[Asterisk-Users] Re: 2 servers

[altus]

I think your's are better and work
Thanks 

David Cook writes: 

Quoting [EMAIL PROTECTED]:
From: "Kanuri, Seshu" <[EMAIL PROTECTED]> 

Dave, 

I am implementing this solution and would appreciate if you can send
me the doc at this email address - [EMAIL PROTECTED] 

Thanks
Seshu Kanuri 

Enough people have asked me for this that I will try and condense it for
the list. I admit I wanted to put it on the wiki and couldn't figure
out how to start a new page!!! (Maybe I'm just thick ;-( 

There is also another document on the wiki about the subject at
http://www.voip-info.org/wiki-Asterisk+-+dual+servers 

Anyhow, here is mine: 

Method 1
Rec'g Svr
iax.conf
[REC_SERVER]
type=user
host=my.calling.server.ca
secret=mysecret
context=local
trunk=yes 

Send'g Svr
extensions.conf
[mycontext]
exten =>
_5XXX,1,Dial(IAX2/REC_SERVER:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten => _5XXX,2,Hangup
exten => _5XXX,102,Hangup
Any call in the mycontext context on Calling Server to extensions
5000-5999 (mapped by extension _5XXX) will get sent to receiving server
(my.receiving.server.ca) into the local context on the receiving
server. 

Performing the same configuration in the opposite direction will allow
cross-calls between Asterisk systems.
Pros:
Simple, all references in one file per server.
Cons:
Information in dialing string will appear in logs inclusive of
user:password.
Dial string becomes very long. 

Method 2
Rev'g Svr
iax.conf[REC_SERVER]
type=user
host=my.calling.server.ca
secret=mysecret
context=local
trunk=yes 

Send'g Svr
iax.conf
[REMOTE_SERVER]
type=peer
host=my.receiving.server.ca
secret=mysecret
context=local
extensions.conf
[mycontext]
exten => 5XXX,1,Dial(IAX2/REMOTE_SERVER/${EXTEN})
exten => _5XXX,2,Hangup
exten => _5XXX,102,Hangup
Pros:
User:Password are stored in the calling server?s iax.conf file and not
part of the Dial string. This is more secure in that they are not
recorded in log in files.Dial strings much shorter and concise. 

Cons:
Calling server now must have iax.conf and extensions.conf coordinated
making setup a little more complicated.Must user ?type=? definition
correctly:Caller = ?peer?; Receiver = ?user?Type=friend is a
bi-directional relationship meaning both ?peer? and ?user? at the same
time. 

Unknown IP (Dynamic IP on one server)
Register Command
If the calling server does not have a fixed IP address or DNS namespace
then the iax.conf file description of the calling server located on the
receiving server should specify host=dynamic. 

If the calling server host is specified as dynamic, the calling server
must register with the receiving server with the register command. 

Rec'g Svr
iax.conf
[REC_SERVER]
type=user
host=dynamic
secret=mysecret
context=local
trunk=yes 

Send'g Svr
iax.conf
register => REC_SERVER:[EMAIL PROTECTED]
[REMOTE_SERVER]
type=peer
host=dynamic
context=local
extensions.conf
[mycontext]
exten => 5XXX,1,Dial(IAX2/REMOTE_SERVER/${EXTEN})
exten => _5XXX,2,Hangup
exten => _5XXX,102,Hangup 

I hope this is both accurate and helpful! 

dbc.
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Re: [Asterisk-Users] Perl AGI - no output from agi script to Asterisk

[Adam Goryachev]

On Wed, 2004-08-25 at 10:00, Robert Rozman wrote:
> Hi,
> 
> I'm writting some Perl AGI scripts. I got lucky on some working, but try to
> debug other.
> 
> I put a lot of verbose() or print STDERR commands into them, but no output
> appears on asterisk CLI (I run asterisk -vvvrgc after normal asterisk
> process is started).
> 
> I get all kinds of other messages with higher verbosity, but none from Perl
> AGI script (I also get some response from eagi-test so I guess this is Perl
> AGI problem).
> 
> Anyone using Perl scripts verbosity successfully under RC 2 ?

I used the asterisk-perl module like this:
$AGI->verbose("something to print\n",1);

and it seemed to sortof work... I hope that helps.

Regards,
Adam



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Re: [Asterisk-Users] strange problem PBX-Asterisk

[Kurt Bauer]

Thanks for the hints, 'overlapdial=yes' did the trick.
br,
kurt
--On Tuesday, August 24, 2004 10:08:08 PM +0200 Peter Svensson 
<[EMAIL PROTECTED]> wrote:

On Tue, 24 Aug 2004, Christian Victor wrote:
> maybe I oversee somth. very obvious, but I'm a little puzzled about
> the  following 'error':
>
> When I make a call from the PBX to * I get number not available, but
> on  debug I see, that asterisk is searching just for the first digit
> in the  extension, which of course doesn't exist, eg:
I seems that you PBX uses "Overlap Dial" and transmits the extensien
digit by digit and Asterisk expects the extension to be "en bloc". So
when it receives anything from the PBX (wich is in this case the first
digit) Asterisk thinks that this is the whole block of extension.
Don't know how to fix it though. ;-)
This in case you are on a PRI: see "overlapdial" at
  http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf
It needs to be set on pri links where ovarlap dialing is used, even
incoming towards asterisk.
Without more information on the connections between the systems and the
configuration it is hard to figure out what is wrong.
Peter
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RE: [Asterisk-Users] Swissvoice IP10S and RTP Port Operation

[Florian Overkamp]

Hi Matthew,

> -Original Message-
> What is even better is that this coincides with the 
> "Terminating on result
> 502 from [EMAIL PROTECTED]" error I get in *
> 
> So, I am guessing these are related. Any help here would be 
> greatly appreciated. I am so close to getting this phone 
> working in MGCP mode.

Did you post your phone config to the list ? (mirror pages or something) I
don't really understand why it's so hard for you where my phones did basic
functions (calling/being called) almost right out of the box.

Florian

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[Asterisk-Users] asterisk directory service

[aster esk]

hello 
 any one knows how to configure directory listing in asterisk



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RE: [Asterisk-Users] SIP "unphones"

[Jay Milk]

Actually, right now its proof-of-concept in an existing home, which I
built three years ago, fully wired.  I have a minimum of two cat5 in
each room, as well as attic and garage, all home-run into the basement.
I want to see if I can get this to work, in order to know how to prewire
the next house we're breaking ground on in a few months.

The idea is to get an intercom/PA system going, as well as utilize the
PBX as a baby-monitor (dial ext, connect mic in one -- soon two --
bedrooms to speaker in master bedroom).

I got one Sipura in the office, the rest in the basement -- like I said,
wiring isn't the issue.  I'll research some doorphones -- still think
that's so far the best lead.  The Grandstream doesn't seem like a good
solution, since it inserts that announcement-beep before turning on
speaker.

> -Original Message-
> From: Chris Shaw [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, August 24, 2004 6:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] SIP "unphones"
> 
> 
> Is this for a remodel or a new home? Depending on your home's 
> telephone layout, some homes have a 66-Block or a 110-Block 
> in the basement or crawlspace under the stairs... you could 
> make your setup look really pretty by wiring the intercom 
> into the 66-Block and then wiring the other end of the 
> 66-block to a wall jack in the room where you keep * box, 
> then you can plug the sipura into that...
> 
> -Chris
> 
> - Original Message -
> From: "Jay Milk" <[EMAIL PROTECTED]>
> To: "'Chris Shaw'" <[EMAIL PROTECTED]>; "'Asterisk Users 
> Mailing List - Non-Commercial Discussion'" 
> <[EMAIL PROTECTED]>
> Sent: Tuesday, August 24, 2004 4:08 PM
> Subject: RE: [Asterisk-Users] SIP "unphones"
> 
> 
> > Yep, great idea, that's what's next -- and I have two extra 
> extensions
> > (Sipura)
> >
> > > -Original Message-
> > > From: Chris Shaw [mailto:[EMAIL PROTECTED]
> > > Sent: Tuesday, August 24, 2004 12:03 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [Asterisk-Users] SIP "unphones"
> > >
> > >
> > > Check out my ATA idea though, with a regular cheap analog 
> doorphone 
> > > and a HTX86 or even Sipura, you can program the ATA to dial an 
> > > extension as soon as the button on the intercom is 
> pressed and then 
> > > with some extension logic you can do neat things... You can get a 
> > > doorphone anywhere even radio shack I think and the HTX86 is like 
> > > $60-70...
> > >
> > >
> > > - Original Message -
> > > From: "Jay Milk" <[EMAIL PROTECTED]>
> > > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 
> > > <[EMAIL PROTECTED]>
> > > Sent: Monday, August 23, 2004 7:16 PM
> > > Subject: RE: [Asterisk-Users] SIP "unphones"
> > >
> > >
> > > > Thank you -- funny thing is, I had the same bookmarked, but it 
> > > > just seemed too expensive for the application -- for 
> $300, I can 
> > > > stick a cheap IP phone in a hole in the wall :)  I 
> think it's time 
> > > > to get a Budgetone.
> > > >
> > > > > -Original Message-
> > > > > From: Chris Shaw [mailto:[EMAIL PROTECTED]
> > > > > Sent: Monday, August 23, 2004 3:52 PM
> > > > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > > > Subject: Re: [Asterisk-Users] SIP "unphones"
> > > > >
> > > > >
> > > > > I recently saw something just like this and I had it
> > > bookmarked...
> > > > > It looks like what you're talking about, but I don't
> > > think it uses
> > > > > SIP. Rather some proprietary protocol that transmit
> > > RTP... I could
> > > > > be wrong... Check it out...
> > > > >
> > > > > http://www.digitalacoustics.com/lanplay.htm
> > > > >
> > > > > I would agree that it really should be SIP, you 
> wouldn't want to 
> > > > > have to rip it out of the wall when the protocol becomes
> > > obsolete or
> > > > > when a SIP-Compliant alternative comes out...
> > > > >
> > > > > -Chris
> > > > >
> > > > > - Original Message -
> > > > > From: "Jay Milk" <[EMAIL PROTECTED]>
> > > > > To: <[EMAIL PROTECTED]>
> > > > > Sent: Monday, August 23, 2004 7:55 AM
> > > > > Subject: [Asterisk-Users] SIP "unphones"
> > > > >
> > > > >
> > > > > > Does anyone know if there are additional SIP devices out
> > > > > there which
> > > > > > aren't phones?  I'm basically looking for a fully-automatic 
> > > > > > SIP speakerphone.  I'd like to be able to dial a 
> sip-extension
> > > > > and make an
> > > > > > announcement (PA) and/or simply listen in to a room 
> > > > > > (baby-monitor). Yes, I know, some of the more advanced
> > > phones can
> > > > > > be configured to behave like that, but it seems to 
> a waste of 
> > > > > > money to have
> > > > > all those
> > > > > > fancy displays and keys tucked away behind a 
> speakergrille and 
> > > > > > drywall.
> > > > > >
> > > > > > BTW, I'm not dead-set on SIP, but it seems to be the
> > > most logical
> > > > > > protocol for this app (NOTIFY msg can carr

RE: [Asterisk-Users] Asterisk to Vonage

[Jay Milk]

AFAIK, unlimited -- since the Softphone is metered, you're being charged
for each outgoing minute anyway.  I've had four simultaneous incoming
calls going at the same time, but no extra charge.

> -Original Message-
> From: Richard Cook [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, August 24, 2004 10:01 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Asterisk to Vonage
> 
> 
> Hello,
> 
> Do you know how many simultaneous outbound/inbound 
> connections are allowed through Vonage?
> 
> --
> Richard Cook
> [EMAIL PROTECTED]
> Tel: 705-497-9320
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Assaf Benharoosh
> Sent: Tuesday, August 24, 2004 10:59 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Asterisk to Vonage
> 
>  
> I made it work:
> 
> Sip.conf:
> 
> register => 1:@atlas-east.vonage.net:5060
> 
> [vonage]
> type=friend
> username=1
> secret=
> host=atlas-east.vonage.net
> port=5060
> allow=all
> maxexpirey=15
> dtmfmode=inband
> fromuser=1
> fromdomain=atlas-east.vonage.net
> canreinvite=no
> nat=yes
> context=
> 
> Extensions.conf:
> 
> exten => _1,1,
> 
> Enjoy
> Assaf.
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Paterson, Mark
> Sent: Tuesday, August 24, 2004 7:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Asterisk to Vonage
> 
> Honestly, do you think I would ask for help on the list if I 
> hadn't come up with any successful results on my own??
> 
>  Just asking if anyone has made this work. If so what rev of 
> * were they running and what do their configs look like.
> 
> Thanks.
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
> Sent: Tuesday, August 24, 2004 6:06 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Asterisk to Vonage
> 
> Yes, search google for
> > asterisk vonage working site:lists.digium.com
> 
> > -Original Message-
> > From: Paterson, Mark [mailto:[EMAIL PROTECTED]
> > Sent: Tuesday, August 24, 2004 11:19 AM
> > To: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] Asterisk to Vonage
> > 
> > 
> > I'm trying to connect my Asterisk server via sip using my 
> vonage soft
> > phone account. Has any anyone successfully got to work? I get error 
> > from
> > asterisk saying:  == Parsing '/etc/asterisk/sip.conf':   == Parsing
> > '/etc/asterisk/sip.conf': Found
> > Aug 24 11:01:11 WARNING[1125329600]: acl.c:146 ast_get_ip: 
> > Unable to lookup '216.115.25.199:5061' when trying to register with 
> > the vonage sip proxy. Any examples would be greatly appreciated.
> > 
> > 
> > Rgs,
> > mark
> >  
> > 
> > ___
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> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
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> 
> 
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Re: [Asterisk-Users] SMP Performance

[joachim]

Send me the quad and i'll send you a 200$ pc to do this job.
The quad is heavily overpowered.
Joachim.
At 22:00 24/08/2004, you wrote:
content-class: urn:content-classes:message
Content-Type: multipart/alternative;
boundary="_=_NextPart_001_01C48A15.130BF232"
We're looking at implementing Asterisk in our department in the near 
future, we're looking at anywhere from 15-25 extensions. The machine we 
were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 
1GB of ram. I've heard bad things about running Asterisk on SMP machines? 
Would we be running into any performance issues with this machine?

Tim Jackson
Network Engineer
Angelina County, Texas
(936)639-4827 office
(936)414-6723 mobile
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RE: [Asterisk-Users] Asterisk to Vonage

[Jay Milk]

Honestly, do you think I would reply with a google-search if what you're
looking for wasn't answered in there?

Click here, my lazy friend:
http://lists.digium.com/pipermail/asterisk-users/2004-June/052678.html

Config files all pre-chewed for your consumption.

> -Original Message-
> From: Paterson, Mark [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, August 24, 2004 6:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Asterisk to Vonage
> 
> 
> Honestly, do you think I would ask for help on the list if I 
> hadn't come up with any successful results on my own??
> 
>  Just asking if anyone has made this work. If so what rev of 
> * were they running and what do their configs look like.
> 
> Thanks.
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
> Sent: Tuesday, August 24, 2004 6:06 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Asterisk to Vonage
> 
> Yes, search google for
> > asterisk vonage working site:lists.digium.com
> 
> > -Original Message-
> > From: Paterson, Mark [mailto:[EMAIL PROTECTED]
> > Sent: Tuesday, August 24, 2004 11:19 AM
> > To: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] Asterisk to Vonage
> > 
> > 
> > I'm trying to connect my Asterisk server via sip using my 
> vonage soft
> > phone account. Has any anyone successfully got to work? I get error 
> > from
> > asterisk saying:  == Parsing '/etc/asterisk/sip.conf':   == Parsing
> > '/etc/asterisk/sip.conf': Found
> > Aug 24 11:01:11 WARNING[1125329600]: acl.c:146 ast_get_ip: 
> > Unable to lookup '216.115.25.199:5061' when trying to register with 
> > the vonage sip proxy. Any examples would be greatly appreciated.
> > 
> > 
> > Rgs,
> > mark
> >  
> > 
> > ___
> > Asterisk-Users mailing list
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> > To 
> > UNSUBSCRIBE or update options visit:
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> 
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RE: [Asterisk-Users] desparate for help DEV LITE KIT

[Sean Cheesman]

You know, if you purchased the kit from Digium it includes support
direct from them.  Especially if you're "desparate" for help.  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of steve
Sent: Wednesday, August 25, 2004 12:03 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] desparate for help DEV LITE KIT


Has anyone had any luck with the DEV LITE KIT?
I'm getting very erradic behavior.
I'll start it up and it will answer maybe one or two calls then hang
leaving my phone in an off hook state or giving a loud shrill tone. It
keeps saying "Red Alarm". Once it hangs it will not answer the phone
again until I reboot. Starting and stopping Asterisk or reloading the
drivers does not help.

I thought maybe it was an IRQ conflict so I removed every card in my
system except the X100P and it still hangs. I've swapped the order of
the devices and tried different slots.

I'm baffled. All I want is a simple 1x1 PBX to keep telemarketers from
ringing my phone. If I can't get this working I'm having my phone
disconected. lol

Thanks,
-s

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Re: [Asterisk-Users] desparate for help DEV LITE KIT

[Darryl Ross]

steve wrote:
I'm baffled. All I want is a simple 1x1 PBX to keep telemarketers from
ringing my phone.
If I can't get this working I'm having my phone disconected. lol
Surely the shrill tone would be good for keeping telemarketers away? ;)
--
Darryl Ross
Senior Network Engineer
OEG Australia
Email: [EMAIL PROTECTED]
Phone: 08 81228363
Office: 08 81226361
If you want to live up to the whole "There is more than one way to
do it" slogan, you have to give someone a swiss army chainsaw ...
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[Asterisk-Users] desparate for help DEV LITE KIT

[steve]

Has anyone had any luck with the DEV LITE KIT?
I'm getting very erradic behavior.
I'll start it up and it will answer maybe one or two calls then hang
leaving
my phone in an off hook state or giving a loud shrill tone.
It keeps saying "Red Alarm". Once it hangs it will not answer the phone
again until I reboot.
Starting and stopping Asterisk or reloading the drivers does not help.

I thought maybe it was an IRQ conflict so I removed every card in my
system
except the X100P and it still hangs.
I've swapped the order of the devices and tried different slots.

I'm baffled. All I want is a simple 1x1 PBX to keep telemarketers from
ringing my phone.
If I can't get this working I'm having my phone disconected. lol

Thanks,
-s

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Re: [Asterisk-Users] how to collect user entered digits

[Ryan Courtnage]

John Millican wrote:
Thanks all for your patience. I found my answer in a post on a totally
unrelated question (I new if I kept reading all posts ...)  saw this in a
post for call forwarding.
exten => *72,1,Answer
exten => *72,2,Wait(1)
exten => *72,3,BackGround(allison7/please-enter-your)
exten => *72,4,Playback(extension)
exten => *72,5,Playback(then-press-pound)
exten => *72,6,Playback(beep)
exten => *72,7,Read(fromext)
Had not seen anything on "Read" anywhere else, must have been looking in all
the wrong places. this is a simple solution to my problem.  
FYI - Read() is described on the wiki:
http://voip-info.org/wiki-Asterisk+cmd+Read
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Re: [Asterisk-Users] Dell PowerEdge 750 rackmount

[Craig Guy]

We use KVM for all our servers here.  Will try with nothing plugged in on
next reboot.

Craig
- Original Message - 
From: "Scott Stingel" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[EMAIL PROTECTED]>
Sent: Wednesday, August 25, 2004 10:36 AM
Subject: RE: [Asterisk-Users] Dell PowerEdge 750 rackmount


> Hi Craig-
>
> I'm also interested in the other fellow's question:  do the newer DL 320's
> require the keyboard to be present to pass the power on test and boot up?
>
> Thanks
> Scott
>
> Scott M. Stingel
> President,
> Emerging Voice Technology, Inc.
> Palo Alto California & London England
> www.evtmedia.com
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy
> Sent: Tuesday, August 24, 2004 7:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Dell PowerEdge 750 rackmount
>
> Hi Steven,
>
> We have just built an Asterisk 1U server using a HP DL320 and a TE410p
card.
> Is working well, however we were caught out when it arrived without the
> combo floppy/cdrom which is an expensive 'option'.  We ended up installing
> FC2 via PXE.  Is very very noisy, even with fans set to 'low'.  In fact
with
> fans on normal speed it is louder than the rest of the machines in the
> server room combined.
>
> Craig
>
> - Original Message -
> From: "Scott Stingel" <[EMAIL PROTECTED]>
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <[EMAIL PROTECTED]>
> Sent: Wednesday, August 25, 2004 2:56 AM
> Subject: RE: [Asterisk-Users] Dell PowerEdge 750 rackmount
>
>
> >
> > > Steven Critchfield <[EMAIL PROTECTED]> wrote:
> >
> > > buying a 1u server is much better than building it as there are
> > > a lot of cooling problems to overcome
> >
> > You're right about that - I learned a lot about 1U cooling and
low-profile
> > fans in the last one I built.  It was fun, but now they want 10 boxes,
> > delivered fast!
> >
> > > The big thing to look into is what PCI busses the machine supports.
> > > We were very surprised with our Dell when it came with a PCIX slot
> > > and a 66mhz 64bit slot
> >
> > I noticed the PowerEdge 750 seems to have one of each: 32- and 64-bit
> PCI's,
> > both brought to the rear panel - nice.
> >
> > BUT, I can't get the Dell's fast enough for this customer, so now I'm
> > looking at the HP Proliant DL-320.
> >
> > Regards
> > Scott Stingel
> >
> >
> > Scott M. Stingel
> > President,
> > Emerging Voice Technology, Inc.
> > Palo Alto California & London England
> > www.evtmedia.com
> >
> >
> >
> > ___
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> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] RTP and NAT

[WPJ mail]

I understand that setting "canreinvite = 
yes" in sip.conf allows clients to send RTP streams directly to each 
other and this works only if there is no NAT involved. But if I want to use 
two SIP phones behind firewalls and using NAT to be able to send RTP packets to 
each other bypassing Asterisk, is there a way to do that?
Can this be done if I used IAX instead of SIP? 
What exactly does the "notransfer" option in iax.conf do?
 
Thanks in advance.
Regards,
Mitul
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[Asterisk-Users] Voicepulse incoming / dial extension

[Anil Purohit]

 

 

All:

 

I am trying to use Voicepulse as
my incoming line and want the caller to simply dial the extension of the party
they want to reach.

 

Here is my problem:

 

- 
the first time they dial it works
fine and  I see the following on my
console

 

 

Aug 24 23:14:31 DEBUG[-1126876240]: chan_sip.c:4408 build_route: build_route: Contact
hop: 

    -- Executing Wait("SIP/s00227156-a5ef", "2") in new
stack

    -- Executing Answer("SIP/s00227156-a5ef", "") in new
stack

    -- Executing DigitTimeout("SIP/s00227156-a5ef",
"10") in new stack

    -- Set Digit Timeout to 10

    -- Executing ResponseTimeout("SIP/s00227156-a5ef",
"30") in new stack

    -- Set Response Timeout to
30

    -- Executing BackGround("SIP/s00227156-a5ef",
"welcome-mainmenu") in new stack

Aug 24 23:14:33 DEBUG[-1221325904]: rtp.c:1146 ast_rtp_write: Ooh, format changed from UNKN to ULAW

Aug 24 23:14:33 DEBUG[-1221325904]: channel.c:1101 ast_settimeout: Scheduling timer at 160 sample intervals

    -- Playing 'welcome-mainmenu' (language 'en')

Aug 24 23:14:33 DEBUG[-1126876240]: chan_sip.c:817 __sip_ack: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 103: Found

Aug 24 23:14:37 DEBUG[-1221325904]: rtp.c:186 send_dtmf: Sending dtmf: 50 (2),
at 66.234.228.137

Aug 24 23:14:37 DEBUG[-1221325904]: channel.c:1101 ast_settimeout: Scheduling timer at 0 sample intervals

Aug 24 23:14:37 DEBUG[-1221325904]: pbx.c:1801 ast_pbx_run: Oooh, got something
to jump out with ('2')!

Aug 24 23:14:38 DEBUG[-1221325904]: rtp.c:186 send_dtmf: Sending dtmf: 48 (0),
at 66.234.228.137

Aug 24 23:14:38 DEBUG[-1221325904]: rtp.c:186 send_dtmf: Sending dtmf: 48 (0),
at 66.234.228.137

Aug 24 23:14:39 DEBUG[-1221325904]: rtp.c:186 send_dtmf: Sending dtmf: 49 (1),
at 66.234.228.137

  == CDR updated on SIP/s00227156-a5ef  

    -- Executing Dial("SIP/s00227156-a5ef",
"SIP/2001|60|tr") in new stack

Aug 24 23:14:39 DEBUG[-1221325904]: app_dial.c:468 dial_exec: SIMPLE DIAL (NO URL)

 

 

The problem is that the second time the
caller dials my voicepulse number,
I do not see the “Executing ……. “ debug
statements in my console and it ignores my dtmf and
the call is NOT transferred to the extension. The call is just dropped after
the timeout of 30 secs. Nothing on my console.

 

Any ideas would be greatly appreciated.
Also, please do a reply-all so that I can receive e-mail directly. 

 

Thanks,

 

Anil

 

 






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RE: [Asterisk-Users] Asterisk to Vonage

[Richard Cook]

Hello,

Do you know how many simultaneous outbound/inbound connections are allowed
through Vonage?

--
Richard Cook
[EMAIL PROTECTED]
Tel: 705-497-9320



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Assaf
Benharoosh
Sent: Tuesday, August 24, 2004 10:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk to Vonage

 
I made it work:

Sip.conf:

register => 1:@atlas-east.vonage.net:5060

[vonage]
type=friend
username=1
secret=
host=atlas-east.vonage.net
port=5060
allow=all
maxexpirey=15
dtmfmode=inband
fromuser=1
fromdomain=atlas-east.vonage.net
canreinvite=no
nat=yes
context=

Extensions.conf:

exten => _1,1,

Enjoy
Assaf.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paterson, Mark
Sent: Tuesday, August 24, 2004 7:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk to Vonage

Honestly, do you think I would ask for help on the list if I hadn't come up
with any successful results on my own??

 Just asking if anyone has made this work. If so what rev of * were they
running and what do their configs look like.

Thanks.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Tuesday, August 24, 2004 6:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk to Vonage

Yes, search google for
> asterisk vonage working site:lists.digium.com

> -Original Message-
> From: Paterson, Mark [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, August 24, 2004 11:19 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Asterisk to Vonage
> 
> 
> I'm trying to connect my Asterisk server via sip using my vonage soft 
> phone account. Has any anyone successfully got to work? I get error 
> from
> asterisk saying:  == Parsing '/etc/asterisk/sip.conf':   == Parsing
> '/etc/asterisk/sip.conf': Found
> Aug 24 11:01:11 WARNING[1125329600]: acl.c:146 ast_get_ip: 
> Unable to lookup '216.115.25.199:5061' when trying to register with 
> the vonage sip proxy. Any examples would be greatly appreciated.
> 
> 
> Rgs,
> mark
>  
> 
> ___
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RE: [Asterisk-Users] Asterisk to Vonage

[Assaf Benharoosh]

 
I made it work:

Sip.conf:

register => 1:@atlas-east.vonage.net:5060

[vonage]
type=friend
username=1
secret=
host=atlas-east.vonage.net
port=5060
allow=all
maxexpirey=15
dtmfmode=inband
fromuser=1
fromdomain=atlas-east.vonage.net
canreinvite=no
nat=yes
context=

Extensions.conf:

exten => _1,1,

Enjoy
Assaf.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paterson,
Mark
Sent: Tuesday, August 24, 2004 7:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk to Vonage

Honestly, do you think I would ask for help on the list if I hadn't come
up with any successful results on my own??

 Just asking if anyone has made this work. If so what rev of * were they
running and what do their configs look like.

Thanks.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Tuesday, August 24, 2004 6:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk to Vonage

Yes, search google for
> asterisk vonage working site:lists.digium.com

> -Original Message-
> From: Paterson, Mark [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, August 24, 2004 11:19 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Asterisk to Vonage
> 
> 
> I'm trying to connect my Asterisk server via sip using my vonage soft 
> phone account. Has any anyone successfully got to work? I get error 
> from
> asterisk saying:  == Parsing '/etc/asterisk/sip.conf':   == Parsing
> '/etc/asterisk/sip.conf': Found
> Aug 24 11:01:11 WARNING[1125329600]: acl.c:146 ast_get_ip: 
> Unable to lookup '216.115.25.199:5061' when trying to register with 
> the vonage sip proxy. Any examples would be greatly appreciated.
> 
> 
> Rgs,
> mark
>  
> 
> ___
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> UNSUBSCRIBE or update options visit:
>
http://lists.digium.com/mailman/listinfo/asterisk-users



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RE: [Asterisk-Users] Asterisk to Vonage

[Greg Hill]

On Tue, 24 Aug 2004, Paterson, Mark wrote:

> Honestly, do you think I would ask for help on the list if I hadn't come
> up with any successful results on my own??
>
>  Just asking if anyone has made this work. If so what rev of * were they

> running and what do their configs look like.

Jay does have it working. The search suggestion he gave is likely to bring
up the message he posted to this list some weeks ago. That message does
contain an example for sip.conf which shows how he has his configured.

Greg



>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
> Sent: Tuesday, August 24, 2004 6:06 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Asterisk to Vonage
>
> Yes, search google for
> > asterisk vonage working site:lists.digium.com
>
> > -Original Message-
> > From: Paterson, Mark [mailto:[EMAIL PROTECTED]
> > Sent: Tuesday, August 24, 2004 11:19 AM
> > To: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] Asterisk to Vonage
> >
> >
> > I'm trying to connect my Asterisk server via sip using my vonage soft
> > phone account. Has any anyone successfully got to work? I get error
> > from
> > asterisk saying:  == Parsing '/etc/asterisk/sip.conf':   == Parsing
> > '/etc/asterisk/sip.conf': Found
> > Aug 24 11:01:11 WARNING[1125329600]: acl.c:146 ast_get_ip:
> > Unable to lookup '216.115.25.199:5061' when trying to register with
> > the vonage sip proxy. Any examples would be greatly appreciated.
> >
> >
> > Rgs,
> > mark
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/aster> isk-users
> > To
> > UNSUBSCRIBE or update options visit:
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
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>
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RE: [Asterisk-Users] Perl AGI - no output from agi script to Aste risk

[mattf]

print to standard error output in your perl script:

print STDERR "This is how perl-AGI prints to Asterisk CLI output\n";


MATT---


-Original Message-
From: Robert Rozman [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 24, 2004 8:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Perl AGI - no output from agi script to
Asterisk


Hi,

I'm writting some Perl AGI scripts. I got lucky on some working, but try to
debug other.

I put a lot of verbose() or print STDERR commands into them, but no output
appears on asterisk CLI (I run asterisk -vvvrgc after normal asterisk
process is started).

I get all kinds of other messages with higher verbosity, but none from Perl
AGI script (I also get some response from eagi-test so I guess this is Perl
AGI problem).

Anyone using Perl scripts verbosity successfully under RC 2 ?

Thanks in advance,

Robert.


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Re: [Asterisk-Users] newb question regarding DTMF

[Greg Hill]

On Tue, 24 Aug 2004, Erik Anderson wrote:

> On Tue, 24 Aug 2004 11:46:36 -0600 (MDT), Greg Hill
> <[EMAIL PROTECTED]> wrote:
> > x-lite uses the RFC2833 style for DTMF "out of the box" (it can be set to
> > transmit inband). You need dtmfmode=rfc2833 in [general] or in the section
> > for your x-lite user.
>
> That's what I've read, and I have added dtmfmode=rfc2833 in my
> sip.conf...see this snippet:
>
> [xlite1]
> ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
> ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
> type=friend
> username=xlite
> callerid="Jane Smith" <5678>
> host=dynamic
> nat=yes   ; X-Lite is behind a NAT router
> canreinvite=no; Typically set to NO if behind NAT
> disallow=all
> allow=gsm ; GSM consumes far less bandwidth than ulaw
> allow=ulaw
> allow=alaw
> dtmfmode=rfc2833
>
> I've applied that change and restarted asterisk, but no dice...

Dial the extension, then on the * CLI use 'sip show channels' to get the
name of the active channel. Next use 'sip show channel ___' to get info on
that particular channel (you can type the first few characters and use tab
completion; no need to type the whole string!). Scan through the output to
see whether asterisk is really using rfc2833 for that channel. If it is,
then the problem is likely in the x-lite config. If not, try moving
dtmfmode to the general section of sip.conf

Greg


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RE: [Asterisk-Users] Dell PowerEdge 750 rackmount

[Scott Stingel]

Hi Craig-

I'm also interested in the other fellow's question:  do the newer DL 320's
require the keyboard to be present to pass the power on test and boot up?

Thanks
Scott
 
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California & London England
www.evtmedia.com 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy
Sent: Tuesday, August 24, 2004 7:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dell PowerEdge 750 rackmount

Hi Steven,

We have just built an Asterisk 1U server using a HP DL320 and a TE410p card.
Is working well, however we were caught out when it arrived without the
combo floppy/cdrom which is an expensive 'option'.  We ended up installing
FC2 via PXE.  Is very very noisy, even with fans set to 'low'.  In fact with
fans on normal speed it is louder than the rest of the machines in the
server room combined.

Craig

- Original Message -
From: "Scott Stingel" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[EMAIL PROTECTED]>
Sent: Wednesday, August 25, 2004 2:56 AM
Subject: RE: [Asterisk-Users] Dell PowerEdge 750 rackmount


>
> > Steven Critchfield <[EMAIL PROTECTED]> wrote:
>
> > buying a 1u server is much better than building it as there are
> > a lot of cooling problems to overcome
>
> You're right about that - I learned a lot about 1U cooling and low-profile
> fans in the last one I built.  It was fun, but now they want 10 boxes,
> delivered fast!
>
> > The big thing to look into is what PCI busses the machine supports.
> > We were very surprised with our Dell when it came with a PCIX slot
> > and a 66mhz 64bit slot
>
> I noticed the PowerEdge 750 seems to have one of each: 32- and 64-bit
PCI's,
> both brought to the rear panel - nice.
>
> BUT, I can't get the Dell's fast enough for this customer, so now I'm
> looking at the HP Proliant DL-320.
>
> Regards
> Scott Stingel
>
>
> Scott M. Stingel
> President,
> Emerging Voice Technology, Inc.
> Palo Alto California & London England
> www.evtmedia.com
>
>
>
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RE: [Asterisk-Users] how to collect user entered digits

[John Millican]

Thanks all for your patience. I found my answer in a post on a totally
unrelated question (I new if I kept reading all posts ...)  saw this in a
post for call forwarding.
exten => *72,1,Answer
exten => *72,2,Wait(1)
exten => *72,3,BackGround(allison7/please-enter-your)
exten => *72,4,Playback(extension)
exten => *72,5,Playback(then-press-pound)
exten => *72,6,Playback(beep)
exten => *72,7,Read(fromext)

Had not seen anything on "Read" anywhere else, must have been looking in all
the wrong places. this is a simple solution to my problem.  I have been
reading VOIP-info.org wiki, asterisk handbook, google searches.  Can anyone
point me to some other places that I can search through?
John

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John
Millican
Sent: Tuesday, August 24, 2004 4:13 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] how to collect user entered digits





On Fri, Aug 20, 2004 at 01:19:54PM -0400, John Millican said:
> I have been searching thru all docs that I can find on wiki and such but
can
> not get an answer.  I am trying to collect a date from user input in the
> form of digits dialed from the phone to use in an agi script to do a
> database look up.  I have tried to use "Get Data filename, timeout,
> maxdigits " in the agi script. In * console I get message saying playing
> filename but it exits as soon as it starts. Could I collect the digits
some
> how before going to script and then send as an arg maybe?

Check out the source to the privacy manager for a simple example...

apps/app_privacy.c


Well Mr. Reed, and all others on list who might help, I guess I am not as
smart as I hoped I was:-)

I looked at the apps_privacy file.  I thought I could write something
similar and compile then call from dial plan.  What I get is a lot of errors
about unreferenced ast_app_streamfile, ast_app_getdata, ... among others. I
used all the same includes in Marks script but still no go. I tried to find
where these references came from in the apps_privacy and could not find
them. I typically code in  C++ with QT 3.3 and would like to be able to use
the DB drivers that I have in QT for PostgreSQL since I will be querying an
app on a separate machine for customer info based on callid and a date in
the form of digits dialed by the caller. Any help in this direction would be
greatly appreciated.  My original plan was an AGI script, is this a bad
idea?  Call levels will most likly not be very high.

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[Asterisk-Users] Zaptel/Zapata and SIP relationship

[Steve D.]

In my test configuration, I have a Budgetone, an Iaxy and two computers
running X-Lite.  My server has one X100P in it (no line hooked up yet).
Currently, I can call from any phone to any phone except on one, when
the caller calls me, I can't hear the caller (using an X-Lite) but the
caller can hear me.  If I call him, everything works fine.  If I pick up
another phone while two phones are talking, the connection dies (both
phones go to busy signal).

I am very new to Asterisk and I am reading everything I can get my hands
on but this one is eluding me.  What exactly is the relationship between
Sip to Sip and Sip to IAX and Channels?  Do I need more channels defined
in Zaptel/Zapata?  Everytime I try to add a channel ztcfg errors out.
Does each connection require a separate channel?  One other thing, are
the context names in one file related to context names in another file?

What I am trying to accomplish is putting the Asterisk box at my office
and have 3 or 4 SIP/IAXY phones located at various points on a wireless
network (40 sq miles).

And failing all else, what is the going hourly rate for commercial
support?  I'm sure someone who knows what they are doing could have this
figured out in an hour instead of me banging my head against the wall
for another week.  At least that way, I'd have a working system to
reverse engineer.

Thanks for any help.

Steve D.


Typical entries:

Sip.conf
[2002]
type=friend
username=2002
secret=2002
host=dynamic
context=internal
mailbox=2002
defaultip=192.168.140.51
callerid="Kevin" <2002>
canreinvite=no
reinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
allow=gsm

Iax.conf entry:
[2050]
type=friend
;accountcode=iaxy
host=dynamic
username=2050
auth=md5
secret=2050
context=internal
disallow=all
allow=ulaw
callerid="Iaxy Box" <2050>
trunk=no
mailbox=2050

Extension.conf (typical)
[internal]
exten => 2002,1,Dial(SIP/2002,20)
exten => 2002,2,Voicemail(u2002)
exten => 2002,102,Voicemail(b2002)
exten => 2002,103,Hangup

exten => 2050,1,Dial(IAX2/2050,20)
exten => 2050,2,Voicemail(u2050)
exten => 2050,102,Voicemail(b2050)
exten => 2050,103,Hangup

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Re: [Asterisk-Users] Dell PowerEdge 750 rackmount

[Craig Guy]

Hi Steven,

We have just built an Asterisk 1U server using a HP DL320 and a TE410p card.
Is working well, however we were caught out when it arrived without the
combo floppy/cdrom which is an expensive 'option'.  We ended up installing
FC2 via PXE.  Is very very noisy, even with fans set to 'low'.  In fact with
fans on normal speed it is louder than the rest of the machines in the
server room combined.

Craig

- Original Message - 
From: "Scott Stingel" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[EMAIL PROTECTED]>
Sent: Wednesday, August 25, 2004 2:56 AM
Subject: RE: [Asterisk-Users] Dell PowerEdge 750 rackmount


>
> > Steven Critchfield <[EMAIL PROTECTED]> wrote:
>
> > buying a 1u server is much better than building it as there are
> > a lot of cooling problems to overcome
>
> You're right about that - I learned a lot about 1U cooling and low-profile
> fans in the last one I built.  It was fun, but now they want 10 boxes,
> delivered fast!
>
> > The big thing to look into is what PCI busses the machine supports.
> > We were very surprised with our Dell when it came with a PCIX slot
> > and a 66mhz 64bit slot
>
> I noticed the PowerEdge 750 seems to have one of each: 32- and 64-bit
PCI's,
> both brought to the rear panel - nice.
>
> BUT, I can't get the Dell's fast enough for this customer, so now I'm
> looking at the HP Proliant DL-320.
>
> Regards
> Scott Stingel
>
>
> Scott M. Stingel
> President,
> Emerging Voice Technology, Inc.
> Palo Alto California & London England
> www.evtmedia.com
>
>
>
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Re: [Asterisk-Users] system reboot often?

[Steven Critchfield]

On Tue, 2004-08-24 at 20:42, el Flynn wrote:
> Steven Critchfield wrote:
> > 
> > It isn't advisable to have "many" Zap cards in a machine. If you are
> > adding analog cards, you quickly run out of PCI slots before you get
> > very far. If you are adding T1/E1 cards, you quickly get to a point
> > where it is too risky to have that many circuits on a single x86 PC. The
> > whole point of X86 PCs is that they are cheap enough to put several in
> > use when you need it instead of building one behemoth machine.
> 
> This may have crept up elsewhere in the list but I thought it might be 
> relevant in this thread:
> 
> Would having 2 of the Quad T1/E1 cards in a single machine, handling 
> about 140 ZAP -> ZAP channels be ok? Or would that be overloading the 
> box and/or asterisk?

Zap to Zap shouldn't be too bad except for echo cancel and PCI load. I
think it also depends on how fast you are setting up and tearing down.
Too many variables to say yes or no to your situation.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] system reboot often?

[Scott Stingel]

Flynn:

As far as max capacity, you might want to check out:

http://www.voip-info.org/wiki-Asterisk+dimensioning

Regards
Scott 


Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California & London England
www.evtmedia.com 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of el Flynn
Sent: Tuesday, August 24, 2004 6:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] system reboot often?

Steven Critchfield wrote:
> 
> It isn't advisable to have "many" Zap cards in a machine. If you are 
> adding analog cards, you quickly run out of PCI slots before you get 
> very far. If you are adding T1/E1 cards, you quickly get to a point 
> where it is too risky to have that many circuits on a single x86 PC. 
> The whole point of X86 PCs is that they are cheap enough to put 
> several in use when you need it instead of building one behemoth machine.

This may have crept up elsewhere in the list but I thought it might be
relevant in this thread:

Would having 2 of the Quad T1/E1 cards in a single machine, handling about
140 ZAP -> ZAP channels be ok? Or would that be overloading the box and/or
asterisk?

Flynn

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Re: [Asterisk-Users] system reboot often?

[el Flynn]

Steven Critchfield wrote:
It isn't advisable to have "many" Zap cards in a machine. If you are
adding analog cards, you quickly run out of PCI slots before you get
very far. If you are adding T1/E1 cards, you quickly get to a point
where it is too risky to have that many circuits on a single x86 PC. The
whole point of X86 PCs is that they are cheap enough to put several in
use when you need it instead of building one behemoth machine.
This may have crept up elsewhere in the list but I thought it might be 
relevant in this thread:

Would having 2 of the Quad T1/E1 cards in a single machine, handling 
about 140 ZAP -> ZAP channels be ok? Or would that be overloading the 
box and/or asterisk?

Flynn
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Re: [Asterisk-Users] Asterisk and software Raid

[Chris]

> > Consider hot swappable SCSI RAID 1 instead of IDE. You'll appreciate it
> > every couple of years when you lose a disk but the PBX stays up.
>
> I hot swap on my 3Ware IDE RAID card all the time.

Aye and with SATA hot swapping is even easier provided the driver notifies
the kernel that the disk geometry has changed... This could probably be done
with a /sbin/hotplug kind of thing...

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Re: [Asterisk-Users] Can I have same numbers in different contexts ?

[Steven Critchfield]

On Tue, 2004-08-24 at 19:20, Robert Rozman wrote:
> Hi,
> 
> I'd like to ask simple question. I have IVR menu in its own context. When I
> go into that IVR menu (new context), can I have also extension numbers that
> are same as in other contexts (1-10 for intance)  or must they be unique ?

They only have to be unique in the context you are working in and any
contexts you include into the one you are working in. 

-- 
Steven Critchfield <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] Asterisk and software Raid

[Greg Boehnlein]

On Mon, 23 Aug 2004, William Boehlke wrote:

> Consider hot swappable SCSI RAID 1 instead of IDE. You'll appreciate it
> every couple of years when you lose a disk but the PBX stays up. 

I hot swap on my 3Ware IDE RAID card all the time.

-- 
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] sip to sip calls thru asterisk

[Karl Brose]

It's not clear how you are making the call.
You should be able to call directly from either phone to the other by 
dialing 5011 or 5012, respectively, if
your context "local"  indeed contains the those extensions, which is not 
clear from your configuration excerpts.

But it seems you are calling another user on carolina.net, who is 
registered with that provider from your asterisk,
so the call will loop back to Asterisk.   SIP does not really have a 
good way to handle such loopbacks, and
therefore you get the error.
If you want to make this work you need to load a second SIP channel 
driver on your asterisk listening on a different port,
The changes are not difficult.

It also seems that both phones are sitting on the same IP address and 
port,  how can that be?
Oh, I see the error message is actually coming from the sip phone, and 
it's because those phones
have the same IP address, and therefore a loop is detected there.  Is 
this just ONE phone with two
proxy-accounts or personalities?


Gary Carr wrote:
I have a test box setup and I can make outbound calls on the PSTN thru 
the diguim card, however I can not make a sip user to sip user call by 
dialing the extensions. I am getting the following error.
 
-- Called cisco7960
-- Got SIP response 482 "Loop Detected" back from 208.218.14.123
  == No one is available to answer at this time
 
 
 
CLI> sip show peers
Name/usernameHostDyn Nat ACL Mask Port 
Status
 
cisco7960/5052   208.218.14.123   D   N  255.255.255.255  5060 
OK (1 ms)
garycarr/5011208.218.14.123   D   N  255.255.255.255  5060 
OK (1 ms)
 
 
sip.conf statements
 
register => [EMAIL PROTECTED]/5011 

register => [EMAIL PROTECTED]/5052 

 
[cisco7960]
type=friend
host=dynamic
nat=yes
qualify=200
dtmfmode=rfc2833
canreinvite=no
mailbox=5052
callerid="Cisco 7960"
context=local
 
[garycarr]
type=friend
host=dynamic
nat=yes
qualify=200
dtmfmode=rfc2833
canreinvite=no
mailbox=5011
callerid="Gary Carr"
context=local
 
extensions.conf statements
 
exten => 5011,1,dial(SIP/garycarr,20,tr)
exten => 5052,1,dial(SIP/cisco7960,20,tr)
 
Is this a possible nat issue? I can make a good call from behind the 
firewall doing sip to pstn so it seems 2 way traffic thru the firewall 
is working.
 
 
I am still sifting thru the sip debug info but anyone has any ideas 
that would be great.
 
 
Gary
 


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Re: [Asterisk-Users] system reboot often?

[Steven Critchfield]

On Tue, 2004-08-24 at 19:54, Michael George wrote:
> On Sat, Aug 21, 2004 at 08:35:22AM -0500, Lyle Giese wrote:
> > This doesn't answer your question completely, but I have noticed that
> > inserting and removing the kernal modules doesn't work all that well and
> > that rebooting is a better answer at that point.
> > 
> > Have you verified that you are not IRQ sharing?  * really doesn't like that,
> > even though other applications are ok with it.
> 
> If * doesn't like IRQ sharing, how can I scale up a machine with several Zap
> card?  Or is it okay to share IRQ's with other Digium cards?

It isn't advisable to have "many" Zap cards in a machine. If you are
adding analog cards, you quickly run out of PCI slots before you get
very far. If you are adding T1/E1 cards, you quickly get to a point
where it is too risky to have that many circuits on a single x86 PC. The
whole point of X86 PCs is that they are cheap enough to put several in
use when you need it instead of building one behemoth machine.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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[Asterisk-Users] How can i configure extensions.conf.

[maya mn]

I have TDM40B, TDM04B cards, 4 analog and digital
phones. 

First I want to use 4 analog phones with my TDM40B
card. I would like to dial between 4 analog phones.
The dialing numbers for 4 analog phone will be
800,801,802 and 803.


These are my conf files.

/etc/zaptel.conf

fxsks=1-4
fxoks=5-8
loadzone = us
defaultzone=us
;;;
/etc/asterisk/zapata.conf

[channels]

relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes

rxgain=0.0
txgain=0.0

immediate=no
context=bell
 
signalling=fxs_ks
channel=1-4
signalling=fxo_ks
callerid="Anolag 1"<801>
channel=5
signalling=fxo_ks
callerid="Anolag 2"<802>
channel=6
signalling=fxo_ks
callerid="Anolag 3"<803>
channel=7
signalling=fxo_ks
callerid="Anolag 4"<804>
channel=8
;;

extensions.conf

exten => 800,1,Answer
exten => 800,2,Playback(new/password)
exten => 800,3,AGI(agi-test.agi,${pass})
exten => 800,4,Hangup

exten => 801,1,Playback(new/password)
exten => 801,2,SetMusicOnHold,loud
exten => 801,3,WaitMusicOnHold,30
exten => 801,4,Hangup
;;

I have tried this long time, But I can't do this so I
need some help. Please help.

Best regards.



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Re: [Asterisk-Users] system reboot often?

[Michael George]

On Sat, Aug 21, 2004 at 08:35:22AM -0500, Lyle Giese wrote:
> This doesn't answer your question completely, but I have noticed that
> inserting and removing the kernal modules doesn't work all that well and
> that rebooting is a better answer at that point.
> 
> Have you verified that you are not IRQ sharing?  * really doesn't like that,
> even though other applications are ok with it.

If * doesn't like IRQ sharing, how can I scale up a machine with several Zap
card?  Or is it okay to share IRQ's with other Digium cards?

> - Original Message - 
> From: "Michael George" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Saturday, August 21, 2004 6:42 AM
> Subject: [Asterisk-Users] system reboot often?
> 
> 
> > I just deployed * on my home system last Sunday.  2x since then the Zap
> > hardware seems to have malfunctioned on some way.
> >
> > One time it would just screech out one FXS, even though it would ring.
> The
> > other time * would bridge to my FXO but it never got out on the line.  I
> have
> > a new TDM400 with 3 FXS and 1 FXO.
> >
> > Both times I tried unloading the zaptel drivers (which worked) and
> reloading
> > them, which failed.  A reboot of the system brought everything back.
> >
> > Is this common?  Are there ways to minimize this?  Would a different PCI
> slot
> > possibly make a difference?  Or a different system?  Is this just a
> chronic
> > problem with the Digium hardware?
> >
> > Thanks!
> >
> > -- 
> > -M
> >
> > There are 10 kinds of people in this world:
> > Those who can count in binary and those who cannot.
> > ___
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-- 
-M

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

[Robert Rozman]

Autoreply - I downloaded exact version of mpg123 and it works - but it
worked before also with another version, but ended of no known reason.

Regards,

Robert.

- Original Message - 
From: "Robert Rozman" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, August 25, 2004 12:15 AM
Subject: Re: [Asterisk-Users] MusicOnHold problem


> Hi,
>
> I've checked again and I can run it from command line. mpg123 is in
> /usr/bin.
>
> Is there any way to debug this problem more deeply ?
>
> What files for musiconhold feature should I reinstall and where in case
> something got corrupted ?
>
> Thanks in advance,
>
> Robert.
>
> - Original Message - 
> From: "Chris Stenton" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <[EMAIL PROTECTED]>
> Sent: Monday, August 23, 2004 12:30 PM
> Subject: Re: [Asterisk-Users] MusicOnHold problem
>
>
> > It can't find mpg123 or you  don't have any mp3 files in your moh
> directory.
> >
> > - Original Message - 
> > From: "Robert Rozman" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Monday, August 23, 2004 12:58 AM
> > Subject: [Asterisk-Users] MusicOnHold problem
> >
> >
> > > Hi,
> > >
> > > I had music on hold working but now don't know what happened.
> > >
> > > I get :
> > > WARNING[...] res_musiconhold.c:334 moh0_exec: Unable to start music on
> > hold
> > > (class '') on channel SIP...
> > >
> > > Any ideas what is wrong ?
> > >
> > > Regards,
> > >
> > > Robert.
> > >
> > > ___
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> > > 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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[Asterisk-Users] Can I have same numbers in different contexts ?

[Robert Rozman]

Hi,

I'd like to ask simple question. I have IVR menu in its own context. When I
go into that IVR menu (new context), can I have also extension numbers that
are same as in other contexts (1-10 for intance)  or must they be unique ?

Regards,

Robert.

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[Asterisk-Users] Perl AGI - no output from agi script to Asterisk

[Robert Rozman]

Hi,

I'm writting some Perl AGI scripts. I got lucky on some working, but try to
debug other.

I put a lot of verbose() or print STDERR commands into them, but no output
appears on asterisk CLI (I run asterisk -vvvrgc after normal asterisk
process is started).

I get all kinds of other messages with higher verbosity, but none from Perl
AGI script (I also get some response from eagi-test so I guess this is Perl
AGI problem).

Anyone using Perl scripts verbosity successfully under RC 2 ?

Thanks in advance,

Robert.


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Re: [Asterisk-Users] X100P connected to Meridan-1 system will not disconnect call

[Leo Ann Boon]

Sanity check - what tone zone are you using? Make sure the disconnect 
tone generated by the Meridian matches what Asterisk is expecting. Some 
PBXs have disconnect tones that sound like a dial tone, i.e. a 
continuous tone. Connect an analog phone to the line and make a call, 
then ask the remote end to hangup. What do you hear?

Mark Farquharson wrote:
 

Hi all,
I am hoping that someone has experienced a similar problem to mine. I 
have a X100P connected to a Nortel Meridan-1 PBX system via an analog 
extension. When the extension is called and then hung up the X100P 
refuses to disconnect the call. I have to reload Asterisk to free up 
the channel. Also is there a way to make the X100P answer the call in 
less than the default tow rings?

Below are the settings that I am currently using in my configuration 
files:

 

In Zapata.conf:
 

Busydetect=yes
Callprogress=yes
Signaling=fxs_ls
 

Zaptel.conf
 

 

Signaling=fxsls
 

I have also tried using ground start but it does not work.

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Re: [Asterisk-Users] RC2 and Netmeeting 3.01 ?

[Zineddin Karzazi]

 --- Robert Rozman <[EMAIL PROTECTED]> schrieb: 
> Hi,
> 
> I'd kindly ask for any guidance how to setup
> Netmeeting to work with
> Asterisk.
> 
> I've setup Asterisk as Gateway, selected GSM codec,
> and I'm able to call
> local extensions (no calls into PBX functions) but
> get no sound.
> 
> Any hint, advice ?
> 
> Anyone using Netmeeting (maybe also windows
> messenger) with Asterisk
> sucessfully ?
> 
> Thanks in advance,
> 
> regards,
> 
> Robert.

I have same Problems with Netmeeting, just wanted to
test H.323 with Astersik , it rings, but as soon as i
answer it dissconnects.
im getting the Following Error:

oh323_exception: OH323/R27469: Invalid format of RTP
addresses. 
Aug 13 10:19:05 ERROR[524304]: chan_oh323.c:1933
oh323_write: OH323/R27469: Failed to create smoother.

Today i tried Openphone (H.323 Client) and it works.

Zineddin. 







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Re: [Asterisk-Users] Grandstream Budgetone BT-101 and VoipJet

[Chris Shaw]

> Is anyone using this combination successfully?  I have a dell 500sc
> running rh9 and asterisk 1.0rc1.  It is configured with an x100p.  I
> have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone
> BT-101.  I have signed up with Voipjet (they use iax2).  I also have
> an FWD number via iax2.  I can sucessfully call back and forth to all
> devices via zap, sip, and fwd.  I can successfully place calls using
> voipjet with everything except the grandstream.  When I place a
> voipjet call with the grandstream, I can hear the party I'm calling,
> but they can't hear me.  I have tried all the different codecs the
> grandstream supports without luck.  I am running the 1.0.5.10
> firmware.  I've emailed voipjet support about it, but they don't have
> one.

Can you send your * debug output? It's kinda hard to figure out what's going
on without it...

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Re: [Asterisk-Users] Remotely change call forward

[Ryan Courtnage]

Russell Horn wrote:
Is it possible using asterisk to allow someone to dial in and remotely
change where their call is forwarded to?
For example, I'm working from home so I want my calls to go to 555 1234,
now I need to go out for a bit so I'd like to phone the office and using
DTMF tell the asterisk PBX to now forward my calls to my cell phone 555
3456
Has anyone implimented anything like this?
Sure:
; prompting call forward app - forwards entered extension
exten => *72,1,Answer
exten => *72,2,Wait(1)
exten => *72,3,BackGround(allison7/please-enter-your)
exten => *72,4,Playback(extension)
exten => *72,5,Playback(then-press-pound)
exten => *72,6,Playback(beep)
exten => *72,7,Read(fromext)
exten => *72,8,Wait(1)
exten => *72,9,BackGround(ent-target-attendant)
exten => *72,10,Background(then-press-pound)
exten => *72,11,Playback(beep)
exten => *72,12,Read(toext)
exten => *72,13,Wait(1)
exten => *72,14,DBput(CF/${fromext}=${toext})
exten => *72,15,Playback(call-fwd-unconditional)
exten => *72,16,Playback(for)
exten => *72,17,Playback(extension)
exten => *72,18,SayDigits(${fromext})
exten => *72,19,Playback(is-set-to)
exten => *72,20,SayDigits(${toext})
exten => *72,21,Macro(hangupcall)
See dialparites.agi for an example of how the check for CF before 
dialing an extension.

Ryan
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RE: [Asterisk-Users] Asterisk to Vonage

[Paterson, Mark]

Honestly, do you think I would ask for help on the list if I hadn't come
up with any successful results on my own??

 Just asking if anyone has made this work. If so what rev of * were they
running and what do their configs look like.

Thanks.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Tuesday, August 24, 2004 6:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk to Vonage

Yes, search google for
> asterisk vonage working site:lists.digium.com

> -Original Message-
> From: Paterson, Mark [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, August 24, 2004 11:19 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Asterisk to Vonage
> 
> 
> I'm trying to connect my Asterisk server via sip using my vonage soft 
> phone account. Has any anyone successfully got to work? I get error 
> from
> asterisk saying:  == Parsing '/etc/asterisk/sip.conf':   == Parsing
> '/etc/asterisk/sip.conf': Found
> Aug 24 11:01:11 WARNING[1125329600]: acl.c:146 ast_get_ip: 
> Unable to lookup '216.115.25.199:5061' when trying to register with 
> the vonage sip proxy. Any examples would be greatly appreciated.
> 
> 
> Rgs,
> mark
>  
> 
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Re: [Asterisk-Users] SIP "unphones"

[Chris Shaw]

Is this for a remodel or a new home? Depending on your home's telephone
layout, some homes have a 66-Block or a 110-Block in the basement or
crawlspace under the stairs... you could make your setup look really pretty
by wiring the intercom into the 66-Block and then wiring the other end of
the 66-block to a wall jack in the room where you keep * box, then you can
plug the sipura into that...

-Chris

- Original Message -
From: "Jay Milk" <[EMAIL PROTECTED]>
To: "'Chris Shaw'" <[EMAIL PROTECTED]>; "'Asterisk Users Mailing List -
Non-Commercial Discussion'" <[EMAIL PROTECTED]>
Sent: Tuesday, August 24, 2004 4:08 PM
Subject: RE: [Asterisk-Users] SIP "unphones"


> Yep, great idea, that's what's next -- and I have two extra extensions
> (Sipura)
>
> > -Original Message-
> > From: Chris Shaw [mailto:[EMAIL PROTECTED]
> > Sent: Tuesday, August 24, 2004 12:03 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] SIP "unphones"
> >
> >
> > Check out my ATA idea though, with a regular cheap analog
> > doorphone and a HTX86 or even Sipura, you can program the ATA
> > to dial an extension as soon as the button on the intercom is
> > pressed and then with some extension logic you can do neat
> > things... You can get a doorphone anywhere even radio shack I
> > think and the HTX86 is like $60-70...
> >
> >
> > - Original Message -
> > From: "Jay Milk" <[EMAIL PROTECTED]>
> > To: "'Asterisk Users Mailing List - Non-Commercial
> > Discussion'" <[EMAIL PROTECTED]>
> > Sent: Monday, August 23, 2004 7:16 PM
> > Subject: RE: [Asterisk-Users] SIP "unphones"
> >
> >
> > > Thank you -- funny thing is, I had the same bookmarked, but it just
> > > seemed too expensive for the application -- for $300, I can stick a
> > > cheap IP phone in a hole in the wall :)  I think it's time to get a
> > > Budgetone.
> > >
> > > > -Original Message-
> > > > From: Chris Shaw [mailto:[EMAIL PROTECTED]
> > > > Sent: Monday, August 23, 2004 3:52 PM
> > > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > > Subject: Re: [Asterisk-Users] SIP "unphones"
> > > >
> > > >
> > > > I recently saw something just like this and I had it
> > bookmarked...
> > > > It looks like what you're talking about, but I don't
> > think it uses
> > > > SIP. Rather some proprietary protocol that transmit
> > RTP... I could
> > > > be wrong... Check it out...
> > > >
> > > > http://www.digitalacoustics.com/lanplay.htm
> > > >
> > > > I would agree that it really should be SIP, you wouldn't want to
> > > > have to rip it out of the wall when the protocol becomes
> > obsolete or
> > > > when a SIP-Compliant alternative comes out...
> > > >
> > > > -Chris
> > > >
> > > > - Original Message -
> > > > From: "Jay Milk" <[EMAIL PROTECTED]>
> > > > To: <[EMAIL PROTECTED]>
> > > > Sent: Monday, August 23, 2004 7:55 AM
> > > > Subject: [Asterisk-Users] SIP "unphones"
> > > >
> > > >
> > > > > Does anyone know if there are additional SIP devices out
> > > > there which
> > > > > aren't phones?  I'm basically looking for a fully-automatic SIP
> > > > > speakerphone.  I'd like to be able to dial a sip-extension
> > > > and make an
> > > > > announcement (PA) and/or simply listen in to a room
> > > > > (baby-monitor). Yes, I know, some of the more advanced
> > phones can
> > > > > be configured to behave like that, but it seems to a waste of
> > > > > money to have
> > > > all those
> > > > > fancy displays and keys tucked away behind a speakergrille and
> > > > > drywall.
> > > > >
> > > > > BTW, I'm not dead-set on SIP, but it seems to be the
> > most logical
> > > > > protocol for this app (NOTIFY msg can carry directions on
> > > > > mike/speaker/two-way, etc)
> > > > >
> > > > > ___
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> > > >
> > > > ___
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> > > > UNSUBSCRIBE or update options visit:
> > > >
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> > >
> > > ___
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> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
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> > To
> > UNSUBSCRIBE or update options visit:
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> 

Re: [Asterisk-Users] Hardware for PBX with 4 incoming/outgoing lines and 20 phones

[William Suffill]

Andrew,

Sounds like it could be a good fit for your needs. Although that
raises many questions as to how exactly you should deploy it. If you
have a good Internet connection to the office in question you could
perhaps use VOIP termination for your outbound calls instead of the
current 4 PSTN lines. This way if all 20 people on the system could
make calls at once or just 4 and you would only pay per minute on the
usage.

If you have good analog phones already then probably the channel bank
route is the best way to support that many phones. Otherwise you could
get IP Phones such as the Cisco 79xx , Polycom's or Snoms.

Hard to say really what's best without knowing more of how much phone
usage is in question and how much you would like to invest in an
Asterisk/VOIP solution.

Hopefully this along with the other posts you get on the list help you
on your deployment. I'd be curious as to what you finally decide on
after you weight the options.

On Tue, 24 Aug 2004 17:11:36 -0600, Andrew Elchuk
<[EMAIL PROTECTED]> wrote:
> Hi
> 
> I am interested in setting up an Asterisk PBX in my office with digium
> hardware, and I just have a few questions in regards to what I would
> need.  It is my understanding that an FXO card is used to interface with
> an incoming/outgoing phone line, and an FXS card is used for interfacing
> with a phone within the system.  Currently we have 4 incoming/outgoing
> phone lines and would like to have 20 phones in the system.  In order to
> accomodate this, would you either reccommend having 1 TDM04B (4 FXO
> modules on it) for the 4 incoming/outgoing lines, and 5 TDM40B (4 FXS
> modules on each) for the 20 phones we would have in the system.  Or
> would you reccommend 1 TDM04B for the 4 incoming/outgoing lines, and a
> T100P connected to a channel bank of some sort to connect to the
> internal phones?  If you reccommend the T100P and channel bank, where do
> you suggest I get an FXS channel bank?  Please let me know if I got any
> of this mixed up (like if I got the FXO and FXS cards mixed up) and
> thank you in advance for your help in us deciding the hardware we need
> for a new PBX.
> 
> Andrew Elchuk
> 
> P.S.  We are currently using an X-Like software phone with a free world
> dialup account for communication with our other office in a different
> city.  My question is if I can configure the extensions.conf to connect
> to a free world dialup number when executing a dial command, or would I
> need to edit sip.conf as well or some other configs or is that even
> possible?  Thank you again.
> 
> --
> Andrew Elchuk
> Technical Associate
> Cronus Technologies
> 248 - 111 Research Drive
> Saskatoon, SK  S7N 2X8
> Tel: (306) 652-5798 ext. 112
> Fax: (306) 652-5799
> Toll Free: 1-877-655-5798
> http://www.cronustech.com
> 
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Re: [Asterisk-Users] Grandstream Budgetone BT-101 and VoipJet

[Glenn Dalgliesh]

I would make the following changes to your sip.conf and restart asterisk.  I
have seen alot of issues with regard to codec with * and grandstream. both
have a little to do with it but this should keep it working.

[general]
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm


- Original Message - 
From: "William Suffill" <[EMAIL PROTECTED]>
To: "John Week" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion" <[EMAIL PROTECTED]>
Sent: Tuesday, August 24, 2004 7:24 PM
Subject: Re: [Asterisk-Users] Grandstream Budgetone BT-101 and VoipJet


> I had this issue with a grandstream as well a week or so ago and have
> yet to solve the issue. Until I get my Budgetone here physically again
> I won't be able to mess with it hands on. What did you use for
> codec/signaling and did your asterisk box see any warnings or errors?
>
> On Tue, 24 Aug 2004 15:56:38 -0700, John Week <[EMAIL PROTECTED]> wrote:
> > Is anyone using this combination successfully?  I have a dell 500sc
> > running rh9 and asterisk 1.0rc1.  It is configured with an x100p.  I
> > have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone
> > BT-101.  I have signed up with Voipjet (they use iax2).  I also have
> > an FWD number via iax2.  I can sucessfully call back and forth to all
> > devices via zap, sip, and fwd.  I can successfully place calls using
> > voipjet with everything except the grandstream.  When I place a
> > voipjet call with the grandstream, I can hear the party I'm calling,
> > but they can't hear me.  I have tried all the different codecs the
> > grandstream supports without luck.  I am running the 1.0.5.10
> > firmware.  I've emailed voipjet support about it, but they don't have
> > one.
> >
> > thanks,
> >
> > John
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Re: [Asterisk-Users] SIP "unphones"

[William Suffill]

Post some pictures when you are all done. Looks like an interesting
task just no need to dive into it myself at this time.

On Tue, 24 Aug 2004 18:08:28 -0500, Jay Milk <[EMAIL PROTECTED]> wrote:
> Yep, great idea, that's what's next -- and I have two extra extensions
> (Sipura)
> 
> 
> 
> > -Original Message-
> > From: Chris Shaw [mailto:[EMAIL PROTECTED]
> > Sent: Tuesday, August 24, 2004 12:03 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] SIP "unphones"
> >
> >
> > Check out my ATA idea though, with a regular cheap analog
> > doorphone and a HTX86 or even Sipura, you can program the ATA
> > to dial an extension as soon as the button on the intercom is
> > pressed and then with some extension logic you can do neat
> > things... You can get a doorphone anywhere even radio shack I
> > think and the HTX86 is like $60-70...
> >
> >
> > - Original Message -
> > From: "Jay Milk" <[EMAIL PROTECTED]>
> > To: "'Asterisk Users Mailing List - Non-Commercial
> > Discussion'" <[EMAIL PROTECTED]>
> > Sent: Monday, August 23, 2004 7:16 PM
> > Subject: RE: [Asterisk-Users] SIP "unphones"
> >
> >
> > > Thank you -- funny thing is, I had the same bookmarked, but it just
> > > seemed too expensive for the application -- for $300, I can stick a
> > > cheap IP phone in a hole in the wall :)  I think it's time to get a
> > > Budgetone.
> > >
> > > > -Original Message-
> > > > From: Chris Shaw [mailto:[EMAIL PROTECTED]
> > > > Sent: Monday, August 23, 2004 3:52 PM
> > > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > > Subject: Re: [Asterisk-Users] SIP "unphones"
> > > >
> > > >
> > > > I recently saw something just like this and I had it
> > bookmarked...
> > > > It looks like what you're talking about, but I don't
> > think it uses
> > > > SIP. Rather some proprietary protocol that transmit
> > RTP... I could
> > > > be wrong... Check it out...
> > > >
> > > > http://www.digitalacoustics.com/lanplay.htm
> > > >
> > > > I would agree that it really should be SIP, you wouldn't want to
> > > > have to rip it out of the wall when the protocol becomes
> > obsolete or
> > > > when a SIP-Compliant alternative comes out...
> > > >
> > > > -Chris
> > > >
> > > > - Original Message -
> > > > From: "Jay Milk" <[EMAIL PROTECTED]>
> > > > To: <[EMAIL PROTECTED]>
> > > > Sent: Monday, August 23, 2004 7:55 AM
> > > > Subject: [Asterisk-Users] SIP "unphones"
> > > >
> > > >
> > > > > Does anyone know if there are additional SIP devices out
> > > > there which
> > > > > aren't phones?  I'm basically looking for a fully-automatic SIP
> > > > > speakerphone.  I'd like to be able to dial a sip-extension
> > > > and make an
> > > > > announcement (PA) and/or simply listen in to a room
> > > > > (baby-monitor). Yes, I know, some of the more advanced
> > phones can
> > > > > be configured to behave like that, but it seems to a waste of
> > > > > money to have
> > > > all those
> > > > > fancy displays and keys tucked away behind a speakergrille and
> > > > > drywall.
> > > > >
> > > > > BTW, I'm not dead-set on SIP, but it seems to be the
> > most logical
> > > > > protocol for this app (NOTIFY msg can carry directions on
> > > > > mike/speaker/two-way, etc)
> > > > >
> > > > > ___
> > > > > Asterisk-Users mailing list [EMAIL PROTECTED]
> > > > > 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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> > > > [EMAIL PROTECTED]
> > > > http://lists.digium.com/mailman/listinfo/aster> isk-users To
> > > > UNSUBSCRIBE or update options visit:
> > > >
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> > >
> > > ___
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> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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> > To
> > UNSUBSCRIBE or update options visit:
> >
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> 
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Re: [Asterisk-Users] SIP "unphones"

[Adam Goryachev]

On Wed, 2004-08-25 at 03:02, Chris Shaw wrote:
> Check out my ATA idea though, with a regular cheap analog doorphone and a
> HTX86 or even Sipura, you can program the ATA to dial an extension as soon
> as the button on the intercom is pressed and then with some extension logic
> you can do neat things... You can get a doorphone anywhere even radio shack
> I think and the HTX86 is like $60-70...

The Grandstream also supports dial on pickup, but, with some extension
logic, why not use the voicemail button, which seems to have the ability
to make the grandstream go off-hook AND dial an extension all in one
neat package... Of course, IMHO, there are some dis-advantages to this
method, including:

You are sending power and ethernet to a complex device, which is
somewhat difficult to access/replace.
Using an analog doorphone, with the IAXy or ATA device in your computer
room/whatever, you have complete physical access to the 'complex'
devices, and not much should? go wrong with the analog doorphone
portion. Also, probably won't need to run new cable, since you might
already have telephone cable run through.

Of course, it would be nice to use a phone with PoE, but you do still
have the problem of accessing the device if it fails for some reason.

Just my 0.02c

Regards,
Adam


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Re: [Asterisk-Users] Grandstream Budgetone BT-101 and VoipJet

[William Suffill]

I had this issue with a grandstream as well a week or so ago and have
yet to solve the issue. Until I get my Budgetone here physically again
I won't be able to mess with it hands on. What did you use for
codec/signaling and did your asterisk box see any warnings or errors?

On Tue, 24 Aug 2004 15:56:38 -0700, John Week <[EMAIL PROTECTED]> wrote:
> Is anyone using this combination successfully?  I have a dell 500sc
> running rh9 and asterisk 1.0rc1.  It is configured with an x100p.  I
> have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone
> BT-101.  I have signed up with Voipjet (they use iax2).  I also have
> an FWD number via iax2.  I can sucessfully call back and forth to all
> devices via zap, sip, and fwd.  I can successfully place calls using
> voipjet with everything except the grandstream.  When I place a
> voipjet call with the grandstream, I can hear the party I'm calling,
> but they can't hear me.  I have tried all the different codecs the
> grandstream supports without luck.  I am running the 1.0.5.10
> firmware.  I've emailed voipjet support about it, but they don't have
> one.
> 
> thanks,
> 
> John
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Re: [Asterisk-Users] Asterisk to Vonage

[Chris Shaw]

I hold no ill will towards Vonage but I have to say honestly... ewww...

They've already made their feelings quite clear by refusing to allow people
to bring their own devices and taking steps to even hide their SIP servers
(changing the port from the RFC standard 5060 to 5061 for example.) Why not
go with someone who's actually willing to allow you to use Asterisk and any
phone you want like NuFone, BroadVoice, IconnectHere or a host of others
instead of trying to hack Vonage...

Again... Ewww...

-Chris


- Original Message -
From: "Jay Milk" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[EMAIL PROTECTED]>
Sent: Tuesday, August 24, 2004 4:06 PM
Subject: RE: [Asterisk-Users] Asterisk to Vonage


> Yes, search google for
> > asterisk vonage working site:lists.digium.com
>
> > -Original Message-
> > From: Paterson, Mark [mailto:[EMAIL PROTECTED]
> > Sent: Tuesday, August 24, 2004 11:19 AM
> > To: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] Asterisk to Vonage
> >
> >
> > I'm trying to connect my Asterisk server via sip using my
> > vonage soft phone account. Has any anyone successfully got to
> > work? I get error from
> > asterisk saying:  == Parsing '/etc/asterisk/sip.conf':   == Parsing
> > '/etc/asterisk/sip.conf': Found
> > Aug 24 11:01:11 WARNING[1125329600]: acl.c:146 ast_get_ip:
> > Unable to lookup '216.115.25.199:5061' when trying to
> > register with the vonage sip proxy. Any examples would be
> > greatly appreciated.
> >
> >
> > Rgs,
> > mark
> >
> >
> > ___
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> > [EMAIL PROTECTED]
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> > To
> > UNSUBSCRIBE or update options visit:
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
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[Asterisk-Users] Hardware for PBX with 4 incoming/outgoing lines and 20 phones

[Andrew Elchuk]

Hi
I am interested in setting up an Asterisk PBX in my office with digium 
hardware, and I just have a few questions in regards to what I would 
need.  It is my understanding that an FXO card is used to interface with 
an incoming/outgoing phone line, and an FXS card is used for interfacing 
with a phone within the system.  Currently we have 4 incoming/outgoing 
phone lines and would like to have 20 phones in the system.  In order to 
accomodate this, would you either reccommend having 1 TDM04B (4 FXO 
modules on it) for the 4 incoming/outgoing lines, and 5 TDM40B (4 FXS 
modules on each) for the 20 phones we would have in the system.  Or 
would you reccommend 1 TDM04B for the 4 incoming/outgoing lines, and a 
T100P connected to a channel bank of some sort to connect to the 
internal phones?  If you reccommend the T100P and channel bank, where do 
you suggest I get an FXS channel bank?  Please let me know if I got any 
of this mixed up (like if I got the FXO and FXS cards mixed up) and 
thank you in advance for your help in us deciding the hardware we need 
for a new PBX.

Andrew Elchuk
P.S.  We are currently using an X-Like software phone with a free world 
dialup account for communication with our other office in a different 
city.  My question is if I can configure the extensions.conf to connect 
to a free world dialup number when executing a dial command, or would I 
need to edit sip.conf as well or some other configs or is that even 
possible?  Thank you again.

--
Andrew Elchuk
Technical Associate
Cronus Technologies
248 - 111 Research Drive
Saskatoon, SK  S7N 2X8
Tel: (306) 652-5798 ext. 112
Fax: (306) 652-5799
Toll Free: 1-877-655-5798
http://www.cronustech.com

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Re: [Asterisk-Users] SMP Performance

[Steve Szmidt]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On Tuesday 24 August 2004 04:00 pm, Tim Jackson wrote:
> We're looking at implementing Asterisk in our department in the near
> future, we're looking at anywhere from 15-25 extensions. The machine we
> were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/
> 1GB of ram. I've heard bad things about running Asterisk on SMP
> machines? Would we be running into any performance issues with this
> machine?

I'm testing a dual Xeon running 650MHz 768MB and I estimate that it might 
handle about 12 connections on g.729. No SMP issues I'm aware of.
- -- 
Steve

"They that would give up essential liberty for temporary safety deserve
neither liberty nor safety."
Benjamin Franklin

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Version: GnuPG v1.2.4 (GNU/Linux)

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alRl1n7b91TBC2MHah+1TjI=
=E06r
-END PGP SIGNATURE-
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[Asterisk-Users] Grandstream Budgetone BT-101 and VoipJet

[John Week]

Is anyone using this combination successfully?  I have a dell 500sc
running rh9 and asterisk 1.0rc1.  It is configured with an x100p.  I
have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone
BT-101.  I have signed up with Voipjet (they use iax2).  I also have
an FWD number via iax2.  I can sucessfully call back and forth to all
devices via zap, sip, and fwd.  I can successfully place calls using
voipjet with everything except the grandstream.  When I place a
voipjet call with the grandstream, I can hear the party I'm calling,
but they can't hear me.  I have tried all the different codecs the
grandstream supports without luck.  I am running the 1.0.5.10
firmware.  I've emailed voipjet support about it, but they don't have
one.

thanks,

John
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RE: [Asterisk-Users] SIP "unphones"

[Jay Milk]

Yep, great idea, that's what's next -- and I have two extra extensions
(Sipura)

> -Original Message-
> From: Chris Shaw [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, August 24, 2004 12:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] SIP "unphones"
> 
> 
> Check out my ATA idea though, with a regular cheap analog 
> doorphone and a HTX86 or even Sipura, you can program the ATA 
> to dial an extension as soon as the button on the intercom is 
> pressed and then with some extension logic you can do neat 
> things... You can get a doorphone anywhere even radio shack I 
> think and the HTX86 is like $60-70...
> 
> 
> - Original Message -
> From: "Jay Milk" <[EMAIL PROTECTED]>
> To: "'Asterisk Users Mailing List - Non-Commercial 
> Discussion'" <[EMAIL PROTECTED]>
> Sent: Monday, August 23, 2004 7:16 PM
> Subject: RE: [Asterisk-Users] SIP "unphones"
> 
> 
> > Thank you -- funny thing is, I had the same bookmarked, but it just 
> > seemed too expensive for the application -- for $300, I can stick a 
> > cheap IP phone in a hole in the wall :)  I think it's time to get a 
> > Budgetone.
> >
> > > -Original Message-
> > > From: Chris Shaw [mailto:[EMAIL PROTECTED]
> > > Sent: Monday, August 23, 2004 3:52 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [Asterisk-Users] SIP "unphones"
> > >
> > >
> > > I recently saw something just like this and I had it 
> bookmarked... 
> > > It looks like what you're talking about, but I don't 
> think it uses 
> > > SIP. Rather some proprietary protocol that transmit 
> RTP... I could 
> > > be wrong... Check it out...
> > >
> > > http://www.digitalacoustics.com/lanplay.htm
> > >
> > > I would agree that it really should be SIP, you wouldn't want to 
> > > have to rip it out of the wall when the protocol becomes 
> obsolete or 
> > > when a SIP-Compliant alternative comes out...
> > >
> > > -Chris
> > >
> > > - Original Message -
> > > From: "Jay Milk" <[EMAIL PROTECTED]>
> > > To: <[EMAIL PROTECTED]>
> > > Sent: Monday, August 23, 2004 7:55 AM
> > > Subject: [Asterisk-Users] SIP "unphones"
> > >
> > >
> > > > Does anyone know if there are additional SIP devices out
> > > there which
> > > > aren't phones?  I'm basically looking for a fully-automatic SIP 
> > > > speakerphone.  I'd like to be able to dial a sip-extension
> > > and make an
> > > > announcement (PA) and/or simply listen in to a room 
> > > > (baby-monitor). Yes, I know, some of the more advanced 
> phones can 
> > > > be configured to behave like that, but it seems to a waste of 
> > > > money to have
> > > all those
> > > > fancy displays and keys tucked away behind a speakergrille and 
> > > > drywall.
> > > >
> > > > BTW, I'm not dead-set on SIP, but it seems to be the 
> most logical 
> > > > protocol for this app (NOTIFY msg can carry directions on 
> > > > mike/speaker/two-way, etc)
> > > >
> > > > ___
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> > > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > > ___
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> > > UNSUBSCRIBE or update options visit:
> > >
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> >
> >
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Re: [Asterisk-Users] Remotely change call forward

[Chris Shaw]

- Original Message -
From: "Russell Horn" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 24, 2004 2:55 PM
Subject: [Asterisk-Users] Remotely change call forward


> Is it possible using asterisk to allow someone to dial in and remotely
> change where their call is forwarded to?
>
> For example, I'm working from home so I want my calls to go to 555 1234,
> now I need to go out for a bit so I'd like to phone the office and using
> DTMF tell the asterisk PBX to now forward my calls to my cell phone 555
> 3456
>
> Has anyone implimented anything like this?
>
> R.

I see 2 ways of doing this... 1 would be an AGI script that uses MySQL or
PostgreSQL to store and retrieve forwarded numbers..

The 2nd and to me easier way would be to use AstDB and some extension
coolness to store the forwarding number, and then implement some kind of
vertical service code that you can dial when you call, someting like
*78NXXNXX, then feed the NXXNXX part of the extension to
DbPut(features/fwd-) or something like that...

Check these out for some ideas/hints gimme a holler if you need more help!

http://www.voip-info.org/wiki-Asterisk+cmd+DbGet
http://www.voip-info.org/wiki-Asterisk+cmd+DbPut
http://www.voip-info.org/wiki-Asterisk+cmd+AGI

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RE: [Asterisk-Users] Asterisk to Vonage

[Jay Milk]

Yes, search google for
> asterisk vonage working site:lists.digium.com

> -Original Message-
> From: Paterson, Mark [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, August 24, 2004 11:19 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Asterisk to Vonage
> 
> 
> I'm trying to connect my Asterisk server via sip using my 
> vonage soft phone account. Has any anyone successfully got to 
> work? I get error from
> asterisk saying:  == Parsing '/etc/asterisk/sip.conf':   == Parsing
> '/etc/asterisk/sip.conf': Found
> Aug 24 11:01:11 WARNING[1125329600]: acl.c:146 ast_get_ip: 
> Unable to lookup '216.115.25.199:5061' when trying to 
> register with the vonage sip proxy. Any examples would be 
> greatly appreciated.
> 
> 
> Rgs,
> mark
>  
> 
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[Asterisk-Users] [Asterisk Users] Help with SIP Hosted Billing Service

[E Samuels]

Hello All,

 

I am trying to connect to a third party’s SIP
based Hosted Billing service that supports G729 as I have found it difficult to
get hold of prepaid billing solution for Asterisk.  I was able to get an
IAX2 connection working fine with another provider but as I am unable to find a
billing service that supports IAX/IAX2, I need to connect to a hosted billing
service using SIP and G729.

 

I have purchased and installed the G729 licenses
from Digium as required.

My endpoints are Xten Pro and Snom.

 

I am not sure what the problem is but this is my
dial plan in extensions.conf

 

[default]

include => incoming

include => from-iax

include => from-sip

include => iaxtel-outbound

include => sip-pstn

include => iax2-pstn

 

[sip-pstn] **I am not certain if this is correct.

 

exten => _81NXXNXX,1,SetCallerID(123456)

exten => _81NXXNXX,2,SetCIDName("Myself")

exten =>
_81NXXNXX,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)

exten => _81NXXNXX,4,Congestion

 

[iax2-pstn]

exten =>
_0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})

exten =>
_00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})

 

If anyone has any suggestions I would be happy to
listen.

 

Thanks in Advance.

 

Errol

 

 






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Re: [Asterisk-Users] MusicOnHold problem

[Robert Rozman]

Hi,

I've checked again and I can run it from command line. mpg123 is in
/usr/bin.

Is there any way to debug this problem more deeply ?

What files for musiconhold feature should I reinstall and where in case
something got corrupted ?

Thanks in advance,

Robert.

- Original Message - 
From: "Chris Stenton" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Monday, August 23, 2004 12:30 PM
Subject: Re: [Asterisk-Users] MusicOnHold problem


> It can't find mpg123 or you  don't have any mp3 files in your moh
directory.
>
> - Original Message - 
> From: "Robert Rozman" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Monday, August 23, 2004 12:58 AM
> Subject: [Asterisk-Users] MusicOnHold problem
>
>
> > Hi,
> >
> > I had music on hold working but now don't know what happened.
> >
> > I get :
> > WARNING[...] res_musiconhold.c:334 moh0_exec: Unable to start music on
> hold
> > (class '') on channel SIP...
> >
> > Any ideas what is wrong ?
> >
> > Regards,
> >
> > Robert.
> >
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[Asterisk-Users] RE: Parking and Extensions

[Matthew Boehm]

Oh poo.

I forgot to include => parkedcalls in my extensions.conf

Oh well. FYI for someone else.

Matthew
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Re: [Asterisk-Users] Broken Pipe

[Scott Petersen]

On Tue, Aug 24, 2004 at 12:55:50PM -0600, Mike Roberts wrote:
> When I woke up this morning * was giving me the message "broken pipe"
> when I try to log into the CLI... Thats all I know.

That has happened to me several times now over the last few weeks on one of my 
installs. Nothing in the logs that I can find. I have to kill -9 the asterisk process 
and then restart it to get things back to normal. I will be trying a newer version of 
asterisk in the next couple of days that has been stable on the test box up until now. 
However, the previous version was also stable on test. 

The two differences between the boxes are mainboard (both gigabyte but different 
versions) and the production box has a TDM400p with 3 FXO modules. 

Cheers
Scott Petersen

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Re: [Asterisk-Users] Inband DTMF is not supported on codec G.711u-law. Use RFC2833

[Chris Shaw]

Is this message coming from * or from Kphone? That's the stupidest error
I've ever seen, of course In-Band DTMF works over U-Law and A-Law... in fact
it ONLY works on U-Law and A-Law and not reliably on any other...

If this error is coming from *, and I can't believe it is, then you have a
codec problem... If the error is coming from Kphone you will need to
complain to them that their DTMF generation/handling code is FUBAR...

I use In-Band DTMF on my GrandStreams all the time and it works perfectly...
Even in Comedian Mail...

-Chris

- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial
Discussion" <[EMAIL PROTECTED]>
Sent: Tuesday, August 24, 2004 2:30 PM
Subject: Re: [Asterisk-Users] Inband DTMF is not supported on codec
G.711u-law. Use RFC2833


> On Tue, 2004-08-24 at 14:00, Eric Wieling wrote:
> > On Tue, 2004-08-24 at 12:03, Steven Critchfield wrote:
> >
> > > Inband DTMF is not supported, Use RFC2833
> > >
> > > Go search your kphone configs and fic it to use some out of band
> > > signalling of DTMF.
> >
> > kphone does not support RFC2833 DTMF, only inband DTMF.
>
> Then don't use kphone, it will only cause you more grief down the road.
> --
> Steven Critchfield <[EMAIL PROTECTED]>
>
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[Asterisk-Users] Parking and Extensions

[Matthew Boehm]

Lets say that I have 9 phones with extensions 1 thru 9. (All SIP)

Parking extension is set to 700.

When I try to park a call to 700 as soon as I press the 7 key (to start
typing 700) the person at extension 7's phone rings.

So instead of transferring to extension 700, it went to 7.

How can I slow down the response time of * to let me get to 700?

Thanks,
Matthew

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Re: [Asterisk-Users] Asterisk with Adit 600

[Michael Welter]

Craig Neumanns wrote:
Hello,
 
I am connecting Asterisk to an Adit 600 via a T100P. Unfortunately I am 
not able to get any lights on the T100P and the Adit 600 only shows red. 
I have already modprobed the t100p.
 
Has anyone successfully connected these before?
 
I've tried both a cross-over cable and a standard cat 5 cable, but I 
can't seem to get them to work.
 
Any suggestions?
 
Thanks!
Craig

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Take a Cat3 cable:
pins 1-2 on one jack go to 4-5 on the other jack, and visa versa.
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado
+1 303 674 2575
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www.introspect.com
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Re: [Asterisk-Users] Asterisk with Adit 600

[Chris Shaw]

Are you talking about an ETHERNET crossover cable? 
You can't use an ethernet crossover, the pinouts are different! 
 
Look here for a wiring diagram:
 
http://www.gcom.com/home/support/t1crossover.html
 
-Chris
 

  - Original Message - 
  From: 
  Craig 
  Neumanns 
  To: [EMAIL PROTECTED] 
  
  Sent: Tuesday, August 24, 2004 2:31 
  PM
  Subject: [Asterisk-Users] Asterisk with 
  Adit 600
  
  Hello,
   
  I am connecting Asterisk to an Adit 600 via a T100P. Unfortunately I am 
  not able to get any lights on the T100P and the Adit 600 only shows red. 
  I have already modprobed the t100p.
   
  Has anyone successfully connected these before? 
   
  I've tried both a cross-over cable and a standard cat 5 cable, but I 
  can't seem to get them to work. 
   
  Any suggestions?
   
  Thanks!
  Craig
  
  
  Do you Yahoo!?New 
  and Improved Yahoo! Mail - 100MB free storage!
  
  

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[Asterisk-Users] RC2 and Netmeeting 3.01 ?

[Robert Rozman]

Hi,

I'd kindly ask for any guidance how to setup Netmeeting to work with
Asterisk.

I've setup Asterisk as Gateway, selected GSM codec, and I'm able to call
local extensions (no calls into PBX functions) but get no sound.

Any hint, advice ?

Anyone using Netmeeting (maybe also windows messenger) with Asterisk
sucessfully ?

Thanks in advance,

regards,

Robert.

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[Asterisk-Users] Remotely change call forward

[Russell Horn]

Is it possible using asterisk to allow someone to dial in and remotely
change where their call is forwarded to?

For example, I'm working from home so I want my calls to go to 555 1234,
now I need to go out for a bit so I'd like to phone the office and using
DTMF tell the asterisk PBX to now forward my calls to my cell phone 555
3456

Has anyone implimented anything like this?

R.
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Re: [Asterisk-Users] Voicemail & "Couldn't read username" error

[Umar Sear]

The error you are getting suggest that Asterisk can't read the DTMF
being entered. 

Umar

On Tue, 2004-08-24 at 18:49, Bill wrote:
> Hi,
> 
> I have Asterisk running with the VoiceMail. Using the latest CVS. I have
> my extensions.conf setup so that if a SIP caller dials *99 the
> VoicemailMain() as follows:
> 
> exten => *99,1,Wait(1)
> exten => *99,2,VoicemailMain()
> 
> A couple days ago I installed the MySQL/Voicemail support described at
> http://www.voip-info.org/wiki-Asterisk+voicemail+database   Now for some
> reason when I call *99 from a SIP extension I am prompted for a Mailbox
> number but then nothing happens. It used to prompt for a password. I've
> un-installed the MySQL/Voicemail support and it still doesn't work. From the
> Asterisk console I get the following message when I hangup when I am NOT
> prompted for a password.
> 
> Aug 24 12:34:36 WARNING[319509]: app_voicemail.c:3568 vm_execmain: Couldn't
> read username
> 
> Any ideas? I can see on several lists that this is a common problem but
> I haven't found the answer yet.
> 
>   Bill Dunn
> 
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Re: [Asterisk-Users] Re: newb question regarding DTMF

[Umar Sear]

Check what DTMF mode you are using in sip.conf and confirm that it
matches what you have configured on x-lite. 

Search the list archive, this issue has been thrashed out several times
and is probably documented on voip-info.org.

Umar.

On Tue, 2004-08-24 at 17:03, Erik Anderson wrote:
> On Mon, 23 Aug 2004 18:20:58 -0500, Erik Anderson <[EMAIL PROTECTED]> wrote:
> > Hello all - I'm just starting to play around w/ asterisk, and I've run
> > into a seemingly simple problem that has really manged to frustrate
> > me...
> > 
> > I'm running the latest cvs version of *, and am trying to dial in to
> > the default extention 1000 demo using x-lite.  I can dial and hear the
> > greeting no problem, but when I try and send any DTMF tones, I don't
> > get any response.  Is there something specific I need to set in my
> > sip.conf to allow DTMF?
> 
> Bump
> 
> Any advice?
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RE: [Asterisk-Users] call queue help

[defiance]

On Tue, 2004-08-24 at 14:55, Andrew Thompson wrote:
> How about posting us some .conf files?
> 
> -
> Andrew Thompson
> http://aktzero.com/


Here ya go...


[extensions.conf]

[general]

static=yes   ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.

[bogon-calls]

exten => _.,1,Congestion 

[incoming]



exten => s,1,Wait,2 ; Allow for PRI to grab info in facility
exten => s,2,BackGround(new-greeting)
exten => s,3,BackGround(new-mainmenu) 
exten => s,4,Wait,5
exten => s,5,BackGround(new-mainmenu)
exten => s,6,Wait,5
exten => s,7,Hangup


exten => 0,1,Goto(s,2)

exten => 1,1,SetCallerID(Toll Free No Cpub)
exten => 1,2,AGI(openclose.agi)
exten => 1,3,GotoIf($[${STATUS} = closed]?6:4)
exten => 1,4,GotoIf($[${STATUS} = holiday]?8:10)
exten => 1,5,Goto(1,10)
exten => 1,6,BackGround(nighttime-greeting)
exten => 1,7,Hangup
exten => 1,8,BackGround(holiday-greeting)
exten => 1,9,Hangup
exten => 1,10,Goto(tech-context,s,1)

exten => 2,1,Wait,1
exten => 2,2,SetCallerID(TimeIPS)
exten => 2,3,AGI(openclose.agi)
exten => 2,4,GotoIf($[${STATUS} = closed]?7:5)
exten => 2,5,GotoIf($[${STATUS} = holiday]?9:11)
exten => 2,6,Goto(2,11)
exten => 2,7,BackGround(nighttime-greeting)
exten => 2,8,Hangup
exten => 2,9,BackGround(holiday-greeting)
exten => 2,10,Hangup
exten => 2,11,Goto(time-ips-sales,s,1)

exten => 5,1,Wait,1
exten => 5,2,SetCallerID(Sales)
exten => 5,3,AGI(openclose.agi)
exten => 5,4,GotoIf($[${STATUS} = closed]?7:5)
exten => 5,5,GotoIf($[${STATUS} = holiday]?9:11)
exten => 5,6,Goto(5,11)
exten => 5,7,BackGround(nighttime-greeting)
exten => 5,8,Hangup
exten => 5,9,BackGround(holiday-greeting)
exten => 5,10,Hangup
exten => 5,11,Goto(sales-context,s,1)

exten => 7,1,Wait,1
exten => 7,2,Goto(2,1)

include => from-sip
include => tech-context
include => time-ips-sales
include => sales context

[sales]

exten => s,1,Wait,1
exten => s,2,SetCallerID(Sales)
exten => s,3,AGI(openclose.agi)
exten => s,4,GotoIf($[${STATUS} = closed]?7:5)
exten => s,5,GotoIf($[${STATUS} = holiday]?9:11)
exten => s,6,Goto(1,11)
exten => s,7,BackGround(nighttime-greeting)
exten => s,8,Goto(1,28)
exten => s,9,BackGround(holiday-greeting)
exten => s,10,Goto(1,28)
exten => s,11,Dial(SIP/3085,15|m)
exten => s,12,Dial(SIP/3082,15|m)
exten => s,13,Dial(SIP/3006,15|m)
exten => s,14,Dial(IAX2/3007,20|m)
exten => s,15,Background(sales-hold)
exten => s,16,WaitMusicOnHold(60)
exten => s,17,Dial(SIP/3085,15|m)
exten => s,18,Dial(SIP/3082,15|m)
exten => s,19,Dial(SIP/3006,15|m)
exten => s,20,Dial(IAX2/3007,20|m)
exten => s,21,Background(sales-hold)
exten => s,22,WaitMusicOnHold(60)
exten => s,23,Dial(SIP/3085,15|m)
exten => s,24,Dial(SIP/3082,15|m)
exten => s,25,Dial(SIP/3006,15|m)
exten => s,26,Dial(IAX2/3007,20|m)
exten => s,27,Voicemail(u3082)
exten => s,28,Hangup
include => from-sip

[tech]

exten => s,1,Wait,2 ; Allow for PRI to grab info in facility
exten => s,2,SetCallerID(Toll CPUB/Modem)
exten => s,3,AGI(openclose.agi)
exten => s,4,GotoIf($[${STATUS} = closed]?7:5)
exten => s,5,GotoIf($[${STATUS} = holiday]?9:11)
exten => s,6,Goto(s,10)
exten => s,7,BackGround(nighttime-greeting)
exten => s,8,Goto(s,14)
exten => s,9,BackGround(holiday-greeting)
exten => s,10,Goto(s,14)
exten => s,11,BackGround(greeting)
exten => s,12,Playback(tech-intro)
exten => s,13,Queue(tech)
exten => s,14,Hangup
exten => h,1,Hangup
exten => t,1,Hangup
include => from-sip

[from-sip]

;elizabeth chastain
exten => 3001,1,Dial(SIP/3001,20)
exten => 3001,2,Voicemail(u3001)
exten => 3001,3,Hangup

;accouting office
exten => 3003,1,Dial(SIP/3003,20)
exten => 3003,2,Voicemail(u3003)
exten => 3003,3,Hangup

;robert barger line 1
exten => 3005,1,Dial(SIP/3005,20)
exten => 3005,2,Voicemail(u3006)
exten => 3005,3,Hangup

;robert barger line 2
exten => 3006,1,Dial(SIP/3006,20)
exten => 3006,2,Dial(SIP/3005,20)
exten => 3006,3,Voicemail(u3006)
exten => 3006,4,Hangup

;lance chastain
exten => 3007,1,Dial(IAX2/3007,20)
exten => 3007,2,Voicemail(u3007)
exten => 3007,3,Hangup

;darcy latourette
exten => 3008,1,Dial(IAX2/3008,20)
exten => 3008,2,Voicemail(u3008)
exten => 3008,3,Hangup

;chris braun
exten => 3009,1,Dial(SIP/3009,20)
exten => 3009,2,Voicemail(u3009)
exten => 3009,3,Hangup

;julie gile 
exten => 3012,1,Dial(SIP/3012,20)
exten => 3012,2,Voicemail(u3012)
exten => 3012,3,Hangup

;ronda baker
exten => 3014,1,Dial(SIP/3014,20)
exten => 3014,2,Voicemail(u3014)
exten => 3014,3,Hangup

;cary morrison
exten => 3015,1,Dial(SIP/3015,20)
exten => 3015,2,Voicemail(u3015)
exten => 3015,3,Hangup

;dave ginskey
exten => 3030,1,Dial(IAX2/3030,20)
exten => 3030,2,Voicemail(u3030)
exten => 3030,3,Hangup

;darin raaf
exten => 3033,1,Dial(IAX2/3033,20)
exten => 3033,2,Dial(Zap/16/3048733,20)
exten => 3033,3,Voicemail(u3033)
exten => 3033,4,Hangup

;mike georges
exten => 3034,1,Dial(IAX2/3034,20)
exten => 3034,2,Dial(Zap/15/2046084,20)
exten => 3034,3,Voicemail(u3034)
exten => 3034,4,Hangup

;sharon kuhn
exten => 3035,1,Di

[Asterisk-Users] Asterisk with Adit 600

[Craig Neumanns]

Hello,
 
I am connecting Asterisk to an Adit 600 via a T100P. Unfortunately I am not able to get any lights on the T100P and the Adit 600 only shows red. I have already modprobed the t100p.
 
Has anyone successfully connected these before? 
 
I've tried both a cross-over cable and a standard cat 5 cable, but I can't seem to get them to work. 
 
Any suggestions?
 
Thanks!
Craig
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Re: Re: [Asterisk-Users] Inband DTMF is not supported on codec G.711 u-law. Use RFC2833

[Linterra]

If your phone does support RFC2833, edit your sip.conf and add
something like this:

[sip1]
type=friend
host=dynamic
dtmfmode=rfc2833
context=sip

On Tue, 24 Aug 2004 14:00:01 -0500, Eric Wieling <[EMAIL PROTECTED]> wrote:
> On Tue, 2004-08-24 at 12:03, Steven Critchfield wrote:
> 
> > Inband DTMF is not supported, Use RFC2833
> >
> > Go search your kphone configs and fic it to use some out of band
> > signalling of DTMF.
> 
> kphone does not support RFC2833 DTMF, only inband DTMF.
> 
> --
>   Eric Wieling * BTEL Consulting * 504-899-1387 x2111
> "In a related story, the IRS has recently ruled that the cost of Windows
> upgrades can NOT be deducted as a gambling loss."
> 
> 
> 
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Re: [Asterisk-Users] MailMan slowness...

[mastapasta]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Chris Shaw schrieb:
| Ok what the flaming hell is up with the MailMan? It's taking over a day to
| send posts now, before it was at most a couple hours...
|
| Is digium doing maintenance on that server or something??
|
It may be more like the censorship is gaining more control again.
Sincerly mastapasta
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.1 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iD8DBQFBK7A+Mj12xhezBnERAssJAJ9bAGFx6AskOgWxjESE/7NSCHSLzgCgqszu
dWbWMlUvhUNW6NSG8nKrECA=
=edCg
-END PGP SIGNATURE-
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Re: [Asterisk-Users] Inband DTMF is not supported on codec G.711 u-law. Use RFC2833

[Steven Critchfield]

On Tue, 2004-08-24 at 14:00, Eric Wieling wrote:
> On Tue, 2004-08-24 at 12:03, Steven Critchfield wrote:
> 
> > Inband DTMF is not supported, Use RFC2833
> > 
> > Go search your kphone configs and fic it to use some out of band
> > signalling of DTMF.
> 
> kphone does not support RFC2833 DTMF, only inband DTMF.

Then don't use kphone, it will only cause you more grief down the road.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] ex-girlfriend logic not working in latest CVS?

[James Sizemore]

I found _my_ problem, to many "X"s on the patten match.
James Sizemore wrote:
Ex-girlfriend logic not working in latest CVS?
Incoming sip calls don't work. Anyone else seen this
problem?
Extension logic looks good:
exten => 6153248305/_931NXXX,1,Queue(queue1);
exten => 6153248305/_615NXXX,1,Queue(queue2);
;exten => 6153248305,1,Queue(queue3);
show dialplan looks good:
-- Added extension '6153248305' priority 1 (CID match 
'_931NXXX')to vantage
-- Added extension '6153248305' priority 1 (CID match 
'_615NXXX')to vantage

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[Asterisk-Users] Using Lucent/Avaya 64XX sets with asterisk

[Peter Miller]

Hi all,
I am new to asterisk, but experienced with Linux and Avaya/Lucent G3R 
hardware/software.  We currently have a large inventory of spare Lucent 
64XX phone sets.  I'm considering setting up an asterisk system for a 
remote office, and would like to use the Lucent sets because they are 
cheap, and familiar to my end users.  Does anyone know of a card that 
can drive these sets?  I think the underlying protocol is ISDN.  Thanks 
in advance.

--- Peter Miller
Information Technology Department
WGBH Educational Foundation
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Re: [Asterisk-Users] Euro isdn caller id problem - SOLVED

[Asterisk]

I've got BT to enable callerid on the line, and all is now working fine.

Thanks very much.

BTW - I've almost got to the point where I've managed to fully integrate a
Meridian 1 option 11c with asterisk, and I'll be writing up a whitepaper on
what we had to do, with the help that we've had from everyone.

Julian.

- Original Message - 
From: "Peter Svensson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 13, 2004 10:23 PM
Subject: Re: [Asterisk-Users] Euro isdn caller id problem


> On Fri, 13 Aug 2004, Asterisk wrote:
>
> > Ok, I've attached the setup frame below: looks like "unknown number
type" is
> > causing the problem ... yes ?
> > the help like this.
>
> There was no "calling number" element that I could see. Your pstn provider
> probably has not enabled it on your line.
>
> "unknown number type" is isdn-speak for the normal national dialing
> format.
>
> Peter
>
>
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RE: [Asterisk-Users] Dell PowerEdge 750 rackmount

[Tim Jackson]

What I have found are the best servers, are IBM xSeries machines. You
can obtain a Refurb xSeries 305 P4 2.4ghz for around $750-800. I have
never had the first problem out of these boxes. SuperMicro is also good,
but the IBMs tend to be a little smaller (a lot less deep) than
SuperMicro 1U machines.

-Tim

-Original Message-
From: Steve Szmidt [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, August 24, 2004 1:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dell PowerEdge 750 rackmount

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On Tuesday 24 August 2004 01:02 pm, Steven Critchfield wrote:
> The big thing to look into is what PCI busses the machine supports. We
> where very surprised with our Dell when it came with a PCIX slot and a
> 66mhz 64bit slot. The included ethernet card wasn't directly supported
> by our install disks and we couldn't install a cheap card to get the
> install bootstrapped due to the incompatible slots. So after building
a
> special boot disk, all was better except the T100P card I owned
wouldn't
> work in it again due to slot incompatibility. That prompted our TE410P
> card purchase.

After using various Dell's for a few years I'm very weary if I need
something 
not bland. They are very very good at cutting corners at all sorts of
places.
Their intended public is mostly "run of the mill" machines. 

They are the pro's in cutting pennies and turn it into BIG savings.

Too many times have I discovered something odd or less than I would have

expected. 

SuperMicro on the other hand have striken a good balance of price and
quality. 
They are really are going after the market with some very solid hardware
and 
not being the cheapest either. Which is really less important in a
business 
environment anyway. Support and quality being more important. It's
really 
true that you get what you pay for. If you can afford quality it's
cheaper in 
the long run.
- -- 
Steve

"They that would give up essential liberty for temporary safety deserve
neither liberty nor safety."
Benjamin Franklin

-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)

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Re: [Asterisk-Users] VoicePluse DID problem

[Jeremy Bogan]

Cal someone help me. I'm register voiceplus DID i try to config
fllow example but not work. When i test call to number and debug
iax2 in my asterisk not found packet.
You need to specify your DID in extensions.conf:
[voicepulse-incoming]
exten => 123456789,1,Playback(demo-congrats)
exten => h,1,Hangup
exten => i,1,Hangup
exten => t,1,Hangup
And yes you should be answering the call before playing any sound.
--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
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[Asterisk-Users] Re: [Asterisk-Dev] Asterisks

[Greg Hill]

On Tue, 24 Aug 2004, Huddleston, Robert wrote:

> Take mercy on me - I'm a newbie w/ Asterisks... Here's what I'm trying to do
> - and please someone let me know if this can be done...
> We have a commercial VoIP network... The gatekeeper supports MGCP/H.323 and
> allows for calls to be made to the PSTN cloud.
> I would like to build Asterisks with H323 (or MGCP if need be) and have it
> attach to our gatekeeper to access the PSTN.
> Instead of installing a T1/E1 or ISDN or POTS card we would like to use the
> existing VoIP network.
> Anyone ran into this before - can provide some direction?

so you want your Asterisk machine to connect to your gatekeeper "via
existing VOIP" .. you mean over an ethernet-type interface, right? Sure
you can do that.

A better place to ask for this type of help is the asterisk-users list,
though. The asterisk-dev list is for topics related to development of
asterisk itself, not development and implementation of solutions involving
asterisk.

Greg


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RE: [Asterisk-Users] Error compiling meetme2

[Geoff Nordli]

[EMAIL PROTECTED] wrote:
> Hi Geoff,
> 
> Geoff Nordli wrote:
> 
>> 
>> 
>> I was able to compile the module and it loads correctly, but I am
>> still having problems with the app. 
>> 
>> I see all the users in the conference, but I can't kick them out, or
>> change their mode from talk to listen-and-talk.  No errors are
>> showing up anywhere. 
>> 
>> I am not really sure how to troubleshoot this, any ideas?
>> 
>> Thanks,
> 
> Enable register_globals in php.
> 
> You can also put an "extract($_GET);" in the top of the php file.



Thanks Nicolas.

Enabling register_globals did the trick.

Geoff

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Re: [Asterisk-Users] strange problem PBX-Asterisk

[Peter Svensson]

On Tue, 24 Aug 2004, Christian Victor wrote:

> > maybe I oversee somth. very obvious, but I'm a little puzzled about the 
> > following 'error':
> > 
> > When I make a call from the PBX to * I get number not available, but on 
> > debug I see, that asterisk is searching just for the first digit in the 
> > extension, which of course doesn't exist, eg:
> 
> I seems that you PBX uses "Overlap Dial" and transmits the extensien 
> digit by digit and Asterisk expects the extension to be "en bloc". So 
> when it receives anything from the PBX (wich is in this case the first 
> digit) Asterisk thinks that this is the whole block of extension.
> 
> Don't know how to fix it though. ;-)

This in case you are on a PRI: see "overlapdial" at
  http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf
It needs to be set on pri links where ovarlap dialing is used, even 
incoming towards asterisk.

Without more information on the connections between the systems and the 
configuration it is hard to figure out what is wrong.

Peter


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RE: [Asterisk-Users] call queue help

[Andrew Thompson]

defiance wrote:
> Guys I am having some serious issues with my call queue and
> Management is breathing down my neck pretty bad, and I am running out
> of ideas.  
> 
> I have a single queue for my tech support department. I originally
> was using the AgentCallbackLogin for them and it tested out great on
> our testing weekends, but it hasn't worked out since. It would only
> let one of them take calls at a time, no matter what I set the ring
> strategy to.
> 
> So I set an extension up to do a dynamic login for them, this worked
> out for about a day or so and it has started doing the same thing.
> Plus it drops calls sometimes too. I don't have a dropped call
> problem anywhere else in the company so I am assuming it is the
> queue. Can't find anything strange (to me) in the logs, but they
> aren't the easiest on the eyes. 
> 
> So basically I need my queue to let all 4 of my tech guys login and
> take calls, and not drop the calls they do get. 

How about posting us some .conf files?

-
Andrew Thompson
http://aktzero.com/

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[Asterisk-Users] off list

[Chris Clifton]

Hi,

Been off list a while, I checked out the changelog, has there been any major
stability or feature changes since CVS-HEAD-05/19/04-09:07:40 ?

Thanks,
Chris

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[Asterisk-Users] Problem with Asterisk VM + MySQL Database

[TSS Support]

We are trying to use the MySQL database for voicemail logins. I can see that
* connects correctly to the database and is executing queries, so I'm pretty
sure I got the config right. However, it doesn't seem to be picking the
password up properly. Found a couple of old mailing list messages about this
but no solution. Anyone seen this before and can help me figure out what I'm
missing? Here's the info:

Relevent debug statements from *

vm:/etc/mysql# asterisk -vgrc
  == Parsing '/etc/asterisk/asterisk.conf': Found
Asterisk CVS-05/31/04-22:00:51, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
=
Connected to Asterisk CVS-05/31/04-22:00:51 currently running on vm (pid =
30519)
-- Remote UNIX connection
-- Executing Wait("SIP/voicemail-6e98", "1") in new stack
-- Executing VoiceMailMain("SIP/voicemail-6e98", "") in new stack
-- Playing 'vm-login' (language 'en')
-- Playing 'vm-password' (language 'en')
-- Incorrect password '1234' for user '3609152000' (context = )
-- Playing 'vm-incorrect' (language 'en')
vm*CLI>

A snippet from the startup debug messages for *:

vm:~# grep database asterisk.txt
-- Logging into database with user asterisk, password (redacted), and
database asterisk

And from the MySQL log:

040823 13:36:10   3 Connect [EMAIL PROTECTED] on asterisk
040823 13:36:43   3 Query   SELECT
password,fullname,email,pager,options FROM users WHERE context='default' AND
mailbox='3609152000'

And finally, the table in MySQL:

mysql> select * from users;
+-++--+--+-+---+
-++
| context | mailbox| password | fullname | email   | pager |
options | stamp  |
+-++--+--+-+---+
-++
| default | 3609152000 | 1234 |  | [EMAIL PROTECTED] |   |
| 20040713164014 |
| default | 3609152050 | 1234 |  | [EMAIL PROTECTED] |   |
| 20040713164025 |
+-++--+--+-+---+
-++
2 rows in set (0.00 sec)



So what did I miss? It looks like the right data is there, it's selecting
it, and successfully getting it back, but not letting us log in.

-Keith, TSS

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Re: [Asterisk-Users] Dell PowerEdge 750 rackmount

[Gary Carr]

Did you check the dell outlet (refurbed with same warranty) or ebay? You can
typically purchase and ship same day from either of those.



Gary


> I noticed the PowerEdge 750 seems to have one of each: 32- and 64-bit
PCI's,
> both brought to the rear panel - nice.
>
> BUT, I can't get the Dell's fast enough for this customer, so now I'm
> looking at the HP Proliant DL-320.
>
> Regards
> Scott Stingel
>
>
> Scott M. Stingel
> President,
> Emerging Voice Technology, Inc.
> Palo Alto California & London England
> www.evtmedia.com
>
>
>
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[Asterisk-Users] CPU utilization

[Michael Welter]

Does anyone know why my CPU utilization spikes every 20 seconds?  top 
says it is asterisk.  The CPU monitor looks like an EKG.  Is this normal?

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[Asterisk-Users] SMP Performance

[Tim Jackson]

We’re looking at implementing Asterisk in our
department in the near future, we’re looking at anywhere from 15-25
extensions. The machine we were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 1GB of ram. I’ve heard bad
things about running Asterisk on SMP machines? Would we be running into any
performance issues with this machine?

 



Tim Jackson





Network Engineer





Angelina County, Texas





(936)639-4827 office





(936)414-6723 mobile



 






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Re: [Asterisk-Users] Re: Voicemail & "Couldn't read username" error

[Eric Wieling]

On Tue, 2004-08-24 at 14:08, Bill wrote:
> I got this problem fixed by putting the following in the sip.conf file.
> 
> dtmfmode=inband
> 

That only works if your call is using the ulaw or alaw codecs.

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

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Re: [Asterisk-Users] Asterisk WITH Swyx... Any Idea?

[Zineddin Karzazi]

 > > Hi,
 > > 
 > > I'm a student and my thesis work consist in
> testing  
 > > Asterisk with Swyx(SwyxWare).
> > > 
 > > My approach is to declare asterisk as h323
 gateway
 > for
 > > the Swyxserver using oh323 Plugin. 
 > > Is there any possibility to connect Asterisk
 with
 > > Swyx? how?
 > > the outgoing call must pass from Swyxit->to
 > > Swyxserver-> Asterisk->to PTSN
 > 
 > Please look up and READ ALL of this URL...
 > http://www.catb.org/~esr/faqs/smart-questions.html
 > 
 > pay ___EXTRA___ attention to
 >
>
http://www.catb.org/~esr/faqs/smart-questions.tml#homework
 > 
 > 
 > Also, note that page is translated into many
 > languages. English not
 > being a primary language is not an excuse for not
 > reading that page.
 >  
 > -- 
 > Steven Critchfield <[EMAIL PROTECTED]>
 > 
 > 
 
 
 "...It is OK to ask for hints, but not for entire 
 solutions." 
 
 I Just Asked for Hints not the entire Solution.
 
 
 Thanks. 







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RE: [Asterisk-Users] how to collect user entered digits

[John Millican]

On Fri, Aug 20, 2004 at 01:19:54PM -0400, John Millican said:
> I have been searching thru all docs that I can find on wiki and such but
can
> not get an answer.  I am trying to collect a date from user input in the
> form of digits dialed from the phone to use in an agi script to do a
> database look up.  I have tried to use "Get Data filename, timeout,
> maxdigits " in the agi script. In * console I get message saying playing
> filename but it exits as soon as it starts. Could I collect the digits
some
> how before going to script and then send as an arg maybe?

Check out the source to the privacy manager for a simple example...

apps/app_privacy.c


Well Mr. Reed, and all others on list who might help, I guess I am not as
smart as I hoped I was:-)

I looked at the apps_privacy file.  I thought I could write something
similar and compile then call from dial plan.  What I get is a lot of errors
about unreferenced ast_app_streamfile, ast_app_getdata, ... among others. I
used all the same includes in Marks script but still no go. I tried to find
where these references came from in the apps_privacy and could not find
them. I typically code in  C++ with QT 3.3 and would like to be able to use
the DB drivers that I have in QT for PostgreSQL since I will be querying an
app on a separate machine for customer info based on callid and a date in
the form of digits dialed by the caller. Any help in this direction would be
greatly appreciated.  My original plan was an AGI script, is this a bad
idea?  Call levels will most likly not be very high.

---
Outgoing mail is certified Virus Free.
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[Asterisk-Users] Monitor() hangs

[Manuel Wenger]

We use the Monitor() command to record all incoming calls to our call center. After 
about 100 incoming calls, the Monitor() command starts to hang, as follows:
- a call comes in
- Asterisk starts recording, the -in.wav and -out.wav files are created
- the partys talk
- after a while (between 1 and 15 minutes), both partys hear only silence (as in "no 
RTP data flows").
- when this happens, Asterisk stops recording (the wav files don't grow anymore, they 
just sit there open).
- when this happens, all other SIP calls active at that time suffer the same problem 
(ie. the callers don't hear each other anymore)
- the Monitor()ed call's channels remain there, as if the call were still active, even 
after the caller hangs up (Asterisk ignores the hangup)
- if I "soft hangup" the channels, Asterisk finally destroys the channels, writes the 
CDR record, closes the wav files and soxmix'es them.
- the other, non Monitor()ed SIP calls that suffered the "silence" problem are hung up 
correctly if one of the 2 partys hangs up (ie. the channels don't "hang" there forever 
like the Monitor()ed call)
- if I listen to the recorded wav file, the recording just stops abruptly at the end, 
no click, no anything - not even silence. It just stops

I'm quite clueless. A July CVS-HEAD is exhibiting the same behaviour. We are using SIP 
only, nothing fancy. The PSTN gateway is somewhere else, not on Asterisk.

We are using CVS-HEAD-08/24/04-08:09:55 on Fedora Core 1 (Kernel 
2.4.22-1.2188.nptlsmp) on a Dual Xeon server (HT disabled). We never have more than 
about 20 channels active at a time, so I don't think the system is "too loaded".

Anyone?

Thanks
-Manuel


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[Asterisk-Users] Directed Call Pickup

[Mike Meyer]

Hi everyone,

I have been trying to get call pickup to work and am having success
with group pickup by setting the callgroup and pickupgroup in the
zapata.conf and sip.conf files. However, I cannot get directed call
pickup to work. 

According to the little documentation
(http://www.voip-info.org/tiki-index.php?page=PBX+Call+Pickup) that I
found on the subject, the user is supposed to be able to enter the call
pickup extension and then enter the extension to pick up.

Basically *8 doesn't work. Is there some configuration that I am
missing? Do I need to define a new call pickup extension in
extensions.conf that answers the call on the remote extension?

Any direction will help.

FYI; Another caveat I have also run into is that a simple reload after
changing the sip.conf and zapata.conf files did not always pick up the
new group settings and allow me to do group call pickup. Anyone else run
into this? I had to stop and start *.

Thanks much,
Mike Meyer


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Re: [Asterisk-Users] Bell Canada Caller-ID

[Kanwar Ranbir Sandhu]

On Tue, 2004-08-24 at 10:07, Matt G wrote:
> Has anyone gotten CID from Bell Canada to work properly with *?
> 
> We have our * box down at our datacentre in St Louis, and whenever we 
> call it from a Bell Canada Telephone line, all we see is '' for the CID. 
> I did some digging on google and the mailing lists and couldn't find 
> anything pertaining directly to Bell-Canada and * CID, but didn't find 
> much. I did however find :

We have had CID working from the very beginning (well, not quite...had
to insert a Wait(1) after answering, after which CID worked flawlessly).

We have a TDM400P, so it's not quite the same setup as your's.  However,
CID works here, so I'm not sure it's a Bell problem.

We do have hangup detect problems though.  It used to work, but now it
doesn't - completely clued out as to why.

Anyway, perhaps it's just a matter of having to use a "Wait" after
answering the line.

HTH,

Ranbir

-- 
Ranbir
Systems Aligned Inc.
www.systemsaligned.com

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Re: [Asterisk-Users] Dell PowerEdge 750 rackmount

[Benjamin Johnson]

I've used DL320's in the past with great success - and greatly 
sympathise with the Dell availibility issues! ;-)

The one thing I found with the DL320's is that they didn't support 
booting without a keyboard and our only solution was to KVM them all 
(which we didn't want to do). This was back in 2002 so they may be 
better now. Just my 2c.

Cheers,
Benjamin
Scott Stingel wrote:
BUT, I can't get the Dell's fast enough for this customer, so now I'm
looking at the HP Proliant DL-320.
 


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[Asterisk-Users] Trouble with all linux sip softphones.... And asterisk/linphone/kphone SRPMs

[John Morris]

Hi, I've been messing with getting SIP working for days now, with
limited success.  I've got Asterisk set up on a remote server with the
echo test.  Please try it out to verify I've got the server working
right:

sip:[EMAIL PROTECTED]

Running FC1, ThinkPad T22, headset thru the soundcard.  Asterisk is
asterisk-1.0_RC1.  No NAT.  The phones I've tried so far are as follows.

** Linphone:  Check out my SRPM at http://www.bigu.org/SRPMS/
  Sound is fine.  Doesn't seem to pick up anything from the microphone,
though.

** kphone:  Check out my SRPM at http://www.bigu.org/SRPMS/
  Sound is fine.  Picks up sound from the microphone, but the echo-test
repeats it back after passing it through a Mr-Roboto filter.

** tkPhone:
  Sound is fine.  Doesn't seem to be reading from the mic, no traffic
going over the network after the 'demo-echotest' recording finishes. 
The following errors are continuously repeated from tkPhone:

  sent 63426 (3),received 30228 (3);read 615040 write 293120 need
  608000 jitter 38!! Error:
  !!!
  Data won't fit within the current RTP packet size

** SJphone:
  Last night:  sound worked fine.  Actually sent sound from the mic,
which came back after about a 5-second delay, but which sounded quite
good.
  Today:  establishes a connection, but absolutely no sound in or out. 
:P

If anyone's interested in my SRPMs, I'd like to know.  The asterisk RPM
builds the zapata, zaptel, and asterisk sources; you must have your
kernel-source rpm installed for it to build the modules against.

Let me know if there's something obvious I'm missing.  Thanks-

John


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Re: [Asterisk-Users] Telemarketer screening

[Walt Reed]

On Tue, Aug 24, 2004 at 03:34:34PM +1000, david kwok said:
> I have been bugging by a telemarketer who does not take any cue at all.
> 
> So I look up the Asterisk Handbook and send his call with the respect 
> caller id to my voicemail.
> 
> Has any one implemented any of this feature with database for more 
> caller ids to be included??

Yep. I have a database of "idiots" that just don't get a clue. I also
force 800 numbers or no callerID to hit "5", which has eliminated
virtually all telemarketing calls (it's amazing how many ignore the
DoNotCall list. I got tired of reporting them.) Seems that the
autodialers don't pass on that critical bit of information... :-)

I've been thinking of handling all sorts of callers specially. * makes
this so easy!

; house line
exten => s,1,NoOp(${CALLERID})
; check for callerID. If none,  make them hit 5. 
exten => s,2,GotoIf($["${CALLERIDNUM}" = ""]?s|1000)
; Is the person calling an idiot?
exten => s,3,DBGet(temp=idiot/${CALLERIDNUM})
exten => s,4,Goto(s,2001) ; Yep, he's an idiot.

; Credit Services - a totally clueless company
; that won't take "wrong number" for an answer
exten => s,104,GotoIf($["${CALLERIDNUM:0:3}" = "8006848429"]?s|2100)
; If 800 type number, make them hit 5. 
exten => s,105,GotoIf($["${CALLERIDNUM:0:3}" = "800"]?s|1000)
exten => s,106,GotoIf($["${CALLERIDNUM:0:3}" = "866"]?s|1000)
exten => s,107,GotoIf($["${CALLERIDNUM:0:3}" = "888"]?s|1000)
exten => s,108,GotoIf($["${CALLERIDNUM:0:3}" = "877"]?s|1000)
exten => s,109,Goto(s|200)
; All special cases handled. 
; Replace callerID name for well known numbers to be more descriptive
exten => s,200,LookupCIDName
; If daytime, ring through
;   Note: should do DB lookup to bypass time restrictions for 
;   family / friends...
exten => s,201,GotoIfTime(7:00-21:00|*|*|*?5|1)
; else make them hit 5 if after hours...
exten => s,202,Goto(s|1000)

exten => s,1000,Background(press5formadsorwalt)

; Idiots here:
exten => s,2001,Playtones(info)
exten => s,2002,Wait,1
exten => s,2003,Playback(discon-or-out-of-service)
exten => s,2004,Congestion

; Cred Services - because the clueless don't give up
exten => s,2100,Answer
; Play nasty message endlessly...
exten => s,2101,Wait(1)
exten => s,2102,Playback(cred-services)
exten => s,2103,Goto(2101)

exten => 5,1,Dial(SIP/604&SIP/602,25,tr)
exten => 5,2,Background(reached-reeds)
exten => 5,3,Background(press-1-leave-msg)
exten => 5,102,Background(reached-reeds)
exten => 5,103,Background(press-1-leave-msg)

exten => o,1,Background(invalid)
exten => o,2,Goto(8,1)
exten => i,1,Background(invalid)
exten => i,2,Goto(8,1)
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup

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Re: [Asterisk-Users] Asterisk Install on Kernel 2.6.x

[Marcelo Pacheco]

I do too, TDM400P and X100P work ok.
But TDMoE with 2.6 freezes one of my computers, causes non-stop stack faults 
on another, working only on a third.
And wcusb doesn't work on any at all.

Marcelo Pacheco

Em Ter 24 Ago 2004 13:36, Steve Szmidt escreveu:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> On Monday 23 August 2004 03:14 pm, Shawn Parker wrote:
> > i know asterisk itself will install on a linux kernel 2.6.x, but i've
> > seen places say that the zaptel drivers wont?  is this still true?  is
> > it possible to build asterisk/zaptel on a linux 2.6.x kernel?
>
> A number of us already use 2.6.
>
> Remember the wiki is your friend.
> http://voip-info.org/tiki-index.php?page=Asterisk
> - --
> Steve
>
> "They that would give up essential liberty for temporary safety deserve
> neither liberty nor safety."
> Benjamin Franklin
>
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.2.4 (GNU/Linux)
>
> iD8DBQFBK26oljK16xgETzkRAvctAJ9c3WV0NE5dmI7FhoqKSYprxHIzqACfTJu5
> ApNofNxmHUGasGqAl9Lydg4=
> =YkYZ
> -END PGP SIGNATURE-
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[Asterisk-Users] Broken Pipe

[Mike Roberts]

When I woke up this morning * was giving me the message "broken pipe"
when I try to log into the CLI... Thats all I know.
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Re: [Asterisk-Users] ex-girlfriend logic not working in latest CVS?

[Josh Roberson]

Maybe it's just me, but it looks as if you have one too many X's in  
your pattern matching..

615NXX is all you need, i see 615NXXX.  Same for 931.
-twisted
James Sizemore wrote:
Ex-girlfriend logic not working in latest CVS?
Incoming sip calls don't work. Anyone else seen this
problem?
Extension logic looks good:
exten => 6153248305/_931NXXX,1,Queue(queue1);
exten => 6153248305/_615NXXX,1,Queue(queue2);
;exten => 6153248305,1,Queue(queue3);
show dialplan looks good:
-- Added extension '6153248305' priority 1 (CID match 
'_931NXXX')to vantage
-- Added extension '6153248305' priority 1 (CID match 
'_615NXXX')to vantage

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Re: [Asterisk-Users] Dell PowerEdge 750 rackmount

[Peter Svensson]

On Tue, 24 Aug 2004, Michael Welter wrote:

> Scott Stingel wrote:
> > I have an upcoming order for a bunch of asterisk boxes, and I'm considering
> > using an "assembled" package for the server, instead of building them from
> > components as I usually do.
> > 
> > Does anyone have experience with the Dell PowerEdge 750 server, or any other
> > 1U rackmount server for use with asterisk?
> 
> I would verify that the BIOS allows the reassignment of IRQs.  My Dell 
> 400SC BIOS won't allow IRQ manipulation.

We use a Fujitsu Siemens Primergy RX100 (1u with a very shallow 
footprint). It allows manual irq assignment. Nice and compact and has two 
drive slots, one full height and one half-height pci slot and two 
on-board nic:s.

Peter


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