[Asterisk-Users] Asterisk launch on boot

[justin]

Hi,

What's the perfered way to launch Asterisk on boot, on say Redhat?

I tried putting this in my rc.local:

modprobe wcfxo
/usr/bin/screen -d -m /usr/sbin/asterisk -vvvcn

But it doesn't work, and I get this in my asterisk message log:

Aug  6 10:24:27 WARNING[16384]: File codec_g729b.c, Line 500 (load_module): Unable to 
initialize va stuff: -1

BTW, I'm using screen because of a problem in the g729 codec.

Thanks,
Justin


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[Asterisk-Users] Minimum system requirement for ....

[Dan]

Hi,

I need to know from your experience which is the minimum hardware
configuration (proc./memory) to run * with two X100P cards and around 10
internal IP phones (from which 4 are hardware ones and use G.711, the rest
of them with GSM), with a couple of IAX connections with other Asterisk
PBX'es (GSM too).

Have someone tried to build a graph about the dependencies between hardware
configuration and the number of used phones, PSTN connections. It can be
very useful for design purpose.

I have found a very cheap, quiet an small system (a Compaq Deskpro EN SFF)
with PII/350MHz and 128MB RAM and I want to know if it is enouigh  for my
purpose (a small home PBX).

Thanks,
Dan

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[Asterisk-Users] Behind Firewalls, SonicWalls, etc..

[John Sutter]

I've searched the archives a bit and have not really come up with
a good answer to my queries.
I have * running on a RH9 box behind a LinkSys NAT box.  I can talk
with iConnectHere outbound just fine.  I am trying to configure an
inbound Xten softphone from outside.  I have that user set as NAT in
sip.conf (seems to help), but I still cannot establish a full session.
I think the problem comes when * sends info about the RTP connection,
it passes the nonroutable address.  Is there a ways to tell it to use
the firewall IP address for users coming in through that address(and the
firewalls nonroutable address).
On the flip side, I have run out of port forwarding slots on my LinkSys,
so I am trying to resurrect my old startup's SonicWall PRO.  They indicate
they support SIP, but they do not elaborate upon that at all.  Anyone have
comments concerning the SW & SIP?
So far I have an ATA-186, a DG104S, as well as a TDM400P and X100P with
service from iConnectHere.  I hope to ship either the ATA-186 or DG104S
to Australia when I figure out how to deal with them remotely.
Thanks,

-- John

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[Asterisk-Users] Leftover Budgettone issues

[Brian Capouch]

I have my new phone mostly working.  I do have a couple of residuals 
that I cannot find mentioned in the list archives:

1. Is it possible to set the volume in these things?  I hope I didn't 
miss it, but I've looked in the doc, the FAQ, and the asterisk archives 
and don't find anything.  The displays in the pictures all have more 
bars on them than my phone does, and I need a bit more volume than I'm 
getting.

2. This phone does not act like all my others do when I am talking and a 
call comes in.  Instead of the jarring ADSI !!!BOING!!! followed by a 
series of call waiting beeps, instead I get a ringing tone in the 
earpiece which is audible to the other party as well.

Maybe there is some config setting that will handle that?

Thanks in advance for any information that might be out there.

B.

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Re: [Asterisk-Users] Libpri and asterisk fail to install

[Martin Pycko]

well should be ok if you cvs update now.

Martin

On Wed, 6 Aug 2003, Rhys Hopkins wrote:

> Martin Pycko wrote:
> > You're looking for libncurses-dev and in libpri you can remove -Werror
> > from libpri/Makefile or cvs update libpri (it should be fixed)
> >
>
> Thanks for that - I installed ncurses-devel and asterisk now builds ok,
> but libpri still gives the following error:
>
> rhys2:/usr/src/libpri # make clean
> rm -f *.o *.so *.lo
> rm -f testpri libpri.a libpri.so.1.0
> rm -f pritest pridump
> rhys2:/usr/src/libpri # make install
> cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o
> pri.o pri.c
> cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o
> q921.o q921.c
> cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o
> prisched.o prisched.c
> cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o
> q931.o q931.c
> q931.c: In function `q931_handle_ie':
> q931.c:1394: warning: comparison between signed and unsigned
> make: *** [q931.o] Error 1
> rhys2:/usr/src/libpri #
>
>
> Regards,
>
> Rhys.
>
>
> > regards
> > Martin
> >
> > On Wed, 6 Aug 2003, Rhys Hopkins wrote:
> >
> >
> >>Hi,
> >>
> >>I am having trouble building and installing libpri and asterisk on my
> >>system. Zaptel seemed to install OK.
> >>
> >>I am running SuSE 8.2 ( Linux 2.4.20-4GB )
> >>I have made sure I have the prerequisites ( rpm versions shown below )
> >>
> >>rhys2:/usr/src/libpri # uname -a
> >>Linux rhys2 2.4.20-4GB #1 Fri Jul 11 07:33:18 UTC 2003 i686 unknown
> >>unknown GNU/Linux
> >>
> >>rhys2:/opt/libpri-0.3.2 # rpm -q openssl-devel
> >>openssl-devel-0.9.6i-12
> >>rhys2:/opt/libpri-0.3.2 # rpm -q readline-devel
> >>readline-devel-4.3-105
> >>rhys2:/opt/libpri-0.3.2 # rpm -q readline
> >>readline-4.3-105
> >>rhys2:/opt/libpri-0.3.2 # rpm -q openssl
> >>openssl-0.9.6i-12
> >>rhys2:/opt/libpri-0.3.2 # rpm -q kernel-source
> >>kernel-source-2.4.20.SuSE-62
> >>rhys2:/opt/libpri-0.3.2 # rpm -q termcap
> >>termcap-2.0.8-674
> >>
> >>
> >>This is the output from make clean ; make install for libpri:
> >>
> >>rhys2:/usr/src/libpri # make clean
> >>rm -f *.o *.so *.lo
> >>rm -f testpri libpri.a libpri.so.1.0
> >>rm -f pritest pridump
> >>rhys2:/usr/src/libpri # make install
> >>cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o
> >>pri.o pri.c
> >>cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o
> >>q921.o q921.c
> >>cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o
> >>prisched.o prisched.c
> >>cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o
> >>q931.o q931.c
> >>q931.c: In function `ie2str':
> >>q931.c:1210: warning: comparison between signed and unsigned
> >>q931.c: In function `msg2str':
> >>q931.c:1227: warning: comparison between signed and unsigned
> >>q931.c: In function `q931_dumpie':
> >>q931.c:1251: warning: comparison between signed and unsigned
> >>q931.c: In function `add_ie':
> >>q931.c:1334: warning: comparison between signed and unsigned
> >>q931.c: In function `q931_handle_ie':
> >>q931.c:1394: warning: comparison between signed and unsigned
> >>make: *** [q931.o] Error 1
> >>
> >>
> >>
> >>
> >>
> >>This is the end of the output from make install for asterisk ( full
> >>output attached as "asterisk_errors" )
> >>
> >>loading cache ./config.cache
> >>checking for gcc... gcc
> >>checking whether the C compiler (gcc  ) works... yes
> >>checking whether the C compiler (gcc  ) is a cross-compiler...  no
> >>checking whether we are using GNU C... yes
> >>checking whether gcc accepts -g... yes
> >>checking how to run the C preprocessor... gcc -E
> >>checking host system type... i686-pc-linux-gnu
> >>checking for a BSD compatible install... install
> >>checking for ranlib... ranlib
> >>checking for ar... /usr/bin/ar
> >>checking for tgetent in -ltermcap... no
> >>checking for tgetent in -ltinfo... no
> >>checking for tgetent in -lcurses... no
> >>checking for tgetent in -lncurses... no
> >>configure: error: termcap support not found
> >>make: *** [editline/config.h] Error 1
> >>
> >>As can be seen from the rpm queries above, I have termcap installed.
> >>
> >>
> >>
> >>
> >>Any help would be much appreciated.
> >>
> >>Regards,
> >>
> >>Rhys Hopkins
> >>
> >>Systems Administrator
> >>Culver Technologies Limited.
> >>
> >>
> >>
> >
> >
> > ___
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> >
> >
>
>
>
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Re: [Asterisk-Users] Ansering machine/voicemail detect?

[Andy Powell]

Garry,

yes this is possible although it would end up being quite convoluted.

Essentially you could have a cron job that monitors your voicemail directory, or use 
the perl manager interface to check the status. Once it has been established that you 
have message(s) submit a .call file to dial you office number, and start an AGI 
script. In the AGI you ask for your pin number and time out (and hang up) if there is 
no response. If there is a correct response then you could playback the files. You'd 
need to move them to the OLD directory afterwards.

Alternatively, someone has written app_hasvoicemail which checks a vm box for messages 
and acts on that...

Andy



On 07/08/2003 at 13:17 Garry Adkins wrote:

>I've been playing with an asterisk box for about 6 months, (bought an FXO
>card, etc.)...
>
>I was thinking about having the system "deliver" my voicemail from the
>asterisk machine to me at work...  I haven't found anything in the
>documentation to help.
>
>
>It would work something like:
>
>
>Voicemail comes into the asterisk machine,
>* Calls me at work
>Plays message for me to enter PIN for voicemail
>Retrieve Voicemail
>Hangup.
>
>
>However, if it got my voicemail at work (due to being on the phone or out
>of the office), I'd like it to do something like:
>Voicemail in *
>* Calls me at work
>Notices that it's voicemail (Possibly due to no pause at the beginning,
>just continuous talking?)
>Just plays a message that I have voicemail at home.
>Hangs up.
>
>Possible?  How could it tell that it got an answering device?
>
>Thanks!
>-G
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Re: [Asterisk-Users] AgentCallbackLogin

[Jim Friedeck]

I am trying to record all logins and logouts for posterity. The act of 
logging in with AgentCallbackLogin produces the same CDR data as logging 
out. Is there some way to distunguish them in CDR? I also noticed the 
management interface maintains a Unique ID for each call and lets that 
call be traced throughout its life in the PBX. Can that data be added to 
CDR as well to allow for easier call tracking? Should I do it with an 
AGI? We plan to use a commercial report generator attached to MySQL to 
create logs for management at our company. This added field would 
greatly improve the ease with which reports are generated. Thanks!

Jim Friedeck

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Re: [Asterisk-Users] New SIP Phone

[Steve Meyers]

On Wed, 2003-08-06 at 16:20, Andy Powell wrote:
> It's just a proxy service like fwd it will work with asterisk... The phones they are 
> selling
> with the deal are Grandstreams. 

Perhaps that explains why nobody can get to the site to order
Grandstreams right now. :)
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[Asterisk-Users] Sip Trunk config

[David Hindmarsh]

Hi

Is it possible to use a sip gateway as a trunk.

If so,  how would I do this

David Hindmarsh

- Original Message - 
From: "Jamie Carl" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, August 07, 2003 12:14 PM
Subject: Re: [Asterisk-Users] X-Lite <-> Snom200


> Yes, over a LAN.  It does it with both g.711 and GSM which 
> both used to work.  Havn't had a chance to have a REAL 
> good look into it though.
> 
> J
> 
> On Wed, 06 Aug 2003 14:33:47 +
>   "WipeOut ." <[EMAIL PROTECTED]> wrote:
> >*This message was transferred with a trial version of 
> >CommuniGate(tm) Pro*
> >> *This message was transferred with a trial version of 
> >>CommuniGate(tm) Pro*
> >> Dunno what I'm doing wrong here but I just did an 
> >>upgrade to the latest
> >> version and now I get no audio at all!
> >> I havn't changed a single thing.  Is there anything 
> >>special I need to do
> >> to get this to work again?
> >> 
> >> I get a quick 'chirp' of audio, which you can tell is 
> >>what I'm
> >> connecting to, (ie MOH), but then nothing.
> >> 
> >> 
> >> Regards,
> >> 
> >> Jamie Carl
> >> Email:  [EMAIL PROTECTED] 
> >> Phone:  +61 414 365 466
> >> Jabber: [EMAIL PROTECTED]
> >> 
> >
> >Are you connecting to * over a LAN?? I have experienced 
> >the "chirp" when the phone was trying to use G.711 over a 
> >dial up link so there was not enough bandwidth..
> >
> >
> >-- 
> >__
> >http://www.linuxmail.org/
> >Now with e-mail forwarding for only US$5.95/yr
> >
> >Powered by Outblaze
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> 
> Regards,
> 
> Jamie Carl
> Jazz Inc.
> Email:  [EMAIL PROTECTED]
> Web:www.jazz-inc.net
> Phone:  +61-414-365-466
> Jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?

[Martin Stubbs]

Hi Dave,

On Tuesday 05 August 2003 13:53, Dave Wilson wrote:
> Hi all,
>
> I can't seem to find any info on this anywhere on the web, except that BT
> caller ID doesnt use the standard bellcore system in use in the US. So, if
> anyone here in the UK is onlist and using the x100p successfully, please
> let me know.
>
>

Unfortunately the present x100p driver code will not decode the callerid for 2 
reasons

1) the UK protocol is different to the US system. 
I have downloaded the specs and coding it would not be too difficult.

2) in the US the callerid is sent between the first and second rings. In the 
UK the callerid is sent before the first ring. 
I have been unable to determine if we can get a zaptel event when a line 
reversal is received which happens before the UK callerid is sent. Without 
this function or continuously monitoring the line for the tones we don't know 
when to enter the callerid routine.  

Apart from the callerid the x100p works well in the UK but without callerid I 
can't implement lots of things I would like to do.


Martin



> TIA,
> Dave
>
>
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[Asterisk-Users] Festival 1.4.3

[Brian West]

Anyone have any luck setting up festival 1.4.3?  Can someone share some
input.

bkw
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Re: [Asterisk-Users] AgentCallbackLogin

[Armand A. Verstappen]

> out. Is there some way to distunguish them in CDR? I also noticed the 
> management interface maintains a Unique ID for each call and lets that 
> call be traced throughout its life in the PBX. Can that data be added to 
> CDR as well to allow for easier call tracking? 

It looks like if you define MYSQL_LOGUNIQUEID in the top of
cdr/cdr_mysql.c and recompile, it will start logging the unique id you
want.

wkr,

-- 
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[EMAIL PROTECTED]   3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht


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[Asterisk-Users] Standard Analoge modem - can it be used?

[Asterisk - linux - JVB]

All,

Could one use the standard analoge modem to test Asterisk functionality. 
I mean just the phone and line jack OR do I need specific hardware?

Thanks in advance

Jeroen

-

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Re: [Asterisk-Users] AgentCallbackLogin

[Jim Friedeck]

I don't think that's the same unique id. It changes for each record in 
the CDR. I believe the management interface unique id is maintained as 
specific to each incoming or 'original' call. Any ideas?

Jim Friedeck

---

Jim Friedeck wrote:

Thanks! I'm trying that now.

Jim Friedeck



Armand A. Verstappen wrote:

out. Is there some way to distunguish them in CDR? I also noticed 
the management interface maintains a Unique ID for each call and 
lets that call be traced throughout its life in the PBX. Can that 
data be added to CDR as well to allow for easier call tracking?   


It looks like if you define MYSQL_LOGUNIQUEID in the top of
cdr/cdr_mysql.c and recompile, it will start logging the unique id you
want.
wkr,

 

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Re: [Asterisk-Users] Unregister SIP connection?

[Martin Pycko]

use "extensions reload" CLI command

Martin

On Wed, 6 Aug 2003, Steven J. Sobol wrote:

>
> Is there a way to make * forget that SIP phone
> [EMAIL PROTECTED] is registered? I ask because I have a few
> different PSTN numbers that I use for various reasons, and I can reprogram
> my Grandstream, but unless I also restart *, calls to the
> originally-registered number still ring through, and calls to the number
> that I am trying to switch to do not work.
>
> --
> JustThe.net Internet & Multimedia Services
> 22674 Motnocab Road * Apple Valley, CA 92307-1950
> Steve Sobol, Proprietor
> 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED]
>
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Re: [Asterisk-Users] 3xx SIP messages

[Mark Spencer]

He should treat the first part as a local extension.

amark

On Thu, 7 Aug 2003, Michael Ulitskiy wrote:

> Hi,
>
> Does anyone know if asterisk can handle 3xx SIP responces?
> I'm trying make it work with redirect server and it looks like
> asterisk isn't going to send another invite, but treats "302 Moved
> Temporarily" message as "Everyone is busy".
> Thanks.
>
> Michael
>
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Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers

[WipeOut .]

> could you send me the exact syntax for rxgain / txgain?
> I think that might help towards my problem
> becuase i'm having to turn the handset volume all the
> way up ..
> 
> thanks
> Dave

You can use either a percentage or a number IIRC..

Somthing like..

rxgain=5%
txgain=5%

or

rxgain=0.4
txgain=0.4

and I thing that you can use negative values as well..

I am not sure what the minimum and maximum values are I use percntages..

Hope that helps..
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Re: [Asterisk-Users] iax.conf / Registration rejected

[Peer Oliver schmidt]

Dan wrote:
add in the [pos] section :
username=pos
and then if it still doesn't work try to comment the lines:
;deny=0.0.0.0/0.0.0.0
;permit=10.1.3.0/255.255.255.0
;defaultip=10.1.3.2
Did all that, even restarted asterisk (instead of reload via CLI), still
no go. Any other idea?
rgds
pos
I am trying to use the Windows iax client.

My iax.conf looks like this:

[general]
port=5036
bindaddr=10.1.3.111
bandwidth=high
allow=gsm   ; Always allow GSM, it's cool :)
tos=lowdelay
[pos|
type=friend
context=default
auth=plaintext
secret=pos
deny=0.0.0.0/0.0.0.0
permit=10.1.3.0/255.255.255.0
host=dynamic
defaultip=10.1.3.2
In the registrations dialog of the Windows clients I put

pos:[EMAIL PROTECTED]

The * console shows the following messages.

Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: REGREQ
Tx-Frame Retry[-01] -- Seqno: 00  Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: REGREJ
Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: ACK




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RE: [Asterisk-Users] [OT] unsubscribe

[Andy Powell]

Steve

I have to say that some listserv's do allow this .. at least he didn't reply to 20 
messages with

REMOVE

in them

Andy


On 07/08/2003 at 10:10 Steve Meyers wrote:

>On Thu, 2003-08-07 at 10:01, Justin Carlson wrote:
>> unsubscribe
>
>Has anyone ever been on a mailing list where you could unsubscribe
>simply by sending a message with "unsubscribe" in it to the mailing
>list?  I swear, every list I've been on, people try to do that, but it
>doesn't work on any of them.
>
>Steve
>
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[Asterisk-Users] MWI bug ?

[Lee Goodman]

Hi
I don't know if this is a bug or a configuration issue. I have Cisco 7960
phones off of my *.
Voicemail works fine, but I can't get the MWI light to work correctly. I
have the phones deposit the VM in a VM directory that is not the default one
(directory called "sip"). If I push VM files into the VM directory called
"default" the MWI light works fine.

Is there a bug that the MWI part of the Voicemail app only looks in the
directory called "default", or is there a way to make MWI look in another VM
directory.

thanks

lee goodman


voicemail.conf
[general]
format=wav
maxmessage=180
[sip]
1000 => 1000,LG,[EMAIL PROTECTED]
1001 => 1001,TG,[EMAIL PROTECTED]
1002 => 1002,BG,[EMAIL PROTECTED]

extensions.conf

[incoming]
exten => s,1,Background(goodmanmenu)
exten => s,2,DigitTimeout,5
exten => s,3,responsetimeout,10
exten => 1000,1,Goto(sip,1000,1)
exten => 1001,1,Goto(sip,1001,1)
exten => 1002,1,Goto(sip,1002,1)
exten => 8500,1,Goto(sip,8500,1)
[sip]
exten => _X.,1,Dial,Zap/1/${EXTEN}
exten => 8500,1,Wait,1
exten => 8500,2,VoiceMailMain2(${CALLERIDNUM})
exten => 8500,3,Hangup
exten => 1000,1,Dial(SIP/1000,20)
exten => 1000,2,Voicemail2(u1000)
exten => 1000,102,Voicemail2(b1000)
exten => 1000,103,Hangup
exten => 1001,1,Dial(SIP/1001,20)
exten => 1001,2,Voicemail2(u1001)
exten => 1001,102,Voicemail2(b1001)
exten => 1001,103,Hangup
exten => 1002,1,Dial(SIP/1002,20)
exten => 1002,2,Voicemail2(u1002)
exten => 1002,102,Voicemail2(b1002)
exten => 1002,103,Hangup

sip.conf

[1000]
type=friend
insecure=np
username=1000
secret=secret
nat=no
host=dynamic
canreinvite=no
context=sip
mailbox=1000



[1001]
type=friend
insecure=no
username=1001
secret=secret
nat=no
host=dynamic
canreinvite=no
context=sip
mailbox=1001

[1002]
type=friend
insecure=no
username=1002
secret=secret
nat=no
host=dynamic
canreinvite=no
context=sip
mailbox=1002


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[Asterisk-Users] Processor Consuption

[Nathan Littlepage]

Since the DSP is software driven for the Wildcard product. Is there a
benchmark out that depicts how much processor is utilized on TDM calls
per codec that's being used?

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Re: [Asterisk-Users] MWI bug ?

[Lee Goodman]

Thanks, that fixed it.

Now, the problem would be if you wanted to keep voicemail in different
directories for a mulit tenant asterisk server. Anyway of making MWI use a
different directory (on a per user basis)?

Lee Goodman


- Original Message -
From: "Brian West" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, August 07, 2003 10:56 AM
Subject: Re: [Asterisk-Users] MWI bug ?


> put your mailboxes under [default] that you want mwi on... I made that
> mistake also.
>
> bkw
>
> On Thu, 7 Aug 2003, Lee Goodman wrote:
>
> > Hi
> > I don't know if this is a bug or a configuration issue. I have Cisco
7960
> > phones off of my *.
> > Voicemail works fine, but I can't get the MWI light to work correctly. I
> > have the phones deposit the VM in a VM directory that is not the default
one
> > (directory called "sip"). If I push VM files into the VM directory
called
> > "default" the MWI light works fine.
> >
> > Is there a bug that the MWI part of the Voicemail app only looks in the
> > directory called "default", or is there a way to make MWI look in
another VM
> > directory.
> >
> > thanks
> >
> > lee goodman
> >
> >
> > voicemail.conf
> > [general]
> > format=wav
> > maxmessage=180
> > [sip]
> > 1000 => 1000,LG,[EMAIL PROTECTED]
> > 1001 => 1001,TG,[EMAIL PROTECTED]
> > 1002 => 1002,BG,[EMAIL PROTECTED]
> >
> > extensions.conf
> >
> > [incoming]
> > exten => s,1,Background(goodmanmenu)
> > exten => s,2,DigitTimeout,5
> > exten => s,3,responsetimeout,10
> > exten => 1000,1,Goto(sip,1000,1)
> > exten => 1001,1,Goto(sip,1001,1)
> > exten => 1002,1,Goto(sip,1002,1)
> > exten => 8500,1,Goto(sip,8500,1)
> > [sip]
> > exten => _X.,1,Dial,Zap/1/${EXTEN}
> > exten => 8500,1,Wait,1
> > exten => 8500,2,VoiceMailMain2(${CALLERIDNUM})
> > exten => 8500,3,Hangup
> > exten => 1000,1,Dial(SIP/1000,20)
> > exten => 1000,2,Voicemail2(u1000)
> > exten => 1000,102,Voicemail2(b1000)
> > exten => 1000,103,Hangup
> > exten => 1001,1,Dial(SIP/1001,20)
> > exten => 1001,2,Voicemail2(u1001)
> > exten => 1001,102,Voicemail2(b1001)
> > exten => 1001,103,Hangup
> > exten => 1002,1,Dial(SIP/1002,20)
> > exten => 1002,2,Voicemail2(u1002)
> > exten => 1002,102,Voicemail2(b1002)
> > exten => 1002,103,Hangup
> >
> > sip.conf
> >
> > [1000]
> > type=friend
> > insecure=np
> > username=1000
> > secret=secret
> > nat=no
> > host=dynamic
> > canreinvite=no
> > context=sip
> > mailbox=1000
> >
> >
> >
> > [1001]
> > type=friend
> > insecure=no
> > username=1001
> > secret=secret
> > nat=no
> > host=dynamic
> > canreinvite=no
> > context=sip
> > mailbox=1001
> >
> > [1002]
> > type=friend
> > insecure=no
> > username=1002
> > secret=secret
> > nat=no
> > host=dynamic
> > canreinvite=no
> > context=sip
> > mailbox=1002
> >
> >
> > ___
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> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> ___
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RE: [Asterisk-Users] [OT] unsubscribe

[Steve Meyers]

On Thu, 2003-08-07 at 10:01, Justin Carlson wrote:
> unsubscribe

Has anyone ever been on a mailing list where you could unsubscribe
simply by sending a message with "unsubscribe" in it to the mailing
list?  I swear, every list I've been on, people try to do that, but it
doesn't work on any of them.

Steve

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Re: [Asterisk-Users] Problem with the Internet LineJACK ISA card...

[Andy Powell]

You also appear to have a big problem with your clock... 

unless you are from the future.. in which case how are Glaxo stocks doing?

Andy


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Re: [Asterisk-Users] ADSI and SoftKeys

[Jayson Vantuyl]

>  I've taken the liberty to edit your patch, to put back in the
> 'adsi_logo' and the values for adapp and adsec as they are in CVS. As
> far as I can tell those changes have no relation to problem this patch
> solves, they're just local changes to satisfy your local preferences,
> right? I've removed those to ease integration into CVS.

Ooops.  Those were actually the security code to unlock our Aastra
phones.

Please disregard that.

Jayson

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[Asterisk-Users] Vonage ATA 186 Factory Default & use with Asterisk ?

[John Schmerold]

I've canceled my Vonage service because of the requirement to prefix 
every call with a 1.

Vonage has charged me $42 & will refund this when they get my ATA186 
back.  I'm thinking I should keep the 186 for possible use with Asterisk.

Anyone know if this will work out?

As I understand it, the procedure to set the ATA186 back to factory 
defaults is as follows:
Press the function button on the Cisco ATA.
Press the digits 322873738 (FACTRESET) then # on your telephone keypad.
Press 3 on your telephone keypad to confirm ATA reset & hang up phone

TIA

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[Asterisk-Users] Warning Messages

[surajee]

hi,

i have connected a SNOM 200 to the asterisk. here are my settings,

Codecs
---

Default codec - g.711u/g.711a
Packet size - 20ms
Negotiation - Interoperable
Type - 160

DTMF


Inband - negotiate
Outband - negotiate
Payload Type - 101

when a call comes to the SNOM or when making an outdial, following warning
messages are coming on asteisk,

WARNING[1209214400]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 
2 frames
WARNING[1209214400]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 
2 frames
WARNING[1209214400]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 
2 frames
...
WARNING[1209214400]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 
2 frames
WARNING[1209214400]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 
2 frames

these warning messages come a lot, but still u can have a normal voice
conversation.

but this warning messages are very irritative..

does anybody has an idea on this?

Thanks inadvance,
Surajee

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Re: [Asterisk-Users] Leftover Budgettone issues

[Steve Meyers]

On Thu, 2003-08-07 at 01:56, Brian Capouch wrote:
> 2. This phone does not act like all my others do when I am talking and a 
> call comes in.  Instead of the jarring ADSI !!!BOING!!! followed by a 
> series of call waiting beeps, instead I get a ringing tone in the 
> earpiece which is audible to the other party as well.

If you find out, please let me know!  I've tried all sorts of settings
to make it stop that.  I'd like to just make it not support call waiting
at all on the SIP connection, that would be easiest, but I can't find a
way to do it.  The BudgeTone configuration doesn't seem to be able to
turn this off, either.

Hopefully they'll fix this soon...

Steve
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[Asterisk-Users] T-shirt ideas

[Mark Spencer]

Digium is planning to make some Asterisk/Digium t-shirts.  We'd like to
have people submit t-shirt designs from which we might select one in
addition to whatever we might come up with on our own.

1) The t-shirt should be primarily for Asterisk but should contain the
Digium logo somewhere, too.

2) Designs will need to be disclaimed.

Just send your design ideas to [EMAIL PROTECTED] and we'll make a web
site with the ones we like to get some feedback.

Mark

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Re: [Asterisk-Users] Leftover Budgettone issues

[WipeOut .]

> I have my new phone mostly working.  I do have a couple of residuals 
> that I cannot find mentioned in the list archives:
> 
> 1. Is it possible to set the volume in these things?  I hope I didn't 
> miss it, but I've looked in the doc, the FAQ, and the asterisk archives 
> and don't find anything.  The displays in the pictures all have more 
> bars on them than my phone does, and I need a bit more volume than I'm 
> getting.

With the hand set off hook (or with speakerphone active) you adjust the volume using 
the blue up and down arrow buttons on the left just below the screen..

> 
> 2. This phone does not act like all my others do when I am talking and a 
> call comes in.  Instead of the jarring ADSI !!!BOING!!! followed by a 
> series of call waiting beeps, instead I get a ringing tone in the 
> earpiece which is audible to the other party as well.
> 
> Maybe there is some config setting that will handle that?

Don't know about that one.. :)

> 
> Thanks in advance for any information that might be out there.
> 
> B.
> 
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Re: [Asterisk-Users] Newbie just starting out with *

[John Schmerold]

Sorry about taking this OT

How do you go about "you could write a quick LDAP->XML dump with perl"

Please just point me to a howto.

TIA

John Todd wrote:
Nice...so mixing and matching IP and POTS is ok and common then?


Yes.

I do care what the IP phones cost but then again I'm a gadget 
freak..if it has a big enough "cool" factor I may be able to justify 
price. What exactly does the Cisco phone get me over a Grandstream? I 
have an NBX100 (3com voip) at work and while it has its nify features 
3com really dropped the ball on some of the features. Like 2K just to 
make it a true IP phone and then it only works with NT and not a H.323 
gateway..but thats a whole other thing that irks me.


A 79xx Cisco phone is (IMHO) higher quality and has many more features, 
including programmability with the large LCD screen.  The ATA-186 is 
simply an analog-to-ethernet adapter, and you'll need your own phone.  
However, it still has some nice features and has seen much more testing 
than the Grandstream.  The Grandstream is cheap, but I prefer my nice 
Nortel phones or vtech wireless phones on my ATA-186, or the 7960 I use 
as my primary phone.

The one major advantage for a power user on the 7960 is that it has six 
line appearances, which makes inbound/outbound calling with identity 
differentiation possible.

Can any of the IP phones play a wav file as a ring tone? What about 
using an LDAP directory for dialing? I'd LOVE that and that may even 
increase the "wife factor" too hehehe.


- Yes, the Cisco can play converted .wav files as ringtones (not raw 
.wav files, but minor effort required)

- LDAP for dialing: not easily.  Cisco 7960 supports xml directories for 
dialing, though, and it pulls that off a web server, so you could write 
a quick LDAP->XML dump with perl.

JT




WipeOut . wrote:

My .02c...(Purely my own opinions)..

Get the X100P and if you need 4 analog ports get the TDM400P..

I use an X100P and a S100U(2 analog phone cordless system) and then 4 
IP phones..

Your choice of IP phones are..

"I don't care what it costs" - Go for Cisco..

"I do care what it costs but still want something stylish and feature 
packed" - Go for Snom200..

"Cost is very important cos the budget is tight but I still want a 
good usable phone" - Go for Grandstream 101 or 102..

I have 2 Snom200's and 2 GS 102's..

Hope that helps.. shout if you have more questions..

Just a note.. IIRC fax machines sometimes have some issues..

Later..



Hey all...I'm brand new to * and I want to convert my home into a pbx
type setup. I've figured out that I want a Wildcard X100P to bring my
single POTS CO into my Linux box. My problem is that I'm sure sure what
I need to do to get my analog phones connected up into a structured
phone system. It *looks* like I can go the route of the Cisco Analog ->
VOIP for about $100 on ebay. That will get me two analog devices on the
system. If I have four analog devices (2 normal phones, 1 fax and one 4
phone cordless system) is this the best setup? Do I need the TDM10B 
with
the Asterisk TDM Dev Kit or does that just let me do one analog phone
into the system? When converting from analog to VOIP do I get all the
same features that I would if I got a TDM400P (4 ports of analog 
devices)?

As I said I'm new and I would LOVE any pointers, HOWTOs or any good
advice from people who have already done something similar. This 
project
started out because I'm tired of the telemarketers calling and it looks
like this will be the best and most flexable way to get my phone system
wired up. I'm interested in any opinions on any real VOIP phones for a
house (assuming VOIP) is the way to go.

I envision that I could have a phone in every room, be able to do an
intercom, MOH so I can hear music in each room etcideas?
Thanks for the help and your patience,
Chris
--
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Re: [Asterisk-Users] Musiconhold interrupted sound

[Michael Manousos]

Michael Ulitskiy wrote:
Michael,

With all due respect to both of you, it's not related to h.323 driver.
The result is the same whether h.323 channel participates in the 
call or it's pure sip-to-sip call. 
Did you try it without the ztdummy and zaprtc?
I posted to Mark a couple of patches fixing MOH and
prompt playback on machines without zaptel hardware.
They are in CVS since end of July.
I have no problem with H.323, SIP and Quicknet devices.
Michael.

Michael

On Monday 04 August 2003 01:59 pm, Michael Manousos wrote:

Use asterisk-oh323. It works great!

Michael.

Michael Ulitskiy wrote:

Hi again,

Am I really the only one who's having this problem?
Music on hold playing like this is very annoying and 
practically unusable. 
One more detail. The sound interruption happens only
when 2 endpoints are actually connected and one of them
put the other on hold. If I set a special extension with
SetMusicOnHold application it seems to play just fine.
Please help!
Thank you.

Michael

On Friday 01 August 2003 04:58 pm, Michael Ulitskiy wrote:


Hi,

I don't seem to be able to get music on hold to play normally.
The sound gets often interrupted with a few seconds of silence
then starts playing again. I'm using mpg123-0.59r and tried
mp3 files with different sample rates with no luck. If that matters, 
endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323)
Quintum Tenor.
Sometimes it may play fine for a few minutes then getting worse 
again. I've noticed that the following messages look directly
connected to sound interruptions:
on asterisk console:
NOTICE[245776]: File rtp.c, Line 239 (process_rfc3389): RFC3389 support incomplete.  Turn off on client if possible
in debug log:
Aug  1 16:43:58 DEBUG[32771]: File res_musiconhold.c, Line 292 (monmp3thread): Only wrote -1 of 640 bytes to pipe
The music stops playing simultenuously with these messages appearing.
Also I've tried to install ztdummy and zaprtc drivers (no zaptel hardware installed). 
None of them helped.
Asterisk version is CVS-07/30/03-18:53:16
If anybody has any idea on how to fix it or what can be done to further 
troubleshoot it, I'd appreciate hearing from you.
Thanks.

Michael

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Re: [Asterisk-Users] DTMF modes and external IVR systems over ISDN

[James Sizemore]

Tones are to short.

Stefano Finetti wrote:

Mark,

I'm now able to send proper DTMF tones checking on the isdn driver and using
"rfc2833" as dtmf mode for sip.conf and phones.
But there is a question that i think only you can check and answer:

Why * often when calling via outside line some number that has IVR systems,
doesn't recognize the answer?
It stays there, waiting, even if i'm sure the other side of the line has
answered the call (tried in the same time from * and using my mobile phone).
I can't figure out what kind of problem can be, I encountered it in many *
installation...
--
Stefano Finetti
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Re: [Asterisk-Users] Vonage ATA 186 Factory Default & use with Asterisk?

[Bruce Ferrell]

That answers one question for me.  I won't be buying any of those vonage 
ATA-186s being advertised on e-bay

Brian Capouch wrote:
John Schmerold wrote:

I've canceled my Vonage service because of the requirement to prefix 
every call with a 1.

Vonage has charged me $42 & will refund this when they get my ATA186 
back.  I'm thinking I should keep the 186 for possible use with Asterisk.

Anyone know if this will work out?

As I understand it, the procedure to set the ATA186 back to factory 
defaults is as follows:
Press the function button on the Cisco ATA.
Press the digits 322873738 (FACTRESET) then # on your telephone keypad.
Press 3 on your telephone keypad to confirm ATA reset & hang up phone

Not any more.

Vonage got Cisco to include a "password protect the config" in the 
latest version of the firmware, and as far as I know now all the Vonage 
ATAs are forever destined to be used with Vonage and only Vonage.

I have stridently communicated my concern in this matter to them (I have 
been a Vonage customer almost from Day One, and I spend quite a bit with 
them) but they seem resolute that (as far as I know) even though nowhere 
do they tell you that they maintain ownership of the ATA interface in 
perpetuity, that is in fact in practice what all of this means.

I object most stridently to this on "green" grounds: they are purposely 
setting things up so that perfectly good hardware is only fit for the 
dustbin.  I know there are arguments against this perspective, but of 
all the things Vonage does, this one I find to be reprehensible and 
damaging to their ultimate interests.

Time will tell, of course.

B.

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