Re: [asterisk-users] Register Sip extension with out Sip phone

[$$ dave cantera (android asus)]

this is an interesting project, SIP protocol is easy to find, writing a php 
script, perl script, or python would probably work. it would probably work 
better if it was a daemon. what would be connecting to it that you would need a 
SIP connection for?...  interesting...

Dave Cantera
(856)813-7098 mobile/txt
david.cant...@ibsonecall.com

Sent from my ASUS Pad

akhilesh chand  wrote:

>Dear all,
>
>I have two system Sys A and Sys X.
>
>Sys A is normal PC.
>
>Sys X have installed asterisk 1.6 and i want register(or reserved)  sip
>extension(like 4001,4002,4003..)  through Sys A(Sys A have some ip address)
>but i don't use any soft-phone means i want to write Perl or php(any
>language)  script to register sip extension.
>
>Suppose to 4001 is reserved  with Sys A.
>4002 is reserved with Sys B.
>
>
>
>
>
>
>Regards
>Akhilesh
>
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Re: [asterisk-users] Asterisk TON number

[$$ dave cantera (android asus)]

if you are a serious teleco guy, which it seems you are. you might consider 
dumping trixbox in the near future. while trixbox does provide a good entry 
level into the * world, there are limitations that will eventually hold you 
back from enjoying the full breadth of utility that * offers.
food for thought,

Dave Cantera
(856)813-7098 mobile/txt
david.cant...@ibsonecall.com

Sent from my ASUS Pad

Steve Totaro  wrote:

>On Wed, Sep 25, 2013 at 3:22 AM, Endri Stefani wrote:
>
>>  Hi
>>
>> ** **
>>
>> Greeting to all you out there.
>>
>> ** **
>>
>> I am new at asterisk, I have been working with PLMN platforms
>> telecommunication for 5 years with NSN and Huawei.
>>
>> We have recently built an asterisk PBX with Trixbox and connected it to
>> our MSS using Digium E1 cards(ISDN). Everything went smoothly as there are
>> tons of information out there, except for the TON number.
>>
>> If you have worked in Telecommunication you will know the importance of
>> TON flexibility. 
>>
>> All the posts online suggested to update under Chan_dahdi.conf:
>>
>> pridialplan = international
>>
>> prilocaldialplan = international
>>
>> or other TON value ,restart the platform and then trixbox1*CLI> dialplan
>> reload
>>
>> I have already done this with no success. Are there other changes I have
>> to make in order to change dialplan?
>>
>> ** **
>>
>> ** **
>>
>> Br
>>
>> **
>>
>
>So what are you trying to do specifically?
>
>Thanks,
>Steve Totaro
>
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Re: [asterisk-users] test call generator

[|| dave cantera Mobile]

dan, elder,
I have played with scripts to generate calls and track their 
completion,  email me off-list if you have questions.

daveC


Daniel - Asterisk wrote:

Hello Everyone,

I wonder if someone could share a manual about using SIPp for 
Asterisk's testing.


I'll be gratefull


Regards,

Elder Arohuanca
Lima - Peru

On Tue, Sep 30, 2008 at 12:59 PM, zac wolfe > wrote:


Sipp looks pretty good! I don't know how I missed this one.  This
would've saved me tons of time a couple months ago.

I plan on using it to load test using 2 Asterisk servers, one to
initiate the SIP calls, the other to receive. Thanks for the tip Alex.

Zac Wolfe
Safi Systems LLC
www.safisystems.com 


On Sat, Sep 27, 2008 at 5:58 AM, Alex Balashov
mailto:abalas...@evaristesys.com>> wrote:

What you are looking for is SIPP:   http://sipp.sourceforge.net/

It won't intrinsically tell you anything about the data;  it's
up to you
to appropriate the findings.  But it accomplishes the
generation of
traffic (and dummy media!) on a technical level.

Igor Hernandez wrote:

> Sam Tam wrote:
>> Hello everyone
>>
>>
>>
>> I am trying to look for a free test call generator that
will get me some
>> stats like PDD, ASR and call quality etc on each route. As
well as do
>> test at every interval too
>>
>>
>> If you know something like this please enlighten me.
>>
>> Sam
>>
>>
>>

>>
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>
> Hey Sam,
>
> I've been looking for such a tool also. I can't seem to find
a tool that
> does those things.
>
> If nothing comes up in the next couple of weeks I'm going to
code
> something up, I wouldn't mind letting you and anyone else
who might be
> interested have the source once its done.
>
> Let me know if you find anything thats already out there in the
> meantime, might just save me a few hours of work.
>
> Regards,
>
>


--
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Re: [asterisk-users] Asterisk 1.8.4 Now Available

[|| dave cantera Mobile]

danny,
not that it matters, but I agree. if the design is a good design, it 
would not have to be redesigned on every release.  in fact, the modules 
template should also follow this philosophy that way you can concentrate 
on adding functions and not the design...


sometimes, it is smarter to scrap what you have and use the knowledge 
gained to come up with a design that provides for a good, tightly 
integrated, but flexible facility to interact with multiple components.  
for example, the admin interface on * is just such of a system. you can 
add commands as required (I haven't looked at the code to see if it is 
easy to do though). 

the more time you put into design, the less time you have to spend 
programming.

IMHO,
daveC

Danny Nicholas wrote:
 
[Danny Nicholas]  
Paul, this is probably a "dumb question", but why are some (or is it all and

I just don't notice it) modules "fundamentally changed" from release to
release (or version to version)?  As a C-dabbler, it seems to me that if I
do gcc app_voicemail.c (using voicemail as an example) on 1.6 or 1.8, it
should be fundamentally similar to doing the same thing in 1.4.  It seems to
me that the 9500 line module that compiles and runs in 1.4 should be pretty
much the same as what is in 1.6 or 1.8, but on closer inspection it is often
not.  I know that some things have to change to add the new enhanced
functionality, but from a "dinosaur programmer" perspective, the fewer
background changes you have from release to release and version to version,
the less chance you have of something breaking and having unhappy users.

Please elaborate on what is wrong with my thought.

Thanks in advance.


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Re: [asterisk-users] best current version and motherboard/CPU compatibilities

[|| dave cantera Mobile]

paul, doug,
I had several AMD athlons 64bit... no problems running centos, suse.  
they seem  solid on 1.4.xx... had a few intel celerons and P4s. they 
were good as well. guess I was Lucky back then!

thanks for supporting the list!
daveC

Paul Hayes wrote:

On 04/05/11 17:10, || dave cantera Mobile wrote:

doug,
why are you shaking!?!?... do you have a better recommendation?
daveC



AMD K6 CPU brings back some pretty bad memories from me too.


Doug Lytle wrote:

C F wrote:

model name : AMD-K6(tm) 3D processor


*shudder*

Doug





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david.cant...@ibsonecall.com
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Office (856)324-4488
Pers Fax (646)827-7108
Ofc Fax (888)487-7711


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(856)581-8971

Home of the Videophone2009.com


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Re: [asterisk-users] best current version and motherboard/CPU compatibilities

[|| dave cantera Mobile]

doug,
why are you shaking!?!?... do you have a better recommendation?
daveC

Doug Lytle wrote:

C F wrote:

model name  : AMD-K6(tm) 3D processor
   


*shudder*

Doug



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Want to invest in Real Estate?
come out and join over 450 real estate investors
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david.cant...@ibsonecall.com
Mobile (856)813-7098
Office (856)324-4488
Pers Fax (646)827-7108
Ofc Fax (888)487-7711


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(856)581-8971

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[asterisk-users] best current version and motherboard/CPU compatibilities

[|| dave cantera Mobile]

I've been away from asterisk for a while since 1.4.16 and only installed 
1.6 once to run a test... can someone recommend what the best version to 
install is and the recommended CPU/motherboard for an * box these days? 
I'm just running about 20 handsets and 4-8 lines with POTS & SIP mix.


I remember there were some issues with bios a while back and a TDM card 
was required for timing conferencing, etc... are these requirements 
still an issue?


I want to setup another * box and was wondering which CPU/motherboard to 
select...

thanks,
daveC

--
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Want to invest in Real Estate?
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david.cant...@ibsonecall.com
Mobile (856)813-7098
Office (856)324-4488
Pers Fax (646)827-7108
Ofc Fax (888)487-7711


Interlocking Business Solutions, LLC
david.cant...@ibsonecall.com
(856)581-8971

Home of the Videophone2009.com


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Re: [asterisk-users] [asterisk-biz] New York Asterisk Users

[| dave cantera |]

dean,
I am an active member of AUG NYC... you can email me off list for any
info you need.

also, I am preparing to start a south jersey * UG.  the phila group is
waning...

thanks,
daveC

Dean Collins wrote:

  
  
  
  
  
  
  
  This is an email to all New
York
based Asterisk users.
   
  For some time it’s been
bugging me that we don’t
have a local contact point/user community. If you are involved in
Asterisk and
in NY/NJ shoot me an email, I’m going to try and revitalize either
meetup.com or some other shared environment for Asterisk users in NY.
   
  Shoot me an email and
once I get an idea of how many
Asterisk users there are in NY we’ll work out what to do from there.
   
   
  
  
Cheers,
Dean 
   
  
  

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Re: [asterisk-users] Need good voicemail documentation

[dave cantera]

jaap,
found this some time ago... might do the trick...
daveC
http://www.venturevoip.com/vm.pdf



Jaap Winius wrote:

  Hi list,

After wrestling with the voicemail system for a while (Asterisk  
1.4.14, Debian etch), I got it to work, but I still have lots of  
questions, like:

 * Why can't I delete any voicemail messages?
   (Response: "Message undeleted.")
 * Why can't I listen to the messages in the Old folder?
 * Why can't I use the advanced options?
   (Response: "I'm sorry, I did not understand your response.")
 * How come if I put "[EMAIL PROTECTED]" in my phone's
   context of sip.conf, do I get an error?
   (CLI: "...Remote host can't match request NOTIFY to call...")

Unfortunately, none of the books and other documentation I've found on  
the subject goes into enough detail to provide answers to such  
questions. So, can anyone recommend some good Asterisk voicemail  
documentation that goes beyond merely scratching the surface?

Cheers,

Jaap

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Re: [asterisk-users] SIPAddHeader in .call file

[dave cantera]

steve,
thanks for posting this tidbit!
daveC

Steve Johnson wrote:

  Sorry to answer my own post, but I have found a solution which perhaps
others can use too...

In the .call file, instead of specifying a channel line as:

  chan: SIP/140  (for example)

use instead:

  chan: Local/[EMAIL PROTECTED]

and put in extensions.conf

[polycom-paging]
exten => _1XX,1,NoOp(Paging Ext ${EXTEN})
exten => _1XX,n,SIPAddHeader(Alert-Info: Ring Answer)
exten => _1XX,n,Dial(SIP/${EXTEN},20,L(6))
exten => _1XX,n,Hangup


Steve Johnson wrote:
  
  
Hi everyone,

How can I add the equivalent of:

   exten => s,n,SIPAddHeader(Alert-Info: Ring Answer)

in a .call file?  This is to support paging to Polycom phones...

Thanks for all info!

Steve


  
  
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Re: [asterisk-users] Asterisk RFC2833 to SIP INFO DTMF conversion erros.

[dave cantera]

mayur,
did you try inband? with sip?
daveC
;dtmfmode=inband    ; Choices are inband, rfc2833, or
info
;allow=ulaw ; dtmfmode=inband only works with ulaw
or alaw!
;dtmfmode=rfc2833   ; Choices are inband, rfc2833, or info
;allow=ulaw ; dtmfmode=inband only works with ulaw
or alaw!

Mayur wrote:

  
  
  
  
  Hi,
     I am using asterisk
1.4.17 which is connected to a SIP trunk
supporting rfc2833 dtmf events. Asterisk stays in the media path. In
sip.conf I
have set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk
account) and
for SIP clients I have set dtmfmode=info. So when I make a call to a
cell
number using the sip trunk and then press digits I can see the 2833
dtmf events
coming to asterisk in the rtp captures. Asterisk seems to detect those
and give
SIP INFO to the SIP client. However it fails to detect some of the
digits (which
is random) hence the correct sequence of digits is not received at the
SIP
client.
  I have tried setting
relaxdtmf=yes in sip.conf but that does
not seem to help. Can anyone help me out here?
   
  Regards,
  Mayur
  
  

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Re: [asterisk-users] Asterisk ports and CentOS firewall

[dave cantera]

ed,
this may be somewhat liberal but should do the trick...
daveC
-A RH-Firewall-1-INPUT -p tcp -m tcp --dport 69 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp -m udp --dport 69 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp -m udp --dport 5061 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp -m udp --dport 5062 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp -m udp --dport 4569 -j ACCEPT
-A RH-Firewall-1-INPUT -p tcp -m tcp --dport 5038 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp -m udp --dport 5036 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp -m udp --dport 1:2 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp -m udp --dport 5004 -j ACCEPT
#
-A RH-Firewall-1-INPUT -p icmp -m icmp --icmp-type any -j ACCEPT
-A RH-Firewall-1-INPUT -p ipv6-crypt -j REJECT
-A RH-Firewall-1-INPUT -p ipv6-auth -j REJECT
-A RH-Firewall-1-INPUT -d 224.0.0.251 -p udp -m udp --dport 5353 -j
ACCEPT
-A RH-Firewall-1-INPUT -p udp -m udp --dport 631 -j ACCEPT
-A RH-Firewall-1-INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT
-A RH-Firewall-1-INPUT -p tcp -m state --state NEW -m tcp --dport 22 -j
ACCEPT
-A RH-Firewall-1-INPUT -p tcp -m state --state NEW -m tcp --dport 80 -j
ACCEPT
-A RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited



Ed Nunez wrote:

  
  
  

  
  If
I enable the firewall on my Server, which ports should I open
for Asterisk to work properly.  Is it enough to just open the SIP ports?
  
  

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Re: [asterisk-users] :POSSIBLE SPAM: Re: conferencing help

[dave cantera]

steve,
thanks for pointing that out, I forgot the exact reason.  
as for the hearing/audio problem...  if all else works the conferencing
should also... I haven't used freepbx, do they handle the port
filtering?

   # tcpdump -i eth0 udp 

should show if the packets are getting in/out...

I have no experience with sangoma cards.
daveC

Steve Edwards wrote:

  
    dave cantera wrote:


  nhadie,
meetme requires a zaptel timing device... ztdummy is unreliable when
using meetme conferencing.
  

  
  
On Wed, 9 Jan 2008, Nhadie wrote:

  
  
hi dave thank you for the reply. i have loaded zap and using only
ztdummy but still can't hear anything when i dial ti my conference, i
think this explains it already. will a sangoma card do?

  
  
I use ztdummy with meetme conferencing and it works fine on CentOS 4.5. 
Ztdummy is not an issue until you get xx callers in xx conferences.

I think (but have no empirical data to back it up) that a card yields 
better sound quality at higher call levels.

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] tale of two firewalls

[dave cantera]

robert,
with limited info below, are you port forwarding on the router with the
public IP, ports 10,000-20,000, 5004, along with 5060?  and the other
router (internal, I assume)???

how do you have two firewalls configured with one * box?
do you have captures on both sides of the internal (I assume)
router?...  
if you want to call me, just send me an email... will be available till
11p EST.
daveC

Robert Moskowitz wrote:

  I have a server behind a firewall.  It is publicly addressed.  Should 
NOT be trying to NAT (how would I know).

The connection is a SIP trunk to Broadvoice.  I am calling the 
Broadvoice # from my cell and the call is being routed to my server.

With one firewall the INVITE contains information for the RTP session to 
be with broadvoice servers with different addresses.  It works.

With the other firewall the INVITE does NOT contain any other IP 
addresses, and the call goes through but no voice (duh).

I have captures of both.  I would include them in this message, but I am 
a little concerned that if some of you get the Registration, you will 
crack my secret...

PLEASE help me out on this.  I am absolutely pulling out my hair on it 
(and I don't have much).  I have stared at the Wireshark displays and 
just don't see it.  I have turned on logging on the failing firewall, 
and am not seeing any messages being dropped or rejected.



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Re: [asterisk-users] GotoIf() help

[dave cantera]

glenn,
what an interesting way to use GotoIf() and 9.  didn't know you
could do that in GotoIf()!
you could have used (broken out) the individual services 
[trunklocal]
[trunkld]
[trunktollfree]
and just included the above individual context in with the groups that
you allowed a particular class of service to ...
daveC

Glenn Cobb wrote:

  
  
  Greetings all,
   
  I'm not real good with dial
plan programming and need some help. I've looked at the 2nd edition of
the Asterisk book about GotoIf() and have a basic idea what I need to
do but not sure about the correct way or the best way, to set it up. I
need to branch based on whether the dialed number is long distance
(international or not) or not. I have branch offices on SIP and IAX
trunks that have 4 digit extensions and one office has a 1000 range for
their extensions so I have to make sure I don't pick that up as dialing
long distance. I think what I have below will work but it can probably
be cleaned up alot. Any help is greatly appreciated.
   
   
  exten =>
s,n,GotoIf($[${DIAL_NUMBER} = 011. ] ? yescode : steptwo)
   
  exten =>
s,n,(steptwo),GotoIf($[${DIAL_NUMBER} = 9XX. ] ? yescode :
stepthree)
   
  exten =>
s,n,(stepthree),GotoIf($[${DIAL_NUMBER} = 1NXXNX. ] ? yescode : nocode)
  
  exten => s,n,(yescode),Playback(please-enter-the&accounting)
  exten => s,n,Read(account|number|8)
  exten => s,n,SetAccount(${account})
  exten =>
s,n,(nocode),Blah, Blah
  
   
  Thanks,
   
  Glenn
  

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Re: [asterisk-users] :POSSIBLE SPAM: Re: :POSSIBLE SPAM: conferencing help

[dave cantera]

nhadie,
meetme requires a zaptel timing device... ztdummy is unreliable when 
using meetme conferencing... I suggest you spend time elsewhere in * 
until you get a digium tdm400 w/ or w/o any daughter modules...  you 
just need the board for the timing device you don't actually need any 
modules. $195 for tdm400p + one mondule.. developers kit...
daveC

Nhadie wrote:
> hi shane,
>
> thanks for your reply. i actually tried 3 phones dialled to the 
> conference, but cant here anything from those phones. i also enabled the 
> usercount so i can hear something at least. but still no sound.
> i'm using ztdummy, as i dont have a card yet.
>
> regards,
> nhadie
>
> Shane D wrote:
>   
>> Wouldn't you need someone besides yourself in the conference?
>>
>> On 1/7/08, Nhadie <[EMAIL PROTECTED]> wrote:
>> 
>>> Hi All,
>>>
>>> kind of need help on the conference module, i'm using freepbx and
>>> enabled conferencing, i created a conference number, 6000. when i dial
>>> to it, my phone says it is connected but i'm hearing nothing, maybe logs
>>> below can help you.
>>>
>>> also, when i hang up the phone, the conference did not disconnect me.
>>> how can i end a conference? thank you
>>>
>>>  -- Executing Macro("SIP/104-519e", "user-callerid|") in new stack
>>>  -- Executing NoOp("SIP/104-519e", "user-callerid: device 104") in
>>> new stack
>>>  -- Executing Set("SIP/104-519e", "AMPUSER=104") in new stack
>>>  -- Executing GotoIf("SIP/104-519e", "0?report") in new stack
>>>  -- Executing GotoIf("SIP/104-519e", "0?start") in new stack
>>>  -- Executing Set("SIP/104-519e", "REALCALLERIDNUM=104") in new stack
>>>  -- Executing NoOp("SIP/104-519e", "REALCALLERIDNUM is 104") in new
>>> stack
>>>  -- Executing Set("SIP/104-519e", "AMPUSER=104") in new stack
>>>  -- Executing Set("SIP/104-519e", "AMPUSERCIDNAME=104") in new stack
>>>  -- Executing GotoIf("SIP/104-519e", "0?report") in new stack
>>>  -- Executing Set("SIP/104-519e", "AMPUSERCID=104") in new stack
>>>  -- Executing Set("SIP/104-519e", "CALLERID(all)="104" <104>") in
>>> new stack
>>>  -- Executing Set("SIP/104-519e", "REALCALLERIDNUM=104") in new stack
>>>  -- Executing NoOp("SIP/104-519e", "TTL:  ARG1: ") in new stack
>>>  -- Executing GotoIf("SIP/104-519e", "0?continue") in new stack
>>>  -- Executing Set("SIP/104-519e", "__TTL=64") in new stack
>>>  -- Executing GotoIf("SIP/104-519e", "1?continue") in new stack
>>>  -- Goto (macro-user-callerid,s,23)
>>>  -- Executing NoOp("SIP/104-519e", "Using CallerID "104" <104>") in
>>> new stack
>>>  -- Executing Set("SIP/104-519e", "MEETME_ROOMNUM=6000") in new stack
>>>  -- Executing GotoIf("SIP/104-519e", "0?USER") in new stack
>>>  -- Executing Answer("SIP/104-519e", "") in new stack
>>>  -- Executing Wait("SIP/104-519e", "1") in new stack
>>>  -- Executing Set("SIP/104-519e", "MEETME_OPTS=") in new stack
>>>  -- Executing Goto("SIP/104-519e", "STARTMEETME|1") in new stack
>>>  -- Goto (from-internal,STARTMEETME,1)
>>>  -- Executing MeetMe("SIP/104-519e", "6000||") in new stack
>>>
>>>
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>> 
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Re: [asterisk-users] pickup application failed

[dave cantera]

rilawich,
do you have the pickup group defined?
http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups
daveC

Rilawich Ango wrote:
> Below is what I got from CLI
> [Jan  7 23:02:46] NOTICE[3450]: app_directed_pickup.c:159 pickup_exec:
> No target channel found for 111.
>
> On Jan 7, 2008 11:48 PM, Rilawich Ango <[EMAIL PROTECTED]> wrote:
>   
>> I have a TDM400 in the server.  I want to press **1XX to pickup a
>> call.  It is ok if I pickup a call dialled from 1XX to 1YY (internal
>> network call).  However, it is failed to pick up a call from PSTN
>> thro' TDM400 card.  It seems I can't guess the correct context of it.
>> How can I know the context of  the call using CLI?  The default
>> context of the TDM400 is from-pstn but pickup still failed if I add
>> exten => _**1XX,n,Pickup(${EXTEN:[EMAIL PROTECTED]) in the end of context
>> BLF_group_pickup.
>>
>> [BLF_group_pickup]
>> exten => _**1XX,1,Pickup(${EXTEN:[EMAIL PROTECTED])
>> exten => _**1XX,n,Pickup(${EXTEN:[EMAIL PROTECTED])
>>
>> 
>
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[asterisk-users] Polycom IP phones that are brick'd

[dave cantera]

I brick'd two of my polycom phones  trying to get the shoretel phones 
working with *...
does anyone have the equipment to unbrick them?

there is a jtag serial cable that is needed along with the knowledge of 
embedded systems..
that is all I currently know.

polycom wants $180 and 30 days to fix them.
daveC

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Re: [asterisk-users] Asterisk content @ OSCON 2008?

[dave cantera]

brian,
cool, I attended one of you tutorials in baltimore... it was Great!  
would go again because I know I would learn even more this time after 
being exposed to it in greater detail... I could absorb more this time...
daveC



Brian Capouch wrote:
> Martin Smith wrote:
>   
>> Hey folks,
>>
>> Is anyone working on Asterisk (or other) presentation proposals for
>> OSCON 2008 in Portland, OR? Here's the link, in case:
>> http://en.oreilly.com/oscon2008/public/cfp/13
>>
>> 
>
> I'm working on a proposal to do a tutorial there on Asterisk.  I have 
> done such the past three years.
>
> I'm also proposing a session on embedded Asterisk, under openWRT, which 
> is my little fringe niche of the world. . .
>
> B.
>
>   

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Re: [asterisk-users] Using Asterisc for Taking Calls for Radio

[dave cantera]

shane, et. al.
shouldn't the console/dsp work?  I have a handset, let me get it  no 
markings on it, has jacks to plug into your sound card for audio in and 
audio out.. it worked on my laptop with asterisk or maybe a softphone... 
  if your sound board has an audio in/out that might work directly... 
I'm not an electronics guy, you may need a pre-amp or something similar 
to step up/down the signal...  even if you had the softphone running on 
the * box, you could have it auto answer and feed the board directly.  
well, maybe auto answer wouldn't be such a great idea... 
I also talked with a radio talk show host who handled 100s of calls per 
night in a 3 hr segment.  they had 4 people screening and staging calls 
to prep the host..  perhaps more than where you are now but, hey, 
someday! :) three people took calls and entered subject/notes into 
ACT of call things to use, then the producer selected and sequenced the 
calls, (read retyped the subj/notes) for display on the host console...  
also other notes were sent in the same way.. quite an interesting layout...
email off list if you want more information...
daveC



Shane D wrote:
> I had thought of that. And sorry for the misspelling of asterisk. For
> some reason, I thought C.
>
> Anyway, I would prefer not to use a softphone, but I could if need be.
>
> I just want a phone number, located who cares where, that will ring my
> asterisk box. My friend has done this successfully for free. Does
> anyone know of a service for this?
>
> On 1/4/08, Don Fanning <[EMAIL PROTECTED]> wrote:
>   
>> Umm... create your dial plan then use a softphone?  You'll have to work
>> out the audio connections in and out of your computer that feeds a audio
>> channel and outputs the monitor back to the computer but it can be done
>> pretty easily.
>>
>> Shane D wrote:
>> 
>>> Hello Asterisc-Users List,
>>>
>>> I am new to the list. I joined with a question in mind: How would you
>>> set up an asterisc box so that:
>>> (A) Someone dials a number
>>> (B) They are presented with a menu
>>> (C) Entering a number, like 1, connects a call to me.
>>> (D) I am on a mixing board, running an internet radio show. I want to
>>> run asterisc into the board, and run an output from the board to
>>> asterisc. Is that possible using a soundcard? I don't really want to
>>> spend money.
>>> (E) I want the board to start wringing when I get a call, and I want
>>> the call audio to the board as well.
>>>
>>> I also would like it if I could not use my local phone line. I would
>>> prefer something like a free internet based number. The box will not
>>> need to be able to call out, so that's not a problem. A friend of mine
>>> uses asterisc, and has a free internet based number for asterisc. I
>>> would like to do the same.
>>>
>>> I hope this is possible, and thanks in advance.
>>>
>>> Shane
>>>   
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Re: [asterisk-users] OT - GEOPRIV and location based SIP services

[dave cantera]

to all,
I had a similar thought... what I came up with was, not my idea just saw 
it done somewhere else, a small windows binary that was exectued on 
login. registered your login name with a server (content filter in that 
case)... any browser requests were logged for filtering and tracking 
purposes. you could just as well match up the username and hostname to 
an on-site database and know where the person was.

I was told the utility was a common open source utility but you'd have 
to be a windoes guy to know it... 'sam' might be the code 
name...maybe... hmmm...
a quick check on google yielded: registry security accounts manager 
(*sam*) hive from the system volume information folder... couldn't find 
any source code though...

daveC



Tzafrir Cohen wrote:
> On Fri, Jan 04, 2008 at 10:09:06AM +0100, Olivier wrote:
>   
>> Hello Dean,
>>
>> 2008/1/3, Dean Collins <[EMAIL PROTECTED]>:
>> 
>>>  Can you provide more details on what you are trying to do. Your
>>> explanation is a bit confusing – sounds interesting but just want to make
>>> sure I have your idea right.
>>>
>>>   
>> Please, apologize for not being very clear (side effects of new year eve).
>>
>> Here is what I'm trying to do :
>>
>> - I want to develop a web application offering click2call services : you
>> browse directory listing and call a contact simply clicking on its phone
>> number or extension.
>> - when a user log, I want the application to guess where the user is located
>> (from office, home, elsewhere).
>> - knowing user location, web application would be able to guess whether it
>> should preferably use your hardphone (when in the office) or your softphone
>> (anywhere else).
>> 
>
> I suspect that a manual list of IP address ranges for "work" (maing the rest
> "home" would be simpler.
>
> That information is not available from geographic location anyway.
>
>   
> 
>
> No virus found in this incoming message.
> Checked by AVG Free Edition. 
> Version: 7.5.516 / Virus Database: 269.17.13/1208 - Release Date: 01/03/2008 
> 03:52 PM
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Re: [asterisk-users] A thougt

[dave cantera]

dean, fredrik,
when I installed skype, ugh, it asked me if I wanted to link phone 
numbers on the web page to be click2dial... I did it and every phone 
number on a web page was a link... I ended up turning it off... it was 
too annoying...  so there are some plug-ins out there that can do that 
sort of thing...
daveC

Dean Collins wrote:
> I think Snapanumber might be what you are looking for.
>
> Regards,
>
> Dean Collins
> Cognation Pty Ltd
> [EMAIL PROTECTED]
> +1-212-203-4357 Ph
> +61-2-9016-5642 (Sydney in-dial).
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Fredrik Söderlund
>> Sent: Thursday, 3 January 2008 2:44 PM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] A thougt
>>
>> Is there any possibilletys to klick on
>> a telephone nr an it will dail like the case in a mail program if you
>> klick a
>> url://a.b.se it opens a browser
>> and in this case it would open a dailplane ??
>> Is there sucha thing ?
>>
>> Asking just out of curisoty
>>
>> /Fredrik Söderlund
>>
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Re: [asterisk-users] AGI stream file

[dave cantera]

tim,
I found exactly the same thing... hit or miss... I'm using php. 
daveC

Timothy Legge wrote:
> Hi
>
> I have created a rudimentary perl script that does most of what I want
> but occasionally in seems that a file will not play.  I see the
> message getting sent to Asterisk but no reply to say that it
> completed.  In fact, the very next SAY DATE command works and
> everything after that but the previous message seems to be ignored.
>
> In addition, I found in one case that I had to read to STDIN
> immediately before sending a message to get it to work even though
> previous requests had already read the STDIN.
>
> Any ideas?
>
> Tim
>
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Re: [asterisk-users] Asterisk dialplan date and time operations

[dave cantera]

erik,
cool, there are so many functions in *, it takes a long time to learn 
the shortcuts!
thanks erik!
daveC


Erik Wartusch wrote:
> daveC,
>
> No it's even simpler. ( I dont need an IF case)
> I just want to add e.g. 15 minutes to the current date / time:
>
> So simply said:
>
> ${STRFTIME(${EPOCH},,%Y%m%d%H%M)} + 15 minutes!
>
> My question was how can I do that.? Of yourse e.g. if it's 23.57 pm and I add 
> 15 minutes the day should increase +1 and the hours start with 0:x the 
> minutes with 12 ( and not 72 as the normal addition would result).
>
> Kind Regards,
>
> Erik
>
>
> Am Mittwoch, 2. Januar 2008 16:02 schrieb dave cantera:
>   
>>  erik,
>>  you can start here:
>>  http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime
>>  http://www.asteriskguru.com/tutorials/gotoiftime.html
>>  daveC
>>
>>  Erik Wartusch wrote:
>> Hi all,
>>
>> Im using Asterisk 1.4.11 and I want to proceed some time and date
>> operations in my dial plan. (for a time shifted callback).
>>
>> Should look like:
>>
>> CURRENT TIME + x minutes.
>>
>> Of course it should increase the hours for example in this case:
>>
>> 10.59 + 5 minutes = 11.04
>>
>> I guess I've to use the math function in 1.4 but how can I manage easily
>> the time operations?
>>
>> Kind Regards,
>>
>> Erik
>>
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>   

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Re: [asterisk-users] Invalid extensions

[dave cantera]

jared,
Happy New Year too!
excellent point!  two brains are always better than one!
daveC

Jared Smith wrote:

  On Wed, 2008-01-02 at 08:27 -0300, Gilberto Nunes Ferreira wrote:
  
  
First I want to wish for everone a happy new year...

  
  
Happy New Year to you as well!

  
  
Well...

I have run asterisk 1.4.16.1 in a server.
I have this IVR, in extensions.conf:

[ura]

;exten => s, 1, Wait,1
exten => s, 1, Answer()
exten => s, n, Noop()
exten => s, n(debug),DumpChan()
exten => s, n, Set(LANGUAGE()=pt_BR)
exten => s, n,
Set(CALLFILENAME=/var/spool/asterisk/monitor/entrada/)
exten => s, n, Set(DYNAMIC_FEATURES =
hangup#pickupexten#atxfer#blidxfer)
exten => s, n,
MixMonitor(${CALLFILENAME}/${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)}-${EXTEN}.wav|v(0)V(0))
exten => s, n, Wait,1
exten => s, n, Set(TIMEOUT(digit)=3) ; Set Digit
Timeout to 5sec
exten => s, n, Set(TIMEOUT(response)=5) ; Set Response
Timeout to 10sec
exten => s, n, Background(bemvindobit)
exten => s, n, WaitExten
exten => 1, 1, Set(LANGUAGE()=pt_BR)
exten => 1, 2, Queue(8600|tT|||30)
exten => 2, 1, Set(LANGUAGE()=pt_BR)
exten => 2, 2, Queue(8500|tT|||30)
exten => i, 1, Wait,1
exten => i, 2, Playback(invalid)
exten => i, 3, Goto(s,11)
exten => t, 1, Wait,1
exten => t, 2, Dial(SIP/8024,150,tTrwW)
exten => s, n, HangUp()

  
  
The 'i' extension above would probaby work, except I think the extra
whitespace you've added may be causing Asterisk not to work.  Try taking
out the extra space before the priority number and the application name,
like this:

exten => i,1,Wait,1

You can also see if Asterisk is correctly parsing your extensions.conf
file by typing "dialplan show [EMAIL PROTECTED]" in the Asterisk CLI and looking at
the output.

--
Jared Smith
Community Relation Manager
Digium, Inc.


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Re: [asterisk-users] Asterisk dialplan date and time operations

[dave cantera]

erik,
you can start here:
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime
http://www.asteriskguru.com/tutorials/gotoiftime.html
daveC

Erik Wartusch wrote:

  Hi all,

Im using Asterisk 1.4.11 and I want to proceed some time and date operations 
in my dial plan. (for a time shifted callback).

Should look like:

CURRENT TIME + x minutes.

Of course it should increase the hours for example in this case:

10.59 + 5 minutes = 11.04

I guess I've to use the math function in 1.4 but how can I manage easily the 
time operations?

Kind Regards,

Erik

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Re: [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk

[dave cantera]

bilal,
you are right.  you need to add port forwarding (UDP) to your router...
should work nicely then for iax.  also, don't forget you iptables or
firewall port config to accept iax on your * box.
daveC

bilal ghayyad wrote:

  Hi List;

I heared that IAX is good for NATing issues, but I do
not know if it can help me in that senario:

I have two Asterisks machines in different sites and
both are behind NAT (both have private IP address), I
need to link these two asterisks with IAX trunk (if it
help really in such senario), but I do not know if it
will work without doing special routing settings on
the router (like TCP/UDP port mapping or IP
forwarding)? How that will be it if possible? Or I
have to do a kind of port mapping?

If I will need to use port mapping, then I have to map
the TCP and UDP ports that are determined in iax.conf
and rtp.conf files at site A for asterisk ip address
at site A? Or I have to map the TCP and UDP ports that
are in iax.conf and rtp.conf at site B for asterisk ip
address at site A? In other words, if I am at site B
then I have to go for router B and do mapping for
TCP/UDP ports of the asterisk at site B or the
asterisk at site A?

Any help.
Regards
Bilal


  
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Re: [asterisk-users] Invalid extensions

[dave cantera]

gilberto,
check your config files, extensions.conf,  to see if autofallthrough is
set or not
you don't have any extensions in [ura] either...  only choice is 1 or
2... anything else is invalid...
daveC

Gilberto Nunes Ferreira wrote:

  Hi all

First I want to wish for everone a happy new year...

Well...

I have run asterisk 1.4.16.1 in a server.
I have this IVR, in extensions.conf:

[ura]

;exten => s, 1, Wait,1
exten => s, 1, Answer()
exten => s, n, Noop()
exten => s, n(debug),DumpChan()
exten => s, n, Set(LANGUAGE()=pt_BR)
exten => s, n,
Set(CALLFILENAME=/var/spool/asterisk/monitor/entrada/)
exten => s, n, Set(DYNAMIC_FEATURES =
hangup#pickupexten#atxfer#blidxfer)
exten => s, n,
MixMonitor(${CALLFILENAME}/${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)}-${EXTEN}.wav|v(0)V(0))
exten => s, n, Wait,1
exten => s, n, Set(TIMEOUT(digit)=3) ; Set Digit
Timeout to 5sec
exten => s, n, Set(TIMEOUT(response)=5) ; Set Response
Timeout to 10sec
exten => s, n, Background(bemvindobit)
exten => s, n, WaitExten
  



  exten => 1, 1, Set(LANGUAGE()=pt_BR)
exten => 1, 2, Queue(8600|tT|||30)
exten => 2, 1, Set(LANGUAGE()=pt_BR)
exten => 2, 2, Queue(8500|tT|||30)

autofallthrough? 

  exten => i, 1, Wait,1
exten => i, 2, Playback(invalid)
exten => i, 3, Goto(s,11)
exten => t, 1, Wait,1
exten => t, 2, Dial(SIP/8024,150,tTrwW)
exten => s, n, HangUp()
;include => ramais

Well, when I atempt dial to a external number, I goto
this IVR.
When the user get a connection, and when he press any
invalid number, I get this message on CLI:

Invalid extension '3' in context 'ura' on
SIP/bitmixinfo-0825f158

Here, we suppost tha the user press key 3, for a
mistake.
In this point, the call hangup!

Why this happen?

I take some research on google, but nothing...

Thanks so much...



  Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento!
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Re: [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?

[dave cantera]

tzafrir,
thanks for the note... yep, it is useless... 
daveC

Tzafrir Cohen wrote:
> On Tue, Jan 01, 2008 at 11:27:54AM -0500, dave cantera wrote:
>   
>> vincent,
>> here is a script that I used to convert a single wav file or the entire
>> directory... no file specified on launch, converts all files in the current
>> directory...
>> creates a logfile, although trivial...
>> daveC
>>
>> #!/bin/sh
>> #
>> #convert-all.sh
>> #
>> #convert all *.wav files to .gsm .au formats
>> #
>>
>> if [ "null${1}" == "null" ]
>> then
>> FILE_LIST=`ls *.wav`
>> else
>> FILE_LIST=`ls ${1}*.wav`
>> fi
>>
>> LOG="./log_convert.log"
>> echo "=== " >>${LOG}
>> echo "started at `date` " >>${LOG}
>>
>> echo " Removing all current .gsm files..."
>> rm -f *.gsm
>>
>> 
>
> # A note from the Useless Use of ls Committee:
>
> for FNAME in $1*.wav
>
>   
>> for FNAME in ${FILE_LIST}
>> do
>> echo "   ---   - "
>> echo "   " >>${LOG}
>> echo " Processing ${FNAME}... "
>> echo " Processing ${FNAME}... " >>${LOG}
>> BASEFNAME=`echo ${FNAME} | awk '{print substr($0,1,length($0)-4)}'`
>>
>> echo " making ${BASEFNAME}.gsm... "
>> echo " making ${BASEFNAME}.gsm... " >>${LOG}
>> #sox -q -V -c 1  ${FNAME} -r 8000 -c 1 -w ${BASEFNAME}.gsm resample -ql 
>> 2>>${LOG}
>> sox -q -V ${FNAME} -r 8000 -c 1 ${BASEFNAME}.gsm resample -ql  2>>${LOG}
>> echo "   " >>${LOG}
>> echo " making ${BASEFNAME}.au... "
>> echo " making ${BASEFNAME}.au... " >>${LOG}
>> sox -q -V ${FNAME} -t au -r 8000 -c 1 -w ${BASEFNAME}.au resample -ql 
>> 2>>$
>> {LOG}
>> done
>> 
>
>   

-- 
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I should buy her a Videophone2008!

Truly, The Next Best Thing to Being There!
--

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Re: [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?

[dave cantera]

vincent,
here is a script that I used to convert a single wav file or the entire
directory... no file specified on launch, converts all files in the
current directory...
creates a logfile, although trivial... 
daveC

#!/bin/sh
#
#    convert-all.sh
#
#    convert all *.wav files to .gsm .au formats
#

if [ "null${1}" == "null" ]
then
    FILE_LIST=`ls *.wav`
else
    FILE_LIST=`ls ${1}*.wav`
fi

LOG="./log_convert.log"
echo "=== "
>>${LOG}
echo "    started at `date` " >>${LOG}

echo " Removing all current .gsm files..."
rm -f *.gsm

for FNAME in ${FILE_LIST}
do
    echo "   ---   - "
    echo "   " >>${LOG}
    echo " Processing ${FNAME}... "
    echo " Processing ${FNAME}... " >>${LOG}
    BASEFNAME=`echo ${FNAME} | awk '{print substr($0,1,length($0)-4)}'`

    echo " making ${BASEFNAME}.gsm... "
    echo " making ${BASEFNAME}.gsm... " >>${LOG}
    #sox -q -V -c 1  ${FNAME} -r 8000 -c 1 -w ${BASEFNAME}.gsm resample
-ql  2>>${LOG}
    sox -q -V ${FNAME} -r 8000 -c 1 ${BASEFNAME}.gsm resample -ql 
2>>${LOG}
    echo "   " >>${LOG}
    echo " making ${BASEFNAME}.au... "
    echo " making ${BASEFNAME}.au... " >>${LOG}
    sox -q -V ${FNAME} -t au -r 8000 -c 1 -w ${BASEFNAME}.au resample
-ql 2>>${LOG} 
done






Vincent wrote:

  Hello

Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0
and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd
like to play PCM WAV files instead of eg. GSM. Per...

www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk

... I recorded a sample of my voice using XP's Sound Recorder, then
ran the following :

sox test_wav.wav -r 8000 -c 1 -s -w test_wav_out.wav resample -ql

But it seems like I'm missing the codec or something:

===
  -- Executing [EMAIL PROTECTED]:2] Playback("SIP/2000-0871d000",
"/usr/local/lib/asterisk/test_wav_out.wav") in new stack

WARNING[37390]: file.c:563 ast_openstream_full: File
/usr/local/lib/asterisk/test_wav_out.wav does not exist in any format

WARNING[37390]: file.c:866 ast_streamfile: Unable to open
/usr/local/lib/asterisk/test_wav_out.wav (format 0x4 (ulaw)): No such
file or directory
===

Here's what "core show file formats" says:
===
Format Name   Extensions
gsmwav49  WAV|wav49
slin   wavwav
adpcm  voxvox
slin   slnsln|raw
g722   g722   g722
ulaw   au au
alaw   alaw   alaw|al
ulaw   pcmpcm|ulaw|ul|mu
ilbc   iLBC   ilbc
h264   h264   h264
h263   h263   h263
gsmgsmgsm
g729   g729   g729
g726   g726-16g726-16
g726   g726-24g726-24
g726   g726-32g726-32
g726   g726-40g726-40
g723   g723sf g723|g723sf
18 file formats registered.
===

Am I missing something in the configuration files, or maybe I'm
missing some module?

Thank you.


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Re: [asterisk-users] Problem with Polycom Soundpoint IP 320 Hardphone

[dave cantera]

glenn,
check your handset cord... it might be plugged into the wrong port in
the back of the phone.  perhaps the headset jack...
daveC

Glenn Gillen wrote:

  Hey all,

I've setup my asterisk install on a CentOS5 server, I've got a few
IAX2 and SIP softphone clients connected on the same subnet and at
least 1 external IAX2 softphone. However I'm having some difficulty
getting the Polycom hardphone to function correctly. Watching the logs
and debug trace it:

- Registers correctly
- Is able to make calls to other peers

However it is not able to answer calls made to it. That is, the
handset actually rings, but I've no way to answer it. The answer soft
key, picking up the phone, etc. all have no effect. And I'm at a loss
as to what setting should be altered to fix it. Any ideas?

Possibly a tangent, but also affecting this handset, is that trying to
dial out over an external SIP trunk fails on the first attempt. But
calling an internal peer and then trying a second time makes it
mysteriously work.

Any help greatly appreciated,

  


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Truly, The Next Best Thing to Being There!
--

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Re: [asterisk-users] application not load

[dave cantera]

bhrugu,
ok, you used menuselect  but you said it is not loaded...  
did you try to load it manually???  does the module.so file exist in
the /usr/lib/asterisk/modules directory???
daveC

Bhrugu Mehta wrote:

  hi,
thnks 4 reply,
actully i am using asterisk 1.4.15 and that is defined in menuselect
file.(xml file)
so no need to add entry in module.conf

Bhrugu mehta


On Dec 27, 2007 7:37 PM, dave cantera <[EMAIL PROTECTED]> wrote:
  
  
bhrugu,

did you try and load it manually?

Modules are compiled in to shared object (.so) files. They are installed
to /usr/lib/asterisk/modules and can be turned on and off from loading
by editing /etc/asterisk/modules.conf. Modules must include
asterisk/modules.h. Modules must also export several functions. The
following functions generally return 0 on success and non-zero on
failure. Do not define any of these functions as static.

http://www.lobstertech.com/doc/ast-12-func/#funcmod
daveC


Bhrugu Mehta wrote:


  hi, all

I creat new application app_myapp.c for asterisk 1.4.15.
I add this in asterisk/apps dir. to load.

after compiling asterisk app_myapp.o and app_myapp.so has been created but when
i run " show applications" at cli> . my application not displayed.

what's wrong???

any suggestion!!!

thanks
Bhrugu Mehta

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Re: [asterisk-users] Samsung iDCS 500R2 Asterisk 1.4.*

[dave cantera]

william,
post your dialplan section for outbound, sip.conf minus passwords, and
CLI> output  so we can see what is going on... make sure you set
debugging higher or set sip debugging on for iDCS, assuming iDCS is a
sip provider, of course
daveC

William Stillwell (Ki4swy) wrote:

  I'm having trouble getting an Asterisk server to make outbound calls on my iDCS Trunks.

I have idcs station to asterisk station working
I have asterisk station to idcs station working

However, I am unable to get Asterisk to utilize any outbound trunks on my iDCS

Anybody have any ideas?

 





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Re: [asterisk-users] application not load

[dave cantera]

bhrugu,

did you try and load it manually?

Modules are compiled in to shared object (.so) files. They are installed 
to /usr/lib/asterisk/modules and can be turned on and off from loading 
by editing /etc/asterisk/modules.conf. Modules must include 
asterisk/modules.h. Modules must also export several functions. The 
following functions generally return 0 on success and non-zero on 
failure. Do not define any of these functions as static.

http://www.lobstertech.com/doc/ast-12-func/#funcmod
daveC

Bhrugu Mehta wrote:
> hi, all
>
> I creat new application app_myapp.c for asterisk 1.4.15.
> I add this in asterisk/apps dir. to load.
>
> after compiling asterisk app_myapp.o and app_myapp.so has been created but 
> when
> i run " show applications" at cli> . my application not displayed.
>
> what's wrong???
>
> any suggestion!!!
>
> thanks
> Bhrugu Mehta
>
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Re: [asterisk-users] Grandtream Conference issue

[dave cantera]

keshaw,
did you set your sip.conf to only allow g729?

disallow=all
allow=g729

I don't use g729 so the allow= may not be the correct syntax...

here is the config I uise:

disallow=all
allow=ulaw
allow=gsm
allow=alaw

daveC


Keshav K. wrote:
> Hi,
> I'm using Grandstream IP phone GXP2000, with Asterisk 1.4.15
> I'm using g729 codec and want to use only this codec for the calls.
> My normal calls are going fine. But issue is coming when I'm using the 
> conference from the Line1 and Line2 Option.
> When I'm initiating the conference at that time, IP phone is sending 
> the G711ulaw for the conference call, while in my phone I've set the 
> all codec option to PCMU only.
>
> Due to this I'm facing issue.
> Any solution for this problem, please let me know.
>
> Regards,
> Keshav
>
>
>
> Regards,
> Kesh
> " Lets change the future...lets change the world."
>
> 
> Never miss a thing. Make Yahoo your homepage. 
> 
> 
>
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>
> No virus found in this incoming message.
> Checked by AVG Free Edition. 
> Version: 7.5.516 / Virus Database: 269.17.7/1194 - Release Date: 12/23/2007 
> 05:27 PM
>   

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Re: [asterisk-users] Summary: Upgrading to Asterisk 1.4

[dave cantera]

russell,
I breezed through the document on 1.6 releases...  it looks like a Great 
move...
are you thinking, in the future, to moving to a development/release 
schedule like linux, a 1.7.x.y?
daveC

Russell Bryant wrote:
> Olivier wrote:
>   
>> Is this going to be  included in 1.6 ?
>> Any commitment on it ?
>> 
>
> Asterisk 1.6 will be managed in such a way that new features may be introduced
> in any version.  So, while some things may not be in 1.6.0, they may be in
> 1.6.1, .2, .3, etc.
>
> See the following post for a document that describes how Asterisk 1.6 will 
> work.
>  I plan for this process to begin very early in 2008.
>
> http://lists.digium.com/pipermail/asterisk-dev/2007-October/030083.html
>
>   

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Re: [asterisk-users] ISDN BRI support with HFC-PCI cards?

[dave cantera]

the /tmp sometimes gets its sticky bit set...

# ls -ld /tmp

will tell you what permissions are set at

/tmp should read:

[EMAIL PROTECTED] ~]# ls -ld /tmp
drwxrwxrwx  7 root root 4096 Dec 26 08:13 /tmp

it may read, which would not allow file/directory creation (notice the 
't' in the other's permissions, I think it comes up in other anyway...):

drwxrwxrw*t*  7 root root 4096 Dec 26 08:13 /tmp

daveC

Jaap Winius wrote:
> Quoting Tzafrir Cohen <[EMAIL PROTECTED]>:
>
>   
>>> Could someone please point me in the direction of some reasonable   
>>> instructions
>>> for setting up ISDN BRI support for Asterisk 1.2 (Debian etch) with
>>> HFC-PCI cards (Cologne chips) and/or cards with Winbond W6692CF chips?
>>>   
>>   apt-get install asterisk zaptel-source
>>   m-a a-i zaptel
>>   echo "#include zapata-channels.conf" >>/etc/asterisk/zapata.conf
>>   genzaptelconf -sd
>>
>> That should basically do it.
>> 
>
> That looks promising, but am I right that this requires the zaptel package to
> be installed as well? Otherwise there's no genzaptelconf (or one that  
> works). However, after installing the zaptel package, I get these  
> errors:
>
> # genzaptelconf -sd
> Stopping Asterisk PBX: asterisk.
> cat: /tmp/tmp.uiMna12463/span_termtype: No such file or directory
> cat: /tmp/tmp.uiMna12463/span_termtype: No such file or directory
> Starting Asterisk PBX: asterisk.
> #
>
> Looks like it's not creating its temporary directory in /tmp. Any ideas?
>
> Thanks,
>
> Jaap
>
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Re: [asterisk-users] Two lines for outgoing calls

[dave cantera]

dominik,
along with steve's note  /var/log/messages, check
/var/log/asterisk/fullas well
daveC

Steve Totaro wrote:
> Dominik Zalewski wrote:
>   
>> Dear All,
>>
>> I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel
>> 2.6.18. 
>>
>> I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm
>> using below context for dialing out.
>>
>> [outbound-local]
>> exten => _9XXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
>> exten => _9X,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
>> exten => _9ZXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
>> exten => _9ZXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
>> exten => _90900X,1,Dial(Zap/g1/${EXTEN:1},30m,tTr)
>>
>> When Zap/1 is busy and I try to call, it will use Zap/2 which is fine
>> but there is something wrong cause I hear one ring and then a weird
>> sound like a noise or something and then hangup. I have to reload zaptel
>> modules and then everything works fine for a while.
>>
>>-- Executing [EMAIL PROTECTED]:1] Dial("SIP/200-08221590",
>> "Zap/g1/150|30|tTr") in new stack
>> -- Called g1/150
>> -- Zap/2-1 answered SIP/200-08221590
>> -- Hungup 'Zap/2-1'
>>
>> I even thought that second fxo module is broken so I changed it. No
>> results.
>>
>> Any ideas?
>>
>> Thank you in advance,
>>
>> Dominik 
>>
>>
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>>   
>> 
> Not sure why you would have intermittent problems like that.  Is this a 
> Digium card? 
>
> I once had a four port FXO Digium card and thought that I had a bad FXO 
> module but it turned out that it was actually the slot on the card 
> itself.  The issue I was experiencing was large amounts of static so 
> that you could not hear anything.
>
> Maybe you can find something helpful in /var/log/messages that 
> corresponds with the issue.
>
> Anyways, I would contact the manufacturer and see what they can do for you.
>
> Thanks,
> Steve Totaro
>
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Re: [asterisk-users] X100P Woes

[dave cantera]

bob,

look on p20 of 'the book' edition 2, or p16 edition 1
this explains the 3.3v vs 5.0v issue with motherboard slots

http://www.oreilly.com/catalog/9780596510480/
daveC


Bob Smither wrote:
> On Wed, 2007-12-26 at 15:46 +1000, Mattt wrote:
>   
>>   Sounds like a PCI bus version issue ;-)
>> 
>
> Thanks Matt, but could you elaborate?  The card supposedly supports both
> 5 and 3.3 volt PCI slots, and the slots on the older system that works
> with the card appear the same as on the board giving me grief.
>
> The connectors in both systems have a single key while the card has two
> keys.
>
> An further explanation of your comment would be appreciated!
>
>
>
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Re: [asterisk-users] Soundcard necessary on an asterisk server to get output of playback()??

[dave cantera]

this sounds like it might be:

1)  a permissions problem...  
did you install * as root?
is there a .gsm, .ulaw, .alaw file?

2) sip.conf problem
   codec not allowed or not specified, did you allow the proper codec?

daveC

1) permissions

drwxr-xr-x  17 root root 221184 Dec  8 10:14 /var/lib/asterisk/sounds/

[EMAIL PROTECTED] ~]# ls -ld  /var/lib/asterisk/sounds/tt-monkey*
-rw-r--r--  1 root dip 129440 Jun 14  2007 
/var/lib/asterisk/sounds/tt-monkeys.alaw
-rw-r--r--  1 root dip  26697 Jun 14  2007 
/var/lib/asterisk/sounds/tt-monkeys.gsm
-rw-r--r--  1 root dip  18139 Jun 14  2007 
/var/lib/asterisk/sounds/tt-monkeysintro.alaw
-rw-r--r--  1 root dip   3762 Jun 14  2007 
/var/lib/asterisk/sounds/tt-monkeysintro.gsm
-rw-r--r--  1 root dip  18139 Jun 14  2007 
/var/lib/asterisk/sounds/tt-monkeysintro.ulaw
-rw-r--r--  1 root dip  36322 Jun 14  2007 
/var/lib/asterisk/sounds/tt-monkeysintro.wav
-rw-r--r--  1 root dip 129440 Jun 14  2007 
/var/lib/asterisk/sounds/tt-monkeys.ulaw
-rw-r--r--  1 root dip 258924 Jun 14  2007 
/var/lib/asterisk/sounds/tt-monkeys.wav

2) sip.conf
disallow=all
allow=ulaw
allow=gsm



Bhrugu Mehta wrote:
> no , not at all, there is no need to install sound card in asteirsk system.
> I am using asterisk server without soundcard.
> so there may be antoher problem may in configurtion of zapata or other.
> cheers!!!
> Bhrugu mehta
>
> On Dec 3, 2007 11:31 PM, Stefan Guenther <[EMAIL PROTECTED]> wrote:
>   
>> Hi,
>>
>> I' still fighting the problem, that I can talk from one SIP phone to
>> another, but I can't hear the output of the playback or similar
>> applications:
>>
>>  exten => 202,1,ANSWER()
>>  exten => 202,2,PLAYBACK(tt-monkeys)
>>  exten => 202,3,HANGUP()
>>
>> When I dial 202, asterisk show the following on the cli:
>>
>> -- Executing [EMAIL PROTECTED]:1] Answer("SIP/user1-0827ebe8", "") in new 
>> stack
>> -- Executing [EMAIL PROTECTED]:2] Playback("SIP/user1-0827ebe8", 
>> "tt-monkeys")
>> in new stack
>> --  Playing 'tt-monkeys' (language 'de')
>>
>> Yes, the file tt-monkeys exist in /var/lib/asterisk/sounds and the
>> subdirectory de.
>>
>> No, there is no error message even if turn on debugging. :-(
>>
>> Besides this strange behaviour, I was wondering whether the asterisk
>> server needs an soundcard to send the output of e.g. the playback
>> application to the phone.
>>
>> BTW, this is asterisk 1.4.13
>>
>> I would be really happy, if someone has an idea how to solve this problem.
>>
>> Thanks in advance,
>>
>> Stefan
>> --
>>
>> 
>> in-put GbR - Das Linux-Systemhaus
>> Stefan-Michael Guenther
>> Geschaeftsfuehrer
>> Moltkestrasse 49 D-76133 Karlsruhe
>> Tel./Fax : +49 (0)721 / 83044 - 98/93
>> http://www.in-put.de
>> 
>>   Schulungen  Installationen
>>   Beratung   Support
>>Voice-over-IP-Loesungen
>> 
>>
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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[dave cantera]

remco,
I just had the same problem/error on my CLI>  when I added a polycom
shoretel IP-100 phone to my network and enabled mgcp...  couldn't
figure out how to get that working yet...    
I don't think it is related to 1.4 as I have been running 1.4 has been
running for over a year now without that error...  I would look
somewhere else...
daveC


[Dec 21 08:51:32] WARNING[16742]: chan_sip.c:6620
determine_firstline_parts: Bad request protocol
[EMAIL PROTECTED]] MGCP 1.0


Remco Barendse wrote:

  
I wonder if there are any major obstacles for upgrading.

  
  

Just tried an in-place upgrade on my home box :

make[1]: Leaving directory `/usr/src/asterisk-addons-1.4.5'
for x in app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so 
res_config_mysql.so; do /usr/bin/install -c -m 755 $x 
/usr/lib/asterisk/modules ; done
/usr/bin/install: cannot stat `app_addon_sql_mysql.so': No such file or 
directory
/usr/bin/install: cannot stat `cdr_addon_mysql.so': No such file or 
directory
/usr/bin/install: cannot stat `res_config_mysql.so': No such file or 
directory
make: *** [install] Error 1


And the asterisk console is flooded with these errors :

[Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707 
determine_firstline_parts: Bad request protocol Packet
[Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707 
determine_firstline_parts: Bad request protocol Packet

So for the next time to come i'll turn back to 1.2 :)

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Re: [asterisk-users] MeetMeConference

[dave cantera]

also, you want to think about transcoding... if you have different 
technologies, the system load for transcoding would increase...

dean, cool, I didn't know you could hang a few * boxes together with 
meetme...
daveC


Dean Collins wrote:
> as far as I know it's unlimited and only tied to the capacity of a
> single machines processing power.
> Of course then all you need to do is tie multiple machines to the same
> room in a daisy chain and expand from there.
> Someone on the list a few months ago gave the example of chaining
> together 5 servers this way - sorry i cant remember who it was.
>
> Cheers,
> Dean
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of bilal
> ghayyad
> Sent: Thursday, 20 December 2007 2:44 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] MeetMeConference
>
> Hi All;
>
> Is there any limitation on the number of users for MeetMe Conference? In
> other words, how many parties can join to the room and become a member
> of the room?
> Is there any limitation?
>
> Regards
> Bilal
>
>
>  
> 
> 
> Be a better friend, newshound, and
> know-it-all with Yahoo! Mobile.  Try it now.
> http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ 
>
>
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Re: [asterisk-users] All trunk are busy please try your call again later

[dave cantera]

SIP/7871-bb64", "SIP/7874|15|tr") in new stack
-- Called 7874

-- SIP/7874-5b48 is ringing
  == Spawn extension (macro-dial, s, 10) exited non-zero on
 'SIP/7871-bb64' in ma cro 'dial'
  == Spawn extension (macro-dial, s, 10) exited non-zero on

 'SIP/7871-bb64' in ma cro 'exten-vm'
  == Spawn extension (macro-dial, s, 10) exited non-zero on
 'SIP/7871-bb64'
asterisk1*CLI>



THANKS SO MUCH I WILL BE EXPECTING YOUR  REPLY.

  
  
  
  On Dec 20, 2007 5:09 PM, Lolu Gbenga <[EMAIL PROTECTED]> wrote:
  
  Hi
all,
I am grateful for our contribution so far .

I followed dave advise and i have the attached file using the aterisk
-r when a call is made.


I attached two files.

One of the attached file is for the external call,which replied with
the PROBLEM all trunks are busy now,please try your call again later.


The second attachment is when i made internal calls and the phone rang.

Please,i will be expecting your replies for further directions.

Best Regards




On Dec 20, 2007 2:58 PM, Steve Totaro <
[EMAIL PROTECTED]>
wrote:
What
is the output of ztconfig from the Linux command line?  What does
your zaptel.conf and zapata.conf look like?  What is the relevant part
of extensions.conf (the dialout section that fails).  Also from the
CLI,
  
it would be most helpful to post the output you get when dialing out
fails.  I don't think it is a network issue at all, I think your configs
need some work.
  
Thanks,
  Steve Totaro
  
  
  
Lolu Gbenga wrote:
> Good Day
>
  
  > Find attached the relevant portions of the asterisk CLI.
>
> Please,which portion of the extension .conf should i send ?
  
>
> It is connected via RJ 45 connector to an E1 modem to the telco
company.
>
> I use E1 link.
>
> I will appreciate your reply.
>
> Best Regards
>
>
> On Dec 18, 2007 4:02 PM, dave cantera <
  [EMAIL PROTECTED]
  
  
  > [EMAIL PROTECTED]> > wrote:
  
>
>     lolu,
>     sounds more like a telco/itsp problem then *.
>     I would
>        tcpdump -i eth0 port 5060
>     to make sure it is actually going out... change 5060 if you
have
>     changed
>     your port to your itsp, of course.
>     to see what is going on as well as the other debugging notes
mentioned
>     in this thread.
>     daveC
>
>     Lolu Gbenga wrote:
  
>     > Good Day all
>     >
>     > Please I am having some issues on my voip asterisk server
>     >
>     > I make internal calls on extensions configured ie
extension 192 can
  
>     > call extension 195 etc
>     >
>     > But each time i try to make calls outside the extension
ie calling a
>     > GSM or an external line ,i always hear this response "all
trunk
  
>     calls
>     > are busy please try your call again later"
>     >
>     > Please how can i resolve this problem .
>     >
>     > I will appreciate your response.
  
>     >
>     > Best Regards
>     >
>     > Success
>     >
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No virus found in this incoming message.
Checked by AVG Free Edition. 
Version: 7.5.503 / Virus Database: 269.17.2 - Release Date: 12/14/2007 12:00 AM
  


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Re: [asterisk-users] All trunk are busy please try your call again later

[dave cantera]

lolu,
while you are making the call., capture and post your CLI> output
...  this is easy to do since you are using putty.

login to your pbx and start asterisk, use the below command:

# asterisk -vvvr

then make the call.  hilite the text on the putty terminal and paste it
into the body of the email to the list...  
sorry if I'm making these instruction too basic...

pbv01*CLI>
    -- Executing [EMAIL PROTECTED]:1] Wait("SIP/202-b753da18", "1") in new
stack
    -- Executing [EMAIL PROTECTED]:2] Answer("SIP/202-b753da18", "") in
new stack
    -- Executing [EMAIL PROTECTED]:3] NoOp("SIP/202-b753da18", "DEBUG:
CALLERID=") in new stack
    -- Executing [EMAIL PROTECTED]:4] Notify("SIP/202-b753da18",
"800202|x202|300/192.168.15.100") in new stack
    -- Notify: sending '800202|x202|300' to 192.168.15.100:4
    -- Executing [EMAIL PROTECTED]:5] AGI("SIP/202-b753da18",
"agi-callpop4.sh||red") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-callpop4.sh
    -- AGI Script agi-callpop4.sh completed, returning 0
    -- Executing [EMAIL PROTECTED]:6] NoOp("SIP/202-b753da18", "AGISTATUS
is >FAILURE<") in new stack
    -- Executing [EMAIL PROTECTED]:7] NoOp("SIP/202-b753da18", "DEBUG:
EXTEN=300") in new stack
    -- Executing [EMAIL PROTECTED]:8] Dial("SIP/202-b753da18",
"SIP/300|15|rt") in new stack
    -- Called 300
    -- SIP/300-09e062e8 is ringing
  == Spawn extension (local-sip, 300, 8) exited non-zero on
'SIP/202-b753da18'

daveC





Lolu Gbenga wrote:
Thanks 
Please am using putty to again access to my Linux asterisk box.
How can i use tcpdump to get your request on the exact Ethernet port
and port number.
  
I will appreciate your  reply.
  
Best Regards 
  
  
  On Dec 18, 2007 4:02 PM, dave cantera <[EMAIL PROTECTED]>
wrote:
  lolu,
sounds more like a telco/itsp problem then *.
I would
   tcpdump -i eth0 port 5060
to make sure it is actually going out... change 5060 if you have changed
your port to your itsp, of course.
to see what is going on as well as the other debugging notes mentioned

in this thread.
daveC


Lolu Gbenga wrote:
> Good Day all
>
> Please I am having some issues on my voip asterisk server
>
> I make internal calls on extensions configured ie extension 192
can

> call extension 195 etc
>
> But each time i try to make calls outside the extension ie calling
a
> GSM or an external line ,i always hear this response "all trunk
calls
> are busy please try your call again later"

>
> Please how can i resolve this problem .
>
> I will appreciate your response.
>
> Best Regards
>
> Success
>


>
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>
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Re: [asterisk-users] turn off auto-seek extention - force use timeout

[dave cantera]

mojo,
nice suggestion.
daveC

Mojo with Horan & Company, LLC wrote:
> So I'm guessing this is what you're doing:
> --
> [ids]
> exten => s,1,playback(enter your id number)
> exten => s,2,WaitExten(10)
> exten => s,3,Goto(1)
>
> exten => 4768,1,blahblahblah
> exten => 4790,1,blahblahblah
> exten => 4732,1,blahblahblah
>
> exten => i,1,playback(error)
> exten => i,2,goto(s,1)
> --
> So, maybe place the phones in a context that waits for a four-digit id 
> _before_ matching it to the context you were initially trying:
> --
> [getid]
> exten => s,1,playback(enter your id number)
> exten => s,2,WaitExten(10)
> exten => s,3,Goto(1)
>
> exten => _4XXX,1,goto(ids,${exten},1)
>
> [ids]
> exten => 4768,1,blahblahblah
> exten => 4790,1,blahblahblah
> exten => 4732,1,blahblahblah
>
> exten => i,1,playback(error)
> exten => i,2,goto(getid,s,1)
>
> --
>
> Untested: I wonder if one entered an extension that didn't exist, say 
> 4555, when we tried to Goto(ids, 4555, 1) would we get directed to 
> extension i in the extensions context or would the call be dropped 
> completely?
>
> Moj
>
> Justin Killen wrote:
>   
>> I have an application where a call-in user is prompted to enter an 
>> identification number for schedule information. That id number is 
>> setup as an extension, and if that extension doesn’t exist, it tells 
>> them that they are not scheduled, then loops back to ask for the id 
>> number again. My problem is that asterisk pre-emptively goes to the i 
>> extension (invalid) too early depending on available extensions. For 
>> example, if I put in id number 4768, and there is only 4790 and 4732, 
>> it will push to the invalid extension on the 6, then the “not 
>> scheduled” playback message (a cepstral command) gets cancelled out 
>> from the DTMF push of the 8. So, if I put in 4768, I get prompted to 
>> enter an id number. What I would like to do is turn off this feature, 
>> so that the number input does not get evaluated until after the 
>> timeout (preferably configurable from the extensions.conf file).
>>
>> Thanks in advance
>>
>> -Justin
>>
>> 
>>
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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[dave cantera]

dovid...
while this seems like a good idea to have both sip show channels and 
show channels sip having two, three or even four ways to do the same 
thing would confuse/cripple the learning curve... * would turn into a 
microsoft mentality where there are dozens of ways to 
configure/reconfigure some of their products...  word, for example, can 
be configured with or without the tool bars and then you can configure 
hot-keys...  in fact, you can configure some products so that someone 
who learns it with a hacked config, could not possibly use the original 
stock config...  sorry to go on about this but it is one of my hot 
buttons... 
daveC

Dovid B wrote:
> - Original Message - 
> From: "Steve Edwards" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Wednesday, December 19, 2007 5:43 AM
> Subject: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
>
>
>   
>> On Sat, 15 Dec 2007, Johansson Olle E wrote:
>>
>> 
>>> I wonder if there are any major obstacles for upgrading.
>>>   
>> How about the change from a bad command line interface to a really bad
>> command line interface?
>>
>> I mean, Seriously? (in a Grey's Anatomy kind of way...)
>>
>> The old syntax was inconsistent -- "show manager command" vs "sip show
>> channels" and just plain bad -- for example "sip reload" should have been
>> "reload sip."
>>
>> The new syntax continues down the noun-verb path instead of correcting
>> itself and using verb-noun like most other applications (MySQL, GDB,
>> Oracle, etc.)
>>
>> Then, just to make it worse, now I have to learn which commands somebody
>> (arbitrarily) decided are "core" and which are not -- for what benefit?
>> Certainly doesn't make MY job easier!
>>
>> Approach the command line like a noob. I want Asterisk to show me
>> something so I'll start the command line with "show." I'm not quite sure
>> what I'm doing, so I'll press  to see what I can show. Oh, "channel"
>> looks like what I want. Hmm, too much. Maybe I should have qualified what
>> kind of channel I'm looking for BEFORE the word "channel."
>>
>> Here's a suggestion -- stop thinking like a parser and start thinking like
>> a person :)
>>
>> Which makes more sense (at least in English)?
>>
>>  1) show black dogs -- show sip channels
>>  2) black show dogs -- sip show channels
>>  3) dogs black show -- channels sip show
>>  4) show dogs black -- show channels sip
>>  5) black dogs show -- sip channels show
>>  6) dogs show black -- channels show sip
>>
>> Is it too late to fix this for 1.6?
>>
>> Thanks in advance,
>> 
>
> I think as many people have pointed out they are used to a lot of commands 
> out there so changing it yet again would make more people unhappy. But maybe 
> asterisk can have both. Why not sip show channels for the old timers and 
> show channels sip or show sip channels for the n00b's. Why shouldn't 
> asterisk have both options ? 
>
>
>
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[asterisk-users] shoreline IP100 aka Polycom 500 boot problem

[dave cantera]

my client purchased a couple of shoreline ip-100 phones...  I managed
to get them to Not boot up...   shows the polycom logo then goes
blank...   looks like the want mcgp...  oh, mgcp...

is there a solution for this?  besides sending it back to polycom?
daveC







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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[dave cantera]

tzafrir,
thanks for the note. btw, Great docs!
asciidocs looks cool too!
thanks!
daveC

Tzafrir Cohen wrote:

  Hi

On Wed, Dec 19, 2007 at 12:19:08AM -0500, dave cantera wrote:
  
  
ok, here is my $0.02...  I created a script since I had to 
install/update so often and for various reasons...
you can choose to compile automatically or manually...
modify the current release numbers, your repository, and source root... 
all else is automated..
which is unloading zap driver, stopping a running asterisk, getting the 
current release, untar'ng it and compiling it...
enjoy,
daveC

  
  
You can find my take on the subject at
http://updates.xorcom.com/astribank/bristuff/1.4/bristuff-current/
I improved the existing scripts from bristuff to be more potent, as
explained in
http://updates.xorcom.com/astribank/bristuff/1.4/INSTALL.html

The bristuff scripts have a little wrapper install.sh that calls
download.sh (downloads and patches. Kind of like rpmbuild -bp) and
compile.sh (builds and installs).

That separation can reduce some of the need for user interaction in your
script.

If you want to use them, I figure you should just remove the patching
commands and then you should be able to use those scripts mostly
unchanged.

  
  

#!/bin/sh
#
#get_latest_rel.sh
#
# Dave Cantera:  [EMAIL PROTECTED]
#
#get the current asterisk release components, put them in our REPOSITORY
#and unpack them in SRC_ROOT

---< Change to suite between these lines >--
VER_AST="1.4.16"
VER_ZAPTEL="1.4.7.1"
VER_LIBPRI="1.4.3"
VER_ADDONS="1.4.5"

REPOSITORY="/root/tarballs"
SRC_ROOT="/usr/local/src"
---< Change to suite between these lines >--

HTTP_SITE="http://downloads.digium.com"
PUB_DIR="/pub"

TARBALL_AST="/asterisk/releases/asterisk-${VER_AST}.tar.gz"
TARBALL_LIBPRI="/libpri/releases/libpri-${VER_LIBPRI}.tar.gz"
TARBALL_ZAPTEL="/zaptel/releases/zaptel-${VER_ZAPTEL}.tar.gz"
TARBALL_ADDONS="/asterisk/releases/asterisk-addons-${VER_ADDONS}.tar.gz"

COMPONENTS="${HTTP_SITE}${PUB_DIR}${TARBALL_AST}
${HTTP_SITE}${PUB_DIR}${TARBALL_ZAPTEL}
${HTTP_SITE}${PUB_DIR}${TARBALL_LIBPRI}
${HTTP_SITE}${PUB_DIR}${TARBALL_ADDONS} "

echo
echo
echo " we are prepared to get the complete current release "
echo " of asterisk, libpri, zaptel, and addons "
echo " the tarballs will be placed in our REPOSITORY and "
echo " then extracted to our SRC_ROOT "
echo
echo "---< Activity Recap >"
echo
echo " TARBALL REPOSITORY: ${REPOSITORY}"
echo "   SRC_ROOT: ${SRC_ROOT}"
echo "   asterisk tarball: ${TARBALL_AST}"
echo " libpri tarball: ${TARBALL_LIBPRI}"
echo " zaptel tarball: ${TARBALL_ZAPTEL}"
echo " addons tarball: ${TARBALL_ADDONS}"
echo
echo -n " Are You Ready?  Y to procced: "
read ANSWER

if [ "null${ANSWER}" == "nullY" ]

  
  
# a matter of style:
case "$ANSWER" in Y* | y*) :;; 
  *) echo " Aborted by user ";;
  exit 0
esac

# and good bye to unneeded nesting.

  
  
then
echo
echo "-"
echo " stopping asterisk "
echo
echo " choose your poison: "
echo " a) /usr/bin/asterisk -xr stop now"
echo " b) /etc/init.d/asterisk stop "
echo
echo -n "  which one? "
read STOPCMD
if [ "null${STOPCMD}" == "nulla" ]
then
/usr/bin/asterisk -r -x 'stop now'
fi
if [ "null${STOPCMD}" == "nullb" ]
then
/etc/init.d/asterisk stop
fi

echo
echo "-"
echo " get the current asterisk & component releases and put them in 
our repository ${REPOSITORY}"
# lets go to the repository directory
cd ${REPOSITORY}

for TARBALL in `echo ${COMPONENTS}`
do
echo "getting component: ${TARBALL} "
#wget ${TARBALL}

  
  
Err... one needs to uncomment that line, I guess.

I tend to like using 'wget -c' . Otherwise strange things may happen if
I press ctrl-C in the middle of the download.

Sadly, the current downloads.digium.com will make you re-download the
tarballs 

  
  
done
   
TARFILES="
asterisk-${VER_AST}.tar.gz
libpri-${VER_LIBPRI}.tar.gz
zaptel-${VER_ZAPTEL}.tar.gz
asterisk-addons-${VER_ADDONS}.tar.gz "
   
echo
echo "-"
echo " unpack the current asterisk & component tarballs into our 
source root ${SRC_ROOT}"
# lets go to the source root directory
cd ${SRC_ROOT}
for TARBALL in `echo

Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[dave cantera]

ok, here is my $0.02...  I created a script since I had to 
install/update so often and for various reasons...
you can choose to compile automatically or manually...
modify the current release numbers, your repository, and source root... 
all else is automated..
which is unloading zap driver, stopping a running asterisk, getting the 
current release, untar'ng it and compiling it...
enjoy,
daveC


#!/bin/sh
#
#get_latest_rel.sh
#
# Dave Cantera:  [EMAIL PROTECTED]
#
#get the current asterisk release components, put them in our REPOSITORY
#and unpack them in SRC_ROOT

---< Change to suite between these lines >--
VER_AST="1.4.16"
VER_ZAPTEL="1.4.7.1"
VER_LIBPRI="1.4.3"
VER_ADDONS="1.4.5"

REPOSITORY="/root/tarballs"
SRC_ROOT="/usr/local/src"
---< Change to suite between these lines >--

HTTP_SITE="http://downloads.digium.com";
PUB_DIR="/pub"

TARBALL_AST="/asterisk/releases/asterisk-${VER_AST}.tar.gz"
TARBALL_LIBPRI="/libpri/releases/libpri-${VER_LIBPRI}.tar.gz"
TARBALL_ZAPTEL="/zaptel/releases/zaptel-${VER_ZAPTEL}.tar.gz"
TARBALL_ADDONS="/asterisk/releases/asterisk-addons-${VER_ADDONS}.tar.gz"

COMPONENTS="${HTTP_SITE}${PUB_DIR}${TARBALL_AST}
${HTTP_SITE}${PUB_DIR}${TARBALL_ZAPTEL}
${HTTP_SITE}${PUB_DIR}${TARBALL_LIBPRI}
${HTTP_SITE}${PUB_DIR}${TARBALL_ADDONS} "

echo
echo
echo " we are prepared to get the complete current release "
echo " of asterisk, libpri, zaptel, and addons "
echo " the tarballs will be placed in our REPOSITORY and "
echo " then extracted to our SRC_ROOT "
echo
echo "---< Activity Recap >"
echo
echo " TARBALL REPOSITORY: ${REPOSITORY}"
echo "   SRC_ROOT: ${SRC_ROOT}"
echo "   asterisk tarball: ${TARBALL_AST}"
echo " libpri tarball: ${TARBALL_LIBPRI}"
echo " zaptel tarball: ${TARBALL_ZAPTEL}"
echo " addons tarball: ${TARBALL_ADDONS}"
echo
echo -n " Are You Ready?  Y to procced: "
read ANSWER

if [ "null${ANSWER}" == "nullY" ]
then
echo
echo "-"
echo " stopping asterisk "
echo
echo " choose your poison: "
echo " a) /usr/bin/asterisk -xr stop now"
echo " b) /etc/init.d/asterisk stop "
echo
echo -n "  which one? "
read STOPCMD
if [ "null${STOPCMD}" == "nulla" ]
then
/usr/bin/asterisk -r -x 'stop now'
fi
if [ "null${STOPCMD}" == "nullb" ]
then
/etc/init.d/asterisk stop
fi

echo
echo "-"
echo " get the current asterisk & component releases and put them in 
our repository ${REPOSITORY}"
# lets go to the repository directory
cd ${REPOSITORY}

for TARBALL in `echo ${COMPONENTS}`
do
echo "getting component: ${TARBALL} "
#wget ${TARBALL}
done
   
TARFILES="
asterisk-${VER_AST}.tar.gz
libpri-${VER_LIBPRI}.tar.gz
zaptel-${VER_ZAPTEL}.tar.gz
asterisk-addons-${VER_ADDONS}.tar.gz "
   
echo
echo "-"
echo " unpack the current asterisk & component tarballs into our 
source root ${SRC_ROOT}"
# lets go to the source root directory
cd ${SRC_ROOT}
for TARBALL in `echo ${TARFILES}`
do
echo "untar'ng component: ${TARBALL} "
#tar xzf ${TARBALL}
done
   
echo
echo "-"
echo " unloading Zap drivers"
# unload the zaptel drivers
ZAP_MODULES=`lsmod | grep zap | awk '{printf("%s,",$4)}' | sed 's/,/ 
/g'`
   
for MODULE in `echo ${ZAP_MODULES}`
do
echo "unloading zap module: ${MODULE}"
#modprobe -r ${MODULE}
done

echo
echo " now you are ready to compile at ${SRC_ROOT} "
echo

echo -n " Shall I continue with the compile? Y?"
read COMPILE
if [ "null${COMPILE}" == "nullY" ]
then
echo " Compiling Zaptel version ${VER_ZAPTEL}"
cd "${SRC_ROOT}/zaptel-${VER_ZAPTEL}"
make;make; make install

echo " Compiling libpri version ${VER_LIBPRI}"
cd "${SRC_ROOT}/libpri-${VER_LIBPRI}"
make; make install

echo " Compiling Asterisk version ${VER_AST} "
cd "${SRC_ROOT}/asterisk-${VER_AST}"
make; ./configure; make; make install

echo " Compiling Asterisk Addons version ${VER_ADDONS} "
cd "${SRC_R

Re: [asterisk-users] All trunk are busy please try your call again later

[dave cantera]

lolu,
sounds more like a telco/itsp problem then *.
I would
tcpdump -i eth0 port 5060
to make sure it is actually going out... change 5060 if you have changed 
your port to your itsp, of course.
to see what is going on as well as the other debugging notes mentioned 
in this thread.
daveC

Lolu Gbenga wrote:
> Good Day all
>
> Please I am having some issues on my voip asterisk server
>
> I make internal calls on extensions configured ie extension 192 can
> call extension 195 etc
>
> But each time i try to make calls outside the extension ie calling a
> GSM or an external line ,i always hear this response "all trunk calls
> are busy please try your call again later"
>
> Please how can i resolve this problem .
>
> I will appreciate your response.
>
> Best Regards
>
> Success
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>   

-- 
My wife's sister is in California.  
I should buy her a Videophone2008!

Truly, The Next Best Thing to Being There!
--

WorldWideVideoPhones.com
856.380.0894




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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[dave cantera]

phil,
I think you are on to it... the best path is to load a new system up 
with 1.4.x and port your existing dialplan over, test it out, lock it 
down and then roll it out...

I've worked as a UNIX system integrator for 20+ years, worked with open 
source and custom developed C/C++ code, Ada, and a slew of developers 
(150 developers at two gov't projects).on new projects, developers 
don't usually provide enough info in the first releases to even install 
the product.  you always have a gotcha.  upgrades are better by far but 
still lack those things that developers take for granted.  developers 
conceived the idea and they have been talking about them for months...   
to the integrators, the release is new and that is when the difficulty 
arises.  as an integrator, we are charged with making the product stable 
in the target environment whereas developers are charged with making the 
product stable in their development environment...  it is two different 
scenarios.

when an integrator sets up a test/QA environment, things that the 
developers never invisioned come to light.  then it is a find, fix, 
retest cycle until all is well.  it is time consuming but well worth the 
effort as your support/help desk calls are greatly reduced...  so now 
that I am talking about this, perhaps I should offer a 
migration/integration/test lab service :)...  since I've been through it 
a hundred times...
daveC




Phil Knighton wrote:
> Hello
>
> As a person who is somewhat a "newbie" to Asterisk, I have been given
> the task of preparing our 1.2 installation for upgrade.  The thing that
> has slowed me down is some of the gaps in information on the upgrade
> process.  What's on the Wiki might make complete sense to both
> experienced Linux users, and Asterisk users but as someone who is
> feeling there way through - it's a bit daunting!
>
> Considering how important a phone system is to a business, I'm loathed
> to rush the upgrade through and have instead opted to install 1.4 on a
> different box, and "port" our existing setup over to it.  This is a time
> consuming process and has taken quite a low priority.  As Olle says -
> 1.2 works just fine.
>
> Personally speaking, the upgrade process has to be even easier if people
> are going to jump for it. 
>
> Phil
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Johansson
> Olle E
> Sent: 15 December 2007 10:57
> To: Asterisk Non-Commercial Discussion Users Mailing List -
> Subject: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
>
> Friends in the Asterisk community,
>
> I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2
> and 1.4 there's been a lot of important development. New code cleanups,
> optimization, new functions.
>
> I realize that 1.4 at release time wasn't ready for release, but we've
> spent one year polishing it, working hard with bug fixes. The 1.4 that
> is in distribution now is very different from the young and immature
> product that was release before Christmas in 2006.  
> Testing, testing, testing
> and hard work from developers has changed this and the 1.4 personality
> is now much more grown-up and mature :-)
>
> I wonder if there are any major obstacles for upgrading.
>
> - Bugs that are still open?
> - Bugs that are not reported?
> - Not enough reasons to upgrade, since 1.2 really works well
> - Just a bad karma for 1.4
>
> When responding, remember that we don't add new features to 1.4 after
> release, so I'm not looking for a wishlist - that's for the coming
> release. We need to make a released product stable, not add new features
> and potential scary bugs.
>
> Success stories with 1.4 are also welcome. "Upgrading to 1.4 doubled our
> revenues in a month and gave us 200% more quality in the voice channels"
> or "Asterisk 1.4 gave us more reliable pizza deliveries and also fixed
> the bad taste of the coffee in our vending machine". Anything.
>
> Also, I would like input on what you consider the most important new
> feature in 1.4.
> I will try to make a list based on the feedback. Feel free to send
> feedback to the list or in a private e-mail to me directly.
>
> Let's make 1.4 the choice for everyone's PBX - from small home systems
> to large scale carrier platforms!
>
> /Olle
>
> ---
> * Olle E. Johansson - [EMAIL PROTECTED]
> * Asterisk Training http://edvina.net/training/
>
>
>
>
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Re: [asterisk-users] dial, answered and then hangup

[dave cantera]

rilawich,
can you post the CLI output so we can see what is going on?
from the exten, it is doing exactly what you tell it to do...  dial then 
hangup
daveC

Rilawich Ango wrote:
> Hi all,
>
>   I will a TDM card with FXO modules on it.  Below is the dial plan.
> When someone can 9123456, CLI will show dialing to 123456 and
> answered.  After answered, the call hangup.  I would like to know what
> will cause the case to happen.  Anyone can give me some advice to
> solve it?
>
> exten => _9X.,n,Dial(Zap/g0/${EXTEN:1}|${RINGTIMEOUT})
> exten => _9X.,n,Hangup
>
> zapata.conf
> signalling=fxs_ks
> callerid=asreceived
> group=0
> context=from-pstn
> ;context=cs
> channel => 1-8
>
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Re: [asterisk-users] GUI for Asterisk: Call Flow

[dave cantera]

bilal,
flash operator panel (fop) or any of the asterisk gui does this...
asteriskNow for example...
http://www.asternic.org/

daveC

bilal ghayyad wrote:

  Hi All;

Is there an GUI for Asterisk that can help in showing
the call flow (who is in progress, who is connected,
called number, ...)? I was think in AsteriskNow does
this? Any advise?

Regards
Bilal


  
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Re: [asterisk-users] asterisk linkedin group

[dave cantera]

steve,
FYI:  randy randulo already has a voip group at  
http://food4wine.ning.com/
try that, it is already established...
daveC



BerkHolz, Steven wrote:

  
  
  
  
  asterisk
linkedin group
   
  I
have created an asterisk linkedin group for anyone interested.
   
  http://www.linkedin.com/e/gis/45252/66270A773F53
  Thank You,
  
  Steven BerkHolz
- MCSA - MCSE -
Manager of Information Systems
  HIROTEC
  AMERICA
  
Board member of
  Connectech Greater
Detroit
  www.connectech.org
   
  
  
  
  Please visit us on the web
at www.hirotecamerica.com
HIROTEC AMERICA Ph. 248-836-5100 Fx. 248-836-5101
  
This e-mail and any files transmitted with it are intended only for the
person or entity to which it is addressed and may contain confidential
material and/or material protected by law. Any retransmission or use of
this information may be a violation of that law. If you received this
in error, please contact the sender and delete the material.
  
  

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Re: [asterisk-users] Video Conference Or Server

[dave cantera]

here are some snippets from previous posts...  let us know what you like 
the best...

CrossPlatform Linux, Windows, Mac OpenSource WebHuddle at 
http://sourceforge.net/projects/webhuddle

I've tried dimdim and it was ok, but not as good as WiredRed. 

take a look at http://code.google.com/p/blindside/ and click on the 
screencast and Webconference demo


bilal ghayyad wrote:
> Hi All;
>
> Any one can advise for a good stable open source video
> conference or video server?
>
> Regards
> Bilal
>
>
>   
> 
> Never miss a thing.  Make Yahoo your home page. 
> http://www.yahoo.com/r/hs
>
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Re: [asterisk-users] rollback procedure requirements before asterisk upgrade

[dave cantera]

marco,
I use 1.4 exclusively but I would think a minor version would go pretty 
easy if you are installing from sources for the current version as well 
as the upgrade...
I would note (not a mental note, a written note) which source versions 
you are using for libpri, zaptel, *, and addons.  you already compiled 
them and they are working as a unit.
download the new versions, and compile and install as usual..  if you 
added additional modules in the current version, do the same for the new 
version...
you should always be able to drop back by following a standard install 
using the previously noted versions...
that should do it..
daveC

Marco Mouta wrote:
> Dear all,
>
>
> I've a live system that needs to be upgraded but, before I proceed to 
> the upgrade I want to assure the rollback process.
>
> That's why I'm requesting your feedback, in fact this asterisk in live 
> system isn't going so bad but the upgrade is essential
>
> NOTICE that the upgrade will keep the same version 1.2 not from 1.2 to 1.4
>
> Requirements:
>
> -backup /usr/sbin/asterisk
> -backup /usr/lib/asterisk/modules
>
> Download  asterisk 1.2.25 and compile it.
>
> This way I've tested and seems to work fine in my Virtual Machine lab.
>
> The only issue i found was one module that is loaded with my 
> modules.conf that I needed to copy from the backup 
> /usr/lib/asterisk/modules and  give the right permissions.
>
> Am I missing something?
>
>
>
> best regards,
>
> Marco Mouta
>
> -- 
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> 01:11 PM
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Re: [asterisk-users] OT - Callto:// tags options

[dave cantera]

oliver,
portsip.com has an sdk with a softphone applet... you might try googling 
'softphone applet'
there was another java softphone promoted somewhere too, so try 
'softphone sdk java'
could get you closer to a solution
daveC


Olivier wrote:
> Hello,
>
> From a previous thread, I learned Callto:// tags can be used inside 
> web pages to mean "dial a new call to the mentioned phone number".
> Is there any pointer explaining available options ?
>
> I'm after something meaning "transfer ongoing call to the mentioned 
> phone extension" instead of "dial a new call".
>
> Google replied me this :
> http://msdn2.microsoft.com/en-us/library/ms709071.aspx
>
> Anything else relevant ?
>
> Regards
> 
>
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> 06:20 PM
>   

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Re: [asterisk-users] OT - Fax and anti-spam

[dave cantera]

I did some research on spam filter about a year ago.  there are image 
analyzers that can detect human skin tones in images detecting porn.  I 
have seen some examples of how the porn guys speckle the images to 
obscure, somewhat, the naked bodies. 
the OCR idea would be useful but the OCR engine would have to be pretty 
good and there is not many of those.  you would have to apply the same 
filtering techniques that email spam employes, there are several, and it 
would take a considerable effort.  just like the email spammers change 
domain names, the fax spammers change phone numbers...
ugh...
daveC

Tzafrir Cohen wrote:
> On Wed, Dec 12, 2007 at 10:37:45AM +, lotusscript wrote:
>   
>> Never tried this but what about running the fax through an OCR ( gocr ) 
>> then composing a mail with sendmail putting the text recovered from the 
>> fax in the body and appending the original tiff image to the mail.
>> 
>
> How well does this work with non-spam faxes?
>
> I figure that spammers may take extra steps to make OCR difficult (see
> e.g. the wikipedia article or your spam folder). But providing the
> content as text can probably be a useful addition for fax2mail.
>
>   

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Re: [asterisk-users] Pickup over IAX

[dave cantera]

google app_pickup2, i just found it myself...
oh, still have the URL up...  here it is..
http://www.thorsten-knabe.de/linux/asterisk/pickup.jsp

Lukassky wrote:
> Hi everybody again. 
> A week ago I started a new Term about Pickup group over IAX or mISDN. I've
> set all the config up with callgroup and pikcupgroup to everything (IAX;
> SIP; mISDN; ZAP) but I'm not able to pick a call up if it's comes out from a
> SIP account. --> Pickup (${EXTEN}) . It's possible to find out another
> config for this?
> Thanks a lot.
>
> .:Luka§ky :.
>
>
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Re: [asterisk-users] Pickup re-invite

[dave cantera]

tim,
sounds like a problem I had with bandwidth... too many devices
communicating on the same network connection to the internet...
have you tcpdump'd or used a bandwidth tool to see what the usage is?
nat=yes or nat=no?  should be yes..
did you change the router between upgrades?
just some random thoughts..
daveC



Tim St. Pierre wrote:

  Hello Folks.

I'm wondering if anyone has any helpful hints.

I recently upgraded to 1.4.11, and I'm having problems with pickup, both 
directed, and the pickup feature.

My server is on the public internet, and all phones are behind a NAT router, 
somewhere else on the public internet.

When a ringing phone is picked up by another phone, you have audio for a few 
seconds, then the call is dropped.

The console shows "No response to our critical packet"

A SIP debug of the conversation between the phone and the server shows a 
re-invite request right when the call drops.  The phone is of course using 
the internal IP address as it's contact, and it looks to me like the server 
is trying to use it.

I have canreinvite=no for both the general sip.conf, as well as per-peer.

I am using the whole range of Aastra Enterprise IP phones.

Interestingly enough, some phones show their true IP address and port in the 
Asterisk registration database.  I believe this is where the phones have 
successfully communicated with a uPNP router, and discovered their public 
address.  These phones can successfully pickup the call.

If I pipe the pickup call through the Local channel, it works.

Why is asterisk still trying to re-invite even though I have explicitly told 
it not to in the config?

It worked fine in 1.2

Any suggestions, or requests for more information?

Thanks for any help.

-Tim
  


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Re: [asterisk-users] text management

[dave cantera]

silvia,

I don't know how to pickup the message but if it is getting into the
dailplan as a variable, you can send it to an AGI() script as a
parameter...
AGI(my_script.php,${IM_TEXT})
if you give me an example of what you have already, perhaps I can think
on it more...
daveC

cimsi wrote:

  Hi,
I know that Asterisk doesn't support Instant Messaging, but I'm trying to use the AGI function RECEIVE TEXT to implement a kind of IM service.
I have a sip softphone that tries to send a message to an active channel and the AGI script that expect to receive the text through the STDIN. 
Two problems arise:
First: How can I say to asterisk to get the message? (I see on CLI console that the message arrives to asterisk but it drops it)
Second: how can I put this message in STDIN to let the AGI read it? 

Has anybody used this feature? Can someone give an example of how to use it?

Any asuggestion would be appreciated.. thank you.
Silvia

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Re: [asterisk-users] asterisk 1.4 with around 230 SIP connections

[dave cantera]

speaking of multi-casting voice.  since it isn't likely to get the ip 
phones changed, could an app_multicast do the job?
has anyone thought of doing that?
daveC

Kristian Kielhofner wrote:
> On Dec 10, 2007 1:17 PM, Jerry Geis <[EMAIL PROTECTED]> wrote:
>   
>> Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a
>> 64 bit 4200+ box
>> would there be any noticable lag or delay to bring each one of them into
>> a PAGE mode. so one speaker can talk out on all 230 SIP clients for a
>> message.
>>
>> Would this work?
>>
>> Thanks,
>>
>> Jerry
>>
>> 
>
> I would also be really concerned about the ability for the NIC to
> serve up all of those RTP streams...
>
> 50pps x 230 = 11,500pps
>
> It would be nice to have some support for RTP multicast or something.
> Obviously this would require changes in Asterisk AND support in each
> phone, but it would be really cool.  I think I've seen some
> Linksys/Sipura devices support it.
>
>   

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Re: [asterisk-users] Appending two voice files

[dave cantera]

bart,
one way is to write the recorded files to a known directory, then
launch an AGI script to use sox to combine/concatenate the two...
if you cat them into a known filename, just use the playback() cmd to
play it.
do you need specifics?
daveC

Bart Fisher wrote:

  
  
  
  Does anyone know how I can append to
different user recorded voice files within a dial plan?  For example
Asterisk ask caller a question and records the answer, then ask another
question record the answer to the end of the first answer - so when
it's played back, all the answers are in one playback.
   
  TIA
   
  Bart
  

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No virus found in this incoming message.
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Re: [asterisk-users] Pickup cmd

[dave cantera]

rilawich,
in the CLI> type the following:

CLI> dialplan show [EMAIL PROTECTED]

then

CLI> dialplan show [EMAIL PROTECTED]
-or-
CLI> dialplan show [EMAIL PROTECTED]

and see if * recognizes the x100 in either of those...
daveC


Rilawich Ango wrote:

  HI,
  I have tried to add the context but it still doesn't work.

On Dec 9, 2007 11:36 PM, F6HQZ <[EMAIL PROTECTED]> wrote:
  
  
Hi,

Your extension "100" doesn't exist in the context where you have your PickUp
instruction.
You must include the context containing your phones into the context used by
your PickUp instruction or the reverse, or precise the context to use with
PickUp by adding it into the instruction line :

[BLF_group_pickup]
exten => _**1XX,1,Pickup(${EXTEN:[EMAIL PROTECTED])
exten => _**1XX,n,Hangup

Best Regards,
Francois BERGERET
France


  
  
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Re: [asterisk-users] Any idea how making Asterisk "transparent"?

[dave cantera]

artifex,
if you want call recording transparently, check out orecX.com   they 
have a commercial and an open source SIP call recording package...   no 
zap recording but if you are forwarding to sip exensions, you should be 
golden!   saw them at VON 2007 boston...  they have a recorded calls 
database lookup and web interface too...  very interesting...
daveC



Artifex Maximus wrote:
> Hello!
>
> I am using Asterisk as transparent voice recorder for calls (isdn <->
> asterisk <-> pbx). Voice recording (therefore voice forwarding) is
> working great but seems that Asterisk does not route/bridge/forward
> D-Channel messages which means PBX cannot get time synchronization
> answer from provider and tarification impulse too. With direct
> connection PBX works great and use both synchronization and give
> impulse value so there must be problem on Asterisk side.
>
> Machine is using lastest versions of Asterisk 1.2 branch (at time of
> writing: zaptel 1.2.22, libpri 1.2.6, asterisk 1.2.24) on Fedora Core
> 4. I tried with facilityenable=yes as well without success. I do not
> exactly know what facilityenable for.
>
> Is Asterisk capable forwarding D-Channel and making Asterisk box
> totally transparent? If yes which version? If branch 1.2 is capable
> how should I setup it right?
>
> Thanks.
>
> bye,
> a
>
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Re: [asterisk-users] Answering Machine Detection

[dave cantera]

carlos,
you got further than I did... AMD didn't work at all on my release.. I 
think I was using 1.4.11 at the time...
I ended up using the below
daveC

;---< amdtest (ext 13) starts here >
;
; restructure this for the following conditions:
; 13 using waitforsilence(variable set) then play message
; only works when there is an answering machine picking up and it doesn't
; cut off (hangup) before SILENCEDURATION ms
exten => 13,1,NoOp( Starting exten 13 AMD stuff)
exten => 13,n,Wait(1)
exten => 13,n,Set(SILENCEDURATION=4300)
exten => 13,n,Set(SILENCEOCCURANCES="")
exten => 13,n,Set(SILENCETIMEOUT=38)
exten => 13,n,NoOp( SILENCEDURATION=${SILENCEDURATION} )
exten => 13,n,NoOp( SILENCEOCCURANCES=${SILENCEOCCURANCES} )
exten => 13,n,NoOp( SILENCETIMEOUT=${SILENCETIMEOUT} )
;exten => 13,n,Answer
exten => 
13,n,WaitForSilence(${SILENCEDURATION},${SILENCEOCCURANCES},${SILENCETIMEOUT})
exten => 13,n,NoOp(Retnd WAITSTATUS=${WAITSTATUS} )
exten => 13,n,Playback(lax/lax-important-msg-from)
exten => 13,n,PlayBack(lax/to-hear-msg-press-1)
exten => 13,n,Read(CALL_ACK,beep,1,,,3)
exten => 13,n,NoOp(CALL_ACK is >${CALL_ACK}<)
exten => 13,n,GotoIf([${CALL_ACK} = ""]?nak13)
exten => 13,n(ack13),NoOp( Ack )
exten => 13,n,NoOp( log the ACK acknowlegement here calling the AGI script)
;exten => 13,n,AGI(lax/track-laxcalls.sh,${EXTEN},${CALL_ACK})
exten => 13,n,GotoIf([${CALL_ACK} = ""]?play13)
exten => 13,n(nak13),NoOp( Nak )
exten => 13,n,NoOp( log the NAK acknowlegement here calling the AGI script)
;exten => 13,n,AGI(lax/track-laxcalls.sh,${EXTEN},${CALL_ACK})
exten => 13,n(play13),Playback(lax/lax-important-msg-from)
exten => 13,n,Playback(tt-weasels)
exten => 13,n,Playback(tt-monkeysintro)
exten => 13,n,Wait(1)
exten => 13,n,Hangup





Carlos Chavez wrote:
>   I am having a bit of a problem getting AMD to work on a new server.  On
> my regular office server it works like a charm.  I am running Asterisk
> 1.4.13, Zaptel 1.4.5.1 on both machines.  Both servers run CentOS 5 and
> I am using a SIP trunk to send out calls (the same one on both servers).
>
>   Here is the output of a call on my office server:
>
> -- Attempting call on Local/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 
> (Retry
> 1)
> -- Executing [EMAIL PROTECTED]:1]
> Set("Local/[EMAIL PROTECTED],2", "CIDTEMP="1" <5540881644>") in new
> stack
> -- Executing [EMAIL PROTECTED]:2]
> Dial("Local/[EMAIL PROTECTED],2", "SIP/protel-out/0445540881644|
> 25") in new stack
> -- Called protel-out/0445540881644
> -- SIP/protel-out-0934bb28 is making progress passing it to
> Local/[EMAIL PROTECTED],2
> -- SIP/protel-out-0934bb28 answered Local/[EMAIL PROTECTED],2
> -- Executing [EMAIL PROTECTED]:1] NoOp("Local/[EMAIL PROTECTED],1", ""1"
> <5540881644>") in new stack
> -- Executing [EMAIL PROTECTED]:2] AMD("Local/[EMAIL PROTECTED],1", "")
> in new stack
> -- AMD: Local/[EMAIL PROTECTED],1 5540881644 (null) (Fmt: 64)
> -- AMD: initialSilence [2500] greeting [1500] afterGreetingSilence
> [800] totalAnalysisTime [5000] minimumWordLength [100]
> betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold
> [256] 
>   == Spawn extension (CC2, 0445540881644, 2) exited non-zero on
> 'Local/[EMAIL PROTECTED],2'
> -- Executing [EMAIL PROTECTED]:1] DeadAGI("Local/[EMAIL PROTECTED],2",
> "agi://localhost/updateCallStatus.agi?callStatus=hangupcc2") in new
> stack
> -- AGI Script
> agi://localhost/updateCallStatus.agi?callStatus=hangupcc2 completed,
> returning 0
> -- AMD: Word detected. iWordsCount:1
> -- AMD: Changed state to STATE_IN_SILENCE
> -- AMD: HUMAN: silenceDuration:800 afterGreetingSilence:800
> -- Executing [EMAIL PROTECTED]:3] GotoIf("SIP/protel-out-0934bb28", 
> "1?7:4")
> in new stack
> -- Goto (CC,2001,7)
> -- Executing [EMAIL PROTECTED]:7] AGI("SIP/protel-out-0934bb28",
> "agi://localhost/updateCallStatus.agi?callStatus=answered") in new stack
> -- AGI Script
> agi://localhost/updateCallStatus.agi?callStatus=answered completed,
> returning 0
> -- Executing [EMAIL PROTECTED]:8] Set("SIP/protel-out-0934bb28",
> "CALLERID(all)=") in new stack
> -- Executing [EMAIL PROTECTED]:9] MixMonitor("SIP/protel-out-0934bb28",
> "1192468625.7.wav|b") in new stack
> -- Executing [EMAIL PROTECTED]:10] Dial("SIP/protel-out-0934bb28", 
> "SIP/2001|
> 20") in new stack
> -- Called 2001
>   == Begin MixMonitor Recording SIP/protel-out-0934bb28
>
>
>   And here is the output on the new server:
>
>  -- Attempting call on Local/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 (Retry 
> 1)
> -- Executing [EMAIL PROTECTED]:1]
> Set("Local/[EMAIL PROTECTED],2", "CIDTEMP="1" <5540881644>") in new
> stack
> -- Executing [EMAIL PROTECTED]:2]
> Dial("Local/[EMAIL PROTECTED],2", "SIP/protel-out/0445540881644|
> 25") in new stack
> -- Called protel-out/0445540881644
> -- SIP/protel-out-09ce0358 is making progress passing it to
> Local/[EMAIL PROTECTED],2
>   

Re: [asterisk-users] get SIP extension status without calling it

[dave cantera]

vieri,
you can get sip status with the following shell script...   I named it 
'sipshowpeer'...   to execute, chmod 755 sipshowpeers
daveC

--< cut here >-
#!/bin/sh
#   sipshowpeers
#
#   show current asterisk SIP peers

asterisk -r -x 'sip show peers' | awk '
  BEGIN{
#Name/username  HostDyn Nat ACL Port Status
# $1$2  $3  $4   $5  $6   $7
  }
  {
name=$1
host=$2
dyn=$3
nat=$4
acl=$5
port=$6
status=$7
  printf("%14.14s %18.18s %14.14s %14.14s %s\n",$1,$2,$3,$4,$5,$6,$7)
  }
  END{
printf("Done...\n")
  }'
#502(Unspecified)D  0Unmonitored
#501(Unspecified)D  0Unmonitored
#40310.10.15.43 5060 Unmonitored
#40210.10.15.42 5060 Unmonitored
#401/401192.168.15.100   D  5062 Unmonitored
#301/301192.168.15.31D  5060 Unmonitored
#300/300192.168.15.31D  5060 Unmonitored
--< cut here 
 >---








Vieri wrote:
> Hi,
>
> I am trying to get a SIP extension's status without
> actually making a call.
>
> I am using sofia-sip's "options" example utility and
> the sip clients are SJphone softphones.
>
> >From Asterisk I run the "options" utility and query a
> sip extension at 10.215.147.240. I get:
>
> # ./options -1 --all sip:10.215.147.240
> SIP/2.0 501 Not Implemented
> Via: SIP/2.0/UDP
> 10.215.144.27:38098;branch=z9hG4bKUKS02S3F8H8ZS;received=10.215.144.27
> From: ;tag=U3DKgF7HgFKXH
> To: "unknown" ;tag=614733430
> Call-ID: b6968197-1b7d-122b-0ab0-00c09f10e472
> CSeq: 92182805 OPTIONS
> Content-Length: 0
> Server: SJphone/1.65.377a (SJ Labs)
>
> I guess that the softphone should be answering with a
> 2xx code followed by a status description?
> So I tried with the INVITE method and set DND on the
> SIP extension:
>
> # ./options -1 --all --method INVITE
> sip:10.215.147.240
> SIP/2.0 486 Busy Here
> Via: SIP/2.0/UDP
> 10.215.144.27:38098;branch=z9hG4bK5Z3BS8F737t0e;received=10.215.144.27
> From: ;tag=590Z1ND8B6XpN
> To: "unknown" ;tag=1a2d77b524
> Call-ID: 668ad4fa-1b7e-122b-fcb6-00c09f10e472
> CSeq: 92182952 INVITE
> Content-Length: 0
> Server: SJphone/1.65.377a (SJ Labs)
>
> The above would suit me fine because I get a "486 Busy
> Here" response.
> However, if DND is off then I get:
>
> # ./options -1 --all --method INVITE
> sip:10.215.147.240
> SIP/2.0 180 Ringing
>
> and the SIP extension actually "rings", as
> expected.(but this is undesireable)
>
> Now, does someone know another way to get the status
> (ie. does it accept calls or not?) without making the
> extension "ring"?
>
> Thanks
>
> Vieri
>
>
>
>   
> 
> Be a better pen pal. 
> Text or chat with friends inside Yahoo! Mail. See how.  
> http://overview.mail.yahoo.com/
>
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>
>
>   

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I should buy her a Videophone2008!

Truly, The Next Best Thing to Being There!
--

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856.380.0894




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Re: [asterisk-users] Consulting/Integration Services Non-US & US *u

[dave cantera]

steve,
oops, you are right... sorry.. wrong list...
daveC

Steve Edwards wrote:

  On Sat, 1 Dec 2007, dave cantera wrote:

[snip]

You forgot "i don't know what the shift key is" and "i don't understand 
what Non-Commercial Discussion means."

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Truly, The Next Best Thing to Being There!
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856.380.0894






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[asterisk-users] Consulting/Integration Services Non-US & US *u

[dave cantera]

to all,
I am available for work either US or Non-US for * consulting, 
configuring, integration with other business applications.  have been 
working with * for about three years on and off and would like to do 
this full time.   am available for on-site or remote project work.

have 20+ years UNIX/Linux (SuSE, redhat, debian, knoppix, lamppix, 
slackware, cdrouter, etc) system and application integration 
experience.  have ISP experience, medical answering service experience, 
full life cycle software design and development in the gov't, financial, 
and private sectors.   have worked with open source products apache, 
dns, inn, ntp, majordomo, sugarCRM, joomla, and countless others...

please contact me off list for specifics and to discuss potential 
projects.  I am in the philadelphia, US area.
thanks,
daveC

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Re: [asterisk-users] AsteriskNOW and TDM800P

[dave cantera]

rafael,
it should work.  both systems are auto configurable...
daveC

Rafael Canchola wrote:
>
> Hi all
>
> I sold new TDM800P card with 8 FXO ports, someone know if can be use 
> this card on AsteriskNOW or trixbox?
> What can i do for use this card?
>
> Thanks.
>
> 
> */ Rafael/*/Canchola
> //*Product Development Engineer*/*,
> Fonet*Global Inc.
> [EMAIL PROTECTED] 
> http://www.fonetglobal.com
> *Ph. *+ 52 800 022 10 21 ext. 214
>   + 52 442 167 08 00
> *VoIP* 523663899
> *d00d! *cyberalph
> 
>
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> 
>
> No virus found in this incoming message.
> Checked by AVG Free Edition. 
> Version: 7.5.503 / Virus Database: 269.15.17/1103 - Release Date: 11/01/2007 
> 06:01 AM
>   

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I should buy her a Videophone2008!

Truly, The Next Best Thing to Being There!
--

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856.380.0894




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[asterisk-users] Playback() clicking sound at the end of the prompt

[dave cantera]

does anyone know how to stop the clicking sound that happens at the end 
of a playback() command?
is it something I can do in the recording?
I looked in the 'book' but there was only a 'j' option...
thanks,
daveC


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Re: [asterisk-users] [asterisk-biz] Live Conference about asterisk and voip: reminder 12:30 PM EDT Friday

[dave cantera]

for those of you who have not joined the conference call yet, I highly
recommend it.  there is always several interesting tidbits that will
help you in your * implementations...
see you at 12:30p today!
daveC




randulo wrote:

  As usual, we'll be jawing about any and all asterisk-related subjects
with the usual gang and any new people are always welcome, regardless
of your level of expertise. You can even come and ask questions, it's
guaranteed to be a more pleasant experience than it will be on IRC ;)

http://VoipUsersConference.org/topics.php

IRC; Freenode.net #voip-users-conference

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Truly, The Next Best Thing to Being There!
--

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856.380.0894






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Re: [asterisk-users] [asterisk-biz] Live Conference about asterisk and voip: reminder 12:30 PM EDT Friday

[dave cantera]

for those of you who have not joined the conference call yet, I highly
recommend it.  there is always several interesting tidbits that will
help you in your * implementations...
see you at 12:30p today!
daveC




randulo wrote:

  As usual, we'll be jawing about any and all asterisk-related subjects
with the usual gang and any new people are always welcome, regardless
of your level of expertise. You can even come and ask questions, it's
guaranteed to be a more pleasant experience than it will be on IRC ;)

http://VoipUsersConference.org/topics.php

IRC; Freenode.net #voip-users-conference

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--

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856.380.0894






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Re: [asterisk-users] Skills Based Routing

[dave cantera]

nick,
I am actually playing with skills based routing right now... 
how would you propose to send multiple calls requiring different skills 
into a single queue and have agents w/o that particular skill in the 
same queue?
daveC

Nick Brown wrote:
> Morning All,
>
> Has anyone here successfully implemented skills based routing within queues?
>
> The concept behind skills based routing is fairly straight forward, and I
> know I could do it with multiple queues, agent penalties and a bit of AGI to
> put the call into the right queue.
>
> However doing this is going to require the addition of several extra queues
> and isn't a very clean solution.
>
> The other alternative is to write our own queue system with AGI, effort++
> though :-)
>
> TIA.
>
> Cheers,
> Nick.
>
>  
>
>
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>   

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--

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Re: [asterisk-users] Testing Framework

[dave cantera]

matt,
are you looking for unit testing of the * components or systems testing, 
testing the finished product?  or both?
I think you are onto something here...  I hope it takes root.  I would 
say put it in the addons.  it would be Great if digium takes it up. it 
is a smart move for them to foster, cajole, nudge, and support it. 
call volume I would leave to others as different processors, O/S, 
builds, kernel versions, and configurations will have too many variables.

I was playing with the idea of monitoring multiple * systems.  perhaps 
we can start out with testing the components and then migrate the 
project (future) to one pbx monitor the other.  we will need scripts to 
initiate some action, config to make some measurements, the scripts to 
gather the results into a nice neat little summary report.  you will 
want to take the human aspect out of the picture as much as possible.  
for example:

on pbx A

* create a recording in multiple formats .gsm, .wav, etc.
* initiate a script to generate 5,10, or 25 calls to pbx B and
  play the file

on pbx B

* pbx B gets the calls, records them,
* copy the recordings from pbx A to pbx B (or have that already
  done)
* have a wave analyzer compare the recordings to the original
  files (you know I won't be writing that program! :)
* report on anomalies

*call
*   *Technology
*   *recording
delta
*
1
Zap Provider 1
2%
2
VoIP Provider 2
5%
3
VoIP Provider 2
15%
...
VoIP Provider 3
...


let me know what you think!
daveC



Matt Riddell wrote:
> Hash: SHA1
>
> Hi,
>
> So, now that we've all complained about the state of testing of Open
> Source versions of Asterisk, lets do something about it.
>
> I propose we start with a list of things that we think should be tested
> in Asterisk, and means to test them.
>
> Maybe we could run certain tests based on the changes between minor
> versions?
>
> Anyway lets start.
>
> Call Volumes
>
> 1) Call volume up to x channels from SIP to SIP (i.e. sipp)
> 2) Call volume up to x channels from IAX2 to SIP
> 3) Call volume up to x channels from IAX2 to IAX2
>
> Application testing
>
> 4) Connect x calls between techs to Meetme (leave running for 1 hour)
> 5) Connect x concurrent calls to VoiceMail
>
> Call Centre Testing
>
> 6) Send x calls to a queue with no agents in it, leave them holding for
> x minutes
> 7) Run x calls against AMD connected to recorded known good files
>
> Recording
>
> 8) Run x calls recording simultaneously from an automatically generated
> call, play ulaw/alaw - compare outputs.
>
> You get the idea.
>
> If people can add to this list, I can start making a few scripts and
> programs that will test them (as I'm sure others can).
>
> If we end up with a complete list, I'm sure some of our individual QA
> departments can take the responsibility for certain items.
>
> The call volume ones are obviously going to either need a live person to
> dial in at volume and check everything is ok, or a recording which can
> later be checked.
>
> I'm of the opinion that the majority of tests should test individual
> components, but that we should also form some "Application Type"
> frameworks so that we can test integration between Asterisk apps.
>
> Any takers?  Add to the list?  If there is something you believe is
> mission critical to your business, write up a test case for it, and
> we'll all try to code something that can run automatically to test it.
>
> If we try and keep to ANSI C for the testing apps, Digium should be able
> to run them on their multi platform machines as well.
>
> Should these tests be added to Asterisk-Addons or maintained outside of
> the tree?
>
> Anyway, what do you think? Feasible? I already have a few tests here and
> I'm sure others have a few too.  Lets put them all together and get a
> framework going.
>
> - --
> Kind Regards,
>
> Matt Riddell
> Director
> ___
>
> http://www.venturevoip.com (Great new VoIP end to end solution)
> http://www.venturevoip.com/news.php (Daily Asterisk News - html)
> http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.7 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>
> iD8DBQFG1yKhDQNt8rg0Kp4RAv5UAJ48tW28T5lWCQIPTwVimyvlhEPJowCgpnE6
> OF3L2M/6Hc+YBNL1NFx6dzA=
> =OXNn
> -END PGP SIGNATURE-
>
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Truly, The Next Best Thing to Being There!
--

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__

[asterisk-users] check out the cursor movement on this website!!!

[dave cantera]

picturephone -dot- com/





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Re: [asterisk-users] 1and1 dedicated servers have been down for a few hours .

[dave cantera]

my shared webhosting is going strong...
daveC

Asterisk guy wrote:
> 1and1 dedicated server's service  has  been down for a few hours  , 
> unable to reach them by phone or email. do anyone know what is going 
> on there ?
>  
> Mario
> 
>
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>
> No virus found in this incoming message.
> Checked by AVG Free Edition. 
> Version: 7.5.476 / Virus Database: 269.10.22/921 - Release Date: 07/26/2007 
> 11:16 PM
>   

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Re: [asterisk-users] IAX connections broken

[dave cantera]

michael,
this is what I use for centOS 4, but I think its too loose... let me 
know if you don't know where to put it...
daveC

# for asterisk
-A RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp -m udp --dport 4569 -j ACCEPT < IAX
-A RH-Firewall-1-INPUT -p udp -m udp --dport 5036 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp -m udp --dport 1:2 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp -m udp --dport 5004 -j ACCEPT





Michael Munger wrote:
>
> It did change, which is what caused this problem in the first place, 
> but all the updates have been applied, propagated, and are 
> working….well, with the exception of this one.
>
> Does anyone know what the iptables command would be to forward these 
> IAX packets to a specific LAN ip?
>
> Michael Munger
>
> High Powered Help, Inc
>
> [EMAIL PROTECTED] 
>
> 404-438-2128 x 101
>
> 
>
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Dave Bour
> *Sent:* Thursday, July 26, 2007 12:29 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* Re: [asterisk-users] IAX connections broken
>
> Are sites listed by IP or DN. If IP, dumb question but did it change? 
> If DN, can you resolve it from the respective boxea?
>
> Dave Bour
> Desktop Solution Center
> 905.381.0077
> [EMAIL PROTECTED]
>
> For those who just want it to work...
> Giving you complete IT peace of mind.
>
> (Sent via Blackberry - hence message may be shorter than my usual 
> verbose responses)
> PIN 4cc364db (as of March 24, 2007)
>
> - Original Message -
> From: [EMAIL PROTECTED] 
> <[EMAIL PROTECTED]>
> To: [EMAIL PROTECTED] <[EMAIL PROTECTED]>; Asterisk 
> Users Mailing List - Non-Commercial Discussion 
> 
> Sent: Thu Jul 26 10:17:23 2007
> Subject: Re: [asterisk-users] IAX connections broken
>
> Not likely.
> #1, I have a public IP on that firewall.
> #2. If I block 4569 at our firewall, then it goes from closed to
> stealth. If I forward the port, it goes from stealth to closed.
>
> The iaxping tool (http://www.bpvn.com/asterisk/iaxping.zip) has no
> problems pinging the box from the lan, and our test machine can make an
> IAX connection to the box. From outside the network, however, it times
> out.
>
> It has to be a NAT problem, but forwarding doesn't appear to be working.
>
> Yours,
> Michael Munger, dCAP
> 404-438-2128
> [EMAIL PROTECTED]
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Baji
> Panchumarti
> Sent: Thursday, July 26, 2007 10:06 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] IAX connections broken
>
> what if your internet provider is blocking inbound 4569 ?
>
> --
>
> On 7/26/07, Michael Munger wrote:
>
> > Dear All:
> >
> > I have several boxes that up and running just great, then we changed
> > internet equipment due to a lightning strike, now all my inbound IAX
> > connections (iax2 show peers) have unknown status. If I log into the
> > remote boxes, it says "Request sent."
> >
> > The authentications haven't changed at all, and all the iax.conf
> > settings are correct. It looks like a firewall issue, but we've got
> 4569
> > TCP & UDP forwarded to our Asterisk box. When I use Shields up from
> > GRC.com to test the port, it is showing up as "closed" rather than
> open,
> > which normally means the port is open, but the service is not running,
> > yet Asterisk is up and running just fine, and my outbound connections
> to
> > Voicepulse work fine. I see voicepulse, voicepulse sees me.
> >
> > There is something I am not seeing here. Any thoughts?
> >
> > -Michael
> >
> > ___
>
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>
> No virus found in this incoming message.
> Checked by AVG Free Edition. 
> Version: 7.5.476 / Virus Database: 269.10.22/921 - Release Date: 07/26/2007 
> 11:16 PM
>   

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I should buy her a Videophone2008!

Truly, The Next Best Thing to Being There!
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Re: [asterisk-users] Calling to users in other asterisk servers

[dave cantera]

aryjunior,
is your dialplan and registration configured to connect to another * 
server?...include your config so we can analyze it...
daveC

Carlos Rojas wrote:
> Hello,
>
> Do you have porf forwardin for SIP protocol in your firewall?
>
> SIP:  5060  udp 
>
> rtp  1 - 2 udp (default)
>
> and IAX2 4569  udp
>
>
> Best Regards
>
>
> Carlos Rojas
>
> On 7/28/07, *Ary Junior* <[EMAIL PROTECTED] 
> > wrote:
>
> Hi, Im a asterisk newbie and I've configured an asterisk server
> here in my house... in my LAN two users can login and call to each
> other, but when I try to call an user in another asterisk server
> outside my LAN ( sip:[EMAIL PROTECTED]
>  ) it dont work... if the
> person outside is conected on my server it works fine... My
> asterisk server is behind a firewall and portfowarding... it is
> possible?
>  
> Thanks very much!!!
>
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>
> No virus found in this incoming message.
> Checked by AVG Free Edition. 
> Version: 7.5.476 / Virus Database: 269.10.22/921 - Release Date: 07/26/2007 
> 11:16 PM
>   

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I should buy her a Videophone2008!

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Re: [asterisk-users] Locking a device to a codec

[dave cantera]

baji, mhoppes,
remember, if you have Only the g729 codec allowed or if this is the
only allow= entry in the sip.conf file, callers requesting any other
codec will be rejected
daveC

Baji Panchumarti wrote:

  On 7/27/07, Matt <[EMAIL PROTECTED]> wrote:
  
  
Can someone comfirm my logic here?

If I want a phone to use G729 I can set it to use G729... do I
also need to set it in Asterisk?  I'm thinking no... as long as
asterisk WILL do G729... if that's all the device accepts it should go
to that codec, yes?

  
  
  (based on my understanding, take it for what it is worth)

   if  allow=all   orallow=g729   is in your
  asterisk configuration (sip.conf / iax.conf ) then asterisk will
  stream packets in g729 (assuming you have any licesnses
  needed in place).

  -baji.

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Re: [asterisk-users] Asterisk Users Conference Friday at 12:30 PM EDT

[dave cantera]

randulo,
I could not get into the conference today...  the SIP line was busy, no 
matter what I do, the website thinks I'm not logged in and gives me the 
login page.  after I login, anything I want to do brings me back to the 
login page... so I tried to re-setup the account thinking I wasn't 
logging in, and the user name was taken  so I know I'm signed up. 

on the webpage, it claimed there were no conferences scheduled...
any thoughts?
daveC

randulo wrote:
> You can listen or join the Asterisk Users Conference Fridays at  12:30 
> PM EDT
>
> Today's subject suggestions:
>
> FAX capabilities, what's your solution?
> Multiple asterisk server implimentation: ENUM, DUNDI or even two 
> servers connected
> Your subjects?
>
> Share your ideas, ask your questions!
>
> See  http://x2z.eu  for instructions on how to join or listen
>
> irc://irc.freenode.net/asterisk-users-conference
>
> Note that the SIP channel will only be open from about 12:20PM EDT.
> Testing before then will give you the message "your PIN is not valid" 
> but if it answers, you're good.
>
> ; SIP call
> ; exten => AUC,1,Dial( SIP/[EMAIL PROTECTED] 
> ,60,D(22622#${YOUR_PIN}#))
>
> If you would like to talk about services or products your company 
> provides and answer users' questions, contact me off list. Anyone is 
> welcome to be a guest and answer users' questions.
>
> Previous guests have been Teliax, Lumenvox, Digium (duh!), Trixbox, 
> Adhearsion
>
> Listen to the archived recordings here:
>
>  http://x2z.eu/astusers.htm  
>
> The Asterisk Users Conference is independently run and has nothing to 
> do contractually or financially with Digium who owns the Asterisk 
> trademark.
> 
>
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>
> No virus found in this incoming message.
> Checked by AVG Free Edition. 
> Version: 7.5.476 / Virus Database: 269.10.22/921 - Release Date: 07/26/2007 
> 11:16 PM
>   

-- 
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I should buy her a Videophone2008!

Truly, The Next Best Thing to Being There!
--

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Re: [asterisk-users] tdm400p fxs module busy

[dave cantera]

matt,
I just had the same problem...  does your CLI> report   'unable to
create channel Zap/#'

post the CLI> output to help us determine the problem.
daveC

Matt Scott wrote:

  
  
  
  Dear All
   
  The
setup is te110p with an 8 channels PRI to make and receive all calls.
SIP phones throughout the company.
TDM400p with 4 FXS modules to send/receive faxes and make credit card
transactions.
  
I have an analogue phone on the tdm400p for testing.
I can receive calls to the exten. There is a dialing tone.
However, when I try to make a call I get a busy signal.
Asterisk stated busy then hungup zap/32-1
   
  why
wont asterisk supply a resource from the te110p pri card for use by the
tdm400p FXS (fxo signalling)?
  
configs below:
  
  
[EMAIL PROTECTED] etc]# more zaptel.conf
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
  
# It must be in the module loading order
  
  
# Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3/CCS RED
span = 1,0,0,ccs,hdb3,crc4
# termtype: te
bchan=1-8
dchan=16
  
# Span 2: WCTDM/0 "Wildcard TDM400P REV H Board 1"
fxoks=32
fxoks=33
fxoks=34
fxoks=35
  
# Global data
  
loadzone    = uk
defaultzone = uk
  
  
  
[EMAIL PROTECTED] asterisk]# more zapata.conf
[trunkgroups]
  
[channels]
  
language=en
internationalprefix = 00
nationalprefix = 0
context=from-pstn
switchtype=euroisdn
pridialplan=local
priindication=outofband
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
group=1
callgroup=0
pickupgroup=0
immediate=no
echotraining=yes
echocancel=yes
echocancelwhenbridged=no
facilityenable=yes
musiconhold=default
overlapdial=yes
immediate=no
txgain=0.0
rxgain=0.0
signalling = pri_cpe
channel => 1-8
  
faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
  
signalling = fxo_ks
echocancel=yes
pulsedial=yes
channel=32-35
  
  
  
[EMAIL PROTECTED] asterisk]# more extensions.conf
[general]
static=yes
writeprotect=yes
;
[globals]
OUTBOUND = Zap/g1
FAX1 = Zap/32
FAX2 = Zap/33
STREAMLINE1 = Zap/34
STREAMLINE2 = Zap/35
CUSTSERVE1 = SIP/401 ;Teresa
CUSTSERVE2 = SIP/402 ; Louise
;CUSTSERVE3 = SIP/404 ; Helen
QUAD1 = SIP/451 ; Matt
QUAD2 = SIP/452 ; Johan
CUSTSERVE = CUSTSERVE1&CUSTSERVE1
;
FSEXT1 = SIP/400 ; Angela
;FSEXT2 = SIP/403 ; Nigel
FSEXT3 = SIP/410 ; Matt
;
;ELLIS = SIP/411
;QUEENS = SIP/412
;FSSHOPS = ELLIS&QUEENS
;
QUAD = SIP/450
;
LONDONSOLE1 = SIP/421 ; Zoe
;LONDONSOLE2 = SIP/422 ; Laura
;LONDONSOLE = LONDONSOLE1&LONDONSOLE2
;
;PRESS1 = SIP/431 ; Lucy
;PRESS2 = SIP/432 ; Gemma
;PRESSOFFICE = PRESS1&PRESS2
;
[macro-oneline]
exten => s,1,Dial(${ARG1},20,t)
exten => s,2,Voicemail(u${MACRO_EXTEN})
exten => s,3,Hangup
exten => s,102,Voicemail(b${MACRO_EXTEN})
exten => s,103,Hangup
;
[macro-oneline1]
exten => s,1,Dial(${ARG1},20,t)
exten => s,2,Voicemail(u${ARG2})
exten => s,3,Hangup
exten => s,102,Voicemail(b${ARG2})
exten => s,103,Hangup
;
[macro-fax]
exten => s,1,Dial(${ARG1},20,t)
exten => s,3,Hangup
;
[default]
;setupdial out
include => from-pstn
;
;test dialplan
exten => _9xxx,1,SayDigits(${EXTEN:1})
;
;setup the phone extensions
exten => 400,1,Macro(oneline,${FSEXT1})
exten => 401,1,Macro(oneline,${CUSTSERVE1})
exten => 402,1,Macro(oneline,${CUSTSERVE2})
exten => 410,1,Macro(oneline,${FSEXT3})
exten => 421,1,Macro(oneline,${LONDONSOLE1})
exten => 450,1,Macro(oneline,${QUAD})
exten => 451,1,Macro(oneline,${QUAD1})
exten => 452,1,Macro(oneline,${QUAD2})
;
exten => 1000,1,Macro(oneline,${CUSTSERVE})
;exten => 2000,1,Macro(oneline,${FSSHOPS})
;exten => 3000,1,Macro(oneline,${PRESSOFFICE})
;
;record new voice files
Exten => 501,1,Wait(2)
Exten => 501,n,Record(/tmp/asterisk-recording:gsm)
Exten => 501,n,Wait(2)
Exten => 501,n,Playback(/tmp/asterisk-recording)
Exten => 501,n,wait(2)
Exten => 501,n,Hangup
;
;goto voicemail
exten=>*98,1,VoiceMailMain([EMAIL PROTECTED]})
;
[dialphone]
exten => 90,1,Macro(fax,${FAX1})
;
[from-pstn]
;this is linked to zapata.conf and defines where the ddi points
exten => 00,1,Dial(SIP/401&SIP/402,15)
exten => 00,2,Voicemail(1000)
;
exten => 769611,1,Macro(oneline1,${FSEXT1})
exten => 769615,1,Macro(oneline1,${LONDONSOLE1})
;exten => 769616,1,Macro(oneline1,${LONDONSOLE2})
exten => 769636,1,Macro(oneline1,${FSEXT1},${401})
;exten => 769637,1,Macro(oneline1,${NIGEL})
;
exten => _9.,1,Set(CALLERID(number)=00)
exten => _9.,2,Dial(${OUTBOUND}/${EXTEN:1})
exten => _9.,3,Congestion()
exten => _9.,102,Congestion()
;
exten => 999,1,Dial,(${OUTBOUND}/999)
exten => ,1,Dial,(${OUTBOUND}/999)
;
exten => 90,1,Dial(Zap/32,15)
  
  

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No virus found in this incomi

Re: [asterisk-users] WAV49 output in sox

[dave cantera]

eric
try this... 
sox foo.wav -r 8000 foo.gsm resample -ql
# add -c1 to write the file in mono

I can't remember if you have to do something special in the recording 
too  depends on your recorder.. oh, now I remember.  you have set 
the recording to 16bit 14400 hz or something like that...  if I find it 
I'll re-email
daveC



Eric "ManxPower" Wieling wrote:
> Does anyone know what options you need to use with "sox" to output the 
> audio in the WAV49 format that Asterisk uses.
>
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I should buy her a Videophone2008!

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Re: [asterisk-users] Asterisk 1.4.9.tar.gz download fails

[dave cantera]

ed,
do you positively have to have 1.4.0?
just download 1.4.9 or 1.4.8...  1.4.0 is too old...
I can email you 1.4.8, 1.4.5, 1.4.9...
I just downloaded 1.4.9 from:
http://www.digium.com/elqNow/elqRedir.htm?ref=http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.9.tar.gz

daveC

EdPimentl wrote:
> Hello Fellow Asterisk Mailing ListMembers,
>
> When I tried to download the latest version of Asterisk this is what I 
> get:
> http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz 
> 
> Opening fileinfo database failed
>
>
> http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz
> Opening fileinfo database failed
>
> Where are all the latest Asterisk 1.4.x source files?
>
> Thanks in advance,
> -E
>
>
>
> 
>
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>
> No virus found in this incoming message.
> Checked by AVG Free Edition. 
> Version: 7.5.476 / Virus Database: 269.10.12/910 - Release Date: 07/21/2007 
> 03:52 PM
>   

-- 
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I should buy her a Videophone2008!

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Re: [asterisk-users] Add prefix digits in dialplan extention

[dave cantera]

satish,
please clarify... 
do you want people to dial 1171 on the avaya system to get to you?
do you want people to dial 1171 on the * box to get to you?
do you want people to dial 71 on either box to get to you?
daveC


satish patel wrote:
> Dear all
>
>  I have asterisk 1.2 configuration and it is working 
> fine but thing is that i have alread Avaya setup and i have intergrate 
> my Linuxbox asterik with Avaya system avaya already use 4 digit 
> dialplan (1644 example ) and in asterisk i have configure 2 digit 
> dialplan ( 44  example ) now i want to configure 4 digit dialplan in 
> asterik without any change in avaya or asterisk so can i configure 
> dialplan like i use prefix 11 automatically dial and add 2 digit in my 
> local extention
>
> Example
>
>  My extention nuber is 71 now whn ppl want to make a call me they will 
> dial 71 and i will pickup the phone but now i want to add 2 more digit 
> in prefix like 1171 when ppl dial 1171 and i pickup the call and 
> talking is it possible in dialplan extention.conf
>
> Rgd
> satish patel
>
>
>
>
> 
> Get the Yahoo! toolbar and be alerted to new email 
> wherever
>  
> you're surfing.
> 
>
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> Version: 7.5.476 / Virus Database: 269.10.12/910 - Release Date: 07/21/2007 
> 03:52 PM
>   

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I should buy her a Videophone2008!

Truly, The Next Best Thing to Being There!
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[asterisk-users] Answering Machine Beep Detection for *

[dave cantera]

hi,
can anyone point me to answering machine beep detection methods or writeups for 
*?
thanks,
daveC


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Re: [asterisk-users] Wake-Up Call didn't work

[dave cantera]

on the CLI>   type this command:

dialplan show [EMAIL PROTECTED]  
-and-  
dialplan show [EMAIL PROTECTED]

you should see a dialplan returned to you.   if not, which is what I 
expect, you have to include the section [where6009is] in [local] or 
[default]... i.e.

[local]
include => where6009is
...

[default]
include => where6009is
...

[where6009is]
exten 6009,1,wait(2)
exten 6009,n,NoOp("getting to 6009")
...


Asterisk guy wrote:
> 1  there is a correct file in  /var/spool/asterisk/outgoing
>  
> 2  i run  asterisk -r to monitor it  , it gives out the following 
> error
>  
>  -- Attempting call on Local/[EMAIL PROTECTED]  PROTECTED]> 
> for application MusicOnHold() (Retry 1)
>
> Jul 24 08:23:17 NOTICE[21177]: chan_local.c:479 local_alloc: No such 
> extension/context [EMAIL PROTECTED]  creating local 
> channel
>
> Jul 24 08:23:17 NOTICE[21177]: channel.c:2409 __ast_request_and_dial: 
> Unable to request channel Local/[EMAIL PROTECTED]  PROTECTED]>
>  
>  
> ( but i have a extension 6009 login to * ) ,  what is the problem?
>
>
>  
> On 7/23/07, *James FitzGibbon* <[EMAIL PROTECTED] 
> > wrote:
>
> On 7/23/07, *Dovid B* < [EMAIL PROTECTED]
> > wrote:
>  
>
> Can it be that asterisk does not have permission to copy the
> file over ?  Also check your date settings on the server.
>
>
>
> Yes, it's interesting that the page intro includes the sentence
> "Lots of error checking to make sure its done correctly", but the
> final step that makes the process work (ensuring that the callfile
> ends up in the directory that pbx_spool is watching) doesn't have
> any error checking:
>
> touch( $wakefile, $time_wakeup, $time_wakeup );
>
> rename( $wakefile, $callfile );
>
> The fact that you see files in /tmp when all is said and done
> means that at least some of the script is working.  A few things
> to check:
>
> Do the files in /tmp have the correct timestamp (file matches the
> requested wakeup time)?  If so, then everything preceeding the
> rename seems to have worked, so check if the user running the AGI
> can move files from /tmp to /var/spool/asterisk/outgoing.  Though
> given that it's an AGI being run by *, you'd have to have a pretty
> strange setup for that to fail.  Perhaps the outgoing directory
> just doesn't exist (was never created for some reason?)
>
> If the files don't have the correct timestamp, start following the
> logic backwards.  Do they look complete?  Look through the AGI for
> places where the wakeup file is written to (i.e.
> fputs( $wuc, "maxretries: $parm_maxretries\n"); ) and check that
> everything that should be written is being written
>
> Working backwards you should be able to figure out where the
> script is failing, then you can check everything that comes
> afterwards as the user running the AGI to make sure that
> permissions and directories are set up properly.
>
> -- 
> j.
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> 03:52 PM
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Re: [asterisk-users] Debian etch and web voice mail - how to configure it?

[dave cantera]

jim,
asterisk does not provide an httpd itself... asteriskNOW does provide 
lightspeedhttpd.. as tzafrir said in his last email, you would have to 
move the vmail.cgi to the apache2 cgi-bin directory, then write an html 
page to execute it.  I would have to look at the application to give 
further insight.  if the link tzafrir provided is correct, I can do 
that...  just let me know.

what I tend to do is install asteriskNOW and then overwrite * with the 
latest version... doing anything else on that box is quite rough though... 
daveC


Jim Archer wrote:
> --On Sunday, July 22, 2007 1:17 PM -0400 dave cantera 
> <[EMAIL PROTECTED]> wrote:
>
>   
>> the asterisk gui doesn't interact with apache or apache2... it has it's
>> own httpd...  perhaps you can move the vmail.cgi script to the apache2
>> directory structure cgi-bin.  I haven't tried that as of yet so I don't
>> know how that would work.
>> 
>
> Hi Dave, thanks very much.  Well I have no burning desire to use Apache at 
> all.  The Debian package for web voice mail installed it.  I assumed it was 
> required since the package manager included it.  If I don't need it, great. 
> One less thing to maintain.  But, how do I activate the http server in 
> asterisk then?
>   
>
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Re: [asterisk-users] Debian etch and web voice mail - how to configure it?

[dave cantera]

jim,
the asterisk gui doesn't interact with apache or apache2... it has it's 
own httpd...  perhaps you can move the vmail.cgi script to the apache2 
directory structure cgi-bin.  I haven't tried that as of yet so I don't 
know how that would work.
daveC

Jim Archer wrote:
> Hi Everyone...
>
> I am running Asterisk 1.2.13 on Debian "Etch".  I installed it from the 
> package.  I also installed the web voice mail package, which installed 
> Apache2 and a bunch of other stuff.
>
> When I point my browser at my PBX machine, the web page says "It Works!" 
> but of course it does not.  It does not seem that Apache is configured to 
> run the vmail.cgi script.  In the docs directory there is just the change 
> log and googling it has not helped.
>
> Can someone give me a hint as to how to configure this or else point me at 
> some docs?
>
> Thanks very much.
>
>
>
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Re: [asterisk-users] MAKE Menuselect

[dave cantera]

kevin,

make menuselect - creates an xml file...  let me look to see where it is

[EMAIL PROTECTED] asterisk-1.4.5]# ls -l menu*
  Current Directory is /usr/local/src/asterisk-1.4.5
-rw-r--r--  1 root  2065 Jun 25 18:36 menuselect.makedeps
-rw-r--r--  1 root  1654 Jun 25 18:36 menuselect.makeopts
-rw-r--r--  1 root 37350 Jun 25 18:34 *menuselect-tree*

look in menuselect-tree, and...

hmm...  this looks promising for trying to figure it out...
  Current Directory is 
/usr/local/src/asterisk-1.4.5/menuselect
-rw-r--r--  1 root 31131 Aug 19  2006 example_menuselect-tree

daveC


Kevin Kiely wrote:
> Does anyone know a way in Asterisk 1.4 to select the options from the
> menuselect menu from the command line?
>
>
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Re: [asterisk-users] The purpose of DUNDi

[dave cantera]

remco, et al,
could I use dundi where I could use an area code to determine the 
connecting server or dial string?  just like we would use 88XXX to dial 
a 3 digit extension on another server at location 88?  or dial 84XXX for 
a 3 digit extension on a server located at 84?...

thanks,
daveC


Remco Post wrote:

Rilawich Ango wrote:
  

It is quite interesting and I am looking for it.  Could you give me
some more information or website how to set it up?





Have a look at:

http://atlaug.com/stuff/Presentations/Astricon06/JR_Richardson_Whitepaper.pdf

and the two links at:

http://www.voip-info.org/wiki/index.php?page=DUNDi%20Enterprise%20Configuration

  


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Re: [asterisk-users] Log CODECS in CDR's

[dave cantera]

morgan,
I've seen some info on additional variables in the CDR... but haven't 
tried it... look to these pages:

daveC

http://www.asterisk.org/doxygen/1.2/AstCDR.html

In addition, you can set your own extra variables by using Set(CDR(name)=value).
These variables can be output into a text-format CDR by using the cdr_custom
CDR driver; see the cdr_custom.conf.sample file in the configs directory for
an example of how to do this.

-and-

http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List




Morgan Gilroy wrote:

Hi,
Does anyone know how to get the codec that was negotiated for a call
after a dial? I want to log them into CDR but can't find any way to do
it without hacking the code.
It would be good if I could get it in an asterisk variable I can log off
seperatly.

Thanks!
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Re: [asterisk-users] question about more than one drop file

[dave cantera]

shawn,
you can set an archive variable in the .call file to 'yes' and it will 
save it in ./outgoing_done... if there is now outbound line availible, 
the .call file is updated (appended to) as per the status... * will keep 
trying till it completes the calls or the number of retries is reached.  
then it will archive the .call file if archive=yes...  if you drop a ton 
of files in the ./outgoing, it tries to make all the calls at 'almost' 
once.   if you drop 20 .call files in there in about 2 seconds, all 
calls will initiate.  if you have less than 20 outbound lines, the will 
all get stalled (for lack of a better word) and queue up until an 
outbound line is freed up...

daveC

shawn bright wrote:

hello there all,
if i have a script that writes drop files into 
/var/spool/asterisk/outgoing

asterisk picks up the file and initiates the call just fine.
however, what is supposed to happen if more than one gets dropped in 
there
within like a second. Will it wait till the first is complete to 
initiate the second ?

Do they dissapear ?

thanks
shawn


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No virus found in this incoming message.
Checked by AVG Free Edition. 
Version: 7.5.467 / Virus Database: 269.6.2/785 - Release Date: 05/02/2007 02:16 PM
  


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Re: [asterisk-users] GXV-3000 IP Video Phone

[dave cantera]

nitesh,
you are correct.  you need 1.4.x...
daveC

Nitesh Divecha wrote:

Hello All,

I just received some test units of Grandstream GXV-3000 IP Video Phone.

I did some research and looks like Asterisk 1.2 does not support video 
H.264 but Asterisk 1.4 does. Is it correct?


Actually I did try to test with Asterisk 1.2 and video did not 
initialize but voice worked...


Any advice?

Thanks,
Nitesh


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[asterisk-users] auto call out via drop file ERROR: 'OutgoingSpoolFailed'

[dave cantera]

has anyone run into this message?  for some reason, which I can not 
determine, this script stop working and now gives this error.  I googled 
'outgoingspoolfailed' but not too much turned up... only questions, no 
answers... :(


I am mv'ng a .call file to the ./outgoing directory. the call initiates 
then hangs up...  and the reason 0, in the last line below, just doesn't 
help too much...


what it was doing was calling and playing a message regardless of being 
answered (but that is another day's problem)... today the script and 
.call file initiate a call but hangs up whether answered or not in about 
4+/- seconds...  as you can see below, hangup is called immediately and 
the 'failed' extension is then executed...  but why is it now failing?


any thoughts?
daveC

pbv01*CLI>
   -- Hungup 'Zap/4-1'
   -- Executing [EMAIL PROTECTED]:1] 
NoOp("OutgoingSpoolFailed", "Call Failed") in new stack
   -- Executing [EMAIL PROTECTED]:2] 
Set("OutgoingSpoolFailed", "CALL_ACK=failed") in new stack
   -- Executing [EMAIL PROTECTED]:3] 
AGI("OutgoingSpoolFailed", "lax/track-laxcalls.sh|failed|failed") in new 
stack

   -- Launched AGI Script /var/lib/asterisk/agi-bin/lax/track-laxcalls.sh
   -- AGI Script lax/track-laxcalls.sh completed, returning 0
   -- Executing [EMAIL PROTECTED]:4] 
Wait("OutgoingSpoolFailed", "1") in new stack
[May  3 00:59:08] NOTICE[8878]: pbx_spool.c:341 attempt_thread: Call 
failed to go through, *reason 0*



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Re: [asterisk-users] using Playback() to play a random sound file

[dave cantera]

steve,
thats Great... my C is old and ftw operated differently on sysV, 
solaris, sunos, ultrix, and osf...  so I went back to bourne...  
couldn't work through the idiosyncracies of gnu autoconf, etc...  
although I have a many reasons to, I just couldn't get to production 'C' 
coding level...

daveC

Steve Edwards wrote:

Steve Edwards wrote:
On Tue, 1 May 2007, Jay Austad wrote:

I've got a directory under /var/lib/asterisk/sounds which contains 
a bunch

of sound files.  I would like to call the Playback command to play the
files, but I need it to select a file to play randomly.  Is there 
any way

to do this?


I do this with an AGI.


On Wed, 2 May 2007, dave cantera wrote:

here is a way that I solved a similar problem...  have a shell script 
that
runs and indexes all the files in the directory into an ascii flat 
file with

a format of
 filename
0001 directory/tt-weasels
0002 directory/tt-monkeys

in your dialplan use the rand() to pick a number, pass it to the 
shell script
as an arg[], then the shells script grep()'s and cut()'s the filename 
puts it
in a db varaible, the dialplan picks it up and plays it...  as you 
can see, I

haven't done it yet :) but, in theory it works...  you could skip the
dialplan rand() and just use linux rand based on the minutes or 
seconds value

for current time...

you don't have to zero fill the index either, I seem to like nicely 
formated

files, they are easier for humans to read.
daveC


Sounds like a lot of effort to avoid writing an AGI. If you have the 
skills to write the script described above, you have the skills to 
write an AGI -- you can write AGI's in shell scripts, btw.


AGI's accept stuff from Asterisk on stdin and send stuff back to 
Asterisk on stdout -- very simple and elegant actually. Take your 
script and rewrite the "reading arguments" bits to read from stdin and 
change the "write db" bits to write to stdout (set a channel variable) 
and you have an AGI and a much cleaner dialplan.


I write AGI's in C for speed and flexibility. No interpreter (bash, 
perl, php, etc.) to fire up, full access to anything you want to do.


In C, I call ftw() ("ftw - traverse (walk) a file tree"). If I get 
more than 1 file, I choose one randomly.


Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] using Playback() to play a random sound file

[dave cantera]

here is a way that I solved a similar problem...  have a shell script 
that runs and indexes all the files in the directory into an ascii flat 
file with a format of

 filename
0001 directory/tt-weasels
0002 directory/tt-monkeys

in your dialplan use the rand() to pick a number, pass it to the shell 
script as an arg[], then the shells script grep()'s and cut()'s the 
filename puts it in a db varaible, the dialplan picks it up and plays 
it...  as you can see, I haven't done it yet :) but, in theory it 
works...  you could skip the dialplan rand() and just use linux rand 
based on the minutes or seconds value for current time...


you don't have to zero fill the index either, I seem to like nicely 
formated files, they are easier for humans to read.

daveC


Steve Edwards wrote:

On Tue, 1 May 2007, Jay Austad wrote:

I've got a directory under /var/lib/asterisk/sounds which contains a 
bunch of sound files.  I would like to call the Playback command to 
play the files, but I need it to select a file to play randomly.  Is 
there any way to do this?


I do this with an AGI.

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Zaptel kernel module load order

[dave cantera]

mitch,
not that I can answer your problem but is this ver 1.4.1?  I had a 
similiar problem in that zapscan was updating the zaptel.conf and 
nothing would work until I mucked with zaptel.conf.zapscan... I might 
have the filename wrong as I have multiple files now :(...  it has 
zapscan in the filename...

daveC

Mitch Jackson wrote:

Evening,

My latest asterisk box is having a difficult problem.  It is
configured with one TE210P and TDM400P with four FXO modules.  I'm
running FC6.

The TE210P only has a single PRI.

When the system boots, it is completely random what order the zaptel
modules will get loaded in.  Sometimes zttool shows the FXO as the
last span, sometimes as the first.  When it does load as the first,
which happens more often, nothing will initialize properly.  When this
happens, I have to unload all the zaptel modules, and re-load them
over and over again, until the hardware comes up in the correct order.
The order it is loaded is in no way related to what order I load the
modules on the command line.  This problems makes it unlikely that
asterisk will start properly if the system is rebooted.

Is there something I can do to ensure the modules get loaded in the
correct order?

Here's my config files, if they will help...

# cat /etc/zaptel.conf
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
defaultzone=us
loadzone=us

span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
defaultzone=us
loadzone=us

fxoks=49-52
defaultzone=us
loadzone=us

# cat /etc/asterisk/zapata.conf
[channels]
language=en
switchtype=national
context=incoming
faxdetect=none
signalling=pri_cpe
group=1
echocancel=yes
resetinterval=never
channel => 1-23

language=en
switchtype=national
context=incoming
faxdetect=none
signalling=pri_cpe
group=3
echocancel=yes
resetinterval=never
channel => 25-47


signalling=fxo_ks
usecallerid=yes
callerid=Fidelity Reserves
group=2
threewaycalling=no
context=outgoing
channel => 49-52





Thanks for any help,

/Mitch
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Re: [asterisk-users] Unable to find a codec translation path from ilbc to ulaw

[dave cantera]

oliver,
ugh,  it is too obvious... why did it take me so long to figure it
out...

both phones have to have to negotiate the same codec for audio...  as
far as I know, *  is supposed to do automatic translation and your
gateway should be doing translations only on the below codecs.   I
haven't had that experience yet... 

one phone may be connected to your * box, but your other phone is
*not*  connected to *. it is connected to a voip provider...  since
they don't do any translation other than below.  the * connection to
webcalldirect must have one of these codecs in the sip.conf for that
extension, the extension where webcalldirect is coming in, that is...

phoneX -> * -> webcalldirect -> phoneY
which one is phone1 and which is phone2?

phoneX< * <-- webcalldirect <---phoneY
-| -| -
local LAN  Internet    local LAN
some code    no codec control    no codec
control
control  little or no call quality control

the phone connected to * will also select a code that matches up with
the caller (webcalldirect)...   you have no advantage whether or not *
converts the audio to the phone connected to *.    you won't get any
better reception from webcalldirect because you are not changing that
connection.

also,  I would change iLBC to ilbc,  case may make a difference... 
don't know for sure...  perhaps someone else does...
hope that is clearer...
daveC



  

  
   Codecs 
  


  
  G.711
(64 kbps) 


  
  G.726 (32 kbps)


  
  G.729
(8 kbps)


  
  G.723 (5.3 & 6.3 kbps)


  
  GSMFR
(13.2 kbps) Temporarily unavailable due to technical difficulties.

  



Oliver Brandt wrote:

  Hi Dave!

Thank you very much for replying!

  
  
what gateway provider are you referring to?doesn't your sip phone 

  
  
webcalldirect (it does not seam to support iLBC directly)

  
  
connect directly to * as your diagram indicated?

  
  
Yes, my sipphone ist connected directly to * and also the gateway
provider is directly connected to *. My * is on a root server at hosting
provider (high bandwith internet connection to the gateway provider) but
my phone is connected through DSL with a very limited upstream. For this
reason I'd like asterisk to do the codec conversion from iLBC to ulaw.

I bett all I have to do is load the codec or/and the codec translator
for iLBC to ulaw. But when googleing I only find articles the describe,
that * is doing the codec translation automatically. I can't find any 
information on how to load a codec or the translator manually. I'm
probably just using the wrong search string in google...

When * starts translators are beeing loaded, but as far as I can see non
for iLBC to ulaw.

I've put together another test setup with to sip phones to clarify the
problem:

[phone1]
disallow=all
allow=iLBC

[phone2]
disallow=all
allow=ulaw

When calling from one phone to the other I get the following message:

chan_sip.c:2841 sip_call: No audio format found to offer. Cancelling
call to phone2

Thank you very much again!
Oliver

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Re: [asterisk-users] Unable to find a codec translation path from ilbc to ulaw

[dave cantera]

oliver,
what gateway provider are you referring to?doesn't your sip phone 
connect directly to * as your diagram indicated?

DSL providers should not be doing any codec anything!
daveC

Oliver Brandt wrote:

Hi!

As the upstream of my DSL-connection is very slow, I'd like my
sip-phones to use iLBC to connect to my *. My gateway provider only
allows ulaw. Hence, I'd like to use the follwing setup:

SIP-phone <--iLBC--> Asterisk <---ulaw> PSTN-Gateway

I get the following error:

"Unable to find a codec translation path from ilbc to ulaw"

Setup SIP-phone:
disallow=all
allow=ilbc

Setup PSTN-Gateway:
disallow=all
allow=ulaw

I've googled for overn an houre. But no luck. So I'd really apreciate
any help!

Thanks!
Oliver
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