Re: [asterisk-users] Base memory usage

2011-01-01 Thread Gilles
On Fri, 31 Dec 2010 08:11:18 -0600, Danny Nicholas
da...@debsinc.com wrote:
Incidently, is there a sure-fire way (eg. checking error messages in
Asterisk's log file) to know which modules a given Asterisk setup
needs, so we can safely not load unneeded modules?

Check /var/log/asterisk/full from your last 1.6 startup.  The list of
modules from there should be what you need in 1.8.

Thanks for the tip. The appliance doesn't log messages in ./full, but
I'll check how to enable it.


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Re: [asterisk-users] Log and forward calls to cellphone?

2011-01-01 Thread Gilles
On Wed, 29 Dec 2010 16:55:46 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
I wouldn't be one of your friend: when I'm calling you I call a landline 
but finally will be charged for a mobile call (imagine I have free calls 
to landlines from my ISP). I give you an information: in France you 
don't have the right to do this unless you have it precise *before* 
redirection.

I checked with the VOSP: Apparently, it doesn't support getting an SIP
message to forward calls on the fly, and I pay for the forwarded leg
of the call (the caller will pay his part).

Thanks guys.


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

2011-01-01 Thread bilal ghayyad
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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[asterisk-users] Saving the monitor file on new file always using Monitor(wav, Record1, m)

2011-01-01 Thread bilal ghayyad
Dear List;

For each call (in specific case), I need to do a record and save in a spearated 
file, so I am thinking the best thing is to save based on the time.

Monitor(wav,Record1,m)

So, how can I make the file name to be based on the current time (which is 
changed always, or based on the some unique paramter (related to the call it 
self).

Any advise?

Regards
Bilal


  

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Re: [asterisk-users] Base memory usage

2011-01-01 Thread Robert Fantini
did you check :
 /var/log/asterisk/full

On Sat, Jan 1, 2011 at 12:30 PM, Gilles codecompl...@free.fr wrote:

 On Fri, 31 Dec 2010 08:11:18 -0600, Danny Nicholas
 da...@debsinc.com wrote:
 Incidently, is there a sure-fire way (eg. checking error messages in
 Asterisk's log file) to know which modules a given Asterisk setup
 needs, so we can safely not load unneeded modules?
 
 Check /var/log/asterisk/full from your last 1.6 startup.  The list of
 modules from there should be what you need in 1.8.

 Thanks for the tip. The appliance doesn't log messages in ./full, but
 I'll check how to enable it.


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Re: [asterisk-users] DIALSTATUS on CANCEL

2011-01-01 Thread Bryant Zimmerman
Vandar

I know understand what you are saying here. Once I turned on CEL I was able 
to see when and where each hangup was firing for each channel and the order 
of operations here.  I am now moving very aggressively to get to CEL as I 
now see why CDR's are so broken. I have my CEL to CDR translator in testing 
and this is looking very promising.

Thanks for your help.
Bryant


 From: brya...@zktech.com
Sent: Friday, December 24, 2010 9:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DIALSTATUS on CANCEL

If a call is hung up before an answer our h extension is not running in 
our dial macro 

Bryant

On Dec 24, 2010, at 3:47 AM, Vardan Harutyunyan hvarda...@gmail.com 
wrote:

 Hello Bryant
 Extension h is worked in any case of hangup.
 It not important to that the call was answered or no.
 It also be more flexible, if you use instead of ${DIALSTATUS}use 
${HANGUPCAUSE}, while in most of cause ${DIALSTATUS} is returned the same 
return code.
 http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
 
 
 -- 
 Vardan Harutyunyan,
 Senior System Administrator
 
 Enterprise Incubator Foundation
 123 Hovsep Emin Street,
 Yerevan 0051, Republic of Armenia
 Tel: + 374 10 219735
 Fax: + 374 10 219777
 E-mail: i...@eif.am
 www.eif-it.com
 
 Bryant Zimmerman wrote:
 Vardan
 
 I have not use AEL so it is a bit hard to follow with the formatting 
the
 way it is but it looks like correct.
 Please note the h extension only appears to run if a call is 
connected
 so I do not know when the CANCEL would ever be set.
 There may be someone else who can speak to this. It also appears thet
 ${DIALSTATUS} may not be set if the call is not allowed to time out or
 dialed. To me it would make sense to set the inital state of the
 ${DIALSTATUS} to CANCEL and if nothing changes it that would stand, but
 I may be missing the point on this can anyone else speak to it?
 
 Bryant
 
 

 *From*: Vardan Harutyunyan hvarda...@gmail.com
 *Sent*: Thursday, December 23, 2010 2:11 AM
 *To*: asterisk-users@lists.digium.com
 *Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
 
 I have make test in AEL.
 
 context fu {
 
 _000./userN = {
 Dial(SIP/${EXTEN:3...@prov);
 Noop(${DIALSTATUS});
 };
 h = {
 Noop(${DIALSTATUS});
 };
 };
 
 And look CLI
 -- Executing [00018185402...@fu:1] NoOp(SIP/userN-b6317738, )
 in new stack
 -- Executing [00018185402...@fo:2] Dial(SIP/user3-b6317738,
 SIP/18185402...@prov) in new stack
 -- Called 18185402...@prov
 -- SIP/Prov-082a83b8 is making progress passing it to
 SIP/userN-b6317738
 == Spawn extension (fu, 00018185402020, 2) exited non-zero on
 'SIP/user3-b6317738'
 -- Executing [...@fu:1] NoOp(SIP/userN-b6317738, CANCEL) in new stack
 
 I think, I am right
 
 --
 Vardan Harutyunyan,
 Senior System Administrator
 
 Enterprise Incubator Foundation
 123 Hovsep Emin Street,
 Yerevan 0051, Republic of Armenia
 Tel: + 374 10 219735
 Fax: + 374 10 219777
 E-mail: i...@eif.am
 www.eif-it.com
 
 Bryant Zimmerman wrote:
 The Dial Status is not set when accessing it from the h extension.
 
 Bryant
 
 

 *From*: Vardan Harutyunyan hvarda...@gmail.com
 *Sent*: Wednesday, December 22, 2010 10:39 AM
 *To*: asterisk-users@lists.digium.com
 *Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
 
 Try to use h extension
 
 --
 Vardan Harutyunyan,
 Senior System Administrator
 
 Enterprise Incubator Foundation
 123 Hovsep Emin Street,
 Yerevan 0051, Republic of Armenia
 Tel: + 374 10 219735
 Fax: + 374 10 219777
 E-mail: i...@eif.am
 www.eif-it.com
 
 Michael wrote:
  Hi Nikhil,
 
  Both debug and verbose are set to 20. That's all I got, but as you 
can
  see, for the other types of reasons, the DIALSTATUS got a value (and 
we
  see the events). I'm pretty sure it's a bug.
 
  Michael
 
  On Wed, Dec 22, 2010 at 9:01 AM, Nikhil d.nik...@cem-solutions..net
  mailto:d.nik...@cem-solutions.net wrote:
 
  Hi
  Enable debug level to more than 1 ,you may get something.
 
  Thanks
  Nikhil
 
  On 12/22/2010 11:26 AM, Michael wrote:
 
  Spawn extension (incoming-private, , 3) exited non-zero
  on 'SIP/Proxy-0031'
 
 
 
 
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Re: [asterisk-users] Saving the monitor file on new file always using Monitor(wav, Record1, m)

2011-01-01 Thread Nic Colledge
Try using ${UNIQUEID} to get the unique id of the current call. That or 
something like CDR(uniqueid). Forget which off the top of my head.
Nic.
-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: 01 January 2011 17:43
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Saving the monitor file on new file always using 
Monitor(wav, Record1, m)

Dear List;

For each call (in specific case), I need to do a record and save in a spearated 
file, so I am thinking the best thing is to save based on the time.

Monitor(wav,Record1,m)

So, how can I make the file name to be based on the current time (which is 
changed always, or based on the some unique paramter (related to the call it 
self).

Any advise?

Regards
Bilal


  

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Re: [asterisk-users] Saving the monitor file on new file always using Monitor(wav, Record1, m)

2011-01-01 Thread Bryant Zimmerman
Use a combination of ${EPOCH} with a format string and the unique call / 
channel id. 

Example:
 
exten = s,1,Set(MY_TIMEVAR=:${STRFTIME(${EPOCH},,%d%mNaVH:NaVS)}) 
exten = s,n,Monitor(wav,${MY_TIMEVAR}~${CHANNEL},m)


 From: bilal ghayyad bilmar...@yahoo.com
Sent: Saturday, January 01, 2011 1:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Saving the monitor file on new file always using 
Monitor(wav, Record1, m)

Dear List;

For each call (in specific case), I need to do a record and save in a 
spearated file, so I am thinking the best thing is to save based on the 
time.

Monitor(wav,Record1,m)

So, how can I make the file name to be based on the current time (which is 
changed always, or based on the some unique paramter (related to the call 
it self).

Any advise?

Regards
Bilal

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Re: [asterisk-users] Saving the monitor file on new file always using Monitor(wav, Record1, m)

2011-01-01 Thread Steve Edwards

On Sat, 1 Jan 2011, bilal ghayyad wrote:

For each call (in specific case), I need to do a record and save in a 
spearated file, so I am thinking the best thing is to save based on the 
time.


Read up on the STRFTIME function.

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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Base memory usage

2011-01-01 Thread Gilles
On Sat, 1 Jan 2011 13:04:29 -0500, Robert Fantini
robertfant...@gmail.com wrote:
did you check :
 /var/log/asterisk/full

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.


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Re: [asterisk-users] Log and forward calls to cellphone?

2011-01-01 Thread Sebastian

Hi,

On 01/01/2011 05:32 PM, Gilles wrote:

On Wed, 29 Dec 2010 16:55:46 +0100, Administrator TOOTAI
ad...@tootai.net  wrote:

I wouldn't be one of your friend: when I'm calling you I call a landline
but finally will be charged for a mobile call (imagine I have free calls
to landlines from my ISP). I give you an information: in France you
don't have the right to do this unless you have it precise *before*
redirection.


I checked with the VOSP: Apparently, it doesn't support getting an SIP
message to forward calls on the fly, and I pay for the forwarded leg
of the call (the caller will pay his part).


I am, in a way, in a similar situation. I have a POTS/PSTN landline 
connected to my Asterisk server - and Asterisk calls my mobile when a 
call comes in down the POTS line and then bridges the calls for me. This 
is effectively home-brew/DIY call diversion. Instead of asking the phone 
company to divert the calls when I'm not home, I setup Asterisk to do 
that for me. The slight advantage in doing it myself is that I use 
another SIP provider for the outgoing leg of the call - who charges me 
far less per minute then my landline provider would charge me for their 
divert feature. They even charge an extra monthly fee for having the 
divert feature!


I take it the above is your option number one - which you are trying to 
avoid. I'm afraid your option number two doesn't really exist - as far 
as I know. First of all - as the others have pointed out, the incoming 
call has dialled a landline number - and expects to pay for a call to a 
landline number. So any diversion happening would be your responsability 
to pay for. That is of course if you don't live in USA or Canada - where 
I believe calls *to* mobiles are similarly charged as calls *to* 
landlines - and it is the receiving end who gets charged for calls to 
mobiles. So in general - sending any sort of message to phone provider 
and asking them, on the fly, to send the call to another number - 
without you being charged - is most likely impossible - and will stay 
that way.


The closest you will come to this is if you have a call divert with the 
phone company, and a package which allows free calls to a specific 
mobile phone (or free mobile minutes). I used to be with a landline 
provider - who gave me free unlimited calls from my landline to my 
mobile phone. They didn't realised that this would mean I could have 
call diverts from my landline to my mobile free as well - as effectively 
I was being charged as if my house phone would call my mobile! This 
worked for about two years - until I had to move house, and provider.


Anyway - there is a third option - which I have been using with some 
success. I connected my softphone on my laptop to my Asterisk server at 
home (through OpenVPN for extra security - but this is not compulsory). 
Sometime I keep my laptop on when out in the field at clients, with 
Internet connection running - and pick-up incoming calls on the laptop. 
This way the divert part of the call is free - as it is coming through 
the Internet to my laptop. I configured my phone divert (in Asterisk) to 
ring simultaneously my mobile and my softphone when a call comes down 
the landline. I answer on whichever one I want. I don't use Followme - I 
don't like the way it has been implemented (the line gets answered early 
- not when I answer the mobile or softphone).


As a last alternative - a slight improvement on the above. If you can 
get a smartphone with Android - which would let you run SIP over 3G - 
you should have true free voice divert. Everything would be as above - 
the main difference is that the phone (instead of the laptop) would be 
on and connected all the time - even when moving out and about - which 
with a laptop is not feasible. This would allow you to answer your calls 
through the 3G data link - and not be charged per minute. If your mobile 
phone company will let you do that (run SIP over 3G). This is where an 
OpenVPN (or any other VPN) connection again would come in handy - they 
shouldn't be able to tell you are running SIP - if it is inside VPN ;-) 
I haven't trialled this version yet - but this would be my ultimate call 
diversion setup.


Hope the above helps,

Sebastian




Thanks guys.


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

2011-01-01 Thread Sebastian

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

2011-01-01 Thread JP CR

Greetings,

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.

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

2011-01-01 Thread Sebastian

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).


Once you are using OpenVPN - it is a bit easier to setup a failover 
scenario - or even load-balancing - as Openvpn has some inbuit 
functionality in this direction. There has been some recent discussion 
on the OpenVPN list about load balancing and failover setups if I 
remember correctly.


Just a thought,

Sebastian


On 12/25/2010 11:01 PM, Dave George wrote:


Server will have two fix public ips.



Dave

   Original Message 
  Subject: Re: [asterisk-users] load balance with 2 wan connections
  From: Alejandro Imass
  Date: Sat, December 25, 2010 1:58 pm
  To: Asterisk Users Mailing List - Non-Commercial Discussion
 
 
  On Sat, Dec 25, 2010 at 1:18 PM, dave george wrote:
   Need some advise or paid help on running asterisk on two WAN
connection. �I
   need load balancing and failover support.
  
   WAN: 1 DSL + 1 Cable ISP.
  
 
  There are _many_ issues. First outgoing and incoming traffic is
  completely different for what you want to do.
 
  Second SIP is hard enough to NAT and route with a single IP let alone
  2 or more and probably dynamic!
 
  Third, load balancing/fail-over is not a simple matter even doing by
  hand with Linux or BSD, there will still be issues with static routing
  and such. There are some cheap hw that may claim it does, but most
  probably it will not be meant for VoIP, SIP or IAX.
 
  Depending on your budget and needs, if you need reliability and high
  bandwidth, probably a better solution is to host your main pbx in a
  reliable server on a fixed and public IP and then route the calls to a
  local Asterisk using IAX and even SIP. If local bandwidth is limited
  IAX is a better bet. By having a public box routing calls to local
  box(es) on your private LAN, you could load balance with multiple
  local Asterisk servers (easy balance by dialplan, for example). To
  save on hardware, you could use virtualization or FreeBSD Jails for
  example. Dunno how the telephony hw works with virtualization or jails
  (yet, thoug I do have a single Asterisk running on a FBSD jail).
 
  Good luck,
  Alejandro Imass
 
  
   Dave
  
  
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[asterisk-users] CDR Questions

2011-01-01 Thread Mike Diehl
Hi all,

I've got two questions about CDR's.

1.  I'd like to start logging the IP address that a call orginates from. 
I'm sure I can get this into the userfield of my CDR table, but what
variable should I use to get this value?  I looked at the variables page at
voip-info and didn't find anything that clearly indicated what I want.

2. I would like to be able to generate a report that indicates that
so-and-so called such-and-such on a given date.  However, the CDR's seem to
keep the caller and callee in different fields depending on which way the
call went and sometimes the phone numbers are  embedded in a dial string
that would have to be decoded to extract the phone number.  What is everyone
else doing to generate a simple report?

TIA,

Mike

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