[asterisk-users] Firewall Issue

[RSCL Mumbai]

Hi,

I seem to be facing an intrusion issue, inspite of firewall (script attached).

What am I missing ??

Any suggestions / recommendation are welcome pls.


Best regards,
Sans
#!/bin/bash

echo 0 > /proc/sys/net/ipv4/ip_forward


# Clear any existing firewall stuff before we start
/sbin/iptables --flush


# As the default policies, drop all incoming traffic but allow all
# outgoing traffic.  This will allow us to make outgoing connections
# from any port, but will only allow incoming connections on the ports
# specified below.
/sbin/iptables --policy INPUT DROP
/sbin/iptables --policy FORWARD DROP
/sbin/iptables --policy OUTPUT ACCEPT


# Allow all incoming traffic if it is coming from the local loopback device
/sbin/iptables -A INPUT -i lo -j ACCEPT


# Allow icmp input so that people can ping us
/sbin/iptables -A INPUT -p icmp --icmp-type 8 -m state --state NEW -j ACCEPT


# Allow returning packets
/sbin/iptables -A INPUT -i eth1 -m state --state RELATED,ESTABLISHED -j ACCEPT
/sbin/iptables -A INPUT -i eth0 -m state --state RELATED,ESTABLISHED -j ACCEPT


# Allow incoming traffic on port 8000 for web server & 2200 for SSh
/sbin/iptables -A INPUT -p tcp --dport 8000 -j ACCEPT
/sbin/iptables -A INPUT -p tcp --dport 2200 -j ACCEPT



#
## RESTRICTED SIP ACCESS 
#


# LAN
/sbin/iptables -A INPUT -p tcp -i eth0 -s 192.168.1.0/24 --dport 5060:5062   -j 
ACCEPT
/sbin/iptables -A INPUT -p udp -i eth0 -s 192.168.1.0/24 --dport 5060:5062   -j 
ACCEPT
/sbin/iptables -A INPUT -p udp -i eth0 -s 192.168.1.0/24 --dport 1:2 -j 
ACCEPT



# Allow traffic from VoIP Service Provider
/sbin/iptables -A INPUT -p udp -i eth1 -s 11.11.11.11 --dport 5060:5062   -j 
ACCEPT
/sbin/iptables -A INPUT -p tcp -i eth1 -s 11.11.11.11 --dport 5060:5062   -j 
ACCEPT
/sbin/iptables -A INPUT -p udp -i eth1 -s 11.11.11.11 --dport 1:2 -j 
ACCEPT





# Check new packets are SYN packets for syn-flood protection
/sbin/iptables -A INPUT -p tcp ! --syn -m state --state NEW -j DROP

# Drop fragmented packets
/sbin/iptables -A INPUT -f -j DROP

# Drop malformed XMAS packets
/sbin/iptables -A INPUT -p tcp --tcp-flags ALL ALL -j DROP

# Drop null packets
/sbin/iptables -A INPUT -p tcp --tcp-flags ALL NONE -j DROP

# Log and drop any packets that are not allowed. You will probably want to turn 
off the logging
#/sbin/iptables -A INPUT -j LOG --log-level 4
/sbin/iptables -A INPUT -j REJECT
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Re: [asterisk-users] Receptionist Extension cannot be Pickup()'ed

[Cassius Smith]

I neglected to say ­ all the extensions can be picked up remotely by the
other endpoints, EXCEPT the receptionist phone x3100. When calls go to that
station, they cannot be picked up. Sorry for the necessity to post twice.


From:  Cassius Smith 
Date:  Fri, 05 Aug 2011 15:31:14 -0500
To:  Asterisk Users Mailing List - Non-Commercial Discussion

Subject:  Receptionist Extension cannot be Pickup()'ed

> Hello all,
> I am struggling with an annoying problem. I have an installation with a small
> number of Grandstream  GXP2010 endpoints. Each endpoint has all the extensions
> programmed into the phone for BLF - for instant pickup, transfer or speed
> dial.
> 
> Except for the Receptionist phone, which is handled internally via the "0"
> extension. That extension drops into a [day-menu] context with an IVR after
> the receptionist phone rings for 20 seconds.
> 
> The receptionist phone has a BLF field on all the other phones. But when that
> phone rings, I think something is messing with some channel variable that is
> preventing Pickup() from working.
> 
> ALL Other extensions can be picked up. ONLY the extension(s) I ring from the
> day-menu cannot.
> 
> Here is a snip from my dialplan:
> exten => s,1,NoOp()
>  same => n,Verbose(2,"Processing incoming call from ${CALLERID(all)})
>  same => n(daycheck),GotoIfTime(08:30-16:59,mon-fri,*,*?open)
>  same => n,Set(MENU=night-menu)
>  same => n,Goto(night)
>  same => n(open),Set(MENU=day-menu)
>  same => n,Set(__PICKUPMARK=)
>  same => n,Dial(SIP/3100,20) ; 3100 is receptionists phone
> ; go to IVR if no answer
>  same => n,Goto(playmenu)
>  same => n(night),NoOp()
>  same => n(top),Wait(0.5)
>  same => n,GotoIf($[${COUNTER}>=10]?wrong)
>  same => n(playmenu),Background(${MENU})
>  same => n(bypass),WaitExten(10)
> ; go straight to VM if they time out...
>  same => n,Goto(2,1)
>  same => n(wrong),Playback(something-terribly-wrong)
>  same => n,Playback(goodbye)
>  same => ,n,Hangup()
> ; within [day-menu] option 2 is Voicemail, option 1 is Directory.
> =
> Calls come in to the dialplan from the PSTN in the [from-pstn] context:
> [from-pstn]
> ; catch analog phone call incoming, send it to main number
> exten => s,1,Verbose(2,---Processing incoming call for ${EXTEN} -
> in context from-pstn)
>  same => n,Answer() ; Wait for CallerID Spill
>  same => n,Wait(1.5) ; Wait for CallerID Spill
>  same => n,Set(CALLER_ID_INFO_ALL=${CALLERID(all)})
>  same => n,NoOp()
>  same => n,Set(__PICKUPMARK=)
>  same => n,Goto(day-menu,s,1)
> 
> Calls are picked up via this context, included in [users]:
> [BLF_group_pickup]
> 
> exten => _**31XX,1,Verbose(2,BLF Pickup Extension ${EXTEN})
>  same => n,Pickup(${EXTEN:2}@users&${EXTEN:2}@default&${EXTEN:2}@PICKUPMARK)
>  same => n,Hangup()
>  (I have also tried adding "@day-menu" to this, but it didn't work either).
> 
> Oh yes ­ Asterisk v1.8.4.1, DAHDI 2.4.1.2 libpri 1.4.11.5
> 
> Thanks
> Cassius
> 
> 
> 


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Re: [asterisk-users] error: Autodestruct on dialog

[Kevin P. Fleming]

On 08/05/2011 09:43 AM, Christian Pinedo Zamalloa wrote:

Hi all,

I need to wait several seconds in "h" extension. Since Wait
application doesn't work in "h" extension I must use System in the
following way:

exten =>  h,1,
 same =>  n,...
 same =>  n,System(/bin/sleep 25)
 same =>  n,...

But when I use this System command in "h" extension I get the following warning:

[Aug  5 14:19:50] WARNING[23637] chan_sip.c: Autodestruct on dialog
'7249D00-BE9611E0-A8B6C958-F31290CD@10.180.4.1' with owner in place
(Method: BYE)


You are stopping the Asterisk SIP channel driver from doing its job; it 
expects the channel to be dead much sooner than 25 seconds after 
receiving (or sending) a BYE. Why do you need to keep the channel alive 
for so long after it has been hungup?


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Answering machine answers after pickup a phone.

[Jorge Barreiro]

O Venres, 5 de Agosto de 2011 21:20:37 Don Kelly escribiu:
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jorge
> Barreiro Sent: Friday, August 05, 2011 12:35 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Answering machine answers after pickup a
> phone.
> 
> O Venres, 5 de Agosto de 2011 17:42:28 Shaun Ruffell escribiu:
> > On Fri, Aug 05, 2011 at 01:14:58PM +0200, Jorge Barreiro wrote:
> > > Hi,
> > > 
> > > thanks for your time!
> > > 
> > > O Venres, 5 de Agosto de 2011 12:35:05 escribiches:
> > > > Completely normal operation.
> > > > You need to read and understand more basic telephony and analog
> > > > lines to understand why that won't work.
> > > 
> > > I definitely have a lot to learn yet.
> > > 
> > > > Asterisk needs to be in control, and once someone answers a phone
> > > > not under Asterisk control, or the call is abandoned there is
> > > > little you can do.
> > > 
> > > What I pretend is that asterisk detects that it's not under control
> > > and gets out of the way. The same way it detects a remote hangup and
> > > stops the dialplan, it could detect that someone else answered (the
> > > line is not ringing anymore) and discard it the same way it does
> > > when the remote part hangup.
> > > 
> > > I've read comments in forums and tutorials that seem to imply that
> > > this happens, but I couldn't find any confirmation (and indeed, it's
> > > not happening to me).
> > 
> > When I first installed Asterisk in my home I used it in the way that
> > you
> > described: as a glorified answering machine to email to me any voice
> > mail.
> > 
> > I think what you want is the WaitForRing()[1] dial plan application.
> > This function will wait x number of seconds, then look for *another*
> > ring to come in. If someone answered the phone before the timeout to
> > that function Asterisk would stop processing the dial plan.
> > 
> > [1] https://wiki.asterisk.org/wiki/display/AST/Application_WaitForRing
> > 
> > I ran into a couple of issues with WaitForRing(). The first being if
> > someone answered the phone and then quickly hung up *and* a new phone
> > call came in within the timeout period, Asterisk wouldn't know that
> > the line was ringing due to a new call. The second problem was I never
> > got the dial tone detection working so that if I tried to *place* a
> > call from Asterisk while someone was on the house line I would aggravate
> 
> my wife.
> 
> > Since coming to work for Digium I've seen in the data sheets for the
> > FXO interfaces that there is a capability to detect when a parallel
> > device on a line goes off hook. This would allow Asterisk to have a
> > better sense of the state of the line (like it currently can detect
> > when a port is unplugged and there is not battery by generating a red
> > alarm.) but I haven't looked into getting that information off the
> 
> hardware and up into Asterisk.
> 
> > Hope this helps,
> > Shaun
> 
> That application looks like a good solution. I can't test it until Monday,
> but I'll try it and let you know. The drawbacks you mention doesn't seem
> too inconvenient in my case.
> 
> Anyway, I started with this cause I thought it was an easy first step, if
> it gets so complicated I think I'll go forward and put all phones under
> the control of the PBX.
> 
> Thank you everybody for your help.
> 
> 
> I don't think this is a solution to the problem you described. No matter
> how long Asterisk 'waits for ring,' if the call has already been answered
> when Asterisk picks up, things won't work out well.

The idea is that asterisk doesn't pick up if doesn't find the ring. 

> The solution I
> described earlier, adding a simple exclusion device, will preclude
> Asterisk 'stepping on' a call in progress. This is the approach that Shaun
> suggests: "...a capability to detect when a parallel device on a line goes
> off hook."  As it has not been implemented in Asterisk, it can be handled
> by an inexpensive device. This will enable you to do as you planned--test
> your implementation step-by-step, starting with the "answering machine."

Yes, that exclusion device would be more of a solution instead of just a 
workaround. But I'm finding it hard to find where to buy one in Spain (I've 
just 
started to look for them, anyway).


Thanks.

> --Don
> 
> Don Kelly
> 
> PCF Corp
> People Come First
> 651 842-1000
> 651 842-1001 fax
> 
> 
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[asterisk-users] Receptionist Extension cannot be Pickup()'ed

[Cassius Smith]

Hello all,
I am struggling with an annoying problem. I have an installation with a
small number of Grandstream  GXP2010 endpoints. Each endpoint has all the
extensions programmed into the phone for BLF - for instant pickup, transfer
or speed dial.

Except for the Receptionist phone, which is handled internally via the "0"
extension. That extension drops into a [day-menu] context with an IVR after
the receptionist phone rings for 20 seconds.

The receptionist phone has a BLF field on all the other phones. But when
that phone rings, I think something is messing with some channel variable
that is preventing Pickup() from working.

ALL Other extensions can be picked up. ONLY the extension(s) I ring from the
day-menu cannot.

Here is a snip from my dialplan:
exten => s,1,NoOp()
 same => n,Verbose(2,"Processing incoming call from ${CALLERID(all)})
 same => n(daycheck),GotoIfTime(08:30-16:59,mon-fri,*,*?open)
 same => n,Set(MENU=night-menu)
 same => n,Goto(night)
 same => n(open),Set(MENU=day-menu)
 same => n,Set(__PICKUPMARK=)
 same => n,Dial(SIP/3100,20) ; 3100 is receptionists phone
; go to IVR if no answer
 same => n,Goto(playmenu)
 same => n(night),NoOp()
 same => n(top),Wait(0.5)
 same => n,GotoIf($[${COUNTER}>=10]?wrong)
 same => n(playmenu),Background(${MENU})
 same => n(bypass),WaitExten(10)
; go straight to VM if they time out...
 same => n,Goto(2,1)
 same => n(wrong),Playback(something-terribly-wrong)
 same => n,Playback(goodbye)
 same => ,n,Hangup()
; within [day-menu] option 2 is Voicemail, option 1 is Directory.
=
Calls come in to the dialplan from the PSTN in the [from-pstn] context:
[from-pstn]
; catch analog phone call incoming, send it to main number
exten => s,1,Verbose(2,---Processing incoming call for ${EXTEN}
- in context from-pstn)
 same => n,Answer() ; Wait for CallerID Spill
 same => n,Wait(1.5) ; Wait for CallerID Spill
 same => n,Set(CALLER_ID_INFO_ALL=${CALLERID(all)})
 same => n,NoOp()
 same => n,Set(__PICKUPMARK=)
 same => n,Goto(day-menu,s,1)

Calls are picked up via this context, included in [users]:
[BLF_group_pickup]

exten => _**31XX,1,Verbose(2,BLF Pickup Extension ${EXTEN})
 same => n,Pickup(${EXTEN:2}@users&${EXTEN:2}@default&${EXTEN:2}@PICKUPMARK)
 same => n,Hangup()
 (I have also tried adding "@day-menu" to this, but it didn't work either).

Oh yes ­ Asterisk v1.8.4.1, DAHDI 2.4.1.2 libpri 1.4.11.5

Thanks
Cassius





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[asterisk-users] Asterisk 1.6.2.20 Now Available

[Asterisk Development Team]

The Asterisk Development Team announces the release of Asterisk 
1.6.2.20. This

release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.20 resolves a regression that was 
introduced just

prior to the release of Asterisk 1.6.2.19.

* Fix reload crash caused by destroying default parking lot.
  (Closes issue ASTERISK-18103. Reported by 808blogger. Patched by jrose.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.20

Thank you for your continued support of Asterisk!

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Re: [asterisk-users] asterisk-users Digest, Vol 85, Issue 10

[Shyju Kanaprath]

Use service from sms providers like smsglobal.com, they have scripts to do
that.

On Fri, Aug 5, 2011 at 9:00 PM, wrote:

> Send asterisk-users mailing list submissions to
>asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
>http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
>asterisk-users-requ...@lists.digium.com
>
> You can reach the person managing the list at
>asterisk-users-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> Today's Topics:
>
>   1. Re: Assistance sending mass sms to cellphones (Robert Huddleston)
>   2. Re: Assistance sending mass sms to cellphones (Landy Landy)
>
>
> --
>
> Message: 1
> Date: Fri, 5 Aug 2011 12:45:52 -0400
> From: "Robert Huddleston" 
> Subject: Re: [asterisk-users] Assistance sending mass sms to
>cellphones
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>
> Message-ID: <013f01cc538f$2427d240$6c7776c0$@com>
> Content-Type: text/plain;   charset="us-ascii"
>
> Seriously Again?
>
> This is off topic...
>
> Asterisk will not provide you with the ability to SMS random cell phones.
>
> Being able to "transport" the SMS yourself is a grewling process.. Look at
> software like Kamel...
>
> Basically you have three options:
> ( a ) cheat and use the email method - i.e. determine everyone's carrier
> and
> use the email address equivalent ( b ) utilize a 3rd party to transmit the
> sms for you (cost) and they might end up doing ( a ) above without you
> knowing ( c ) spend lots of money and headaches transporting sms yourself.
>
> Either way it's off-topic and not related to Asterisk.
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
> Sent: Friday, August 05, 2011 12:42 PM
> To: asterisk
> Subject: [asterisk-users] Assistance sending mass sms to cellphones
>
> Hello.
>
> I would like to know if is possible to send mass sms with an php agi script
> through asterisk?
>
> For example: I have about 50 cellphone numbers I would like to text
> whenever
> theres a meeting, I should load the numbers from a database and send a
> message via web with php and have asterisk send it.
>
> I've been googling about it but, I get a lot of providers that already do
> this but, I would like to learn how to do it myself since my budget is very
> minimum.
>
> Thanks in advanced for your help and time.
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
>
> Message: 2
> Date: Fri, 5 Aug 2011 09:48:43 -0700 (PDT)
> From: Landy Landy 
> Subject: Re: [asterisk-users] Assistance sending mass sms to
>cellphones
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID:
><1312562923.47115.yahoomailclas...@web126015.mail.ne1.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> Robert.
>
> Thanks for replying.
>
> --- On Fri, 8/5/11, Robert Huddleston  wrote:
>
> > From: Robert Huddleston 
> > Subject: Re: [asterisk-users] Assistance sending mass sms to cellphones
> > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <
> asterisk-users@lists.digium.com>
> > Date: Friday, August 5, 2011, 11:50 AM
> > This is off topic...
> >
> > Asterisk will not provide you with the ability to SMS
> > random cell phones.
>
> We actually have a group of people belonging to a rotary club and we wanted
> to automate the sms process... is not random cell phones.
>
> >
> > Being able to "transport" the SMS yourself is a grewling
> > process.. Look at
> > software like Kamel...
> >
> > Basically you have three options:
> > ( a ) cheat and use the email method - i.e. determine
> > everyone's carrier and
> > use the email address equivalent
> > ( b ) utilize a 3rd party to transmit the sms for you
> > (cost) and they might
>
> Looks like this is the easiest option but, very expensive for what we
> really want to do.
>
> > end up doing ( a ) above without you knowing
> > ( c ) spend lots of money and headaches transporting sms
> > yourself.
> >
> > Either way it's off-topic and not related to Asterisk.
> >
>
> Sorry, didn't think this wasnt an asterisk related question.
>
> >
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com]
> > On Behalf Of L

Re: [asterisk-users] Answering machine answers after pickup a phone.

[Don Kelly]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jorge Barreiro
Sent: Friday, August 05, 2011 12:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Answering machine answers after pickup a
phone.

O Venres, 5 de Agosto de 2011 17:42:28 Shaun Ruffell escribiu:
> On Fri, Aug 05, 2011 at 01:14:58PM +0200, Jorge Barreiro wrote:
> > Hi,
> > 
> > thanks for your time!
> > 
> > O Venres, 5 de Agosto de 2011 12:35:05 escribiches:
> > > Completely normal operation.
> > > You need to read and understand more basic telephony and analog 
> > > lines to understand why that won't work.
> > 
> > I definitely have a lot to learn yet.
> > 
> > > Asterisk needs to be in control, and once someone answers a phone 
> > > not under Asterisk control, or the call is abandoned there is 
> > > little you can do.
> > 
> > What I pretend is that asterisk detects that it's not under control 
> > and gets out of the way. The same way it detects a remote hangup and 
> > stops the dialplan, it could detect that someone else answered (the 
> > line is not ringing anymore) and discard it the same way it does 
> > when the remote part hangup.
> > 
> > I've read comments in forums and tutorials that seem to imply that 
> > this happens, but I couldn't find any confirmation (and indeed, it's 
> > not happening to me).
> 
> When I first installed Asterisk in my home I used it in the way that 
> you
> described: as a glorified answering machine to email to me any voice mail.
> 
> I think what you want is the WaitForRing()[1] dial plan application.  
> This function will wait x number of seconds, then look for *another* 
> ring to come in. If someone answered the phone before the timeout to 
> that function Asterisk would stop processing the dial plan.
> 
> [1] https://wiki.asterisk.org/wiki/display/AST/Application_WaitForRing
> 
> I ran into a couple of issues with WaitForRing(). The first being if 
> someone answered the phone and then quickly hung up *and* a new phone 
> call came in within the timeout period, Asterisk wouldn't know that 
> the line was ringing due to a new call. The second problem was I never 
> got the dial tone detection working so that if I tried to *place* a 
> call from Asterisk while someone was on the house line I would aggravate
my wife.
> 
> Since coming to work for Digium I've seen in the data sheets for the 
> FXO interfaces that there is a capability to detect when a parallel 
> device on a line goes off hook. This would allow Asterisk to have a 
> better sense of the state of the line (like it currently can detect 
> when a port is unplugged and there is not battery by generating a red 
> alarm.) but I haven't looked into getting that information off the
hardware and up into Asterisk.
> 
> Hope this helps,
> Shaun


That application looks like a good solution. I can't test it until Monday,
but I'll try it and let you know. The drawbacks you mention doesn't seem too
inconvenient in my case.

Anyway, I started with this cause I thought it was an easy first step, if it
gets so complicated I think I'll go forward and put all phones under the
control of the PBX.

Thank you everybody for your help.


I don't think this is a solution to the problem you described. No matter how
long Asterisk 'waits for ring,' if the call has already been answered when
Asterisk picks up, things won't work out well. The solution I described
earlier, adding a simple exclusion device, will preclude Asterisk 'stepping
on' a call in progress. This is the approach that Shaun suggests: "...a
capability to detect when a parallel device on a line goes off hook."  As it
has not been implemented in Asterisk, it can be handled by an inexpensive
device. This will enable you to do as you planned--test your implementation
step-by-step, starting with the "answering machine."

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
651 842-1001 fax


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Re: [asterisk-users] Answering machine answers after pickup a phone.

[John Novack]

Jorge Barreiro wrote:

O Venres, 5 de Agosto de 2011 17:42:28 Shaun Ruffell escribiu:

On Fri, Aug 05, 2011 at 01:14:58PM +0200, Jorge Barreiro wrote:

Hi,

thanks for your time!

O Venres, 5 de Agosto de 2011 12:35:05 escribiches:

Completely normal operation.
You need to read and understand more basic telephony and analog lines
to understand why that won't work.

I definitely have a lot to learn yet.


Asterisk needs to be in control, and once someone answers a phone not
under Asterisk control, or the call is abandoned there is little you
can do.

What I pretend is that asterisk detects that it's not under control and
gets out of the way. The same way it detects a remote hangup and stops
the dialplan, it could detect that someone else answered (the line is
not ringing anymore) and discard it the same way it does when the remote
part hangup.

I've read comments in forums and tutorials that seem to imply that this
happens, but I couldn't find any confirmation (and indeed, it's not
happening to me).

When I first installed Asterisk in my home I used it in the way that you
described: as a glorified answering machine to email to me any voice mail.

I think what you want is the WaitForRing()[1] dial plan application.  This
function will wait x number of seconds, then look for *another* ring to
come in. If someone answered the phone before the timeout to that function
Asterisk would stop processing the dial plan.

[1] https://wiki.asterisk.org/wiki/display/AST/Application_WaitForRing

I ran into a couple of issues with WaitForRing(). The first being if
someone answered the phone and then quickly hung up *and* a new phone call
came in within the timeout period, Asterisk wouldn't know that the line
was ringing due to a new call. The second problem was I never got the dial
tone detection working so that if I tried to *place* a call from Asterisk
while someone was on the house line I would aggravate my wife.

Since coming to work for Digium I've seen in the data sheets for the FXO
interfaces that there is a capability to detect when a parallel device on a
line goes off hook. This would allow Asterisk to have a better sense of the
state of the line (like it currently can detect when a port is unplugged
and there is not battery by generating a red alarm.) but I haven't looked
into getting that information off the hardware and up into Asterisk.

Hope this helps,
Shaun


That application looks like a good solution. I can't test it until Monday, but
I'll try it and let you know. The drawbacks you mention doesn't seem too
inconvenient in my case.

Anyway, I started with this cause I thought it was an easy first step, if it
gets so complicated I think I'll go forward and put all phones under the
control of the PBX.

Thank you everybody for your help.

the situation gets more complex if Caller ID is sent and one wants to act upon 
it, and Asterisk doesn't handle the call

In the US the data is sent between the first and second ring, and if the call 
isn't answered by Asterisk, it thinks that another call has arrived WITHOUT the 
information, and the situation falls apart rapidly or one needs to have some 
logic to figure out if this is the third ring on call #1 or a new call

Though the chipset in the X100P supports looking at the output port, I don't 
believe the driver fully supports it

using a TDM 4xx or even better a single port T1 card with a channel bank gives 
a lot more  ports for the cost

John Novack

--

Dog is my Co-pilot


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Re: [asterisk-users] Ring delay problem

[Alejandro Cabrera Obed]

Warrem thanksa lotI'll test next monday and I'll tell you.

Regards

2011/8/5 Warren Selby 

> On Fri, Aug 5, 2011 at 7:53 AM, Alejandro Cabrera Obed 
> wrote:
>
>> Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and
>> Celeron), and last days when I call from one extension to another of the
>> same PBX after I dial the number the rings sound after 20 seconds.
>>
>> In the CLI log, when I debug the AGI, I see always goes good until
>> dialparties.agi, and after that there are 20 seconds without any log, and so
>> the ring sound.
>>
>>
> I've had this issue before.  Try moving the /etc/php.d/imap.so file out of
> the /etc/php.d directory and see if that helps.  It's been a while but I may
> have had to restart the machine when I did the file move.  It may have also
> just been a DNS timeout issue, I don't recall the specifics.  I believe I
> used this thread as a reference:
> http://www.fonality.com/trixbox/forums/trixbox-forums/help/suddenly-everything-slow
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com 
>
>
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aco1...@gmail.com
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[asterisk-users] No more CDR record for simple Hangup?

[J Gao]

I am using the new 1.8.5 and I just found out that Asterisk won't record 
the call if the call just hangup. I did a test like this:


exten => 1009, 1, Hangup()

Then I called 1009:

 == Using UDPTL CoS mark 5
  == Using SIP RTP CoS mark 5
-- Executing [1009@init-1005:1] Hangup("SIP/1007-003c", "") in 
new stack
  == Spawn extension (init-1005, 1009, 1) exited non-zero on 
'SIP/1005-003c'


I am not sure why now Asterisk doesn't write this into CDR.  In the 
previous version Asterisk record the hangup call.


Is there anyway I can have the hangup write into the CDR?



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Re: [asterisk-users] Ring delay problem

[Warren Selby]

On Fri, Aug 5, 2011 at 7:53 AM, Alejandro Cabrera Obed wrote:

> Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and
> Celeron), and last days when I call from one extension to another of the
> same PBX after I dial the number the rings sound after 20 seconds.
>
> In the CLI log, when I debug the AGI, I see always goes good until
> dialparties.agi, and after that there are 20 seconds without any log, and so
> the ring sound.
>
>
I've had this issue before.  Try moving the /etc/php.d/imap.so file out of
the /etc/php.d directory and see if that helps.  It's been a while but I may
have had to restart the machine when I did the file move.  It may have also
just been a DNS timeout issue, I don't recall the specifics.  I believe I
used this thread as a reference:
http://www.fonality.com/trixbox/forums/trixbox-forums/help/suddenly-everything-slow

-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com 
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Re: [asterisk-users] Send Refer with replaces from asterisk

[Kevin P. Fleming]

On 08/05/2011 05:55 AM, Nikhil wrote:


How to send REFER with replaces from asterisk (Sending out) for doing
attended transfer.


The Transfer() application can be used from the dialplan to initiate a 
transfer of the channel it is executed on. There's no way to do an 
'automated attended transfer' though, because that doesn't really make 
any sense. Whether chan_sip will use REFER with a Replaces header or not 
to effect the transfer I can't say for sure, but it will cause a blind 
transfer of the channel to the destination specified.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Answering machine answers after pickup a phone.

[Jorge Barreiro]

O Venres, 5 de Agosto de 2011 17:42:28 Shaun Ruffell escribiu:
> On Fri, Aug 05, 2011 at 01:14:58PM +0200, Jorge Barreiro wrote:
> > Hi,
> > 
> > thanks for your time!
> > 
> > O Venres, 5 de Agosto de 2011 12:35:05 escribiches:
> > > Completely normal operation.
> > > You need to read and understand more basic telephony and analog lines
> > > to understand why that won't work.
> > 
> > I definitely have a lot to learn yet.
> > 
> > > Asterisk needs to be in control, and once someone answers a phone not
> > > under Asterisk control, or the call is abandoned there is little you
> > > can do.
> > 
> > What I pretend is that asterisk detects that it's not under control and
> > gets out of the way. The same way it detects a remote hangup and stops
> > the dialplan, it could detect that someone else answered (the line is
> > not ringing anymore) and discard it the same way it does when the remote
> > part hangup.
> > 
> > I've read comments in forums and tutorials that seem to imply that this
> > happens, but I couldn't find any confirmation (and indeed, it's not
> > happening to me).
> 
> When I first installed Asterisk in my home I used it in the way that you
> described: as a glorified answering machine to email to me any voice mail.
> 
> I think what you want is the WaitForRing()[1] dial plan application.  This
> function will wait x number of seconds, then look for *another* ring to
> come in. If someone answered the phone before the timeout to that function
> Asterisk would stop processing the dial plan.
> 
> [1] https://wiki.asterisk.org/wiki/display/AST/Application_WaitForRing
> 
> I ran into a couple of issues with WaitForRing(). The first being if
> someone answered the phone and then quickly hung up *and* a new phone call
> came in within the timeout period, Asterisk wouldn't know that the line
> was ringing due to a new call. The second problem was I never got the dial
> tone detection working so that if I tried to *place* a call from Asterisk
> while someone was on the house line I would aggravate my wife.
> 
> Since coming to work for Digium I've seen in the data sheets for the FXO
> interfaces that there is a capability to detect when a parallel device on a
> line goes off hook. This would allow Asterisk to have a better sense of the
> state of the line (like it currently can detect when a port is unplugged
> and there is not battery by generating a red alarm.) but I haven't looked
> into getting that information off the hardware and up into Asterisk.
> 
> Hope this helps,
> Shaun


That application looks like a good solution. I can't test it until Monday, but 
I'll try it and let you know. The drawbacks you mention doesn't seem too 
inconvenient in my case.

Anyway, I started with this cause I thought it was an easy first step, if it 
gets so complicated I think I'll go forward and put all phones under the 
control of the PBX.

Thank you everybody for your help.



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Re: [asterisk-users] Assistance sending mass sms to cellphones

[Robert Huddleston]

When you say expensive... You are talking about pennies per SMS... Again -
if you want to cheat and go the email route - that would be free... It's
unreliable and requires some thought...

If you want more information / consulting contact me off-list.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
Sent: Friday, August 05, 2011 12:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Assistance sending mass sms to cellphones

Robert.

Thanks for replying.

--- On Fri, 8/5/11, Robert Huddleston  wrote:

> From: Robert Huddleston 
> Subject: Re: [asterisk-users] Assistance sending mass sms to cellphones
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

> Date: Friday, August 5, 2011, 11:50 AM
> This is off topic...
> 
> Asterisk will not provide you with the ability to SMS
> random cell phones.

We actually have a group of people belonging to a rotary club and we wanted
to automate the sms process... is not random cell phones.

> 
> Being able to "transport" the SMS yourself is a grewling
> process.. Look at
> software like Kamel...
> 
> Basically you have three options:
> ( a ) cheat and use the email method - i.e. determine
> everyone's carrier and
> use the email address equivalent
> ( b ) utilize a 3rd party to transmit the sms for you
> (cost) and they might

Looks like this is the easiest option but, very expensive for what we really
want to do.

> end up doing ( a ) above without you knowing
> ( c ) spend lots of money and headaches transporting sms
> yourself.
> 
> Either way it's off-topic and not related to Asterisk.
> 

Sorry, didn't think this wasnt an asterisk related question.

> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com]
> On Behalf Of Landy Landy
> Sent: Friday, August 05, 2011 11:42 AM
> To: asterisk
> Subject: [asterisk-users] Assistance sending mass sms to
> cellphones
> 
> Hello.
> 
> I would like to know if is possible to send mass sms with
> an php agi script
> through asterisk?
> 
> For example: I have about 50 cellphone numbers I would like
> to text whenever
> theres a meeting, I should load the numbers from a database
> and send a
> message via web with php and have asterisk send it.
> 
> I've been googling about it but, I get a lot of providers
> that already do
> this but, I would like to learn how to do it myself since
> my budget is very
> minimum.
> 
> Thanks in advanced for your help and time.
> 
> 
> --
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Re: [asterisk-users] Assistance sending mass sms to cellphones

[Landy Landy]

Robert.

Thanks for replying.

--- On Fri, 8/5/11, Robert Huddleston  wrote:

> From: Robert Huddleston 
> Subject: Re: [asterisk-users] Assistance sending mass sms to cellphones
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 
> 
> Date: Friday, August 5, 2011, 11:50 AM
> This is off topic...
> 
> Asterisk will not provide you with the ability to SMS
> random cell phones.

We actually have a group of people belonging to a rotary club and we wanted to 
automate the sms process... is not random cell phones.

> 
> Being able to "transport" the SMS yourself is a grewling
> process.. Look at
> software like Kamel...
> 
> Basically you have three options:
> ( a ) cheat and use the email method - i.e. determine
> everyone's carrier and
> use the email address equivalent
> ( b ) utilize a 3rd party to transmit the sms for you
> (cost) and they might

Looks like this is the easiest option but, very expensive for what we really 
want to do.

> end up doing ( a ) above without you knowing
> ( c ) spend lots of money and headaches transporting sms
> yourself.
> 
> Either way it's off-topic and not related to Asterisk.
> 

Sorry, didn't think this wasnt an asterisk related question.

> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com]
> On Behalf Of Landy Landy
> Sent: Friday, August 05, 2011 11:42 AM
> To: asterisk
> Subject: [asterisk-users] Assistance sending mass sms to
> cellphones
> 
> Hello.
> 
> I would like to know if is possible to send mass sms with
> an php agi script
> through asterisk?
> 
> For example: I have about 50 cellphone numbers I would like
> to text whenever
> theres a meeting, I should load the numbers from a database
> and send a
> message via web with php and have asterisk send it.
> 
> I've been googling about it but, I get a lot of providers
> that already do
> this but, I would like to learn how to do it myself since
> my budget is very
> minimum.
> 
> Thanks in advanced for your help and time.
> 
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> every Thurs:
>            
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>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
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Re: [asterisk-users] Assistance sending mass sms to cellphones

[Robert Huddleston]

Seriously Again?

This is off topic...

Asterisk will not provide you with the ability to SMS random cell phones.

Being able to "transport" the SMS yourself is a grewling process.. Look at
software like Kamel...

Basically you have three options:
( a ) cheat and use the email method - i.e. determine everyone's carrier and
use the email address equivalent ( b ) utilize a 3rd party to transmit the
sms for you (cost) and they might end up doing ( a ) above without you
knowing ( c ) spend lots of money and headaches transporting sms yourself.

Either way it's off-topic and not related to Asterisk.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
Sent: Friday, August 05, 2011 12:42 PM
To: asterisk
Subject: [asterisk-users] Assistance sending mass sms to cellphones

Hello.

I would like to know if is possible to send mass sms with an php agi script
through asterisk?

For example: I have about 50 cellphone numbers I would like to text whenever
theres a meeting, I should load the numbers from a database and send a
message via web with php and have asterisk send it.

I've been googling about it but, I get a lot of providers that already do
this but, I would like to learn how to do it myself since my budget is very
minimum.

Thanks in advanced for your help and time.


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Re: [asterisk-users] Assistance sending mass sms to cellphones

[Eric Wieling]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Landy Landy
> Sent: Friday, August 05, 2011 12:42 PM
> To: asterisk
> Subject: [asterisk-users] Assistance sending mass sms to cellphones
> 
> Hello.
> 
> I would like to know if is possible to send mass sms with an php agi script
> through asterisk?
> 
> For example: I have about 50 cellphone numbers I would like to text
> whenever theres a meeting, I should load the numbers from a database and
> send a message via web with php and have asterisk send it.
> 
> I've been googling about it but, I get a lot of providers that already do this
> but, I would like to learn how to do it myself since my budget is very
> minimum.

Asterisk's SMS application only supports LANDLINE SMS.  Please read the 
voip-info Asterisk SMS page.

>From http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms:

The SMS module for asterisk was developed by Adrian Kennard, and is an 
implementation of the ETSI specification for landline SMS, ETSI ES 201 912, 
which is available from www.etsi.org. Landline SMS is starting to be available 
in various parts of Europe, and is available from BT in the UK. However, 
asterisk would allow gateways to be created in other locations such as the US, 
and use of SMS capable phones such as the Magic Messenger. SMS works using 
analogue or ISDN lines.

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[asterisk-users] Assistance sending mass sms to cellphones

[Landy Landy]

Hello.

I would like to know if is possible to send mass sms with an php agi script 
through asterisk?

For example: I have about 50 cellphone numbers I would like to text whenever 
theres a meeting, I should load the numbers from a database and send a message 
via web with php and have asterisk send it.

I've been googling about it but, I get a lot of providers that already do this 
but, I would like to learn how to do it myself since my budget is very minimum.

Thanks in advanced for your help and time.


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Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?

[Kevin P. Fleming]

On 08/03/2011 09:33 PM, Bruce B wrote:

Can you please elaborate on how to apply the patch?
Also, is the repository updated with the new code?


No, of course not. RPMs and DEBs are not patched, they are produced when 
new releases are made. If you haven't seen a new release made that 
includes the patch in question, then RPMs and DEBs don't have it either.


--
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Re: [asterisk-users] Assistance sending mass sms to cellphones

[Robert Huddleston]

This is off topic...

Asterisk will not provide you with the ability to SMS random cell phones.

Being able to "transport" the SMS yourself is a grewling process.. Look at
software like Kamel...

Basically you have three options:
( a ) cheat and use the email method - i.e. determine everyone's carrier and
use the email address equivalent
( b ) utilize a 3rd party to transmit the sms for you (cost) and they might
end up doing ( a ) above without you knowing
( c ) spend lots of money and headaches transporting sms yourself.

Either way it's off-topic and not related to Asterisk.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
Sent: Friday, August 05, 2011 11:42 AM
To: asterisk
Subject: [asterisk-users] Assistance sending mass sms to cellphones

Hello.

I would like to know if is possible to send mass sms with an php agi script
through asterisk?

For example: I have about 50 cellphone numbers I would like to text whenever
theres a meeting, I should load the numbers from a database and send a
message via web with php and have asterisk send it.

I've been googling about it but, I get a lot of providers that already do
this but, I would like to learn how to do it myself since my budget is very
minimum.

Thanks in advanced for your help and time.


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Re: [asterisk-users] Increasing volume ?

[isrlgb]

Well even in my example there is a mistake in the second line change the 1 to a 
2

exten =>_.,1,Set(VOLUME(TX)=10)
exten =>_.,2,Set(VOLUME(RX)=10)

-Original Message-
From: Zeeshan Ali Shah 
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 5 Aug 2011 14:52:52 
To: Asterisk Users Mailing List - Non-Commercial 
Discussion
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] Increasing volume ?

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Re: [asterisk-users] Answering machine answers after pickup a phone.

[Shaun Ruffell]

On Fri, Aug 05, 2011 at 01:14:58PM +0200, Jorge Barreiro wrote:
> Hi,
> 
> thanks for your time!
> 
> O Venres, 5 de Agosto de 2011 12:35:05 escribiches:
> > Completely normal operation.
> > You need to read and understand more basic telephony and analog lines to
> > understand why that won't work.
> 
> I definitely have a lot to learn yet. 
> 
> > Asterisk needs to be in control, and once someone answers a phone not under
> > Asterisk control, or the call is abandoned there is little you can do.
> 
> What I pretend is that asterisk detects that it's not under control and gets 
> out of the way. The same way it detects a remote hangup and stops the 
> dialplan, it could detect that someone else answered (the line is not ringing 
> anymore) and discard it the same way it does when the remote part hangup.
> 
> I've read comments in forums and tutorials that seem to imply that this 
> happens, but I couldn't find any confirmation (and indeed, it's not happening 
> to 
> me).

When I first installed Asterisk in my home I used it in the way that you
described: as a glorified answering machine to email to me any voice mail.

I think what you want is the WaitForRing()[1] dial plan application.  This
function will wait x number of seconds, then look for *another* ring to come
in. If someone answered the phone before the timeout to that function Asterisk
would stop processing the dial plan.

[1] https://wiki.asterisk.org/wiki/display/AST/Application_WaitForRing

I ran into a couple of issues with WaitForRing(). The first being if someone
answered the phone and then quickly hung up *and* a new phone call came in
within the timeout period, Asterisk wouldn't know that the line was ringing
due to a new call. The second problem was I never got the dial tone detection
working so that if I tried to *place* a call from Asterisk while someone was
on the house line I would aggravate my wife.

Since coming to work for Digium I've seen in the data sheets for the FXO
interfaces that there is a capability to detect when a parallel device on a
line goes off hook. This would allow Asterisk to have a better sense of the
state of the line (like it currently can detect when a port is unplugged and
there is not battery by generating a red alarm.) but I haven't looked into
getting that information off the hardware and up into Asterisk.

Hope this helps,
Shaun

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[asterisk-users] Assistance sending mass sms to cellphones

[Landy Landy]

Hello.

I would like to know if is possible to send mass sms with an php agi script 
through asterisk?

For example: I have about 50 cellphone numbers I would like to text whenever 
theres a meeting, I should load the numbers from a database and send a message 
via web with php and have asterisk send it.

I've been googling about it but, I get a lot of providers that already do this 
but, I would like to learn how to do it myself since my budget is very minimum.

Thanks in advanced for your help and time.


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[asterisk-users] error: Autodestruct on dialog

[Christian Pinedo Zamalloa]

Hi all,

I need to wait several seconds in "h" extension. Since Wait
application doesn't work in "h" extension I must use System in the
following way:

exten => h,1,
same => n,...
same => n,System(/bin/sleep 25)
same => n,...

But when I use this System command in "h" extension I get the following warning:

[Aug  5 14:19:50] WARNING[23637] chan_sip.c: Autodestruct on dialog
'7249D00-BE9611E0-A8B6C958-F31290CD@10.180.4.1' with owner in place
(Method: BYE)


If i run in the CLI "sip show channels" there are a lot of SIP dialogs
that haven't finished yet and that are hold by Asterisk:

asterisk*CLI> sip show channels
10.180.4.1   652  5648af9721df9cc  0x0 (nothing)No
  Rx: BYEcme01
10.180.4.1   650  E546BE0-BEA411E  0x0 (nothing)No
  Rx: BYEcme01
10.180.4.1   699095244BAC5BF87-BC2811  0x0 (nothing)No
  Rx: BYEcme01
636 active SIP dialogs


But they aren't active channels:

asterisk*CLI> core show channels
Channel  Location State   Application(Data)
0 active channels
0 active calls
1013 calls processed

Could this be a bug or am I doing something bad??? Thanks,

-- 
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PGP keyID: 0x828D0C80
Fingerprint: 7BFF 4105 F46B 7977 BD96  348C 1007 4FF8 828D 0C80

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[asterisk-users] Fax Detection with chan_ss7

[AC]

Hi all,

http://www.voip-info.org/wiki/view/Asterisk+fax shows that it is possible to
get asterisk to automatically detect a fax. However, I am using chan_ss7
instead of chan_dahdi or chan_zap. Is it possible to detect a fax with
chan_ss7?

Thanks,
Amish
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Re: [asterisk-users] ASterisk is Going stop whenever restart the server

[Richard Mudgett]

> I am using goautodial, I am using 20channels telcom PRI line and in my
> server DIgium TE120 PRI card which is for 31 channel. with this
> configuration
> I am able to call from server . but problem whenever i restarted the
> server that time is Asterisk is stop then I am not able to call
> outside.
> how to resolve this issue. every time whenever restart the server this
> asterisk it going stop.

I don't know what you mean by the stop references.  If you mean the call
fails, then that could be because the call attempted to use a channel
not provisioned for your PRI line.  Your configuration provisions all
30 channels for an E1 line when you said you have 20 channels.

> 
> inthis server my hardware configuration is
> /etc/asterisk/dahdi-channels.conf
> group=0,11
> context=default
> switchtype = euroisdn
> signalling = pri_cpe
> channel => 1-15,17-31
^^^ This line is creating all of the channels for an E1 so chan_dahdi
thinks it has all 30 channels available.  You said you only have 20
channels available so the channels line is wrong here.  It should be
something like
channel => 1-15,17-21

Also anything after a channel line does not apply to the channels created
by a channel line.

> context = default
> group = 63
> 
> /etc/asterisk/chan_dahdi.conf
> [trunkgroups]
> 
> [channels]
> #include dahdi-channels.conf
> language=en
> context=default
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> restictcid=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> echotraining=800
> relaxdtmf=yes
> rxgain=0.0
> txgain=0.0
> ;group=1
> ;callgroup=1
> ;pickupgroup=1
> busydetect=yes
> busycount=6
> immediate=no
> resetinterval=never
> switchtype=euroisdn
> signalling=pri_cpe
> pridialplan=unknown
> prilocaldialplan=unknown
> group=0
> channel => 1-20
^^^ This channel line is wrong.  It attempts to redefine channels
you have already defined earlier by the dahdi-channels.conf include.
You should have gotten warning messages when Asterisk loaded about
channels already defined.

> 
> 
> /etc/dahdi/system.conf
> span=1,1,0,ccs,hdb3,crc4
> # termtype: te
> bchan=1-15,17-31
> dchan=16
> echocanceller=mg2,1-15,17-31

Richard

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Re: [asterisk-users] Know the number of concurrent dial ?

[Danny Nicholas]

The portion of the script I posted only gathers the data
Add #!/usr/local/bin/perl to the top
And 
my $daymax=0;
my %hourmax;
for (my $i=0;$i<24;$i++) {
   $hourmax{$i}=0;
   }
for (my $i=0;$i<$call_max;$i++) {
   if ($call_con{$i}>$daymax) {
  $daymax=$call_con{$i};
  }
   if ($call_con{$i}>$hourmax{$call_conhour{$i}}) {
  $hourmax{$call_conhour{$i}}=$call_con{$i};
  }
   }
Print STDOUT "maximum calls for day was $daymax\n";
For (my $i=0;$i<24;$i++) {
  Print STDOUT "maximum calls for hour $i was ",$hourmax{$i},"\n";
 }
Exit;

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
CALVANO
Sent: Friday, August 05, 2011 12:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Know the number of concurrent dial ?

Hi

Thanks for your answer and help but your script don't work on my server

best regards
olivier


2011/8/4 Danny Nicholas :
> I did a PERL routine to get this information out of Master.csv
> (/var/log/asterisk/cdr-csv)
> open (my $cdr_in, "<", "/var/log/asterisk/cdr-csv/Master.csv");
> my %call_start;
> my %call_end;
> my %call_con;
> my $call_max=0;
> while (<$cdr_in>) {
>   my (@cdr_data) = split /\,/, $_;
>   my $day=unpack("x1 a10", $cdr_data[9]);
>   my $dx=sprintf("%20s", $cdr_data[9]);
>   my $hour=unpack("x12 a2", $dx);
>   my $xx= $cdr_data[11];
>   my $dur= $cdr_data[12];
>  my $syr=unpack("x1 a4", $dx);
>   my $smon=unpack("x6 a2", $dx);
>   my $sday=unpack("x9 a2", $dx);
>   my $shr=unpack("x12 a2", $dx);
>   $shr = $shr * 1;
>   my $smin=unpack("x15 a2", $dx);
>   my $ssec=unpack("x18 a2", $dx);
>   if ($smon == 1 && $sday > 27) {
>      $sday=27;
>      }
>   if ($sday == 31 && ($smon == 3 || $smon == 10 || $smon == 5 || $smon 
> ==
> 8)) {
>      $sday=30;
>      }
>
>   my $stime = timelocal($ssec,$smin,$shr,$sday,$smon,$syr);
>   my $etime = $stime + $dur;
>   $call_start{$call_max}=$stime;
>   $call_end{$call_max}=$etime;
>   $call_conhour{$call_max}=$shr;
>   $call_con{$call_max}=0;
>   $call_max++;
>   }
> close $cdr_in;
> for (my $i=0;$i<$call_max;$i++) {
>   my $test_start=$call_start{$i}-1;
>   my $test_end=$call_end{$i}+1;
>   for (my $j=0;$j<$call_max;$j++) {
>      if ($call_start{$j}>$test_start && $call_end{$j} < $test_end) {;
>         $call_con{$i}++;
>         }
>      }
>   }
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier 
> CALVANO
> Sent: Wednesday, August 03, 2011 4:56 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Know the number of concurrent dial ?
>
> Hi
>
> I connected Asterisk 1.6 has several SIP provider, Do you know a tool 
> to make a graph of the number of simultaneous calls incoming and 
> outgoing ? and know the max outgoing call in same time ?
>
> thanks
> Olivier.
>
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[asterisk-users] Ring delay problem

[Alejandro Cabrera Obed]

Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and
Celeron), and last days when I call from one extension to another of the
same PBX after I dial the number the rings sound after 20 seconds.

In the CLI log, when I debug the AGI, I see always goes good until
dialparties.agi, and after that there are 20 seconds without any log, and so
the ring sound.

I've read dialparties.agi use PHP interpreter, so I notice when I use PHP
command to execute a .php script, it is too slow !!! So I think the problem
es PHP.

After that I reinstall php* packages with "yum reinstall php*", but I have
the same problem: the called extension rings after 20 seconds.

Any idea please ???

Really thanks.
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Re: [asterisk-users] Increasing volume ?

[Zeeshan Ali Shah]

hmm no effect. may be i shd read  asterisk book for knowing the flow and
underlying architecture first.

/Zee


On Thu, Aug 4, 2011 at 4:06 PM, Israel Gottlieb  wrote:

> Set(VOLUME(TX)=10) is correct but you arent putting it in a context so
> asterisk doesnt know how to deal with it
>
>
> do this
>
> [bigbluebutton]
> exten => _.,1,Set(VOLUME(TX)=10)
> exten => _.,1,Set(VOLUME(RX)=10)
> exten => _.,n,Goto(start-dialplan,s,1)
> exten => _.,n,Hangup
>
>
>
> On Thu, Aug 4, 2011 at 4:33 PM, Zeeshan Ali Shah 
> wrote:
>
>> any hint since it seems asterisk treat it as unknown directive
>>
>>
>> On Thu, Aug 4, 2011 at 12:22 PM, Zeeshan Ali Shah <
>> zees...@infoshield.info> wrote:
>>
>>> but got these as well
>>>
>>> [Aug  4 12:21:08] WARNING[3082]: pbx_config.c:1588 pbx_load_config:
>>> ==!!== Unknown directive: Set(VOLUME(TX) at line 9 -- IGNORING!!!
>>> [Aug  4 12:21:08] WARNING[3082]: pbx_config.c:1588 pbx_load_config:
>>> ==!!== Unknown directive: SetGlobalVar(SetVOLUME(TX) at line 10 --
>>> IGNORING!!!
>>> [Aug  4 12:21:08] WARNING[3082]: pbx_config.c:1588 pbx_load_config:
>>> ==!!== Unknown directive: SetGlobalVar(SetVOLUME(RX) at line 11 --
>>> IGNORING!!!
>>>
>>>
>>> On Thu, Aug 4, 2011 at 12:20 PM, Zeeshan Ali Shah <
>>> zees...@infoshield.info> wrote:
>>>
 Yes i tried the followings one by one with differnet values..
 Set(VOLUME(TX)=10)
 ;SetGlobalVar(VOLUME(TX)=10)
 ;SetGlobalVar(SetVOLUME(RX)=10)


  , but no improvement..  dont i have to change something in dialplan ?


 On Thu, Aug 4, 2011 at 11:17 AM, Matt Riddell wrote:

> On 4/08/11 9:16 PM, Zeeshan Ali Shah wrote:
>
>>
>> Tried below, but it still no improvement
>>
>>
>> Zeeshan
>> SetGlobalVar(VOLUME(TX)=10)
>> SetGlobalVar(VOLUME(RX)=10)
>>
>
> Have you tried just doing
>
> Set(VOLUME(TX)=10)
>
> and then 5 etc to make sure you are actually changing the volume?
>
>
> --
> Cheers,
>
> Matt Riddell
> __**_
>
> http://www.venturevoip.com/**news.php(Daily
>  Asterisk News)
> http://www.venturevoip.com/**exchange.php(Full
>  ITSP Solution)
> http://www.venturevoip.com/cc.**php(Call
>  Centre Solutions)
>
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>>>
>>
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[asterisk-users] Audio when a call is on hold.

[AC]

Hi All,

When asterisk bridges a call between 2 peers and peer-A's user puts the call
on hold, then peer-A sends a INVITE with recvonly in the SDP. Asterisk
responds to peer-A with sendonly in the SDP and asterisk sends an INVITE to
peer-B with recvonly in the SDP. Peer-B then responds with a sendonly in the
SDP.

I've noticed in the above scenario that peer-B contiutes to send audio to
peer-A. What is the point in having audio from peer-B to peer-A as the user
at peer-A has put the phone on hold?

Alternatively, in the above scenario is it possible for me to either:
1) generate MOH music to peer-B and send no audio to peer-A, or
2) not allow peer-B to send audio to peer-A until the user returns to the
call and I get a INVITE with sendrecv from peer-A?

Thanks,
AC
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[asterisk-users] Asterisk 1.8.5 eventfilter not working in manager.conf

[Ishfaq Malik]

Hi

Today I upgraded my test server from 1.8.3.2 to 1.8.5 with all the
config files staying the same.

I have the line 
eventfilter=!Event: FullyBooted
in my manager.conf

but since the upgrade this event is not being filtered out.
Has anyone else noticed this or is able to replicate it?

Thanks

Ish
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] Answering machine answers after pickup a phone.

[Doug Lytle]

Don Kelly wrote:

There are analog phones connected to the same PSTN.


And that is why what you want to do won't work.  To have it do what 
you'd like, you'd need to have Asterisk as the only one receiving the call.


Then your other analog phones would need to be on something like a ATAs 
that would allow them to be connected to your phone system as extensions.


Incoming calls could then be set to ring all of the phones and the one 
that answered first would get the call.


The way you have it setup now, as soon as a ring is detected on the FXO, 
Asterisk will try to process it, when you're picking up a phone that's 
on the same wire as the FXO, Asterisk is still going to process that call.


Doug

--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


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Re: [asterisk-users] REGISTER forwarding problem

[Alex Balashov]

On 08/04/2011 01:45 PM, Baybal Ni wrote:


I see 401, but asterisk has my proxy in its trunk list. Can this be
caused by anything else?


Some sort of failure to match the proxy to the sip.conf peer.


Is there any way to do it without using path extension?


We do it by having the proxy rewrite the Contact header somewhat 
steganographically.  For instance, if the REGISTER comes into the proxy 
with a Contact of , we extract those URI 
particles and do:


  remove_hf("Contact");
  append_hf("Contact: \r\n");

  (s, xxx.xxx.xxx.xxx and 5060 are filled in by PVs)

On the inbound leg, the request URI of the initial INVITE is parsed and 
these elements are selected back out.


--
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Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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Re: [asterisk-users] Answering machine answers after pickup a phone.

[Don Kelly]

I'll try to explain myself better. The PBX has only one FXO card, connected
to the PSTN. There is no other phones connected to the PBX nor SIP
extensions. 
There are analog phones connected to the same PSTN.

What I try to do is that, when there is an incoming call from the ouside, if
someone answers on a phone, then the PBX won't answer.



If you want to be certain that the Asterisk system won't interfere with an
active call, you can install an exclusion device between the PSTN and the
FXO card.

Google "telephone exclusion device".

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
651 842-1001 fax


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Re: [asterisk-users] display name

[A J Stiles]

On Friday 05 Aug 2011, salaheddine elharit wrote:
> My question if there is any way to display a name to the customer when i
> call from my Number using asterisk 1.4
>
> Exp: I call from my number 0520 and I want when the customer recive I
> call from 0520 he sees in his phone a name “test”

You can *try*
Set(CALLERID(name))=Test
as a dialplan step, before the Dial() statement obviously.

Whether or not it will have any effect in practice depends entirely on 
equipment downstream of you, and over which you have no control.  Firstly, 
your telco almost certainly will not let you use any ident you're not 
entitled to  (otherwise it would be too easy to generate fake caller-IDs);  
and secondly, the receiving subscriber's apparatus may only be capable of 
displaying numbers, not names.

-- 
AJS

Answers come *after* questions.

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[asterisk-users] ASterisk is Going stop whenever restart the server

[mahesh katta]

Hi,

I am using goautodial, I am using 20channels telcom PRI line and in my
server DIgium TE120 PRI card which is for 31 channel. with this
configuration
I am able to call from server . but problem whenever i restarted the server
that time is Asterisk is stop then I am not able to call outside.
how to resolve this issue. every time whenever restart the server this
asterisk it going stop.

inthis server my hardware configuration is
/etc/asterisk/dahdi-channels.conf
group=0,11
context=default
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
context = default
group = 63

/etc/asterisk/chan_dahdi.conf
[trunkgroups]

[channels]
#include dahdi-channels.conf
language=en
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
restictcid=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
relaxdtmf=yes
rxgain=0.0
txgain=0.0
;group=1
;callgroup=1
;pickupgroup=1
busydetect=yes
busycount=6
immediate=no
resetinterval=never
switchtype=euroisdn
signalling=pri_cpe
pridialplan=unknown
prilocaldialplan=unknown
group=0
channel => 1-20


/etc/dahdi/system.conf
span=1,1,0,ccs,hdb3,crc4
# termtype: te
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31


Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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[asterisk-users] Call Park announcing to Caller rather than callee.

[Jeff Johnson]

We are having several issues with call parking in Asterisk 1.8.5.
First, when a call is parked it is announcing the park location to the
caller rather than the callee.  We also are experiencing an issue
whereby if you attempt to retrieve a parked call when a new call is
incoming the new caller and the parked caller are connected together.

 

 

 

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Re: [asterisk-users] Answering machine answers after pickup a phone.

[Jorge Barreiro]

Hi,

thanks for your time!

O Venres, 5 de Agosto de 2011 12:35:05 escribiches:
> Completely normal operation.
> You need to read and understand more basic telephony and analog lines to
> understand why that won't work.

I definitely have a lot to learn yet. 

> Asterisk needs to be in control, and once someone answers a phone not under
> Asterisk control, or the call is abandoned there is little you can do.

What I pretend is that asterisk detects that it's not under control and gets 
out of the way. The same way it detects a remote hangup and stops the 
dialplan, it could detect that someone else answered (the line is not ringing 
anymore) and discard it the same way it does when the remote part hangup.

I've read comments in forums and tutorials that seem to imply that this 
happens, but I couldn't find any confirmation (and indeed, it's not happening 
to 
me).

If you confirm me that this is the normal behavior, then I at least I know my 
solution is in the dialplan and not a card/line/driver problem.

> Sounds like a task for a simple answering machine from Wal-Mart
> All you other phones should be connected to FXS ports, or you need to be
> smarter in your dialplan. Once you answer, Asterisk is behaving normally

Yes, it's a really simple task, but this should be just a starting point. The 
plan is to start migrating services to the PBX little by little, and the 
voicemail looked like the easier thing to start. I wanted to maintain the 
current analog phones until I feel confident with the asterisk configuration. 
Maybe it wasn't such a great idea, and I should start by moving the phones to 
FXS ports in the PBX.


> 
> John Novack
> 
> Jorge Barreiro wrote:
> > Hi again,
> > 
> > thanks for your answer, but it didn't solve the problem. That Dial
> > command returns inmediately, so I don't even have the delay.
> > 
> > I'll try to explain myself better. The PBX has only one FXO card,
> > connected to the PSTN. There is no other phones connected to the PBX nor
> > SIP extensions. There are analog phones connected to the same PSTN.
> > 
> > What I try to do is that, when there is an incoming call from the ouside,
> > if someone answers on a phone, then the PBX won't answer.
> > 
> > 
> > Thanks.
> > 
> > O Venres, 5 de Agosto de 2011 00:04:02 Ruben Rögels escribiu:
> >> Hi,
> >> 
> >> your concept using Wait() won't work here.
> >> Try it like this:
> >> 
> >> [incoming]
> >> exten =>  s,1,Dial(DAHDI/1234,30) ; This will ring the phone 30s
> >> exten =>  s,n,BackGround(wellcome-message)
> >> exten =>  s,n,Voicemail(1234)
> >> exten =>  #,1,Hangup()
> >> 
> >> So, of you answer the call within 30s, you'll get the call on your
> >> phone. After 30s, the Voicemail will answer the phone.
> >> 
> >> 
> >> regards,
> >> Ruben
> >> 
> >> Am 04.08.2011 21:39, schrieb Jorge Barreiro:
> >>> Hello,
> >>> 
> >>> I'm configuring an Asterisk PBX to use as an answering machine. I have
> >>> a FXO card connected to the line, and other analog telephones
> >>> connected to the same line. The PBX answers and redirects you to the
> >>> voicemail after a delay.
> >>> 
> >>> The problem is that even if I pickup any analog phone in the line
> >>> before the PBX does, it answers after the delay anyway. And I couldn't
> >>> find how to prevent this, or even if this is supposed to happen.
> >>> 
> >>> My FXO card is a cheap X100P (source of problems, I know), and I'm
> >>> using the Asterisk version included in Debian Squeeze (1.6.2.9).
> >>> My dial plan looks like this:
> >>> 
> >>> [incoming]
> >>> exten =>  s,1,Wait(8)
> >>> exten =>  s,2,Answer
> >>> exten =>  s,3,BackGround(wellcome-message)
> >>> exten =>  s,4,Voicemail(1234)
> >>> exten =>  #,1,Hangup
> >>> 
> >>> I don't know if this is related, but I'm in Spain and I had to add:
> >>> hanguponpolarityswitch=yes
> >>> to the chan_dahdi.conf file so that asterisk detects the remote hangup.
> >>> I also added:
> >>> answeronpolarityswitch=yes
> >>> but this didn't help. It seems to be used just to detect the answer
> >>> when you are calling, not when receiving a call.
> >>> 
> >>> 
> >>> I'd appreciate any help you could provide.
> >>> 
> >>> Thanks!
> >>> 
> >>> --
> >>> _
> >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >>> 
> >>> New to Asterisk? Join us for a live introductory webinar every Thurs:
> >>> http://www.asterisk.org/hello
> >>> 
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> >>> 
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> >>> http://lists.digium.com/mailman/listinfo/asterisk-users
> >> 
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> >> 
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> >> 
> >> To UNSUBSCRIBE or update o

Re: [asterisk-users] Answering machine answers after pickup a phone.

[Roger Burton West]

On Fri, Aug 05, 2011 at 10:59:03AM +0200, Jorge Barreiro wrote:

>What I try to do is that, when there is an incoming call from the ouside, if 
>someone answers on a phone, then the PBX won't answer.

I have a couple of VoIP phones fed through Asterisk, as well as analogue
phones linked directly to the line. In this case, picking up the
analogue phone stops the VoIP phones ringing (after ten seconds or so).
I don't know whether this would be achievable with the Asterisk
console and soundcard drivers...

Roger

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[asterisk-users] (no subject)

[Jeff Johnson]

We are having several issues with call parking in Asterisk 1.8.5.
First, when a call is parked it is announcing the park location to the
caller rather than the callee.  We also are experiencing an issue
whereby if you attempt to retrieve a parked call when a new call is
incoming the new caller and the parked caller are connected together.

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Re: [asterisk-users] display name

[bakko]

Hello,

I use asteirsk 1.6 but i think you can set the callerid variable en asterisk 
1.4 to.

CALLERID(num)=test

before de dial application.

Regards

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[asterisk-users] Send Refer with replaces from asterisk

[Nikhil]

Hi
How to send REFER with replaces from asterisk (Sending out) for 
doing attended transfer.


Thanks
Nikhil

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[asterisk-users] display name

[salaheddine elharit]

hello list

My question if there is any way to display a name to the customer when i
call from my Number using asterisk 1.4



Exp: I call from my number 0520 and I want when the customer recive I
call from 0520 he sees in his phone a name “test”





Thanks and regards
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Re: [asterisk-users] Answering machine answers after pickup a phone.

[Dan Journo]

> Maybe you can try with to detect an answered call with BackGroundDetect()



> exten => s,1,Answer()

> exten => s,n,BackGroundDetect(silence/10)

> exten => s,n,Voicemail(1234)



I thought about that kind of solution, however...



The problem with this is that the caller will hear silence because Asterisk 
will pick up the phone as soon as it starts ringing.

In addition, if the caller happens to be talking before the call is answered, 
BackGroundDetect(silence/10) won't work.




Dan Journo
Kesher Communications (UK)
Business Phone Systems | Hosted 
PBX




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Re: [asterisk-users] Answering machine answers after pickup a phone.

[isrlgb]

You should change in dahdi conf the amount of time (rings) it should wait 
before answering

The dialplan doesn't handle that 
-Original Message-
From: Ruben Rögels 
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 05 Aug 2011 12:36:46 
To: Asterisk Users Mailing List - Non-Commercial 
Discussion
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] Answering machine answers after pickup a phone.

Hi!

I'm sorry that I have misundertood your question, didn't read it
carefully enough.
So you have your asterisk and your phone conntected to the same incoming
line.

Maybe you can try with to detect an answered call with BackGroundDetect()

exten => s,1,Answer()
exten => s,n,BackGroundDetect(silence/10)
exten => s,n,Voicemail(1234)

exten => talk,1,HangUp()


I can't try it for your setup with a POTS line, but I think this might
work, especially when you tune the time values for BackGroundDetect().

Quote of the manual:

--- SNIP ---
 -= Info about application 'BackgroundDetect' =-

[Synopsis]
Background a file with talk detect

[Description]
  BackgroundDetect(filename[|sil[|min|[max]]]):  Plays  back  a  given
filename, waiting for interruption from a given digit (the digit must
start the beginning of a valid extension, or it will be ignored).
During the playback of the file, audio is monitored in the receive
direction, and if a period of non-silence which is greater than 'min' ms
yet less than 'max' ms is followed by silence for at least 'sil' ms then
the audio playback is aborted and processing jumps to the 'talk' extension
if available.  If unspecified, sil, min, and max default to 1000, 100, and
infinity respectively.
--- SNAP ---

Hope this helps.

regards,
Ruben


Am 05.08.2011 10:59, schrieb Jorge Barreiro:
> Hi again,
> 
> thanks for your answer, but it didn't solve the problem. That Dial command 
> returns inmediately, so I don't even have the delay.
> 
> I'll try to explain myself better. The PBX has only one FXO card, connected 
> to 
> the PSTN. There is no other phones connected to the PBX nor SIP extensions. 
> There are analog phones connected to the same PSTN.
> 
> What I try to do is that, when there is an incoming call from the ouside, if 
> someone answers on a phone, then the PBX won't answer.
> 
> 
> Thanks.
> 
> O Venres, 5 de Agosto de 2011 00:04:02 Ruben Rögels escribiu:
>> Hi,
>>
>> your concept using Wait() won't work here.
>> Try it like this:
>>
>> [incoming]
>> exten => s,1,Dial(DAHDI/1234,30) ; This will ring the phone 30s
>> exten => s,n,BackGround(wellcome-message)
>> exten => s,n,Voicemail(1234)
>> exten => #,1,Hangup()
>>
>> So, of you answer the call within 30s, you'll get the call on your
>> phone. After 30s, the Voicemail will answer the phone.
>>
>>
>> regards,
>> Ruben
>>
>> Am 04.08.2011 21:39, schrieb Jorge Barreiro:
>>> Hello,
>>>
>>> I'm configuring an Asterisk PBX to use as an answering machine. I have a
>>> FXO card connected to the line, and other analog telephones connected to
>>> the same line. The PBX answers and redirects you to the voicemail after
>>> a delay.
>>>
>>> The problem is that even if I pickup any analog phone in the line before
>>> the PBX does, it answers after the delay anyway. And I couldn't find how
>>> to prevent this, or even if this is supposed to happen.
>>>
>>> My FXO card is a cheap X100P (source of problems, I know), and I'm using
>>> the Asterisk version included in Debian Squeeze (1.6.2.9).
>>> My dial plan looks like this:
>>>
>>> [incoming]
>>> exten => s,1,Wait(8)
>>> exten => s,2,Answer
>>> exten => s,3,BackGround(wellcome-message)
>>> exten => s,4,Voicemail(1234)
>>> exten => #,1,Hangup
>>>
>>> I don't know if this is related, but I'm in Spain and I had to add:
>>> hanguponpolarityswitch=yes
>>> to the chan_dahdi.conf file so that asterisk detects the remote hangup.
>>> I also added:
>>> answeronpolarityswitch=yes
>>> but this didn't help. It seems to be used just to detect the answer when
>>> you are calling, not when receiving a call.
>>>
>>>
>>> I'd appreciate any help you could provide.
>>>
>>> Thanks!
>>>
>>> --
>>>_
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>>
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> --
>>_
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users


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Re: [asterisk-users] Answering machine answers after pickup a phone.

[Carlos M Cruz]

John is absolutly right. You should connect your phones to an FXS port.

Otherwise you can't do what you wan't.

Regards,

Carlos M Cruz
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Re: [asterisk-users] Answering machine answers after pickup a phone.

[John Novack]

Completely normal operation.
You need to read and understand more basic telephony and analog lines to 
understand why that won't work.

Asterisk needs to be in control, and once someone answers a phone not under 
Asterisk control, or the call is abandoned there is little you can do.
Sounds like a task for a simple answering machine from Wal-Mart
All you other phones should be connected to FXS ports, or you need to be 
smarter in your dialplan.
Once you answer, Asterisk is behaving normally


John Novack



Jorge Barreiro wrote:

Hi again,

thanks for your answer, but it didn't solve the problem. That Dial command
returns inmediately, so I don't even have the delay.

I'll try to explain myself better. The PBX has only one FXO card, connected to
the PSTN. There is no other phones connected to the PBX nor SIP extensions.
There are analog phones connected to the same PSTN.

What I try to do is that, when there is an incoming call from the ouside, if
someone answers on a phone, then the PBX won't answer.


Thanks.

O Venres, 5 de Agosto de 2011 00:04:02 Ruben Rögels escribiu:

Hi,

your concept using Wait() won't work here.
Try it like this:

[incoming]
exten =>  s,1,Dial(DAHDI/1234,30) ; This will ring the phone 30s
exten =>  s,n,BackGround(wellcome-message)
exten =>  s,n,Voicemail(1234)
exten =>  #,1,Hangup()

So, of you answer the call within 30s, you'll get the call on your
phone. After 30s, the Voicemail will answer the phone.


regards,
Ruben

Am 04.08.2011 21:39, schrieb Jorge Barreiro:

Hello,

I'm configuring an Asterisk PBX to use as an answering machine. I have a
FXO card connected to the line, and other analog telephones connected to
the same line. The PBX answers and redirects you to the voicemail after
a delay.

The problem is that even if I pickup any analog phone in the line before
the PBX does, it answers after the delay anyway. And I couldn't find how
to prevent this, or even if this is supposed to happen.

My FXO card is a cheap X100P (source of problems, I know), and I'm using
the Asterisk version included in Debian Squeeze (1.6.2.9).
My dial plan looks like this:

[incoming]
exten =>  s,1,Wait(8)
exten =>  s,2,Answer
exten =>  s,3,BackGround(wellcome-message)
exten =>  s,4,Voicemail(1234)
exten =>  #,1,Hangup

I don't know if this is related, but I'm in Spain and I had to add:
hanguponpolarityswitch=yes
to the chan_dahdi.conf file so that asterisk detects the remote hangup.
I also added:
answeronpolarityswitch=yes
but this didn't help. It seems to be used just to detect the answer when
you are calling, not when receiving a call.


I'd appreciate any help you could provide.

Thanks!

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Re: [asterisk-users] Answering machine answers after pickup a phone.

[Ruben Rögels]

Hi!

I'm sorry that I have misundertood your question, didn't read it
carefully enough.
So you have your asterisk and your phone conntected to the same incoming
line.

Maybe you can try with to detect an answered call with BackGroundDetect()

exten => s,1,Answer()
exten => s,n,BackGroundDetect(silence/10)
exten => s,n,Voicemail(1234)

exten => talk,1,HangUp()


I can't try it for your setup with a POTS line, but I think this might
work, especially when you tune the time values for BackGroundDetect().

Quote of the manual:

--- SNIP ---
 -= Info about application 'BackgroundDetect' =-

[Synopsis]
Background a file with talk detect

[Description]
  BackgroundDetect(filename[|sil[|min|[max]]]):  Plays  back  a  given
filename, waiting for interruption from a given digit (the digit must
start the beginning of a valid extension, or it will be ignored).
During the playback of the file, audio is monitored in the receive
direction, and if a period of non-silence which is greater than 'min' ms
yet less than 'max' ms is followed by silence for at least 'sil' ms then
the audio playback is aborted and processing jumps to the 'talk' extension
if available.  If unspecified, sil, min, and max default to 1000, 100, and
infinity respectively.
--- SNAP ---

Hope this helps.

regards,
Ruben


Am 05.08.2011 10:59, schrieb Jorge Barreiro:
> Hi again,
> 
> thanks for your answer, but it didn't solve the problem. That Dial command 
> returns inmediately, so I don't even have the delay.
> 
> I'll try to explain myself better. The PBX has only one FXO card, connected 
> to 
> the PSTN. There is no other phones connected to the PBX nor SIP extensions. 
> There are analog phones connected to the same PSTN.
> 
> What I try to do is that, when there is an incoming call from the ouside, if 
> someone answers on a phone, then the PBX won't answer.
> 
> 
> Thanks.
> 
> O Venres, 5 de Agosto de 2011 00:04:02 Ruben Rögels escribiu:
>> Hi,
>>
>> your concept using Wait() won't work here.
>> Try it like this:
>>
>> [incoming]
>> exten => s,1,Dial(DAHDI/1234,30) ; This will ring the phone 30s
>> exten => s,n,BackGround(wellcome-message)
>> exten => s,n,Voicemail(1234)
>> exten => #,1,Hangup()
>>
>> So, of you answer the call within 30s, you'll get the call on your
>> phone. After 30s, the Voicemail will answer the phone.
>>
>>
>> regards,
>> Ruben
>>
>> Am 04.08.2011 21:39, schrieb Jorge Barreiro:
>>> Hello,
>>>
>>> I'm configuring an Asterisk PBX to use as an answering machine. I have a
>>> FXO card connected to the line, and other analog telephones connected to
>>> the same line. The PBX answers and redirects you to the voicemail after
>>> a delay.
>>>
>>> The problem is that even if I pickup any analog phone in the line before
>>> the PBX does, it answers after the delay anyway. And I couldn't find how
>>> to prevent this, or even if this is supposed to happen.
>>>
>>> My FXO card is a cheap X100P (source of problems, I know), and I'm using
>>> the Asterisk version included in Debian Squeeze (1.6.2.9).
>>> My dial plan looks like this:
>>>
>>> [incoming]
>>> exten => s,1,Wait(8)
>>> exten => s,2,Answer
>>> exten => s,3,BackGround(wellcome-message)
>>> exten => s,4,Voicemail(1234)
>>> exten => #,1,Hangup
>>>
>>> I don't know if this is related, but I'm in Spain and I had to add:
>>> hanguponpolarityswitch=yes
>>> to the chan_dahdi.conf file so that asterisk detects the remote hangup.
>>> I also added:
>>> answeronpolarityswitch=yes
>>> but this didn't help. It seems to be used just to detect the answer when
>>> you are calling, not when receiving a call.
>>>
>>>
>>> I'd appreciate any help you could provide.
>>>
>>> Thanks!
>>>
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>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
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>>>
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>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
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Re: [asterisk-users] Answering machine answers after pickup a phone.

[Jorge Barreiro]

Hi again,

thanks for your answer, but it didn't solve the problem. That Dial command 
returns inmediately, so I don't even have the delay.

I'll try to explain myself better. The PBX has only one FXO card, connected to 
the PSTN. There is no other phones connected to the PBX nor SIP extensions. 
There are analog phones connected to the same PSTN.

What I try to do is that, when there is an incoming call from the ouside, if 
someone answers on a phone, then the PBX won't answer.


Thanks.

O Venres, 5 de Agosto de 2011 00:04:02 Ruben Rögels escribiu:
> Hi,
> 
> your concept using Wait() won't work here.
> Try it like this:
> 
> [incoming]
> exten => s,1,Dial(DAHDI/1234,30) ; This will ring the phone 30s
> exten => s,n,BackGround(wellcome-message)
> exten => s,n,Voicemail(1234)
> exten => #,1,Hangup()
> 
> So, of you answer the call within 30s, you'll get the call on your
> phone. After 30s, the Voicemail will answer the phone.
> 
> 
> regards,
> Ruben
> 
> Am 04.08.2011 21:39, schrieb Jorge Barreiro:
> > Hello,
> > 
> > I'm configuring an Asterisk PBX to use as an answering machine. I have a
> > FXO card connected to the line, and other analog telephones connected to
> > the same line. The PBX answers and redirects you to the voicemail after
> > a delay.
> > 
> > The problem is that even if I pickup any analog phone in the line before
> > the PBX does, it answers after the delay anyway. And I couldn't find how
> > to prevent this, or even if this is supposed to happen.
> > 
> > My FXO card is a cheap X100P (source of problems, I know), and I'm using
> > the Asterisk version included in Debian Squeeze (1.6.2.9).
> > My dial plan looks like this:
> > 
> > [incoming]
> > exten => s,1,Wait(8)
> > exten => s,2,Answer
> > exten => s,3,BackGround(wellcome-message)
> > exten => s,4,Voicemail(1234)
> > exten => #,1,Hangup
> > 
> > I don't know if this is related, but I'm in Spain and I had to add:
> > hanguponpolarityswitch=yes
> > to the chan_dahdi.conf file so that asterisk detects the remote hangup.
> > I also added:
> > answeronpolarityswitch=yes
> > but this didn't help. It seems to be used just to detect the answer when
> > you are calling, not when receiving a call.
> > 
> > 
> > I'd appreciate any help you could provide.
> > 
> > Thanks!
> > 
> > --
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> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > 
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> >http://www.asterisk.org/hello
> > 
> > asterisk-users mailing list
> > 
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
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> 
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[asterisk-users] Custom Dialplan

[Richard Zulu]

Hey,

I have been using asterisk on slackware and had thus come up with my own
dialplan.

I would like to import my dialplan into freepbx+asterisk since I am
switching to that...how can I create my own custom dialplan in freepbx?

Thanks

Richard Zulu

Twitter
www.twitter.com/richardzulu

Skype: zulu.richard
*
*
*There is no place like 127.0.0.1*
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Re: [asterisk-users] pickupgroup

[Ishfaq Malik]

On Thu, 2011-08-04 at 11:20 -0400, Dan Journo wrote:
> Hi, 
> 
>  
> 
> Using 1.4, I see that pickupgroup can only be between 1 and 63.  
> 
> We run a hosted PBX service and need to give our client access to the
> call pickup feature. 
> 
> I thought that I could simply use the client's ID number for the
> pickupgroup number. 
> 
> The client's ID number is generated by our CRM software and is unique
> for each client.
> 
> That way, I can ensure clients can only pick up calls from extensions
> on their account. 
> 
> However, since pickupgroup can only be between 1 and 63, this causes a
> problem. 
> 
> I was wondering what people have done to get around this. 
> 
> Any feedback is greatly appreciated. 
> 
> Many thanks
> 
> Dan 
> 

> 
Just want to say I'm totally with Dan on this one, we run a hosted
service and the limit on pickup groups will be getting problematic soon.

I know one can use Pickup application within a dialplan but in an office
with lots of phones, it can be hard to discern which extension it is
that is ringing...

-- 
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