[Asterisk-Users] "Call progress" when making call using ATA via iconnecthere

[Brian Capouch]

Just curious as to whether or not there's anyone else out there who is 
using iconnecthere behind a completely-NATted asterisk system.

My system, using code from approx a week ago, works just fine with 
iconnect, except that I get no "call progress" information (if that's 
the correct terminology).  I dial, wait a bit, and if the person on the 
other end picks up I just suddenly hear them, without hearing any ring 
tones first.

I fake this with "r" in the context, but I would much rather have it be 
meaningful, since this way I get ringing tones even when iconnect has 
sent one of its infernal occasional "480" messages.

Thx.

B.

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RE: [Asterisk-Users] iconnect & caller ID

[Steve Radich]

Mark - I beg to differ.  Generally callerid works with Deltathree; but
sometimes they seem to reject it / mess it up.

I (see arhives) set callerid to the extension dialed at work based on giving
friends one extension, customers another, really important people another -
About 75% of the time deltathree passes my coded caller id (it's 4
characters, so definitely not a valid caller id) and that appears on my cell
phone for the call. 

About 25% of the time their local POP # shows up (I guess), a 202-xxx number
(Washington, DC, which is the area I'm both in and calling to).

I basically have a dial plan for my extensions of ring Zap extension, SNOM
phone via SIP and my cell phone via SIP through IConnect (to avoid dialing
out from Asterisk on a normal line, and to have an excuse to play with more
VoIP stuff).

My exten is (numbers changed)

Exten => 911,1,SetCallerID,911
Exten => 911,2,Dial,Zap/xx&SIP/[EMAIL PROTECTED]&SIP/[EMAIL PROTECTED]|30|t
Exten => 911,3,VoiceMail,.

My sip.conf has:

[iconnect]
type=friend
username=#
password=
host=213.137.73.178
callerid="1-301-345-6789"

As I said, about 75% of the time IConnect passes it through.  I've kind of
guessed it's extra circuits they have don't handle caller id right - not
Asterisk as it seems to work always during off peak hours and about 1/2 the
time during "normal" business hours.

Steve Radich - Colocation / Virtual Dedicated / Dedicated Servers 
BitShop, Inc. - http://www.bitshop.com - $149/month colo special


-Original Message-
From: Mark Spencer [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, March 12, 2003 11:49 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] iconnect & caller ID

The friend would only happen if the "From: " was iconnect.  Unfortuantely
SIP does not differentiate a user from Caller*ID. The only way to make the
peer match would be if we matched the peer based on IP address.

Mark

On Wed, 12 Mar 2003, Jim Archer wrote:

> Hi All...
>
> We have found that the caller ID information presented to some one we call
> from Asterisk using iconenct is not predictable.  The caller ID will be
> unavailable or else deltathree.  I have in sip.conf:
>
> [iconnect]
> type=friend
> username=41306756
> password=2264
> host=natrelay.deltathree.com
> callerid="My COmpany, Inc." <1 401 nnn >
> txgain = 5.0;
> rxgain = 5.0;
>
> Has anyone else seen this problem?
>
>
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[Asterisk-Users] about new sip.conf options

[Anton Yurchenko]

Hello,

I see on the mailing all kinds of new sip.conf options appearing like 
dtmfmode , etc. Could anyone describe them, anything that is not in the 
version 2 Asterisk Handbook, the sip.conf.pdf.

Thanks,

--

Anton Yurchenko<[EMAIL PROTECTED]>
Digital Generation
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Re: [Asterisk-Users] Help With Music On Hold

[Walt Davis]

No but how about I try installing it? Thanks mark!

> Did you remember to move to mpg123 and not use mpg321 which is often
> aliased to mpg123?
>
> Mark
>
> On Wed, 12 Mar 2003, Walt Davis wrote:
>
>> It used to work just fine and I can only think of three possibilities:
>>
>> 1) Moved asterisk onto a different server and that server does not
>> have a sound card?
>>
>> 2) Rebuilt and hardened asterisk on a Mandrake 9.0 distro, so perhaps
>> I left out some modules or something.
>>
>> 3) Upgraded from the developer kit light to a single span T1.
>>
>> Can anybody confirm or deny these as possibilities?
>>
>> > Asterisk -rvvv
>> > -- Starting simple switch on 'Zap/7-1'
>> > -- Executing MusicOnHold("Zap/7-1", "") in new stack
>> > -- Started music on hold, class 'default', on Zap/7-1
>> > -- Stopped music on hold on Zap/7-1
>> >   == Spawn extension (home, 6, 1) exited non-zero on 'Zap/7-1'
>> > -- Hungup 'Zap/7-1'
>> >
>> > As you can see MusicOnHold is starting and stopping but I dont hear
>> anything, just dead silence.
>> >
>> > My music on hold config:
>> > default => mp3:/var/lib/asterisk/mohmp3, -z.
>> >
>> > All MP3 Files in this dirctory have global read access.
>> >
>> > I'm getting a lot of these in the debug log:
>> > Mar 11 14:12:10 DEBUG[3076]: File res_musiconhold.c, Line 240
>> > (monmp3thread): Read 100 bytes of audio while expecting 640
>> >
>> > and these in the Message log:
>> > Mar 11 14:10:24 WARNING[1024]: File chan_oss.c, Line 419
>> > (soundcard_init)
>> >
>> > Any ideas?
>> >
>> > --
>> > Walt Davis
>> > www.waltdavis.net
>> >
>> >
>> > ___
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>>
>>
>>
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>
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Re: [Asterisk-Users] Build a complex IVR?

[Jim Gottlieb]

On 2003-03-13 at 15:39, "it" <[EMAIL PROTECTED]> wrote:

>I would like to know if Asterisk could be used to build a IVR with
>complex flow?

Yes, we have done this.  We use menu files that define what each
keypress does so we can build complex menus with full flexibility.

A short example (since you asked):

default = 0
prompt
{
name = 3046
}

0
{
action = debitcard
declines = debitcard-declines
pin-prompt = 5122
number-prompt = 5123
thankyou = 5124
}

1
{
action = debitcardbalance
declines = debitcard-declines
pin-prompt = 5122
}

2
{
action = debitcardgenerate
declines = debitcard-declines
cardtype = 420
}

3
{
action = vmail
userno = 3641234
}

Once it goes into vmail, we have a whole hard-coded voicemail app that
is basically modeled on the Centigram VoiceMemo system including
message sending, forwarding, paging, transferring to attendant.
Likewise for other functions.  It also does queries into our backend
billing database and credit card charging systems.  

I wish I could convince my company to release the source code, but I
don't think that's ever going to happen, as they consider this code to
be their strategic advantage.  It's been developed over many years and
used to run with Dialogic hardware but we have recently ported it to
asterisk.

So yes, you can build almost any IVR application under asterisk.  Not
only is the hardware one-eighth the cost, but the tech support is far
far superior.  We used to wait years for Dialogic to fix bugs.  And
best of all, asterisk provides all the core functions so you can spend
your time on your custom features and not on coding the basic
telephony functions, message playing, etc.
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[Asterisk-Users] Help using Clarisys USB phone with Asterisk?

[Roy Sigurd Karlsbakk]

hi

I just received two Clarisys USB phones. These look pretty straight forward, 
but I've no idea of how to configure them with gnophone or anything.

Any good tips? Anyone that knows the tech specs?

roy
-- 
Roy Sigurd Karlsbakk, Datavaktmester
ProntoTV AS - http://www.pronto.tv/
Tel: +47 9801 3356

Computers are like air conditioners.
They stop working when you open Windows.


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Re: [Asterisk-Users] chan_capi version 0.1.0 released

[Roy Sigurd Karlsbakk]

> hi alaw (and now also ulaw) folks,
>
> version 0.1.0 is out now. thanks to Bicster for beating
> eicon capi to support ulaw and echo cancellation!!
>
> http://www.junghanns.net/chan_capi.html

Can I suggest something?

If you package the tar balls with version number in the directory name, such 
as recent linux kernel versions do it, it's a LOT easier to unpack, test, and 
roll back to older versions.

[EMAIL PROTECTED] /usr/src># tar tzf chan_capi.0.1.0.tar.gz | head -2
chan_capi/
chan_capi/Makefile

if this could be changed to

[EMAIL PROTECTED] /usr/src># tar tzf chan_capi.0.1.0.tar.gz | head -2
chan_capi-0.1.0/
chan_capi-0.1.0/Makefile

I'd be very grateful ;-)

and - do you have this in CVS? Do you need CVS?

thanks for the good work

roy
-- 
Roy Sigurd Karlsbakk, Datavaktmester
ProntoTV AS - http://www.pronto.tv/
Tel: +47 9801 3356

Computers are like air conditioners.
They stop working when you open Windows.


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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Liaan van der Merwe]

Hallo
Thanks very much for great work
Just quick Q.
Why would asterisk keep on telling me :" NOTICE[8192]: File chan_capi.c,
Line 1984 (load_module): CAPI not installed!" when starting???
Thanks


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
Junghanns
Sent: 12 Maart 2003 19:09
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] chan_capi version 0.1.0 released

hi alaw (and now also ulaw) folks,

version 0.1.0 is out now. thanks to Bicster for beating
eicon capi to support ulaw and echo cancellation!!

http://www.junghanns.net/chan_capi.html

regards
kapejod

--
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705390
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]


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R: [Asterisk-Users] Different ring pattern for extensions possible?

[Matteo Brancaleoni]

But where these patterns can bve found/defined?

(BUG: using Dial(Zap/Xr3) make * crash ;< )

Matteo.

> -Messaggio originale-
> Da: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] Per conto di 
> Mike Reiling
> Inviato: giovedì 13 marzo 2003 1.31
> A: [EMAIL PROTECTED]
> Oggetto: Re: [Asterisk-Users] Different ring pattern for 
> extensions possible?
> 
> 
> look at the 'r' option for the Dial command.  Example exten => 
> 333,1,Dial(Zap/1r2)
> 
> --Mike
> 
> On Wednesday, March 12, 2003, at 04:03  PM, Jim Archer wrote:
> 
> > Hi All...
> >
> > Is it possible to change the ringing pattern of an 
> extension?  I would
> > like to make my extension ring differently based upon what number 
> > (distinctive ring or different channel) I am being called from.
> >
> > The best use of this I think would be to let internal calls ring
> > differently than external calls.  Also, I would like to 
> give my family 
> > a number that only they know and that rings my extension in 
> a special 
> > way.
> >
> > I would appreciate any suggestions! Thanks!
> >
> > ___
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> >
> 
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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Klaus-Peter Junghanns]

Hi Liaan,

looks like you dont have capi4linux installed. What card
are you trying to use?

regards
kapejod


-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705390
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]



Am Don, 2003-03-13 um 10.23 schrieb Liaan van der Merwe:
> Hallo
> Thanks very much for great work
> Just quick Q.
> Why would asterisk keep on telling me :" NOTICE[8192]: File chan_capi.c,
> Line 1984 (load_module): CAPI not installed!" when starting???
> Thanks
> 
> 

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[Asterisk-Users] Beginning of voicemail missed by sip phone

[T Aksoy]

Hi,
 
We are testing a number of sip phones from 
different manufacturers. With one phone in particular, when I dial the asterisk 
voicemail, it misses around half a second from the beginning of the 
announcement. I don't have this problem with the snom 200 or 100.
 
Does anyone know why this happens? Is it a sync 
issue? How do I delay the start of the voicemail announcement? (Maybe that will 
fix the problem).
 
Thanks
Tan Aksoy
Telappliant Solutions
 

Re: [Asterisk-Users] Build a complex IVR?

[it]

Hi,Jim,
Thank you very much for your reply. But I think the menu file you
designed must be interpreted by another program writen by your but not
asterisk.

- Original Message -
From: "Jim Gottlieb" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, March 13, 2003 12:21 AM
Subject: Re: [Asterisk-Users] Build a complex IVR?


> On 2003-03-13 at 15:39, "it" <[EMAIL PROTECTED]> wrote:
>
> >I would like to know if Asterisk could be used to build a IVR
with
> >complex flow?
>
> Yes, we have done this.  We use menu files that define what each
> keypress does so we can build complex menus with full flexibility.
>
> A short example (since you asked):
>
> default = 0
> prompt
> {
> name = 3046
> }
>
> 0
> {
> action = debitcard
> declines = debitcard-declines
> pin-prompt = 5122
> number-prompt = 5123
> thankyou = 5124
> }
>
> 1
> {
> action = debitcardbalance
> declines = debitcard-declines
> pin-prompt = 5122
> }
>
> 2
> {
> action = debitcardgenerate
> declines = debitcard-declines
> cardtype = 420
> }
>
> 3
> {
> action = vmail
> userno = 3641234
> }
>
> Once it goes into vmail, we have a whole hard-coded voicemail app that
> is basically modeled on the Centigram VoiceMemo system including
> message sending, forwarding, paging, transferring to attendant.
> Likewise for other functions.  It also does queries into our backend
> billing database and credit card charging systems.
>
> I wish I could convince my company to release the source code, but I
> don't think that's ever going to happen, as they consider this code to
> be their strategic advantage.  It's been developed over many years and
> used to run with Dialogic hardware but we have recently ported it to
> asterisk.
>
> So yes, you can build almost any IVR application under asterisk.  Not
> only is the hardware one-eighth the cost, but the tech support is far
> far superior.  We used to wait years for Dialogic to fix bugs.  And
> best of all, asterisk provides all the core functions so you can spend
> your time on your custom features and not on coding the basic
> telephony functions, message playing, etc.
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

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[Asterisk-Users] DTMF settings

[T Aksoy]

Hi,
 
We are trialling a new sip phone and I am having 
difficulty getting the voicemail to recognise the dtmf tones. I have tried 
setting the dtmf mode on the phone to out-of-band but still no 
luck.
 
How can I determine which end (asterisk or phone) 
has the problem?

ThanksTanTelappliant 
Solutions

[Asterisk-Users] (no subject)

[Lars Abelius]

unsubscribe
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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Liaan van der Merwe]

Hallo
To be honest.. no card as yet.
Truying to find out whether external TA will work.
I'll install latest i4l and see what happens
Thanks



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
Junghanns
Sent: 13 Maart 2003 11:54
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released

Hi Liaan,

looks like you dont have capi4linux installed. What card
are you trying to use?

regards
kapejod


--
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705390
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]



Am Don, 2003-03-13 um 10.23 schrieb Liaan van der Merwe:
> Hallo
> Thanks very much for great work
> Just quick Q.
> Why would asterisk keep on telling me :" NOTICE[8192]: File chan_capi.c,
> Line 1984 (load_module): CAPI not installed!" when starting???
> Thanks
>
>

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Re: [Asterisk-Users] SIP and MWI 7960

[billp]

On Wed, Mar 12, 2003 at 11:33:31PM -0600, Mark Spencer wrote:
> > Is anyone using MWI on 7960's when Asterisk is ONLY being used
> > for voicemail, and not for a gatekeeper?
> >
> > If anyone has MWI working successfully with 7960's, would it
> > be possible to get a dump of a successful NOTIFY message that
> > turns a light on/off?
> 
> Just put "mailbox=1234" in your sip friend/peer for the phone.

I am not using Asterisk as the gatekeeper, only for voicemail.
My phones do not register with Asterisk.

Or am I missing something here?

This is my sip.conf:

  [general]
  port = 5110 ; Port to bind to
  bindaddr = 206.165.202.3; Address to bind to
  context = default   ; Default for incoming calls

bill
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[Asterisk-Users] splitting the asterisk list?

[Roy Sigurd Karlsbakk]

hi

Is it only me, or is the asterisk mailing list growing rapidly?

I've got an idea, although I don't know how good it is ;-)

how about splitting the asterisk list between good old telephony with channel 
banks and telephones, and IP based telephony?

roy
-- 
Roy Sigurd Karlsbakk, Datavaktmester
ProntoTV AS - http://www.pronto.tv/
Tel: +47 9801 3356

Computers are like air conditioners.
They stop working when you open Windows.


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[Asterisk-Users] X100P in Europe/Austria

[Klaus Darilion]

Hello!

Does the X100P card supports the european phone standards, especially
the one in Austria? Does someone have ever tried it and succeeded?

Regards,
Klaus

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R: [Asterisk-Users] X100P in Europe/Austria

[Matteo Brancaleoni]

yes! works w/o problems.

Matteo in italy ;-)


> -Messaggio originale-
> Da: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] Per conto di 
> Klaus Darilion
> Inviato: giovedì 13 marzo 2003 11.49
> A: [EMAIL PROTECTED]
> Oggetto: [Asterisk-Users] X100P in Europe/Austria
> 
> 
> Hello!
> 
> Does the X100P card supports the european phone standards, 
> especially the one in Austria? Does someone have ever tried 
> it and succeeded?
> 
> Regards,
> Klaus
> 
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Re: [Asterisk-Users] X100P in Europe/Austria

[Michael Bielicki]

It works in UK, DE and in Pakistan.

On Thursday 13 March 2003 10:49, Klaus Darilion shaped the electrons to say:
> Hello!
>
> Does the X100P card supports the european phone standards, especially
> the one in Austria? Does someone have ever tried it and succeeded?
>
> Regards,
> Klaus
>
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-- 
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Managing Director
TAAN Consultants Ltd
http://www.global-gateway.net/

--

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Re: [Asterisk-Users] X100P in Europe/Austria

[Florian Overkamp]

Hi

At 11:49 13-3-2003 +0100, you wrote:
Does the X100P card supports the european phone standards, especially
the one in Austria? Does someone have ever tried it and succeeded?
Yes it does, although I don't think its certified for use in australia yet...

Florian

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Re: [Asterisk-Users] splitting the asterisk list?

[Florian Overkamp]

At 11:36 13-3-2003 +0100, you wrote:
Is it only me, or is the asterisk mailing list growing rapidly?

I've got an idea, although I don't know how good it is ;-)

how about splitting the asterisk list between good old telephony with channel
banks and telephones, and IP based telephony?
Hmm, I think that would suck for those with mixed mode environments (such 
as me :-).

Best regards,
Florian
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[Asterisk-Users] CAPI errors...

[Roy Sigurd Karlsbakk]

hi

I keep getting these errors all the time:

ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
DATA_B3_REQ (error=0x1102, datalen=160)
ERROR[3076]: File chan_capi.c, Line 1081 (pipe_msg): error sending 
DATA_B3_RESP (error=0x1102)
ERROR[3076]: File chan_capi.c, Line 1081 (pipe_msg): error sending 
DATA_B3_RESP (error=0x1102)
ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
DATA_B3_REQ (error=0x1102, datalen=160)
ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
DATA_B3_REQ (error=0x1102, datalen=160)
ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
DATA_B3_REQ (error=0x1102, datalen=160)
ERROR[3076]: File chan_capi.c, Line 1081 (pipe_msg): error sending 
DATA_B3_RESP (error=0x1102)
ERROR[3076]: File chan_capi.c, Line 1081 (pipe_msg): error sending 
DATA_B3_RESP (error=0x1102)
ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
DATA_B3_REQ (error=0x1102, datalen=160)
ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
DATA_B3_REQ (error=0x1102, datalen=160)

anyone that might know why? Klaus?

roy
-- 
Roy Sigurd Karlsbakk, Datavaktmester
ProntoTV AS - http://www.pronto.tv/
Tel: +47 9801 3356

Computers are like air conditioners.
They stop working when you open Windows.


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Re: [Asterisk-Users] X100P in Europe/Austria

[Iain Stevenson]

The preceding comments probably apply to direct analogue PSTN connections. 
You may have problems if the line you are connecting to is from a PBX or 
ISDN terminal adapter.

 Iain



--On Thursday, March 13, 2003 11:49 am +0100 Klaus Darilion 
<[EMAIL PROTECTED]> wrote:

Hello!

Does the X100P card supports the european phone standards, especially
the one in Austria? Does someone have ever tried it and succeeded?
Regards,
Klaus
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Re: [Asterisk-Users] splitting the asterisk list?

[Gary]

On Thu, 13 Mar 2003 12:20:03 +0100, Florian Overkamp wrote:

>At 11:36 13-3-2003 +0100, you wrote:
>>Is it only me, or is the asterisk mailing list growing rapidly?
>>
>>I've got an idea, although I don't know how good it is ;-)
>>
>>how about splitting the asterisk list between good old telephony with channel
>>banks and telephones, and IP based telephony?
>
>Hmm, I think that would suck for those with mixed mode environments (such 
>as me :-).
>
>Best regards,
>Florian

I tend to agree, it would be nice if their was a message board
enviroment which also sent out emails to the list labeled from the
various forums, but then I don't know if anyone has done one like that,
of course messages on the list have to migrated back into the correct
forum as well .

Gary
.



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Re: [Asterisk-Users] splitting the asterisk list?

[Michael Bielicki]

yeah what would we all carrier guys do who have completely mixed environments 
? create a asterisk-carrier and a asterisk-oem and a asterisk-sysintegrator 
list ?

On Thursday 13 March 2003 11:20, Florian Overkamp shaped the electrons to say:
> At 11:36 13-3-2003 +0100, you wrote:
> >Is it only me, or is the asterisk mailing list growing rapidly?
> >
> >I've got an idea, although I don't know how good it is ;-)
> >
> >how about splitting the asterisk list between good old telephony with
> > channel banks and telephones, and IP based telephony?
>
> Hmm, I think that would suck for those with mixed mode environments (such
> as me :-).
>
> Best regards,
> Florian
>
> ___
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-- 
Michael Bielicki
Managing Director
TAAN Consultants Ltd
http://www.global-gateway.net/

--

This correspondence is for the named person's use only. It may contain
confidential or legally privileged information or both. No confidentiality
or privilege is waived or lost by any mistransmission. If you receive this
correspondence in error, please immediately delete it from your system and
notify the sender. You must not disclose, copy or rely on any part of this
correspondence if you are not the intended recipient.

Any opinions expressed in this message are those of the individual sender.
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[Asterisk-Users] Asterisk, SIP, and Registrations

[Steve Woolley]

I am very new to asterisk and SIP so these questions may be fairly
basic.
I have setup an asterisk system based on Red Hat 8.0 Linux with a few
analog cards from Digium (X101P's). I have also purchased a SNOM 200
phone and a couple of soft SIP clients for windows. Everything (so far)
is working nicely.

As I understand it, the asterisk server performs as a SIP proxy. The
sip.conf file sets up  the SIP user agents (clients) connections --
whether they are dynamically addressed, are they "friendly", etc. The
name given in the [context] is what is used by the extensions.conf file
to associate an extension with this user agent.

 Example:
   ;
   ; SIP Configuration for Asterisk
   ;
   [general]
   port = 5060 ; Port to bind to
   bindaddr = 172.16.1.155 ; Address to bind to
   context = default   ; Default for incoming calls
   ;
   [ext]
   type=friend
   secret=1234
   host=dynamic
   defaultip=172.16.14.1

   ;
   ; excerpt from extensions.conf for Asterisk
   ;
   exten => ,1,Dial,SIP/ext|20
   exten => ,2,Voicemail,u
   exten => ,102,Voicemail,b

Now based on my ability to configure the user agent(s), which is
sometimes difficult because it seems every user agent vendor seems to
have a slightly different SIP vocabulary and sparse documentation, I
point my SIP clients at asterisk, dial the extension, and it works.

Here are my questions:

1) What purpose does the "defaultip" entry in the sip.conf file serve if
it is set to host=dynamic?


2) If my asterisk server has just one IP address, can I use bindaddr =
0.0.0.0 instead of specifically setting the IP address?

3) I assume the use of username and password in the sip.conf is strictly
for the ability to authenticate a user agent to the asterisk server --
so some yahoo just doesn't stick any old user agent on your network?

4) The BIG question: it is my understanding of SIP that the really cool
feature (of SIP) is the ability to register a user such as
(sip:[EMAIL PROTECTED]) at the user agent level and some type of
registration service would associate sip:[EMAIL PROTECTED] with the
user agent (example: [EMAIL PROTECTED]). All the SIP clients I have dealt
with have a line for "register as" with username and password. 

  Can asterisk as a registration server?

  I noticed "sip show registrations" command inside asterisk, but no
entries ever show up?

  If asterisk can act as a registration server, how is the
feature/service started and configured?

  
Thanks for some insight.


Steve Woolley
[EMAIL PROTECTED]

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Re: [Asterisk-Users] splitting the asterisk list?

[Roy Sigurd Karlsbakk]

join both

On Thursday 13 March 2003 12:57, Michael Bielicki wrote:
> yeah what would we all carrier guys do who have completely mixed
> environments ? create a asterisk-carrier and a asterisk-oem and a
> asterisk-sysintegrator list ?
>
> On Thursday 13 March 2003 11:20, Florian Overkamp shaped the electrons to 
say:
> > At 11:36 13-3-2003 +0100, you wrote:
> > >Is it only me, or is the asterisk mailing list growing rapidly?
> > >
> > >I've got an idea, although I don't know how good it is ;-)
> > >
> > >how about splitting the asterisk list between good old telephony with
> > > channel banks and telephones, and IP based telephony?
> >
> > Hmm, I think that would suck for those with mixed mode environments (such
> > as me :-).
> >
> > Best regards,
> > Florian
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Roy Sigurd Karlsbakk, Datavaktmester
ProntoTV AS - http://www.pronto.tv/
Tel: +47 9801 3356

Computers are like air conditioners.
They stop working when you open Windows.


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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Liaan van der Merwe]

I just installed all the latest isdn4linux stuff..
Still same error
Maybe asterisk wont work with external isdn devices.. any ideas?
Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
Junghanns
Sent: 13 Maart 2003 11:54
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released

Hi Liaan,

looks like you dont have capi4linux installed. What card
are you trying to use?

regards
kapejod


--
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705390
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]



Am Don, 2003-03-13 um 10.23 schrieb Liaan van der Merwe:
> Hallo
> Thanks very much for great work
> Just quick Q.
> Why would asterisk keep on telling me :" NOTICE[8192]: File chan_capi.c,
> Line 1984 (load_module): CAPI not installed!" when starting???
> Thanks
>
>

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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Sander Striker]

> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Liaan van der
> Merwe
> Sent: Thursday, March 13, 2003 1:20 PM

> I just installed all the latest isdn4linux stuff..
  ^^

You need capi4linux, not isdn4linux (although it is part of the
isdn4linux CVS store).

> Still same error
> Maybe asterisk wont work with external isdn devices.. any ideas?

What device are you trying to use?

A bit more info would be helpfull.

Sander
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Re: [Asterisk-Users] SIP and MWI 7960

[James Sizemore]

Message waiting indication work fine you
just need to set up DDNS for phones Asterisks
Needs to know how to reach the phones!
[2114]
type=peer
username=2114
insecure=yes   
canreinvite=no 
context=default
mailbox=2114
host=SIP003094C274B3.bna01.isdn.net

You will need to have no voice mail be for
you start to test this.
Also your phones control port needs to be
5060, Asterisk will not use another port.
billp wrote:

Two issues-- is anyone using Asterisk as a gatekeeper with
cisco 7960 phones and cisco gateways?  Experiences, thoughts,
etc appreciated.  If anyone has moved from/to ser to/from 
Asterisk, I would be interested in hearing experiences...

We have been trying to get message waiting indication working
on our 7960's without luck.
Is anyone using MWI on 7960's when Asterisk is ONLY being used
for voicemail, and not for a gatekeeper?
If anyone has MWI working successfully with 7960's, would it
be possible to get a dump of a successful NOTIFY message that
turns a light on/off?
thanks
bill
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Re: [Asterisk-Users] Cisco 7960

[James Sizemore]

I send example XML files a week or so ago, as well as
an impotent  line for you dhcp server.
Mike Reiling wrote:

Anyone know if it is possible to load your own XML scripts on to the 
phone, bypassing the Cisco CallManager?  I am still waiting for my 
phone to arrive, but I have been playing with Cisco's phone services 
emulator, and that doesn't seem to like anything I pass to it.

If it is possible, anyone want to share any sample scripts they have.

--Mike

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Re: [Asterisk-Users] CAPI errors...

[Klaus-Peter Junghanns]

Morning Roy,

capi says that the send queue is full. what did you stick in there? ;-)

regards
kapejod

Am Don, 2003-03-13 um 12.35 schrieb Roy Sigurd Karlsbakk:
> hi
> 
> I keep getting these errors all the time:
> 
> ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
> DATA_B3_REQ (error=0x1102, datalen=160)
> ERROR[3076]: File chan_capi.c, Line 1081 (pipe_msg): error sending 
> DATA_B3_RESP (error=0x1102)
> ERROR[3076]: File chan_capi.c, Line 1081 (pipe_msg): error sending 
> DATA_B3_RESP (error=0x1102)
> ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
> DATA_B3_REQ (error=0x1102, datalen=160)
> ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
> DATA_B3_REQ (error=0x1102, datalen=160)
> ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
> DATA_B3_REQ (error=0x1102, datalen=160)
> ERROR[3076]: File chan_capi.c, Line 1081 (pipe_msg): error sending 
> DATA_B3_RESP (error=0x1102)
> ERROR[3076]: File chan_capi.c, Line 1081 (pipe_msg): error sending 
> DATA_B3_RESP (error=0x1102)
> ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
> DATA_B3_REQ (error=0x1102, datalen=160)
> ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
> DATA_B3_REQ (error=0x1102, datalen=160)
> 
> anyone that might know why? Klaus?
> 
> roy
> -- 
> Roy Sigurd Karlsbakk, Datavaktmester
> ProntoTV AS - http://www.pronto.tv/
> Tel: +47 9801 3356
> 
> Computers are like air conditioners.
> They stop working when you open Windows.
> 
> 
> ___
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Re: [Asterisk-Users] chan_capi version 0.1.0 released

[Michael Bielicki]

You are making a basic mistake here. External TA's are treated by linux like 
modems. isdn4linux faq tells you that explicitly.

It won't work.

On Thursday 13 March 2003 12:19, Liaan van der Merwe shaped the electrons to 
say:
> I just installed all the latest isdn4linux stuff..
> Still same error
> Maybe asterisk wont work with external isdn devices.. any ideas?
> Thanks
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
> Junghanns
> Sent: 13 Maart 2003 11:54
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released
>
> Hi Liaan,
>
> looks like you dont have capi4linux installed. What card
> are you trying to use?
>
> regards
> kapejod
>
>
> --
> Klaus-Peter Junghanns
>
> CEO,CTO
> Junghanns.NET GmbH
> Breite Strasse 13 - 12167 Berlin - Germany
> fon:+49 30 79705390
> fax:+49 30 79705391
> iaxtel: 1-700-157-8753
> email:  [EMAIL PROTECTED]
>
> Am Don, 2003-03-13 um 10.23 schrieb Liaan van der Merwe:
> > Hallo
> > Thanks very much for great work
> > Just quick Q.
> > Why would asterisk keep on telling me :" NOTICE[8192]: File chan_capi.c,
> > Line 1984 (load_module): CAPI not installed!" when starting???
> > Thanks
>
> ___
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> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ___
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> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Michael Bielicki
Managing Director
TAAN Consultants Ltd
http://www.global-gateway.net/

--

This correspondence is for the named person's use only. It may contain
confidential or legally privileged information or both. No confidentiality
or privilege is waived or lost by any mistransmission. If you receive this
correspondence in error, please immediately delete it from your system and
notify the sender. You must not disclose, copy or rely on any part of this
correspondence if you are not the intended recipient.

Any opinions expressed in this message are those of the individual sender.
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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Klaus-Peter Junghanns]

Hi,

this is funny :)
you expect it to work with no card 
i dont think that there are external TAs with capi 2.0
linux drivers. isdn4linux wont help much...you want
capi4linux, but without a card it's not much fun ;-)

regards
kapejod


Am Don, 2003-03-13 um 11.11 schrieb Liaan van der Merwe:
> Hallo
> To be honest.. no card as yet.
> Truying to find out whether external TA will work.
> I'll install latest i4l and see what happens
> Thanks
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
> Junghanns
> Sent: 13 Maart 2003 11:54
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released
> 
> Hi Liaan,
> 
> looks like you dont have capi4linux installed. What card
> are you trying to use?
> 
> regards
> kapejod
> 
> 
> --
> Klaus-Peter Junghanns
> 
> CEO,CTO
> Junghanns.NET GmbH
> Breite Strasse 13 - 12167 Berlin - Germany
> fon:+49 30 79705390
> fax:+49 30 79705391
> iaxtel: 1-700-157-8753
> email:  [EMAIL PROTECTED]
> 
> 
> 
> Am Don, 2003-03-13 um 10.23 schrieb Liaan van der Merwe:
> > Hallo
> > Thanks very much for great work
> > Just quick Q.
> > Why would asterisk keep on telling me :" NOTICE[8192]: File chan_capi.c,
> > Line 1984 (load_module): CAPI not installed!" when starting???
> > Thanks
> >
> >
> 
> ___
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> 
> ___
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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Liaan van der Merwe]

Oops... see. I need to learn bit more about this whole capi thing.
It will be a planet external isdn TA.. capi2.0 compatible.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Sander Striker
Sent: 13 Maart 2003 14:36
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released

> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Liaan van der
> Merwe
> Sent: Thursday, March 13, 2003 1:20 PM

> I just installed all the latest isdn4linux stuff..
  ^^

You need capi4linux, not isdn4linux (although it is part of the
isdn4linux CVS store).

> Still same error
> Maybe asterisk wont work with external isdn devices.. any ideas?

What device are you trying to use?

A bit more info would be helpfull.

Sander
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Re: [Asterisk-Users] E1 with R2 signaling

[Steve Underwood]

OK, let me rephrase that.

MFC/R2 isn't "a" protocol. Its a whole family - one for almost every 
country, and two for Colombia. :-)

Which variant do you need? That is, which country are you trying to install something in?

Regards,
Steve


Claudio Aznar wrote:

The most common R2 version, MFC R2. How are working in this project ?

Regards,
Peter
On Tue, 11 Mar 2003 21:43:46 +0800
Steve Underwood <[EMAIL PROTECTED]> wrote:
 

Claudio Aznar wrote:

   

Hello,

I'm testing the E400P with PRI signaling and work fine, but I want to test it 
with R2 signaling.
My question are:
1. Can the E400P work with R2 signaling ?
2. If the firs question is yes, HOWTO?
Thank in advance
Peter
 

Is it available? No.
Is there any work in progress? Yes.
R2 isn't "a" protocol. Its a whole family - one for almost every 
country, and two for Colombia. :-)

Which variant do you need?

Regards,
Steve
   



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Re: [Asterisk-Users] splitting the asterisk list?

[Florian Overkamp]

At 13:16 13-3-2003 +0100, you wrote:
join both
But then, if you have a question that has an overlap, should you post and 
crosspost to both lists ? And comments that return on that question ?

Best regards,
Florian
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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Liaan van der Merwe]

That what I thought...
Mmmm


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael Bielicki
Sent: 13 Maart 2003 15:01
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] chan_capi version 0.1.0 released

You are making a basic mistake here. External TA's are treated by linux like
modems. isdn4linux faq tells you that explicitly.

It won't work.

On Thursday 13 March 2003 12:19, Liaan van der Merwe shaped the electrons to
say:
> I just installed all the latest isdn4linux stuff..
> Still same error
> Maybe asterisk wont work with external isdn devices.. any ideas?
> Thanks
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
> Junghanns
> Sent: 13 Maart 2003 11:54
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released
>
> Hi Liaan,
>
> looks like you dont have capi4linux installed. What card
> are you trying to use?
>
> regards
> kapejod
>
>
> --
> Klaus-Peter Junghanns
>
> CEO,CTO
> Junghanns.NET GmbH
> Breite Strasse 13 - 12167 Berlin - Germany
> fon:+49 30 79705390
> fax:+49 30 79705391
> iaxtel: 1-700-157-8753
> email:  [EMAIL PROTECTED]
>
> Am Don, 2003-03-13 um 10.23 schrieb Liaan van der Merwe:
> > Hallo
> > Thanks very much for great work
> > Just quick Q.
> > Why would asterisk keep on telling me :" NOTICE[8192]: File chan_capi.c,
> > Line 1984 (load_module): CAPI not installed!" when starting???
> > Thanks
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ___
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> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users

--
Michael Bielicki
Managing Director
TAAN Consultants Ltd
http://www.global-gateway.net/

--

This correspondence is for the named person's use only. It may contain
confidential or legally privileged information or both. No confidentiality
or privilege is waived or lost by any mistransmission. If you receive this
correspondence in error, please immediately delete it from your system and
notify the sender. You must not disclose, copy or rely on any part of this
correspondence if you are not the intended recipient.

Any opinions expressed in this message are those of the individual sender.
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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Liaan van der Merwe]

Ok.
So... my guess.. stuff the external isdn idea and get internal card.??


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
Junghanns
Sent: 13 Maart 2003 15:00
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released

Hi,

this is funny :)
you expect it to work with no card 
i dont think that there are external TAs with capi 2.0
linux drivers. isdn4linux wont help much...you want
capi4linux, but without a card it's not much fun ;-)

regards
kapejod


Am Don, 2003-03-13 um 11.11 schrieb Liaan van der Merwe:
> Hallo
> To be honest.. no card as yet.
> Truying to find out whether external TA will work.
> I'll install latest i4l and see what happens
> Thanks
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
> Junghanns
> Sent: 13 Maart 2003 11:54
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released
>
> Hi Liaan,
>
> looks like you dont have capi4linux installed. What card
> are you trying to use?
>
> regards
> kapejod
>
>
> --
> Klaus-Peter Junghanns
>
> CEO,CTO
> Junghanns.NET GmbH
> Breite Strasse 13 - 12167 Berlin - Germany
> fon:+49 30 79705390
> fax:+49 30 79705391
> iaxtel: 1-700-157-8753
> email:  [EMAIL PROTECTED]
>
>
>
> Am Don, 2003-03-13 um 10.23 schrieb Liaan van der Merwe:
> > Hallo
> > Thanks very much for great work
> > Just quick Q.
> > Why would asterisk keep on telling me :" NOTICE[8192]: File chan_capi.c,
> > Line 1984 (load_module): CAPI not installed!" when starting???
> > Thanks
> >
> >
>
> ___
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Re: [Asterisk-Users] CAPI errors...

[Roy Sigurd Karlsbakk]

this comes all the time, and has been there since I started using CAPI. I 
don't know what it might mean ...

On Thursday 13 March 2003 13:52, Klaus-Peter Junghanns wrote:
> Morning Roy,
>
> capi says that the send queue is full. what did you stick in there? ;-)
>
> regards
> kapejod
>
> Am Don, 2003-03-13 um 12.35 schrieb Roy Sigurd Karlsbakk:
> > hi
> >
> > I keep getting these errors all the time:
> >
> > ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending
> > DATA_B3_REQ (error=0x1102, datalen=160)
> > ERROR[3076]: File chan_capi.c, Line 1081 (pipe_msg): error sending
> > DATA_B3_RESP (error=0x1102)
> > ERROR[3076]: File chan_capi.c, Line 1081 (pipe_msg): error sending
> > DATA_B3_RESP (error=0x1102)
> > ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending
> > DATA_B3_REQ (error=0x1102, datalen=160)
> > ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending
> > DATA_B3_REQ (error=0x1102, datalen=160)
> > ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending
> > DATA_B3_REQ (error=0x1102, datalen=160)
> > ERROR[3076]: File chan_capi.c, Line 1081 (pipe_msg): error sending
> > DATA_B3_RESP (error=0x1102)
> > ERROR[3076]: File chan_capi.c, Line 1081 (pipe_msg): error sending
> > DATA_B3_RESP (error=0x1102)
> > ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending
> > DATA_B3_REQ (error=0x1102, datalen=160)
> > ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending
> > DATA_B3_REQ (error=0x1102, datalen=160)
> >
> > anyone that might know why? Klaus?
> >
> > roy
> > --
> > Roy Sigurd Karlsbakk, Datavaktmester
> > ProntoTV AS - http://www.pronto.tv/
> > Tel: +47 9801 3356
> >
> > Computers are like air conditioners.
> > They stop working when you open Windows.
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ___
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> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Roy Sigurd Karlsbakk, Datavaktmester
ProntoTV AS - http://www.pronto.tv/
Tel: +47 9801 3356

Computers are like air conditioners.
They stop working when you open Windows.


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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Lars Kneschke(priv.)]

Liaan van der Merwe <[EMAIL PROTECTED]> schrieb: 
>Oops... see. I need to learn bit more about this whole capi thing.
>It will be a planet external isdn TA.. capi2.0 compatible.

But not under Linux. The only card that support capi under linux are cards
from AVM and Eicon.

It want work with external TA. If you need a external box, there will be USB
ISDN card from AVM available, which will be supported by capi.

Cu
--
Lars Kneschke
http://www.kneschke.de
written with FeLaMiMail
http://www.saunalahti.fi/sakarit/kerro-lisaa/bart.gif



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[Asterisk-Users] E1 yellow alarms

[Michiel Betel]

About every hour I see the yellow alarms on all or a number of channels of
my PRI which is connected to the dutch telephony network, Asterisk keeps
on working fine
Here's an example where channel 1-24 went into alarm:

WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event): Detected
alarm on channel 1: Yellow Alarm
WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event): Detected
alarm on channel 2: Yellow Alarm
...
...
WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event): Detected
alarm on channel 23: Yellow Alarm
WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event): Detected
alarm on channel 24: No Alarm


And right after that they are cleared again:
NOTICE[90124]: File chan_zap.c, Line 4134 (handle_init_event): Alarm
cleared on
channel 1
NOTICE[90124]: File chan_zap.c, Line 4134 (handle_init_event): Alarm
cleared on
channel 2


ending with:
WARNING[81931]: File chan_zap.c, Line 5137 (zt_pri_error): PRI: Read on 69
failed: Unknown error 500

Zaptel.conf has:
# E1 card
span=1,1,0,ccs,hdb3,crc4
# T1 card
span=1,0,0,d4,ami
# E1
bchan=1-15
dchan=16
bchan=17-31
# T1
fxoks=32-55
#

Any ideas on how to get rid of these alarms?

Thanks!

Michiel Betel

-- 

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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Liaan van der Merwe]

Ok...
Got the capi4linux.. install everything.. (no real updates done)
How do I know whether is was installed correctly for it still says capi not
installed...
Thanks
Ps: the following modules are all loaded

- capi
- capidrv
- capifs
- kernelcapi
- capiutil

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
Junghanns
Sent: 13 Maart 2003 15:00
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released

Hi,

this is funny :)
you expect it to work with no card 
i dont think that there are external TAs with capi 2.0
linux drivers. isdn4linux wont help much...you want
capi4linux, but without a card it's not much fun ;-)

regards
kapejod


Am Don, 2003-03-13 um 11.11 schrieb Liaan van der Merwe:
> Hallo
> To be honest.. no card as yet.
> Truying to find out whether external TA will work.
> I'll install latest i4l and see what happens
> Thanks
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
> Junghanns
> Sent: 13 Maart 2003 11:54
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released
>
> Hi Liaan,
>
> looks like you dont have capi4linux installed. What card
> are you trying to use?
>
> regards
> kapejod
>
>
> --
> Klaus-Peter Junghanns
>
> CEO,CTO
> Junghanns.NET GmbH
> Breite Strasse 13 - 12167 Berlin - Germany
> fon:+49 30 79705390
> fax:+49 30 79705391
> iaxtel: 1-700-157-8753
> email:  [EMAIL PROTECTED]
>
>
>
> Am Don, 2003-03-13 um 10.23 schrieb Liaan van der Merwe:
> > Hallo
> > Thanks very much for great work
> > Just quick Q.
> > Why would asterisk keep on telling me :" NOTICE[8192]: File chan_capi.c,
> > Line 1984 (load_module): CAPI not installed!" when starting???
> > Thanks
> >
> >
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ___
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> [EMAIL PROTECTED]
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Re: [Asterisk-Users] chan_capi version 0.1.0 released

[Roy Sigurd Karlsbakk]

On Thursday 13 March 2003 14:38, Lars Kneschke(priv.) wrote:
> It want work with external TA. If you need a external box, there will be
> USB ISDN card from AVM available, which will be supported by capi.

You _don't_ want to work with isdn4linux.
It's horrible when it comes to latency and echo.
-- 
Roy Sigurd Karlsbakk, Datavaktmester
ProntoTV AS - http://www.pronto.tv/
Tel: +47 9801 3356

Computers are like air conditioners.
They stop working when you open Windows.


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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Liaan van der Merwe]

Ok... now I'm lost
What is the idea then behind capi2.0 standard??
Is this not a "language" to talk to certain modems?


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lars
Kneschke(priv.)
Sent: 13 Maart 2003 15:39
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released

Liaan van der Merwe <[EMAIL PROTECTED]> schrieb:
>Oops... see. I need to learn bit more about this whole capi thing.
>It will be a planet external isdn TA.. capi2.0 compatible.

But not under Linux. The only card that support capi under linux are cards
from AVM and Eicon.

It want work with external TA. If you need a external box, there will be USB
ISDN card from AVM available, which will be supported by capi.

Cu
--
Lars Kneschke
http://www.kneschke.de
written with FeLaMiMail
http://www.saunalahti.fi/sakarit/kerro-lisaa/bart.gif



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RE: [Asterisk-Users] iconnect & caller ID

[Mark Spencer]

> Mark - I beg to differ.  Generally callerid works with Deltathree; but
> sometimes they seem to reject it / mess it up.

Nevermind, I was thinking this was incoming Caller*ID.

Mark

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Re: [Asterisk-Users] splitting the asterisk list?

[Mark Spencer]

The more logical breakup is asterisk-users and asterisk-dev which are both
there, and everyone who subscribed to [EMAIL PROTECTED] is now subscribed
to both the users and devel list.

Mark

On Thu, 13 Mar 2003, Roy Sigurd Karlsbakk wrote:

> join both
>
> On Thursday 13 March 2003 12:57, Michael Bielicki wrote:
> > yeah what would we all carrier guys do who have completely mixed
> > environments ? create a asterisk-carrier and a asterisk-oem and a
> > asterisk-sysintegrator list ?
> >
> > On Thursday 13 March 2003 11:20, Florian Overkamp shaped the electrons to
> say:
> > > At 11:36 13-3-2003 +0100, you wrote:
> > > >Is it only me, or is the asterisk mailing list growing rapidly?
> > > >
> > > >I've got an idea, although I don't know how good it is ;-)
> > > >
> > > >how about splitting the asterisk list between good old telephony with
> > > > channel banks and telephones, and IP based telephony?
> > >
> > > Hmm, I think that would suck for those with mixed mode environments (such
> > > as me :-).
> > >
> > > Best regards,
> > > Florian
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> Roy Sigurd Karlsbakk, Datavaktmester
> ProntoTV AS - http://www.pronto.tv/
> Tel: +47 9801 3356
>
> Computers are like air conditioners.
> They stop working when you open Windows.
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Lars Kneschke(priv.)]

Liaan van der Merwe <[EMAIL PROTECTED]> schrieb: 
>Ok... now I'm lost
>What is the idea then behind capi2.0 standard??
>Is this not a "language" to talk to certain modems?
>
Capi is the interface for the software. You still need a driver which
translates the CAPI commands to the hardware commands. 

CAPI is not like Hayes.

Any ISDN-Card has capi driver under Windows. But for Linux we hard first
isdn4linux(wich is not capi). Then some german man, developed the capi
interface for AMV B1(active card) and latter AVM released capi-drivers for
all it's isdn card. Eicon did it too.

But no other vendor did release Linux-CAPI drivers. 

You can get most card working, using isdn4linux. But that's not the Capi.

Cu
--
Lars Kneschke
http://www.kneschke.de
written with FeLaMiMail
http://www.saunalahti.fi/sakarit/kerro-lisaa/bart.gif



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Re: [Asterisk-Users] chan_capi version 0.1.0 released

[Jeff Noxon]

The capi drivers are not enough.  You need to actually have CAPI hardware
working in your system.

On Thu, Mar 13, 2003 at 04:12:25PM +0200, Liaan van der Merwe wrote:
> Ok...
> Got the capi4linux.. install everything.. (no real updates done)
> How do I know whether is was installed correctly for it still says capi not
> installed...
> Thanks
> Ps: the following modules are all loaded
> 
> - capi
> - capidrv
> - capifs
> - kernelcapi
> - capiutil
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[Asterisk-Users] Proposed IAX2 Name

[Mark Spencer]

What do you all think of renaming IAX2 as:

Telephony Authentication, Signalling, and Transport Exchange (TASTE)

"TASTE" is easy to remember and has a sort of ironic relation to "SIP".
Is it took hoaky?

Mark

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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Klaus-Peter Junghanns]

> Liaan van der Merwe <[EMAIL PROTECTED]> schrieb: 
> >Oops... see. I need to learn bit more about this whole capi thing.
> >It will be a planet external isdn TA.. capi2.0 compatible.
> 
> But not under Linux. The only card that support capi under linux are cards
> from AVM and Eicon.
> 
> It want work with external TA. If you need a external box, there will be USB
> ISDN card from AVM available, which will be supported by capi.

somebody tried that usb AVM thing, without much success,
please forget this usb thing ;-)

regards
kapejod

> 
> Cu
> --
> Lars Kneschke
> http://www.kneschke.de
> written with FeLaMiMail
> http://www.saunalahti.fi/sakarit/kerro-lisaa/bart.gif
> 
> 


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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Liaan van der Merwe]

Aaa.. ok
Now it makes sense...
No I'm back to square 1.
Thanks
Cheers


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lars
Kneschke(priv.)
Sent: 13 Maart 2003 16:59
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released

Liaan van der Merwe <[EMAIL PROTECTED]> schrieb:
>Ok... now I'm lost
>What is the idea then behind capi2.0 standard??
>Is this not a "language" to talk to certain modems?
>
Capi is the interface for the software. You still need a driver which
translates the CAPI commands to the hardware commands.

CAPI is not like Hayes.

Any ISDN-Card has capi driver under Windows. But for Linux we hard first
isdn4linux(wich is not capi). Then some german man, developed the capi
interface for AMV B1(active card) and latter AVM released capi-drivers for
all it's isdn card. Eicon did it too.

But no other vendor did release Linux-CAPI drivers.

You can get most card working, using isdn4linux. But that's not the Capi.

Cu
--
Lars Kneschke
http://www.kneschke.de
written with FeLaMiMail
http://www.saunalahti.fi/sakarit/kerro-lisaa/bart.gif



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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Klaus-Peter Junghanns]

CAPI == Common isdn API

it's not about talking to certain modems, it's a hardware
independent interface to access all kinds of isdn controllers
in the same way.

regards
kapejod

Am Don, 2003-03-13 um 15.28 schrieb Liaan van der Merwe:
> Ok... now I'm lost
> What is the idea then behind capi2.0 standard??
> Is this not a "language" to talk to certain modems?
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Lars
> Kneschke(priv.)
> Sent: 13 Maart 2003 15:39
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released
> 
> Liaan van der Merwe <[EMAIL PROTECTED]> schrieb:
> >Oops... see. I need to learn bit more about this whole capi thing.
> >It will be a planet external isdn TA.. capi2.0 compatible.
> 
> But not under Linux. The only card that support capi under linux are cards
> from AVM and Eicon.
> 
> It want work with external TA. If you need a external box, there will be USB
> ISDN card from AVM available, which will be supported by capi.
> 
> Cu
> --
> Lars Kneschke
> http://www.kneschke.de
> written with FeLaMiMail
> http://www.saunalahti.fi/sakarit/kerro-lisaa/bart.gif
> 
> 
> 
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-- 
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CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705390
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]


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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Liaan van der Merwe]

Ok
Change the line in capi_chan.c( it think) to return 0 when capi not
installed.. now everything starts and loads fine.. m

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
Junghanns
Sent: 13 Maart 2003 15:00
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released

Hi,

this is funny :)
you expect it to work with no card 
i dont think that there are external TAs with capi 2.0
linux drivers. isdn4linux wont help much...you want
capi4linux, but without a card it's not much fun ;-)

regards
kapejod


Am Don, 2003-03-13 um 11.11 schrieb Liaan van der Merwe:
> Hallo
> To be honest.. no card as yet.
> Truying to find out whether external TA will work.
> I'll install latest i4l and see what happens
> Thanks
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
> Junghanns
> Sent: 13 Maart 2003 11:54
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released
>
> Hi Liaan,
>
> looks like you dont have capi4linux installed. What card
> are you trying to use?
>
> regards
> kapejod
>
>
> --
> Klaus-Peter Junghanns
>
> CEO,CTO
> Junghanns.NET GmbH
> Breite Strasse 13 - 12167 Berlin - Germany
> fon:+49 30 79705390
> fax:+49 30 79705391
> iaxtel: 1-700-157-8753
> email:  [EMAIL PROTECTED]
>
>
>
> Am Don, 2003-03-13 um 10.23 schrieb Liaan van der Merwe:
> > Hallo
> > Thanks very much for great work
> > Just quick Q.
> > Why would asterisk keep on telling me :" NOTICE[8192]: File chan_capi.c,
> > Line 1984 (load_module): CAPI not installed!" when starting???
> > Thanks
> >
> >
>
> ___
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> [EMAIL PROTECTED]
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>
> ___
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RE: [Asterisk-Users] Proposed IAX2 Name

[Benjamin Miller]

Mark, you're a darn good coder. . ...


;-)


How about "TIP", Telephony Internet Protocol.
Or is that to close to Sip?
Ben

-Original Message-
From: Mark Spencer [mailto:[EMAIL PROTECTED] 
Sent: Thursday, March 13, 2003 10:19 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Proposed IAX2 Name


What do you all think of renaming IAX2 as:

Telephony Authentication, Signalling, and Transport Exchange (TASTE)

"TASTE" is easy to remember and has a sort of ironic relation to "SIP".
Is it took hoaky?

Mark

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Re: [Asterisk-Users] Proposed IAX2 Name

[Martin Pycko]

IAX is short and I like it. Besides if that additional '2' irritates you
then anyways in the near future when IAX2 is working fine ppl will switch
eventually to IAX2 and then we'll refer to IAX2 as IAX 

Martin

On Thu, 13 Mar 2003, Mark Spencer wrote:

> What do you all think of renaming IAX2 as:
>
> Telephony Authentication, Signalling, and Transport Exchange (TASTE)
>
> "TASTE" is easy to remember and has a sort of ironic relation to "SIP".
> Is it took hoaky?
>
> Mark
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

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[Asterisk-Users] Sip call pickup ?

[Matteo Brancaleoni]

Hi, I have a mixed sip-zap evironment
in my office. I was wondering if is
possible to do remote call pickup
from a sip phone, like from zap.

Any hint?

Matteo Brancaleoni
[EMAIL PROTECTED]
Emmegi System Administrator
 
EspiA - EMMEGI Srl - e*solution provider
Uffici: Via Pascoli, 37
20129 Milano - Italy
Sede Legale: Corso Sempione 67
20149 Milano - Italy
Tel. +39 0270633354
Fax. +39 0245487890
http://www.espia.it

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Re: [Asterisk-Users] Proposed IAX2 Name

[Eric Wieling]

I like it.  It's better than STUN. *grin*

On Thu, Mar 13, 2003 at 09:19:11AM -0600, Mark Spencer wrote:
> What do you all think of renaming IAX2 as:
> 
> Telephony Authentication, Signalling, and Transport Exchange (TASTE)
> 
> "TASTE" is easy to remember and has a sort of ironic relation to "SIP".
> Is it took hoaky?
> 
> Mark
> 
> ___
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> 
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Re: [Asterisk-Users] H323 on and on

[Alejandro Olchik]

I didn't see any message regarding this subject
during the last days and I'm also interested in
pricing info to obtain G.723.1 and G.729 licences
for *.

If anyone has such this information, please send.

Tks,
Alejandro Olchik


Krzysztof Bujak said:
> Sorry for bothering you Jeremy.
>
> So asterisk can support only:
> G.711, G. 0610 and adpcm?
>
> Which means that in both cases with G.723.1 and G.729 I would have to
> get the licence.
> But where can I get codecs themselves? Are they ready to compile with
> asterisk?
> Where to find pricing info?
>
> Best regards,
> Krzysztof Bujak
>
> - Original Message -
>
> From: Jeremy McNamara 
> To: [EMAIL PROTECTED]
> 
> Sent: Monday, March 10, 2003 5:57 AM
> Subject: Re: [Asterisk-Users] H323 on and on
>
> No. G.723.1is encumbered with international patents and their associated
> royalty fee's, so no, G.723.1 is not part of the standard Asterisk
> distribution.
>
>
> Jeremy McNamara
>
>
>
> Krzysztof Bujak wrote:
>
>
> So I understand that G.723 is included in asterisk distribution?
> Sorry for lame:-(
>
> Best regards,
> KRzysztof Bujak
>
> - Original Message -
> From: Jeremy McNamara 
> To: [EMAIL PROTECTED]
> 
> Sent: Sunday, March 09, 2003 11:43 PM
> Subject: Re: [Asterisk-Users] H323 on and on
>
> chan_h323 works with any codec that Asterisk can deal with. chan_h323
> lets Asterisk deal with the audio directly, instead of attempting to
> emulate a sound card.
>
> If you want G.729 you are going to need to contact Greg Vance at Digium
> to purchase a license.
>
>
> Jeremy McNamara
>
>
>
> Krzysztof Bujak wrote:
>
>
> Could you please list codecs that are supported with this channel
> driver?
>
>
>
> The case is that, voip provider I wanted to use for routing
> international
>
> calls
>
> supports only those 723, 729.
>
> Is it possible to use it with asterisk?
>
>
>
> Best regards,
>
> Krzysztof Bujak
>
>
>
> - Original Message -
>
> From: "Jeremy McNamara"    <[EMAIL PROTECTED]>
>
> To:   
> <[EMAIL PROTECTED]>
>
> Sent: Sunday, March 09, 2003 8:52 AM
>
> Subject: Re: [Asterisk-Users] H323 on and on
>
>
>
>
>
>
>
> Well, someone is attempting to delete a object that has never been
>
> allocated.
>
>
>
> You might give chan_h323 a try, you might have better luck.
>
>
>
>   http://asterisk.nufone.net 
>
>
>
>
>
> Jeremy McNamara
>
>
>
>
>
>
>
> Ben Clark wrote:
>
>
>
>
>
> this is what it printed
>
>
>
> (gdb) bt
>
> #0  0x40120cb3 in chunk_free () from /lib/libc.so.6
>
> #1  0x40120c53 in free () from /lib/libc.so.6
>
> #2  0x403e7977 in __builtin_delete (ptr=0x80e41f8) from
>
> /usr/local/lib/liboh323wrap.so
>
> #3  0x4103a27a in PContainer::Destruct () from
>
> /usr/local/lib/libpt_linux_x86_r.so.1
>
> #4  0x4102adfa in PThread::PX_ThreadStart () from
>
> /usr/local/lib/libpt_linux_x86_r.so.1
>
> #5  0x40030e67 in pthread_start_thread () from /lib/libpthread.so.0
>
>
>
> On Sunday, March 9, 2003, at 01:16 AM, Jeremy McNamara wrote:
>
>
>
>
>
> type bt, press enter and send what gets printed out
>
>
>
> Jeremy McNamara
>
>
>
>
>
>
>
> Ben Clark wrote:
>
>
>
>
>
> this is the end of the output I get from gdb...
>
>
>
> Reading symbols from /usr/lib/asterisk/modules/chan_oh323.so...done.
>
> Loaded symbols for /usr/lib/asterisk/modules/chan_oh323.so
>
> Reading symbols from /usr/local/lib/liboh323wrap.so...done.
>
> Loaded symbols for /usr/local/lib/liboh323wrap.so
>
> Reading symbols from /usr/local/lib/libh323_linux_x86_r.so.1...done.
>
> Loaded symbols for /usr/local/lib/libh323_linux_x86_r.so.1
>
> Reading symbols from /usr/local/lib/libpt_linux_x86_r.so.1...done.
>
> Loaded symbols for /usr/local/lib/libpt_linux_x86_r.so.1
>
> Reading symbols from /usr/lib/libstdc++-libc6.2-2.so.3...done.
>
> Loaded symbols for /usr/lib/libstdc++-libc6.2-2.so.3
>
> ---Type  to continue, or q  to quit---
>
> #0  0x40120cb3 in chunk_free () from /lib/libc.so.6
>
>
>
> On Saturday, March 8, 2003, at 07:59 PM, Martin Pycko wrote:
>
>
>
>
>
> asterisk -vvvcg
>
>
>
> when the segfault happens
>
> gdb ./asterisk core.[pid]
>
>
>
> regards
>
> Martin
>
>
>
> On Sat, 8 Mar 2003, Ben Clark wrote:
>
>
>
>
>
> I am also getting a seg fault when asterisk tries to load
>
> chan_oh323.so.  What should I try to get it to work?
>
>
>
> [chan_oh323.so] => (OpenH323 Channel Driver)
>
>== Parsing '/etc/asterisk/oh323.conf': Found
>
> Segmentation fault
>
>
>
>
>
> On Saturday, March 8, 2003, at 09:06 AM, Krzysztof Bujak wrote:
>
>
>
>
>
> ...I tried asterisk open h323 but it seg faults:-(...
>
>
>
> ___
>
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>
> [EMAIL PROTECTED] 
>
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
>
>
>
>
>
> __

Re: [Asterisk-Users] Proposed IAX2 Name

[ron]

Quoting Mark Spencer <[EMAIL PROTECTED]>:

> What do you all think of renaming IAX2 as:
> 
> Telephony Authentication, Signalling, and Transport Exchange (TASTE)
> 
> "TASTE" is easy to remember and has a sort of ironic relation to "SIP".
> Is it took hoaky?
> 
> Mark
> 

I still live "LIVE"...

LIghtweight
Voice over IP
Exchange

Ron Gage
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Re: [Asterisk-Users] Beginning of voicemail missed by sip phone

[Martin Pycko]

You can playback a second or two of silence ...

regards
Martin

On Thu, 13 Mar 2003, T Aksoy wrote:

> Hi,
>
> We are testing a number of sip phones from different manufacturers. With one phone 
> in particular, when I dial the asterisk voicemail, it misses around half a second 
> from the beginning of the announcement. I don't have this problem with the snom 200 
> or 100.
>
> Does anyone know why this happens? Is it a sync issue? How do I delay the start of 
> the voicemail announcement? (Maybe that will fix the problem).
>
> Thanks
> Tan Aksoy
> Telappliant Solutions
>

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Re: [Asterisk-Users] Sip call pickup ?

[Mark Spencer]

This feature is in development currently.

Mark

On Thu, 13 Mar 2003, Matteo Brancaleoni wrote:

> Hi, I have a mixed sip-zap evironment
> in my office. I was wondering if is
> possible to do remote call pickup
> from a sip phone, like from zap.
>
> Any hint?
>
> Matteo Brancaleoni
> [EMAIL PROTECTED]
> Emmegi System Administrator
>
> EspiA - EMMEGI Srl - e*solution provider
> Uffici: Via Pascoli, 37
> 20129 Milano - Italy
> Sede Legale: Corso Sempione 67
> 20149 Milano - Italy
> Tel. +39 0270633354
> Fax. +39 0245487890
> http://www.espia.it
>
> ___
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>

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Re: [Asterisk-Users] Proposed IAX2 Name

[Matthew Farley]

Finally, a question on this list I feel qualified to answer :D

I like it. Keeps the name from being confused with other
computer-related acronyms (a buddy of mine once asked why on earth I
kept referring to AIX as a VoIP thing... Surely, he said, I had to know
that it is an Operating System. It took us a few minutes to realize that
he had misheard my references to IAX).

TASTE... Sounds good to me. Opens the door to new odd conversational
elements like "I can surf the web, but I can't TASTE anything". I like
the ironic angle... And no computer acronym is too hoaky, so long as it
is memorable and innofensive (IMHO).

-Matthew Farley

On Thu, 2003-03-13 at 09:19, Mark Spencer wrote:
> What do you all think of renaming IAX2 as:
> 
> Telephony Authentication, Signalling, and Transport Exchange (TASTE)
> 
> "TASTE" is easy to remember and has a sort of ironic relation to "SIP".
> Is it took hoaky?
> 
> Mark
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Matthew Farley <[EMAIL PROTECTED]>

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[Asterisk-Users] Asterisk Clients

[Liaan van der Merwe]

Hallo

What soft clients can all be used
with asterisk?

Thanks

liaan

[Asterisk-Users] Asterisk Clients

[Liaan van der Merwe]

Please disregard last

Will RTFM

RE: [Asterisk-Users] iconnect & caller ID

[Jim Archer]

Still, I can't get it to work. Maybe its specific to my area.  I think I am 
giving up on iconnect for a while.  Thanks.

--On Thursday, March 13, 2003 8:45 AM -0600 Mark Spencer 
<[EMAIL PROTECTED]> wrote:

Mark - I beg to differ.  Generally callerid works with Deltathree; but
sometimes they seem to reject it / mess it up.
Nevermind, I was thinking this was incoming Caller*ID.
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Re: [Asterisk-Users] Proposed IAX2 Name

[Steve Kann]

TEA : Telephony data Exchange with Authentication

goes well with SIP also :)

or, 

RSTP:  Really Simple Telephony Protocol
(gets out that unlike h323 and SIP, this is designed to be
simple, but way to easy to confuse with RTSP, or others).

-SteveK


On Thu, Mar 13, 2003 at 10:50:57AM -0500, Benjamin Miller wrote:
> Mark, you're a darn good coder. . ...
> 
> 
> ;-)
> 
> 
> How about "TIP", Telephony Internet Protocol.
> Or is that to close to Sip?
> Ben
> 
> -Original Message-
> From: Mark Spencer [mailto:[EMAIL PROTECTED] 
> Sent: Thursday, March 13, 2003 10:19 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Proposed IAX2 Name
> 
> 
> What do you all think of renaming IAX2 as:
> 
> Telephony Authentication, Signalling, and Transport Exchange (TASTE)
> 
> "TASTE" is easy to remember and has a sort of ironic relation to "SIP".
> Is it took hoaky?
> 
> Mark
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
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Re: [Asterisk-Users] Build a complex IVR?

[Martin Pycko]

[deeper]
exten => s,1,Playback,you-re-in-the-deepest-menu
exten => s,2,Goto,options|s|1
[options]
exten => s,1,Background,prompt-1-deeper-2-back
exten => 1,1,Goto,deeper|s|1
exten => 2,1,Goto,sales|s|1
[sales]
exten => 
s,1,Background,prompt-1-information-2-connect-or-stay-on-the-line-0-operator-*-to-go-back
exten => s,2,Goto,2
exten => 0,1,Dial,Zap/2
exten => 1,1,PlayBack,information
exten => 2,1,Dial,Zap/1
exten => *,1,Goto,Menu|s|1
[Menu]
exten => s,1,Background,prompt-1-sales-2-options
exten => 1,1,Goto,sales|s|1
exten => 2,1,Goto,options|s|1
exten => i,1,Goto,s
exten => t,1,Goto,s

[incoming]
exten => s,1,Goto,Menu|s|1


regards
Martin

On Thu, 13 Mar 2003, it wrote:

> Hi,Jim,
> Thank you very much for your reply. But I think the menu file you
> designed must be interpreted by another program writen by your but not
> asterisk.
>
> - Original Message -
> From: "Jim Gottlieb" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Thursday, March 13, 2003 12:21 AM
> Subject: Re: [Asterisk-Users] Build a complex IVR?
>
>
> > On 2003-03-13 at 15:39, "it" <[EMAIL PROTECTED]> wrote:
> >
> > >I would like to know if Asterisk could be used to build a IVR
> with
> > >complex flow?
> >
> > Yes, we have done this.  We use menu files that define what each
> > keypress does so we can build complex menus with full flexibility.
> >
> > A short example (since you asked):
> >
> > default = 0
> > prompt
> > {
> > name = 3046
> > }
> >
> > 0
> > {
> > action = debitcard
> > declines = debitcard-declines
> > pin-prompt = 5122
> > number-prompt = 5123
> > thankyou = 5124
> > }
> >
> > 1
> > {
> > action = debitcardbalance
> > declines = debitcard-declines
> > pin-prompt = 5122
> > }
> >
> > 2
> > {
> > action = debitcardgenerate
> > declines = debitcard-declines
> > cardtype = 420
> > }
> >
> > 3
> > {
> > action = vmail
> > userno = 3641234
> > }
> >
> > Once it goes into vmail, we have a whole hard-coded voicemail app that
> > is basically modeled on the Centigram VoiceMemo system including
> > message sending, forwarding, paging, transferring to attendant.
> > Likewise for other functions.  It also does queries into our backend
> > billing database and credit card charging systems.
> >
> > I wish I could convince my company to release the source code, but I
> > don't think that's ever going to happen, as they consider this code to
> > be their strategic advantage.  It's been developed over many years and
> > used to run with Dialogic hardware but we have recently ported it to
> > asterisk.
> >
> > So yes, you can build almost any IVR application under asterisk.  Not
> > only is the hardware one-eighth the cost, but the tech support is far
> > far superior.  We used to wait years for Dialogic to fix bugs.  And
> > best of all, asterisk provides all the core functions so you can spend
> > your time on your custom features and not on coding the basic
> > telephony functions, message playing, etc.
> > ___
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> >
>
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Re: [Asterisk-Users] Proposed IAX2 Name

[Florian Overkamp]

At 09:19 13-3-2003 -0600, you wrote:
What do you all think of renaming IAX2 as:

Telephony Authentication, Signalling, and Transport Exchange (TASTE)

"TASTE" is easy to remember and has a sort of ironic relation to "SIP".
Is it took hoaky?
Sounds good. It slightly reminds me of a local frat-club, but I'll get over 
that :-)

Florian

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Re: [Asterisk-Users] Proposed IAX2 Name

[Mark Spencer]

> LIghtweight
> Voice over IP
> Exchange

Or:

Lightweight
Internet
Voice
Exchange

Mark

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Re: [Asterisk-Users] Build a complex IVR?

[Chris Albertson]

What is "complex"?  If all you need is a menus tree with static
menus (that is, the options are fixed) then the menus system
built-into Asterisk would work.

But if you want to serve up dynamic content, like custom menus
for each caller based on products they may have purchced from you
or what accounts they have with you that you querry out of a
database based on caller ID then you need more.

I don't concider staic menu trees "complex"  For complex
IVR I'm looking into VXML or "Voice XML".  It doesn't look to hard to
integrate a VXML processor into Asterisks and one you've got that,
serving up dynamic content is like serving up dynamic, multi-teir
web pages.


--- it <[EMAIL PROTECTED]> wrote:
> Hi,every one! 
> I would like to know if Asterisk could be used to build a IVR
> with complex flow? To provide a complex sample would be appreciated.
> 
>Regards.
> 
>john
> 


=
Chris Albertson
  Home:   310-376-1029  [EMAIL PROTECTED]
  Cell:   310-990-7550
  Office: 310-336-5189  [EMAIL PROTECTED]
  KG6OMK

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Re: [Asterisk-Users] Cisco 7960

[Mike Reiling]

Morning,
	Actually the phone will be running SIP, but from what I have read,  
most people use the call manager for the services_url stuff.  I read  
the url mentioned below, and wrote some simple xml, but the phone  
services emulator doesn't like it.  My guess is that just doesn't work  
correctly.

With all that in mind, once I get my phone (hopefully tomorrow), I am  
planning on writing some XML files to be sent to the phone.  What  
things would people like to see developed.  My first script will be for  
the directory, using voicemail.conf.

Thanks,
Mike
On Wednesday, March 12, 2003, at 08:40  PM, Stephen Webb wrote:

What mode are you running the Phone in? SIP, MCGP, or SCCP (Skinny)

You mentioned Call Manager so I will assume SCCP. If that is the case I
do not know.
However if you are running it in SIP, All you have to do is set
# XML URLs
services_url: ""  ; URL for external Phone Services
directory_url: "" ; URL for external Directory location
logo_url: ""  ; URL for branding logo to be used on phone display
These in you configuration and point it to a webserver.
The xml format can be found here.
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/ 
bxtml.htm

Hope this helps!

Stephen

On Wed, 2003-03-12 at 18:59, Mike Reiling wrote:
Anyone know if it is possible to load your own XML scripts on to the
phone, bypassing the Cisco CallManager?  I am still waiting for my
phone to arrive, but I have been playing with Cisco's phone services
emulator, and that doesn't seem to like anything I pass to it.
If it is possible, anyone want to share any sample scripts they have.

--Mike

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RE: [Asterisk-Users] Beginning of voicemail missed by sip phone

[Benjamin Miller]

Title: Message



Actually I've seen this exact issue with my Cisco 
7960.
And 
it's any voice prompt I dial.  I loose just the very first .5 seconds of 
the audio for whatever reason.
So the 
sip users hear "eedian Mail" and "nk you for calling".
Any 
one else dealing with this?
Any 
ideas?

  
  -Original Message-From: T Aksoy 
  [mailto:[EMAIL PROTECTED] Sent: Thursday, March 13, 2003 4:54 
  AMTo: [EMAIL PROTECTED]Subject: 
  [Asterisk-Users] Beginning of voicemail missed by sip 
  phone
  Hi,
   
  We are testing a number of sip phones from 
  different manufacturers. With one phone in particular, when I dial the 
  asterisk voicemail, it misses around half a second from the beginning of the 
  announcement. I don't have this problem with the snom 200 or 100.
   
  Does anyone know why this happens? Is it a sync 
  issue? How do I delay the start of the voicemail announcement? (Maybe that 
  will fix the problem).
   
  Thanks
  Tan Aksoy
  Telappliant Solutions
   

Re: [Asterisk-Users] Proposed IAX2 Name

[Jeremy McNamara]

Denon mentioned to me:

SVP -- Streamlined Voice Protocol


Jeremy 

Mark Spencer wrote:

  
LIghtweight
Voice over IP
Exchange

  
  
Or:

Lightweight
Internet
Voice
Exchange

Mark
  

Re: [Asterisk-Users] Proposed IAX2 Name

[Matteo Brancaleoni]

TASTE... tastes good for me.

but have you thinked 'bout LISP ? ;-)

Lightweight
Internet
Signalling
Protocol

or SIVEX? Simple Internet Voice EXchange

WONVE : Working Over Nat Voice Exchange ;-)

think that any combination could be done...

matteo

Il gio, 2003-03-13 alle 17:40, Mark Spencer ha scritto:
> > LIghtweight
> > Voice over IP
> > Exchange
> 
> Or:
> 
> Lightweight
> Internet
> Voice
> Exchange
> 
> Mark
> 
> ___
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-- 
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Espia - Emmegi Srl

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Re: [Asterisk-Users] Proposed IAX2 Name

[denon]

I still like my original idea, SVP - Streamlined Voice Protocol.

I think the trick to naming IAX2, is to encourage the thought of 
streamlined and efficiency.  If people want a feature-rich and bloated 
protocol, they'll run H.323, SIP, etc .. IAX is all about performance and 
resource utilization.

denon

At 09:19 AM 3/13/2003 -0600, you wrote:
What do you all think of renaming IAX2 as:

Telephony Authentication, Signalling, and Transport Exchange (TASTE)

"TASTE" is easy to remember and has a sort of ironic relation to "SIP".
Is it took hoaky?
Mark

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Re: [Asterisk-Users] Proposed IAX2 Name

[John Todd]

I'm in favor of "TASTE" myself, though Mark's take on  "LIVE" has the 
all-important "I", to establish the use of this protocol over IP 
networks, which is an important part of the protocol and conceptual 
structure, yes?

Perhaps "ITASTE" with the "I" standing for the obvious "Internet".

JT


What do you all think of renaming IAX2 as:

Telephony Authentication, Signalling, and Transport Exchange (TASTE)

"TASTE" is easy to remember and has a sort of ironic relation to "SIP".
Is it took hoaky?
Mark
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Re: [Asterisk-Users] Beginning of voicemail missed by sip phone

[Lele Forzani]

On Thursday 13 March 2003 18:00, Benjamin Miller wrote:

> Actually I've seen this exact issue with my Cisco 7960.
> And it's any voice prompt I dial.  I loose just the very first .5
> seconds of the audio for whatever reason.
> So the sip users hear "eedian Mail" and "nk you for calling".
> Any one else dealing with this?
> Any ideas?

Same here. But it seems to be somewhat related with the hardware: the 7960 
looses about .5 seconds, and an old siemens optipoint 100 I have around 
looses up to 2 seconds. 

But there's no delay with the ATA186 (sip) and with the Pingtel Expressa. With 
the ATA I can hear the change in the background noise shortly before the 
beginning of the recording, which is probably the connection of the rtp 
stream.

lele

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Re: [Asterisk-Users] Proposed IAX2 Name

[Howard White]

Mark and List:

I agree with the earlier post that said keep the name IAX.  No we don't
need to go through a progression like Algol60 -> Pascal -> Modula2 ->
Modula3...

To our multi-lingual listers - do IAX or TASTE have any non-English
complications?  Remember how much fun General Motors had with the "Nova"
in Spanish (or Romance language) speaking countries?

Howard White

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Re: [Asterisk-Users] Proposed IAX2 Name

[Chris Albertson]

Before you prepend the "I", does IAX2 really depend on IP?
For example couldn't you send TASTE over ATM, FDDI, X25 or a
simple serial cable?  My bet is that most VoIP would work better
if it weren't for the IP part.  Yes I do have use for other
transports.


--- John Todd <[EMAIL PROTECTED]> wrote:
> 
> I'm in favor of "TASTE" myself, though Mark's take on  "LIVE" has the
> 
> all-important "I", to establish the use of this protocol over IP 
> networks, which is an important part of the protocol and conceptual 
> structure, yes?
> 
> Perhaps "ITASTE" with the "I" standing for the obvious "Internet".
> 
> JT
> 
> 
> >What do you all think of renaming IAX2 as:
> >
> >Telephony Authentication, Signalling, and Transport Exchange (TASTE)
> >
> >"TASTE" is easy to remember and has a sort of ironic relation to
> "SIP".
> >Is it took hoaky?
> >
> >Mark
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users


=
Chris Albertson
  Home:   310-376-1029  [EMAIL PROTECTED]
  Cell:   310-990-7550
  Office: 310-336-5189  [EMAIL PROTECTED]
  KG6OMK

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[Asterisk-Users] Does anyone know.

[James O. Sizemore III]

Out of band DTMF for SIP seems to be broke.
I tried switching to dtmfmode=inband this
works fine for local phones, but any phones
over non LAN link, can not enter digits
without duplicates showing up, this is most
sever for the user name prompt in voice mail
main.
Is anyone working on getting out of band DTMF
working again?
Thanks,

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RE: [Asterisk-Users] iconnect & caller ID

[William Walsh]

On Thu, 2003-03-13 at 08:55, Jim Archer wrote:
> Still, I can't get it to work. Maybe its specific to my area.  I think I am 
> giving up on iconnect for a while.  Thanks.


 I presume the problem is in the CallerIDName that shows up when you
call a regular number through iconnecthere?

If so, that's because, despite what you send through iconnecthere (or
the other SIP to POTS providers), the callerid system at the telcos do
not automatically trust that.

Technically the phone numbers "belong" to Deltathree, and in the telco
systems that is how they are registered, and that is why  your
CallerIDName does not show up.  I really do not know that
deltathree/iconnecthere can do anything about that.  It's a design issue
in the system.  Your system's sent data can't override the telco
systems.

-- 
William Walsh <[EMAIL PROTECTED]>
Jabber: [EMAIL PROTECTED]


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RE: [Asterisk-Users] splitting the asterisk list?

[Steve Radich]

There should maybe be a few reminders sent out to posters saying this should
have been in the dev mailing list.

I would recommend though that something like a initial setup list be created
for the questions like getting extensions working, setting up initial VoIP,
config voicemail, initial channel bank configs, I found phone model xyz that
works, etc.  As the product gets more and more popular the newbie questions,
and those just wanting to get started and test it out, will get to be more
and more traffic which is useless to many of the users.  As long as plenty
of people keep supporting the newbie list that would work - if people quit
supporting it then keeping it in one list is needed.

Steve Radich - Colocation / Virtual Dedicated / Dedicated Servers 
BitShop, Inc. - http://www.bitshop.com - $149/month colo special


-Original Message-
From: Mark Spencer [mailto:[EMAIL PROTECTED] 
Sent: Thursday, March 13, 2003 9:59 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] splitting the asterisk list?

The more logical breakup is asterisk-users and asterisk-dev which are both
there, and everyone who subscribed to [EMAIL PROTECTED] is now subscribed
to both the users and devel list.

Mark

On Thu, 13 Mar 2003, Roy Sigurd Karlsbakk wrote:

> join both
>
> On Thursday 13 March 2003 12:57, Michael Bielicki wrote:
> > yeah what would we all carrier guys do who have completely mixed
> > environments ? create a asterisk-carrier and a asterisk-oem and a
> > asterisk-sysintegrator list ?
> >
> > On Thursday 13 March 2003 11:20, Florian Overkamp shaped the electrons
to
> say:
> > > At 11:36 13-3-2003 +0100, you wrote:
> > > >Is it only me, or is the asterisk mailing list growing rapidly?
> > > >
> > > >I've got an idea, although I don't know how good it is ;-)
> > > >
> > > >how about splitting the asterisk list between good old telephony with
> > > > channel banks and telephones, and IP based telephony?
> > >
> > > Hmm, I think that would suck for those with mixed mode environments
(such
> > > as me :-).
> > >
> > > Best regards,
> > > Florian
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> Roy Sigurd Karlsbakk, Datavaktmester
> ProntoTV AS - http://www.pronto.tv/
> Tel: +47 9801 3356
>
> Computers are like air conditioners.
> They stop working when you open Windows.
>
>
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>

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Re: [Asterisk-Users] Proposed IAX2 Name

[Karl Putland]

What about ITP

Internet/IP
Telephony
Protocol

On Thu, 2003-03-13 at 09:40, Mark Spencer wrote:
> > LIghtweight
> > Voice over IP
> > Exchange
> 
> Or:
> 
> Lightweight
> Internet
> Voice
> Exchange
> 
> Mark
> 
> ___
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> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Karl Putland <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Proposed IAX2 Name

[Jared Smith]

I like the name IAX, only because I want to get "call piggybacking"
working and call it pigbIAX (pronounced "pig beaks").  :-)

If I were to chose a new name, TASTE is probably the one I like the
best...




On Thu, 2003-03-13 at 08:19, Mark Spencer wrote:
> What do you all think of renaming IAX2 as:
> 
> Telephony Authentication, Signalling, and Transport Exchange (TASTE)
> 
> "TASTE" is easy to remember and has a sort of ironic relation to "SIP".
> Is it took hoaky?
> 
> Mark
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Jared Smith ([EMAIL PROTECTED])
Infrastructure Programmer
Discovery Research Group
6975 Union Park Center, Suite 450
Midvale, UT 84047
1-800-678-3748 ext. 124



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RE: [Asterisk-Users] Proposed IAX2 Name

[Steve Radich]

When you get tired of SIPping you can upgrade to TASTEing...  How do we go
from that to Asterisk?

Searching for taste though isn't going to return favorable VoIP results, not
that sip is great - but iax / iax2 is pretty distinct.

I do like taste though.

Steve Radich - Colocation / Virtual Dedicated / Dedicated Servers 
BitShop, Inc. - http://www.bitshop.com - $149/month colo special


-Original Message-
From: John Todd [mailto:[EMAIL PROTECTED] 
Sent: Thursday, March 13, 2003 1:12 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Proposed IAX2 Name


I'm in favor of "TASTE" myself, though Mark's take on  "LIVE" has the 
all-important "I", to establish the use of this protocol over IP 
networks, which is an important part of the protocol and conceptual 
structure, yes?

Perhaps "ITASTE" with the "I" standing for the obvious "Internet".

JT


>What do you all think of renaming IAX2 as:
>
>Telephony Authentication, Signalling, and Transport Exchange (TASTE)
>
>"TASTE" is easy to remember and has a sort of ironic relation to "SIP".
>Is it took hoaky?
>
>Mark
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Re: [Asterisk-Users] Proposed IAX2 Name

[Martin Pycko]

or Packet Telephony (Simple) Protocol


On 13 Mar 2003, Karl Putland wrote:

> What about ITP
>
> Internet/IP
> Telephony
> Protocol
>
> On Thu, 2003-03-13 at 09:40, Mark Spencer wrote:
> > > LIghtweight
> > > Voice over IP
> > > Exchange
> >
> > Or:
> >
> > Lightweight
> > Internet
> > Voice
> > Exchange
> >
> > Mark
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> --
> Karl Putland <[EMAIL PROTECTED]>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

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RE: [Asterisk-Users] splitting the asterisk list?

[Steven Critchfield]

On Thu, 2003-03-13 at 13:23, Steve Radich wrote:
> There should maybe be a few reminders sent out to posters saying this should
> have been in the dev mailing list.
> 
> I would recommend though that something like a initial setup list be created
> for the questions like getting extensions working, setting up initial VoIP,
> config voicemail, initial channel bank configs, I found phone model xyz that
> works, etc.  As the product gets more and more popular the newbie questions,
> and those just wanting to get started and test it out, will get to be more
> and more traffic which is useless to many of the users.  As long as plenty
> of people keep supporting the newbie list that would work - if people quit
> supporting it then keeping it in one list is needed.

Newbie questions should be where the user list excels. This is users
helping users fix config issues. The whole naming of IAX2 might have
done better on the dev list. Patch submissions and talk about feature
implementation should be on dev. This way we don't confuse normal users
with information that isn't ready for general consumption.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Beginning of voicemail missed by sip phone

[Matteo Brancaleoni]

I can confirm that.
With the snom, I get no delay.
with a sip-fxs gw, I get 2 seconds delay.

Matteo

Il gio, 2003-03-13 alle 19:59, Lele Forzani ha scritto:
> On Thursday 13 March 2003 18:00, Benjamin Miller wrote:
> 
> > Actually I've seen this exact issue with my Cisco 7960.
> > And it's any voice prompt I dial.  I loose just the very first .5
> > seconds of the audio for whatever reason.
> > So the sip users hear "eedian Mail" and "nk you for calling".
> > Any one else dealing with this?
> > Any ideas?
> 
> Same here. But it seems to be somewhat related with the hardware: the 7960 
> looses about .5 seconds, and an old siemens optipoint 100 I have around 
> looses up to 2 seconds. 
> 
> But there's no delay with the ATA186 (sip) and with the Pingtel Expressa. With 
> the ATA I can hear the change in the background noise shortly before the 
> beginning of the recording, which is probably the connection of the rtp 
> stream.
> 
> lele
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Matteo Brancaleoni <[EMAIL PROTECTED]>
Espia - Emmegi Srl

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RE: [Asterisk-Users] iconnect & caller ID

[Jim Archer]

--On Thursday, March 13, 2003 11:11 AM -0800 William Walsh 
<[EMAIL PROTECTED]> wrote:

 I presume the problem is in the CallerIDName that shows up when you
call a regular number through iconnecthere?
It usually presents no caller ID, resulting in an anonymous call rejection 
message.

Technically the phone numbers "belong" to Deltathree, and in the telco
systems that is how they are registered, and that is why  your
CallerIDName does not show up.  I really do not know that
deltathree/iconnecthere can do anything about that.  It's a design issue
in the system.  Your system's sent data can't override the telco
systems.
Fair enough.  But most of the people we call are customers at their homes. 
Verizon and other carriers have given everyone "anonymous call rejection" 
for free and, at least in the case of Verizon they have turned it on by 
default.  The result is that we can't call people through iconnect.  I have 
tried sending the code to enable passing caller ID but no luck.

I just got to the point where I had dumped a lot of time into it and was 
becoming convinced that I was not going to be able to get it to work.  I 
was also seeing issues with call connect reliability.  It seems often that 
Asterisk initiates a connection to iconnect but iconnect either fails to 
notice or says it is temporarily unavailable.

I didn't even look at the issue of whether or not we can make 2 or more 
calls out through iconnect at the same time.  That may work fine, I just 
have not tried it.

I am happy to keep testing if it will be useful to anyone else, especially 
the developers.

Jim

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[Asterisk-Users] Intergrate with MySQL

[Ajit Kallingal]

Hello All,
Can the current Asterisk be integrated with mySQL to query a database ?
I am looking at a  typical IVR scenario where the user punches a "product
code" and the database query will determine if the product is available or
not. The reply would be number of items available , else none.

Thanks and Regards
Ajit

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Re: [Asterisk-Users] Beginning of voicemail missed by sip phone

[Mark Spencer]

Can somebody look at the RTP packets with "ethereal" and tell me if they
notice any difference between what we send and what we receive?  Perhaps
we're starting out with values that are too high or something?

Mark

On 13 Mar 2003, Matteo Brancaleoni wrote:

> I can confirm that.
> With the snom, I get no delay.
> with a sip-fxs gw, I get 2 seconds delay.
>
> Matteo
>
> Il gio, 2003-03-13 alle 19:59, Lele Forzani ha scritto:
> > On Thursday 13 March 2003 18:00, Benjamin Miller wrote:
> >
> > > Actually I've seen this exact issue with my Cisco 7960.
> > > And it's any voice prompt I dial.  I loose just the very first .5
> > > seconds of the audio for whatever reason.
> > > So the sip users hear "eedian Mail" and "nk you for calling".
> > > Any one else dealing with this?
> > > Any ideas?
> >
> > Same here. But it seems to be somewhat related with the hardware: the 7960
> > looses about .5 seconds, and an old siemens optipoint 100 I have around
> > looses up to 2 seconds.
> >
> > But there's no delay with the ATA186 (sip) and with the Pingtel Expressa. With
> > the ATA I can hear the change in the background noise shortly before the
> > beginning of the recording, which is probably the connection of the rtp
> > stream.
> >
> > lele
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> --
> Matteo Brancaleoni <[EMAIL PROTECTED]>
> Espia - Emmegi Srl
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

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RE: [Asterisk-Users] iconnect & caller ID

[William Walsh]

On Thu, 2003-03-13 at 12:36, Jim Archer wrote:
> --On Thursday, March 13, 2003 11:11 AM -0800 William Walsh 
> <[EMAIL PROTECTED]> wrote:
> 
> >  I presume the problem is in the CallerIDName that shows up when you
> > call a regular number through iconnecthere?
> 
> It usually presents no caller ID, resulting in an anonymous call rejection 
> message.

The # is getting passed in all of my calls through them, but not always
the correct name I set, depending on other things like I mentioned
earlier.

> > Technically the phone numbers "belong" to Deltathree, and in the telco
> > systems that is how they are registered, and that is why  your
> > CallerIDName does not show up.  I really do not know that
> > deltathree/iconnecthere can do anything about that.  It's a design issue
> > in the system.  Your system's sent data can't override the telco
> > systems.
> 
> Fair enough.  But most of the people we call are customers at their homes. 
> Verizon and other carriers have given everyone "anonymous call rejection" 
> for free and, at least in the case of Verizon they have turned it on by 
> default.  The result is that we can't call people through iconnect.  I have 
> tried sending the code to enable passing caller ID but no luck.
> 
> I just got to the point where I had dumped a lot of time into it and was 
> becoming convinced that I was not going to be able to get it to work.  I 
> was also seeing issues with call connect reliability.  It seems often that 
> Asterisk initiates a connection to iconnect but iconnect either fails to 
> notice or says it is temporarily unavailable.

I see this occasionally myself.  For the cheap price of the service, I
chalk it up to a small price to pay normally.  I do notice that
sometimes you have to wait a minute between placing calls as well.

> I didn't even look at the issue of whether or not we can make 2 or more 
> calls out through iconnect at the same time.  That may work fine, I just 
> have not tried it.

One outgoing call at a time per account.  A silly restriction when you
are paying for minutes, IMO, but one that does not appear to be unique
to iconnect/deltathree in the VoIP termination industry.

> I am happy to keep testing if it will be useful to anyone else, especially 
> the developers.

I think you just have to decide if the limitations make it worthwhile or
not.  I wouldn't depend on iconnecthere as a sole means of making calls.

I've been looking myself at signing up for packet8.net's service
myself.  For about twice as much as iconnecthere's service, you can get
unlimited calling (with a 1 year contract).  But that also is 1 outgoing
call at a time.  

If you are looking at some serious call volume, you may want to approach
deltathree's business development service for a custom arrangement that
better meets your needs.  I know a couple people on this list are
working on IAX termination services as well that may be worthwhile
checking out for business volume calling.


-- 
William Walsh <[EMAIL PROTECTED]>
Jabber: [EMAIL PROTECTED]


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Re: [Asterisk-Users] Intergrate with MySQL

[Steven Critchfield]

On Thu, 2003-03-13 at 14:26, Ajit Kallingal wrote:
> Hello All,
> Can the current Asterisk be integrated with mySQL to query a database ?
> I am looking at a  typical IVR scenario where the user punches a "product
> code" and the database query will determine if the product is available or
> not. The reply would be number of items available , else none.

You can write an AGI app and have your app do the queries to the
database where your connection method is totally up to you. This is
probably prefered as perl and/or other languages will let you do better
error handling if your connection is down to the database.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Detecting channel status

[Steven Critchfield]

On Thu, 2003-03-13 at 15:31, [EMAIL PROTECTED] wrote:
> In the extensions.conf file, is there a way to detect if it is on hook 
> or off hook (without dialing it)?

I do not believe so. What are you trying to accomplish?
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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[Asterisk-Users] Detecting channel status

[david . anderson]

In the extensions.conf file, is there a way to detect if it is on hook 
or off hook (without dialing it)?

Dave

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Re: [Asterisk-Users] Intergrate with MySQL

[Roderick Montgomery]

According to Ajit Kallingal:
> Hello All,
> Can the current Asterisk be integrated with mySQL to query a database ?
> I am looking at a  typical IVR scenario where the user punches a "product
> code" and the database query will determine if the product is available or
> not. The reply would be number of items available , else none.

Sure, something like the following AGI script would work. Just grab the Perl
AGI module from http://asterisk.gnuinter.net/>, drop the following in
your extensions.conf...

exten => s,1,Answer
exten => s,2,Wait,1
exten => s,3,AGI,script.agi

...and put the following as "script.agi" in /var/lib/asterisk/agi-bin:

---
#!/usr/bin/perl

use Asterisk::AGI;
use DBI;

my $AGI = new Asterisk::AGI;
my %input = $AGI->ReadParse();

### Play "welcome.gsm", greeting the caller 

$AGI->stream_file('welcome');

### Play "product-code-prompt.gsm", instructing the caller to enter the 
### product code, then allow ten seconds to enter four-digits

$prodcode = $AGI->get_data('product-code-prompt',1,4);

$quantity = &quan_by_code($prodcode);

if ($quantity != 0) {
$AGI->stream_file('there-are');
$AGI->say_digits($quantity);
$AGI->stream_file('available');
} else {
$AGI->stream_file('none-available');
}

$AGI->stream_file('goodbye');
$AGI->hangup;
exit;




sub quan_by_code {
### Takes a product code as input, then returns the quantity available.
my $code = shift;
my $dbh = open_connection();
my $sql = "SELECT code, quantity FROM product WHERE code=\'$code\' LIMIT 1";
my $sth = $dbh->prepare($sql);
$sth->execute or die "Unable to execute SQL query: $dbh->errstr\n";
my $row = $sth->fetchrow_arrayref;
$sth->finish;
$dbh->disconnect;
if ( ($code == $row->[0]) && ($code != 0) ) {
return $row[1];
} else {
return 0;
}
}

sub open_connection {
my $dsn = "mysql:dbname:localhost";
my $username = 'dbusername';
my $password = 'dbpassword';

return DBI->connect("DBI:$dsn",$username,$password) or die $DBI::errstr;
}
---


Of course, you'll need to record prompts and responses for:
  welcome
  product-code-prompt
  there-are
  available
  none-available
  goodbye


Hope this get you started,
rm
-
 Roderick Montgomery   [EMAIL PROTECTED]   http://thecomplex.com/>
the fool stands only to fall, but the wise trip on grace... [Sarah Masen]
-
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Re: [Asterisk-Users] Detecting channel status

[James Golovich]

What type of channel are you trying to do this for?  I was thinking of
writing an app ZapIsAvail, that would check if a certain zap channel is in
use and if so continue on with priority+1, otherwise jump to priority+101

James


On Thu, 13 Mar 2003 [EMAIL PROTECTED] wrote:

> In the extensions.conf file, is there a way to detect if it is on hook 
> or off hook (without dialing it)?
> 
> Dave
> 
> ___
> Asterisk-Users mailing list
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> 

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[Asterisk-Users] Logging errors to syslog

[Jean-Pierre Denis]

Hi,

is there a way to log errors to syslog ?

Currently logger.conf is logging everything in /var/log/asterisk/messages
but I would like to see the errors in my /dev/tty12 console on my server.

[logfiles]
console => notice,warning,error
messages => notice,warning,error

Thanks,

Jean-Pierre Denis
jp at msfree dot ca


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