Re: [asterisk-users] FTC Bans Prerecorded Telemarketing Drivel

[Alex Balashov]

I would be curious to know where, in this classification, fall various 
telemarketing schemes that are technically not cold-calls, but are 
generated from leads that come from customer-provided information, but 
where the customer does not know explicitly that they are signing up to 
receive calls.

For instance, this is common in a number of industries such as financial 
services.  You do a search to get a quote on something, and provide your 
phone number in the process, although the phone number bears no relation 
to the submission and is just an ancillary required item.  Several 
places' telemarketing organisations call you back in response.  For 
example, lendingtree.com.

Is this a solicited call?

Michael Collins wrote:

 Gives us legitimate telemarketers a bad damn name.  :-)
 
 Isn't legitimate telemarketers an oxymoron?

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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[asterisk-users] DSS1 vs SS7

[mark morreny]

Hi,

I am requesting for a E1 connection from my telco.  They are asking if I
want DSS1 or SS7, and I am stuck here.  Could someone tell me the difference
between the two?  How should I decide which one to use?

Thanks in advance for your help.

Mark
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Re: [asterisk-users] DSS1 vs SS7

[Alex Balashov]

Use DSS1.  It's European ISDN and would give you the equivalent of a 
North American PRI.

You don't want SS7.

mark morreny wrote:

 Hi,
 
 I am requesting for a E1 connection from my telco.  They are asking if I 
 want DSS1 or SS7, and I am stuck here.  Could someone tell me the 
 difference between the two?  How should I decide which one to use?


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] The problem of the ${CALLERID(num)} for the fxo

[Mr Shunz]

Hi,

There is a question about the fxo of the zaptel card which is set a
 number to use as common analog phone. When I use ${CALLERID(num)}to get it's
 number, it could'n be done. But ${CALLERID(num)} could get the other number
 of the SIP or IAX . Could you tell me the reason, and how I could get the
 number of the fxo which is used as a common analog phone?

maybe your PSTN provider doesn't send you CID info?
often it's an option (and with a cost  too!)

cheers

-- 

Daniele Santi   .o.
[EMAIL PROTECTED] ..o () ascii ribbon campaign
Linux User #415108  ooo /\ www.asciiribbon.org


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Re: [asterisk-users] Is there a way to encrypt passwords stored in the realtime database?

[Tim Panton]

On 20 Aug 2008, at 18:00, Eric Chamberlain wrote:

 We are exploring using Asterisk for a project and we are looking for a
 way to encrypt/decrypt the peer passwords stored in the realtime
 database (postrges).

 Ideally, we want to use a public key to encrypt the passwords before
 they go into the database and have Asterisk use a private key to
 decrypt the password as part of the call out process.

 Has anyone developed something like this?

I haven't done this in asterisk, but we did do a selective
encryption layer for a database on a non-voip project.

First - understand what you are protecting against:
We wanted to be sure that if the backup/sever/tapes/disk were
stolen then the personal data in the database would not be
accessible without the private key.

The way this worked was a bit oracle specific, but
the same concepts are available in postgress.

Basically you have a base table containing the encrypted fields,
this is what is stored on the disk. You then layer on a view (with
appropriate triggers/stored procedures) and the application
(asterisk realtime in your case) uses this view.

The view takes the encrypted fields from the base table and decrypts
them before returning the data to the application.

The trick is that the key is stored in the user's login session (ie in  
memory)
and is initialized at startup (either by typing or from somewhere that  
isn't the
disk - think of a flash drive superglued to the wall :-) with asterisk  
I'd
be tempted to have it call me and I have to dtmf the key in! 

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[asterisk-users] Any chance this is related to fastagi: received mini-frame before full voice frame

[Novak Joe]

Hi,
  I have recently been having difficulty with cmd record where calls
are not being recorded.  I would like to know whether it is possible
that my fastagi script is the root cause of the problem.
  I am using a fastagi script written in python to answer the calls,
and the dialogue interaction seems to work fine, the call is
successfully answered, and I can always hear the audio prompts,
however recently I have been seeing a high incidence of failure to
record the user generated audio, which is invariably accompanied by
iax2 debug messages: received mini-frame before full voice frame.
  Another list user kindly explained to me in a previous message that
this error message means that, if the first full voice frame never
comes through, there is no data on which to base the changes being
conveyed in the mini frames, and thus record doesn't have any way of
knowing what to record.
  My understanding of agi is that it simply passes text commands
back-and-forth between asterisk and the agi/fastagi/eagi script, over
stdin.  This would seem to imply that the record failures I've
described can't be related to the agi script as the actual recording
is not done there anyway.  If possible I'd like to rule out agi as the
culprit categorically in an effort to reduce the problem space a bit.
  Thanks

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Re: [asterisk-users] Asterisk build-environment in Xen-DomU

[Thorolf Godawa]

Hi,

thanks a lot for your answer!

 If you just need Astrerisk for building Zaptel, you don't need the
 kernel modules installed.
I don't need Asterisk for buildung zaptel, I need zaptel running to be
able to compile Asterisk WITH meetme-module (and some others) to build a
RPM that can be installed on other, physical machines.

Or an other possibility that Asterisk is compiled WITH zaptel-support
even if it is not there!
-- 

Chau y hasta luego,

Thorolf

-- 

Chau y hasta luego,

Thorolf

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[asterisk-users] IVR question

[Szasz Szabolcs]

Hi!

I'm setting up my IVR system, how can I register in a mysql database the 
IVR menus accessed by the clients ?

Thanks a lot,

Szasz Szabolcs

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Re: [asterisk-users] How can I determine if IAX trunking is being used and how many calls are being trunked?

[Shaun Wingrin]

Thanks

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Re: [asterisk-users] IVR question

[Yves Räber]

You could use func_odbc in your dialplan, check here : 

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

Yves.

On Thu, 2008-08-21 at 14:57 +0300, Szasz Szabolcs wrote:
 Hi!
 
 I'm setting up my IVR system, how can I register in a mysql database the 
 IVR menus accessed by the clients ?
 
 Thanks a lot,
 
 Szasz Szabolcs
 
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Re: [asterisk-users] IVR question

[Christian Victor]

 I'm setting up my IVR system, how can I register in a mysql database the 
 IVR menus accessed by the clients ?

Just use the MYSQL-Functions in the dialplan to write the menues name
(and datetime maybe) in a table.

To access MYSQL from the dialplan you need to have the asterisk-addons.

Christian

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Re: [asterisk-users] IVR question

[Yves Räber]

Sorry, maybe I misunderstood your question.

If you want the dialplan to be in a MySQL dabtase, check here : 
http://www.voip-info.org/wiki/view/Asterisk+configuration+from+database

Works great, but the documentation is sometimes a bit outdated.

Good luck.

Yves.

On Thu, 2008-08-21 at 14:57 +0300, Szasz Szabolcs wrote:
 Hi!
 
 I'm setting up my IVR system, how can I register in a mysql database
 the 
 IVR menus accessed by the clients ?
 
 Thanks a lot,
 
 Szasz Szabolcs
 
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[asterisk-users] Changing callerID in a context

[Andy Dixon]

Hello,

I am trying to alter the outbound callerID for extensions within a  
context I have created.

I wrote the following:

exten = _9.,2,ExecIf($[$[${REALCALLERIDNUM} = 360] | $[$ 
{REALCALLERIDNUM} = 670]]|Set|CALLERID(num)=581560)
exten = _9.,3,ExecIf($[$[${REALCALLERIDNUM} = 361] | $[$ 
{REALCALLERIDNUM} = 671]]|Set|CALLERID(num)=581561)
exten = _9.,4,ExecIf($[$[${REALCALLERIDNUM} = 362] | $[$ 
{REALCALLERIDNUM} = 672]]|Set|CALLERID(num)=581562)
exten = _9.,5,ExecIf($[$[${REALCALLERIDNUM} = 363] | $[$ 
{REALCALLERIDNUM} = 673]]|Set|CALLERID(num)=581563)
exten = _9.,6,ExecIf($[$[${REALCALLERIDNUM} = 364] | $[$ 
{REALCALLERIDNUM} = 674]]|Set|CALLERID(num)=581564)
exten = _9.,7,ExecIf($[$[${REALCALLERIDNUM} = 365] | $[$ 
{REALCALLERIDNUM} = 675]]|Set|CALLERID(num)=581565)
exten = _9.,8,ExecIf($[$[${REALCALLERIDNUM} = 366] | $[$ 
{REALCALLERIDNUM} = 676]]|Set|CALLERID(num)=581566)
exten = _9.,9,ExecIf($[$[${REALCALLERIDNUM} = 367] | $[$ 
{REALCALLERIDNUM} = 677]]|Set|CALLERID(num)=581567)
exten = _9.,10,ExecIf($[$[${REALCALLERIDNUM} = 368] | $[$ 
{REALCALLERIDNUM} = 678]]|Set|CALLERID(num)=581568)
exten = _9.,11,ExecIf($[$[${REALCALLERIDNUM} = 369] | $[$ 
{REALCALLERIDNUM} = 679]]|Set|CALLERID(num)=581569)
exten = _9.,12,ExecIf($[$[${REALCALLERIDNUM} = 700] | $[$ 
{REALCALLERIDNUM} = 701]]|Set|CALLERID(num)=581557)
exten = _9.,13,ExecIf($[$[${REALCALLERIDNUM} = 100] | $[$ 
{REALCALLERIDNUM} = 101]]|Set|CALLERID(num)=581500)


This *should* change the callerID for (for example) 700 and 701 to be  
581557, and any extensions not listed above, it should leave them alone.

If I call from extension 666, I get the correct outbound number (as it  
does exist), but the rules above are not being followed.

I have tried to use Set(CALLERID(num)=581500) which works okay  
slightly further down.

I am aiming for any numbers starting with a 9 to follow the rules  
above, and then to follow a further rule (eg if the number starts 901,  
or 907)

I'm stuck.. If anyone could help, I would be eternally grateful..


Thanks!

Andy

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Re: [asterisk-users] Two peers, same IP and port

[Drew Gibson]

Chris Hastie wrote:
 Is it possible to have two peers register to Asterisk from the same
 IP/port combination?

 I have a Zoom 5821 two port ATA that can support up to 4 VOIP accounts.
 I want to use it to provide two different extensions on an Asterisk
 system. In the past I have configured two port ATAs to use a different
 SIP local port for each account, but the Zoom unit does not appear to
 allow the SIP local port to be specified on an account by account basis.
 Can I get the unit to register two separate accounts on Asterisk from
 the same port and IP?

   
Hi Chris,

from testing I did a  year ago with 1.2, I would say that his is not 
possible. Asterisk was tracking the registration by IP and could not 
differentiate the accounts by port number alone.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


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[asterisk-users] OT - Asterisk-Stats - Billsec instead of Duration

[Olivier]

Hi,

To check telco billing, I'm usinfg Asterisk-Stats from
http://www.areski.net/asterisk-stat-v2/about.php .

How can you tweak this application to display graphics and data that use
Billsec instead of Duration ?

Regards
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Re: [asterisk-users] Changing callerID in a context

[Philipp Kempgen]

Andy Dixon schrieb:

 I am trying to alter the outbound callerID for extensions within a  
 context I have created.
 
 I wrote the following:
 
 exten = _9.,2,ExecIf($[$[${REALCALLERIDNUM} = 360] | $[$ 
 {REALCALLERIDNUM} = 670]]|Set|CALLERID(num)=581560)
 exten = _9.,3,ExecIf($[$[${REALCALLERIDNUM} = 361] | $[$ 
 {REALCALLERIDNUM} = 671]]|Set|CALLERID(num)=581561)
 exten = _9.,4,ExecIf($[$[${REALCALLERIDNUM} = 362] | $[$ 
 {REALCALLERIDNUM} = 672]]|Set|CALLERID(num)=581562)
 exten = _9.,5,ExecIf($[$[${REALCALLERIDNUM} = 363] | $[$ 
 {REALCALLERIDNUM} = 673]]|Set|CALLERID(num)=581563)
 exten = _9.,6,ExecIf($[$[${REALCALLERIDNUM} = 364] | $[$ 
 {REALCALLERIDNUM} = 674]]|Set|CALLERID(num)=581564)
 exten = _9.,7,ExecIf($[$[${REALCALLERIDNUM} = 365] | $[$ 
 {REALCALLERIDNUM} = 675]]|Set|CALLERID(num)=581565)
 exten = _9.,8,ExecIf($[$[${REALCALLERIDNUM} = 366] | $[$ 
 {REALCALLERIDNUM} = 676]]|Set|CALLERID(num)=581566)
 exten = _9.,9,ExecIf($[$[${REALCALLERIDNUM} = 367] | $[$ 
 {REALCALLERIDNUM} = 677]]|Set|CALLERID(num)=581567)
 exten = _9.,10,ExecIf($[$[${REALCALLERIDNUM} = 368] | $[$ 
 {REALCALLERIDNUM} = 678]]|Set|CALLERID(num)=581568)
 exten = _9.,11,ExecIf($[$[${REALCALLERIDNUM} = 369] | $[$ 
 {REALCALLERIDNUM} = 679]]|Set|CALLERID(num)=581569)
 exten = _9.,12,ExecIf($[$[${REALCALLERIDNUM} = 700] | $[$ 
 {REALCALLERIDNUM} = 701]]|Set|CALLERID(num)=581557)
 exten = _9.,13,ExecIf($[$[${REALCALLERIDNUM} = 100] | $[$ 
 {REALCALLERIDNUM} = 101]]|Set|CALLERID(num)=581500)
 
 
 This *should* change the callerID for (for example) 700 and 701 to be  
 581557, and any extensions not listed above, it should leave them alone.
 
 If I call from extension 666, I get the correct outbound number (as it  
 does exist), but the rules above are not being followed.
 
 I have tried to use Set(CALLERID(num)=581500) which works okay  
 slightly further down.
 
 I am aiming for any numbers starting with a 9 to follow the rules  
 above, and then to follow a further rule (eg if the number starts 901,  
 or 907)
 
 I'm stuck.. If anyone could help, I would be eternally grateful..

This would be much more readable in AEL.
Or in an external script.
But maybe all you really need is fromuser in sip.conf or similar.

Grüße,
Philipp Kempgen
-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998

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Re: [asterisk-users] How can I determine if IAX trunking is being used and how many calls are being trunked?

[Doug Lytle]

Shaun Wingrin wrote:
 Thanks


There will be a (T) after the iax entry:

asterisk.cw  192.168.200.2   (D)  255.255.255.255  4569 (T)  OK 
(76 ms)
asterisk.liv 192.168.102.15  (D)  255.255.255.255  4569 (T)  OK 
(77 ms)
asterisk.bc  192.168.104.10  (D)  255.255.255.255  4569 (T)  OK 
(40 ms)

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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[asterisk-users] Anyone using asterisk on centos 4.X without hardware cards and using console/dsp

[Jerry Geis]

I am using centos 4.6 i586.

I have compiles zaptel 1.4.11 ztdummy.
When I load ztdummy the /proc/interupts  rtc does not increment.

centos runs 2.6.9 kernel.
I'm not sure ztdummy.c uses RTC by default in this case.

Anyone using centos 4.X successfully with console/dsp and not internal 
cards.
How did you do it?

Thanks,

Jerry

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Re: [asterisk-users] Two peers, same IP and port

[Tariq ..]

I have LinkSYS PAP2t and it worked the way you discribed it.. Asterisk simply 
assigns a different port for the peer automaticaly.

 

 Date: Wed, 20 Aug 2008 20:09:32 +0100 From: [EMAIL PROTECTED] To: 
 asterisk-users@lists.digium.com Subject: [asterisk-users] Two peers, same IP 
 and port  Is it possible to have two peers register to Asterisk from the 
 same IP/port combination?  I have a Zoom 5821 two port ATA that can 
 support up to 4 VOIP accounts. I want to use it to provide two different 
 extensions on an Asterisk system. In the past I have configured two port 
 ATAs to use a different SIP local port for each account, but the Zoom unit 
 does not appear to allow the SIP local port to be specified on an account by 
 account basis. Can I get the unit to register two separate accounts on 
 Asterisk from the same port and IP?  --  Chris Hastie  
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[asterisk-users] USB ISDN TA Help requested

[Tariq ..]

Hello 
I have a SendTEK UNIK-22 USB ISDN TA unit attached to my Asterisk
i it possible to use it to make and receive calls with asterisk? and if so can 
anyone help me? or at least give me some hints? i tried but couldn't manage it
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[asterisk-users] A Suggestion To Asterisk Appliance Developers

[mgraves]

Yesterday I blogged a post about some ideas that I think will help
Asterisk appliances further penetrate SMB/SOHO sites in ways that are
not presently being addressed.

http://blog.mgraves.org/2008/08/20/a-suggestion-to-asterisk-appliance-developers/

Michael Graves
mgraves at mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
FWD 54245




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Re: [asterisk-users] A Suggestion To Asterisk Appliance Developers

[EdPimentl]

Please google VoIP2.0 apps... this is old old news... even Cisco has
marketed this going back to 2001.
-E
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Re: [asterisk-users] 1st call after some time has one way speech, but calls after that are fine..

[Shaun Wingrin]

Hi, 

Hoping someone can help with this most frustrating situation.

I have a Linksys PAP2T registering with ADSL to my asterisk server which also 
sits behind a Mikrotik router.

Thanks
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Re: [asterisk-users] USB ISDN TA Help requested

[Tzafrir Cohen]

On Thu, Aug 21, 2008 at 02:38:49PM +, Tariq .. wrote:
 Hello 
 I have a SendTEK UNIK-22 USB ISDN TA unit attached to my Asterisk
 i it possible to use it to make and receive calls with asterisk? and if so 
 can anyone help me? or at least give me some hints? i tried but couldn't 
 manage it

Generally with Linux if you ask about a random USB device, in addition
to the name of the device, the output of lsusb can be of help.

Also: What version of Asterisk do you have? What operating system? (e.g:
what Linux distribution)?

(I suspect that this device is supported by mISDN and hence could be
used with chan_misdn in Asterisk)

-- 
   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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[asterisk-users] Asterisk Realtime pounds MySQL

[J . M .]

I am running Asterisk 1.4.21.2 with Realtime.  I have a phone setup in the
database and when I connect that phone to Asterisk there are suddenly an
endless number of SELECT * FROM sip WHERE name = '1001' AND host =
'dynamic' queries being run.  The only way to stop the flood of queries
coming from Asterisk to restart the Asterisk process.  Even disconnecting
the phone doesn't stop Asterisk from running the queries.

Has anyone seen this before?  Why would Asterisk do that and does anyone
know the fix?

The phone I am using is the softphone X-Lite.
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Re: [asterisk-users] Anyone using asterisk on centos 4.X without hardware cards and using console/dsp

[Tzafrir Cohen]

On Thu, Aug 21, 2008 at 10:29:18AM -0400, Jerry Geis wrote:
 I am using centos 4.6 i586.
 
 I have compiles zaptel 1.4.11 ztdummy.
 When I load ztdummy the /proc/interupts  rtc does not increment.

does ztdummy itself tick?

  try zttest

If it does not stay hung there, it's working.

-- 
   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] FTC Bans Prerecorded Telemarketing Drivel

[Jay R. Ashworth]

On Thu, Aug 21, 2008 at 02:15:58AM -0400, Alex Balashov wrote:
 I would be curious to know where, in this classification, fall various 
 telemarketing schemes that are technically not cold-calls, but are 
 generated from leads that come from customer-provided information, but 
 where the customer does not know explicitly that they are signing up to 
 receive calls.
 
 For instance, this is common in a number of industries such as financial 
 services.  You do a search to get a quote on something, and provide your 
 phone number in the process, although the phone number bears no relation 
 to the submission and is just an ancillary required item.  Several 
 places' telemarketing organisations call you back in response.  For 
 example, lendingtree.com.
 
 Is this a solicited call?

In order to classify that as a solicited call, I believe, you have to
have language *on the form the customer fills out* that says they're
authorizing you to call, and you have to be able to produce
ink-on-paper if the FTC ever calls you on it.

IANAL.  YMMV.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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Re: [asterisk-users] DSS1 vs SS7

[Jay R. Ashworth]

On Thu, Aug 21, 2008 at 02:38:19AM -0400, Alex Balashov wrote:
 Use DSS1.  It's European ISDN and would give you the equivalent of a 
 North American PRI.
 
 You don't want SS7.

I would assume that means SS7 protocol over a link not routed directly
to the SS7 backbone.

At least I hope it means that.  shudder

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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Re: [asterisk-users] Anyone using asterisk on centos 4.X without hardware cards and using console/dsp

[Jerry Geis]

On Thu, Aug 21, 2008 at 10:29:18AM -0400, Jerry Geis wrote:
/ I am using centos 4.6 i586.
// 
// I have compiles zaptel 1.4.11 ztdummy.

// When I load ztdummy the /proc/interupts  rtc does not increment.
/
does ztdummy itself tick?

  try zttest

If it does not stay hung there, it's working.

  



zttest
Opened pseudo zap interface, measuring accuracy...
99.978424% 99.961136% 99.971382% 99.972458% 99.970802% 99.971672% 
99.971092%

99.972565% 99.966499% 99.972946% 99.972458% 99.973045%


This is what I get from zttest - cat /proc/interrupts is not 
incrementing rtc.


Jerry
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[asterisk-users] callfiles/manager api originate call fails

[Rizwan Hisham]

Hi all,
asterisk is giving me tough time. its been 3 days I am trying to originate
outgoing call using manager api/callfiles. both seem to work fine when i
originate a call for a local peer, but if i try originating a call outside
using a trunk thats when everything goes wrong. It does originate the call
but the call does not go through to the desired endpoint. The trunk
configuration is correct as all the other calls from users are fine. Am here
for any suggestion. How can i make it work. If anyone knows anyother
technique to originate auto calls from asterisk i'll be happy to try them
out.

 I am using the following manager command,

fputs($socket, Action: Originate\r\n);

//fputs($socket, Channel: SIP/abc\r\n);

fputs($socket, Channel: SIP/.$txt_your_number.@TRUNK-OUT\r\n);

fputs($socket, Context: webcall\r\n);

fputs($socket, Exten:
932\r\n);
fputs($socket, Priority: 1\r\n);

fputs($socket, CallerID:
WebCall932\r\n);
fputs($socket, Timeout: 3\r\n);

fputs($socket, Variable: ID= . $id . |accountcode=7:0:webcall|sec= .
$min . |dialnum= . $txt_to_number . |source_num= . $txt_your_number .
|calldate= . date(Y-m-d H:i:s) . \r\n\r\n);


and my callfile contents are:

Channel: SIP/TRUNK-OUT/$DIALNUM
CallerID: Webcall932
MaxRetries: 2
RetryTime: 10
WaitTime: 35
Account: 7:0:webcall
Context: webcall
Extension: 932
Priority: 1
Set: ID=.$id.
Set: accountcode=7:0:webcall
Set: sec=.$allowed_secs.
Set: dialnum=.$dialnum.\
et: source_num=.$srcnum.
Set: calldate=.$calldate. .$calltime.\n;

-- 
Best Regards
Rizwan Hisham
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Re: [asterisk-users] Anyone using asterisk on centos 4.X without hardware cards and using console/dsp

[Tzafrir Cohen]

On Thu, Aug 21, 2008 at 11:19:22AM -0400, Jerry Geis wrote:
 
 On Thu, Aug 21, 2008 at 10:29:18AM -0400, Jerry Geis wrote:
 / I am using centos 4.6 i586.
 // 
 // I have compiles zaptel 1.4.11 ztdummy.
 // When I load ztdummy the /proc/interupts  rtc does not increment.
 /
 does ztdummy itself tick?
 
   try zttest
 
 If it does not stay hung there, it's working.
 
   
 
 
 zttest
 Opened pseudo zap interface, measuring accuracy...
 99.978424% 99.961136% 99.971382% 99.972458% 99.970802% 99.971672% 
 99.971092%
 99.972565% 99.966499% 99.972946% 99.972458% 99.973045%
 
 
 This is what I get from zttest - cat /proc/interrupts is not 
 incrementing rtc.
 
 Jerry

Zaptel timing works just fine. 

Maybe you have the volume set to 0 or something . 

-- 
   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] A Suggestion To Asterisk Appliance Developers

[Michael Graves]

I Googled as you suggest and nothing even vaugely related is returned.
In fact, VOIP 2.0 as a term doesn't seem to relate.

What I'm suggesting is that smaller PBX systems should embrace a larger
role in the end users operation. 

I don't see CCM is small companies or home offices. This is all about
the SMB/SOHO sector where the PBX could easily me more central to the
users business.

Michael

--Original Message Text---
From: EdPimentl
Date: Thu, 21 Aug 2008 11:02:22 -0400

Please google VoIP2.0 apps... this is old old news... even Cisco has
marketed this going back to 2001.
-E

Internal Virus Database is out of date.
Checked by AVG - http://www.avg.com 
Version: 8.0.138 / Virus Database: 270.6.4/1617 - Release Date:
8/17/2008 12:58 PM


--
Michael Graves
mgravesatmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves

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[asterisk-users] After Dial execution, using DIALEDTIME, ANSWEREDTIME

[selmak se]

 Hi,







 I noticed that when dial terminates it does not return to the dialplan, 
and therefore can not execute any entry after Dial().



 Is there any trick to overcome this limitation ?





 How I am supposed to handle the returned vales DIALEDTIME, ANSWEREDTIME if 
I can not execute anything after Dial()?





 I made a workaround with DeadAGI (below) but it is unreliable: if 2 calls 
end nearly at the same time I do not know to whom belongs the ANSWEREDTIME 
value :



 exten = h,1,DeadAGI(myagi.agi,0,${DIALEDTIME},${ANSWEREDTIME},00)



 Any comments?





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[asterisk-users] Automatic call to voicemail on login?

[Stefan Gofferje]

Hi,

I would like to arrange that when an IAX client logs in / registers with
my * AND there are unread voicemails, this IAX client will be
automatically called and connected to the respective voicemail box.

One possibility is to have a cronjob that creates a callfile - let's say
- every five minutes which checks ChanIsAvail and connect to the
voicebox if new messages are there.

But with lots of IAX clients, this does not exactly scale very well.

If there any other native way to execute an action on login or logout of
a client?

Terve,
Stefan

-- 
Last words of a stormchaser:
Where is that rotation on the radar?!


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Re: [asterisk-users] After Dial execution, using DIALEDTIME, ANSWEREDTIME

[Darren Sessions]

We recently discussed DeadAGI on the list - I'd check the archives  
first.


I just finished doing a write up on DeadAGI and Perl on my website if  
you're interested.


DeadAGI *can* be very reliable if done properly.

- Darren


_

[EMAIL PROTECTED]
http://www.darrensessions.com
_




On Aug 21, 2008, at 9:35 AM, selmak se wrote:



Hi,



I noticed that when dial terminates it does not return to the  
dialplan, and therefore can not execute any entry after Dial().


Is there any trick to overcome this limitation ?


How I am supposed to handle the returned vales DIALEDTIME,  
ANSWEREDTIME if I can not execute anything after Dial()?



I made a workaround with DeadAGI (below) but it is unreliable: if 2  
calls end nearly at the same time I do not know to whom belongs the  
ANSWEREDTIME value :


exten = h,1,DeadAGI(myagi.agi,0,${DIALEDTIME},${ANSWEREDTIME},00)

Any comments?



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Re: [asterisk-users] FTC Bans Prerecorded Telemarketing Drivel

[Anthony Francis]

Jay R. Ashworth wrote:
 On Thu, Aug 21, 2008 at 02:15:58AM -0400, Alex Balashov wrote:
   
 I would be curious to know where, in this classification, fall various 
 telemarketing schemes that are technically not cold-calls, but are 
 generated from leads that come from customer-provided information, but 
 where the customer does not know explicitly that they are signing up to 
 receive calls.

 For instance, this is common in a number of industries such as financial 
 services.  You do a search to get a quote on something, and provide your 
 phone number in the process, although the phone number bears no relation 
 to the submission and is just an ancillary required item.  Several 
 places' telemarketing organisations call you back in response.  For 
 example, lendingtree.com.

 Is this a solicited call?
 

 In order to classify that as a solicited call, I believe, you have to
 have language *on the form the customer fills out* that says they're
 authorizing you to call, and you have to be able to produce
 ink-on-paper if the FTC ever calls you on it.

 IANAL.  YMMV.

 Cheers,
 -- jra
   
Actually in the US all you have to do is provide some proof of a 
business relationship with them. Companes get away with calling you if 
you have ever bought even one item from them.

-- 
Thank you and have any kind of day you want,

Anthony Francis



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Re: [asterisk-users] callfiles/manager api originate call fails

[Anthony Francis]

Rizwan Hisham wrote:
 Hi all,
 asterisk is giving me tough time. its been 3 days I am trying to 
 originate outgoing call using manager api/callfiles.

I would say remove the @TRUNK-OUT part and make sure that the context 
you send the call to knows about sending calls to the outside world.

-- 
Thank you and have any kind of day you want,

Anthony Francis




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Re: [asterisk-users] Automatic call to voicemail on login?

[Martin Smith]

Hi Stefan,

I'd expect there's a Manager event that is fired when an IAX client
login happens. You could watch for that and initiate your call if
there's voicemail at that time.

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Stefan Gofferje
 Sent: Thursday, August 21, 2008 11:39 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Automatic call to voicemail on login?
 
 Hi,
 
 I would like to arrange that when an IAX client logs in / 
 registers with
 my * AND there are unread voicemails, this IAX client will be
 automatically called and connected to the respective voicemail box.
 
 One possibility is to have a cronjob that creates a callfile 
 - let's say
 - every five minutes which checks ChanIsAvail and connect to the
 voicebox if new messages are there.
 
 But with lots of IAX clients, this does not exactly scale very well.
 
 If there any other native way to execute an action on login 
 or logout of
 a client?
 
 Terve,
 Stefan
 
 -- 
 Last words of a stormchaser:
 Where is that rotation on the radar?!
 
 
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[asterisk-users] Asterisk and Huawei SoftX3000

[Gustavo A Gonzalez]

Hi folks! I have a problem with our Sip provider that have a Softswitch
Huawei SoftX3000 to send us SIP calls to our Asterisk PBX, we are working
with G711 with them. They start sending calls to our pbx, some time after
they start to receive 408 messages from asterisk and some time after this
they start to complete calls normally, I don’t know what can be wrong.
Someone has configured asterisk to wok with this Softswitch? Thanks for any
help!

 

Cheers!

 

 

Gustavo A. González
Dto. de Infraestructura
Despegar.com, Inc.
 mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] 

 

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Re: [asterisk-users] After Dial execution, using DIALEDTIME, ANSWEREDTIME

[Ruddy Gbaguidi]

First, if you want to use that, you may want pass the call tracknum to 
the myagi.agi,
so you will know which call the dialedtime and answeredtime belongs to.

But you can use the Dial 'g' option that doesn't hangup up both legs of 
the call when the called party hangs up.



selmak se wrote:


 Hi,




 I noticed that when dial terminates it does not return to the 
 dialplan, and therefore can not execute any entry after Dial().


 Is there any trick to overcome this limitation ?



 How I am supposed to handle the returned vales DIALEDTIME, 
 ANSWEREDTIME if I can not execute anything after Dial()?



 I made a workaround with DeadAGI (below) but it is unreliable: if 2 
 calls end nearly at the same time I do not know to whom belongs the 
 ANSWEREDTIME value :


 exten = h,1,DeadAGI(myagi.agi,0,${DIALEDTIME},${ANSWEREDTIME},00)


 Any comments?




 

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 Internal Virus Database is out of date.
 Checked by AVG. 
 Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 
 7:42 PM
   


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Re: [asterisk-users] After Dial execution, using DIALEDTIME, ANSWEREDTIME

[selmak se]

Thank you for your answer,





Is the call tracknum stored in some variable?



Could you let me know how to  pass a call tracknum to an AGI. 





Se.-





 - Original Message -
 From: Ruddy Gbaguidi
 
 First, if you want to use that, you may want pass the call tracknum to 
 the myagi.agi,
 so you will know which call the dialedtime and answeredtime belongs to.
 
 But you can use the Dial 'g' option that doesn't hangup up both legs of 
 the call when the called party hangs up.
 
 
 
 selmak se wrote:
 
 
  Hi,
 
 
 
 
  I noticed that when dial terminates it does not return to the 
  dialplan, and therefore can not execute any entry after Dial().
 
 
  Is there any trick to overcome this limitation ?
 
 
 
  How I am supposed to handle the returned vales DIALEDTIME, 
  ANSWEREDTIME if I can not execute anything after Dial()?
 
 
 
  I made a workaround with DeadAGI (below) but it is unreliable: if 2 
  calls end nearly at the same time I do not know to whom belongs the 
  ANSWEREDTIME value :
 
 
  exten = h,1,DeadAGI(myagi.agi,0,${DIALEDTIME},${ANSWEREDTIME},00)
 
 
  Any comments?
 
 
 
 
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Re: [asterisk-users] FTC Bans Prerecorded Telemarketing Drivel

[Jay R. Ashworth]

On Thu, Aug 21, 2008 at 09:44:50AM -0600, Anthony Francis wrote:
 Jay R. Ashworth wrote:
  On Thu, Aug 21, 2008 at 02:15:58AM -0400, Alex Balashov wrote:
  I would be curious to know where, in this classification, fall various 
  telemarketing schemes that are technically not cold-calls, but are 
  generated from leads that come from customer-provided information, but 
  where the customer does not know explicitly that they are signing up to 
  receive calls.
 
  For instance, this is common in a number of industries such as financial 
  services.  You do a search to get a quote on something, and provide your 
  phone number in the process, although the phone number bears no relation 
  to the submission and is just an ancillary required item.  Several 
  places' telemarketing organisations call you back in response.  For 
  example, lendingtree.com.
 
  Is this a solicited call?
 
  In order to classify that as a solicited call, I believe, you have to
  have language *on the form the customer fills out* that says they're
  authorizing you to call, and you have to be able to produce
  ink-on-paper if the FTC ever calls you on it.
 
  IANAL.  YMMV.

 Actually in the US all you have to do is provide some proof of a 
 business relationship with them. Companes get away with calling you if 
 you have ever bought even one item from them.

Which doesn't actually speak to the situation about which Alex asked,
and I posited.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

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 Those who count the vote decide everything.
   -- (Josef Stalin)

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Re: [asterisk-users] DSS1 vs SS7

[Alex Balashov]

Jay R. Ashworth wrote:
 On Thu, Aug 21, 2008 at 02:38:19AM -0400, Alex Balashov wrote:
 Use DSS1.  It's European ISDN and would give you the equivalent of a 
 North American PRI.

 You don't want SS7.
 
 I would assume that means SS7 protocol over a link not routed directly
 to the SS7 backbone.
 
 At least I hope it means that.  shudder

Indeed, it is certainly private SS7.  :-)  But that does not mean it is 
any more desirable for the user.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] FTC Bans Prerecorded Telemarketing Drivel

[Alex Balashov]

Anthony Francis wrote:

 Actually in the US all you have to do is provide some proof of a 
 business relationship with them. Companes get away with calling you if 
 you have ever bought even one item from them.

So, what if you never bought anything, but ended up as a lead in their 
system through some voluntary action?

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Asterisk and Huawei SoftX3000

[Alex Balashov]

Gustavo A Gonzalez wrote:

 Hi folks! I have a problem with our Sip provider that have a Softswitch 
 Huawei SoftX3000 to send us SIP calls to our Asterisk PBX, we are 
 working with G711 with them. They start sending calls to our pbx, some 
 time after they start to receive 408 messages from asterisk and some 
 time after this they start to complete calls normally, I don’t know what 
 can be wrong. Someone has configured asterisk to wok with this 
 Softswitch? Thanks for any help!

A packet capture illustrating the problem would be of utmost utility.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] DSS1 vs SS7

[Jay R. Ashworth]

On Thu, Aug 21, 2008 at 12:21:36PM -0400, Alex Balashov wrote:
  I would assume that means SS7 protocol over a link not routed directly
  to the SS7 backbone.
  
  At least I hope it means that.  shudder
 
 Indeed, it is certainly private SS7.  :-)  But that does not mean it is 
 any more desirable for the user.

Thank ghod.  :-)
-- j
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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Re: [asterisk-users] After Dial execution, using DIALEDTIME, ANSWEREDTIME

[Tony Mountifield]

In article [EMAIL PROTECTED], selmak se [EMAIL PROTECTED] wrote:
  I noticed that when dial terminates it does not return to the dialplan, 
 and therefore can not execute any entry after Dial().
 
  Is there any trick to overcome this limitation ?

You can give the 'g' option to Dial, but that might not be the best way.

  How I am supposed to handle the returned vales DIALEDTIME, ANSWEREDTIME if 
 I can not execute anything after Dial()?

In the 'h' extension, as you mentioned below.

  I made a workaround with DeadAGI (below) but it is unreliable: if 2 calls 
 end nearly at the same time I do not know to whom belongs the ANSWEREDTIME 
 value :
 
  exten = h,1,DeadAGI(myagi.agi,0,${DIALEDTIME},${ANSWEREDTIME},00)

Use the value of ${UNIQUEID} to distinguish between the channels. It will
be unique for every channel in a system.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] FTC Bans Prerecorded Telemarketing Drivel

[Mark Adams]

Not to sound arrogant but the law is one thing and enforcement is another,
these types of calls have been illegal for a long time and 9 times out of 10
the only penalty one receives is a civil suit by some back yard attorney
looking for a couple thousand bucks. 

Unless that is you are a serious violator and then that's a different story.


Cases I have ever read about are from yahoos that cant maintain a do not
call list or don't even bother to try. 

My 2 cents

Mark 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: Thursday, August 21, 2008 12:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FTC Bans Prerecorded Telemarketing Drivel

Anthony Francis wrote:

 Actually in the US all you have to do is provide some proof of a 
 business relationship with them. Companes get away with calling you if 
 you have ever bought even one item from them.

So, what if you never bought anything, but ended up as a lead in their 
system through some voluntary action?

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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[asterisk-users] Question: Soft phone for ACD agents?

[Michael Collins]

To those running call centers I have a question: what kinds of soft
phones, if any, do you use? I'm wondering what is out there that has
some hooks for custom applications or host system integration, etc.
OTOH, do you prefer a desk phone for any reason?  If so, why?

 

Thanks for your thoughts,

Michael

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[asterisk-users] 5 min limitation on phone calls! how to!

[RoLaNd RoLaNd]

Hello all! 
 
my last month's phone bill sky rocketed after i setup asterisk with softphones 
all over the house!

could someone help me set up a limitation for my wife and kids not to be able 
to talk for more than 5 min at a time!
or like 20 min per week! or whtever limitation i could set for this!

any help would trull be appreciated:)

_
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[asterisk-users] Siemens Gigaset IP in USA (S685 IP in particular)

[Paul Chambers]

For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP 
range in the U.S. I'm particularly interested in the Gigaset S685 IP. 
Since it's DECT 6.0, and there's an English (UK) version, I'm thinking 
it should work just fine, after dealing with the walwart issue (and 
maybe caller ID signalling).

Anyone imported one from the UK and using it in the US? for how long? 
impressions? anything not working?

Have you purchased additional US-spec handsets and used them with the UK 
basestation?

Thanks in advance,

Paul

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[asterisk-users] ultramonkey and asterisk

[ronald]

hi all,

has anyone able to configure ultramonkey for sip (namely asterisk).
i tried from this tutorial:

http://blog.iclutton.com/2008/01/load-balancing-and-high-availablity.html

i have this on my ldirectord.cf:

virtual=123.45.67.155:5060
  real=123.45.67.130:5060 gate
  real=123.45.67.131:5060 gate
  service=sip
  scheduler=rr
  protocol=udp
  checktype=negotiate
  persistent=1

i was able to make my http and https to work but not sip.
hope someon could help me. thanks

regards,
nhadie

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Re: [asterisk-users] Siemens Gigaset IP in USA (S685 IP in particular)

[Drew Gibson]

Paul Chambers wrote:
 For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP 
 range in the U.S. I'm particularly interested in the Gigaset S685 IP. 
 Since it's DECT 6.0, and there's an English (UK) version, I'm thinking 
 it should work just fine, after dealing with the walwart issue (and 
 maybe caller ID signalling).

 Anyone imported one from the UK and using it in the US? for how long? 
 impressions? anything not working?

 Have you purchased additional US-spec handsets and used them with the UK 
 basestation?

 Thanks in advance,

 Paul
   

The original DECT standard uses 1880-1900MHz, as implemented in Europe.

The US FCC designated 1920-1930MHz. This is marketed as DECT 6.0.

The FCC might get angry at you for using regular DECT phones in the US.

And your neighbours with iPhones (GSM) might also get angry...

regards,

Drew


-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


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Re: [asterisk-users] 5 min limitation on phone calls! how to!

[Lyle Giese]

RoLaNd RoLaNd wrote:

Hello all!
 
my last month's phone bill sky rocketed after i setup asterisk with 
softphones all over the house!


could someone help me set up a limitation for my wife and kids not to 
be able to talk for more than 5 min at a time!

or like 20 min per week! or whtever limitation i could set for this!

any help would trull be appreciated:)
I would check the CDR logs first and make sure that a hacker did not get 
into your * box and is making calls on your dime.


Lyle

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Re: [asterisk-users] 5 min limitation on phone calls! how to!

[Stefan Gofferje]

RoLaNd RoLaNd schrieb:
 Hello all!
  
 my last month's phone bill sky rocketed after i setup asterisk with
 softphones all over the house!
 
 could someone help me set up a limitation for my wife and kids not to be
 able to talk for more than 5 min at a time!
 or like 20 min per week! or whtever limitation i could set for this!

Set(TIMEOUT(absolute)=seconds)

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


Terve,
Stefan

-- 
Last words of a stormchaser:
Where is that rotation on the radar?!


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Re: [asterisk-users] 5 min limitation on phone calls! how to!

[Steve Totaro]

On Thu, Aug 21, 2008 at 12:50 PM, RoLaNd RoLaNd [EMAIL PROTECTED] wrote:
 Hello all!

 my last month's phone bill sky rocketed after i setup asterisk with
 softphones all over the house!

 could someone help me set up a limitation for my wife and kids not to be
 able to talk for more than 5 min at a time!
 or like 20 min per week! or whtever limitation i could set for this!

 any help would trull be appreciated:)


You could check out ASTPP or ASTCC and give them their own accounts.

Thanks,
Steve Totaro

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Re: [asterisk-users] Question: Soft phone for ACD agents?

[Jay R. Ashworth]

On Thu, Aug 21, 2008 at 09:40:04AM -0700, Michael Collins wrote:
To those running call centers I have a question: what kinds of soft phones,
if any, do you use? I’m wondering what is out there that has some hooks 
 for
custom applications or host system integration, etc.  OTOH, do you prefer a
desk phone for any reason?  If so, why?

The experience in the center I manage the network for was that
softphones didn't work out that well, and that regular phones on ATAs
weren't much of a win either: ATAs apparently weren't built for 8+
hr/day service; they'd melt down.

What we ended up with was Panasonic hardphones with headset jacks (the
KX-TS105) plugged into Zhone zPlex 10 FXS channel banks, then T-1 into
quad cards on our VICIdial diallers.  That gives you 3 spans per room,
which works out pretty well even for automatic outbound, though we
only do manual these days.

I'm in the market for a better station: I don't need the dial, or the
handset, or even the ringer (a neon bulb would be fine), but I *do*
want something more rugged than those Panasonic's.  I'm not sure why no
one seems to build a ruggedized agent phone.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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[asterisk-users] Reacting to an event in the dialplan (Was RE:Automatic call to voicemail on login?)

[Martin Smith]

That's a good point. I don't know, honestly, if you can react to a SIP
register or an IAX login from within the dialplan. To anyone else: 

Is there a way to act in the dialplan on a manager event?

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Stefan Gofferje
 Sent: Thursday, August 21, 2008 1:40 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Automatic call to voicemail on login?
 
 Martin Smith schrieb:
  I'd expect there's a Manager event that is fired when an IAX client
  login happens. You could watch for that and initiate your call if
  there's voicemail at that time.
 
 That would mean, I would need to write an application opening a socket
 and such. Not exactly easy. Isn't there anything I could do with the
 dialplan or similar; i.e. from asterisk itself?
 
 Terve,
 Stefan
 
 -- 
 Last words of a stormchaser:
 Where is that rotation on the radar?!
 
 
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Re: [asterisk-users] 5 min limitation on phone calls! how to!

[Eric ManxPower Wieling]

Steve Totaro wrote:
 On Thu, Aug 21, 2008 at 12:50 PM, RoLaNd RoLaNd [EMAIL PROTECTED] wrote:
 Hello all!

 my last month's phone bill sky rocketed after i setup asterisk with
 softphones all over the house!

 could someone help me set up a limitation for my wife and kids not to be
 able to talk for more than 5 min at a time!
 or like 20 min per week! or whtever limitation i could set for this!

 any help would trull be appreciated:)

 
 You could check out ASTPP or ASTCC and give them their own accounts.

The L() option of Dial is used for this sort of thing.

-- 
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, 
T-1, PRI, Frame Relay, Linux, and network design.  Based near 
Birmingham, AL.  Now accepting clients worldwide.

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Re: [asterisk-users] Question: Soft phone for ACD agents?

[Tim Panton]

On 21 Aug 2008, at 18:44, Jay R. Ashworth wrote:

 On Thu, Aug 21, 2008 at 09:40:04AM -0700, Michael Collins wrote:
   To those running call centers I have a question: what kinds of  
 soft phones,
   if any, do you use? I’m wondering what is out there that has  
 some hooks for
   custom applications or host system integration, etc.  OTOH, do  
 you prefer a
   desk phone for any reason?  If so, why?

 The experience in the center I manage the network for was that
 softphones didn't work out that well, and that regular phones on ATAs

Interesting  - What were the problems with softphones ?
Or alternatively , what would you want from an ideal softphone ?

Tim.
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Re: [asterisk-users] 5 min limitation on phone calls! how to!

[Karl Fife]

This has got to be one of the funniest threads ever to grace this forum.
Sorry honey! ...CLICK.
In my house, this might require a more 'diplomatic' approach :-)
-Karl 



On Thu, 21 Aug 2008 21:41:40 +0300, RoLaNd RoLaNd
[EMAIL PROTECTED] said:
 
 i tried that before.. it didnt actually work! it the call kept on going
 well beyound the allowed test seconds...
 heres my extensions.conf:
 
 
 [sipura-line]
 exten = 301,1,Answer() ; Answer inbound calls
 exten = 301,2,Playback(silence/1)
 exten = 301,3,Background(simzy1) ; input an extension
 exten = 301,4,WaitExten(8)
 exten = 301,5,Dial(SIP/100,15) ; goes to operator
 exten = 301,4,Wait(8)
 include = spa
 exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
 exten = 301,n,Hangup()
 
 
 
 
 [spa]
 exten =_301,1,GoTo(sipura-line,${EXTEN},1)
 exten = _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds so
 it will ring 3 times
 exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if
 line is busy or unavailable
 exten = _1XX,3,HangUp()
 exten = _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds so
 it will ring 3 times
 exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if
 line is busy or unavailable
 exten = _2XX,3,HangUp()
 exten = _3XX,1,Dial(SIP/${EXTEN},15) ; each ring equals to 5 seconds so
 it will ring 3 times
 exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if
 line is busy or unavailable
 exten = _3XX,3,HangUp()
 exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
 exten =_01,2,Set(TIMEOUT(absolute)=5)
 exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
 exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer
 exten =_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte
 exten =_05,1,Dial(SIP/$(EXTEN)@307) ; conference
 exten =_06,1,Dial(SIP/$(EXTEN)@308) ; line 4
 exten = 303,1,VoicemailMain ; voicemail box to be redirected to
 
 
 
 
  Date: Thu, 21 Aug 2008 20:26:48 +0300
  From: [EMAIL PROTECTED]
  To: asterisk-users@lists.digium.com
  Subject: Re: [asterisk-users] 5 min limitation on phone calls! how to!
  
  RoLaNd RoLaNd schrieb:
   Hello all!

   my last month's phone bill sky rocketed after i setup asterisk with
   softphones all over the house!
   
   could someone help me set up a limitation for my wife and kids not to be
   able to talk for more than 5 min at a time!
   or like 20 min per week! or whtever limitation i could set for this!
  
  Set(TIMEOUT(absolute)=seconds)
  
  http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AbsoluteTimeout
  
  
  Terve,
  Stefan
  
  -- 
  Last words of a stormchaser:
  Where is that rotation on the radar?!
  
  
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Re: [asterisk-users] 5 min limitation on phone calls! how to!

[Singer XJ Wang]

Phone Guy: NO PHONE FOR YOU!

Karl Fife wrote:

This has got to be one of the funniest threads ever to grace this forum.
Sorry honey! ...CLICK.
In my house, this might require a more 'diplomatic' approach :-)
-Karl 




On Thu, 21 Aug 2008 21:41:40 +0300, RoLaNd RoLaNd
[EMAIL PROTECTED] said:
  

i tried that before.. it didnt actually work! it the call kept on going
well beyound the allowed test seconds...
heres my extensions.conf:


[sipura-line]
exten = 301,1,Answer() ; Answer inbound calls
exten = 301,2,Playback(silence/1)
exten = 301,3,Background(simzy1) ; input an extension
exten = 301,4,WaitExten(8)
exten = 301,5,Dial(SIP/100,15) ; goes to operator
exten = 301,4,Wait(8)
include = spa
exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
exten = 301,n,Hangup()




[spa]
exten =_301,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds so
it will ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if
line is busy or unavailable
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds so
it will ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if
line is busy or unavailable
exten = _2XX,3,HangUp()
exten = _3XX,1,Dial(SIP/${EXTEN},15) ; each ring equals to 5 seconds so
it will ring 3 times
exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if
line is busy or unavailable
exten = _3XX,3,HangUp()
exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
exten =_01,2,Set(TIMEOUT(absolute)=5)
exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer
exten =_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte
exten =_05,1,Dial(SIP/$(EXTEN)@307) ; conference
exten =_06,1,Dial(SIP/$(EXTEN)@308) ; line 4
exten = 303,1,VoicemailMain ; voicemail box to be redirected to






Date: Thu, 21 Aug 2008 20:26:48 +0300
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] 5 min limitation on phone calls! how to!

RoLaNd RoLaNd schrieb:
  

Hello all!
 
my last month's phone bill sky rocketed after i setup asterisk with

softphones all over the house!

could someone help me set up a limitation for my wife and kids not to be
able to talk for more than 5 min at a time!
or like 20 min per week! or whtever limitation i could set for this!


Set(TIMEOUT(absolute)=seconds)

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


Terve,
Stefan

--
Last words of a stormchaser:
Where is that rotation on the radar?!


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Re: [asterisk-users] 5 min limitation on phone calls! how to!

[mgraves]

You're not kidding. I would have to investigate cheaper routing.
Truncating my wife's calls would be met with harsh reaction at best.
Maybe painful, too.

Michael Graves
mgraves at mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
FWD 54245


  Original Message 
 Subject: Re: [asterisk-users] 5 min limitation on phone calls! how to!
 From: Singer XJ Wang [EMAIL PROTECTED]
 Date: Thu, August 21, 2008 2:42 pm
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 
 Phone Guy: NO PHONE FOR YOU!
 
 Karl Fife wrote:
  This has got to be one of the funniest threads ever to grace this forum.
  Sorry honey! ...CLICK.
  In my house, this might require a more 'diplomatic' approach :-)
  -Karl 
 
 
 
  On Thu, 21 Aug 2008 21:41:40 +0300, RoLaNd RoLaNd
  [EMAIL PROTECTED] said:

  i tried that before.. it didnt actually work! it the call kept on going
  well beyound the allowed test seconds...
  heres my extensions.conf:
 
 
  [sipura-line]
  exten = 301,1,Answer() ; Answer inbound calls
  exten = 301,2,Playback(silence/1)
  exten = 301,3,Background(simzy1) ; input an extension
  exten = 301,4,WaitExten(8)
  exten = 301,5,Dial(SIP/100,15) ; goes to operator
  exten = 301,4,Wait(8)
  include = spa
  exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
  exten = 301,n,Hangup()
 
 
 
 
  [spa]
  exten =_301,1,GoTo(sipura-line,${EXTEN},1)
  exten = _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds so
  it will ring 3 times
  exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if
  line is busy or unavailable
  exten = _1XX,3,HangUp()
  exten = _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds so
  it will ring 3 times
  exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if
  line is busy or unavailable
  exten = _2XX,3,HangUp()
  exten = _3XX,1,Dial(SIP/${EXTEN},15) ; each ring equals to 5 seconds so
  it will ring 3 times
  exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if
  line is busy or unavailable
  exten = _3XX,3,HangUp()
  exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
  exten =_01,2,Set(TIMEOUT(absolute)=5)
  exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
  exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer
  exten =_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte
  exten =_05,1,Dial(SIP/$(EXTEN)@307) ; conference
  exten =_06,1,Dial(SIP/$(EXTEN)@308) ; line 4
  exten = 303,1,VoicemailMain ; voicemail box to be redirected to
 
 
 
 
  
  Date: Thu, 21 Aug 2008 20:26:48 +0300
  From: [EMAIL PROTECTED]
  To: asterisk-users@lists.digium.com
  Subject: Re: [asterisk-users] 5 min limitation on phone calls! how to!
 
  RoLaNd RoLaNd schrieb:

  Hello all!
   
  my last month's phone bill sky rocketed after i setup asterisk with
  softphones all over the house!
 
  could someone help me set up a limitation for my wife and kids not to be
  able to talk for more than 5 min at a time!
  or like 20 min per week! or whtever limitation i could set for this!
  
  Set(TIMEOUT(absolute)=seconds)
 
  http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AbsoluteTimeout
 
 
  Terve,
  Stefan
 
  -- 
  Last words of a stormchaser:
  Where is that rotation on the radar?!
 
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users

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  http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE
  
 
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Re: [asterisk-users] 5 min limitation on phone calls! how to!

[Singer XJ Wang]

Heck, I was going to say I probably be on the sofa that night and the 
next...


[EMAIL PROTECTED] wrote:

You're not kidding. I would have to investigate cheaper routing.
Truncating my wife's calls would be met with harsh reaction at best.
Maybe painful, too.

Michael Graves
mgraves at mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
FWD 54245


  

 Original Message 
Subject: Re: [asterisk-users] 5 min limitation on phone calls! how to!
From: Singer XJ Wang [EMAIL PROTECTED]
Date: Thu, August 21, 2008 2:42 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com


Phone Guy: NO PHONE FOR YOU!

Karl Fife wrote:


This has got to be one of the funniest threads ever to grace this forum.
Sorry honey! ...CLICK.
In my house, this might require a more 'diplomatic' approach :-)
-Karl 




On Thu, 21 Aug 2008 21:41:40 +0300, RoLaNd RoLaNd
[EMAIL PROTECTED] said:
  
  

i tried that before.. it didnt actually work! it the call kept on going
well beyound the allowed test seconds...
heres my extensions.conf:


[sipura-line]
exten = 301,1,Answer() ; Answer inbound calls
exten = 301,2,Playback(silence/1)
exten = 301,3,Background(simzy1) ; input an extension
exten = 301,4,WaitExten(8)
exten = 301,5,Dial(SIP/100,15) ; goes to operator
exten = 301,4,Wait(8)
include = spa
exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
exten = 301,n,Hangup()




[spa]
exten =_301,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds so
it will ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if
line is busy or unavailable
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds so
it will ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if
line is busy or unavailable
exten = _2XX,3,HangUp()
exten = _3XX,1,Dial(SIP/${EXTEN},15) ; each ring equals to 5 seconds so
it will ring 3 times
exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if
line is busy or unavailable
exten = _3XX,3,HangUp()
exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
exten =_01,2,Set(TIMEOUT(absolute)=5)
exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer
exten =_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte
exten =_05,1,Dial(SIP/$(EXTEN)@307) ; conference
exten =_06,1,Dial(SIP/$(EXTEN)@308) ; line 4
exten = 303,1,VoicemailMain ; voicemail box to be redirected to







Date: Thu, 21 Aug 2008 20:26:48 +0300
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] 5 min limitation on phone calls! how to!

RoLaNd RoLaNd schrieb:
  
  

Hello all!
 
my last month's phone bill sky rocketed after i setup asterisk with

softphones all over the house!

could someone help me set up a limitation for my wife and kids not to be
able to talk for more than 5 min at a time!
or like 20 min per week! or whtever limitation i could set for this!



Set(TIMEOUT(absolute)=seconds)

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


Terve,
Stefan

--
Last words of a stormchaser:
Where is that rotation on the radar?!


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http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE



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Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man

[Darren Sessions]

I'd run top on the server to see if the CPU utilization is going  
through the roof. If you use AGI, make sure there aren't any orphaned  
processes consuming resources.


If all else fails on the software side of things, I'd restart the  
server.



_

Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_





On Aug 21, 2008, at 4:03 PM, bilal ghayyad wrote:


Hi All;

My asterisk version is 1.4.19.2 and it contains one digium card of 2  
fxs and 2 fxo ports, it was working great for more than one month  
without any problem.


Suddenly, any call will be done, then voice becoming like robot (or  
sick man), it slow and cutting.


I restarted the machine, but it is the same !!!

I checked the RAM which is 1 GB and I found a lot of space.

Any advise what could be the problem?
Regards
Bilal






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Re: [asterisk-users] DSS1 vs SS7

[Alex Balashov]

Saul Bejarano wrote:

 With SS7 They will have to define point code and stuff like that, it is
 usually granted to carriers which are members of the SS7 network, I have
 not seen a carrier offering SS7 as a home service.
 Go for standard DSS1 which as somebody said will be the equivalent in
 signlaing to an ISDN trunk.

Some carriers now do offer private SS7 instead of ISDN.  But there is 
absolutely no reason why you should be doing this with Asterisk. 
Asterisk-SS7 is quite tenuous at best.  Unless you have some specific 
reason to be using it, don't.

If your reason is TCAP (LNP, LIDB, CNAM), don't bother using Asterisk at 
all.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man

[bilal ghayyad]

Thanks a lot for your kindly help and advise.

1) I did restart for the machine and it stayed the same.
2) If I call to the Asterisk via the PSTN, and the IVR answer and then I enter 
the extension of the IP Phone which is in another country, the voice is nice 
and no problem, but If I call from the same IP Phone that is connected in 
another country, then voice become robot and very slow (sick man).
3) What is the command to know my partitions size, and which partition really 
need to be increased for this problem?
4) What is the command to know the processor utilization (live) while the calls 
are running?

Note: could my asterisk corrupted and need to be re compiled?

Any help?
Regards
Bilal


--- On Thu, 8/21/08, Darren Sessions [EMAIL PROTECTED] wrote:

 From: Darren Sessions [EMAIL PROTECTED]
 Subject: Re: [asterisk-users] Suddenly the voice become like robot (cutting), 
 like sick man
 To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial 
 Discussion asterisk-users@lists.digium.com
 Date: Thursday, August 21, 2008, 6:13 PM
 I'd run top on the server to see if the CPU utilization
 is going  
 through the roof. If you use AGI, make sure there
 aren't any orphaned  
 processes consuming resources.
 
 If all else fails on the software side of things, I'd
 restart the  
 server.
 
 
 _
 
 Darren Sessions
 [EMAIL PROTECTED]
 http://www.darrensessions.com
 _
 
 
 
 
 
 On Aug 21, 2008, at 4:03 PM, bilal ghayyad wrote:
 
  Hi All;
 
  My asterisk version is 1.4.19.2 and it contains one
 digium card of 2  
  fxs and 2 fxo ports, it was working great for more
 than one month  
  without any problem.
 
  Suddenly, any call will be done, then voice becoming
 like robot (or  
  sick man), it slow and cutting.
 
  I restarted the machine, but it is the same !!!
 
  I checked the RAM which is 1 GB and I found a lot of
 space.
 
  Any advise what could be the problem?
  Regards
  Bilal
 
 
 
 
 
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users


  

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Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man

[bilal ghayyad]

Dear Darren;

I discovered that calling from the Asterisk to the IP Phone Extension (like 
calling from mobile to digium and then enter the IP Phone extension, or calling 
from fxs to the IP Phone extension), it goes very good without any problem.

But calling from the same IP Phone to another IP Phone or to any mobile (via 
fxo port) or to the fxs, it cause the problem (voice become very very bad, like 
robot with weak battery or sick man).

Another way for the problem, if I called from another Asterisk PBX to our 
Asterisk PBX (that has the problem) and the call was via IAX, and I was need to 
reach to the IP Phone, then I hear the voice like robot with weak battery.

So, the problem appear if the call originator was IP and not TDM. What could be 
the reason for the problem? No one did any change, I am sure, it suddenly 
become like this.

Any help?
Regards
Bilal


--- On Thu, 8/21/08, Darren Sessions [EMAIL PROTECTED] wrote:

 From: Darren Sessions [EMAIL PROTECTED]
 Subject: Re: [asterisk-users] Suddenly the voice become like robot (cutting), 
 like sick man
 To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial 
 Discussion asterisk-users@lists.digium.com
 Date: Thursday, August 21, 2008, 6:13 PM
 I'd run top on the server to see if the CPU utilization
 is going  
 through the roof. If you use AGI, make sure there
 aren't any orphaned  
 processes consuming resources.
 
 If all else fails on the software side of things, I'd
 restart the  
 server.
 
 
 _
 
 Darren Sessions
 [EMAIL PROTECTED]
 http://www.darrensessions.com
 _
 
 
 
 
 
 On Aug 21, 2008, at 4:03 PM, bilal ghayyad wrote:
 
  Hi All;
 
  My asterisk version is 1.4.19.2 and it contains one
 digium card of 2  
  fxs and 2 fxo ports, it was working great for more
 than one month  
  without any problem.
 
  Suddenly, any call will be done, then voice becoming
 like robot (or  
  sick man), it slow and cutting.
 
  I restarted the machine, but it is the same !!!
 
  I checked the RAM which is 1 GB and I found a lot of
 space.
 
  Any advise what could be the problem?
  Regards
  Bilal
 
 
 
 
 
 
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Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man

[bilal ghayyad]

Dear All;

I start beleive that if I did asterisk compilation again, then the problem 
might be removed as I start beleive it is related to corrupt happened in some 
files.

The questions here are:

1) Which could these files?
2) How can I know the reason for the corrupt?
3) I have another Asterisks running same version of this machine, can I copy 
files from these other asterisk to that machine that has a problem? Or I have 
to recompile?

I decided not to recompile before hearing from the list the advise.
Regards
Bilal





--- On Thu, 8/21/08, Darren Sessions [EMAIL PROTECTED] wrote:

 From: Darren Sessions [EMAIL PROTECTED]
 Subject: Re: [asterisk-users] Suddenly the voice become like robot (cutting), 
 like sick man
 To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial 
 Discussion asterisk-users@lists.digium.com
 Date: Thursday, August 21, 2008, 6:13 PM
 I'd run top on the server to see if the CPU utilization
 is going  
 through the roof. If you use AGI, make sure there
 aren't any orphaned  
 processes consuming resources.
 
 If all else fails on the software side of things, I'd
 restart the  
 server.
 
 
 _
 
 Darren Sessions
 [EMAIL PROTECTED]
 http://www.darrensessions.com
 _
 
 
 
 
 
 On Aug 21, 2008, at 4:03 PM, bilal ghayyad wrote:
 
  Hi All;
 
  My asterisk version is 1.4.19.2 and it contains one
 digium card of 2  
  fxs and 2 fxo ports, it was working great for more
 than one month  
  without any problem.
 
  Suddenly, any call will be done, then voice becoming
 like robot (or  
  sick man), it slow and cutting.
 
  I restarted the machine, but it is the same !!!
 
  I checked the RAM which is 1 GB and I found a lot of
 space.
 
  Any advise what could be the problem?
  Regards
  Bilal
 
 
 
 
 
 
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[asterisk-users] How to block incoming calls on PRI

[Egbert]

Hi,

I have several asterisk pstn gateways running, each with at least 2 e1 
pri circuits connected. I wonder if there's a way to block incoming 
calls on the pri's, in such way that my telco sends the call to one of 
the other pri's (all the pri's are together in a 'hunt group', calls get 
evenly distributed).
The cli command 'stop gracefully' did not work. Calls send to that 
gateway díd get blocked, but I heard a message from my telco saying 'the 
number you have dialed is not in use', instead of getting connnected to 
the next gateway.
I read some old discussion regarding the same, or related issue, but did 
not see any results.  ('How to busy out PRI channels?' on asterisk-dev, 
end 2006)


thanks,
Egbert Groot.

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[asterisk-users] The problem of the fxs

[larry]

HI

   Here  is a question about the fxs of the zaptel card which is set a
number to use in the inter as common analog phone. When I also use
${CALLERID(num)}to get it's number, it also could not be done. At this time
,the fxs phone does not get any relation with the outbound which is like
PSTN and so forth. It just set the phone number and extension in the * for
inter used . Could you tell me the reason, and how I could get the number of
the fxs?

  Thanks

  Larry 

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Re: [asterisk-users] How to block incoming calls on PRI

[Dwayne Hubbard]

- Egbert [EMAIL PROTECTED] wrote:

 I have several asterisk pstn gateways running, each with at least 2 e1
 pri circuits connected. I wonder if there's a way to block incoming 
 calls on the pri's, 

I have been working on this functionality and have development branches that 
are ready for testing before they get committed to trunk.  I'd love to hear 
some feedback if you can take the time to test the branches.  Here are the 
links to the branches:

libpri - http://svn.digium.com/svn/libpri/team/dhubbard/issue3450_to_commit
Asterisk - http://svn.digium.com/svn/asterisk/team/group/issue3450

you will need DAHDI:

http://svn.digium.com/svn/dahdi/linux/trunk
http://svn.digium.com/svn/dahdi/tools/trunk

If you test the branches, please provide feedback via 
http://bugs.digium.com/view.php?id=3450

I also want to reiterate that the libpri and Asterisk branches above are 
development branches, so be careful in a production environment.  This 
functionality will be available in Asterisk 1.6.2.  To disable a channel via 
the CLI type 'pri service disable channel chan' and to enable the channel 
type 'pri service enable channel chan'.

good luck!

Dwayne Hubbard


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[asterisk-users] Linksys - Sipura VMWI splash ring

[Joseph]

I'm trying to configure Linksys 3102 for a short splash ring when someone 
leaves a message.
in my sip.conf I have
mailbox=number

I have can see a visual indicator (light blinking on the phone) but there is no 
short splash ring)

Linksys setting:

Regional - tab
Ring and Call Waiting Tone Spec
Ring Waveform: Trapezoids   Ring Frequency: 25
VMWI Refresh Intvl: 30  (was 0 I changed to 30 makes no difference)

User 1 - tab

Ring Settings:
VMWI Ring Splash Len: 0.5

Did I miss any settings? Why isn't it working?

-- 
#Joseph

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[asterisk-users] Problem with modem data calls and xorcom astribanks

[Col Ferguson]

Hello all,
I have a system at a motel that is mostly analog phones with 2 32 port
astribanks.

I am having problems getting a modem data call to connect.
There are many travelling salesmen that require this functionality to work
to dial direct into their company systems.

I am using Asterisk 1.4.18.1, and Zaptel 1.4.9.2 and freePBX 2.4.0.1 and
Oslec echo can.

I have now the simplest dialplan I can come up with and get a 4800
connection about 1 in 10 times. This should bypass any smarts that freePBX
is adding in.

The dialplan is
[outbound-allroutes-custom]
exten = 791,1,Dial(Zap/69/ww019830,300)
exten = 791,n,Hangup

In Hyperterminal I do
atdt791

The number dialled is for a large dialup ISP.
ww is needed to get a dialtone for the modem. Could this be causing the
problem ?

The log file shows
[Aug 22 13:17:32] DEBUG[748] chan_zap.c: Deferring dialing...
[Aug 22 13:17:32] VERBOSE[748] logger.c: -- Called 69/ww019830
[Aug 22 13:17:36] VERBOSE[748] logger.c: -- Zap/69-1 answered Zap/67-1
[Aug 22 13:17:36] DEBUG[748] chan_zap.c: Took Zap/67-1 off hook
[Aug 22 13:17:36] DEBUG[748] chan_zap.c: master: 67, slave: 69, nothingok: 0
[Aug 22 13:17:36] DEBUG[748] chan_zap.c: Stopping tones on 67/0 talking to
69/0
[Aug 22 13:17:36] DEBUG[748] chan_zap.c: Stopping tones on 69/0 talking to
67/0
[Aug 22 13:17:36] DEBUG[748] chan_zap.c: Making 69 slave to master 67 at 0
[Aug 22 13:17:36] DEBUG[748] chan_zap.c: Added 18 to conference 9/67
[Aug 22 13:17:36] DEBUG[748] chan_zap.c: Added 76 to conference 9/69
[Aug 22 13:17:36] VERBOSE[748] logger.c: -- Native bridging Zap/67-1 and
Zap/69-1
[Aug 22 13:20:00] DEBUG[748] chan_zap.c: Unlinking slave 69 from 67
[Aug 22 13:20:00] DEBUG[748] chan_zap.c: Removed 18 from conference 9/67
[Aug 22 13:20:00] DEBUG[748] chan_zap.c: Removed 76 from conference 9/69
[Aug 22 13:20:00] VERBOSE[748] logger.c: -- Hungup 'Zap/69-1'

Does anyone know if there is some type of native echo canceller in the
astribanks that could be affecting this ? Or anything else I could try ?
Looking at /proc/oslec/info shows that oslec is not being used at the time.

If I have the modem connected directly into the phone line, and completely
bypass the astribank, I get a 50666 connection every time.

Any suggestions gratefully accepted.

Thanks,
Col


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