Re: [asterisk-users] Chatterbug

[Robert Goodyear]

Wow. How on EARTH do these people stay in business? Just running the  
law of averages and hoping it works out?

$10 a month for unlimited routing through their 800 number seems like  
a risky gamble for them.

On Nov 12, 2007, at 9:21 PM, Paul Hales wrote:

>
> http://www.oldskoolphreak.com/tfiles/voip/chatter_bug.pdf
>
> PaulH
>
>
> On Mon, 2007-11-12 at 21:07 -0800, Robert Goodyear wrote:
>> Does anyone know anything about the Chatterbug product? I can't tell
>> if it's an ATA with a modem or some sort of LCR proxy or somesuch.
>>
>> Anyone?
>>
>>
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Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files

[Greg Oliver]

On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote:
> Hello List, 
> 
> Does anyone have access to the soft key configuration files for the
> Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and
> didn’t find much up there.  
> 
> Thanks
> 

Softkeys running both SCCP and SIP firmware are both sent through the
protocols themselves.  I have done packet captures to prove it out from
CCM 5.x and 6.0.  Sorry, no xml files to accomplish it.  Maybe one day
they will be less of basterds?!?!?!?!?

-Greg

 
> 
> Roy Anciso 
> 
> Director of Technology
> 
> Manistee Intermediate School District
> 
> 1710 Merkey Road
> 
> Manistee, MI 49660
> 
> Ph: 231-723-4264
> 
> Fx: 231-723-1690
> 
> [EMAIL PROTECTED]
> 
>  
> 
> 
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Re: [asterisk-users] Toshiba DK - Asterisk Integration

[Indika Wasala]

Hi All,

Interfaces of my PBX are as follows,

Toshiba dk28 

CO Lines (to telcos) : 12 - (2 free)
Digital extensions    : 8 - (full)
Analog extensions    :18 - (full)

Toshiba dk280 

CO Lines (to telcos) : 8 - (1 free)
Digital extensions    : 16 - (5 Free)
Analog extensions   : 16 - (1 free)

Toshiba dk8

CO Lines (to telcos) : 4 - (1 free)
Digital extensions    : 8 - (2 free)
Analog extensions   : 2

Non of the systems have T1 interfaces and also it seems these systems
does not support T1. What you mean is if I need 5 IP phones (sip
extensions) I need 5 POTS interfaces. Please advice.

Thanks
Indika.

Tony Plack wrote:

  
  
  Indika,
  The question of interface depends
on how your Strata PBX are connected to the telco currently and what
interfaces your Strata supports.
   
  If all you have is POTS
interfaces to the telco, your integration may be limited because every
SIP extension will require a separate POTS line to the Strata.  But if
you have a T1 interface, you should be able to have trunked
lines/multiple extensions.
   
  So we need more details.
   
  Tony Plack
  
  
  > Hi All,
  >
  > I am new to
both Asterisk and PBX stuff. I have 3 Tohiba PBXs in 3
  > separate
offices as follows,
  >
  > Toshiba Strata
dk28
  > Toshiba Strata
dk280
  > Toshiba Strata
dk8
  >
  > I need to
install 3 Asterisk servers in these 3 locations and
  > integrate them
with each of the Toshiba PBX s. This is to give IP
  > Phones/soft
phones to the users and to route these VOIP calls
  > through the PBX
to POTS. What are the Digium cards I should use in
  > each of these
cases and How should I integrate Asterisk with above
  > systems.
  >
  > I read the
article in
  > http://www.voipinfo.org/wiki/index.php?page=Asterisk-ToshibaStrata
  > and not sure
whether that scenario fits mine. Also it is bit
  > confusing to
identify what Digium cards should I need for my cases.
  >
  > Any help is
highly appreciated.
  >
  > Thanks,
  > Indika.
  >
  >
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[asterisk-users] Asterisk ignoring manager events when busy

[Nick Adams]

Hello,

I currently have a pretty standard 1.2.21 Asterisk system running purely
SIP termination (no zap/IAX/H323..etc).

We have an auto-dialing system that generates calls via the manager API.

The system runs beautifully until it gets to about 200 calls. I can
generate these calls in quite literally seconds if desired (or minutes).

The kicker: I can't seem to get past this 200 call point even though the
system is/seems very idle. Low CPU utilisation (~40%), plenty of RAM,
close to zero disk usage (I/O Wait ~2%), bandwidth is plentiful (1Gbit)
and the current calls are crystal clear.

I get the feeling Asterisk is ignoring or dumping manager events. We
watch for the "Success" from the originate command and it comes if there
is less than roughly 200 calls but is unreliable after that.

I could understand if this was happening at 100% cpu but not at the
under-utilised state the box is at.

Any hints?


Thanks,

Nick.


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Re: [asterisk-users] asterisk and installing chan_h323.so rpm

[Bincy K. Philip]

Hello,

 While trying to install H323 support for asterisk, I missed one step.
After compiling files in channel/h323, need to select chanh323 from the menu 
and compile and install asterisk.


cd asterisk

*  ./configure

 * make menuconfig

 channel drivers->chanh323

 save the setting by giving x.


*  make 
   make

* make install



Hope this will help someone.



Thanks & Regards
Bincy K Philip


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, November 13, 2007 2:56 PM
To: asterisk-users@lists.digium.com
Subject: asterisk-users Digest, Vol 40, Issue 34


Send asterisk-users mailing list submissions to
asterisk-users@lists.digium.com

To subscribe or unsubscribe via the World Wide Web, visit
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or, via email, send a message with subject or body 'help' to
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When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-users digest..."


Today's Topics:

   1. Re: asterisk and  installing chan_h323.so rpm (Dovid B)
   2. Re: Asterisk direct dialing (Dovid B)
   3. Re: Help: Static and dropped calls (Dovid B)
   4. Re: How to delete voice mail messages? (Dovid B)
   5. Re: asterisk and  installing chan_h323.so rpm (David Boyd)
   6. Re: ztdummy and BackGround (Tony Plack)
   7. MOH Codec Issue (Nick Brown)
   8. Re: RTP traffic not being forwarded (Ryan Newington)
   9. Re: MOH Codec Issue (Paul Hales)
  10. Chatterbug (Robert Goodyear)
  11. Re: MOH Codec Issue (Nick Brown)
  12. Re: Chatterbug (Paul Hales)
  13. Re: MOH Codec Issue (Paul Hales)
  14. Re: MOH Codec Issue (Paul Hales)
  15. Stress-Testing Asterisk (Jeng Yu)
  16. Re: Stress-Testing Asterisk (Suity Zsolt)
  17. Re: Stress-Testing Asterisk (Tzafrir Cohen)
  18. Toshiba DK - Asterisk Integration (Indika Wasala)
  19. Fwd: Re:  Grandstream GXP2020 + Asterisk 1.4.11 (Erik Wartusch)


--

Message: 1
Date: Tue, 13 Nov 2007 03:00:22 +0200
From: "Dovid B" <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] asterisk and  installing chan_h323.so
rpm
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=original


- Original Message - 
From: "Bincy K. Philip" <[EMAIL PROTECTED]>
To: 
Sent: Thursday, November 08, 2007 2:13 PM
Subject: [asterisk-users] asterisk and installing chan_h323.so rpm


> Hello,
>
> When I tried to install chan_h323-1.0.1-module.i386 RPM i got these 
> errors.
>
>
> Failed dependencies:
>libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386
>   libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386
>
> But i found the same files in
>
> /usr/lib/libh323_linux_x86_r.so.1
> /usr/lib/libpt_linux_x86_r.so.1
>
>
> What to do for asterisk to detect the same files?
>
I had this issue in the past. I do not remember what I did to resolve it. In 
the end I went with the h323 channel driver located in the asterisk add-ons. 
It was a lot easier to work with and worked with out any issues. 





--

Message: 2
Date: Tue, 13 Nov 2007 03:02:08 +0200
From: "Dovid B" <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Asterisk direct dialing
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=original


- Original Message - 
From: "Gopal krishnan" <[EMAIL PROTECTED]>
To: 
Sent: Saturday, November 10, 2007 12:46 PM
Subject: [asterisk-users] Asterisk direct dialing


> Hi,
> I am using Asterisk 1.2.24, I have written my dialplan to land
> with an IVR with the same time if the customer knows the parties
> extensions they can dial directly, but what happens is sometimes its
> working and sometime its not working.
> My extensions.conf as follows,
>
> [incoming]
> exten => 052477302,1,Wait(2)
> exten => 052477302,2,NoOp(${CALLERIDNUM})
> exten => 052477302,3,Goto(from-internal,s,1)
>
> [from-internal]
> exten => s,1,Answer()
> exten => s,2,Background(welcome_pride5)
> exten => 501,1,Dial(Zap/1)
> exten => 502,1,Dial(Zap/2)
> exten => 503,1,Dial(Zap/3)
> exten => 504,1,Dial(Zap/4)
> exten => 505,1,Dial(Zap/5)
>
> I dont know what could be the reason. Is there any other way that i can 
> use.
>
> -- 
> Thank you, with regards,
> Gopal,
>

What comes up in the CLI when it does not work ? 





--

Message: 3
Date: Tue, 13 Nov 2007 03:10:36 +0200
From: "Dovid B" <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Help: Static and dropped calls
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Message-ID: <[EMAIL PROTECTED]>
Content

Re: [asterisk-users] Asterisk ignoring manager events when busy

[Doug Lytle]

Nick Adams wrote:
> The kicker: I can't seem to get past this 200 call point even though the
>   

What does your console show at this time?

When testing, I've noted the 200 call limit was because I had too many 
open files.  I had to increase this by typing ulimit -n 4096 before 
starting Asterisk.  The default is 1024.

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] Nortel digital FXO channel bank? Exists?

[Rupert Utteridge - Digital Techniques (Asia) Limited]

Citel SIP handset gateways at www.dtasia.net

Rupert Utteridge
Director - Sales & Marketing
Digital Techniques (Asia) Limited
4 The Lee
Middle Cove, NSW, 2068
Australia
 
Tel:  +61 2 9037 4191
Mobile:  +61 424 373 516
 
Web:  www.dtasia.net

Message: 1
Date: Tue, 13 Nov 2007 23:00:54 -0500
From: Jon Pounder <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Nortel digital FXO channel bank?
Exists?
To: asterisk-users@lists.digium.com
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain;   charset=ISO-8859-1; DelSp="Yes";
format="flowed"

Quoting Michelle Dupuis <[EMAIL PROTECTED]>:

> We have a client with a Nortel PBX with digital phone sets.  Due to T1
> problems (old firmware), we are interested in trying a FXO channel bank.
>
> Is there a channel bank (or equivalent) which emulates Meridian digital
> phone sets?  In order words, an FXO channel bank that's Meridian digital?

I think the basic wireline signalling is isdn bri for that, but with  
non-standard protocols.

ie the channel bank would talk to the ksu/pbx, but nothing but a  
reverse of the same hardware would understand anything on the other  
end of the t1.

ps - this sort of channel bank would be pricey at best, rediculous  
probably in reality.

I would try other solutions before even attempting this sort of thing  
since the odds of success are probably not too high.



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[asterisk-users] Linksys 942 Call Transfer

[Jon Farmer]

Hi

I have a customer who is using Linksys 942 phones.
When they try to transfer a call the Asterisk CLI
reports that  both legs of the call must exist on the
server. The call they are trying to transfer then
drops.

Does anyone know why this is and how to fix it?

TIA

Regards

Jon






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[asterisk-users] Using php exec() in agi script

[Andre Courchesne]

Hi,

  Any reason why I can not get the php exec() function to execute a shell 
command inside an agi script?

  Thanks.

Andre
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Re: [asterisk-users] What is wrong with this mailing list

[Baji Panchumarti]

  On Nov 14, 2007 12:52 AM, Erik Anderson  wrote:
> On Nov 13, 2007 11:44 PM, Mohammad Shokuie  wrote:
> >
> > HI Erik,
> >
> > thanks for your post, Actually im sending new posts not
> > replying but if you see them correct, how come its wrongly
> > viewed for me. Are you using a speciall software to view
> > mailing lists? Im just using firefox not a special one!
>
> You're using firefox?  How so?  I'd recommend either a good
> email client (Thunderbird) or a good web email interface (gmail).
>
> (I'm using gmail's web interface)

 I believe Mohammad is accessing the list pretty much the
 way many of us are, only diff is that his provider is hotmail.

 There could be two other explanations, one could be that
 many spammers use real or fake hotmail addresses and
 digium list server is aggresively filtering hotmail addresses
 for that reason. The second reason could be, that the IP
 block he is in is blacklisted either at digium or elsewhere.

 --

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Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files

[Anciso, Roy]

The Cisco Documentation states that you can modify standard and
nonstandard softkey templates.  They may not be xml files. I just
assumed they were xml since that is what is used to configure the phone.

Here is snip from the 7911G documentation that states you can configure
the private key (which is really all I need) and modifying the softkeys:

"Configuring Softkey Templates

Using Cisco Unified CallManager Administration, you can manage softkeys
associated with applications that are supported by the Cisco Unified IP
Phone 7906G and 7911G. Cisco Unified CallManager supports two types of
softkey templates: standard and nonstandard. Standard softkey templates
include Standard User, Standard Feature, Standard IPMA Assistant,
Standard IPMA Manager, and Standard IPMA Shared Mode Manager An
application that supports softkeys can have one or more standard softkey
templates associated with it. You can modify a standard softkey template
by making a copy of it, giving it a new name, and making updates to that
copied softkey template. You can also modify a nonstandard softkey
template.

To configure softkey templates, select Device > Device Settings >
Softkey Template from Cisco Unified CallManager Administration. To
assign a softkey template to a phone, use the Softkey Template field in
the Cisco Unified CallManager Administration Phone Configuration page.
Refer to Cisco Unified CallManager Administration Guide, Cisco Unified
CallManager System Guide for more information."

Here is the link to configuring a 7911G phone with SIP and instructions
for modifying the softkeys. 

http://cisco.com/en/US/customer/products/hw/phones/ps379/products_admini
stration_guide_chapter09186a0080798562.html#wp1058838

So again, if anyone can post the softkey templates I would greatly
appreciate it.  

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
Oliver
Sent: Wednesday, November 14, 2007 3:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML
Files

On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote:
> Hello List, 
> 
> Does anyone have access to the soft key configuration files for the
> Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and
> didn't find much up there.  
> 
> Thanks
> 

Softkeys running both SCCP and SIP firmware are both sent through the
protocols themselves.  I have done packet captures to prove it out from
CCM 5.x and 6.0.  Sorry, no xml files to accomplish it.  Maybe one day
they will be less of basterds?!?!?!?!?

-Greg

 
> 
> Roy Anciso 
> 
> Director of Technology
> 
> Manistee Intermediate School District
> 
> 1710 Merkey Road
> 
> Manistee, MI 49660
> 
> Ph: 231-723-4264
> 
> Fx: 231-723-1690
> 
> [EMAIL PROTECTED]
> 
>  
> 
> 
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[asterisk-users] Problem with AGI Script

[Matt]

I have asterisk 1.2.18 running on a new system we just installed.   Although
I've used AGIs many times in the past, I'm stumped on this one.  It may just
be a simple issue that I need another eyeset to look at.

My AGI does the following:
#!/usr/bin/perl

#Load a few modules...
use Asterisk::AGI;
use DBI;

$AGI = new Asterisk::AGI;

#Grab input from Asterisk
my %input = $AGI->ReadParse();


#Some Debugging
$AGI->exec('SayDigits',$ARGV[0]);
exit;

All seems fine.  If I run the script from the command line it works as
expected:
[EMAIL PROTECTED] agi-bin]# ./GetEmailFromDID.agi 333
EXEC SayDigits "333"

However, when actually running in practice I get:
   -- Executing AGI("Zap/23-1", "GetEmailfromDID.agi|5706016716") in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/GetEmailfromDID.agi
-- AGI Script GetEmailfromDID.agi completed, returning 0

extensions.conf
[macro-faxreceive]
exten => s,1,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten => s,2,agi(GetEmailfromDID.agi|${CALLERID(number)})
exten => s,3,rxfax(${FAXFILE})
exten => s,104,Set([EMAIL PROTECTED])
exten => s,105,Goto(3)


Any thoughts on why asterisk doesn't seem to be passing anything to the
script and the script doesn't seem to be passing anything back?  When I call
I do not hear the digits read to me, instead I just get thrown to the next
object after the digit reading.
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Re: [asterisk-users] Problem with AGI Script

[Brett Crapser]

On Wed, 14 Nov 2007, Matt wrote:
> I have asterisk 1.2.18 running on a new system we just installed.
> Although I've used AGIs many times in the past, I'm stumped on this one.
> All seems fine.  If I run the script from the command line it works as
> expected:
> However, when actually running in practice I get:
>   -- Executing AGI("Zap/23-1", "GetEmailfromDID.agi|5706016716") in new
> stack
>-- Launched AGI Script /var/lib/asterisk/agi-bin/GetEmailfromDID.agi
>-- AGI Script GetEmailfromDID.agi completed, returning 0
> 
> Any thoughts on why asterisk doesn't seem to be passing anything to the
> script and the script doesn't seem to be passing anything back?

Permissions?

Brett

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Re: [asterisk-users] Nortel digital FXO channel bank? Exists?

[Michelle Dupuis]

I looks like this gateway is the FXS - it allows Meridian handsets to talk
to a SIP pbx. 
I need a an FXO gateway - which allows Meridian PBX digital lines (not
trunks) to talk to a PC/SIP/etc.

MD

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Rupert Utteridge - Digital Techniques (Asia) Limited
> Sent: Wednesday, November 14, 2007 7:27 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Nortel digital FXO channel bank? Exists?
> 
> Citel SIP handset gateways at www.dtasia.net
> 
> Rupert Utteridge
> Director - Sales & Marketing
> Digital Techniques (Asia) Limited
> 4 The Lee
> Middle Cove, NSW, 2068
> Australia
> 
> Tel:  +61 2 9037 4191
> Mobile:  +61 424 373 516
> 
> Web:  www.dtasia.net
> 
> Message: 1
> Date: Tue, 13 Nov 2007 23:00:54 -0500
> From: Jon Pounder <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Nortel digital FXO channel bank?
> Exists?
> To: asterisk-users@lists.digium.com
> Message-ID:
> <[EMAIL PROTECTED]>
> Content-Type: text/plain;   charset=ISO-8859-1; DelSp="Yes";
> format="flowed"
> 
> Quoting Michelle Dupuis <[EMAIL PROTECTED]>:
> 
> > We have a client with a Nortel PBX with digital phone sets. 
>  Due to T1 
> > problems (old firmware), we are interested in trying a FXO 
> channel bank.
> >
> > Is there a channel bank (or equivalent) which emulates Meridian 
> > digital phone sets?  In order words, an FXO channel bank 
> that's Meridian digital?
> 
> I think the basic wireline signalling is isdn bri for that, 
> but with non-standard protocols.
> 
> ie the channel bank would talk to the ksu/pbx, but nothing 
> but a reverse of the same hardware would understand anything 
> on the other end of the t1.
> 
> ps - this sort of channel bank would be pricey at best, 
> rediculous probably in reality.
> 
> I would try other solutions before even attempting this sort 
> of thing since the odds of success are probably not too high.
> 
> 
> 
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[asterisk-users] Can I connect device on FXS of Sipura 3000 to internet virtually ? - it can only call ISPs numbers on POTS line

[Robert Rozman]

Hi,

I have an older phone with touch screen from Philips. It have ti connected 
to Sipura 3000 FXS port and majority of features work ok.

But phone also has touchscreen and web browser that I'd love to use for 
accessing my local web pages. But the phone only allows me to setup ISP 
phone number and it wants to call it to get to Internet. Since it is 
connected to Sipura3000, call can come to Asterisk and I'd love to somehow 
fool that device and connect it to local web pages ? I guess I could somehow 
mimic ISP "internet calling" feature on local Asterisk server, but have no 
clue even where to start searching ...

Any advice ?

Thanks in advance,

regards,

Rob.


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Re: [asterisk-users] Nortel digital FXO channel bank? Exists?

[Jon Pounder]

Quoting Michelle Dupuis <[EMAIL PROTECTED]>:


there were the meridian ATA boxes that plug into a digital extension  
port and provide a standard analog extension - you would probably be  
better off with several of those - that is what they were actually  
intended for (analog fax, voicemail servers, etc)

I think I even have 2 tucked away someplace if you are interested in  
making an offer on them.




> I looks like this gateway is the FXS - it allows Meridian handsets to talk
> to a SIP pbx.
> I need a an FXO gateway - which allows Meridian PBX digital lines (not
> trunks) to talk to a PC/SIP/etc.
>
> MD
>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of
>> Rupert Utteridge - Digital Techniques (Asia) Limited
>> Sent: Wednesday, November 14, 2007 7:27 AM
>> To: asterisk-users@lists.digium.com
>> Subject: Re: [asterisk-users] Nortel digital FXO channel bank? Exists?
>>
>> Citel SIP handset gateways at www.dtasia.net
>>
>> Rupert Utteridge
>> Director - Sales & Marketing
>> Digital Techniques (Asia) Limited
>> 4 The Lee
>> Middle Cove, NSW, 2068
>> Australia
>>
>> Tel:  +61 2 9037 4191
>> Mobile:  +61 424 373 516
>>
>> Web:  www.dtasia.net
>>
>> Message: 1
>> Date: Tue, 13 Nov 2007 23:00:54 -0500
>> From: Jon Pounder <[EMAIL PROTECTED]>
>> Subject: Re: [asterisk-users] Nortel digital FXO channel bank?
>> Exists?
>> To: asterisk-users@lists.digium.com
>> Message-ID:
>> <[EMAIL PROTECTED]>
>> Content-Type: text/plain;   charset=ISO-8859-1; DelSp="Yes";
>> format="flowed"
>>
>> Quoting Michelle Dupuis <[EMAIL PROTECTED]>:
>>
>> > We have a client with a Nortel PBX with digital phone sets.
>>  Due to T1
>> > problems (old firmware), we are interested in trying a FXO
>> channel bank.
>> >
>> > Is there a channel bank (or equivalent) which emulates Meridian
>> > digital phone sets?  In order words, an FXO channel bank
>> that's Meridian digital?
>>
>> I think the basic wireline signalling is isdn bri for that,
>> but with non-standard protocols.
>>
>> ie the channel bank would talk to the ksu/pbx, but nothing
>> but a reverse of the same hardware would understand anything
>> on the other end of the t1.
>>
>> ps - this sort of channel bank would be pricey at best,
>> rediculous probably in reality.
>>
>> I would try other solutions before even attempting this sort
>> of thing since the odds of success are probably not too high.
>>
>>
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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Jon Pounder

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_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
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Re: [asterisk-users] Problem with AGI Script

[Matt]

-rwxrw-r--  1 asterisk asterisk  1053 Nov 14 08:54 GetEmailFromDID.agi

asterisk  3348  0.0  0.2 27024 9380 ?Sl   Oct01   0:01
/usr/sbin/asterisk -U asterisk -G asterisk -v -g -p -U asterisk -G asterisk


On Nov 14, 2007 9:17 AM, Brett Crapser <[EMAIL PROTECTED]> wrote:

>
> On Wed, 14 Nov 2007, Matt wrote:
> > I have asterisk 1.2.18 running on a new system we just installed.
> > Although I've used AGIs many times in the past, I'm stumped on this one.
> > All seems fine.  If I run the script from the command line it works as
> > expected:
> > However, when actually running in practice I get:
> >   -- Executing AGI("Zap/23-1", "GetEmailfromDID.agi|5706016716") in new
> > stack
> >-- Launched AGI Script /var/lib/asterisk/agi-bin/GetEmailfromDID.agi
> >-- AGI Script GetEmailfromDID.agi completed, returning 0
> > 
> > Any thoughts on why asterisk doesn't seem to be passing anything to the
> > script and the script doesn't seem to be passing anything back?
>
> Permissions?
>
> Brett
>
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Re: [asterisk-users] Using php exec() in agi script

[Moises Silva]

No.

AGI script is just like any other PHP script, the only difference is
that STDOUT/IN are connected with the Asterisk thread that launched
the script.

You should run the script directly and see what the problem is.

- Moy

On Nov 14, 2007 6:51 AM, Andre Courchesne <[EMAIL PROTECTED]> wrote:
> Hi,
>
>   Any reason why I can not get the php exec() function to execute a shell 
> command inside an agi script?
>
>   Thanks.
>
> Andre
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-- 
"Within C++, there is a much smaller and cleaner language struggling
to get out."

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Re: [asterisk-users] route INVITE sip:[EMAIL PROTECTED]

[Marc LEURENT]

We are using 2 different incoming trunks.
The first one is alsion.com and is sending INVITE with phone number in
the INVITE line whereas plugandtel put the callee number only inside the
To: Section.



Marco Mouta a écrit :
> Could you describe in detail how did you fall into this situation, I mean
> the real example which SIP phone sends this invite? Is registered in
> asterisk? it is a non-registered sip phone trying to dial a sip user at your
> * box?
> 
> If this is an issue with a specific hardware outside of your asterisk,  may
> be something not well configured ... describe it a bit more in detail.
> 
> If you don't have anyworkaround for this Invite format I would use OpenSER
> in front of Asterisk to handle this invites and replace to SIP URI with info
> from the tag TO: ...
> 
> Any way if you provide more details may be someone in the Mailing list is
> able to help u out;)
> 
> Best regards
> MoutaPT
> 
> On Nov 13, 2007 6:14 PM, Marc LEURENT <[EMAIL PROTECTED]> wrote:
> 
>> Good evening!
>> I was wondering one thing,
>> I'm using freepbx to configure my asterisk server and I have a problem
>> with some inbound calls.
>>
>> When I receive a call to an INVITE sip:[EMAIL PROTECTED] I an set an
>> inbound route! It matches a DID number.
>>
>> How can I route an INVITE sip:[EMAIL PROTECTED] The number only appear in the
>> To: Section.
>>
>> Thanks!
>>
>> Example:
>>
>> With this one, I cannot route it (there is only the number to be reached
>> in the To: section)
>> #
>> U 217.36.112.145:5060 -> 192.168.95.235:5060
>> INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0.
>> Allow: UPDATE,REFER,INFO.
>> Call-ID: [EMAIL PROTECTED]
>> Contact: .
>> Content-Type: application/sdp.
>> CSeq: 34878212 INVITE.
>> From: "0614740696"
>> ;tag=02975-US-0223ae6e-67d6c4495.
>> Max-Forwards: 31.
>> To: .
>> User-Agent: Cirpack/v4.41c (gw_sip).
>> Via: SIP/2.0/UDP 217.36.112.145:5060;branch=z9hG4bK-744D-33B812.
>> Content-Length: 303.
>> .
>>
>>
>>
>> Whereas with this one I can do it! (there is a number in the INVITE)
>> #
>> U 87.98.202.114:5060 -> 192.168.95.235:5060
>> INVITE sip:[EMAIL PROTECTED] SIP/2.0.
>> Via: SIP/2.0/UDP 87.98.202.114:5060;branch=z9hG4bK1fd2c6b4;rport.
>> From: "0158136741" ;tag=as25391ca7.
>> To: .
>> Contact: .
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 102 INVITE.
>> User-Agent: Asterisk PBX.
>> Max-Forwards: 70.
>> Date: Tue, 13 Nov 2007 18:07:00 GMT.
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
>> Content-Type: application/sdp.
>> Content-Length: 233.
>> .
>>
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>>
> 
> 
> 
> 
> 
> 
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[asterisk-users] IVR Tree Best Practices

[Matt]

Does anyone have any solid documented evidence for best practices for IVR
Trees?  I'm just curious.  I did some google searching this morning, but
only found one article from TMC.What I'm looking for are studies
showing, if a customer must go to an IVR, what they prefer, over what they
prefer to not have in the trees.
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Re: [asterisk-users] Problem with AGI Script

[Moises Silva]

what does agi debug says?
what if you run the script from the command line and you fake the
asterisk input?

Regards,

On Nov 14, 2007 8:33 AM, Matt <[EMAIL PROTECTED]> wrote:
> -rwxrw-r--  1 asterisk asterisk  1053 Nov 14 08:54 GetEmailFromDID.agi
>
> asterisk  3348  0.0  0.2 27024 9380 ?Sl   Oct01   0:01
> /usr/sbin/asterisk -U asterisk -G asterisk -v -g -p -U asterisk -G asterisk
>
>
>
>
> On Nov 14, 2007 9:17 AM, Brett Crapser <[EMAIL PROTECTED]> wrote:
> >
> >
> > On Wed, 14 Nov 2007, Matt wrote:
> > > I have asterisk 1.2.18 running on a new system we just installed.
> > > Although I've used AGIs many times in the past, I'm stumped on this one.
> >
> > > All seems fine.  If I run the script from the command line it works as
> > > expected:
> >
> > > However, when actually running in practice I get:
> > >   -- Executing AGI("Zap/23-1", " GetEmailfromDID.agi|5706016716") in new
> > > stack
> > >-- Launched AGI Script /var/lib/asterisk/agi-bin/GetEmailfromDID.agi
> > >-- AGI Script GetEmailfromDID.agi completed, returning 0
> > > 
> >
> > > Any thoughts on why asterisk doesn't seem to be passing anything to the
> > > script and the script doesn't seem to be passing anything back?
> >
> > Permissions?
> >
> > Brett
> >
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> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
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-- 
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[asterisk-users] "Whats New at Digium the Asterisk Company" -- Junk?

[Philipp Kempgen]

Is the "Whats New at Digium the Asterisk Company" message I got from
[EMAIL PROTECTED] really from Digium?
If so I suggest to send it from digium.com and not to use those
shady Eloqua redirect URLs.

Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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Re: [asterisk-users] Problem with AGI Script

[Mindaugas Kezys]

Make sure /usr/bin/perl can be reached.

 

Also try in your CLI:

 

agi debug

 

Same case happens when I do not have php-cli installed for php AGI scripts.

 

Mindaugas Kezys

http://www.kolmisoft.com

MOR - Advanced Billing for Asterisk PBX

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Wednesday, November 14, 2007 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Problem with AGI Script

 

I have asterisk 1.2.18 running on a new system we just installed.   Although
I've used AGIs many times in the past, I'm stumped on this one.  It may just
be a simple issue that I need another eyeset to look at.

My AGI does the following:
#!/usr/bin/perl

#Load a few modules...
use Asterisk::AGI;
use DBI;

$AGI = new Asterisk::AGI;

#Grab input from Asterisk
my %input = $AGI->ReadParse();


#Some Debugging
$AGI->exec('SayDigits',$ARGV[0]);
exit;

All seems fine.  If I run the script from the command line it works as
expected:
[EMAIL PROTECTED] agi-bin]# ./GetEmailFromDID.agi 333 
EXEC SayDigits "333"

However, when actually running in practice I get:
   -- Executing AGI("Zap/23-1", "GetEmailfromDID.agi|5706016716") in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/GetEmailfromDID.agi 
-- AGI Script GetEmailfromDID.agi completed, returning 0

extensions.conf
[macro-faxreceive]
exten => s,1,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten => s,2,agi(GetEmailfromDID.agi|${CALLERID (number)})
exten => s,3,rxfax(${FAXFILE})
exten => s,104,Set([EMAIL PROTECTED])
exten => s,105,Goto(3)


Any thoughts on why asterisk doesn't seem to be passing anything to the
script and the script doesn't seem to be passing anything back?  When I call
I do not hear the digits read to me, instead I just get thrown to the next
object after the digit reading. 

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Re: [asterisk-users] Problem with AGI Script

[Matt]

[EMAIL PROTECTED] agi-bin]# /usr/bin/perl -v

This is perl, v5.8.5 built for i386-linux-thread-multi


Debug shows nothing:
-- Launched AGI Script /var/lib/asterisk/agi-bin/GetEmailfromDID.agi
AGI Tx >> agi_request: GetEmailfromDID.agi
AGI Tx >> agi_channel: Zap/23-1
AGI Tx >> agi_language: en
AGI Tx >> agi_type: Zap
AGI Tx >> agi_uniqueid: 1195061174.4
AGI Tx >> agi_callerid: 5706016716
AGI Tx >> agi_calleridname: Test Networks
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 33
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: 5706010280
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: macro-faxreceive
AGI Tx >> agi_extension: s
AGI Tx >> agi_priority: 2
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
-- AGI Script GetEmailfromDID.agi completed, returning 0

Just returned with a 0 and doesn't do anything it is suppose to do.  I'm
kind of at a loss.


On Nov 14, 2007 11:40 AM, Mindaugas Kezys <[EMAIL PROTECTED]> wrote:

>  Make sure /usr/bin/perl can be reached.
>
>
>
> Also try in your CLI:
>
>
>
> agi debug
>
>
>
> Same case happens when I do not have php-cli installed for php AGI
> scripts.
>
>
>
> Mindaugas Kezys
>
> http://www.kolmisoft.com
>
> MOR – Advanced Billing for Asterisk PBX
>
>
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Matt
> *Sent:* Wednesday, November 14, 2007 4:00 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Problem with AGI Script
>
>
>
> I have asterisk 1.2.18 running on a new system we just installed.
> Although I've used AGIs many times in the past, I'm stumped on this one.  It
> may just be a simple issue that I need another eyeset to look at.
>
> My AGI does the following:
> #!/usr/bin/perl
>
> #Load a few modules...
> use Asterisk::AGI;
> use DBI;
>
> $AGI = new Asterisk::AGI;
>
> #Grab input from Asterisk
> my %input = $AGI->ReadParse();
>
>
> #Some Debugging
> $AGI->exec('SayDigits',$ARGV[0]);
> exit;
> 
> All seems fine.  If I run the script from the command line it works as
> expected:
> [EMAIL PROTECTED] agi-bin]# ./GetEmailFromDID.agi 333
> EXEC SayDigits "333"
>
> However, when actually running in practice I get:
>-- Executing AGI("Zap/23-1", "GetEmailfromDID.agi|5706016716") in new
> stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/GetEmailfromDID.agi
> -- AGI Script GetEmailfromDID.agi completed, returning 0
> 
> extensions.conf
> [macro-faxreceive]
> exten => s,1,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
> exten => s,2,agi(GetEmailfromDID.agi|${CALLERID (number)})
> exten => s,3,rxfax(${FAXFILE})
> exten => s,104,Set([EMAIL PROTECTED])
> exten => s,105,Goto(3)
>
>
> Any thoughts on why asterisk doesn't seem to be passing anything to the
> script and the script doesn't seem to be passing anything back?  When I call
> I do not hear the digits read to me, instead I just get thrown to the next
> object after the digit reading.
>
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Re: [asterisk-users] "Whats New at Digium the Asterisk Company" -- Junk?

[Jesse Molina]

I received this message as well.  I considered it spam, as I¹ve not
voluntarily signed up to receive mailings from Digium.



On 11/14/07 8:47 AM, "Philipp Kempgen" <[EMAIL PROTECTED]> wrote:

> Is the "Whats New at Digium the Asterisk Company" message I got from
> [EMAIL PROTECTED] really from Digium?
> If so I suggest to send it from digium.com and not to use those
> shady Eloqua redirect URLs.
> 
> Regards,
>   Philipp Kempgen
> 
> --
> amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
> Let's use IT to solve problems and not to create new ones.
>   Asterisk? -> http://www.das-asterisk-buch.de
> 
> Geschäftsführer: Stefan Wintermeyer
> Handelsregister: Neuwied B 14998
> 
> ___
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>http://lists.digium.com/mailman/listinfo/asterisk-users
> 


-- 
# Jesse Molina
# The Translational Genomics Research Institute
# http://www.tgen.org
# Mail = [EMAIL PROTECTED]
# Desk = 1.602.343.8459
# Cell = 1.602.323.7608
 


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[asterisk-users] PBX Testing Framework

[IQ Labs VoIP Team]

IQ Labs announces the release of PBX Testing Framework.

This software is intended to test existing call-center PBX, and is
distributed under GPL license.

Currently it allows SIP testing, but implementing IAX (and even Zap)
shouldn't be a problem, as the framework is based on Asterisk, and can
do anything the Asterisk does.

Please see README file included for configuration and scripting samples.

You can download it from:
http://ftp.iq-labs.net/pbx-test/pbx-test-0.1.0.tar.gz

Please give us a short feedback if you're successfully using it.

For any questions and improvements you may contact [EMAIL PROTECTED]

Regards,
Atis Lezdins
VoIP Developer
IQ Labs Inc.

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[asterisk-users] Help in getting a dialplan to produce the right CDR info

[Robert Moskowitz]

I have been shaking down a dialplan for SIP fax to efax.

The basic senario is an ATA on the same subnet as the Asterisk 1.2 box 
(avoid RTP packet lose and thus fax crash), calling a 'fax extension' 
and envoking rxfax then email.

I leverage off of context: from-internal-additional-custom, so as not to 
have it overwritten by FreePBX.

In extension_custom.conf I have:

[from-internal-additional-custom]

include => ext-local-fax-custom

#include extensions_fax.conf

[custom-fax1]
exten => s,1,Answer
exten => s,n,StopPlayTones
exten => s,n,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten => s,n,rxfax(${FAXFILE})
exten => s,n,Hangup
exten => h,1,system(/var/lib/asterisk/bin/fax-process.pl --to 
${FAX_TO_EMAIL} --from ${FAX_RX_FROM} --subject "Fax from 
${URIENCODE(${CALLERID(number)})} ${URIENCODE(${CALLERID(name)})}" 
--attachment fax_${URIENCODE(${CALLERID(number)})}.pdf --type 
application/pdf --file ${FAXFILE});
exten => h,2,Hangup()


And extensions_fax.conf has:

exten => 29809,1,Set([EMAIL PROTECTED])
exten => 29809,n,Macro(just-user-callerid,)
exten => 29809,n,Goto(custom-fax1,s,1)

[macro-just-user-callerid]
exten => s,1,Set(AMPUSER=${DB(DEVICE/${CALLERID(number)}/user)})
exten => s,n,Set(AMPUSERCIDNAME=${DB(AMPUSER/${AMPUSER}/cidname)})
exten => s,n,Set(CALLERID(all)=${AMPUSERCIDNAME} <${AMPUSER}>)

=

I had to add the just-user-callerid macro to pick up the callerid name.

But the goto custom-fax1 results in the destination extension for the 
CDR record to be 's'.

If I make custom-fax1 a macro instead, I get the proper value for the 
CDR record, but the Hangup logic does not work.

I suspect that before executing rxfax, I could SET a variable with the 
destination extension, and then in the Hangup logic push it back in, but 
I need some help.

Or a better way.

And yes, I know that Asterisk 1.4 supports T.38, and I have every 
intension to get things set with 1.4


Thank you.

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[asterisk-users] Error: inserting return line in dialing strings

[Alejandro Lengua]

My calls provider has suspended my account, because he says that I am
send bad formating call strings.
According to the email he sent me a line return is beign inserted
after the number.
One string he sent me is the following:
"BADCALL","101339","0115712550727
","reseller","""cesar  reategui"" <101339>","IAX2/101339-496","", ...

Any idea on how to solve the issue?, what can cause it?
I am using Trixbox 2.0 and Zopier softphone for placing the calls ...

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Re: [asterisk-users] route INVITE sip:[EMAIL PROTECTED]

[Marco Mouta]

Hi,

I would suggest you to use Asterisk Application SIPGetHeader in your
Dialplan for incoming calls from "plugandtel".

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPGetHeader

Something like
*
exten=>_[a-z].,1,SIPGetHeader(Var_TO=To)
exten=>_[a-z].,2,Dial(SIP/${Var_TO})

Please be aware that this application as a different name in asterisk 1.4.

Learn from CLI > show application ?

Hope it helps,

Best regards,
MoutaPT


*
On Nov 14, 2007 3:02 PM, Marc LEURENT <[EMAIL PROTECTED]> wrote:

> We are using 2 different incoming trunks.
> The first one is alsion.com and is sending INVITE with phone number in
> the INVITE line whereas plugandtel put the callee number only inside the
> To: Section.
>
>
>
> Marco Mouta a écrit :
> > Could you describe in detail how did you fall into this situation, I
> mean
> > the real example which SIP phone sends this invite? Is registered in
> > asterisk? it is a non-registered sip phone trying to dial a sip user at
> your
> > * box?
> >
> > If this is an issue with a specific hardware outside of your asterisk,
>  may
> > be something not well configured ... describe it a bit more in detail.
> >
> > If you don't have anyworkaround for this Invite format I would use
> OpenSER
> > in front of Asterisk to handle this invites and replace to SIP URI with
> info
> > from the tag TO: ...
> >
> > Any way if you provide more details may be someone in the Mailing list
> is
> > able to help u out;)
> >
> > Best regards
> > MoutaPT
> >
> > On Nov 13, 2007 6:14 PM, Marc LEURENT <[EMAIL PROTECTED]> wrote:
> >
> >> Good evening!
> >> I was wondering one thing,
> >> I'm using freepbx to configure my asterisk server and I have a problem
> >> with some inbound calls.
> >>
> >> When I receive a call to an INVITE sip:[EMAIL PROTECTED] I an set an
> >> inbound route! It matches a DID number.
> >>
> >> How can I route an INVITE sip:[EMAIL PROTECTED] The number only appear in 
> >> the
> >> To: Section.
> >>
> >> Thanks!
> >>
> >> Example:
> >>
> >> With this one, I cannot route it (there is only the number to be
> reached
> >> in the To: section)
> >> #
> >> U 217.36.112.145:5060 -> 192.168.95.235:5060
> >> INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0.
> >> Allow: UPDATE,REFER,INFO.
> >> Call-ID: [EMAIL PROTECTED]
> >> Contact: .
> >> Content-Type: application/sdp.
> >> CSeq: 34878212 INVITE.
> >> From: "0614740696"
> >>  ;user=phone>;tag=02975-US-0223ae6e-67d6c4495.
> >> Max-Forwards: 31.
> >> To: .
> >> User-Agent: Cirpack/v4.41c (gw_sip).
> >> Via: SIP/2.0/UDP 217.36.112.145:5060;branch=z9hG4bK-744D-33B812.
> >> Content-Length: 303.
> >> .
> >>
> >>
> >>
> >> Whereas with this one I can do it! (there is a number in the INVITE)
> >> #
> >> U 87.98.202.114:5060 -> 192.168.95.235:5060
> >> INVITE sip:[EMAIL PROTECTED] SIP/2.0.
> >> Via: SIP/2.0/UDP 87.98.202.114:5060;branch=z9hG4bK1fd2c6b4;rport.
> >> From: "0158136741" ;tag=as25391ca7.
> >> To: .
> >> Contact: .
> >> Call-ID: [EMAIL PROTECTED]
> >> CSeq: 102 INVITE.
> >> User-Agent: Asterisk PBX.
> >> Max-Forwards: 70.
> >> Date: Tue, 13 Nov 2007 18:07:00 GMT.
> >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> >> Content-Type: application/sdp.
> >> Content-Length: 233.
> >> .
> >>
> >> ___
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> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >
> >
> >
> > 
> >
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[asterisk-users] Routing Anonymous Callerid

[C. Duncan Hudson]

I had (I thought) a generic inbound route that was to handle all calls - 
dumping them all into my queue after doing a bunch of time logic.  Once 
in the queue all the extensions are rung.  The exception seems to be any 
call that has blocked their callerid - those calls make it to the queue, 
but they're only ringing one extension not the entire ring group like 
normal calls.  What do I need to do to handle anonymous calls like every 
other call?  Thanks,

Dunc

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Re: [asterisk-users] Asterisk ignoring manager events when busy

[Nick Adams]

Doug Lytle wrote:
> Nick Adams wrote:
>> The kicker: I can't seem to get past this 200 call point even though the
>>   
> 
> What does your console show at this time?
> 
> When testing, I've noted the 200 call limit was because I had too many 
> open files.  I had to increase this by typing ulimit -n 4096 before 
> starting Asterisk.  The default is 1024.
> 
> Doug

Thanks for the reply Doug. The console doesn't reveal anything too
suspicious. I do get some "RTP Read too short" but I don't think that
has anything to do with the problem.

I'll adjust the open files limit and see how it goes. Thanks for your
advice.


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[asterisk-users] Real Time CDR

[Tony Plack]

Every once in a while (like 2 out of 7 times), I get the following message:

[Nov 14 12:49:02] NOTICE[6855]: cdr.c:434 ast_cdr_free: CDR on channel 
'SIP/5000-082508f0' not posted

I look in the cdr table in mySQL and indeed, the record is not posted for that 
call.

This makes me want to create hard file and a compare script between the file 
cdr and the odbc cdr, but I was wondering if anyone else has seen this error or 
if my config is off.

mySQL 5
Asterisk branch/1.4 current rev

cdr.conf
[general]
enabled=yes
batch=no
size=100
time=300
scheduleronly=no
endbeforehexten=no
safeshutdown=yes


cdr_manager.conf
[general]
enabled=yes

cdr_odbc.conf
[global]
dsn=asterisk
username=user
password=password
loguniqueid=yes
dispositionstring=yes
table=cdr;"cdr" is default 
table name
;usegmtime=no; set to "yes" to log in GMT

loaded files
cdr_manager.so
cdr_odbc.so
cdr_csv.so

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[asterisk-users] pip tones in Monitor or MixMonitor

[Tony Plack]

Is there a way to enable the pip tones (beep) indicating that a call is being 
recorded?

I know that ChanSpy does beep (unless q option is chosen) once, but not quite 
the same.

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Re: [asterisk-users] pip tones in Monitor or MixMonitor

[Mindaugas Kezys]

exten => _X.,1,Playback(beep)
exten => _X.,2,MixMonitor.

If you are starting the recording using some DTMF code sequence described in
features.conf make sure you use "caller", "callee" or "both" value to play
sound to correct line end.

Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony Plack
Sent: Wednesday, November 14, 2007 11:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] pip tones in Monitor or MixMonitor

Is there a way to enable the pip tones (beep) indicating that a call is
being recorded?

I know that ChanSpy does beep (unless q option is chosen) once, but not
quite the same.

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Re: [asterisk-users] asterisk-stat problem

[Erik Anderson]

On Nov 14, 2007 4:15 PM, Richard Cahilig <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I installed asterisk-addons and asterisk-stats, Its working now except
> of one problem. The problem is there is no call logs when you open the
> cdr report. The message is when you open the cdr report is:  - Call
> Logs -   Back to Top
> No data found !!!
> 1 / 1
> Did I missed something in the configuration of mysql-addons or
> asterisk-stat? Here is my asterisk-stats page:
> http://203.115.187.91/cdr, the username is admin and the password is
> password. Thank you very much.

Richard - just click "search" when you go to one of the report pages.
It doesn't do the query manually.

-erik

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[asterisk-users] asterisk-stat problem

[Richard Cahilig]

Hi,

I installed asterisk-addons and asterisk-stats, Its working now except
of one problem. The problem is there is no call logs when you open the
cdr report. The message is when you open the cdr report is:  - Call
Logs -   Back to Top
No data found !!!
1 / 1
Did I missed something in the configuration of mysql-addons or
asterisk-stat? Here is my asterisk-stats page:
http://203.115.187.91/cdr, the username is admin and the password is
password. Thank you very much.

-- 
Richard R. Cahilig
http://chr05210084.com

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Re: [asterisk-users] asterisk-stat problem

[Anthony Messina]

On Wednesday 14 November 2007 04:15:38 pm Richard Cahilig wrote:
> Hi,
>
> I installed asterisk-addons and asterisk-stats, Its working now except
> of one problem. The problem is there is no call logs when you open the
> cdr report. The message is when you open the cdr report is:  - Call
> Logs - Back to Top
> No data found !!!
> 1 / 1
> Did I missed something in the configuration of mysql-addons or
> asterisk-stat? Here is my asterisk-stats page:
> http://203.115.187.91/cdr, the username is admin and the password is
> password. Thank you very much.

you have to click on the "search" button.  if you want all results, just leave 
all the fields blank.

-- 
Anthony -  http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E


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Re: [asterisk-users] function voicemailmain

[[EMAIL PROTECTED]]

You need some experiance with the ANSI C programming language. Once
you have acquired that the rest is pretty straightforward.

On Nov 14, 2007 2:21 AM, Rilawich Ango <[EMAIL PROTECTED]> wrote:
> You mean modify the source?  Could you give me an example, say I wrong
> to remove advance option?
>
>
> On Nov 14, 2007 1:59 PM, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> > vi app_voicemail.c
> >
> >
> >
> > On Nov 13, 2007 10:34 PM, Rilawich Ango <[EMAIL PROTECTED]> wrote:
> > > Hi all,
> > >
> > >   Can I simply the voicemailmain IVR?  I just only want some of the
> > > option in voicemailmain, ie read or delete messages.  Is it possible
> > > to configure that function?
> > >
> > > Ango
> > >
> >
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> >
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>
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Re: [asterisk-users] Linksys 942 Call Transfer

[[EMAIL PROTECTED]]

did you try

canreinvite=no

in your sip.conf file

It would also help to:
1) Post the relevant configuration files (phone AND Asterisk)
2) Post the EXACT message from column 1 to EOL
3) What version of Asterisk? Stock? From a certain distribution? Patches?

Or I could just say "There is a problem with your configuration,
transfer of calls from an SPA-phone works fine for me." (it really
does!)

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Re: [asterisk-users] function voicemailmain

[Baji Panchumarti]

  On Nov 14, 2007 6:31 PM, joakimsen  wrote:

> You need some experiance with the ANSI C programming language.
> Once you have acquired that the rest is pretty straightforward.

http://www.amazon.com/C-Programming-Language-2nd-Ed/dp/0131103709/

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[asterisk-users] asterisk integration with panasonic analog pbx

[jorain]

Hi all, 

I have an existing panasonic analog pbx in use and a asterisk server with 
digium tdm400p(2 fxs and 2 fxo). 

channel 1 -> fxs -> telephone 
channel 2 -> fxs -> telephone 
channel 3 -> fxo -> extension 15 at panasonic pbx 
channel 4 -> fxo -> phone line from telco 

We call in to fxo (channel 4) and enter the ivr which prompt us to enter the 
extension number. After that asterisk will dialout from fxo(channel 3) to other 
extensions at panasonic pbx(eg ext 16,17,18 etc) 

The problem is when the extension 16 at panasonic pbx answers the call, 
asterisk can only detect the ANSWER after few seconds(range from 2 to 20 
seconds). 

If the extension 16 is busy, caller gets a busy tone (approximately 10 seconds) 
before asterisk detects the BUSY status and run the following commands in 
dialplan. 

I have tried 
- to call from telephone at fxs(channel 1) to pstn number using channel 4, it 
can detect the status immediately. 
- testing the outcome of settings in zapata.conf(busydetect, busycount, 
answeronpolarity, hanguponpolarity) 


Another question, is there any asterisk utility which can detect the tone on a 
channel so that we can get its pattern? 



Regards,
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Re: [asterisk-users] function voicemailmain

[Tzafrir Cohen]

On Wed, Nov 14, 2007 at 11:34:31AM +0800, Rilawich Ango wrote:
> Hi all,
> 
>   Can I simply the voicemailmain IVR?  I just only want some of the
> option in voicemailmain, ie read or delete messages.  Is it possible
> to configure that function?

What about minivm?

-- 
   Tzafrir Cohen   
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] asterisk-stat problem

[Paul Hales]

All looks fine to me - and hopefully nobody does anything nasty to your
server.

PaulH


On Thu, 2007-11-15 at 06:15 +0800, Richard Cahilig wrote:
> Hi,
> 
> I installed asterisk-addons and asterisk-stats, Its working now except
> of one problem. The problem is there is no call logs when you open the
> cdr report. The message is when you open the cdr report is:  - Call
> Logs - Back to Top
> No data found !!!
> 1 / 1
> Did I missed something in the configuration of mysql-addons or
> asterisk-stat? Here is my asterisk-stats page:
> http://203.115.187.91/cdr, the username is admin and the password is
> password. Thank you very much.
> 


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[asterisk-users] Integration of Asterisk with MS Dynamics CRM

[Robert Roach]

Have a request from a customer to integrate Asterisk into an MS CRM
environment.  From quick research it seems the way to do this is through
a CTI interface and it requires 3rd party middleware.

I have zero experience with MS CRM, so looking for some tips on if/how
this is possible, specifically what MS CRM is capable of.  I think it
should be possible with a basic TAPI exchange and no need for CTI.

Thanks!


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Re: [asterisk-users] Problem with AGI Script

[Benjamin Jacob]

well.. if nothings working.. try putting in debug lines urself in the 
code.. say
use system calls to write some debugging data into some temporary file 
in ur perl code.

let us know..

Matt wrote:

> [EMAIL PROTECTED] agi-bin]# /usr/bin/perl -v
>
> This is perl, v5.8.5 built for i386-linux-thread-multi
>
>
> Debug shows nothing:
> -- Launched AGI Script /var/lib/asterisk/agi-bin/GetEmailfromDID.agi
> AGI Tx >> agi_request: GetEmailfromDID.agi
> AGI Tx >> agi_channel: Zap/23-1
> AGI Tx >> agi_language: en
> AGI Tx >> agi_type: Zap
> AGI Tx >> agi_uniqueid: 1195061174.4
> AGI Tx >> agi_callerid: 5706016716
> AGI Tx >> agi_calleridname: Test Networks
> AGI Tx >> agi_callingpres: 0
> AGI Tx >> agi_callingani2: 0
> AGI Tx >> agi_callington: 33
> AGI Tx >> agi_callingtns: 0
> AGI Tx >> agi_dnid: 5706010280
> AGI Tx >> agi_rdnis: unknown
> AGI Tx >> agi_context: macro-faxreceive
> AGI Tx >> agi_extension: s
> AGI Tx >> agi_priority: 2
> AGI Tx >> agi_enhanced: 0.0
> AGI Tx >> agi_accountcode:
> AGI Tx >>
> -- AGI Script GetEmailfromDID.agi completed, returning 0
>
> Just returned with a 0 and doesn't do anything it is suppose to do. 
> I'm kind of at a loss.
>
>
> On Nov 14, 2007 11:40 AM, Mindaugas Kezys <[EMAIL PROTECTED] 
> > wrote:
>
> Make sure /usr/bin/perl can be reached.
>
> Also try in your CLI:
>
> agi debug
>
> Same case happens when I do not have php-cli installed for php AGI
> scripts.
>
> Mindaugas Kezys
>
> http://www.kolmisoft.com
>
> MOR – Advanced Billing for Asterisk PBX
>
> *From:* [EMAIL PROTECTED]
> 
> [mailto:[EMAIL PROTECTED]
> ] *On Behalf Of *Matt
> *Sent:* Wednesday, November 14, 2007 4:00 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Problem with AGI Script
>
> I have asterisk 1.2.18 running on a new system we just installed.
> Although I've used AGIs many times in the past, I'm stumped on
> this one. It may just be a simple issue that I need another eyeset
> to look at.
>
> My AGI does the following:
> #!/usr/bin/perl
>
> #Load a few modules...
> use Asterisk::AGI;
> use DBI;
>
> $AGI = new Asterisk::AGI;
>
> #Grab input from Asterisk
> my %input = $AGI->ReadParse();
>
>
> #Some Debugging
> $AGI->exec('SayDigits',$ARGV[0]);
> exit;
> 
> All seems fine. If I run the script from the command line it works
> as expected:
> [EMAIL PROTECTED] agi-bin]# ./GetEmailFromDID.agi 333
> EXEC SayDigits "333"
>
> However, when actually running in practice I get:
> -- Executing AGI("Zap/23-1", "GetEmailfromDID.agi|5706016716") in
> new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/GetEmailfromDID.agi
> -- AGI Script GetEmailfromDID.agi completed, returning 0
> 
> extensions.conf
> [macro-faxreceive]
> exten => s,1,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
> exten => s,2,agi(GetEmailfromDID.agi|${CALLERID (number)})
> exten => s,3,rxfax(${FAXFILE})
> exten => s,104,Set([EMAIL PROTECTED]
> )
> exten => s,105,Goto(3)
>
>
> Any thoughts on why asterisk doesn't seem to be passing anything
> to the script and the script doesn't seem to be passing anything
> back? When I call I do not hear the digits read to me, instead I
> just get thrown to the next object after the digit reading.
>
>
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Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files

[Patrick]

On Wed, 2007-11-14 at 09:06 -0500, Anciso, Roy wrote:
> The Cisco Documentation states that you can modify standard and
> nonstandard softkey templates.  They may not be xml files. I just
> assumed they were xml since that is what is used to configure the phone.

Just bumped into some info about this:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_programming_usage_guide_chapter09186a00807a35b9.html#wp1040919


Hope this helps.

Regards,
Patrick


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Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files

[Olivier]

2007/11/14, Greg Oliver <[EMAIL PROTECTED]>:
>
> On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote:
> > Hello List,
> >
> > Does anyone have access to the soft key configuration files for the
> > Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and
> > didn't find much up there.
> >
> > Thanks
> >
>
> Softkeys running both SCCP and SIP firmware are both sent through the
> protocols themselves.


How ?
In SIP mode, is it using RegEvents (rfc3680) ?

regards

  I have done packet captures to prove it out from
> CCM 5.x and 6.0.  Sorry, no xml files to accomplish it.  Maybe one day
> they will be less of basterds?!?!?!?!?
>
> -Greg
>
>
> >
> > Roy Anciso
> >
> > Director of Technology
> >
> > Manistee Intermediate School District
> >
> > 1710 Merkey Road
> >
> > Manistee, MI 49660
> >
> > Ph: 231-723-4264
> >
> > Fx: 231-723-1690
> >
> > [EMAIL PROTECTED]
> >
> >
> >
> >
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[asterisk-users] Queue

[Bhrugu Mehta]

Hi all
I want to create Ivrs using dialplan and aslo want to transfer call to
agent using Queue app in asterisk.
Is there any way to get IP ADDRESS of free agent  which is found by asterisk
thnks ,
Bhrugu mehta

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