RE: [Asterisk-Users] Alchemy branch integration, one way audio

[Stuart Mackintosh]

Does this mean i will need a vocoder card to make this work?

> The vocoder card has the ability to do both alaw and ulaw 64 codecs ala
> asterisk

> 
> I am attempting to connect an asterisk system into an existing Network
> Alchemy branch. This system supports h.323 and has an optional vocoder card
> (Very expensive!) to enable other codecs.
> 
> I have achieved one-way audio in either direction but cannot get 2way. I
> feel the problem is codec related. I have selected Transparent 64k as the
> codec (I believe the only one supported without the vocoder card) but unsure
> what I should set at the asterisk end.
> 
> Using recent asterisk, oh323 0.63b and oh323 1.13
> 

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Re: [Asterisk-Users] Sip Outbound Proxy

[Olle E. Johansson]

Chad Brown wrote:
How do you configure an outbound proxy for Asterisk? If the sip call is 
not local I want everything to go to a designated sip proxy.
In the standard chan_sip, there's no support for outbound proxy.
In my chan_sip2 test channel, I have that support. Please test!
If I get enough positive feedback, I might try to port this to the standard
chan_sip after Astricon and 1.0.
chan_sip2 is to be found in bugs.digium.com
/Olle
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Re: [Asterisk-Users] Suggested Motherboard for TE410P

[Adam Goryachev]

On Sat, 2004-09-11 at 14:25, Kevin P. Fleming wrote:
> Adam Goryachev wrote:
> 
> > PS, in case you are wondering, I (and my supplier) have spent hours
> > looking at different motherboard specs, and so far haven't been able to
> > find anything suitable (except a dual opteron motherboard and just using
> > a single CPU).
> 
> VIA K8T800 Pro chipset boards that support Athlon 64 and 3.3v/5v PCI:
> 
> MSI K8T Neo2-F
> 
> nVidia nForce 3 (or )Ultra chipset boards that support Athlon 64 and 
> 3.3v/5v PCI (PCI-2.3 even):
> 
> MSI K8N Neo Platinum
> MSI K8N Neo2 Platinum

Except, if you look at the motherboard detailed picture of all these,
and compare to the picture on the digium website showing the physical
PCI bus requirements, the 3.3V PCI version, won't fit into the 3.3V PCI
slots on these motherboards.

Have you actually used them? Is the picture just wrong?

Thanks,
Adam


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RE: [Asterisk-Users] Suggested Motherboard for TE410P

[Adam Goryachev]

On Sat, 2004-09-11 at 13:37, Scott Stingel wrote:
> Hi Adam-
> 
> I'm wondering if the TE405P might be a better choice, since it's 5 volt PCI
> and may allow you to consider a wider selection of motherboards.  Sounds
> like you may not need the latest and fastest motherboards, which often use
> the 64bit 3.3v slots as you've probably found.

One reason I am keen to use the 3.3V version is that I currently have
for my own office, the 5V version. Since I first installed it, some
remote parties always complain of poor quality phone lines. ie, it is
always the same people that complain, but not everyone complains. It
doesn't seem to matter whether I use an analog phone connected to a
TDM40B, or I use a budgetone, or even a polycom IP 600.
One person suggested that I try swapping to the 3.3V card, as it may
resolve my issue, so, that is firstly what I am using for my customer,
but in-between I will test it for myself and see if it helps.
Also, the 5V version is not approved for use in Australia.

I haven't been able to hear the problem in calls recorded using monitor,
even though the person complained. However, I sent some fax audio files
created from rxfax, to coppice, and he said:
"The audio is quite distorted, and it takes a bit longer for the
adaptive filter to settle than on most data I have looked at. There are
no audio skips, though. This is just more distorted than usual. I think
the next version of spandsp will deal with your problems. This is the
first audio of this nature I have received from a user. Thanks."

So, it would seem that I have something unusual happening, possibly due
to my configuration/telco circuits/something, but I haven't been able to
find anything that really makes any difference.

PS, it is easily tested/re-producible, to the point where I can call
some customers, and they know it is me before they even finish saying
"hello". Usually, they say "hell, oh, hi adam..."

Regards,
Adam


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Re: [Asterisk-Users] Re: Line death not recognized on TDM400P?

[matt . riddell]

On 10 Sep 2004 at 16:36, Jinsong Liao wrote:
> Just tried the latest CVS version with "#define ZAP_CHECK_HOOKSTATE"
> in chan_zap.c.  I am using a TDM400P with 4 FXO modules.  Only port 1
> is connected to a phone line.
> 
> When * starts, I cannot make any outgoing call.  All 4 Zap channels
> are unavailable.  "zap show channel 1" indicated "Actual Hook State:
> Onhook" just like "zap show channel 2 (or 3, 4)" , even though channel
> 1 is the one with battery.
> 
> If I unplug the RJ11 cable and then plug it back in, or if I make an
> incoming call then hang up, "zap show channel 1" will indicate "Actual
> Hook State: Offhook".  After this I can make outgoing calls as Zap/1-1
> is now available.
> 
> Is there anyway to make * correctly recognize the hook state upon
> startup?  Thanks in advance.

Well not really a solution, but if you're going to use the 
ZAP_CHECK_HOOKSTATE you may as well start everything up with no 
cables plugged in.  That way your state and the software's state will 
always match no matter how many lines you plug in or unplug (which is 
what my customer had needed).

Kind regards,

Matt Riddell
http://www.sineapps.com
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[Asterisk-Users] Making the Old PABX work with new * box

[alexb]

Afternoon All,

We are looking at replacing our Goldstar
GDK-162 (digital) system with a * box, however I would like to know if
there is know migration paths.

We have 4 x ISDN 2 services (8lines)
with 100 in dial numbers coming into out GDK-162 system. What I would like
to do and not sure if this can be done, 

would be to install the * server as
a bridge which could then filter out number which have been moved from
hard phones to softphones.

eg:
ISDN --|*|--|GDK-162|

or is there any other way of having
two systems (* and GDK) working side by side??

After typing this out I believe that
I have answered my own question with a simple NO but I live in hope.

Cheers
Alex

Technical Support Manager
http://www.sirca.org.au
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Re: [Asterisk-Users] asterisk make

[Thomas Niesel]

GDay

On Tue, Sep 14, 2004 at 12:36:01PM +0800, Dinesh wrote:
> cd ../asterisk
> # make clean; make install
> 
> 
> Hello when I do a make clean and make install, I get this error message on
> my asterisk box.
> 
> bdb1.a stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv   -lssl
> /usr/bin/ld: cannot find -lssl
   ^
You need some ssl(ssl-dev) support
Check your distro for more details on how it is called.

> collect2: ld returned 1 exit status
> make: *** [asterisk] Error 1
> 
> Any ideas?
> 
> Dinesh.
> 
> 
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-- 
Tho/\/\as
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Re: [Asterisk-Users] Astricon tutorials :: Open for registration again

[matt . riddell]

> You are free to choose any tutorial, as long as we have seats
> available. We are also researching the possibility to tape the
> tutorials to make them available on line or on dvd later on.

What was the end result of streaming?  Is this not happening now?

Cheers,

Matt Riddell
http://www.sineapps.com
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[Asterisk-Users] asterisk make

[Dinesh]

cd ../asterisk
# make clean; make install


Hello when I do a make clean and make install, I get this error message on
my asterisk box.

bdb1.a stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv   -lssl
/usr/bin/ld: cannot find -lssl
collect2: ld returned 1 exit status
make: *** [asterisk] Error 1

Any ideas?

Dinesh.


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[Asterisk-Users] asterisk make

[Dinesh]

cd ../asterisk
# make clean; make install


Hello when I do a make clean and make install, I get this error message on
my asterisk box.

bdb1.a stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv   -lssl
/usr/bin/ld: cannot find -lssl
collect2: ld returned 1 exit status
make: *** [asterisk] Error 1

Any ideas?

Dinesh.


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RE: [Asterisk-Users] PABX & VOIP Gateway

[Phil Stevens]

Thanks for the info.

>I don't quite understand what you need Austel compliance for if all you
want to do is link to a PBX. You don't need Austel >approval for that,
only if you connect directly to Telstra.

Can anyone please confirm this?




-Original Message-
From: Benjamin on Asterisk Mailing Lists
[mailto:[EMAIL PROTECTED] 
Sent: Tuesday, September 14, 2004 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] PABX & VOIP Gateway


On Tue, 14 Sep 2004 10:34:51 +1000, Phil Stevens
<[EMAIL PROTECTED]> wrote:
> I'm researching the possibility of using VOIP (SIP) with an existing 
> PABX system. Ideally, the setup would be to dial an outside line 
> through the PABX (that would actually link to the the VOIP gateway).
> 
> At this point I would prefer not to purchase a hardware-based VOIP 
> gateway. I would prefer to use a software-based gateway for research &

> testing purposes. Could anyone please describe a simple setup?
> 
> Naturally the connection to the gateway would have to be Austel 
> approved. I have seen references to the Netjet ISDN cards? I am having

> difficulties in finding information with regards to Austel compliance.

I don't quite understand what you need Austel compliance for if all you
want to do is link to a PBX. You don't need Austel approval for that,
only if you connect directly to Telstra.

Anyway, Voicetronix is an Aussie company that makes 4, 6 and 12 port
analog boards which work with Asterisk ...

http://www.voicetronix.com.au

I also seem to remember having read about some Digium board having
obtained Austel approval quite a while ago, so you may want to ask one
of Digium's resellers in AU about that ...

http://www.austechpartnerships.com/atp/

rgds
benjk

-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

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[Asterisk-Users] Galaxy Voice Release with RC2

[Kevin]

As a follow up to this post of the problem that I was unable to receive
inbound calls with Galaxy Voice. I had recently upgraded to asterisk RC2
when the problem started.  I downgraded back to a May release of the CVS
and things work fine.  Not sure if this is of interest to the
contributors of the RC2 code but I also had the same problem on another
box with RC2 and Galaxy Voice.



-Original Message-
From: Eric Wieling [mailto:[EMAIL PROTECTED] 
Sent: Monday, September 13, 2004 10:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Galaxy Voice Configuration Question

On Sun, 2004-09-12 at 21:25, Kevin wrote:
> I am using Galaxy Voice until recently I can receive any inbound
calls.
> If I remove the [galaxy voice] context in my sip file the call rings
in
> but I obviously can't make any outgoing calls.  Any suggestions?

Don't remove the [galaxyvoice] entry from your sip.conf

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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RE: [Asterisk-Users] Server load capabilities

[Eric Wieling]

On Mon, 2004-09-13 at 13:06, C. David Kading wrote:
> I Have addressed the scaling issue, according to one of the test cases I
> read, a single server should be capable of handling 50 concurrent voice
> calls. My main concern is its ability to run 15 different instances of * for
> 15 different customers simultaneously. Each client site should be able
> manage only their own connections and clients.

You cannot run 15 instances of Asterisk on the same system.  Heck, you
can't run more than one instance of Asterisk.  What you do is use
#include or include => lines to include customer managed files into
Asterisk's config files or use a database for this stuff (which is
non-trivial and I keep hearing of annoyances with database driven
configs)
-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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[Asterisk-Users] Read command without #

[bagattin jerome]

Hi, 

For my IVR, I use Read command. It works fine when
ending bu # but I can't get anything without ending by
# 
The wiki tell me is it possible with maxdigit option
but it don't work for me.

my command :

exten => 3,1,Read(ILE,as/iles,1)

Anybody can tell me howto do thanks

Another question about read command: 
Howto sup file option and keep maxdigits options ?
exten => 3,1,Read(ILE,1)  
give me :
Unable to open 1 (format UNKN): No such file or
directory  :-(

Thanks for all






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Re: [Asterisk-Users] compiling zaptel

[ivlok]

try making a symbolic link 
ln -s /usr/src/kernel-source-2.6.8 /usr/src/linux 
and see if it finds the source. 

On Mon, 2004-09-13 at 18:06, [EMAIL PROTECTED] wrote:
> Hello List!
> I am currently trying to compile zaptel.
> 
> A "make" gives me this error:
> In file included from /usr/src/zaptel/zaptel.c:40:
> /usr/src/zaptel/zconfig.h:10:27: linux/version.h: No such file or directory
> .
> .
> .
> Where in the Makefile do i have to change the path to my include files?
> I have the following version.h files:
> 
> /usr/src/kernel-headers-2.6.8-1-386/include/config/isdn/diversion.h
> /usr/src/kernel-headers-2.6.8-1-386/include/linux/version.h
> /usr/src/kernel-headers-2.6.8-1/include/linux/dvb/version.h
> /usr/src/kernel-headers-2.6.8-1/include/linux/version.h
> /usr/src/kernel-headers-2.6.8-1/include/pcmcia/version.h
> /usr/src/kernel-headers-2.6.8-1/include/sound/version.h
> /usr/src/kernel-source-2.6.8/arch/i386/math-emu/version.h
> /usr/src/kernel-source-2.6.8/drivers/net/sk98lin/h/skversion.h
> /usr/src/kernel-source-2.6.8/drivers/scsi/qla2xxx/qla_version.h
> /usr/src/kernel-source-2.6.8/fs/xfs/linux-2.6/xfs_version.h
> /usr/src/kernel-source-2.6.8/include/linux/dvb/version.h
> /usr/src/kernel-source-2.6.8/include/pcmcia/version.h
> /usr/src/kernel-source-2.6.8/include/sound/version.h
> 
> 
> 
> Thanks, Mario
> 
> 
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Re: [Asterisk-Users] Extending E1's over a Satellite link

[Arinze Izukanne]

OK. What I am thinking is, It is possible to install a server running Asterisk on each end of the Satellite link, with an E1 card in each Asterisk; and the boxes are configured to pick up incomming calls  on the E1 channels and pass them to the other box across the link to the other Asterisk PBX which in turn completes the call via the PSTN E1 on that side.
 
Basically the aim is to provide a solution that works similar to RAD's VMUX but using Asterisk as the engine.
 
I expect there would be codec translation along the way.
 
I have this concept but how to implement it is another issue. I need someone who has experience with Asterisk already to tell me how feasible this is and to what extent it would achieve the aim of reducing bandwidth and maintaining minimal latency.
 
 
Best regards
 
Arinze
Kannaiyan Natesan <[EMAIL PROTECTED]> wrote:




asterisk is a pbx software. I don't think there is a compression and uncompression utility except codec conversions. Even in those cases you can be sure that there will be loss of data as there is no lossless compression.
 
If you have any satellite transmission and reception card which can be interfaced for voice communication with asterisk kindly share your views here, we can discuss more on that.
 
Asterisk sends calls over internet very efficiently by reducing the ethernet overhead and the channel overheads and that is only in IAX protocol. May be this you can consider to extend for satellite communication.
 
If you have more information kindly let us know about it whether anything of the above matches to your interest.
 
-Kannaiyan
 

- Original Message - 
From: Arinze Izukanne 
To: [EMAIL PROTECTED] 
Sent: Monday, September 13, 2004 11:41 PM
Subject: [Asterisk-Users] Extending E1's over a Satellite link

hi
 I want to compress and trunk 2E1 capacity over a satellite SCPC link using asterisk. I am nw to asterisk and I need suggestions on how to implement this.
 
Best regards
 
Arinze 


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Re: [Asterisk-Users] New BudgeTone

[bagattin jerome]

 --- "Sys.Concept" <[EMAIL PROTECTED]> a écrit : 
> Is BudgeTone planning on releasing new model of
> their sip phone? Are
> there any better alternative in that price range?

Zultis zip 2 ?

> 
> -- 
> #Joseph
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[Asterisk-Users] agents and *8 pickupgroups

[Sam Tilders]

Hi folks,

Recently we assigned our users agent id's and switched to
having them use agentcallbacklogin instead of just ringing the phones
directly it's been going well for the most part.

Before we gave the users agent id's we had their sip configuration
set to incominglimit=1 for one line on their phone, to prevent
call waiting beeps.

If they wanted to pick up while already on a call they had to use *8, 
which we set speeddials for and gave them a second line per phone.

This worked well.

But since they have been logging on as agents the *8 has stopped working.
The log file just says "Nothing to pick up".

However, someone who is not logged in as an agent, can use *8 to
pick up the same call that didn't work for someone who was logged in.

Anyone else seen this sort of behaviour? 

The pickupgroup, callgroup configuration hasn't changed since it worked
before we used agents.

- Sam

-- 
-- 
Sam Tilders
[EMAIL PROTECTED]
(Move to Jupiter)
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Re: [Asterisk-Users] Extending E1's over a Satellite link

[Kannaiyan Natesan]

asterisk is a pbx software. I don't think there is 
a compression and uncompression utility except codec conversions. Even in 
those cases you can be sure that there will be loss of data as there is no 
lossless compression.
 
If you have any satellite transmission and reception card 
which can be interfaced for voice communication with asterisk kindly share your 
views here, we can discuss more on that.
 
Asterisk sends calls over internet very efficiently by 
reducing the ethernet overhead and the channel overheads and that is only in IAX 
protocol. May be this you can consider to extend for satellite 
communication.
 
If you have more information kindly let us know about it 
whether anything of the above matches to your interest.
 
-Kannaiyan
 

  - Original Message - 
  From: 
  Arinze 
  Izukanne 
  To: [EMAIL PROTECTED] 
  
  Sent: Monday, September 13, 2004 11:41 
  PM
  Subject: [Asterisk-Users] Extending E1's 
  over a Satellite link
  
  hi
   I want to compress and trunk 2E1 capacity over a satellite 
  SCPC link using asterisk. I am nw to asterisk and I need suggestions on 
  how to implement this.
   
  Best regards
   
  Arinze 
  
  
  ALL-NEW Yahoo! 
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  fun! 
  
  

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RE: [Asterisk-Users] Aasterisk SIP<->SIP No audio

[Larry Shields]

If you are running a software based firewall on either client be sure to
open the correct ports (or disable if possible).  I ran into a similar
problem after I upgraded my WinXP Pro clients to SP2 which turns on the FW
by default.  My X-LITE clients would initiate a call... But no audio. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dev gnu
Sent: Monday, September 13, 2004 9:08 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Aasterisk SIP<->SIP No audio

Hello,

I am an * beginner and trying to get SIP-SIP configuration work. Everything
is fine but no audio.

I am using Asterisk RC2 and the SIP clients are XLite softphones.

The error message i am getting is 

Attempting native bridge of SIP/X and SIP/Y.

I am trying to follow the guidelines from the Onlamp article by John todd.

Every help will put me step ahead.

Thanks all.
Dev




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[Asterisk-Users] Aasterisk SIP<->SIP No audio

[dev gnu]

Hello,

I am an * beginner and trying to get SIP-SIP
configuration work. Everything is fine but no audio.

I am using Asterisk RC2 and the SIP clients are XLite
softphones.

The error message i am getting is 

Attempting native bridge of SIP/X and SIP/Y.

I am trying to follow the guidelines from the Onlamp
article by John todd.

Every help will put me step ahead.

Thanks all.
Dev




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Re: [Asterisk-Users] PABX & VOIP Gateway

[Benjamin on Asterisk Mailing Lists]

On Tue, 14 Sep 2004 10:34:51 +1000, Phil Stevens
<[EMAIL PROTECTED]> wrote:
> I'm researching the possibility of using VOIP (SIP) with an existing
> PABX system. Ideally, the setup would be to dial an outside line through
> the PABX (that would actually link to the the VOIP gateway).
> 
> At this point I would prefer not to purchase a hardware-based VOIP
> gateway. I would prefer to use a software-based gateway for research &
> testing purposes. Could anyone please describe a simple setup?
> 
> Naturally the connection to the gateway would have to be Austel
> approved. I have seen references to the Netjet ISDN cards? I am having
> difficulties in finding information with regards to Austel compliance.

I don't quite understand what you need Austel compliance for if all
you want to do is link to a PBX. You don't need Austel approval for
that, only if you connect directly to Telstra.

Anyway, Voicetronix is an Aussie company that makes 4, 6 and 12 port
analog boards which work with Asterisk ...

http://www.voicetronix.com.au

I also seem to remember having read about some Digium board having
obtained Austel approval quite a while ago, so you may want to ask one
of Digium's resellers in AU about that ...

http://www.austechpartnerships.com/atp/

rgds
benjk

-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

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[Asterisk-Users] PABX & VOIP Gateway

[Phil Stevens]

Hello,

I'm researching the possibility of using VOIP (SIP) with an existing
PABX system. Ideally, the setup would be to dial an outside line through
the PABX (that would actually link to the the VOIP gateway).

At this point I would prefer not to purchase a hardware-based VOIP
gateway. I would prefer to use a software-based gateway for research &
testing purposes. Could anyone please describe a simple setup?

Naturally the connection to the gateway would have to be Austel
approved. I have seen references to the Netjet ISDN cards? I am having
difficulties in finding information with regards to Austel compliance.

Has anyone had experiences with such setups?

Any help would be much appreciated.

Thank you.
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Re: [Asterisk-Users] chan_capi module

[Dave Cotton]

On Mon, 2004-09-13 at 22:22 +0200, [EMAIL PROTECTED] wrote:

> I dont have a chan_modem_chan_capi.so module, only a chan_modem.o.
> I am using chan_capi from: 
> http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.5.tar.gz
> My modem.conf:
> ---
> [interfaces]
> context=remote
> driver=chan_capi
> stripmsd=0
> dialtype=tone
> mode=immediate
> group=1
> 

And there seems to be your error chan_capi  does not need anything in
modem.conf. Just follow the directions in INSTALL


-- 
Dave Cotton <[EMAIL PROTECTED]>

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[Asterisk-Users] chan_sip2 Install Question

[Chad Brown]

It looks like chan_sip2 may solve my problem with
outboundproxy support. However, I am having problems getting the solution
installed. From what I understand these are the tasks…

 

 

Add
chan_sip2 to the channels/Makefile
* Rename the file downloaded to chan_sip2.c
* make / make install
* Change your modules.conf 
  Add "noload=chan_sip.so" if you want to run chan_sip2
* Restart asterisk

 

However, I can’t
seem to get a basic make to work. What is involved with “Add chan_sip2 to
the channels/Makefile directory?” Is it more than just adding
chan_sip2.so to the end of CHANNEL_LIBS=?

 

Chad






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[Asterisk-Users] Sipura-3000 Assistant for Asterisk on MacOSX? Well, maybe, with your help!

[Benjamin on Asterisk Mailing Lists]

Hi

We are getting more and more email from Mac users asking how they can
connect their MacOSX based Asterisk server to a PSTN phone line. This
has led to two ideas ...

1) Mid term: set up a donation fund to sponsor the development of
Zaptel drivers for MacOSX

Note: if everyone who downloaded the Asterisk installation package for
MacOSX during the last two months since its release would donate only
5 USD, there would be a sponsorship fund far in excess of 5000 USD.
Wouldn't that get us Zaptel drivers for MacOSX?!

2) Short term: create an assistant to configure Asterisk to use a
Sipura-3000 to use a PSTN phone line

Unfortunately, Sipura have not been very helpful and their attitude
has now led to a lack of determination to buy a device to be shipped
to Japan. In fact we have now cancelled an order that was never
fulfilled.

This is not about the money but about the attitude Sipura have shown.
It is not very encouraging to put in 200-300 hours of your time to
create a free tool for the community without asking anything in return
only to see the company that will get more benefit than anybody else
to give you an attitude like a 1970s IBM.

My first thoughts were "Who are those guys anyway?" and "What have
they done that they think they are better than the rest of us?", but
then I thought "Hey, we are part of a community. Let's see if
community spirit can change the attitude of those Sipura folks."

So, I am asking anybody who has got an interest in a Sipura-3000
Assistant for Asterisk on MacOSX to consider sending an email to
[EMAIL PROTECTED] and encourage them to support a) Asterisk and its
developers and b) Mac users who want to use Asterisk with GUI tools.

If Sipura don't change their attitude, I am probably going to end up
writing the assistant anyway after going to the US and picking up a
device in person, but we will then make this assistant donation ware
with mandatory contributions to the above mentioned Zaptel on MacOSX
sponsorship fund. Probably a good way to deal with products from
vendors who don't like us, I guess.

rgds
benjk

-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

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[Asterisk-Users] Extending E1's over a Satellite link

[Arinze Izukanne]

hi
 I want to compress and trunk 2E1 capacity over a satellite SCPC link using asterisk. I am nw to asterisk and I need suggestions on how to implement this.
 
Best regards
 
Arinze 
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Re: [Asterisk-Users] Caller ID "forwarded" to analog phone?

[Iassen Hristov]

I have done some sifting trough information about adaptors for my purposes
and for my money here is the short list of candidates.

1) Sipura SPA-2000 Phone Adaptor
   + provides connection for 2 phones - means $50/line
   - no second RJ45 port
   $95 at 

2) Grandstream HandyTone 286 (SIPphone Mini 286)
   - no second RJ45 port
   $60 at 

3) Grandstream HandyTone 486 (SIPphone Mini 486)
   +- includes a build-in switch (second RJ45, but it is 10 Mbps)
   $70 at 

As you can see from the documentation of the products both the HandyTone
and the Sipura support Caller-ID.


--On Tuesday, September 14, 2004 6:15 AM +0900 Benjamin on Asterisk Mailing
Lists <[EMAIL PROTECTED]> wrote:

> On Mon, 13 Sep 2004 14:57:08 -0400, Walt Reed <[EMAIL PROTECTED]>
> wrote:
>> Check the asterisk Wiki for hardware compatability. I did not see an IP
>> -> FXO (or FXS) device on the pcphoneline site.  They had a couple IP
>> phones that may or may not work with * (check the wiki) and a USB FXS
>> adaptor that may or may not work with *.
> 
> I don't know how to put this diplomatically, but PCphoneline are one
> of those few companies which would appear to be somewhat abusive of
> the Wiki. I had to repeatedly remove their entries from categories
> that the listed product clearly did not belong to. Quite a few people
> ended up buying the wrong product. I know this because some folks
> contacted me for help when they couldn't configure the stuff they
> bought.
> 
> My advice to anyone looking for fxo/fxs gateways would be this: Be
> *extremely* careful about products listed on the VoIP Gateway and ATA
> Wikis if the product listing sticks out with marketing language,
> something along the lines of ...
> 
> "ONLY 59.99 ! Does everything - Works with everything - Needs no
> drivers and no software"
> 
> If it sounds too good to be true, there is a high chance it probably is.
> 
> USB devices for example will almost certainly require some software
> that is only available for Windoze and some softphone, it won't work
> directly with Asterisk. The only USB device that can be expected to
> work with Asterisk is the S100U from Digium and that was only ever
> intended for development and testing purposes, not recommended for
> real usage.
> 
> rgds
> benjk



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RE: [Asterisk-Users] chan_capi module

[Craig Waddington]

Go into modules.conf

Comment out chan_modem.so=yes

Make it look like this:

[global]
chan_capi.so=yes
chan_modem.so=yes
;space here


Hope that helps


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 13 September 2004 21:22
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] chan_capi module

Hi!

I am trying to start Asterisk 1.0-RC1 with chan_capi.
Here the error:
---
Parsing '/etc/asterisk/modem.conf': Found
  == Loading modem driver chan_modem_chan_capi.soSep 13 22:14:08
  WARNING[1076968064]: loader.c:242 ast_load_resource:
  /usr/lib/asterisk/modules/chan_modem_chan_capi.so: cannot open shared
  object file: No such file or directorySep 13 22:14:08
ERROR[1076968064]: chan_modem.c:954 load_module: Failed to
load driver chan_modem_chan_capi.so  == Unregistered channel type
'Modem'
Sep 13 22:14:08 WARNING[1076968064]: loader.c:328 ast_load_resource:
chan_modem.so: load_module failed, returning -1  == Unregistered channel
type 'Modem'
Sep 13 22:14:08 WARNING[1076968064]: loader.c:374 load_modules: Loading
module chan_modem.so failed!---

I dont have a chan_modem_chan_capi.so module, only a chan_modem.o.
I am using chan_capi from: 
http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.5.tar.gz
My modem.conf:
---
[interfaces]
context=remote
driver=chan_capi
stripmsd=0
dialtype=tone
mode=immediate
group=1



Capiinfo:
---
02  (49.18)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
   internal controller supported
   DTMF supported
   Supplementary Services supported
   channel allocation supported (leased lines)
B1 protocols support: 0x411f
   64 kbit/s with HDLC framing
   64 kbit/s bit-transparent operation
   V.110 asynconous operation with start/stop byte framing
   V.110 synconous operation with HDLC framing
   T.30 modem for fax group 3
   Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x0b1b
   ISO 7776 (X.75 SLP)
   Transparent
   LAPD with Q.921 for D channel X.25 (SAPI 16)
   T.30 for fax group 3
   ISO 7776 (X.75 SLP) with V.42bis compression
   V.120 asyncronous mode
   V.120 bit-transparent mode
B3 protocols support: 0x80bf
   Transparent
   T.90NL, T.70NL, T.90
   ISO 8208 (X.25 DTE-DTE)
   X.25 DCE
   T.30 for fax group 3
   T.30 for fax group 3 with extensions
   Modem

  0100
  0200
  3900
  1f010040
  1b0b
  bf80
       
  0101 0002   

Supplementary services support: 0x03ff
   Hold / Retrieve
   Terminal Portability
   ECT
   3PTY
   Call Forwarding
   Call Deflection
   MCID
   CCBS


Any help/hints/tips would be great!
Thanks!

Mario




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[Asterisk-Users] compiling zaptel

[asterisk]

Hello List!
I am currently trying to compile zaptel.

A "make" gives me this error:
In file included from /usr/src/zaptel/zaptel.c:40:
/usr/src/zaptel/zconfig.h:10:27: linux/version.h: No such file or directory
.
.
.
Where in the Makefile do i have to change the path to my include files?
I have the following version.h files:

/usr/src/kernel-headers-2.6.8-1-386/include/config/isdn/diversion.h
/usr/src/kernel-headers-2.6.8-1-386/include/linux/version.h
/usr/src/kernel-headers-2.6.8-1/include/linux/dvb/version.h
/usr/src/kernel-headers-2.6.8-1/include/linux/version.h
/usr/src/kernel-headers-2.6.8-1/include/pcmcia/version.h
/usr/src/kernel-headers-2.6.8-1/include/sound/version.h
/usr/src/kernel-source-2.6.8/arch/i386/math-emu/version.h
/usr/src/kernel-source-2.6.8/drivers/net/sk98lin/h/skversion.h
/usr/src/kernel-source-2.6.8/drivers/scsi/qla2xxx/qla_version.h
/usr/src/kernel-source-2.6.8/fs/xfs/linux-2.6/xfs_version.h
/usr/src/kernel-source-2.6.8/include/linux/dvb/version.h
/usr/src/kernel-source-2.6.8/include/pcmcia/version.h
/usr/src/kernel-source-2.6.8/include/sound/version.h



Thanks, Mario


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[Asterisk-Users] voicepulse problems since new configs

[Steve Totaro]

Voicepulse has ignored 
four emails over the course of two weeks.
 
Anyone have any ideas of whats wrong?
 
- Executing Dial("IAX2/[EMAIL PROTECTED]:4569/7", 
"IAX2/acctname:[EMAIL PROTECTED]/14109649073") in new 
stack
    -- Called 
acctname:[EMAIL PROTECTED]/14109649073
Sep 13 22:48:25 WARNING[131080]: chan_iax2.c:5375 socket_read: Call rejected by 66.234.228.160: No such 
context/extension
    -- Hungup 'IAX2/vpconnect-t01/8'
  == No one is available to answer at this 
time
    -- Executing Dial("IAX2/[EMAIL PROTECTED]:4569/7", 
"IAX2/acctname:[EMAIL PROTECTED]/14109649073") in new 
stack
    -- Called 
acctname:[EMAIL PROTECTED]/14109649073
Sep 13 22:48:26 WARNING[131080]: chan_iax2.c:5375 socket_read: Call rejected by 66.234.228.166: No such 
context/extension
    -- Hungup 
'IAX2/66.234.228.166:4569/9'
  == No one is available to answer at this 
time
 
I am running today’s cvs 
head
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[Asterisk-Users] Codec usage in iax.conf

[Larry Shields]

I am running  
Asterisk CVS-HEAD-08/25/04-20:11:31.  I have two IAX accounts that I use, 
FWD and VoicePulse.  FWD requires you use ULAW codec to connect.  
VoicePulse will work with most all the codecs Asterisk supports.  

 
I recently tried to 
force VoicePulse connections to use ILBC or GSM and found it would not work so 
long as ULAW was enabled anywhere in the iax.conf file.  Below is my 
iax.conf file and although I have explicitly disallowed ULAW in the VoicePulse 
section, it is still used when I make a VP call.  
 
Only when I disallow 
all ULAW usage will I connect to VP with ILBC or GSM... but when I do this FWD 
will not work as it requires ULAW.  Is this a bug or am I not setting the 
parameters correctly.  Any help is appreciated.
 
 
; 
iax.conf
[general]port=5036bindaddr=0.0.0.0
 
; Specify bandwidth of low, medium, or high to 
control which codecs are used; in 
general.;bandwidth=low
 
; You can also fine tune codecs here using "allow" 
and "disallow" clauses; with specific codecs.  Use "all" to represent 
all formats.;disallow=all
 
allow=gsmallow=ilbcallow=ulaw
 
jitterbuffer=no
; FWD IAX2 Accountregister => aa:pp@iax2.fwdnet.net
 
; VoicePulse Accountregister => :ppp@gwiax-in-01.voicepulse.com
 
[iaxfwd]type=usercontext=from-iaxfwddeny=0.0.0.0/0.0.0.0permit=65.39.205.0/255.255.255.0disallow=gsmdisallow=ilbcallow=ulaw
 
[voicepulse-in-01]type=usercontext=from-voicepulseauth=rsainkeys=voicepulse01disallow=ulawallow=gsmallow=ilbc
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[Asterisk-Users] Sip Outbound Proxy

[Chad Brown]

How do you configure an outbound proxy for Asterisk? If the
sip call is not local I want everything to go to a designated sip proxy.

 

Thanks,

 

Chad






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Re: [Asterisk-Users] chan_capi module

[asterisk]

> Hi!
>
> I am trying to start Asterisk 1.0-RC1 with chan_capi.
> Here the error:
> ---
> Parsing '/etc/asterisk/modem.conf': Found
>  == Loading modem driver chan_modem_chan_capi.soSep 13 22:14:08
>  WARNING[1076968064]: loader.c:242 ast_load_resource:
>  /usr/lib/asterisk/modules/chan_modem_chan_capi.so: cannot open shared
>  object file: No such file or directory

I just got a little further. I changed my modules.conf, which looks like
this now:
---
[modules]
autoload=yes

noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
noload => pbx_kdeconsole.so

noload => app_intercom.so

;load => chan_modem.so
load => chan_capi.so
load => res_musiconhold.so

noload => chan_alsa.so

[global]
chan_capi.so=yes
---


And i get this error now:
---
 [WaitExten]
  == Registered application 'WaitExten'
Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf': Found
 [chan_capi.so]Sep 13 23:16:26 WARNING[1076968064]: loader.c:242
 ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined
 symbol: ast_pthread_createSep 13 23:16:26 WARNING[1076968064]: loader.c:374 
load_modules: Loading
module chan_capi.so failed!



Thanks, Mario


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Re: [Asterisk-Users] Caller ID "forwarded" to analog phone?

[Benjamin on Asterisk Mailing Lists]

On Mon, 13 Sep 2004 14:57:08 -0400, Walt Reed <[EMAIL PROTECTED]> wrote:
> Check the asterisk Wiki for hardware compatability. I did not see an IP
> -> FXO (or FXS) device on the pcphoneline site.  They had a couple IP
> phones that may or may not work with * (check the wiki) and a USB FXS
> adaptor that may or may not work with *.

I don't know how to put this diplomatically, but PCphoneline are one
of those few companies which would appear to be somewhat abusive of
the Wiki. I had to repeatedly remove their entries from categories
that the listed product clearly did not belong to. Quite a few people
ended up buying the wrong product. I know this because some folks
contacted me for help when they couldn't configure the stuff they
bought.

My advice to anyone looking for fxo/fxs gateways would be this: Be
*extremely* careful about products listed on the VoIP Gateway and ATA
Wikis if the product listing sticks out with marketing language,
something along the lines of ...

"ONLY 59.99 ! Does everything - Works with everything - Needs no
drivers and no software"

If it sounds too good to be true, there is a high chance it probably is.

USB devices for example will almost certainly require some software
that is only available for Windoze and some softphone, it won't work
directly with Asterisk. The only USB device that can be expected to
work with Asterisk is the S100U from Digium and that was only ever
intended for development and testing purposes, not recommended for
real usage.

rgds
benjk

-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
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RE: [Asterisk-Users] Arrgh, Broadvoice, SIP.conf

[Richard Cook]

Hello Duncan,

Check your config against this one:
http://www.aspworld.com/telco/bv_sip.htm

 
--
Richard Cook
[EMAIL PROTECTED]
Tel: 705-497-9320 - ext 2010
 
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, September 13, 2004 4:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Arrgh, Broadvoice, SIP.conf


Greetings,

A few days ago, I set up asterisk-1.0-RC2 on a Redhat 9 box. 

I also signed up for broadvoice and am now trying to get asterisk to talk to
broadvoice and so on.

I've tried setting up my sip.conf in two ways:


--
register => [240xxx]:[EMAIL PROTECTED]


[Broadvoice]
type=peer
username=[240xxx]
fromuser=[240xxx]
secret=[my_password]
host=sip.broadvoice.com
context=incoming
fromdomain=sip.broadvoice.com
nat=no
canreinvite=no
dtmfmode=inband
--

This results in complaints from asterisk in the logs:

Sep 13 15:14:16 NOTICE[1087249200]: Registration for
'[EMAIL PROTECTED]' timed out, trying again

And at the CLI:

-- parse_srv: SRV mapped to host proxy.lax.broadvoice.com, port 5060
-- Got SIP response 404 "Not found" back from 147.135.8.128

Sep 13 16:07:40 NOTICE[1087249200]: chan_sip.c:3922 sip_reg_timeout: 
Registration for '[EMAIL PROTECTED]' timed out, trying again
-- parse_srv: SRV mapped to host proxy.dca.broadvoice.com, port 5060
-- Got SIP response 404 "Not found" back from 147.135.0.128

and a busy message when I attempt to dial into the box.

If I move the "register" line down into the "[Broadvoice]" section of the
sip.conf file, the complaints in the logs and at the CLI stop, but I still
get the busy message when I try to dial into the box:

--

[Broadvoice]   
register => [240xxx]:[EMAIL PROTECTED]
type=peer
username=[240xxx]
fromuser=[240xxx]
secret=[my_password]
host=sip.broadvoice.com
context=incoming
fromdomain=sip.broadvoice.com
nat=no
canreinvite=no
dtmfmode=inband
--

TIA for any assistance/hints/pointers/clues...

--Duncan

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RE: [Asterisk-Users] Arrgh, Broadvoice, SIP.conf

[Marty Mastera]

> 
> I've tried setting up my sip.conf in two ways:
> 
> 
> --
> register => [240xxx]:[EMAIL PROTECTED]
> 
> 
> [Broadvoice]
> type=peer
> username=[240xxx]
> fromuser=[240xxx]
> secret=[my_password]
> host=sip.broadvoice.com
> context=incoming
> fromdomain=sip.broadvoice.com
> nat=no
> canreinvite=no
> dtmfmode=inband
> --
> 
> This results in complaints from asterisk in the logs:
> 
> Sep 13 15:14:16 NOTICE[1087249200]: Registration for 
> '[EMAIL PROTECTED]' timed out, trying again
> 
> And at the CLI:
> 
> -- parse_srv: SRV mapped to host proxy.lax.broadvoice.com, port 5060
> -- Got SIP response 404 "Not found" back from 147.135.8.128
> 
> Sep 13 16:07:40 NOTICE[1087249200]: chan_sip.c:3922 sip_reg_timeout: 
> Registration for '[EMAIL PROTECTED]' timed out, 
> trying again
> -- parse_srv: SRV mapped to host 
> proxy.dca.broadvoice.com, port 5060
> -- Got SIP response 404 "Not found" back from 147.135.0.128
> 
> and a busy message when I attempt to dial into the box.
> 
> If I move the "register" line down into the "[Broadvoice]" 
> section of the sip.conf file, the complaints in the logs and 
> at the CLI stop, but I still get the busy message when I try 
> to dial into the box:
> 


The register line needs to be under the general section of
sip.conf...you may not get the same errors when you move it into the
broadvoice section, but that's b/c it's not processed and no
registration is ever attempted.  Check your formatting for the username
and passwordthere shouldn't be any brackets surrounding either (ie
the register line should look like: register =>
240XXX:[EMAIL PROTECTED] and the same for those entries
in the broadvoice section). If you're sure you have the username and
password correct in your register statement, and you have SRV lookups
enabled in the general section and you're still getting registration
timeouts with "404 Not Found" I would think you need to call Broadvoice
and let them know about the "404" errors to make sure your account is
provisioned correctly.  Until you get the registration to work
succesfully, you should expect problems when dialing your BV number.

Marty
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Re[2]: [Asterisk-Users] sipphone dial out problems..

[Danny Zak]

Hello Marconi,

still the same

got the following when setting up call

-- Executing Dial("SIP/home-422a", "SIP/[EMAIL PROTECTED]") in new
stack Sep 13 22:52:48 WARNING[376851]: chan_sip.c:590 __sip_xmit:
sip_xmit of 0x815e6e4 (len 677) to 198.65.166.131 returned -1: Invalid
argument -- Called [EMAIL PROTECTED] Sep 13 22:52:49

WARNING[114695]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x815e6e4 (len
677) to 198.65.166.131 returned -1: Invalid argument Sep 13 22:52:50

WARNING[114695]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x815e6e4 (len
677) to 198.65.166.131 returned -1: Invalid argument

strange .. since all other calls ( within the * )

-- 
Best regards,
 Dannymailto:[EMAIL PROTECTED]

belGOnet.com  a  Euro-pictures  division - internet solutions
place princesse elisabeth 9/11   -   1030 Brussels  - Belgium
Tel : +32-(0)2-215.67.65  -  Fax : +32-(0)2-215.66.65

domains - hosting - hardware - VoiP - consultancy - backuping
CISCO - HP/COMPAQ - SUN - EMC - JUNIPER - IBM - DELL - NORTEL


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neither  is  belGOnet.com committed to them. The content of this email
is  exclusively  intended  for  adressee(s)  and information purposes.
belGOnet.com  accepts  no  liability for any damage resulting from the
use and/or acceptation of the content of this email.


Sunday, September 12, 2004, 8:12:04 PM, you wrote:

MR> On Sun, 12 Sep 2004 14:27:09 +0200, Danny Zak <[EMAIL PROTECTED]> wrote:
>> hello;
>> 
>> i got this each time when trying to dialout
>> 
>> Sep 12 12:49:27 WARNING[114695]: chan_sip.c:590 __sip_xmit: sip_xmit
>> of 0x811f88c (len 407) to 198.65.166.131 returned -1: Invalid argument
>> 
>> sip.conf
>> 
>> register => 1747668417x:[EMAIL PROTECTED]/1747668417x
>> 
>> [sipphone]
>> type=peer
>> secret=
>> username=1747668417x

MR> Insert this:
MR> fromuser = 1747668417x

>> host=proxy01.sipphone.com
>> nat=yes
>> qualify=no
>> host=dynamic
>> reinvite=no
>> canreinvite=no
>> dtmfmode=inband
>> 
>> extensions.conf
>> 
>> exten => _1747.,1,SetCallerID,1747668417x
>> exten => _1747.,2,Dial,SIP/${EXTEN:[EMAIL PROTECTED]

MR> comment that, and put:

MR>  exten => _1747.,1,Dial(SIP/[EMAIL PROTECTED])

>> 
>> anyone a resolving issue on this ?
>> 
>> --
>> Best regards,
>> Danny  mailto:[EMAIL PROTECTED]

MR> Hope that solves your problem.

MR> Regards,
MR> Marconi.

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[Asterisk-Users] IAXy loud static problem

[Brad Ediger]

I just bought an IAXy and have been using it with * and NuFone. Around 
once every 24-48 hours of operation (not off hook, just powered up), 
when I pick up the phone I hear loud static instead of dialtone. If 
there's an incoming call, the phone will ring and * CLI will show that 
it's trying iaxy, but when picking up--no answer to * and loud static. 
The incoming call goes to VM. Cycling power on the IAXy fixes the 
problem. Has anyone else had this issue? I'm using a fairly new device 
(2 wks. old) with Digium's power supply, and Asterisk 1.0RC2 on FC2.

Thanks,
Brad
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[Asterisk-Users] chan_capi module

[asterisk]

Hi!

I am trying to start Asterisk 1.0-RC1 with chan_capi.
Here the error:
---
Parsing '/etc/asterisk/modem.conf': Found
  == Loading modem driver chan_modem_chan_capi.soSep 13 22:14:08
  WARNING[1076968064]: loader.c:242 ast_load_resource:
  /usr/lib/asterisk/modules/chan_modem_chan_capi.so: cannot open shared
  object file: No such file or directorySep 13 22:14:08 ERROR[1076968064]: 
chan_modem.c:954 load_module: Failed to
load driver chan_modem_chan_capi.so  == Unregistered channel type 'Modem'
Sep 13 22:14:08 WARNING[1076968064]: loader.c:328 ast_load_resource:
chan_modem.so: load_module failed, returning -1  == Unregistered channel type 'Modem'
Sep 13 22:14:08 WARNING[1076968064]: loader.c:374 load_modules: Loading
module chan_modem.so failed!---

I dont have a chan_modem_chan_capi.so module, only a chan_modem.o.
I am using chan_capi from: 
http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.5.tar.gz
My modem.conf:
---
[interfaces]
context=remote
driver=chan_capi
stripmsd=0
dialtype=tone
mode=immediate
group=1



Capiinfo:
---
02  (49.18)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
   internal controller supported
   DTMF supported
   Supplementary Services supported
   channel allocation supported (leased lines)
B1 protocols support: 0x411f
   64 kbit/s with HDLC framing
   64 kbit/s bit-transparent operation
   V.110 asynconous operation with start/stop byte framing
   V.110 synconous operation with HDLC framing
   T.30 modem for fax group 3
   Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x0b1b
   ISO 7776 (X.75 SLP)
   Transparent
   LAPD with Q.921 for D channel X.25 (SAPI 16)
   T.30 for fax group 3
   ISO 7776 (X.75 SLP) with V.42bis compression
   V.120 asyncronous mode
   V.120 bit-transparent mode
B3 protocols support: 0x80bf
   Transparent
   T.90NL, T.70NL, T.90
   ISO 8208 (X.25 DTE-DTE)
   X.25 DCE
   T.30 for fax group 3
   T.30 for fax group 3 with extensions
   Modem

  0100
  0200
  3900
  1f010040
  1b0b
  bf80
       
  0101 0002   

Supplementary services support: 0x03ff
   Hold / Retrieve
   Terminal Portability
   ECT
   3PTY
   Call Forwarding
   Call Deflection
   MCID
   CCBS


Any help/hints/tips would be great!
Thanks!

Mario




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[Asterisk-Users] Dial-plan transfer

[Matt Hohman]

Uniden's UIP300 transfer key is pretty much just a function key that can be assigned a DTMF value. How could I have asterisk monitor the channel for lets say *#*# then wait for a extension timeout then Start a consultive transfer? Through extensions.conf? Is this even possible? 

Any help would be great!
Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office: 360.694.4985  Fax: 360.694.0219
Email: [EMAIL PROTECTED]
On Sep 11, 2004, at 5:47 PM, Matt Hohman wrote:

We have purchased a couple of Uniden UIP300 phones (h.323 only). We love them all features work except hold and transfer. The hold function is local unless you define a DTMF hold sequence... Does Asterisk support this? I had heard that # Was transfer when I hit the pound key I get... " I am sorry that's not a valid extension" before i get a chance to enter anything.

Any idea's?

Thanks,
Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office: 360.694.4985  Fax: 360.694.0219
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[Asterisk-Users] Arrgh, Broadvoice, SIP.conf

[buffalo]

Greetings,

A few days ago, I set up asterisk-1.0-RC2 on a Redhat 9 box. 

I also signed up for broadvoice and am now trying to get asterisk to talk 
to broadvoice and so on.

I've tried setting up my sip.conf in two ways:


--
register => [240xxx]:[EMAIL PROTECTED]


[Broadvoice]
type=peer
username=[240xxx]
fromuser=[240xxx]
secret=[my_password]
host=sip.broadvoice.com
context=incoming
fromdomain=sip.broadvoice.com
nat=no
canreinvite=no
dtmfmode=inband
--

This results in complaints from asterisk in the logs:

Sep 13 15:14:16 NOTICE[1087249200]: Registration for 
'[EMAIL PROTECTED]' timed out, trying again

And at the CLI:

-- parse_srv: SRV mapped to host proxy.lax.broadvoice.com, port 5060
-- Got SIP response 404 "Not found" back from 147.135.8.128

Sep 13 16:07:40 NOTICE[1087249200]: chan_sip.c:3922 sip_reg_timeout: 
Registration for '[EMAIL PROTECTED]' timed out, trying again
-- parse_srv: SRV mapped to host proxy.dca.broadvoice.com, port 5060
-- Got SIP response 404 "Not found" back from 147.135.0.128

and a busy message when I attempt to dial into the box.

If I move the "register" line down into the "[Broadvoice]" section of the
sip.conf file, the complaints in the logs and at the CLI stop, but I still
get the busy message when I try to dial into the box:

--

[Broadvoice]   
register => [240xxx]:[EMAIL PROTECTED]
type=peer
username=[240xxx]
fromuser=[240xxx]
secret=[my_password]
host=sip.broadvoice.com
context=incoming
fromdomain=sip.broadvoice.com
nat=no
canreinvite=no
dtmfmode=inband
--

TIA for any assistance/hints/pointers/clues...

--Duncan

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Re: [Asterisk-Users] festival

[Seth Remington]

On Mon, 2004-09-13 at 15:31, Rich Allen wrote:
> iH
> 
> have trouble getting the festival command to work. when i dial 
> extension i have set up i get
> 
> Sep 13 11:25:33 WARNING[344080]: app_festival.c:440 festival_exec: Festival 
> returned ER
> 
> festival does work correctly when i use it from the unix command line. my 
> festival.conf file is set up per the wiki

Did you apply the patch in /usr/src/asterisk/contrib? The
/usr/src/asterisk/contrib/README.festival document might have some info
to help you.

-Seth

-- 
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559

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Re: [Asterisk-Users] allowing/disallowing codecs in dialplan?

[Kannaiyan Natesan]

You said what is possible and in exists.
Can you try with the settings what you have mentioned.
-Kannaiyan
- Original Message - 
From: "Andreas Greulich" <[EMAIL PROTECTED]>
To: "Asterisk-Users" <[EMAIL PROTECTED]>
Cc: "Greulich, Andreas" <[EMAIL PROTECTED]>
Sent: Monday, September 13, 2004 8:35 PM
Subject: [Asterisk-Users] allowing/disallowing codecs in dialplan?

Hi all,
Is there a possibility to set the codecs Asterisk will choose in the 
dialplan
("exten=>" statements or their contexts) instead of sip.conf?

My problem is that I connect my SIP phone with several providers (Nikotel,
Sipgate, Stanaphone) for icoming and outgoing calls. Not all of these 
providers
offer the same set of codecs. I'd like Asterisk to use the same codec for 
the
provider side as well as of the device side, to prevent codec translation.
Unfortunately, Asterisk seems to negotiate the codec for the device and for 
the
peer independently, so it often happens it sets them differently.

I could of course just allow ulaw, bit I'd like to use ilbc where possible 
(less
dropouts). So, I'd like to configure Asterisk in a way that, depending on 
what
extension is chosen, it negotiates another codec with the phone. Of course I
considered setting the allow statements in the sections of sip.conf. But it
seems these settings are only considered on incoming calls, but not on 
outgoing
calls (initiated by a Dial application with the sip.conf section as its 
argument).

So, when I have something like
[general]
disallow=all
allow=alaw
allow=ulaw
allow=ilbc
[SipgateAccount]
type=friend
...
disallow=all
allow=ilbc
then these settings are only considered when I get an incoming call, but not
when I make an outgoing call using
"exten=>...,Dial/SIP/[EMAIL PROTECTED],60,)"
Thanks in advance, Andy
--
Andreas Greulich
E-Mail: [EMAIL PROTECTED]
Skype: klaymen-neverhood
Sermo datur cunctis, animi sapientia paucis.
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Re: [Asterisk-Users] DevKit TDM400P module won't load

[Colin Haxton]

Yep.  I was on a 2.4 kernel to start with and that failed so I moved to
the 2.6 kernel hoping that would fix it.

Basically it fails in both.  :(

Colin

Michael George wrote:
> 
> On Fri, Sep 10, 2004 at 02:34:45PM +1200, Colin Haxton wrote:
> > Okay.  I read that it should be wcfxs in a list somewhere, makes sense
> > that it's the card not the module though.  :)
> >
> > If I load it under a 2.4 kernel the pci comes out as
> > -
> >   Bus  0, device   8, function  0:
> > Network controller: Tiger Jet Network Inc. Intel 537 (rev 0).
> >   IRQ 17.
> >   Master Capable.  Latency=32.  Min Gnt=1.Max Lat=128.
> >   I/O at 0xdc00 [0xdcff].
> >   Non-prefetchable 32 bit memory at 0xe400 [0xe4000fff].
> > -
> > if I load it under a 2.6 kernel it says
> > -
> >   Bus  0, device   8, function  0:
> > Network controller: Individual Computers - Jens Schoenfeld Intel 537
> > (rev 0).
> >   IRQ 17.
> >   Master Capable.  Latency=32.  Min Gnt=1.Max Lat=128.
> >   I/O at 0xdc00 [0xdcff].
> >   Non-prefetchable 32 bit memory at 0xe400 [0xe4000fff].
> > -
> > any ideas why it would say that?  or is that a red hearing?
> 
> My 2.6 system reports the same way and mine works fine.
> 
> > The X100P comes up as a "Communication controller: Tiger Jet Network
> > Inc. Intel 537 (rev 0)." on a 2.4 kernel.
> >
> > Yeah, I just wish I could put my finger on the problem.  I have been
> > banging my head against it for a few days now and it's getting rather
> > frustrating.  :(
> 
> Do you also have problems with the 2.4 kernel and driver loading?
> 
> --
> -M
> 
> There are 10 kinds of people in this world:
> Those who can count in binary and those who cannot.
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[Asterisk-Users] Registering asterisk with FWD

[Rodolfo Grave]

Hi.
I have a x100p card installed and also asterisk, but I just dont get 
asterisk to register with my sip provider (FWD)... when I start asterisk 
using the following command I get the following messages (first, a lot 
of messages show up immediatly after starting up: I'read this is normal, 
then the CLI console comes out and this messages appear):

NOTICE[229390]: chan_sip.c:3922 sip_reg_timeout: Registration for
[EMAIL PROTECTED] has timed out, trying again
WARNING[229390]: chan_sip.c: 590 __sip_xmit: sip_xmit of 0x81445dc (len
366) to 192.246.69.223 returned -1: Bad file descriptor
WARNING[229390]: chan_sip.c: 590 __sip_xmit: sip_xmit of 0x81445dc (len
366) to 192.246.69.223 returned -1: Bad file descriptor
WARNING[229390]: chan_sip.c: 590 __sip_xmit: sip_xmit of 0x81445dc (len
366) to 192.246.69.223 returned -1: Bad file descriptor
WARNING[229390]: chan_sip.c: 590 __sip_xmit: sip_xmit of 0x81445dc (len
366) to 192.246.69.223 returned -1: Bad file descriptor
WARNING[229390]: chan_sip.c: 590 __sip_xmit: sip_xmit of 0x81445dc (len
366) to 192.246.69.223 returned -1: Bad file descriptor
WARNING[229390]: chan_sip.c: 673 retrans_pkt: maximum retries exceeded
on call [EMAIL PROTECTED] for seqno 104 (Critical Request)
I'm using the sample configuration files at www.fnords.org/~eric/asterisk
Please, help...
---
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Virus Database (VPS): 0437-1, 09/09/2004
Tested on: 13/09/2004 21:37:07
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http://www.avast.com

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[Asterisk-Users] allowing/disallowing codecs in dialplan?

[Andreas Greulich]

Hi all,

Is there a possibility to set the codecs Asterisk will choose in the dialplan
("exten=>" statements or their contexts) instead of sip.conf?

My problem is that I connect my SIP phone with several providers (Nikotel,
Sipgate, Stanaphone) for icoming and outgoing calls. Not all of these providers
offer the same set of codecs. I'd like Asterisk to use the same codec for the
provider side as well as of the device side, to prevent codec translation.
Unfortunately, Asterisk seems to negotiate the codec for the device and for the
peer independently, so it often happens it sets them differently.

I could of course just allow ulaw, bit I'd like to use ilbc where possible (less
dropouts). So, I'd like to configure Asterisk in a way that, depending on what
extension is chosen, it negotiates another codec with the phone. Of course I
considered setting the allow statements in the sections of sip.conf. But it
seems these settings are only considered on incoming calls, but not on outgoing
calls (initiated by a Dial application with the sip.conf section as its argument).

So, when I have something like 

[general]
disallow=all
allow=alaw
allow=ulaw
allow=ilbc

[SipgateAccount]
type=friend
...
disallow=all
allow=ilbc

then these settings are only considered when I get an incoming call, but not
when I make an outgoing call using
"exten=>...,Dial/SIP/[EMAIL PROTECTED],60,)"

Thanks in advance, Andy

-- 

Andreas Greulich
E-Mail: [EMAIL PROTECTED]
Skype: klaymen-neverhood

Sermo datur cunctis, animi sapientia paucis.

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Re: [Asterisk-Users] AstriCon Reminder: Please register today

[Brian Wilkins]

Quick question:

Are the hotel rooms registered automatically upon receipt of payment for 
AstriCon plus hotel room costs ? Or do we have to register manually by 
calling the hotel? Thanks -

On Monday 30 August 2004 10:07 pm, Steven Sokol wrote:
> Just a brief reminder to everyone who wishes to attend AstriCon 2004: We
> need your registrations ASAP, especially if you plan on staying on-site at
> the conference hotel.  We have to present the hotel with a solid count of
> rooms on Wednesday, so please take a few minutes and sign up at:
>
> http://www.astricon.net/
>
> As of today we have 189 registrations and the response is growing.  We will
> soon have to cut off registration for the Tutorials due to limited space
> and materials.  If you wish to attend the tutorials you should sign up now.
>
> Thanks,
>
> Steve & Olle
>
>
>
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-- 
--
Heritage Communications Corporation
  Melbourne, FL USA 32935
http://www.hcc.net
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[Asterisk-Users] festival

[Rich Allen]

iH

have trouble getting the festival command to work. when i dial 
extension i have set up i get

Sep 13 11:25:33 WARNING[344080]: app_festival.c:440 festival_exec: Festival 
returned ER

festival does work correctly when i use it from the unix command line. my 
festival.conf file is set up per the wiki

thanks,
- hcir
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Re: [Asterisk-Users] Caller ID "forwarded" to analog phone?

[Walt Reed]

On Mon, Sep 13, 2004 at 02:32:29PM -0400, Andrew said:
> 
> I'm a bit new to the terminology.  Let me ask my question more simply, even though I 
> think you already answered that "it should
> work"
> 
> I want to receive calls into the Asterisk PBX via a cheap POTS->PBX method, such as 
> a WinModem or other FXS endpoint on the Asterisk
> PBX.
> 
> I want the caller ID & call-waiting information that comes in on that line to be 
> forwarded by the Asterisk PBX to a connected IP
> phone.
> 
> Most importantly:  I also want to connect a POTS phone via an IP<->FXO jack, such as 
> the one supplied by www.pcphoneline.com and I
> want to know if the caller ID/call-waiting is likely to work on that POTS phone.

I think you are consfused on terminology.

You hook a phone line to a FXO port on your asterisk server, and a
normal analog telephone to a FXS port on your server. A winmodem can't
be an FXS interface. The digium X100P is a specific winmodem card that *
supports as an FXO port. Other winmodems probably won't work (unless
they are based on the exact same chipset, and not even all those work.)

Digium products come with support that can help you get it going. Other
generic products are unlikely to come with that support.

Check the asterisk Wiki for hardware compatability. I did not see an IP
-> FXO (or FXS) device on the pcphoneline site.  They had a couple IP
phones that may or may not work with * (check the wiki) and a USB FXS
adaptor that may or may not work with *. I really don't think you want
to mess with an FXS to FXO converter.

There are several inexpensive IP -> FXO / FXS adaptors that are known to
work with * listed on the Wiki. I would suggest one of those. I've heard
good things about the Sipura products.

CallerID is forwarded to all * supported products AFAIK.

When dealing with VOIP phones, you generally get what you may for.
Careful of low-end cheap stuff unless the Wiki indicates that it works
very well with *.
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RE: [Asterisk-Users] Alchemy branch integration, one way audio

[asterisk]

The vocoder card has the ability to do both alaw and ulaw 64 codecs ala
asterisk


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stuart
Mackintosh
Sent: 13 September 2004 18:27
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Alchemy branch integration, one way audio


I am attempting to connect an asterisk system into an existing Network
Alchemy branch. This system supports h.323 and has an optional vocoder card
(Very expensive!) to enable other codecs.

I have achieved one-way audio in either direction but cannot get 2way. I
feel the problem is codec related. I have selected Transparent 64k as the
codec (I believe the only one supported without the vocoder card) but unsure
what I should set at the asterisk end.

Using recent asterisk, oh323 0.63b and oh323 1.13

Has anyone had this working?


Thanks,
Stuart.


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RE: [Asterisk-Users] VoIP Telephony with Asterisk by Paul Mahler

[William Boehlke]

We're hoping to have a digital rights managed ebook version of VoIP
Telephony with Asterisk by the end of 2004.

William



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hank smith
Sent: Saturday, September 11, 2004 11:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoIP Telephony with Asterisk by Paul Mahler

is it in ebook format at all?
I am a blind computer user and have no way  of getting it scanned in to my 
computer even if I were to purchase it.
thanks
hank
- Original Message - 
From: "Sys. Concept Inc." <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, September 11, 2004 10:08 PM
Subject: [Asterisk-Users] VoIP Telephony with Asterisk by Paul Mahler


> Does anybody have the book:  VoIP Telephony with Asterisk by Paul
> Mahler.
> Is it for beginners or advanced users?
> --
> #Joseph
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[Asterisk-Users] Dialplan transfer. (h323 transfer)

[Matt Hohman]

Uniden's UIP300 transfer key is pretty much just a function key that can be assigned a DTMF value. How could I have asterisk monitor the channel for lets say *#*# then wait for a extension timeout then Start a consultive transfer? Through extensions.conf? Is this even possible? 

Any help would be great!
Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office: 360.694.4985  Fax: 360.694.0219
Email: [EMAIL PROTECTED]
On Sep 11, 2004, at 5:47 PM, Matt Hohman wrote:

We have purchased a couple of Uniden UIP300 phones (h.323 only). We love them all features work except hold and transfer. The hold function is local unless you define a DTMF hold sequence... Does Asterisk support this? I had heard that # Was transfer when I hit the pound key I get... " I am sorry that's not a valid extension" before i get a chance to enter anything.

Any idea's?

Thanks,
Matt Hohman
New Heights Church
http://www.newheights.org
7913 NE 58th Ave. Vancouver, WA 98665
Office: 360.694.4985  Fax: 360.694.0219
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RE: [Asterisk-Users] Caller ID "forwarded" to analog phone?

[Andrew]

I'm a bit new to the terminology.  Let me ask my question more simply, even though I 
think you already answered that "it should
work"

I want to receive calls into the Asterisk PBX via a cheap POTS->PBX method, such as a 
WinModem or other FXS endpoint on the Asterisk
PBX.

I want the caller ID & call-waiting information that comes in on that line to be 
forwarded by the Asterisk PBX to a connected IP
phone.

Most importantly:  I also want to connect a POTS phone via an IP<->FXO jack, such as 
the one supplied by www.pcphoneline.com and I
want to know if the caller ID/call-waiting is likely to work on that POTS phone.

A.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Marcelo
Pacheco
Sent: Monday, September 13, 2004 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Caller ID "forwarded" to analog phone?


US/Canada caller id can be forwarded between FXO to FXS devices.
On call waiting, I believe you need this: When I'm on my FXS and need to flash
the FXO device, I do flash then *0 on my phone. Works like a charm for me, I
use an asterisk behind another PBX, so when I need to transfer a call coming
from the other PBX to another extension on the same PBX, I do flash, *0 then
dial the extension number on the other PBX I need to transfer to.

Marcelo

Em Seg 13 Set 2004 15:09, Andrew escreveu:
> Folks:
>
> I'd like to install Asterisk for use in my home.  However, I'd like to
> continue using wireless phones in a couple of locations.  The cheapest way
> to do this is to continue to use analog phone devices via an FXO/FXS box.
> However, I am not clear on whether I can expect these devices to provide
> "call waiting" features and caller ID features to the connected analog
> phone.  Ideally I would use one of the cheaper devices such as that
> available from www.pcphoneline.com
>
> Any clues?
>
> A.
>
>
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RE: [Asterisk-Users] Server load capabilities

[Steven Critchfield]

On Mon, 2004-09-13 at 13:06, C. David Kading wrote:
> I Have addressed the scaling issue, according to one of the test cases I
> read, a single server should be capable of handling 50 concurrent voice
> calls. My main concern is its ability to run 15 different instances of * for
> 15 different customers simultaneously. Each client site should be able
> manage only their own connections and clients.

You seem to have missed the big picture from the last message. You need
to spend a lot of time learning before you go about thinking of
deploying. Your questions don't just appear to be homework questions,
they are outright requests to do your work for you. 

Your suggestions that you would run 15 instances proves you are not yet
very knowledgable about how network apps work let alone how PSTN apps
work. You don't run 15 instances of apache when you want 15 websites,
you run 1 instance of apache and set up 15 virtual servers.

In your case, you run a single asterisk instance on your coloed machine
and if you want each company to edit their own configs, you will need to
set a machine in each of their offices that service the company and let
them log back to the colo machine. 
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Caller ID "forwarded" to analog phone?

[Marcelo Pacheco]

US/Canada caller id can be forwarded between FXO to FXS devices.
On call waiting, I believe you need this: When I'm on my FXS and need to flash 
the FXO device, I do flash then *0 on my phone. Works like a charm for me, I 
use an asterisk behind another PBX, so when I need to transfer a call coming 
from the other PBX to another extension on the same PBX, I do flash, *0 then 
dial the extension number on the other PBX I need to transfer to.

Marcelo

Em Seg 13 Set 2004 15:09, Andrew escreveu:
> Folks:
>
> I'd like to install Asterisk for use in my home.  However, I'd like to
> continue using wireless phones in a couple of locations.  The cheapest way
> to do this is to continue to use analog phone devices via an FXO/FXS box. 
> However, I am not clear on whether I can expect these devices to provide
> "call waiting" features and caller ID features to the connected analog
> phone.  Ideally I would use one of the cheaper devices such as that
> available from www.pcphoneline.com
>
> Any clues?
>
> A.
>
>
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RE: [Asterisk-Users] TDMoE questions

[Senad Jordanovic]

[EMAIL PROTECTED] wrote:
> Hello,
>
> thanks for the answers!!! You mentionned to use the switch command. I
> read about it in the WIKI, but I couldn't find enought information to
> understand what it is actually doing. Can someone point me to the
> right direction?
>

I would do this:
Create a single numbering plan for all users on your network
Get each server to have extensions from 1-999. Then assign to each server
its code.
Depending on how many servers you have (or will have) you could use 1-9 or
11-99 or more but
you must have server's code number of digits same accross your network.

Now , get all these servers link/register to each other using IAX. Stephen,
is right DO NOT use TDMoE for this
purpose. As to TDMoE, my understanding (I have not tried it) is that it uses
FULL bandwidth assigned at all times. Can your
IP connectivity cope with that?


Once you have this setup, your local users (users registered to same server)
can call each by dialing local extension(1-999). Network users on the other
hand need to know FULL dialing digits for a network user. For example a user
from server A need to know FULL number ($SERVER CODE + LOCAL EXTENSION) for
all other network users. This number users will give each other or you can
make it available to all users by some other method.

Ta
SJ




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RE: [Asterisk-Users] Server load capabilities

[Senad Jordanovic]

[EMAIL PROTECTED] wrote:
> I Have addressed the scaling issue, according to one of the test
> cases I read, a single server should be capable of handling 50
> concurrent voice calls. My main concern is its ability to run 15
> different instances of * for 15 different customers simultaneously.
> Each client site should be able manage only their own connections and
> clients.

Maybe you need commercial asterisk GUI complete solution as provided
by http://www.bicomsystems.com/products/C/SC/319/131/
PBXware allows you to manage unlimited number of servers from one screen in
addition to auto
creation of users, conferences, queues, auto attendants etc...

Ta
SJ

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[Asterisk-Users] Caller ID "forwarded" to analog phone?

[Andrew]

Folks:

I'd like to install Asterisk for use in my home.  However, I'd like to continue using 
wireless phones in a couple of locations.  The
cheapest way to do this is to continue to use analog phone devices via an FXO/FXS box. 
 However, I am not clear on whether I can
expect these devices to provide "call waiting" features and caller ID features to the 
connected analog phone.  Ideally I would use
one of the cheaper devices such as that available from www.pcphoneline.com

Any clues?

A.


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RE: [Asterisk-Users] Server load capabilities

[C. David Kading]

I Have addressed the scaling issue, according to one of the test cases I
read, a single server should be capable of handling 50 concurrent voice
calls. My main concern is its ability to run 15 different instances of * for
15 different customers simultaneously. Each client site should be able
manage only their own connections and clients.

CD Kading
Northwind Computers, Inc.
[EMAIL PROTECTED]
208.424.0125
 
-Original Message-
From: Eric Wieling [mailto:[EMAIL PROTECTED] 
Sent: Monday, September 13, 2004 11:22 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Server load capabilities

On Mon, 2004-09-13 at 12:08, C. David Kading wrote:
> I am trying to asses the possibility of setting up Asterisk at a
collocation
> service provider (ELI) for supplying VOIP to about 15 different
> customers/locations. Is it possible to use one server to handle this load?

It depends on how many calls you will be having at a time, if you are
using analog or VoIP phones, what codecs you use, etc.  You didn't look
at the mailing list archive discussions or the Wiki pages about Asterisk
Scaling, did you?

> Is there a call management GUI (Windows), which will work well in this
> situation and allow administration? Is there a call client GUI (Windows),
> which will work well in this situation?

No.  There are several proto-projects to do this.  Look in the mailing
list archives.  Operator Panel (also see the mailing list archives) is
pretty good as an operator console.

> Has anyone implemented such an installation and if so how do you handle
QOS
> when using DSL or Cable at the customer location.

For the most part you don't.  You can control QoS on OUTBOUND packets
from the remote site by using a router or a linux box (acting as a
router) that provides QoS, but there's no way to make the ISP do QoS
when sending you packets.  This is less of an issue than it seems at
first because most of the time you have 1Mbps or more bandwidth from ISP
-> USER, but usually only 256K from USER -> ISP.

TEST TEST TEST.  If you do not build a prototype system to work all the
issues your project will fail.

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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[Asterisk-Users] WhoIsIt -- a contributed utility

[Steve Murphy]

Hello, everyone--

I've just posted a package on the Asterisk wiki that, when installed,
will allow you to announce incoming callers over the computer speakers,
based on their CID.

It's pretty simple, uses the linux /usr/bin/play (which on redhat, plays
gsm files just fine over the speakers), and also uses festival to play
the CID name string over the speaker, if you don't already have an
"announcment" for the incoming caller, and if there is any usable CID
information.

It can generate gsm announcments for all the area codes in NA, and the
country codes for the rest of the world, using festival, and will play
these for incoming long-distance calls if there are no "announcements".

There's a README in the package that explains everything in great and
gory detail, including what to put in your extensions.conf to make it
run. You'll need to compile the small .c file with gcc; there's a
makefile included to aid in the compile and installation.

We've been using it on the home system to allow the correct person to
gravitate to the phone to answer the call.

See: 

http://www.voip-info.org/wiki-Asterisk+WhoIsIt

You can get it from the main wiki page via "tips and tricks"

murf



-- 
Steve Murphy <[EMAIL PROTECTED]>
Electronic Tools Company

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[Asterisk-Users] Alchemy branch integration, one way audio

[Stuart Mackintosh]

I am attempting to connect an asterisk system into an existing Network
Alchemy branch. This system supports h.323 and has an optional vocoder
card (Very expensive!) to enable other codecs.

I have achieved one-way audio in either direction but cannot get 2way. I
feel the problem is codec related. I have selected Transparent 64k as
the codec (I believe the only one supported without the vocoder card)
but unsure what I should set at the asterisk end.

Using recent asterisk, oh323 0.63b and oh323 1.13

Has anyone had this working?


Thanks,
Stuart.


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Re: [Asterisk-Users] Re: Astersk as AVAYA IVR

[Scott Lykens]

On Mon, 13 Sep 2004 10:07:30 -0600, Jason Kawakami 
> will require a path back to the Index.  not sure what you mean by
> 'tromboning' but it may be in reference to using the same b-channel on the

On tromboning and anti-tromboning, this is a feature that is part of
the Nortel ITG system we are currently using on my voice network.
(Which I am hoping to replace with * soon) I believe ITG has had this
feature since its start, it certainly did when we first deployed it
four years ago.

What it basically is: It is a feature to detect if a call is being
sent back to the system it originated from and to, in effect, transfer
the call to its destination on the originating system and tear down
the voice path to the remote system rather than opening a new voice
path and having the call use two voice paths on a particular link when
no one at the distant end is involved in the call anymore.

The feature is very nice for use in a multi site environment where
calls are transferred around or where latency is large. It saves on
bandwidth and simultaneous path planning and helps a lot when sending
calls back to a main office from a remote facility located on the
other side of the world.

To the original poster, you might want to see if you can configure *
and the Avaya to allow you to transfer calls back to the Avaya system
as though the * system was nothing but a group of extensions off the
Avaya. You might not be able to do this with PRI but if the Avaya has
the capability to do Off-Premise Extensions (OPX) via T1 you might be
able to use that. Something like determine the route for the call and
then transfer it to that queue.

This should mean that you only need enough capacity between the Avaya
and * systems to handle the maximum number of simultaneous IVR calls
you expect at one time. This keeps * out of the middle of the call and
means it is only used to determine a queue to enter.
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[Asterisk-Users] Re:RE:Re: Astersk as AVAYA IVR

[Jason Kawakami]

- Original Message - 

> You may be thinking of PRI '2 B Channel Transfer' facility - see
> http://www.voip-info.org/wiki-Asterisk+bounty+PRI+2B+channel+transfer
>
that is exactly what i was thinking.  if it is on the bounty list i guess we
will wait for someone smarter than me to complete it.

so it will end up being

AVAYA--->*(ZAP/G1)--->*IVR logic--->*(ZAP/G2)--->AVAYA-ACD

passing call control back and forth from system to system.  if the IVR was
to front all of the calls into the ACD.

i would think it would be better served as a spur to the ACD like.


AVAYA
|
AVAYA-ACD
|
Option to IVR>*(ZAP/G1)--->*IVR logic--->*IVR option to speak to an
agent
|
|
AVAYA-ACD<*(ZAP/G2)

wouldn't bottle neck the trunks through the * and would only require
multiple channels between if the caller needed to speak to an agent after
going throught the IVR logic.

whatever, could be done either way.

Jason Kawakami
www.optellabs.com



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Re: [Asterisk-Users] Server load capabilities

[Eric Wieling]

On Mon, 2004-09-13 at 12:08, C. David Kading wrote:
> I am trying to asses the possibility of setting up Asterisk at a collocation
> service provider (ELI) for supplying VOIP to about 15 different
> customers/locations. Is it possible to use one server to handle this load?

It depends on how many calls you will be having at a time, if you are
using analog or VoIP phones, what codecs you use, etc.  You didn't look
at the mailing list archive discussions or the Wiki pages about Asterisk
Scaling, did you?

> Is there a call management GUI (Windows), which will work well in this
> situation and allow administration? Is there a call client GUI (Windows),
> which will work well in this situation?

No.  There are several proto-projects to do this.  Look in the mailing
list archives.  Operator Panel (also see the mailing list archives) is
pretty good as an operator console.

> Has anyone implemented such an installation and if so how do you handle QOS
> when using DSL or Cable at the customer location.

For the most part you don't.  You can control QoS on OUTBOUND packets
from the remote site by using a router or a linux box (acting as a
router) that provides QoS, but there's no way to make the ISP do QoS
when sending you packets.  This is less of an issue than it seems at
first because most of the time you have 1Mbps or more bandwidth from ISP
-> USER, but usually only 256K from USER -> ISP.

TEST TEST TEST.  If you do not build a prototype system to work all the
issues your project will fail.

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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[Asterisk-Users] Server load capabilities

[C. David Kading]

I am trying to asses the possibility of setting up Asterisk at a collocation
service provider (ELI) for supplying VOIP to about 15 different
customers/locations. Is it possible to use one server to handle this load?
Is there a call management GUI (Windows), which will work well in this
situation and allow administration? Is there a call client GUI (Windows),
which will work well in this situation?

Has anyone implemented such an installation and if so how do you handle QOS
when using DSL or Cable at the customer location.

 
CD Kading
Northwind Computers, Inc.
[EMAIL PROTECTED]
208.424.0125
 


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RE: [Asterisk-Users] Re: Astersk as AVAYA IVR

[Kris Boutilier]

> -Original Message-
> From: Jason Kawakami [mailto:[EMAIL PROTECTED]
> Sent: September 13, 2004 9:08 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Re: Astersk as AVAYA IVR
> 
{clip}
> 
> Your assumptions on routing are correct.  a path out of the 
> Index to the *
> will require a path back to the Index.  not sure what you mean by
> 'tromboning' but it may be in reference to using the same 
> b-channel on the
> pri to route the call back to the Index.  not sure about 
> that, would require
> some analysis of d-channel messaging.  most systems that support PRI
> networking have some special d-channel signalling and it may just be a
> question of building something in * that the AVAYA needs to see to
> accomplish this.
> 

You may be thinking of PRI '2 B Channel Transfer' facility - see
http://www.voip-info.org/wiki-Asterisk+bounty+PRI+2B+channel+transfer

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RE: [Asterisk-Users] Playback Fileformats

[Brian West]

Its correct.. they all result in SLINR output.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Eric Wieling
> Sent: Monday, September 13, 2004 10:51 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Playback Fileformats
> 
> On Mon, 2004-09-13 at 10:21, Brian West wrote:
> > Update your asterisk install them because you must have an old one
> >
> > asterisk*CLI> show file formats
> > Format Name   Extensions
> > SLINR  slnsln|raw
> > SLINR  wavwav
> > SLINR  mp3mp3
> 
> Is this correct or a bug in the formats list?
> 
> --
>   Eric Wieling * BTEL Consulting * 504-899-1387 x2111
> "In a related story, the IRS has recently ruled that the cost of Windows
> upgrades can NOT be deducted as a gambling loss."
> 
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[Asterisk-Users] Re: Astersk as AVAYA IVR

[Jason Kawakami]

- Original Message - 
>snip>
I've got 2x PRI 30 lines coming in to the Index, and I have 4 spare PRI
cards in the Index.  I was thinking about using a QUAD PRI card from Digium
and having the calls come into the Index then transfer to Asterisk for IVR
then back to the Index.   That way if we get 60 inbound calls we'd in
essence be using all 6 PRI cards in the Index.  (2 for termination to PSTN,
2 for outbound calls from Index to Asterisk and 2 for calls back from
Asterisk to Index).
>
> Is this feasible?  Can anyone offer some tips/advice based on their
experience?  I've had a look around and I can't find anything relating to
tromboning or anti-tromboning so I suppose each call will have to take the
path of:


so you have ACD on the AVAYA?  are you thinking of front ending all calls to
the ACD with the IVR?  I would think that you would set the IVR up as an
option within the ACD instead of front ending all calls to the ACD.  The
flexibility of * is great for this and I have built several IVR's for
customers using * as the telephony engine.  does require knowledge of some
scripting language to run the AGI scripts in (PHP, PERL etc.)

Your assumptions on routing are correct.  a path out of the Index to the *
will require a path back to the Index.  not sure what you mean by
'tromboning' but it may be in reference to using the same b-channel on the
pri to route the call back to the Index.  not sure about that, would require
some analysis of d-channel messaging.  most systems that support PRI
networking have some special d-channel signalling and it may just be a
question of building something in * that the AVAYA needs to see to
accomplish this.

Good Luck

Jason Kawakami
www.optellabs.com


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Re: [Asterisk-Users] Simple Caller-ID match and block and/or play voice file saying you're calling too much or don't call!

[Joseph Finley]

Low, Adam wrote:
Ironic, Im just working on something similar myself, you can either use the 
appropriately named ex-girlfriend feature or I use GotoIf statements to match the 
caller id and maybe a timer or something to route to another context.
; note page search in girlfriend
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
; note page search on CALLERIDNUM
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf
; found this too but havent used it
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetGroup
-Original Message-
From: Joseph Finley [mailto:[EMAIL PROTECTED]
Sent: 13 September 2004 16:23
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Simple Caller-ID match and block and/or play
voice file saying you're calling too much or don't call!

The subject says it all.  A couple of my sons have very annoying friends 
that tend to call ALOT.  I usually don't like to answer the phone but 
these kids keep calling back with in 2 minutes of calling.  I'm sure 
someone else has this problem and maybe using * to do a callerID match 
and block?  Even add logic that if they called so many times in an hour? 
  Or in my case, make it a month

Joe

Ok, I forgot the term "ex-girlfriend" because I was looking on wiki and 
didn't see any matches.  Been married too long to remember the 
ex-girlfriend!  Thanks!

Joe

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[Asterisk-Users] iax2 transfer and CDRs

[Matthew Simpson]

Does IAX2 properly update call records for transferred calls to another IAX2
server?  Or should I still be using notransfer=yes ?

Example:

SERVER1 calls SERVER2 which transfers call to SERVER3

If Call records are pulled from Server2 will that call have proper CDRs?
The Wiki says no.

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Re: [Asterisk-Users] Playback Fileformats

[Eric Wieling]

On Mon, 2004-09-13 at 10:20, Christian Victor wrote:
> I am running Asterisk CVS-09/12/04-16:39:35. I am quite new to CVS. Is 
> CVS_HEAD something different?

There was a difference at one time, not any more.

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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RE: [Asterisk-Users] Playback Fileformats

[Eric Wieling]

On Mon, 2004-09-13 at 10:21, Brian West wrote:
> Update your asterisk install them because you must have an old one
> 
> asterisk*CLI> show file formats
> Format Name   Extensions
> SLINR  slnsln|raw
> SLINR  wavwav
> SLINR  mp3mp3

Is this correct or a bug in the formats list?

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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[Asterisk-Users] CVS lock directory still not fixed?

[Matthew Boehm]

I've seen others complain about this error with no result. Any idea on this?

cvs server: failed to create lock directory for
`/usr/cvsroot/asterisk-addons/format_mp3'
(/usr/cvsroot/asterisk-addons/format_mp3/#cvs.lock): Permission denied
cvs server: failed to obtain dir lock in repository
`/usr/cvsroot/asterisk-addons/format_mp3'
cvs [server aborted]: read lock failed - giving up

Don't think this is a local issue since it says "cvs server" and the path is
different from mine.

-Matthew

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Re: [Asterisk-Users] Problem with stuttering on TE410P

[Chad Scott]

Is this problem more-or-less continuous or does it happen occasionally?
If the latter, does it always happen on channel one?
At first glance, this looks like some sort of line hit, maybe a loss of 
sync?

On Sep 10, 2004, at 8:58 AM, Claus Futtrup wrote:
Hi Guys,
Im having some problems with a Wildcard TE410P card.. During a call I 
get
some strange messages and the voice drops out:
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[Asterisk-Users] Astricon tutorials :: Open for registration again

[Olle E. Johansson]

We're now opening up registrations for the Astricon tutorials again.
We've been able to move to new conference rooms within the same hotel.
Register on line at http://www.astricon.net
We're sorry for the inconvienience our recent closing of the tutorials
may have caused you. You are  welcome to contact us at [EMAIL PROTECTED]
to change your reservation to include the tutorials. There are a lot of
things going on now, so we may not be able to answer your phone call.
Even though we now have larger rooms, we want to emphasize that the
seats in the tutorials are served on a first come basis. If everyone
wants to attend the same tutorial, there simply won't be enough
seats available.
You are free to choose any tutorial, as long as we have seats available.
We are also researching the possibility to tape the tutorials to make
them available on line or on dvd later on.
We will be well over 300 people in Atlanta.
See you there!
Best regards,
/Olle and Steven
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RE: [Asterisk-Users] Playback Fileformats

[Brian West]

Update your asterisk install them because you must have an old one

asterisk*CLI> show file formats
Format Name   Extensions
SLINR  slnsln|raw
ILBC   iLBC   ilbc
G726   g726-16g726-16
G726   g726-24g726-24
G726   g726-32g726-32
G726   g726-40g726-40
H263   h263   h263
ALAW   alaw   alaw|al
G729A  g729   g729
ULAW   pcmpcm|ulaw|ul|mu
GSMwav49  WAV|wav49
SLINR  wavwav
GSMgsmgsm
SLINR  mp3mp3

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Christian Victor
> Sent: Monday, September 13, 2004 10:05 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Playback Fileformats
> 
> Brian West schrieb:
> 
> > At the cli do
> >
> > show file formats
> 
> Sorry - that does not work. And SHOW AUDIO CODECS shows me the codecs
> but not the file formats (.wav etc) supported.
> 
> Christian
> 
> >>I wonder what sondfile formats Playback() could play. I know it plays
> >>GSM but to save the CPU time I will avoid converting GSM > alaw for my
> >>E1 and user alaw compressed wavs.
> >>
> >>Unfortunately the wiki does not list the supported filetypes but I know
> >>that there are a few.
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Re: [Asterisk-Users] Playback Fileformats

[Christian Victor]

Eric Wieling schrieb:
At the cli do
show file formats
Sorry - that does not work. And SHOW AUDIO CODECS shows me the codecs 
but not the file formats (.wav etc) supported.
Upgrade your Asterisk.  "show file formats" works for Asterisk
CVS-HEAD-07/18/04-11:25:14
I am running Asterisk CVS-09/12/04-16:39:35. I am quite new to CVS. Is 
CVS_HEAD something different?

Christian
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[Asterisk-Users] Zaprtc help

[asterisk]

I am still stuck, anyone got any ideas ?


Hi,

Having no digium hardware in my box and two cpus and a ohci usb bus im
forced to use zaprtc.

I have recompiled the kernel and removed enhanced rtc support.
When I attempt to compile zaprtc I get the following error.

zaprtc.c:1077: warning: implicit declaration of function `barrier'
zaprtc.c:1078: warning: implicit declaration of function `cpu_relax'
zaprtc.c: At top level:
zaprtc.c:109: storage size of `rtc_irq_timer' isn't known
zaprtc.c:719: storage size of `rtc_fops' isn't known
zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never
defined
make: *** [zaprtc.o] Error 1

Can anyone offer advice on where to start .

Thanks

David

Counting the days to astricon.

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Re: [Asterisk-Users] Playback Fileformats

[Eric Wieling]

On Mon, 2004-09-13 at 10:05, Christian Victor wrote:
> Brian West schrieb:
> 
> > At the cli do
> > 
> > show file formats
> 
> Sorry - that does not work. And SHOW AUDIO CODECS shows me the codecs 
> but not the file formats (.wav etc) supported.

Upgrade your Asterisk.  "show file formats" works for Asterisk
CVS-HEAD-07/18/04-11:25:14

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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[Asterisk-Users] Astersk as AVAYA IVR

[Matt]

I'm thinking about using asterisk as an IVR system with an existing avaya index 
system.  

I've got 2x PRI 30 lines coming in to the Index, and I have 4 spare PRI cards in the 
Index.  I was thinking about using a QUAD PRI card from Digium and having the calls 
come into the Index then transfer to Asterisk for IVR then back to the Index.   That 
way if we get 60 inbound calls we'd in essence be using all 6 PRI cards in the Index.  
(2 for termination to PSTN, 2 for outbound calls from Index to Asterisk and 2 for 
calls back from Asterisk to Index).

Is this feasible?  Can anyone offer some tips/advice based on their experience?  I've 
had a look around and I can't find anything relating to tromboning or anti-tromboning 
so I suppose each call will have to take the path of:

PSTN
|
Index
|
Asterisk
|
Index
|
Agent


Regards

Matt
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Re: [Asterisk-Users] Playback Fileformats

[Christian Victor]

Brian West schrieb:
At the cli do
show file formats
Sorry - that does not work. And SHOW AUDIO CODECS shows me the codecs 
but not the file formats (.wav etc) supported.

Christian
I wonder what sondfile formats Playback() could play. I know it plays
GSM but to save the CPU time I will avoid converting GSM > alaw for my
E1 and user alaw compressed wavs.
Unfortunately the wiki does not list the supported filetypes but I know
that there are a few.
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RE: [Asterisk-Users] Unknown RTP codec 72 received

[Elman Efendiyev]

Thanks for the hint Eric, but yes, before sending a message to the list
I checked google and wiki and NO - I didn't find an answer/solution/any
info on this subject.
There was couple of the same questions on the list but none of them
answered

--
Sincerely,
Elman Efendiyev
[EMAIL PROTECTED] 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling
Sent: Monday, September 13, 2004 5:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Unknown RTP codec 72 received


On Mon, 2004-09-13 at 06:13, Elman Efendiyev wrote:

> I get "Unknown RTP codec 72 received" message in console when call in 
> progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN

None of the 19 hits I saw on Google about this were helpful?

To search the Asterisk mailing list archive go to www.google.com
and put site:lists.digium.com in addition to your other query
terms.

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a
related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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RE: [Asterisk-Users] Unknown RTP codec 72 received

[Low, Adam]

According to IANA's list of RTP payload types 
(http://www.iana.org/assignments/rtp-parameters) RTP payload type 72 fulls within the 
following range:

  72--76 reserved for RTCP conflict avoidance [RFC3550]

I can't find much else in RFC3550 that defines it further but this should start you on 
the right path I hope.

-Original Message-
From: Eric Wieling [mailto:[EMAIL PROTECTED]
Sent: 13 September 2004 16:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Unknown RTP codec 72 received


On Mon, 2004-09-13 at 06:13, Elman Efendiyev wrote:

> I get "Unknown RTP codec 72 received" message in console when call in
> progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN



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[Asterisk-Users] CDR database.

[Jefferson Carvalho]

Hello list,
I'm developing a front-end for asterisk cdr-mysql and i need
a database for tests.
Could someone send me a mysql data for a test ?
I'm looking for a table with at least 5000 call records.
Best regards ,
-Jefferson Carvalho



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[Asterisk-Users] test membership

[Pasha]

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RE: [Asterisk-Users] Playback Fileformats

[Brian West]

At the cli do

show file formats

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Christian Victor
> Sent: Monday, September 13, 2004 9:42 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Playback Fileformats
> 
> Hi!
> 
> I wonder what sondfile formats Playback() could play. I know it plays
> GSM but to save the CPU time I will avoid converting GSM > alaw for my
> E1 and user alaw compressed wavs.
> 
> Unfortunately the wiki does not list the supported filetypes but I know
> that there are a few.
> 
> Thanks in davance
> Christian
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[Asterisk-Users] Post to list

[George Zecheru]

 
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[Asterisk-Users] Playback Fileformats

[Christian Victor]

Hi!
I wonder what sondfile formats Playback() could play. I know it plays 
GSM but to save the CPU time I will avoid converting GSM > alaw for my 
E1 and user alaw compressed wavs.

Unfortunately the wiki does not list the supported filetypes but I know 
that there are a few.

Thanks in davance
Christian
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[Asterisk-Users] Asterisk daemon start errors

[Trevor Morrison]

Hi,

I have a RH 9 box just freshly installed on a Dell optiplex GX1.  I also
have an quicknet phone jack isa card installed and is seen by the OS.  I
checked out the latest cvs from the Asterisk site and compiled and installed
fine.  I also downloaded the mpg123 tar.gz and compiled and installed fine.
My problem is when I start Asterisk with the asterisk -c command a lot
of information scrolls by and it stops with following error message:

res_musiconhold.c:306 monmp3thread: Request to schedule in the past?

I have searched the archives and did not find anything in this exact error.
Any help is appreciated.

TIA,

Trevor

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RE: [Asterisk-Users] Simple Caller-ID match and block and/or play voice file saying you're calling too much or don't call!

[Low, Adam]

Ironic, Im just working on something similar myself, you can either use the 
appropriately named ex-girlfriend feature or I use GotoIf statements to match the 
caller id and maybe a timer or something to route to another context.

; note page search in girlfriend
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf

; note page search on CALLERIDNUM
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf

; found this too but havent used it
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetGroup

-Original Message-
From: Joseph Finley [mailto:[EMAIL PROTECTED]
Sent: 13 September 2004 16:23
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Simple Caller-ID match and block and/or play
voice file saying you're calling too much or don't call!




The subject says it all.  A couple of my sons have very annoying friends 
that tend to call ALOT.  I usually don't like to answer the phone but 
these kids keep calling back with in 2 minutes of calling.  I'm sure 
someone else has this problem and maybe using * to do a callerID match 
and block?  Even add logic that if they called so many times in an hour? 
  Or in my case, make it a month

Joe

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[Asterisk-Users] IAXy DHCP lease not renewing

[Glenn A. Thompson]

Hi,
I have an IAXy which *appears* not to renew it's DHCP lease.
The DHCP server is a Solaris box running the native Solaris DHCP server.
Is there any known DHCP issues I should be aware of?
Is there anyway to get the IAXy to log it's DHCP activity?
Thanks,
Glenn
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[Asterisk-Users] Simple Caller-ID match and block and/or play voice file saying you're calling too much or don't call!

[Joseph Finley]

The subject says it all.  A couple of my sons have very annoying friends 
that tend to call ALOT.  I usually don't like to answer the phone but 
these kids keep calling back with in 2 minutes of calling.  I'm sure 
someone else has this problem and maybe using * to do a callerID match 
and block?  Even add logic that if they called so many times in an hour? 
 Or in my case, make it a month

Joe
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Re: [Asterisk-Users] Unknown RTP codec 72 received

[Eric Wieling]

On Mon, 2004-09-13 at 06:13, Elman Efendiyev wrote:

> I get "Unknown RTP codec 72 received" message in console when call in
> progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN

None of the 19 hits I saw on Google about this were helpful?

To search the Asterisk mailing list archive go to www.google.com
and put site:lists.digium.com in addition to your other query
terms.

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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Re: [Asterisk-Users] unavail and busy.

[Eric Wieling]

On Mon, 2004-09-13 at 09:02, Jozeph Brasil wrote:
> Hi guys,
> 
>  
> 
> What is different and the “context” to play unavail message and busy
> message?
> 
> When a SIP connection is unregistered, voicemail will play unavail
> message, right?

No.  If there's a n+101 Asterisk will jump to that if the destination is
busy, unavailable, lagged, or not registered.  If you want to know the
ACTUAL cause of the the call failing you need to look at ${DIALSTATUS}
and figgle with dialplan logic.

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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Re: [Asterisk-Users] (no subject)

[Eric Wieling]

On Mon, 2004-09-13 at 05:39, Murali wrote:
> hi all,
> 
> can anyone give solution for this.
> 
>   wct1xxp -  Digium Wildcard T100P T1/PRI Card 0
> 
> 
> 
> 
> 
> zttool gives
> 
> RED Digium Wildcard T100P T1/PRI Card 0   
> 
> 
> 
> 
> my zaptel.conf look like this
> 
> span=1,1,0,esf,b8zs
> bchan=1-23
> dchan=24
> loadzone=us
> defaultzone=us
> 
> the above 5 lines only placed in my zaptel.conf file

RED usually means "I don't see any voltage on the port and therefore
there is no line installed."

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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Re: [Asterisk-Users] Galaxy Voice Configuration Question

[Eric Wieling]

On Sun, 2004-09-12 at 21:25, Kevin wrote:
> I am using Galaxy Voice until recently I can receive any inbound calls.
> If I remove the [galaxy voice] context in my sip file the call rings in
> but I obviously can't make any outgoing calls.  Any suggestions?

Don't remove the [galaxyvoice] entry from your sip.conf

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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[Asterisk-Users] unavail and busy.

[Jozeph Brasil]

Hi guys,

 

What is different and the “context” to
play unavail message and busy message?

When a SIP connection is unregistered, voicemail will
play unavail message, right?






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[Asterisk-Users] Agentlogin incorrect

[Stig Thune]

Followed; http://www.voip-info.org/wiki-Asterisk+Agents
 
agents.conf 
 [agents] 
 agent => 1001,4321,Ben Dover 
queues.conf 
 [queue1] 
 member => Agent/1001 
extensions.conf 
 exten => 28,1,AgentLogin(1001) 
 exten => 29,1,Queue(queue1) 
 
 
But when I call number 28, I get:
"Please enter your password followed by the pound 
key"..
 
but when I enter the the password, 4321, I get: 
login incorrect.
 
CLI>

    -- Executing 
AgentLogin("SIP/stigcompaq-f8be", "1001") in new stack    -- 
Playing 'agent-pass' (language 'en')    -- Playing 
'agent-incorrect' (language 'en')
 
/ Stig Henning
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RE: [Asterisk-Users] music on hold not strting

[Wiley E. Siler]

I never stripped my tags and things work fine for me.  I had problems at
first too with MOH.  My problem was due to how I was copying over the
files.  I was copy via FTP using the command line in Linux.  However, if
you do not explicitly state binary as the copy method, it will copy the
files over using ASCII.  Doing so mungs the whole MOH player and never
worked right.  Issueing the binary command at the ftp command prompt
prior to pulling the files to my asterisk box solved the problem for me.
I do not know your method of copying to your server but this may be your
problem if you are using FTP.

Cheers,
Wiley
 

-Original Message-
From: Andreas Roedl [mailto:[EMAIL PROTECTED] 
Sent: Monday, September 13, 2004 6:34 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] music on hold not strting

Hello!

Am Montag, 13. September 2004 14:40 schrieb Altus Snyman:
> In the howto it tells me I should strip the ID3 tags How do I do that?

  http://fibiger.org/mp3tag.html


Andi
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