Re: [asterisk-users] Callback form to place on site for customers. Recomendation to achieve this.

2011-01-02 Thread Randy R
On Sun, Jan 2, 2011 at 1:04 AM, JP CR jprollersk...@hotmail.com wrote:
 I want to place a form on my site so customers can recieve an mmediate
 callback and the PBX should connect them to a cell sales agent.

 Are there anfree modules available for this, or one should code this from
 scratch?

 What I want is when a potential client submits his number... the PBX dials
 the number makes an announcement and dials an extension (which is actually a
 cellhopne dahdi member) and makes the connection.

You could look at http://phono.com for the PhonoSDK. It can connect
you to skype, sip or PSTN destinations via Tropo applications.

I wonder though if/how you are going to have a way to check if the
number given isn't a joke. What prevents people from just typing in
all kinds of phone numbers at 6AM in California or something? Is
anyone working on that problem? Do you restrict it to hours that are
safe all over the USA? Is there a check of area codes?

Color me just curious...
/r

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[asterisk-users] Forward voicemail not working

2011-01-02 Thread duane . larson
I have asterisk 1.8.0 installed and I am not able to forward a voicemail  
from one users mailbox to another user.


I have the user log into their mailbox
press 8 to forward a message
enter the extension of the user I wish to forward too
I don't prepend a audio message
and press # to send the message to the other user

from a debug perspective I don't see any errors. The only message I see is
==  
Saving '/home/asterisk/asterisk-bin/spool/asterisk/voicemail/irock.com/9XX2XX2009/Old/msg.txt':  
[Jan 2 11:24:18] NOTICE[17036]: app_voicemail.c:5154 copy_message: Copying  
message from 9xx2xx2...@irock.com to 2...@irock.com



Yet when I look in 2011 spool directory I don't see any message at all. It  
is just not being copied. What could be the issue?
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Re: [asterisk-users] Base memory usage

2011-01-02 Thread Steve Edwards

On Sat, 1 Jan 2011 13:04:29 -0500, Robert Fantini
robertfant...@gmail.com wrote:

did you check :
/var/log/asterisk/full


On Sat, 1 Jan 2011, Gilles wrote:

Yup, but this file doesn't exist. It's an appliance with not much 
RAM/NAND memory, so it makes sense to disable logging to save space.


1) Configure Asterisk to syslog on another host.

2) Start Asterisk from the command line with lots of '-d' and '-v' and 
watch for error messages.


Personally, I prefer to use 'autoload=no' and explicitly load the modules 
I need on production systems.


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Re: [asterisk-users] Base memory usage

2011-01-02 Thread Gilles
On Sun, 2 Jan 2011 10:04:44 -0800 (PST), Steve Edwards
asterisk@sedwards.com wrote:
Personally, I prefer to use 'autoload=no' and explicitly load the modules 
I need on production systems.

Thanks Steve.


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[asterisk-users] incoming

2011-01-02 Thread Thomas Perron
Is it possible to have
Calls incoming to different DIDs?
I want an AA that handles 100s of businesses.

[Incoming-pizza]
Exten = 4045551212,1,Goto(pizza,s,1)

[Incoming-hvac]
Exten = 8085551212,1,Goto(hvac,s,1)

[Incoming-gutter]
Exten = 6175551212,1,Goto(gutter,s,1)

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Re: [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?

2011-01-02 Thread Bruce B
Thanks for the input. I have the latest drivers but it seems that there is
some serious incompatibility issue with the kernel as when the FLASHING
happens even if the system is restarted it's still not detected. One has to
re-plug it in and then it shows in wanrouter hwprobe.

It could also be that the atom board is not compatible with the driver in my
case. It your /var/log/messages upon restart do you any line that might
match this:
*
*
*cd /var/log/*
*grep -o device not accepting address **
*grep -o USB device is disconnected **
*dmesg | egrep device not accepting address *
*dmesg | egrep USB device is disconnected *

If not then you are not experiencing the same issue. If you have then it's a
universal issue and not hardware specific.

I would really appreciate it if you look into your logs and let me know.

Regards,
Bruce


On Sat, Jan 1, 2011 at 6:57 PM, Sebastian s...@open-t.co.uk wrote:

 Hi Bruce,


 On 12/28/2010 10:51 PM, Bruce B wrote:

 Thanks for the input. I can not replicate the situation as it happens
 randomely or maybe over the weekend. However I have sent you all the
 requested command and logs in a separate e-mail for your analyzes. The
 only thing that stood out at me was the output of lsusb -v at the very
 end where it timed out.

 Since all lines didn't work I am to assume that both module went down
 but per my diagnoses with hwprobe I could see one unit connected and
 the other was not when the problem happened. Simply
 connecting/disconnecting that unit or connecting it to another port
 solved the problem and it showed up in hwprobe

 This is an Acer Aspire Revo mini PC. I am wondering if the U100s draw
 too much power? The only other USB connected device is the thumb size
 wireless connector for the keyboard.

 Acer computer:

 http://reviews.cnet.com/desktops/acer-aspire-revo-ar1600/4505-3118_7-33777218.html


 Don't know if this will help - but I will butt in with what I have :-)

 I've been using a Sangoma U100 adapter for about 2 years now. It is
 connected to a Compaq V2120 laptop (Celeron M 1.4GHz processor) which serves
 as my home server. It is actually the main reason I went for the U100 - as I
 couldn't add a PCI or PCIe card to a laptop to get the FXO ports I needed. I
  have to say I really like the U100 - I believe it is the only low(ish) cost
 USB based FXO interface on the market.

 I have had occasional problems with it. I remember it used to just stop
 working - and the lights would start flashing. If I remember correctly - I
 went to Sangoma's website and downloaded the latest wanpipe drivers,
 compiled and installed them - and everything was ok after that. At the
 moment I'm running Asterisk 1.6.2.9 and wanpipe 3.5.11. I can't remember
 what version of wanpipe was giving problems, I'm afraid.

 I also found that mine doesn't really like to be hot-plugged - it just
 freezes the system with strange characters on the screen. But that was a
 while ago. Since I've learned it's foibles - it must be at least one year
 since I had to look at it.

 Sebastian



 Looking forward to your analysis.

 Regards,
 Bruce

 On Tue, Dec 28, 2010 at 3:58 PM, Moises Silva moises.si...@gmail.com
 mailto:moises.si...@gmail.com wrote:

On Tue, Dec 28, 2010 at 11:33 AM, Bruce B bruceb...@gmail.com
mailto:bruceb...@gmail.com wrote:


I appreciate your feedback and let me know what info I can post
here that may help resolve the issue (such as output from dmesg
or lspci?).


Hi Bruce,

The following would be useful for starters:

1. cat /etc/wanpipe/*.conf

2. ifconfig -a (from a working and non-working situation)

3. lspci -v and lsusb -v (from a working and non-working situation)

4. wanrouter hwprobe verbose (from a working and non-working situation)

5. /var/log/messages (near the date the problem happened)

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. m...@sangoma.com mailto:m...@sangoma.com


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Re: [asterisk-users] Cisco IP Phones and AVAYA IP Phones: How to configure in Asterisk

2011-01-02 Thread Cassius Smith
CallFwd should be one of the soft keys on your Cisco phones. Are you
re-flashing the Cisco phones with SIP?
-Cassius

On 1/2/11 3:50 AM, bilal ghayyad bilmar...@yahoo.com wrote:

Hi All;

How to configure the buttons in the Cisco IP Phones to be used for
different functionalities like Call Forward, Call Pickup, ... etc?

For example, if I need to assign one of the buttons existed at Cisco IP
Phone to be used for CallFrw, how to do this in Asterisk?

Regards
Bilal


  





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Re: [asterisk-users] incoming

2011-01-02 Thread Rick Hall
Yes, I don't see why not.  You just need to setup an IVR for each business
and then assign each individual DID to the appropriate IVR.

This may help:

http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu

Cheers!

Rick

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On Sun, Jan 2, 2011 at 11:50 AM, Thomas Perron thomas.per...@gmail.comwrote:

 Is it possible to have
 Calls incoming to different DIDs?
 I want an AA that handles 100s of businesses.

 [Incoming-pizza]
 Exten = 4045551212,1,Goto(pizza,s,1)

 [Incoming-hvac]
 Exten = 8085551212,1,Goto(hvac,s,1)

 [Incoming-gutter]
 Exten = 6175551212,1,Goto(gutter,s,1)

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Re: [asterisk-users] incoming

2011-01-02 Thread Thomas Perron
Cool.  So, one Asterisk machine handling up to 100 DID numbers, correct?
Yes. I will have unique IVR flows/plans for each.
I assume that the DID mumbers dialed would be the exaxt match needed
to start the respective context.  Correct?

On 1/3/11, Rick Hall r...@readywire.com wrote:
 Yes, I don't see why not.  You just need to setup an IVR for each business
 and then assign each individual DID to the appropriate IVR.

 This may help:

 http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu

 Cheers!

 Rick

 --
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 Senior Vice President
 ReadyWire Multimedia Solutions

 Affordable Website  Reseller Hosting
 http://www.readywire.com/
 (312) 278-4446 x5446

 Technical Support:
 24 hours a day / 7 days a week

 Customer Login...: https://secure.readywire.com/
 Server Notices.: http://status.readywire.com/
 Support Center: https://secure.readywire.com/
 Twitter.: http://twitter.com/readywire
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 This message contains confidential information and is intended only for the
 individual named. If you are not the named addressee you should not
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 this e-mail from your system. E-mail transmission cannot be guaranteed to be
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 therefore does not accept liability for any errors or omissions in the
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 Multimedia Solutions. PO BOX 811061, Chicago, IL, USA, 60681.
 www.readywire.com.



 On Sun, Jan 2, 2011 at 11:50 AM, Thomas Perron
 thomas.per...@gmail.comwrote:

 Is it possible to have
 Calls incoming to different DIDs?
 I want an AA that handles 100s of businesses.

 [Incoming-pizza]
 Exten = 4045551212,1,Goto(pizza,s,1)

 [Incoming-hvac]
 Exten = 8085551212,1,Goto(hvac,s,1)

 [Incoming-gutter]
 Exten = 6175551212,1,Goto(gutter,s,1)

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Re: [asterisk-users] load balance with 2 wan connections

2011-01-02 Thread Hans Witvliet
On Sun, 2011-01-02 at 00:10 +, Sebastian wrote:
 Hi,
 
 One possibility that you might want to explore is OpenVPN. If your VoIP 
 clients support OpenVPN (either through a local Openvpn client on the 
 clients network being used as an OpenVPN gateway, or through individual 
 clients supporting OpenVPN (laptops with softphones, or I've heard of 
 one particular hardphone supporting OpenVPN - but can't remember which).
 

afaicr, snom supports vpn.
Though probaby just plain setup, no special configs like load
balancing/H.A.


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Re: [asterisk-users] incoming

2011-01-02 Thread Steve Edwards

On Sun, 2 Jan 2011, Thomas Perron wrote:


Is it possible to have Calls incoming to different DIDs?


Yes*, depending on whether your provider 'provides' the DID in the call 
setup.



*) Better subjects attract more readers. More detail yields better 
answers.


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Re: [asterisk-users] incoming

2011-01-02 Thread Roger Burton West
On Mon, Jan 03, 2011 at 02:41:36AM +0400, Thomas Perron wrote:
Cool.  So, one Asterisk machine handling up to 100 DID numbers, correct?

As many as you like, modulo memory and CPU requirements.

I assume that the DID mumbers dialed would be the exaxt match needed
to start the respective context.  Correct?

Depends on how they're presented to you by the DID provider.


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Re: [asterisk-users] incoming

2011-01-02 Thread Steve Edwards

On Mon, 3 Jan 2011, Thomas Perron wrote:


So, one Asterisk machine handling up to 100 DID numbers, correct?


The number of DIDs is not limited. You could handle a bazillion DIDs with 
a simple dial plan like:


exten = _!.,1,  verbose(1,[${ext...@${context}])
exten = _!.,n,  playback(demo-congrats)
exten = _!.,n,  hangup()


I assume that the DID mumbers dialed would be the exaxt match needed
to start the respective context. Correct?


The exten does not determine which context is started. The provider 
configuration does.


Matching is facilitated by patterns.

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Newline  Fax: +1-760-731-3000

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[asterisk-users] Realtime SIP, multiple AX servers question

2011-01-02 Thread Bryan Field-Elliot
We have several Asterisk servers (1.6.2.15) all configured for Realtime, all 
backed by the same database. The Asterisk servers are all listed under DNS SRV 
records, and SIP ATAs find us this way.

Normally, no matter which Asterisk server an ATA connects to, we get our 
database fields filled out correctly, such as regseconds, lastms, ipadr, 
etc. However, with some ATA's we are experiencing a problem as follows:

1. ATA reaches its re-registration timeout, which we typically configure to 
be 60 minutes.
2. ATA re-queries DNS SRV record, and ends up re-registering with a different 
AX server than it was on previously.
3. The new AX server updates the Realtime DB fields (regseconds, lastms, etc).
4. The old AX server, after a few more minutes, notices that the ATA has 
vanished, and therefore clears out these same fields.

Is there any way to fix this problem? We need to know if ATA's go offline, but, 
we don't want them caught in this endless loop where our multiple AX servers 
are out-guessing eachother and overwriting valid data in the database.

Our realtime options in sip.conf are as follows:

rtcachefriends=yes
rtsavesysname=yes
;rtautoclear=yes
;ignoreregexpire=yes

Because we are using rtsavesysname, a perfect solution seems like it might be 
along the lines of If an ATA disappears, empty out the RT database fields ONLY 
if it's last regserver was this one. Is this possible?


Thank you,

Bryan


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