Re: [Asterisk-Users] Ok, Im confused

[Asterisk]

Not sure specifically about FWD specifically but you can get * sip to
register from behind NAT in general but look at this link and work with the
NAT options in the sip.conf , general section, if you Asterisk box in behind
a NAT device such as a linksys. In some cases in may be neccessay to plug
port on your NAT device. General 5060 and 1 thru 2 to the internal
ip of your Asterisk box. FYI if you external IP changes on your NAT device
you will need account for that in your sip.conf . Also, in this
configuration it will work best with canreinvite=no for all sip devices.

http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions

- Original Message - 
From: "Scott Weis" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, April 22, 2004 1:54 AM
Subject: Re: [Asterisk-Users] Ok, Im confused


> The simple answer probably is, If you have a NAT firewall (like a linksys,
> netgear, dlink, etc) it will not work.
>
> If your linux machine is directly connected follow the instructions on the
> wiki and it will work no problems.  I could not get FWD to work at all
until
> I made my linux box  the outside edge of my network.
>
> Scott
> - Original Message - 
> From: "James H. Thompson" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, April 21, 2004 8:58 PM
> Subject: Re: [Asterisk-Users] Ok, Im confused
>
>
> > You can post your .conf files.
> >
> > But here is a guess at what you may need
> > replace "FWD##"  with your freeworlddialup number and "mypassword" with
> your freeworlddialup
> > password.
> >
> > in sip.conf
> >
> > context = from-fwd
> > register=FWD##:[EMAIL PROTECTED]/FWD##
> >
> > [fwd]
> > type=friend
> > secret=mypassword
> > username=FWD##
> > host=fwd.pulver.com
> >
> > in phone.conf
> > ...
> > context=from-phone
> > ...
> >
> > in extensions.conf
> >
> > [from-fwd]
> > exten => FWD##,1,Dial(Phone/phone0)
> > exten => i,2,Playback(invalid)
> >
> > [from-phone]
> > exten => _.,1,SetCallerID(FWD##)
> > exten => _.,2,SetCIDName(FWD##)
> > exten => _.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
> > exten => _.,4,Playback(invalid)
> > exten => _.,5,Hangup
> >
> >
> >
> >
> >
> > Jim
> >
> > James H. Thompson
> > [EMAIL PROTECTED]
> >
> > - Original Message - 
> > From: "tmpm" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Wednesday, April 21, 2004 1:24 PM
> > Subject: Re: [Asterisk-Users] Ok, Im confused
> >
> >
> > > Thanks Jim,
> > > But that page started my trip off to confusionbeen theretried
it
> 10
> > > different ways...still no joy.
> > > I'll go through it once again, maybe Im missing something, I dont
know.
> Im
> > > about ready to boot the penguin to the curb...
> > > I know its in there...I think Ive got it all configured, and I dial
the
> > > outbound strings, and get a fast busy...I know one stinking letter
off,
> and
> > > its whacked...
> > > HOW for example do I specify my one and only extension is the Internet
> > > phone jack? Phone0?
> > > Somehow theres got to be a tie-in...I cant find it.
> > > been thru extensions.conf, phones.conf, sip.conf..etc.
> > > groan..
> > >
> > > At 18:40 4/21/2004, you wrote:
> > > >Look here:
> > > > http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
> > > >
> > > >Jim
> > > >
> > > >James H. Thompson
> > > >[EMAIL PROTECTED]
> > > >
> > > >- Original Message -
> > > >From: "tmpm" <[EMAIL PROTECTED]>
> > > >To: <[EMAIL PROTECTED]>
> > > >Sent: Wednesday, April 21, 2004 11:50 AM
> > > >Subject: [Asterisk-Users] Ok, Im confused
> > > >
> > > >
> > > > > Im totally a newbee at *
> > > > >
> > > > > Im confused.
> > > > > Ive got a FWD account, and it works on the winboxen. Ive got * up
> and can
> > > > > do the echotest etc, so its working.
> > > > >
> > > > > I want to get FWD working, and all the pages ive seen on setup are
> most
> > > > > confusing.
> > > > > Is FWD setup like IAXTEL? Do i plug in my FWD info in the same
> places as
> > > > > the IAXTEL stuff?
> > > > > Ive been trying for a week now, and Im more lost than before.
> > > > >
> > > > > Ive got a Internet phonejack card in the penguin, phone0, and all
I
> > > > want to
> > > > > do at this point is make and receive calls thru FWD using that
> jackIll
> > > > > plug the house in later...Ill learn the other stuff later. No
> > > > voicemail, no
> > > > > BS, no dial thru least cost routing, or nightlines just make
it
> > > > work as
> > > > > a phone.
> > > > >
> > > > > Im either more stupid than I think, or Im missing something major
> here.
> > > > >
> > > > > Ive got to the point the CLI shows me connected to FWD fine.(I
> think)
> > > > > Sip show users
> > > > >
> > > > > Username Secret Authen Def. Context a/c
> > > > > fwd.pulver.com secret md5,plaintext default no
> > > > >
> > > > > Need some basic, stupidly simple scripts here...I dont need or
want
> to
> > > > dial
> > > > > 1-700 or *9 or any other crap, just make it work like the stupid
> winbox
> > > > > phone for now...Ill r

[Asterisk-Users] Asterisk & RedHat Enterprise

[Asterisk]

Are their any issues with Asterisk and Redhat 
Enterprise? I have see one or two posts with issues concerning compiling zaptel 
drivers but that is about it. Just looking for some consensus to if any problems 
exist with it.

RE: [Asterisk-Users] Asttapi

[Florian Overkamp]

Hi, 

> -Original Message-
> Instruction's can be found at www.omniis.com/asttapi, 
> including where to
> download it from. This is update 0.02, this now includes a little
> feedback from Asterisk so that when click to dial has occurred then it
> is indicated at the start and the end of the call.
> 
> Now working on inbound calls.
> 
> Any question, please send to me.

Cool program!

What I would like is feedback on this:

You are now using the manager interface to initiate calls. This means there
can be no valid callerid to either side of the call, and billing
(accountcodes) might break. Would it make more sense to signal to a small
daemon that can write spool files ? It might be more flexible. 

Also, we now set the channel to dial with, and it gets the number appended.
However, if you were to use chan_local, a suffix would be desirable (to set
context other than default).

Best regards,
Florian

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[Asterisk-Users] PC based Switchboard application files??

[Altus Snyman]

Good day all
I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard)
But in the pdf howto they speak about a swb.txt and I dont have that
file

Thanks
Altus

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Re: [Asterisk-Users] PC based Switchboard application files??

[Pertti Pikkarainen]

I'm so sorry. The file is now there.
Please download it.
Thanks !
Best regards Pertti

Altus Snyman wrote:

Good day all
I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard)
But in the pdf howto they speak about a swb.txt and I dont have that
file
Thanks
Altus
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Re: [Asterisk-Users] Asterisk & RedHat Enterprise

[George Pajari]

> Are their any issues with Asterisk and Redhat Enterprise?

None so far. We are running two Asterisk servers on RHEL 3 WS, one has 2
X100P, the other a T100P.

You can call 604 484 4020 (a DID on the T100P pri) and listen to the dulcet
tones of Allison Smith.

George Pajari
netVOICE communications

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Re: [Asterisk-Users] PC based Switchboard application files??

[Altus Snyman]

Thanks


On Thu, 2004-04-22 at 09:16, Pertti Pikkarainen wrote:
> I'm so sorry. The file is now there.
> Please download it.
> Thanks !
> 
> Best regards Pertti
> 
> 
> Altus Snyman wrote:
> 
> >Good day all
> >I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard)
> >But in the pdf howto they speak about a swb.txt and I dont have that
> >file
> >
> >Thanks
> >Altus
> >
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> >  
> >
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Re: [Asterisk-Users] PC based Switchboard application files??

[Altus Snyman]

Is this error ok? When I insert txt file into the db,Im loged in as
postgres

CREATE TABLE
INSERT 16984 1
CREATE TABLE
CREATE TABLE
INSERT 17003 1
CREATE TABLE
CREATE TABLE
CREATE TABLE
INSERT 17020 1
INSERT 17021 1
NOTICE:  CREATE TABLE will create implicit sequence 'cdr_acctid_seq' for
SERIAL column 'cdr.acctid'
CREATE TABLE
CREATE TABLE
NOTICE:  ALTER TABLE / ADD PRIMARY KEY will create implicit index
'cdr_pkey' for table 'cdr'
ALTER TABLE
 setval

454
(1 row)
 
ERROR:  Relation "asterikssettings" does not exist
GRANT
GRANT
GRANT
GRANT
GRANT
GRANT
GRANT
GRANT





On Thu, 2004-04-22 at 09:16, Pertti Pikkarainen wrote:
> I'm so sorry. The file is now there.
> Please download it.
> Thanks !
> 
> Best regards Pertti
> 
> 
> Altus Snyman wrote:
> 
> >Good day all
> >I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard)
> >But in the pdf howto they speak about a swb.txt and I dont have that
> >file
> >
> >Thanks
> >Altus
> >
> >___
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> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >  
> >
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Re: [Asterisk-Users] Ok, Im confused

[tmpm]

Thanks Jim will do...

At 20:58 4/21/2004, you wrote:
You can post your .conf files.

But here is a guess at what you may need
replace "FWD##"  with your freeworlddialup number and "mypassword" with 
your freeworlddialup
password.

in sip.conf

context = from-fwd
register=FWD##:[EMAIL PROTECTED]/FWD##
[fwd]
type=friend
secret=mypassword
username=FWD##
host=fwd.pulver.com
in phone.conf
...
context=from-phone
...
in extensions.conf

[from-fwd]
exten => FWD##,1,Dial(Phone/phone0)
exten => i,2,Playback(invalid)
[from-phone]
exten => _.,1,SetCallerID(FWD##)
exten => _.,2,SetCIDName(FWD##)
exten => _.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _.,4,Playback(invalid)
exten => _.,5,Hangup




Jim

James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: "tmpm" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, April 21, 2004 1:24 PM
Subject: Re: [Asterisk-Users] Ok, Im confused
> Thanks Jim,
> But that page started my trip off to confusionbeen theretried it 10
> different ways...still no joy.
> I'll go through it once again, maybe Im missing something, I dont know. Im
> about ready to boot the penguin to the curb...
> I know its in there...I think Ive got it all configured, and I dial the
> outbound strings, and get a fast busy...I know one stinking letter off, and
> its whacked...
> HOW for example do I specify my one and only extension is the Internet
> phone jack? Phone0?
> Somehow theres got to be a tie-in...I cant find it.
> been thru extensions.conf, phones.conf, sip.conf..etc.
> groan..
>
> At 18:40 4/21/2004, you wrote:
> >Look here:
> > http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
> >
> >Jim
> >
> >James H. Thompson
> >[EMAIL PROTECTED]
> >
> >- Original Message -
> >From: "tmpm" <[EMAIL PROTECTED]>
> >To: <[EMAIL PROTECTED]>
> >Sent: Wednesday, April 21, 2004 11:50 AM
> >Subject: [Asterisk-Users] Ok, Im confused
> >
> >
> > > Im totally a newbee at *
> > >
> > > Im confused.
> > > Ive got a FWD account, and it works on the winboxen. Ive got * up 
and can
> > > do the echotest etc, so its working.
> > >
> > > I want to get FWD working, and all the pages ive seen on setup are most
> > > confusing.
> > > Is FWD setup like IAXTEL? Do i plug in my FWD info in the same 
places as
> > > the IAXTEL stuff?
> > > Ive been trying for a week now, and Im more lost than before.
> > >
> > > Ive got a Internet phonejack card in the penguin, phone0, and all I
> > want to
> > > do at this point is make and receive calls thru FWD using that 
jackIll
> > > plug the house in later...Ill learn the other stuff later. No
> > voicemail, no
> > > BS, no dial thru least cost routing, or nightlines just make it
> > work as
> > > a phone.
> > >
> > > Im either more stupid than I think, or Im missing something major here.
> > >
> > > Ive got to the point the CLI shows me connected to FWD fine.(I think)
> > > Sip show users
> > >
> > > Username Secret Authen Def. Context a/c
> > > fwd.pulver.com secret md5,plaintext default no
> > >
> > > Need some basic, stupidly simple scripts here...I dont need or want to
> > dial
> > > 1-700 or *9 or any other crap, just make it work like the stupid winbox
> > > phone for now...Ill read the wiki for a couple years, and then 
maybe I can
> > > do voicemail or whatever...
> > >
> > > frustrated...and I know its showing...sri
> > >
> > > ___
> > > 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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> >http://lists.digium.com/mailman/listinfo/asterisk-users
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
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>

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Re: [Asterisk-Users] Ok, Im confused

[tmpm]

Its that now, its the machine facing my 4mb cablemodem...thanks..

At 01:54 4/22/2004, you wrote:
The simple answer probably is, If you have a NAT firewall (like a linksys,
netgear, dlink, etc) it will not work.
If your linux machine is directly connected follow the instructions on the
wiki and it will work no problems.  I could not get FWD to work at all until
I made my linux box  the outside edge of my network.
Scott
- Original Message -
From: "James H. Thompson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, April 21, 2004 8:58 PM
Subject: Re: [Asterisk-Users] Ok, Im confused
> You can post your .conf files.
>
> But here is a guess at what you may need
> replace "FWD##"  with your freeworlddialup number and "mypassword" with
your freeworlddialup
> password.
>
> in sip.conf
>
> context = from-fwd
> register=FWD##:[EMAIL PROTECTED]/FWD##
>
> [fwd]
> type=friend
> secret=mypassword
> username=FWD##
> host=fwd.pulver.com
>
> in phone.conf
> ...
> context=from-phone
> ...
>
> in extensions.conf
>
> [from-fwd]
> exten => FWD##,1,Dial(Phone/phone0)
> exten => i,2,Playback(invalid)
>
> [from-phone]
> exten => _.,1,SetCallerID(FWD##)
> exten => _.,2,SetCIDName(FWD##)
> exten => _.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
> exten => _.,4,Playback(invalid)
> exten => _.,5,Hangup
>
>
>
>
>
> Jim
>
> James H. Thompson
> [EMAIL PROTECTED]
>
> - Original Message -
> From: "tmpm" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, April 21, 2004 1:24 PM
> Subject: Re: [Asterisk-Users] Ok, Im confused
>
>
> > Thanks Jim,
> > But that page started my trip off to confusionbeen theretried it
10
> > different ways...still no joy.
> > I'll go through it once again, maybe Im missing something, I dont know.
Im
> > about ready to boot the penguin to the curb...
> > I know its in there...I think Ive got it all configured, and I dial the
> > outbound strings, and get a fast busy...I know one stinking letter off,
and
> > its whacked...
> > HOW for example do I specify my one and only extension is the Internet
> > phone jack? Phone0?
> > Somehow theres got to be a tie-in...I cant find it.
> > been thru extensions.conf, phones.conf, sip.conf..etc.
> > groan..
> >
> > At 18:40 4/21/2004, you wrote:
> > >Look here:
> > > http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
> > >
> > >Jim
> > >
> > >James H. Thompson
> > >[EMAIL PROTECTED]
> > >
> > >- Original Message -
> > >From: "tmpm" <[EMAIL PROTECTED]>
> > >To: <[EMAIL PROTECTED]>
> > >Sent: Wednesday, April 21, 2004 11:50 AM
> > >Subject: [Asterisk-Users] Ok, Im confused
> > >
> > >
> > > > Im totally a newbee at *
> > > >
> > > > Im confused.
> > > > Ive got a FWD account, and it works on the winboxen. Ive got * up
and can
> > > > do the echotest etc, so its working.
> > > >
> > > > I want to get FWD working, and all the pages ive seen on setup are
most
> > > > confusing.
> > > > Is FWD setup like IAXTEL? Do i plug in my FWD info in the same
places as
> > > > the IAXTEL stuff?
> > > > Ive been trying for a week now, and Im more lost than before.
> > > >
> > > > Ive got a Internet phonejack card in the penguin, phone0, and all I
> > > want to
> > > > do at this point is make and receive calls thru FWD using that
jackIll
> > > > plug the house in later...Ill learn the other stuff later. No
> > > voicemail, no
> > > > BS, no dial thru least cost routing, or nightlines just make it
> > > work as
> > > > a phone.
> > > >
> > > > Im either more stupid than I think, or Im missing something major
here.
> > > >
> > > > Ive got to the point the CLI shows me connected to FWD fine.(I
think)
> > > > Sip show users
> > > >
> > > > Username Secret Authen Def. Context a/c
> > > > fwd.pulver.com secret md5,plaintext default no
> > > >
> > > > Need some basic, stupidly simple scripts here...I dont need or want
to
> > > dial
> > > > 1-700 or *9 or any other crap, just make it work like the stupid
winbox
> > > > phone for now...Ill read the wiki for a couple years, and then maybe
I can
> > > > do voicemail or whatever...
> > > >
> > > > frustrated...and I know its showing...sri
> > > >
> > > > ___
> > > > 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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> >
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Re: [Asterisk-Users] Install Timer to run MeetMe service

[PTCHEN]

(1) error message after make
zaptel.c:5937: storage size of `zt_fops' isn't known
/usr/include/linux/proc_fs.h:193: warning: `create_proc_read_entry' declared
`st
atic' but never defined
make: *** [zaptel.o] Error 1

(2) error messgae after make install
zaptel.c:5937: storage size of `zt_fops' isn't known
/usr/include/linux/proc_fs.h:193: warning: `create_proc_read_entry' declared
`st
atic' but never defined
make: *** [zaptel.o] Error 1

- Original Message -
From: "Dave Cotton" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, April 22, 2004 2:54 PM
Subject: Re: [Asterisk-Users] Install Timer to run MeetMe service


> On Thu, 2004-04-22 at 14:41 +0800, PTCHEN wrote:
>
> > I get zaptel.tar.gz from ftp.asterisk.org , uncomment ztdummy,
> > then make -> make install, but it fail with errors.
> > Where and How can I get (not thru cvs) ztdummy module, and get it
> > compiled ?
>
> To be able to help at all we must know what the errors are.
> Post your error report.
> --
> Dave Cotton
>
> http://www.linuxautrement.com
>
>
>
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Re: [Asterisk-Users] PC based Switchboard application files??

[Pertti Pikkarainen]

You are ok already.
It should work even if that privilege line is missing.
But to be sure later you can easily fix that.
The error is due to a typo in the end of the file
Run the first  GRANT command again with 'asterisksettings' and not 
'asterikssettings'

I just fixed the download file.

Best regards Pertti



Altus Snyman wrote:

Is this error ok? When I insert txt file into the db,Im loged in as
postgres
CREATE TABLE
INSERT 16984 1
CREATE TABLE
CREATE TABLE
INSERT 17003 1
CREATE TABLE
CREATE TABLE
CREATE TABLE
INSERT 17020 1
INSERT 17021 1
NOTICE:  CREATE TABLE will create implicit sequence 'cdr_acctid_seq' for
SERIAL column 'cdr.acctid'
CREATE TABLE
CREATE TABLE
NOTICE:  ALTER TABLE / ADD PRIMARY KEY will create implicit index
'cdr_pkey' for table 'cdr'
ALTER TABLE
setval

   454
(1 row)
ERROR:  Relation "asterikssettings" does not exist
GRANT
GRANT
GRANT
GRANT
GRANT
GRANT
GRANT
GRANT




On Thu, 2004-04-22 at 09:16, Pertti Pikkarainen wrote:
 

I'm so sorry. The file is now there.
Please download it.
Thanks !
Best regards Pertti

Altus Snyman wrote:

   

Good day all
I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard)
But in the pdf howto they speak about a swb.txt and I dont have that
file
Thanks
Altus
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--

**
Nordic LAN&WAN Communication Oy
Pertti Pikkarainen
vp of engineering 
WWW: http://www.lanwan.fi
E-Mail: [EMAIL PROTECTED]
tel: +358-9-4243 
fax: +358-9-5023840
gsm: +358-500-511467

Sinikalliontie 16
02630 Espoo
FINLAND
**

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RE: [Asterisk-Users] Ok, Im confused

[William J Mandra]

Take a look at my website http://www.mandra.homeip.net/asterisk the config
files are also posted on the wiki. If you are still having problems call me
at FWD 290805.

Bill

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of tmpm
Sent: Thursday, April 22, 2004 ONYX 3:27 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Ok, Im confused


Its that now, its the machine facing my 4mb cablemodem...thanks..

At 01:54 4/22/2004, you wrote:
>The simple answer probably is, If you have a NAT firewall (like a linksys,
>netgear, dlink, etc) it will not work.
>
>If your linux machine is directly connected follow the instructions on the
>wiki and it will work no problems.  I could not get FWD to work at all
until
>I made my linux box  the outside edge of my network.
>
>Scott
>- Original Message -
>From: "James H. Thompson" <[EMAIL PROTECTED]>
>To: <[EMAIL PROTECTED]>
>Sent: Wednesday, April 21, 2004 8:58 PM
>Subject: Re: [Asterisk-Users] Ok, Im confused
>
>
> > You can post your .conf files.
> >
> > But here is a guess at what you may need
> > replace "FWD##"  with your freeworlddialup number and "mypassword" with
>your freeworlddialup
> > password.
> >
> > in sip.conf
> >
> > context = from-fwd
> > register=FWD##:[EMAIL PROTECTED]/FWD##
> >
> > [fwd]
> > type=friend
> > secret=mypassword
> > username=FWD##
> > host=fwd.pulver.com
> >
> > in phone.conf
> > ...
> > context=from-phone
> > ...
> >
> > in extensions.conf
> >
> > [from-fwd]
> > exten => FWD##,1,Dial(Phone/phone0)
> > exten => i,2,Playback(invalid)
> >
> > [from-phone]
> > exten => _.,1,SetCallerID(FWD##)
> > exten => _.,2,SetCIDName(FWD##)
> > exten => _.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
> > exten => _.,4,Playback(invalid)
> > exten => _.,5,Hangup
> >
> >
> >
> >
> >
> > Jim
> >
> > James H. Thompson
> > [EMAIL PROTECTED]
> >
> > - Original Message -
> > From: "tmpm" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Wednesday, April 21, 2004 1:24 PM
> > Subject: Re: [Asterisk-Users] Ok, Im confused
> >
> >
> > > Thanks Jim,
> > > But that page started my trip off to confusionbeen theretried
it
>10
> > > different ways...still no joy.
> > > I'll go through it once again, maybe Im missing something, I dont
know.
>Im
> > > about ready to boot the penguin to the curb...
> > > I know its in there...I think Ive got it all configured, and I dial
the
> > > outbound strings, and get a fast busy...I know one stinking letter
off,
>and
> > > its whacked...
> > > HOW for example do I specify my one and only extension is the Internet
> > > phone jack? Phone0?
> > > Somehow theres got to be a tie-in...I cant find it.
> > > been thru extensions.conf, phones.conf, sip.conf..etc.
> > > groan..
> > >
> > > At 18:40 4/21/2004, you wrote:
> > > >Look here:
> > > > http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
> > > >
> > > >Jim
> > > >
> > > >James H. Thompson
> > > >[EMAIL PROTECTED]
> > > >
> > > >- Original Message -
> > > >From: "tmpm" <[EMAIL PROTECTED]>
> > > >To: <[EMAIL PROTECTED]>
> > > >Sent: Wednesday, April 21, 2004 11:50 AM
> > > >Subject: [Asterisk-Users] Ok, Im confused
> > > >
> > > >
> > > > > Im totally a newbee at *
> > > > >
> > > > > Im confused.
> > > > > Ive got a FWD account, and it works on the winboxen. Ive got * up
>and can
> > > > > do the echotest etc, so its working.
> > > > >
> > > > > I want to get FWD working, and all the pages ive seen on setup are
>most
> > > > > confusing.
> > > > > Is FWD setup like IAXTEL? Do i plug in my FWD info in the same
>places as
> > > > > the IAXTEL stuff?
> > > > > Ive been trying for a week now, and Im more lost than before.
> > > > >
> > > > > Ive got a Internet phonejack card in the penguin, phone0, and all
I
> > > > want to
> > > > > do at this point is make and receive calls thru FWD using that
>jackIll
> > > > > plug the house in later...Ill learn the other stuff later. No
> > > > voicemail, no
> > > > > BS, no dial thru least cost routing, or nightlines just make
it
> > > > work as
> > > > > a phone.
> > > > >
> > > > > Im either more stupid than I think, or Im missing something major
>here.
> > > > >
> > > > > Ive got to the point the CLI shows me connected to FWD fine.(I
>think)
> > > > > Sip show users
> > > > >
> > > > > Username Secret Authen Def. Context a/c
> > > > > fwd.pulver.com secret md5,plaintext default no
> > > > >
> > > > > Need some basic, stupidly simple scripts here...I dont need or
want
>to
> > > > dial
> > > > > 1-700 or *9 or any other crap, just make it work like the stupid
>winbox
> > > > > phone for now...Ill read the wiki for a couple years, and then
maybe
>I can
> > > > > do voicemail or whatever...
> > > > >
> > > > > frustrated...and I know its showing...sri
> > > > >
> > > > > ___
> > > > > Asterisk-Users mailing list
> > > > > [EMAIL PROTECTED]
> > > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > > To UNS

Re: [Asterisk-Users] PC based Switchboard application files??

[Altus Snyman]

Thanks
I'll have a look now just waiting for the war file to download
As far as I can remember this is not open-source? How much will you
charge for it?
Altus



On Thu, 2004-04-22 at 09:50, Pertti Pikkarainen wrote:
> You are ok already.
> It should work even if that privilege line is missing.
> 
> But to be sure later you can easily fix that.
> The error is due to a typo in the end of the file


> Run the first  GRANT command again with 'asterisksettings' and not 
> 'asterikssettings'
> 
> I just fixed the download file.
> 
> Best regards Pertti
> 
> 
> 
> Altus Snyman wrote:
> 
> >Is this error ok? When I insert txt file into the db,Im loged in as
> >postgres
> >
> >CREATE TABLE
> >INSERT 16984 1
> >CREATE TABLE
> >CREATE TABLE
> >INSERT 17003 1
> >CREATE TABLE
> >CREATE TABLE
> >CREATE TABLE
> >INSERT 17020 1
> >INSERT 17021 1
> >NOTICE:  CREATE TABLE will create implicit sequence 'cdr_acctid_seq' for
> >SERIAL column 'cdr.acctid'
> >CREATE TABLE
> >CREATE TABLE
> >NOTICE:  ALTER TABLE / ADD PRIMARY KEY will create implicit index
> >'cdr_pkey' for table 'cdr'
> >ALTER TABLE
> > setval
> >
> >454
> >(1 row)
> > 
> >ERROR:  Relation "asterikssettings" does not exist
> >GRANT
> >GRANT
> >GRANT
> >GRANT
> >GRANT
> >GRANT
> >GRANT
> >GRANT
> >
> >
> >
> >
> >
> >On Thu, 2004-04-22 at 09:16, Pertti Pikkarainen wrote:
> >  
> >
> >>I'm so sorry. The file is now there.
> >>Please download it.
> >>Thanks !
> >>
> >>Best regards Pertti
> >>
> >>
> >>Altus Snyman wrote:
> >>
> >>
> >>
> >>>Good day all
> >>>I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard)
> >>>But in the pdf howto they speak about a swb.txt and I dont have that
> >>>file
> >>>
> >>>Thanks
> >>>Altus
> >>>
> >>>___
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> >>>
> >>>  
> >>>
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> >>
> >>
> >
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> >  
> >

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Re: [Asterisk-Users] Asterisk & RedHat Enterprise

[Fran Boon]

Asterisk wrote:
Are their any issues with Asterisk and Redhat Enterprise? I have see one 
or two posts with issues concerning compiling zaptel drivers but that is 
about it. Just looking for some consensus to if any problems exist with it.
Works perfectly for me :)

F
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[Asterisk-Users] Asterisk with UUI support ?

[jean-marie . goupil]

Hi there,

Is it possible to manage UUI with asterisk and ISDN (T0 Fritz card).
Basically, is it possible to send User to User Information using the
D-channel, while making a call?

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Re: [Asterisk-Users] PC based Switchboard application files??

[Altus Snyman]

It comes up with the index page but when you login with admin,admin it
says error logging on to database,postgresql is running as postgres user
and the db has been added with the txt file,I did the change in tomcat
folder.

Must postgresql run as postgres user? 
Any Ideas

Thanks
Altus

On Thu, 2004-04-22 at 09:31, Altus Snyman wrote:
> Is this error ok? When I insert txt file into the db,Im loged in as
> postgres
> 
> CREATE TABLE
> INSERT 16984 1
> CREATE TABLE
> CREATE TABLE
> INSERT 17003 1
> CREATE TABLE
> CREATE TABLE
> CREATE TABLE
> INSERT 17020 1
> INSERT 17021 1
> NOTICE:  CREATE TABLE will create implicit sequence 'cdr_acctid_seq' for
> SERIAL column 'cdr.acctid'
> CREATE TABLE
> CREATE TABLE
> NOTICE:  ALTER TABLE / ADD PRIMARY KEY will create implicit index
> 'cdr_pkey' for table 'cdr'
> ALTER TABLE
>  setval
> 
> 454
> (1 row)
>  
> ERROR:  Relation "asterikssettings" does not exist
> GRANT
> GRANT
> GRANT
> GRANT
> GRANT
> GRANT
> GRANT
> GRANT
> 
> 
> 
> 
> 
> On Thu, 2004-04-22 at 09:16, Pertti Pikkarainen wrote:
> > I'm so sorry. The file is now there.
> > Please download it.
> > Thanks !
> > 
> > Best regards Pertti
> > 
> > 
> > Altus Snyman wrote:
> > 
> > >Good day all
> > >I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard)
> > >But in the pdf howto they speak about a swb.txt and I dont have that
> > >file
> > >
> > >Thanks
> > >Altus
> > >
> > >___
> > >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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> > 
> 
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Re: [Asterisk-Users] Cisco 7940/7960 SIP functionality questions

[Vic Cross]

On Thu, 22 Apr 2004, Louis van Dompselaar wrote:

> > There are two SCCP modules, but I haven't heard about anyone using  
> > 7940/60s with SCCP and Asterisk.
> 
> Let me be the first to say that I do, then.  Three 7940g over Skinny 
> without any problems.
> Some initial setup trouble as this setup isn't really documented 
> anywhere.  But no Cisco login,
> so no SIP and no choice...

And let me be the second ;)  A 7960, using chan_sccp from Lambda 
Solutions.  Multiple line presentations work well (except things go a 
little weird when you're on a call on one line and a call comes in on 
another).  I echo Louis' comments that documentation is quite sparse, but 
I really didn't need any more than the well-commented sample sccp.conf.

I have a second 7960 that I'll probably use for SIP playing, if I ever 
find a Cisco reseller who will take my meager business (one or two 
Smartnets for phones is obviously beneath them).

Cheers,
Vic Cross
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RE: [Asterisk-Users] Ser and Asterisk together

[Barry Flanagan]

I am finally making some progress on this.

I now have SER passing off PSTN calls to * OK. Calls are being
connected, however, I don't hear anything on the SIP end, and asterisk
gives the following error:

WARNING[9226]: chan_sip.c:457 __sip_xmit: sip_xmit of 0x80e2dec (len
642) to 212.17.32.215 returned -1: Operation not permitted


Below is the context of this. I am using nathelper on SER, but I am not
at all confident of my config file (it being a patchwork of bits from
different examples. I attach my SER conf at the end of this message.

Should * be talking directly with the SIP UA, or should it be talking to
SER? 

Any help would be appreciated! Even better would be a sample ser.cfg
which supports nathelper and using * for VM and PSTN!!


to 212.17.32.215:3568
Apr 22 12:31:38 WARNING[9226]: chan_sip.c:457 __sip_xmit: sip_xmit of
0x80e2dec (len 642) to 212.17.32.215 returned -1: Operation not
permitted
Retransmitting #2 (no NAT):
INVITE sip:[EMAIL PROTECTED]:3568 SIP/2.0
Via: SIP/2.0/UDP 212.17.35.184:5060;branch=z9hG4bK4c8263f2
From: ;tag=as4e38a4ab
To: "Ray Naughton"
;tag=e64bcbbe63564744
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 164

v=0
o=root 21443 21445 IN IP4 213.137.65.251
s=session
c=IN IP4 213.137.65.251
t=0 0
m=audio 16670 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

 to 212.17.32.215:3568
Apr 22 12:31:39 WARNING[9226]: chan_sip.c:457 __sip_xmit: sip_xmit of
0x80e2dec (len 642) to 212.17.32.215 returned -1: Operation not
permitted
zeppelin*CLI>


= ser.cfg 

#
# $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $
#
# simple quick-start config script
#

# --- global configuration parameters 

debug=7 # debug level (cmd line: -dd)
fork=no
log_stderror=yes # (cmd line: -E)


listen=213.159.144.8
#listen=127.0.0.1

# hostname matching an alias will satisfy the condition uri==myself".
alias=voip.edo.ie
alias=avmx.edo.ie



# Uncomment these lines to enter debugging mode 
/*
debug=7
fork=no
log_stderror=yes
*/

check_via=no# (cmd. line: -v)
dns=no   # (cmd. line: -r)
rev_dns=no  # (cmd. line: -R)
port=5060
children=4
fifo="/tmp/ser_fifo"
alias=voip.edo.ie avmx.edo.ie localhost 

# -- module loading --

# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/ser/modules/mysql.so"

loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"

# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"

# load the voicemail module
#loadmodule "/usr/local/lib/ser/modules/vm.so"

# load the enum module
loadmodule "/usr/local/lib/ser/modules/enum.so"

# load the group module, to verify if a user forwards to voicemail
loadmodule "/usr/local/lib/ser/modules/group.so"

# load the nathelper module
loadmodule "/usr/local/lib/ser/modules/nathelper.so"

# - setting module-specific parameters ---

# -- registrar parameter
# special NAT flag indicates that a registered client is behind NAT
modparam("registrar", "nat_flag", 6)

# -- usrloc params --

#modparam("usrloc", "db_mode",   0)

# Uncomment this if you want to use SQL database 
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
#modparam("usrloc", "db_url", "mysql://ser:[EMAIL PROTECTED]/ser")
modparam("usrloc|auth_db|acc|group|msilo|uri","db_url","mysql://ser:[EMAIL 
PROTECTED]/ser")

# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config), 
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
#modparam("auth_db", "db_url", "mysql://ser:[EMAIL PROTECTED]/ser")

# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)

# -- voicemail params --
#modparam("voicemail", "db_url","mysql://ser:[EMAIL PROTECTED]/ser")

# -- voicemail params --
#modparam("group", "db_url","mysql://serro:[EMAIL PROTECTED]/ser")

# -- nathelper params --
modparam("nathelper", "natping_interval", 60)
modparam("nathelper", "ping_nated_only", 1)

modparam("tm", "fr_inv_timer", 30 )
#modparam("tm", "fr_inv_timer", 8 )

# -  request routing logic ---

# main routing logic

route{
log(1, "---\n");
log(1, "entering main loop\n");

[Asterisk-Users] asterisk no card

[Altus Snyman]

Good day all
Is it possible to run asterisk and sip without any
cards,(t100,voicetronix)
Just a plain linux server,running mail and web, and add asterisk
At the moment they are running msn?
Tanks
Altus

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Re: [Asterisk-Users] PSTN incoming - both SIP & H323 always arrive in default context :-?

[Kelvin Chua]

this is something i just recently noticed.
have you found any info on how to manage incoming calls through
chan_h323? it doesn't seem to match any entity you define, it always
uses the default context...

On Sat, 2004-01-24 at 02:39, Fran Boon wrote:
> Some of you may remember seeing my issue using SIP for incoming calls 
> from the PSTN:
> http://voip-info.org/wiki-Asterisk+cisco+FXO
> 
> i.e. all incoming calls arrive in the default 'bogon-calls' context.
> 
> 
> Well, I tried again using H.323 & get exactly the same result (both for 
> chan_h323 & chan_oh323)
> 
> i.e. all attempts to put a type=peer in sip.conf or a type=user in 
> h323.conf for my host are ignored/bypassed.
> 
> Is this a bug?
> 
> 
> Luckily for me, I can firewall off the H.323 port to all bar this one 
> IP, so I now have a workable solution...until I want to extend the H.323 
> gateway to other devices...
> 
> Anyone get host=x.x.x.x to be able to bypass the default contexts with 
> either SIP or H.323?
> 
> Cheers,
> Fran.
> 
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Re: [Asterisk-Users] asterisk no card

[Fran Boon]

Altus Snyman wrote:
Is it possible to run asterisk and sip without any
cards,(t100,voicetronix)
Just a plain linux server,running mail and web, and add asterisk
Yes

F
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Re: [Asterisk-Users] PC based Switchboard application files??

[Pertti Pikkarainen]

After restarting postgres you need to stop and start
$CATALINA_HOME/bin/shutdown.sh
$CATALINA_HOME/bin/startup.sh
Or something is wrong with the postgre access rights.
Did you remember  modify
/usr/local/pgsql/data/pg_hba.conf
If a new start doesn't help
please, send me $CATALINA_HOME/logs/catalina.out
Best regards Pertti

Altus Snyman wrote:

It comes up with the index page but when you login with admin,admin it
says error logging on to database,postgresql is running as postgres user
and the db has been added with the txt file,I did the change in tomcat
folder.
Must postgresql run as postgres user? 
Any Ideas

Thanks
Altus
On Thu, 2004-04-22 at 09:31, Altus Snyman wrote:
 

Is this error ok? When I insert txt file into the db,Im loged in as
postgres
CREATE TABLE
INSERT 16984 1
CREATE TABLE
CREATE TABLE
INSERT 17003 1
CREATE TABLE
CREATE TABLE
CREATE TABLE
INSERT 17020 1
INSERT 17021 1
NOTICE:  CREATE TABLE will create implicit sequence 'cdr_acctid_seq' for
SERIAL column 'cdr.acctid'
CREATE TABLE
CREATE TABLE
NOTICE:  ALTER TABLE / ADD PRIMARY KEY will create implicit index
'cdr_pkey' for table 'cdr'
ALTER TABLE
setval

   454
(1 row)
ERROR:  Relation "asterikssettings" does not exist
GRANT
GRANT
GRANT
GRANT
GRANT
GRANT
GRANT
GRANT




On Thu, 2004-04-22 at 09:16, Pertti Pikkarainen wrote:
   

I'm so sorry. The file is now there.
Please download it.
Thanks !
Best regards Pertti

Altus Snyman wrote:

 

Good day all
I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard)
But in the pdf howto they speak about a swb.txt and I dont have that
file
Thanks
Altus
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-

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[Asterisk-Users] Flash panel

[Altus Snyman]

Good day all
Did someone get the new ver0.5 flash panel working
Is it suppose not to show who the caller is calling,like on ver0.2?
And how do I change the language
Thanks
Altus


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[Asterisk-Users] ALSA help required !

[Adnan Shah]

 I have just installed the Alsa drivers
 for my 2.4.18-14 kernel (RH8). I have configured
 the sound card ok with alsaconf and tested
 with the aplay , works fine. But when I run
 asterisk it says..
 ---
 [chan_alsa.so] => (ALSA Console Channel Driver)
 Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init:
 snd_pcm_open failed: No such device or address
 Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init:
 snd_pcm_open failed: No such device or address
 Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:474 soundcard_init: Problem
 opening alsa I/O devices
 
   == No sound card detected -- console channel will be unavailable
 
   == Turn off ALSA support by adding 'noload=chan_alsa.so' in
 /etc/asterisk/modules.conf
 
 --
earlier when using the OSS, the playback was choppy not smooth,
I added some more RAM (total 256 on Intel PIII 600 processor), but the
problem was still there so I turned to the Alsa drivers.Asterisk doesn't
seem to work with it what might be wrong, any ideas ?



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Re: [Asterisk-Users] Ok, Im confused

[Brian Cuthie]

Not exactly :-)

Actually, you can get FWD to work through a NAT. SIP will send its 
registration requests out the same port it expects a response on. This 
will create a mapping in your NAT so that when INVITEs appear from FWD 
the NAT can figure out which local host to send them to. Same thing 
happens with SIP between the caller and the local machine. RTP does the 
same trick too. I do this all the time with a Linksys router without any 
difficulty.

Now things get a little more complicated if you have more than one SIP 
device behind the firewall that's trying to talk to the outside world.  
You'll need to configure each to use a unique SIP port, as well as a 
unique range of RTP ports. If not, the NAT will see more than one 
local_ip:proto:port tuple and will have to remap the port as the packet 
leaves the router. But since the VIA header in the SIP packets will 
refer to the original port, incoming SIP traffic will end up at the 
wrong local host. With * you can avoid these problems by having all 
traffic go through the * box. Do this by adding "reinvite=no" to your 
sip.conf, and configure your SIP phones to use * as a proxy. Do not turn 
on NAT in the SIP phones.

Actual firewalls (as apposed to just NATs like on a Linksys) may pose 
additional problems depending on how they're configured. But that's 
probably now what you're experiencing.

Cheers,

-brian

Scott Weis wrote:

The simple answer probably is, If you have a NAT firewall (like a linksys,
netgear, dlink, etc) it will not work.
If your linux machine is directly connected follow the instructions on the
wiki and it will work no problems.  I could not get FWD to work at all until
I made my linux box  the outside edge of my network.
Scott
- Original Message - 
From: "James H. Thompson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, April 21, 2004 8:58 PM
Subject: Re: [Asterisk-Users] Ok, Im confused

 

You can post your .conf files.

But here is a guess at what you may need
replace "FWD##"  with your freeworlddialup number and "mypassword" with
   

your freeworlddialup
 

password.

in sip.conf

context = from-fwd
register=FWD##:[EMAIL PROTECTED]/FWD##
[fwd]
type=friend
secret=mypassword
username=FWD##
host=fwd.pulver.com
in phone.conf
...
context=from-phone
...
in extensions.conf

[from-fwd]
exten => FWD##,1,Dial(Phone/phone0)
exten => i,2,Playback(invalid)
[from-phone]
exten => _.,1,SetCallerID(FWD##)
exten => _.,2,SetCIDName(FWD##)
exten => _.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _.,4,Playback(invalid)
exten => _.,5,Hangup




Jim

James H. Thompson
[EMAIL PROTECTED]
- Original Message - 
From: "tmpm" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, April 21, 2004 1:24 PM
Subject: Re: [Asterisk-Users] Ok, Im confused

   

Thanks Jim,
But that page started my trip off to confusionbeen theretried it
 

10
 

different ways...still no joy.
I'll go through it once again, maybe Im missing something, I dont know.
 

Im
 

about ready to boot the penguin to the curb...
I know its in there...I think Ive got it all configured, and I dial the
outbound strings, and get a fast busy...I know one stinking letter off,
 

and
 

its whacked...
HOW for example do I specify my one and only extension is the Internet
phone jack? Phone0?
Somehow theres got to be a tie-in...I cant find it.
been thru extensions.conf, phones.conf, sip.conf..etc.
groan..
At 18:40 4/21/2004, you wrote:
 

Look here:
   http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
Jim

James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: "tmpm" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, April 21, 2004 11:50 AM
Subject: [Asterisk-Users] Ok, Im confused
   

Im totally a newbee at *

Im confused.
Ive got a FWD account, and it works on the winboxen. Ive got * up
 

and can
 

do the echotest etc, so its working.

I want to get FWD working, and all the pages ive seen on setup are
 

most
 

confusing.
Is FWD setup like IAXTEL? Do i plug in my FWD info in the same
 

places as
 

the IAXTEL stuff?
Ive been trying for a week now, and Im more lost than before.
Ive got a Internet phonejack card in the penguin, phone0, and all I
 

want to
   

do at this point is make and receive calls thru FWD using that
 

jackIll
 

plug the house in later...Ill learn the other stuff later. No
 

voicemail, no
   

BS, no dial thru least cost routing, or nightlines just make it
 

work as
   

a phone.

Im either more stupid than I think, or Im missing something major
 

here.
 

Ive got to the point the CLI shows me connected to FWD fine.(I
 

think)
 

Sip show users

Username Secret Authen Def. Context a/c
fwd.pulver.com secret md5,plaintext default no
Need some basic, stupidly simple scripts here...I dont need or want
 

to
 

dial
   

1-700 or *9 or any other crap, just make it w

RE: [Asterisk-Users] Ser and Asterisk together

[Dawid Mielnik]

In my setup * is talking to sip us through ser - this is done by setting the
record route parameter in ser routing logic. A laso pass the media stream
thorugh ser - this is done through the rtpproxy module (ser).

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Barry
Flanagan
Sent: Thursday, April 22, 2004 1:46 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Ser and Asterisk together


I am finally making some progress on this.

I now have SER passing off PSTN calls to * OK. Calls are being
connected, however, I don't hear anything on the SIP end, and asterisk
gives the following error:

WARNING[9226]: chan_sip.c:457 __sip_xmit: sip_xmit of 0x80e2dec (len
642) to 212.17.32.215 returned -1: Operation not permitted


Below is the context of this. I am using nathelper on SER, but I am not
at all confident of my config file (it being a patchwork of bits from
different examples. I attach my SER conf at the end of this message.

Should * be talking directly with the SIP UA, or should it be talking to
SER?

Any help would be appreciated! Even better would be a sample ser.cfg
which supports nathelper and using * for VM and PSTN!!


to 212.17.32.215:3568
Apr 22 12:31:38 WARNING[9226]: chan_sip.c:457 __sip_xmit: sip_xmit of
0x80e2dec (len 642) to 212.17.32.215 returned -1: Operation not
permitted
Retransmitting #2 (no NAT):
INVITE sip:[EMAIL PROTECTED]:3568 SIP/2.0
Via: SIP/2.0/UDP 212.17.35.184:5060;branch=z9hG4bK4c8263f2
From: ;tag=as4e38a4ab
To: "Ray Naughton"
;tag=e64bcbbe63564744
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 164

v=0
o=root 21443 21445 IN IP4 213.137.65.251
s=session
c=IN IP4 213.137.65.251
t=0 0
m=audio 16670 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

 to 212.17.32.215:3568
Apr 22 12:31:39 WARNING[9226]: chan_sip.c:457 __sip_xmit: sip_xmit of
0x80e2dec (len 642) to 212.17.32.215 returned -1: Operation not
permitted
zeppelin*CLI>


= ser.cfg 

#
# $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $
#
# simple quick-start config script
#

# --- global configuration parameters 

debug=7 # debug level (cmd line: -dd)
fork=no
log_stderror=yes # (cmd line: -E)


listen=213.159.144.8
#listen=127.0.0.1

# hostname matching an alias will satisfy the condition uri==myself".
alias=voip.edo.ie
alias=avmx.edo.ie



# Uncomment these lines to enter debugging mode
/*
debug=7
fork=no
log_stderror=yes
*/

check_via=no# (cmd. line: -v)
dns=no   # (cmd. line: -r)
rev_dns=no  # (cmd. line: -R)
port=5060
children=4
fifo="/tmp/ser_fifo"
alias=voip.edo.ie avmx.edo.ie localhost

# -- module loading --

# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/ser/modules/mysql.so"

loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"

# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"

# load the voicemail module
#loadmodule "/usr/local/lib/ser/modules/vm.so"

# load the enum module
loadmodule "/usr/local/lib/ser/modules/enum.so"

# load the group module, to verify if a user forwards to voicemail
loadmodule "/usr/local/lib/ser/modules/group.so"

# load the nathelper module
loadmodule "/usr/local/lib/ser/modules/nathelper.so"

# - setting module-specific parameters ---

# -- registrar parameter
# special NAT flag indicates that a registered client is behind NAT
modparam("registrar", "nat_flag", 6)

# -- usrloc params --

#modparam("usrloc", "db_mode",   0)

# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
#modparam("usrloc", "db_url", "mysql://ser:[EMAIL PROTECTED]/ser")
modparam("usrloc|auth_db|acc|group|msilo|uri","db_url","mysql://ser:[EMAIL PROTECTED]
calhost/ser")

# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
#modparam("auth_db", "db_url", "mysql://ser:[EMAIL PROTECTED]/ser")

# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)

# -- voicemail params --
#modparam("voicemail", "db_url","mysql://ser:[EMAIL PROTECTED]/ser")

# -- voicemail params --
#modparam("group", "db_url","mysql:/

RE: [Asterisk-Users] Help choosing a UK IAX provider

[Azher Amin]

Hi,

I have recently used the service from Magrathea, a UK IAX provider. So
far Voice quality is perfect.

You can mail to:

[EMAIL PROTECTED]


Regards
Azher
---
http://www.consulttech.com.pk


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of tmpm
Sent: Thursday, April 22, 2004 12:30 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Help choosing a UK IAX provider

Yeah, primarily fired at them from large telcos with infinite
bandwidth...

At 11:01 4/21/2004, you wrote:
>Wait till DDOS/extortion scams start hitting voip providers!
>
>Panny

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[Asterisk-Users] Double echo cancellation disable?

[Tais M. Hansen]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi,

I was going through the logs on one Asterisk server and found this:

Apr 22 14:46:16 gw-1 asterisk[22088]: DEBUG[132972557]: chan_zap.c:1153 in 
zt_disable_ec: disabled echo cancellation on channel 2
Apr 22 14:46:16 gw-1 asterisk[22088]: DEBUG[132972557]: chan_zap.c:1153 in 
zt_disable_ec: disabled echo cancellation on channel 2
Apr 22 14:46:40 gw-1 asterisk[22081]: DEBUG[132890638]: chan_zap.c:1153 in 
zt_disable_ec: disabled echo cancellation on channel 3
Apr 22 14:46:40 gw-1 asterisk[22081]: DEBUG[132890638]: chan_zap.c:1153 in 
zt_disable_ec: disabled echo cancellation on channel 3
Apr 22 14:46:41 gw-1 asterisk[22098]: DEBUG[133136401]: chan_zap.c:1153 in 
zt_disable_ec: disabled echo cancellation on channel 25
Apr 22 14:46:41 gw-1 asterisk[22098]: DEBUG[133136401]: chan_zap.c:1153 in 
zt_disable_ec: disabled echo cancellation on channel 25
Apr 22 14:47:01 gw-1 asterisk[22079]: DEBUG[132857868]: chan_zap.c:1153 in 
zt_disable_ec: disabled echo cancellation on channel 1
Apr 22 14:47:01 gw-1 asterisk[22079]: DEBUG[132857868]: chan_zap.c:1153 in 
zt_disable_ec: disabled echo cancellation on channel 1

Is it intentional that it seems to be disabling EC on the same channel twice?

(CVS-04/09/04-23:00:24-CEST)

- -- 
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374

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Version: GnuPG v1.2.3 (GNU/Linux)

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W4laWw8uQnH5dbLW3iYSkdY=
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[Asterisk-Users] Detecting Distinctive ring on a POTS line

[Andre Normandin]

Hello,

I have a POTS with distinctive ring on it from the phone company. It looks
like there is a way with asterisk to create distinctive rings, something
like dial(ZAP/1rx) but is there a way with asterisk (using an X100P PCI
card) to detect a distinctive ring from a POTS line?

Thanks for your help,
   - Andre


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Re: [Asterisk-Users] Flash panel

[Nicolas Gudino]

- Original Message - 
From: "Altus Snyman" <[EMAIL PROTECTED]>


> Good day all
> Did someone get the new ver0.5 flash panel working
> Is it suppose not to show who the caller is calling,like on ver0.2?
> And how do I change the language
> Thanks
> Altus

Hi Altus,

There is a mailing list for the flash operator panel. Please use that
mailing list for discussing the application. You can subscribe by sending an
empty email to:

[EMAIL PROTECTED]

The panel works if its properly configured. You can translate or change the
text information by editing the op_server.pl. Good luck,


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RE: [Asterisk-Users] Ser and Asterisk together

[Barry Flanagan]

On Thu, 2004-04-22 at 13:47, Dawid Mielnik wrote:
> In my setup * is talking to sip us through ser - this is done by setting the
> record route parameter in ser routing logic. A laso pass the media stream
> thorugh ser - this is done through the rtpproxy module (ser).
> 

Any chance of seeing your ser.cfg file?

Thanks.

-- 
-Barry Flanagan

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[Asterisk-Users] Asterisk with 3rd party voicemail

[Jason Jessico]

Is anyone using Asterisk with a 3rd party voicemail system perhaps one that
uses the SMDI interface?

Thanks,
Jason 
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[Asterisk-Users] * INSTRUCTIONS FOR NEW MEMBERS OF THE COMMUNITY * Please read

[Olle E. Johansson]

Welcome to the Asterisk users community!

These are exiting times for Asterisk.org. We're getting close to a
1.0 release, working hard to fix all reported bugs in Asterisk.
At the same time, the community is growing and we're having a lot
of interaction, on the IRC and on the mailing lists.
It's great to have you participating in this Open Source project
- building an Open Source PBX. Here are a few things to know and
remember while working with the project.
** The mailing list is growing

The lead programmer of Asterisk, Mark Spencer at Digium, inc, writes:
The Asterisk community is growing at a remarkable pace.  I know there are
thousands of you out there -- in fact there are over eight *thousand*
subscribers to asterisk-users alone, and almost one *thousand* registered
users on the bug tracker.  
This means that everything anyone write to this mailing list, is sent to over
8.000 mailboxes that is already flowing over with messages.
I would like to stress the fact that you have to think before you send a
message to such a big list. Do *not* send out personal replies on the list.
If you offer services to someone, do *not* CC: or reply to the list, it
will annoy more potential customers than get you new customers. If you
send out a message by mistake, you don't have to apologize to all of us,
we understand you're embarassed. We will get more annoyed by your apology
than over your first message.
** Try finding the answer first, then ask the list

The Asterisk Wiki at http://www.voip-info.org project is an important
knowledge base for the project.
Go there to find your answer first, then search the mailing list
archives (Google or http://search.voip-forum.com) and then
go to the IRC channel. The IRC channel is populated with Asterisk gurus
around the clock (literally) and they'll help you move forward.
* IRC info: http://www.asterisk.org/index.php?menu=support#irc
* There's many links to Asterisk web pages on the documentation
  page at http://www.asterisk.org
* The Asterisk FAQ is found on the wiki
  http://www.voip-info.org/wiki-Asterisk+FAQ
* The Asterisk documentation project (which needs your help)
  is at http://www.asteriskdocs.org
Finally, if you don't find the answer elsewhere, try the list.

** Mailing lists
For developers, there is a developer's list. You'll find it
on http://lists.digium.com, which is the address where you manage
your subscription to this list as well.
** Reporting bugs
If you think you have found a bug, report it. We need bug reports.
Read this document http://www.digium.com/bugtracker.html and then
go to the bugtracker http://bugs.digium.com to file a report.
If you are unsure, find a bug marshal on the IRC channel to help
you. They're appointed to support you with how to handle bugs.
Please check the bugtracker thoroughly before posting a new bug;
often, your bug or feature already exists but is simply slowly
making it's way through the system.  Duplicate reports slow things
down for everyone, so please spend a few minutes searching first.
The bug tracker is also a place where you add your contribution
to Asterisk. If you have coded extra functionality, make sure you
give it back to the project so it can be added to the code base.
This is how Asterisk grows, free contributions and consultants
that are paid to add functionality on a case by case basis.
** Remember: It's Open Source, it's voluntary
Asterisk.org is a Open Source project. This means you can't request
help from people, demand new functions or support. However, there
are many individuals and companies out there that are offering
services based on Asterisk, from VoIP service providers to
consultants all over the world.
* See http://www.voip-info.org/wiki-Asterisk%20consultants

Of course, this is also part of Digium's business, so you have
plenty of help if your willing to pay. Digium is to be found at
http://www.digium.com. Service providers and consultants are
listed on the wiki, where you'll find companies all over the globe
that are willing to set up your PBX and get you connected to either
the PSTN or the growing telephony network on the Internet.
Again, welcome to the Asterisk.org Open Source PBX Project!

Meet you on the IRC channel :-)

/oej

-
PS. This message will be sent regularly. If you have any
corrections or additional information that needs to be
included, mail me * off list *. Thank you!
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Re: [Asterisk-Users] PC based Switchboard application files??

[Altus Snyman]

Does it any difference that I'm using a already running tomcat?
Here is the output
Thanks again for your help

DBCP borrowObject failed: Connection refused. Check that the hostname
and port are correct and that the postmaster is accepting TCP/IP
connections.
Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources

INFO: Initializing, config='org.apache.struts.taglib.bean.LocalStrings',
returnNull=true
Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources

INFO: Initializing, config='org.apache.struts.util.LocalStrings',
returnNull=true
Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources

INFO: Initializing, config='org.apache.struts.taglib.html.LocalStrings',
returnNull=true
Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources

INFO: Initializing, config='org.apache.struts.taglib.html.LocalStrings',
returnNull=true
Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources

INFO: Initializing, config='org.apache.struts.taglib.html.LocalStrings',
returnNull=true
Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources

INFO: Initializing, config='org.apache.struts.taglib.html.LocalStrings',
returnNull=true
Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources

INFO: Initializing, config='org.apache.struts.taglib.html.LocalStrings',
returnNull=true
DBCP borrowObject failed: Connection refused. Check that the hostname
and port are correct and that the postmaster is accepting TCP/IP
connections.
Stopping service Tomcat-Standalone



On Thu, 2004-04-22 at 14:30, Pertti Pikkarainen wrote:
> After restarting postgres you need to stop and start
> $CATALINA_HOME/bin/shutdown.sh
> $CATALINA_HOME/bin/startup.sh
> 
> Or something is wrong with the postgre access rights.
> Did you remember  modify
>  /usr/local/pgsql/data/pg_hba.conf
> 
> If a new start doesn't help
> please, send me $CATALINA_HOME/logs/catalina.out
> 
> Best regards Pertti
> 
> 
> Altus Snyman wrote:
> 
> >It comes up with the index page but when you login with admin,admin it
> >says error logging on to database,postgresql is running as postgres user
> >and the db has been added with the txt file,I did the change in tomcat
> >folder.
> >
> >Must postgresql run as postgres user? 
> >Any Ideas
> >
> >Thanks
> >Altus
> >
> >On Thu, 2004-04-22 at 09:31, Altus Snyman wrote:
> >  
> >
> >>Is this error ok? When I insert txt file into the db,Im loged in as
> >>postgres
> >>
> >>CREATE TABLE
> >>INSERT 16984 1
> >>CREATE TABLE
> >>CREATE TABLE
> >>INSERT 17003 1
> >>CREATE TABLE
> >>CREATE TABLE
> >>CREATE TABLE
> >>INSERT 17020 1
> >>INSERT 17021 1
> >>NOTICE:  CREATE TABLE will create implicit sequence 'cdr_acctid_seq' for
> >>SERIAL column 'cdr.acctid'
> >>CREATE TABLE
> >>CREATE TABLE
> >>NOTICE:  ALTER TABLE / ADD PRIMARY KEY will create implicit index
> >>'cdr_pkey' for table 'cdr'
> >>ALTER TABLE
> >> setval
> >>
> >>454
> >>(1 row)
> >> 
> >>ERROR:  Relation "asterikssettings" does not exist
> >>GRANT
> >>GRANT
> >>GRANT
> >>GRANT
> >>GRANT
> >>GRANT
> >>GRANT
> >>GRANT
> >>
> >>
> >>
> >>
> >>
> >>On Thu, 2004-04-22 at 09:16, Pertti Pikkarainen wrote:
> >>
> >>
> >>>I'm so sorry. The file is now there.
> >>>Please download it.
> >>>Thanks !
> >>>
> >>>Best regards Pertti
> >>>
> >>>
> >>>Altus Snyman wrote:
> >>>
> >>>  
> >>>
> Good day all
> I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard)
> But in the pdf howto they speak about a swb.txt and I dont have that
> file
> 
> Thanks
> Altus
> 
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>  
> 
> 
> 
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> >>>  
> >>>
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> >>
> >>
> >
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> >  
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> 
> -
> 
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[Asterisk-Users] Re: Help choosing a UK IAX provider

[Maron Kristófersson]

Well, I've been testing the service from Iceland for the last few days, 
calling both UK based numbers and some numbers in Sweden, Can't say that 
I've had any problems, actually, the phone calls all had excellent 
quality, except one, but at that time I was downloading a linux .iso, 
without QoS enabled on my end.

Best regards,

Maron Kristofersson
Reykjavik
Iceland
Craig Waddington wrote:
Hahahhaaa your right there Tan.

List, don't get me wrong, voiptalk are very good, service, support,
price, I am just having some issues which may be my end.
I was just wanting to try some iax providers out to see what worked best
for us.
Hopefully will get sorted.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 21 April 2004 16:02
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Help choosing a UK IAX provider
In the UK, with the sort of equipment that BT has in its network, you're
lucky to even get adsl going through! ISPs can only provide QoS up to a
certain boundary. After that it is out of their control!
Tan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: 21 April 2004 15:44
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Help choosing a UK IAX provider
On Wed, 2004-04-21 at 09:32, Steve Kennedy wrote:

That's the trouble with running VoIP over contended "public" Internet.


Find someone who can offer you connectivity with QoS and then has QoS 
across their network for VoIP traffic.


LOL!  I've not found any providers that offer QoS on their network other
than a small regional ISP that put QoS on their network when we waved
enough money at them.
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Re: [Asterisk-Users] ALSA help required !

[creslin]

On Thu, Apr 22, 2004 at 05:38:17PM +0600, Adnan Shah wrote:
>  I have just installed the Alsa drivers
>  for my 2.4.18-14 kernel (RH8). I have configured
>  the sound card ok with alsaconf and tested
>  with the aplay , works fine. But when I run
>  asterisk it says..
>  ---
>  [chan_alsa.so] => (ALSA Console Channel Driver)
>  Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init:
>  snd_pcm_open failed: No such device or address
>  Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init:
>  snd_pcm_open failed: No such device or address
>  Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:474 soundcard_init: Problem
>  opening alsa I/O devices
>  
>== No sound card detected -- console channel will be unavailable
>  
>== Turn off ALSA support by adding 'noload=chan_alsa.so' in
>  /etc/asterisk/modules.conf
>  
>  --
> earlier when using the OSS, the playback was choppy not smooth,
> I added some more RAM (total 256 on Intel PIII 600 processor), but the
> problem was still there so I turned to the Alsa drivers.Asterisk doesn't
> seem to work with it what might be wrong, any ideas ?

Ok, this may seems like a silly question, but do you have your sound
card already opened by another application, such as xmms, artsd, etc?

Check for any running applications that may be using the sound card.  If
I remember correctly, if your card doesn't do any kind of stream mixing
on board (like the emu10k1) only one application can have control over
it at any given time.  If that doesn't work, get a hold of me off list
and I'll see if I can fix it.  I'm Cresl1n on #asterisk (though I'm
often away from my IRC console).  Thanks.


Matthew Fredrickson
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Re: [Asterisk-Users] Very basic questions

[C. Maj]

On Wed, 21 Apr 2004, Laurent BURGY waxed:

> Hi,
> I am new in asterisk and i've bought a X100p and a TDM400...

First, you are probably eligible for support from digium
directly if you bought the hardware from them.

> First of all, how can i verify my config files ?

You could try attaching them to the email so people can take
a look.  zapata.conf, zaptel.conf, extensions.conf,
indications.conf are probably the minimum that you need to
alter.

> Secondly, when i'm trying to pass a call to the outside, i ve a Notice 
> about appdial.c (l 554) telling me: unable to create channel of type Zap 
> ...and i don't understand...

That's a problem, but possibly for lots of reasons.  No
zaptel kernel module, no asterisk driver, bad extensions.conf format, etc.

> Finally, when i plug my analog phones in RJ45 of my TDM400, there is no 
> tonality ( i'm not sure that it is the right word in english , but i 
> can't hear any tut-tut or any noise...) ...

Did you compile the kernel modules in the zaptel source
directory for that card ?  What does lsmod show you ?

> Maybe, it's obvious but i can't succeed...

No, it's still quite difficult, especially if english is not
your first language.  I am completely unaware of any
non-english documentation.

Good luck,
--Chris


-- 
Chris Maj, Rochester
cmaj_at_freedomcorpse_dot_com
Pronunciation Guide: Maj == May
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[Asterisk-Users] How to get call back when transfer fails

[Andrew Kohlsmith]

I searched the 22490 messages I have in my own personal asterisk-users archive 
and have not found the answer, and it also does not appear on the wiki.

I have a SIP phone and a regular phone on a TDM400P FXS interface.  Extensions 
are 100 and 101, respectively.

On the SIP phone I can hit #, get the "Transfer" prompt and enter an extension 
I want to transfer to.  No problem.  I can do the same thing on the FXS port.

My question is does anyone have a dialplan that will bring the call I 
transfered back to me if the transfer fails (i.e. busy extension or an 
extesion that does not answer)?  I see many examples of going to voicemail 
but I have no idea how to get the call back to me.

The general idea: 
- I call someone or I receive a call.  
- I want to transfer said call, so I hit # and enter the extension
- if the extension answers, it's all good.
- if the extension is busy or does not answer, give me the call again.  

The dialplan is pretty straightforward:
exten = 100,1,Dial(SIP/224,10,t)
exten = 100,2,Congestion
exten = 101,1,Dial(Zap/1,10,t)
exten = 101,2,Congestion

I do not wish to have the transferred call go to voicemail.  I want the call 
back.  I didn't see any kind of application which would connect me back to 
the call, and simply dropping off the dialplan just hangs up.  

So far the only thing I can think of is to park the call but that is 
suboptimal. 

Anyone?

-A.
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Re: [Asterisk-Users] A few questions

[C. Maj]

On Wed, 21 Apr 2004, Ben Merrills waxed:

> Hi,
>  
> I have a couple of questions about MeetMe and call queues. I'm still
> pretty new to Asterisk, but already having to write a Service Center
> call manager for it (which I might add, our director has agreed to make
> open source!).

That's great news.

> MeetMe:
> 
> How can I get MeetMe (does it even do this) to ask the user to speak
> their name first, and play that as the new member announcement. It seems
> like a common feature in most hardware PBX systems we've used that
> support Call Conferences.
>  
> Has anyone found a way of doing this? Is there an alternative to MeetMe
> that would support this feature (that's as good if not better?).

I don't think this is currently supported, could be wrong,
tho.  Would take some modification to app_meetme.c, or else
just have people say there name when it beeps them in. :)

Sort of the flip side, but maybe it would be more helpful to
have the person entering the conference hear the name of
everyone already in it.  That could be done via Record and
Playback apps, before executing MeetMe.  Every time someone
enters, have Record take their name.  Then, run Playback for
each of the Recorded files.

> Queues:
>  
> I'm running the 1.0 stable from the cvs server, and I've added the queue
> status announcement directives to the queues.conf - yet asterisk gives
> me the following errors:
>  
> Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales':
> monitor-format at line 9 of queue.conf

I think this only works in development, not stable, CVS. :(

--Chris


-- 
Chris Maj, Rochester
cmaj_at_freedomcorpse_dot_com
Pronunciation Guide: Maj == May
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RE: [Asterisk-Users] Ser and Asterisk together

[Dawid Mielnik]

Barry,

below my ser.cfg

Dave

#
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
#
# simple quick-start config script
#

# --- global configuration parameters 

#debug=3 # debug level (cmd line: -dd)
#fork=yes
#log_stderror=no# (cmd line: -E)

/* Uncomment these lines to enter debugging mode
debug=9
fork=no
log_stderror=yes
*/

check_via=no# (cmd. line: -v)
dns=no   # (cmd. line: -r)
rev_dns=no  # (cmd. line: -R)
#port=5060
#children=4
fifo="/tmp/ser_fifo"

# -- module loading --

# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/ser/modules/mysql.so"

loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"

# accounting
loadmodule "/usr/local/lib/ser/modules/acc.so"

# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
#loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/auth_radius.so"

# Nathelper
loadmodule "/usr/local/lib/ser/modules/nathelper.so"

# - setting module-specific parameters ---

# -- usrloc params --

#modparam("usrloc", "db_mode",   0)

# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)

# -- auth params --
# Uncomment if you are using auth module
#
#modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
#modparam("auth_db", "password_column", "password")

# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)

# -- nathelper params --
modparam("registrar", "nat_flag", 6)
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)

# -- acc params --
modparam("acc", "radius_config",
"/usr/local/etc/radiusclient/radiusclient.conf")
modparam("acc", "log_level", 1)
modparam("acc", "radius_flag", 1)
modparam("acc", "report_ack", 0)

# -  request routing logic ---

# main routing logic

route{

# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
if (nat_uac_test("3")) {

if (method == "REGISTER" || !search("^Record-Route:")) {
log("LOG: Kolejny NATowiec...\n");

fix_nated_contact();
if (method == "INVITE") {
fix_nated_sdp("1");
};
force_rport();
setflag(6);
};
};

# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
record_route();
# loose-route processing
if (loose_route()) {
route(1);
#t_relay();
break;
};

# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
#   if (uri=="yyy.yyy.yyy.yyy") {

if (method=="REGISTER") {

# Uncomment this if you want to use digest authentication
#if (!www_authorize("xxx.xxx.xxx.xxx", "subscriber")) {
if (!radius_www_authorize("xxx.xxx.xxx.xxx")) {
www_challenge("xxx.xxx.xxx.xxx", "0");
break;
};

save("location");
break;
};
#setflag(1);
# native SIP destinations are handled using our USRLOC DB
# going to our sip users ?
if (uri=~"sip:32679*" || uri=~"sip:58279*") {

if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
# going to pstn
} else {
#   };
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP

# coming from fax ?
if (search("(f|From): [EMAIL PROTECTED]")) { # fax numbers
# forward to fax gw
  

[Asterisk-Users] inbound calls better quality than outbound calls on X100P

[Chris Stenton]

I have a strange problem in that when I receive a call through the X100P
which is forwarded to my budgetone 100 then the voice quality is perfect
both directions. However, if I make a call out from the budgetone to the
same caller via the X100P the sound level is a lot lower and the quality a
lot poorer. I've had to set the rx tx gain to 1.5 or I can hardly hear at
all.

Any ideas what is wrong, I'm using the latest zaptel and asterisk from the
cvs head as of today.


Chris


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[Asterisk-Users] no signal, mgcp

[Arkadiusz Murzyn]

Hi,

I'm trying to connect asterisk on FreeBSD and mtas.
I have registered endpoints, but after I pick up the receiver I hear
nothing, there is no signal.
There is a message:

-- MGCP mgcp_new( endpoint ) created in state : Down

I'm concerned about messages like
chan_mgcp.c:372 __mgcp_xmit: mgcp_xmit returned -1: Address family not
supported by protocol family

Can asterisk work without soundcard.?
Can anyone help ?

Thanks

Arkdiusz Murzyn
[EMAIL PROTECTED]

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[Asterisk-Users] Re: VoiceTronix Openline 4

[Chris Tooley]

Ron,

VoiceTronix worked with me last week and backported the new chan_vpb
that's in the current HEAD CVS tree to the 1.0 stable branch.  So if you
are using 0.7.2 or 0.9.0 you can go to the VoiceTronix website now (note
that this is very new) and download an Asterisk package.  When you
download it there is an new chan_vpb.c that needs to be put in your
asterisk source code tree.  Also there is a vpb.conf that should go
in /etc/asterisk/ and it has examples for all of the different
VoiceTronix FXO hardware available.  The developer at VoiceTronix, Ben,
was very nice and helped me out a lot.  I now have all 4 of my FXO ports
working, with Caller ID.

There is a modification to be made to the chan_vpb.c if you live in the
US.  There are ring tones for the Australia and ring tones for the USA
and at the moment you have to comment out the Australian ring tones and
uncomment the USA ones.  This should help get you started.  I'm going to
post this to the asterisk mailing list as well so new information can be
provided to everyone.

Chris Tooley

On Wed, 2004-04-21 at 22:47 -0500, [EMAIL PROTECTED] wrote:
> Chris,
> 
> In your posting to the Asterisk Wiki, you mentioned using the VoiceTronix
> OpenLine4 fxo card.  I am just starting out setting up my first Asterisk
> PBX using a Digium TDM40B (4 port fxs) card and a VoiceTronix OpenLine4
> fxo card. I have my fxs card working and my Sayson phones ringing each
> other.  I have not been able to find much info on the conf file settings
> to
> get the VoiceTronix card working.  I have done all of the /dev/vpb0 stuff
> and my RedHat 9 system sees the card.  I just don't know what other files
> besides the vpb.conf to edit and the required information for this card. 
> Any help that you could give would be greatly appreciated.
> 
> 
> Sincerely,
> 
> 
> Ron McDaniel
> Southern Computer Services,Inc.
> [EMAIL PROTECTED]
> (251) 294-1202 cell

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RE: [Asterisk-Users] inbound calls better quality than outbound calls on X100P

[David J Carter]

I have my RX at 4.0 ant TX at 8.0,
I get slight echo for the first 5-6 seconds then all OK.


Regards

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Stenton
Sent: 22 April 2004 17:07
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] inbound calls better quality than outbound
calls on X100P


I have a strange problem in that when I receive a call through the X100P
which is forwarded to my budgetone 100 then the voice quality is perfect
both directions. However, if I make a call out from the budgetone to the
same caller via the X100P the sound level is a lot lower and the quality a
lot poorer. I've had to set the rx tx gain to 1.5 or I can hardly hear at
all.

Any ideas what is wrong, I'm using the latest zaptel and asterisk from the
cvs head as of today.


Chris


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[Asterisk-Users] Sound Problem

[Dan]

I have some incoming did`s from a sip provider..  The problem i`m having is when
no one is talking it sounds like someone is faxing or sending Data.

I can`t figure out for the life of me what the problem is  I`m not receiving any
errors from AsteriskCould someone please direct me to a way to fix this


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Re: [Asterisk-Users] How to get call back when transfer fails

[Steven Critchfield]

On Thu, 2004-04-22 at 09:58, Andrew Kohlsmith wrote:
> I searched the 22490 messages I have in my own personal asterisk-users archive 
> and have not found the answer, and it also does not appear on the wiki.
> 
> I have a SIP phone and a regular phone on a TDM400P FXS interface.  Extensions 
> are 100 and 101, respectively.
> 
> On the SIP phone I can hit #, get the "Transfer" prompt and enter an extension 
> I want to transfer to.  No problem.  I can do the same thing on the FXS port.
> 
> My question is does anyone have a dialplan that will bring the call I 
> transfered back to me if the transfer fails (i.e. busy extension or an 
> extesion that does not answer)?  I see many examples of going to voicemail 
> but I have no idea how to get the call back to me.
> 
> The general idea: 
> - I call someone or I receive a call.  
> - I want to transfer said call, so I hit # and enter the extension
> - if the extension answers, it's all good.
> - if the extension is busy or does not answer, give me the call again.  

Well you could do a supervised transfer, or 3 way call. Basically, you
place the one leg of the call on hold, dial the extension you are to
transfer to, then if you successfully connect, bring the call to three
way, and then excuse yourself.

Or you could create a kind of Macro for transfers where it stores the
originating part of the transfer, and upon failed connect, does a return
dial. This would bypass normal call routing where a direct call would go
to voicemail if it misses a person at the end. 
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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[Asterisk-Users] Channel Bank - New * install

[Jon Brandon]

I am looking at installing * as the PBX in a new office and have a few
questions that I hope someone can help me with. The installation will be small
at first with about 8 internal extensions, but will grow to 24 within a year or
so.

 

First is there any benefit to using VoIP
phones instead of installing a channel bank and analog business phones? 

 

If not, what are some good analog business phones that people have used?


How about channel banks, can I get some suggestions?

 

Thanks

 

 

-

-Jon Brandon

VP of Technology

Monsoon

Add me to your contacts: http://www.monsoonretail.com/vcards/JonBrandon.vcf

 

[Asterisk-Users] Cisco phones

[Nick Knight]

Hello all,

 

I haven’t used cisco phones as yet – do they
work with asterisk, are they good which models are the best?

 

I am after a starting point!

 

Thanks

 

Nick

Re: [Asterisk-Users] PC based Switchboard application files??

[Pertti Pikkarainen]

in /usr/local/pgsql/data/pg_hba.conf   this kind of  information is given
# To allow TCP/IP access, even from localhost, the postmaster must also be
# started with the -i option or the option TCPIP_SOCKET must be set in
# /etc/postgresql/postgresql.conf.
In the end of  my/usr/local/pgsql/data/postgresql.conf

# TCP/IP access is allowed by default, but the default access given in
# pg_hba.conf will permit it only from localhost, not other machines.
tcpip_socket = 1
I have postgres and tomcat in the same computer.

--Pertti



Altus Snyman wrote:

Does it any difference that I'm using a already running tomcat?
Here is the output
Thanks again for your help
DBCP borrowObject failed: Connection refused. Check that the hostname
and port are correct and that the postmaster is accepting TCP/IP
connections.
Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources

INFO: Initializing, config='org.apache.struts.taglib.bean.LocalStrings',
returnNull=true
Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources

INFO: Initializing, config='org.apache.struts.util.LocalStrings',
returnNull=true
Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources

INFO: Initializing, config='org.apache.struts.taglib.html.LocalStrings',
returnNull=true
Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources

INFO: Initializing, config='org.apache.struts.taglib.html.LocalStrings',
returnNull=true
Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources

INFO: Initializing, config='org.apache.struts.taglib.html.LocalStrings',
returnNull=true
Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources

INFO: Initializing, config='org.apache.struts.taglib.html.LocalStrings',
returnNull=true
Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources

INFO: Initializing, config='org.apache.struts.taglib.html.LocalStrings',
returnNull=true
DBCP borrowObject failed: Connection refused. Check that the hostname
and port are correct and that the postmaster is accepting TCP/IP
connections.
Stopping service Tomcat-Standalone


On Thu, 2004-04-22 at 14:30, Pertti Pikkarainen wrote:
 

After restarting postgres you need to stop and start
$CATALINA_HOME/bin/shutdown.sh
$CATALINA_HOME/bin/startup.sh
Or something is wrong with the postgre access rights.
Did you remember  modify
/usr/local/pgsql/data/pg_hba.conf
If a new start doesn't help
please, send me $CATALINA_HOME/logs/catalina.out
Best regards Pertti

Altus Snyman wrote:

   

It comes up with the index page but when you login with admin,admin it
says error logging on to database,postgresql is running as postgres user
and the db has been added with the txt file,I did the change in tomcat
folder.
Must postgresql run as postgres user? 
Any Ideas

Thanks
Altus
On Thu, 2004-04-22 at 09:31, Altus Snyman wrote:

 

Is this error ok? When I insert txt file into the db,Im loged in as
postgres
CREATE TABLE
INSERT 16984 1
CREATE TABLE
CREATE TABLE
INSERT 17003 1
CREATE TABLE
CREATE TABLE
CREATE TABLE
INSERT 17020 1
INSERT 17021 1
NOTICE:  CREATE TABLE will create implicit sequence 'cdr_acctid_seq' for
SERIAL column 'cdr.acctid'
CREATE TABLE
CREATE TABLE
NOTICE:  ALTER TABLE / ADD PRIMARY KEY will create implicit index
'cdr_pkey' for table 'cdr'
ALTER TABLE
setval

  454
(1 row)
ERROR:  Relation "asterikssettings" does not exist
GRANT
GRANT
GRANT
GRANT
GRANT
GRANT
GRANT
GRANT




On Thu, 2004-04-22 at 09:16, Pertti Pikkarainen wrote:
  

   

I'm so sorry. The file is now there.
Please download it.
Thanks !
Best regards Pertti

Altus Snyman wrote:



 

Good day all
I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard)
But in the pdf howto they speak about a swb.txt and I dont have that
file
Thanks
Altus
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-

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Re: [Asterisk-Users] Cisco phones

[John Fraizer]

Nick Knight wrote:

I haven’t used cisco phones as yet – do they work with asterisk, are 
they good which models are the best?
I use a 7960 with Asterisk and absolutely love it.  It blows the snot 
out of the Nortel phone I used to use.

John
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Re: [Asterisk-Users] How to get call back when transfer fails

[Andrew Kohlsmith]

> Well you could do a supervised transfer, or 3 way call. Basically, you
> place the one leg of the call on hold, dial the extension you are to
> transfer to, then if you successfully connect, bring the call to three
> way, and then excuse yourself.

Ok, I understand that one.  Overkill but it would work.  I know how to do this 
on a Zap interface (hookflashes) , but how does one do it on a SIP or IAX 
interface?

> Or you could create a kind of Macro for transfers where it stores the
> originating part of the transfer, and upon failed connect, does a return
> dial. This would bypass normal call routing where a direct call would go
> to voicemail if it misses a person at the end.

I thought of that too, but I don't think it'll work:

exten 101,1,Setvar(myexten)
exten 101,2,Dial(Zap/1,10,t)
exten 101,103,getvar(myexten)
exten 101,104,Dial(myexten)

If extension 101 is busy it will immediately try to dial me, but I'm still on 
the phone since the transfer didn't complete...

Or did I misunderstand "return-dial" ?

-A.
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Re: [Asterisk-Users] Channel Bank - New * install

[Steven Critchfield]

On Thu, 2004-04-22 at 12:50, Jon Brandon wrote:
> I am looking at installing * as the PBX in a new office and have a few
> questions that I hope someone can help me with. The installation will
> be small at first with about 8 internal extensions, but will grow to
> 24 within a year or so.
> 
>  
> First is there any benefit to using VoIP phones instead of installing
> a channel bank and analog business phones? 
> 
>  
> If not, what are some good analog business phones that people have
> used? 
> 
> How about channel banks, can I get some suggestions?

Dude, drop the HTML, and remember why google exists.

VoIP phones have the benefit of linear growth cost. A phone costs $X,
and for the most part will cost $X no matter how many lines you roll
out. So a new extension is just $X increase, and your system is just $X
x N extensions to deploy. Also VoIP can be deployed pretty much
anywhere.

Analog has the benefit of cheaper phones, and what I consider a better
service record. There isn't really a problem of what is and isn't
supported, or supported to what extent. Draw backs are you can either
deploy in multiples of 4 with the TDM400 or go T1 and deploy in
multiples of 24. Either way, it makes the first step beyond the current
block slightly expensive, but then the increment is a small amount till
you fully deploy your current block.

24 Budgetones would be ~$1800(assuming you find them for $75 each).
24 analog AT&T phones, $1720(assuming a channel bank from ebay at $500
and $30 phones).

So you can see where the 25th phone goes back to the VoIP phones as the
25th phone on analog will run you another $1030 for the T1 port and
channel bank.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Cisco phones

[Gregory Junker]

Well, before someone jumps down your throat for asking a question that
gets asked multiple times a week... ;)

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



On a related note...whoever maintains the page at 

http://www.asteriskpbx.com/index.php?menu=support

may want to think about putting the link to the Wiki in a much more
prominent place, and make it clear that new users should look there
first, since they often go through the hassle of signing up for the
mailing list only to post a question for which they could have found the
answer in much less time.

Greg

On Thu, 2004-04-22 at 18:55 +0100, Nick Knight wrote:
> Hello all,
> 
>  
> 
> I havenât used cisco phones as yet â do they work with asterisk, are
> they good which models are the best?
> 
>  
> 
> I am after a starting point!
> 
>  
> 
> Thanks
> 
>  
> 
> Nick
> 
> 

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[Asterisk-Users] Music on Music on Hold Distorted

[David Liu]

Hi there,

I just tried today's CVS: 4/23/2004 version and found a strange loise
with music on hold.  Basically, when on hold you hear very distorted
music as if it was very loud.  This is the exact same problem described
last year at:

http://lists.digium.com/pipermail/asterisk-users/2003-April/009735.html
http://lists.digium.com/pipermail/asterisk-users/2003-May/011688.html

No answers on the above.  Perhaps it is something very trivial?  

mpg123 version is: 0.59s-r2  on Gentoo with kernel 2.6

Any ideas would be great!

David
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Re: [Asterisk-Users] Cisco phones

[Scott Laird]

On Apr 22, 2004, at 11:09 AM, John Fraizer wrote:

Nick Knight wrote:

I haven’t used cisco phones as yet – do they work with asterisk, are 
they good which models are the best?
I use a 7960 with Asterisk and absolutely love it.  It blows the snot 
out of the Nortel phone I used to use.
I have a 7940, which is basically a 7960 with 4 fewer buttons, and it 
works perfectly.  The 7905/7912s are supposed to work, too, but I 
haven't used one.

Scott

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[Asterisk-Users] D/41 ESC dialogic ISA CARD

[Alejandro Acosta]

Hello,
  I just wanted to know if any of you has successfully (or know about) 
installed the Dialogic 4xFXs ISA CARD D/41 ESC? Does it work with *?, if 
so, how is the driver called?

Thanks a lot for your comments.

Alejandro Acosta,-
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Re: [Asterisk-Users] How to get call back when transfer fails

[Steven Critchfield]

On Thu, 2004-04-22 at 13:13, Andrew Kohlsmith wrote:
> > Well you could do a supervised transfer, or 3 way call. Basically, you
> > place the one leg of the call on hold, dial the extension you are to
> > transfer to, then if you successfully connect, bring the call to three
> > way, and then excuse yourself.
> 
> Ok, I understand that one.  Overkill but it would work.  I know how to do this 
> on a Zap interface (hookflashes) , but how does one do it on a SIP or IAX 
> interface?

I don't know about SIP, and I think IAX should work similarly to ZAP.

> > Or you could create a kind of Macro for transfers where it stores the
> > originating part of the transfer, and upon failed connect, does a return
> > dial. This would bypass normal call routing where a direct call would go
> > to voicemail if it misses a person at the end.
> 
> I thought of that too, but I don't think it'll work:
> 
> exten 101,1,Setvar(myexten)
> exten 101,2,Dial(Zap/1,10,t)
> exten 101,103,getvar(myexten)
> exten 101,104,Dial(myexten)
> 
> If extension 101 is busy it will immediately try to dial me, but I'm still on 
> the phone since the transfer didn't complete...

As I think this example over more, I'm not sure it will work. 2 problems
plague it. Specifically, when does it receive the information about the
call. IF you dial out, should part of the dial out statements store your
local interface? On a dial in, you would have to store it also. Okay,
these two don't sound too difficult to implement.

Of course, I think this will dovetail into the commentary about passing
variables from one call leg to another. On an outbound call, I think you
would be creating a new call leg to transfer. 

Anyways, to fix the problem of being on the phone when the call comes
back, just put a wait in there that is reasonable enough for you to
hangup, or place a prompt in there that indicates the transfer is going
back to the originator. 

Of course you run into the problem of what happens if you receive a call
in the meantime.  Your line goes back to busy.


OF course you could just use call parking and it has a feature for
returning the call to you if it isn't picked up within a certain amount
of time. It also allows you to either go face to face brief the other
party, or just call them. 

-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] How to get call back when transfer fails

[Jeremy Hall]

>I thought of that too, but I don't think it'll work:
>
>exten 101,1,Setvar(myexten)
>exten 101,2,Dial(Zap/1,10,t)
>exten 101,103,getvar(myexten)
>exten 101,104,Dial(myexten)
>
>If extension 101 is busy it will immediately try to dial me, but I'm
still >on the phone since the transfer didn't complete...
>
>Or did I misunderstand "return-dial" ?
>
>-A.

Try something like this.  Record a message (if one doesn't already
exist) saying "Please hold while your call is being transferred."

Then add it to your dialplan as such:

exten 101,1,Setvar(myexten)
exten 101,2,Dial(Zap/1,10,t)
exten 101,103,getvar(myexten)
exten 101,104,playback(hold-transfer)
exten 101,105,Wait(3)
exten 101,106,Dial(myexten)

The hold message would take about 3 seconds, then with the additional 3
second wait, that would give you 6 seconds to hang up your phone, which
if you are transferring a call is very reasonable.  If it goes through,
then great, you are done.  If not, your phone will ring again in a few
seconds and you get the call back.

Hope that helps!

Jeremy

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Re: [Asterisk-Users] Cisco phones

[Alric]

> Nick Knight wrote:
>
> > I haven’t used cisco phones as yet – do they work with asterisk, are
> > they good which models are the best?
>
> I use a 7960 with Asterisk and absolutely love it.  It blows the snot
> out of the Nortel phone I used to use.
>
> John

I know the 7905, 7912, 7940, and 7960 all work with Asterisk using the SIP
protocol.

I'm currently using a 7905G at home, and it works great.

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RE: [Asterisk-Users] D/41 ESC dialogic ISA CARD

[Storer, Darren]

Hi Alejandro,

from memory only the newer JCT series of Dialogic cards are supported by
special drivers for Asterisk (obtained under license from Digium direct).
Please check if your D/41 card has a JCT suffix.

HTH

Darren
--
Comgate UK
Telco>Internetmailto:[EMAIL PROTECTED] Behalf Of Alejandro
Acosta
Sent: 22 April 2004 19:29
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] D/41 ESC dialogic ISA CARD


Hello,
   I just wanted to know if any of you has successfully (or know about)
installed the Dialogic 4xFXs ISA CARD D/41 ESC? Does it work with *?, if
so, how is the driver called?

Thanks a lot for your comments.

Alejandro Acosta,-
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[Asterisk-Users] Russia calling

[Olle E. Johansson]

http://bugs.digium.com/bug_view_page.php?bug_id=0001471

Indications for Russia

Please check if this works for you - if you know how it should sound in
Russia and other countries that have the same phone system. If it works
or if you object, please confirm or add your comments to the bug tracker.
Thank you!

/O
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[Asterisk-Users] Looking for IAX/SIP termination in Israel

[Mark Johnston]

I'm looking for IAX (preferred) or SIP termination in Israel, with DIDs.  Can 
anyone give me a price, or better yet, point me to a resource for 
buying/selling minutes?

Thanks,
Mark
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[Asterisk-Users] Echo Cancellation Feature

[Brent Franks]

I do feel the echo cancellation does need some work.

Currently, other than listening to users, there is no way to benchmark or
trouble shoot echo problems.

We find that 2 to 3 out of every 20 calls will experience echo.  While
echo is a problem that naturally occurs from SIP - PSTN and vice versa, I
am still baffled by the fact that the cancellation works randomly.

When doing a zap show channel X, it will also report that the cancellation
is still on.  We experience the most echo with a T100P --> Adtran TA 750
FXO modules.  While I understand these do not have impedence matching, I
wonder to myself why echo cancellation works sometimes, and others not.

Looking at Network util, processor util, and memory utilization, they do
not provide any clear indication as to why /when it occurs.

Is there anything more that can be done to debug echo cancellation, and
further are other users experiencing this random echo.  I know it was
discussed before, but the support folks at digium aren't able to offer
anymore help.

Asterisk is truly a great piece of open soruce software, and I commend the
authors/contributors for their hardware and attention to detail.

We're just a few items a way from making this thing absolutely kill the
traditional PBX market :).

Long live Asterisk,

Brent

On Thu, 22 Apr 2004, David J Carter wrote:

> I have my RX at 4.0 ant TX at 8.0,
> I get slight echo for the first 5-6 seconds then all OK.
> 
> 
> Regards
> 
> Dave
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Chris Stenton
> Sent: 22 April 2004 17:07
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] inbound calls better quality than outbound
> calls on X100P
> 
> 
> I have a strange problem in that when I receive a call through the X100P
> which is forwarded to my budgetone 100 then the voice quality is perfect
> both directions. However, if I make a call out from the budgetone to the
> same caller via the X100P the sound level is a lot lower and the quality a
> lot poorer. I've had to set the rx tx gain to 1.5 or I can hardly hear at
> all.
> 
> Any ideas what is wrong, I'm using the latest zaptel and asterisk from the
> cvs head as of today.
> 
> 
> Chris
> 
> 
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[Asterisk-Users] Prepaid

[Julio]

Hello,

somebody has an example of the base with all the loaded data?my
prepaid this running but does not finish the calls...

-- Starting simple switch on 'Zap/2-1'
-- Executing Prepaid("Zap/2-1", "") in new stack
-- Playing 'prepaid-enter-card-num' (language 'en')
-- Playing 'prepaid-you-have' (language 'en')
-- Playing 'digits/70' (language 'en')
-- Playing 'prepaid-dollars' (language 'en')
-- Playing 'prepaid-enter-dest' (language 'en')
-- Playing 'prepaid-dest-blocked' (language 'en')

Thks

Julio


- Original Message -
From: "Alejandro Acosta" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, April 22, 2004 12:29 PM
Subject: [Asterisk-Users] D/41 ESC dialogic ISA CARD


> Hello,
>I just wanted to know if any of you has successfully (or know about)
> installed the Dialogic 4xFXs ISA CARD D/41 ESC? Does it work with *?, if
> so, how is the driver called?
>
> Thanks a lot for your comments.
>
> Alejandro Acosta,-
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[Asterisk-Users] Interfacing with an existing phone system

[Joel Duffield]

We want to use asterisk to extend our current phone system. It is a
regular plain old system. Has anyone done this before? We would be
adding about 4 SIP (probably Cisco) phones to use with asterisk. What
kind of card will I need to use for this, FXS or FXO.

Also does anyone have any ideas what the best way to go about this is,
should I just forward existing lines to specific phones (just to save on
running new telephone cabling) or would there be any simple ways to make
a small menu and just put one more layer before they get through?

Thanks
 
Joel Duffield
Near North Business Machines
www.NearNorthBusiness.com
 

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[Asterisk-Users] Modems compatible with NTL caller id

[Alex Brett]

I'm looking at using a modem to provide caller ID info on my NTL line 
following the steps in the article posted by Darren Poulson:

http://www.22balmoralroad.net/modules.php?name=Sections&op=viewarticle&artid=1

I was wondering if anybody had experience of using any modems (can be 
pci/isa/external) with an ntl line (in an ex. Cable & Wireless area if 
it makes a difference) and would know which modems will pick up the 
caller id.  Failing anybody having direct experience, does anybody know 
what caller id standard NTL use so I can find a modem that supports that 
standard or even maybe get Asterisk working with it directly (I haven't 
actually looked yet to see if Caller ID is enabled on the line I'm using 
so it could be as simple as just getting ntl to switch it on).

Thanks,
Alex Brett
[EMAIL PROTECTED]
+44 (0)870 744 2170
http://www.loho.co.uk/
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Re: [Asterisk-Users] Asterisk with UUI support ?

[John Todd]

At 11:54 AM +0200 on 4/22/04, [EMAIL PROTECTED] wrote:
Hi there,

Is it possible to manage UUI with asterisk and ISDN (T0 Fritz card).
Basically, is it possible to send User to User Information using the
D-channel, while making a call?


No, though others have mentioned it and even developed patches for 
Asterisk for displaying the information, but nobody has created a 
complete patch to pass information into and out of the UUI fields in 
PRI D channel.

If you feel like adding something, feel free to write the patches and 
put into the bugs.digium.com interface after you've disclaimed the 
code.

For reference:

http://lists.digium.com/pipermail/asterisk-dev/2003-September/001751.html

JT
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Re: [Asterisk-Users] Echo Cancellation Feature

[Ryan Thrash]

I would also offer feedback that we too have random calls with echo on 
our end, that can't be traced to a reproducible event. It's very odd 
and can be frustrating, as it's a big distraction for those that don't 
know better. Like a bad cell phone connection when you hear yourself 
talk. For us, this happens in a pure SIP environment on a network 
switch dedicated to Asterisk, a T1 PRI and 18 SIP handsets.

HTH,
Ryan
On Apr 22, 2004, at 1:37 PM, Brent Franks wrote:

I do feel the echo cancellation does need some work.

Currently, other than listening to users, there is no way to benchmark 
or
trouble shoot echo problems.

We find that 2 to 3 out of every 20 calls will experience echo.  While
echo is a problem that naturally occurs from SIP - PSTN and vice 
versa, I
am still baffled by the fact that the cancellation works randomly.


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Re: [Asterisk-Users] weird IAX2 things going on

[Mark Phillips]

I'm using the guest account that comes in the default setup files and I
don't register any of the machines.


> On Wed, 2004-04-21 at 17:31, Mark Phillips wrote:
>> Hi all,
>>
>> I have 3 * boxes all running the same OS and software version. Machine A
>> has an X100P card, machines B and C do not. They all have the same
>> dialplan.
>>
>> I can dial from machine A to either of the other 2 with no problem. I
>> can
>> dial from either of the other 2 to machine A with no problem. I cannot
>> dial from B to C or vice versa.
>>
>> What's really wierd is that I can dial from machine B through machine A
>> to
>> machine C. The IAX2 session then drops machine A from the picture and
>> continues directly between B and C. The same happens in reverse.
>>
>> Why can I not dial directly from B to C? Do B and C require X100P cards
>> before IAX2 will work correctly? I don't think this is the case because
>> the call can be passed to them after it has been setup via A.
>>
>> Its really bugging me. Any ideas?
>
> Is B and C registering to each other like they are to A?
> --
> Steven Critchfield  <[EMAIL PROTECTED]>
>
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G7LTT/KC2ENI
Mark Phillips
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Re: [Asterisk-Users] Music on Music on Hold Distorted

[Eric Wieling]

I saw a similar report on #asterisk the other day.  When the guy
switched to Version 0.59r (1999/Jun/15). Written and copyrights by
Michael Hipp everything was fine.

On Thu, 2004-04-22 at 13:17, David Liu wrote:
> Hi there,
> 
> I just tried today's CVS: 4/23/2004 version and found a strange loise
> with music on hold.  Basically, when on hold you hear very distorted
> music as if it was very loud.  This is the exact same problem described
> last year at:
> 
> http://lists.digium.com/pipermail/asterisk-users/2003-April/009735.html
> http://lists.digium.com/pipermail/asterisk-users/2003-May/011688.html
> 
> No answers on the above.  Perhaps it is something very trivial?  
> 
> mpg123 version is: 0.59s-r2  on Gentoo with kernel 2.6
> 
> Any ideas would be great!
> 
> David
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-- 
  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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[Asterisk-Users] Trouble Compiling "zaptel"

[Sam Bacsa]

So I went to go compile the Zaptel library from the HEAD CVS and I get
some really really odd errors which don't make any sense.  I've attached
the console output ... any idea why this is going on and how to fix
this?

Thanks,
Sam Bacsa


 SNIP 

[EMAIL PROTECTED] zaptel]# make
gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__
-DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I.
-Wstrict-prototypes -fomit-frame-pointer
-I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include
-I/usr/src/linux/include/net -DMODVERSIONS -include
/usr/src/linux-2.4/include/linux/modversions.h  -DSTANDALONE_ZAPATA -c
tor2.c
In file included from tor2.c:30:
/usr/src/linux-2.4/include/linux/kernel.h:60: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:60: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:60: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:61: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:61: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:62: `panic_R_ver_str' declared
as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:62: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:68: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:68: `simple_strtoul_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:68: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:69: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:69: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:69: `simple_strtol_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:69: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:70: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:70: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:70:
`simple_strtoull_R_ver_str' declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:70: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:72: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:72: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:73: `sprintf_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:73: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:74: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:74: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:74: `vsprintf_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:74: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:75: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:75: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:76: `snprintf_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:76: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:77: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:77: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:77: `vsnprintf_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:77: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:79: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:79: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:80: `sscanf_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:80: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:81: `vsscanf_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:81: warning: parameter names
(without types) in function declaration
/usr/src/linux-2.4/include/linux/kernel.h:83: `get_option_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:83: warning: parameter names
(without types) in function declaration
/usr/src/linux-2.4/include/linux/kernel.h:84: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:84: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:84: `get_options_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:84: warning: function
declaration isn't a prototype
/usr/src/linux

Re: [Asterisk-Users] Asterisk & RedHat Enterprise

[Raymond McKay]

> Are their any issues with Asterisk and Redhat Enterprise? I have see one
or two posts with issues concerning compiling zaptel ?
> drivers but that is about it. Just looking for some consensus to if any
problems exist with it.

I have one production system running with the T1 card with no issues.  Its
been running for about 3 months now and I haven't had any issues to speak
of.


Raymond McKay
President
RAYNET Technologies LLC
(860) 833-9720
http://www.raynettech.com

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[Asterisk-Users] Avoiding IAX destroy deadlock

[kc2eni]

On one of my 3 * servers I get this after 2 or 3 IAX2 calls 

Apr 22 15:54:39 NOTICE[1150495040]: chan_iax2.c:1271 iax2_destroy: Avoiding IAX
destroy deadlock

And as if that wasn't enough I get a never ending stream of this error flying
off the top of the screen. At which point I can no longer make any calls into
or out of the box. Any commands issued at the CLI prompt are ignored so I have
to do a service asterisk restart now to get it back into service again.

Ideas?
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RE: [Asterisk-Users] Trouble Compiling "zaptel"

[Jeremy Hall]

I have the same thing on my test system here at work.  I haven't had the
time to look into it but the last time I saw a similar error it was
because the kernel source didn't match the installed kernel.  I wonder
if Up2Date snuck in a kernel update even though I have them ignored?
But then I didn't think it would download any updates without asking.
Odd.  Let me know if you find something out, I won't have a chance to
look into it until tomorrow.

Jeremy

-Original Message-
From: Sam Bacsa [mailto:[EMAIL PROTECTED] 
Sent: Thursday, April 22, 2004 2:23 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Trouble Compiling "zaptel"

So I went to go compile the Zaptel library from the HEAD CVS and I get
some really really odd errors which don't make any sense.  I've attached
the console output ... any idea why this is going on and how to fix
this?

Thanks,
Sam Bacsa


 SNIP 

[EMAIL PROTECTED] zaptel]# make
gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__
-DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I.
-Wstrict-prototypes -fomit-frame-pointer
-I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include
-I/usr/src/linux/include/net -DMODVERSIONS -include
/usr/src/linux-2.4/include/linux/modversions.h  -DSTANDALONE_ZAPATA -c
tor2.c
In file included from tor2.c:30:
/usr/src/linux-2.4/include/linux/kernel.h:60: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:60: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:60: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:61: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:61: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:62: `panic_R_ver_str' declared
as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:62: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:68: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:68: `simple_strtoul_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:68: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:69: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:69: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:69: `simple_strtol_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:69: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:70: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:70: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:70:
`simple_strtoull_R_ver_str' declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:70: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:72: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:72: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:73: `sprintf_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:73: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:74: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:74: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:74: `vsprintf_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:74: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:75: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:75: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:76: `snprintf_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:76: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:77: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:77: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:77: `vsnprintf_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:77: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:79: invalid suffix on integer
constant
/usr/src/linux-2.4/include/linux/kernel.h:79: parse error before numeric
constant
/usr/src/linux-2.4/include/linux/kernel.h:80: `sscanf_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:80: warning: function
declaration isn't a prototype
/usr/src/linux-2.4/include/linux/kernel.h:81: `vsscanf_R_ver_str'
declared as function returning a function
/usr/src/linux-2.4/include/linux/kernel.h:81: warning: parameter names
(

Re: [Asterisk-Users] How to get call back when transfer fails

[Andrew Kohlsmith]

> Try something like this.  Record a message (if one doesn't already
> exist) saying "Please hold while your call is being transferred."
>
> Then add it to your dialplan as such:
>
> exten 101,1,Setvar(myexten)
> exten 101,2,Dial(Zap/1,10,t)
> exten 101,103,getvar(myexten)
> exten 101,104,playback(hold-transfer)
> exten 101,105,Wait(3)
> exten 101,106,Dial(myexten)
>
> The hold message would take about 3 seconds, then with the additional 3
> second wait, that would give you 6 seconds to hang up your phone, which
> if you are transferring a call is very reasonable.  If it goes through,
> then great, you are done.  If not, your phone will ring again in a few
> seconds and you get the call back.

That would indeed work.  A little hackish but yeah, definitely.

I've been playing around a little bit this afternoon.  I added a PARKEDAT 
variable to res_parking.c so that I could get the parked extension in the 
dialplan.  I'm hoping to do something like this:

exten 100,1,ChanIsAvail(Zap/1)
exten 100,2,Transfer(Zap/1)
exten 100,3,Hangup
exten 100,102,Park
exten 100,103,Play(unavailable, press 1 to return to the call or 2 to try 
someone else)
exten 1,1,ParkedCall(${PARKEDAT})
exten 2,whatever...

obviously in a macro but that's the idea...  I was originally farting around 
with quiet meetme conferences but that wasn't quite right.  

Am I on the right track here?  Hopefully I can take res_parking and create a 
park() that can incorporate a parkandannounce so that you can select whether 
you want it audible or in a script (for putting the parking # on a sip phone 
or something).

What do you think?

-A.
exten 1,1,
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Re: [Asterisk-Users] Echo Cancellation Feature

[Rich Adamson]

> I do feel the echo cancellation does need some work.
> 
> Currently, other than listening to users, there is no way to benchmark or
> trouble shoot echo problems.

Sure there are, it's just that 99% of the asterisk implementors don't
have the test equipment to do it, and a good share probably wouldn't
know how to do it if they had access to the equipment.

> We find that 2 to 3 out of every 20 calls will experience echo.  While
> echo is a problem that naturally occurs from SIP - PSTN and vice versa, I
> am still baffled by the fact that the cancellation works randomly.
> 
> When doing a zap show channel X, it will also report that the cancellation
> is still on.  We experience the most echo with a T100P --> Adtran TA 750
> FXO modules.  While I understand these do not have impedence matching, I
> wonder to myself why echo cancellation works sometimes, and others not.
> 
> Looking at Network util, processor util, and memory utilization, they do
> not provide any clear indication as to why /when it occurs.

Not likely to have any impact whatsoever.
 
> Is there anything more that can be done to debug echo cancellation, and
> further are other users experiencing this random echo.  I know it was
> discussed before, but the support folks at digium aren't able to offer
> anymore help.

You've probably read enough from previous postings to know there are
several different locations within an end-to-end voice call where echo can
creap into a system. In very general terms, any place where an end-to-end 
channel incures a two-wire to four-wire conversion (whether done in hardware
or software), echo can creap in due to lots of different reasons. Since
asterisk provides us with lots of configuration choices, hardly any two
systems are alike. Therefore, don't know that anyone is going to write
* code anytime soon to correct something that can't be pointed to, etc.

Someone mentioned they have echo on sip to sip calls (presumably the call
was processed by a single * system). If they do, the problem is likely
in the sip phone as there are no echo cancallation needs in that four-wire
end-to-end call from an * perspective.

I've got a fair amount of test equipment (and 20+ years telephony 
background), and am planning to assemble a document identifying some of 
the pstn issues, level settings, and other things impacting a reasonable 
system implementation. Unless someone wants to UPS a transmission test 
set to me quickly, the document won't be completed for several weeks. 
(The only test set I have access to will not be released for a couple 
of weeks due to classes, etc.)

I'm also expecting these tests to point out a number of other transmission
issues within asterisk that we'll get documented with real numbers, etc.

Rich


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RE: [Asterisk-Users] How to get call back when transfer fails

[Jeremy Hall]

exten 100,1,ChanIsAvail(Zap/1)
exten 100,2,Transfer(Zap/1)
exten 100,3,Hangup
exten 100,102,Park
exten 100,103,Play(unavailable, press 1 to return to the call or 2 to
try 
someone else)
exten 1,1,ParkedCall(${PARKEDAT})
exten 2,whatever...

obviously in a macro but that's the idea...  I was originally farting
around 
with quiet meetme conferences but that wasn't quite right.  

Am I on the right track here?  Hopefully I can take res_parking and
create a 
park() that can incorporate a parkandannounce so that you can select
whether 
you want it audible or in a script (for putting the parking # on a sip
phone 
or something).

What do you think?

-A.
exten 1,
-
I think that is a great idea.  Rather than the phrase you used above,
something along the lines of " is unavailable.  To be
transferred to voice mail, press 1.  To return to the previous person
you spoke with, press 2."  There has to be a more elegant way to say it,
but you get the idea.

You could take it to the next step and have a single operator queue, and
have another option of "To be placed on hold and wait for the person to
become available, press 3."  Then if they choose that option, "You will
be connected as soon as the person is available.  Maximum wait time is 3
minutes, at which time you will be transferred to voice mail.  You may
press the * key at any time to be immediately transferred to voicemail."

The options you are opening up with your idea are tremendous, and will
help asterisk fit in with and replace some other existing PBX solutions.

Jeremy

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[Asterisk-Users] Adtran TA750 Noise

[Greg Scasny]

All,

I need help.

I have an (actually 2) Adtran TA750’s with 8 FXO
ports. I get a terrible buzz on every FXO port. If I unplug the Adtran and put
an analog phone on each incoming line, I have no buzz.

I also have 2 Carrier Access Access Bank I’s with 12
FXO ports. When I plug the same analog lines into either one of those, no noise
or buzz whatsoever.

 

I went so far as to move the TA750 to within 5 feet of the demark
and ran a short CAT5 cable from the demark to the TA705 and still lots of buzz.


 

I have duplicates of every part for the TA750’s and
swapped every component and cannot get rid of the hum on either unit. I have
the power supply of the Adtran grounded. I am out of ideas L.

Any assistance would be greatly appreciated.

 

 

Thanks in advance…..>>Greg

 

Gregory P. Scasny

Golden Technologies Inc.

http://www.golden-tech.com

219-462-7200

 

RE: [Asterisk-Users] Echo Cancellation Feature

[Jeremy Hall]

I've got a fair amount of test equipment (and 20+ years telephony 
background), and am planning to assemble a document identifying some of 
the pstn issues, level settings, and other things impacting a reasonable

system implementation. Unless someone wants to UPS a transmission test 
set to me quickly, the document won't be completed for several weeks. 
(The only test set I have access to will not be released for a couple 
of weeks due to classes, etc.)

I'm also expecting these tests to point out a number of other
transmission
issues within asterisk that we'll get documented with real numbers, etc.

Rich


___

Rich,

One suggestion I would like to make, is where possible, tell us how to
replicate the tests as best we can if we don't have the proper
equipment.  I'd venture to say most of us have a good, fairly sensitive,
digital VOM.  I know not all tests can be made with that, but I'm sure
some of them can.

There are fairly accurate tone generator programs that work with a sound
card, same with data decoding.  My point is, as you said before, not
everyone has a multi-thousand dollar test set, but would still like to
do what they can to properly implement a telephone system.

Thanks,

Jeremy

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Re: [Asterisk-Users] Echo Cancellation Feature

[Tom]

At 02:30 PM 4/22/2004, you wrote:
I would also offer feedback that we too have random calls with echo on our 
end, that can't be traced to a reproducible event. It's very odd and can 
be frustrating, as it's a big distraction for those that don't know 
better. Like a bad cell phone connection when you hear yourself talk. For 
us, this happens in a pure SIP environment on a network switch dedicated 
to Asterisk, a T1 PRI and 18 SIP handsets.

HTH,
Ryan
We have three Cisco 7940 SIP phones and 1 POTS phone connected to our * 
server with  TDM10B fxs card.  Our * server is connected to the pstn with 3 
X100P cards.  We have similar echo problems but only on our SIP phones.  We 
do not have any echo problems with the POTS phone.

We just purchased a Polycom IP 500 SIP phone to test but I expect similar 
echo problems.

The recent thread which addressed milliwatt generators and gain adjustments 
helped to reduce our echo but the thread was never completed.  We are not 
sure how to test the TX gain adjustment and where on the graph to set the 
RX gain when using a milliwatt generator tone.

Tom

Tom


On Apr 22, 2004, at 1:37 PM, Brent Franks wrote:

I do feel the echo cancellation does need some work.

Currently, other than listening to users, there is no way to benchmark or
trouble shoot echo problems.
We find that 2 to 3 out of every 20 calls will experience echo.  While
echo is a problem that naturally occurs from SIP - PSTN and vice versa, I
am still baffled by the fact that the cancellation works randomly.

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[Asterisk-Users] Re: Trouble Compiling "zaptel"

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Sam Bacsa <[EMAIL PROTECTED]> wrote:
> So I went to go compile the Zaptel library from the HEAD CVS and I get
> some really really odd errors which don't make any sense.  I've attached
> the console output ... any idea why this is going on and how to fix
> this?

Uninstall and re-install the kernel sources, and make sure the link
/usr/src/linux2.4 points to the correct source tree that matches the
running kernel.

Cheers,
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Cisco phones

[Paul Tyreman]

I have bough a cisco phone on eBay to use with 
Asterisk, but according to that website, you need a contract with Cisco systems 
to upgrade the phone to work with SIP.
 
I am guessing the phone that I get won't come with 
that as it was used with the cisco call manager software in the past.  Can 
I still use this phone with Asterisk, or have I waited my money ?
 
Thanks, Paul.
 
 
 
-Original Message-From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory 
JunkerPosted At: 22 April 2004 19:16Posted To: 
Asterisk-UsersConversation: [Asterisk-Users] Cisco phonesSubject: Re: 
[Asterisk-Users] Cisco phones
 
Well, before someone jumps down your throat for asking a question that 
gets asked multiple times a week... ;)
 
http://www.voip-info.org/wiki-Cisco+Phones
 
 
 
On a related note...whoever maintains the page at 
 
http://www.asteriskpbx.com/index.php?menu=support
 
may want to think about putting the link to the Wiki in a much more 
prominent place, and make it clear that new users should look there first, since 
they often go through the hassle of signing up for the mailing list only to post 
a question for which they could have found the answer in much less time.
 
Greg
 
 
On Thu, 2004-04-22 at 18:55 +0100, Nick Knight wrote:> Hello 
all,> >  > > I haven’t used cisco phones as yet 
– do they work with asterisk, are > they good which models are the 
best?> >  > > I am after a starting 
point!> >  > > Thanks> >  
> > Nick> > 
 
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RE: [Asterisk-Users] Echo Cancellation Feature

[Brent Franks]

> We have three Cisco 7940 SIP phones and 1 POTS phone connected to our
*
> server with  TDM10B fxs card.  Our * server is connected to the pstn
with
> 3
> X100P cards.  We have similar echo problems but only on our SIP
phones.
> We do not have any echo problems with the POTS phone.
> 
> We just purchased a Polycom IP 500 SIP phone to test but I expect
similar
> echo problems.
> 

Don't expect the IP 500's to do anything as you stated, we are using
these now and are having the problem I described earlier with these
phones.

- B

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Re: [Asterisk-Users] Adtran TA750 Noise

[Michael Welter]

I'm having the _exact_ same problem.  Is your POTS line, by chance, a 
fiber circuit from the CO and then analog to your location?

Mike

Greg Scasny wrote:


All,

I need help.

I have an (actually 2) Adtran TA750’s with 8 FXO ports. I get a terrible 
buzz on every FXO port. If I unplug the Adtran and put an analog phone 
on each incoming line, I have no buzz.

I also have 2 Carrier Access Access Bank I’s with 12 FXO ports. When I 
plug the same analog lines into either one of those, no noise or buzz 
whatsoever.

 

I went so far as to move the TA750 to within 5 feet of the demark and 
ran a short CAT5 cable from the demark to the TA705 and still lots of buzz.

 

I have duplicates of every part for the TA750’s and swapped every 
component and cannot get rid of the hum on either unit. I have the power 
supply of the Adtran grounded. I am out of ideas L .

Any assistance would be greatly appreciated.

 

 

Thanks in advance…..>>Greg

 

Gregory P. Scasny

Golden Technologies Inc.

http://www.golden-tech.com 

219-462-7200

 

--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado
+1 303 674 2575
[EMAIL PROTECTED]
www.introspect.com
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Re: [Asterisk-Users] Echo Cancellation Feature

[Chris Maresca]

The single most usefull tool that anyone outside telco consultants is
likely to have is ztmonitor.

If you do a ztmonitor [channel number] -v you will get a visual of the
sound strengths and it's pretty easy to see when rx or tx are out of
balance...

Now, if only that would help fix the low-level static noise I have on the
x100p, that would be great.

Chris.

On Thu, 22 Apr 2004, Rich Adamson wrote:

> > I do feel the echo cancellation does need some work.
> > 
> > Currently, other than listening to users, there is no way to benchmark or
> > trouble shoot echo problems.
> 
> Sure there are, it's just that 99% of the asterisk implementors don't
> have the test equipment to do it, and a good share probably wouldn't
> know how to do it if they had access to the equipment.
> 
> > We find that 2 to 3 out of every 20 calls will experience echo.  While
> > echo is a problem that naturally occurs from SIP - PSTN and vice versa, I
> > am still baffled by the fact that the cancellation works randomly.
> > 
> > When doing a zap show channel X, it will also report that the cancellation
> > is still on.  We experience the most echo with a T100P --> Adtran TA 750
> > FXO modules.  While I understand these do not have impedence matching, I
> > wonder to myself why echo cancellation works sometimes, and others not.
> > 
> > Looking at Network util, processor util, and memory utilization, they do
> > not provide any clear indication as to why /when it occurs.
> 
> Not likely to have any impact whatsoever.
>  
> > Is there anything more that can be done to debug echo cancellation, and
> > further are other users experiencing this random echo.  I know it was
> > discussed before, but the support folks at digium aren't able to offer
> > anymore help.
> 
> You've probably read enough from previous postings to know there are
> several different locations within an end-to-end voice call where echo can
> creap into a system. In very general terms, any place where an end-to-end 
> channel incures a two-wire to four-wire conversion (whether done in hardware
> or software), echo can creap in due to lots of different reasons. Since
> asterisk provides us with lots of configuration choices, hardly any two
> systems are alike. Therefore, don't know that anyone is going to write
> * code anytime soon to correct something that can't be pointed to, etc.
> 
> Someone mentioned they have echo on sip to sip calls (presumably the call
> was processed by a single * system). If they do, the problem is likely
> in the sip phone as there are no echo cancallation needs in that four-wire
> end-to-end call from an * perspective.
> 
> I've got a fair amount of test equipment (and 20+ years telephony 
> background), and am planning to assemble a document identifying some of 
> the pstn issues, level settings, and other things impacting a reasonable 
> system implementation. Unless someone wants to UPS a transmission test 
> set to me quickly, the document won't be completed for several weeks. 
> (The only test set I have access to will not be released for a couple 
> of weeks due to classes, etc.)
> 
> I'm also expecting these tests to point out a number of other transmission
> issues within asterisk that we'll get documented with real numbers, etc.
> 
> Rich
> 
> 
> 

--
chris maresca
  senior partner - www.olliancegroup.com

linux, up 17 days


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RE: [Asterisk-Users] Cisco phones

[Sam Bacsa]

You can get an upgrade contract with Cisco for like $8 or 
something to download the SIP firmware for your phone.
 
So no, no waste of money -- unless you bought the wrong 
phone.
 
- Sam


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Paul 
TyremanSent: Thursday, April 22, 2004 2:49 PMTo: 
[EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Cisco 
phones

I have bough a cisco phone on eBay to use with 
Asterisk, but according to that website, you need a contract with Cisco systems 
to upgrade the phone to work with SIP.
 
I am guessing the phone that I get won't come with 
that as it was used with the cisco call manager software in the past.  Can 
I still use this phone with Asterisk, or have I waited my money ?
 
Thanks, Paul.
 
 
 
-Original Message-From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory 
JunkerPosted At: 22 April 2004 19:16Posted To: 
Asterisk-UsersConversation: [Asterisk-Users] Cisco phonesSubject: Re: 
[Asterisk-Users] Cisco phones
 
Well, before someone jumps down your throat for asking a question that 
gets asked multiple times a week... ;)
 
http://www.voip-info.org/wiki-Cisco+Phones
 
 
 
On a related note...whoever maintains the page at 
 
http://www.asteriskpbx.com/index.php?menu=support
 
may want to think about putting the link to the Wiki in a much more 
prominent place, and make it clear that new users should look there first, since 
they often go through the hassle of signing up for the mailing list only to post 
a question for which they could have found the answer in much less time.
 
Greg
 
 
On Thu, 2004-04-22 at 18:55 +0100, Nick Knight wrote:> Hello 
all,> >  > > I haven’t used cisco phones as yet 
– do they work with asterisk, are > they good which models are the 
best?> >  > > I am after a starting 
point!> >  > > Thanks> >  
> > Nick> > 
 
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[Asterisk-Users] MWI indicator on SNOM200 doesn't disappear

[Ian White]

On recent releases of the snom200 firmware, the MWI indicator will turn 
on, but won't turn off when the message has been checked. It works on 
firmware 2.03o, but not in 2.04g or newer. I filed a bug report with 
snom, but they're claiming it is an asterisk issue and that it should 
have been resolved. They suggested that I ask on the list.

"Anyway, Asterisk had a bug where it didn't send the NOTIFY correctly to
turn off the MWI.  The message doesn't contain the line so the phone
doesn't know which line to apply the messages to.
Basically the NOTIFY message should contain something like the
following:
NOTIFY sip:[EMAIL PROTECTED];line=34n34jed SIP/2.0
There was a bugfix for this in Asterisk for this problem, do you have
that applied?"
I am running the current CVS version, and don't see anything in the 
code that looks like this has been touched, and I haven't seen 
reference to it on this list. They are right in that the line 
information isn't being sent, looking at the SIP debugs on both ends. 
Anybody have ideas?

Ian

--
Ian White
South Island Community Access Network (SICAN)
email: [EMAIL PROTECTED]
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Re: [Asterisk-Users] Cisco phones

[Hermann Wecke]

On Thu, 22 Apr 2004, Paul Tyreman wrote:
> I am guessing the phone that I get won't come with that as it was used
> with the cisco call manager software in the past.  Can I still use this
> phone with Asterisk, or have I waited my money ?

Every Cisco software embedded with their hardware is valid only for the
first owner. When someone sells the equipment, the software license is not
transfered. The new buyer must buy a new license.

I bought a Cisco 7960G over eBay also, and I bought their SIP software
later. I paid US$ 105. The original was a SCCP.

Check also the list history. You will find several messages regarding this
same issue (cisco hardware X software X upgrade). You can find the
archives here: http://lists.digium.com/pipermail/asterisk-users/
(actually, use Google with this query:
"cisco sip upgrade site:lists.digium.com"
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RE: [SPAM] - Re: [Asterisk-Users] Adtran TA750 Noise - Email found in subject

[Greg Scasny]

I believe it is not fiber, but I am not sure. I am going to take one of
them home tonight and hook it to my POTS line there, which for sure is
not fiber.

Gregory P. Scasny

Golden Technologies Inc.

http://www.golden-tech.com

219-462-7200


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Welter
Sent: Thursday, April 22, 2004 5:02 PM
To: [EMAIL PROTECTED]
Subject: [SPAM] - Re: [Asterisk-Users] Adtran TA750 Noise - Email found
in subject

I'm having the _exact_ same problem.  Is your POTS line, by chance, a 
fiber circuit from the CO and then analog to your location?

Mike

Greg Scasny wrote:
> 
> 
> All,
> 
> I need help.
> 
> I have an (actually 2) Adtran TA750's with 8 FXO ports. I get a
terrible 
> buzz on every FXO port. If I unplug the Adtran and put an analog phone

> on each incoming line, I have no buzz.
> 
> I also have 2 Carrier Access Access Bank I's with 12 FXO ports. When I

> plug the same analog lines into either one of those, no noise or buzz 
> whatsoever.
> 
>  
> 
> I went so far as to move the TA750 to within 5 feet of the demark and 
> ran a short CAT5 cable from the demark to the TA705 and still lots of
buzz.
> 
>  
> 
> I have duplicates of every part for the TA750's and swapped every 
> component and cannot get rid of the hum on either unit. I have the
power 
> supply of the Adtran grounded. I am out of ideas L .
> 
> Any assistance would be greatly appreciated.
> 
>  
> 
>  
> 
> Thanks in advance.>>Greg
> 
>  
> 
> Gregory P. Scasny
> 
> Golden Technologies Inc.
> 
> http://www.golden-tech.com 
> 
> 219-462-7200
> 
>  
> 

-- 
Michael Welter
Introspect Telephony Corp.
Denver, Colorado
+1 303 674 2575
[EMAIL PROTECTED]
www.introspect.com


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RE: [Asterisk-Users] Echo Cancellation Feature

[Chris Maresca]

I've got a really cheap analog phone connected to a Sipura SIP adaptor,
and have zero echo problems...

Just static problems, but it may be related.

Chris.

On Thu, 22 Apr 2004, Brent Franks wrote:

> > We have three Cisco 7940 SIP phones and 1 POTS phone connected to our
> *
> > server with  TDM10B fxs card.  Our * server is connected to the pstn
> with
> > 3
> > X100P cards.  We have similar echo problems but only on our SIP
> phones.
> > We do not have any echo problems with the POTS phone.
> > 
> > We just purchased a Polycom IP 500 SIP phone to test but I expect
> similar
> > echo problems.
> > 
> 
> Don't expect the IP 500's to do anything as you stated, we are using
> these now and are having the problem I described earlier with these
> phones.
> 
> - B
> 
> 

--
chris maresca
  senior partner - www.olliancegroup.com

linux, up 17 days


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Re: [Asterisk-Users] MWI indicator on SNOM200 doesn't disappear

[Geert Nijpels]

Ian White wrote:

On recent releases of the snom200 firmware, the MWI indicator will 
turn on, but won't turn off when the message has been checked. It 
works on firmware 2.03o, but not in 2.04g or newer. I filed a bug 
report with snom, but they're claiming it is an asterisk issue and 
that it should have been resolved. They suggested that I ask on the list.

"Anyway, Asterisk had a bug where it didn't send the NOTIFY correctly to
turn off the MWI.  The message doesn't contain the line so the phone
doesn't know which line to apply the messages to.
Basically the NOTIFY message should contain something like the
following:
NOTIFY sip:[EMAIL PROTECTED];line=34n34jed SIP/2.0
There was a bugfix for this in Asterisk for this problem, do you have
that applied?"
I am running the current CVS version, and don't see anything in the 
code that looks like this has been touched, and I haven't seen 
reference to it on this list. They are right in that the line 
information isn't being sent, looking at the SIP debugs on both ends. 
Anybody have ideas?

Ian

This is a problem I have been digging into a bit. In my case asterisk 
did not send out the NOTIFY with the header Content-Type: 
"application/simple-message-summary", but with "Content-Type: 
text/plain", so the NOTIFY is treated as a txt message. In result, when 
I pressed the MWI button, I saw the text from asterisk stating the 
amount of messages I have. I changed it to work, and now asterisk calls 
the extension the message is sent from ([EMAIL PROTECTED]). After 
calling this the MWI indication disappears, I'm not sure if it also 
disappears after calling from another phone.

I'm using chan_sip2 and I changed some stuff, so I'm not sure if this is 
also a problem with standard chan_sip (the txt vs vm issue).

Kind regards,

Geert
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Re: [Asterisk-Users] Adtran TA750 Noise

[Rich Adamson]

> I have an (actually 2) Adtran TA750s with 8 FXO ports. I get a terrible buzz on 
> every FXO 
port. If I unplug the Adtran and put an analog phone
> on each incoming line, I have no buzz.
> 
> I also have 2 Carrier Access Access Bank Is with 12 FXO ports. When I plug the same 
> analog 
lines into either one of those, no noise or buzz
> whatsoever.
> 
>  
> 
> I went so far as to move the TA750 to within 5 feet of the demark and ran a short 
> CAT5 cable 
from the demark to the TA705 and still lots of buzz.
> 
>  
> 
> I have duplicates of every part for the TA750s and swapped every component and 
> cannot get rid 
of the hum on either unit. I have the power
> supply of the Adtran grounded. I am out of ideas L.
> 
> Any assistance would be greatly appreciated.

Well, you're the second one in a rather short period of time that
has complained about the exact same hum/buzz when the Adtran 750
is used with FXO interfaces and with X100P cards.

There was another posting earlier today in which the individual made
a comment the 750 FXO interface does not support impedence matching.
I thought the statement was rather strange, but since I don't own one
of these units, I went to the Adtran site in search of a technical
description of the FXO card. I couldn't find one, and for that matter,
it appears Adtran has little reference to the 750 being used with any
FXO interface. (Its almost like they know there is a problem and
removed the 750 FXO options. Selling as FXS only now.)

>From what I'm hearing/understanding, its all beginning to make sense
(believe or not). If the no-impedence-matching is true (or even if the
technical words are slightly/somewhat incorrect), then its beginning
to appear the Adtran FXO interface is not presenting a "balanced" 
interface to the tip & ring pstn line. In other words, one side of 
the line must have some internal electronics hanging on it that 
disturbs the balance needed for pstn lines, and that imbalance is 
causing induced AC power (which is extremely common on most pstn 
lines) to be heard.

This is going to be rather difficult to explain without a drawing, but
I'll give it a try. The pstn line (all the way from the CO or fiber
mux cabinet) is nothing more then twisted copper pairs, that have a
very specific number of twists per unit of length. The twists are
actually built into the cables to ensure that whatever outside electrical
influence exists (such as AC Power), that outside source influences 
both tip and ring in "exactly" the same amount. At the end of that cable
(whether its in your house or business) if you attached a perfectly
balanced piece of equipment, it doesn't make any difference whether
that outside influence (in this case, AC power) is ten volts or fifty
volts, that influence is cancelled out and not heard. But, its because
the attached device (usually an analog phone) presents an equal load
to both the tip and ring. (That should be fairly obvious since the
typical analog phone doesn't have any real way to create an imbalance
since it doesn't have access to ground or AC power. For the real
technical types, its the differential voltage between tip and ring
that creates the sound.)

If one would connect an analog phone to the pstn line that you're 
having the hum on, and then attach a resister from one side of the
line to ground (say, from the tip to ground), you are artifically
creating the imbalance that I'm talking about. The analog phone will
now have the hum that you're hearing via the Adtran & asterisk because
of the imbalanced line. The size of the resister (whether 100 ohms or 
1,000,000 ohms) will impact the loudness of the hum; the smaller the 
value the louder the hum.

In the olden days of telephony, we use to install "repeat coils" to
isolate the imbalanced equipment (usually customer owned stuff).
(Here comes the harder part to describe in words. Really need a visual
schematic for this.)

Repeat coils were absolutely nothing more then a basic audio transformer
with two primary windings and two secondary windings. A couple of 2 ufd
capacitors and the repeat coil was all that was needed to isolate the
imbalanced piece of equipment from the pstn line, pass the DC component
needed for supervision, and elimate the hum. In the '60s and '70's, 
those repeat coils were almost the size of a beer can, and one use to
buy them from Western Electric (and others) mounted on rails that
fit a 19" or 23" rack, along with the capacitors on another set of rails.

In very unusual cases of imbalance, we use to purchase the same set of
repeat coils and capacitors implemented in a box with switches on the
front panel. The switches would allow the technician to add resistive
values onto one side of the line or the other, in an attempt to "match"
the imperfections created by water logged cables, etc. If a cable pair
had a problem that indicated 10,000 ohms of resistence to ground on 
the tip side of the line, this box would attempt to add 10,000 ohms
to the ring side 

RE: [Asterisk-Users] Cisco phones

[Simon Brown]

Are you using the 7905 with a SIP image?

Simon

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alric
Sent: Friday, 23 April 2004 4:37
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco phones



> Nick Knight wrote:
>
> > I haven't used cisco phones as yet - do they work with asterisk, are 
> > they good which models are the best?
>
> I use a 7960 with Asterisk and absolutely love it.  It blows the snot 
> out of the Nortel phone I used to use.
>
> John

I know the 7905, 7912, 7940, and 7960 all work with Asterisk using the SIP
protocol.

I'm currently using a 7905G at home, and it works great.

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[Asterisk-Users] SIP/IAX termination provider in NZ

[Simon Brown]

I am looking for a SIP/IAX termination provider in New Zealand.  Does anyone
know of one?

TIA

Simon
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Re: [Asterisk-Users] * INSTRUCTIONS FOR NEW MEMBERS OF THE COMMUNITY * Please read

[Leo Ann Boon]

These are exiting times for Asterisk.org. We're getting close to a

Ehh, you surely don't mean 'exiting' time :)? Not a good idea to tell 
new members to jump off the ship just after they boarded.

Suggestion, can this message be sent as part of the list welcome message?

Cheers.

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Re: [Asterisk-Users] How to get call back when transfer fails

[Andrew Kohlsmith]

> I think that is a great idea.  Rather than the phrase you used above,
> something along the lines of " is unavailable.  To be
> transferred to voice mail, press 1.  To return to the previous person
> you spoke with, press 2."  There has to be a more elegant way to say it,
> but you get the idea.

Well I was looking at it more as what the receptionist would hear trying to 
send a call to someone's extension.  I like what you just explained though -- 
it almost eliminates the need for assisted transfer and makes it very 
friendly to the person on the other side.

It's also paving the way to assisted transfer (where you page the extension, 
say "xyz is on the line" and then when you hang up they're connected).

> You could take it to the next step and have a single operator queue, and
> have another option of "To be placed on hold and wait for the person to
> become available, press 3."  Then if they choose that option, "You will
> be connected as soon as the person is available.  Maximum wait time is 3
> minutes, at which time you will be transferred to voice mail.  You may
> press the * key at any time to be immediately transferred to voicemail."

Yes that is very nice!

> The options you are opening up with your idea are tremendous, and will
> help asterisk fit in with and replace some other existing PBX solutions.

Exactly why I'm trying to do it.  :-)

As I said I am hoping to modify res_parking and app_parkandannounce to create 
a more generic app_park, where it returns ${PARKEDAT}, jumps to n+101 if it 
can't park and last but not least OPTIONALLY announces the park location.

I have to say I am really pleased with how modular asterisk is.  Most of the 
apps are under a thousand lines of code, and the whole thing seems to be 
geared for maximum modularity. 

Regards,
Andrew
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RE: [Asterisk-Users] Echo Cancellation Feature

[Rich Adamson]

> 
> 
> I've got a fair amount of test equipment (and 20+ years telephony 
> background), and am planning to assemble a document identifying some of 
> the pstn issues, level settings, and other things impacting a reasonable
> 
> system implementation. Unless someone wants to UPS a transmission test 
> set to me quickly, the document won't be completed for several weeks. 
> (The only test set I have access to will not be released for a couple 
> of weeks due to classes, etc.)
> 
> I'm also expecting these tests to point out a number of other
> transmission
> issues within asterisk that we'll get documented with real numbers, etc.



> One suggestion I would like to make, is where possible, tell us how to
> replicate the tests as best we can if we don't have the proper
> equipment.  I'd venture to say most of us have a good, fairly sensitive,
> digital VOM.  I know not all tests can be made with that, but I'm sure
> some of them can.
> 
> There are fairly accurate tone generator programs that work with a sound
> card, same with data decoding.  My point is, as you said before, not
> everyone has a multi-thousand dollar test set, but would still like to
> do what they can to properly implement a telephone system.

I'm about 95% confident I can measure and describe several different
issues that truly have been impacting interfaces to pstn lines, etc.
But, I need to validate the steps before running off at the mouth with
misrepresentations, etc. I don't believe these issues are asterisk
related at all, but rather outside influences that many are feeling
but can't see (or deal with).

Assuming I'm correct, the instrument needed to identify at least "some"
of these issues retails for about $300 US. Example, www.triplett.com
click on Test Equipment, Telco Testers, Model 2 to 7, don't know for
sure as yet. Lots of other venders as well. To prove the point (and
write the documentation), several other pieces of test equipment will
be likely for me, but not needed by you. The unit I'm borrowing sells
for over $3,000 but does lots of other stuff not needed in typical
asterisk deployments.

Just about every telephone installer in the US (there are exceptions)
is carrying something similar to the above. They use it to measure levels
"to" the milliwatt generator, and they use the noise measurement side
"to" the quiet termination. The milliwatt generator and quiet termination
are often times implemented in the same piece of telco hardware, but
they have different telephone numbers assigned to them. (Don't think 
we have a quiet termination in asterisk as yet, but should be easy to
implement if it is actually needed.)

The standard VOM isn't going to cut it for these tests as they are no
where near sensitive enough to accurately measure levels, noise, AC
induction, etc. But, if we're all going to play in the telephony 
business then we better buy (and understand how to use) the tools 
necessary to play in that business. I think I can help.

I'm kind of thinking that "if" this can be described accurately, I'd
guess one of the developers can add some code to help measure/identify
the issues. If that impression is correct, then the test set won't be
required at all. Let's see how it goes!

Hopefully I can help that process and understanding even though I'm 
not a coder.

Rich


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Re: [Asterisk-Users] Echo Cancellation Feature

[Rich Adamson]

> At 02:30 PM 4/22/2004, you wrote:
> >I would also offer feedback that we too have random calls with echo on our 
> >end, that can't be traced to a reproducible event. It's very odd and can 
> >be frustrating, as it's a big distraction for those that don't know 
> >better. Like a bad cell phone connection when you hear yourself talk. For 
> >us, this happens in a pure SIP environment on a network switch dedicated 
> >to Asterisk, a T1 PRI and 18 SIP handsets.
> >
> >HTH,
> >Ryan
> 
> We have three Cisco 7940 SIP phones and 1 POTS phone connected to our * 
> server with  TDM10B fxs card.  Our * server is connected to the pstn with 3 
> X100P cards.  We have similar echo problems but only on our SIP phones.  We 
> do not have any echo problems with the POTS phone.
> 
> We just purchased a Polycom IP 500 SIP phone to test but I expect similar 
> echo problems.
> 
> The recent thread which addressed milliwatt generators and gain adjustments 
> helped to reduce our echo but the thread was never completed.  We are not 
> sure how to test the TX gain adjustment and where on the graph to set the 
> RX gain when using a milliwatt generator tone.

The thread truly has not been completed and will likely take a few more
weeks to do it. I do believe we'll get to the bottom of it though, just
hand tight.

Rich


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Re: [Asterisk-Users] MWI indicator on SNOM200 doesn't disappear

[Rich Adamson]

FWIW, the snom 200 also had a problem when two or more lines were
registering with *. Since the MWI is shared between all lines, a message
left on line 1 would turn it on, followed by a notify on line 2 that
no messages existed and turned if off. On/off/on/off and most of the 
time you wouldn't even notice the change. Not a very cool multi-line
phone.


> On recent releases of the snom200 firmware, the MWI indicator will turn 
> on, but won't turn off when the message has been checked. It works on 
> firmware 2.03o, but not in 2.04g or newer. I filed a bug report with 
> snom, but they're claiming it is an asterisk issue and that it should 
> have been resolved. They suggested that I ask on the list.
> 
> "Anyway, Asterisk had a bug where it didn't send the NOTIFY correctly to
> turn off the MWI.  The message doesn't contain the line so the phone
> doesn't know which line to apply the messages to.
> 
> Basically the NOTIFY message should contain something like the
> following:
> NOTIFY sip:[EMAIL PROTECTED];line=34n34jed SIP/2.0
> 
> There was a bugfix for this in Asterisk for this problem, do you have
> that applied?"
> 
> I am running the current CVS version, and don't see anything in the 
> code that looks like this has been touched, and I haven't seen 
> reference to it on this list. They are right in that the line 
> information isn't being sent, looking at the SIP debugs on both ends. 
> Anybody have ideas?
> 
> Ian
> 
> -- 
> Ian White
> South Island Community Access Network (SICAN)
> email: [EMAIL PROTECTED]
> 
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