Re: [Asterisk-Users] super high bandwidth codec

[Tzafrir Cohen]

On Tue, Jul 26, 2005 at 10:42:35PM -0700, Andrew C. Brown wrote:
> > 
> > A recent blog entry indicated that GIPS was issuing licenses for its
> > technology from a mere $50k for "unlimited licenses" with respect to an
> > agreement with Microsoft. I don't have a huge concern about bandwidth
> > limits. If I could get better quality than G.711 in the same bandwidhth
> > that would be great.
> > 
> > However, since I'm using IAX2 based DIDs and termination would it
> > really matter? That is, if the ITSPs are connection to the PSTN via TDM
> > interconnects wouldn't any calls be limited to G.711 quality anyway?
> 
> IAX2 is a protocol, not a codec, so has little impact on sampling
> quality. But the second assumption is correct. If you are going to PSTN
> at any point in the chain, you are back to 8kHz sample rate and that
> extra spectrum you put over iSAC or whatever is tossed out the window.

And also when you use MeetMe, right?

-- 
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http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
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[Asterisk-Users] re: switch statement in dialplan

[Yair Hakak]

hi all,
 is there a switch statement in the dialplan? or do i have to
daisy-chain GoToIf statements? i don't see a switch statement on the
wiki, but you never know...

thanks
 yair
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Re: [Asterisk-Dev] RE: [Asterisk-Users] Zaptel update, Asterisk 1.2 janitor projects

[Dave Cotton]

On Tue, 2005-07-26 at 20:11 +0200, Michiel van Baak wrote:
> On 18:30, Tue 26 Jul 05, Dave Cotton wrote:
> > On Tue, 2005-07-26 at 18:11 +0300, Tzafrir Cohen wrote:
> > 
> > > I suppose you refer to:
> > > http://lists.digium.com/pipermail/asterisk-cvs/2005-July/007125.html
> > > 
> > > How do I track only the changes to the stable branch? For a user of
> > > Stable most of the messages on the CVS list are rather irrelevant.
> > > 
> > > There seems to be a '  Tag: v1-0' in the message but it is in the body.
> > 
> > Can't mutt filter on the body contents?
> >  
> 
> it can.

So the original problem can be solved :).


-- 
Dave Cotton <[EMAIL PROTECTED]>

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[Asterisk-Users] Supervised transfer over SIP to outside POTS lines

[Damon Brown]

Hello all, 
I am trying to complete my dial plan and have come up with an
interesting situation.  My configuration is set up with 12 xlite SIP
clients on SUSE linux workstation.  They are calling out via 10 analog
lines, TE110P->rhino 24 fxo.  

It all works and dials out great ... but ... this unit was brought in to
handle the "global" office.  So the help desk support on the Suse
machines need to transfer a call to an available local rep in another
state.  I thought this was possible  until I realized the "transfer"
only works on xPRO, which isn't available for linux.

So I cant rely on SIP to handle this, I set up my extensions.conf have
transfers, ie: 

[sip-exten]
exten => 1001,1,Dial(SIP/1001,20,Trt)
exten => 1001,2,Hangup

And features.conf is :

[featuremap]
blindxfer => *1; Blind transfer
;disconnect => *0   ; Disconnect
;automon => *1  ; One Touch Record
atxfer => *2   ; Attended transfer


OK each analog phone has three way calling on it ... can I set up a
"flash" command?  How would that be done???.  

Thanks so much!!
D



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Re: [Asterisk-Users] super high bandwidth codec

[Andrew C. Brown]

Michael Graves wrote:
> On Tue, 26 Jul 2005 18:50:11 -0700, Andrew C. Brown wrote:
> 
> 
>>Brian Capouch wrote:
>>
>>>Geoff Manning wrote:
>>>
>>>
>>>Skype uses wideband-ilbc.
>>>
>>
I don't think thats right. I think it just uses iLBC over it's own
proprietary Voip protocol.
http://www.skype.com/help/faq/technical.html
How much bandwidth does Skype use while I'm in a call?
Skype automatically selects the best codec depending on the
connection
between yourself and the person you are calling. On average, Skype uses
between 3-16 kilobytes/sec depending on bandwidth available for other
party,
network conditions in between, callers CPU performance, etc.
>>>
>>>
>>>I don't think that's correct.
>>>
>>>I don't have the link to the Columbia paper where they tried (with only
>>>mixed success) to figure out what all nefarious stuff Skype does
>>>(hijacking port 80 being the most pernicious) but I'm pretty sure they
>>>have figured out that if possible, it will use the (proprietary)
>>>wideband version of iLBC.
>>>
>>
>>FYI: One can find the columbia paper link by going to the VoIP wiki's
>>Skype page.
>>
>>According to GIPS datasheets, GIPS offers two proprietary wideband
>>codecs. iPCM-wb and iSAC. Both have 16kHZ sample rate* which is double
>>the 8kHz of PSTN, iLBC and most of the other codecs, hence the
>>relatively wonderous sound quality which I, among others, covet for
>>Asterisk.
>>
>>The channel bit rate is respectively (it varies dynamically)
>>iLBC (free)   13-15kbps
>>iSAC ($)  10-30kbps
>>iPCM-wb ($)   80kbps
>>
>>iPCM-wb doesn't seem to offer any outright fidelity advantage over iSAC
>>since they are the same sample rate. I presume all those extra bits are
>>redundancy to make the quality more robust.
>>
>>* Ref: http://www.globalipsound.com/solutions/solutions_Codecs.php
>>
> 
> 
> A recent blog entry indicated that GIPS was issuing licenses for its
> technology from a mere $50k for "unlimited licenses" with respect to an
> agreement with Microsoft. I don't have a huge concern about bandwidth
> limits. If I could get better quality than G.711 in the same bandwidhth
> that would be great.
> 
> However, since I'm using IAX2 based DIDs and termination would it
> really matter? That is, if the ITSPs are connection to the PSTN via TDM
> interconnects wouldn't any calls be limited to G.711 quality anyway?

IAX2 is a protocol, not a codec, so has little impact on sampling
quality. But the second assumption is correct. If you are going to PSTN
at any point in the chain, you are back to 8kHz sample rate and that
extra spectrum you put over iSAC or whatever is tossed out the window.

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Re: [Asterisk-Users] mpg123 - two processes

[El Flynn]

Ronald_Wiplinger wrote:

Brian West wrote:

If you use mp3nb from the sample configs you will have exactly 1 per  
class.




Great!
Where can I read more details about it?
(musiconhold.conf)




in musiconhold.conf:

[classes]
default => mp3nb:/var/lib/asterisk/mohmp3

Flynn

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Re: [Asterisk-Users] MozIAX phone on FC4/Firefox 1.6

[Jean-Denis Girard]

Jason Walker a écrit :

Has anyone had any luck with MozIAX (Mozphone) on FC4 with Firefox
1.6? jslib and moziax install through Firefox correctly - at least
that is the message I get.


I guess this is Firefox-1.0.6, or I must have been sleeping to long ;)



I am able to log into the IAX Phone on Windows, however I get an error stating:

--
FATAL ERROR: no connection to "network_client".

MozPhone will stop now!
--

I am able to connect with the same connection settings on a Windows
2000 PC running Firefox 1.6 with MozPhone. Calls are successful in
that environment.

Any ideas? 


network_client uses port 9998, maybe there is already a service running 
on that port which prevents network_client from starting. Could you 
start firefox -phone from a console ? You should see some messages like 
the following:


Setting up network input
ProxIAX network_client waiting on port 9998
Client connection...
Greeting client with  now...
Connecting|Looking up asterisk.sysnux.pf

gui_hide_doing
Connecting...|Connecting to asterisk.sysnux.pf
gui_hide_doing
Logging in|Logging in astman '...
gui_hide_doing
device,0,/dev/dsp,63
.

I use Mozphone all day long with Firefox-1.0.6 on Mandriva-10.2.


Thanks for using MozPhone,
--
Jean-Denis Girard

SysNux  Systèmes Linux en Polynésie française
http://www.sysnux.pf/   Tél: +689 483 527 / GSM: +689 797 527
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Re: [Asterisk-Users] super high bandwidth codec

[Tzafrir Cohen]

On Tue, Jul 26, 2005 at 10:18:35PM -0500, Michael Graves wrote:

> A recent blog entry indicated that GIPS was issuing licenses for its
> technology from a mere $50k for "unlimited licenses" with respect to an
> agreement with Microsoft. I don't have a huge concern about bandwidth
> limits. If I could get better quality than G.711 in the same bandwidhth
> that would be great.

Try speex with a license price of 0$ per year.

> 
> However, since I'm using IAX2 based DIDs and termination would it
> really matter? That is, if the ITSPs are connection to the PSTN via TDM
> interconnects wouldn't any calls be limited to G.711 quality anyway?

Please, not another patented algorithm. We have enough troubles from
those already.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] Fritz PCI card in ptp mode with chan_misdn

[Craig Guy]

The mISDN Fritz! driver supports PTP mode.  In your startup script where you
load the mISDN drivers call the fritz driver thusly:

modprobe avmfritz protocol=34

Bit 5 sets PTP mode, bits 3-0 set the D-channel protocol ID (set bit one for
DSS1).

Craig

- Original Message - 
From: "Michiel van Baak" <[EMAIL PROTECTED]>
To: 
Sent: Wednesday, July 27, 2005 12:35 AM
Subject: Re: [Asterisk-Users] Fritz PCI card in ptp mode with chan_misdn


> On 18:21, Mon 25 Jul 05, Johann Steinwendtner wrote:
> > Hello !
> >
> > I would like to get working a Fritz PCI card using chan_misdn
> > operating in ptp mode.
>
> As far as I know the fritz cards do not support ptp mode.
> We tried all the possible config file options with chan_capi
> and in the end we trashed them and installed a junghanns
> QuadBRI.
>
> If you get it working in ptp mode, please tell me how you
> did it.
> -- 
> Michiel van Baak
> http://michiel.vanbaak.info
> [EMAIL PROTECTED]
> GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D
>
> "Why is it drug addicts and computer afficionados are both called users?"
> ___
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Re: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem

[Tzafrir Cohen]

Hi

On Tue, Jul 26, 2005 at 09:09:27PM +0200, Walid Azab wrote:
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Neil Cherry
> Sent: Tuesday, July 26, 2005 6:00 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem
> 
> > Walid Azab wrote:
> > 
> > > Thanks to all of you guys. I managed to fix it. It turned out to be 
> > > that the ZIP file has to be extracted inside the TFTP root not outside 
> > > then copied to the TFTP root. It is working now.
> > 
> > Walid, you should be able to unzip it anywhere and copy it into the
> > directory. It sounds like a permissions problem when you copied it. In the
> > future just make sure that files copied into the tftp directory have at
> > least read permission for everyone (user, group and other). Since it's
> > working now you don't need to fool with it.
> > Just information for the future.
>
> Yep you are right , I usually do a chmod 777.

755 would have been enough. 777 allows everyone who happen to get access
to your network to change that firmware using simply tftp. Anyone feels
like trojaning cisco phones?

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] IAX over HTTP

[asterisk]

On Fri, Jul 22, 2005 at 01:48:09PM +0200, Jerry Glomph Black wrote:
> 
> [Have had good results with UDP-based secure tunnel transport of IAX 
> traffic (CIPE and OpenVPN)]

Have you tried making it use port 53? Many otherwise restrictive
firewalls will obligingly let that through.
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Re: [Asterisk-Users] mpg123 - two processes

[Ronald_Wiplinger]

Brian West wrote:

If you use mp3nb from the sample configs you will have exactly 1 per  
class.



Great!
Where can I read more details about it?
(musiconhold.conf)


bye

Ronald



/b

On Jul 26, 2005, at 9:38 PM, MF Hulber wrote:


Yes, I always have two.

MARK.

Billy Dunn wrote:


Does everyone have two processes running for mpg123?  I always  have 
them when I'm running an idle Asterisk box.  No calls going  in or 
out and nothing off hook.  Is this normal?  Thanks!


5008 ?S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f  
4096 fpm-calm-ri

5015 ?S  0:00 /usr/sbin/asterisk
5061 ?S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f  
4096 fpm-calm-ri


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RE: [Asterisk-Users] To anyone seeking 911 Service Providers"stayaway!!!"

[Rusty Shackleford]

> 
> The "mandatory" part that is due right now is the section of 
> the law that deals with "informing" the voip user of their 
> current E911 status. That part is not in a comment phase.

Actually, that part takes effect July 29. The access requirements will
probably hit at the end of November.

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Re: [Asterisk-Users] super high bandwidth codec

[Michael Graves]

On Tue, 26 Jul 2005 18:50:11 -0700, Andrew C. Brown wrote:

>Brian Capouch wrote:
>> Geoff Manning wrote:
>> 
>> Skype uses wideband-ilbc.
>>
>
>>>
>>> I don't think thats right. I think it just uses iLBC over it's own
>>> proprietary Voip protocol.
>>> http://www.skype.com/help/faq/technical.html
>>> How much bandwidth does Skype use while I'm in a call?
>>> Skype automatically selects the best codec depending on the
>>> connection
>>> between yourself and the person you are calling. On average, Skype uses
>>> between 3-16 kilobytes/sec depending on bandwidth available for other
>>> party,
>>> network conditions in between, callers CPU performance, etc.
>> 
>> 
>> I don't think that's correct.
>> 
>> I don't have the link to the Columbia paper where they tried (with only
>> mixed success) to figure out what all nefarious stuff Skype does
>> (hijacking port 80 being the most pernicious) but I'm pretty sure they
>> have figured out that if possible, it will use the (proprietary)
>> wideband version of iLBC.
>> 
>
>FYI: One can find the columbia paper link by going to the VoIP wiki's
>Skype page.
>
>According to GIPS datasheets, GIPS offers two proprietary wideband
>codecs. iPCM-wb and iSAC. Both have 16kHZ sample rate* which is double
>the 8kHz of PSTN, iLBC and most of the other codecs, hence the
>relatively wonderous sound quality which I, among others, covet for
>Asterisk.
>
>The channel bit rate is respectively (it varies dynamically)
>iLBC (free)13-15kbps
>iSAC ($)   10-30kbps
>iPCM-wb ($)80kbps
>
>iPCM-wb doesn't seem to offer any outright fidelity advantage over iSAC
>since they are the same sample rate. I presume all those extra bits are
>redundancy to make the quality more robust.
>
>* Ref: http://www.globalipsound.com/solutions/solutions_Codecs.php
>

A recent blog entry indicated that GIPS was issuing licenses for its
technology from a mere $50k for "unlimited licenses" with respect to an
agreement with Microsoft. I don't have a huge concern about bandwidth
limits. If I could get better quality than G.711 in the same bandwidhth
that would be great.

However, since I'm using IAX2 based DIDs and termination would it
really matter? That is, if the ITSPs are connection to the PSTN via TDM
interconnects wouldn't any calls be limited to G.711 quality anyway?

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



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RE: [Asterisk-Users] ASTCC: different incriments

[Rusty Shackleford]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Ronald Wiplinger
> Sent: Tuesday, July 26, 2005 4:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] ASTCC: different incriments
> 
> 
> How can I fulfill that?
> 
> *Billing Increments*
> Continental USA: six (6) second increments.
> International: thirty (30) seconds minimum and six (6) 
> seconds thereafter.
> Mexico: sixty (60) seconds minimum and six (6) seconds thereafter.

The billing increment is set in the "brands" table. When you create
cards, this value is copied into the "inc" column in the "cards" table.
(I'll spare us the rant on normalization here...)

The per call minimum is set in the "includedseconds" column, in the
"routes" table. This value, along with the value of the "connectcost"
column for a given record (route) is used to compute the cost of the
call.

So, in theory, you set all your cards for 6 second increments, and you
set your routes to 6, 30, or 60 "includedseconds". 

That's the theory, but the stock ASTCC code has a bug in the way it
makes this computation. Darren has reopened the bug report. 

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[Asterisk-Users] What does pbx-wilcalu.so do and why does it keep crashing my * box?

[Mark Phillips]

I downloaded the latest CVS a few days ago. It all compiled nicely on my 
new AAH platform. However, it won't start up.


Investigation of my log files produces this;

Jul 26 22:59:18 VERBOSE[31473] logger.c:  [pbx_wilcalu.so]

Jul 26 22:59:18 VERBOSE[31473] logger.c:  [pbx_wilcalu.so]

Jul 26 22:59:18 WARNING[31473] loader.c: 
/usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: 
ast_pthread_create


Jul 26 22:59:18 WARNING[31473] loader.c: Loading module pbx_wilcalu.so 
failed!


If I "noload" the module in modules.conf it all runs fine.

Mark

--

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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RE: [Asterisk-Users] Re: [Asterisk-Dev] Cluecon - Who's going ?

[Kevin Walsh]

Greg Boehnlein [EMAIL PROTECTED] wrote:
> I'll be REALLY interested in your talk! Please make sure that you have
> take-away notes available so it doesn't evaporate into thin air after the
> conference! :)
>
Perhaps also a drive-through lane at the side of the venue for people
who don't have the time to find a seat, let alone sit and take notes.
:-)

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

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Re: [Asterisk-Users] mpg123 - two processes

[Brian West]

If you use mp3nb from the sample configs you will have exactly 1 per  
class.


/b

On Jul 26, 2005, at 9:38 PM, MF Hulber wrote:


Yes, I always have two.

MARK.

Billy Dunn wrote:


Does everyone have two processes running for mpg123?  I always  
have them when I'm running an idle Asterisk box.  No calls going  
in or out and nothing off hook.  Is this normal?  Thanks!


5008 ?S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f  
4096 fpm-calm-ri

5015 ?S  0:00 /usr/sbin/asterisk
5061 ?S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f  
4096 fpm-calm-ri


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Re: [Asterisk-Users] Melting TDM card

[Kevin P. Fleming]

Robert Christian wrote:


Still, nothing was touching the card, it was in the right slot, and it was
installed as instructed.  Beats me what happened.  I'm just not sure I like
spending hundreds of dollars on a card that arcs and melts when I install it
- even if it continues to work.


Call Digium support, it's a brand new card and the mailing list 
certainly can't help you with it.

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[Asterisk-Users] Are busy and congestion behaving differently than documented?

[Steve Gladden]

I am using asterisk (2 week old CVS) am for the first time have
been starting to experiment with busy and congestion.

At this point I am only using sip endpoints PAP2-NA devices.
All testing of this is being done on a local network.

my test extension looks like this:


exten => ,1,Answer
exten => ,2,busy(35)
exten => ,3,Hangup



Or like this:

exten => ,1,Answer
exten => ,2,Playtones(busy)
exten => ,2,busy(35)
exten => ,3,Hangup


In both cases I seem to be getting the EXACT behaviour.

Asterisk plays a 'busy' signal for 35 seconds
I have also tried this with congestion (instead of busy).

What is strange is that in either case, the busy
tones are coming from asterisk and *not* being locally generated
by the PAP2-NA.

In example #1 I only send a busy status to the PAP2 where I'd expect it
to actually generate the busy audible signal.

It's easily verified by changing the busy tones (regional settings)
in the PAP2 to something other than standard

No matter what I try I still get the 'standard' busy tones from
asterisk  even if I do not execute Playtones.

The documentation (wiki) suggests that you HAVE TO execute playtones
in order to hear busy tones if the local device does not generate them.

The two strrange thigs I see happening here are

#1 the PAP2-NA is not generating the busy tones
#2 asterisk *is* generating them regardless of executing Playtones
:-)

I'm confused!

--Steve



Busy()

Sends a signal to inform the channel of a busy condition. This command
waits for the user to hang up; it does not continue execution of further
commands.

Note that this command does not actually play a busy tone to the user. If
you wish to do that, call Playtones(busy) before calling this command. For
most channels, the channel or the end device will take care of playing the
busy signal to the caller - in particular, with SIP phones they will play
their own Busy tone even if you have already called Playtones - it is
often better to just call Playtones(Busy) and not Busy();

exten => s,1,Answer  ; can't Playtones unless we answer first
exten => s,2,Playtones(busy) ; send the audio sequence that humans
understand means "busy"
exten => s,3,Busy; signal the other end that the line is busy.
Wait for hangup.





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Re: [Asterisk-Users] Melting TDM card

[Andrew Kohlsmith]

On Tuesday 26 July 2005 21:27, Robert Christian wrote:
> Still, nothing was touching the card, it was in the right slot, and it was
> installed as instructed.  Beats me what happened.  I'm just not sure I like
> spending hundreds of dollars on a card that arcs and melts when I install
> it - even if it continues to work.

This is clearly a case for Digium tech support.  Do you honestly think that 
this is normal?  I've installed dozens of these cards over the past year or 
so and never run into it.

It almost sounds like a grounding issue between the telco lines and the PC.

-A.
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Re: [Asterisk-Users] mpg123 - two processes

[MF Hulber]

Yes, I always have two.

MARK.

Billy Dunn wrote:

Does everyone have two processes running for mpg123?  I always have 
them when I'm running an idle Asterisk box.  No calls going in or out 
and nothing off hook.  Is this normal?  Thanks!


5008 ?S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 
fpm-calm-ri

5015 ?S  0:00 /usr/sbin/asterisk
5061 ?S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 
fpm-calm-ri


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[Asterisk-Users] TE110P Cable Pin Out

[Paul Dracevich]

I have just got a TE110P card, and I need the cable pin out.

 

Thanks

 








--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 7/25/2005
 
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RE: [Asterisk-Users] To anyone seeking 911 Service Providers "stayaway!!!"

[Rich Adamson]

> > There where people saying this Vonage issue was all FUD, anyway, side 
> > effect was...Seems this is 911 for VOIP is "FCC mandatory" 
> > now in USA ? Not sure, I use * at my home and have DSL, so I 
> > just route my 911 to the 
> > landline outbound, I would not expect the outbound IAX providers to 
> > offer 911 to me :-)
> 
> The FCC regulation is still in the comment phase. When it goes into
> effect it will require "interconnected" VOIP lines (defined as those
> enabling calls both to and from the PSTN) to provide access to the
> appropriate local PSAP, via the appropriate selective router (where on
> exists for that location). The regulation further requires providers of
> interconnected VOIP service to provide one or more methods for end-users
> to update their location. At least one of those methods must be
> accessible via nothing more than the CPE. 

The "mandatory" part that is due right now is the section of the law
that deals with "informing" the voip user of their current E911 status.
That part is not in a comment phase.


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[Asterisk-Users] Real-time for H.323?

[Ronald_Wiplinger]

Matthew,

can we use real-time also for H.323 phones? (h323_buddies) ???


bye

Ronald

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Re: [Asterisk-Users] super high bandwidth codec

[Andrew C. Brown]

Brian Capouch wrote:
> Geoff Manning wrote:
> 
> Skype uses wideband-ilbc.
>

>>
>> I don't think thats right. I think it just uses iLBC over it's own
>> proprietary Voip protocol.
>> http://www.skype.com/help/faq/technical.html
>> How much bandwidth does Skype use while I'm in a call?
>> Skype automatically selects the best codec depending on the
>> connection
>> between yourself and the person you are calling. On average, Skype uses
>> between 3-16 kilobytes/sec depending on bandwidth available for other
>> party,
>> network conditions in between, callers CPU performance, etc.
> 
> 
> I don't think that's correct.
> 
> I don't have the link to the Columbia paper where they tried (with only
> mixed success) to figure out what all nefarious stuff Skype does
> (hijacking port 80 being the most pernicious) but I'm pretty sure they
> have figured out that if possible, it will use the (proprietary)
> wideband version of iLBC.
> 

FYI: One can find the columbia paper link by going to the VoIP wiki's
Skype page.

According to GIPS datasheets, GIPS offers two proprietary wideband
codecs. iPCM-wb and iSAC. Both have 16kHZ sample rate* which is double
the 8kHz of PSTN, iLBC and most of the other codecs, hence the
relatively wonderous sound quality which I, among others, covet for
Asterisk.

The channel bit rate is respectively (it varies dynamically)
iLBC (free) 13-15kbps
iSAC ($)10-30kbps
iPCM-wb ($) 80kbps

iPCM-wb doesn't seem to offer any outright fidelity advantage over iSAC
since they are the same sample rate. I presume all those extra bits are
redundancy to make the quality more robust.

* Ref: http://www.globalipsound.com/solutions/solutions_Codecs.php

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RE: [Asterisk-Users] mpg123 - two processes

[Cullin J. Wible]

I always have two as well - not sure why though. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Billy Dunn
Sent: Tuesday, July 26, 2005 6:25 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] mpg123 - two processes

Does everyone have two processes running for mpg123?  I always have them 
when I'm running an idle Asterisk box.  No calls going in or out and 
nothing off hook.  Is this normal?  Thanks!

5008 ?S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 
fpm-calm-ri
5015 ?S  0:00 /usr/sbin/asterisk
5061 ?S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 
fpm-calm-ri

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Re: [Asterisk-Users] Good day everyone, i need firmware for the ATA186.

[Derek Whitten]

There is a link to the 3.x firmware on voip-info.org



On Tue, 2005-07-26 at 16:23, Otto Krumm Hernández wrote:
> Hi everyone,
>  
> I need the firmware for the cisco ATA 186 (SIP), anyone please?
>  
> Thanks Otto.
>  
> 
> __
> ___
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-- 
-BEGIN GEEK CODE BLOCK-
Version: 3.1
GCS/GIT> d-@ s+:+ a? C+++ BLHIS$ U+++ P+> L+++ !E W+++$ N++ o+ K w--
PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+> h r+++ y 
 --END GEEK CODE BLOCK--


signature.asc
Description: This is a digitally signed message part
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Re: [Asterisk-Users] Some more VOICEMAILMAIN issue...

[Tim Litwiller]

this works for me

exten => 100,1,Wait(1)
exten => 100,2,VoicemailMain([EMAIL PROTECTED])

change 100 to whatever you want the common voicemail extension to be. -  
the wait keeps the first part of the voice from being chopped off.


Mauro Zanin wrote:


Done both reload and Linux box re-load, after a power down.
Still have request for Box & Password!
 
Regards and thanks

Mauro



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[Asterisk-Users] Melting TDM card

[Robert Christian]

Yesterday FedEx brought me my new TDM400P with 4 FXS
modules.  I installed it (in the correct type PCI slot) , plugged in the
power, and fired up the system.  A few minutes later everyone in the
office is complaining about something burning.  I open the server again,
and the top of the Digium card is black, slightly deformed, and looks and
smells of melted plastic.  The funny thing is…the card still works
just fine (as far as my testing has revealed, anyway.  I haven’t
tried ringing the lines, just placing calls from them).  Maybe this is how
Digium knows if a card is returned used or not?

 

Still, nothing was touching the card, it was in the right
slot, and it was installed as instructed.  Beats me what happened.  I’m
just not sure I like spending hundreds of dollars on a card that arcs and melts
when I install it – even if it continues to work.

 

Has anyone else had this problem with the TDM cards?

 

 

- Robert






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Re: [Asterisk-Users] super high bandwidth codec

[Andrew C. Brown]

Brian Capouch wrote:
> Geoff Manning wrote:
> 
> Skype uses wideband-ilbc.
>

>>
>> I don't think thats right. I think it just uses iLBC over it's own
>> proprietary Voip protocol.
>> http://www.skype.com/help/faq/technical.html
>> How much bandwidth does Skype use while I'm in a call?
>> Skype automatically selects the best codec depending on the
>> connection
>> between yourself and the person you are calling. On average, Skype uses
>> between 3-16 kilobytes/sec depending on bandwidth available for other
>> party,
>> network conditions in between, callers CPU performance, etc.
> 
> 
> I don't think that's correct.
> 
> I don't have the link to the Columbia paper where they tried (with only
> mixed success) to figure out what all nefarious stuff Skype does
> (hijacking port 80 being the most pernicious) but I'm pretty sure they
> have figured out that if possible, it will use the (proprietary)
> wideband version of iLBC.

BTW, with all this talk about "wideband iLBC", we should probably start
using correct terms since iLBC actually stands for "internet Low
Bandwidth Codec". So "wb iLBC" is an oxymoron. GIPS has a few
proprietary codecs. Probably the one we are referring to is "iPCM-wb",
which according to GIPS (globalipsound.com) "...is a wide band codec
with extreme robustness against packet loss [that] provides
better-than-PSTN quality across ... the public Internet." They also have
iSAC which supposedly "...makes VoIP communications possible using a
dial-up modem, automatically adjusting transmission rates to deliver
better-than-PSTN voice quality."

Skype is listed as a customer of GIPS on GIPS' website, but I don't know
which codec Skype uses. It must be one of the proprietary ones though.
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RE: [Asterisk-Users] Automatic setup of calls between two externallines

[Cullin J. Wible]

I think you could accomplish this with EAGI or the manager interface. You
should also read:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out 

On a side note, we spent lots of time when we setup our Asterisk system
dealing with the answer detection for PSTN calls. We use Teliax for VoIP
traffic as well as a 4-port Digium FXO card. We have come to understand that
POTS lines can not accurately handle answer detection and that some of the
800 numbers (e.g. American Airlines, Staples, etc.) also pose problems even
for all-digital services (including T1 carriers).

We have had the most success placing PSTN calls if we Answer() the call
first and then Dial() the PSTN telephone number. This also allows VoIP users
to accurately hear telco messages (such as this number is not in service)
that happen before the channel is answered.

After weeks of debugging and testing I think that accurate answer detection
on the PSTN will be the most complicated part of what you are attempting to
do. Granted if you are calling a limited set of numbers you can probably
test it and see how it works but I would proceed with caution.

And just remember that this has nothing to do with Asterisk it is an
unfortunate part of the PSTN.

The better solution would be to call user 1 and play a repeated prompt
saying "Press # to initiate your connection". Then on DTMF# you would
initiate the second call. It's not elegant but it's bullet proof.

Hope that helps,

Cullin


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Billy Dunn
Sent: Tuesday, July 26, 2005 6:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Automatic setup of calls between two
externallines

Rob Scott wrote:

>Is it possible to automatically set up a call between two external
>lines?
>I would like Asterisk is call a cellphone number, wait for it to answer,
>and then call another cellphone, when that answers connect the two
>together.
>I assume it is possible but can someone point me how to do it.
>
>Thanks.
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>
>  
>
I think this is possible, but the conversation will look like this:

1st cell phone -- asterisk -- 2nd cell phone



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RE: [Asterisk-Users] Generate ring while waiting for SIP connection toinitiate

[Cullin J. Wible]

Use the "r" option in the Dial() command.

Cullin 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nick
Kartsioukas
Sent: Tuesday, July 26, 2005 4:57 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Generate ring while waiting for SIP connection
toinitiate

We're passing PSTN traffic on to a SIP proxy.  The SIP phone customers
have voicemail that will answer if their phone isn't picked up in a
certain amount of time.  However, if their phone is not on the network,
a caller will get nothing but dead air as Asterisk keeps attempting to
initiate the SIP connection.  Is there a way to generate a ringtone for
the caller while Asterisk is trying to make the SIP connection?

-- 
Nick Kartsioukas
Sky Way Networks, LLC
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Re: [Asterisk-Users] Call forwarding

[Adrian A]

Turns out it should actually be:
exten => s,1,AGI(agi-script.agi|arg1|arg2)

On 7/26/05, Adrian A <[EMAIL PROTECTED]> wrote:
> Thanks, that actually helps a lot.
> One problem I have (kind of unrelated) is with the AGI script
> requiring two arguments.  You have:
> exten => s,1,AGI(forward-get.agi,internal,${MACRO_EXTEN})
> On my Asterisk installation that somehow passes the two arguments
> internal and ${MACRO_EXTEN} as one argument to the bash script causing
> the blank check for ${exten} to exit the script.  I have even tried
> other suggestions such as:
> exten => s,1,AGI(forward-get.agi|internal&${MACRO_EXTEN}) or
> exten => s,1,AGI,forward-get.agi,internal ${MACRO_EXTEN}
> I'm running a recent version of CVS HEAD.
> Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686
> running Linux on 2005-07-07 18:42:16
> 
> 
> 
> On 7/25/05, Cullin J. Wible <[EMAIL PROTECTED]> wrote:
> > 1) You could use asterisk realtime and a mysql database.
> >
> > 2) You could use an asterisk database and allow users to set call forwarding
> > by calling an extension.
> >
> > 3) You could write some scripts to use an external database (what we did)
> > and either allow users to update their forwarding options via a web page or
> > telephone.
> >
> > I have attached some simple shell AGI-scripts and parts of our dial-plan so
> > you can see how it all works. We authenticate against the mysql voicemail
> > database and then our standard extension macro checks the database, possibly
> > adding another channel to the dial command.
> >
> > I hope this helps.
> >
> > Cullin
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Adrian A
> > Sent: Monday, July 25, 2005 4:15 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [Asterisk-Users] Call forwarding
> >
> > Is there an easy way to allow the users to go to a webpage or dial an
> > extension and enter a phone number that their extension can be
> > forwarded to?
> > I'm using SER+Asterisk so doing this in sip.conf for example would not
> > work since all users are registered to SER.  Currently in
> > extensions.conf I have:
> > exten => s,2,Dial(SIP/[EMAIL PROTECTED],20)
> > Is there a way to check that the user at ${ARG1} has setup forwarding
> > and retrieve the forwarding destination?
> > ___
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> >
> >
> >
>
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[Asterisk-Users] Upgrade to [EMAIL PROTECTED] 1.3 "Problems with Background files"

[Hatzis, Michael (Icon)]

Title: Upgrade to [EMAIL PROTECTED] 1.3 "Problems with Background files"







Hi All,


I have upgraded from asterisk 0.7 to [EMAIL PROTECTED] 1.3 and found the following problems:


I use distinctive ring to direct calls to the office or home. I have 3 gsm file for 3 scenarios, this now does not work.

Office hours gsm

After hours gsm

Home gsm


 

Symptoms


If I have 1 gsm file in the exenstion.conf all works fine, no matter which file.


But when I have 2 or more files I will not hear the voice of the second file added, only silence. But all the standard sound work??


Has anyone had this problem.



I have reinstalled [EMAIL PROTECTED] incase it was something simple, with no luck.


Regards


Mike



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Re: [Asterisk-Users] best way to dial and connect two users

[tom fielding]

Is this considered a 3PCC (3rd party call control) call where Asterisk is in the middle? I can set up two .call files to make two outgoing calls, but how do I bridge them?  Or is this the wrong way of going about calling and connecting two users?
 
Thanks,Tomtom fielding <[EMAIL PROTECTED]> wrote:

Hello all,
 
I'm trying to set up a call between two users - basically
 
1) dial user A
2) dial user B
3) connect the two calls legs
 
I've read the Wiki and looks like there may be a few things I can try, but wanted to get the list's opinion on the best/easiest way to do this?
 
Thanks very much for the advice,
Tom


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[Asterisk-Users] Good day everyone, i need firmware for the ATA186.

[Otto Krumm Hernández]

    Hi everyone,
 
I need the firmware for the cisco ATA 186 
(SIP), anyone please?
 
    Thanks Otto.
 
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Re: [Asterisk-Users] Perl AGI

[Waldo Rubinstein]

This worked great. Thanks,

Waldo

On Jul 26, 2005, at 12:15 PM, Nathan Pralle wrote:

To deal with this problem, I've used DeadAGI in the h, t, and i  
events (as appropriate) to detect and call the script again.


As such:
exten => h,1,DeadAgi,perlscript.pl|${UNIQUEID}|hangup

I pass the word "hangup" as the second word to flag to the AGI  
script that this is a hangup event, not a normal one.  I track the  
call using UNIQUEID so I can refer to it again after the call.


I have not tried any callback'ing.  This seems to work relatively  
well.


Nathan


I'm wondering if there is a way using AGI or the Perl AGI library  
to  catch the event of a call drop or hangup so I create the MySQL  
row  and properly close all database connections. I tried using  
the $AGI-  >setcallback() and putting code in the referenced  
callback function  but that doesn't seem to do it. Am I missing  
something?





--
-
Nathan E. Pralle
Give the Director a Serpent Deflector
www.nathanpralle.com
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Re: [Asterisk-Users] super high bandwidth codec

[Brian Capouch]

Geoff Manning wrote:

Skype uses wideband-ilbc.





I don't think thats right. I think it just uses iLBC over it's own
proprietary Voip protocol. 


http://www.skype.com/help/faq/technical.html
How much bandwidth does Skype use while I'm in a call?
Skype automatically selects the best codec depending on the connection
between yourself and the person you are calling. On average, Skype uses
between 3-16 kilobytes/sec depending on bandwidth available for other party,
network conditions in between, callers CPU performance, etc.


I don't think that's correct.

I don't have the link to the Columbia paper where they tried (with only 
mixed success) to figure out what all nefarious stuff Skype does 
(hijacking port 80 being the most pernicious) but I'm pretty sure they 
have figured out that if possible, it will use the (proprietary) 
wideband version of iLBC.


B.
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[Asterisk-Users] mpg123 - two processes

[Billy Dunn]

Does everyone have two processes running for mpg123?  I always have them 
when I'm running an idle Asterisk box.  No calls going in or out and 
nothing off hook.  Is this normal?  Thanks!


5008 ?S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 
fpm-calm-ri

5015 ?S  0:00 /usr/sbin/asterisk
5061 ?S  0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 
fpm-calm-ri


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[Asterisk-Users] cannot find channel_pvt.h

[wassim darwish]

when i tried to compile asterisk-oh323 i get an error
that channel_pvt.h is missing,where i can find and
download  it and in which directory i must put it.

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RE: [Asterisk-Users] Polycom digitmap question

[Cullin J. Wible]

It is my understanding that the purpose digitmap is to determine when the
phone should transmit the digits entered to the server. I do not believe
that it has any method for changing the dialstring.

However, you could place the Polycom phones in their own context which would
perform this mangling for you.

Cullin

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
aka ManxPower
Sent: Tuesday, July 26, 2005 4:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom digitmap question

[EMAIL PROTECTED] wrote:
> via google, I found the reference regarding digit maps for the Polycom 
> phones:
> http://lists.digium.com/pipermail/asterisk-users/2005-January/082884.html
> 
> But I don't see any instruction for prepending digits to the number 
> dialed.  Does anyone know how to prepend a digit to the number dialed 
> (from the Polycom side, not Asterisk)? I can do this pretty easily on a 
> Sipura. i.e. Say I want to add the digit "9" to what the user dials 
> 1xx, the Polycom should actually send 91xx to Asterisk.

I've never seen this option in the Polycom Admin Guide (which doesn't 
say much about the digitmap and refers you to the MGCP RFC digitmap 
handling.

--Eric
-- 
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
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[Asterisk-Users] problems with compiling asterisk-oh323-0.6.5

[jonny hashem]

after editing the Makefile according to my directories
ive compiled asterisk-oh323-0.6.5 but i got these
errors can any body help me with this :

[EMAIL PROTECTED]@net asterisk-oh323-0.6.5]# make
for x in wrapper asterisk-driver; do make -C $x build
|| exit 1 ; done
make[1]: Entering directory
`/home/wassim/asterisk-oh323-0.6.5/wrapper'
./check_ver /home/wassim/pwlib pwlib
./check_ver /home/wassim/openh323 openh323
ar rc liboh323wrap_s.a wrapper_misc.o asteriskaudio.o
wrapconnection.o wrapendpoint.o wrapper.o wrapcaps.o
wrapgkserver.o
make[1]: Leaving directory
`/home/wassim/asterisk-oh323-0.6.5/wrapper'
make[1]: Entering directory
`/home/wassim/asterisk-oh323-0.6.5/asterisk-driver'
gcc -Wall -pipe -Wall -Wstrict-prototypes
-Wmissing-prototypes -Wmissing-declarations
-D_REENTRANT -D_GNU_SOURCE -I/usr/src/asterisk/include
-I../wrapper -g -c -o chan_oh323.o chan_oh323.c
chan_oh323.c:37:34: asterisk/channel_pvt.h: No such
file or directory
chan_oh323.c: In function `oh323_exception':
chan_oh323.c:1145: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_indicate':
chan_oh323.c:1326: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_digit':
chan_oh323.c:1388: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_text':
chan_oh323.c:1410: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_call':
chan_oh323.c:1434: error: dereferencing pointer to
incomplete type
chan_oh323.c:1453: error: structure has no member
named `callerid'
chan_oh323.c:1455: error: structure has no member
named `callerid'
chan_oh323.c:1457: error: structure has no member
named `callerid'
chan_oh323.c: In function `oh323_hangup':
chan_oh323.c:1613: error: dereferencing pointer to
incomplete type
chan_oh323.c:1721: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_read':
chan_oh323.c:1749: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_write':
chan_oh323.c:2050: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_answer':
chan_oh323.c:2242: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_fixup':
chan_oh323.c:2286: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `ast_oh323_new':
chan_oh323.c:2518: error: dereferencing pointer to
incomplete type
chan_oh323.c:2527: error: dereferencing pointer to
incomplete type
chan_oh323.c:2529: error: dereferencing pointer to
incomplete type
chan_oh323.c:2536: error: dereferencing pointer to
incomplete type
chan_oh323.c:2537: error: dereferencing pointer to
incomplete type
chan_oh323.c:2538: error: dereferencing pointer to
incomplete type
chan_oh323.c:2539: error: dereferencing pointer to
incomplete type
chan_oh323.c:2540: error: dereferencing pointer to
incomplete type
chan_oh323.c:2541: error: dereferencing pointer to
incomplete type
chan_oh323.c:2542: error: dereferencing pointer to
incomplete type
chan_oh323.c:2543: error: dereferencing pointer to
incomplete type
chan_oh323.c:2544: error: dereferencing pointer to
incomplete type
chan_oh323.c:2545: error: dereferencing pointer to
incomplete type
chan_oh323.c:2546: error: dereferencing pointer to
incomplete type
chan_oh323.c:2547: error: dereferencing pointer to
incomplete type
chan_oh323.c:2548: error: dereferencing pointer to
incomplete type
chan_oh323.c:2549: error: dereferencing pointer to
incomplete type
chan_oh323.c:2550: error: dereferencing pointer to
incomplete type
chan_oh323.c:2551: error: dereferencing pointer to
incomplete type
chan_oh323.c:2552: error: dereferencing pointer to
incomplete type
chan_oh323.c:2579: error: structure has no member
named `dnid'
chan_oh323.c:2589: error: structure has no member
named `callerid'
chan_oh323.c:2590: error: structure has no member
named `callerid'
chan_oh323.c:2591: error: structure has no member
named `callerid'
chan_oh323.c:2596: error: structure has no member
named `callerid'
chan_oh323.c:2597: error: structure has no member
named `callerid'
chan_oh323.c:2598: error: structure has no member
named `callerid'
chan_oh323.c:2600: error: structure has no member
named `callerid'
chan_oh323.c:2605: error: structure has no member
named `callerid'
chan_oh323.c:2606: error: structure has no member
named `callerid'
chan_oh323.c:2608: error: structure has no member
named `callerid'
chan_oh323.c:2610: error: structure has no member
named `callerid'
chan_oh323.c:2614: error: structure has no member
named `callerid'
chan_oh323.c:2617: error: structure has no member
named `ani'
chan_oh323.c:2617: error: structure has no member
named `callerid'
chan_oh323.c:2623: error: structure has no member
named `callerid'
chan_oh323.c:2624: error: structure has no member
named `callerid'
chan_oh323.c: In function `oh323_request':
chan_oh323.c:2741: error: dereferencing pointer to
incomplete type
chan_oh323.c:2743: error: dereferencing pointer to
incomplete type
chan

[Asterisk-Users] Re: [Asterisk-Dev] Cluecon - Who's going ?

[Greg Boehnlein]

On Mon, 25 Jul 2005, Brian West wrote:

> I'm going to be speaking about how to use valgrind, gdb and strace to  
> help debug issues... it can be applied to more than just asterisk.

Given the following from one of my Client's boxes...

pbx*CLI> show memory summary 

[DELETED]

  7084 bytes in   435 allocations in file 'res_indications.c'
   223 bytes in24 allocations in file 'chanvars.c'
  51734730 bytes in 186815 allocations in file 'frame.c'
51993128 bytes allocated 188799 units total


I'll be REALLY interested in your talk! Please make sure that you have 
take-away notes available so it doesn't evaporate into thin air after the 
conference! :)

-- 
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] Registration failed problems/Polycom 500/maybe nat problem?

[Chris Mason (Lists)]

I have sent a Polycom IP500 phone to an overseas remote user who has a 
Speedtouch adsl modem/router. The phone is connected to the 4 port 
router which performs NAT for the network.
The phone rboots, find the server, downloads it's config and tried to 
register. Then it immediately becomes UNREACHABLE.


Jul 26 17:44:41 NOTICE[21026]: chan_sip.c:9138 handle_request_register: 
Registration from '' failed for '209.59.xxx.xxx'


816/816209.59.xxx.xxx D   N  
255.255.255.255  10979UNREACHABLE


I have tried everything I can think of. Is it possible the ISP is 
blocking the SIP packets going back to the phone?


The entry in sip.conf is:
[816]
type=friend
host=dynamic
dtmfmode=rfc2833
username=816
secret=mysecret
canreinvite=no
callerid="Me" <816>
mailbox=816
disallow=all
allow=g729
allow=ulaw
nat=yes
qualify=yes
context=default

--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


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[Asterisk-Users] What are SIP proxies and H323 Gatekeepers

[Obelix]

I know this must look like a very trivial question, but what are SIP Proxies and
H323 Gatekeepers, and what do they add to Asterisk?

Why should Asterisk need them?

/Obelix



This message was sent using IMP, the Internet Messaging Program.

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[Asterisk-Users] ASTCC: different incriments

[Ronald Wiplinger]

How can I fulfill that?

*Billing Increments*
Continental USA: six (6) second increments.
International: thirty (30) seconds minimum and six (6) seconds thereafter.
Mexico: sixty (60) seconds minimum and six (6) seconds thereafter.


bye

Ronald Wiplinger

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[Asterisk-Users] E-911

[JOAO CARLOS MOURA]

Hi there,
Somebody knows a solution E911 for Asterisk?
I can implement the E911 with a AGI?
We have some customers with a DID in our termination into our Asterisk in 
different areas...

Somebody know how can I send the address?
Today if some of those users we have dial 911 the address we will appear 
will be our address for the PSAP...

Some help??

Thanks in advance
João Carlos Moura 



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Re: [Asterisk-Users] CRITICAL PATCH for anyone using the L option in dial.

[Lists]

On Tuesday 26 July 2005 13:39, Brian West wrote:
> http://bugs.digium.com/view.php?id=4760
>
> If you use the L() option on dial and say the latest CVS-HEAD in the
> past month you're potentially getting screwed out of a lot of money.
>
> We originally wrote the L() option for dial and it worked great till
> someone came along and hijacked the timer for something else thus
> causing the L option to fail/reset the timer to zero thus causing it
> to never timeout if someone were to say press a DTMF digit.

This is not the right list probably, and I do appreciate the warning. But 
maybe you can educate me on how it can be hijacked, provided the code is 
fully modular, which I assume projects like these are? 

I've seen other things which I don't get, of course I've not looked through 
the code to get a feel for how it's written as I'm not much of a C 
programmer. So I'm hoping that someone could give me at least a conceptual 
understanding of how this goes... Are people simply too sloppy or is it just 
me? : )

-- 

List Manager
Network Voice Communications, Inc.
netwvcom.com
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Re: [Asterisk-Users] Method not allowed error

[Paul Traue, Jr.]

One idea would be to convert this extension into a queue extension, and 
then set the strategy to be ringall.


Paul

Lenwood S. Sawyer, III wrote:
I d/l CVS HEAD today and am getting the same error with a strange 
behavior.  If I ring a number that calls two sip extensions and pick up 
the call on one of the extensions, then the other extension continues to 
ring indefinitely.


Any ideas?

Thanks,
Lenny Sawyer

Afzaal Mirza wrote:


Hi,

 

I am getting *“Got SIP response 405” Method not allowed”* error on 
CLI. I am also getting *Port restricted Cone NAT* error on my SJ phone.


 


*Please help!*

* *

* *

*Afzaal*




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[Asterisk-Users] Channel restarted en E1 Card

[Alex Ternero]

Every hour the 
channel in the E1 is restarted, and all active calls drop.
 
Is there a 
configuration to disable the restart parameters, configured in zapata.conf 
file??
 
There is a parameter 
: resetinterval = 3600 in zapata.conf
 
 
How can i avoid the 
continuos restart of channels??
 
 
Thanks
 
Alex
 
 
 



  
  

  


  

  
  

  


  
  Insert a catchy tag line 
  here

  

  


  Alex Ternero H.Gerente 
Tecnico 
  Winet PeruAntequera 777 Piso 10 
  

  [EMAIL PROTECTED] 
  

  
  
tel: fax: 
  mobile: 
511-2228000511-2228000511-97522329 

  
  

  


  Add me to your address book...
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Re: [Asterisk-Users] Some more VOICEMAILMAIN issue...

[C F]

You sure you not trying to polish your shoes when you do that?
As far as I can see from the info you gave that is the problem.


On 7/26/05, Mauro Zanin <[EMAIL PROTECTED]> wrote:
> Done both reload and Linux box re-load, after a power down.
> Still have request for Box & Password!
>  
> Regards and thanks
> Mauro
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Re: [Asterisk-Users] Method not allowed error

[dbruce]

I've been looking at the issue... but haven't found anything obvious... and
I can't reconfigure my server till after midnight...

Can someone set verbose 4 and set debug 4 and capture a more detailed SIP
debug for a call in this situation...

Regards,
Derek

- Original Message -
From: "Matt Loretitsch" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Tuesday, July 26, 2005 2:50 PM
Subject: RE: [Asterisk-Users] Method not allowed error


Same method not allowed here on the CLI.  Call from outside continues to
ring my internal polycom ip501 even after hanging up incoming call.  CVS
from this morning.  No solution yet.  Hopefully tomorrow things will be
better.

-Matt

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lenwood S.
Sawyer, III
Sent: Tuesday, July 26, 2005 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Method not allowed error


I d/l CVS HEAD today and am getting the same error with a strange
behavior.  If I ring a number that calls two sip extensions and pick up
the call on one of the extensions, then the other extension continues to

ring indefinitely.

Any ideas?

Thanks,
Lenny Sawyer

Afzaal Mirza wrote:
> Hi,
>
>
>
> I am getting *"Got SIP response 405" Method not allowed"* error on
> CLI.
> I am also getting *Port restricted Cone NAT* error on my SJ phone.
>
>
>
> *Please help!*
>
> * *
>
> * *
>
> *Afzaal*
>
>
> --
> --
>
> ___
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--
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Sawyer Real Estate - Commercial Division
2501 13th Street
Gulfport, MS 39501
Telephone:  228.314.5023
Cellular:   228.547.6918
Fax:228.863.0632
Email:  [EMAIL PROTECTED]
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Re: [Asterisk-Users] Automatic setup of calls between two external lines

[Billy Dunn]

Rob Scott wrote:


Is it possible to automatically set up a call between two external
lines?
I would like Asterisk is call a cellphone number, wait for it to answer,
and then call another cellphone, when that answers connect the two
together.
I assume it is possible but can someone point me how to do it.

Thanks.
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I think this is possible, but the conversation will look like this:

1st cell phone -- asterisk -- 2nd cell phone



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Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?

[C F]

If you download it then you agree to the EULA.

On 7/24/05, Steve Totaro <[EMAIL PROTECTED]> wrote:
> Just because you give your email address does not allow for unsolicited
> emails unless you agree in the EULA or terms and conditions.
> 
> 
> - Original Message -
> From: "C F" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Sunday, July 24, 2005 3:13 PM
> Subject: Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?
> 
> 
> Interesting, I've been on this list for almost a year now, and I
> didn't recieve this spam. Are you sure you didn't download any sip
> softphones and gave your email address?
> In which case it is NOT spam, you gave your email address to them.
> 
> 
> On 7/22/05, Jay Milk <[EMAIL PROTECTED]> wrote:
> > > -Original Message-
> > > From: Jerry Glomph Black [mailto:[EMAIL PROTECTED]
> > > Sent: Friday, July 22, 2005 2:18 AM
> > > Subject: Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?
> > >
> > > It's not a spam.  They are not yokels.  Don't know about you
> > >
> > > Gizmo is basically a different front end offered by the
> > > Sipphone.com people, to offer an alternative to Skype which
> > > is not a closed jail (interoperates with all
> > > SIP devices, asterisk, etc.).
> > >
> > > I think they sent the mail to all registered sipphone.com users.
> >
> > Dear Glomph,
> >
> > Thanks, I know what the Gizmo project is.  I'm just not interested.
> >
> > And yes, it's spam:
> > 1. It's unsolicited -- I'm not registered on Sipphone.com.  It came in
> > through an email address I use exclusively for the asterisk-users list
> > and a small volume of personal mail.
> > 2. It's commercial -- #5 of "10 more things to do" reads "Add $10 in
> > CallOut credit to your account and talk for hours".  So, they're trying
> > to sell me something.
> >
> > Yokel: "A clumsy, unsophisticated person".  I think that would be their
> > preferred label... Why?  Because if you send unsolicited commercial
> > email in the US without offering a way to opt out of future mailings,
> > you're either clumsy and unsophisticated, or you're a criminal.  I gave
> > them the benefit of the doubt and called them "yokels".  If that term
> > offends you in any way, then let's use the more factual "criminals."
> >
> > ___
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> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] CRITICAL PATCH for anyone using the L option indial.

[Steve Totaro]

This effects ASTCC
$dialstr = "Zap/$res->{path}/$phone|30|HL(" . ($maxtime * 60 * 1000) .
":6:3)";

- Original Message - 
From: "Brian West" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Cc: "Asterisk Developers Mailing List" 
Sent: Tuesday, July 26, 2005 10:39 AM
Subject: [Asterisk-Users] CRITICAL PATCH for anyone using the L option
indial.


> http://bugs.digium.com/view.php?id=4760
>
> If you use the L() option on dial and say the latest CVS-HEAD in the
> past month you're potentially getting screwed out of a lot of money.
>
> We originally wrote the L() option for dial and it worked great till
> someone came along and hijacked the timer for something else thus
> causing the L option to fail/reset the timer to zero thus causing it
> to never timeout if someone were to say press a DTMF digit.
>
> So if you use this please test this and report back to the bug ASAP.
>
> Thanks,
> Brian West
>
>
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[Asterisk-Users] Generate ring while waiting for SIP connection to initiate

[Nick Kartsioukas]

We're passing PSTN traffic on to a SIP proxy.  The SIP phone customers
have voicemail that will answer if their phone isn't picked up in a
certain amount of time.  However, if their phone is not on the network,
a caller will get nothing but dead air as Asterisk keeps attempting to
initiate the SIP connection.  Is there a way to generate a ringtone for
the caller while Asterisk is trying to make the SIP connection?

-- 
Nick Kartsioukas
Sky Way Networks, LLC
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[Asterisk-Users] Re: [Asterisk-Dev] Cluecon - Who's going ?

[Greg Boehnlein]

On Mon, 25 Jul 2005, Brian West wrote:

> I'm going to be speaking about how to use valgrind, gdb and strace to  
> help debug issues... it can be applied to more than just asterisk.

Given the following from one of my Client's boxes...

pbx*CLI> show memory summary 

[DELETED]

  7084 bytes in   435 allocations in file 'res_indications.c'
   223 bytes in24 allocations in file 'chanvars.c'
  51734730 bytes in 186815 allocations in file 'frame.c'
51993128 bytes allocated 188799 units total


I'll be REALLY interested in your talk! Please make sure that you have 
take-away notes available so it doesn't evaporate into thin air after the 
conference! :)

-- 
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] Method not allowed error

[Brian Capouch]

Lenwood S. Sawyer, III wrote:
I d/l CVS HEAD today and am getting the same error with a strange 
behavior.  If I ring a number that calls two sip extensions and pick up 
the call on one of the extensions, then the other extension continues to 
ring indefinitely.


Any ideas?


It's a bug in the current CVS-HEAD code.

B.
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Re: [Asterisk-Users] Call forwarding

[Adrian A]

Thanks, that actually helps a lot.  
One problem I have (kind of unrelated) is with the AGI script
requiring two arguments.  You have:
exten => s,1,AGI(forward-get.agi,internal,${MACRO_EXTEN})
On my Asterisk installation that somehow passes the two arguments
internal and ${MACRO_EXTEN} as one argument to the bash script causing
the blank check for ${exten} to exit the script.  I have even tried
other suggestions such as:
exten => s,1,AGI(forward-get.agi|internal&${MACRO_EXTEN}) or
exten => s,1,AGI,forward-get.agi,internal ${MACRO_EXTEN}
I'm running a recent version of CVS HEAD.
Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686
running Linux on 2005-07-07 18:42:16



On 7/25/05, Cullin J. Wible <[EMAIL PROTECTED]> wrote:
> 1) You could use asterisk realtime and a mysql database.
> 
> 2) You could use an asterisk database and allow users to set call forwarding
> by calling an extension.
> 
> 3) You could write some scripts to use an external database (what we did)
> and either allow users to update their forwarding options via a web page or
> telephone.
> 
> I have attached some simple shell AGI-scripts and parts of our dial-plan so
> you can see how it all works. We authenticate against the mysql voicemail
> database and then our standard extension macro checks the database, possibly
> adding another channel to the dial command.
> 
> I hope this helps.
> 
> Cullin
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Adrian A
> Sent: Monday, July 25, 2005 4:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Call forwarding
> 
> Is there an easy way to allow the users to go to a webpage or dial an
> extension and enter a phone number that their extension can be
> forwarded to?
> I'm using SER+Asterisk so doing this in sip.conf for example would not
> work since all users are registered to SER.  Currently in
> extensions.conf I have:
> exten => s,2,Dial(SIP/[EMAIL PROTECTED],20)
> Is there a way to check that the user at ${ARG1} has setup forwarding
> and retrieve the forwarding destination?
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> 
>
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[Asterisk-Users] TO: M.G. Ref: Dial using URI(web) or using FORM(web)

[Joe McConnaughey]

Does the SugarCRM included with AAH 1.3 not meet 
this criteria for you?
 
--Message: 3Date: Tue, 26 Jul 2005 
17:48:08 +0100From: "JunkMail" <[EMAIL PROTECTED]>Subject: 
[Asterisk-Users] Dial using URI(web) or using FORM(web)To: Message-ID: 
<[EMAIL PROTECTED]>Content-Type: 
text/plain; charset="iso-8859-1"Hello!I have an [EMAIL PROTECTED] instalation with 7 users working 
OK, and I'ld liketo implement either a-- Web dial feature, where the 
user would fill one form field with a phonenumber and a connection would be 
created between his extention and theentered number.OR-- Dial using 
an URI (callto:x link in a web page), having AstTapiinstalled and 
configured in all workstations.Considering we spend all day manually 
dialing numbers from our database, oneof those solutions would be GREAT 
!The second solution should be easier but the windows dialer app just 
won'tdo...Thanks in 
advanceM.G.--
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RE: [Asterisk-Users] Linksys PAP2-NA failures...

[Steve Gladden]

We had 2 fail at the office...
Red light and no talk battery on line #1 or line #2 port.
Unit completly responsive...
Not able to factory default due to unit being completly unresponsive.
It's some kind of hardware or total firmware fail.

They failed here before my very face and there were no issues
of anyone plugging them into phone wiring as each had a single
phone plugged in with a 3 foot cord :-)

Steve







> I've had some of these fail with the red light. I think customers may be
> plugging them into the wall with a pstn line still connected.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Craig Guy
> Sent: Tuesday, July 19, 2005 11:34 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Linksys PAP2-NA failures...
>
>
> I have found my PAP2-NA's to be picky about their DHCP server.  My PAP2-NA
> appears to lockup if it is set for DHCP and the server is my Netgear RP614
> Websafe router.  The fix for this is to unplug the ethernet from the unit,
> plug in an analog handset, power it on, using the handset perform a
> factory
> reset and then set it for static IP settings.
>
> Craig
>
> - Original Message -
> From: "Kris Stark" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Wednesday, July 20, 2005 6:24 AM
> Subject: [Asterisk-Users] Linksys PAP2-NA failures...
>
>
>> Has anybody else experienced problems with the Linksys PAP2-NA's?
>>
>> I've now had two of them fail unexpectedly, with no apparent rhyme or
>> reason, having gone into a RED power LED, with a solid blue ethernet
>> LED.  No response from the device either on the network or from the
>> phone  To make matters even crazier, the one that now failed was the
>> one I received as a replacement for the previous dead one - and no, they
>> were never installed in the same location either
>>
>> Grrr
>>
>> Kris
>>
>>
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RE: [Asterisk-Users] Method not allowed error

[Matt Loretitsch]

Same method not allowed here on the CLI.  Call from outside continues to
ring my internal polycom ip501 even after hanging up incoming call.  CVS
from this morning.  No solution yet.  Hopefully tomorrow things will be
better.

-Matt

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lenwood S.
Sawyer, III
Sent: Tuesday, July 26, 2005 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Method not allowed error


I d/l CVS HEAD today and am getting the same error with a strange 
behavior.  If I ring a number that calls two sip extensions and pick up 
the call on one of the extensions, then the other extension continues to

ring indefinitely.

Any ideas?

Thanks,
Lenny Sawyer

Afzaal Mirza wrote:
> Hi,
> 
>  
> 
> I am getting *"Got SIP response 405" Method not allowed"* error on 
> CLI.
> I am also getting *Port restricted Cone NAT* error on my SJ phone.
> 
>  
> 
> *Please help!*
> 
> * *
> 
> * *
> 
> *Afzaal*
> 
> 
> --
> --
> 
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-- 
Lenwood (Lenny) S. Sawyer III
Sawyer Real Estate - Commercial Division
2501 13th Street
Gulfport, MS 39501
Telephone:  228.314.5023
Cellular:   228.547.6918
Fax:228.863.0632
Email:  [EMAIL PROTECTED]
Visit our Web Site at: http://www.sawyercommercial.com
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Re: [Asterisk-Users] existing ISDN PBX <-> asterisk <-> 2xBRI for IVR and SIP

[Peter Svensson]

On Tue, 26 Jul 2005, Alex Ongena wrote:

> I'am new to * and googled/read a lot, but did not find (yet)
> a lot of info to do the above.
> 
> Some months ago, I did find a 'story' from somebody having
> put * between his PRI and current PBX as IVR, but I can not
> find it back :-(

We have an Asterisk pbx between the PSTN and our old pbx. Lots and lots of 
details to get right, but just take your time and work through them.

* Thing through your dialplan in advance. 
   * Which number goes where. 
   * How do you dial from one system to another. (This depends to a large 
 degree on your existing pbx)
   * Most user prefer the dialplan at each pbx to be prefix-free (see
 http://en.wikipedia.org/wiki/Prefix-free_code) i.e. that you dont
 have both 345 and 3456 as valid extensions at the same time.

* Get the details on the isdn signalling
   * How many digits and which numbering plan are incoming/ougoing 
 called numbers and callerid delivered/expected.
   * Which signalling format (since you use BRI I expect it to be 
 EuroISDN).

* Get the timing right. 
   * You will almost certainly accept timing from the PSTN and provide 
 it to the other pbx.
   * Multi-bri cards can take the timing from the PSTN and provide an
 identical clock to the other pbx. This preevents bit-slips and is 
 a good thing.

Peter


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Re: [Asterisk-Users] Polycom digitmap question

[Eric Wieling aka ManxPower]

[EMAIL PROTECTED] wrote:
via google, I found the reference regarding digit maps for the Polycom 
phones:

http://lists.digium.com/pipermail/asterisk-users/2005-January/082884.html

But I don't see any instruction for prepending digits to the number 
dialed.  Does anyone know how to prepend a digit to the number dialed 
(from the Polycom side, not Asterisk)? I can do this pretty easily on a 
Sipura. i.e. Say I want to add the digit "9" to what the user dials 
1xx, the Polycom should actually send 91xx to Asterisk.


I've never seen this option in the Polycom Admin Guide (which doesn't 
say much about the digitmap and refers you to the MGCP RFC digitmap 
handling.


--Eric
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
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Re: [Asterisk-Users] Dial using URI(web) or using FORM(web)

[Michiel van Baak]

On 17:48, Tue 26 Jul 05, JunkMail wrote:
> Hello!
> 
> I have an [EMAIL PROTECTED] instalation with 7 users working OK, and I'ld like
> to implement either a
> -- Web dial feature, where the user would fill one form field with a phone
> number and a connection would be created between his extention and the
> entered number.
> OR
> -- Dial using an URI (callto:x link in a web page), having AstTapi
> installed and configured in all workstations.
> 
> Considering we spend all day manually dialing numbers from our database, one
> of those solutions would be GREAT !
> The second solution should be easier but the windows dialer app just won't
> do...
> 
> Thanks in advance
> 
> M.G.

Hi,

We have implemented something that looks like option 1
We have a webbased addressbook. As soon as you login the
system will know your extension.
When you click on a phonenr in the webapp it will connect to
the manager API and originate a call for you. Your phone
rings, you pick up, and the system calls the number.
Works like a charm and never failed on us.
If this is what you are looking for I can provide some more
indepth info :)))
-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Why is it drug addicts and computer afficionados are both called users?"
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Re: [Asterisk-Users] 7960 from SIP to SKINNY

[izo]

On 7/26/05, Walid Azab <[EMAIL PROTECTED]> wrote:
> Anyone tried reverting to SKINNY from SIP. I have a problem I cannot fix and
> need to get back to SCCP to be able to use the phone. 
>   

I managed to do that yesterday from SIP 7.4 to SCCP 7.2

regards
m.
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Re: [Asterisk-Users] Method not allowed error

[Lenwood S. Sawyer, III]

I d/l CVS HEAD today and am getting the same error with a strange 
behavior.  If I ring a number that calls two sip extensions and pick up 
the call on one of the extensions, then the other extension continues to 
ring indefinitely.


Any ideas?

Thanks,
Lenny Sawyer

Afzaal Mirza wrote:

Hi,

 

I am getting *“Got SIP response 405” Method not allowed”* error on CLI. 
I am also getting *Port restricted Cone NAT* error on my SJ phone.


 


*Please help!*

* *

* *

*Afzaal*




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--
Lenwood (Lenny) S. Sawyer III
Sawyer Real Estate - Commercial Division
2501 13th Street
Gulfport, MS 39501
Telephone:  228.314.5023
Cellular:   228.547.6918
Fax:228.863.0632
Email:  [EMAIL PROTECTED]
Visit our Web Site at: http://www.sawyercommercial.com
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RE: [Asterisk-Users] super high bandwidth codec

[Geoff Manning]

 
>>> Skype uses wideband-ilbc.
>>> 
>> 

I don't think thats right. I think it just uses iLBC over it's own
proprietary Voip protocol. 

http://www.skype.com/help/faq/technical.html
How much bandwidth does Skype use while I'm in a call?
Skype automatically selects the best codec depending on the connection
between yourself and the person you are calling. On average, Skype uses
between 3-16 kilobytes/sec depending on bandwidth available for other party,
network conditions in between, callers CPU performance, etc.
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RE: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem

[Walid Azab]

Yep you are right , I usually do a chmod 777.

Thanks anyway :)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Neil Cherry
Sent: Tuesday, July 26, 2005 6:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem

Walid Azab wrote:

> Thanks to all of you guys. I managed to fix it. It turned out to be 
> that the ZIP file has to be extracted inside the TFTP root not outside 
> then copied to the TFTP root. It is working now.

Walid, you should be able to unzip it anywhere and copy it into the
directory. It sounds like a permissions problem when you copied it. In the
future just make sure that files copied into the tftp directory have at
least read permission for everyone (user, group and other). Since it's
working now you don't need to fool with it.
Just information for the future.

-- 
Linux Home Automation Neil Cherry   [EMAIL PROTECTED]
http://home.comcast.net/~ncherry/   (Text only)
http://hcs.sourceforge.net/ (HCS II)
http://linuxha.blogspot.com/My HA Blog
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RE: [Asterisk-Users] 7960 from SIP to SKINNY

[Walid Azab]

I just succeeded in doing so. My problem was also with obtaining the Skinny
image. But I managed to get one off the existing Cisco Call Manager that we
have. 

Kind Regards,
Walid

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Tuesday, July 26, 2005 7:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 7960 from SIP to SKINNY

On Tue, 2005-07-26 at 18:31 +0200, Walid Azab wrote:
> Hello,
>  
> Anyone tried reverting to SKINNY from SIP. I have a problem I cannot 
> fix and need to get back to SCCP to be able to use the phone.

It works fine.

You can edit your SIPxxx file and add a line:

image_version:P00307010200

This will change it to sccp. 
You will need to then use the SEPxxx to change it back.

--
respectfully, Joseph


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Re: [Asterisk-Users] Stumped on vMail problem, any ideas?

[Andrew C. Brown]

Oh right. That is a good point. Really your issue is more with
[EMAIL PROTECTED] (and/or AMP) config than with Asterisk. You aren't really
intended to mess with the .conf files if you are using either of those
setups, I believe. If you do want to mess with them anyway, then you are
more in the [EMAIL PROTECTED] debug business. Either way, I think your best
bet is to take it to the respective lists in a decending order. First
[EMAIL PROTECTED], second AMP, third Asterisk-Users, fourth Asterisk-devel.

See http://voip-info.org/tiki-index.php?page=Asterisk+at++Home

Good luck.

Howard Leadmon wrote:
> Oops sorry about that, as I said very new to this stuff, and guessing most of
> this is stuff put in place by AAH.   Here is what I show in my extensions.conf
> for the dial macro, let me know if I missed anything else:
> 
> ; Rings one or more extensions.  Handles things like call forwarding and DND
> ; We don't call dial directly for anything internal anymore.
> ; ARGS: $TIMER, $OPTIONS, $EXT1, $EXT2, $EXT3, ...
> ; Use a Macro call such as the following: 
> ;  Macro(dial,$DIAL_TIMER,$DIAL_OPTIONS,$EXT1,$EXT2,$EXT3,...)
> [macro-dial]
> exten => s,1,GotoIf($[ "${MACRO_CONTEXT}" = "macro-rg-group" ]?4:2)  ; if this
> is from rg-group, don't strip prefix
> exten => s,2,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}} !=
> ${RGPREFIX}]?4:3)  ; check for ring-group prefix
> exten => s,3,SetCIDName(${CALLERIDNAME:${LEN(${RGPREFIX})}}) ; strip off
> prefix
> exten => s,4,AGI,dialparties.agi
> exten => s,5,NoOp(Returned from dialparties with no extensions to call)
> exten => s,6,SetVar(DIALSTATUS=BUSY)
> exten => s,10,Dial(${ds})   ; dialparties will set
> the priority to 10 if $ds is not null
> 
> 
> 
> ---
> Howard Leadmon - [EMAIL PROTECTED]
> http://www.leadmon.net 
> 
> 
> 
>>-Original Message-
>>From: [EMAIL PROTECTED] [mailto:asterisk-users-
>>[EMAIL PROTECTED] On Behalf Of Andrew C. Brown
>>Sent: Tuesday, July 26, 2005 8:34 AM
>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>Subject: Re: [Asterisk-Users] Stumped on vMail problem, any ideas?
>>
>>Howard Leadmon wrote:
>>
>>>Hello all,
>>>
>>> I think I have most of my AAH 1.3 setup running (Asterisk 1.0.9), but
>>
>>somehow
>>
>>>something is not quite right with my vMail setup.  I would have sworn this
>>
>>was
>>
>>>all working, but maybe I was just dreaming.
>>>
>>> Anyway here is what is happening, say I am on extension 200 and I want to
>>>call to extension 201.   If extension 201 is no connected, then it rolls
>>
>>right
>>
>>>into vMail with the message the user is not available.   If I am talking
>>
>>on
>>
>>>201 then vMail responds with the user is currently on the phone, which is
>>>great.   Where this goes wrong is if 201 is on hook, and ringing, in this
>>
>>case
>>
>>>it will just keep on ringing 201 forever, and never transfer into the
>>
>>vMail
>>
>>>system.   Needless to say this is NOT what I am looking to have happen, I
>>>would like it to ring for say 30 seconds, and then transfer over to the
>>>mailbox if the person doesn't answer.
>>>
>>>Looking at my AAH extensions_additional.conf I see:
>>>
>>>[ext-local]
>>>include => ext-local-custom
>>>exten => 200,1,Macro(exten-vm,[EMAIL PROTECTED],200)
>>>exten => ${VM_PREFIX}200,1,Macro(vm,200)
>>>exten => 201,1,Macro(exten-vm,[EMAIL PROTECTED],201)
>>>exten => ${VM_PREFIX}201,1,Macro(vm,201)
>>>
>>>
>>>Looking under my AAH extensions.conf I see the following marcro's:
>>>
>>>; Ring an extension, if the extension is busy or there is no answer send
>>
>>it
>>
>>>; to voicemail
>>>; ARGS: $VMBOX, $EXT
>>>[macro-exten-vm]
>>>exten => s,1,Setvar(FROMCONTEXT=exten-vm)
>>>exten => s,2,Macro(record-enable,${ARG2},IN)
>>>exten => s,3,Macro(dial,${RINGTIMER},${DIAL_OPTIONS},${ARG2})
>>>exten => s,4,GotoIf($[${CHANNEL:0:5} = Local]?s-${DIALSTATUS},1) ; if the
>>>channel is Local, then do not go to voicemail.  This is primarily to avoid
>>
>>vm
>>
>>>for call-forwarded extensions in ring groups
>>>exten => s,5,GotoIf($[${ARG1} = novm]?s-${DIALSTATUS},1) ; no voicemail in
>>
>>use
>>
>>>for this extension
>>>exten => s,6,NoOp(Sending to Voicemail box ${ARG1})
>>>exten => s,7,Macro(vm,${ARG1},${DIALSTATUS})
>>>exten => s-BUSY,1,NoOp(Extension is reporting BUSY and has no Voicemail)
>>>exten => s-BUSY,2,Busy()
>>>exten => s-BUSY,3,Wait(60)
>>>exten => s-BUSY,4,NoOp()
>>>exten => _s-.,1,Congestion()
>>>
>>>[macro-vm]
>>>exten => s,1,Goto(s-${ARG2},1)
>>>exten => s-BUSY,1,Voicemail(b${ARG1})   ; Voicemail Busy message
>>>exten => s-BUSY,2,Hangup()
>>>exten => _s-.,1,Voicemail(u${ARG1}) ; Voicemail Unavailable message
>>>exten => _s-.,2,Hangup()
>>>exten => o,1,Background(one-moment-please)  ; 0 during vm message will
>>>hangup
>>>exten => o,2,GotoIf($["foo${FROM_DID}" =
>>>"foo"]?from-pstn,s,1:from-pstn,${FROM_DID},1)
>>>exten => a,1,VoiceMailMain(${ARG1})
>>>exten => a,2,Hangup
>>>
>>>
>>>Also looking I see the following globals defined that appear to

Re: [Asterisk-Users] DISA disconnects

[Min Hwan Chang]

Thanks for the reply Andres.  I'm not very familiar with goto commands
so what I did was get rid of the goto and just wrote this instead.  It
works now, thanks for helping out.

[aa_1]
exten => 400,1,Answer
exten => 400,2,DigitTimeout(5)
exten => 400,3,ResponseTimeout(10)
exten => 400,4,Authenticate()
exten => 400,5,DISA(no-password|from-internal)


On 7/25/05, Andres Tello Abrego <[EMAIL PROTECTED]> wrote:
> The issue, is that when you make goto(disa_custom|s|1) the line is
> already answered...
> 
> Jump directly to disa_custom|s|4 or remove from 1 to 3 steps and re-number..
> 
> 
> Min Hwan Chang wrote:
> > DISA is currently disconnecting when I dial  to access DISA.
> > Below is my extensions.conf file from [EMAIL PROTECTED] and some lines 
> > which shows
> > the disconnect. Should DISA be loaded as a module in modules.conf?
> > When I do a 'show applications' i see that DISA is there.  Help!
> >
> > --
> > ;Asterisk CLI as I placed a call from cell into the system.
> >
> > Playing 'custom/aa_1' (language 'en')
> > ==CDR updated on Zap/3-1
> > Executing Goto("Zap/3-1", "disa_custom|s|1") in new stack
> > Goto (disa_custom,s,1)
> > Hungup 'Zap/3-1'
> >
> > -
> > ; MY EXTENSIONS.CONF
> >
> > [aa_1]
> > exten => ,1,Goto(disa_custom,s,1)
> >
> > [disa-custom]
> > exten => s,1,Answer
> > exten => s,2,DigitTimeout(5)
> > exten => s,3,ResponseTimeout(10)
> > exten => s,4,Authenticate()
> > exten => s,5,DISA(no-password|from-internal)
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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Re: [Asterisk-Users] TNT and SIP problem

[Dave Weis]

On Tue, 26 Jul 2005, Kevin P. Fleming wrote:

Eric Wieling aka ManxPower wrote:
Thank you for the correction.  Is this new to CVS-HEAD, or does it apply to 
1.0.x as well?  1.0.x sip.conf.sample only lists the "host" option for a 
peer, not for a user.


It only applies to 'peer' entries, but the important point is that 'peer' 
entries _are_ allowed to place calls through Asterisk, not just receive them. 
That applies to both 1.0.x and CVS HEAD. In CVS HEAD _every_ config option 
allowed on a 'user' entry is also supported on a 'peer', although that's not 
the case in 1.0.x (and won't be changed there, for obvious reasons). 
Realistically, this means that 'type=user' and 'type=friend' are of little 
value any more in CVS HEAD, except for some unusual cases.


I did have it set up as friend, but I see now why that didn't work.

Thanks
dave


--
Dave Weis "I believe there are more instances of the abridgment
[EMAIL PROTECTED]   of the freedom of the people by gradual and silent
  encroachments of those in power than by violent
  and sudden usurpations."- James Madison
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Re: [Asterisk-Users] Any experience with Sixtel--tollfreedirect--iax.cc?

[Erik Espinoza]

The Good:
I've had sixTel since December 2004. I've had surprisingly good call
quality and very short ping times. Overall the service has been good.
Response times for support were great when I needed it in the past. I
got a $10 rebate for my toll free DID because it was a couple of weeks
late, however I was kept in the loop from beginning to end until it
was active. I was informed that the provider they used lagging. I was
informed how to determine if the DID was routed to them yet or not,
etc.

The Bad:
My buddy, based on my recommendation, signed up for sixTel + toll free
DID on May 2005. To this day he has not recieved a response from
sixTel or been able to contact them. I have attemped to contact them
on his behalf myself, since I was so successful before, with no luck.
He is chalking this up to a bad luck and looking for a new provider.

The Ugly:
My buddy is stuck in a predicament, his custom toll free DID is
currently in the possesion of sixTel and he can't move it to a new
provider. In addition, during the few times I have attempted to
communicate with them I have observed the following:

1) All calls to sixTel's toll free number lead to a voiceprompt that
tells you to use the web site, no one is ever available.
2) sixTel's own toll free number has disappeared on more than one
occasion, where you don't even get the voice prompt telling you to use
the web site.
3) The web site's emergency support that supposedly pages a tech
doesn't get a response.

In short, I myself will be looking for a new provider once the money I
have deposited in my sixTel account runs out. I'd like to have a cheap
per minute rate similar to sixTel, and I'd like to be able to LNP my
current numbers away from sixTel. If anyone knows of a provider that
meets this criteria please shoot me a note.

Thanks,
Erik

On 7/26/05, Jay Milk <[EMAIL PROTECTED]> wrote:
> There was a thread about sixtel not too long ago.  Quality is ok, when
> it's working.  DID requests can take weeks or months, some are never
> answered.  Termination usually works, and with decent quality.  Looks,
> feels and smells like a one-pony show -- and the pony needs a vet.
> 
> Try this: http://www.google.com/search?q=sixtel+site%3Adigium.com
> 
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 26, 2005 8:31 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Any experience with
> Sixtel--tollfreedirect--iax.cc?
> 
> 
> Appears to be a diversified company (Google of it shows that they do web
> design, VOIP, etc.). But this link caused me concern.
> 
> http://www.ripoffreport.com/reports/ripoff145784.htm
> 
> Thoughts? Experiences?
> 
> ___
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Re: [Asterisk-Users] A TDM issue..

[chouck]

Oooh.. That makes sense now..  I have only used fxo for a pots line and 
would issue dial like so:


exten=blah,1,Dial(ZAP/1/blah)

But I never really thought about just a simple dial without a number.. so 
essentially the card will just pick up the channel and bridge them together?


Thanks alot everyone,

-Chad

- Original Message - 
From: "Andrew Kohlsmith" <[EMAIL PROTECTED]>

To: 
Sent: Tuesday, July 26, 2005 12:22 PM
Subject: Re: [Asterisk-Users] A TDM issue..



On Tuesday 26 July 2005 11:31, chouck wrote:

That is what I was worried about.  Let me try and explain the situation a
bit better... Basically I have an existing P.A. system that works by 
simply
picking up the line that it is connected to.  So basically I just pick up 
a
phone and I am instantly talking over the P.A.  So what I wanted to do 
was

hook the line into an fxs port (instead of a standard phone) and dial an
extension from anywhere and have it instantly bridge me to the line.  I
figured this would be simple and I thought that the answer command would 
do

it for me.. for instance


Ok let me make sure I understand.

You have a PA system that takes a regular phone.  When you pick up the 
phone

the PA keys and you talk.

The PA system is acting like a phone company -- it provides battery 
voltage to

the regular phone.

You want to replace the regular phone with Asterisk.

Asterisk needs to look like a phone to the PA system.

That's a FXO port, not an FXS port.

So... Get yourself an FXO port and Dial(Zap/1) -- it will work exactly as
expected.


exten=1234,1,Answer(ZAP/1)


exten => 1234,1,Dial(Zap/1)

No immediate mode needed.  It was just a misconception of how the system
interfaced.  :-)

-A.
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Re: [Asterisk-Dev] RE: [Asterisk-Users] Zaptel update, Asterisk 1.2 janitor projects

[Tzafrir Cohen]

On Tue, Jul 26, 2005 at 06:30:49PM +0200, Dave Cotton wrote:
> On Tue, 2005-07-26 at 18:11 +0300, Tzafrir Cohen wrote:
> 
> > I suppose you refer to:
> > http://lists.digium.com/pipermail/asterisk-cvs/2005-July/007125.html
> > 
> > How do I track only the changes to the stable branch? For a user of
> > Stable most of the messages on the CVS list are rather irrelevant.
> > 
> > There seems to be a '  Tag: v1-0' in the message but it is in the body.
> 
> Can't mutt filter on the body contents?

Why mutt? I want to filter it on the server-side.

Yes, procmail can filter by body. But:
a. it means more tests on *every* message from that list
b. much more error-prone. 

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Dev] RE: [Asterisk-Users] Zaptel update, Asterisk 1.2 janitor projects

[Michiel van Baak]

On 18:30, Tue 26 Jul 05, Dave Cotton wrote:
> On Tue, 2005-07-26 at 18:11 +0300, Tzafrir Cohen wrote:
> 
> > I suppose you refer to:
> > http://lists.digium.com/pipermail/asterisk-cvs/2005-July/007125.html
> > 
> > How do I track only the changes to the stable branch? For a user of
> > Stable most of the messages on the CVS list are rather irrelevant.
> > 
> > There seems to be a '  Tag: v1-0' in the message but it is in the body.
> 
> Can't mutt filter on the body contents?
>  

it can.

-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Why is it drug addicts and computer afficionados are both called users?"
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Re: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem

[Tom Rymes]

One caveat with the unzipping process: copy the ZIP file to the linux  
box before unzipping. Do NOT unzip on your Windows desktop and copy  
the individual files. This might not always be a problem, but it is a  
good way to avoid issues. (easier, too!)


Tom

On Jul 26, 2005, at 12:00 PM, Neil Cherry wrote:


Walid Azab wrote:


Thanks to all of you guys. I managed to fix it. It turned out to  
be that the
ZIP file has to be extracted inside the TFTP root not outside then  
copied to

the TFTP root. It is working now.



Walid, you should be able to unzip it anywhere and copy it into
the directory. It sounds like a permissions problem when you copied
it. In the future just make sure that files copied into the tftp
directory have at least read permission for everyone (user, group
and other). Since it's working now you don't need to fool with it.
Just information for the future.

--
Linux Home Automation Neil Cherry   [EMAIL PROTECTED]
http://home.comcast.net/~ncherry/   (Text only)
http://hcs.sourceforge.net/ (HCS II)
http://linuxha.blogspot.com/My HA Blog
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Re: [Asterisk-Users] Polycom 600 Presence indications - ALWAYS OFF-HOOK? - SOLVED

[Billy Dunn]

Billy Dunn wrote:

I am having one problem with the Polycom 600 phones.  All phones on 
the local network are fine and indicate presence to other phones 
perfectly.  One phone that is outside the network can see presence 
indications of the other phones correctly, but that phone always shows 
as off the hook to certain phones on the network.  Any ideas?


Thanks!
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Nevermind.  We resolved the issues by loading new firmware and/or 
rebooting those certain phones.

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[Asterisk-Users] Load Balancing with SER

[Kamran Ahmad]

hello

Can we use SER in front of 10 Asterisk for load
balancing. any idea

Thanks in advance
Kamran




Start your day with Yahoo! - make it your home page 
http://www.yahoo.com/r/hs 
 
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[Asterisk-Users] Polycom digitmap question

[asterisk]

via google, I found the reference regarding digit maps for the Polycom phones:
http://lists.digium.com/pipermail/asterisk-users/2005-January/082884.html

But I don't see any instruction for prepending digits to the number 
dialed.  Does anyone know how to prepend a digit to the number dialed (from 
the Polycom side, not Asterisk)? I can do this pretty easily on a Sipura. 
i.e. Say I want to add the digit "9" to what the user dials 1xx, 
the Polycom should actually send 91xx to Asterisk.


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[Asterisk-Users] If voice volume level too low, it is been cut

[Alexandre Leclerc]

Hi all,

I am at my first installation of [EMAIL PROTECTED] and this works almost out
of the... cd.

When I call phones on PSTN or when people call me and they do not speak
lound enought, all the sound is being cut and I hear nothing.

How can I reduce the *loud level* so that the sound is not cut?

(I use XList/SIP with [EMAIL PROTECTED] and an extarnal line... I did no
special config.)

Thanks for any pointers.

Regards.

-- 
Alexandre Leclerc

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RE: [Asterisk-Users] To anyone seeking 911 Service Providers "stayaway!!!"

[Rusty Shackleford]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Julio Arruda
> Sent: Tuesday, July 26, 2005 6:01 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] To anyone seeking 911 Service 
> Providers "stayaway!!!"
> 
> There where people saying this Vonage issue was all FUD, anyway, side 
> effect was...Seems this is 911 for VOIP is "FCC mandatory" 
> now in USA ? Not sure, I use * at my home and have DSL, so I 
> just route my 911 to the 
> landline outbound, I would not expect the outbound IAX providers to 
> offer 911 to me :-)

The FCC regulation is still in the comment phase. When it goes into
effect it will require "interconnected" VOIP lines (defined as those
enabling calls both to and from the PSTN) to provide access to the
appropriate local PSAP, via the appropriate selective router (where on
exists for that location). The regulation further requires providers of
interconnected VOIP service to provide one or more methods for end-users
to update their location. At least one of those methods must be
accessible via nothing more than the CPE. 

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 07/25/2005
 

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[Asterisk-Users] Unicall problem

[Jose Chiantera]

Hi Steve



Denis said me, you can help us with unicall problem, let me explain.

I have currently installed

supertone.0.0.2

spandsp.0.0.2

mfcr2.0.0.3

unicall.0.0.3

asterisk CVS

country  ve,10,4



I run the program testcall from unicall with 2 E1 30 channels in and 30 
channels out, work fine; I receive DNIS and ANI make calls and answer calls.




When start Asterisk I receive call form outside correctly including DNIS and 
ANI


but when try to make a call receive the messages:



Jul 26 17:45:29 WARNING[28636]: chan_unicall.c:635 unicall_report: MFC/R2 
UniCall/1 Call control(1)
Jul 26 17:45:29 WARNING[28636]: chan_unicall.c:635 unicall_report: MFC/R2 
UniCall/1 Make call
Jul 26 17:45:29 WARNING[28636]: chan_unicall.c:635 unicall_report: MFC/R2 
UniCall/1 Making a new call with CRN 32769
Jul 26 17:45:29 WARNING[28636]: chan_unicall.c:635 unicall_report: MFC/R2 
UniCall/1 0001  ->  [1/   1/Idle  /Idle ]

   -- Called g1/7168
Jul 26 17:45:29 WARNING[28636]: chan_unicall.c:2662 handle_uc_event: 
Unicall/1 event Dialing
Jul 26 17:45:29 WARNING[28636]: channel.c:1567 ast_read: Exception flag set 
on 'UniCall/1-1', but no exception handler
Jul 26 17:45:29 WARNING[28636]: channel.c:1567 ast_read: Exception flag set 
on 'UniCall/1-1', but no exception handler
Jul 26 17:45:29 WARNING[28636]: channel.c:1567 ast_read: Exception flag set 
on 'UniCall/1-1', but no exception handler
Jul 26 17:45:29 WARNING[28636]: channel.c:1567 ast_read: Exception flag set 
on 'UniCall/1-1', but no exception handler
Jul 26 17:45:29 WARNING[28636]: channel.c:1567 ast_read: Exception flag set 
on 'UniCall/1-1', but no exception handler
Jul 26 17:45:29 WARNING[28636]: channel.c:1567 ast_read: Exception flag set 
on 'UniCall/1-1', but no exception handler
Jul 26 17:45:29 WARNING[28636]: channel.c:1567 ast_read: Exception flag set 
on 'UniCall/1-1', but no exception handler

.. until I kill asterisk



I think the problem is in channels/chan_unicall.c , the effort to obtain 
what is the exception not handled have no result, but I not familiar with 
program structure.


If you have any idea that can help let me know, I continue trying solving 
the problem.




Thanks

Jose













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[Asterisk-Users] CRITICAL PATCH for anyone using the L option in dial.

[Brian West]

http://bugs.digium.com/view.php?id=4760

If you use the L() option on dial and say the latest CVS-HEAD in the  
past month you're potentially getting screwed out of a lot of money.


We originally wrote the L() option for dial and it worked great till  
someone came along and hijacked the timer for something else thus  
causing the L option to fail/reset the timer to zero thus causing it  
to never timeout if someone were to say press a DTMF digit.


So if you use this please test this and report back to the bug ASAP.

Thanks,
Brian West


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[Asterisk-Users] Automatic setup of calls between two external lines

[Rob Scott]

Is it possible to automatically set up a call between two external
lines?
I would like Asterisk is call a cellphone number, wait for it to answer,
and then call another cellphone, when that answers connect the two
together.
I assume it is possible but can someone point me how to do it.

Thanks.
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RE: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem

[Joseph]

On Tue, 2005-07-26 at 18:30 +0200, Walid Azab wrote:
> I went from 5.3 to 6.3 then from 6.3 t 7.5 directly. However, I have
> the warning message (Protocol Application Invalid)
>  
> Please any help.
>  
> Walid
> 

You may have this: P003-07-5-00
instead of this: P0S3-07-5-00

Note the S

Also, note that if you use the SEPxx.cnf.xml you can update the 7960 all
the way from the old version to the latest version.

-- 
respectfully, Joseph


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[Asterisk-Users] sip+oh323 - no voice at sip side

[bartek]

Hello,
I have something like this:
SIPUSER <-sip-> ASTERISK <-oh323-> AUDIOCODEC <-e1-> PSTN

After calling from SIP to PSTN (and from PSTN to SIP too)
I can't hear anything only in my SIPUSER. At the PSTN side everything is OK.

I have another network with another h323/sip (in the place of asterisk)
and there everything is OK.

In AUDIOCODES logs I see that everything goes fine with asterisk, but SIPUSER
can't hear the PSTN user.

sip.conf:
[general]
disallow=all

[48224789000]
type=friend
username=48224789000
secret=xx
host=dynamic
nat=yes
qualify=100
disallow=all
allow=g729
context=intern

Here is sip debug:
SIP Debugging Enabled
-- Inbound H.323 call 'ip$10.0.0.3:61804/23122' detected.
  == Starting OH323/datacom1234,@10.0.0.3-9de1 at voip-h323,224789000,1 failed 
so falling back to exten 's'
  == Starting OH323/datacom1234,@10.0.0.3-9de1 at voip-h323,s,1 still failed so 
falling back to context 'default'
-- Executing Dial("OH323/datacom1234,@10.0.0.3-9de1", "SIP/48224789000") in 
new stack
-- Inbound H.323 call 'ip$10.0.0.3:61804/23122', channel 
'OH323/datacom1234,@10.0.0.3-9de1'.
We're at 192.168.0.252 port 15278
Answering/Requesting with root capability 0x100 (g729)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.252:5060;branch=z9hG4bK411177bc;rport
From: "datacom1234, 224782479 " ;tag=as195b9c0f
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 26 Jul 2005 17:18:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 224

v=0
o=root 15298 15298 IN IP4 192.168.0.252
s=session
c=IN IP4 192.168.0.252
t=0 0
m=audio 15278 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (NAT) to 194.246.106.22:5060
-- Called 48224789000


Sip read:
SIP/2.0 100 Trying
To: 
From: "datacom1234, 224782479 " ;tag=as195b9c0f
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.252:5060;branch=z9hG4bK411177bc
Server: Sipura/SPA1000-2.0.13(g)
Content-Length: 0


8 headers, 0 lines


Sip read:
SIP/2.0 180 Ringing
To: ;tag=b857823bdad08738i0
From: "datacom1234, 224782479 " ;tag=as195b9c0f
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.252:5060;branch=z9hG4bK411177bc
Server: Sipura/SPA1000-2.0.13(g)
Content-Length: 0


8 headers, 0 lines
-- SIP/48224789000-8290 is ringing


Sip read:
SIP/2.0 200 OK
To: ;tag=b857823bdad08738i0
From: "datacom1234, 224782479 " ;tag=as195b9c0f
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.252:5060;branch=z9hG4bK411177bc
Contact: 48224789000 
Server: Sipura/SPA1000-2.0.13(g)
Content-Length: 236
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 59832 59832 IN IP4 172.16.13.169
s=-
c=IN IP4 172.16.13.169
t=0 0
m=audio 16418 RTP/AVP 18 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

12 headers, 12 lines
Found RTP audio format 18
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 172.16.13.169:16418
Found description format G729a
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), 
combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: 
set_destination: Parsing  for address/port to send 
to
set_destination: set destination to 172.16.13.169, port 5060
Transmitting:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.252:5060;branch=z9hG4bK6bacf8a7;rport
From: "datacom1234, 224782479 " ;tag=as195b9c0f
To: ;tag=b857823bdad08738i0
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 194.246.106.22:5060
-- SIP/48224789000-8290 answered OH323/datacom1234,@10.0.0.3-9de1
   > H.323 call 'ip$10.0.0.3:61804/23122', exception CALL_ESTABLISHED.
Destroying call '[EMAIL PROTECTED]'


Sip read:
BYE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 172.16.13.169:5060;branch=z9hG4bK-c77a5b50
From: ;tag=b857823bdad08738i0
To: "datacom1234, 224782479 " ;tag=as195b9c0f
Call-ID: [EMAIL PROTECTED]
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Sipura/SPA1000-2.0.13(g)
Content-Length: 0


9 headers, 0 lines
Sending to 172.16.13.169 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
172.16.13.169:5060;branch=z9hG4bK-c77a5b50;received=194.246.106.22;rport=5060
From: ;tag=b857823bdad08738i0
To: "datacom1234, 224782479 " ;tag=as195b9c0f
Call-ID: [EMAIL PROTECTED]
CSeq: 101 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to 194.246.106.22:5060
  == Spawn exten

[Asterisk-Users] Call Quality Reporting

[Nathan Pralle]

From what I've searched in the archives, having Asterisk report call 
quality statistics for each call has been discussed and mulled a little 
bit, but it is not implemented in * and doesn't have a plan for it at 
the moment.  For the time being, such things are a pipe dream.


Is this correct?  I'm having difficulty finding anymore information on it.

Helpful pointers appreciated...if it's not implemented, I'm fairly 
interested in looking into making it work.  I think that'd be an 
infinitely handy thing to be able to extract from Asterisk.


Nathan


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[Asterisk-Users] CVS HEAD behavior change: Beware!

[Kevin P. Fleming]

I have just committed some changes to CVS HEAD that make the effort to 
eliminate 'priority jumping' applications sooner vs. later...


Basically, there is now a global option, settable in extensions.conf, to 
disable all priority jumping. The only application that has been updated 
to respect this option is app_dial, but I will update the "janitor 
project" list to reflect what needs to be done. The 'j' option to Dial() 
now has the exact opposite behavior it had before: if priority jumping 
has been disabled globally, the 'j' option will cause that instance of 
Dial() to do jumping. This will be the model for all applications to be 
converted over to.


Anyone who does not change the global setting in extensions.conf (it 
defaults to 'on' in the source code) will not experience any change in 
behavior unless they were using 'j' to suppress jumping in Dial(). In 
that case, you will need to global suppress jumping, and then enable it 
in any Dial() calls where you need it. As more applications get 
converted to use this new option, you may need to add more 'j' options 
in different parts of your dialplan.


New users of CVS HEAD (anyone who uses the 'make samples' starter 
configuration file) will not have priority jumping on by default, since 
it is explicitly turned off in the sample configuration file. We will 
try to get the remaining apps converted over as quickly as possible, so 
that their behavior will be consistent.

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Re: [Asterisk-Users] 7960 from SIP to SKINNY

[Joseph]

On Tue, 2005-07-26 at 18:31 +0200, Walid Azab wrote:
> Hello,
>  
> Anyone tried reverting to SKINNY from SIP. I have a problem I cannot
> fix and need to get back to SCCP to be able to use the phone. 

It works fine.

You can edit your SIPxxx file and add a line:

image_version:P00307010200

This will change it to sccp. 
You will need to then use the SEPxxx to change it back.

-- 
respectfully, Joseph


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Re: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem

[Steve Blair]

There are several postings about this on the web. I don't have the 
details handy

anymore but a google search (or search of Cisco's site) should turn up the
answer. I remember seeing this with v7.0 code because of a problem with
the image released from Cisco. If you don't find the answer email me
back and I'll try to dig up what we did.

-Steve

Walid Azab wrote:

I went from 5.3 to 6.3 then from 6.3 t 7.5 directly. However, I have 
the warning message (Protocol Application Invalid)
 
Please any help.
 
Walid



*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of 
*Watkins, Bradley

*Sent:* Tuesday, July 26, 2005 4:12 PM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* RE: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem

I believe you have to upgrade to 5.3 in order to go from unsigned to 
signed executables.  Once you're at 5.3, you can go directly to 7.5.  
I did this recently with a couple of 7960s I had in the lab and it 
worked perfectly.
 
Regards,

- Brad

-Original Message-
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of
*Walid Azab
*Sent:* Tuesday, July 26, 2005 10:29 AM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem

Hi,
 
I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then

will to go up to 7.5
 
However in my first attempt to go from V.5.1 to 6.0 this is hat

happens:
 
- The phone reboots

- The phone then reads the file OS79XX.TXT from the TFP server. In
the file I added the version "P0S3-06-0-00"
- It starts upgrading firmware
- Then I get the following message: (Upgrade Failed - Unauthorized)
 
Any ideas? Please find below my conf files.
 
*SIP.CONF*

[300]
username=300
type=friend
secret=cisco
record_out=On-Demand
record_in=On-Demand
qualify=no
port=5060
nat=never
[EMAIL PROTECTED] 
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid="" <300>
 
*SIP000CCE351C07.cnf*

# SIP Configuration Generic File (start)
 
# Line 1 Settings

line1_name: "300" ; Line 1 Extension\User ID
line1_displayname: "300"   ; Line 1 Display Name
line1_authname: "300" ; Line 1 Registration Authentication
line1_password: "cisco" ; Line 1 Registration Password
 
# Line 2 Settings

line2_name: ""; Line 2 Extension\User ID
line2_displayname: ""; Line 2 Display Name
line2_authname: "UNPROVISIONED" ; Line 2 Registration
Authentication
line2_password: "UNPROVISIONED" ; Line 2 Registration Password
 
# Line 3 Settings

line3_name: ""  ; Line 3 Extension\User ID
line3_displayname: ""   ; Line 3 Display Name
line3_authname: "UNPROVISIONED" ; Line 3 Registration
Authentication
line3_password: "UNPROVISIONED" ; Line 3 Registration Password
 
# Line 4 Settings

line4_name: ""  ; Line 4 Extension\User ID
line4_displayname: ""   ; Line 4 Display Name
line4_authname: "UNPROVISIONED" ; Line 4 Registration
Authentication
line4_password: "UNPROVISIONED" ; Line 4 Registration Password
 
# Line 5 Settings

line5_name: ""  ; Line 5 Extension\User ID
line5_displayname: ""   ; Line 5 Display Name
line5_authname: "UNPROVISIONED" ; Line 5 Registration
Authentication
line5_password: "UNPROVISIONED" ; Line 5 Registration Password
 
# Line 6 Settings

line6_name: ""  ; Line 6 Extension\User ID
line6_displayname: ""   ; Line 6 Display Name
line6_authname: "UNPROVISIONED" ; Line 6 Registration
Authentication
line6_password: "UNPROVISIONED" ; Line 6 Registration Password
 
# NAT/Firewall Traversal

nat_address: ""
voip_control_port: "5060"
start_media_port: "16384"
end_media_port:  "32766"
 


# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "WaZaB-SIP"; Has no effect on SIP messaging
 
# Time Zone phone will reside in

time_zone: EST
 
# Phone prompt/password for telnet/console session

phone_prompt: "Cisco7960"  ;
Telnet/Console Prompt
phone_password: "abc"  ; Telnet/Console
Password
 
# SIP Configuration Generic File (stop)

*SIPDefault.cnf*
# Image Version
image_version: "P0S3-06-0-00"
 
# Proxy Server

prox

RE: [Asterisk-Users] Any experience with Sixtel--tollfreedirect--iax.cc?

[Richard Cook]

 
>> Appears to be a diversified company (Google of it shows that they do web
design, VOIP, etc.). But this link caused me concern. 
>> http://www.ripoffreport.com/reports/ripoff145784.htm
> Thoughts? Experiences?


My recommendation would be to stay away.  They are unresponsive and they are
thieves.

--
Richard Cook
[EMAIL PROTECTED]
T: 705-223-2000  ext 2010
 


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RE: [Asterisk-Users] Some more VOICEMAILMAIN issue...

[Tarpo, Louie]

You 
need to match the extensions you have in voicemail.conf to the callerid you're 
passing to voicemailmain().   For 5 digit extensions it would be 
exten => 22999,1,VoiceMailMain(s${CALLERIDNUM:-5)
 
 
Louie

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Mauro 
  ZaninSent: Tuesday, July 26, 2005 12:56 AMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] Some more 
  VOICEMAILMAIN issue...
  Hi everybody,
  I have corrected this line in extensions.conf by 
  stripping spaces off and now it executes:
   
  
  exten => 
  22999,1,VoiceMailMain(s${CALLERIDNUM})
  when it runs, the mail box number is asked and 
  password too. I expected no question were made, because I inserted 
  CALLERIDNUMBER and s in front of box number.
  Anybody knows why?
  Thank to you all, very kind members of this 
  list!
  Ciao
  Mauro
   
   
   
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[Asterisk-Users] Sound Quality Problems

[Robert Christian]

Thanks for reading this.  I’ve been pulling hair
for days trying to resolve this, and any help someone can give me would be very
much appreciated.

 

I have an Asterisk box that is basically a P4-3GHz, a
Digium-recommended SuperMicro X5SSE-GM motherboard, 2GB RAM, 250GB IDE
hard-drive with UDMA, a SoundBlaster Live! 24 sound card, a Digium Wildcard
TE110P, and a Digium Wildcard TDM400P with 4 FXS modules.  Right now the
Ethernet is hooked up directly to a single VoIP phone (Grandstream
GXP-2000).  I’m running kernel 2.6.12.3.

 

THE PROBLEM:  The problem I am having is when using the
speakers hooked into the SB Live! 24 card.  When I call through like an
intercom (my intention) and have the dialplan play an announcement (for testing
purposes, I have used both the included “you-sound-cute” and
“lots-o-monkeys”), it’s very choppy – to the point of
being laughably unacceptable.  Then when I talk over the
“intercom,” my voice sounds just as choppy.  Interestingly
enough, when I go to the console and dial the demo included in the Asterisk
sample configuration files (which I left in for testing), the woman’s
voice sounds fine (for 8KHz anyway).  But the demo is an IVR menu, and
when I dial “2” for more information (for example), it starts
playing the next message – all choppy.

 

I have read hundreds of mailing list posts and dozens of
how-tos and diagnostics suggestions online.  I’ve tweaked out
motherboard settings, kernel options, hard drive parameters, etc. with no
success.  I’ve tried literally dozens of ALSA configurations
thinking that the problem could be there.  I’ve focused on kernel
and IRQ optimizing, ALSA configuration, Asterisk configuration, and Zaptel
configuration (in case it’s a timing problem) with no success.  I
have been completely unable to resolve this problem.  I have a PRI line on
its way, but since “zttest” suggests the internal timing of the
TE110P is accurate within Digium specs, I don’t think external
synchronization will help anything.

 

Maybe the SB Live! 24 isn’t the best card to use for
the console intercom on this system.  But supposedly ALSA has full support
for it.

 

Any help at all is greatly appreciated.

Thank you.

 

- Robert






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Re: [Asterisk-Users] 100% CPU with Unicall and * head

[Denis Galvão - iSolve]

But which packages are you using?

libunicall
spandsp
asterisk
zaptel

D e n i s   G a l v ã o
iSolve - Solve Is Our Business
Av. Candido de Abreu, 526 1206B
CEP: 80530-000 - Curitiba - PR
+55 41 3252-2977   r 
http://www.isolve.com.br




On 26 de jul de 2005, at 12:27, Jose Chiantera wrote:


Hi denis

I am using Country ve,10,4Venezuela 10 ani 4 dnis

please let me know if I can do some test, or anything to help

Thanks

- Original Message - From: "Denis Galvão - iSolve"  
<[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"  


Sent: Monday, July 25, 2005 11:39 PM
Subject: Re: [Asterisk-Users] 100% CPU with Unicall and * head




Hi Jose.

What is the packages version that are you using? What MFCR2  
variant  are you using, I mean, wich country?


Maybe Steve could help us on it. I told him about this problem.

Keep in touch.

Denis.



On 25 de jul de 2005, at 15:36, Jose Chiantera wrote:



Hi,

I got the same error, when call from IP to digital link using   
MFCR2, I thinks the problem is a event not managed, If you find  
a  correction for this problem please let me know.
Maybe the error in the program channel.c, but I am not sure, now  
I  put some traces to try find what kind of event is.


regards
Jose


- Original Message - From: "Denis Galvão - iSolve"  
<[EMAIL PROTECTED]>

To: "Asterisk Users" 
Sent: Monday, July 25, 2005 9:47 AM
Subject: [Asterisk-Users] 100% CPU with Unicall and * head





Hi all.

When I place a call Im getting this error:

Jul 25 09:50:07 WARNING[3200] channel.c: Exception flag set on  
'UniCall/13-1', but no exception handler


Lots of this messages appeared on my Asterisk full log and the   
CPU  got 100%.


Topology:
Analog Phone - Hicom300H E1 - E1 Asterisk - IP Trunk

Problem:
1. Calls from Analog Phone through Asterisk is ok, but the   
messages appeared.
2. Calls from IP Trunk to Analog Phone is not ok andd the  
messages appeared too.


System:
Fedora Core 2 - Asterisk CVS HEAD(last week) - unicall-0.0.3pre3  
- spandsp-0.0.2pre18


unicall.conf
[channels]
language=br
context=from-internal
usecallerid=yes
hidecallerid=no
immediate=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
loglevel=255

protocolclass=mfcr2
protocolvariant=br,10,13
protocolend=co
group=1
callerid=asreceived
channel=>1-15
channel=>17-31
--

zaptel.conf
loadzone = us
defaultzone=us
span=1,0,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
--

Thanks.

Denis Galvão





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[Asterisk-Users] Dial using URI(web) or using FORM(web)

[JunkMail]

Hello!

I have an [EMAIL PROTECTED] instalation with 7 users working OK, and I'ld like
to implement either a
-- Web dial feature, where the user would fill one form field with a phone
number and a connection would be created between his extention and the
entered number.
OR
-- Dial using an URI (callto:x link in a web page), having AstTapi
installed and configured in all workstations.

Considering we spend all day manually dialing numbers from our database, one
of those solutions would be GREAT !
The second solution should be easier but the windows dialer app just won't
do...

Thanks in advance

M.G.

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[Asterisk-Users] Polycom 600 Presence indications - ALWAYS OFF-HOOK?

[Billy Dunn]

I am having one problem with the Polycom 600 phones.  All phones on the 
local network are fine and indicate presence to other phones perfectly.  
One phone that is outside the network can see presence indications of 
the other phones correctly, but that phone always shows as off the hook 
to certain phones on the network.  Any ideas?


Thanks!
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[Asterisk-Users] Asterisk 1.2 Release Plans

[Kevin P. Fleming]

As previously mentioned on the lists by Olle Johannson, we are actively 
trying to get Asterisk in shape for a 1.2 release within the next 60 
days. To accomplish this, we need a few things to happen:


1) A feature freeze - This will occur at the end of this month, with no 
new feature submissions accepted after July 31st. Any _pending_ feature 
patches in Mantis that have passed architecture review and functionality 
testing before August 1st can be accepted into 1.2, if they make it 
through the remainder of the review processes and are able to be merged 
before August 15th.


2) Progress on open bugs - There are a number of bugs open in Mantis 
that are waiting for the poster to provide additional information, test 
results, call traces, etc. We would much prefer to not release 1.2 with 
suspected problems already identified, but we cannot solve them without 
adequate input from you. If you have an open bug and are not in a 
position to continue providing assistance in solving it, please post a 
message to the mailing lists asking for volunteers to help replicate the 
problem so it can get resolved.


3) Testing - We need a _lot_ of help testing. If you have not previously 
tested CVS HEAD, please download it, read the UPGRADE.txt file and 
install it on one or more systems to play around with. Please do _not_ 
put it into a production environment unless you are willing to accept 
the consequences of that action. If you do find a bug or other issue, 
when you open a bug in Mantis, please try to provide _all_ the 
configuration information, call traces, etc. that the bug guidelines 
request, so that we don't waste 3-4 days just going back and forth 
requesting more information from you. If possible, join the #asterisk or 
#asterisk-dev IRC channel to find out exactly what debugging information 
will be required and how to produce it, if you don't already have that 
knowledge.


4) Release Candidates - I will produce the first release candidate on 
August 20th, with followup versions produced every week until we deem 
the release ready for public consumption. I expect it will require at 
least three -RC releases for us to get things in shape, so that means 
that 1.2 itself may be ready by September 15th.


We are very thankful for the community's help and support, and we want 
Asterisk 1.2 to be as important a release as 1.0 itself was. The number 
of new features, performance improvements, bug fixes and 
interoperability enhancements in CVS HEAD is astonishing, and a very 
large percentage of them came directly from community contributions. We 
hope that all of the 'non-developers' in the community will be able to 
help us 'shake out' the bugs and problems remaining in the code, so we 
can be assured of the most stable 1.2 release possible :-)

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