date:20061201

No, no menuslect on system beside *

I unzipped it, ran configure, then make (or make menuselect) they both
give the same immediate error 3.

From what I see with 1.4.x  it might be good to have a completely seperste
list. I suspect there will be tons of email volume once it's use or
attempt of use ramps up!

Doug

On Fri, 1 Dec 2006, Tim Panton wrote:


 On 1 Dec 2006, at 03:49, Doug Crompton wrote:

  no - make menuselect -  does the same thing.

 Have you got a (non asterisk) binary or shell script called
 menuselect in your path?

 try

 which menuselect

 Tim Panton

 www.mexuar.net
 www.westhawk.co.uk/



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Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety.  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



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[asterisk-users] spa3k dtmf problem asterisk 1.2.x

Anyone that uses the spa3k with Asterisk knows about the dtmf issues of
not being able to get tones properly to an IVR after call completion. You
can make it work by eliminating ALL special keys - transfer, etc. in the
dial and using inband signaling. This has been beat to death over the last
year.

My question is that there were patches to rtp.c that were an attempt to
correct. I tried a few to no avail. Does anyone have a patch that works?
I am currently using 1.2.13

My understand from googling this is that the problem is both a Sipura and
Asterisk problem, although more of the blame is put on Asterisk.

Also the rtp in 1.4 has been completely reworked. Has anyone tested this
with the spa3k? Unfortunately 1.4 is a significant change that involves a
great deal of time to test and is not at all like doing an upgrade within
1.2. So I am not inclined to go that route yet unless it fixes this
problem.

Doug

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Re: [asterisk-users] How to park calls on a specific extension

2006-12-01 Thread Tom Rymes


On Nov 30, 2006, at 8:55 PM, Brad Templeton wrote:


On Thu, Nov 30, 2006 at 02:50:21PM -0500, Tom Rymes wrote:

for example: In your example above where they can't figure out how to
transfer, why don't you edit features.conf and define the transfer
key as # or something. Then, when they have a call for Bill across
they way, they can do this:


In this case don't they need to have a t in every Dial as well?
And then there's the other direction.  Sometimes (actually quite
often) I like to transfer a call that I dialed, which requires
the T but means you interfere with typing touch tones to IVRs
that you call.


I'd have to double-check on that, but I am fairly certain the you  
would indeed need to put the 'T' and 't' options in the dial string.  
I think you would want to arrange your dialplan to make sure that you  
don't inadvertently make it so that incoming callers can transfer  
their own calls coming in. In other words, Dial command for incoming  
calls would have 'T' and for outgoing calls would have 't'. Double  
check the docs, though, because I am pulling this from (spotty)  
memory. As for not interfering with IVRs, I would suggest a two  
character transfer, such as #8, which reads #T if you look at the  
keypad. Tell your users, Press #T to transfer a call. I also use #7  
for pickup, as it is #P. Same sort of idea as using *86 or *VM  
for voicemail.



No, almost all IP phones have a transfer button, the nice thing
would be if somehow the UI for that could have been standardized,
at least for the phones that don't have screens and soft buttons
(which can extend the interface because they can show it to you).


Better configurability of the SIP phones would be nice. We use Cisco  
7940 phones, and changing the soft buttons on the bottom of the  
screen to hold commonly used functions would be nice, but AFAIK, that  
cannot be done in the SIP firmware.


Tom
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[asterisk-users] server specs / hardware

2006-12-01 Thread Richard Minshaw



Hi all,

We're looking at replacing our ailing Avaya phone system, potentially 
with *. I've been asked to draw up costs and so on. The advice on 
hardware seems to be sketchy, generally, or I'm looking in the wrong place!


We're looking at 8 PRI lines coming in, handling up to 300 calls per 15 
mins (what does that average to? er... 20 per minute). We have a call 
centre environment, with many NGN (Non-Geographic Numbers - 0845 types) 
coming in. These need branding before routing to certain phones / hunt 
groups. We also need to run IVR and call recording on some calls.


Based on what I've read thus far, I'm thinking of splitting it over two 
beefy servers (dual processor, lots of RAM) with 4 port TE412P cards in 
each.


Does this sounds feasible?

Please feel free to request more info if I've not provided enough.

TIA,

Rich
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[asterisk-users] No caller ID, no incoming call

2006-12-01 Thread jezzzz .

Is it possible to reject all incoming calls that do
not have a CID?

Could I do something like that (modified version from
the book): 

exten = 123,1,GotoIf($[${CALLERIDNUM} = ]?20:10)
exten = 123,10,Dial(Zap/4)
exten = 123,20,Playback(abandon-all-hope)
exten = 123,21,Hangup(

Alternatively, what's the privacy.conf file for? What
does it mean for a user to have to chances to 'enter
his CID' else his call is rejected?

Thanks,
Jez


 

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RE: [asterisk-users] Answer Supervision problem

2006-12-01 Thread shadowym

Hi Timur,

I just had to deal with a similar realization myself.  We wanted to set up
something similar to what Vonage does with simuring where Asterisk would
ring a local extension and a cell phone or landline at the same time and
route to the call to which ever answered first.  Seemed simple enough.
However, Asterisk automatically set the outgoing zap channel to Answered
as soon as it started ringing.  Upon further investigation I found that this
is not an exception but a rule when it comes to plain analog POT's lines.
At least in N. America.  They generally do NOT have any type of answer
supervision.  It is possible to get lines with polarity reversal for answer
supervision but that may be a special order and not available in all areas.
You would have to talk to your telco provider about this  

When your dealing with Telco's, the 2 things you DO NOT want to deal with
are any requests involving the words special and not available in all
areas.  They have a hard enough time getting 100 year old POT's lines
right.  At least in my area.  You may have to speak with several different
people at the telco before you find someone who even knows what answer
supervision means.

To my knowledge, the only thing the analog cards+zaptel+Asterisk can detect
for answer supervision with Analog POT's is polarity reversal.  There is a
feature in Asterisk called callprogress which attempts to listen to the
line and determine answer/hangup/busy status but it is NOT reliable at all
so not recommended.

-Original Message-
From: Timur S. Sattarov [mailto:[EMAIL PROTECTED] 
Sent: Thursday, November 30, 2006 10:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Answer Supervision problem

Does anybody work with zaptel and analog lines ?
how to configure it in ideal case, if I really got correct answer
supervision from CO ?

regards
Tim

Timur S. Sattarov wrote:
 Hi !

 I've Linux Debian/testing, asterisk-1.2.13, zaptel 1.2.10 and Digium, 
 Inc. Wildcard TDM2400P with 24 FXO ports.
 Our equipment is located in US. Lines and phone numbers from ATT.
 How can I detect answers on these lines ?
 We've discovered that during answer voltage falls and after hangup - 
 rises.
 During remote hangup there is kernel message from zaptel driver  - NO 
 BATTERY on 2/24 and then BATTERY on 2/24 (-)! and 
 hanguponponlarityswitch option in zapata.conf works fine.
 There is no such message during answer. But we can see voltage change. 
 Is there any way to detect answer like we have on hangup ?
 Or if there any possibility to detect answer through voltage changes 
 with zaptel ?
 Please help me.

 Thanks in advance.

 With best regards
 Timur.
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Re: [asterisk-users] No caller ID, no incoming call

2006-12-01 Thread Julio Arruda



Try to search for the PrivacyManager application.
It does 'check' if the CallerID is present, if not, it will play an 
announcement to ask the person to 'type' their phone number, and it will 
 allow you to then accept it.



je . wrote:

Is it possible to reject all incoming calls that do
not have a CID?

Could I do something like that (modified version from
the book): 


exten = 123,1,GotoIf($[${CALLERIDNUM} = ]?20:10)
exten = 123,10,Dial(Zap/4)
exten = 123,20,Playback(abandon-all-hope)
exten = 123,21,Hangup(

Alternatively, what's the privacy.conf file for? What
does it mean for a user to have to chances to 'enter
his CID' else his call is rejected?


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[asterisk-users] CALL TRANSFER

2006-12-01 Thread omar parihuana


Hi Guys,

I'm implementing my Asterisk step by step, so far the communications between
softphones, hardphones with Gateways, voice mail, are working fine. Rightnow
I would like to enable Call tranfer (like Traditional PBX) in Blind Transfer
and AttendXFER, I'm reading features.conf in accordance to voip-info.org but
the transfer doesn't work!  Please if you can provide me some examples will
be very appreciate.

Rgds.

--
Omar E.P.T
-
Certified Networking Professionals make better Connections!

http://omarept.blogspot.com/

 Usysnet Corp
Open Source Solutions
www.usysnet.com.pe
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Re: [asterisk-users] Asterisk: SIP Gateway or Proxy

2006-12-01 Thread David Thomas


Asterisk can actually act as a Gateway and a SIP Proxy. This is where
a lot of confusion comes in. It can do pretty much any voip function
you throw at it. Definitely search the archives if you still have
questions.

try site:lists.digium.com keyword in google to search the mail archives.

David

On 12/1/06, yusuf [EMAIL PROTECTED] wrote:

Hi,

I realise this might be an insane noob question, but I'm on a huge brain 
freeze, and I'm trying to
decide this:

Is Asterisk a SIP Gateway or SIP proxy?


--
thanks,
yusuf

--
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Re: [asterisk-users] Cisco IAXmodem HylaFAX

2006-12-01 Thread Lee Howard


[EMAIL PROTECTED] wrote:


In my company a have a 100% IP Telephony implemented (Cisco Phones, Routers,
Switches, etc). 
We are connected to our operator by two E1 lines. The PBX is on operator

side (a H.323 Gateway is configured on call manager).

I study the possibility to implement a Hylafax server. In some sites I read
that with the IAXmodem is possible to have virtual modems, with don't so buy
voip-fax-gateway or an expensive modem card with many channels.

The how-to that I read is with Asterix. Is possible to use a HylaFax Server
with IAXmodem and configure to work with Cisco Call Manager?



I think that you would be better-of in using T.38 via OpenH323's 
t38modem in this case.


Lee.
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Re: [asterisk-users] No caller ID, no incoming call

2006-12-01 Thread jezzzz .

Thanks so much for the swift response, this is exactly
what I was looking for. If I set maxretries to 0, all
callers with no caller ID will be forwarded to the
voicemail immediately, sounds good.

Thanks

Jez

--- Julio Arruda [EMAIL PROTECTED] wrote:

 
 Try to search for the PrivacyManager application.
 It does 'check' if the CallerID is present, if not,
 it will play an 
 announcement to ask the person to 'type' their phone
 number, and it will 
   allow you to then accept it.
 
 
 je . wrote:
  Is it possible to reject all incoming calls that
 do
  not have a CID?
  
  Could I do something like that (modified version
 from
  the book): 
  
  exten = 123,1,GotoIf($[${CALLERIDNUM} = ]?20:10)
  exten = 123,10,Dial(Zap/4)
  exten = 123,20,Playback(abandon-all-hope)
  exten = 123,21,Hangup(
  
  Alternatively, what's the privacy.conf file for?
 What
  does it mean for a user to have to chances to
 'enter
  his CID' else his call is rejected?
  
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[asterisk-users] app_sql_postgres gone in 1.4

2006-12-01 Thread Gavin Hamill

Hi,

I'm putting together a system to manage agents with Realtime, and
without chan_agent. In 1.2.13, there's a handy (although marked as deprecated 
in apps/Makefile) PGSQL application to let me do this:

macro queue-addremove(queuename,penalty) {
switch(${MACRO_EXTEN:0:1})
{
case I:  // Login
PGSQL(Connect connid host=XXX user=XXX password=XXX dbname=XXX);
PGSQL(Query resultid ${connid} INSERT INTO queue_member_table 
VALUES (\'${queuename}\'\,\'Local/${MACRO_EXTEN:[EMAIL 
PROTECTED]'\,${penalty}));
PGSQL(Clear ${resultid});
PGSQL(Disconnect ${connid});
break;

I do this because AddQueueMember does not INSERT the new agent into the table 
defined in extconfig.conf (I even have ReadOnly in odbc.ini set to No). In this 
way, I can preserve the state of agents between Asterisk restarts.

However, I notice in 1.4 beta3, this application has gone. Can anyone suggest 
what would be the best alternative?

I have thought of System(psql -h xxx -U  -P xxx ...) but that's just 
horrendous :)

gdh
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[asterisk-users] Problem with agent AgentCallbackLogin()

2006-12-01 Thread gc

When I use AgentCallbackLogin() function to login an agent, I got following 
warning message saying the agent is not valid for auto login while my other 
extensions work fine for this function.
Does anybody know why?  This extension has the same settings as the other ones 
like agents.conf and queues.conf.

Here is the message from asterisk:
-- Executing AgentCallbackLogin(SIP/gc3-08c7ad58, 611222||[EMAIL 
PROTECTED]) in new stack
-- Playing 'agent-pass' (language 'en')
Dec  1 11:03:09 WARNING[30060]: chan_agent.c:1844 __login_exec: Extension 
'611222' is not valid for automatic login of agent '611222'


gary

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[ignore] Re: [asterisk-users] app_sql_postgres gone in 1.4

2006-12-01 Thread Gavin Hamill

On Fri, 1 Dec 2006 16:26:49 +
Gavin Hamill [EMAIL PROTECTED] wrote:

slump Just found func_odbc.conf - problem solved. :)

Hurrah, it's Friday afternoon, and not a moment too soon

gdh
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Re: [asterisk-users] Asterisk: SIP Gateway or Proxy

2006-12-01 Thread olivier.taylor


Not a proxy, of course, or proxy for a very small amount of users.
Pbx and gateway is the best usage.



David Thomas a écrit :

Asterisk can actually act as a Gateway and a SIP Proxy. This is where
a lot of confusion comes in. It can do pretty much any voip function
you throw at it. Definitely search the archives if you still have
questions.

try site:lists.digium.com keyword in google to search the mail 
archives.


David

On 12/1/06, yusuf [EMAIL PROTECTED] wrote:

Hi,

I realise this might be an insane noob question, but I'm on a huge 
brain freeze, and I'm trying to

decide this:

Is Asterisk a SIP Gateway or SIP proxy?


--
thanks,
yusuf

--
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[asterisk-users] App_Swift

2006-12-01 Thread Hall, Eric M.

Group
I have app_swift working on our asterisk server running 1.4-Beta3.
My question is can you read variables with it? Like reading back
callerid number ${CALLERID(number) 
 
 

Eric Hall


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Re: [asterisk-users] Asterisk: SIP Gateway or Proxy

2006-12-01 Thread Peter Bowyer


On 01/12/06, yusuf [EMAIL PROTECTED] wrote:

Hi,

I realise this might be an insane noob question, but I'm on a huge brain 
freeze, and I'm trying to
decide this:

Is Asterisk a SIP Gateway or SIP proxy?



http://www.voip-info.org/wiki-Asterisk+SIP+not-proxy



--
Peter Bowyer
Email: [EMAIL PROTECTED]
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Re: [asterisk-users] No caller ID, no incoming call

2006-12-01 Thread John Novack

Perhaps it has been worked over since I last looked into this, but it 
USED to pass CLID that weren't, and would only come into play if the 
CLID were blank, Don't know if that is still the case,but I had to add 
gotoif's for Anonymous, asterisk or private. It also passed CLID's that 
were abridged extension numbers.

Has this been fixed? Or is it considered a feature?

John Novack



Julio Arruda wrote:


Try to search for the PrivacyManager application.
It does 'check' if the CallerID is present, if not, it will play an 
announcement to ask the person to 'type' their phone number, and it 
will  allow you to then accept it.



je . wrote:

Is it possible to reject all incoming calls that do
not have a CID?

Could I do something like that (modified version from
the book):
exten = 123,1,GotoIf($[${CALLERIDNUM} = ]?20:10)
exten = 123,10,Dial(Zap/4)
exten = 123,20,Playback(abandon-all-hope)
exten = 123,21,Hangup(

Alternatively, what's the privacy.conf file for? What
does it mean for a user to have to chances to 'enter
his CID' else his call is rejected?


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[asterisk-users] feel free to add to the bounty for issue 8064

I have posted a bounty in the dev mailing list for this issue, if you
are similarly impacted feel free to join in. This is a 2 month old issue
with significant impact to those of us using callback agents with SIP
transfers in 1.2;

 

http://bugs.digium.com/view.php?id=8064

 

I would like to offer a $300(US) bounty for a fix and subsequent merge
for issue 8064 in 1.2.12.1 and 1.2.13.

 

If anyone else can add to the bounty please feel free.

 

I think that the source of the bug has been clearly identified on the
bug tracker, including the revision and exact code that caused it. The
reporters, including myself (sohosys), do not have enough experience
with the code to provide a patch. We simply do not understand the reason
that the code was changed in the first place so we can not be sure we do
not regress. There are also other related bugs that must be considered,
at one point in the related changes a deadlock in the queue app was
experienced when a call was transferred and even a segfault after doing
a show channels after the failed transfer. The latter two issues do
not seem to be present in 1.2.12.1 and 1.2.13.

 

Terms;

 

Programmer must review and understand the code added in revisions 31520
37361, the patch must resolve the new issue as well as not regress on
the original need for the changes. A summary of how this was
accomplished as well as the patch for at least 1.2.12.1 and 1.2.13 must
be posted to the bug tracker. If the code requires a different patch for
trunk it will also need to be posted.

 

Disclaimers must be filed, and the patch and notes must be posted on the
bug tracker.

 

$150 will be paid via PayPal within 2 days of the delivery of a workable
patch (as tested by us on several high volume production servers), and
an addition $150 will be paid when (and if) the patch is merged into to
code tree for the 1.2 branch, which will obviously require favorable
peer review.

 

If this issue appears larger than a couple of hours work I welcome the
feedback.

 

Damon Estep

[EMAIL PROTECTED]

 

 

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RE: [asterisk-users] CALL TRANSFER

Your dial string must have either the t or T option set.

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of omar
parihuana
Sent: Friday, December 01, 2006 9:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] CALL TRANSFER

 

Hi Guys,

 

I'm implementing my Asterisk step by step, so far the communications
between softphones, hardphones with Gateways, voice mail, are working
fine. Rightnow I would like to enable Call tranfer (like Traditional
PBX) in Blind Transfer and AttendXFER, I'm reading features.conf in
accordance to voip-info.org but the transfer doesn't work!  Please if
you can provide me some examples will be very appreciate.

 

Rgds.

-- 
Omar E.P.T
-
Certified Networking Professionals make better Connections!

http://omarept.blogspot.com/

  Usysnet Corp 
Open Source Solutions
www.usysnet.com.pe 

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[asterisk-users] Recommendation for FXO

2006-12-01 Thread Martin Joseph


Ok,

I am back from my thanksgiving holiday,  and I find there was a big 
snow storm here in Seattle.  Apparently during the storm there where 
multiple brown out/black outs.


I have struggled since day one to get a high quality PSTN gateway 
configured with my very long loop and Mac based asterisk.


I originally tried the HT-488, which had multiple issues, and was 
unacceptable.  I then purchased the wellgate 3701a, which was much 
better, but lacked ANY support from the manufacturer, and had some 
other semi-minor problems (rfc2833 didn't work, never got caller id 
working, etc.).


Now the power problems seems to have done something bad to the PSTN 
gateway, as it appears to be up and running,  but the gains are really 
whacked, and it's almost impossible to conduct a call through at this 
point.


I tried hooking a handset directly to the PSTN line and that sounds fine.

So,  I would like to purchase another PSTN gateway which WORKS WELL 
with asterisk.  I need it to hook up via ethernet, since my platform of 
choice (mac OSX) has no PCI card support.  I only have one PSTN line, 
and already have other ATA's for FXSs, so I really only need one FXO 
port, although I realize there is no such animal.


Any positive experiences with FXO gateways that connect via ethernet?  
Especially with a long loop/echo issues (ie not SPA3000)?


Thanks in advance.
Marty

PS I am ready to spend to buy something quality.


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Re: [asterisk-users] 1.4beta3 help

2006-12-01 Thread Anthony Rodgers


IIRC, menuselect requires ncurses-devel (or your distro's equivalent).

CP

On Dec 1, 2006, at 7:05 AM, Doug Crompton wrote:


No, no menuslect on system beside *

I unzipped it, ran configure, then make (or make menuselect) they both
give the same immediate error 3.

From what I see with 1.4.x  it might be good to have a completely 
seperste

list. I suspect there will be tons of email volume once it's use or
attempt of use ramps up!

Doug

On Fri, 1 Dec 2006, Tim Panton wrote:


 On 1 Dec 2006, at 03:49, Doug Crompton wrote:

  no - make menuselect -  does the same thing.

 Have you got a (non asterisk) binary or shell script called
 menuselect in your path?

 try

 which menuselect

 Tim Panton

 www.mexuar.net
 www.westhawk.co.uk/



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safety

 deserve neither liberty nor safety.  -- Ben Franklin (1759)


*  Doug Crompton       *
*  Richboro, PA 18954      *
*  215-431-6307        *
*              *
* [EMAIL PROTECTED] *
* http://www.crompton.com  *



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Re: [asterisk-users] Recommendation for FXO

I have an spa3000 and in general it is OK. I have the echo at acceptable
levels and often non-existent. The big grip is the failed rfc-2833. BUT it
might be more the fault of Asterisk - See

http://forum.voxilla.com/linksys-sipura-spa-users-group/dtmf-rfc2833-incompatibility-between-spa3000-asterisk-12306.html

and my previous (today) message to this list on spa3k

So other (external) devices may also not play well because of this
problem. Digium it seems is not real excited to do anything about this as
it is not an issue with their internal hardware and also 1.2.x is not on
the front burner with 1.4 out. 1.4 wwith it's complete rewrite of rtp code
might solve this issue. It is not clear as I have not seen an answer to my
previous question on that.

See...

http://www.voipsupply.com/index.php?cPath=96

for a listing of the external ata's. Grandstream has one, as does linksys
and others. I think they all have some issue. It is just picking the one
that has the least or the most bearable for you. Unfortunately this may
not become apparent until you get it and use it.

I too would be interested in trying another fxo/fxs as my only experience
is with Sipura.

Doug

On Fri, 1 Dec 2006, Martin Joseph wrote:

Ok,

I am back from my thanksgiving holiday, and I find there was a big
snow storm here in Seattle. Apparently during the storm there where
multiple brown out/black outs.

I have struggled since day one to get a high quality PSTN gateway
configured with my very long loop and Mac based asterisk.

I originally tried the HT-488, which had multiple issues, and was
unacceptable. I then purchased the wellgate 3701a, which was much
better, but lacked ANY support from the manufacturer, and had some
other semi-minor problems (rfc2833 didn't work, never got caller id
working, etc.).

Now the power problems seems to have done something bad to the PSTN
gateway, as it appears to be up and running, but the gains are really
whacked, and it's almost impossible to conduct a call through at this
point.

I tried hooking a handset directly to the PSTN line and that sounds fine.

So, I would like to purchase another PSTN gateway which WORKS WELL
with asterisk. I need it to hook up via ethernet, since my platform of
choice (mac OSX) has no PCI card support. I only have one PSTN line,
and already have other ATA's for FXSs, so I really only need one FXO
port, although I realize there is no such animal.

Any positive experiences with FXO gateways that connect via ethernet?
Especially with a long loop/echo issues (ie not SPA3000)?

Thanks in advance.
Marty

PS I am ready to spend to buy something quality.

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deserve neither liberty nor safety. -- Ben Franklin (1759)

* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
* *
* [EMAIL PROTECTED]*
* http://www.crompton.com *

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Re: [asterisk-users] 1.4beta3 help

Ok Mine is probably old and not the right version.

locate ncurses

/lib/libncurses.so.4
/lib/libncurses.so.4.2
/lib/libncurses.so.5
/lib/libncurses.so.5.2
/usr/include/ncurses.h
/usr/include/ncurses_dll.h
/usr/lib/libncurses++.a
/usr/lib/libncurses.a
/usr/lib/libncurses.so
/usr/lib/libncurses.so.1.9
/usr/lib/libncurses.so.1.9.7a
/usr/lib/libncurses.so.2.1

as I am using a SUSE 7.3 system.

It is time for a system rebuild and upgrade and I will probably wait until
then to upgrade to 1.4 once there is an official release.

Just playing now!

Doug

On Fri, 1 Dec 2006, Anthony Rodgers wrote:

 IIRC, menuselect requires ncurses-devel (or your distro's equivalent).

 CP

 On Dec 1, 2006, at 7:05 AM, Doug Crompton wrote:

  No, no menuslect on system beside *
 
  I unzipped it, ran configure, then make (or make menuselect) they both
  give the same immediate error 3.
 
  From what I see with 1.4.x? it might be good to have a completely
  seperste
  list. I suspect there will be tons of email volume once it's use or
  attempt of use ramps up!
 
  Doug
 
  On Fri, 1 Dec 2006, Tim Panton wrote:
 
  
   On 1 Dec 2006, at 03:49, Doug Crompton wrote:
  
no - make menuselect -? does the same thing.
  
   Have you got a (non asterisk) binary or shell script called
   menuselect in your path?
  
   try
  
   which menuselect
  
   Tim Panton
  
   www.mexuar.net
   www.westhawk.co.uk/
  
  
  
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  ??? http://lists.digium.com/mailman/listinfo/asterisk-users
  
 
 
  Those that sacrifice essential liberty to obtain a little temporary
  safety
  ?deserve neither liberty nor safety.? -- Ben Franklin (1759)
 
  
  *? Doug Crompton??? ?? *
  *? Richboro, PA 18954?? ?? *
  *? 215-431-6307 ??? ?? *
  *?? ??? ? ? ?? *
  * [EMAIL PROTECTED] *
  * http://www.crompton.com? *
  
 
 
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 deserve neither liberty nor safety.  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



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[asterisk-users] direct IP calling with extension

2006-12-01 Thread Jerry Geis


All,

If I have video phones behind an asterisk server (with 2 network cards)
and all the phones have extensions. Internally everything works great.

Now for people that want to call my video phones external to my office
is there a way to do that? On the extenal persons phone enter an IP/EXTEN
where IP is my server and not the phone? Can that work?

Would I have to have PUBLIC IP address for every phone NAT'ed through my
server to make the call?

Thanks,

Jerry
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RE: [asterisk-users] Hold calling channel and ask called channel beforeconnect???

2006-12-01 Thread Nigel J. Terry

I posted this a week ago and have had no response.  Can someone tell me if I
am asking a stupid question, i.e. is the answer either obvious or
impossible?

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nigel J. Terry
Sent: Wednesday, November 22, 2006 10:27 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hold calling channel and ask called channel
beforeconnect???

I am a newbie.  Just got my Asterisk working and I love it.

I want to do the following, believe it should be possible, but can't work
out how:

When I get an incoming call, I want to answer and just send ringing to the
calling channel.
Then I want to call the destination channel, send a message asking if they
will accept the call, get a response (1 or 2) and then either connect the
parties (1) or send the calling channel to voicemail (2).

Any ideas, thanks

Nigel

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[asterisk-users] setcallerpres not working

2006-12-01 Thread Patrick Fortin


Hi

I have the following setup

phone - mta - asterisk - patton_sn2400 - PRI

I am trying to program *67 to block caller id name and number

To do this correctly I have to leave the fields callerid name and number 
unchanged and only set the flag callerpres to restricted


The problem seems to be that Asterisk replace the name and number to 
unknown and then send the call to my Patton box.


How can I make this setup work ?

Patrick

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Re: [asterisk-users] direct IP calling with extension

Am Freitag, den 01.12.2006, 13:44 -0500 schrieb Jerry Geis:
 All,
 
 If I have video phones behind an asterisk server (with 2 network cards)
 and all the phones have extensions. Internally everything works great.
 
 Now for people that want to call my video phones external to my office
 is there a way to do that? On the extenal persons phone enter an IP/EXTEN
 where IP is my server and not the phone? Can that work?
 
 Would I have to have PUBLIC IP address for every phone NAT'ed through my
 server to make the call?

AFAIR, define a context in the global section of sip.conf. Any
incoming SIP connections that are not identified to belong to any other
context (registration) will come thru that extensions.conf context.
Inside, just forward through to your proper local extensions:

[locals]
exten = 200,1,Dial(SIP/myphone1)
...
[sipfromoutside]
exten = johndoe,1,Goto(locals,200,1)
...

Then direct IP calling as   [EMAIL PROTECTED]
should work. To get calling at a hostname working as well, you will need
a few records in your DNS setup.

example.com. IN A 123.45.67.89
sip.example.com. IN A 123.45.67.89
example.com. IN NAPTR 60 50 s SIP+D2U  _sip.udp.example.com.
_sip._udp.example.com. IN SRV 10 10 5060 sip.example.com.

Which will allow for [EMAIL PROTECTED]
I suspect the first example.com line is not necessary (such that you
can host your domain on a different server than that which runs
Asterisk), but I did not test.

HTH
Anselm

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[asterisk-users] Fwd: Trouble using 2 IAX2 DiDs provided by different ITSPs

2006-12-01 Thread hugolivude


Asterisk 1.2.7
Redhat 9

I have DiDs from two different ITSP both set up as IAX2.  Each one
works when it's the only one in my iax.conf, but when I have them both
defined in iax.conf at the same time, only one will work.  My iax.conf
is provided below.

Any ideas how to fix?  I'd like to use both DiDs!

Thanks,
H

My iax.conf is below.  When I dial the DiD provided by ITSP_B, the
other ITSP seems to reject it.  For example when I call the ITSP_B
DiD, I get the following error message:

Nov 29 21:50:17 NOTICE[23106]: chan_iax2.c:7203 socket_read: Host IP
failed to authenticate as ITSP_A

iax.conf
==
[general]
register = my UserID:my password@ITSP A Server #1 domain
register = my UserID:my password@ITSP A Server #2 domain
register = my UserID:my password@ITSP B #1 domain
notransfer=yes
bindport=4569
bindaddr=0.0.0.0
bandwidth=low
disallow=all
allow=ulaw
allow=g729
jitterbuffer=yes
forcejitterbuffer=no
tos=lowdelay
autokill=yes

[ITSP_B]
context=incoming-iax
type=friend
qualify=2000
host=ITSP B #1 domain
user=my UserID
username=my UserID
auth=md5
secret=my password
disallow=all
allow=ulaw
;
; *** ITSP_A Inbound ***
[ITSP_A]
context=incoming-iax
type=user
auth=md5
username=my UserID
secret=my password
disallow=all
allow=ulaw
;
; *** ITSP_A Outbound ***
[ITSP_A-Out]
type=peer
host=ITSP A Server #1 domain
auth=md5
username=my UserID
secret=my password
disallow=all
qualify=yes
allow=ulaw
;
[ITSP_A-Out2]
type=peer
host=ITSP A Server #2 domain
auth=md5
username=my UserID
secret=my password
disallow=all
qualify=yes
allow=ulaw
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Re: [asterisk-users] Caller ID Rewrite

Am Freitag, den 01.12.2006, 14:26 + schrieb David Bath:
 So onto the problem… I’m trying to write a quick on-liner which will
 fix up incoming UK format numbers (0 + 10digits) into (44 + 9 digits).
 I got as far as this:
 
  
 
 exten = incoming extension,1,Set(foo=${IF(REGEX(^0[1-9][0-9]\{9
 \}$ {CALLERID(number)})?Set(CALLERID(number)=44
 ${CALLERID(number):1})})

I would try something like

exten = _0[1-9]X.,1,Goto(0044${EXTEN:1},1)
(All numbers beginning null not-null will be rewritten to 0044 plus
the number without the leading zero)

Hth
Anselm

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[asterisk-users] Linksys PAP2T - problem

2006-12-01 Thread Radu Padure


Hi everybody,

Somebody had work experiences with Linksys PAP2T (firmware 5.1.1) and
asterisk 1.2.13 (realtime sip account) ?

I found after a some time I lose SIP registration and I cannot make any
calls. Any other extension works fine.

There are some special configuration (option) ?

Thanks a lot,
Radu 

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RE: [asterisk-users] Caller ID Rewrite

2006-12-01 Thread David Bath

Hi Anselm,

Thanks for the help...

I'm slightly confused as to your response.

Wouldn't that look for a /dialled/ number in the format _0number try
and jump to another extension 0044number with priority 1?

If so, that's not what I'm trying to achieve.

I have an external SIP provider, and the extension is a 6 digit number,
e.g. 123456.  When calls come in, they are always TO: this 6 digit
number..

Hence, the dialplan has 

exten = 123456,1,Goto(sipinternal,myphoneextension,1)

at the moment, all incoming calls are forwarded directly to my
deskphone.

What I'm trying to do is first mangle the incoming caller id (i.e. the
FROM: field) so that all numbers come in countrycode + number.

I've made a bit more progress... and my current diaplan entry looks like
this:

exten=123456,1,Set(${IF(REGEX(^0[1-9][0-9]\{9\}$
{CALLERID(number)})?CALLERID(number)=44${CALLERID(number):1})})
exten = 123456,2,Goto(sipinternal,101,1)

BUT! There's a very odd problem (and I'm sure it's my fault..) the
second callerid function is not being evaluated...

I.e if the true condition is met in the IF statement, the command
should distil down to 

Set(CALLERID(number)=44${CALLERID(number):1}

Which it does... but, 44${CALLERID(number):1} appears as a string,
instead of being evaluated!

Any ideas why ??

Cheers,

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin Hoffmeister
Sent: 01 December 2006 19:40
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID Rewrite

Am Freitag, den 01.12.2006, 14:26 + schrieb David Bath:
 So onto the problem... I'm trying to write a quick on-liner which will
 fix up incoming UK format numbers (0 + 10digits) into (44 + 9 digits).
 I got as far as this:
 
  
 
 exten = incoming extension,1,Set(foo=${IF(REGEX(^0[1-9][0-9]\{9
 \}$ {CALLERID(number)})?Set(CALLERID(number)=44
 ${CALLERID(number):1})})

I would try something like

exten = _0[1-9]X.,1,Goto(0044${EXTEN:1},1)
(All numbers beginning null not-null will be rewritten to 0044 plus
the number without the leading zero)

Hth
Anselm

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Re: [asterisk-users] Fwd: Trouble using 2 IAX2 DiDs provided by different ITSPs

2006-12-01 Thread Andrew Joakimsen


Your configuration is wrong, IAX is sort of tricky to master. Perhaps you
should stop flooding the list with your messages every few hours? Play
around with user= fromuser= and peer and friend and it should work with a
few tries :) Better yet maybe the ITSP provide sample configfiles?

On 12/1/06, hugolivude [EMAIL PROTECTED] wrote:


Asterisk 1.2.7
Redhat 9

I have DiDs from two different ITSP both set up as IAX2.  Each one
works when it's the only one in my iax.conf, but when I have them both
defined in iax.conf at the same time, only one will work.  My iax.conf
is provided below.

Any ideas how to fix?  I'd like to use both DiDs!

Thanks,
H

My iax.conf is below.  When I dial the DiD provided by ITSP_B, the
other ITSP seems to reject it.  For example when I call the ITSP_B
DiD, I get the following error message:

Nov 29 21:50:17 NOTICE[23106]: chan_iax2.c:7203 socket_read: Host IP
failed to authenticate as ITSP_A

iax.conf
==
[general]
register = my UserID:my password@ITSP A Server #1 domain
register = my UserID:my password@ITSP A Server #2 domain
register = my UserID:my password@ITSP B #1 domain
notransfer=yes
bindport=4569
bindaddr=0.0.0.0
bandwidth=low
disallow=all
allow=ulaw
allow=g729
jitterbuffer=yes
forcejitterbuffer=no
tos=lowdelay
autokill=yes

[ITSP_B]
context=incoming-iax
type=friend
qualify=2000
host=ITSP B #1 domain
user=my UserID
username=my UserID
auth=md5
secret=my password
disallow=all
allow=ulaw
;
; *** ITSP_A Inbound ***
[ITSP_A]
context=incoming-iax
type=user
auth=md5
username=my UserID
secret=my password
disallow=all
allow=ulaw
;
; *** ITSP_A Outbound ***
[ITSP_A-Out]
type=peer
host=ITSP A Server #1 domain
auth=md5
username=my UserID
secret=my password
disallow=all
qualify=yes
allow=ulaw
;
[ITSP_A-Out2]
type=peer
host=ITSP A Server #2 domain
auth=md5
username=my UserID
secret=my password
disallow=all
qualify=yes
allow=ulaw
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RE: [asterisk-users] Caller ID Rewrite

Am Freitag, den 01.12.2006, 20:03 + schrieb David Bath:
 Hi Anselm,
 
 Thanks for the help...
 
 I'm slightly confused as to your response.
 
 Wouldn't that look for a /dialled/ number in the format _0number try
 and jump to another extension 0044number with priority 1?
 
 If so, that's not what I'm trying to achieve.

Sorry, my brain is in need for a weekend off work. I obviously
understood your question wrong. My fault.

 I've made a bit more progress... and my current diaplan entry looks like
 this:
 
 exten=123456,1,Set(${IF(REGEX(^0[1-9][0-9]\{9\}$
 {CALLERID(number)})?CALLERID(number)=44${CALLERID(number):1})})
 exten = 123456,2,Goto(sipinternal,101,1)
 
 BUT! There's a very odd problem (and I'm sure it's my fault..) the
 second callerid function is not being evaluated...

I _think_ the IF is a string evaluation, so the format should be like
SET MYVARIABLE =  [IF condition? value1 : value2]

(see http://www.voip-info.org/wiki/index.php?page=Asterisk+func+if )

Try
exten=123456,1,Set(CALLERID(number)=
${IF(REGEX(^0[1-9][0-9]\{9\}$ ${CALLERID(number)})?
44${CALLERID(number):1}:${CALLERID(number)})})

(two linebreaks to be removed)

HTH,

Anselm

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[asterisk-users] direct IP calling with extension

2006-12-01 Thread Jerry Geis


Am Freitag, den 01.12.2006, 13:44 -0500 schrieb Jerry Geis:

/ All,
// 
// If I have video phones behind an asterisk server (with 2 network cards)

// and all the phones have extensions. Internally everything works great.
// 
// Now for people that want to call my video phones external to my office

// is there a way to do that? On the extenal persons phone enter an IP/EXTEN
// where IP is my server and not the phone? Can that work?
// 
// Would I have to have PUBLIC IP address for every phone NAT'ed through my

// server to make the call?
/

AFAIR, define a context in the global section of sip.conf. Any
incoming SIP connections that are not identified to belong to any other
context (registration) will come thru that extensions.conf context.
Inside, just forward through to your proper local extensions:



[locals]
exten = 200,1,Dial(SIP/myphone1)
...
[sipfromoutside]
exten = johndoe,1,Goto(locals,200,1)
...



Then direct IP calling as   johndoe at 123.45.67.89 
http://lists.digium.com/mailman/listinfo/asterisk-users
should work. To get calling at a hostname working as well, you will need
a few records in your DNS setup.



example.com. IN A 123.45.67.89
sip.example.com. IN A 123.45.67.89
example.com. IN NAPTR 60 50 s SIP+D2U  _sip.udp.example.com.
_sip._udp.example.com. IN SRV 10 10 5060 sip.example.com.



Which will allow for johndoe at example.com 
http://lists.digium.com/mailman/listinfo/asterisk-users
I suspect the first example.com line is not necessary (such that you
can host your domain on a different server than that which runs
Asterisk), but I did not test.



HTH
Anselm



THanks, this seems to almost get me there... Once I call into the server
and goes to my locals I no longer get Video.

When I call the extension directly I get video no problem.

When I first call the server at my IP address then it routes to my local
I no longer get video.

Any ideas why that might be?

THanks,

Jerry

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[asterisk-users] Audiocodes MP104-FXO - Transfer the call only after 3 rings

2006-12-01 Thread Yuval Yogev

Hi all,
Has anybody a clue how to pass the call immidiatly ?
At the MP104 I put at the autodial a number of a sip extention.

Also, I noticed that after those 3 rings the extention is ringing but
not passing the call to voicemail as should be after a while like it
happens when you call from other sip extention.

We run Trixbox 1.2.3

Thanks
Yuval


 

Do you Yahoo!?
Everyone is raving about the all-new Yahoo! Mail beta.
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Re: [asterisk-users] Hold calling channel and ask called channel beforeconnect???

2006-12-01 Thread Moises Silva


I think you can, using the manager Originate action and the Bridge()
application on the dial plan. The bridge application is still not in
trunk ( AFAIK ), or you can try to test the framework were working on
to have complete control over the asterisk channels using a PHP
routing daemon. The page of the project is
http://opencallmanager.ivsol.net/

Best Regards

d have had no response.  Can someone tell me if I

am asking a stupid question, i.e. is the answer either obvious or
impossible?

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nigel J. Terry
Sent: Wednesday, November 22, 2006 10:27 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hold calling channel and ask called channel
beforeconnect???

I am a newbie.  Just got my Asterisk working and I love it.

I want to do the following, believe it should be possible, but can't work
out how:

When I get an incoming call, I want to answer and just send ringing to the
calling channel.
Then I want to call the destination channel, send a message asking if they
will accept the call, get a response (1 or 2) and then either connect the
parties (1) or send the calling channel to voicemail (2).

Any ideas, thanks

Nigel

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--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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RE: [asterisk-users] Caller ID Rewrite

2006-12-01 Thread David Bath

Hi,

Thanks for quick response.

I changed it as you suggested, but it has the same effect:

In the console I get:

--Executing
Set(SIP/604625-b79140a8,CALLERID(number)=44${CALLERID(number)}) in
new stack

It's running the IF code correctly, but in the true it's just not
evaluating the variable...

Cheers,

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin Hoffmeister
Sent: 01 December 2006 20:26
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Caller ID Rewrite

Am Freitag, den 01.12.2006, 20:03 + schrieb David Bath:
 Hi Anselm,
 
 Thanks for the help...
 
 I'm slightly confused as to your response.
 
 Wouldn't that look for a /dialled/ number in the format _0number try
 and jump to another extension 0044number with priority 1?
 
 If so, that's not what I'm trying to achieve.

Sorry, my brain is in need for a weekend off work. I obviously
understood your question wrong. My fault.

 I've made a bit more progress... and my current diaplan entry looks
like
 this:
 
 exten=123456,1,Set(${IF(REGEX(^0[1-9][0-9]\{9\}$
 {CALLERID(number)})?CALLERID(number)=44${CALLERID(number):1})})
 exten = 123456,2,Goto(sipinternal,101,1)
 
 BUT! There's a very odd problem (and I'm sure it's my fault..) the
 second callerid function is not being evaluated...

I _think_ the IF is a string evaluation, so the format should be like
SET MYVARIABLE =  [IF condition? value1 : value2]

(see http://www.voip-info.org/wiki/index.php?page=Asterisk+func+if )

Try
exten=123456,1,Set(CALLERID(number)=
${IF(REGEX(^0[1-9][0-9]\{9\}$ ${CALLERID(number)})?
44${CALLERID(number):1}:${CALLERID(number)})})

(two linebreaks to be removed)

HTH,

Anselm

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[asterisk-users] Interesting CALLERID behavior

2006-12-01 Thread Bruce Ferrell


I've found an interesting behavior in callerid handling.

I have very long callerid coming in or maybe just improperly combined 
information.  In any case the result is that the caller ID is set to 
asterisk on the outgoing leg.  Has anyone else seen this before?  Is 
there a solution for it?



Thanks in advance

--
One day at a time, one second if that's what it takes

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Re: [asterisk-users] direct IP calling with extension

Am Freitag, den 01.12.2006, 15:27 -0500 schrieb Jerry Geis:
 Am Freitag, den 01.12.2006, 13:44 -0500 schrieb Jerry Geis:
 THanks, this seems to almost get me there... Once I call into the server
 and goes to my locals I no longer get Video.
 
 When I call the extension directly I get video no problem.
 
 When I first call the server at my IP address then it routes to my local
 I no longer get video.
 
 Any ideas why that might be?

I suspect that is caused by asterisk not knowing about your video codecs
(I have never done video over IP myself, so just a guess).

If your devices talk to each other,
#   device = device
they can use all the codecs both of them know

As soon as asterisk is involved,
#   device = asterisk = device
the codec negotiation goes through asterisk. So asterisk must at least
_know_ the codec that those devices want to use.

I think I saw some info about that in the www.voip-wiki.org

BR
Anselm


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Re: [asterisk-users] How to park calls on a specific extension

2006-12-01 Thread John Novack




Tom Rymes wrote:

On Nov 30, 2006, at 8:55 PM, Brad Templeton wrote:


On Thu, Nov 30, 2006 at 02:50:21PM -0500, Tom Rymes wrote:

for example: In your example above where they can't figure out how to
transfer, why don't you edit features.conf and define the transfer
key as # or something. Then, when they have a call for Bill across
they way, they can do this:
One of the basic flaws in the current Asterisk, which has been mentioned 
before, is Transfer. Asterisk seems to have been designed on an obsolete 
PBX model that has been obsolete for the last 20 years, where users had 
a POTS phone and most everything was done with flash and feature codes.
In a modern phone system, attended transfer and blind transfer are not 
different functions
In most hybrid business systems one does NOT place a call on hold, but 
begins a transfer, either a specific function button or intercom button 
which automatically places the call on hold, gives a new dialtone and 
another extension is dialed. IF the called party answers, the 
transferrer can announce the call, and if the called party wants to 
accept the call, they simply hang up. Blind transfer is done the same 
way, but the transferrer doesn't wait for the called party to answer. If 
no one is home, the call goes to VM in the prescribed amount of time.
Problem with all of this is lack of line or loop keys, so if the 
transferrer needs to get back to the original party, there isn't a real 
way. Asterisk doesn't yet support what is called a shared line appearance.


Parking a call could work with a little training, and many hybrid 
systems support that as well. Put a call into a park orbit, announce the 
call and move on. IF the call stays in orbit for a period of time, does 
the call come back to the person who parked it?



Users really don't care if it is an Open Source effort or not. Users 
want something easy to use and reliable. Users want buttons and lights
Developers want new wiz bang features. They don't want to go back and 
fix or document what they have done
Look at your old Legend, Partner, Panasonic and NEC for models of a 
decent hybrid system, and build on that.


JMO

John Novack


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[asterisk-users] sip address in voicemail emails

2006-12-01 Thread Mark Price


hi,

I am using asterisk 1.2.10.
I am trying to send sip links in asterisk voicemail, so that users can
easily reply to emails.
This does not seem to be straightforward.
First, there seems to be no variable that prints out the domain name of the
sip call, since I am including every variable mentioned on
http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
Second, if i include in the emailbody variable sip:[EMAIL PROTECTED] (on
the wrong assumption that the phone call was from a telephone number), gmail
responds by assuming it is a mailto link, and outlook treats it as plain
text.  Having examined the emails being sent in mutt, it appears that the
message has no mime type (neither text nor text/html).  Unsurprisingly,
then, enclosing it in an href, i.e. a href=sip:[EMAIL PROTECTED]click
herea/ is of no help.

Is there  a workaround for this?

Perhaps it is being addressed in a later version of asterisk?  I have seen
no mention of it in the release notes for 1.4.

Thanks
Mark Price
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Re: [asterisk-users] Cisco 7970 SIP upgrade issues

2006-12-01 Thread Lacy Moore - Aspendora


On 11/29/06, Paul A Brown [EMAIL PROTECTED] wrote:


 Hi Mattias,

That is what I did for my 7960 and what I need to do for this. However my
problem is when I un tar the cisco file it won't run. I think it needs call
manager :-(




You apparently downloaded the wrong version.  I don't know what version you
downloaded.  You need the zip version of cmterm-7970-7971-sccp-7.0-3.  Unzip
it to your tftp directory.  There is no setup file.
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[asterisk-users] video for call attendant systems

2006-12-01 Thread Jerry Geis


Presently I have a wav file (voice) for my call attendant.

How do I specify a video file for a call attendant for video phones?

Thanks,

Jerry
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[asterisk-users] Re: sip address in voicemail emails

2006-12-01 Thread Mark Price


Hi,

Item zero, Thanks for hosting and participating in this list, digium, and
all the developers involved.

First, I realize that my first post probably did not belong on
asterisk-dev.  I had intended to remove that address from my recipients
list, but did not
Secondly,  I figure that now that my neck is extended, I should write to
clarify one thing and correct another in my first email.

On 12/1/06, Mark Price [EMAIL PROTECTED] wrote:


hi,

I am using asterisk 1.2.10.
I am trying to send sip links in asterisk voicemail, so that users can
easily reply to emails.
This does not seem to be straightforward.
First, there seems to be no variable that prints out the domain name of
the sip call, since I am including every variable mentioned on
http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf



To clarify to myself, I made a call from a different sip domain from a
username that does not exist on the asterisk box, and found out that it is
true: VM_CIDNUM contains the username, but not the domain name of the call.
Therefore, as long as the username is a telephone number, we can work around
that, but the message printed to describe a non-telephone-number phone call
will be incorrect.

Second, if i include in the emailbody variable sip:[EMAIL PROTECTED] (on

the wrong assumption that the phone call was from a telephone number), gmail
responds by assuming it is a mailto link, and outlook treats it as plain
text.  Having examined the emails being sent in mutt, it appears that the
message has no mime type (neither text nor text/html).



I thought I should point out I was incorrect.  The text portion of the email
is given mime-type text.  Therefore it appears to be impossible to send a
sip link in the email unless the receiving email client knows how to
recognize them (as many know how to recognize http:// and mailto: links).

Thanks,
Mark Price
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[asterisk-users] Re: No sound: X-Lite - Asterisk - VoIP Provider - Cellphone

2006-12-01 Thread Vincent Delporte


At 17:01 01/12/2006 +0100, Noah Miller [EMAIL PROTECTED] wrote:
Just to double check - have you limited the RTP ports on the asterisk 
server to 8000-8019 (in rtp.conf)?


Thanks. That what was missing. In rtp.conf, I fixed ports 1-10019 and 
mapped those ports on the router, and it worked.


Also, Xlite uses (or used to use) a silence suppresion mechanism that 
doesn't work too well with asterisk.  According to the WIKI:

Turn off Silence Supression (to avoid RFC3389 warnings on Asterisk console):
Menu | Advanced System Settings | Audio Settings | Silence Settings |
Transmit Silence: Yes

OK. However, the person on the other end tells me that my voice was very 
low, barely audible. Do you know what could be done about it? Are there 
voice-related settings in Asterisk that I should look at?


Would playing with canreinvite to remove Asterisk from the loop and have 
RTP packets go directly from the VoIP provider to my X-Lite client at home 
make a difference? What should I do if canreinvite=yes means that the VoIP 
provider doesn't use the RTP ports that I expect to use on my side?


Thank you.

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RE: [asterisk-users] Interesting CALLERID behavior

Caller ID should always be either ANI + CNAM (where available) on
inbound, or anonymous (No ANI).

If you are getting anything different from your Telco something is
wrong.

For SIP originated calls the CID is derived from the INVITE

Outbound caller ID is as you set it in your peer/user config. Name is
useless for calls handed to the PSTN over PRI or other digital TDM
interfaces since CNAM is looked up at the terminating end, not sent in
the signaling. Only the ANI is passed form you to the telco.

Unless you use 10 digit extensions, you have to specify callerid= for
each of your extensions.

If you are getting asterisk it is because the channels that originated
the calls have no callerid= in their config

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Bruce Ferrell
 Sent: Friday, December 01, 2006 2:33 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Interesting CALLERID behavior
 
 I've found an interesting behavior in callerid handling.
 
 I have very long callerid coming in or maybe just improperly combined
 information.  In any case the result is that the caller ID is set to
 asterisk on the outgoing leg.  Has anyone else seen this before?  Is
 there a solution for it?
 
 
 Thanks in advance
 
 --
 One day at a time, one second if that's what it takes
 
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Re: [asterisk-users] Caller ID Rewrite

2006-12-01 Thread Edwin Lam


David Bath wrote:

Hi,

Thanks for quick response.

I changed it as you suggested, but it has the same effect:

In the console I get:

--Executing
Set(SIP/604625-b79140a8,CALLERID(number)=44${CALLERID(number)}) in
new stack

It's running the IF code correctly, but in the true it's just not
evaluating the variable...


since the REGEX returns 1 if match. try this instead:

exten=123456,1,Set(CALLERID(number)=
${IF($[REGEX(^0[1-9][0-9]\{9\}$ ${CALLERID(number)}) = 1]?
44${CALLERID(number):1}:${CALLERID(number)})})



Cheers,

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin Hoffmeister
Sent: 01 December 2006 20:26
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Caller ID Rewrite

Am Freitag, den 01.12.2006, 20:03 + schrieb David Bath:


Hi Anselm,

Thanks for the help...

I'm slightly confused as to your response.

Wouldn't that look for a /dialled/ number in the format _0number try
and jump to another extension 0044number with priority 1?

If so, that's not what I'm trying to achieve.



Sorry, my brain is in need for a weekend off work. I obviously
understood your question wrong. My fault.



I've made a bit more progress... and my current diaplan entry looks


like


this:

exten=123456,1,Set(${IF(REGEX(^0[1-9][0-9]\{9\}$
{CALLERID(number)})?CALLERID(number)=44${CALLERID(number):1})})
exten = 123456,2,Goto(sipinternal,101,1)

BUT! There's a very odd problem (and I'm sure it's my fault..) the
second callerid function is not being evaluated...



I _think_ the IF is a string evaluation, so the format should be like
SET MYVARIABLE =  [IF condition? value1 : value2]

(see http://www.voip-info.org/wiki/index.php?page=Asterisk+func+if )

Try
exten=123456,1,Set(CALLERID(number)=
${IF(REGEX(^0[1-9][0-9]\{9\}$ ${CALLERID(number)})?
44${CALLERID(number):1}:${CALLERID(number)})})

(two linebreaks to be removed)

HTH,

Anselm

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--
Edwin Lam [EMAIL PROTECTED]
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

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Re: [asterisk-users] How to park calls on a specific extension

2006-12-01 Thread Steve Sobol

On Wed, 29 Nov 2006, Ira wrote:

 Either write what you want, or learn to use what we have and hope 

I am not averse to coding something. I'm actually looking at the source 
for parkandannounce and I've looked at the source for valetparking. 

Valetparking looks like it would do fine, but I can't get it to work. But 
I am going to do a little more research before I complain to the list that 
it's broken. :)

-- 
Steve Sobol, Professional Geek ** Java/VB/VC/PHP/Perl ** Linux/*BSD/Windows
Victorville, California PGP:0xE3AE35ED

It's all fun and games until someone starts a bonfire in the living room.

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Re: [asterisk-users] How to park calls on a specific extension

2006-12-01 Thread Steve Sobol

On Thu, 30 Nov 2006, Brad Templeton wrote:

 Problem there is only some phones have line buttons, and when they have
 them they are scarce and there's many things you might like to do with them,
 and dedicating them to this would be low on my list.   Dedicating one speed

Eventually I am going to do a little sleuthing to find out what my 
GXP-2000s' HOLD buttons send to Asterisk, and I'm going to make the HOLD 
button park a call. :)

Until then, I'm going to have to use an interim solution.

Isn't there a separate hold dialplan context?

-- 
Steve Sobol, Professional Geek ** Java/VB/VC/PHP/Perl ** Linux/*BSD/Windows
Victorville, California PGP:0xE3AE35ED

It's all fun and games until someone starts a bonfire in the living room.

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[asterisk-users] no tx audio

2006-12-01 Thread Justin Findlay

I have an asterisk server that connects to my voip provider over iax2.
Some of the POTS phone numbers I've called consistently get no tx audio.
This behavior happens with kiax on the same machine as the server and
with iaxcomm on an ubuntu desktop, a windows desktop, and a windows
laptop.

The ISP's tech came today and I used iaxcomm on his laptop to connect to
my asterisk server and called one of the problem numbers and the tx
audio worked.  After he left I tried iaxcomm on my roomate's windows
desktop and a friend's laptop, neither of which gave any tx audio.

I've been trying to debug this for weeks and still have no idea how to
proceed from here.


Justin
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Re: [asterisk-users] CALL TRANSFER

2006-12-01 Thread omar parihuana


Thanks!!!

I forget Tt option! (too basis!!)


On 12/1/06, Damon Estep [EMAIL PROTECTED] wrote:


 Your dial string must have either the t or T option set.


  --

*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *omar parihuana
*Sent:* Friday, December 01, 2006 9:10 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] CALL TRANSFER



Hi Guys,



I'm implementing my Asterisk step by step, so far the communications
between softphones, hardphones with Gateways, voice mail, are working fine.
Rightnow I would like to enable Call tranfer (like Traditional PBX) in Blind
Transfer and AttendXFER, I'm reading features.conf in accordance to
voip-info.org but the transfer doesn't work!  Please if you can provide me
some examples will be very appreciate.



Rgds.

--
Omar E.P.T
-
Certified Networking Professionals make better Connections!

http://omarept.blogspot.com/

  Usysnet Corp
Open Source Solutions
www.usysnet.com.pe

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--
Omar E.P.T
-
Certified Networking Professionals make better Connections!

http://omarept.blogspot.com/

 Usysnet Corp
Open Source Solutions
www.usysnet.com.pe
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RE: [asterisk-users] CALL TRANSFER

Be careful, if you set both T and t you might be allowing the wrong
party to transfer the call! In MOST cases you would want T or t, not T
and t, although there are some cases where you might want both.

 

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of omar
parihuana
Sent: Friday, December 01, 2006 5:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CALL TRANSFER

 

Thanks!!!

 

I forget Tt option! (too basis!!)

 

On 12/1/06, Damon Estep [EMAIL PROTECTED] wrote: 

Your dial string must have either the t or T option set.

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] ] On Behalf Of omar
parihuana
Sent: Friday, December 01, 2006 9:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Subject: [asterisk-users] CALL TRANSFER

 

Hi Guys,

 

I'm implementing my Asterisk step by step, so far the communications
between softphones, hardphones with Gateways, voice mail, are working
fine. Rightnow I would like to enable Call tranfer (like Traditional
PBX) in Blind Transfer and AttendXFER, I'm reading features.conf in
accordance to voip-info.org http://voip-info.org/  but the transfer
doesn't work!  Please if you can provide me some examples will be very
appreciate. 

 

Rgds.

-- 
Omar E.P.T
-
Certified Networking Professionals make better Connections!

http://omarept.blogspot.com/

  Usysnet Corp 
Open Source Solutions
www.usysnet.com.pe http://www.usysnet.com.pe/  


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-- 
Omar E.P.T
-
Certified Networking Professionals make better Connections!

http://omarept.blogspot.com/

  Usysnet Corp
Open Source Solutions
www.usysnet.com.pe 

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RE: [asterisk-users] How to park calls on a specific extension

2006-12-01 Thread Ken Williams

I was able to set a program to speed dial the park extension.  Then a user just 
hits TNFR followed by the line I've programmed to speed dial park.  

If you get the HOLD button to do this, I'd love to hear how :).  

From: Steve Sobol
Sent: Fri 12/1/2006 5:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to park calls on a specific extension

On Thu, 30 Nov 2006, Brad Templeton wrote:

 Problem there is only some phones have line buttons, and when they have
 them they are scarce and there's many things you might like to do with them,
 and dedicating them to this would be low on my list.   Dedicating one speed

Eventually I am going to do a little sleuthing to find out what my 
GXP-2000s' HOLD buttons send to Asterisk, and I'm going to make the HOLD 
button park a call. :)

Until then, I'm going to have to use an interim solution.

Isn't there a separate hold dialplan context?

-- 
Steve Sobol, Professional Geek ** Java/VB/VC/PHP/Perl ** Linux/*BSD/Windows
Victorville, California PGP:0xE3AE35ED

It's all fun and games until someone starts a bonfire in the living room.

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Re: [asterisk-users] How to park calls on a specific extension

2006-12-01 Thread Brad Templeton

On Fri, Dec 01, 2006 at 04:55:51PM -0500, John Novack wrote:
 In most hybrid business systems one does NOT place a call on hold, but 
 begins a transfer, either a specific function button or intercom button 
 which automatically places the call on hold, gives a new dialtone and 
 another extension is dialed. IF the called party answers, the 
 transferrer can announce the call, and if the called party wants to 
 accept the call, they simply hang up. Blind transfer is done the same 


Alas, it's not possible in these days to make it that simple.  Today, almost
all calls will be answered (by a voice mail) if not by the person.  So you
need an additional UI for attended transfer, which allows you to say No,
I got the voice mail, disconnect the voice mail and bring me back to my
call.   I guess you needed that for endless ring in the old days too.

You're right that there is no big difference between attended and
unattended in the UI when it works, but when the attended call fails
with voicemail or unlimited ring, or for that matter busy signal, you need
a means to go back to the caller.

That's one thing soft buttons are good for, you can create soft buttons
for specialty functions like this.   If you have line buttons on
your phone, normally the original call is on one line button, and the
2nd call on another line button, so you just press the first line button
to abandon the call attempt.

On my Asterisk system, I have done another thing which is handy.  My
extension macro looks at the caller-id.  Calls within the house do not
ever go to voicemail.  Calls from outside (including ones transferred)
will go to voicemail after the timeout.  So I never get voicemail but
I do get endless ring.

Many PBXs also offered a feature that if you blind transfer, and the call
goes into endless ring that it transfers back to you after some timeout.
Today, voicemail has largely eliminated that.
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Re: [asterisk-users] How to park calls on a specific extension

2006-12-01 Thread Brad Templeton

On Fri, Dec 01, 2006 at 07:37:35PM -0700, Ken Williams wrote:
 I was able to set a program to speed dial the park extension.  Then a user 
 just hits TNFR followed by the line I've programmed to speed dial park.  
 
 If you get the HOLD button to do this, I'd love to hear how :).  

Oh, that would require new code in Asterisk, a new commmand that
is able to get all channels that are currently on hold, and connect
to one if only one or give a menu and connect to one if more than
one.   Don't know if it would require any fancy changes to holding
itself.

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Re: [asterisk-users] 1.4beta3 help