Re: [Asterisk-Users] Grandstream

[Rob Lith]

Menu - Display " -- rESEt --", please be very CAREFUL hereA> Key in the physical / MAC address on back of thephone, Press Menu, phone will be reset back toFACTORY DEFAULT setting, all your settingwill be erased and gone.
B> Press Menu without key in anything, phone willfunction the same as power cycle / rebootThen it will be back to adminRegardsRobOn 9/17/05, 
Jason Walker <[EMAIL PROTECTED]> wrote:
That's what I have used...works until you change it. ;)-Original Message-From: [EMAIL PROTECTED][mailto:
[EMAIL PROTECTED]] On Behalf Of Rene KluwenSent: Friday, September 16, 2005 4:51 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Grandstream
admin?-Original Message-From: [EMAIL PROTECTED][mailto:
[EMAIL PROTECTED]]On Behalf Of Joshua AbbottSent: zaterdag 17 september 2005 1:45To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Grandstream
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[Asterisk-Users] Odd SpanDSP behavior

[Chris]

    I've been working with the 
SpanDSP to get sending and receive working on the asterisk 
box.    I noticed in ztmonitor the RX was low so when I boost it 
with rxgain=15.0 (which puts it about midway on the scale) the Txfax works 
great.
 
    At that point the RXfax doesn't 
work at all and it fails during the training.   If I set rxgain=0.0 
I've been able to receive one fax (out of 24 attempts). 
It seems to work better on Rx if the gain is set to 0.
 
    Is there any further debuging I 
can try?
 
 
Chris
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[Asterisk-Users] How to make Basic authenticatuion in Asterisk server.

[shubhendu mishra]

Hi all,
 
I am using Asterisk server with my SIP phones.
I want to set Asterisk server in Basic authentication so that when SIP phone sends the register request to asterisk server , asterisk server will send Basic authentication challenge to SIP phones.Presently asterisk is sending only digest challenge to SIP phones.
 
Please tell me how can we set basic mode in Asterisk.what are the correct configuration for this problem.
 
Please suggest as early as possible.
 
Thanks and regard's 
Shubhendu Mohan
		 
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Re: [Asterisk-Users] TDM400P Dialing Out - "Cannot be completed as dialed"

[John Novack (port)]

MF Hulber wrote:

I can't see anything immediately wrong here.  Maybe it's something in 
your dialplan?  It sounds like Asterisk is dialing out but the PSTN 
doesn't like the number you are dialing.  Are you in an area that 
requires 10 digits or does not like if you dial 11 for local calls?


From what I can tell, the 805 area code requires 11 digits when dialing.

MARK.

Of course check your dialplan first, but if that fails, monitor the PSTN 
line with a buttset and/or digitgrabber, you probably will find that 
Asterisk is dialing without waiting for dialtone, so the first digit is 
not detected at the CO.
Some versions will allow a "w" to wait 1/2 second before starting to 
dial, bur it doesn't always work, and there seems to be little interest 
in detecting dialtone before dialing.
The NANP was SUPPOSED to require  7/10 digit local and 11 digit toll 
dialing, but since that is now a state by state determination, there is 
little "plan" left in the NANP


John Novack


Barry King wrote:


I've tried to google this issue with no resolution.

I'm having the same issue as this person:
http://lists.digium.com/pipermail/asterisk-users/2004-August/058280.html

Basically, anytime I try to dial out on my TDM400P w/ FXO, I get "we're
sorry, but your call cannot be completed as dialed."

When I "debug channel Zap/x-x", I get a whole bunch of this:  [ TYPE:
Null Frame (5) SUBCLASS: N/A (0) ] [Zap/2-1]

I recieve calls just fine.

I've tried dialing the Zap group and the specific channel, all with the
same results.

Any ideas?  zapata and zaptel configs follow.

zapata.conf:
[channels]
context=local-in
relaxdtmf=yes
language=en
signalling=fxs_ks
;rxwink=300
usecallerid=yes
useincomingcalleridonzaptransfer=yes
callerid=asreceived
group=1
immediate=yes
echocancel=64
echocancelwhenbridged=no
echotraining=800
rxgain=9.0
txgain=1.0
channel => 1-4

zaptel.conf:

fxsks=1-4
defaultzone=us
loadzone=us

Regards,
Barry King
King Computer Solutions
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Re: [Asterisk-Users] wav instead of gsm for vm-sounds?

[Kevin P. Fleming]

Damon Estep wrote:


Having a set of files with the .ul extension present on the system will
result in asterisk picking those files when the call is g.711u or zap.


There is no 'zap' channel format :-)

Zaptel channels can operate in G.711 u-law or A-law format, depending on 
their configuration.

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RE: [Asterisk-Users] wav instead of gsm for vm-sounds?

[Damon Estep]

> > Damon Estep wrote:
> >
> > > What I can not decipher is what file name extension should be
used,
> > > should it be a .wav file encoded at 8k/8b/mono? Or are you telling
> me
> > > that it should be .ulaw
> >
> > What I listed were file name extensions as recognized by Asterisk's
> > format modules and sox (and other tools).
> >
> > WAV at 8KHz 8-bit mono is _not_ the same thing as ulaw; WAV is not
> > compressed (normally), so that would be 8 bits of dynamic range.
ulaw
> > and alaw are 8-bit compressed formats, with 13/14 bits of dynamic
> range
> > (this is all documented on the wiki too, I think).
> 

For others that may find this archived in the future!

If using zaptel and g.711u you would want sound files saved in the ulaw
format, so record at a higher quality, maybe 16bit, 8khz, mono WAV and
the use sox to convert to ulaw

Sox does not recognize .ulaw as a format so use .ul - both asterisk and
sox recognize that extension

# sox sound.wav sound.ul

The quality, as Kevin mentioned, is far better than raw 8bit pcm.

Having a set of files with the .ul extension present on the system will
result in asterisk picking those files when the call is g.711u or zap.


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Re: [Asterisk-Users] TDM400P Dialing Out - "Cannot be completed as dialed"

[MF Hulber]

I can't see anything immediately wrong here.  Maybe it's something in 
your dialplan?  It sounds like Asterisk is dialing out but the PSTN 
doesn't like the number you are dialing.  Are you in an area that 
requires 10 digits or does not like if you dial 11 for local calls?


From what I can tell, the 805 area code requires 11 digits when dialing.

MARK.

Barry King wrote:


I've tried to google this issue with no resolution.

I'm having the same issue as this person:
http://lists.digium.com/pipermail/asterisk-users/2004-August/058280.html

Basically, anytime I try to dial out on my TDM400P w/ FXO, I get "we're
sorry, but your call cannot be completed as dialed."

When I "debug channel Zap/x-x", I get a whole bunch of this:  [ TYPE:
Null Frame (5) SUBCLASS: N/A (0) ] [Zap/2-1]

I recieve calls just fine.

I've tried dialing the Zap group and the specific channel, all with the
same results.

Any ideas?  zapata and zaptel configs follow.

zapata.conf:
[channels]
context=local-in
relaxdtmf=yes
language=en
signalling=fxs_ks
;rxwink=300
usecallerid=yes
useincomingcalleridonzaptransfer=yes
callerid=asreceived
group=1
immediate=yes
echocancel=64
echocancelwhenbridged=no
echotraining=800
rxgain=9.0
txgain=1.0
channel => 1-4

zaptel.conf:

fxsks=1-4
defaultzone=us
loadzone=us

Regards,
Barry King
King Computer Solutions
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RE: [Asterisk-Users] wav instead of gsm for vm-sounds?

[Damon Estep]

> Damon Estep wrote:
> 
> > What I can not decipher is what file name extension should be used,
> > should it be a .wav file encoded at 8k/8b/mono? Or are you telling
me
> > that it should be .ulaw
> 
> What I listed were file name extensions as recognized by Asterisk's
> format modules and sox (and other tools).
> 
> WAV at 8KHz 8-bit mono is _not_ the same thing as ulaw; WAV is not
> compressed (normally), so that would be 8 bits of dynamic range. ulaw
> and alaw are 8-bit compressed formats, with 13/14 bits of dynamic
range
> (this is all documented on the wiki too, I think).


Thanks Kevin!
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Re: [Asterisk-Users] wav instead of gsm for vm-sounds?

[Kevin P. Fleming]

Damon Estep wrote:


What I can not decipher is what file name extension should be used,
should it be a .wav file encoded at 8k/8b/mono? Or are you telling me
that it should be .ulaw


What I listed were file name extensions as recognized by Asterisk's 
format modules and sox (and other tools).


WAV at 8KHz 8-bit mono is _not_ the same thing as ulaw; WAV is not 
compressed (normally), so that would be 8 bits of dynamic range. ulaw 
and alaw are 8-bit compressed formats, with 13/14 bits of dynamic range 
(this is all documented on the wiki too, I think).

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RE: [Asterisk-Users] wav instead of gsm for vm-sounds?

[Damon Estep]

> 
> Damon Estep wrote:
> 
> > Is there a reference of codec to preferred format somewhere? What is
the
> > best format match for g.711u?
> 
> They are all pretty obvious:
> 
> G.711 mu-law: ulaw (or ul)
> G.711 A-law: alaw (or al)
> GSM: gsm
> G.729: g729
> Signed Linear (raw): sln (or raw)
> 
> Just look at the source files in the formats directory to see which
> extensions map to which formats.

Sorry for being so dense, but I am a little lost on this. I know that
both TDM (zaptel) and g.711u are 8khz, 8bit, mono, raw PCM, so it makes
sense that the sound file is best stored as a pcm file, 8khz, 8bit,
mono.

What I can not decipher is what file name extension should be used,
should it be a .wav file encoded at 8k/8b/mono? Or are you telling me
that it should be .ulaw

Patience please... I know I should get this, bit I don't!

Thanks a million.

Damon
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Re: [Asterisk-Users] wav instead of gsm for vm-sounds?

[Kevin P. Fleming]

Damon Estep wrote:


Is there a reference of codec to preferred format somewhere? What is the
best format match for g.711u?


They are all pretty obvious:

G.711 mu-law: ulaw (or ul)
G.711 A-law: alaw (or al)
GSM: gsm
G.729: g729
Signed Linear (raw): sln (or raw)

Just look at the source files in the formats directory to see which 
extensions map to which formats.

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RE: [Asterisk-Users] wav instead of gsm for vm-sounds?

[Damon Estep]

> Every attempt to play a file in Asterisk (that doesn't specify the
> extension) will use the 'best' format it can find for the call's
format.
> The applications that ship with Asterisk do not specify any particular
> format, so your best bet is to provide the files in all the formats
you
> expect your callers to be using, so Asterisk never has to transcode a
> file playback. The same thing is true for voicemail recording; you
> should record in every format your users might call in with, unless
the
> extra disk space consumption would be an issue.

Thanks Kevin,

Is there a reference of codec to preferred format somewhere? What is the
best format match for g.711u?
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Re: [Asterisk-Users] wav instead of gsm for vm-sounds?

[Kevin P. Fleming]

Damon Estep wrote:

Do you simply replace the .gsm files with .wav files and it plays them
in these apps, or is there more to it?

I am talking about the built in functionality of vm, queues, agents --
not the playback app.


Every attempt to play a file in Asterisk (that doesn't specify the 
extension) will use the 'best' format it can find for the call's format. 
The applications that ship with Asterisk do not specify any particular 
format, so your best bet is to provide the files in all the formats you 
expect your callers to be using, so Asterisk never has to transcode a 
file playback. The same thing is true for voicemail recording; you 
should record in every format your users might call in with, unless the 
extra disk space consumption would be an issue.

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[Asterisk-Users] Re: Sipura 2k voice quality

[Matthew Harrell]

Darn it.  I'll swear I had changed those but maybe I forgot to save that
setting.  That definitely helped with the volume problem.  Now I just need
to tweak the echo training some on asterisk since there is still somewhat
of an echo and a sligt time lag


: Raise both gains from -3 to 5 that solves volume problem,
: log in, click admin, advanced, I guess is on the sip tab,
: 
: Matthew Harrell wrote:
: 
: >When I have voip conversations over asterisk through my computer the voice
: >quality is nice and loud and quite clear.  When I go through my Sipura 2K
: >then the conversations are typically very muted and my responses sound 
: >somewhat
: >delayed.  I've tried fiddling with settings under asterisk and the 2K but 
: >haven't had any luck.  Can someone point me in the right direction to get
: >this configured right?
: >
: > 
: >
: 
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-- 
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  Bit Twiddlers, Inc.   Then again, so does my cat.
  [EMAIL PROTECTED] 
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Re: [Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?

[Steven Sokol]

On 9/16/05, Brian Roy <[EMAIL PROTECTED]> wrote:
> 
> 
>  
> On 9/16/05, Steven Sokol <[EMAIL PROTECTED]> wrote: 
> > Hi,
> > 
> > I'm taking a straw-poll to see who out there is planning on going to
> > AstriCon. 
>  
>   
> Enjoyed it last year, but putting it on the west coast seems to be pretty
> restrictive. I won't be making it. Atlanta was a good compromise. 
>   
> Maybe consider moving it to a more central location next year and I'll be
> back. 
>   

Well, I know it's a bit of a flight from coast-to-coast, but the
Californians, Oregonians, and Washingtonians (sp?) did it last year,
so we figured it would be good to give them a break.  We also hoped to
get some Asterisk users from Asia to make the hop.

We're thinking of a central location for 2006 -- Dallas and Denver
(two places that are central and fairly easy to get to by air) are
currently at the top of the list.  What central city sounds good to
you?

Thanks,

Steve



> -Brian 
> 
>   


-- 
Steven Sokol
CEO/Manager
Sokol & Associates, LLC

Ask Me About AstriCon 2005!
http://www.astricon.net/
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RE: [Asterisk-Users] wav instead of gsm for vm-sounds?

[Damon Estep]

Do you simply replace the .gsm files with .wav files and it plays them
in these apps, or is there more to it?

I am talking about the built in functionality of vm, queues, agents --
not the playback app.

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Joshua Abbott
> Sent: Friday, September 16, 2005 7:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] wav instead of gsm for vm-sounds?
> 
> Asterisk allows wav automatically
> 
> 
> Damon Estep wrote:
> 
> >Is there a way to get * to use wav files instead of gsm files for the
> >voicemail, agents, and queues applications?
> >
> >Gsm does not give all the quality we would like to have, and we use
no
> >low bit rate codecs.
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> >
> >
> 
> 
> --
> =
> Joshua Abbott, Support Technician
> http://www.successfulhosting.com/
> Direct Line: PENDING
> Phone: (866) 494-5096 x1207
> 
> E-Fax: (419) 858-3241
> Alt E-Fax: (801) 217-1123
> [EMAIL PROTECTED]
> =
> The Success behind your web site!
> =
> 
> CONFIDENTIAL:  This e-mail and any attachments are confidential and
may
> be privileged.  If you are not a named recipient, please notify the
> sender immediately and do not disclose the contents to another party,
> use it for any purpose, or store or copy the information in any
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Re: [Asterisk-Users] wav instead of gsm for vm-sounds?

[Joshua Abbott]

Asterisk allows wav automatically


Damon Estep wrote:


Is there a way to get * to use wav files instead of gsm files for the
voicemail, agents, and queues applications?

Gsm does not give all the quality we would like to have, and we use no
low bit rate codecs.
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=
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http://www.successfulhosting.com/
Direct Line: PENDING
Phone: (866) 494-5096 x1207

E-Fax: (419) 858-3241
Alt E-Fax: (801) 217-1123
[EMAIL PROTECTED]
=
The Success behind your web site!
=

CONFIDENTIAL:  This e-mail and any attachments are confidential and may
be privileged.  If you are not a named recipient, please notify the 
sender immediately and do not disclose the contents to another party, 
use it for any purpose, or store or copy the information in any medium.


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Re: [Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?

[Brian Roy]

On 9/16/05, Steven Sokol <[EMAIL PROTECTED]> wrote:
Hi,I'm taking a straw-poll to see who out there is planning on going toAstriCon. 

 
Enjoyed it last year, but putting it on the west coast seems to be pretty restrictive. I won't be making it. Atlanta was a good compromise.
 
Maybe consider moving it to a more central location next year and I'll be back.
 
-Brian
 
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[Asterisk-Users] wav instead of gsm for vm-sounds?

[Damon Estep]

Is there a way to get * to use wav files instead of gsm files for the
voicemail, agents, and queues applications?

Gsm does not give all the quality we would like to have, and we use no
low bit rate codecs.
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Re: [Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?

[Matthew Simpson]

I will be there.

Matthew Simpson
TxLink-Commpartners



Steven Sokol wrote:

Hi,

I'm taking a straw-poll to see who out there is planning on going to
AstriCon.  I would like to hear from both new members of the community
and gurus.  What kinds of things would you like to see at an Asterisk
Conference?  What topics are good BOF (Birds Of a Feather - informal
discussion group) fodder?  What parts of Asterisk require the most
attention?

FYI - AstriCon is October 12 - 14 in Anaheim.  For more information on
what we currently have planned, see the web site (listed below).

Thanks,

Steve

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RE: [Asterisk-Users] Grandstream

[Jason Walker]

That's what I have used...works until you change it. ;)

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rene Kluwen
Sent: Friday, September 16, 2005 4:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Grandstream

admin?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Joshua Abbott
Sent: zaterdag 17 september 2005 1:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Grandstream


Where do I find or what is the default password for a GrandStream BT 101 for
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[Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?

[Steven Sokol]

Hi,

I'm taking a straw-poll to see who out there is planning on going to
AstriCon.  I would like to hear from both new members of the community
and gurus.  What kinds of things would you like to see at an Asterisk
Conference?  What topics are good BOF (Birds Of a Feather - informal
discussion group) fodder?  What parts of Asterisk require the most
attention?

FYI - AstriCon is October 12 - 14 in Anaheim.  For more information on
what we currently have planned, see the web site (listed below).

Thanks,

Steve
-- 
Steven Sokol
Sokol & Associates/AstriCon

Ask Me About AstriCon 2005!
http://www.astricon.net/
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[Asterisk-Users] free IAX calling platform

[Matthew Simpson]

Hello all,

I have set up a free IAX calling platform similar to 
FreeWorldDialup/IAXtel.  You can access it at http://www.goiax.com/


The website is still very beta but it will allow you to sign up for a 
virtual phone number, and you can make outgoing calls to US toll-free 
numbers.  There is also a conference bridge set up. Codecs are G.711 and 
GSM.


I intend to allow users to choose united states DIDs for free in the 
future after I get the website polished up a little bit.


The backend is using Asterisk Realtime.

If anybody has any ideas for improvements let me know.

yours,
Matthew
TxLink
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[Asterisk-Users] Re: Double Ring

[Matt]

It almost seems like I'm getting an inbound ring from my PRI/IAX
terminator and asterisk is also generating a ring.   If I put an 'r'
in my dial statement I get only one ring.   But is there any issue to
be taken with putting an r in?  And apparently I have to put a
'timeout' value in?

On 9/16/05, Matt <[EMAIL PROTECTED]> wrote:
> Hi,
> It seems like my ATA is making a ringing noise... (as it used to), but
> now (After the upgrade from 1.0.7 to 1.2) asterisk also is either
> making the ringing, or passing the PRI ringing from the telco on to
> me.   Any suggestions on how to fix this?
>
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Fwd: [Asterisk-Users] Seperate Incoming calls on TDM02?

[C. Hatton Humphrey]

> I have 4 FXO ports, 2 on one number and 2 on another and want to have
> different incoming rules\IVR depending upon channel called.
> Is it as simple as changing the contexts in the zapata.conf or is there more
> to it.

Here is what my experience was.  Understand when reading it that I am
running [EMAIL PROTECTED] version 1.5.  I have 6 VoIP DIDs, two point to
one auto attendant (aa_1 in my dialplan as created by AAH), two point
to a second IVR (aa_2 in my dialplan as created by AAH) and the last
two point to a direct extension.  The "From PSTN" rules point all
incoming calls to aa_1.

AAH includes a "context=" in the main zapata.conf file.  The rules for
zapata are odd, basically you define a set of parameters and load it
into a channel.  If you don't redefine them and create another channel
the parameters already set are included in the second channel.
Therefore you can set the main setting parameters once and then change
things as needed.

The other thing that gets created by AAH is a file called
zapata-auto.conf; this file is created by the "genzaptel" script and
automatically sets your Zaptel channels up.  If you open this script
you'll see each of your incoming ports with a parameter setting of
"context=from_pstn".  What I did (against the suggestion of the file)
was change the context in the zapata-auto.conf file to point to the
correct IVR.

The point that I missed originally that I figured out was that the
reload command in asterisk does NOT redefine the channels.  You have
to execute a "restart" command, either "restart now" or "restart when
convenient".  That will reread the zapata.conf file and redefine the
zaptel channels.

Hope this helps,
Hatton
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[Asterisk-Users] Double Ring

[Matt]

Hi,
It seems like my ATA is making a ringing noise... (as it used to), but
now (After the upgrade from 1.0.7 to 1.2) asterisk also is either
making the ringing, or passing the PRI ringing from the telco on to
me.   Any suggestions on how to fix this?
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Re: [Asterisk-Users] Re: g729 to asterisk to g729 voip provider

[Enzo Michelangeli]

- Original Message - 
From: "Christian B" <[EMAIL PROTECTED]>
To: 
Sent: Saturday, September 17, 2005 5:44 AM
Subject: Re: [Asterisk-Users] Re: g729 to asterisk to g729 voip provider

> On Fri, 16 Sep 2005 16:09:37 -0500
> Erick Perez <[EMAIL PROTECTED]> wrote:
>
> > Hi, your project is indeed interesting, however for learning purposes
> > i do need to know the answer of at least:
>
> it is not my project.
>
> > 1- Using sipura sip/g729 to connect to an asterisk server that will
> > server as a gateway to a VOIP provider(g729), all in g729 will require
> > to purchase codecs from Digium?
>
> read the page, it provides you with a free version of the g729

Erick's question is better answered here:

http://www.voip-info.org/wiki-Asterisk+G.729+pass-thru

Enzo

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RE: [Asterisk-Users] Grandstream

[Rene Kluwen]

admin?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Joshua
Abbott
Sent: zaterdag 17 september 2005 1:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Grandstream


Where do I find or what is the default password for a GrandStream BT 101 
for the web interface
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[Asterisk-Users] Grandstream

[Joshua Abbott]

Where do I find or what is the default password for a GrandStream BT 101 
for the web interface

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Re: [Asterisk-Users] How to IGNORE distinctive ring

[Steven Premeau]

The way I accomplished this is to leave my default context empty, then 
define the distinctive ring in Asterisk, but send it to the empty 
default context.


Asterisk will generate a warning that it doesn't know what to do, but it 
will also do nothing with the call.You can just do this with an 
empty non-default context , because * will fall back to default when it 
hits a completely empty context.


You will need to ensure that your real incoming line go to a different 
context, but otherwise it should be pretty straightforward.


(I don't have the configuration that I used handy, but can dig for it if 
my description isn't making sense.)


Steve.

Brad Jacobs wrote:


PSI System Admin-Message-ID: <[EMAIL PROTECTED]>

I'll give that a try in the meantime. If you find the right way to do it let
me know or if I find something I'll get back to you!

Thanks
Brad

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shawn Porter
Sent: Tuesday, September 13, 2005 19:02
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] How to IGNORE distinctive ring

Brad,

 I posted a similar question on voipuser, no response yet, but I ended up
making a separate extension Its not perfect, but it does technically ignore
the call.

[Home]
exten => s,1,Wait(30)
exten => s, 2, Hangup


Shawn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brad Jacobs
Sent: Tuesday, September 13, 2005 5:53 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] How to IGNORE distinctive ring

PSI System Admin-Message-ID: <[EMAIL PROTECTED]>

Hi list members,

I'm sure this question has been posted before but I am still unable to find
the answer. I have a TDM 400P line card and I would like to set it up to
IGNORE the distinctive ring pattern that I have for a fax machine.

Many thanks
Brad


 



--
Steven Premeau [EMAIL PROTECTED]
-
Gebt mir endlich einen Computer, der das tut, was ich will 
   (und nicht was ich sage...)!


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[Asterisk-Users] TDM400P Dialing Out - "Cannot be completed as dialed"

[Barry King]

I've tried to google this issue with no resolution.

I'm having the same issue as this person:
http://lists.digium.com/pipermail/asterisk-users/2004-August/058280.html

Basically, anytime I try to dial out on my TDM400P w/ FXO, I get "we're
sorry, but your call cannot be completed as dialed."

When I "debug channel Zap/x-x", I get a whole bunch of this:  [ TYPE:
Null Frame (5) SUBCLASS: N/A (0) ] [Zap/2-1]

I recieve calls just fine.

I've tried dialing the Zap group and the specific channel, all with the
same results.

Any ideas?  zapata and zaptel configs follow.

zapata.conf:
[channels]
context=local-in
relaxdtmf=yes
language=en
signalling=fxs_ks
;rxwink=300
usecallerid=yes
useincomingcalleridonzaptransfer=yes
callerid=asreceived
group=1
immediate=yes
echocancel=64
echocancelwhenbridged=no
echotraining=800
rxgain=9.0
txgain=1.0
channel => 1-4

zaptel.conf:

fxsks=1-4
defaultzone=us
loadzone=us

Regards,
Barry King
King Computer Solutions
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Re: [Asterisk-Users] mpg123 on x86_64 (Opteron MP)

[Paul]

Joseph wrote:


Why do you need to compile it?
Isn't it available as an rpm package?

 


I will assume he knows why he needs to compile it.

See if the source for the rpm, deb, or whatever from the distro you are 
running will build for you. That will often get your system to the point 
where any header, tools and libraries needed are now installed. That's 
how I do it when I want to use something from cvs.


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Re: [Asterisk-Users] How to access * thru router when ip address is not known

[Paul]

I came up with a solution a few years ago that only required a web 
browser to get the current IP address.


It requires an account with ssh access allowed on a web host.

1) Install your public ssh key on the account where the web pages live.

2) If you are concerned about others getting your IP address create a 
password protected web directory where dynamic IP host data will be stored.


3) create a script the is executed on the appropriate ip up condition. 
On debian this script would be installed in /etc/ppp/ip-up.d directory 
if you wanted it to run whenever a ppp link comes up. The script takes 
the output of ifconfig and uses scp to update the web host. Just browse 
and refresh to get the current IP address.


This provides the new address right away.  No need to wait for dyndns or 
email propagation.


I should mention that I also have cron run the script every 20 minutes. 
I did this after I had a situation where the dsl at the web host was 
down for an hour or so. During that time  2 remotes changed address but 
were unable to update my web host. Now I will get the new address within 
20 minutes of recovering from a network outage.


Colin Anderson wrote:


Assuming your Asterisk mail client is set up correctly, cron this twice a
day on your Asterisk box:

ifconfig eth0 | grep "inet addr" | mail -s "My Asterisk IP address"
[EMAIL PROTECTED]

Check your mail before you call and verify that the IP address has not
changed, if it has, modify your client accordingly. 


-Original Message-
From: Robert Rawlinson [mailto:[EMAIL PROTECTED]
Sent: Friday, September 16, 2005 1:52 PM
To: Asterisk
Subject: [Asterisk-Users] How to access * thru router when ip address is
not known


I am going to be traveling and I wanted to be able to get on the 
internet and call thru * to make calls. The problem is I do not have a 
fixed ip address. How do you make this work? I will be using IAXCOMM. TIA

Bob
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RE: [Asterisk-Users] How to access * thru router when ip address is not known

[Colin Anderson]

Another method comes to mind similar to my last posted method, but for
*behind* a firewall if you have a static hostname but dynamic ip (lots of
ISP's do this):

traceroute -m1 my.statichostname.net | mail -s "My Asterisk IP address"
[EMAIL PROTECTED]

Yet another method is,  a lot of Linksys / SMC bluebox routers also display
the external IP in their little management web site that they have. You
could use this in a shell script as follows:

getmyexternalip.sh

rm -f /etc/myexternalip.txt
wget --cache=off --http-user=admin --http-passwd=admin
--output-document=/etc/myexternalip.txt http://192.168.1.1/status.html
grep "WAN IP Address" /etc/myexternalip.txt | mail -s "My Asterisk IP
address" [EMAIL PROTECTED]

Otherwise, DyDNS.org is the way to go. 

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RE: [Asterisk-Users] Seperate Incoming calls on TDM02?

[Matt Love]

Hi Hatton,

Could you provide some examples of the config files for this. Im trying to
do the same. Im confused with some of the other posts (its not hard to
confuse me!) Some say its just the zapata and some say theres way more to
it.
I have 4 FXO ports, 2 on one number and 2 on another and want to have
different incoming rules\IVR depending upon channel called.
Is it as simple as changing the contexts in the zapata.conf or is there more
to it.

Thanks 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C. Hatton
Humphrey
Sent: 16 September 2005 15:17
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Seperate Incoming calls on TDM02?

> > Yeah, in your zapata.conf just give each channel a different context
> > setting.
> 
> It's slightly more complicated with [EMAIL PROTECTED] and/or AMP, you need to 
> use the
> zapata_custom.conf file, instead. You also need to use the
> extensions_custom.conf file, too, though there might be a better way I
don't
> know about.

I'm fighting with this right now and I'm hitting a serious frustration
point - right now all incoming calls are getting handled by the
from_pstn context which is how it honestly should be according to the
current conf files.  However when I change the context from from_pstn
to aa_1 and aa_2 respectively it doesn't change anything in the way
the system is answering the lines.

Hatton

zapata.conf:
;
; Zapata telephony interface
;
; Configuration file

[trunkgroups]

[channels]

language=en
context=from-pstn
signalling=fxs_ks
rxwink=300  ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes

usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=no
threewaycalling=no
transfer=no
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no

;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

;Include AMP configs
#include zapata_additional.conf

;Include genzaptelconf configs
#include zapata-auto.conf

zapata_additional.conf is empty
zapata-auto.conf:
; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is intended 
; to be #include-d by /etc/zapata.conf that will include the global settings
;
callerid=asreceived

; Span 1: WCTDM/0 "Wildcard TDM400P REV E/F Board 1" 
; channel 1, WCTDM, inactive.
; channel 2, WCTDM, inactive.
signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 3
context=from_pstn
group=0
channel => 3

signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4
context=from_pstn
group=0
channel => 4
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Re: [Asterisk-Users] mpg123 on x86_64 (Opteron MP)

[Joseph]

Why do you need to compile it?
Isn't it available as an rpm package?

-- 
#Joseph

On Fri, 2005-09-16 at 08:56 -0700, [EMAIL PROTECTED] wrote:
> Has anyone successfully compiled mpg123 in the 1.0.x or 1.2beta1  
> distributions (I'm running FC3 linux on an Opteron 2 processor  
> system)?  Are there any patches out there to make it work?
> 
> gcc -DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX  - 
> DREAD_MMAP -DOSS -DTERM_CONTROL-Wall -O2 - 
> m486 -fomit-frame-pointer -funroll-all- 
> loops  -finline-functions -ffast-math   -c -o  
> dct64_i386.o dct64_i386.c
> as   -o decode_i586.o decode_i586.s
> decode_i586.s: Assembler messages:
> decode_i586.s:44: Error: suffix or operands invalid for `push'
> 
> decode_i586.s:161: Error: suffix or operands invalid for `pop'
> 


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[Asterisk-Users] linux sip or iax phone that will autoanswer and route to console

[Jerry Geis]

Is there a linux sip or iax phone that will autoanswer
and connect to the
console or soundcard?

I found linphonec but it does not autoanswer from what I can tell.

Jerry




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[Asterisk-Users] Orinoco Injectors

[Darren Wright]

Has anyone gotten the Lucent / Orinoco injectors (AE-1, AE-6, AE-12) to
work with the Cisco 79* series phones?

I'm not sure if the are the statndard POE or not

-Darren

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RE: [Asterisk-Users] How to access * thru router when ip address is not known

[razza]

Colin Anderson Wrote:
>Assuming your Asterisk mail client is set up correctly, cron this twice
a day on your Asterisk box:
>ifconfig eth0 | grep "inet addr" | mail -s "My Asterisk IP address"
[EMAIL PROTECTED]
>Check your mail before you call and verify that the IP address has not
changed, if it has, modify your client accordingly. 
%<- SNIP! ->%

A more elegant method, which also works behind a firewall (if using
DDNS) would be the solution discussed in the 'NAT and SIP.conf update'
thread, whereby Wilson Pickett wrote:

>What I did was to have the ip checker write a one line file called
"externip.conf" containing the line: 
>externip = nnn.nnn.nnn.nnn ; this is the new ip address then in
sip.conf, 
>#include "externip.conf" ; replace the externip= line in sip.conf with
this include

>Now in your ip checker script (run every n minutes with crontab),check
the ip against an the last saved ip (in file >"old_ip" for example) if
ips are the same, exit, nothing to do if different, write the
externip.conf file and 
> do a "sip reload" command
> save  the ip to the old_ip file
>Exit

I have requested he send me/post his checker, as I am sure there are
plenty who would find this useful and if like me do not posses the
necessary skills to craft this myself.




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Re: [Asterisk-Users] Re: g729 to asterisk to g729 voip provider

[Christian B]

On Fri, 16 Sep 2005 16:09:37 -0500
Erick Perez <[EMAIL PROTECTED]> wrote:

> Hi, your project is indeed interesting, however for learning purposes
> i do need to know the answer of at least:

it is not my project.

> 
> 1- Using sipura sip/g729 to connect to an asterisk server that will
> server as a gateway to a VOIP provider(g729), all in g729 will require to
> purchase codecs from Digium?

read the page, it provides you with a free version of the g729

> 
> 2- also, in this scenario the transcoding is almost non-existent right?

since all codecs are the same, nothing has to be transcoded of course.

> 
> 3- I have read many documents about the type of codecs, and g729 seems to
> be a good trade between almost-toll quality and low bandwith usage
> right?

yes, it offers a good trade between both. however, voice quality is not superb 
but satisfying.

regards
christian


> On 9/16/05, ChB <[EMAIL PROTECTED]> wrote:
> > Hello Erik!
> > 
> > check out this website: 
> > http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
> > i have both codecs running on gentoo without problems(working with cisco 
> > 7960 and snom 190). regarding your hardware question i don't have enough 
> > experience yet, sorry.
> > 
> > regards
> > christian
> > 
> > 
> > On Fri, 16 Sep 2005 15:01:32 -0500
> > Erick Perez <[EMAIL PROTECTED]> wrote:
> > 
> > > anyone with some info on this?
> > >
> > > thanks again.
> > >
> > > On 9/14/05, Erick Perez <[EMAIL PROTECTED]> wrote:
> > > > Using sipura sip/g729 to connect to an asterisk server that will
> > > > server as a gateway to a VOIP provider, all in g729 will require to
> > > > purchase codecs from Digium?
> > > >
> > > > also, in this scenario the transcoding is almost non-existent right?
> > > > I have read many documents about the type of codecs, and g729 seems to
> > > > be a good trade between almost-toll quality and low bandwith usage
> > > > right?
> > > >
> > > >
> > > > A P4 (HT) 3.2 Ghz with 1gb ram in a 800mhz board with 3 raid0 disk can
> > > > sustain more than 100 calls or up to a 100?
> > > > I just looking at hardware capacity, since the machine will be located
> > > > at an ISP with more than needed bandwith.
> > > >
> > > > There is no need for voicemail, web interfaces or anything else, since
> > > > the * box will only function as a gateway to a US-based VOIP provider.
> > > >
> > > > The machine in question runs Centos4 Linux (Redhat enterprise 4) and
> > > > CDR logging only.
> > > >
> > > > Thanks,
> > > >
> > >
> > >
> > > --
> > >
> > > ---
> > > Erick Perez
> > > Linux User 376588
> > > http://counter.li.org/  (Get counted!!!)
> > > Panama, Republic of Panama
> > > ___
> > > --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
> > >
> > 
> 
> 
> -- 
> 
> ---
> Erick Perez
> Linux User 376588
> http://counter.li.org/  (Get counted!!!)
> Panama, Republic of Panama
> 
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Re: [Asterisk-Users] Re: g729 to asterisk to g729 voip provider

[Erick Perez]

Hi, your project is indeed interesting, however for learning purposes
i do need to know the answer of at least:

1- Using sipura sip/g729 to connect to an asterisk server that will
server as a gateway to a VOIP provider(g729), all in g729 will require to
purchase codecs from Digium?

2- also, in this scenario the transcoding is almost non-existent right?

3- I have read many documents about the type of codecs, and g729 seems to
be a good trade between almost-toll quality and low bandwith usage
right?

On 9/16/05, ChB <[EMAIL PROTECTED]> wrote:
> Hello Erik!
> 
> check out this website: 
> http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
> i have both codecs running on gentoo without problems(working with cisco 7960 
> and snom 190). regarding your hardware question i don't have enough 
> experience yet, sorry.
> 
> regards
> christian
> 
> 
> On Fri, 16 Sep 2005 15:01:32 -0500
> Erick Perez <[EMAIL PROTECTED]> wrote:
> 
> > anyone with some info on this?
> >
> > thanks again.
> >
> > On 9/14/05, Erick Perez <[EMAIL PROTECTED]> wrote:
> > > Using sipura sip/g729 to connect to an asterisk server that will
> > > server as a gateway to a VOIP provider, all in g729 will require to
> > > purchase codecs from Digium?
> > >
> > > also, in this scenario the transcoding is almost non-existent right?
> > > I have read many documents about the type of codecs, and g729 seems to
> > > be a good trade between almost-toll quality and low bandwith usage
> > > right?
> > >
> > >
> > > A P4 (HT) 3.2 Ghz with 1gb ram in a 800mhz board with 3 raid0 disk can
> > > sustain more than 100 calls or up to a 100?
> > > I just looking at hardware capacity, since the machine will be located
> > > at an ISP with more than needed bandwith.
> > >
> > > There is no need for voicemail, web interfaces or anything else, since
> > > the * box will only function as a gateway to a US-based VOIP provider.
> > >
> > > The machine in question runs Centos4 Linux (Redhat enterprise 4) and
> > > CDR logging only.
> > >
> > > Thanks,
> > >
> >
> >
> > --
> >
> > ---
> > Erick Perez
> > Linux User 376588
> > http://counter.li.org/  (Get counted!!!)
> > Panama, Republic of Panama
> > ___
> > --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
> >
> 


-- 

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!)
Panama, Republic of Panama
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[Asterisk-Users] Anyone using iPlan Networks in Argentina?

[Ilan Rabinovitch]

Hello,

Is anyone successfully working with iPlan Networks in Argentina for
telephony service?
I'm interested in hearing about people's experience with their service
and support.

Regards,
Ilan
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RE: [Asterisk-Users] How to access * thru router when ip address is not known

[Colin Anderson]

whoops, should have prefaced that with the Asterisk box has to be forward of
any firewall, otherwise it's going to return a 10.X.X.X or 192.168.X.X IP
address. 

-Original Message-
From: Colin Anderson [mailto:[EMAIL PROTECTED]
Sent: Friday, September 16, 2005 2:28 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] How to access * thru router when ip
address is not known


Assuming your Asterisk mail client is set up correctly, cron this twice a
day on your Asterisk box:

ifconfig eth0 | grep "inet addr" | mail -s "My Asterisk IP address"
[EMAIL PROTECTED]

Check your mail before you call and verify that the IP address has not
changed, if it has, modify your client accordingly. 

-Original Message-
From: Robert Rawlinson [mailto:[EMAIL PROTECTED]
Sent: Friday, September 16, 2005 1:52 PM
To: Asterisk
Subject: [Asterisk-Users] How to access * thru router when ip address is
not known


I am going to be traveling and I wanted to be able to get on the 
internet and call thru * to make calls. The problem is I do not have a 
fixed ip address. How do you make this work? I will be using IAXCOMM. TIA
Bob
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Re: [Asterisk-Users] USB Phones for use with Asterisk

[Derek Whitten]

There is a standalone linux version of xlite available on their
homepage.. http://www.xten.com


On Fri, 2005-09-16 at 11:38, Guillermo Salas M wrote:
> On Fri, 2005-09-16 at 11:54 +0100, c waddy wrote:
> > Are there drivers available for Xten Softphones?
> >  
> > Do they work with X-lite/Eyebeam Softphones?
> >  
> 
> And work on Linux (kernel 2.6)?
> 
> >  
> > 
> > 
> >  
> > On 9/16/05, Bill McCready (PCPhoneline.com)
> > <[EMAIL PROTECTED]> wrote: 
> > > Hi all,
> > >
> > > I have a question that I was hoping someone could answer for
> > me.
> > >
> > > I would like to find a USB phone that works with Asterisk...
> > preferably 
> > > one that does not have any issues with NAT.
> > 
> > RESPONSE:  This is Bill McCready from PCPhoneline.com .   I
> > suggest you give
> > our VTA1000 Skype+SIP+IAX2 Gateway which is USB based with a
> > good cordless
> > phone. 
> > 
> > We offer several versions of the VTA1000 but this is the
> > latest one we have
> > developed and it supports Skype + 10 x SIP + 10 x IAX2
> > networks all at the
> > same time and plugs into the USB port of a Windows 2000 or
> > Windows XP 
> > computer.   You can optionally disable Skype and/or the SIP
> > calling
> > capabilities if you don't need them.   The VTA1000 is designed
> > to work even
> > if your computer screen is powered off using it with a
> > cordless phone of 
> > course gives you the added advantage of not needing to be tied
> > to your
> > computer to place and receive calls.   If you don't mind being
> > tethered to a
> > computer we also have the VPT1000 phone available with the
> > same capability 
> > set.
> > 
> > You can find them in our Ebay store at
> > http://stores.ebay.com/pcphoneline .
> > If you need a bunch of these and want a build specially
> > tailored and branded
> > please private e-mail me with your requirements and we will do
> > our best to
> > try to accomodate you.
> > 
> > Best regards...Bill
> > 
> > 
> > >
> > > Can anyone point me to something suitable ?  We are
> > essentially planning 
> > > to do a beta for a few hundred customers on our network and
> > would like
> > > them to be able to call one another @ no charge.
> > >
> > > Because these are going to be end users who have no idea
> > about Voice / 
> > > VOIP stuff... it needs to be extremely easy to use.
> > >
> > > Can someone point me in the right direction ?
> > >
> > > Cheers,
> > >
> > > Callum
> > > ___ 
> > > --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
> > >
> > >
> > 
> > 
> > ___
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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
> > 
> > ___
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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
-- 
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PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+> h r+++ y 
 --END GEEK CODE BLOCK--


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RE: [Asterisk-Users] How to access * thru router when ip address is not known

[Colin Anderson]

Assuming your Asterisk mail client is set up correctly, cron this twice a
day on your Asterisk box:

ifconfig eth0 | grep "inet addr" | mail -s "My Asterisk IP address"
[EMAIL PROTECTED]

Check your mail before you call and verify that the IP address has not
changed, if it has, modify your client accordingly. 

-Original Message-
From: Robert Rawlinson [mailto:[EMAIL PROTECTED]
Sent: Friday, September 16, 2005 1:52 PM
To: Asterisk
Subject: [Asterisk-Users] How to access * thru router when ip address is
not known


I am going to be traveling and I wanted to be able to get on the 
internet and call thru * to make calls. The problem is I do not have a 
fixed ip address. How do you make this work? I will be using IAXCOMM. TIA
Bob
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Re: [Asterisk-Users] wiki down?

[ChB]

voip-info is down from time to time, guess more spending for their server 
hardware is needed.

On Fri, 16 Sep 2005 19:26:18 +0200 (CEST)
"Francesco Peeters" <[EMAIL PROTECTED]> wrote:

> I'm unable to connect to voip-info.org... Anybody else have the same
> issues, ro is it just me?
> 
> -- 
> Francesco Peeters
> 
> GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
> If your program doesn't recognize my signature, please visit
> http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
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Re: [Asterisk-Users] Re: g729 to asterisk to g729 voip provider

[ChB]

Hello Erik!

check out this website: 
http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
i have both codecs running on gentoo without problems(working with cisco 7960 
and snom 190). regarding your hardware question i don't have enough experience 
yet, sorry.

regards
christian


On Fri, 16 Sep 2005 15:01:32 -0500
Erick Perez <[EMAIL PROTECTED]> wrote:

> anyone with some info on this?
> 
> thanks again.
> 
> On 9/14/05, Erick Perez <[EMAIL PROTECTED]> wrote:
> > Using sipura sip/g729 to connect to an asterisk server that will
> > server as a gateway to a VOIP provider, all in g729 will require to
> > purchase codecs from Digium?
> > 
> > also, in this scenario the transcoding is almost non-existent right?
> > I have read many documents about the type of codecs, and g729 seems to
> > be a good trade between almost-toll quality and low bandwith usage
> > right?
> > 
> > 
> > A P4 (HT) 3.2 Ghz with 1gb ram in a 800mhz board with 3 raid0 disk can
> > sustain more than 100 calls or up to a 100?
> > I just looking at hardware capacity, since the machine will be located
> > at an ISP with more than needed bandwith.
> > 
> > There is no need for voicemail, web interfaces or anything else, since
> > the * box will only function as a gateway to a US-based VOIP provider.
> > 
> > The machine in question runs Centos4 Linux (Redhat enterprise 4) and
> > CDR logging only.
> > 
> > Thanks,
> > 
> 
> 
> -- 
> 
> ---
> Erick Perez
> Linux User 376588
> http://counter.li.org/  (Get counted!!!)
> Panama, Republic of Panama
> ___
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] wiki down?

[Francesco Peeters]

On Fri, September 16, 2005 19:53, Wiley Siler said:
> I got right in just fine...
>
> W
>
>
Me too now.  :-/

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
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[Asterisk-Users] Re: g729 to asterisk to g729 voip provider

[Erick Perez]

anyone with some info on this?

thanks again.

On 9/14/05, Erick Perez <[EMAIL PROTECTED]> wrote:
> Using sipura sip/g729 to connect to an asterisk server that will
> server as a gateway to a VOIP provider, all in g729 will require to
> purchase codecs from Digium?
> 
> also, in this scenario the transcoding is almost non-existent right?
> I have read many documents about the type of codecs, and g729 seems to
> be a good trade between almost-toll quality and low bandwith usage
> right?
> 
> 
> A P4 (HT) 3.2 Ghz with 1gb ram in a 800mhz board with 3 raid0 disk can
> sustain more than 100 calls or up to a 100?
> I just looking at hardware capacity, since the machine will be located
> at an ISP with more than needed bandwith.
> 
> There is no need for voicemail, web interfaces or anything else, since
> the * box will only function as a gateway to a US-based VOIP provider.
> 
> The machine in question runs Centos4 Linux (Redhat enterprise 4) and
> CDR logging only.
> 
> Thanks,
> 


-- 

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!)
Panama, Republic of Panama
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Re: [Asterisk-Users] How to access * thru router when ip address is not known

[mike.jennings]

dyndns.org
> 
> From: Robert Rawlinson <[EMAIL PROTECTED]>
> Date: 2005/09/16 Fri PM 03:51:56 EDT
> To: Asterisk 
> Subject: [Asterisk-Users] How to access * thru router when ip address is not
>   known
> 
> I am going to be traveling and I wanted to be able to get on the 
> internet and call thru * to make calls. The problem is I do not have a 
> fixed ip address. How do you make this work? I will be using IAXCOMM. TIA
> Bob
> ___
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> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] How to access * thru router when ip address is not known

[Robert Rawlinson]

I am going to be traveling and I wanted to be able to get on the 
internet and call thru * to make calls. The problem is I do not have a 
fixed ip address. How do you make this work? I will be using IAXCOMM. TIA

Bob
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[Asterisk-Users] asterisk mixing sound card with anybody?

[Jerry Geis]

I am trying to get asterisk to MIX micely with alsa or oss.

Anybody doing that? if so can you share with me how you did it.


When I set modules.conf to load alsa and noload oss asterisk
starts and binds the sound port. nothing else will play.
If I stop asterisk other things play.

If I put a file /etc/asound.conf asterisk doesnt like it (tried a couple 
ones)

and will not use the sound port at all. This same config file allows other
applications to mix nicely like mplayer and aplay at the same time.

I dont want to have to stop asterisk to play a mplayer file or an aplay 
file.


I then tried noload on alsa and load on oss. Pretty much with the same 
results.

Can get anything else to play if asterisk is playing..

You help is appreciated.

Thanks,

Jerry

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[Asterisk-Users] Weird behaviour

[andrutto]

Hi,

 I noticed this weird behavior - in my office I use mixed "phone technology". I 
use Sip and Zap phones, analog and ISDN. I also defined a pickup feature and 
everything works prima to the time when I want to pickup call with ISDN phone. 
The console says (when I press my pickup extension *6) no such extension. Why? 
Other phones do not need special definition in extension.conf. When I added 
extra line in my dialplan - exten => *6,1,Pickup(group) - the ISDN phone was 
able to pickup all channels instead of Sip. To be certain the Sip channel was 
picked up, but the phone did not stop to ring. All phones are in the same 
pickup and call group. 
Do I need something special in my zapata.conf to normally pickup all channels 
or do I have to add some application to asterisk? Or I just did something wrong?

Regards
Andrutto


--
Prawie 40.000 samochodow na sprzedaz! >>> http://link.interia.pl/f18b2

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RE: R: [Asterisk-Users] direct sip call pickup

[Joshua Colp - Asterlink]

Hello Everyone,

For regular call pickup you can't really specify a pickup group number...
that's why it's set in the configuration.

For directed call pickup you need to have the latest CVS head as it uses an
API call that Kevin put in espically for me to use lastnight.

Joshua Colp

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan & Company, LLC
Sent: Friday, September 16, 2005 2:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: R: [Asterisk-Users] direct sip call pickup

What you were trying to do, *8#exten, is almost right I think.  Look at 
it like this instead, though.  The # is a pickup group number:

*8x
where x is the pickup group you want to pick up a call from.  I could be 
wrong but that's how I understood it.

Mojo

Giordano Grandis wrote:
> I cannot use CVS, is there anoyher way to use direct pickup ?
> 
>  
> 
> Thanks again
> 
>  
> 
> **Giordano**
> 
> 
> 
> *Da:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *Per conto di 
> *Alexander Lopez
> *Inviato:* venerdì 16 settembre 2005 17.53
> *A:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Oggetto:* RE: [Asterisk-Users] direct sip call pickup
> 
>  
> 
> On CVS head there is app_directed_pickup
> 
>  
> 
> It will let you pickup a ringing extension directly without having to 
> worry about pickup groups etc.
> 
>  
> 
>  
> 
>  
> 
>

> 
> *From:* [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] *On Behalf Of
> *Giordano Grandis
> *Sent:* Friday, September 16, 2005 11:19 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [Asterisk-Users] direct sip call pickup
> 
> Hi, i’m working about sip call pick and *8 works very fine but I
> pickup ringing phone on the same group. What happen if I have more
> than one ringing call?
> 
> I tryied *8#exten, *8eten# but it doesn’t wotk.
> 
> Is it correct? How it does work ?
> 
>  
> 
> Thanks
> 
>  
> 
> **Giordano**
> 
> 
> 
> 
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-- 
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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[Asterisk-Users] Re: Polycom randomly fails outbound calls,

[Noah Miller]

Hi Andres -

I am not using [EMAIL PROTECTED], it didn't work well for me, I just using AMP  
over asterisk, and yes, sip are 100% tweakable,

how do you configure your system, all by hand?


Yeah, by hand.  When I first started doing this there was no such  
thing as AMP.  Plus, I've got some wacky dialplan stuff that probably  
wouldn't work out too well with AMP.



Well, the two weird things I see here are the type setting and  
the  host.  Type is set to peer, but there doesn't seem to be a   
corresponding user definition (AFAIK, all peers have to have  
users).   You might try changing it to "type=friend" instead (like  
201).


I did it, it was set to peer just because I red somewhere that  
Polys didn't like friend type,



For the host setting, this is the address of the sip device, and  
not  the asterisk server.  If you have the Polycom set to a static  
address  of 192.168.1.18, all is well.  If your Polycom is set to  
DHCP (this  is the default), you should use "host=dynamic"


it's fixed to 18


Well, if you've got all this stuff and the phones still aren't  
working, I'd say there's either something funny going on with AMP or  
Asterisk.  I would try new/different versions of both.  The rest of  
your configs look good to me.  Maybe somebody else can spot an  
inconsistency?


Just to cover all the bases, have you tried configuring the Polycom  
without AMP?


- Noah
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Re: [Asterisk-Users] queue_log on mysql

[lenz]

Thanks, is there a standard schema for queue_log or can I define it myself?
Thanks
l.


In data Fri, 16 Sep 2005 18:48:13 +0200, William Lloyd <[EMAIL PROTECTED]>  
ha scritto:



Best to log directly to MySQL.  Add in ODBC code.

You are not the first to ask for it.

-bill

On 16-Sep-05, at 11:06 AM, lenz wrote:


Hello,
is there a best practice to upload queue_log file into MySQL? or -  
better

- to have Asterisk log the queue_log straight to MySQL?
Is it worth doing?
Thanks
l.

--Assum est, versa et manduca.
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--
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Re: [Asterisk-Users] USB Phones for use with Asterisk

[Guillermo Salas M]

On Fri, 2005-09-16 at 11:54 +0100, c waddy wrote:
> Are there drivers available for Xten Softphones?
>  
> Do they work with X-lite/Eyebeam Softphones?
>  

And work on Linux (kernel 2.6)?

>  
> 
> 
>  
> On 9/16/05, Bill McCready (PCPhoneline.com)
> <[EMAIL PROTECTED]> wrote: 
> > Hi all,
> >
> > I have a question that I was hoping someone could answer for
> me.
> >
> > I would like to find a USB phone that works with Asterisk...
> preferably 
> > one that does not have any issues with NAT.
> 
> RESPONSE:  This is Bill McCready from PCPhoneline.com .   I
> suggest you give
> our VTA1000 Skype+SIP+IAX2 Gateway which is USB based with a
> good cordless
> phone. 
> 
> We offer several versions of the VTA1000 but this is the
> latest one we have
> developed and it supports Skype + 10 x SIP + 10 x IAX2
> networks all at the
> same time and plugs into the USB port of a Windows 2000 or
> Windows XP 
> computer.   You can optionally disable Skype and/or the SIP
> calling
> capabilities if you don't need them.   The VTA1000 is designed
> to work even
> if your computer screen is powered off using it with a
> cordless phone of 
> course gives you the added advantage of not needing to be tied
> to your
> computer to place and receive calls.   If you don't mind being
> tethered to a
> computer we also have the VPT1000 phone available with the
> same capability 
> set.
> 
> You can find them in our Ebay store at
> http://stores.ebay.com/pcphoneline .
> If you need a bunch of these and want a build specially
> tailored and branded
> please private e-mail me with your requirements and we will do
> our best to
> try to accomodate you.
> 
> Best regards...Bill
> 
> 
> >
> > Can anyone point me to something suitable ?  We are
> essentially planning 
> > to do a beta for a few hundred customers on our network and
> would like
> > them to be able to call one another @ no charge.
> >
> > Because these are going to be end users who have no idea
> about Voice / 
> > VOIP stuff... it needs to be extremely easy to use.
> >
> > Can someone point me in the right direction ?
> >
> > Cheers,
> >
> > Callum
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> >
> >
> 
> 
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-- 
Guillermo Salas M.
Telconet S.A. Manta
Calle 15 y Av. 24 Esq.
Phone : 593 5 262 8071
Mobile: 593 9 985 5138
SIP   : [EMAIL PROTECTED]
e-mail: [EMAIL PROTECTED]
www   : http://www.telconet.net
http://www.telcocarrier.net

Linux User: 255902
Soporte en Linea en http://www.manta.telconet.net

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

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[Asterisk-Users] lastest spandsp-0.03pre1 don't compile

[Raymond Chen]

Dear all,

 

Anyone get the lastest spandsp with udptl.c and tpkt.c
compile in Fedora 3?

 

tpkt.c: In function `accept_thread':

tpkt.c:140: error: `TCP_NODELAY' undeclared (first use
in this function)

tpkt.c:140: error: (Each undeclared identifier is
reported only once

tpkt.c:140: error: for each function it appears in.)

tpkt.c:144: error: invalid application of `sizeof' to
incomplete type `mansession' 

tpkt.c:148: error: invalid application of `sizeof' to
incomplete type `mansession' 

tpkt.c:149: error: dereferencing pointer to incomplete
type

tpkt.c:151: error: `block_sockets' undeclared (first
use in this function)

tpkt.c:156: error: dereferencing pointer to incomplete
type

tpkt.c:157: error: dereferencing pointer to incomplete
type

tpkt.c:158: error: dereferencing pointer to incomplete
type

tpkt.c:159: error: `sessionlock' undeclared (first use
in this function)

tpkt.c:160: error: dereferencing pointer to incomplete
type

tpkt.c:160: error: `sessions' undeclared (first use in
this function)

tpkt.c:163: error: `t' undeclared (first use in this
function)

tpkt.c:163: error: `session_do' undeclared (first use
in this function)

tpkt.c:164: warning: implicit declaration of function
`destroy_session'

tpkt.c: At top level:

tpkt.c:171: warning: no previous prototype for
'init_tpkt'

tpkt.c: In function `init_tpkt':

tpkt.c:173: warning: passing arg 1 of `pthread_kill'
makes integer from pointer without a cast

tpkt.c:177: error: `addr' undeclared (first use in
this function)

tpkt.c:178: error: `portno' undeclared (first use in
this function)

tpkt.c:179: error: `ba' undeclared (first use in this
function)

tpkt.c:180: error: `val' undeclared (first use in this
function)

tpkt.c:180: error: `cfg' undeclared (first use in this
function)

tpkt.c:188: warning: `return' with a value, in
function returning void

tpkt.c:190: error: `x' undeclared (first use in this
function)

tpkt.c:195: warning: `return' with a value, in
function returning void

tpkt.c:201: warning: `return' with a value, in
function returning void

tpkt.c:205: error: `t' undeclared (first use in this
function)

tpkt.c: In function `tpkt_rx_packet':

tpkt.c:241: warning: implicit declaration of function
`decode_open_type'

tpkt.c:211: warning: unused variable `stat2'

tpkt.c:212: warning: unused variable `i'

tpkt.c:213: warning: unused variable `j'

 

 

udptl.c: In function `udptl_process_packet':

udptl.c:150: warning: no return statement in function
returning non-void

udptl.c: In function `udptl_build_packet':

udptl.c:541: warning: implicit declaration of function
`udptl_debug_test_addr'

udptl.c:541: error: `udptl' undeclared (first use in
this function)

udptl.c:541: error: (Each undeclared identifier is
reported only once

udptl.c:541: error: for each function it appears in.)

udptl.c:543: error: `iabuf' undeclared (first use in
this function)

udptl.c:544: error: `payload' undeclared (first use in
this function)

udptl.c:544: error: `res' undeclared (first use in
this function)

udptl.c:544: error: `hdrlen' undeclared (first use in
this function)

udptl.c: At top level:

udptl.c:571: warning: static declaration of
'udptl_debug_test_addr' follows non-static declaration

udptl.c:541: warning: 'udptl_debug_test_addr' declared
inline after being called

udptl.c:541: warning: previous implicit declaration of
'udptl_debug_test_addr' was here

udptl.c: In function `ast_udptl_read':

udptl.c:643: error: `payloadtype' undeclared (first
use in this function)

udptl.c:643: error: `timestamp' undeclared (first use
in this function)

udptl.c:643: error: `hdrlen' undeclared (first use in
this function)

udptl.c:649: error: `AST_FORMAT_T38' undeclared (first
use in this function)

udptl.c: In function `ast_udptl_new_with_bindaddr':

udptl.c:706: error: `s' undeclared (first use in this
function)

udptl.c: In function `ast_udptl_write':

udptl.c:820: error: `codec' undeclared (first use in
this function)

udptl.c:820: error: structure has no member named
`lastts'

udptl.c:820: error: `hdrlen' undeclared (first use in
this function)

udptl.c: In function `ast_udptl_reload':

udptl.c:1084: warning: implicit declaration of
function `ast_load'

udptl.c:1084: warning: assignment makes pointer from
integer without a cast

udptl.c:1138: warning: implicit declaration of
function `ast_destroy'

 

ray

 






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[Asterisk-Users] Re: Polycom randomly fails outbound calls,

[Andres Paglayan]

Noah Miller wrote:


Hi Andres -
The two that we have are just used as lobby phones.  They're good  
little phones, but if you have the money, I'd definitely recommend  
the IP501 instead.  The screen is MUCH better, and having full  
speakerphone is great!  Plus the 500/501 just feels a little more solid.


Yeah, I think it was a wrong move going for the 301 instead,




Hmm.  I'm not sure either.  I've never used AMP before (except for a  
quick glance at [EMAIL PROTECTED]).  If you can change the sip settings,  
I don't think it should matter.


I am not using [EMAIL PROTECTED], it didn't work well for me, I just using AMP over 
asterisk, and yes, sip are 100% tweakable,

how do you configure your system, all by hand?



Well, the two weird things I see here are the type setting and the  
host.  Type is set to peer, but there doesn't seem to be a  
corresponding user definition (AFAIK, all peers have to have users).   
You might try changing it to "type=friend" instead (like 201).


I did it, it was set to peer just because I red somewhere that Polys 
didn't like friend type,




For the host setting, this is the address of the sip device, and not  
the asterisk server.  If you have the Polycom set to a static address  
of 192.168.1.18, all is well.  If your Polycom is set to DHCP (this  
is the default), you should use "host=dynamic"


it's fixed to 18



A couple of things that I know you don't need:
nat=never
qualify=no


I took them off too, I got them from the only "how to" I found about amp 
and the polycom,





- Noah



Thanks ,
I hope I can help you same day,

Andres
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Re: [Asterisk-Users] Sipura 2k voice quality

[Andres Paglayan]

Raise both gains from -3 to 5 that solves volume problem,
log in, click admin, advanced, I guess is on the sip tab,

Matthew Harrell wrote:


When I have voip conversations over asterisk through my computer the voice
quality is nice and loud and quite clear.  When I go through my Sipura 2K
then the conversations are typically very muted and my responses sound somewhat
delayed.  I've tried fiddling with settings under asterisk and the 2K but 
haven't had any luck.  Can someone point me in the right direction to get

this configured right?

 



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RE: [Asterisk-Users] wiki down?

[Wiley Siler]

I got right in just fine...

W
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Francesco
Peeters
Sent: Friday, September 16, 2005 10:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] wiki down?

I'm unable to connect to voip-info.org... Anybody else have the same
issues, ro is it just me?

--
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704 If your
program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA
certificate.
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[Asterisk-Users] didn't get a frame from channel

[Andres Paglayan]

This is an excerpt from the log file,

My problem is that randomly, 1out of 3 or 1 out of 2, some calls are
not going out and this is the message in the log file,
The device that should provide the frame is a Sipura 3000 which has its
FXO providing outside connectivity,

  24185 Sep 16 10:35:40 DEBUG[17604]: Didn't get a frame from
channel: SIP/200-b635
  24186 Sep 16 10:35:40 DEBUG[17604]: Bridge stops bridging channels
SIP/200-b635 and SIP/pstn_1-a971
  24187 Sep 16 10:35:40 DEBUG[17604]: update_user_counter(ww9863038) -
decrement outUse counter
  24188 Sep 16 10:35:40 DEBUG[17604]: ww9863038 is not a local user
  24189 Sep 16 10:35:40 DEBUG[17604]: Exiting with DIALSTATUS=ANSWER.

I can provide the whole call log but it's many lines long.

Any clue on where to look for anything?

Thanks

Andres



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[Asterisk-Users] wiki down?

[Francesco Peeters]

I'm unable to connect to voip-info.org... Anybody else have the same
issues, ro is it just me?

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
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[Asterisk-Users] Zap failed

[Ugur GUNCER]

I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to
start:

Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>

 [chan_zap.so] => (Zapata Telephony w/PRI)
  == Parsing '/etc/asterisk/zapata.conf': Found
Sep 16 20:36:51 ERROR[6750]: chan_zap.c:6246 mkintf: Unable to get
parameters
Sep 16 20:36:51 ERROR[6750]: chan_zap.c:9191 setup_zap: Unable to register
channel '1-15'
Sep 16 20:36:51 WARNING[6750]: loader.c:345 ast_load_resource: chan_zap.so:
load_module failed, returning -1
  == Unregistered channel type 'Tor'
  == Unregistered channel type 'Zap'
Sep 16 20:36:51 WARNING[6750]: loader.c:440 load_modules: Loading module
chan_zap.so failed!
sip:/etc # Ouch ... error while writing audio data: : Broken pipe




Here is the /proc/zaptel/1 file(seems to be correct, and log Messages too
indicates the initialization is correct.):

 cat /proc/zaptel/1
Span 1: WCT1/0 "Digium Wildcard E100P E1/PRA Card 0" HDB3/CCS ClockSource

   1 WCT1/0/1 Clear
   2 WCT1/0/2 Clear
   3 WCT1/0/3 Clear
   4 WCT1/0/4 Clear
   5 WCT1/0/5 Clear
   6 WCT1/0/6 Clear
   7 WCT1/0/7 Clear
   8 WCT1/0/8 Clear
   9 WCT1/0/9 Clear
  10 WCT1/0/10 Clear
  11 WCT1/0/11 Clear
  12 WCT1/0/12 Clear
  13 WCT1/0/13 Clear
  14 WCT1/0/14 Clear
  15 WCT1/0/15 Clear
  16 WCT1/0/16 HDLCFCS
  17 WCT1/0/17 Clear
  18 WCT1/0/18 Clear
  19 WCT1/0/19 Clear
  20 WCT1/0/20 Clear
  21 WCT1/0/21 Clear
  22 WCT1/0/22 Clear
  23 WCT1/0/23 Clear
  24 WCT1/0/24 Clear
  25 WCT1/0/25 Clear
  26 WCT1/0/26 Clear
  27 WCT1/0/27 Clear
  28 WCT1/0/28 Clear
  29 WCT1/0/29 Clear
  30 WCT1/0/30 Clear
  31 WCT1/0/31 Clear

Ztcfg -vv
Zaptel Configuration
==

SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: Individual Clear channel (Default) (Slaves: 03)
Channel 04: Individual Clear channel (Default) (Slaves: 04)
Channel 05: Individual Clear channel (Default) (Slaves: 05)
Channel 06: Individual Clear channel (Default) (Slaves: 06)
Channel 07: Individual Clear channel (Default) (Slaves: 07)
Channel 08: Individual Clear channel (Default) (Slaves: 08)
Channel 09: Individual Clear channel (Default) (Slaves: 09)
Channel 10: Individual Clear channel (Default) (Slaves: 10)
Channel 11: Individual Clear channel (Default) (Slaves: 11)
Channel 12: Individual Clear channel (Default) (Slaves: 12)
Channel 13: Individual Clear channel (Default) (Slaves: 13)
Channel 14: Individual Clear channel (Default) (Slaves: 14)
Channel 15: Individual Clear channel (Default) (Slaves: 15)
Channel 16: D-channel (Default) (Slaves: 16)
Channel 17: Individual Clear channel (Default) (Slaves: 17)
Channel 18: Individual Clear channel (Default) (Slaves: 18)
Channel 19: Individual Clear channel (Default) (Slaves: 19)
Channel 20: Individual Clear channel (Default) (Slaves: 20)
Channel 21: Individual Clear channel (Default) (Slaves: 21)
Channel 22: Individual Clear channel (Default) (Slaves: 22)
Channel 23: Individual Clear channel (Default) (Slaves: 23)
Channel 24: Individual Clear channel (Default) (Slaves: 24)
Channel 25: Individual Clear channel (Default) (Slaves: 25)
Channel 26: Individual Clear channel (Default) (Slaves: 26)
Channel 27: Individual Clear channel (Default) (Slaves: 27)
Channel 28: Individual Clear channel (Default) (Slaves: 28)
Channel 29: Individual Clear channel (Default) (Slaves: 29)
Channel 30: Individual Clear channel (Default) (Slaves: 30)
Channel 31: Individual Clear channel (Default) (Slaves: 31)

31 channels configured.


/etc/zaptel file:
span=1,1,0,ccs,hdb3

bchan=1-15,17-31
dchan=16

loadzone=us
defaultzone=us

/etc/asterisk/zapata.conf

[channels]
switchtype = euroisdn
signalling = pri_net
pridialplan = local
language=en
context=ivr-in
overlapdial=yes
usecallerid=yes
hidecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
musiconhold=default
group=1
channel => 1-15
channel => 17-31 



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[Asterisk-Users] Sipura 2k voice quality

[Matthew Harrell]

When I have voip conversations over asterisk through my computer the voice
quality is nice and loud and quite clear.  When I go through my Sipura 2K
then the conversations are typically very muted and my responses sound somewhat
delayed.  I've tried fiddling with settings under asterisk and the 2K but 
haven't had any luck.  Can someone point me in the right direction to get
this configured right?

-- 
  Matthew Harrell  Beauty is in the eye of the beer
  Bit Twiddlers, Inc.   holder.
  [EMAIL PROTECTED] 
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Re: R: [Asterisk-Users] direct sip call pickup

[Mojo with Horan & Company, LLC]

What you were trying to do, *8#exten, is almost right I think.  Look at 
it like this instead, though.  The # is a pickup group number:


*8x
where x is the pickup group you want to pick up a call from.  I could be 
wrong but that's how I understood it.


Mojo

Giordano Grandis wrote:

I cannot use CVS, is there anoyher way to use direct pickup ?

 


Thanks again

 


**Giordano**



*Da:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *Per conto di 
*Alexander Lopez

*Inviato:* venerdì 16 settembre 2005 17.53
*A:* Asterisk Users Mailing List - Non-Commercial Discussion
*Oggetto:* RE: [Asterisk-Users] direct sip call pickup

 


On CVS head there is app_directed_pickup

 

It will let you pickup a ringing extension directly without having to 
worry about pickup groups etc.


 

 

 




*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of
*Giordano Grandis
*Sent:* Friday, September 16, 2005 11:19 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [Asterisk-Users] direct sip call pickup

Hi, i’m working about sip call pick and *8 works very fine but I
pickup ringing phone on the same group. What happen if I have more
than one ringing call?

I tryied *8#exten, *8eten# but it doesn’t wotk.

Is it correct? How it does work ?

 


Thanks

 


**Giordano**




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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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RE: [Asterisk-Users] RxFax/TxFax - Compile Problem

[Alexander Lopez]

Maybe I should ask this question that I know has been 
discussed to death.
 
"stable" = 1.0 release
"CVS HEAD' = 1.1 release
 
Is this a correct statment
 
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of David 
  SampsonSent: Thursday, September 15, 2005 12:17 PMTo: 
  [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: RE: [Asterisk-Users] RxFax/TxFax - Compile 
  Problem
  
  
  I used the latest 
  version (.3) and also the previous .2 ver (pre20).  The spandsp seems to 
  compile but when I download the rxfax/txfax .c files and drop them in the apps 
  directory that is where I get the compile error.
  
  Dave
   
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of BJ WeschkeSent: Wednesday, September 14, 2005 2:43 
  PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [Asterisk-Users] RxFax/TxFax 
  - Compile Problem
   
   What version of spandsp are you 
  attempting to compile in to the 1.0.9 tree?
  
  On 9/14/05, David 
  Sampson <[EMAIL PROTECTED]> 
  wrote: 
  
  Anyone know how to fix 
  this?gcc -shared -Xlinker -x -o app_rxfax.so app_rxfax.c -lspandsp 
  -ltiff
  In file included from 
  app_rxfax.c:14:
  /usr/include/asterisk/lock.h: In 
  function `ast_mutex_init':
  /usr/include/asterisk/lock.h:302: 
  error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this 
  function)
  /usr/include/asterisk/lock.h:302: 
  error: (Each undeclared identifier is reported only 
  once
  /usr/include/asterisk/lock.h:302: 
  error: for each function it appears in.)
  app_rxfax.c: In function 
  `rxfax_exec':
  app_rxfax.c:263: warning: passing 
  arg 1 of `fax_init' from incompatible pointer 
type
  app_rxfax.c:264: error: structure 
  has no member named `verbose'
  app_rxfax.c:325: warning: passing 
  arg 1 of `fax_release' from incompatible pointer 
  type
  make[1]: *** [app_rxfax.so] Error 
  1
  make[1]: Leaving directory 
  `/usr/src/asterisk/asterisk-1.0.9/apps'
  make: *** [subdirs] Error 
  1
   
   
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[Asterisk-Users] Easier way for end user to change main greeting?

[Doug]

Hi,

Has someone figured out how to change the main
autoattendant message easily?

Right now, you call *77 and record the message.
Then you have to get into the Unix/Linux command
line to get that message over to where it will
be used.  Is there a simpler way?

Thanks for your help.

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Re: [Asterisk-Users] Re: T.38 ATA

[VoIP Newbie]

PA-122TI from www.broad-tel.com supports T.38 and Fax pass-thru.
On 9/15/05, Rosario Pingaro <[EMAIL PROTECTED]> wrote:
about spa-2100, the t38 stream is on UDPTL and so asterisk passthroughdoesn't work.- Original Message -
From: "Nenad Radosavljevic" <[EMAIL PROTECTED]>To: Sent: Wednesday, September 14, 2005 12:16 PM
Subject: [Asterisk-Users] Re: T.38 ATA> Hi !>> First of all thank you all for fast response on matter of T.38 capable> ATAs.>> I have asked a UK VoIP suplier to check with manufacterers of various ATAs
> they sell, do they support T.38 and here is what they/I have got as a> result:>> 1. Sipura SPA-2100 only and with firmware 3.2.1 is T.38 capable (no> information on type of T.28 support UDPTL/TPKT)
>> 2. All Gradnstream Handytone ATAs with firmware grater than 1.0.6.x are> T.38 capable and they use UDPTL T.38>> Regards,>Nenad>> The newest 2100 firmware has 
T.38.>> ___> --Bandwidth and Colocation sponsored by Easynews.com -->> Asterisk-Users mailing list
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Re: [Asterisk-Users] queue_log on mysql

[William Lloyd]

Best to log directly to MySQL.  Add in ODBC code.

You are not the first to ask for it.

-bill

On 16-Sep-05, at 11:06 AM, lenz wrote:


Hello,
is there a best practice to upload queue_log file into MySQL? or -  
better

- to have Asterisk log the queue_log straight to MySQL?
Is it worth doing?
Thanks
l.

--
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RE: [Asterisk-Users] RxFax/TxFax - Compile Problem

[Dave Cotton]

On Fri, 2005-09-16 at 09:36 -0400, David Sampson wrote:
> If I understand correctly you are supposed to patch the Makefile in the apps 
> directory and then run the main Makefile.  I've tried both ways - the patch 
> failed on the main Makefile.  Should I try to make that work?

There's next to nothing in the patchfile so edit the Makefile manually.


-- 
Dave Cotton <[EMAIL PROTECTED]>

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[Asterisk-Users] Asterisk as a gateway. 'flash for transfers transparency?'

[felipe hangen]

Hi,
I have 2 asterisk boxes as Gateway, in this arrangement.


(PANASONIC PBX) -  [ASTERISK1] - network - [ASTERISK2] - (ANALOG PHONE)


everything works great, in both directions (receiving and making calls),
but when i get a call on the (ANALOGPHONE), I haven't been able to 
transfer it to another extension of the PANASONIC PBX  using the flash key.

I've tried the using the t T options on the 'Dial(' command of the extensions,  but no
luck.  I assume that's more for transfering on the asterisk domain.  I just want to pass
the flash key through the 2 asterisk boxes to the panasonic PBX. (Call transfer, blind i 
would say).

anyone have some insight on how to do this? ( i haven't been able to locate any documentation
on the web)

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Re: use of automon sequence (was Re: [Asterisk-Users] Call recording between SIP phones)

[John covici]

Thanks -- it works very nicely -- I will have to try that filename,
but otherwise it seems to be just fine.

on Friday 09/16/2005 Kevin Bockman([EMAIL PROTECTED]) wrote
 > John covici wrote:
 > > OK, I wonder if I have something wrong -- I have the *1 in my
 > > features.conf for the one touch record -- now I called a number, and
 > > when the call was answered flashed the hook and pressed *1 and went
 > > back tothe call, but nothing happpened.  I am using CVS from 8/26 --
 > > is this too old or am I doing something wrong?  I did add wW to the
 > > dial options of the call and that part did work, but there is nothing
 > > in the log about automon.  Also, what kind of file does this produce,
 > > etc?
 > > 
 > > Thanks in advance for any help.
 > I'm not sure when it started, but as long as you have it in the example 
 > features.conf in your asterisk source/configs/features.conf file.
 > 
 > You don't do a flash to do the record.  You press the sequence WHILE YOU 
 > ARE ON THE CALL.  It will record a file to /var/spool/monitor.  To 
 > change the filename, you probably have to set the MONITOR_FILENAME 
 > variable before the dial.
 > 
 > 
 > Kevin
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-- 
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How do
you spend it?

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 [EMAIL PROTECTED]
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Re: R: [Asterisk-Users] direct sip call pickup

[Iqbal]

see if it compiles into ur install

Iqbal

Giordano Grandis wrote:


I cannot use CVS, is there anoyher way to use direct pickup ?

Thanks again

**Giordano**



*Da:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *Per conto di 
*Alexander Lopez

*Inviato:* venerdì 16 settembre 2005 17.53
*A:* Asterisk Users Mailing List - Non-Commercial Discussion
*Oggetto:* RE: [Asterisk-Users] direct sip call pickup

On CVS head there is app_directed_pickup

It will let you pickup a ringing extension directly without having to 
worry about pickup groups etc.




*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of
*Giordano Grandis
*Sent:* Friday, September 16, 2005 11:19 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [Asterisk-Users] direct sip call pickup

Hi, i’m working about sip call pick and *8 works very fine but I
pickup ringing phone on the same group. What happen if I have more
than one ringing call?

I tryied *8#exten, *8eten# but it doesn’t wotk.

Is it correct? How it does work ?

Thanks

**Giordano**



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R: [Asterisk-Users] direct sip call pickup

[Giordano Grandis]

I cannot use CVS, is
there anoyher way to use direct pickup ?

 

Thanks again

 



Giordano











Da:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di Alexander Lopez
Inviato: venerdì 16 settembre 2005
17.53
A: Asterisk Users Mailing List -
Non-Commercial Discussion
Oggetto: RE: [Asterisk-Users]
direct sip call pickup



 

On CVS head there is app_directed_pickup

 

It will let you pickup a ringing extension
directly without having to worry about pickup groups etc.

 

 



 







From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis
Sent: Friday, September 16, 2005
11:19 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] direct
sip call pickup

Hi, i’m working about sip call pick and *8
works very fine but I pickup ringing phone on the same group. What happen if I
have more than one ringing call?

I tryied *8#exten, *8eten# but it doesn’t wotk.

Is it correct? How it does work ?

 

Thanks

 

Giordano








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Re: [Asterisk-Users] Caller Name: Asterisk reading too fast

[steve]

On Fri, 16 Sep 2005, J Thomas wrote:

> I asked my telco to release caller name on the PRI. Earlier they were
> releasing only the phone number. 
> 
> I still did not see the name, but only the number in caller id. Actually
> I now see number twice. When I inquired with them this is the response I
> got:
> 
>   "I ran a trace on your TG.  I see that your switch is
>   picking up the call so fast that it is not able to pick
>   up the name. The name is being sent, but I suspect after
>   it is too late.  This is something that will need to be
>   corrected in your switch. 
> 
>   I have attached a sample call out of the trace I performed
>   this morning."
> 
> They have sent me the trace file.
> 
> Is there a way as it is in Asterisk so that it reads the caller name
> properly?


Sheesh - you should really look after that support contact at your telco.

Than ran a trace, looked at it, understood it and gave you great advice on 
how to solve the problem.

That must be 1 in a 1000 support!

Steve

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[Asterisk-Users] New version of idefisk softphone released.

[Zoa]

We just uploaded the latest and greatest version of the idefisk iax2
softphone,  version 1.24

Freely downloadable at: http://www.asteriskguru.com/tools/idefisk_beta.php

Changes since the last release include:

- history panel is working
- receiving messages and urls (sendtext command in asterisk)
- some bugfixes (the annoying hangup bug is finally gone!).

A big thanks to everybody who sent us comments and suggestions !

Zoa


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[Asterisk-Users] mpg123 on x86_64 (Opteron MP)

[asterisk]

Has anyone successfully compiled mpg123 in the 1.0.x or 1.2beta1  
distributions (I'm running FC3 linux on an Opteron 2 processor  
system)?  Are there any patches out there to make it work?


gcc -DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX  - 
DREAD_MMAP -DOSS -DTERM_CONTROL-Wall -O2 - 
m486 -fomit-frame-pointer -funroll-all- 
loops  -finline-functions -ffast-math   -c -o  
dct64_i386.o dct64_i386.c

as   -o decode_i586.o decode_i586.s
decode_i586.s: Assembler messages:
decode_i586.s:44: Error: suffix or operands invalid for `push'

decode_i586.s:161: Error: suffix or operands invalid for `pop'

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RE: [Asterisk-Users] Caller Name: Asterisk reading too fast

[Alexander Lopez]

 
This is because Caller ID name is being send in the FACILITY messages
instead of in the SETUP.

Wait(1) is the solution, There is nothing wrong with your install or
asterisk. 

Asterisk picks up on SETUP as it should. When it recieves the Name via
FACILITY it propigates it in the channel, however the other channel has
already started to ring so it does not have the name when it gets built
and since ZAP, IAX, and SIP, etc. do not support Caller ID after the
setup you never get it

Search on the Mantis for this


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jeremy Gault
> Sent: Friday, September 16, 2005 11:29 AM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Caller Name: Asterisk reading too fast
> 
> As was already suggested, Wait() is your friend.
> 
> We had the same problem when our PRI was installed.  It was 
> supposed to include Caller ID Name delivery, but it seemed to 
> be hit-or-miss as to if it would work.
> 
> This is what I found: When people call our auto-attendant and 
> dial an extension, the name would work fine.  However, if 
> they dialed one of our DIDs (each extension has its own DID) 
> then the name did not work.
> 
> Inserting a Wait() command on the DIDs for extensions seemed 
> to do the trick, like this:
> 
> exten => 423303,1,Wait(0.75)
> exten => 423303,2,Dial(SIP/xxx)
> 
> Here's my take on what happened: A call comes in to the 
> auto-attendant, which immediately answers and begins the 
> greeting.  Shortly after, the telco sends the name down the 
> PRI, and Asterisk associates it to that call.  Then the 
> caller punches in the extension and it goes through with 
> name.  (Keep in mind it only takes a fraction of a second for 
> the name to show up on the PRI, so by the time a normal human 
> realizes the auto-attendant has answered and dials an 
> extension, plenty of time has passed for the name to come in.)
> 
> However, the DIDs were spwaning Dial() *immediately* when the 
> call came in.  Of course, at that point the name was not 
> present yet.  (The telco was still doing the lookup.)  But, 
> Dial() has to send CID immediately when invoked, so it sent 
> the number twice.  Adding the 0.75 second delay gave time for 
> the name to arrive before spawning Dial().
> 
> I've found 0.75 to be a fairly reliable delay.  It misses a 
> name here or there, but not often enough to be a problem.  
> Most callers won't really notice the delay.  If I increased 
> this to one full second, it would be more noticeable but 
> would probably fix the few ones we miss.  You may have to 
> play with this value to find the optimum setting for your setup.
> 
> Another test you should be able to do: Have a call come in 
> (when it shows only the number twice) and answer it.  Then 
> login to the Asterisk console, do "show channels" and find 
> out which Zap channel it's on.  Do a "show channel 
> Zap/whatever" (whichever channel it is) and see if the name 
> has shown up.  IIRC, before I added the Wait, the "show channel" 
> would still see the name, even though the phone didn't.
> 
> Good luck!
> 
>   Jeremy
> 
> 
> 
> J Thomas wrote:
> 
> >I asked my telco to release caller name on the PRI. Earlier 
> they were 
> >releasing only the phone number.
> >
> >I still did not see the name, but only the number in caller id. 
> >Actually I now see number twice. When I inquired with them 
> this is the 
> >response I
> >got:
> >
> >  "I ran a trace on your TG.  I see that your switch is
> > picking up the call so fast that it is not able to pick
> > up the name. The name is being sent, but I suspect after
> > it is too late.  This is something that will need to be
> > corrected in your switch. 
> >
> > I have attached a sample call out of the trace I performed
> > this morning."
> >
> >They have sent me the trace file.
> >
> >Is there a way as it is in Asterisk so that it reads the caller name 
> >properly?
> >
> >Thanks,
> >-- jt
> >
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> >  
> >
> 
> -- 
> Jeremy Gault, KD4NED<[EMAIL PROTECTED]>
> Network Administrator, WinWorld Corporation
> Member: Bradley County ACS/RACES/SkyWarn
> voice: +1.423.473.8084  fax: +1.423.472.9465
> fwd: 461771 msn msgr: [EMAIL PROTECTED]
> 
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RE: [Asterisk-Users] direct sip call pickup

[Alexander Lopez]

On CVS head there is 
app_directed_pickup
 
It will let you pickup a ringing extension directly 
without having to worry about pickup groups etc.
 
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Giordano 
  GrandisSent: Friday, September 16, 2005 11:19 AMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] direct sip 
  call pickup
  
  
  Hi, i’m working about sip call 
  pick and *8 works very fine but I pickup ringing phone on the same group. What 
  happen if I have more than one ringing call?
  I tryied *8#exten, *8eten# but it 
  doesn’t wotk.
  Is it correct? How it does work 
  ?
   
  Thanks
   
  Giordano
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Re: [Asterisk-Users] Caller Name: Asterisk reading too fast

[Jeremy Gault]

As was already suggested, Wait() is your friend.

We had the same problem when our PRI was installed.  It was supposed to 
include Caller ID Name delivery, but it seemed to be hit-or-miss as to 
if it would work.


This is what I found: When people call our auto-attendant and dial an 
extension, the name would work fine.  However, if they dialed one of our 
DIDs (each extension has its own DID) then the name did not work.


Inserting a Wait() command on the DIDs for extensions seemed to do the 
trick, like this:


exten => 423303,1,Wait(0.75)
exten => 423303,2,Dial(SIP/xxx)

Here's my take on what happened: A call comes in to the auto-attendant, 
which immediately answers and begins the greeting.  Shortly after, the 
telco sends the name down the PRI, and Asterisk associates it to that 
call.  Then the caller punches in the extension and it goes through with 
name.  (Keep in mind it only takes a fraction of a second for the name 
to show up on the PRI, so by the time a normal human realizes the 
auto-attendant has answered and dials an extension, plenty of time has 
passed for the name to come in.)


However, the DIDs were spwaning Dial() *immediately* when the call came 
in.  Of course, at that point the name was not present yet.  (The telco 
was still doing the lookup.)  But, Dial() has to send CID immediately 
when invoked, so it sent the number twice.  Adding the 0.75 second delay 
gave time for the name to arrive before spawning Dial().


I've found 0.75 to be a fairly reliable delay.  It misses a name here or 
there, but not often enough to be a problem.  Most callers won't really 
notice the delay.  If I increased this to one full second, it would be 
more noticeable but would probably fix the few ones we miss.  You may 
have to play with this value to find the optimum setting for your setup.


Another test you should be able to do: Have a call come in (when it 
shows only the number twice) and answer it.  Then login to the Asterisk 
console, do "show channels" and find out which Zap channel it's on.  Do 
a "show channel Zap/whatever" (whichever channel it is) and see if the 
name has shown up.  IIRC, before I added the Wait, the "show channel" 
would still see the name, even though the phone didn't.


Good luck!

 Jeremy



J Thomas wrote:


I asked my telco to release caller name on the PRI. Earlier they were
releasing only the phone number. 


I still did not see the name, but only the number in caller id. Actually
I now see number twice. When I inquired with them this is the response I
got:

 "I ran a trace on your TG.  I see that your switch is
picking up the call so fast that it is not able to pick
up the name. The name is being sent, but I suspect after
it is too late.  This is something that will need to be
	corrected in your switch. 


I have attached a sample call out of the trace I performed
this morning."

They have sent me the trace file.

Is there a way as it is in Asterisk so that it reads the caller name
properly?

Thanks,
-- jt

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--
Jeremy Gault, KD4NED<[EMAIL PROTECTED]>
Network Administrator, WinWorld Corporation
Member: Bradley County ACS/RACES/SkyWarn
voice: +1.423.473.8084  fax: +1.423.472.9465
fwd: 461771 msn msgr: [EMAIL PROTECTED]

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[Asterisk-Users] direct sip call pickup

[Giordano Grandis]

Hi, i’m working about sip call pick and *8
works very fine but I pickup ringing phone on the same group. What happen if I have
more than one ringing call?

I tryied *8#exten, *8eten# but it doesn’t wotk.

Is it correct? How it does work ?

 

Thanks

 

Giordano






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[Asterisk-Users] Asterisk don't start

[[EMAIL PROTECTED]]

Asterisk don't start, because show this message:


Sep 16 17:04:59 WARNING[1075709024]: cli.c:702 ast_cli_register: Command 
'showparkedcalls' already registered (or something close enough)
  == Parsing '/etc/asterisk/features.conf': Found
-- Added extension '700' priority 1 to parkedcalls
/usr/sbin/asterisk: relocation error: 
/usr/lib/asterisk/modules/res_features.so: undefined symbol: ast_pthread_create


THANKS!

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[Asterisk-Users] queue_log on mysql

[lenz]

Hello,
is there a best practice to upload queue_log file into MySQL? or - better
- to have Asterisk log the queue_log straight to MySQL?
Is it worth doing?
Thanks
l.

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[Asterisk-Users] alsa issue with asound.conf

[Jerry Geis]

I am using alsa with asterisk.
The asound.conf is below.

When I start asterisk with /etc/asound.conf present I get errors on
the console that:
chan_alsa.c:304 alsa_card_init: snd_pcm_open failed: Invalid argument

If I remove the asound.conf asterisk starts up and works. However
I NEED the asound.conf for another application.

What might be the issue here?

THanks
jerry


/etc/asound.conf
-
# This is /etc/asound.conf
#
# this makes legacy OSS apps use alsa software mixing dmix
pcm.dsp0 {
   type plug
   slave.pcm "dmix"
}

# mixer0 can stay unchanged, because it isn't used anyway, I guess ? ;)
ctl.mixer0 {
   type hw
   card 0
}

# this makes native ALSA apps default to using dmix
pcm.!default {
   type plug
   slave.pcm "dmix"
}
---

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RE: [Asterisk-Users] RxFax/TxFax - Compile Problem

[Alexander Lopez]

I had sone trouble with this also.

But I got it.

I used the latest SpanDSP
0.0.9pre1
I used the applications app)_txfax and app_rxfax from the previous release 
0.0.2pre8

I have included my makefile and the two app fiels that complied on the Latest 
CVS (last night) on FC3.

Apps/MakeFile  
#
# Asterisk -- A telephony toolkit for Linux.
#
# Makefile for PBX applications
#
# Copyright (C) 1999-2005, Digium
#
# Mark Spencer <[EMAIL PROTECTED]>
#
# This program is free software, distributed under the terms of
# the GNU General Public License
#

APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_mp3.so\
 app_system.so app_echo.so app_record.so app_image.so app_url.so 
app_disa.so \
 app_adsiprog.so app_getcpeid.so app_milliwatt.so \
 app_zapateller.so app_setcallerid.so app_festival.so \
 app_queue.so app_senddtmf.so app_parkandannounce.so app_striplsd.so \
 app_setcidname.so app_lookupcidname.so app_substring.so app_macro.so \
 app_authenticate.so app_softhangup.so app_lookupblacklist.so \
 app_waitforring.so app_privacy.so app_db.so app_chanisavail.so \
 app_enumlookup.so app_transfer.so app_setcidnum.so app_cdr.so \
 app_hasnewvoicemail.so app_sayunixtime.so app_cut.so app_read.so \
 app_setcdruserfield.so app_random.so app_ices.so app_eval.so \
 app_nbscat.so app_sendtext.so app_exec.so app_sms.so \
 app_groupcount.so app_txtcidname.so app_controlplayback.so \
 app_talkdetect.so app_alarmreceiver.so app_userevent.so app_verbose.so \
 app_test.so app_forkcdr.so app_math.so app_realtime.so \
 app_dumpchan.so app_waitforsilence.so app_while.so app_setrdnis.so \
 app_md5.so app_readfile.so app_chanspy.so app_settransfercapability.so \
 app_dictate.so app_externalivr.so app_directed_pickup.so app_rxfax.so 
app_txfax.so

#
# Obsolete things...
#
#APPS+=app_sql_postgres.so
#APPS+=app_sql_odbc.so

#
# Experimental things
#
#APPS+=app_ivrdemo.so
#APPS+=app_skel.so
#APPS+=app_rpt.so

ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/linux/zaptel.h)$(wildcard 
$(CROSS_COMPILE_TARGET)/usr/local/include/zaptel.h),)
  APPS+=app_zapras.so app_meetme.so app_flash.so app_zapbarge.so app_zapscan.so
endif

ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/osp/osp.h),)
APPS+=app_osplookup.so
endif

ifeq ($(findstring BSD,${OSARCH}),BSD)
CFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include 
-L$(CROSS_COMPILE_TARGET)/usr/local/lib
endif

CURLLIBS=$(shell $(CROSS_COMPILE_BIN)curl-config --libs)
ifneq ($(shell if [[ 0x`$(CROSS_COMPILE_BIN)curl-config --vernum` -ge 0x70907 
]]; then echo "OK" ; fi),)
  ifneq (${CURLLIBS},)
APPS+=app_curl.so
  endif
endif

CFLAGS+=-fPIC

#
# If you have UnixODBC you can use ODBC voicemail
# storage
#
# Uncomment to use ODBC storage
#CFLAGS+=-DUSE_ODBC_STORAGE
# Uncomment for extended ODBC voicemail storage
#CFLAGS+=-DEXTENDED_ODBC_STORAGE
# See doc/README.odbcstorage for more information

all: $(APPS)

clean:
rm -f *.so *.o look .depend

%.so : %.o
$(CC) $(SOLINK) -o $@ $<

app_rpt.so : app_rpt.o
$(CC) $(SOLINK) -o $@ $< -ltonezone

install: all
for x in $(APPS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(MODULES_DIR) ; 
done
rm -f $(DESTDIR)$(MODULES_DIR)/app_datetime.so
rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so

app_curl.so: app_curl.o
$(CC) $(SOLINK) -o $@ $< $(CURLLIBS)

app_rxfax.so : app_rxfax.o
$(CC) $(SOLINK) -o $@ $< -lspandsp -ltiff

app_txfax.so : app_txfax.o
$(CC) $(SOLINK) -o $@ $< -lspandsp -ltiff

app_sql_postgres.o: app_sql_postgres.c
$(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o 
app_sql_postgres.o app_sql_postgres.c

app_sql_postgres.so: app_sql_postgres.o
$(CC) $(SOLINK) -o $@ $< -L/usr/local/pgsql/lib -lpq

app_sql_odbc.so: app_sql_odbc.o
$(CC) $(SOLINK) -o $@ $< -lodbc

look:   look.c
$(CC) -pipe -O6 -g look.c -o look -lncurses

ifeq (SunOS,$(shell uname))
app_chanspy.so: app_chanspy.o
$(CC) $(SOLINK) -o $@ $< -lrt
endif


ifneq ($(wildcard .depend),)
 include .depend
endif

depend: .depend

.depend:
../build_tools/mkdep $(CFLAGS) `ls *.c`

env:
env



rtxfax.c


I had to delete a routine that checked for chan_callerid in an old release,

And bingo it complied.

I do not know if it is stable howeer I will test over the weekend...

Alex





> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> David Sampson
> Sent: Friday, September 16, 2005 9:37 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] RxFax/TxFax - Compile Problem
> 
> If I understand correctly you are supposed to patch the 
> Makefile in the apps directory and then run the main 
> Makefile.  I've tried both ways - the patch failed on the 
> main Makefile.  Should I try to make that work?
> 
> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EM

Re: [Asterisk-Users] Seperate Incoming calls on TDM02?

[C. Hatton Humphrey]

> I'm fighting with this right now and I'm hitting a serious frustration
> point - right now all incoming calls are getting handled by the
> from_pstn context which is how it honestly should be according to the
> current conf files.  However when I change the context from from_pstn
> to aa_1 and aa_2 respectively it doesn't change anything in the way
> the system is answering the lines.

Found the source of the problem and it wasn't in the config files,
rather in the way that Asterisk "reloads" when issued that command -
it does NOT rebuild the Zapata channel table.  I had to perform a
complete restart of Asterisk to get the changes (which I made in my
zapata-auto.conf file) into the system.  After I did that everything
started running right as rain!

Thanks for the assistance, hopefully this will make it where people
will see it in the future.

Hatton
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Re: use of automon sequence (was Re: [Asterisk-Users] Call recording between SIP phones)

[Kevin Bockman]

John covici wrote:

OK, I wonder if I have something wrong -- I have the *1 in my
features.conf for the one touch record -- now I called a number, and
when the call was answered flashed the hook and pressed *1 and went
back tothe call, but nothing happpened.  I am using CVS from 8/26 --
is this too old or am I doing something wrong?  I did add wW to the
dial options of the call and that part did work, but there is nothing
in the log about automon.  Also, what kind of file does this produce,
etc?

Thanks in advance for any help.
I'm not sure when it started, but as long as you have it in the example 
features.conf in your asterisk source/configs/features.conf file.


You don't do a flash to do the record.  You press the sequence WHILE YOU 
ARE ON THE CALL.  It will record a file to /var/spool/monitor.  To 
change the filename, you probably have to set the MONITOR_FILENAME 
variable before the dial.



Kevin
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RE: [Asterisk-Users] 7 digit dialing to e.164 format

[Jonathan k. Creasy]

You could name your peers by the full e.164 number associated with them
then parse the area code from that to append to the number they dialed
as a 7 digit number possibly. 

-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Schulte
Sent: Friday, September 16, 2005 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] 7 digit dialing to e.164 format

All, I've asked this once a long time ago and got a vague response, any
suggestions? I'm wanting to convert for example a 7 digit extension
(whether it be via dialplan or agi) to e.164. This is for the sake of
getting everything outbound into e164 format. The issue I see you will
need to append the areacode of the calling party to the 7 digits, from
there adding a +1 is of course easy. example:

Customer dials 5610680, his callerid (from) is +16364424591.. I want to
take the 636 and add it to the 561 number so the result would be
6365610680 .. Any thoughts? I've been looking at the internal ast
commands with no luck. I'm sure people have run into this before, it
would be nice to add a ${e164_fromNPA} variable!!! ;-)

Matt
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[Asterisk-Users] SIP port assignment for user agents registering to Asterisk.

[Steve Lane]

I was wondering if anyone knows why when I register a user
agent like XLite with Asterisk I am noticing that the
port assignment on the “sip show peers” command shows the port to
be different than any of the other user agents. The other user agents are
logging in from different networks from all over the internet. Here is a sample
of my table when I issue the command:

Ipbx1*CLI>sip show peers

 

Name/username
    Host  
  Dyn Nat ACL
Mask
  Port Status

cydalgw1/216.90.228.254    216.90.228.254   255.255.255.255      5060 OK (14 ms)

steve/steve  
  68.113.199.172   D     N 
 255.255.255.255      62784   Unmonitored

hooman7/hooman7   
  68.88.66.248      D 
   N 
 255.255.255.255      5060 UNREACHABLE

 

I am using the xlite soft phone. I
tried other phones but got the same result. Can anyone give me a clue as to
what is going on? I figure my laptop might be screwed up or something. 

 

Steve 






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Re: [Asterisk-Users] Caller Name: Asterisk reading too fast

[Juan Jose Comellas]

Have you tried placing a Wait(1) before Answer() in your dialplan?


On Friday 16 September 2005 11:23, J Thomas wrote:
> I asked my telco to release caller name on the PRI. Earlier they were
> releasing only the phone number.
>
> I still did not see the name, but only the number in caller id. Actually
> I now see number twice. When I inquired with them this is the response I
> got:
>
>   "I ran a trace on your TG.  I see that your switch is
>   picking up the call so fast that it is not able to pick
>   up the name. The name is being sent, but I suspect after
>   it is too late.  This is something that will need to be
>   corrected in your switch.
>
>   I have attached a sample call out of the trace I performed
>   this morning."
>
> They have sent me the trace file.
>
> Is there a way as it is in Asterisk so that it reads the caller name
> properly?
>
> Thanks,
> -- jt
>
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([EMAIL PROTECTED])

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RE: [Asterisk-Users] USB Phones for use with Asterisk

[Kanuri, Seshu \(Company IT\)]

USB phone and NAT - What has USB Phpne got to do with NAT?

USB Phone is just a hardware piece that pipes the audio output from your
softphone.

Your softphone has to take care of that.

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Callum
McGillivray
Sent: Thursday, September 15, 2005 8:41 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] USB Phones for use with Asterisk

Hi all,

I have a question that I was hoping someone could answer for me.

I would like to find a USB phone that works with Asterisk... preferably
one that does not have any issues with NAT.

Can anyone point me to something suitable ?  We are essentially planning
to do a beta for a few hundred customers on our network and would like
them to be able to call one another @ no charge.

Because these are going to be end users who have no idea about Voice /
VOIP stuff... it needs to be extremely easy to use.

Can someone point me in the right direction ?

Cheers,

Callum
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NOTICE: If received in error, please destroy and notify sender.  Sender does 
not waive confidentiality or privilege, and use is prohibited.
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[Asterisk-Users] Caller Name: Asterisk reading too fast

[J Thomas]

I asked my telco to release caller name on the PRI. Earlier they were
releasing only the phone number. 

I still did not see the name, but only the number in caller id. Actually
I now see number twice. When I inquired with them this is the response I
got:

  "I ran a trace on your TG.  I see that your switch is
picking up the call so fast that it is not able to pick
up the name. The name is being sent, but I suspect after
it is too late.  This is something that will need to be
corrected in your switch. 

I have attached a sample call out of the trace I performed
this morning."

They have sent me the trace file.

Is there a way as it is in Asterisk so that it reads the caller name
properly?

Thanks,
-- jt

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[Asterisk-Users] Asterisk don't start

[[EMAIL PROTECTED]]

Asterisk don't running, because show this message


WARNING[1075709024]: cli.c:702 ast_cli_register: Command 'showparkedcalls' 
already registered (or something close enough)
  == Parsing '/etc/asterisk/features.conf': Found
-- Added extension '700' priority 1 to parkedcalls
/usr/sbin/asterisk: relocation error: 
/usr/lib/asterisk/modules/res_features.so: undefined symbol: ast_pthread_create


Thanks!


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[Asterisk-Users] Asterisk don't start

[[EMAIL PROTECTED]]

Asterisk don't running, because show this message


WARNING[1075709024]: cli.c:702 ast_cli_register: Command 'showparkedcalls' 
already registered (or something close enough)
  == Parsing '/etc/asterisk/features.conf': Found
-- Added extension '700' priority 1 to parkedcalls
/usr/sbin/asterisk: relocation error: 
/usr/lib/asterisk/modules/res_features.so: undefined symbol: ast_pthread_create


Thanks!


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Re: [Asterisk-Users] Seperate Incoming calls on TDM02?

[C. Hatton Humphrey]

> > Yeah, in your zapata.conf just give each channel a different context
> > setting.
> 
> It's slightly more complicated with [EMAIL PROTECTED] and/or AMP, you need to 
> use the
> zapata_custom.conf file, instead. You also need to use the
> extensions_custom.conf file, too, though there might be a better way I don't
> know about.

I'm fighting with this right now and I'm hitting a serious frustration
point - right now all incoming calls are getting handled by the
from_pstn context which is how it honestly should be according to the
current conf files.  However when I change the context from from_pstn
to aa_1 and aa_2 respectively it doesn't change anything in the way
the system is answering the lines.

Hatton

zapata.conf:
;
; Zapata telephony interface
;
; Configuration file

[trunkgroups]

[channels]

language=en
context=from-pstn
signalling=fxs_ks
rxwink=300  ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes

usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=no
threewaycalling=no
transfer=no
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no

;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

;Include AMP configs
#include zapata_additional.conf

;Include genzaptelconf configs
#include zapata-auto.conf

zapata_additional.conf is empty
zapata-auto.conf:
; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is intended 
; to be #include-d by /etc/zapata.conf that will include the global settings
;
callerid=asreceived

; Span 1: WCTDM/0 "Wildcard TDM400P REV E/F Board 1" 
; channel 1, WCTDM, inactive.
; channel 2, WCTDM, inactive.
signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 3
context=from_pstn
group=0
channel => 3

signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4
context=from_pstn
group=0
channel => 4
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[Asterisk-Users] Extension Restrictions

[Waldo Rubinstein]

Is it possible to define an extension that is not allowed to make or  
receive calls, unless an agent logs in? Obviously it would require  
that the extension be able to dial the # for the agent to log in.


Thanks,
Waldo

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[Asterisk-Users] broadvoice incoming caller ID is wierd when calling from voipjet

[Paul]

Calling bv from pstn phones my log shows correct caller id number and name

Calling from voipjet with cid set to 10 digit number(207826) my logs 
show bv adding a leading + and setting the name to egypt as a result


Everything else I call from voipjet gets caller id correct

Could be they are using the same carrier for caller ID to be mishandled 
this way.


Anyone else noticed this behavior?

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Re: [Asterisk-Users] Call Forward - 7940 Asterisk - Help

[c waddy]

Hi
 
I am only interested in using SIP, i would like to setup simple call forwarding either by the phone or Asterisk, it is a common legacy PBX feature and i am sure it is available from Asterisk.
 
What would be the best way to do it?
 
Is it hard coded into Asterisk?
 
Why do the SIP Cisco 7940's call forward feature not work with Asterisk?
 
 
On 9/16/05, Stefan Gofferje <[EMAIL PROTECTED]> wrote:
Hi DerekDerek Conniffe schrieb:> Hi Stefan,>> Has chan_sccp matured a lot?  I remember (maybe a year ago or so) that I
> had a lot of problems with chan_sccp and chan_skinny (one thing is that> I remember with chan_sccp is that the VM button didn't work and trying> to answer multiple incoming calls tended to make the phone go into a
> weird state where I had to power cycle it to get it back right again).> I upgraded the phoen to SIP and never looked back - but then maybe I'm> not getting all the fetures like call forwarding (which sounds very
> useful to me)..I am talking about a complete rewrite. About half a year ago, Sergio waswriting patches for Julien's chan_sccp (chan-sccp.sf.net).For some reason, Sergio decided to do a project split and started his
own chan_sccp. After a few weeks he stopped patching Julien's work anddid a complete rewrite. Meanwhile, chan_sccp (chan-sccp.berlios.de or
chan-sccp.org) is not only stable but also very feature reach!Sergio does the coding and me and some other people do very heavytesting and the project is making great progress.Highlights are:- Line status monitoring on 7960/7914 (you see what other - not only
  SCCP - extensions are doing)- Good hardware support (7940/7960/7914 almost 100%, 7905 almost  90%, 7920 almost 80%, 7970 currently under heavy development)- Support for call waiting, call forwarding, ...
- Intercom / autoanswer - controlled by dial applicationand a lot more...Have a look at http://chan-sccp.org/ ...Slan go foil,Stefan--(o_   Stefan Gofferje| SCLT, MCP
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Re: [Asterisk-Users] Call Forward - 7940 Asterisk - Help

[Stefan Gofferje]

Hi Derek

Derek Conniffe schrieb:

Hi Stefan,

Has chan_sccp matured a lot?  I remember (maybe a year ago or so) that I 
had a lot of problems with chan_sccp and chan_skinny (one thing is that 
I remember with chan_sccp is that the VM button didn't work and trying 
to answer multiple incoming calls tended to make the phone go into a 
weird state where I had to power cycle it to get it back right again).
I upgraded the phoen to SIP and never looked back - but then maybe I'm 
not getting all the fetures like call forwarding (which sounds very 
useful to me)..


I am talking about a complete rewrite. About half a year ago, Sergio was 
writing patches for Julien's chan_sccp (chan-sccp.sf.net).
For some reason, Sergio decided to do a project split and started his 
own chan_sccp. After a few weeks he stopped patching Julien's work and 
did a complete rewrite. Meanwhile, chan_sccp (chan-sccp.berlios.de or 
chan-sccp.org) is not only stable but also very feature reach!
Sergio does the coding and me and some other people do very heavy 
testing and the project is making great progress.


Highlights are:
- Line status monitoring on 7960/7914 (you see what other - not only
  SCCP - extensions are doing)
- Good hardware support (7940/7960/7914 almost 100%, 7905 almost
  90%, 7920 almost 80%, 7970 currently under heavy development)
- Support for call waiting, call forwarding, ...
- Intercom / autoanswer - controlled by dial application

and a lot more...

Have a look at http://chan-sccp.org/ ...

Slan go foil,
Stefan

--
 (o_   Stefan Gofferje| SCLT, MCP
 //\   Reg'd Linux User #247167   | VCP #2263
 V_/_  Heckler & Koch - the original point and click interface

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