[Asterisk-Users] Keytouch without effect

[Hans-Peter Straub]

Hello anybody,

i have a problem on connecting an innovaphone ip202 to theAsterisk-PBX. When i 
dial in the PBX with the standard (make samples) configuration with the ip202 
the connection is fine, but to push any Key on the keypad dosn't take any 
effect. Is for H323-Phones a special DTMF config necessary?

Thanks

Hans-Peter Straub


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[Asterisk-Users] CPU load

[Anders Svensson]

Hi!

Here comes a newbi question.

I now that transcoding of codecs take a lot of cpu
load. But if I want to receive all traffic as IAX and then want to send it out
as SIP. Is it the same? Requires a lot of CPU and RAM?

 

 

 

Regards

Anders Svensson 

 






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RE: [Asterisk-Users] custom ring tone

[Marko Rakar]

but how to take advantage of this within asterisk?



Two atoms bump into each other. One says 
"I think I lost an electron!" The other 
asks, "Are you sure?", to which the 
first replies, "I'm positive."

mailto:[EMAIL PROTECTED]
http://printel.hr 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Vishnev
Sent: Friday, September 23, 2005 3:47 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] custom ring tone


Yes, sometime audio is both ways. Sometimes, it is just one way. This
only works in digital network (T1/E1 to T1/E1 (CAS handoff) - the window
is much shorter even in feature group "D". or T1/E1 to T1/E1 PRI
signaling where the window could be as large as 90 seconds. Again, that
depends on country, provider, switch software. You can't get this if you
are calling POTS lines.

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Re: [Asterisk-Users] Asterisk + GNUGK + Asterisk-Addons ooh323

[brett]

On 9/23/2005, "Brian C. Fertig" <[EMAIL PROTECTED]> wrote:

> I am having a slight issue.  I am trying to register 2 asterisk boxes with
> GNUGK and when I try to add the 2nd it gets denied cause of it saying its
> a duplicate.  How do I change the configs to allow more than one asterisk
> box register to the same GK?
> 
> brian

Don't 'quote me' on this but...  Look in the h323.conf/s and see if you
have two different h323id strings for the servers.  I think it defaults
to 'Asterisk PBX' but the current sample makes it 'ObjSysAsterisk'. I
am
pretty sure they have to have different names or GNUGK is going to think
they are the same.

Brett
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Re: [Asterisk-Users] Problems with sipura 1001's and 2002's

[Peter Wemm]

On Wednesday 21 September 2005 12:31 pm, Phil Allred wrote:
> I'm having lots of problems with sipura spa1001's and spa2002's. 
> Asterisk claims they are busy when they aren't.  Other times, it
> claims to be ringing them, but they aren't really ringing.  I have
> done the following to try to resolve the problem:
>
>   1)   I upgraded all my spa1001's and 2002's to their latest
> firmware (3.1.5).  This lessened, but did not resolve completely the
> problem.

Make sure you use the dual line firmware for the 1001.  A few months ago 
when I was tearing my hair out with phantom rings, I discovered that I 
had to reflash the firmware to the special dual-line one.  The default 
firmware wouldn't generate an actual ring.

-- 
Peter Wemm - [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED]
"All of this is for nothing if we don't go to the stars" - JMS/B5
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RE: [Asterisk-Users] SNOM 190 '486/Busy here' after upgrade to firmware 3.60s

[Christian Stredicke]

Looks like this phone has redirection or DND set. Anything on the
display? If it still a mystery send us the settings of the phone, then
it should become clear.

BTW if you have a snom trouble ticket, you can also go to
http://www.snom.com/onlinesupport.html (scroll down to set up an
account).

CS 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Colin Anderson
> Sent: Friday, September 23, 2005 1:31 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] SNOM 190 '486/Busy here' after 
> upgrade to firmware 3.60s
> 
> I have a Snom 190 that refuses to accept calls after upgrade 
> to firmware 3.60s, latest. I get "SIP 486/Busy Here". No 
> change in the dialplan, nor settings in the phone. Calls out 
> fine. I did 30 other phones yesterday with 3.60s with no 
> problem, this is the only one. In the phone's log I get:
> 
> [5]22/9/2005 17:01:00: timeout::callback: Registering with 
> timeout of 0 ms
> [5]22/9/2005 17:05:45: Match challenge for user=9002, realm=asterisk
> [2]22/9/2005 17:07:31: Denying call id=-4 
> reason=unconditional <--wtf does this mean?
> [5]22/9/2005 17:07:31: Dialog -4/2 going to terminated
> [5]22/9/2005 17:07:31: timeout::callback: Registering with 
> timeout of 0 ms
> 
> Any Snom factory guys on the list care to comment?
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> 
> 
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[Asterisk-Users] Asterisk + GNUGK + Asterisk-Addons ooh323

[Brian C. Fertig]

I am having a slight issue.  I am trying to register 2 
asterisk boxes with GNUGK
and when I try to add the 2nd it gets denied cause 
of it saying its a duplicate.  How
do I change the configs to allow more than one 
asterisk box register to the same GK?
 
brian
 

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Re: [Asterisk-Users] externpass

[Tzafrir Cohen]

On Thu, Sep 22, 2005 at 05:27:15PM -0300, Sebastian Kühner wrote:
> Hello,
> 
> I found another solution - a script that does the following steps:
> 
> 1.  Get the old password from the db
> 
> 2.  Update database (subscriberàpassword) with new password (that's for 
> my web interface and ser)
> 
> 3.  Replace old with new password in the file 
> /etc/asterisk/additional_ser_voicemail.conf
> 
> 4.  Restart asterisk with asterisk manager (php 
> /usr/local/etc/ser/scripts/ast-reload/reload_asterisk.php)

You mean that every time someone calls and changes a password,  reload
is triggered? (I hope you meant "reload" and not "restart").

-- 
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http://tzafrir.org.il |   | a Mutt's  
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RE: [Asterisk-Users] sipuras 841 bad sound

[Alex Ternero]

Upgrade the Firmware.

Alex

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Juan Jose
Comellas
Enviado el: martes, 20 de septiembre de 2005 23:23
Para: asterisk-users@lists.digium.com
Asunto: Re: [Asterisk-Users] sipuras 841 bad sound


Have you tried upgrading the firmware? I had several problems with the
outbound volume of these phones until I upgraded them.


On Tuesday 20 September 2005 20:46, Anton Krall wrote:
> Hi Guys!
>
> I have a problems with some sipuras 841 and asterisk 1.0.9.
>
> Im using 841 with asterisk 1.0.9 with a digium card (single e1 span) with
> steve's unicall.
>
> Everything compiled fine and in fact I can make and receive calls but I
> have a problem with bad sound when the sipuras call the outside E1's
lines.
> I can listen to the caller without problems but they heard me with a
choppy
> sound as if you were losing frames or cutting off. Calls between internal
> sipuras sound good (eventhough the speaker and headset sound comes and
> goes, for example, when you start talking, seems as if the sipuras takes a
> few seconds to catch up with you on volume so the remote user listen to
you
> as if the first words and the last were at low volume and the conversation
> in the middle sound good, any had that problem?)
>
> So, internal calls sound good between 841's but sound volume is weird at
> the start and end of a sentence.
> Calling the outside lines via E1's, I can listen to people without
problems
> but they heard me as choppy or cut off.
>
> Anybody had issues like this? Is it asterisk or the phones or what?
>
> Hope you can help Guys, Im really banging my head against the wall here.
>
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--
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([EMAIL PROTECTED])

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Re: [Asterisk-Users] Recently reported ASTCC audio issues

[Darren Wiebe]

I just tested this on Asterisk CVS-v1-0-09/22/05-22:23:34 built by 
[EMAIL PROTECTED] on a i686 running Linux

and it still works perfectly.

Darren Wiebe
[EMAIL PROTECTED]

Darren Wiebe wrote:

I'm running Asterisk CVS-v1-0-06/06/05-17:29:02 built by 
[EMAIL PROTECTED] on a i686 running Linux.


I just spent some time in testing this.  I tested the local and IAX2 
trunks.  Both worked flawlessly.


Any comments?

Darren Wiebe
[EMAIL PROTECTED]


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[Asterisk-Users] Re: SayUnixTime in CVS?

[Brian McEntire]

I've partially figured it out:

The voip-info.org wiki for SayUnixTime suggests that if you don't specify the 3rd parameter, it will default to:
  ABdY 'digits/at' IMp

For some reason, for my setup, the default wasn't working when I left
the parameter blank. When I specified the format in the extensions
line, it works fine:
  exten => 400,n,SayUnixTime(,EST5EDT,ABdY \'digits/at\' IMp )


I looked at the code and it seems to attempt to set the default format
string based on chan->language ... maybe my language isn't set up
correctly.

Anyway, hope that may be useful to someone.

On 9/22/05, Brian McEntire <[EMAIL PROTECTED]> wrote:
Can anyone tell me what I missed? I'm trying to setup a simple
extension (400) that reports the time when it is dialed. I searched the
threads and it seems like this should work...

Here's what's in my extensions.conf:

exten => 400,1,Answer()
exten => 400,n,Wait,1
exten => 400,n,SayUnixTime(,EST5EDT,)
exten => 400,n,Playback(tt-weasels)


[BTW, tt-weasels is hillarious!  ;-)  Props to whoever made that. ]


When I call the extension, it answers and immediately falls through to
tt-weasels, which I hear fine. It's like SayUnixTime gets jumped over
or returns nothing (very quickly.) I checked and the "at" sound which
is used by the default SayUnixTime format string is in place. Also,
SayDigits and SayNumber seem to work okay.

When asterisk starts up in verbose mode, it looks like SayUnixTime gets loaded okay:

[app_sayunixtime.so]Sep 23 03:24:04 VERBOSE[3854] logger.c:  [app_sayunixtime.so] => (Say time)
 == Registered application 'SayUnixTime'
 == Registered application 'DateTime'


This is what is output from asterisk -vvv when I dial 400:

    -- Starting simple switch on 'Zap/1-1'
    -- Executing Answer("Zap/1-1", "") in new stack
    -- Executing Wait("Zap/1-1", "1") in new stack
    -- Executing SayUnixTime("Zap/1-1", "|EST5EDT|") in new stack
    -- Executing Playback("Zap/1-1", "tt-weasels") in new stack
    -- Playing 'tt-weasels' (language 'en')
    -- Executing Wait("Zap/1-1", "5") in new stack


Any ideas?


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Re: [Asterisk-Users] Sip Groups

[Kevin Bockman]

Tracy Peek wrote:
I have 5 broadvoice accounts and have set up some database gets/puts to 
determine of the account is in use when someone needs to make an 
outbound call. I have tried adding "group=6" in sip.conf and then 
dial(sip/G6) in extensions.conf to no avail. Should the group 
declaration work in sip.conf?

No.  For -HEAD, use:
${GROUP_COUNT()}
and Set(GROUP())

Example:
GotoIf($[ ${GROUP_COUNT(g1)} > 22 ]?g2)
Set(GROUP()=g1)
NoOp(${GROUP_COUNT(g1)})

Otherwise, I think it corresponds to CheckGroup and SetGroup.



Kevin
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[Asterisk-Users] CVS-HEAD and Caller ID -- Pulling my hair out!

[pbx]

I have looked into this callerid problem now for a few hours.

1) Caller id on a sipura-2000 now shows:

cidname
2000

Where cidname is the new outputted formate from the cid_rewrite agi script
and 2000 is the exten number.

In looking at the Dial() application,

option "o"
  'o' -- Original (inbound) Caller*ID should be placed on the outbound
leg of the call
 instead of using the destination extension (old style
asterisk behavior)

I tried using this in my Dial string as an option but it wouldn't work, i
was getting the same Information..

Before I would have

Ben Cell
1234567890

appear on the sipura-2000 attached telephone.

The only thing i can get now is just

1234567890
2000 to appear on the phone

And when I try to use the new function set(CallerId(Name)="name") it says
that that function is not registered..

However when I use the SetCidName and SetCidNum i get this:

Sep 22 21:15:03 WARNING[25515]: app_setcidnum.c:72 setcallerid_exec:
SetCIDNum is deprecated, please use Set(CALLERID(number)=value) instead.
-- Executing SetCIDName("IAX2/66.234.228.170:4569-2", "xx") in
new stack
Sep 22 21:15:03 WARNING[25515]: app_setcidname.c:70 setcallerid_exec:
SetCIDName is deprecated, please use Set(CALLERID(name)=value) instead.


ARGH!  it's a never ending Circle

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Re: [Asterisk-Users] SOHO Survey / Creative Asterisk Solutions

[Brian McEntire]

Hehe... that's awesome  :)  I laughed out loud when I read it.

Someone else replied that they are going to use * to control their entry gate system by cell phone. Nice. 

Thanks for the examples! While reading over at voip-info.org, I found
the auto-dial feature that can be combined with .call files. That
should be perfect for an idea I had -- use cron and POP3 to check my
e-mail account for any new messages from the transit authority... if
there are any, dial/ring home phones at 6am and playback a message to
check e-mail for possible morning commute problems. Not quite as good
as drunkdial though  :)
On 9/22/05, Tom Hayden <[EMAIL PROTECTED]> wrote:
Well, there are about a billion DIY * projects out there.  Myslightly-insane use of asterisk on a tiny budget is running thesite:www.drunkdial.orgPeople call in - leave drunk messages then comment on them later.
Cheers,--Tom HaydenOn 9/22/05, Brian McEntire <[EMAIL PROTECTED]> wrote:> I hope the subject isn't too buzzword compliant  :)
>>  I'm just curious: What have people done with Asterisk? I'm particularly> interested in DIY projects and things that can be done on a small/home> office (or even hobbiest's) budget. If you have clever hacks or creative
> functionality you've implemented, I'd love to hear what a few people have> come up with.>>  Thanks!  -Brian>> ___> --Bandwidth and Colocation sponsored by 
Easynews.com -->> Asterisk-Users mailing list> Asterisk-Users@lists.digium.com> 
http://lists.digium.com/mailman/listinfo/asterisk-users> To UNSUBSCRIBE or update options visit:>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>--Tom
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[Asterisk-Users] SayUnixTime in CVS?

[Brian McEntire]

Can anyone tell me what I missed? I'm trying to setup a simple
extension (400) that reports the time when it is dialed. I searched the
threads and it seems like this should work...

Here's what's in my extensions.conf:

exten => 400,1,Answer()
exten => 400,n,Wait,1
exten => 400,n,SayUnixTime(,EST5EDT,)
exten => 400,n,Playback(tt-weasels)


[BTW, tt-weasels is hillarious!  ;-)  Props to whoever made that. ]


When I call the extension, it answers and immediately falls through to
tt-weasels, which I hear fine. It's like SayUnixTime gets jumped over
or returns nothing (very quickly.) I checked and the "at" sound which
is used by the default SayUnixTime format string is in place. Also,
SayDigits and SayNumber seem to work okay.

When asterisk starts up in verbose mode, it looks like SayUnixTime gets loaded okay:

[app_sayunixtime.so]Sep 23 03:24:04 VERBOSE[3854] logger.c:  [app_sayunixtime.so] => (Say time)
 == Registered application 'SayUnixTime'
 == Registered application 'DateTime'


This is what is output from asterisk -vvv when I dial 400:

    -- Starting simple switch on 'Zap/1-1'
    -- Executing Answer("Zap/1-1", "") in new stack
    -- Executing Wait("Zap/1-1", "1") in new stack
    -- Executing SayUnixTime("Zap/1-1", "|EST5EDT|") in new stack
    -- Executing Playback("Zap/1-1", "tt-weasels") in new stack
    -- Playing 'tt-weasels' (language 'en')
    -- Executing Wait("Zap/1-1", "5") in new stack


Any ideas?
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Re: [Asterisk-Users] anyone know about this company? www.blue-wireless.net

[C. Hatton Humphrey]

> I've got a customer who is thinking of installing this
>
> www.blue-wireless.net

Know nothing about the software but if it's anything like their
website then I'd be wary.

Hatton
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[Asterisk-Users] Sip Groups

[Tracy Peek]

I have 5 broadvoice accounts and have set up some database gets/puts to 
determine of the account is in use when someone needs to make an 
outbound call. I have tried adding "group=6" in sip.conf and then 
dial(sip/G6) in extensions.conf to no avail. Should the group 
declaration work in sip.conf?


Tracy Peek
[EMAIL PROTECTED]
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Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

[Matt Florell]

On 9/22/05, Matt Roth <[EMAIL PROTECTED]> wrote:
Matt,- Have you tried recording directly to GSM format? It will help reduce the
- bottleneck on disk IO although it will use more CPU cycles(in your case- on a RAM drive this may not help at all)We don't want to do any transcoding on the Asterisk server because ofthe drain on CPU. The RAM disk seems to be the best solution for our
scenario.
That's pretty much what I though, wanted to mention it though. 
- (Regarding "Avoided deadlock" warnings and their effects)- There aren't always audio skips but they do happen more when you get more
- ast_channel_walk warnings. The audio gaps are usually less than a quarter- second in our experience but can be upto a second depending on the- severity of the IO problem at that instance. It's very hard to test for
- until you get into production and you have real conversations and real- people listening to them that can hear the audio skips.Thank you for this information. It seems like the warnings won't causeany unacceptable results, but we'll be listening to the audio regularly
after we go into production.
When we traced the recording skips back to an exact time there is
always a ast_channel_walk warning in the logs, although there is not
always an audio skip if the warning appears, but the more of those
warnings, the more skips we got.

- We mostly do outbound and the volume is split across several servers, and- for inbound we do have forwarding to other servers if the defined
- capacity is exceeded a certain point.How are you calculating this capacity?
Trial and error,  astGUIclient/VICIDIAL has a very detailed 
system performance logging utility built into it that tracks open
asterisk channels, ram, CPU use %, system load and a few other things
and we were best able to find a good and safe capacity at which to
limit our systems to ensure reliability:
http://astguiclient.sourceforge.net/VDreports/performance_new.gif

- As for phone calls at one time: for inbound we almost never exceed 50- agents on a single server with no more than 72 incoming lines live at
once.- Our average is actually much less than that. For outbound we usually have- about 15-40 agents per server with upto 96 lines dialing outconcurrently.- At our main office location we've had over 100 agents on at one time
across- 6 Asterisk servers handling over 350 calls at once with a total ofmore than- 550 live channels on our Asterisk servers(includes recording, client and- trunk channels).I hope you don't mind answering some questions about your setup.
It seems like you have really solid numbers to work with. We'reimplementing our own reporting (both real-time and next-day) usingAsterisk's call detail records (stored in a MySQL database) andinformation captured from the management interface (via AstManProxy).
Are we reinventing the wheel? Do you have any tips for us in regards toour reporting software?
We don't use any Asterisk-generated logs, they didn't offer enough
information for us, so we created several of our own MySQL logs for
calls and Manager actions that we use to track every action that
happens across all systems as well as all live calls that are occuring
in real-time on all systems. This allows VICIDIAL to have real-time
status screens and datetime-level statistical performance reports
available any time you want to pull them. As for tips, just log
everything, you may not think you need it now but you will eventually
want just about every little piece of call and action information that
you can get your hands on for reporting.
I'm assuming you are using VICIDIAL as a predictive dialer inconjunction with Asterisk for your outbound calling. We are looking for
an outbound dialing solution. Could you please provide a list of theabilities and limitations of VICIDIAL?
Well, since we wrote VICIDIAL we are a little biased :) it is free and
GPL so you won't get many of the pretty features you get with those
expensive dialer solutions, but you can tinker with it all you like.
We've been using it for about 2 years in production and now have it on
over 200 seats across 4 locations for our company. There are also over
50 production VICIDIAL installations in over a dozen countries that we
know about. It works better for us than any of the dialer systems we
used to use and we have total control over it which makes it easier to
change things on the fly.  There are limitations and things that
we have not gotten around to writing yet, but there is an active
community around it and we are developing new features for it all the
time and releasing a new version every 2-8 weeks. One limitation is
Answering Machine detection. We haven't found a good method of doing
this and we don't like the delay that all commercial systems have when
doing AM detection so we just do a dial timeout by default, about 4-5
rings will eliminate 90% of the answering machines you receive without
any delay in the calls that are answered by a human. For other features
see the product page:
h

[Asterisk-Users] Recently reported ASTCC audio issues

[Darren Wiebe]

I'm running Asterisk CVS-v1-0-06/06/05-17:29:02 built by 
[EMAIL PROTECTED] on a i686 running Linux.


I just spent some time in testing this.  I tested the local and IAX2 
trunks.  Both worked flawlessly.


Any comments?

Darren Wiebe
[EMAIL PROTECTED]


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RE: [Asterisk-Users] custom ring tone

[Alex Vishnev]

Yes, sometime audio is both ways. Sometimes, it is just one way. This only
works in digital network (T1/E1 to T1/E1 (CAS handoff) - the window is much
shorter even in feature group "D". or T1/E1 to T1/E1 PRI signaling where the
window could be as large as 90 seconds. Again, that depends on country,
provider, switch software. You can't get this if you are calling POTS lines.



Hth
Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter
http://www.0xdecafbad.com
Sent: Thursday, September 22, 2005 9:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] custom ring tone

Audio both ways?  Sure would beat the collect call game :P



On Thu, 2005-09-22 at 21:15 -0400, Alex Vishnev wrote:
> Actually that is not true. You can have a short time where audio path is
> open prior to answering of the call. This depends on the provider, switch
> and software. I think the largest window I have seen is 90 seconds.
> 
> Alex
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of trixter
> http://www.0xdecafbad.com
> Sent: Thursday, September 22, 2005 4:22 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] custom ring tone
> 
> On Thu, 2005-09-22 at 22:01 +0200, Marko Rakar wrote:
> > yes, yes
> > 
> > the thing is that local telco uses this feature for their customer
> > support line and also one of wireless providers now also offers ability
> > to customize your ring tone
> > 
> > I was told that if you have analog or even ISDN BRI line that ring tone
> > is generated in your local teclo exchange, but if you have connection
> > like E1 that it is generated localy in your PBX (explanation being that
> 
> So in short you can have a toll free info line without actually paying
> for the toll free.  While its not interactive, by not sending answering
> supervision the caller is not charged.  Interesting concept they have
> there, sure beats the 10k resistor trick from the analog switch days
> (although then you could talk to the other person).
> 
> 
-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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RE: [Asterisk-Users] custom ring tone

[trixter http://www.0xdecafbad.com]

Audio both ways?  Sure would beat the collect call game :P



On Thu, 2005-09-22 at 21:15 -0400, Alex Vishnev wrote:
> Actually that is not true. You can have a short time where audio path is
> open prior to answering of the call. This depends on the provider, switch
> and software. I think the largest window I have seen is 90 seconds.
> 
> Alex
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of trixter
> http://www.0xdecafbad.com
> Sent: Thursday, September 22, 2005 4:22 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] custom ring tone
> 
> On Thu, 2005-09-22 at 22:01 +0200, Marko Rakar wrote:
> > yes, yes
> > 
> > the thing is that local telco uses this feature for their customer
> > support line and also one of wireless providers now also offers ability
> > to customize your ring tone
> > 
> > I was told that if you have analog or even ISDN BRI line that ring tone
> > is generated in your local teclo exchange, but if you have connection
> > like E1 that it is generated localy in your PBX (explanation being that
> 
> So in short you can have a toll free info line without actually paying
> for the toll free.  While its not interactive, by not sending answering
> supervision the caller is not charged.  Interesting concept they have
> there, sure beats the 10k resistor trick from the analog switch days
> (although then you could talk to the other person).
> 
> 
-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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[Asterisk-Users] anyone know about this company? www.blue-wireless.net

[Dean Collins]

I’ve got a customer who is thinking of installing this

www.blue-wireless.net

 

Anyone know anything about them? Strengths weaknesses etc?

 

Cheers,

Dean

 

 

 

 






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RE: [Asterisk-Users] custom ring tone

[Alex Vishnev]

Actually that is not true. You can have a short time where audio path is
open prior to answering of the call. This depends on the provider, switch
and software. I think the largest window I have seen is 90 seconds.

Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter
http://www.0xdecafbad.com
Sent: Thursday, September 22, 2005 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] custom ring tone

On Thu, 2005-09-22 at 22:01 +0200, Marko Rakar wrote:
> yes, yes
> 
> the thing is that local telco uses this feature for their customer
> support line and also one of wireless providers now also offers ability
> to customize your ring tone
> 
> I was told that if you have analog or even ISDN BRI line that ring tone
> is generated in your local teclo exchange, but if you have connection
> like E1 that it is generated localy in your PBX (explanation being that

So in short you can have a toll free info line without actually paying
for the toll free.  While its not interactive, by not sending answering
supervision the caller is not charged.  Interesting concept they have
there, sure beats the 10k resistor trick from the analog switch days
(although then you could talk to the other person).


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] Re: How can i call to a cellphone here in Mexico?

[Claudio Canseco]

Hi all,

I'm still on the same, i tried to changed as some of you told me, but
no changes, all my calls going to a cellphone are redirected to 066.
Could it be that it is ignore the 44 part of my dial extension??

exten => _904466,1,NoOp("Call for "(${EXTEN:1})  ;Llamadas a Celular
exten => _904466,2,Dial(Zap/1/ww${EXTEN:1})

I dont know what else i could do, I hope soon we´ll be buying an
TDM422B pci card, do you think this would get fixed using this
pseudo-tdm interface card?, instead of the current x100p clone i have?

Regards,
Claudio
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Re: [Asterisk-Users] [Fwd: ASTCC speaks and cut RTP channel, => Kind of solution...

[Darren Wiebe]

I will look into this and post back what I find.

Darren Wiebe
[EMAIL PROTECTED]

Ricardo Poppi wrote:


Yes Darren. The problem is the same using Zap or SIP. I had no
oportunity to verify that using IAX or E1/T1.

Rgds, Ricardo Poppi.

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Re: [Asterisk-Users] Tux/Asterisk logo for Cisco phones

[David McNett]

I made http://slacker.com/~nugget/stuff/asterisk-cow-real.bmp for my
non-Linux asterisk servers.

-- 
David McNett <[EMAIL PROTECTED]>
http://slacker.com/~nugget/
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[Asterisk-Users] Re: Set Log Level for Messages log file

[Arik Funke]

Thank you very much for clarifying this. Once I saw "for win32 only" I 
never tried whether it worked for me...


I will take a close look at logger.conf now.

Thanks,
Arik


Jesse Keating wrote:

On Fri, 2005-09-23 at 01:10 +0200, Arik Funke wrote:


Hi Jesse,

According to voip-info logger.conf is "for Win32 platform only". Is this 
wrong? If not, I do not quite understand how it will help with a linux 
system. I am a bit at a loss here...


Thanks for the effort!




That must be outdated info.  I use logger.conf to process my logging on
a CentOS4 x86_64 system.  Thats not windows (:

pandora*CLI> logger reload
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger restarted

pandora*CLI> logger show channels
Channel Type StatusConfiguration
---  ---
/var/log/asterisk/full.pandora.game File Enabled- Debug Verbose Warning 
Notice Error
/var/log/asterisk/messages.pandora. File Enabled- Warning Notice Error
Console  Enabled- Warning Notice Error



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[Asterisk-Users] Re: Set Log Level for Messages log file

[Jesse Keating]

On Fri, 2005-09-23 at 01:10 +0200, Arik Funke wrote:
> Hi Jesse,
> 
> According to voip-info logger.conf is "for Win32 platform only". Is this 
> wrong? If not, I do not quite understand how it will help with a linux 
> system. I am a bit at a loss here...
> 
> Thanks for the effort!
> 

That must be outdated info.  I use logger.conf to process my logging on
a CentOS4 x86_64 system.  Thats not windows (:

pandora*CLI> logger reload
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger restarted

pandora*CLI> logger show channels
Channel Type StatusConfiguration
---  ---
/var/log/asterisk/full.pandora.game File Enabled- Debug Verbose Warning 
Notice Error
/var/log/asterisk/messages.pandora. File Enabled- Warning Notice Error
Console  Enabled- Warning Notice Error


-- 
Jesse Keating
GameHouse -- Systems Engineer

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[Asterisk-Users] Extended SIP registration failures

[Mark Hulber]

I had some complaints today that one of my incoming SIP numbers was 
failing for several hours.  I looked at my console and didn't see 
anything unusual but SIP show registry confirmed that my registrations 
were in a failed state.  I did a SIP reload and saw this in the output:


Sep 22 18:54:48 NOTICE[9997]: sched.c:296 ast_sched_del: Attempted to 
delete nonexistent schedule entry 312896!
Sep 22 18:54:48 NOTICE[9997]: sched.c:296 ast_sched_del: Attempted to 
delete nonexistent schedule entry 313687!



After the reload the registrations were successful.  Without knowing 
much about it, apparently they were scheduled to retry but that failed 
for some reason.  Any ideas?


MARK.
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Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

[Matt Roth]

Hi everyone,

This is just another attempt to address everybody in one place and 
consolidate the thread.




Zoa,

- I think its the best you can do.
If something pops into your head, even if it's off the wall, don't 
hesitate to share it.  Any idea is better than no idea if things go south.


- Maybe there should be some option to be set for the monitor command to
- buffer, with a warning that it will eat memory.
- Its also not needed to buffer the complete call at once, just buffering
- and writing to disk every 10 seconds would already be a big improvement,
- and since the calls won't all start at the same time, the writing to
- disk will be spread over time.
This would be a nice option. res_monitor.c is using ast_writefile() to 
handle writing each call leg to file. Maybe a more specialized file 
writing function that includes buffering or threading is all that is needed.


- Good luck with the implementation, let us know how it turns out!
- Try to find out what happens if the nfs share is gone, or overloaded too.
Thanks. The information provided by members of this list is invaluable, 
so I'll be sure to keep you informed of our progress. The plan is to 
have the application that is called via the MONITOR_EXEC hook verify the 
NFS mount before the copy. If it's not there it'll have to write locally 
until the problem is resolved.


The NFS mount will have it's own dedicated network interface (an Intel 
Pro 1000 MT) communicating to the digital recording server over a 
Gigabit connection.  That should be sufficient.


- If there is no zaptel hardware involved, i don't think you have to be
- too concerned about the network cards interrupts.
Nope, we're terminating our Ts in a Cisco AS5400HPX Universal Gateway 
and using ztdummy as a timing source for music on hold. Our Asterisk 
server (a Dell PowerEdge 6850) has configurable IRQs, so I can fix any 
conflicts that may arise with other hardware relatively easily.




Matt,

- Have you tried recording directly to GSM format? It will help reduce the
- bottleneck on disk IO although it will use more CPU cycles(in your case
- on a RAM drive this may not help at all)
We don't want to do any transcoding on the Asterisk server because of 
the drain on CPU. The RAM disk seems to be the best solution for our 
scenario.


- (Regarding "Avoided deadlock" warnings and their effects)
- There aren't always audio skips but they do happen more when you get more
- ast_channel_walk warnings. The audio gaps are usually less than a quarter
- second in our experience but can be upto a second depending on the
- severity of the IO problem at that instance. It's very hard to test for
- until you get into production and you have real conversations and real
- people listening to them that can hear the audio skips.
Thank you for this information. It seems like the warnings won't cause 
any unacceptable results, but we'll be listening to the audio regularly 
after we go into production.


- We have sevaral call centers as well, and we just restrict a single server
- to 50 recordings at once and then we would pass the next recording as an
- IAX2 channel to another recording server. It's a scalable system for us
- that is relatively cheap and works well since we can mix and gsm-encode
- the audio on these multiple servers at night when not in production
- leaving the NSF server just for storage and not audio processing.
That is a smart solution. Thanks for sharing it. If we have any problems 
with our setup we may start looking in this direction.


- (Regarding queue management, call volume, and system architecture of
-  his Asterisk setup)
- We wrote VICIDIAL(part of the GPL astGUIclient suite
- http://astguiclient.sf.net) for our call center operations. Yes it does
- use a central queue and does not use Asterisk queues or agents. The
- system is based on a MySQL database and meetme rooms for the agents that
- use a web-client app for lead information and call control.
We'll have to take a look at your work. We have to manage multiple 
queues and plan on using Asterisk queues and agents on our single 
Asterisk server.


- We mostly do outbound and the volume is split across several servers, and
- for inbound we do have forwarding to other servers if the defined
- capacity is exceeded a certain point.
How are you calculating this capacity?

- As for our distributed recording approach, it's easy with meetme rooms to
- record a call on one server from another server, you just drop a call into
- the meetme room that is a monitor exten on another server over IAX2, TDMoE
- or a crossover Zap T1 connection.
Great idea. Once again, if we have problems with our implementation it's 
great to have this alternative path to explore.


- As for phone calls at one time: for inbound we almost never exceed 50
- agents on a single server with no more than 72 incoming lines live at 

[Asterisk-Users] SNOM 190 '486/Busy here' after upgrade to firmwa re 3.60s

[Colin Anderson]

I have a Snom 190 that refuses to accept calls after upgrade to firmware
3.60s, latest. I get "SIP 486/Busy Here". No change in the dialplan, nor
settings in the phone. Calls out fine. I did 30 other phones yesterday with
3.60s with no problem, this is the only one. In the phone's log I get:

[5]22/9/2005 17:01:00: timeout::callback: Registering with timeout of 0 ms
[5]22/9/2005 17:05:45: Match challenge for user=9002, realm=asterisk
[2]22/9/2005 17:07:31: Denying call id=-4 reason=unconditional <--wtf does
this mean?
[5]22/9/2005 17:07:31: Dialog -4/2 going to terminated
[5]22/9/2005 17:07:31: timeout::callback: Registering with timeout of 0 ms

Any Snom factory guys on the list care to comment?
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[Asterisk-Users] Re: Set Log Level for Messages log file

[Arik Funke]

Hi Jesse,

According to voip-info logger.conf is "for Win32 platform only". Is this 
wrong? If not, I do not quite understand how it will help with a linux 
system. I am a bit at a loss here...


Thanks for the effort!

- Arik

-
Original Message:
Date: Thu, 22 Sep 2005 13:46:35 -0700
From: Jesse Keating <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Set Log Level for Messages log file
To: asterisk-users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain

On Thu, 2005-09-22 at 22:33 +0200, Arik Funke wrote:

>>
>> is there a way to set the log level to the equivalent of:
>>
>> asterisk -vdc
>>
>> Or anything similar? I have I problem I cannot really trace with the
>> standard log level.


Have you looked at logger.conf and the full variable?

-- Jesse Keating GameHouse
-- Systems Engineer
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Re: [Asterisk-Users] Codec conversion

[Asterisk guy]

for sip calls,   asterisk is able to convert a incoming g729 cal to a outgoing G.711 call. 
 
For oh323,  I am unable to get asterisk to convert a incoming g729 call to a outgoing G711 call .
 
 
my question is : For h323,   how to configure asterisk to convert a incoming h323/g729 calls to a outgoing h323 g.711 call  ?
 
any suggest are welcome.
 
 
On 1/17/05, Helder Rogério [MICROREDE] <[EMAIL PROTECTED]> wrote:

Hi!
 
Is there any way to receive in * server a call from a Terminal adapter in G.723/
G.729 and then convert it to G.711?
 
I'm wondering this because I can only place all thru Broadvoice in G.711 but most of customers have ADSL connection with 128k upstream, so the result is that they can hear in excellent conditions but can't be heard very well the sound is all choppy. even directly to broadvoice thru Xten sip client.

 
So the idea was to act as "proxy" and "codec converter" so that the communication coming out their router is the smaller it can get. I've mentioned G729 or 
G.723 becuase their routers have it, (Draytek 2600V).
 
Thanks in advance for your suggestions
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[Asterisk-Users] priindication passthru TE410P EuroISDN?

[Bruno Voigt]

Hi all,
I have to asterisk-1.0.9BRIstuffed-0.2.0n boxes each equipped with a TE410P.

Box A is connected with pri1 to the PSTN.
Box B is connected with pri1 (cpe) to the Box A at pri2 (net).

Now I want Box B to dial out to the PSTN tunneled thru Box A
and have it get all ISDN indications in case of call failure, eg.
unallocated destination number etc.

But currently Box B always gets only "normal clearing".
Interestingly on Box A the NoOp Output of PRI_CAUSE and CAUSECODE after
the dial attempt
to an unallocated number is also empty.

How to access/query the reason for a failed ISDN dial attempt from the
dial plan,
I expected at last PRI_CAUSE to be filled.

Is it possible at all with asterisk - what modification to my attached
configs would you suggest?
Which version is better suited ? BRIstuffed or plain asterisk with
regards to this issue.
What about asterisk 1.2 - are there any enhancements in this area?

Thanks in advance for all hints,
Bruno


Log output from Box A:

< Protocol Discriminator: Q.931 (8)  len=46
< Call Ref: len= 2 (reference 31/0x1F) (Originator)
< Message type: SETUP (5)
< [04 03 80 90 a3]
< Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Speech (0)
<  Ext: 1  Trans mode/rate: 64kbps,
circuit-mode (16)
<  Ext: 1  User information layer 1: A-Law (35)
< [18 03 a9 83 9f]
< Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
 Protocol Discriminator: Q.931 (8)  len=14
> Call Ref: len= 2 (reference 32799/0x801F) (Terminator)
> Message type: SETUP ACKNOWLEDGE (13)
> [18 03 a9 83 9f]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
>ChanSel: Reserved
>   Ext: 1  Coding: 0   Number Specified   Channel
Type: 3
>   Ext: 1  Channel: 31 ]
> [1e 02 81 82]
> Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)
0: 0   Location: Private network serving the local user (1)
>   Ext: 1  Progress Description: Called
equipment is non-ISDN. (2) ]
-- Accepting overlap voice call from '01722270201' to
'041090007' on channel 0/31, span 2
-- Starting simple switch on 'Zap/124-1'
-- Executing Dial("Zap/124-1", "Zap/r1/041090007") in new stack
-- Making new call for cr 32770
-- Requested transfer capability: 0x00 - SPEECH
> Protocol Discriminator: Q.931 (8)  len=46
> Call Ref: len= 2 (reference 2/0x2) (Originator)
> Message type: SETUP (5)
> [04 03 80 90 a3]
> Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Speech (0)
>  Ext: 1  Trans mode/rate: 64kbps,
circuit-mode (16)
>  Ext: 1  User information layer 1: A-Law (35)
> [18 03 a9 83 81]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
>ChanSel: Reserved
>   Ext: 1  Coding: 0   Number Specified   Channel
Type: 3
>   Ext: 1  Channel: 1 ]
> [6c 0d 41 83 30 31 37 32 32 32 37 30 32 30 31]
> Calling Number (len=15) [ Ext: 0  TON: Subscriber Number (4)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>   Presentation: Presentation allowed of
network provided number (3) '01722270201' ]
> [70 0e c1 30 34 31 30 39 30 30 30 30 30 30 30 37]
> Called Number (len=16) [ Ext: 1  TON: Subscriber Number (4)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '041090007' ]
-- Called r1/041090007
> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 32799/0x801F) (Terminator)
> Message type: PROGRESS (3)
> [1e 02 81 88]
> Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)
0: 0   Location: Private network serving the local user (1)
>   Ext: 1  Progress Description: Inband
information or appropriate pattern now available. (8) ]
< Protocol Discriminator: Q.931 (8)  len=10
< Call Ref: len= 2 (reference 32770/0x8002) (Terminator)
< Message type: SETU

[Asterisk-Users] problems with sending fax from SIP channels

[Andy Kuo]

Hi All,
 
I'm having problem sending fax from SIP extensions (Linksys PAP2) through Asterisk Zap channels (ISDN PRI).
The SIP extensions can receive fax without problems, but sending fax fails most of the time.
 
Does anyone have this problem?
 
Please advice.
Thank you.
AK
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Re: [Asterisk-Users] Get SIP to work over very limited network access

[steve]

On Wed, 21 Sep 2005, Sebastian A. Espindola wrote:

> > I've got a friend who's spending 6 months on the other side of the world. So
> > before he left I configured him a softphone on his laptop to connect to my
> > asterisk so he can call home free of charge.
> >
> > Unfortunately, he just found out he has horrible internet connection.

I hope he enjoys his stay in South Africa ;-)

Steve

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Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

[steve]

> - The reason i recommended you to use a ramdisk is because i think the
> - problem with recording to disk is saving 20ms of stream 1, then 20 ms of
> - stream 2, then 20ms of stream 3 etc etc meaning you write everytime
> - very small things. (with a lot of seeking).


I was thinking about this idly in the car yesterday, and imagined that we 
could send all the Monitor audio out in a single multiplexed stream into a 
single file.  

Then we can either dumultiplex after, or reassemble the audio stream on
the fly during playback.  Generally more is recorded than listened to.

Steve

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Re: [Asterisk-Users] bristuff-0.2.0RC8m

[Arik Funke]

Hello,

I have not been following the asterisk list the last few weeks but from
the number of requests I have received asking this question I guess it
is best to send my answer to the list for reference.

So here is what I did to compile bristuff on Fedora Core 4.

Make the following changes to these files:
zaphfc/zaphfc.c, line 1000, remnove the leading "(ulong)"
channel.c, line 64 (AST_MUTEX_DEFINE_STATIC(uniquelock);), comment out
channel.c, line 302 (ast_mutex_lock(&uniquelock);), comment out
channel.c, line 304 (ast_mutex_unlock(&uniquelock);), comment out

I have to caution however that I have only a vague idea of the effects
these changes to the source code actually have on the behaviour of
asterisk. In any case I would carefully check before using it on any
production mode system. This said, I have been running my asterisk
server (a small one) with this for a few weeks/months without
registering any problems that I have traced to this hack.

I have sent this hack to [EMAIL PROTECTED] but never got an answer back.
Either I have simply adressed myself to the wrong contact, or ...

Anyway, I hope this helps everybody who encounters the same problem as
me. I am of course happy to help anybody if these instructions do not
solve the problem - if I can. If anybody finds out more about this
please send me a CC. Thanks.

With best regards,
Arik Funke


PS: I have the suspicion that the problems arise from a change to gcc.
Did anybody at all encounter this problem with a gcc with version before
4.0.1 or even 3.xx? (try gcc --version for version info)


Jens P. Hansen wrote:

Hi there,

Did you ever get below fixed ? (Have same problem here..).

Best regards
Jens P. Hansen


Arik Funke wrote:


Hi,
I have following problems compiling bristuff-0.2.0RC8m. Has anybody 
seen them before and can point me in the right direction?


zaphfc/make all:
--
/usr/src/asterisk/bristuff-0.2.0-RC8m/zaphfc/zaphfc.c: In function 
'hfc_findCards':
/usr/src/asterisk/bristuff-0.2.0-RC8m/zaphfc/zaphfc.c:1000: error: 
invalid lvalue in assignment

make[2]: *** [
/usr/src/asterisk/bristuff-0.2.0-RC8m/zaphfc/zaphfc.o] Error 1
make[1]: *** [_module_
/usr/src/asterisk/bristuff-0.2.0-RC8m/zaphfc] Error 2
make[1]: Leaving directory '/usr/src/linux-2.6.12'
make: *** [linux26] Error 2

asterisk-1.0.9/make all:

channel.c64: error: static declaration of 'uniquelock' follows 
non-static declaration
include/asterisk/channel.h:58: error: previous declaration of 
'uniquelock' was here

make: *** [channel.o] Error 1


I am using following:
-
Kernel: Fedora Core: 2.6.12-1.1398_FC4-i686
CPU: AMD Sempron 2800+
GCC: 4.0.1

Thanks very much for the effort.

Cheers,
Arik
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[Asterisk-Users] rtp problems

[Anton Krall]

Guys.

Im having some audio problems.

My asterisk box is the router for my network but remote softphone cant
connect to it. I tried running 1.2beta1 and cvs-head with the same luck and
etheral throws this:

577.358145 201.138.93.80 -> 201.129.249.28 SIP Request: REGISTER
sip:voip.intruder.com.mx:5060
577.359698 201.129.249.28 -> 201.138.93.80 SIP Status: 100 Trying(1
bindings)
577.359723 201.129.249.28 -> 201.138.93.80 SIP Status: 401 Unauthorized
(1 bindings)
577.420875 201.129.249.28 -> 201.138.93.80 SIP Request: OPTIONS
sip:[EMAIL PROTECTED]:5060
577.424962 201.138.93.80 -> 201.129.249.28 SIP Request: REGISTER
sip:voip.intruder.com.mx:5060
577.426443 201.129.249.28 -> 201.138.93.80 SIP Status: 100 Trying(1
bindings)
577.466468 201.138.93.80 -> 201.129.249.28 SIP Status: 405 Method Not
Allowed
577.559090 201.129.249.28 -> 201.138.93.80 SIP Status: 200 OK(1
bindings)
584.530456 201.138.93.80 -> 201.129.249.28 UDP Source port: 5061
Destination port: 5060
590.052883 201.138.93.80 -> 201.129.249.28 SIP Request: OPTIONS
sip:voip.intruder.com.mx:5060
590.054351 201.129.249.28 -> 201.138.93.80 SIP Status: 404 Not Found
594.538727 201.138.93.80 -> 201.129.249.28 UDP Source port: 5061
Destination port: 5060
604.555242 201.138.93.80 -> 201.129.249.28 UDP Source port: 5061
Destination port: 5060
610.036528 201.138.93.80 -> 201.129.249.28 SIP Request: OPTIONS
sip:voip.intruder.com.mx:5060
610.037968 201.129.249.28 -> 201.138.93.80 SIP Status: 404 Not Found

Can anybody shed al ight on this messgaes for me please?


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Re: [Asterisk-Users] Set Log Level for Messages log file

[Jesse Keating]

On Thu, 2005-09-22 at 22:33 +0200, Arik Funke wrote:
> 
> is there a way to set the log level to the equivalent of:
> 
> asterisk -vdc
> 
> Or anything similar? I have I problem I cannot really trace with the 
> standard log level.

Have you looked at logger.conf and the full variable?

-- 
Jesse Keating
GameHouse -- Systems Engineer

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RE: [Asterisk-Users] custom ring tone

[Fernando Herrera]

If you use FXO you must answer before sending media. If you use BRI, I think
you should be able to send media after sending Progress/Alerting. 

But, watch out!! As trixter said in a previous mail

>> Check your local laws on that

Local regulations may limit the amount of information you can send before
connect. Here in Argentina, you have 1 minute of "wait for answer" time.

Kind regards, 


Fernando Herrera

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de John Hill
Enviado el: Jueves, 22 de Septiembre de 2005 17:25
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: RE: [Asterisk-Users] custom ring tone

I was thinking of PSTN over FXO cards. When I see PSTN I think pots.

You mentioned BRI whould PRI do as well?

--john

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Fernando 
> Herrera
> Sent: Thursday, September 22, 2005 3:07 PM
> To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - 
> Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] custom ring tone
> 
>  
> John,
> >> Ringback is provided by your PSTN provider until answer by
> asterisk.
> >> You have no control until you answer
> 
> Generally the ringback tone is sent by the last ClassV/Class IV switch 
> in the telephony path. This is for Telco's to send inband 
> error/progress/information announcements. However, some telcos just 
> send back the relase indicating a certain Release Cause Value and 
> letting you (in case you are another Telco) decide whether to play an 
> announcement or not.
> 
> Marko,
> I think that the DIAL command will match your needs. When you get an 
> incoming call to your asterisk (through any channel, let's say, just 
> as an example, the incoming call comes from an ITSP through a SIP
> channel) you
> configure the Asterisk to send the Music On Hold as a ring back tone 
> (Dial(SIP/1234|90|m)). Though, when you got an incoming call, this 
> will
> happen:
> 
> 1. The ITSP sends an INVITE to your asterisk 2. Asterisk answers with 
> a TRYING 3. Then. Asterisk will send a 183 (Session Progress) and you 
> start transmiting RTP. Normally, you will send the RTP for ring back 
> tone (tuuu tuuu). Here, you will send music on hold through the RTP 
> channel.
> 4. At this very same moment, the asterisk's end user's phone starts 
> ringing.
> 
> 
> 
> You will be able to implement such thing with SIP or H.323 channels if 
> you connect to PSTN through an ITSP. In case your asterisk is 
> connected to PSTN through POTS, you will only be able to do it if you 
> use ISDN.
> If you are
> using FXS/FXO, you won't be able to do it, since in this case the 
> ringback tone is generated by the TELCO's Class V switch.
> 
> Kind regards,
> 
> Fernando Herrera
> 
> -Mensaje original-
> De: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] En nombre de John 
> Novack Enviado el: Jueves, 22 de Septiembre de 2005 16:46
> Para: Asterisk Users Mailing List - Non-Commercial Discussion
> Asunto: Re: [Asterisk-Users] custom ring tone
> 
> 
> 
> Marko Rakar wrote:
> 
> >I am not interested in Dial app, I want the callers who are calling 
> >FROM pstn TO asterisk to hear different kind of ring tone (wav, mp3, 
> >gsm or whatever)
> >
> >  
> >
> 
> ??
> Ringback is provided by your PSTN provider until answer by asterisk.
> You have no control until you answer
> Then you go to IVR, VM or ??
> 
> John Novack
> 
> >For users within asterisk domain who actually use Dial
> command it does
> >not matter and I know that I can have full control over them
> >
> >
> >
> >  
> >
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[Asterisk-Users] Set Log Level for Messages log file

[Arik Funke]

Hi,

is there a way to set the log level to the equivalent of:

asterisk -vdc

Or anything similar? I have I problem I cannot really trace with the 
standard log level.


Thanks,
Arik
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Re: [Asterisk-Users] externpass

[Sebastian Kühner]

Hello,
 
I found another solution - a script that does the following steps:
 

1.  
Get the old password from 
the db
2.  
Update database 
(subscriberàpassword) with new password (that's for my web 
interface and ser)
3.  
Replace old with new 
password in the file /etc/asterisk/additional_ser_voicemail.conf
4.  
Restart asterisk with 
asterisk manager (php 
/usr/local/etc/ser/scripts/ast-reload/reload_asterisk.php)
 
It's working very good...
 
Thanks for your help!
 
Sebastian
 
 
 
 
- Original Message - 
From: "Tzafrir Cohen" <[EMAIL PROTECTED]>
To: 
Sent: Thursday, September 22, 2005 3:46 PM
Subject: Re: [Asterisk-Users] externpass
> On Thu, Sep 22, 2005 at 02:25:02PM -0300, Sebastian Kühner 
wrote:> > Hello,> > > > If I change the password 
in the Voicemail-Menu without externpass, the> > password gets changed 
correctly. But if I use a extern script, the password> > is still the 
same as before.> > > > What do I have to do in my external 
script to change the password in the> > asterisk's memory, 
too?> > Is http://bugs.digium.com/view.php?id=4602 
relevant?> > -- > Tzafrir 
Cohen | [EMAIL PROTECTED] | VIM 
is> http://tzafrir.org.il 
|   
| a Mutt's  > [EMAIL PROTECTED] 
|   
|  best> ICQ# 
16849755 
|   
| friend> ___> 
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http://lists.digium.com/mailman/listinfo/asterisk-users> 
To UNSUBSCRIBE or update options visit:>    http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] custom ring tone

[trixter http://www.0xdecafbad.com]

On Thu, 2005-09-22 at 14:02 -0600, Colin Anderson wrote:
> >??
> >Ringback is provided by your PSTN provider until answer by asterisk.
> >You have no control until you answer
> >Then you go to IVR, VM or ??
> 
> OP said T1/E1, so (usually) there's no ringing delay until Asterisk picks up
> like in a POTS line. Answering a T1/PRI line is transparent to the caller
> and then you can fake any ringtone you want. So:

Check your local laws on that, in America there was a telephone company
many many years ago that got into trouble for doing that type of stuff
as the caller was billed but got what they thought was a ring or busy
but never an answer.  I dont know exactly which state or what set of
laws the telco was found to have violated, but the call was not answered
technically yet answering supervision was sent.  The laws may have
changed by now or been obsoleted by new ones I have no idea.  


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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RE: [Asterisk-Users] custom ring tone

[John Hill]

I was thinking of PSTN over FXO cards. When I see PSTN I think pots.

You mentioned BRI whould PRI do as well?

--john

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Fernando Herrera
> Sent: Thursday, September 22, 2005 3:07 PM
> To: [EMAIL PROTECTED]; 'Asterisk Users Mailing 
> List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] custom ring tone
> 
>  
> John,
> >> Ringback is provided by your PSTN provider until answer by 
> asterisk.
> >> You have no control until you answer
> 
> Generally the ringback tone is sent by the last ClassV/Class 
> IV switch in
> the telephony path. This is for Telco's to send inband
> error/progress/information announcements. However, some 
> telcos just send
> back the relase indicating a certain Release Cause Value and 
> letting you (in
> case you are another Telco) decide whether to play an 
> announcement or not. 
> 
> Marko,
> I think that the DIAL command will match your needs. When you get an
> incoming call to your asterisk (through any channel, let's 
> say, just as an
> example, the incoming call comes from an ITSP through a SIP 
> channel) you
> configure the Asterisk to send the Music On Hold as a ring back tone
> (Dial(SIP/1234|90|m)). Though, when you got an incoming call, 
> this will
> happen:
> 
> 1. The ITSP sends an INVITE to your asterisk
> 2. Asterisk answers with a TRYING
> 3. Then. Asterisk will send a 183 (Session Progress) and you start
> transmiting RTP. Normally, you will send the RTP for ring 
> back tone (tuuu
> tuuu). Here, you will send music on hold through the RTP channel. 
> 4. At this very same moment, the asterisk's end user's phone 
> starts ringing.
> 
> 
> 
> You will be able to implement such thing with SIP or H.323 
> channels if you
> connect to PSTN through an ITSP. In case your asterisk is 
> connected to PSTN
> through POTS, you will only be able to do it if you use ISDN. 
> If you are
> using FXS/FXO, you won't be able to do it, since in this case 
> the ringback
> tone is generated by the TELCO's Class V switch. 
> 
> Kind regards, 
> 
> Fernando Herrera
> 
> -Mensaje original-
> De: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] En nombre de 
> John Novack
> Enviado el: Jueves, 22 de Septiembre de 2005 16:46
> Para: Asterisk Users Mailing List - Non-Commercial Discussion
> Asunto: Re: [Asterisk-Users] custom ring tone
> 
> 
> 
> Marko Rakar wrote:
> 
> >I am not interested in Dial app, I want the callers who are calling 
> >FROM pstn TO asterisk to hear different kind of ring tone (wav, mp3, 
> >gsm or whatever)
> >
> >  
> >
> 
> ??
> Ringback is provided by your PSTN provider until answer by asterisk.
> You have no control until you answer
> Then you go to IVR, VM or ??
> 
> John Novack
> 
> >For users within asterisk domain who actually use Dial 
> command it does 
> >not matter and I know that I can have full control over them
> >
> >
> >
> >  
> >
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Re: [Asterisk-Users] Will Digium Wildard work with PCI-X or PCI Express

[Matt Roth]

Just correcting myself.  The 3 PCI-X slots are one 64-bit 133 MHz and 
two 64-bit 100 MHz.


Matt

Matt Roth wrote:

Don't bank on it.  We were going to use a Wildcard as a timing source 
on our Dell PowerEdge 6850 and the BIOS didn't see it.  Depending on 
the PCI-X slot I installed it in, sometimes the box wouldn't even 
boot.  For perspective the 6850 has 4 PCI-e slots, and 3 PCI-X slots 
(one 64-bit 133 MHz, two 32-bit 100 MHz).


I believe the timing is only needed for music on hold, IAX trunking, 
and MeetMe conferencing.  We're not doing trunking or conferencing 
(for now) so we're going with ztdummy.  If the timing isn't perfect 
only our music on hold will suffer, which is no big deal.  If we run 
into other problems, we might try popping our quad-span card in there 
just to see if it works.


Keep in mind that Digium no longer produces Wildcards.  I'm not sure 
why they don't work with our 6850 and the techs at Dell didn't know 
either.  Maybe they are not 100% PCI compliant.


Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

Kevin Bockman wrote:


Chuck Bunn wrote:

Does anyone know if the Digium Wildcard will work on a PCI Express 
or PCI-X motherboard. Specifically I am looking at the Dell 850 1U 
rack server for use with Asterisk.




They will work in PCI-X of course  but not PCI Express.  They are 
totally different.


You will need the 3.3v cards.


Kevin
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RE: [Asterisk-Users] custom ring tone

[trixter http://www.0xdecafbad.com]

On Thu, 2005-09-22 at 22:01 +0200, Marko Rakar wrote:
> yes, yes
> 
> the thing is that local telco uses this feature for their customer
> support line and also one of wireless providers now also offers ability
> to customize your ring tone
> 
> I was told that if you have analog or even ISDN BRI line that ring tone
> is generated in your local teclo exchange, but if you have connection
> like E1 that it is generated localy in your PBX (explanation being that

So in short you can have a toll free info line without actually paying
for the toll free.  While its not interactive, by not sending answering
supervision the caller is not charged.  Interesting concept they have
there, sure beats the 10k resistor trick from the analog switch days
(although then you could talk to the other person).


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] Will Digium Wildard work with PCI-X or PCI Express

[Matt Roth]

Don't bank on it.  We were going to use a Wildcard as a timing source on 
our Dell PowerEdge 6850 and the BIOS didn't see it.  Depending on the 
PCI-X slot I installed it in, sometimes the box wouldn't even boot.  For 
perspective the 6850 has 4 PCI-e slots, and 3 PCI-X slots (one 64-bit 
133 MHz, two 32-bit 100 MHz).


I believe the timing is only needed for music on hold, IAX trunking, and 
MeetMe conferencing.  We're not doing trunking or conferencing (for now) 
so we're going with ztdummy.  If the timing isn't perfect only our music 
on hold will suffer, which is no big deal.  If we run into other 
problems, we might try popping our quad-span card in there just to see 
if it works.


Keep in mind that Digium no longer produces Wildcards.  I'm not sure why 
they don't work with our 6850 and the techs at Dell didn't know either.  
Maybe they are not 100% PCI compliant.


Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

Kevin Bockman wrote:


Chuck Bunn wrote:

Does anyone know if the Digium Wildcard will work on a PCI Express or 
PCI-X motherboard. Specifically I am looking at the Dell 850 1U rack 
server for use with Asterisk.



They will work in PCI-X of course  but not PCI Express.  They are 
totally different.


You will need the 3.3v cards.


Kevin
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RE: Re: [Asterisk-Users] custom ring tone

[John Hill]

Forgot. You must answer the line first.

Other than that Asterisk is not involved with the external pstn until it is
answered.

--john 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Marko Rakar
> Sent: Thursday, September 22, 2005 2:31 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: Re: [Asterisk-Users] custom ring tone
> 
> I am not interested in Dial app, I want the callers who are 
> calling FROM
> pstn TO asterisk to hear different kind of ring tone (wav, mp3, gsm or
> whatever)
> 
> For users within asterisk domain who actually use Dial command it does
> not matter and I know that I can have full control over them
> 
> 
> 
> Two atoms bump into each other. One says 
> "I think I lost an electron!" The other 
> asks, "Are you sure?", to which the 
> first replies, "I'm positive."
> 
> mailto:[EMAIL PROTECTED]
> http://printel.hr 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> John Hill
> Sent: Thursday, September 22, 2005 9:26 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: Re: [Asterisk-Users] custom ring tone
> 
> 
> Look at the dial app. I think it has several options.
> 
> Most custom 'TONES' are wav, acc, mp3 etc. files.  
> If you can set a different MOH class or perhaps a playback file in the
> dial app that plays a file that is a 'RING TONE' that may work.
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RE: Re: [Asterisk-Users] custom ring tone

[John Hill]

I have a [pstn-inbound] that calls a dial app that plays music to the pstn
caller.

--john
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Marko Rakar
> Sent: Thursday, September 22, 2005 2:31 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: Re: [Asterisk-Users] custom ring tone
> 
> I am not interested in Dial app, I want the callers who are 
> calling FROM
> pstn TO asterisk to hear different kind of ring tone (wav, mp3, gsm or
> whatever)
> 
> For users within asterisk domain who actually use Dial command it does
> not matter and I know that I can have full control over them
> 
> 
> 
> Two atoms bump into each other. One says 
> "I think I lost an electron!" The other 
> asks, "Are you sure?", to which the 
> first replies, "I'm positive."
> 
> mailto:[EMAIL PROTECTED]
> http://printel.hr 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> John Hill
> Sent: Thursday, September 22, 2005 9:26 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: Re: [Asterisk-Users] custom ring tone
> 
> 
> Look at the dial app. I think it has several options.
> 
> Most custom 'TONES' are wav, acc, mp3 etc. files.  
> If you can set a different MOH class or perhaps a playback file in the
> dial app that plays a file that is a 'RING TONE' that may work.
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RE: [Asterisk-Users] custom ring tone

[Fernando Herrera]

 
John,
>> Ringback is provided by your PSTN provider until answer by asterisk.
>> You have no control until you answer

Generally the ringback tone is sent by the last ClassV/Class IV switch in
the telephony path. This is for Telco's to send inband
error/progress/information announcements. However, some telcos just send
back the relase indicating a certain Release Cause Value and letting you (in
case you are another Telco) decide whether to play an announcement or not. 

Marko,
I think that the DIAL command will match your needs. When you get an
incoming call to your asterisk (through any channel, let's say, just as an
example, the incoming call comes from an ITSP through a SIP channel) you
configure the Asterisk to send the Music On Hold as a ring back tone
(Dial(SIP/1234|90|m)). Though, when you got an incoming call, this will
happen:

1. The ITSP sends an INVITE to your asterisk
2. Asterisk answers with a TRYING
3. Then. Asterisk will send a 183 (Session Progress) and you start
transmiting RTP. Normally, you will send the RTP for ring back tone (tuuu
tuuu). Here, you will send music on hold through the RTP channel. 
4. At this very same moment, the asterisk's end user's phone starts ringing.



You will be able to implement such thing with SIP or H.323 channels if you
connect to PSTN through an ITSP. In case your asterisk is connected to PSTN
through POTS, you will only be able to do it if you use ISDN. If you are
using FXS/FXO, you won't be able to do it, since in this case the ringback
tone is generated by the TELCO's Class V switch. 

Kind regards, 

Fernando Herrera

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de John Novack
Enviado el: Jueves, 22 de Septiembre de 2005 16:46
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] custom ring tone



Marko Rakar wrote:

>I am not interested in Dial app, I want the callers who are calling 
>FROM pstn TO asterisk to hear different kind of ring tone (wav, mp3, 
>gsm or whatever)
>
>  
>

??
Ringback is provided by your PSTN provider until answer by asterisk.
You have no control until you answer
Then you go to IVR, VM or ??

John Novack

>For users within asterisk domain who actually use Dial command it does 
>not matter and I know that I can have full control over them
>
>
>
>  
>
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Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

[Matt Florell]

We are 100% Asterisk on the VOIP side. We use SIP, IAX and Zap(channelbanks) for phones and Zap T1s for telco termination.

MATT---On 9/22/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
Hi Matt,Is your solution 100% Asterisk or are you using other "helpers" such as SER or XXXproxy or whatever?Thanks,Waldo
On Sep 21, 2005, at 12:45 PM, Matt Roth wrote:
All, 
 
This
message has generated a lot of responses, so I'm going to address each
of them here in an attempt to consolidate the thread. 
 
 
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RE: [Asterisk-Users] custom ring tone

[Colin Anderson]

>??
>Ringback is provided by your PSTN provider until answer by asterisk.
>You have no control until you answer
>Then you go to IVR, VM or ??

OP said T1/E1, so (usually) there's no ringing delay until Asterisk picks up
like in a POTS line. Answering a T1/PRI line is transparent to the caller
and then you can fake any ringtone you want. So:

1. Create your ringtone, call it MyCustomRingtone.mp3 & copy it to
/var/lib/asterisk/mohmp3

2. Edit /etc/asterisk/musiconhold.conf:

[classes]

default => MyCustomRingtone:/var/lib/asterisk/mohmp3

3. Modify extensions.conf as follows:

[my-inbound-context]

exten => s,1,Wait(1) 'Wait 1 so PRI has time to send Caller ID name fully
exten => s,2,Answer()
exten => s,3,Dial(SIP/blabla,40,m) <--the 'm' parameter plays music on hold
whilst dialling. In a perfect world, the 'MyCustomRingtone.mp3' file will
play, faking a ringtone to the caller

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RE: [Asterisk-Users] custom ring tone

[Marko Rakar]

yes, yes

the thing is that local telco uses this feature for their customer
support line and also one of wireless providers now also offers ability
to customize your ring tone

I was told that if you have analog or even ISDN BRI line that ring tone
is generated in your local teclo exchange, but if you have connection
like E1 that it is generated localy in your PBX (explanation being that
when you hev E1 that telco is giving control over call to your PBX when
you dial last number in the phone number which determines that the rest
of the range is yours; so if you have tel number 4800600 and have a
range of 100 numbers, then when you dial 48006 whatever happens after
that is generated on your local PBX because when you press that "6"
telco transfers everything to you because od DID and other
functionality)



Two atoms bump into each other. One says 
"I think I lost an electron!" The other 
asks, "Are you sure?", to which the 
first replies, "I'm positive."

mailto:[EMAIL PROTECTED]
http://printel.hr 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan & Company, LLC
Sent: Thursday, September 22, 2005 9:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] custom ring tone


I think Dial will work for you too, though: In your incoming from pstn 
context, Answer, then Dial providing music on hold or Backgrounding an 
audio file.

I assume, however, you don't want to answer the line at all.  You don't 
want the remote caller to be billed for this call, and you just want 
them to receive an informational message, am I correct?
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Re: [Asterisk-Users] Tux/Asterisk logo for Cisco phones

[Mark Phillips]

Excellent!!!

Thanks fellas.

Mark


David Mallwitz wrote:

Mark Phillips wrote:

I was at VON in Boston today and saw on the Digium stand a Cisco 7960 
with a picture of Tux and the Asterisk log on its display. I WANT IT!


Anyone know where I can download this file please?




http://www.loligo.com/asterisk/cisco/79xx/2003-04-27.examples/asterisk-tux.bmp 


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--

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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Re: [Asterisk-Users] Polycom IP500 Quickstart page or files?

[Andres Paglayan]

After dealing with a Poly 301 I rather use the FTP server and config 
files, even for a single phone,

download the manual and stuff from freedomphones.net/polycom

Tom Vile wrote:


but you do not get all of the features via a web browser to customize.

On 9/22/05, *Tom Hayden* <[EMAIL PROTECTED]  
> wrote:


Huh? You can easily configure an IP500 via a web browser. Just point
the URL to the IP addr of the telephone.

--
Tom Hayden


On 9/22/05, Wilson Pickett <[EMAIL PROTECTED]
> wrote:
> Hi,
>
> I just got my ip500 back after months of waiting. Is there an
easy way
> to get it hooked up to asterisk without [t]ftp servers and all
that or
> is there a quickstart page somewhere?
>
> tia
>
> r
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--
Tom
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
Phone: 518-631-2855 x205
Fax: 518-631-2856



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Re: [Asterisk-Users] custom ring tone

[Mojo with Horan & Company, LLC]

I think Dial will work for you too, though: In your incoming from pstn 
context, Answer, then Dial providing music on hold or Backgrounding an 
audio file.


I assume, however, you don't want to answer the line at all.  You don't 
want the remote caller to be billed for this call, and you just want 
them to receive an informational message, am I correct?


Marko Rakar wrote:

I am not interested in Dial app, I want the callers who are calling FROM
pstn TO asterisk to hear different kind of ring tone (wav, mp3, gsm or
whatever)

For users within asterisk domain who actually use Dial command it does
not matter and I know that I can have full control over them



Two atoms bump into each other. One says 
"I think I lost an electron!" The other 
asks, "Are you sure?", to which the 
first replies, "I'm positive."


mailto:[EMAIL PROTECTED]
http://printel.hr 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Hill
Sent: Thursday, September 22, 2005 9:26 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: Re: [Asterisk-Users] custom ring tone


Look at the dial app. I think it has several options.

Most custom 'TONES' are wav, acc, mp3 etc. files.  
If you can set a different MOH class or perhaps a playback file in the

dial app that plays a file that is a 'RING TONE' that may work.
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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] custom ring tone

[John Novack]

Marko Rakar wrote:


I am not interested in Dial app, I want the callers who are calling FROM pstn 
TO asterisk to hear different kind of ring tone (wav, mp3, gsm or whatever)

 



??
Ringback is provided by your PSTN provider until answer by asterisk.
You have no control until you answer
Then you go to IVR, VM or ??

John Novack


For users within asterisk domain who actually use Dial command it does
not matter and I know that I can have full control over them



 


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Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

[Waldo Rubinstein]

Hi Matt,Is your solution 100% Asterisk or are you using other "helpers" such as SER or XXXproxy or whatever?Thanks,WaldoOn Sep 21, 2005, at 12:45 PM, Matt Roth wrote:All,  This message has generated a lot of responses, so I'm going to address each of them here in an attempt to consolidate the thread.   ___
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Re: [Asterisk-Users] Call getting disconnected in queue

[Rajkumar S]

Bump!

raj

Rajkumar S wrote:

Hi,

I have a small call center with 4 Zap lines and 4 agents. Agents login 
using sip phones with AgentCallbackLogin. I occasionally gets a 
complaint that when customers call the call center, after the initial 
greeting is over the call gets cut after playing the thank you message. 
I started investigating and found that that happens when the call gets 
transferred to an agent who is making an outbound call (either calling 
customers or logging out). The debug logs of one such conversation is 
given below:


As you can read below, the call gets fwd to agent 1005 at SIP/1004. But 
he is trying to log off at the same time, and call gets disconnected.


Any help to fix this will be very much appreciated.

regards,

raj

   -- Executing Answer("Zap/2-1", "") in new stack
-- Executing Goto("Zap/2-1", "MainMenu|s|1") in new stack
-- Goto (MainMenu,s,1)
-- Executing BackGround("Zap/2-1", "Welcome") in new stack
-- Playing 'Welcome' (language 'en')
-- Playing 'agent-incorrect' (language 'en')
  == Spawn extension (from-sip, 2002, 1) exited non-zero on 'SIP/1004-cc2f'
-- Executing Queue("Zap/2-1", "callcenter|tT|||300") in new stack
-- Started music on hold, class 'default', on Zap/2-1
-- Stopped music on hold on Zap/2-1
-- Playing 'queue-youarenext' (language 'en')
-- Executing AgentCallbackLogin("SIP/1004-e376", "|l") in new stack
-- Playing 'agent-user' (language 'en')
-- Told Zap/2-1 in callcenter their queue position (which was 1)
-- Playing 'queue-thankyou' (language 'en')
-- Started music on hold, class 'default', on Zap/2-1
-- outgoing agentcall, to agent '1005', on 'Local/[EMAIL PROTECTED],1'
-- Executing Dial("Local/[EMAIL PROTECTED],2", "SIP/1004") in new 
stack
Sep 21 11:56:39 ERROR[12633]: chan_sip.c:1615 update_user_counter: Call 
from user '1004' rejected due to usage limit of 1

-- Couldn't call 1004
  == Everyone is busy/congested at this time
-- Called Agent/1005
-- Playing 'agent-incorrect' (language 'en')
  == Spawn extension (from-sip, 2002, 1) exited non-zero on 'SIP/1004-e376'
-- Timeout on Local/[EMAIL PROTECTED],2
  == CDR updated on Local/[EMAIL PROTECTED],2
-- Executing BackGround("Local/[EMAIL PROTECTED],2", "vm-goodbye") 
in new stack

-- Playing 'vm-goodbye' (language 'en')
-- Agent/1005 answered Zap/2-1
-- Stopped music on hold on Zap/2-1
-- Executing Hangup("Local/[EMAIL PROTECTED],2", "") in new stack
  == Spawn extension (from-sip, t, 2) exited non-zero on 
'Local/[EMAIL PROTECTED],2'
monitor executing ( nice -n 19 soxmix 
"/var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-in.wav" 
"/var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-out.wav" 
"/var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010.wav" 
 && rm -f 
"/var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-"* 
) &

  == Spawn extension (MainMenu, s, 2) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'

sip.conf entry for the phone is

[1004]
host=dynamic
type=friend
dtmfmode=RFC2833
username=1004
secret=password
context =  from-sip
disallow=all
allow=speex
allow=gsm
incominglimit=1


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RE: Re: [Asterisk-Users] custom ring tone

[Marko Rakar]

I am not interested in Dial app, I want the callers who are calling FROM
pstn TO asterisk to hear different kind of ring tone (wav, mp3, gsm or
whatever)

For users within asterisk domain who actually use Dial command it does
not matter and I know that I can have full control over them



Two atoms bump into each other. One says 
"I think I lost an electron!" The other 
asks, "Are you sure?", to which the 
first replies, "I'm positive."

mailto:[EMAIL PROTECTED]
http://printel.hr 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Hill
Sent: Thursday, September 22, 2005 9:26 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: Re: [Asterisk-Users] custom ring tone


Look at the dial app. I think it has several options.

Most custom 'TONES' are wav, acc, mp3 etc. files.  
If you can set a different MOH class or perhaps a playback file in the
dial app that plays a file that is a 'RING TONE' that may work.
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RE: [Asterisk-Users] Cisco AS5XXX + CallerID Name

[B. J. Bomar]

Try putting the command "timers buffer-invite 5000" in the sip-ua config.
This works on both our 3640 and 7206.  I'm not sure if this command is
available in the 12.3 series as I have 12.3T on my equipment.

B. J.




-Original Message-
From: Max Braz [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, September 21, 2005 20:29
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco AS5XXX + CallerID Name

Hi guys.

We have currently Asterisk CVS-v1-0-08/15/05-15:53:48
connected in SIP with a Cisco AS5300 (IOS 12.3). One
PRI is connected to the Cisco gateway. 

The problem we have is that on incoming PSTN calls to
the AS5300, relayed in SIP to Asterisk, the callerID
name is not being transmitted. We received the
callerID number but no name. I know we are receiving
the name from the PRI in hexadecimal format (running
in debug mode) but the AS53003 doesn't do anything
with it.

We tried to setup remote-party-id in the Cisco and
setting trustripd=yes in sip.com but that did not
help. Also, I know we can send names to Asterisk from
the Cisco since I'm able to force the name on all
incoming calls from the PSTN -->router(config-sip-ua)#
calling-info pstn-to-sip from name set foo<-- The
callerid name foo is then displayed with the correct
callerid number for all incoming calls.

Debug mode in Asterisk shows that we are in fact
getting the remote-party-id header but no names in it.
Any ideas or suggestions?

Thanks,

Max








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RE: Re: [Asterisk-Users] custom ring tone

[John Hill]

Look at the dial app. I think it has several options.

Most custom 'TONES' are wav, acc, mp3 etc. files.  
If you can set a different MOH class or perhaps a playback file in the dial
app that plays a file that is a 'RING TONE' that may work.


-John
> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Domjan Attila
> Sent: Thursday, September 22, 2005 2:16 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: Re: [Asterisk-Users] custom ring tone
> 
> Dial(SIP/1234|90|m)
> the caller will hear music on hold while SIP/1234 is ringing
> 
> On Thu, 2005-09-22 at 20:18 +0200, Marko Rakar wrote:
> > yes, but I want this feature to be turned on for people who 
> are calling
> > my asterisk from PSTN
> > 
> > 
> > 
> > Two atoms bump into each other. One says 
> > "I think I lost an electron!" The other 
> > asks, "Are you sure?", to which the 
> > first replies, "I'm positive."
> > 
> > mailto:[EMAIL PROTECTED]
> > http://printel.hr 
> > 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Domjan
> > Attila
> > Sent: Thursday, September 22, 2005 7:45 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: PS: Re: [Asterisk-Users] custom ring tone
> > 
> > 
> > Hi,
> > Dial application with m option, if the telco accept from you.
> > 
> > On Thu, 2005-09-22 at 19:10 +0200, Marko Rakar wrote:
> > > Few weeks back local telco introduced option of custom 
> ring tones. I 
> > > am not talking about your phone ring tones but about ring 
> tones you 
> > > hear in your headset while phone is ringing on the other end.
> > > 
> > > If I understand correctly, ringing tone is generated localy on 
> > > asterisk if you are connected to phone network with E1/T1 
> connection. 
> > > Which means that instead of regular beep-beep tone we could send 
> > > something else to the caller in PSTN (like mp3 music).
> > > 
> > > Is there a way of customizing ring tone in asterisk and 
> if yes how?
> > ___
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> > 
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> -- 
> Domjan Attila <[EMAIL PROTECTED]>
> 
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Will Digium Wildard work with PCI-X or PCI Express

[Kevin Bockman]

Chuck Bunn wrote:
Does anyone know if the Digium Wildcard will work on a PCI Express or 
PCI-X motherboard. Specifically I am looking at the Dell 850 1U rack 
server for use with Asterisk.


They will work in PCI-X of course  but not PCI Express.  They are 
totally different.


You will need the 3.3v cards.


Kevin
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RE: Re: [Asterisk-Users] custom ring tone

[Domjan Attila]

Dial(SIP/1234|90|m)
the caller will hear music on hold while SIP/1234 is ringing

On Thu, 2005-09-22 at 20:18 +0200, Marko Rakar wrote:
> yes, but I want this feature to be turned on for people who are calling
> my asterisk from PSTN
> 
> 
> 
> Two atoms bump into each other. One says 
> "I think I lost an electron!" The other 
> asks, "Are you sure?", to which the 
> first replies, "I'm positive."
> 
> mailto:[EMAIL PROTECTED]
> http://printel.hr 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Domjan
> Attila
> Sent: Thursday, September 22, 2005 7:45 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: PS: Re: [Asterisk-Users] custom ring tone
> 
> 
> Hi,
> Dial application with m option, if the telco accept from you.
> 
> On Thu, 2005-09-22 at 19:10 +0200, Marko Rakar wrote:
> > Few weeks back local telco introduced option of custom ring tones. I 
> > am not talking about your phone ring tones but about ring tones you 
> > hear in your headset while phone is ringing on the other end.
> > 
> > If I understand correctly, ringing tone is generated localy on 
> > asterisk if you are connected to phone network with E1/T1 connection. 
> > Which means that instead of regular beep-beep tone we could send 
> > something else to the caller in PSTN (like mp3 music).
> > 
> > Is there a way of customizing ring tone in asterisk and if yes how?
> ___
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> 
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Domjan Attila <[EMAIL PROTECTED]>

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[Asterisk-Users] Will Digium Wildard work with PCI-X or PCI Express

[Chuck Bunn]

Hi,

Does anyone know if the Digium Wildcard will work on a PCI Express or 
PCI-X motherboard. Specifically I am looking at the Dell 850 1U rack 
server for use with Asterisk.


Thanks
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[Asterisk-Users] logging in problem

[prashant yadav]

  
i have registered on teliax service and i m using a hathway internet connection.with X-lite phone it is not logging in .it says login timed out whereas the phone with same X-lite and service settings i m getting logged in all other internet connection and the phone also works perfectly. the issue only remains with hathway internet connection.what is the nature of problem i need immediate help regarding it 
thanking you 
prashant



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Re: [Asterisk-Users] Polycom IP500 Quickstart page or files?

[Tom Vile]

but you do not get all of the features via a web browser to customize.On 9/22/05, Tom Hayden <[EMAIL PROTECTED]
> wrote:Huh? You can easily configure an IP500 via a web browser. Just pointthe URL to the IP addr of the telephone.
--Tom HaydenOn 9/22/05, Wilson Pickett <[EMAIL PROTECTED]> wrote:> Hi,>> I just got my ip500 back after months of waiting. Is there an easy way
> to get it hooked up to asterisk without [t]ftp servers and all that or> is there a quickstart page somewhere?>> tia>> r> ___
> --Bandwidth and Colocation sponsored by Easynews.com -->> Asterisk-Users mailing list> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users> To UNSUBSCRIBE or update options visit:>
http://lists.digium.com/mailman/listinfo/asterisk-users>--Tom___--Bandwidth and Colocation sponsored by Easynews.com
 --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP TelephonyPhone: 518-631-2855 x205Fax: 518-631-2856
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Re: [Asterisk-Users] POP3 and TTS (Festival?)

[Tzafrir Cohen]

In short: use the right tool for the joob. See below.

On Wed, Sep 21, 2005 at 05:34:23PM -0300, Alchaemist wrote:
> Hi,
> 
> Has anybody seen a non commercial, or freeware, or GPL, or even 
> CHEAP... POP/IMAP to Text-to-speech?

Pull pop3/imap/imaps/whatever with fetchmail (or getmail, or whatever).

deliver it to a script using procmail.

Now you are left with the problem of TTS from a local mail message. What
is the expected format of the message?

BTW: you can plug just about anything else instead of POP3: e.g: use
gotmail to fetch messages from hotmail.

> 
> I have a working version for POP3 using festival. It DOES 
> work... it even cleans the email contents to get the actual content. It 
> works great with Outlook emails and similar, and skips non 
> multipart/alternative (that would be mainly SPAM, where the email is just 
> html or multipart without the text/plain alternative).

Pass them first through spamassasin or whatever. You can easily
integrate spam filtering in procmail. Or in the MTA.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] Cisco AS5XXX + CallerID Name

[[EMAIL PROTECTED]]

You need to run a late 12.3T to get name passed via SIP.

Max Braz wrote:

Hi guys.

We have currently Asterisk CVS-v1-0-08/15/05-15:53:48
connected in SIP with a Cisco AS5300 (IOS 12.3). One
PRI is connected to the Cisco gateway. 


The problem we have is that on incoming PSTN calls to
the AS5300, relayed in SIP to Asterisk, the callerID
name is not being transmitted. We received the
callerID number but no name. I know we are receiving
the name from the PRI in hexadecimal format (running
in debug mode) but the AS53003 doesn't do anything
with it.

We tried to setup remote-party-id in the Cisco and
setting trustripd=yes in sip.com but that did not
help. Also, I know we can send names to Asterisk from
the Cisco since I'm able to force the name on all
incoming calls from the PSTN -->router(config-sip-ua)#
calling-info pstn-to-sip from name set foo<-- The
callerid name foo is then displayed with the correct
callerid number for all incoming calls.

Debug mode in Asterisk shows that we are in fact
getting the remote-party-id header but no names in it.
Any ideas or suggestions?

Thanks,

Max








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Re: [Asterisk-Users] Polycom IP500 Quickstart page or files?

[Tom Hayden]

Huh? You can easily configure an IP500 via a web browser. Just point
the URL to the IP addr of the telephone.

--
Tom Hayden


On 9/22/05, Wilson Pickett <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I just got my ip500 back after months of waiting. Is there an easy way
> to get it hooked up to asterisk without [t]ftp servers and all that or
> is there a quickstart page somewhere?
>
> tia
>
> r
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--
Tom
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(solved) Re: [Asterisk-Users] ISDN-forwarding to intern without cost?

[Oliver Rath]

Joerg Lauer schrieb:


Hi,

I think I had the same problem and I think the error was that the dial 
statement had to be:


exten => s,1,Dial(Zap/1/,60,tT)

I may remeber wrong, though.

Btw: It may be a better idea to use Zap/g2/,60,tT), this way both 
B-Channels of the HFC card may be used.


Thanks! The problem was the ISDN-Phone, which explicitely wanted to get 
his msn. So the right entry was:



exten => 12345678,1,Dial(Zap/g2/12345678,60,tT)

TfH

Oliver


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Re: [Asterisk-Users] AgentRecord In and Out streams

[Mojo with Horan & Company, LLC]

make sure the program 'sox' is installed, the in and out streams get 
muxed automatically (not quite on the fly but after recording has stopped)


Crystal Stream, Incorporated wrote:

How do I combined these in and out wav files on the
fly through asterisk to where I hear the whole
conversation and only have one wav-file 
(i.e. :

agent-1001-asterisk-478-1127389080-17-in_out.wav)

agent-1001-asterisk-478-1127389080-17-in.wav
agent-1001-asterisk-478-1127389080-17-out.wav

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--
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(907) 747- x112
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Re: [Asterisk-Users] Polycom IP500 Quickstart page or files?

[Mojo with Horan & Company, LLC]

It's best for you to set up an ftp server instead of a tftp server, but 
I don't think you'll enjoy setting up a soundpoint phone without either 
of them.  The Polycom Phones page in the wiki was pretty much all I 
needed to set mine up: 
http://www.voip-info.org/tiki-index.php?page=Polycom+Phones


More specifically, the link  most of the way down:
http://www.krisk.org/asterisk/pcom/ was the starting point for my whole 
configuration.


Good luck!  Soundpoint phones, in my opinion, are worth every second 
spent setting them up.


Mojo


Wilson Pickett wrote:

Hi,

I just got my ip500 back after months of waiting. Is there an easy way
to get it hooked up to asterisk without [t]ftp servers and all that or
is there a quickstart page somewhere?

tia

r
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Re: [Asterisk-Users] externpass

[Tzafrir Cohen]

On Thu, Sep 22, 2005 at 02:25:02PM -0300, Sebastian Kühner wrote:
> Hello,
> 
> If I change the password in the Voicemail-Menu without externpass, the
> password gets changed correctly. But if I use a extern script, the password
> is still the same as before.
> 
> What do I have to do in my external script to change the password in the
> asterisk's memory, too?

Is http://bugs.digium.com/view.php?id=4602 relevant?

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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[Asterisk-Users] Windows Clients problem

[Mostafa]

Dear All,


I have tested more than 4 windows clients to asterisk 

I am testing them remotely through adsl at the client and adsl at the
sever.  

there is a big problem in the QOS from * to the client is OK while from
the client to * is very . I have investigated the problem using
ethereal . I found that the packets from the client to * is not tagged
DSCP value is 0 . I think this is a problem with the windows client TOS
is not working . As I have tested ping -v which is made to test the TOS
it also does not work.

Is there any client that has resolved this issue IAX or SIP ?

 

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RE: [Asterisk-Users] custom ring tone

[Marko Rakar]

similar service is when you call our local telco customer service, when
you dial it then you first hear "you have reached customer service,
plase wait" and while you hear that your call is still not connected and
therefore it is free

in addition you might have similar message to the caller that "operating
hours are from x till xx" so he can hear that instead of ring tone

I can think of many uses of that ring tone customisation



Two atoms bump into each other. One says 
"I think I lost an electron!" The other 
asks, "Are you sure?", to which the 
first replies, "I'm positive."

mailto:[EMAIL PROTECTED]
http://printel.hr 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Simpson
Sent: Thursday, September 22, 2005 7:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] custom ring tone


Yeah. It's a brilliant idea because I believe they would probably return

answer supervision to play these "custom ring tones" therefore creating 
more revenue from the incoming calls.


Marko Rakar wrote:
> Few weeks back local telco introduced option of custom ring tones. I 
> am not talking about your phone ring tones but about ring tones you 
> hear in your headset while phone is ringing on the other end.
> 
> If I understand correctly, ringing tone is generated localy on 
> asterisk if you are connected to phone network with E1/T1 connection. 
> Which means that instead of regular beep-beep tone we could send 
> something else to the caller in PSTN (like mp3 music).
> 
> Is there a way of customizing ring tone in asterisk and if yes how?
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RE: [Asterisk-Users] WaitExten

[Sherwood McGowan]

Try forcing the dtmf mode, such as 
Exten=EXTEN,1,SipDTMFMode(inband)

That worked for me, but you'll need to only do it on the SIP calls, so route
accordingly 

->-Original Message-
->From: [EMAIL PROTECTED] 
->[mailto:[EMAIL PROTECTED] On Behalf Of 
->Andrew Nowrot
->Sent: Thursday, September 22, 2005 1:50 PM
->To: asterisk-users@lists.digium.com
->Subject: [Asterisk-Users] WaitExten
->
->Hi,
->
->In my dialplan I'm using a WaitExten() command. It works only 
->with Zap phones. When I dial this command with Sip phone 
->asterisk do nothing.
->Should I put extra definition in sip.conf to make this work 
->with Sip phones?
->
->Thanks in advance
->
->Cheers
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->http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: Re: [Asterisk-Users] custom ring tone

[Marko Rakar]

yes, but I want this feature to be turned on for people who are calling
my asterisk from PSTN



Two atoms bump into each other. One says 
"I think I lost an electron!" The other 
asks, "Are you sure?", to which the 
first replies, "I'm positive."

mailto:[EMAIL PROTECTED]
http://printel.hr 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Domjan
Attila
Sent: Thursday, September 22, 2005 7:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: PS: Re: [Asterisk-Users] custom ring tone


Hi,
Dial application with m option, if the telco accept from you.

On Thu, 2005-09-22 at 19:10 +0200, Marko Rakar wrote:
> Few weeks back local telco introduced option of custom ring tones. I 
> am not talking about your phone ring tones but about ring tones you 
> hear in your headset while phone is ringing on the other end.
> 
> If I understand correctly, ringing tone is generated localy on 
> asterisk if you are connected to phone network with E1/T1 connection. 
> Which means that instead of regular beep-beep tone we could send 
> something else to the caller in PSTN (like mp3 music).
> 
> Is there a way of customizing ring tone in asterisk and if yes how?
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[Asterisk-Users] Polycom IP500 Quickstart page or files?

[Wilson Pickett]

Hi,

I just got my ip500 back after months of waiting. Is there an easy way
to get it hooked up to asterisk without [t]ftp servers and all that or
is there a quickstart page somewhere?

tia

r
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Re: [Asterisk-Users] SOHO Survey / Creative Asterisk Solutions

[Tom Vile]

try [EMAIL PROTECTED] it has some builtin features like the weather script
and stuff and if you are just playing around and have fun then this
would be the way to get some nice features and such.On 9/22/05, Brian McEntire <[EMAIL PROTECTED]
> wrote:Yes, and it is a fantastic resource! I don't think I could have gotten up and running without it.


I assume you pointed me there because there are some articles about
what people are doing with *. I'll dig for them. I previously read in
an issue of ;Login: several columns discussing things you could do with
Asterisk.

Articles are good, I was just hoping to get a couple of quick ideas
about what people on the list have accomplished. Things like "I set up
time and temperature for whenever I dial *22" ... or something like
that. Figured it might be a change for people to show off a bit and for
me to get some ideas/get inspired   ;-)

Thanks again for the voip-info ref, I'll work my way through that wiki.On 9/22/05, Damon Estep <
[EMAIL PROTECTED]
> wrote:












Have you discovered 

www.voip-info.org yet?

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of 
Brian McEntire
Sent: Thursday, September 22, 2005
6:18 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SOHO
Survey / Creative Asterisk Solutions



 

I hope the subject isn't too buzzword compliant  :)

I'm just curious: What have people done with Asterisk? I'm particularly
interested in DIY projects and things that can be done on a small/home office
(or even hobbiest's) budget. If you have clever hacks or creative functionality
you've implemented, I'd love to hear what a few people have come up with.

Thanks!  -Brian











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http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP TelephonyPhone: 518-631-2855 x205
Fax: 518-631-2856
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[Asterisk-Users] WaitExten

[Andrew Nowrot]

Hi,

In my dialplan I'm using a WaitExten() command. It works only with Zap
phones. When I dial this command with Sip phone asterisk do nothing.
Should I put extra definition in sip.conf to make this work with Sip
phones?

Thanks in advance

Cheers
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Re: [Asterisk-Users] custom ring tone

[Domjan Attila]

Hi,
Dial application with m option, if the telco accept from you.

On Thu, 2005-09-22 at 19:10 +0200, Marko Rakar wrote:
> Few weeks back local telco introduced option of custom ring tones. I am
> not talking about your phone ring tones but about ring tones you hear in
> your headset while phone is ringing on the other end.
> 
> If I understand correctly, ringing tone is generated localy on asterisk
> if you are connected to phone network with E1/T1 connection. Which means
> that instead of regular beep-beep tone we could send something else to
> the caller in PSTN (like mp3 music).
> 
> Is there a way of customizing ring tone in asterisk and if yes how?
> 
> 
> 
> Long, long ago in a galaxy far away, in General Hospital born I was, and
> quite happy were my parents, but when a youngling still I was, moved we
> did.
> 
> mailto:[EMAIL PROTECTED]
> http://printel.hr  
> ___
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> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Domjan Attila <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] custom ring tone

[Matthew Simpson]

Yeah. It's a brilliant idea because I believe they would probably return 
answer supervision to play these "custom ring tones" therefore creating 
more revenue from the incoming calls.



Marko Rakar wrote:

Few weeks back local telco introduced option of custom ring tones. I am
not talking about your phone ring tones but about ring tones you hear in
your headset while phone is ringing on the other end.

If I understand correctly, ringing tone is generated localy on asterisk
if you are connected to phone network with E1/T1 connection. Which means
that instead of regular beep-beep tone we could send something else to
the caller in PSTN (like mp3 music).

Is there a way of customizing ring tone in asterisk and if yes how?



Long, long ago in a galaxy far away, in General Hospital born I was, and
quite happy were my parents, but when a youngling still I was, moved we
did.

mailto:[EMAIL PROTECTED]
http://printel.hr  
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[Asterisk-Users] OT: Sangoma A102u available

[Christian M. Watts]

Please forgive the intrusion ...

We have a Sangoma A102u dual-span T1/E1/J1 card coming available as we are
upgrading our equipment. We have the original packaging, cables, manuals and
software that came with this card. It has only been used for 3 months 
in a data

center environment and is in perfect condition.

If interested, please email off-list and we can discuss options.

Thanks,
Christian
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RE: [Asterisk-Users] Cisco AS5XXX + CallerID Name

[Benjamin Lawetz]

 
I seem to recall this problem on the mailing list a couple of months ago,
I'd point you towards, but can't seem to find it.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Max Braz
Sent: September 21, 2005 9:29 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco AS5XXX + CallerID Name

Hi guys.

We have currently Asterisk CVS-v1-0-08/15/05-15:53:48 connected in SIP with
a Cisco AS5300 (IOS 12.3). One PRI is connected to the Cisco gateway. 

The problem we have is that on incoming PSTN calls to the AS5300, relayed in
SIP to Asterisk, the callerID name is not being transmitted. We received the
callerID number but no name. I know we are receiving the name from the PRI
in hexadecimal format (running in debug mode) but the AS53003 doesn't do
anything with it.

We tried to setup remote-party-id in the Cisco and setting trustripd=yes in
sip.com but that did not help. Also, I know we can send names to Asterisk
from the Cisco since I'm able to force the name on all incoming calls from
the PSTN -->router(config-sip-ua)# calling-info pstn-to-sip from name set
foo<-- The callerid name foo is then displayed with the correct callerid
number for all incoming calls.

Debug mode in Asterisk shows that we are in fact getting the remote-party-id
header but no names in it.
Any ideas or suggestions?

Thanks,

Max








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magasinage.yahoo.ca
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[Asterisk-Users] Re: Asterisk with iptel.org

[Sebastian Milioto]

With sip show registry I can see I my asterisk is registered:

Host: iptel.org:5060
Username:84565616
Refresh: 145
State: Registered

The following is part of sip.conf:

[general]

port = 5060   ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
allow=g729
allow=alaw
allow=ulaw
context = from-sip
callerid = Unknown


register => 84565616:[EMAIL PROTECTED]/200  ;iptel register

[200]
username=200
type=friend
secret=otro_password
record_out=On-Demand
record_in=On-Demand
qualify=no
port=5060
nat=never
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid="Sebastian" <200>

[iptel]
type=friend
username=84565616
secret=password_iptel
fromdomain=iptel.org
host=iptel.org


And the following is part of extensions.conf

[outbound-allroutes]
include => outbound-allroutes-custom
include => outrt-001-llamadasSIP

[outrt-001-llamadasSIP]
include => outrt-001-llamadasSIP-custom
;exten => _9.,1,Macro(dialout-trunk,2,${EXTEN:1},00323) before,
this line was uncommented and the following line doesn't existed. But
it doesn't work, so I changed it by the following line

exten => _9.,1,Macro(dialout-SIPiptel,2,${EXTEN:1},00323)
exten => _9.,2,Macro(outisbusy) ; No available circuits


[macro-dialout-SIPiptel]
exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r) ;To take the
SIP trunk I dial 9

exten => 200,1,Aswer
exten => 200,3,Hangup



; dialout using a trunk, using pattern matching (don't strip any prefix)
; arg1 = trunk number, arg2 = number, arg3 = route password
[macro-dialout-trunk]
exten => s,1,GotoIf($[foo${ARG3} = foo]?3:2))   ; arg3 is pattern password
exten => s,2,Authenticate(${ARG3})
exten => s,3,Macro(record-enable,${CALLERIDNUM},OUT)
exten => s,4,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?7)  ;check for
CID override for exten
exten => s,5,SetCallerID(${ECID${CALLERIDNUM}})
exten => s,6,Goto(9)
exten => s,7,GotoIf($[foo${OUTCID_${ARG1}} = foo]?9)  ;check for CID
override for trunk
exten => s,8,SetCallerID(${OUTCID_${ARG1}})
exten => s,9,SetGroup(OUT_${ARG1})
exten => s,10,CheckGroup(${OUTMAXCHANS_${ARG1}})
; if we've used up the max channels, continue at 109 (n+101)
exten => s,11,SetVar(DIAL_NUMBER=${ARG2})
exten => s,12,SetVar(DIAL_TRUNK=${ARG1})
exten => s,13,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the
proper dial string for this trunk
exten => s,14,SetVar(OUTNUM=${OUTPREFIX_${ARG1}}${DIAL_NUMBER})  ;
OUTNUM is the final dial number
exten => s,15,Cut(custom=OUT_${ARG1},:,1)  ; Custom trunks are
prefixed with "AMP:"
exten => s,16,GotoIf($[${custom} = AMP]?19)
exten => s,17,Dial(${OUT_${ARG1}}/${OUTNUM})  ; Regular Trunk Dial
exten => s,18,Goto(s-${DIALSTATUS},1)

; This is a custom trunk.  Substitute $OUTNUM$ with the actual number
and rebuild the dialstring
; example trunks: "AMP:CAPI/:b$OUTNUM$,30,r",
"AMP:OH323/[EMAIL PROTECTED]:"
exten => s,19,Cut(pre_num=OUT_${ARG1},$,1)
exten => s,20,Cut(the_num=OUT_${ARG1},$,2)  ; this is where we expect
to find string OUTNUM
exten => s,21,Cut(post_num=OUT_${ARG1},$,3)
exten => s,22,GotoIf($[${the_num} = OUTNUM]?23:24) ; if we didn't find
"OUTNUM", then skip to Dial
exten => s,23,SetVar(the_num=${OUTNUM}) ; replace "OUTNUM" with the
actual number to dial
exten => s,24,Dial(${pre_num:4}${the_num}${post_num})
exten => s,25,Goto(s-${DIALSTATUS},1)

exten => s,111,Noop(max channels used up)
exten => s-BUSY,1,NoOp(Trunk is reporting BUSY)
exten => s-BUSY,2,Busy()
exten => s-BUSY,3,Wait(60)
exten => s-BUSY,4,NoOp()

Hope you can help me. I'm afraid I make more trouble trying to fix it

Regards,

Sebastian
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Re: [Asterisk-Users] SOHO Survey / Creative Asterisk Solutions

[Brian McEntire]

Yes, and it is a fantastic resource! I don't think I could have gotten up and running without it.

I assume you pointed me there because there are some articles about
what people are doing with *. I'll dig for them. I previously read in
an issue of ;Login: several columns discussing things you could do with
Asterisk.

Articles are good, I was just hoping to get a couple of quick ideas
about what people on the list have accomplished. Things like "I set up
time and temperature for whenever I dial *22" ... or something like
that. Figured it might be a change for people to show off a bit and for
me to get some ideas/get inspired   ;-)

Thanks again for the voip-info ref, I'll work my way through that wiki.On 9/22/05, Damon Estep <[EMAIL PROTECTED]
> wrote:












Have you discovered 
www.voip-info.org yet?

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of 
Brian McEntire
Sent: Thursday, September 22, 2005
6:18 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SOHO
Survey / Creative Asterisk Solutions



 

I hope the subject isn't too buzzword compliant  :)

I'm just curious: What have people done with Asterisk? I'm particularly
interested in DIY projects and things that can be done on a small/home office
(or even hobbiest's) budget. If you have clever hacks or creative functionality
you've implemented, I'd love to hear what a few people have come up with.

Thanks!  -Brian










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[Asterisk-Users] Re: Re: Asterisk and a SPA3000behindNATpeerregistration

[Alchaemist]

Hi,

Lets see... dynamic IP, means mainly two options:

1- PPPoE in the same machine as asterisk
In that case, you can get the IP locallyin the shell
2- Whatever protocol, in a router
In that case you must rely in querying your router or an external system 
(like www.myipaddress.com)

Now, in the confs, you need in sip.conf a line like "#include 
sip_nat.conf"
Then, in your script you would need something like this php with 
wrappers enabled
And finally you need to setup the right cron with the script.

CASE 2 - External IP source
  Put the right file path in $ConfFile, and make this script run in your 
cron, often... the interval you setup will be the maximum "SIP trouble time" 
you will have.
  Just in case don't put it with an interval too low, because it might be 
something nasty to do to a website like this.
  Also, if you want to change it for your router webpage, then all you have 
to do, is to update the URL and the regex. If it has password, all you have 
to do is to use an URL like http://user:[EMAIL PROTECTED]/path/to/page
#-
#!/usr/bin/php -q
http://www.myipaddress.com');
  $Matches = array();
  preg_match('/.*IP address is\:\*<\/a>\s*(\d+\.\d+\.\d+\.\d+)<\/b>/i',$Webpage,$Matches);
  $IP = @ $Matches[1];
  if ($IP){
   echo "Your IP is: $IP\n";
   if ($FH = fopen($ConfFile,'w')){
fputs($FH,"externip=$IP\n");
fclose($FH);
$Reload = `asterisk -rx \"sip reload\"`;
echo "Reload Status: $Reload\n";
   }
   else{
echo "Unable to save file.\nSomething odd is going on, check permissions 
and paths.\n";
   }
  }
  else{
   echo "Unable to detect IP.\nPerhaps you are not connected, the website is 
down, or it changed its HTML.\n";
  }
#-



CASE 1 - Similar to case 2, except that you get the IP from a local command, 
I've seen an SH script that does this work, in this very list, somewhere.
Another difference is that instead of letting the cron regularly run it, you 
can/must run it every time the os detects an IP change/interface reload, 
init.d something, I cannot recall it now. Anyway case 2 will work as well.


Cheers!

Alchaemist

- Original Message - 
From: "razza" <[EMAIL PROTECTED]>
Newsgroups: gmane.comp.telephony.pbx.asterisk.user
Sent: Thursday, September 22, 2005 5:24 AM
Subject: RE: Re: Asterisk and a SPA3000behindNATpeerregistration


>
> Alchaemist Wrote:
>>Now... if you have dynamic IP in the asterisk... things change
> because
>>Asterisk must know in sip.conf the external IP.
>>I think I read in this list, that the best (only?) way to get arround,
> is to
>>place a script that detects the external IP when it changes,updates
> sip.conf
>>(preferably another conf like sip_nat.conf that is included into
> sip.conf),
>>and then uses the CLI to reload the sip configuration, you have a
> command I
>>cannot recall right now to only reload that part instead of the whole
> bunch
>>of confs.
>
> The problem is no one has come back with a script which does the job :o(
> The command to reload the sip.conf is -
> asterisk -rx "sip reload"



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[Asterisk-Users] externpass

[Sebastian Kühner]

Hello,

If I change the password in the Voicemail-Menu without externpass, the
password gets changed correctly. But if I use a extern script, the password
is still the same as before.

What do I have to do in my external script to change the password in the
asterisk's memory, too?

Thanks for your help!!

Sebastian

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[Asterisk-Users] cdr_custom?

[Sherwood McGowan]

I have a need to use 
cdr_custom and would like to know if anyone has gotten it to work with a mysql 
cdr backend, and any examples if possible
 
 
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Re: [Asterisk-Users] AGI Script to interact with ACCESS Databse and Set CID info on the fly.

[Tom Hayden]

If you used a perl or PHP agi script, you could probably use some kind
of ODBC drivers to communicate between the two.

--
Tom Hayden

On 9/22/05, Tim King <[EMAIL PROTECTED]> wrote:
>
>
>
> Well guys here comes the fun part. I have a Microsoft access (VBA)
> application that interfaces with my SQL database. This app pulls of info
> from the SQL record and than picks up the phone and dials that locations
> number. I have purchased a few hundred NpaNxx's for my own use. I want get
> into too much detail there but no worries this is legal. I need to change my
> CID info on the fly. So I am thinking it should be easy to make an AGI
> script that just sets the CID info on a particular line using two variables
> being passed to it $Line_No to tell it what line to set and than $CID to be
> the number to set on that extension for that call. It also should be
> relatively simple to have the access app take a look at the area code and
> phone number for the location being called and pull a phone number from the
> NUMBERS table which has all of my numbers in it and pass that over. The real
> question is how do we get Access to speak to an AGI script. Has anyone done
> anything like this? Thanks a lot for reading but this will be a fun one.
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>


--
Tom
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Re: [Asterisk-Users] AGI Script to interact with ACCESS Databse and Set CID info on the fly.

[pbx]

ACCESS supports ODBC driven connections..



> Well guys here comes the fun part. I have a Microsoft access (VBA)
> application that interfaces with my SQL database. This app pulls of info
> from the SQL record and than picks up the phone and dials that locations
> number. I have purchased a few hundred NpaNxx's for my own use. I want get
> into too much detail there but no worries this is legal. I need to change
> my

> question is how do we get Access to speak to an AGI script. Has anyone
> done
> anything like this? Thanks a lot for reading but this will be a fun one.
>


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[Asterisk-Users] custom ring tone

[Marko Rakar]

Few weeks back local telco introduced option of custom ring tones. I am
not talking about your phone ring tones but about ring tones you hear in
your headset while phone is ringing on the other end.

If I understand correctly, ringing tone is generated localy on asterisk
if you are connected to phone network with E1/T1 connection. Which means
that instead of regular beep-beep tone we could send something else to
the caller in PSTN (like mp3 music).

Is there a way of customizing ring tone in asterisk and if yes how?



Long, long ago in a galaxy far away, in General Hospital born I was, and
quite happy were my parents, but when a youngling still I was, moved we
did.

mailto:[EMAIL PROTECTED]
http://printel.hr  
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Re: [Asterisk-Users] AGI Script to interact with ACCESS Databse and Set CID info on the fly.

[Paul]

Tim King wrote:

Well guys here comes the fun part. I have a Microsoft access (VBA) 
application that interfaces with my SQL database. This app pulls of 
info from the SQL record and than picks up the phone and dials that 
locations number. I have purchased a few hundred NpaNxx’s for my own 
use. I want get into too much detail there but no worries this is 
legal. I need to change my CID info on the fly. So I am thinking it 
should be easy to make an AGI script that just sets the CID info on a 
particular line using two variables being passed to it $Line_No to 
tell it what line to set and than $CID to be the number to set on that 
extension for that call. It also should be relatively simple to have 
the access app take a look at the area code and phone number for the 
location being called and pull a phone number from the NUMBERS table 
which has all of my numbers in it and pass that over. The real 
question is how do we get Access to speak to an AGI script. Has anyone 
done anything like this? Thanks a lot for reading but this will be a 
fun one.


Use odbc to update a database on the same server the agi runs on. The 
agi script can get the latest data that way.


I did this before to export data from access to postgresql. It works.

I showed a customer how to do it. First he converted his vb programs so 
that all databases were on the postgre server. Then he wrote a php 
web-based app that replaced the vb app.


You can use something besides postgres if there is win odbc support for it.



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RE: [Asterisk-Users] AGI Script to interact with ACCESS Databse a nd Set CID info on the fly.

[Colin Anderson]

lol 
just posted this yesterday, it's for any ODBC DSN so Access or SQL or an Excel 
spreadsheet, as long as it's set as a DSN. This will work with outgoing Caller 
ID as well, it's just how you set it up in your dialplan. If you want I can 
email you the .agi since email will undoubtedly mangle the script. 
hth.
 
HOWTO: A simple AGI application to modify incoming CallerID on 
the fly using SQL Server (or any ODBC DSN)
 
Requirements: 
 
1. http://sourceforge.net/projects/odbcsock
 
2. The 
SQL server must be in the same subnet as your * server
 
Howto:
 
1. 
Install ODBCSocketServer on your SQL server and verify connection withthe 
included VB COM app from a Windows box.
 
2. 
Decide how you want to transform the Caller ID. In my case, I want to doa 
lookup of the CallerID number in SQL server and prepend the Caller ID witha 
"job number" which is a unique ID we assign to each 
customer.
 
3. In 
the case of a large database with lots of fields it's a good idea tocreate a 
view in SQL server that has *only* the records you want, then youcan filter 
from there. Note the view name.
 
4. 
Create an ODBC system DSN on the SQL server that points to your SQLserver 
DB
 
5. 
Modify the following PHP script to your taste. Ensure your PHP.ini in/etc 
has error and warning suppression ON or else the AGI will returninvalid 
characters:
 
#!/usr/bin/php -q
 
 class ODBCSocketServer 
{    var 
$sHostName; //name of the host to connect to  var $nPort; //port 
to connect to  var $sConnectionString; //connection string to 
use  
  //function to parse the SQL 
 
function ExecSQL($sSQL) {
 
   $fToOpen = fsockopen($this->sHostName, 
$this->nPort,&$errno, &$errstr, 30);   if 
(!$fToOpen)   {//contruct error 
string to return$sReturn = 
"version=\"1.0\"?>\r\nstate=\"failure\">\r\n$errstr\r\n\r\n";   }   else   {//construct 
XML to send//search and replace HTML chars in SQL 
first$sSQL = 
HTMLSpecialChars($sSQL);$sSend = 
"version=\"1.0\"?>\r\n\r\n$this->sConnectionString\r\n$sSQL\r\n\r\n";//write 
request   fputs($fToOpen, 
$sSend);//now read 
responsewhile 
(!feof($fToOpen)){ $sReturn 
= $sReturn .fgets($fToOpen, 
128);}fclose($fToOpen);   }   return 
$sReturn;  } 
}//class
 
 //Here is the code that uses this class.  First we create the 
class $oTest = new ODBCSocketServer;
 
 //Set the Hostname, port, and connection 
string  $oTest->sHostName = 
"192.168.1.17"; $oTest->nPort = 
9628; $oTest->sConnectionString = 
"DSN=intranet;UID=sa;PWD=12345;";
 
//It 
is bad practice to use the SA account; in sane installations you use 
anon-priviledge elevated user //now exec the SQL $sResult 
= $oTest->ExecSQL("SELECT * FROM AsteriskCallerID wherehomephonecd like 
'".$argv[1]."'"); 
 
//the 
$argv[x] variable array contains any arguments you pass to the script.The 
array is // $argv[0] the script itself, $argv[1] the first argument, 
$argv[2] thesecond, etc.
 
$p = 
xml_parser_create();xml_parse_into_struct($p, $sResult, $vals, 
$index);
 
//xml_parse_into_struct takes the returned XML and parses it out into 
avariable array
 
xml_parser_free($p); //clean up
 
//the 
print statement returns your SQL data to Asterisk using the SETVARIABLE 
statement. //$vals[x][value] returns the contents of the variable based on 
it'svariable index, you will//have to play with the index to determine 
which one has the data you wantto return
 
print 
"SET VARIABLE LANDMARKCID \"".$vals[4][value];print " 
".$vals[2][value].":\"";
 
?>
 
6. 
Rename this script to transformcallerid.agi and drop it 
into/var/lib/asterisk/agi-bin, & chmod 755 it.
 
7. 
Modify your dialplan with the AGI script called in a distinct context 
andthat context returns control to the original context when 
done.
 
[my-inbound-context-with-did]
 
'Assumption here is that you have several DID's in this context but 
shouldbe able to work without a DID
 
exten 
=> 3078,1,SetVar(CURRENTEXTEN=3078)'Set a variable with the 
currentextension being processedexten => 
3078,2,Goto(Transform-CallerID,s,1))exten => 
3078,3,DoRegualarDialplanStuffHere
 
[Transform-CallerID]
 
exten 
=> s,1,SetVar(CIDPREFIX=${CALLERIDNUM:3:3})exten => 
s,2,SetVar(CIDSUFFIX=${CALLERIDNUM:6:10})exten => 
s,3,agi(transformcallerid.agi|${CIDPREFIX}-${CIDSUFFIX}) 'We keepphone 
numbers in XXX- formatexten => s,4,NoOp(${LANDMARKCID}) 'Display the 
returned variable fordebugging purposesexten => 
s,5,Gotoif($["${LANDMARKCID}" = " :" ]?6:7) 'If the AGI didn't fonda match 
in the databaseexten => s,6,SetVar(LANDMARKCID=UNKNOWN:) 'Prepend the 
caller ID withUNKNOWN:exten => 
s,7,SetCallerID(${LANDMARKCID}${CALLERIDNUM:3:10}) 'Otherwiseprepend with 
the returned SQL dataexten => 
s,8,Goto(my-inbound-context-with-did,${CURRENTEXTEN},3) 'Returncontrol to 
calling context
 
 

  -Original Message-From: Tim King 
  [mailto:[EMAIL PROTECTED]Sent: Thursday, September 22, 2005 
  10:52 AMTo: asterisk-users@lists.dig

[Asterisk-Users] Re: POP3 and TTS (Festival?)

[Alchaemist]

Hi,

I plan, if I have time, to work on it this weekend, so in that case 
I will post the code, no problem at all.
I want to add reply with record functionality first.
Regards!
Alchaemist

"Michiel van Baak" <[EMAIL PROTECTED]> wrote in message 
news:[EMAIL PROTECTED]
> On 17:34, Wed 21 Sep 05, Alchaemist wrote:
>> Hi,
>>
>> Has anybody seen a non commercial, or freeware, or GPL, or 
>> even
>> CHEAP... POP/IMAP to Text-to-speech?
>>
>> I have a working version for POP3 using festival. It DOES
>> work... it even cleans the email contents to get the actual content. It
>> works great with Outlook emails and similar, and skips non
>> multipart/alternative (that would be mainly SPAM, where the email is just
>> html or multipart without the text/plain alternative).
>>
>> BUT, festival does really sound BAD when reading...
>>
>> AND... IMAP would be preferable, because you can mark emails 
>> as
>> read, instead I have to go over all the emails I have. So if I can get 
>> past
>> this without further coding... would be great.
>>
>> I tested with Cepstral.. it seems quite good... perhaps 
>> somebody
>> has another option, preferable as cheap as cepstral...
>>
>> In, short... any experiences with this?
>
> Hi,
>
> Care to share the code you already have for POP3 using
> festival ? If so I can help to port it to IMAP (I'm using
> IMAP here for mail).
>
> Greetz
>
> -- 
> Michiel van Baak
> http://michiel.vanbaak.info
> [EMAIL PROTECTED]
> GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D
>
> "Why is it drug addicts and computer afficionados are both called users?"
>
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> 



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[Asterisk-Users] AGI Script to interact with ACCESS Databse and Set CID info on the fly.

[Tim King]

Well guys here comes the fun part. I have a Microsoft access
(VBA) application that interfaces with my SQL database. This app pulls of info from
the SQL record and than picks up the phone and dials that locations number. I
have purchased a few hundred NpaNxx’s for my own use. I want get into too
much detail there but no worries this is legal. I need to change my CID info on
the fly. So I am thinking it should be easy to make an AGI script that just sets
the CID info on a particular line using two variables being passed to it
$Line_No to tell it what line to set and than $CID to be the number to set on
that extension for that call. It also should be relatively simple to have the
access app take a look at the area code and phone number for the location being
called and pull a phone number from the NUMBERS table which has all of my
numbers in it and pass that over. The real question is how do we get Access to
speak to an AGI script. Has anyone done anything like this? Thanks a lot for
reading but this will be a fun one. 






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[Asterisk-Users] Hardware Recommendations for Junghanns card QuadBRI PCI.

[Dpto . Técnico .]

Good afternoon,
 
I have to develop an Asterisk system in Spain that 
can support up to 8 simultaneos ISDN calls (4 BRI), initially only 6 simultaneos 
calls.
 
I have been looking into VOIP-INFO and I found 
that the Junghanns card QuadBRI is a very good option. But when I look for a 
server to develop the system, the information is a bit more 
confuse.
 
Any one have experience with this card? Which 
server do you will recommend?
 
Regards.
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RE: [Asterisk-Users] SOHO Survey / Creative Asterisk Solutions

[Damon Estep]

Have you discovered www.voip-info.org yet?

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian McEntire
Sent: Thursday, September 22, 2005
6:18 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SOHO
Survey / Creative Asterisk Solutions



 

I hope the subject isn't too buzzword compliant  :)

I'm just curious: What have people done with Asterisk? I'm particularly
interested in DIY projects and things that can be done on a small/home office
(or even hobbiest's) budget. If you have clever hacks or creative functionality
you've implemented, I'd love to hear what a few people have come up with.

Thanks!  -Brian








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