[asterisk-users] help
Derek Kruger
dkruger at ci.safford.az.us
Wed Oct 4 15:13:32 MST 2006
-----Original Message-----
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Sent: Wednesday, October 04, 2006 11:03 AM
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Subject: asterisk-users Digest, Vol 27, Issue 16
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Today's Topics:
1. Re: Spandsp and tif (Steve Underwood)
2. Zaptel problems (Shea, Matt)
3. Call Interception (Delca)
4. RE: Zaptel problems (Colin Anderson)
5. Asterisk 1.4 moh - mohsuggest (Douglas Garstang)
6. Re: Zaptel problems (Bernardo Vieira)
7. Re: Call Interception (Bernardo Vieira)
8. Re: Digium TDM or SPA-3000? (Jay R. Ashworth)
9. Oneway audio (Giordano Grandis)
10. digium compatibility notes (marek cervenka)
11. Re: Call Interception (Jay R. Ashworth)
12. Transfer feature - howto? (Mike)
13. Re: Call Interception (Time Bandit)
14. New tutorial - peering two * servers using IAX (lenz)
15. Re: Call Interception (Steve Edwards)
16. snom 360: how to make record button working ? (noro kamen)
17. Re: Call Interception (Don)
18. SIP client that runs on Linux or Solaris through X Windows?
(Joe)
19. Re: Where is the PlayDTMF command? (Frank Church)
20. Wouldn't Tri-tone detection in Dial() be cool? (Steve Murphy)
21. Re: T1 incoming connects, but no sound (Nathan Bell)
22. Intel Chipset 945p compatible? (R.R. Libera)
23. RE: New tutorial - peering two * servers using IAX
(Douglas Garstang)
24. Need USA DID + trunk provider (R.R. Libera)
25. Need USA DID + trunk provider (R.R. Libera)
----------------------------------------------------------------------
Message: 1
Date: Thu, 05 Oct 2006 00:22:15 +0800
From: Steve Underwood <steveu at coppice.org>
Subject: Re: [asterisk-users] Spandsp and tif
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <4523DFB7.2040605 at coppice.org>
Content-Type: text/plain; charset=UTF-8; format=flowed
Giedrius Augys wrote:
> Hi,
> Now I'm testing faxes with spandsp. I have problems that spandsp do
> not add headers to fax page: LOCALHEADERINFO.
> Please help me.
There is a bug in adding page header with spandsp-0.0.2pre26. I have
fixed this in the development code, but I haven't yet put the fix into
the 0.0.2prexx series.
Steve
------------------------------
Message: 2
Date: Wed, 4 Oct 2006 12:27:02 -0400
From: "Shea, Matt" <Matt.Shea at ONSTAR.com>
Subject: [asterisk-users] Zaptel problems
To: <asterisk-users at lists.digium.com>
Message-ID:
<E251C4244455254CBF8F7F03465B43A30175772B at USRN4EX0ONS01.onstar.ad.gm.com>
Content-Type: text/plain; charset="us-ascii"
I'm running Asterisk/Zaptel on a Fedora Core 4 machine. The software
runs ok with one exception. Zaptel appears to load OK on bootup, but
when you check it on login, zttool still shows red/nop alarms on the T1
lines. I have to manually start it again for the alarms to disappear
and the T1 lines to function properly. I've updated the drivers to
1.2.9.1 and double checked my configuration files to no effect. Any
suggestions will be much appreciated.
Matt
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Message: 3
Date: Wed, 4 Oct 2006 16:31:51 +0000
From: Delca <delcas at gmail.com>
Subject: [asterisk-users] Call Interception
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<9d0acd5b0610040931t357d6bbel438f680cbd6cd08a at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi,
I'm deploying an asterisk PBX for a Call Center and i was ordered to
check if the Customer Support Supervisor could intercept the calls so
they can check how they employees work with Asterisk.
I read http://www.voip-info.org/wiki/view/Intercepting+SIP+Calls and i
got a clue about intercepting calls. But actually i wanted to know if
someone have experience with this sort of things.
Cheers!
Santiago
------------------------------
Message: 4
Date: Wed, 4 Oct 2006 10:36:07 -0600
From: Colin Anderson <ColinA at landmarkmasterbuilder.com>
Subject: RE: [asterisk-users] Zaptel problems
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
<asterisk-users at lists.digium.com>
Message-ID:
<E251506D3758AA4882130317D52ACD324A9D1D at land-edm-hs2.landmarkhomes.net>
Content-Type: text/plain; charset="iso-8859-1"
Had the same problem in fc2. Solution was to chkconfig zaptel off chkconfig
asterisk off then in rc.local modprobe wct1xxp (i think) then ztcfg then
start safe_asterisk. Dunno why.
Hey, is OnStar using Asterisk? Details, please.
-----Original Message-----
From: Shea, Matt [mailto:Matt.Shea at ONSTAR.com]
Sent: Wednesday, October 04, 2006 10:27 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Zaptel problems
I'm running Asterisk/Zaptel on a Fedora Core 4 machine. The software runs
ok with one exception. Zaptel appears to load OK on bootup, but when you
check it on login, zttool still shows red/nop alarms on the T1 lines. I
have to manually start it again for the alarms to disappear and the T1 lines
to function properly. I've updated the drivers to 1.2.9.1 and double
checked my configuration files to no effect. Any suggestions will be much
appreciated.
Matt
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Message: 5
Date: Wed, 4 Oct 2006 10:37:00 -0600
From: "Douglas Garstang" <dgarstang at oneeighty.com>
Subject: [asterisk-users] Asterisk 1.4 moh - mohsuggest
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<645FEC31A18FE54A8721500CDD55A7B6035D0B45 at mail.oneeighty.com>
Content-Type: text/plain; charset="iso-8859-1"
I'm trying to get moh working correctly in Asterisk 1.4. A complete lack of
documentation isn't helping much.
I have this in sip.conf:
[3254101]
type=friend
...
mohsuggest=class1
[3254102]
type=friend
...
mohsuggest=class2
A call is bridged between the two extensions. When 3254102 puts 3254101 on
hold, 3254101 hears moh class 'class2' which is correct. However, when
3254101 puts 3254102 on hold, the 3254102 hears the default music class.
Why?
Doug.
------------------------------
Message: 6
Date: Wed, 04 Oct 2006 13:39:47 -0300
From: Bernardo Vieira <bernardo.vieira at terra.com.br>
Subject: Re: [asterisk-users] Zaptel problems
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <4523E3D3.5040603 at terra.com.br>
Content-Type: text/plain; charset=ISO-8859-1
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Is ztcfg running at boot after the zaptel modules have been loaded?
What's the output of ztcfg?
Shea, Matt wrote:
> I'm running Asterisk/Zaptel on a Fedora Core 4 machine. The software
> runs ok with one exception. Zaptel appears to load OK on bootup, but
> when you check it on login, zttool still shows red/nop alarms on the T1
> lines. I have to manually start it again for the alarms to disappear
> and the T1 lines to function properly. I've updated the drivers to
> 1.2.9.1 and double checked my configuration files to no effect. Any
> suggestions will be much appreciated.
>
>
>
> Matt
>
>
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
- --
"What most profoundly divides two men is a different sense and degree of
cleanliness. What help is all honesty and mutual utility, what help is
all the good will for each other: in the end the fact remains-they can't
stand each other?s smell!"
- - Nietzsche
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------------------------------
Message: 7
Date: Wed, 04 Oct 2006 13:45:21 -0300
From: Bernardo Vieira <bernardo.vieira at terra.com.br>
Subject: Re: [asterisk-users] Call Interception
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <4523E521.3070900 at terra.com.br>
Content-Type: text/plain; charset=ISO-8859-1
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Do you need to actively intercept the call (i.e. participate in the
conversation) or just listen in the channel? For the latter you can just
use the ChanSpy application.
Delca wrote:
> Hi,
>
> I'm deploying an asterisk PBX for a Call Center and i was ordered to
> check if the Customer Support Supervisor could intercept the calls so
> they can check how they employees work with Asterisk.
>
> I read http://www.voip-info.org/wiki/view/Intercepting+SIP+Calls and i
> got a clue about intercepting calls. But actually i wanted to know if
> someone have experience with this sort of things.
>
>
> Cheers!
> Santiago
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
- --
"What most profoundly divides two men is a different sense and degree of
cleanliness. What help is all honesty and mutual utility, what help is
all the good will for each other: in the end the fact remains-they can't
stand each other?s smell!"
- - Nietzsche
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------------------------------
Message: 8
Date: Wed, 4 Oct 2006 12:48:10 -0400
From: "Jay R. Ashworth" <jra at baylink.com>
Subject: Re: [asterisk-users] Digium TDM or SPA-3000?
To: asterisk-users at lists.digium.com
Message-ID: <20061004164810.GD17956 at cgi.jachomes.com>
Content-Type: text/plain; charset=us-ascii
On Tue, Oct 03, 2006 at 09:41:04PM -0600, Joseph wrote:
> Since you are just planning it, keep in mind to select something that
> will be IPv6 ready.
I don't know that this is necessary, actually.
If I understood the OP correctly, he's terminating line/trunk
appearances which arrive at his switch analog, so the IP side of a
media gateway would be on a private LAN, and therefore IPv6 would be
entirely unnecessary, no?
Cheers,
-- jra
--
Jay R. Ashworth
jra at baylink.com
Designer Baylink RFC
2100
Ashworth & Associates The Things I Think '87
e24
St Petersburg FL USA http://baylink.pitas.com +1 727 647
1274
"That's women for you; you divorce them, and 10 years later,
they stop having sex with you." -- Jennifer Crusie; _Fast_Women_
------------------------------
Message: 9
Date: Wed, 4 Oct 2006 18:46:53 +0200
From: "Giordano Grandis" <g.grandis at invidea.it>
Subject: [asterisk-users] Oneway audio
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<62A244761070314C8DC5860B0ED131FD0232AB at homer.tecnojest.it>
Content-Type: text/plain; charset="us-ascii"
Hi list,
I'm testing transfer with sip re-invite and bristuff-0.0.8-RCn using an
HFC pci card connetced directly to telco; this is what happen:
1. SIP phone calls a mobile phone (or another residential phone)
2. The called party answers the call
3. Now the sip phone puts on hold the call and calls another sip phone
4. They speak normally
5. Now hte phone that called the mobile transfer the session to the
second one phone
6. The sip phone can hear the mobile phone, but not viceversa.
This works perfectly if i try a blind transfer.
Whaere could be the problem? On the phone....on asterisk ?
Anyone can help me?
Thanks in advance
Giordano
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Message: 10
Date: Wed, 4 Oct 2006 18:53:38 +0200 (CEST)
From: marek cervenka <cervajs at fpf.slu.cz>
Subject: [asterisk-users] digium compatibility notes
To: asterisk-users at lists.digium.com
Message-ID: <Pine.LNX.4.61.0610041847220.28620 at axpsu.fpf.slu.cz>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
hi,
what is mean by "partially incompatible" in
http://www.digium.com/en/docs/misc/compatibility_notes.php
i have server with E7221+te110p mobo and i think i dont have any problems
thanks
---------------------------------------
Marek Cervenka
=======================================
------------------------------
Message: 11
Date: Wed, 4 Oct 2006 12:57:19 -0400
From: "Jay R. Ashworth" <jra at baylink.com>
Subject: Re: [asterisk-users] Call Interception
To: asterisk-users at lists.digium.com
Message-ID: <20061004165719.GF17956 at cgi.jachomes.com>
Content-Type: text/plain; charset=us-ascii
On Wed, Oct 04, 2006 at 04:31:51PM +0000, Delca wrote:
> I'm deploying an asterisk PBX for a Call Center and i was ordered to
> check if the Customer Support Supervisor could intercept the calls so
> they can check how they employees work with Asterisk.
The call center bix calls that "Service Observing", and I believe
that yeah, you can do that with *. I base that thought on some
things I've read on the mailing list this week and last; if you've
just subscribed, you might want to scan the archives.
Cheers,
-- jra
--
Jay R. Ashworth
jra at baylink.com
Designer Baylink RFC
2100
Ashworth & Associates The Things I Think '87
e24
St Petersburg FL USA http://baylink.pitas.com +1 727 647
1274
"That's women for you; you divorce them, and 10 years later,
they stop having sex with you." -- Jennifer Crusie; _Fast_Women_
------------------------------
Message: 12
Date: Wed, 4 Oct 2006 13:01:50 -0400
From: "Mike" <list at virtutel.ca>
Subject: [asterisk-users] Transfer feature - howto?
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <003001c6e7d6$ca15ed10$0a01a8c0 at MIKE>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
My setup is the following: Voip provider-----------(SIP
DID)----------->Asterisk box----(SIP through a termination
provider)--------------->multiple cell phones.
The cell phones each have their extension (201,202,203,204) and I'd like to
be able to have them transfer a call to somebody else. Ex: Prospect calls
extension 201, talks to the salesgy, who forwards him to the tech guru
somehow.
My guess is I have to use the transfer feature found in feature.conf. I
tried, no success. What Wiki page do I need to look at to get details on
this?
Mike
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Message: 13
Date: Wed, 4 Oct 2006 13:07:21 -0400
From: "Time Bandit" <timebandit001 at gmail.com>
Subject: Re: [asterisk-users] Call Interception
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<1e2050d50610041007q6195362eh5b4047612caa56df at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> I'm deploying an asterisk PBX for a Call Center and i was ordered to
> check if the Customer Support Supervisor could intercept the calls so
> they can check how they employees work with Asterisk.
have a look at these :
http://www.voip-info.org/wiki-Asterisk+cmd+ZapBarge
and
http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy
hth
------------------------------
Message: 14
Date: Wed, 04 Oct 2006 19:10:35 +0200
From: lenz <lenz-ml at oinko.net>
Subject: [asterisk-users] New tutorial - peering two * servers using
IAX
To: "asterisk-users at lists.digium.com"
<asterisk-users at lists.digium.com>
Message-ID: <op.tgwpzxz5uxa8ts at smtp.ngi.it>
Content-Type: text/plain; format=flowed; delsp=yes;
charset=iso-8859-15
Hi list,
today I have been teaching a class on * and have found that many students
find it quite hard to understand how setting up IAX peering between two
servers may work. So I prepared a little step by step tutorial hoping it
might be useful to someone in the future.
See it at http://astrecipes.net/index.php?n=204
Comments and corrections are welcome. The site is a wiki, so feel free to
modify and improve.
l.
--
Home of QueueMetrics - http://queuemetrics.loway.it
------------------------------
Message: 15
Date: Wed, 4 Oct 2006 10:20:39 -0700 (PDT)
From: Steve Edwards <asterisk.org at sedwards.com>
Subject: Re: [asterisk-users] Call Interception
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <Pine.LNX.4.64.0610040959230.16990 at fs.sedwards.com>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
Check out meetme.
We create a meetme conference for each agent when the agent logs in. As
customer's call in, the call is matched (by DNIS and IVR) to the "longest
idle" agent with the required skill (or any agent if no agent with the
matching skill is available).
The supervisors can join any conference "pre-muted" by entering the agent
ID. If needed, they can "un-mute" and contribute to the call or kick the
agent and take the call.
It took a couple of AGI's and some tweaks to app_meetme.c for custom
whispers at the start of the call to tell the agent the type of call while
the customer hears "ring" and kicking the agents, but we're pretty happy
at this point.
On Wed, 4 Oct 2006, Delca wrote:
> Hi,
>
> I'm deploying an asterisk PBX for a Call Center and i was ordered to
> check if the Customer Support Supervisor could intercept the calls so
> they can check how they employees work with Asterisk.
>
> I read http://www.voip-info.org/wiki/view/Intercepting+SIP+Calls and i
> got a clue about intercepting calls. But actually i wanted to know if
> someone have experience with this sort of things.
>
>
> Cheers!
> Santiago
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
Thanks in advance,
------------------------------------------------------------------------
Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
------------------------------
Message: 16
Date: Wed, 4 Oct 2006 19:33:50 +0200
From: "noro kamen" <noroast at gmail.com>
Subject: [asterisk-users] snom 360: how to make record button working
?
To: asterisk-users at lists.digium.com
Message-ID:
<6f66c2f50610041033m39c64d5cibaee7b46232b5d50 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi,
I'd like to make record button working on snom 320/360 + asterisk.
As I learned from wireshark output, the phone produces SIP info
message "Record: on", while record button pressed.
Can anybody give me an advice, how to teach asterisk to understand
that SIP info message and start recording ?
TIA
noro
------------------------------
Message: 17
Date: Wed, 4 Oct 2006 13:34:30 -0400
From: "Don" <sales at xwebfactor.com>
Subject: Re: [asterisk-users] Call Interception
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID: <004301c6e7db$5a6dfed0$1d01a8c0 at shizznit2000>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=original
If they are just trying to listen in you can use zapbarge
----- Original Message -----
From: "Jay R. Ashworth" <jra at baylink.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, October 04, 2006 12:57 PM
Subject: Re: [asterisk-users] Call Interception
> On Wed, Oct 04, 2006 at 04:31:51PM +0000, Delca wrote:
>> I'm deploying an asterisk PBX for a Call Center and i was ordered to
>> check if the Customer Support Supervisor could intercept the calls so
>> they can check how they employees work with Asterisk.
>
> The call center bix calls that "Service Observing", and I believe
> that yeah, you can do that with *. I base that thought on some
> things I've read on the mailing list this week and last; if you've
> just subscribed, you might want to scan the archives.
>
> Cheers,
> -- jra
> --
> Jay R. Ashworth
> jra at baylink.com
> Designer Baylink RFC
> 2100
> Ashworth & Associates The Things I Think '87
> e24
> St Petersburg FL USA http://baylink.pitas.com +1 727 647
> 1274
>
> "That's women for you; you divorce them, and 10 years later,
> they stop having sex with you." -- Jennifer Crusie; _Fast_Women_
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.1.407 / Virus Database: 268.12.12/462 - Release Date: 10/3/2006
>
>
------------------------------
Message: 18
Date: Wed, 4 Oct 2006 12:35:42 -0500
From: Joe <joe.uelk at gmail.com>
Subject: [asterisk-users] SIP client that runs on Linux or Solaris
through X Windows?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<9d6632e0610041035p2d34fac4g5412041ab050af0c at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello,
I'm looking for a SIP client that work with Asterisk that will run on
Linux or Solaris and will work with X Windows. I know X won't all the
media to work but I'm really only interested in SIP signaling.
Thanks!
Joe
------------------------------
Message: 19
Date: Wed, 4 Oct 2006 18:36:04 +0100
From: "Frank Church" <voipfc at googlemail.com>
Subject: Re: [asterisk-users] Where is the PlayDTMF command?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<84b7c6460610041036n693e9f97l1fdf7ff386f22d9b at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Moises, do you know if the DTMF event in bug 6082 made it into version 1.4?
When I last tried to compile that branch it needed the latest version
of make 3.81, which trunk did not, and caused me to wonder if it had
been committed to trunk.
The DTMF detection events in trunk did not also function, and made we
wonder if they had been taken out or required some additional post
install configuration, as they worked well before
That bug thread seems to has gone rather quiet now.
On 10/4/06, Moises Silva <moises.silva at gmail.com> wrote:
> You are just not loading the module. Connect to Asterisk terminal
>
> # asterisk -vr
>
> and load the module
>
> CLI> load app_senddtmf.so
>
>
> Best Regards.
>
> On 10/4/06, Jan du Toit <jan.du.toit at decisionworx.com> wrote:
> > Hi all.
> >
> > I'm confused. On http://www.voip-info.org/wiki-Asterisk+manager+API it
> > says that the PlayDTMF command is available since version 1.2.8. I
> > upgraded to version 1.2.12.1 but I cant find it if I type in "show
> > manager commands" there is no PlayDTMF command. According to resources
> > on the internet this action links to the send dtmf application. I
> > checked the source code under the apps folder and it is their!
> > |apps/app_senddtmf.c
> >
> > |Is it not compiling? Why is this function not available to me?
> >
> > Please help.
> > Thanks.||
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> --
> "Su nombre es GNU/Linux, no solamente Linux, mas info en
http://www.gnu.org"
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
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>
------------------------------
Message: 20
Date: Wed, 04 Oct 2006 11:35:53 -0600
From: Steve Murphy <murf at digium.com>
Subject: [asterisk-users] Wouldn't Tri-tone detection in Dial() be
cool?
To: asterisk-users at lists.digium.com
Message-ID: <1159983353.3638.58.camel at monster>
Content-Type: text/plain; charset="us-ascii"
To: Whom it may Concern:
Well, it hit me last night as I was falling asleep... Asterisk (in the
app Zapateller)
can emit the tri-tone (you know beep-Beep-BEEP... The number you have
dialed is no longer
in service. Please check the number and...blah, blah)
Well, it occurred to me that, for the sake of orthogonality, wouldn't it
be cool if
Asterisk's Dial function also detected that tone, with an option to
immediately
hang up if it occurred, with a result code of WRONGNUMBER or NOSERVICE
or whatever?
It also occurred to me that this **might** only be useful to the hated
and dreaded
autodialers that telemarketers use. Even so, it wouldn't hurt me any
more than normal
to have asterisk-based autodialers detect that and get me off their call
lists!
Hah, I'm not trying to imply that I have the skill set right now to
implement this,
nor am I trying to convince anyone right now to do it. The idea just hit
me, and
I wonder if it has already been done somewhere?
murf
--
Steve Murphy
Software Developer
Digium
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Message: 21
Date: Wed, 04 Oct 2006 11:45:03 -0600
From: Nathan Bell <nathanb at actarg.com>
Subject: Re: [asterisk-users] T1 incoming connects, but no sound
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <4523F31F.5080806 at actarg.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Mark Farver wrote:
> Nathan Bell wrote:
>
>> extensions.conf:
>> [from-ptsn]
>> exten => s,1,Answer()
>> exten => s,2,Playback(vm-goodbye)
>> exten => s,3,Hangup()
>>
> You might try adding a "wait(3)" command after the answer. Some
> analog lines do not pass audio immediately after being answered.
> (Something to do with how toll processing is handled)
>
> Mark
>
After adding in a "wait(3)" to the extensions.conf, at s,2, I still get
no audo be passed to me. However, I noticed that each time I place a
call, asterisk thinks that two calls are happening. Here's the log
output of what happens (all with no audio):
Oct 3 17:16:48 NOTICE[10763] chan_zap.c: Got event 18 (Ring Begin)...
Oct 3 17:16:48 VERBOSE[10763] logger.c: -- Executing
Answer("Zap/1-1", "") in new stack
Oct 3 17:16:48 DEBUG[10763] chan_zap.c: Took Zap/1-1 off hook
Oct 3 17:16:48 DEBUG[10763] chan_zap.c: Enabled echo cancellation on
channel 1
Oct 3 17:16:48 DEBUG[10763] chan_zap.c: Engaged echo training on channel 1
Oct 3 17:16:48 VERBOSE[10763] logger.c: -- Executing
Wait("Zap/1-1", "3") in new stack
Oct 3 17:16:48 VERBOSE[10766] logger.c: -- Starting simple switch
on 'Zap/2-1'
Oct 3 17:16:51 VERBOSE[10763] logger.c: -- Executing
Playback("Zap/1-1", "vm-goodbye") in new stack
Oct 3 17:16:51 DEBUG[10763] channel.c: Scheduling timer at 160 sample
intervals
Oct 3 17:16:51 VERBOSE[10763] logger.c: -- Playing 'vm-goodbye'
(language 'en')
Oct 3 17:16:52 DEBUG[10763] channel.c: Scheduling timer at 0 sample
intervals
Oct 3 17:16:52 DEBUG[10763] channel.c: Scheduling timer at 0 sample
intervals
Oct 3 17:16:52 VERBOSE[10763] logger.c: -- Executing
Hangup("Zap/1-1", "") in new stack
Oct 3 17:16:52 VERBOSE[10763] logger.c: == Spawn extension
(from-ptsn, s, 4) exited non-zero on 'Zap/1-1'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '(null)'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '(null)'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is 's'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is 'from-ptsn'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is 'Zap/1-1'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '(null)'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is 'Hangup'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '(null)'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '2006-10-03 17:16:48'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '2006-10-03 17:16:48'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '2006-10-03 17:16:52'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '4'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '4'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is 'ANSWERED'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is 'DOCUMENTATION'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '(null)'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '1159917403.6'
Oct 3 17:16:52 DEBUG[10763] pbx.c: Function result is '(null)'
Oct 3 17:16:52 DEBUG[10763] chan_zap.c: Hangup: channel: 1 index = 0,
normal = 17, callwait = -1, thirdcall = -1
Oct 3 17:16:52 DEBUG[10763] chan_zap.c: disabled echo cancellation on
channel 1
Oct 3 17:16:52 DEBUG[10763] chan_zap.c: Set option TDD MODE, value:
OFF(0) on Zap/1-1
Oct 3 17:16:52 DEBUG[10763] chan_zap.c: Updated conferencing on 1, with
0 conference users
Oct 3 17:16:52 VERBOSE[10763] logger.c: -- Hungup 'Zap/1-1'
Oct 3 17:16:53 NOTICE[10766] chan_zap.c: Got event 18 (Ring Begin)...
Oct 3 17:16:53 VERBOSE[10766] logger.c: -- Executing
Answer("Zap/2-1", "") in new stack
Oct 3 17:16:53 DEBUG[10766] chan_zap.c: Took Zap/2-1 off hook
Oct 3 17:16:53 DEBUG[10766] chan_zap.c: Enabled echo cancellation on
channel 2
Oct 3 17:16:53 DEBUG[10766] chan_zap.c: Engaged echo training on channel 2
Oct 3 17:16:53 VERBOSE[10766] logger.c: -- Executing
Wait("Zap/2-1", "3") in new stack
Oct 3 17:16:56 VERBOSE[10766] logger.c: -- Executing
Playback("Zap/2-1", "vm-goodbye") in new stack
Oct 3 17:16:56 DEBUG[10766] channel.c: Scheduling timer at 160 sample
intervals
Oct 3 17:16:56 VERBOSE[10766] logger.c: -- Playing 'vm-goodbye'
(language 'en')
Oct 3 17:16:57 DEBUG[10766] channel.c: Scheduling timer at 0 sample
intervals
Oct 3 17:16:57 DEBUG[10766] channel.c: Scheduling timer at 0 sample
intervals
Oct 3 17:16:57 VERBOSE[10766] logger.c: -- Executing
Hangup("Zap/2-1", "") in new stack
Oct 3 17:16:57 VERBOSE[10766] logger.c: == Spawn extension
(from-ptsn, s, 4) exited non-zero on 'Zap/2-1'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '(null)'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '(null)'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is 's'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is 'from-ptsn'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is 'Zap/2-1'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '(null)'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is 'Hangup'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '(null)'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '2006-10-03 17:16:53'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '2006-10-03 17:16:53'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '2006-10-03 17:16:57'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '4'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '4'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is 'ANSWERED'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is 'DOCUMENTATION'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '(null)'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '1159917408.7'
Oct 3 17:16:57 DEBUG[10766] pbx.c: Function result is '(null)'
Oct 3 17:16:57 DEBUG[10766] chan_zap.c: Hangup: channel: 2 index = 0,
normal = 18, callwait = -1, thirdcall = -1
Oct 3 17:16:57 DEBUG[10766] chan_zap.c: disabled echo cancellation on
channel 2
Oct 3 17:16:57 DEBUG[10766] chan_zap.c: Set option TDD MODE, value:
OFF(0) on Zap/2-1
Oct 3 17:16:57 DEBUG[10766] chan_zap.c: Updated conferencing on 2, with
0 conference users
Oct 3 17:16:57 VERBOSE[10766] logger.c: -- Hungup 'Zap/2-1'
Notice that "Starting simple switch on 'Zap/2-1'" happens almost
immediatly after it starts on Zap/1-1. Caling in multiple times results
in nearly identical output, with the only difference being which line
Zap/2-1 starts at.
>
>> Testing my setup from a channel bank seems to work just fine
>> (slightly different zaptel.conf and zapata.conf).
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------
Message: 22
Date: Wed, 04 Oct 2006 15:00:23 -0300
From: "R.R. Libera" <astecomm at gmail.com>
Subject: [asterisk-users] Intel Chipset 945p compatible?
To: Asterisk-Users List <asterisk-users at lists.digium.com>
Message-ID: <4523F6B7.5060603 at gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello,
I had recently install an Asterisk PBX into a brand new PC: Intel
Pentium D 3.4GHz Dual-Core + P5LD2 motherboard + SATA HDD.
I´m planning to handle one E1 with a TE110P interface and I want to know
the compatibility between TE110P and Intel 945P chipset. I already buy
the hardware and the only thing I got into account was the compatibility
between the hardware selected and Debian Sarge (the distro I selected).
I´ll accept any suggestion, advice or comment. Thanks in advance.
R.R. Libera
------------------------------
Message: 23
Date: Wed, 4 Oct 2006 12:01:20 -0600
From: "Douglas Garstang" <dgarstang at oneeighty.com>
Subject: RE: [asterisk-users] New tutorial - peering two * servers
using IAX
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<645FEC31A18FE54A8721500CDD55A7B6035D0B47 at mail.oneeighty.com>
Content-Type: text/plain; charset="iso-8859-15"
How about preparing a step by step guide to DUNDi? Good luck with that though
because base DUNDi docs are rarer than periodic element #114 in the known
universe.
Doug.
> -----Original Message-----
> From: lenz [mailto:lenz-ml at oinko.net]
> Sent: Wednesday, October 04, 2006 11:11 AM
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] New tutorial - peering two *
> servers using IAX
>
>
>
> Hi list,
> today I have been teaching a class on * and have found that
> many students
> find it quite hard to understand how setting up IAX peering
> between two
> servers may work. So I prepared a little step by step
> tutorial hoping it
> might be useful to someone in the future.
>
> See it at http://astrecipes.net/index.php?n=204
>
> Comments and corrections are welcome. The site is a wiki, so
> feel free to
> modify and improve.
> l.
>
>
>
>
> --
> Home of QueueMetrics - http://queuemetrics.loway.it
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
------------------------------
Message: 24
Date: Wed, 04 Oct 2006 15:03:59 -0300
From: "R.R. Libera" <astecomm at gmail.com>
Subject: [asterisk-users] Need USA DID + trunk provider
To: Asterisk-Users List <asterisk-users at lists.digium.com>
Message-ID: <4523F78F.3010506 at gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello,
I need an USA DID + 15 b-channels. The only option I already have is
OpenVox and I want to see some alternatives. Sound quality is my
priority. Thanks in advance.
R.R Libera
------------------------------
Message: 25
Date: Wed, 04 Oct 2006 15:05:26 -0300
From: "R.R. Libera" <astecomm at gmail.com>
Subject: [asterisk-users] Need USA DID + trunk provider
To: Asterisk-Users List <asterisk-users at lists.digium.com>
Message-ID: <4523F7E6.4040401 at gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Sorry, when I said "OpenVox" I should say "VoxBone".
Regards,
------------------------------
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