Re: [Asterisk-Users] Throroughly confused about SetCallerID

[beonice]

--- Dan Journo <[EMAIL PROTECTED]> wrote:

> Hi Beonice,
>  Just told i was dealing with the same problem.
>  [incoming]
> exten => _!,1,GotoIf($["${CALLERID}" =
> "unknown"]?2:5)
> exten => _!,2,Set(CALLERID(name)=Withheld Number)
> exten => _!,3,Set(CALLERID(number)=00)
> exten => _!,4,Goto(8)
> exten => _!,5,GotoIf($["${CALLERID}" =
> "asterisk"]?2)
> exten => _!,6,Set(CALLERID(name)=${CALLERID})
> exten => _!,7,Set(CALLERID(number)=${CALLERID})
> exten => _!,8,Dial(SIP/${EXTEN})
>  Maybe you can use that to solve your problem.
> With regards to voicemail etc, i cant help you with
> that.
>  Take care
> Dan
> www.sippal.com <http://www.sippal.com>


Thanks, Dan. I figured out that I needed to use the
GotoIf variety just as soon as I finished posting. It
does work great now. I guess it works the same on all
incoming calls, whether or not it gets sent off to
voicemail after that is irrelevant.

Thanks again!

Maya





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Re: [Asterisk-Users] Throroughly confused about SetCallerID

[beonice]

I forgot to state that this is only for INCOMING
calls. I'm not making outgoing calls, so I really
don't care what the outgoing caller id is. 

I'm running Asterisk 1.0.5 stable ... it's a
production environment, and the users are getting
really confused about the caller id strings on their
incoming voicemail notifications and voicemail
envelopes!

Cheers,
Maya


--- beonice <[EMAIL PROTECTED]> wrote:

> Folks,
> 
> I've been trying to handle the problem where 
> blocked callerids appear as coming from
> asterisk 
> on the email notification, and the message 
> envelope simply doesn't say anything (does not
> actually play the vm-unknown message).
> 
> So, following the tip provided by several
> previous posters, I tried putting this in my
> extensions.conf (the xx's are my DID, obviously:
> 
> exten => xxx,1,SetCallerID();
> exten => xxx,2,Goto(veeport_context,s,1);
> 
> Okay. This works find for BLOCKED callerids. 
> But the problem is that now all callerids are
> being treated as blocked, not just the really 
> blocked ones!Help.
> 
> What I want to do is:
>   For calls that don't have blocked caller id, 
> I want asterisk to do what it does normally: tell
> me the call is from phonenumber xx
>   For calls that have a blocked callerid, I want
> asterisk to indicate that the callerid is unknown,
> both in the email and in the message envelope on
> the actual voicemail.
> 
> Is there any way to achieve both these phenomena?
> I tried editing chan_sip.c to change the
> default_callerid there, but that causes its own set
> of
> problems.
> 
> Any help would be much appreciated!
> 
> Thanks,
> Maya
> 
> 
> 
> 
> 
>   
>   
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[Asterisk-Users] Throroughly confused about SetCallerID

[beonice]

Folks,

I've been trying to handle the problem where 
blocked callerids appear as coming from
asterisk 
on the email notification, and the message 
envelope simply doesn't say anything (does not
actually play the vm-unknown message).

So, following the tip provided by several
previous posters, I tried putting this in my
extensions.conf (the xx's are my DID, obviously:

exten => xxx,1,SetCallerID();
exten => xxx,2,Goto(veeport_context,s,1);

Okay. This works find for BLOCKED callerids. 
But the problem is that now all callerids are
being treated as blocked, not just the really 
blocked ones!Help.

What I want to do is:
  For calls that don't have blocked caller id, 
I want asterisk to do what it does normally: tell
me the call is from phonenumber xx
  For calls that have a blocked callerid, I want
asterisk to indicate that the callerid is unknown,
both in the email and in the message envelope on
the actual voicemail.

Is there any way to achieve both these phenomena?
I tried editing chan_sip.c to change the
default_callerid there, but that causes its own set of
problems.

Any help would be much appreciated!

Thanks,
Maya







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[Asterisk-Users] Unknown or blocked ID now shows up as "asterisk"

[beonice]

Hello, all. 

This is rather annoying. I know that this _used_ to
work in my production system.

I'm running Asterisk 1.0.5 and am using IAX
exclusively. 

Used to be, when someone called in from a
caller-id-blocked number, the notification email said:
"From an unknown caller", and the message envelope
when I listened to it also said "From an unknown
caller".

Something seems to have changed. I have NOT upgraded
asterisk versions, but I did recompile asterisk
recently.  Did a full make clean; make; make install
thing. 

Any clues? The stuff I'm seeing on previous
discussions talks about editing the conf files to
recognise blocked callers, but I don't think I did
that before, so what's likely to have changed?

I'm not seeing the forest for the trees, obviously.

Any help would be appreciated!

Thanks,
Maya



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Re: [Asterisk-Users] asterisk showing more than once on ps

[beonice]

--- [EMAIL PROTECTED] wrote:

> Actually this is on an enterprise system - I have
> never seen more  
> than one. Just checked now and there were several
> calls up and only  
> one ps.
> 
> 
> On Jul 1, 2005, at 8:33 PM, Luki wrote:
> 
> >> Do not know why, but have noticed redhat = 1, and
> debian = many
> >>
> > Not quite. RedHat Enterprise also = many at times,
> depending on number
> > of concurrent calls; usually one when idle. Maybe
> it has something to
> > do with kernel 2.4 vs 2.6 and how threads show up
> in ps.
> >
> > --Luki
> > ___
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What I've noticed (running the auto-start script on
White-Box Enterprise Linux ... basically the same as
RedHat Enterprise Linux of some vintage) is that when
the auto-start executes safe_asterisk, I see multiple
instances of asterisk under ps. However, if I use
"asterisk -r" to connect, then stop and restart
safe_asterisk manually, ps drops to showing exactly 1
instance of asterisk running. 

For what it's worth ...
Maya




 
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Re: [Asterisk-Users] Unable to connect to remote asterisk

[beonice]

Jason, when I've seen this, it's usually meant that I
was logged in as a user who didn't have the right
permissions. :)

Check your notes to figure out which user you gave the
permission to (root, maybe?) and log in as that user,
then try to connect using "asterisk -r".

Hope that helps,
Maya


--- Jason Greene <[EMAIL PROTECTED]> wrote:

> Hello,
>   I'm trying to figure out why the asterisk service
> starts fine, but 
> when i try to connect by typing asterisk -r I get:
> Unable to connect to remote asterisk
> 
> The service is running and lists under ps -ef as:
> asterisk -vvvg -c
> 
> any help is appreciated
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Re: [Asterisk-Users] Asterisk server with remote monitoring capabilities

[beonice]

--- Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

> Hi
> 
> On Thu, Jun 23, 2005 at 05:52:44PM -0700, beonice
> wrote:
> > 
> > Okay, so what makes more sense:
> >   1) a remote management card that will let me
> > actually log in to the machine to monitor it as
> well
> > as to reboot it
> > vs.
> >   2) a remote-accessible powerstrip that will
> allow me
> > to remotely reboot the server?
> 
> Linux also has a software watchdog module. Maybe it
> could work here
> without extra hardware to set up.
> 
> > 
> > I'm realising that sometimes the problem may
> simply be
> > processes out of control, and may be something
> that
> > doesn't require killing the entire machine, but
> just
> > some processes. 
> > 
> > In my current setup (an ordinary PIII 1.someting
> GHz
> > machine, not a server-class machine), when a
> process
> > goes haywire, I lose remote access via SSH, so I
> drive
> > to the colo, log in, sigh in frustration, and
> reboot
> > "because I'm already here, so why not?".
> 
> Because you destroy any evidence of the problem.
> 
> What processes are taking much CPU time? Are there
> any relevant log
> messages? Is this a case of over-swapping? (not 100%
> CPU usage, but
> rather large swap usage, CPU spends too much time at
> "system", though 
> the latter may be probably normal for an Asterisk
> server).
> 
> Could you login from the console? Did you manage to
> move between virtual
> consoles?
> 
> Install the package sysstat and run sar to get some
> stats. Consider
> adding a cron job to gather more relevant stats
> every 5 minutes or every
> minute.
> 
> BTW: does asterisk run with real-time priority? try
> removing it, so at
> least asterisk won't hang the whole system. Though I
> doubt it if this
> would help.
> 
> > 
> > Some of the problems were caused by my old router
> ...
> > since I replaced it, the need to drive the 40
> miles
> > each way has gone down significantly ... in fact,
> to
> > pretty much zero. So I have time to contemplate my
> > options here. :)
> 
> You have an extra router there? I recall that there
> was a kernel patch
> to reboot the system upon recieving a specific ICMP
> packet.
> 
> -- 
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Thanks, everyone, for all the responses and
suggestions. I already do have remote access via SSH.
The problem is that, occasionally, the box simply
won't let me SSH in, so I have to drive over and
reboot.

Several of those times, it turned out to be simply the
router acting up, so power-cycling the router would
bring things back to life, including SSH. As I
mentioned, the router has now been replaced and the
new one seems a lot more stable.

However, on a _few_ occasions, when I went into the
colo, it turned out that the box itself was not
responding to input, even on the console. At those
times, the ONLY thing I could do was a hard reboot ...
and yes, I'm aware of the potential hazards involved
in a hard reboot. :)

The frustrating thing is that since I upgraded the
router, the box hasn't crashed at all ... now I'm
thinking back and wondering if I imagined those
occasions. :)

Based on the suggestions provided by all of you, I
think I'm definitely going to try to get better stats
on what exactly is going on (thanks for the tip that
there are production Asterisk servers with
months/years uptime ... that was an eye-opener!)
before I invest in any new hardware.

Thanks again, everyone! 

Cheers,
Maya




 
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Re: [Asterisk-Users] Asterisk server with remote monitoring capabilities

[beonice]

--- Michael Welter <[EMAIL PROTECTED]> wrote:

> William Boehlke wrote:
> > Dell sells a remote management card for under $400
> that enables remote
> > reboots. I know there are others out there but
> have no experience with them.
> > 
> >  
> > William Boehlke
> > Signate
> > 
> > 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED]
> On Behalf Of beonice
> > 
> > I'm tired of having to drive out to the colocation
> facility each time my
> > dedicated asterisk server craps out, just to press
> the button to do a hard
> > reboot.
> > (I'm running 1.05 stable at present, no telephony
> hardware, as this is
> > mainly a system that receives calls, no dial-out
> ability is needed.) 
> > 
> APC makes a power strip with a web server.  Each
> socket has its own IP 
> address.  All you have to do to power cycle is
> access the IP address 
> from your web browser and give the power cycle
> command.  It is sooo cool.

Thanks for your responses, folks.

Okay, so what makes more sense:
  1) a remote management card that will let me
actually log in to the machine to monitor it as well
as to reboot it
vs.
  2) a remote-accessible powerstrip that will allow me
to remotely reboot the server?

I'm realising that sometimes the problem may simply be
processes out of control, and may be something that
doesn't require killing the entire machine, but just
some processes. 

In my current setup (an ordinary PIII 1.someting GHz
machine, not a server-class machine), when a process
goes haywire, I lose remote access via SSH, so I drive
to the colo, log in, sigh in frustration, and reboot
"because I'm already here, so why not?".

Some of the problems were caused by my old router ...
since I replaced it, the need to drive the 40 miles
each way has gone down significantly ... in fact, to
pretty much zero. So I have time to contemplate my
options here. :)

Cheers,
Maya





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[Asterisk-Users] Asterisk server with remote monitoring capabilities

[beonice]

Hello, all.

I'm tired of having to drive out to the colocation
facility each time my dedicated asterisk server craps
out, just to press the button to do a hard reboot.
(I'm running 1.05 stable at present, no telephony
hardware, as this is mainly a system that receives
calls, no dial-out ability is needed.) 

I've been looking at the fancy xeon-based systems
listed on ebay for a couple of hundred dollars, in the
hope that some of them have remote reboot
capabilities, but most of the sellers don't mention
this ability, and by the time I send out email, the
item is already taken anyway. :)

So, to cut the long story short, has anyone used one
of these server-class machines with remote reboot
capability, and does it really help? Are there any
particular configurations to stay away from? 

The wiki doesn't talk specifically about issues
regarding dual-CPU machines, but in following the chat
here on asterisk-users, it seems there are definitely
issues there ... can anyone elaborate? I don't want to
spend money on a fancy system that turns out to be
useless for my purposes.

Thanks for any insight!

Cheers,
Maya




 
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[Asterisk-Users] Asterisk, SIP and NAT: Help needed!

[beonice]

I've been googling and talking with Libretel about my
setup and the fact that incoming calls to my asterisk
box through the Libretel number reach my box (I hear
the greeting being played) but then don't accept DTMF.

Here is a rough diagram of my setup:

   Asterisk   | 
server| NAT<  Libretel
  | router 
  |

Note that there are NO SIP devices, just the Asterisk
box itself.

Apparently my asterisk server is trying to accept
audio on port 31347 at 192.168.0.2 (which is,
obviously, the internal IP address). I'm already
forwarding port 5060, so I presume that's what's
allowing the incoming call to hear my recorded
greeting.

>From google, I see that I can:
  a) Set my externip address to the address of the
 NAT router
  b) Set nat=yes 
  c) Somehow mangle headers???
  d) change the dtmfmode in [general]

Okay, so I've done all of the above:
  a) I've set the externip to the static IP address
 of the router
  b) I've set nat=yes in the [general] context (I'm
 only interested in incoming calls right now)
  d) I've tried each of the dtmfmode options in the
 [general] context

As for c) I'm not sure what needs to be done there. 

I've also, additionally forwarded ports 1 through
4 from the router to the asterisk box (Libretel
said the box is listening to audio on port 31347, so I
needed at least to ensure that's open, and got a
little generous there).

I'm really bewildered. Especially as IAX was so
straightforward to set up and get running! 

Thanks for any help!

Cheers,
Maya

Attached is my current sip.conf:
--
sip.conf

;
; SIP Configuration for Asterisk
;
; Syntax for specifying a SIP device in
extensions.conf is
; SIP/devicename where devicename is defined in a
section below.
;
; You may also use
; SIP/[EMAIL PROTECTED] to call any SIP user on the
Internet
; (Don't forget to enable DNS SRV records if you want
to use this)
;
; If you define a SIP proxy as a peer below, you may
call
; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED]
; where the proxyhostname is defined in a section
below
;
; Useful CLI commands to check peers/users:
;   sip show peers  Show all SIP peers
(including friends)
;   sip show users  Show all SIP users
(including friends)
;   sip show registry   Show status of hosts
we register with
;
;   sip debug   Show all SIP messages
;

[general]
;context=default; Default
context for incoming
calls
context=from-sip; Default context for incoming
calls
; After all, we don't
want any random
; incoming calls to
have access to
outbound
; calling - Maya, May
1, 2005
;recordhistory=yes  ; Record SIP history
by default
; (see sip history /
sip no history)
;realm=mydomain.tld ; Realm for digest
authentication
; defaults to
"asterisk"
; Realms MUST be
globally unique
according to RFC 3261
; Set this to your
host name or domain
name
port=5060   ; UDP Port to bind to
(SIP standard port
is 5060)
bindaddr=192.168.0.2; IP address to
bind to (0.0.0.0 binds
to all)
srvlookup=yes   ; Enable DNS SRV
lookups on outbound
calls
; Note: Asterisk only
uses the first
host
; in SRV records
; Disabling DNS SRV
lookups disables the
; ability to place SIP
calls based on
domain
; names to some other
SIP users on the
Internet

;pedantic=yes   ; Enable slow,
pedantic checking for
Pingtel
; and multiline
formatted headers for
strict
; SIP compatibility
(defaults to "no")
;tos=184; Set IP QoS to either
a keyword or
numeric val
;tos=lowdelay   ;
lowdelay,throughput,reliability,mincost,none
;maxexpirey=3600; Max length of
incoming registration we
allow
;defaultexpirey=120 ; Default length of
incoming/outoing registration
defaultexpirey=600  ; length of
incoming/outoing registration
;notifymimetype=text/plain  ; Allow overriding of
mime type in MWI NOTIFY
;videosupport=yes   ; Turn on support for
SIP video

disallow=all   ; First disallow all
codecs
allow=ulaw ; Allow codecs in order
of preference
allow=ilbc ; Note: codec order is
respected only in

Re: [Asterisk-Users] Voicemail Passwords

[beonice]

--- Jeff Heath <[EMAIL PROTECTED]> wrote:
> 
> On Tue, 2005-05-10 at 21:25, BJ Weschke wrote:
> >  voicemail.conf
> > 
> >  edit that file and issue a reload to change them.
> 
> I tried this, but I still can't get access to
> voicemail from one of the
> phones.  
> 
> This is a test system that I setup about a month
> ago.  Got busy and am
> just now getting back to it.  I have 2 SIP phones
> and the Asterisk
> server.  The default voicemail password is 1234 for
> both extensions.  I
> changed the password for one of them and (doh!)
> forgot/lost it.
> 
> Since this is a test system, I tried an experiment. 
> I went into the
> phone where I can get access to voicemail, and I
> manually changed the
> password from 1234 to 4567.  Then I issued a reload
> (the default
> passwords in voicemail.conf are 1234).  Then I
> accessed voicemail again,
> and the password is 4567 not 1234.

When you say manually, what do you mean? :) By editing
the voicemail.conf file? See below:

   --- snipped ---
> voicemail.conf
> 
> [general]
> format = wav49|gsm|wav
> serveremail = asterisk
> attach = yes
> maxmessage = 180
> maxgreet = 60
> skipms = 3000
> maxsilence = 10
> silencethreshold = 128
> maxlogins = 3
> 
> [default]
> 4009 => 1234,Jeff
> 4035 => 1234,Pam

 Right there in the [default] context is where you
would change 1234 to whatever else you want. :) Then,
even if you restart, you should have the new password.
You don't ever _need_ to remember the password. :)

Hope that helps.

Cheers,
Maya




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[Asterisk-Users] Getting DTMF to work with SIP?

[beonice]

Folks, from googling, I see that the dtmfmode
parameter is not valid in the [general] context. 

My problem is that my overseas DID through Libretel
seems  to want to come into the [general] context!
And, having done that, I get my welcome message, but
then the DID does not accept the DTMF when I try to
dial an extension! It plays the welcome message,
waits, and then times out (and hangs up nicely, yes).

I've actually tried setting the dtmfmode parameter in
the [general] context, obviously with no success.

Also, I note that when the incoming call uses SIP, a
physical hangup by the caller is not recognised ... it
basically goes through my timeout routine and hangs up
at some point.

The same context in extensions.conf can be reached
through my California DID which uses IAX, and in that
case I have no problem with DTMF or hangup.

Any ideas on how to solve this? I've got a question
open at Libretel, too, but was hoping someone here may
have already dealt with this?

Thanks,
Maya




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Re: [Asterisk-Users] Voice mail Greetings

[beonice]

--- Michael D Schelin <[EMAIL PROTECTED]> wrote:
> Hi all, What would cause the greetings not to play.
> The u command is 
> supposed to play the unavailable greeting. It
> doesn't work. with this 
> setup. Maybe I'm missing something. The voice
> prompts play well.  What 
> do you think? Thanks
> 
> exten => 9007,1,VoicemailMain
> exten => _.,2,Voicemail(u${EXTEN})

I'm guessing if you try changing the priority of the
above line to 1, you may get somewhere. Haven't tried
it, but there's not much else obvious to me. :)

> exten => _.,2,AbsoluteTimeout(180)
> exten => _.,4,Congestion 

  ^^ change priority to 3
> exten => _.,5,Hangup  
  ^^ change priority to 4

I'm not sure what asterisk does when priorities skip
numbers in sequence, but it's my guess that this would
be part of the problem.

Hope that helps ...

Cheers,
Maya




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[Asterisk-Users] Forwarding incoming calls via SIP

[beonice]

   I accidentally posted this on the biz list,
but it probably belongs here.

---

Folks, 

Has anyone successfully got a Libretel DID forwarded
to a Voicepulse Connect number for incoming calls?

I have signed up for an overseas DID with Libretel. My
main VOIP number is a VoicepulseConnect number, which
I've been happily using for incoming calls via IAX.

Libretel insists on SIP (or IAX via FWD ... but that's
adding another point of failure, potentially, so I
don't want to go there). The choices for forwarding
seem to be FWD, Generic SIP and Broadvoice. So, I
selected Generic SIP, and gave my address as:
   [EMAIL PROTECTED] (the same userid and
srvr I  use to register asterisk at voicepulse
connect).

I see on the asterisk console that it successfully
registers asterisk on the Voicepulse SIP server.

When I call my Libretel DID, however, all I get is a
busy signal. Libretel tells me that when they try to
forward a call through to my VPC SIP-address, they get
a "user not known". VPC tells me they are not set up
for IP-forwarding incoming calls, that calls must be
dialed with a phone or PBX.

So, what are my options? Has anyone successfully got
Libretel and Voicepulse interacting? Any ideas on what
to do next?

Thanks,
Maya


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RE: [Asterisk-Users] Set up SIP, now I'm getting a busy tone and weird (to me)messages on the console

[beonice]

> --- Tim Connolly <[EMAIL PROTECTED]> wrote:
> 
> > Is NAT=yes on, are you behind a firewall? Give us
> > some connectivity details.
> > Usually when you see maximum retries, its because
> > you have one-way
> > communications with the far end for some reason.
> Are
> > you setting "externip"
> > statically?

THANKS, Tim! That nat=yes bit did the trick as far as
the console messages go. Now I see a "registration
successful" message on the console. (Is it normal for
this to happen every few minutes? I haven't timed it
but it looks like the registration is recurring every
couple of minutes, maybe even once every minute.)

Unfortunately, I still get only a busy tone when I
dial the Canadian DID. When I dial the Canadian
number, absolutely nothing happens on the console!

Interestingly, even when I dial the local California
number (I'm assuming it's still using IAX at that
point), it seems to go into the unwelcome-calls
extension (which I thought was sip-specific) and,
according to the console, plays the congestion stuff
which I never hear.

Thanks again,
Maya


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RE: [Asterisk-Users] Set up SIP, now I'm getting a busy tone and weird (to me)messages on the console

[beonice]

--- Tim Connolly <[EMAIL PROTECTED]> wrote:

> Is NAT=yes on, are you behind a firewall? Give us
> some connectivity details.
> Usually when you see maximum retries, its because
> you have one-way
> communications with the far end for some reason. Are
> you setting "externip"
> statically?

To answer your questions, yes, I am behind a firewall.
The asterisk server is the only device connected to a
cheapo Netgear 4-port router/firewall. I'm not setting
externip myself, so whatever the default is, it's
getting used. I'm also NOT making outgoing calls, and
there are no actual SIP devices attached ... I'm just
trying to receive incoming calls forwarded from a
different provider via SIP.

Here is a complete sip.conf file ... do I need to
provide anything else?

sip.conf:
;
; SIP Configuration for Asterisk
;
; Syntax for specifying a SIP device in
extensions.conf is
; SIP/devicename where devicename is defined in a
section below.
;
; You may also use
; SIP/[EMAIL PROTECTED] to call any SIP user on the
Internet
; (Don't forget to enable DNS SRV records if you want
to use this)
;
; If you define a SIP proxy as a peer below, you may
call
; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED]
; where the proxyhostname is defined in a section
below
;
; Useful CLI commands to check peers/users:
;   sip show peers  Show all SIP peers
(including friends)
;   sip show users  Show all SIP users
(including friends)
;   sip show registry   Show status of hosts
we register with
;
;   sip debug   Show all SIP messages
;

[general]
;context=default; Default
context for incoming calls
context=unwelcome-calls ; Default context for
incoming calls
; After all, we don't
want any random
; incoming calls to
have access to outbound
; calling 
;recordhistory=yes  ; Record SIP history
by default
; (see sip history /
sip no history)
;realm=mydomain.tld ; Realm for digest
authentication
; defaults to
"asterisk"
; Realms MUST be
globally unique according to RFC 3261
; Set this to your
host name or domain name
port=5060   ; UDP Port to bind to
(SIP standard port is 5060)
bindaddr=0.0.0.0; IP address to bind
to (0.0.0.0 binds to all)
srvlookup=yes   ; Enable DNS SRV
lookups on outbound calls
; Note: Asterisk only
uses the first host
; in SRV records
; Disabling DNS SRV
lookups disables the
; ability to place SIP
calls based on domain
; names to some other
SIP users on the Internet

;pedantic=yes   ; Enable slow,
pedantic checking for Pingtel
; and multiline
formatted headers for strict
; SIP compatibility
(defaults to "no")
;tos=184; Set IP QoS to either
a keyword or numeric val
;tos=lowdelay   ;
lowdelay,throughput,reliability,mincost,none
;maxexpirey=3600; Max length of
incoming registration we allow
;defaultexpirey=120 ; Default length of
incoming/outoing registration
;notifymimetype=text/plain  ; Allow overriding of
mime type in MWI NOTIFY
;videosupport=yes   ; Turn on support for
SIP video

;disallow=all   ; First disallow all
codecs
;allow=ulaw ; Allow codecs in
order of preference
;allow=ilbc ; Note: codec order is
respected only in [general]
;musicclass=default ; Sets the default
music on hold class for all SIP calls
; This may also be set
for individual users/peers
;language=en; Default language
setting for all users/peers
; This may also be set
for individual users/peers
;relaxdtmf=yes  ; Relax dtmf handling
;rtptimeout=60  ; Terminate call if 60
seconds of no RTP activity
; when we're not on
hold
;rtpholdtimeout=300 ; Terminate call if
300 seconds of no RTP activity
; when we're on hold
(must be > rtptimeout)
;trustrpid = no ; If Remote-Party-ID
should be trusted
;progressinband=no  ; If we should
generate in-band ringing always
;useragent=Asterisk PBX ; Allows you to change
the user agent string
;nat=no ; NAT settings
; yes = Always ignore
info and assume NAT
; no = Use NAT mode
only according to RFC3581
; never = Never
attempt NAT mod

[Asterisk-Users] Set up SIP, now I'm getting a busy tone and weird (to me) messages on the console

[beonice]

Folks,

I'm hoping someone has already run into this ... the
only other complaint I've seen is here:

http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000640.html

and that basically was a problem with the /etc/hosts
... my server is definitely described in my hosts
file.

I've been using asterisk with IAX and a voicepulse
connect number. No problems at all receiving calls.

Now, I've just purchased a DID in Canada from another
provider, and their proxy only supports SIP. So,
following the generic instructions I've found off the
web, I set up my SIP.conf to point to voicepulse's
server, and set up the other DID to point into this
newly defined sip context, i.e., to
uid:[EMAIL PROTECTED]/888

The problem? The remote DID, when called, simply gives
me a busy signal. Also, on the asterisk console, I'm
seeing these messages that don't tell me anything:

May  1 18:37:09 WARNING[12065]: chan_sip.c:695
retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno
103 (Critical Request)
May  1 18:37:23 NOTICE[12065]: chan_sip.c:4036
sip_reg_timeout:-- Registration for
'[EMAIL PROTECTED]' timed out, trying again
May  1 18:37:23 DEBUG[12065]: chan_sip.c:4150
transmit_register: Scheduled a registration timeout #
5
May  1 18:37:29 WARNING[12065]: chan_sip.c:695
retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno
104 (Critical Request)
May  1 18:37:43 NOTICE[12065]: chan_sip.c:4036
sip_reg_timeout:-- Registration for
'[EMAIL PROTECTED]' timed out, trying again
May  1 18:37:43 DEBUG[12065]: chan_sip.c:4150
transmit_register: Scheduled a registration timeout #
7




It looks like the remote DID is failing to register
with the voicepulse server. Any hints on what could be
the problem?

If it helps, here is the relevant portion of my
sip.conf file.

[general]
;context=default; Default
context for incoming calls
context=unwelcome-calls ; Default context for
incoming calls
; After all, we don't
want any random
; incoming calls to
have access to outbound
; calling - Maya
Kurup, May 1, 2005
;recordhistory=yes  ; Record SIP history
by default
; (see sip history /
sip no history)
;realm=mydomain.tld ; Realm for digest
authentication
; defaults to
"asterisk"
; Realms MUST be
globally unique according to RFC 3261
; Set this to your
host name or domain name
port=5060   ; UDP Port to bind to
(SIP standard port is 5060)
bindaddr=0.0.0.0; IP address to bind
to (0.0.0.0 binds to all)

...

register => uid:[EMAIL PROTECTED]

; We need to allow at least incoming calls to
; accept calls via libretel, etc.
; So, let's add a context for that:

[888]; For incoming calls ONLY
type=user ; This device takes incoming calls
username=uid ; Username on device
secret=secret ; Password for device
host=srvr.voicepulse.com  ; This host will not 
  ; change frequently
context=allowed_context  ; Inbound calls from 
 ; this host go
 ; to the normal context

 -

and I have allowed_context described in my
extensions.conf, it's the same one I'm using for
regular IAX incoming calls, and works fine.
The context for unwelcome-calls is as follows:

[unwelcome-calls]
;
; Take unknown callers that may have found
; our system, and send them to a re-order tone.
; The string "_." matches any dialed sequence, so all
; calls will result in the Congestion tone application
; being called. They'll get bored and hang up
eventually.
;

exten => _.,1,Congestion
---


Any help would be appreciated.

Thanks,
Maya


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Re: [Asterisk-Users] VoicpulseConnect problems?

[beonice]

--- Tim Burt <[EMAIL PROTECTED]> wrote:
> 
> I was having problems with voicepulse about a week
> or two ago...
> 
> Incoming calls would fail, and one incoming call,
> would block all outgoing
> calls.
> 
> Then one day, the DTMF tones stopped working.
> 
> I could call into Asterisk, but I could not navigate
> because my tones were
> being ignored.
> 
> I ran debug, and Asterisk was not seeing the tones.
> 
> I emailed the voicepulse folks, and they fixed it
> the next day.  It was
> like they knew the bug, and reset whatever needed
> fixing.
> 
> Their reply was terse, but the fix was good.
> 
> Aside from this event, voicepulse has been very good
> for me!
> 
> Try to email their support

I did open a ticket on my account at voicepulse. What
you had sounds exactly like what I'm experiencing. If
I don't get a fix by tomorrow, I will probably just
call them and ask them when they plan to get to it. :)

Thanks for the feedback!

Cheers,
Maya


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[Asterisk-Users] VoicpulseConnect problems?

[beonice]

Folks, I'm having trouble with my voicepulse numbers.
Over the past week, incoming calls have been very
slow to be answered, but they seem fine while the call
is in progress. When the caller hangs up, asterisk
takes a while (over 2 minutes in some cases). This
system does not make outgoing calls.

Today, after rebooting my machine and rotating the log
files, I have absolutely NO incoming calls being
received. My cell phone dials the number, tells me
it's connected, and then happily hangs up 10 to 12
seconds later, while asterisk (and the logs) show no
indication at all of any incoming calls.

Looking at my syslog and asterisk messages, the only
thing I'm seeing over the past week that did not use
to happen is this message in the asterisk logs:

Apr 28 10:06:45 WARNING[4282]: Host
'gwiax-in-01.voicepulse.com' not found at line 72

But that's been happening for about the same time as
the slow-down issue, and still calls _were_ being
answered, albeit slowly.

I'm HoSed. :) Has anyone else run into this? Got any
ideas on what's up at VPConnect? Do I need to placate
the rain-god or something?

Any help would be appreciated!

Thanks,
Maya


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[Asterisk-Users] Has anyone used Libretel DIDs with Asterisk?

[beonice]

Hello, all.

Google searches reveal that a couple of users have
used Libretel port-of-call service via FWD using
Asterisk. I don't have a FWD account. I don't have a
SIP phone (no SIP setup at all, in fact).

I just purchased a DID in Brazil from Libretel and
discovered that they assume that I either have a SIP
phone or a VOIP number through BroadVoice. My provider
is actually VoicepulseConnect in California. My server
here is an Asterisk box connecting to a Voicepulse DID
using IAX.

HELP. :( I was assuming everyone supports IAX. I guess
I should have made sure first, huh?

Any help would be much appreciated.

Thanks,
Maya


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Re: [Asterisk-Users] Help using wav files for IVR

[beonice]

--- "trixter http://www.0xdecafbad.com";
<[EMAIL PROTECTED]> wrote:
> This script does mp3-> wav -> gsm, if you omit the
> first part it should
> work for you.  requires mpg123 and sox for wav->gsm
> conversion and
> mp3-decoder to start from mp3.
> 
> 
> will process all .mp3 files in the current
> directory..
> 
> #!/bin/sh
> for i in *.mp3; do
>   file=`echo $i | sed -e s/\.mp3$//g`
>   mp3-decoder -w "$file.wav" "$i"
>   normalize-audio "$file.wav"
>   sox "$file.wav" -r 8000 "decafbad-$file.gsm"
>   rm "$file.wav"
> done
> 
> 
> 
> for ease will do wav->gsm for your pleasure..  
> 
> #!/bin/sh
> for i in *.wav; do
>   file=`echo $i | sed -e s/\.wav$//g`
>   normalize-audio "$i"
>   sox "$i" -r 8000 "$file.gsm"
> done

Thanks! This script (modified to just look at the .wav
files) did exactly what I hoped it would ... I now
have decent-sounding gsm files that I can use. 

I still don't know why the original wav files were not
played back properly, though.

Thanks again,
Maya




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[Asterisk-Users] Help using wav files for IVR

[beonice]

Hi, all.

We needed to add a few custom menus to the IVR, so got
some recordings done professionally over the past
week. Now, the files (very good sound quality) have
been delivered to us, in .wav format. From googling at
the digium site, I'm seeing that people in the past
have successfully used wav files for IVRs.

I've tried sticking them into the
/var/libs/asterisk/sounds directory (after removing
corresponding .gsm files where necessary). However, if
I leave the extension as .wav (the format it came to
me in), then I get the error message:
ast_streamfile: Unable to open test-welcome (format
ulaw): No such file or directory


and if I just change the extension to .WAV, I get the
message:
ast_streamfile failed on
IAX2/[EMAIL PROTECTED]:4569/5 for
test-welcome

Any ideas what I'm doing wrong? It looks like (from
the messages above) the .WAV is the preferred format,
but it's obviously not working.

Help!

Thanks,
Maya




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Re: [Asterisk-Users] Asterisk@Home Install Problem

[beonice]

--- Scheda <[EMAIL PROTECTED]> wrote:
> Whenever I try to install [EMAIL PROTECTED], I get this
> error at about 43%
> 
>  There was an error installing
> rpmdb-redhat-3.4-0.20050105. This
> can indicate media failure, lack of disk space,
> and/or hardware
> problems. This is a fatal error and your install
> will be aborted.
> Please verify your media and try your install again.
> 
> I've gotten this on 3 different cd's I've burned. To
> make sure it
> wasn't a hardware problem of some sort, I installed
> Debain last night.
> Went without a hitch, but [EMAIL PROTECTED] doesn't like it...
> Any ideas as to
> why?

Scheda,

You _did_ verify that you have sufficient disk space,
right? I recently installed Linux on an old 6 GB hard
drive and forgot to tell it to take the whole drive,
so it only took what was left over after a Win2000
installation (not much was left, obviously), and I got
the same error before the OS finished installing.

Doesn't hurt to double check using df.

Other than that, I don't know ... looks like it's not
the media, unless those CDs won't let you install on a
different Linux box either.

Hardware problems can be a pain to isolate ... good
luck!

Cheers,
Maya




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[Asterisk-Users] SetDigitTimeout question

[beonice]

Folks, 

I'm trying to slow down the speed at which Asterisk
decides I've finished typing in an extension for
forwarding voicemail. I've tried using

exten => s,2,DigitTimeout(5);
exten => s,3,ResponseTimeout(5) ;

in my extensions.conf, but it still seems only about 2
seconds (or less!) before the "forward" functionality
decides that I've had enough time to type in the next
digit in the extension. Given that I occasionally have
brain-freeze when I look at an extension, this isn't
making me a happy camper today.

Any suggestions? A _quick_ look in the app_voicemail.c
didn't give any strong clues about timeouts at all.

HELP! Wild bears are ... oh, never mind, it's just an
asterisk.

Thanks,
Maya




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Re: [Asterisk-Users] SetCallerID({$NEWCALLERID})

[beonice]

--- Steven Frazier <[EMAIL PROTECTED]> wrote:
> I am trying to SetCallerID to a variable I have
> defined.  This obviously is
> wrong.  It actually sets the caller ID to
> $NEWCALLERID.  I have search
> through the examples on wiki but wasn't able to find
> something similar to
> see what I was doing wrong.  Could someone tell me
> the correct way to
> SetCallerID to a defined variable?
> 
>  exten => 2125551212,5,SetCallerID({$NEWCALLERID})

  --- snipped the rest ---

Off-hand, not having actually tested this, I'd guess
that you have the $ in the wrong place. Move it one
character to the left.

Cheers,
Maya




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Re: [Asterisk-Users] Question with email notification

[beonice]

--- J P Edmund <[EMAIL PROTECTED]> wrote:
> I have been searching all over for the answer on all
> sources online and 
> have come to the conclusion that it must be
> rudimentary or I am asking 
> the wrong question.
> 
> I cannot figure out how to configure the box to set
> the "from" address 
> to a correct domain, as my outgoing isp will not
> pass mail from 
> [EMAIL PROTECTED], as I expect it wouldn't.
> 
> Any help is appreciated, even just what should I be
> looking for to find 
> the correct information.
> 

This is more of a straight Linux question ... you
probably need to either set hostname for your
machine(s) or configure sendmail (assuming you're
using the default mailer) properly to set the correct
"from" address. You won't find the answer on the
asterisk wiki or mailing-list archives, I suspect. :)

Try reading the man pages for hostname and sendmail
... or do a google search. I remember having to do
something like that myself a while ago when I was
setting up my Linux boxen. 

Cheers,
Maya





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Re: [Asterisk-Users] Question about AGI vs. FastAGI vs. straight C/DB development

[beonice]

--- Steven Critchfield <[EMAIL PROTECTED]> wrote:
> On Mon, 2005-03-07 at 14:21 -0800, beonice wrote:
 --- snip ---
> > Would it make more sense to write a custom
> application
> > in C instead, designing it to work sort-of like
> the
> > built-in app_voicemail.c and others? I do know C,
> but
> > is it worth the effort in terms of ROI? :) 
> > 
> > I'd like to figure this out in advance rather than
> > after all my customers (hopefully there _will_ be
> > customers) start yelling at me about performance
> > issues. :)
> 
> You said a magic word there, customers. You need to
> become versed in the
> ins and outs of licensing. If you write it in C and
> distribute it, you
> must also offer to those you distribute asterisk
> your modifications and
> linked in code.

Hmm. I'm not going to be distributing this, it'll be
an in-house application, for my eyes only. :) The
customers will merely be using the in-house server. So
GPL or not would not be an issue. Robustness,
scaleability and ROI definitely are the issues I'm
interested in.

Again, the licensing issues for AGI or FastAGI using
any Asterisk libraries should not be an issue either.

   --- snipped some of Steven's response ---

> FastAGI offers faster initial response times than
> AGI as it will most
> likely already be running and ready to accept a new
> call. FastAGI should
> also be capable of being load balanced if that
> process is doing too much
> work to be responsive upon scaling. AGI is not
> capable of being run from
> a different machine.

Okay, so it looks like AGI itself will not be worth
pursuing. That's okay, at least I have a handle on
what I'm doing now. 

So, FastAGI or straight C? I just want something
that's easy to write. FastAGI will have to be wrapped
around Asterisk commands, so I'm anticipating a slight
overhead there, but if it's really not that much
slower, it may be faster to get the IVR code up and
running.

Opinions? Would any of you bother writing an IVR-only
application (like the app_voicemail application) in C?
Why or why not?

Thanks,
Maya





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[Asterisk-Users] Question about AGI vs. FastAGI vs. straight C/DB development

[beonice]

Folks,

I want to build a custom IVR for my setup. I've got it
working (well, the bells and whistles are not there
yet, but the basic stuff works) using AGI, but I'm
worried about how well this will scale.

I've seen references to FastAGI, and presumably this
will be more efficient. 

Question, though: how well do either of these (AGI or
FastAGI) scale if my system is handling a large number
of simultaneous calls? I'm assuming that even with
FastAGI, there's going to be some CPU overhead that
will increase as the number of simultaneous calls
increases. 

Would it make more sense to write a custom application
in C instead, designing it to work sort-of like the
built-in app_voicemail.c and others? I do know C, but
is it worth the effort in terms of ROI? :) 

I'd like to figure this out in advance rather than
after all my customers (hopefully there _will_ be
customers) start yelling at me about performance
issues. :)

Cheers,
Maya





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RE: [Asterisk-Users] Asterisk & MySQL Blobs

[beonice]

--- Steven Critchfield <[EMAIL PROTECTED]> wrote:
> On Mon, 2005-03-07 at 13:30 -0800, beonice wrote:
> >
  ... snip ...
> > Would it help to split the db off to a separate
> server
> > (that should reduce the CPU load on the asterisk
> > server)?
> > 
> > Any other alternatives? Anyone verified whether
> the
> > BLOB storage solution breaks down first or the
> number
> > of inodes runs out first? :)
> 
> On a 40gig drive I have, there are 4.7million
> inodes. 
> 
> On an 80gig drive I have there are 9.4million
> inodes.
> 
> Block size is (I think) 32k on these machines. At
> 32k I could only use
> 2.6 million inodes pointing to minimum sized files
> on the 80gig drive.
> 
> If you are worried about inodes, I believe it is xfs
> that dynamically
> creates inodes as needed and won't run out.

Thanks, Steven. This is exactly what I was hoping to
find out. DB performance issues aside, I was worried
about running out of resources on the filesystem
itself if I used the present vm storage mechanism.

Cheers,
Maya





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RE: [Asterisk-Users] Asterisk & MySQL Blobs

[beonice]

--- Colin Anderson <[EMAIL PROTECTED]>
wrote:
> >Has anyone had production experience using * w/
> MySQL Blobs to store sound
> files?  The >application I am working on requires
> all user data resides in a
> database.   I am currently >reading/writing the
> files to disk via a phpagi
> scripts but I would love to read the blob into a>
> variable in the dial plan,
> etc.  It seems like a waste of resources to write
> and delete the file.   
> 
> Too bad your requirement is to have everything in
> the DB, 'cause you will be
> asking for trouble in the long run. BLOBs are
> probably the fastest way to
> kill your DB once you scale. I did an experiment a
> few years ago to stream
> faxes as BLOB's into a SQL server and performance
> beyond a few thousand
> records was to put it mildly crap. 

Fair enough, I was thinking the same thing myself.

> IMO, use filesystem for files. Use DB for DB. Put a
> pointer in a field to
> the file. Your DB will love you for it. 

The problem I suspect will arise is the number of
inodes allowed by the file system. I don't know the
exact size of the typical inode-max, but this will
also presumably become an issue when the user tries to
scale to really large amounts of faxes or voicemail!
Would it help to split the db off to a separate server
(that should reduce the CPU load on the asterisk
server)?

Any other alternatives? Anyone verified whether the
BLOB storage solution breaks down first or the number
of inodes runs out first? :)

Cheers,
Maya





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Re: [Asterisk-Users] Weird behaviour on incoming DIDs

[beonice]

--- Michael Loftis <[EMAIL PROTECTED]> wrote:

> 
> 
> --On Monday, February 28, 2005 08:46 -0800 beonice
> <[EMAIL PROTECTED]> 
> wrote:
> 
> >  -- snipped --
> > When _I_ dial
> > either DID, I get exactly the same behaviour that
> I
> > have specified (the call is answered, and then I
> play
> > my own welcome mesage, then handle any extension
> > dialed).
> >
> > However, when one of my friends dials in, the 415
> DID
> > consistently works as designed, but the 650 DID
> > sometimes just tells him goodbye, and then hangs
> up on
> > him!
> >
 ... snipped
> 
> How are these DIDs being delivered?  SIP, IAX, PRI? 
> Either way I would 
> connect to my asterisk console in verbose mode (set
> verbose 255) and get 
> someone to call and then see whats different about
> the failed call versus 
> the successful calls.
> 

The DIDs are delivered via IAX. Yes, I've now set
verbose to extremely high, but of course, the problem
isn't happening right now. :)

I'll keep monitoring it ... it looks like there may be
something happening when I call that 650 DID that
clears up the problem, then repeated calls from
outside cause the weirdness again. Of course, then he
tells me it's broken, so I try calling it, which
clears up whatever the problem was. At least that is
what the _symptom_ is. :)

Thanks,
Maya





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[Asterisk-Users] Weird behaviour on incoming DIDs

[beonice]

Folks,

I have a problem here. I have 2 DIDs, one a 415 number
and the other a 650 number. I have my extensions.conf
set up to handle both of them exactly the same way,
passing them to an internal context. When _I_ dial
either DID, I get exactly the same behaviour that I
have specified (the call is answered, and then I play
my own welcome mesage, then handle any extension
dialed).

However, when one of my friends dials in, the 415 DID
consistently works as designed, but the 650 DID
sometimes just tells him goodbye, and then hangs up on
him! 

I don't know if it matters, but I am calling from the
650 area code and he's calling sometimes from the 415
area code and sometimes from 408. No, there is no
pattern as to which incoming call gets hung-up!

Here are the relevant sections of my extensions.conf:

[incoming_context] 
  ; This is the incoming call/DID context only.
exten => 415xxx,1,Goto(internal_context,s,1)
exten => 650xxx,1,Goto(internal_context,s,1)
 ; munged numbers, obviously
exten => i,1,Background(invalid)
exten => #,1,Background(goodbye)
exten => #,2,Wait(2)
exten => #,3,Hangup
exten => t,1,Background(goodbye)
exten => t,2,Wait(2)
exten => t,3,Hangup
exten => h,1,Hangup
exten => 1000,1,Background(goodbye);
exten => 1000,2,Wait(2);
exten => 1000,3,Hangup

#include 

And, my other_extensions.conf has:
[internal_context]
exten => #,1,Goto(incoming_context,1000,1)  
exten => *,1,VoiceMailMain()   
exten => *,2,Background(demo-congrats)
exten => h,1,Goto(incoming_context,h,1)) 
exten => i,1,Background(invalid)   
exten => s,1,Answer()  
exten => s,2,Background(test-welcome)   
exten => t,1,Goto(incoming_context,1000,1)   
exten => _[1-9]XX,1,VoiceMail(u${EXTEN}) 
exten => _[1-9]XX,2,Goto(incoming_context,1000,1)

Also, when this happens, I don't see Asterisk logging
the hung-up call in Master.csv. Other calls seem to be
logged fine.

Can this by any chance be caused by Master.csv getting
too large? If so, how come the same asterisk can still
handle calls coming in on the other DID with no
problems?

Thanks,
Maya
  
~




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Re: [Asterisk-Users] Custom Menu Not Working

[beonice]

I'll let someone else speak to the missing .conf
files. 

If you could post your extensions.conf and
extensions_additional.conf, it would be easier to help
you debug this. The suggestion from timebandit
_should_ have worked given your original post.

Personally, for learning my way around Asterisk, I
found that AMP actually got in the way. If you browse
the wiki, you'll find lots of helpful examples of the
config files, which is really what gets you going with
Asterisk.

Cheers,
Maya

--- Chris Blake <[EMAIL PROTECTED]> wrote:

> On Tue, 2005-02-22 at 10:48, [EMAIL PROTECTED]
> wrote:
> > > When adding the details in AMP for when caller
> dials 3, I have
> > > referenced it using 'custom-myapp,s,1', and if I
> go to
> > > 'extensions_additional.conf' I see the following
> line under the rest of
> > > menu item info that was created :
> > > 
> > > "exten => 3,1,Goto(custom-myapp,s,1) ;"
> > > 
> > > and in the extensions_custom.conf file I have
> > > 
> > > [custom-myapp]
> > > exten => 3,1,SayDigits(1234)
> > > exten => 3,2,Hangup()
> > 
> > Change to this
> >  [custom-myapp]
> >  exten => s,1,SayDigits(1234)
> >  exten => s,2,Hangup()
> > 
> > since you send it to "s,1"
> > 
> > hth
> 
> Howdy TimeBandit, thanks for replying...
> 
> I tried your suggestions and it made no difference,
> so I went back to
> AMP and removed entry pointing to custom-myapp,
> leaving the other two
> that I had also created earlier, which worked.
> 
> Now when I dial in the digital receptionist doesn`t
> do anything, and the
> call just disconnects. Here is the output of the CLI
> from the time the
> call is initiated, to termination :
> 
> ==
>  Starting simple switch on 'Zap/4-1'
> -- Executing GotoIf("Zap/4-1",
> "0?from-pstn-reghours|s|1:") in new
> stack
> -- Executing GotoIf("Zap/4-1",
> "0?from-pstn-afthours|s|1:") in new
> stack
> -- Executing GotoIfTime("Zap/4-1",
> "||*|*?from-pstn-reghours|s|1:")
> in new stack
> -- Goto (from-pstn-reghours,s,1)
> -- Executing GotoIf("Zap/4-1",
> "0?from-pstn-reghours-nofax|s|1:2")
> in new stack
> -- Goto (from-pstn-reghours,s,2)
> -- Executing Answer("Zap/4-1", "") in new stack
> -- Executing Wait("Zap/4-1", "1") in new stack
> -- Executing SetVar("Zap/4-1", "intype=") in new
> stack
> -- Executing Cut("Zap/4-1", "intype=intype|-|1")
> in new stack
> -- Executing GotoIf("Zap/4-1", "0?7:9") in new
> stack
> -- Goto (from-pstn-reghours,s,9)
> -- Executing GotoIf("Zap/4-1", "0?10:12") in new
> stack
> -- Goto (from-pstn-reghours,s,12)
> -- Executing Goto("Zap/4-1", "|s|1") in new
> stack
> -- Goto (,s,1)
> -- Hungup 'Zap/4-1'
> 
> 
> So I reload, and I notice that there are a few
> entries which seem a bit
> suspect :
> 
> 
> 
> == Parsing '/etc/asterisk/extconfig.conf': Not found
> (No such file or
> directory)
> 
> == Parsing '/etc/asterisk/adsi.conf': Not found (No
> such file or
> directory)
> 
> Reloading MGCP
>   == Parsing '/etc/asterisk/mgcp.conf': Not found
> (No such file or
> directory)
> 
> 
> 
> Based on the recommendations on the AMP site, I
> should not manually edit
> the extensions.conf, extensions_custom.conf and
> extensions_additional.conf files, so if removing the
> entries via AMP
> does not solve the problem, how do I go about fixing
> this ?
> 
> Commenting out the lines inserted by AMP into the
> above files does not
> make a difference.
> 
> Does the custom-myapp entry I listed earlier (with
> corrections of
> course) have to be in extensions_additional.conf as
> well?
> 
> Regards
> 
> --
> Chris Blake 
> Cell: 082 775 1492
> Work: +27 11 782 0840
> Fax : +27 11 782 0841
> Mail: [EMAIL PROTECTED]
> 
> Enzymes are things invented by biologists that
> explain things which
> otherwise require harder thinking. -- Jerome Lettvin
> 
> 
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Re: [Asterisk-Users] Custom Menu Not Working

[beonice]

Try this:


In the extensions_custom.conf file set:
 
[custom-myapp]
exten => s,1,SayDigits(1234)
exten => s,2,Hangup()


You're telling it to go to priority s, 
   ("exten => 3,1,Goto(custom-myapp,s,1)  
but in the custom-myapp context, you have priority 3
instead. 

Hope that helps.

Maya

--- Chris Blake <[EMAIL PROTECTED]> wrote:

> Greetings *`s,
> 
> I am having what appears to be a small problem, but
> the frustration is
> erally getting to me, what am I doing wrong here ?
> 
> I used AMP to set up a custom menu, so if caller
> presses 1 it goes to
> ext200, if caller presses 2 it goes to ext201 etc
> etc...
> 
> Now I have created a third option that when the
> caller presses 3 it must
> play a sound and hang up.
> No rocket science yet.
> 
> When adding the details in AMP for when caller dials
> 3, I have
> referenced it using 'custom-myapp,s,1', and if I go
> to
> 'extensions_additional.conf' I see the following
> line under the rest of
> menu item info that was created :
> 
> "exten => 3,1,Goto(custom-myapp,s,1) ;"
> 
> and in the extensions_custom.conf file I have 
> 
> [custom-myapp]
> exten => 3,1,SayDigits(1234)
> exten => 3,2,Hangup()
> 
> But when you call and press option 3, it hangs up
> immediately.
> I have followed examples from the documentation, and
> this should be
> working.
> 
> Any other places I can check where something is
> perhaps missing ?
> 
> Regards
> 
> --
> Chris Blake 
> Cell: 082 775 1492
> Work: +27 11 782 0840
> Fax : +27 11 782 0841
> Mail: [EMAIL PROTECTED]
> 
> When the ax entered the forest, the trees said, "The
> handle is one of
> us!" -- Turkish proverb
> 
> 
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Re: [Asterisk-Users] voice recognition xml

[beonice]

Dean,

I'd be very interested in helping with this effort.
I've worked with both SGML and XML in the past (I used
to work at SoftQuad in Toronto, one of the original
providers of SGML and HTML tools), and have written
several DTDs, both for SGML and XML.

I think it would be fun to work on an XML interchange
design for voice recognition ... please let me know if
your contact would be interested.

Cheers,
Maya Kurup

--- dean collins <[EMAIL PROTECTED]> wrote:

> Anyone here technical enough to design a voice
> recognition voice xml
> interchange for asterisk please email me; I've been
> speaking with a
> contact of mine that is in the voice recognition
> space and he is
> interested in 'donating' some technical support to
> the Asterisk
> community to assist with this project.
> 
>  
> 
> This can only help benefit the Asterisk Community if
> this comes off.
> 
>  
> 
> If this got up and running it would mean that
> Asterisk users would be
> able to offer voice recognition capabilities to
> their clients (or on
> their own installations) in an on-net ASP
> capability.
> 
>  
> 
> Email me and I'll send you the details of the
> working group.
> 
>  
> 
>  
> 
> Cheers,
> 
> Dean
> 
>  
> 
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Re: [Asterisk-Users] Anyone having trouble with VoicePulse Connect?

[beonice]

Thanks, Robert. Yes, I _finally_ figured out why I
need multiple extension contexts. I'm now one happy
camper.

Thanks again,
Maya

--- Robert Hajime Lanning
<[EMAIL PROTECTED]> wrote:

> 
> 
> > Robert, thank you very much for that informative
> > write-up. Of course, I now have more questions.
> The
> > first is really basic. I thought "extension" meant
> > something the caller dials _after_ reaching
> asterisk.
> > How come incoming DIDs have to be handled as if
> they
> > are extensions?
> 
> Actually the answer is better shown when looking at
> DID analog trunks.  Analog trunks are just POTS
> lines.
> With inbound DID analog trunks, the side that
> provides
> the voltage and dialtone are reversed.  The CO looks
> like a phone and your PBX looks like the CO.
> 
> So when a call comes in, the CO picks up the "phone"
> to your PBX and dials the DID number.  In this case,
> there is no callerID available.
> 
> >> [DID]
> >> exten => _X.,1,Goto(PublicExtensions,8001,1)
> >
> >
> > But won't this match every single number possible?
> > Including the extensions I set up? So, for
> example, if
> > I had an extension '1234', it would also be sent
> to
> > PublicExtensions extension 8001 with priority 1,
> as
> > would someone calling extension 8001, as well as
> > someone calling in from any DID. I'm totally
> confused
> > about the way this would behave.
> 
> This is why you use a seperate context ("[DID]") for
> DID's.  DID trunks go into the DID context.  Your
> station
> channels go into some other "internal" context, that
> contains your real extension numbers.  The DID
> context
> will match DID's to extensions that exist in other
> contexts.
> In my example, I have "PublicExtensions" which is a
> list of
> extensions that are available via my main phone
> number and
> it gets included in my "PrivateExtensions" context
> which is
> where my internal channels drop into.
> 
> Just remember that contexts seperate out extensions.
> Extension "8001" in context "internal" has nothing
> to
> do with extension "8001" in context "inbound".  You
> can
> definately point on towards the other, but there is
> no
> intrinsic relationship between the two.
> 
> -- 
> END OF LINE
>-MCP
> 
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Re: [Asterisk-Users] Still asterisk startup crash plz help

[beonice]

It looks like it's breaking at the iax.conf file. Have
you set up your iax.conf with the registration info
your service provider gave you? It should look
something like this:

register => iaxid:[EMAIL PROTECTED]

So, in my case, I have a line that says
register => myid:[EMAIL PROTECTED]

where myid, mypwd and the provider string were all
supplied to me by voicepulse when I signed up for a
DID with them.

If you're using a SIP phone, you'll also have to set
up sip.conf for your specific device. Also, you'll
need to modify extensions.conf and voicemail.conf to
suit your specific configuration.

I hope that helps.

Cheers,
Maya

--- Edward Banfa <[EMAIL PROTECTED]> wrote:

> Hi,
> First i would like to thank the kind people of the
> list who have
> answered my previuos mail, but i am still stuck as
> asterisk still
> crashes upon startup, i have read the install
> article at
>
http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
>  and i have search the asterisk archives, but i
> still cant get asterisk
> to work, i have tried reinstalling asterisk but it
> still complains and
> exits with the output below, i have trie editing the
> config files but to
> no avail, plz i would appreciate any pointers to
> anything that can help
> me here.
> Once again thank u
> 
> Edward
> 
> ASTERISK OUTPUT
> 
> 
> [EMAIL PROTECTED] asterisk-1.0.5]# /usr/sbin/asterisk
> -c
>   == Parsing '/etc/asterisk/asterisk.conf': Found
>   == Parsing '/etc/asterisk/extconfig.conf': Found
> Asterisk 1.0.5, Copyright (C) 1999-2004 Digium.
> Written by Mark Spencer <[EMAIL PROTECTED]>
>
=
>   == Parsing '/etc/asterisk/logger.conf': Found
> Asterisk Event Logger Started
> /var/log/asterisk/event_log
>   == Manager registered action Ping
>   == Manager registered action Events
>   == Manager registered action Logoff
>   == Manager registered action Hangup
>   == Manager registered action Status
>   == Manager registered action Setvar
>   == Manager registered action Getvar
>   == Manager registered action Redirect
>   == Manager registered action Originate
>   == Manager registered action MailboxStatus
>   == Manager registered action Command
>   == Manager registered action ExtensionState
>   == Manager registered action AbsoluteTimeout
>   == Manager registered action MailboxCount
>   == Manager registered action ListCommands
>   == Parsing '/etc/asterisk/manager.conf': Found
>   == Parsing '/etc/asterisk/rtp.conf': Found
>   == RTP Allocating from port range 1 -> 2
> Asterisk PBX Core Initializing
> Registering builtin applications:
>  [AbsoluteTimeout]
>   == Registered application 'AbsoluteTimeout'
>  [Answer]
>   == Registered application 'Answer'
>  [BackGround]
>   == Registered application 'BackGround'
>  [Busy]
>   == Registered application 'Busy'
>  [Congestion]
>   == Registered application 'Congestion'
>  [DigitTimeout]
>   == Registered application 'DigitTimeout'
>  [Goto]
>   == Registered application 'Goto'
>  [GotoIf]
>   == Registered application 'GotoIf'
>  [GotoIfTime]
>   == Registered application 'GotoIfTime'
>  [Hangup]
>   == Registered application 'Hangup'
>  [NoOp]
>   == Registered application 'NoOp'
>  [Prefix]
>   == Registered application 'Prefix'
>  [Progress]
>   == Registered application 'Progress'
>  [ResetCDR]
>   == Registered application 'ResetCDR'
>  [ResponseTimeout]
>   == Registered application 'ResponseTimeout'
>  [Ringing]
>   == Registered application 'Ringing'
>  [SayNumber]
>   == Registered application 'SayNumber'
>  [SayDigits]
>   == Registered application 'SayDigits'
>  [SayAlpha]
>   == Registered application 'SayAlpha'
>  [SayPhonetic]
>   == Registered application 'SayPhonetic'
>  [SetAccount]
>   == Registered application 'SetAccount'
>  [SetAMAFlags]
>   == Registered application 'SetAMAFlags'
>  [SetGlobalVar]
>   == Registered application 'SetGlobalVar'
>  [SetLanguage]
>   == Registered application 'SetLanguage'
>  [SetVar]
>   == Registered application 'SetVar'
>  [StripMSD]
>   == Registered application 'StripMSD'
>  [Suffix]
>   == Registered application 'Suffix'
>  [Wait]
>   == Registered application 'Wait'
>  [WaitExten]
>   == Registered application 'WaitExten'
> Asterisk Dynamic Loader Starting:
>   == Parsing '/etc/asterisk/modules.conf': Found
>  [chan_modem.so] => (Generic Voice Modem Driver)
>   == Parsing '/etc/asterisk/modem.conf': Found
>   == Loading modem driver chan_modem_aopen.so =>
> (A/Open (Rockwell
> Chipset) ITU-2 VoiceModem Driver)
>   == Registered channel type 'Modem' (Generic Voice
> Modem Channel
> Driver)
>  [res_musiconhold.so] => (Music On Hold Resource)
>   == Parsing '/etc/asterisk/musiconhold.conf': Found
>   == Registered application 'MusicOnHold'
>   == Registered application 'WaitMusicOnHold'
>   == Registered application 'SetMusicOnHold'
>  [res_adsi.so] => (ADSI Resource)
>   == Parsing '/etc/asterisk/adsi.conf': Found
>  [res_features.so] 

Re: [Asterisk-Users] Anyone having trouble with VoicePulse Connect?

[beonice]

Andrew, thanks for the explanation ... see more
questions below. :)

--- Andrew Thompson <[EMAIL PROTECTED]> wrote:

> beonice wrote:
--- snipped some ---
> > 
> > I guess the fundamental question is "why is a call
> > coming in from a DID any different?" And, of
> course,
> > "does a call coming in _not_ from a DID (maybe via
> an
> > SIP device? I don't know what the options are!)
> get
> > automagically handled by the 's' handler without
> > special mappings?"
> 
> I think you've confused your DID with inbound
> callerid.
> 
> Unless you have a international DID, the
> exten=>_NXXNXX pattern 
> should always accept any call bound for you from
> that context.
> 
> Even if someone calls you from an international
> location, voicepulse 
> *should* always present a unique DID to you the same
> way every time.

Okay. I think you unintentionally cleared up a
visualization problem for me. I was treating ALL my
extensions the same, including the inbound DID number.
So, I had:

[voicepulse_connect_context]
exten=>_NXXNXX,... 
exten => s,...
exten => i,...
exten => h,...
exten => t,...
exten => 1000,...
exten => 1001,...
etc. 

As you can see, all were lumped in the same context.
So (bear with me here, I'm trying to reason my way
through this), if I split up the internal extensions
into a different context, like so:

[voicepulse_connect_context]
exten=> 9995551212 Goto(my_internal_extensions, s, 1)
; where 9995551212 is the incoming DID
exten => s,...
exten => i,...
exten => h,...
exten => t,...

[my_internal_extensions]
exten => s,1,Answer
exten => #,1,Hangup
exten => 1000,...
exten => 1001,...
exten => 9995551212,... ; where 9995551212 is 
; an internal extension
, etc.

then I'd basically be able to treat the DID value
differently from what I want to do when the actual
extension 9995551212 is dialed?? All calls coming in
from that DID would go to the "my_internal_extensions"
handler for 's'. After that, if the caller dials an
extension, the "my_internal_extensions" context takes
over and handles the actual extensions.

Am I misunderstanding this? I'm going to go try it
out. :)

> 
> Now, if you want to do processing of an inbound call
> diferently based on 
> it's origination number(it's callerid), you handle
> that afterwards.
-- snipped an interesting example --

Nh. That wasn't what I was trying to achieve. I
just wasn't at all understanding what that "let's
handle an incoming DID as an extension" achieved.

If my "new" understanding of contexts is correct, then
I can see lots of possibilities. :)

Thanks a bunch!

Cheers,
Maya




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Re: [Asterisk-Users] Anyone having trouble with VoicePulse Connect?

[beonice]

--- Martijn van Oosterhout <[EMAIL PROTECTED]>
wrote:
 --- snipped my quote of what 's' is --
> > I guess it
> > implies that calls coming from DIDs have digits
> > associated with them.
> 
> Correct. On ISDN lines, E1, T1 and related digital
> protocols, details
> such as CallerID, Dialled Number, CLI Presentation,
> etc are passed as
> part of the call setup, before there is any
> discussion of "ringing". So
> Asterisk can go straight into the part of the script
> that matches.

Ah. I see.
> 
> However, on an analog line, you start with ringing
> and you still know
> nothing about the call. CallerID comes later and
> Dialled number is
> generally never sent at all. So you always start in
> "s".
> 
> Hope this helps,

It did clear up why an incoming DID is treated as an
extension. :) 

Thanks!

Cheers,
Maya




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Re: [Asterisk-Users] Anyone having trouble with VoicePulse Connect?

[beonice]

Robert, thank you very much for that informative
write-up. Of course, I now have more questions. The
first is really basic. I thought "extension" meant
something the caller dials _after_ reaching asterisk.
How come incoming DIDs have to be handled as if they
are extensions?

More questions follow:

--- Robert Hajime Lanning
<[EMAIL PROTECTED]> wrote:

  -- snip --
> "s" is the "start" extension if there is not one
> already
> provided.  When DID comes in, the channel is kindof
> "predialed".
> This is with most digital calls, SIP, IAX, H323,
> ISDN/PRI...

So the DID sort of dials its own number as an
extension? I'm not sure I understand, but that seems
to be the only way it would make sense to treat an
incoming DID as an extension.

> 
> You purchase a DID.  When a call comes in it says,
> "This is the
> number they were calling, you do your own matching
> to whatever
> extension you want."

Okay, that sort of helps me get a mental picture of
what's going on.

-- snip --
> 
> Usually you do not use wildcards for DIDs.  This is
> because
> people normally purchase more than one.  So, you
> need to
> distinguish between phone numbers.
> 
> I currently have two numbers from VoicePulse, so my
> extensions.conf
> has this: (numbers are changed to avoid crank calls)
> 
> [DID]
> exten => _4157611829,1,Goto(PublicExtensions,8001,1)
> exten => _4157611763,1,Goto(PublicExtensions,8003,1)
> 
> So, all inbound calls from VoicePulse goto this
> context.
> I jump from here to the extension I want the
> external phone
> number mapped to.
> 

OKAY. That example really helped. So you're basically
saying that I can treat each and every incoming DID
differently. That makes sense. Especially that I
realised that the wild-card approach would mean that
if I were to actually allocate a 10-digit extension,
it would get treated exactly the same as the incoming
DID-id'ed call. I tried it, and that's exactly what
happened. Your example is a better solution.

> If you get multiple numbers (say regional numbers)
> and you
> want all of them to goto the same place, you can
> wildcard
> like this: (gets past the international numbering
> differences)
> 
> [DID]
> exten => _X.,1,Goto(PublicExtensions,8001,1)


But won't this match every single number possible?
Including the extensions I set up? So, for example, if
I had an extension '1234', it would also be sent to
PublicExtensions extension 8001 with priority 1, as
would someone calling extension 8001, as well as
someone calling in from any DID. I'm totally confused
about the way this would behave.

Thanks again,
Maya


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RE: [Asterisk-Users] Anyone having trouble with VoicePulse Connect?

[beonice]

--- beonice <[EMAIL PROTECTED]> wrote:

  ... snipped ... this thread is getting long ...
> > 
> 
> Hmm. I guess that would be one way to do it. 
> 
> But we still have the issue of what happens when
> calls
> come in from DIDs in other countries. How are our
> colleagues in Europe and Asia handling this? Are you
> all creating handlers that special-case your
> incoming
> DID pattern and then map it to the handler for 's'
> as
> Robert demonstrated above?
> 
> I guess the fundamental question is "why is a call
> coming in from a DID any different?" And, of course,
> "does a call coming in _not_ from a DID (maybe via
> an
> SIP device? I don't know what the options are!) get
> automagically handled by the 's' handler without
> special mappings?"
> 
> Any deep insight into this issue would be welcome. I
> hate not knowing WHY I have to do magic
> incantations.

To answer my own question, at least partially, here is
a quote from the Asterisk Configuration chapter in
Paul Mahler's book "VoIP Telephony With Asterisk":

Table 1. Reserved Extension Names
--
Character   NameUsage
-   -   --
  s Start   A call that does not have
digits associated with it,
for example a loopstart
analog line, begins at the
"s" extension

Interesting. I don't understand it fully, but I'm sure
I will if I stare at it long enough. :) I guess it
implies that calls coming from DIDs have digits
associated with them.

Cheers,
Maya





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RE: [Asterisk-Users] Anyone having trouble with VoicePulse Connect?

[beonice]

--- Robert Webb <[EMAIL PROTECTED]> wrote:

> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED]
> On Behalf Of
> > Robert Goodyear
> > Sent: Thursday, February 17, 2005 8:36 PM
> > To: Asterisk Users Mailing List - Non-Commercial
> Discussion
> > Subject: Re: [Asterisk-Users] Anyone having
> trouble with
> > VoicePulse Connect?
> >
> >
> > On Feb 17, 2005, at 4:32 PM, Andrew Thompson
> wrote:
> >
> > > beonice wrote:
> > >> The culprit? Me. I'd commented out the line:
> > >> exten => _NXXNXX,1,Background(welcome) ;
> which is apparently a
> > >> critical one. I was under the impression that
> exten => s,1,Answer
> > >
> > > Will s be traveled if a call arrives at it with
> a DID?
> > >
> > > The pattern you have above matches any US did
> that arrives
> > into that
> > > context.
> >
> > I can't get the default handler (s) to work
> either; I'm using
> > VP Connect and I RTFMed a lot.
> >
> 
> I had to do a little trickery to get mine to work.
> Here is what I have:
> 
> iax.conf
> 
> [general]
> register =>
> dqo75qMa71:[EMAIL PROTECTED]
> 
> [voicepulse-in-01]
> type=user
> context=fromVoicePulse
> auth=rsa
> inkeys=voicepulse01
> 
> 
> extensions.conf
> 
> [fromVoicePulse]
> exten => _NXXNXX,1,Goto(fromPSTN,s,1)
> 
> [fromPSTN]
> 
> exten => s,1,LookupBlacklist
> exten => s,2,DigitTimeout(3)
> exten => s,3,ResponseTimeout(5)
> exten => s,4,Wait(1)
> exten => s,5,Background(custom/pls-wait)
> exten => s,102,Goto(blacklisted,s,1)
> 
> Etc...
> 
> 
> For me, I have all incoming lines going to the
> "fromPSTN" context. Since
> I am using it at home I treat most all incoming
> calls the same. But
> hopefully this will give you an idea of the way I
> had to do it. Not sure
> why...
> 
> Robert
> 

Hmm. I guess that would be one way to do it. 

But we still have the issue of what happens when calls
come in from DIDs in other countries. How are our
colleagues in Europe and Asia handling this? Are you
all creating handlers that special-case your incoming
DID pattern and then map it to the handler for 's' as
Robert demonstrated above?

I guess the fundamental question is "why is a call
coming in from a DID any different?" And, of course,
"does a call coming in _not_ from a DID (maybe via an
SIP device? I don't know what the options are!) get
automagically handled by the 's' handler without
special mappings?"

Any deep insight into this issue would be welcome. I
hate not knowing WHY I have to do magic incantations.

Cheers,
Maya





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Re: [Asterisk-Users] Anyone having trouble with VoicePulse Connect?

[beonice]

--- Andrew Thompson <[EMAIL PROTECTED]> wrote:

> beonice wrote:
> > The culprit? Me. I'd commented out the line:
> > exten => _NXXNXX,1,Background(welcome) ;
> > which is apparently a critical one. I was under
> the
> > impression that 
> > 
> > exten => s,1,Answer
> 
> Will s be traveled if a call arrives at it with a
> DID?
> 
> The pattern you have above matches any US did that
> arrives into that 
> context.

If I understood the little documentation I found on
's', it's supposed to be a catchall for ALL incoming
calls. That's why I assumed it would catch a DID as
well. If that's not the case, it really should be
updated in some meta-doc somewhere. :)

So what happens if the DID is _not_ a US DID? I've
seen users here from Europe and Asia as well ... does
each country need its own mapping to catch the
appropriate incoming DID? 

Now, what about the folks who are trying to call other
countries, and potentially be called by other DIDs
themselves? I'm assuming this sort of thing is very
likely.

How are others solving this issue?

Really wondering what's going on ...
Maya




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Re: [Asterisk-Users] Having trouble with extensions in an include file and retrieve_extensions_from_mysql.pl

[beonice]

--- Andrew Thompson <[EMAIL PROTECTED]> wrote:

 ... snip ...
> Back to your original post...
> 
>  > As of yesterday, though, when I have this format,
>  > asterisk won't accept incoming calls. It barfs
> with
>  > the message:
>  > Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757
>  > socket_read: Rejected connect attempt from
>  > 66.234.228.170, request
>  > '[EMAIL PROTECTED]' does not
> exist
> 
> So, where is this voicepulse_connect_context
> context?

Ah. When I posted, I shortened it to vp_context. This
is what used to be in the main extensions.conf, then
that got changed to main_vp_context while
voicepulse_connect_context moved into
extensions_from_mysql.conf.

I should be more careful when I post. :)

Cheers,
Maya



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Re: [Asterisk-Users] Having trouble with extensions in an include file and retrieve_extensions_from_mysql.pl

[beonice]

I've finally figured the problem. It turns out that it
had nothing to do with the include file. When I was
commenting out unnecessary extensions in the main
extensions.conf, I commented out the line:
exten => _NXXNXX,1,Background(welcome) ;

I did not realise (it's not mentioned anywhere) that
this particular extension NEEDS to be handled, but as
soon as I added this line back in, it fixed the
problem. Now I have asterisk reading multiple contexts
happily from the included file, answering the phone
when it rings, and letting me leave myself voicemail.
Very cool.

Could someone explain why this line is so important?
I'm assuming that incoming calls in other countries
will have different patterns for the incoming dialed
number. How does asterisk handle that? Can we add the
explanation to that question into the FAQ somewhere?

Thanks again, everyone!

Cheers,
Maya

--- beonice <[EMAIL PROTECTED]> wrote:

> --- Andrew Thompson <[EMAIL PROTECTED]>
> wrote:
> 
>  --- snip ---
> > 
> > The only thing that seems out of place to me is
> your
> > #include in 
> > [main_vp_context]. It looks to me like you intend
> > for the s, #, t, and i 
> > extensions to be in [main_vp_context]. The way you
> > layed out this 
> > example, that's not what is happenning.
> > 
> > I think you wanted this:
> > 
> > Your extensions_from_mysql.conf should still look
> > like:
> > 
> > [vp_context]
> > exten => 1000,1,Record(/tmp/rec:gsm);
> > exten => 1000,2,Playback(/tmp/rec)  ;
> > exten => 1000,3,Background(goodbye) ;
> > exten => 1000,4,Hangup();
> > 
> > 
> > Then, in extensions.conf:
> > 
> > #include 
> > 
> > [main_vp_context]
> > exten => s,1,Answer
> > exten => #,1,Background(goodbye)   ; Notify caller
> > exten => #,2,Hangup() ; Hang up
> > exten => t,1,Hangup() ; Hang up if timeout
> > exten => i,1,Playback(invalid) ; Play "invalid
> > ; extension" if
> > caller
> > ; misdials an
> > extension
> > include => vp_context
> > 
> > This way, you define both contexts, and include
> the
> > extensions that were 
> > defined in [vp_context] into [main_vp_context].
> > 
> > I don't know if this will resolve your other
> > problem, but I believe this 
> > is the dialplan you were trying to build.
> > 
> Hi, Andrew.
> 
> Yes, I see what you are saying. This sounds
> backwards,
> but it's actually doing what I _want_ it to do. :)
> 
> >From what I see in the dialplan, what asterisk does
> is, it loads the handlers for '#', 't' and 'i' as
> part
> of vp_context, not as part of main_vp_context. That
> actually happens to be as I wanted it.
> 
> main_vp_context is simply a place-holder for when I
> am
> testing without the include file, and in those
> cases,
> I simply comment out my include file and voila,
> those
> handlers now handle the main_vp_context incoming
> cases.
> 
> I know, I'm weird. :)
> 
> I'm seriously concerned that my problem may be
> caused
> by some interaction between asterisk and voicepulse:
> at the time of writing this, even with a simple
> extensions.conf that has no included files at all, I
> cannot dial in to the asterisk box ... all calls are
> being rejected.
> 
> Now I've spent a few minutes on (non-toll-free) hold
> with Voicepulse, sent them copies of my
> extensions.conf and iax.conf and am waiting for a
> response. Life really is exciting on the bleeding
> edge.
> 
> Cheers,
> Maya
> 
> 
> 
>   
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Re: [Asterisk-Users] Anyone having trouble with VoicePulse Connect?

[beonice]

Robert,

Thanks for the suggestion. I also came to the same
conclusion, but Asterisk wasn't showing any errors.
Then I decided to go back to basics, so I backed up my
extensions.conf and used the sample one provided by
voicepulse  voila, it worked. 

The culprit? Me. I'd commented out the line:
exten => _NXXNXX,1,Background(welcome) ;
which is apparently a critical one. I was under the
impression that 

exten => s,1,Answer

was all that was needed to make sure Asterisk answers
the phone ... obviously not. :) 

The importance of this line probably belongs in a FAQ
somewhere. I'll see if I can figure out how to update
the information on the wiki somehow.

By the way, what _is_ the known bug in #include files
that I need to look out for (for future debugging
hints ... I'm sure I'll run into in somewhere down the
line!)?

Thanks again,
Maya

--- Robert Goodyear <[EMAIL PROTECTED]> wrote:

> Look at an EXTENSIONS RELOAD and make sure the
> include is being parsed 
> -- and not throwing file not found errors. I broke
> my include 
> functionality last week by reMAKEing and not paying
> attention to a 
> known bug in the #INCLUDE function that existed in
> non-HEAD versions.
> 
> /rg




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Re: [Asterisk-Users] Having trouble with extensions in an include file and retrieve_extensions_from_mysql.pl

[beonice]

--- Andrew Thompson <[EMAIL PROTECTED]> wrote:

 --- snip ---
> 
> The only thing that seems out of place to me is your
> #include in 
> [main_vp_context]. It looks to me like you intend
> for the s, #, t, and i 
> extensions to be in [main_vp_context]. The way you
> layed out this 
> example, that's not what is happenning.
> 
> I think you wanted this:
> 
> Your extensions_from_mysql.conf should still look
> like:
> 
> [vp_context]
> exten => 1000,1,Record(/tmp/rec:gsm);
> exten => 1000,2,Playback(/tmp/rec)  ;
> exten => 1000,3,Background(goodbye) ;
> exten => 1000,4,Hangup();
> 
> 
> Then, in extensions.conf:
> 
> #include 
> 
> [main_vp_context]
> exten => s,1,Answer
> exten => #,1,Background(goodbye)   ; Notify caller
> exten => #,2,Hangup() ; Hang up
> exten => t,1,Hangup() ; Hang up if timeout
> exten => i,1,Playback(invalid) ; Play "invalid
> ; extension" if
> caller
> ; misdials an
> extension
> include => vp_context
> 
> This way, you define both contexts, and include the
> extensions that were 
> defined in [vp_context] into [main_vp_context].
> 
> I don't know if this will resolve your other
> problem, but I believe this 
> is the dialplan you were trying to build.
> 
Hi, Andrew.

Yes, I see what you are saying. This sounds backwards,
but it's actually doing what I _want_ it to do. :)

>From what I see in the dialplan, what asterisk does
is, it loads the handlers for '#', 't' and 'i' as part
of vp_context, not as part of main_vp_context. That
actually happens to be as I wanted it.

main_vp_context is simply a place-holder for when I am
testing without the include file, and in those cases,
I simply comment out my include file and voila, those
handlers now handle the main_vp_context incoming
cases.

I know, I'm weird. :)

I'm seriously concerned that my problem may be caused
by some interaction between asterisk and voicepulse:
at the time of writing this, even with a simple
extensions.conf that has no included files at all, I
cannot dial in to the asterisk box ... all calls are
being rejected.

Now I've spent a few minutes on (non-toll-free) hold
with Voicepulse, sent them copies of my
extensions.conf and iax.conf and am waiting for a
response. Life really is exciting on the bleeding
edge.

Cheers,
Maya




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[Asterisk-Users] Having trouble with extensions in an include file and retrieve_extensions_from_mysql.pl

[beonice]

Folks,

I've been running asterisk successfully using the
extensions.conf and voicemail.conf.

Now that I've got asterisk happily looking up MySQL
tables for the VM configuration, I decided to try out
the contributed script
 
/usr/src/asterisk/contrib/scripts/retrieve_extensions_from_mysql.pl

I edited the script so that its output goes to a
separate  extensions_from_mysql.conf file.

The resulting extensions_from_mysql.conf file looks
something like this:
[vp_context]
exten => 1000,1,Record(/tmp/rec:gsm);
exten => 1000,2,Playback(/tmp/rec)  ;
exten => 1000,3,Background(goodbye) ;
exten => 1000,4,Hangup();

I decided to #include this in my main extensions.conf,
like so:

[main_vp_context]
exten => s,1,Answer
#include 
exten => #,1,Background(goodbye)   ; Notify caller
exten => #,2,Hangup() ; Hang up
exten => t,1,Hangup() ; Hang up if timeout
exten => i,1,Playback(invalid) ; Play "invalid
   ; extension" if caller
   ; misdials an extension

Basically, I expect asterisk to load the two as
separate contexts, and I could swear that it used to.

In fact, when I set the verbosity higher, asterisk is
definitely still loading them as separate contexts.

As of yesterday, though, when I have this format,
asterisk won't accept incoming calls. It barfs with
the message:
Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757
socket_read: Rejected connect attempt from
66.234.228.170, request
'[EMAIL PROTECTED]' does not exist

The only way to get asterisk to receive calls again is
to edit the included file to ensure it does not have a
context line in it. So I commented out the line where
the retrieve_extensions_from_mysql.pl sticks the
context information into the created file.

Now, it all works fine.

But it's no good.

What about when I want to have a sip.conf and have a
list of extensions that do different things in the sip
context? I really like the contributed script for its
ability to add multiple context sections.

Anyone see a possible reason for the problem? Do you
have any ideas how to use an include file which
contains multiple contexts? Or will I have to generate
multiple include files, one per included context,
without the context lines in these files?

Thanks for any help!

Cheers,
Maya








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Re: [Asterisk-Users] Anyone having trouble with VoicePulse Connect?

[beonice]

I was doing some testing and it seems to be related to
my extensions.conf. 

I have a #include  that
was working fine yesterday:

[voicepulse_connect_context2]
exten => s,1,Answer
exten => s,2,NoOp,${CALLERID}
#include 

and extensions_from_mysql.conf is:
[voicepulse_connect_context]
exten => 1234567890,1,VoiceMail(u1234567890);
exten => 1234567890,2,Background(goodbye)() ;
exten => 1234567890,3,Hangup()  ;
exten => 1234567890,102,VoiceMail(b1234567890)  ;
exten => 1234567890,103,Background(goodbye)()   ;
exten => 1234567890,104,Hangup();

Today this _will_NOT_ work. Today, the only way I can
get it to work is to set the context in
extensions.conf back to 

[voicepulse_connect_context]

exten => s,1,Answer
exten => s,2,NoOp,${CALLERID}
; stick my extensions in here
;
;#include 

Note that I've had to comment out the #include!

Has anyone run into this weird behaviour? I am not
convinced as to where the problem is, given that the
message still seems to indicate somehow that the
interaction with voicepulse is the problem.

Thanks,
BeOnIce.

--- beonice <[EMAIL PROTECTED]> wrote:

> I've been using my voicepulse connect number for
> over
> a month now, but today it simply won't connect. My
> partner and I each have a number, both are mapped in
> my iax.conf and extensions.conf files. This has been
> working fine.
> 
> Today, either number gives this message:
> Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757
> socket_read: Rejected connect attempt from
> 66.234.228.170, request
> '[EMAIL PROTECTED]' does not
> exist
> 
> where ## is the number I dialed. Obviously
> chan_iax2 knows which of the numbers I dialed, but
> for
> some reason is unable to connect my machine to
> voicepulse?
> 
> Any troubleshooting tips? This is frustrating! I
> _have_ asked a question via my VoicePulse Connect
> web
> account, but I don't know when they will respond. 
> 
> Thanks,
> BeOnIce.
> 
> 
> 
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[Asterisk-Users] Anyone having trouble with VoicePulse Connect?

[beonice]

I've been using my voicepulse connect number for over
a month now, but today it simply won't connect. My
partner and I each have a number, both are mapped in
my iax.conf and extensions.conf files. This has been
working fine.

Today, either number gives this message:
Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757
socket_read: Rejected connect attempt from
66.234.228.170, request
'[EMAIL PROTECTED]' does not exist

where ## is the number I dialed. Obviously
chan_iax2 knows which of the numbers I dialed, but for
some reason is unable to connect my machine to
voicepulse?

Any troubleshooting tips? This is frustrating! I
_have_ asked a question via my VoicePulse Connect web
account, but I don't know when they will respond. 

Thanks,
BeOnIce.



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Re: [Asterisk-Users] Still stuck trying to make Asterisk read MySQL (SOLVED)

[beonice]

Thanks to everyone who responded. I submitted a bug
report to digium
(http://bugs.digium.com/bug_view_page.php?bug_id=0003580),
and markster responded, suggesting that I get an
updated version of stable asterisk from CVS. I did,
and now it's all working fine. I must have initially
downloaded a not-quite-stable stable asterisk. :)

The only glitch I seem to notice is that it demands
that the VM context be 'default'. :) I set it to
something else, and the generated SQL code is still
looking for "context=default". Oh well, at least
that's easy to work around.

Thanks again,
Maya

--- Joe Dennick <[EMAIL PROTECTED]> wrote:

> I've been working with RealTime configuration from
> MySQL Server, and have had
> good results.  You might check it out. You can do a
> search for 'realtime' on
> the Wiki and get some good documentation on how to
> set it up.  I think in the
> extconfig.conf file, not only do you need to
> identify the engine (ODBC in your
> case), but you also need to identify the actual
> table you used for your
> Voicemail configuration.  If I recall correctly, the
> default is a table named
> 'voicemail' and since you are using a different
> name, you need to specify the
> name in the extconfig.conf file so it can find it.
> 
> beonice ([EMAIL PROTECTED]) wrote:
> >
> > I've been continuing to experiment with MySQL. I'm
> > having absolutely no luck getting asterisk to read
> > voicemail configuration data and mailbox
> configuration
> > data from mysql tables instead of from
> voicemail.conf.
> >
> >
> > The default Asterisk setup that reads from
> > voicemail.conf and extensions.conf works fine. I'm
> > using
> > Asterisk CVS-v1-0-12/12/04-15:58:29 on a Whitebox
> > Enterprise Linux box. I'm not using any telephony
> > hardware or SIP phones. I've just got a voicepulse
> DID
> > talking to asterisk via IAX.
> >
> > I've got mysql downloaded and installed and have
> > successfully got the contributed script reading
> from
> > my asterisk_vm database to set up the
> extensions.conf,
> > as per the instructions at:
> >
>
http://www.voip-info.org/wiki-Asterisk+extensions+from+mysql
> >
> > Now I'm trying to get Asterisk to look up
> voicemail
> > configs from the asterisk_vm database. In order to
> do
> > this, I've been following the instructions at:
> >
>
http://www.voip-info.org/wiki-Asterisk+voicemail+database
> >
> > So, I've:
> > 1) Updated the /usr/src/asterisk/apps/Makefile to
> have
> > USE_MYSQL_VM_INTERFACE=1 and recompiled asterisk,
> with
> > make clean; make; make install
> >
> > 2) Updated voicemail.conf to have the appropriate
> > entries:
> > dbuser=username ;; Yes I changed this to my
> username
> > dbpass=password ;; Yes I changed this to my
> password
> > dbhost=localhost
> > dbname=asterisk_vm
> >
> >
> > 3) Created the users table in the asterisk_vm
> > database.
> >
>
+-++--+--+---+---+++
> > | context | mailbox| password | fullname |
> > email | pager | options| stamp
> >  |
> >
>
+-++--+--+---+---+++
> > | default |    | 1234 | Moron Tester |
> > [EMAIL PROTECTED] |   | attach=yes |
> 20050211131641
> > |
> >
>
+-++--+--+---+---+++
> >
> > 4) Updated extensions.conf to have the following
> line:
> > exten => ,1,VoiceMail(u)
> >
> > I tried restarting asterisk at this point, called
> in
> > and tried to leave voicemail for extension (and
> > mailbox) . Here's the message I get:
> >
> > *CLI> Feb 11 13:21:36 WARNING[18393]:
> > app_voicemail.c:1539 leave_voicemail: No entry in
> > voicemail config file for ''
> >
> >
> > So I dug around some more and found
> > http://www.voip-info.org/wiki-Asterisk+res_config
> >
> > Decided to try these instructions as well. So:
> >
> > 5) I created the ast_config table as directed:
> > Here is the data:
> >
> >
>
++++---++--+--+-+
> > | id | cat_metric | var_metric | commented |
> filename
> >  | category | var_name | var_val |
> >
>
++++-

Re: [Asterisk-Users] ast_data does not patch

[beonice]

Heh. Good point, Kevin. I didn't realise that ast_data
was also a third party add-on. :)

So I submitted a bug report to digium with my gdb
trace
(http://bugs.digium.com/bug_view_page.php?bug_id=0003580),
and markster there suggested that I should update to
the latest stable asterisk from CVS. I did. And now
the core asterisk can see my voicemail configuration
in MySQL just fine. I must have originally retrieved a
buggy version of the stable asterisk. :)

Thanks, everyone, for all your help!

Cheers,
Maya

--- "Kevin P. Fleming" <[EMAIL PROTECTED]>
wrote:

> beonice wrote:
> 
> > I don't know whether RealTime PostgreSQL, but I
> can't
> > upgrade to RealTime anyway ... I need a stable
> version
> > of asterisk, and the current stable version does
> not
> > include RealTime. :(
> 
> You need a "stable" version of Asterisk, but you're
> willing to patch 
> with an unsupported change like ast_data? Seems a
> little contradictory 
> to me :-)
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Re: [Asterisk-Users] ast_data does not patch

[beonice]

Matthew,

I believe you're the original developer on the
RealTime code ... do you know if it will work with
Asterisk 1.0.5 or whatever the stable version of
Asterisk is? 

Thanks,
Maya

--- Matthew Boehm <[EMAIL PROTECTED]> wrote:

> Why not just use the built-in database features to
> do what you want? Its
> called RealTime. Lots of info on it on the wiki.
> 
> -Matthew
> 
> - Original Message - 
> From: <[EMAIL PROTECTED]>
> To: "'Asterisk Users Mailing List - Non-Commercial
> Discussion'"
> 
> Sent: Saturday, February 12, 2005 4:56 PM
> Subject: [Asterisk-Users] ast_data does not patch
> 
> 
> > Hello all,
> >
> > I have just been trying to install the latest
> ast_data from:
> >
> > http://svn.asteriskdocs.org/res_data/ast_data/
> >
> > into my cvs version of Asterisk and have found
> that the install patching
> > fails.
> > -
> >
> > patching file contrib/scripts/sip-friends.sql
> > patching file contrib/scripts/iax-friends.sql
> > patching file apps/app_voicemail.c
> > patching file apps/app_directory.c
> > patching file channels/chan_sip.c
> > Hunk #2 succeeded at 621 (offset 9 lines).
> > Hunk #3 FAILED at 1480.
> > Hunk #4 succeeded at 1549 (offset 11 lines).
> > Hunk #5 succeeded at 1617 (offset 18 lines).
> > Hunk #6 succeeded at 1972 (offset 11 lines).
> > 1 out of 6 hunks FAILED -- saving rejects to file
> channels/chan_sip.c.rej
> > patching file channels/chan_iax2.c
> > Hunk #2 succeeded at 593 with fuzz 2 (offset 13
> lines).
> > Hunk #3 FAILED at 944.
> > Hunk #4 succeeded at 4441 (offset 57 lines).
> > Hunk #5 FAILED at 5234.
> > 2 out of 5 hunks FAILED -- saving rejects to file
> channels/chan_iax2.c.rej
> > patching file Makefile
> > patching file pbx.c
> > Hunk #6 succeeded at 1390 (offset 18 lines).
> > Hunk #8 succeeded at 1439 (offset 18 lines).
> > Hunk #10 succeeded at 1508 (offset 18 lines).
> > patching file asterisk.c
> > Hunk #2 succeeded at 1922 (offset 76 lines).
> >
> > --
> >
> > Does anyone know how to get in touch with the
> developer or have another
> > viable and working option that will allow me to
> dynamically place my users
> > information in a MySQL database?
> >
> > Thanks,
> > Lonnie
> >
> >
> > ___
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> > Asterisk-Users@lists.digium.com
> >
>
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> >   
>
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Re: [Asterisk-Users] ast_data does not patch

[beonice]

Lonnie, 

If you look at:

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

it says that MySQL _is_ supported.

I don't know whether RealTime PostgreSQL, but I can't
upgrade to RealTime anyway ... I need a stable version
of asterisk, and the current stable version does not
include RealTime. :(

I am hoping to be able to use MySQL as well ... I've
got all the config files set up correctly, I believe,
but I get nowhere because:
  a) ast_data seems to be currently broken
  b) the shipped version of asterisk app_voicemail.c
seems to have a bug in its select statement ...
MySQL's log shows that it's connecting correctly to
the database, but then it issues a command "SEL"
instead of the command "SELECT * from users WHERE ..."

Very frustrating. I don't really want to debug this
code, but it looks like I'm going to give it a shot
anyway. (No, not the ast_data code, the basic
app_voicemail.c code.)

Lonnie, if you have any luck getting ast_data to work,
or find a contact address for rgagnon (who seems to
have done the original development and presumably is
still maintaining it), please forward that to me.

Cheers,
Maya

--- [EMAIL PROTECTED] wrote:

> Thanks
> 
> I'll look into it, but from the little that I read
> on RealTime, I was
> under the impression that it did not use MySQL or
> PostgreSQL which is a
> database feature that I was hoping to use.
> 
> --Lonnie
> 
> 
> > Why not just use the built-in database features to
> do what you want? Its
> > called RealTime. Lots of info on it on the wiki.
> >
> > -Matthew
> >
> > - Original Message -
> > From: <[EMAIL PROTECTED]>
> > To: "'Asterisk Users Mailing List - Non-Commercial
> Discussion'"
> > 
> > Sent: Saturday, February 12, 2005 4:56 PM
> > Subject: [Asterisk-Users] ast_data does not patch
> >
> >
> >> Hello all,
> >>
> >> I have just been trying to install the latest
> ast_data from:
> >>
> >> http://svn.asteriskdocs.org/res_data/ast_data/
> >>
> >> into my cvs version of Asterisk and have found
> that the install patching
> >> fails.
> >> -
> >>
> >> patching file contrib/scripts/sip-friends.sql
> >> patching file contrib/scripts/iax-friends.sql
> >> patching file apps/app_voicemail.c
> >> patching file apps/app_directory.c
> >> patching file channels/chan_sip.c
> >> Hunk #2 succeeded at 621 (offset 9 lines).
> >> Hunk #3 FAILED at 1480.
> >> Hunk #4 succeeded at 1549 (offset 11 lines).
> >> Hunk #5 succeeded at 1617 (offset 18 lines).
> >> Hunk #6 succeeded at 1972 (offset 11 lines).
> >> 1 out of 6 hunks FAILED -- saving rejects to file
> >> channels/chan_sip.c.rej
> >> patching file channels/chan_iax2.c
> >> Hunk #2 succeeded at 593 with fuzz 2 (offset 13
> lines).
> >> Hunk #3 FAILED at 944.
> >> Hunk #4 succeeded at 4441 (offset 57 lines).
> >> Hunk #5 FAILED at 5234.
> >> 2 out of 5 hunks FAILED -- saving rejects to file
> >> channels/chan_iax2.c.rej
> >> patching file Makefile
> >> patching file pbx.c
> >> Hunk #6 succeeded at 1390 (offset 18 lines).
> >> Hunk #8 succeeded at 1439 (offset 18 lines).
> >> Hunk #10 succeeded at 1508 (offset 18 lines).
> >> patching file asterisk.c
> >> Hunk #2 succeeded at 1922 (offset 76 lines).
> >>
> >> --
> >>
> >> Does anyone know how to get in touch with the
> developer or have another
> >> viable and working option that will allow me to
> dynamically place my
> >> users
> >> information in a MySQL database?
> >>
> >> Thanks,
> >> Lonnie
> >>
> >>
> >> ___
> >> Asterisk-Users mailing list
> >> Asterisk-Users@lists.digium.com
> >>
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> >> To UNSUBSCRIBE or update options visit:
> >>   
>
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> >>
> >
> > ___
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> > Asterisk-Users@lists.digium.com
> >
>
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> >   
>
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> >
> 
> 
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[Asterisk-Users] Is no one using MySQL on stable asterisk?

[beonice]

I'm still (doggedly) trying to get asterisk to read my
voicemail configuration from MySQL. I'm using the
stable release of Asterisk, from back in December,
before realtime was included.

If anyone has got it to work, please contact me ...
I've posted details, but everyone who's responded so
far has been working with the newer version that uses
realtime.

Unfortunately, this is going to be a production
server, so I need a stable release, hence the
reluctance to upgrade.

Any help or pointers would be deeply appreciated. My
original post is at:

http://lists.digium.com/pipermail/asterisk-users/2005-February/088949.html

Thanks,
Maya
(I _need_ to get this working. I hate being defeated
by a bunch of bits and bytes!)





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[Asterisk-Users] Still stuck trying to make Asterisk read MySQL

[beonice]

I've been continuing to experiment with MySQL. I'm
having absolutely no luck getting asterisk to read
voicemail configuration data and mailbox configuration
data from mysql tables instead of from voicemail.conf.


The default Asterisk setup that reads from
voicemail.conf and extensions.conf works fine. I'm
using 
Asterisk CVS-v1-0-12/12/04-15:58:29 on a Whitebox
Enterprise Linux box. I'm not using any telephony
hardware or SIP phones. I've just got a voicepulse DID
talking to asterisk via IAX.

I've got mysql downloaded and installed and have
successfully got the contributed script reading from
my asterisk_vm database to set up the extensions.conf,
as per the instructions at:
http://www.voip-info.org/wiki-Asterisk+extensions+from+mysql

Now I'm trying to get Asterisk to look up voicemail
configs from the asterisk_vm database. In order to do
this, I've been following the instructions at: 
http://www.voip-info.org/wiki-Asterisk+voicemail+database

So, I've:
1) Updated the /usr/src/asterisk/apps/Makefile to have
USE_MYSQL_VM_INTERFACE=1 and recompiled asterisk, with
make clean; make; make install

2) Updated voicemail.conf to have the appropriate
entries:
dbuser=username ;; Yes I changed this to my username
dbpass=password ;; Yes I changed this to my password
dbhost=localhost 
dbname=asterisk_vm 


3) Created the users table in the asterisk_vm
database.
+-++--+--+---+---+++
| context | mailbox| password | fullname |
email | pager | options| stamp
 |
+-++--+--+---+---+++
| default |    | 1234 | Moron Tester |
[EMAIL PROTECTED] |   | attach=yes | 20050211131641
|
+-++--+--+---+---+++

4) Updated extensions.conf to have the following line:
exten => ,1,VoiceMail(u)

I tried restarting asterisk at this point, called in
and tried to leave voicemail for extension (and
mailbox) . Here's the message I get:

*CLI> Feb 11 13:21:36 WARNING[18393]:
app_voicemail.c:1539 leave_voicemail: No entry in
voicemail config file for ''


So I dug around some more and found 
http://www.voip-info.org/wiki-Asterisk+res_config

Decided to try these instructions as well. So:

5) I created the ast_config table as directed:
Here is the data:

++++---++--+--+-+
| id | cat_metric | var_metric | commented | filename 
 | category | var_name | var_val |
++++---++--+--+-+
|  1 |  0 |  0 | 0 |
voicemail.conf | default  |  | |
++++---++--+--+-+

6) I edited /etc/asterisk/configs/res_odbc.conf to
contain:
[mysql1]
dsn = MySQL-asterisk
username = myuser
password = mypass
pre-connect = yes
[mysql1]
dsn = asterisk_vm
username = myuser ;; changed to my userid on mysql
password = mypass ;; changed to my password on mysql
pre-connect = yes

[mysql2]
dsn = MySQL2-asterisk
username = myuser2
password = mypass2
enabled = no

[ENV] 
VAR=VALUE 

7) Inserted glue to tell asterisk where to look:
; /etc/asterisk/res_config_odbc.conf
[settings]
table = ast_config
connection = mysql1 

8) Rerouted Asterisk's config engine:
; /etc/asterisk/extconfig.conf
[settings]
;queues.conf => odbc 
voicemail.conf => odbc 

9) I modified the sample script load_res_config.pl and
ran it, it successfully updated my ast_config table,
stuffing in all the settings that I'm used to seeing
in voicemail.conf.

10) I restarted asterisk _again_.
I get the exact same message.
Feb 11 14:18:40 WARNING[18528]: app_voicemail.c:1539
leave_voicemail: No entry in voicemail config file for
''

I'm totally out of ideas now. Anyone else got a clue
to lend me?

Thanks,
Maya




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Re: [Asterisk-Users] Asterisk-MySQL: Not loading voicemail config fromMySQL

[beonice]

By the way, I did fix the typo in my users table so
now the context is 'voicepulse_connect_context', just
like in the extensions.conf. That didn't fix the
problem.

Cheers,
Maya

--- Matthew Boehm <[EMAIL PROTECTED]> wrote:

> What version of asterisk?
> 
> -Matthew
> 
> ----- Original Message - 
> From: "beonice" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial
> Discussion"
> 
> Sent: Friday, February 11, 2005 10:36 AM
> Subject: [Asterisk-Users] Asterisk-MySQL: Not
> loading voicemail config
> fromMySQL
> 
> 
> > Folks,
> >
> > I'm trying to get Asterisk to load my voicemail
> > configuration from MySQL. I've followed the
> > instructions at:
> >
> >
>
http://www.voip-info.org/wiki-Asterisk+voicemail+database
> >
> > I restarted Asterisk, but no luck: the
> voicemail.conf
> > does not get updated. I started with a sample
> > voicemail.conf that I found on the Wiki. Or was it
> > from Voicepulse? I can't remember. For initial
> > testing, I added extensions  and 100 in the
> > [voicepulse_connect_context] with appropriate
> settings
> > in extensions.conf to direct incoming calls to
> those
> > mailboxes, and that works. I was expecting that
> after
> > I added in the db details, reloading or restarting
> > Asterisk would add the new extension from MySQL's
> > 'users' table into the voicemail.conf. It doesn't.
> >
> > As soon as I type  (the beginning of the
> mailbox
> > and also the extension number), I get the message:
> >
> > *CLI> Feb 11 08:38:38 WARNING[5224]:
> > app_voicemail.c:1539 leave_voicemail: No entry in
> > voicemail config file for ''
> >
> > If I add a line for  into my voicemail.conf,
> all
> > works well.
> >
> > Please help me understand what is going on here!
> >
> > Thanks,
> > Maya.
> >
> > --- My configuration ---
> >
> > My 'users' table has 1 row only, for testing
> purposes:
> >
> >
>
+---++--+--+
> ---+---+++
> > | context   | mailbox|
> password |
> > fullname | email | pager | options
> > | stamp  |
> >
>
+---++--+--+
> ---+---+++
> > | voicemail_connect_context |    | 1234   
>  |
> > Moron Tester | [EMAIL PROTECTED]  |   |
> attach=yes
> > | 00 |
> >
>
+---++--+--+
> ---+---+++
> >
> > --
> > The appropriate settings from extensions.conf:
> >
> > [voicepulse_connect_context]  ; <-- Should match
> the
> > context you have
> >   ; under [voicepulse-in-01]
> in
> > iax.conf
> >
> > exten => 100,1,Playback(tt-monkeys)
> > exten => 100,2,Record(/tmp/asterisk-recording:gsm)
> > ;exten => 100,3,Wait(2)
> > exten => 100,3,Playback(/tmp/asterisk-recording)
> > ;exten => 100,5,Wait(2)
> > exten => ,1,Playback(transfer,skip)
> > exten => ,2,VoiceMail,u
> > exten => ,102,VoiceMail,b
> > exten => ,1,VoiceMail,u
> > --
> >
> > My complete voicemail.conf looks like this:
> > ;
> > ; Voicemail Configuration
> > ;
> > [general]
> > ; Default formats for writing Voicemail
> > ;format=g723sf|wav49|wav
> > format=wav49|gsm|wav
> > ; Who the e-mail notification should appear to
> come
> > from
> > [EMAIL PROTECTED]
> > ; Should the email contain the voicemail as an
> > attachment
> > attach=yes
> > ; Maximum length of a voicemail message in seconds
> > ;maxmessage=180
> > ; Minimum length of a voicemail message in seconds
> > ;minmessage=3
> > ; Maximum length of greetings in seconds
> > ;maxgreet=60
> > ; How many miliseconds to skip forward/back when
> > rew/ff in message playback
> > skipms=3000
> > ; How many seconds of silence before we end the
> > recording
> > maxsilence=10
> > ; Silence threshold (what we consider silence, the
> > lower, the more sensitive)
> > silencethreshold=128
> > ; Max number of failed login attempts
> > maxlogins=3
> > ; If you need to have an external program, i.e.
>

Re: [Asterisk-Users] Asterisk-MySQL: Not loading voicemail config fromMySQL

[beonice]

--- Matthew Boehm <[EMAIL PROTECTED]> wrote:

> What version of asterisk?
> 
> -Matthew
> 

Asterisk CVS-v1-0-12/12/04-15:58:29 built by
[EMAIL PROTECTED] on a i686 running WhiteBox
Enterprise Linux

By the way, I _have_ created an ast_config db and the
content of my ast_config table is:
++++---++--+--+-+
| id | cat_metric | var_metric | commented | filename 
 | category | var_name | var_val |
++++---++--+--+-+
|  1 |  0 |  0 | 0 |
voicemail.conf | default  |  | |
++++---++--+--+-+

I've also created etc/asterisk/configs/res_odbc.conf
as described in: 

http://voip-info.org/wiki-Asterisk+res_config



My extconfig.conf says:
[settings]

;uncomment to load queues.conf via the db engine.
;queues.conf => odbc

voicemail.conf => odbc


Unfortunately, I'm not sure what values to put in for 
[mysql1]
dsn = MySQL-asterisk
username = myuser
password = mypass
pre-connect = yes
 and for 
[ENV]
VAR=VALUE
 
I suspect this MAY be the problem. :) I'm unable to
guess what to substitute as an appropriate dsn value
and what to put into the [ENV] section.

Thanks,
Maya




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[Asterisk-Users] Asterisk-MySQL: Not loading voicemail config from MySQL

[beonice]

Folks,

I'm trying to get Asterisk to load my voicemail
configuration from MySQL. I've followed the
instructions at:

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

I restarted Asterisk, but no luck: the voicemail.conf
does not get updated. I started with a sample
voicemail.conf that I found on the Wiki. Or was it
from Voicepulse? I can't remember. For initial
testing, I added extensions  and 100 in the
[voicepulse_connect_context] with appropriate settings
in extensions.conf to direct incoming calls to those
mailboxes, and that works. I was expecting that after
I added in the db details, reloading or restarting
Asterisk would add the new extension from MySQL's
'users' table into the voicemail.conf. It doesn't.

As soon as I type  (the beginning of the mailbox
and also the extension number), I get the message:

*CLI> Feb 11 08:38:38 WARNING[5224]:
app_voicemail.c:1539 leave_voicemail: No entry in
voicemail config file for ''

If I add a line for  into my voicemail.conf, all
works well.

Please help me understand what is going on here!

Thanks,
Maya.

--- My configuration ---

My 'users' table has 1 row only, for testing purposes:

+---++--+--+---+---+++
| context   | mailbox| password |
fullname | email | pager | options   
| stamp  |
+---++--+--+---+---+++
| voicemail_connect_context |    | 1234 |
Moron Tester | [EMAIL PROTECTED]  |   | attach=yes
| 00 |
+---++--+--+---+---+++

--
The appropriate settings from extensions.conf:

[voicepulse_connect_context]  ; <-- Should match the
context you have
  ; under [voicepulse-in-01] in
iax.conf

exten => 100,1,Playback(tt-monkeys)
exten => 100,2,Record(/tmp/asterisk-recording:gsm)
;exten => 100,3,Wait(2)
exten => 100,3,Playback(/tmp/asterisk-recording)
;exten => 100,5,Wait(2)
exten => ,1,Playback(transfer,skip)
exten => ,2,VoiceMail,u
exten => ,102,VoiceMail,b
exten => ,1,VoiceMail,u
--

My complete voicemail.conf looks like this:
;
; Voicemail Configuration
;
[general]
; Default formats for writing Voicemail
;format=g723sf|wav49|wav
format=wav49|gsm|wav
; Who the e-mail notification should appear to come
from
[EMAIL PROTECTED]
; Should the email contain the voicemail as an
attachment
attach=yes
; Maximum length of a voicemail message in seconds
;maxmessage=180
; Minimum length of a voicemail message in seconds
;minmessage=3
; Maximum length of greetings in seconds
;maxgreet=60
; How many miliseconds to skip forward/back when
rew/ff in message playback
skipms=3000
; How many seconds of silence before we end the
recording
maxsilence=10
; Silence threshold (what we consider silence, the
lower, the more sensitive)
silencethreshold=128
; Max number of failed login attempts
maxlogins=3
; If you need to have an external program, i.e.
/usr/bin/myapp
; called when a voicemail is left, delivered, or your
voicemailbox
; is checked, uncomment this:
;externnotify=/usr/bin/myapp
; If you need to have an external program, i.e.
/usr/bin/myapp
; called when a voicemail password is changed,
; uncomment this:
;externpass=/usr/bin/myapp
; For the directory, you can override the intro file
if you want
;directoryintro=dir-intro
; The character set for voicemail messages can be
specified here
;charset=ISO-8859-1
; The ADSI feature descriptor number to download to
;adsifdn=000F
; The ADSI security lock code
;adsisec=9BDBF7AC
; The ADSI voicemail application version number.
;adsiver=1
; Skip the "[PBX]:" string from the message title
pbxskip=yes
; Change the From: string
fromstring=The Asterisk PBX
;
;Change the From: string for pager messages
;pagerfromstring=The Asterisk PBX
;
; Change the email body and/or subject, variables:
; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX,
VM_CALLERID, VM_DATE
;
; Note: The emailbody config row can be up to 512
characters due to a limitation in
;   asterisk config files.
;emailsubject=New VM (${VM_MSGNUM}) - ${VM_DUR} long
in mailbox ${VM_MAILBOX} from ${VM_CALLERID}
emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you
know you were just left a ${VM_DUR} long message
(number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from
${VM_CALLERID}, on ${VM_DATE} so you might\nwant to
check it when you get a chance.  Thanks!
;
; You can override the default program to send e-mail
if you wish, too
;
;mailcmd=/usr/sbin/sendmail -t
;
; Users may be located in different timezones, or may
have different
; message announcements for their introductory message
when they enter
; the voicemail system. Set the message and the
timezone each user
; hears here. Set the user into one of these zones
with the tz= attri

Re: [Asterisk-Users] Wireless LANs and Asterisk

[beonice]

Mike,

I'm using Asterisk over a wireless LAN (Netgear
something or the other). I use my desktop as the
server and a laptop for remote administration and
testing. So far, I haven't had any major problems that
I would attribute to the wireless connectivity.

Cheers,
BeOnIce

--- Mike Meyer <[EMAIL PROTECTED]> wrote:

> Has anyone had any experience with wireless LANs and
> Asterisk?
> 
> We have and here are my impressions.
> 
> We configured an Asterisk in the office as a
> precaution to see how it
> would work for our own retail customers. Our office
> is open space, about
> 800 sq ft. (20x40 area). We use Snom200 and
> Grandstream SIP phones.
> 
> Using the latest Linksys wireless access point
> (WAP54g) and 3 wireless
> bridges (WET54g), I have found that it works most of
> the time with WPA
> encryption on, but will occasionally drop voice
> (loosing packets). With
> no encryption on the WLAN it seems to work without a
> hitch! Using a less
> CPU intense encryption such as 64bit WEP, things
> also work fine. There
> must be too much delay with higher rate encryption.
> 
> Also we had one bridge that seemed to be a week
> puppy in the litter. It
> could only muster 60-70% signal strength. It seemed
> to have problems
> under all configurations. Finally we positioned it
> such that it too
> works well running WEP 64b. I wonder if having 3
> wireless bridges in
> close proximity would have anything to do with the
> signal strength? I
> would doubt it though.
> 
> Anyone else with other experiences to share
> regarding wireless LANs and
> encryption? I'd me interested to hear them.
> 
> 
> Thanks,
> Mike Meyer
> GenDesign Corporation
> 
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[Asterisk-Users] Why does Asterisk Hangup cause server to freeze?

[beonice]

Hello all.

I'm still investigating the cause of freezes on my
asterisk server. It's a minimal installation: the only
things I remember running are httpd, sshd, sendmail
and asterisk itself. I have a DID from Voicepulse. No
telephony cards or SIP phones ... I'm just trying to
figure out the voicemail issues at this point. So a
call comes in, and the caller can type a voicemail
number and leave a message. So far so good.

I'm using Asterisk CVS-v1-0-12/12/04-15:58:29 on a
i686 running WhiteBox Enterprise Linux.

Unfortunately, Asterisk has been crashing multiple
times a day. When it crashes, it takes the whole
machine with it ... I can't make the server accept any
keystrokes or mouse events, no virtual consoles,
nothing. Only a hard reboot satisfies this beast. When
I reboot, there's nothing particularly interesting in
/var/log/asterisk/messages either.

After a LOT of reboots, I finally realised that the
occasions when the machine would freeze were exactly
the occasions when my dialplan indicated that the next
priority should cause a Hangup. For example, I had:

exten => ,1,Playback(transfer,skip)
exten => ,2,VoiceMail,u
exten => ,3,Hangup
exten => ,102,VoiceMail,b
exten => t,1,Hangup
exten => i,1,Hangup


And this would reliably freeze the machine after I
left a voicemail in mailbox , or pressed an
invalid key, or allowed Asterisk to timeout.
Basically, all I wanted was for Asterisk to hangup on
the caller. :)

Now I've changed all instances of Hangup to
SoftHangup, and the problem does not occur any more.
This is unsatisfactory, though, because I DO want
asterisk to hang up the connection when it sees fit.

Does anyone have any ideas why Hangup isn't working
for me? It seems a simple enough command, and others
are obviously using it! Has anyone else encountered
problems with Hangup? Some folks had suggested that
the freeze could be indicative of a hardware problem,
but it doesn't seem likely given that changing Hangup
to SoftHangup seems to remove the problem.

Any help would be much appreciated.

Thanks,
BeOnIce.




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RE: [Asterisk-Users] Asterisk causing server to hang ... any hints?

[beonice]

By the way, in case it matters, I don't have any
digium cards or SIP phones attached to the Asterisk
setup. Right now, it's all IAX, just receiving
incoming calls.

Cheers,
BeOnIce.

--- "Chamberland-Larose, Guillaume" <[EMAIL PROTECTED]>
wrote:

> If the asterisk process is hung up you should be
> able to debug it. If
> the whole machine is hung up this is a totally
> different issue isn't it.
> If you're running linux and the machine locks up
> that often, you must
> have a hardware problem. Or maybe you just think the
> machine is locked
> up while it actually isn't. 
> 
> You might want to look on google and the wiki about
> deadlocks, I've seen
> a lot of information around on how to debug them and
> report them.
> 
> Guills
> 
> > -Original Message-
> > From: beonice [mailto:[EMAIL PROTECTED] 
> > Sent: Tuesday, February 08, 2005 3:41 PM
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] Asterisk causing server
> to hang ... 
> > any hints?
> > 
> > I am trying to set up a simple Asterisk server.
> All it's 
> > going to do for now is to act as my voicemail box.
> I've got a 
> > DID from Voicepulse, and am using IAX (I'll get to
> SIP 
> > someday when I want to circumvent the phone
> company for 
> > long-distance, but for now I'd be happy to get a
> trial 
> > version of Asterisk running).
> > 
> > So far, I've managed to set up voicemail.conf, 
> > extensions.conf and iax.conf so that incoming
> calls are 
> > answered and a voicemail can be left. Initially I
> did not 
> > have a handler for the timeout case, and Asterisk
> would 
> > complain about the missing handler and
> occasionally would 
> > crash the server after several messages had been
> left (I'm 
> > the only one testing it, and the server has
> nothing else 
> > running, just Asterisk), requiring a hard reboot. 
> > 
> > To avoid this, I put in a handler for the timeout.
> > Now, Asterisk crashes the server as soon as I
> leave myself a 
> > message! This is irritating. Hard reboots
> everytime someone 
> > leaves me a voicemail is not going to be something
> I can 
> > trust if I go on vacation (hopefully someday
> soon).
> > 
> > Any ideas what I am doing wrong?
> > 
> > Here's the change I made to my extension file
> (this used to 
> > crash occasionally):
> > [old-context]
> > exten => ,2,VoiceMail,u
> > exten => ,3,Hangup
> > exten => ,102,VoiceMail,b
> > exten => ,3,Hangup
> > exten => ,103,Hangup
> > 
> > 
> > [new-context] ;; crashes all the time!
> > exten => ,2,VoiceMail,u
> > exten => ,3,Hangup
> > exten => ,102,VoiceMail,b
> > exten => ,3,Hangup
> > exten => ,103,Hangup
> > exten => ,1,Playback(transfer,skip)
> > exten => ,2,Ringing
> > exten => ,3,Wait(2)
> > exten => ,4,VoiceMail,u
> > exten => ,104,VoiceMail,b
> > exten => 1001,1,Ringing
> > exten => 1001,2,Wait(2)
> > exten => 1001,3,VoicemailMain
> > exten => _NXXNXX,1,Background(beep)
> > ;exten => _NXXNXX,2,SayDigits(${EXTEN}) ;exten
> => 
> > _NXXNXX,3,Goto(testdtmf|s|1) exten =>
> t,1,Ringing exten 
> > => t,2,Hangup exten => i,1,Ringing exten =>
> i,2,Hangup exten 
> > => a,1,VoicemailMain,EXTEN exten => a,2,Hangup
> > 
> > I thought that using the
> > exten => t,2,Hangup and the
> > exten => i,2,Hangup
> > would cause Asterisk to hang up on timeout, but
> obviously I 
> > misunderstood. By the way, what _does_ the 'a'
> handler do, 
> > anyway? I believe 'i' is for invalid key and 't'
> is for 
> > timeout. I'm not sure what 'a'
> > does.
> > 
> > The last time I tried to leave myself voicemail,
> here's the 
> > output I got. And yes, it crashed. I've got
> Asterisk running 
> > at a level of 4 in terms of verbosity, i.e., I
> start it with 
> > /usr/sbin/asterisk -cp
> > 
> > -- Playing 'beep' (language 'en')
> > Feb  8 14:07:29 DEBUG[4195]: chan_iax2.c:5310
> > socket_read: Ooh, voice format changed to 4
> >   == CDR updated on
> > IAX2/[EMAIL PROTECTED]:4569/1
> > -- Executing
> >
>
Playback("IAX2/[EMAIL PROTECTED]:4569/1",
> > "transfer|skip")

RE: [Asterisk-Users] Asterisk causing server to hang ... any hints?

[beonice]

Guillaume,

The problem definitely is that the whole machine locks
up. Even at the console itself, I cannot get it to
accept typing! Even a Ctrl-C or Ctrl-Alt-Del or any
other such combination doesn't work.

I'll check the wiki for information about deadlocks,
thanks.

Cheers,
BeOnIce.
--- "Chamberland-Larose, Guillaume" <[EMAIL PROTECTED]>
wrote:

> If the asterisk process is hung up you should be
> able to debug it. If
> the whole machine is hung up this is a totally
> different issue isn't it.
> If you're running linux and the machine locks up
> that often, you must
> have a hardware problem. Or maybe you just think the
> machine is locked
> up while it actually isn't. 
> 
> You might want to look on google and the wiki about
> deadlocks, I've seen
> a lot of information around on how to debug them and
> report them.
> 
> Guills
> 
> > -Original Message-
> > From: beonice [mailto:[EMAIL PROTECTED] 
> > Sent: Tuesday, February 08, 2005 3:41 PM
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] Asterisk causing server
> to hang ... 
> > any hints?
> > 
> > I am trying to set up a simple Asterisk server.
> All it's 
> > going to do for now is to act as my voicemail box.
> I've got a 
> > DID from Voicepulse, and am using IAX (I'll get to
> SIP 
> > someday when I want to circumvent the phone
> company for 
> > long-distance, but for now I'd be happy to get a
> trial 
> > version of Asterisk running).
> > 
> > So far, I've managed to set up voicemail.conf, 
> > extensions.conf and iax.conf so that incoming
> calls are 
> > answered and a voicemail can be left. Initially I
> did not 
> > have a handler for the timeout case, and Asterisk
> would 
> > complain about the missing handler and
> occasionally would 
> > crash the server after several messages had been
> left (I'm 
> > the only one testing it, and the server has
> nothing else 
> > running, just Asterisk), requiring a hard reboot. 
> > 
> > To avoid this, I put in a handler for the timeout.
> > Now, Asterisk crashes the server as soon as I
> leave myself a 
> > message! This is irritating. Hard reboots
> everytime someone 
> > leaves me a voicemail is not going to be something
> I can 
> > trust if I go on vacation (hopefully someday
> soon).
> > 
> > Any ideas what I am doing wrong?
> > 
> > Here's the change I made to my extension file
> (this used to 
> > crash occasionally):
> > [old-context]
> > exten => ,2,VoiceMail,u
> > exten => ,3,Hangup
> > exten => ,102,VoiceMail,b
> > exten => ,3,Hangup
> > exten => ,103,Hangup
> > 
> > 
> > [new-context] ;; crashes all the time!
> > exten => ,2,VoiceMail,u
> > exten => ,3,Hangup
> > exten => ,102,VoiceMail,b
> > exten => ,3,Hangup
> > exten => ,103,Hangup
> > exten => ,1,Playback(transfer,skip)
> > exten => ,2,Ringing
> > exten => ,3,Wait(2)
> > exten => ,4,VoiceMail,u
> > exten => ,104,VoiceMail,b
> > exten => 1001,1,Ringing
> > exten => 1001,2,Wait(2)
> > exten => 1001,3,VoicemailMain
> > exten => _NXXNXX,1,Background(beep)
> > ;exten => _NXXNXX,2,SayDigits(${EXTEN}) ;exten
> => 
> > _NXXNXX,3,Goto(testdtmf|s|1) exten =>
> t,1,Ringing exten 
> > => t,2,Hangup exten => i,1,Ringing exten =>
> i,2,Hangup exten 
> > => a,1,VoicemailMain,EXTEN exten => a,2,Hangup
> > 
> > I thought that using the
> > exten => t,2,Hangup and the
> > exten => i,2,Hangup
> > would cause Asterisk to hang up on timeout, but
> obviously I 
> > misunderstood. By the way, what _does_ the 'a'
> handler do, 
> > anyway? I believe 'i' is for invalid key and 't'
> is for 
> > timeout. I'm not sure what 'a'
> > does.
> > 
> > The last time I tried to leave myself voicemail,
> here's the 
> > output I got. And yes, it crashed. I've got
> Asterisk running 
> > at a level of 4 in terms of verbosity, i.e., I
> start it with 
> > /usr/sbin/asterisk -cp
> > 
> > -- Playing 'beep' (language 'en')
> > Feb  8 14:07:29 DEBUG[4195]: chan_iax2.c:5310
> > socket_read: Ooh, voice format changed to 4
> >   == CDR updated on
> > IAX2/[EMAIL PROTECTED]:4569/1
> &

[Asterisk-Users] Asterisk causing server to hang ... any hints?

[beonice]

I am trying to set up a simple Asterisk server. All
it's going to do for now is to act as my voicemail
box. I've got a DID from Voicepulse, and am using IAX
(I'll get to SIP someday when I want to circumvent the
phone company for long-distance, but for now I'd be
happy to get a trial version of Asterisk running).

So far, I've managed to set up voicemail.conf,
extensions.conf and iax.conf so that incoming calls
are answered and a voicemail can be left. Initially I
did not have a handler for the timeout case, and
Asterisk would complain about the missing handler and
occasionally would crash the server after several
messages had been left (I'm the only one testing it,
and the server has nothing else running, just
Asterisk), requiring a hard reboot. 

To avoid this, I put in a handler for the timeout.
Now, Asterisk crashes the server as soon as I leave
myself a message! This is irritating. Hard reboots
everytime someone leaves me a voicemail is not going
to be something I can trust if I go on vacation
(hopefully someday soon).

Any ideas what I am doing wrong?

Here's the change I made to my extension file (this
used to crash occasionally):
[old-context]
exten => ,2,VoiceMail,u
exten => ,3,Hangup
exten => ,102,VoiceMail,b
exten => ,3,Hangup
exten => ,103,Hangup


[new-context] ;; crashes all the time!
exten => ,2,VoiceMail,u
exten => ,3,Hangup
exten => ,102,VoiceMail,b
exten => ,3,Hangup
exten => ,103,Hangup
exten => ,1,Playback(transfer,skip)
exten => ,2,Ringing
exten => ,3,Wait(2)
exten => ,4,VoiceMail,u
exten => ,104,VoiceMail,b
exten => 1001,1,Ringing
exten => 1001,2,Wait(2)
exten => 1001,3,VoicemailMain
exten => _NXXNXX,1,Background(beep)
;exten => _NXXNXX,2,SayDigits(${EXTEN})
;exten => _NXXNXX,3,Goto(testdtmf|s|1)
exten => t,1,Ringing
exten => t,2,Hangup
exten => i,1,Ringing
exten => i,2,Hangup
exten => a,1,VoicemailMain,EXTEN
exten => a,2,Hangup

I thought that using the 
exten => t,2,Hangup and the
exten => i,2,Hangup
would cause Asterisk to hang up on timeout, but
obviously I misunderstood. By the way, what _does_ the
'a' handler do, anyway? I believe 'i' is for invalid
key and 't' is for timeout. I'm not sure what 'a'
does.

The last time I tried to leave myself voicemail,
here's the output I got. And yes, it crashed. I've got
Asterisk running at a level of 4 in terms of
verbosity, i.e., I start it with /usr/sbin/asterisk
-cp

-- Playing 'beep' (language 'en')
Feb  8 14:07:29 DEBUG[4195]: chan_iax2.c:5310
socket_read: Ooh, voice format changed to 4
  == CDR updated on
IAX2/[EMAIL PROTECTED]:4569/1
-- Executing
Playback("IAX2/[EMAIL PROTECTED]:4569/1",
"transfer|skip") in new stack
-- Playing 'transfer' (language 'en')
-- Executing
VoiceMail("IAX2/[EMAIL PROTECTED]:4569/1",
"u") in new stack
Feb  8 14:07:42 DEBUG[4195]: app_voicemail.c:1381
leave_voicemail:
voicemail/voicepulse_connect_context//unavail
doesn't exist, doing what we can
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
Feb  8 14:07:53 DEBUG[4195]: app.c:549
ast_play_and_record: play_and_record: ,
/var/spool/asterisk/voicemail/voicepulse_connect_context//INBOX/msg0001,
'wav49|gsm|wav'
Feb  8 14:07:53 DEBUG[4195]: app.c:566
ast_play_and_record: Recording Formats: sfmts=wav49
-- x=0, open writing: 
/var/spool/asterisk/voicemail/voicepulse_connect_context//INBOX/msg0001
format: wav49, 0x814ded0
-- x=1, open writing: 
/var/spool/asterisk/voicemail/voicepulse_connect_context//INBOX/msg0001
format: gsm, 0x814dff0
-- x=2, open writing: 
/var/spool/asterisk/voicemail/voicepulse_connect_context//INBOX/msg0001
format: wav, 0x814e100
-- User ended message by pressing #
-- Playing 'auth-thankyou' (language 'en')
localhost*CLI>

At this point, the server itself seems to hang. I can
do nothing but a physical reboot of the machine.

HELP.

Thanks,
BeOnIce.




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Re: [Asterisk-Users] Newbie question: Can't start up asterisk

[beonice]

Thanks, Brian. I'm now a happy camper!

Cheers,
BeOnIce

--- Brian Dingman <[EMAIL PROTECTED]> wrote:

> It has to do with spandsp and receiving incoming
> faxes. This should
> probably be updated in the documentation.
> 
> 
> On Tue, 18 Jan 2005 17:09:04 -0800 (PST), beonice
> <[EMAIL PROTECTED]> wrote:
> > 
> > --- Brian Dingman <[EMAIL PROTECTED]> wrote:
> > 
> > > Put /usr/local/lib in /etc/ld.so.conf then run
> > > ldconfig.
> > >
> > 
> > Hmm. I don't understand what that did, but THANKS,
> > Brian. It seems to have at least got Asterisk to
> > successfully run. Now I can have fun with
> configuring
> > it!
> > 
> > Thanks a bunch!
> > 
> > BeOnIce.
> ___
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>
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Re: [Asterisk-Users] Newbie question: Can't start up asterisk

[beonice]

--- Brian Dingman <[EMAIL PROTECTED]> wrote:

> Put /usr/local/lib in /etc/ld.so.conf then run
> ldconfig.
> 

Hmm. I don't understand what that did, but THANKS,
Brian. It seems to have at least got Asterisk to
successfully run. Now I can have fun with configuring
it!

Thanks a bunch!

BeOnIce.




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Re: [Asterisk-Users] Newbie question: Can't start up asterisk

[beonice]

--- Denis Galvão - iSolve <[EMAIL PROTECTED]> wrote:

> Did you install mpg123 from source!? Or you're using
> a distro native 
> version!?
> 
> You have to get the mpg123 from its website and then
> get it compiled to your 
> suystem.

I believe I have a freshly compiled version. I've got
version 0.59r, is what running mpg123 tells me.

Cheers,
BeOnIce




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Re: [Asterisk-Users] Newbie question: Can't start up asterisk

[beonice]

--- Matt Riddell <[EMAIL PROTECTED]> wrote:

> Colin Anderson wrote:
> >>Ouch ... error while writing audio data: : Broken
> pipe
> > 
> > 
> > Did you run "make samples" from /usr/src/asterisk?
> 
> Yeah he has config files.
> 
> What is the response you get running mpg123?
> 
> I.E. what are the first four lines?
> 
> I.E. what version?
> 
> :)
> 
> -- 
> Cheers,
> 
> Matt Riddell

Well, I just tried running mpg123. Here is the version
information:


High Performance MPEG 1.0/2.0/2.5 Audio Player for
Layer 1, 2 and 3.
Version 0.59r (1999/Jun/15). Written and copyrights by
Michael Hipp.
Uses code from various people. See 'README' for more!
THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE
AT YOUR OWN RISK!




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RE: [Asterisk-Users] Newbie question: Can't start up asterisk

[beonice]

--- Colin Anderson <[EMAIL PROTECTED]>
wrote:

> >Ouch ... error while writing audio data: : Broken
> pipe
> 
> Did you run "make samples" from /usr/src/asterisk?

Hmm. I thought I had, but I've attempted this Asterisk
installation so many times that I seem to have skipped
it on this incarnation. I just did the "make samples",
and now I get this output (note that it still dies
with the same message, but now there is more
context!):

Set to realtime thread
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk CVS-v1-0-12/12/04-15:58:29, Copyright (C)
1999-2004 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
=
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started
/var/log/asterisk/event_log
  == Manager registered action Ping
  == Manager registered action Events
  == Manager registered action Logoff
  == Manager registered action Hangup
  == Manager registered action Status
  == Manager registered action Setvar
  == Manager registered action Getvar
  == Manager registered action Redirect
  == Manager registered action Originate
  == Manager registered action MailboxStatus
  == Manager registered action Command
  == Manager registered action ExtensionState
  == Manager registered action AbsoluteTimeout
  == Manager registered action MailboxCount
  == Manager registered action ListCommands
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/rtp.conf': Found
  == RTP Allocating from port range 1 -> 2
Asterisk PBX Core Initializing
Registering builtin applications:
 [AbsoluteTimeout]
  == Registered application 'AbsoluteTimeout
'
 [Answer]
  == Registered application 'Answer'
 [BackGround]
  == Registered application 'BackGround'
 [Busy]
  == Registered application 'Busy'
 [Congestion]
  == Registered application 'Congestion'
 [DigitTimeout]
  == Registered application 'DigitTimeout'
 [Goto]
  == Registered application 'Goto'
 [GotoIf]
 == Registered application 'GotoIf'
 [GotoIfTime]
  == Registered application 'GotoIfTime'
 [Hangup]
  == Registered application 'Hangup'
 [NoOp]
  == Registered application 'NoOp'
 [Prefix]
  == Registered application 'Prefix'
 [Progress]
  == Registered application 'Progress'
 [ResetCDR]
  == Registered application 'ResetCDR'
 [ResponseTimeout]
  == Registered application 'ResponseTimeout
'
 [Ringing]
  == Registered application 'Ringing'
 [SayNumber]
  == Registered application 'SayNumber'
 [SayDigits]
  == Registered application 'SayDigits'
[SayAlpha]%)
  == Registered application 'SayAlpha'
 [SayPhonetic]
  == Registered application 'SayPhonetic'
 [SetAccount]
  == Registered application 'SetAccount'
 [SetAMAFlags]
  == Registered application 'SetAMAFlags'
 [SetGlobalVar]
  == Registered application 'SetGlobalVar'
 [SetLanguage]
  == Registered application 'SetLanguage'
 [SetVar]
  == Registered application 'SetVar'
 [StripMSD]
  == Registered application 'StripMSD'
 [Suffix]
  == Registered application 'Suffix'
 [Wait]
  == Registered application 'Wait'
 [WaitExten]
  == Registered application 'WaitExten'
Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf': Found
 [chan_modem.so] => (Generic Voice Mo
dem Driver)
  == Parsing '/etc/asterisk/modem.conf': Found
  == Loading modem driver chan_modem_aopen.so =>
(A/Ope
n (Rockwell Chipset) ITU-2 VoiceModem Driver)
  == Registered channel type 'Modem' (Generic Voice
Modem Chan
nel Driver)
 [res_musiconhold.so] => (Music On Ho
ld Resource)
  == Parsing '/etc/asterisk/musiconhold.conf': Found
  == Registered application 'MusicOnHold'
  == Registered application 'WaitMusicOnHold
'
  == Registered application 'SetMusicOnHold'
 [res_adsi.so] => (ADSI Resource
)
  == Parsing '/etc/asterisk/adsi.conf': Found
 [res_features.so] => (Call Parking R
esource)
  == Parsing '/etc/asterisk/features.conf': Found
-- Registered extension context 'parkedcalls'
-- Added extension '700' priority 1 to parkedcalls
  == Registered application 'ParkedCall'
  == Registered application 'Park'
  == Manager registered action ParkedCalls
 [res_crypto.so] => (Cryptographic Di
gital Signatures)
-- Loaded PUBLIC key 'iaxtel'
-- Loaded PUBLIC key 'freeworlddialup'
-- Loaded PUBLIC key 'voicepulse01'
 [res_indications.so] => (Indications
 Configuration)
  == Parsing '/etc/asterisk/indications.conf': Found
-- Registered indication country 'cl'
-- Registered indication country 'tw'
-- Registered indication country 'us'
-- Registered indication country 'au'
-- Registered indication country 'fr'
-- Registered indication country 'de'
-- Registered indication country 'nl'
-- Registered indication country 'uk'
-- Registered indication country 'fi'
-- Registered indication country 'no'
-- Registered indication country 'br'
-- Registered indication country 'za'
-- Registered indica

Re: [Asterisk-Users] Newbie question: Can't start up asterisk

[beonice]

Denis wrote:

--- Denis Galvão - iSolve <[EMAIL PROTECTED]> wrote:

> Em Ter 18 Jan 2005 20:43, Matt Riddell escreveu:
> > beonice wrote:
> > > "Ouch ... error while writing audio data: :
> Broken
> > > pipe"
> >
> > What are the messages before this?
> 
> Matt I think that is something related to mpg123...
> 
> -- 
> D e n i s   G a l v ã o
> iSolve - Solve Is Our Business
> Av. Candido de Abreu, 526 1206B
> CEP: 80530-000 - Curitiba - PR
> +55 41 252-2977
> http://www.isolve.com.br

That _seems_ to be a possibility. But I'm not really
sure. I made sure that there is a symbolic link in
/usr/bin to mpg123 ... the actual version is in
/usr/local/bin.

Thanks. By the way, I accidentally created a new post
with the details of the output instead of responding
to  Matt's question right here ... but here is the
output again:

[EMAIL PROTECTED] asterisk]# /usr/sbin/asterisk -cp
Set to realtime thread
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Not found
(No such file or directory)
Asterisk CVS-v1-0-12/12/04-15:58:29, Copyright (C)
1999-2004 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
=
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started
/var/log/asterisk/event_log
  == Manager registered action Ping
  == Manager registered action Events
  == Manager registered action Logoff
  == Manager registered action Hangup
  == Manager registered action Status
  == Manager registered action Setvar
  == Manager registered action Getvar
  == Manager registered action Redirect
  == Manager registered action Originate
  == Manager registered action MailboxStatus
  == Manager registered action Command
  == Manager registered action ExtensionState
  == Manager registered action AbsoluteTimeout
  == Manager registered action MailboxCount
  == Manager registered action ListCommands
  == Parsing '/etc/asterisk/manager.conf': Found
Asterisk Management interface listening on port 5038
  == Parsing '/etc/asterisk/rtp.conf': Found
  == RTP Allocating from port range 1 -> 2
Asterisk PBX Core Initializing
Registering builtin applications:
 [AbsoluteTimeout]
  == Registered application 'AbsoluteTimeout'
 [Answer]
  == Registered application 'Answer'
 [BackGround]
  == Registered application 'BackGround'
 [Busy]
  == Registered application 'Busy'
 [Congestion]
  == Registered application 'Congestion'
 [DigitTimeout]
  == Registered application 'DigitTimeout'
 [Goto]
  == Registered application 'Goto'
 [GotoIf]
  == Registered application 'GotoIf'
 [GotoIfTime]
  == Registered application 'GotoIfTime'
 [Hangup]
  == Registered application 'Hangup'
 [NoOp]
  == Registered application 'NoOp'
 [Prefix]
  == Registered application 'Prefix'
 [Progress]
  == Registered application 'Progress'
 [ResetCDR]
  == Registered application 'ResetCDR'
 [ResponseTimeout]
  == Registered application 'ResponseTimeout'
 [Ringing]
  == Registered application 'Ringing'
 [SayNumber]
  == Registered application 'SayNumber'
 [SayDigits]
  == Registered application 'SayDigits'
 [SayAlpha]
  == Registered application 'SayAlpha'
 [SayPhonetic]
  == Registered application 'SayPhonetic'
 [SetAccount]
  == Registered application 'SetAccount'
 [SetAMAFlags]
  == Registered application 'SetAMAFlags'
 [SetGlobalVar]
  == Registered application 'SetGlobalVar'
 [SetLanguage]
  == Registered application 'SetLanguage'
 [SetVar]
  == Registered application 'SetVar'
 [StripMSD]
  == Registered application 'StripMSD'
 [Suffix]
  == Registered application 'Suffix'
 [Wait]
  == Registered application 'Wait'
 [WaitExten]
  == Registered application 'WaitExten'
Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf': Found
 [chan_modem.so] => (Generic Voice Modem Driver)
  == Parsing '/etc/asterisk/modem.conf': Found
  == Loading modem driver chan_modem_aopen.so =>
(A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver)
  == Registered channel type 'Modem' (Generic Voice
Modem Channel Driver)
 [res_musiconhold.so] => (Music On Hold Resource)
  == Parsing '/etc/asterisk/musiconhold.conf': Found
Junk at the beginning 49443303
Warning, flexibel rate not heavily tested!
  == Registered application 'MusicOnHold'
  == Registered application 'WaitMusicOnHold'
  == Registered application 'SetMusicOnHold'
 [res_adsi.so] => (ADSI Resource)
  == Parsing '/etc/asterisk/adsi.conf': Not f

Re: [Asterisk-Users] Newbie question: Can't start up asterisk

[beonice]

Here is the entire output until it dies:

[EMAIL PROTECTED] asterisk]# /usr/sbin/asterisk -cp
Set to realtime thread
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Not found
(No such file or directory)
Asterisk CVS-v1-0-12/12/04-15:58:29, Copyright (C)
1999-2004 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
=
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started
/var/log/asterisk/event_log
  == Manager registered action Ping
  == Manager registered action Events
  == Manager registered action Logoff
  == Manager registered action Hangup
  == Manager registered action Status
  == Manager registered action Setvar
  == Manager registered action Getvar
  == Manager registered action Redirect
  == Manager registered action Originate
  == Manager registered action MailboxStatus
  == Manager registered action Command
  == Manager registered action ExtensionState
  == Manager registered action AbsoluteTimeout
  == Manager registered action MailboxCount
  == Manager registered action ListCommands
  == Parsing '/etc/asterisk/manager.conf': Found
Asterisk Management interface listening on port 5038
  == Parsing '/etc/asterisk/rtp.conf': Found
  == RTP Allocating from port range 1 -> 2
Asterisk PBX Core Initializing
Registering builtin applications:
 [AbsoluteTimeout]
  == Registered application 'AbsoluteTimeout'
 [Answer]
  == Registered application 'Answer'
 [BackGround]
  == Registered application 'BackGround'
 [Busy]
  == Registered application 'Busy'
 [Congestion]
  == Registered application 'Congestion'
 [DigitTimeout]
  == Registered application 'DigitTimeout'
 [Goto]
  == Registered application 'Goto'
 [GotoIf]
  == Registered application 'GotoIf'
 [GotoIfTime]
  == Registered application 'GotoIfTime'
 [Hangup]
  == Registered application 'Hangup'
 [NoOp]
  == Registered application 'NoOp'
 [Prefix]
  == Registered application 'Prefix'
 [Progress]
  == Registered application 'Progress'
 [ResetCDR]
  == Registered application 'ResetCDR'
 [ResponseTimeout]
  == Registered application 'ResponseTimeout'
 [Ringing]
  == Registered application 'Ringing'
 [SayNumber]
  == Registered application 'SayNumber'
 [SayDigits]
  == Registered application 'SayDigits'
 [SayAlpha]
  == Registered application 'SayAlpha'
 [SayPhonetic]
  == Registered application 'SayPhonetic'
 [SetAccount]
  == Registered application 'SetAccount'
 [SetAMAFlags]
  == Registered application 'SetAMAFlags'
 [SetGlobalVar]
  == Registered application 'SetGlobalVar'
 [SetLanguage]
  == Registered application 'SetLanguage'
 [SetVar]
  == Registered application 'SetVar'
 [StripMSD]
  == Registered application 'StripMSD'
 [Suffix]
  == Registered application 'Suffix'
 [Wait]
  == Registered application 'Wait'
 [WaitExten]
  == Registered application 'WaitExten'
Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf': Found
 [chan_modem.so] => (Generic Voice Modem Driver)
  == Parsing '/etc/asterisk/modem.conf': Found
  == Loading modem driver chan_modem_aopen.so =>
(A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver)
  == Registered channel type 'Modem' (Generic Voice
Modem Channel Driver)
 [res_musiconhold.so] => (Music On Hold Resource)
  == Parsing '/etc/asterisk/musiconhold.conf': Found
Junk at the beginning 49443303
Warning, flexibel rate not heavily tested!
  == Registered application 'MusicOnHold'
  == Registered application 'WaitMusicOnHold'
  == Registered application 'SetMusicOnHold'
 [res_adsi.so] => (ADSI Resource)
  == Parsing '/etc/asterisk/adsi.conf': Not found (No
such file or directory)
 [res_features.so] => (Call Parking Resource)
  == Parsing '/etc/asterisk/features.conf': Not found
(No such file or directory)
  == Parsing '/etc/asterisk/parking.conf': Found
-- Registered extension context 'default'
-- Added extension '700' priority 1 to default
  == Registered application 'ParkedCall'
  == Registered application 'Park'
  == Manager registered action ParkedCalls
 [res_crypto.so] => (Cryptographic Digital Signatures)
-- Loaded PUBLIC key 'iaxtel'
-- Loaded PUBLIC key 'freeworlddialup'
-- Loaded PUBLIC key 'voicepulse01'
 [res_indications.so] => (Indications Configuration)
  == Parsing '/etc/asterisk/indications.conf': Found
-- Registered indication country 'us'
-- Registered indication country 'au'
-- Registered indication country 'fr'
-- Registered indication country 'de'
-- Registered indication country 'nl'
-- Registered indication country 'uk'
-- Registered indication country 'fi'
-- Registered indication country 'no'
-- Registered indication country 'br'
-- Registered indication country 'za'
-- Registered indication country 'it'
-- Setting default indication country to 'us'
  == Registered application 'Playtones'
  == Registered application 'StopPlaytones'
 [res_monit

[Asterisk-Users] Newbie question: Can't start up asterisk

[beonice]

Folks,

I've just successfully set up Asterisk (as part of the
Asterisk Management Portal installation). When I say
"successfully", I mean that I have gone through all
the steps detailed for the installation of AMP and not
hit any snags there. I can connect to my asterisk
server via ssh and can also connect via Http to the
portal to change settings in AMP.

Now I'm trying to start up Asterisk itself. When I
type in "/usr/sbin/asterisk -cp", it does a  bunch
of stuff, then exits with the message 
"Ouch ... error while writing audio data: : Broken
pipe"

>From a Google search, this doesn't seem to be anything
obvious. Can anyone walk me through this? I'd really
like to get asterisk running.

My environment: White Box Enterprise Linux, a
VoicePulse Connect! DID, no telephony cards. I've
tried to edit the system to use the ztdummy, but am
not sure it successfully registered that change.

Any and all help would be much appreciated!

Thanks!

BeOnIce




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