Gary

On Sat, 28 Aug 2004 17:08:18 +1000, Jeremy Bogan wrote:

>> http://www.dse.com.au/cgi-bin/dse.storefront/ 
>> 412ff6210573f994273fc0a87f9
>> c0726/Product/View/M9917
>
>I emailed Dick Smith with the requirements but none of the power  
>supplies they have can do the 1500mA.
>
>-- 
>jeremy bogan[ [EMAIL PROTECTED] ]
>segment publishing - design.develop.host
>
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.


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[Asterisk-Users] FXO probs in Aus. Should I give up?

[Jamie Carl]

Hey all,

I've been trying to get my X101P working again as of late (it used to
work great) and before I decide to trash the card I thought I'd post up
my symptoms to see if anyone has any ideas.

My old working config was basically 1 channel running fxsks signalling.
It was working great with no echo, busy detect worked well and I was
very impressed considering this is all off and Australian PSTN line for
which the X101P is not certified.  (hhh).  So one day I update the
zaptel drivers (not sure if this caused it however), and now it cannot
go off-hook on it's own.

Outbound Symptoms are:

Placing a call from a SIP softphone, * will cease the zap channel and
look like it's working, but no audio can be heard (ring tone, etc) on
the softphone. Now, if I go off-hook on a POTS phone running parallel to
the X101p suddenly everything comes to life.  If I go off-hook on the
parallel phone before the X101p tries to dial, everything works fine.
But on it's own, it's a no go.

Inbound Symptoms:

The zap channel detects ring, ceases the channel and begins normal call
flow and in my test setup going straight to voicemail.  The caller can
hear the call is answered but again, no audio.  Going off-hook again on
the parallel phone kicks everything back into life.

Now here's the kicker.  I have an old frame-relay voice switch I
'borrowed' from an ex-employer and have configured some slots for FXS to
run back-to-back with the X101p.  It works first time, every time.  Only
difference I can think of between them is that the voice switch is from
the US and therefore uses US tones, etc.  ??

I have tried both Loopstart and Koolstart signalling.  Groundstart will
not load when I use 'ztcfg' for some reason.  So is there something I'm
missing.  This used to work fine.  Has something changed in the zaptel
driver? Are there any undocumented settings I can tweak to possible get
this working again?

I'm about to chuck the card and go for a SIP or MGCP gateway but if I
can not spend the cash, I will.  Anyone with ideas?

Thanks heaps.

Regards,

Jamie Carl
Chief 'Stuff' Officer
J-Code
Web:http://www.j-code.net
Email:  [EMAIL PROTECTED]



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Re: [Asterisk-Users] IAXy Power in Australia?

[Jeremy Bogan]

http://www.dse.com.au/cgi-bin/dse.storefront/ 
412ff6210573f994273fc0a87f9
c0726/Product/View/M9917
I emailed Dick Smith with the requirements but none of the power  
supplies they have can do the 1500mA.

--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
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