[asterisk-users] Re: Can you explain why multiple registration is an important (missing) feature ?

[Tomislav Parčina]

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> And there is your problem.  Using the extension as the SIP User ID does 
> not scale, is confusing, and limits your thinking about devices and 
> extensions.  There are several reasons this is a bad idea.  Multiple 
> extension numbers ringing on the same device / line appearance is the 
> most common.
> 
> We use the MAC address of the device as the SIP User ID.  We append a 
> -a, -b, -c, etc to the MAC address for each line appearance.  This does 
> not work well for Softphone, but since All Softphones Suck(TM), we don't 
> really care about this limitation.
> 
> Users seldom need to know their SIP User ID.

Can you please tell me more about this. I don't follow you weary well. I 
understand that we need to treat phone and users different, but I don't thing 
that is easy to do with Asterisk 1.2. Maybe something will change, but till 
then...



--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] RE: Asterisk 1.4 Docs

[Tomislav Parčina]

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> One of the providers that I use already offers this feature via a macro
> in the dail plan
> http://connect.voicepulse.com/FlexRate.aspx

Hi Jason!

This is interested, although it's not related to AOC messages.


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] Noob question: Packet size

[Avi Miller]

Hi guys,

I have what is probably a very noob question. I've tried to search the 
wiki, but my lack of knowledge is hindering me in finding the right 
keywords:


I'd like to know what the packet size of an IAX2 packet is, if its using 
the ilbc codec.


Now I'll tell you why, so you can tell me what I really want to know. :)

I'm experiencing packet loss on my inter office network, so I installed 
SmokePing to determine the extent of the loss. However, I'm not sure 
what the best size packet to test would be.


Any advice/suggestions would be great.

Thanks,
Avi

--
National Manager - Special Projects

< Sydney / Melbourne / Canberra / Hobart / London />
  2/340 Gore Street  T: 1 3000 SQUIZ (77849)
  Fitzroy, VIC   T: +61 (0) 3 9235 5400
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RE: [asterisk-users] Polycom Expansion Module

[Douglas Garstang]

As far as I know, it's 12.

-Original Message- 
From: Noah Miller [mailto:[EMAIL PROTECTED] 
Sent: Sun 9/17/2006 10:27 AM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Cc: 
Subject: Re: [asterisk-users] Polycom Expansion Module



Hi Kevin -

> Has anyone used the Polycom expansion module with multiple lines?
>
> My application is for 20 lines and read there was a limit of 7 at one 
point.

I heard rumors that the newest version of the polycom sip firmware
(2.01) would lift the limit of 7.  It just came out, and I haven't had
time to test it yet, but you can give it a try.

- Noah
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[asterisk-users] problem installing func_odbc on asterisk 1.2 ...

[Vince Daneff]

Hi,
I have a problem installing func_odbc on asterisk 1.2.12.1 the message
that I received is:

./astxs func_odbc.c
make[1]: *** No rule to make target `apps_env'.  Stop.
 -I/usr/src/asterisk -I/usr/src/asterisk/include  -c func_odbc.c -o func_odbc.o
make: *** [func_odbc.so] Error 255

then I've changed the file astxs the symbol (`) with (") and then the
output was and still is:

./astxs func_odbc.c
 -I/usr/src/asterisk -I/usr/src/asterisk/include  -c func_odbc.c -o func_odbc.o
make: *** [func_odbc.so] Error 255

i.e. the same

and I have these files which I downloaded from here
http://svncommunity.digium.com/view/func_odbc/1.2/
-
astxs
func_array.c
func_odbc.c
func_odbc.conf.sample
Makefile
README
separate.h
---
  please advice

--=-\/-=--
 Vince

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RE: [asterisk-users] Does a "HST Saphir III ML PCI" work with Asterisk?

[James Harper]

I tried one of these and pretty much got it working under visdn. If you
do decide to try one, make sure you get the HFC version. Earlier ones
used another chipset and definitely weren't supported using open sourced
drivers.

Please post back if you do get one and get it going though.

Thanks

James

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Patrick Cervicek
> Sent: Sunday, 17 September 2006 21:46
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Does a "HST Saphir III ML PCI" work with
> Asterisk?
> 
> I am looking for Infos & Tutorials for installing "ISDN Karte PCI HST
> Saphir III ML". Does sombody have infos & links for me?
> 
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Re: [asterisk-users] Cisco Distinctive ring using alert-info

[Rich Adamson]

There has been several different hardware versions of the phone, but to 
the best of my knowledge, the ringer has not changed.  The cisco 
documentation suggests there is a way to create your own ring tones, but 
I've not tried that either.


The stock 7960 "sip" phone's built in ring tones are not very 
impressive, and as I recall, are basically limited to sounds such as 
one-long, one-long & one short, etc.



Lacy Moore - Aspendora wrote:

Do some 7960s perform differently?

On 9/15/06, *Eric ManxPower Wieling* <[EMAIL PROTECTED] 
> wrote:


Rich Adamson wrote:
>  Julian Lyndon-Smith wrote:
> > I've got a cisco 7960, with (amongst many others) the following
in the
> > RINGLIST.DAT file
> >
> > Foghorn foghorn.raw
> >
> > I can manually select this for the ringtone. However, I was
wanting to
> > use a normal ringtone, with foghorn being used if the call was
coming
> > in from the girlfriend/wife/mother-in-law etc ;)
> >
> > I was trying to use the following:
> >
> > exten => 5711,1,SIPAddHeader("Alert-Info: ")
> > exten => 5711,n,Dial(SIP/5711)
> > exten => 5711,n,Hangup()
> >
> > However, not matter what I try, I get the standard ringtone. If I use
> >
> > exten => 5711,1,SIPAddHeader("Alert-Info: ")
>
>  Our past experience indicates the  approach is the only
>  one that works.

That TOTALLY depends on the phone.
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--
Lacy Moore
Aspendora, Inc.




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Re: [asterisk-users] blf aastra 9133i working but can't pickup calls

[Jean-Louis curty]

I'm using bristuff which patches lot of things, I did not try to use this patch ... may be I shouldjl2006/9/15, Gareth Owen <[EMAIL PROTECTED]
>:













I got a chance to patch my Asterisk server
this afternoon and was able to confirm that the directed call pickup feature is
working (at least for me).  I'm running Asterisk 1.2.12.1 and used
the latest pickup patch (http://bugs.digium.com/view.php?id=5014,
pickup-mgernoth-2006-07-28.patch.txt)

 

One thing I did notice (and am about to
fix) is that the configuration parameter name for this feature is wrong on the wiki. 
It should be "directed call pickup", not "call pickup". 
Could this have been the problem?

 

 

Gareth

 

 

 



-Original Message-
From: [EMAIL PROTECTED]

[mailto:[EMAIL PROTECTED]] On Behalf Of 
Jean-Louis curty
Sent: 08 September,
 2006 5:25 AM

To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] blf
aastra 9133i working but can't pickup calls

 

Hi Dave , long time no
ear  :-)

still progressing on * , I have now my own distribution on a compact flash
including bristuff ...

i m fighting with this nice 9133i, lastest firmware 1.4 , blf is turned on and
working ( light flashing when calls comes in ) but I can not pick it up, 

ps I followed the procedure to enable it from the voip-info website 

anybody suceeded ???
jl



2006/9/7, Dave Cotton <[EMAIL PROTECTED] 
>:

On Thu, 2006-09-07 at 11:14 -0400, Gareth Owen wrote:
> The directed call pickup functionality is turned off by default –
you 
> have to explicitly enable it.  Instructions can be found at
> http://www.voip-info.org/wiki/index.php?page=Asterisk+and+Aastra

> +Phones#DirectedCallPickup
>

I'd forgotten about that patch, nice to see the manufacturer's people on
the list. Gives confidence in their product.

--
Dave Cotton <
[EMAIL PROTECTED]>

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Re: [asterisk-users] amr codec

[Net Nut]

Well this would not be for comercial use.. I just want it for my own
cell phone to talk on my own asterisk system.
is that ok?

Tim Panton wrote:
>
> On 16 Sep 2006, at 20:38, Net Nut wrote:
>
>> So with that said, can anyone recommend a way that I can get a sip
>> client on a cell phone that uses H.263 and amr to talk to an asterisk
>> system?
>> Is it just not possible because of licensing? It sounds kind of lame to
>> have a sip client that can't talk to anything else because of codecs..
>
> Well Asterisk does not _have_ to have an amr codec for you
> to be able to use your handset. If you have several of these
> handsets or other devices that support amr, then asterisk can
> route calls between them, just passing the stream through.
>
> If you want any of the interesting asterisk features, then
> it will need to transcode, and then Steve's right, not only do you have
> to add codec code to Asterisk (which is almost certainly a GPL violation)
> you also have to pay the patent holder for any commercial use of the
> codec.
>
> Your best hope is if a few of us can persuade Digium to support amr
> in the same way that they license g729 .
>
> Tim.
> Tim Panton
>
> www.mexuar.com
>
>
>
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Re: [asterisk-users] Cisco Distinctive ring using alert-info

[Lacy Moore - Aspendora]

Do some 7960s perform differently?
On 9/15/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
Rich Adamson wrote:> Julian Lyndon-Smith wrote:>> I've got a cisco 7960, with (amongst many others) the following in the
>> RINGLIST.DAT file Foghorn foghorn.raw I can manually select this for the ringtone. However, I was wanting to>> use a normal ringtone, with foghorn being used if the call was coming
>> in from the girlfriend/wife/mother-in-law etc ;) I was trying to use the following: exten => 5711,1,SIPAddHeader("Alert-Info: ")
>> exten => 5711,n,Dial(SIP/5711)>> exten => 5711,n,Hangup() However, not matter what I try, I get the standard ringtone. If I use exten => 5711,1,SIPAddHeader("Alert-Info: ")
>> Our past experience indicates the  approach is the only> one that works.That TOTALLY depends on the phone.___--Bandwidth and Colocation provided by 
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-- Lacy MooreAspendora, Inc. 
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Re: [asterisk-users] A1200+fxo, anyone using this?

[Darrick Hartman]

Nick Ellson wrote:


I know it's not a digium product, but the 12 port A1200P card with a 
single FXO module at pbxeq.com at first glance would seem to be the 
way to get started for me with an in-system controller card. 4 ports 
seems too small for expansion, the huge 24 port card a tad too big 
(and spendy).


So has anyone used this card with Asterisk? I googled for reviews and 
have not found anything, and I am tryingto find a way to search the 
archives without looking at each month one at a time.


Nick


I have one.  I used it some when I had an analog line at my main 
office.  It worked the same as the TDM400 card.  I even received faxes 
on it via iaxmodem and hylafax.


I got the card more or less to evaluate it.  Since then, Sangoma's card 
was released and is the way I'm going on commercial installations.  If 
you'd be interested in the card I have, contact me off list and we could 
probably work something out.


Darrick

--
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DJH Solutions, LLC
http://www.djhsolutions.com
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Re: [asterisk-users] Issues with AGI+Dial command

[Steve Totaro]

Brian Rogan wrote:

Hi,

I am still working on trying to figure out why I cannot use the Dial
command from my AGI script.  Can anyone tell me what I can do to get
more information about what's going on.  I've tried asterisk -v with as
many v's as I can put on one line (like 40), and I was wondering if
there is anything that I can do to debug this problem.

Thanks a lot,

--Brian

On Fri, Sep 15, 2006 at 09:51:01AM -0400, Brian Rogan wrote:
  

Hello,

I am trying to write an AGI application that will transfer the caller to
a phone number on certain conditions.  From what I understand (from the
astcc application and voip-info wiki), I should just be able to EXEC the
dial command.  I'm having problems with this though.  I send asterisk
the following:

EXEC Dial "Zap/g1/8475881188|30"

I get back:

200 result=-1

On the asterisk console I see:

-- AGI Script Executing Application: (Dial) Options:
(Zap/g1/8475881188|30)
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/8475881188
-- Hungup 'Zap/1-1'


The HANGUPCAUSE variable is set to 0.  When I put this dial in my
dialplan as Dial(Zap/g1/8475881188|30), the call goes through fine, so I
don't think that its the T card or any configuration.

Has anyone ever seen this before?

Thanks,

--Brian
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__

Type AGI debug in the console

Thanks,
Steve Totaro
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Re: [asterisk-users] Issues with AGI+Dial command

[Doug Lytle]

Brian Rogan wrote:

Hi,

many v's as I can put on one line (like 40), and I was wondering if
there is anything that I can do to debug this problem.
  


/etc/asterisk/logger.conf

Uncomment the full and restart Asterisk.

You'll find the log in:

/var/log/asterisk/full

Doug



-- Ben Franklin quote: "Those who would give up Essential Liberty to 
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Re: [asterisk-users] Issues with AGI+Dial command

[Brian Rogan]

Hi,

I am still working on trying to figure out why I cannot use the Dial
command from my AGI script.  Can anyone tell me what I can do to get
more information about what's going on.  I've tried asterisk -v with as
many v's as I can put on one line (like 40), and I was wondering if
there is anything that I can do to debug this problem.

Thanks a lot,

--Brian

On Fri, Sep 15, 2006 at 09:51:01AM -0400, Brian Rogan wrote:
> Hello,
> 
> I am trying to write an AGI application that will transfer the caller to
> a phone number on certain conditions.  From what I understand (from the
> astcc application and voip-info wiki), I should just be able to EXEC the
> dial command.  I'm having problems with this though.  I send asterisk
> the following:
> 
> EXEC Dial "Zap/g1/8475881188|30"
> 
> I get back:
> 
> 200 result=-1
> 
> On the asterisk console I see:
> 
> -- AGI Script Executing Application: (Dial) Options:
> (Zap/g1/8475881188|30)
> -- Requested transfer capability: 0x00 - SPEECH
> -- Called g1/8475881188
> -- Hungup 'Zap/1-1'
> 
> 
> The HANGUPCAUSE variable is set to 0.  When I put this dial in my
> dialplan as Dial(Zap/g1/8475881188|30), the call goes through fine, so I
> don't think that its the T card or any configuration.
> 
> Has anyone ever seen this before?
> 
> Thanks,
> 
> --Brian
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Re: [asterisk-users] Polycom programmable buttons

[Noah Miller]

Hi Again Ron -


Yeah i was messing around last night and saw that! Now if I can only get the
other caller to not hear the DTMF digits id be set! I didnt know you could
remap the keys to DTMF digits, but since I can do that this will work
perfect for the most part!


Let me qualify by saying that you can remap a key to a DTMF digit
(notice the singular).  You can now (as of sip firmware 2.01) remap a
key to a speed dial for multiple DTMF digits, but this may start a new
channel.  I'm not sure as I haven't yet tested this.

- Noah
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Re: [asterisk-users] Polycom programmable buttons

[Ron McCarthy]

Yeah i was messing around last night and saw that! Now if I can only get the other caller to not hear the DTMF digits id be set! I didnt know you could remap the keys to DTMF digits, but since I can do that this will work perfect for the most part!
Thanks for the info!On 9/17/06, Noah Miller <[EMAIL PROTECTED]> wrote:
Hi Ron -> Is there a way to program one of the buttons on the 501 (Like the services
> button) to do on the fly call recording? So in the middle of the phonecall> you can record the call without have to do a transfer type of setup. Ive> looked at the manual but cant seem how to do that, I only see how you can
> "remap" a key function.You can use the asterisk "automon" feature along with Polycom keyremapping.  In features.conf, just set the DTMF code you'd like to useautomon, and then in the Polycom 
sip.cfg file, remap your Services keyto this DTMF code.The recordings will be placed in /var/spool/asterisk/automon- Noah___--Bandwidth and Colocation provided by 
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[asterisk-users] Termination Rates

[broadbandvoice]

I saw this termination company, www.BuyMin.com the rates looks good. Has anyone any experience with this company? I use Gafachi, very reliable but expensive.

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Re: [asterisk-users] Asterisk Server Down

[broadbandvoice]

Thanks everyone it is working now.
 
-- Original message -- From: Tzafrir Cohen <[EMAIL PROTECTED]> > On Sun, Sep 17, 2006 at 10:40:16AM -0400, Steve Totaro wrote: > > > >you're right, one should proof, under which user asterisk runs... > > >Besides security reasons, running asterisk as root, doesn't it allow a > > >higher prioritization of asterisk processes? > > This is why we let asterisk setuid itself to user asterisk, and don't > let the wrappr script handle that. Asterisk sets scheduling priority > before running setuid/setgid . > > > I can see a problem with security issues but is it a bad thing to allow > > higher priority of the asterisk process? Not sure that it does anyways, > > but I don't see how that is a bad thing? > > It can help the qu
 ality 
of Audio. On the downside, it means that a 100% > CPU loop in asterisk is a pain to recover from. Security implications: > if someone can inject you one line to the dialpan, they can (under the > right circumstances) get your system stuck very badly . Unless you have > a manager connection availble. > > -- > Tzafrir Cohen sip:[EMAIL PROTECTED] > icq#16849755 iax:[EMAIL PROTECTED] > +972-50-7952406 jabber:[EMAIL PROTECTED] > [EMAIL PROTECTED] http://www.xorcom.com > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users 

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Re: [asterisk-users] Starting out

[Tim Panton]

On 17 Sep 2006, at 12:25, Timothy Parez wrote:


Hi,

I have to decide on hardware to buy real fast (being rocketed into  
the situation).


We have 1 computer, we'll install hardware from digium in there to  
connect with the ISDN phone lines (2)
It's a normal computer, I have no idea what type of card to take  
and about the 3.3v vs 5v  PCI.


The  idea is the following.
We'll have about 10 internal phones.
One of the phones should be like a central station, where all  
other  calls can be monitored (if possible)
and from that phone the user should be able to press a button to  
take over a call which is rining on another phone.


Then we need less advanced phones for the rest of us, but we should  
still be able to  pick up calls that are rining

on a phone in the same room. (if possible)

I live in Belgium and we are using ISDN lines.
If I had to select phones from this page: http:// 
www.voipsolutions.be/index.php/cPath/54_24

What whould you sugest and why ?

Also from this page: http://www.asterisk.org/hardware
What would you sugest and why ?

Stuff we need
- Call forwarding (to another internal phone, to a classic phone  
number)

- Call take over (picking up a phone that is rining somewhere else)
- Menu system (got this working)
- Voicemail (got this working)
- Allowing a employee who's in a hotel somewhere to phone the  
internal numbers using his softphone over the internet
- Allowing that same employee to use his sotphone in order to make  
phonecalls to normal landlines through our server

- Call monitoring/recording (got this working)

I know it's a lot to ask and a lot of it is probably documented  
somewhere (although I couldn't find it in the asterisk manual draft)

but like I said I have very little time to decide

Thank you for any information you might be able to provide.

Tim.


All the things you want are not too hard to do in asterisk, we have a  
similar sort of set-up, except that

we happen (for historical reasons) to use a PRI isdn line.

The only thing on the list I see any problem with is forwarding - it  
works fine,
but if you mean forwarding an incoming PSTN call to a second PSTN  
number, you will
find that you end up using 2 channels unless you pick an ISDN card  
and driver
that can send the magic message to the providers switch to get it to  
do the work.
- Mine doesn't (Hairpinning I think it is called). With only 4 lines  
for 10

people consuming 2 for a call might be inconvenient.

My other bit of advice is _don't_ hurry the choice of phone.
Phones are a trade off - size/price/features etc. There is no single
'best' phone. My current favorite for an office SIP phone is the  
elmeg 290,

but others will tell you different.

You _can't_ tell what a phone is like until you live with it for a  
day or 2.
So set up your asterisk, make a shortlist of 3 phones and buy one of  
each.

Try them out, pick the one you like, then buy 9 more of them.

Tim Panton

www.mexuar.com



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[asterisk-users] Register message received from realtime peer crashes Asterisk

[kjcsb]

When Asterisk (1.2.12.1) receives a SIP register message for a realtime
peer, the CLI reports "Disconnected from Asterisk server". Asterisk has
disappeared:
asterisk -r
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl
exist?)

A look at the full log doesn't reveal much:
Sep 17 06:11:25 DEBUG[11011] acl.c: # Testing 60.234.nnn.nnn with
192.168.1.0
Sep 17 06:11:25 DEBUG[11011] chan_sip.c: Target address 60.234.nnn.nnn is
not local, substituting externip
Sep 17 06:11:25 DEBUG[11011] res_config_mysql.c: MySQL RealTime: Retrieve
SQL: SELECT * FROM sip_buddies WHERE name = '6000'
Sep 17 06:11:25 DEBUG[11011] res_config_mysql.c: MySQL RealTime: Everything
is fine.

Asterisk then restarts (it gets a new pid) and will continue running happily
until a new register request is received for a realtime peer. Note that
Asterisk operates normally in all other respects until the register is
received e.g. sip peers in sip.conf can register and make calls
successfully. Only when a register is received from a peer that exists in
sip_buddies does Asterisk crash.

I can run the query successfully on mysql command line:
SELECT * FROM sip_buddies WHERE name = '6000';

1 row in set (0.32 sec)

A review of syslog and the mysql log reveals little:
mysql log
060917  6:54:31  14 Init DB asterisk
  14 Query   SELECT * FROM sip_buddies WHERE name = '6000'

syslog
Nothing report at the time of the crash (06:54).

extconfig.conf
[settings]
sipusers => mysql,asterisk,sip_buddies
sippeers => mysql,asterisk,sip_buddies

res_mysql.conf
[general]
dbhost = 127.0.0.1
dbname = asterisk
dbuser = root
dbpass = password
dbport = 3306

Could anyone advise what's going on or further checking that I could do to
analyse the problem?

Thanks

Cameron

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Re: [asterisk-users] How to install HUDLite Server

[Nicolás Gudiño]

As for FOP, when clients come to meet you after seeing attractive interfaces
from other proprietary systems, its just embarrassing to show them such an
ugly interface like FOP.


FOP interface, altough it has some limitations (fixed button positions
and size for the flash client), it is pretty much configurable. It can
look ugly or impresive, depending on the time you put to create your
own look. You can use custom background for the whole screen, or for
each button, etc.

Using the dhtml panel there is no limitations on the layout, you can
make it look as anything, including HUD. It is limited to your html
skills. Have you taken a look at http://www.asternic.org live demo?

Anyways, attractive is not everything, sometimes the information
displayed is more important. The most advanced call center reports I
saw are just plain text with standard formatting. I guess the
important thing on this interfaces should be usability and flexibility
and not only looks.

Best regards,


--
Nicolás Gudiño
Buenos Aires - Argentina
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Re: [asterisk-users] A1200+fxo, anyone using this?

[Doug Lytle]

Nick Ellson wrote:



So has anyone used this card with Asterisk? I googled for reviews and 
have not found anything, and I am tryingto find a way to search the 
archives without looking at each month one at a time.




I was able to download the .gz files, extract them into SeaMonkey's mail 
directory and use them directly.


Doug



-- Ben Franklin quote: "Those who would give up Essential Liberty to 
purchase a little Temporary Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] Why not g726-32?

[Rich Adamson]

RR wrote:

On 9/16/06, Rich Adamson <[EMAIL PROTECTED]> wrote:

RR wrote:
> All,
>
> is there anyone who uses g726-32 ? If not, then does anyone know why
> don't people use it?

I use g726 on iax links between systems and to teliax.com for LD calls.
Have no idea if its -32 or what though. What ships with asterisk (in
terms of g726) has been working very well for us with the exception of a
period of time where all g726 calls via teliax were not usable. Teliax
had to have had a problem or was playing around as that was the only iax
link that had bad audio.


Thanks Rich for the positive email about g726. Just FYI, (*) supports
only g726-32 AFAIK so that's probably what you've been using. I don't
have the worry of Teliax as I'd probably never be using them or at
least not in the immediate/near future. Like I said, all I want to do
is avoid usage of license fees, save bandwidth, and not stress out my
systems with cpu intensive codecs like g729 and maybe find something
that can still deliver comparable quality.

I'm still confused as to why more people and carriers don't use g726
however. 


I can only guess that many itsp's actually support it, but don't 
advertise its availability, just like they don't advertise ilbc, etc. 
I'd also have to guess that phone manufacturers haven't implemented it 
(in the past) due to limits on memory, etc.



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Re: [asterisk-users] Polycom Expansion Module

[Noah Miller]

Hi Kevin -


Has anyone used the Polycom expansion module with multiple lines?

My application is for 20 lines and read there was a limit of 7 at one point.


I heard rumors that the newest version of the polycom sip firmware
(2.01) would lift the limit of 7.  It just came out, and I haven't had
time to test it yet, but you can give it a try.

- Noah
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Re: [asterisk-users] DTMF Tone Not Passing Help

[Moises Silva]

Hi, dont expect too much help providing only the Asterisk version you
are using. You need to tell us the call path, DTMF mode (inband,
outband, SIP INFO etc) used and call technologies involved (SIP, ZAP,
IAX2 etc).

Regards

On 9/15/06, Nitesh Divecha <[EMAIL PROTECTED]> wrote:

Hello All,

Can anyone help me with this DTMF tone problem.

I am running Asterisk 1.2.9.1 svn with couple of Polycom and Snom
phones.

None of the phones are passing the DTMF tones to remote IVR.

Called 1-800-FLOWERS, when asked to press 1, none of the phones were
able to transmit the digit.

Can anyone please guide me to correct path on how to solve this problem.

Thanks,
Nitesh


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--
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
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Re: [asterisk-users] Polycom programmable buttons

[Noah Miller]

Hi Ron -


Is there a way to program one of the buttons on the 501 (Like the services
button) to do on the fly call recording? So in the middle of the phonecall
you can record the call without have to do a transfer type of setup. Ive
looked at the manual but cant seem how to do that, I only see how you can
"remap" a key function.


You can use the asterisk "automon" feature along with Polycom key
remapping.  In features.conf, just set the DTMF code you'd like to use
automon, and then in the Polycom sip.cfg file, remap your Services key
to this DTMF code.

The recordings will be placed in /var/spool/asterisk/automon

- Noah
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[asterisk-users] RE: FollowMe question

[Hall, Eric M.]

I got the config working. Not sure if someone has 
pre-recorded sounds for this app or not. Looked all over for them and I'm unable 
to locate them.If anyone has sound file they would like to share that would help 
me greatly.
 
Thanks
 


 Sent: Friday, September 15, 
2006 5:23 PMTo: 'asterisk-users@lists.digium.com'Subject: 
FollowMe question

Group
 Does anyone 
have the FollowMe sound files? Do I need to record them?
Also does anyone 
have a working followme.conf file that they would share?
Thanks!
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[asterisk-users] A1200+fxo, anyone using this?

[Nick Ellson]

I know it's not a digium product, but the 12 port A1200P card with a 
single FXO module at pbxeq.com at first glance would seem to be the way to 
get started for me with an in-system controller card. 4 ports seems too 
small for expansion, the huge 24 port card a tad too big (and spendy).


So has anyone used this card with Asterisk? I googled for reviews and have 
not found anything, and I am tryingto find a way to search the archives 
without looking at each month one at a time.


Nick


--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.

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Re: [asterisk-users] Asterisk Server Down

[Tzafrir Cohen]

On Sun, Sep 17, 2006 at 10:40:16AM -0400, Steve Totaro wrote:

> >you're right, one should proof, under which user asterisk runs...
> >Besides security reasons, running asterisk as root, doesn't it allow a
> >higher prioritization of asterisk processes?

This is why we let asterisk setuid itself to user asterisk, and don't
let the wrappr script handle that. Asterisk sets scheduling priority
before running setuid/setgid .

> I can see a problem with security issues but is it a bad thing to allow 
> higher priority of the asterisk process?  Not sure that it does anyways, 
> but I don't see how that is a bad thing?

It can help the quality of Audio. On the downside, it means that a 100%
CPU loop in asterisk is a pain to recover from. Security implications:
if someone can inject you one line to the dialpan, they can (under the
right circumstances) get your system stuck very badly . Unless you have
a manager connection availble.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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Re: [asterisk-users] Asterisk Server Down

[Steve Totaro]

Guido Hecken wrote:

-Ursprüngliche Nachricht-
Von: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
Gesendet: Sonntag, 17. September 2006 15:56
An: asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] Asterisk Server Down

On Sun, Sep 17, 2006 at 03:54:46PM +0200, Guido Hecken wrote:


Hi,

obviously asterisk doesn't start with the installed(?) start script.
Try to start it manually and watch the cli for informations with
asterisk -vvvc
  

One warning: if your system is normally configured to run as non-root,
this may cause it to write some fiels as root, and not start properly
next time you start it with the standard script.

With the Debian packages, use:

/etc/init.d/asterisk debug

Which is normally just a glorified:

  asterisk -U asterisk -vv



Tzafrir,

you're right, one should proof, under which user asterisk runs...
Besides security reasons, running asterisk as root, doesn't it allow a
higher prioritization of asterisk processes?

Guido

  
I can see a problem with security issues but is it a bad thing to allow 
higher priority of the asterisk process?  Not sure that it does anyways, 
but I don't see how that is a bad thing?

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Re: [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy

[Steve Totaro]

Much higher, maybe double but that is when the agents start to complain 
that their conversations start "cutting in and out".


This is the main reason I am looking into building a re-invite solution 
that moves the call into "recording/cdr" server's media path.  Then we 
just cap these servers at 50-70 calls and keep track of it in a DB.  I 
think my developers could re-write asterisk to handle this.  It will 
also allow the main "Queue" server(s) to go down, so long as the 
"recording/cdr" server is still up no ongoing calls would get lost.


Thanks,
Steve Totaro

Matt Florell wrote:

To maintain high recording quality with no audio skips we have found
that you should not go over 50 conversations being recorded on a
single server. What have you found is your limit while maintaining
very good audio quality?

MATT---

On 9/16/06, Steve Totaro <[EMAIL PROTECTED]> wrote:

Right now we are all inbound and every call is recorded.

Matt Florell wrote:
> Hello,
>
> At this point in time VICIDIAL is more focused on outbound features,
> but inbound and blended capcbilities have been part of VICIDIAL for
> about two years. The most I have done inbound-only with it is 3 T1s
> with 60 agents. But for outbound and inbound agents together we have
> had upto 120 agents on one setup with 20 T1s(spread across 8 Asterisk
> servers, 2 web servers and 1 MySQL server) handling over 200,000 calls
> a day(mostly outbound of course).
>
> The inbound portion of VICIDIAL does not have customized hold music or
> periodic announcements yet, but we plan on adding those features in a
> future version as we begin to focus more on inbound in the project.
>
> We do use meetme rooms in VICIDIAL. This allows for easy third-party
> calls and multiple monitoring/manager intrusion into an agent session.
>
> The load balancing works by having the Database keep track of agents
> and calls for all of the servers and then use IAX to move the calls
> from where they originated to whoever the next available agent is, no
> matter what server they are on. So the calls can come from anywhere
> and the agents can be logged in anywhere.
>
> If you receive inbound callerID on these calls you will also be able
> to have caller information appear on the Agent's web interface. And if
> they are a customer that is already in the system it will bring up
> their existing record.
>
> As for reliability, we have not had a total system failure in the last
> 2 years(aside from long power outages and hurricane interruptions).
> MySQL can handle a tremendous volume and it is the only
> total-system-single-point-of-failure in VICIDIAL, ours never crashes.
> The web servers can be load balanced(no need for session-awareness)
> and you can use any Apache/PHP webserver that may be on your system to
> serve the AJAX-based agent web interface. As for Asterisk, we have had
> servers crash periodically(a couple crashes a month across 8 servers),
> but that is to be expected when you push tens of thousands of calls
> through each one per day.
>
> Will you be recording all calls in this setup?
>
> MATT---
>
> On 9/16/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
>> Matt,
>>
>> I am sure this is a RTFM and I am pretty sure you are using meetme
>> rooms.  Just not too sure how you do the magic.
>>
>> 28 T1s with NFAS so 95 channels per trunk group, seven trunk groups =
>> 665 lines.  My client's call volume has shot from 5,000 to about 
10,000
>> calls a day.  Due to recent product offerings/advertising, I 
expect to
>> be eating up 6 T1 (peak) by the end  of October.  They will 
eventually
>> have every channel in use during peaks, whether that is in 
November or
>> December, I am not sure.  I just know it can't break at that point 
due

>> the the sheer expense of revenue lost for downtime.
>>
>> Thanks,
>> Steve
>>
>>
>> Matt Florell wrote:
>> > How many lines and agents are you looking at?
>> >
>> > What kind of call volume?
>> >
>> > Average expected hold time?
>> >
>> > VICIDIAL could be an option for you since it does not use Asterisk
>> > Queues and can already easily scale across many servers.
>> >
>> > MATT---
>> >
>> >
>> > On 9/15/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
>> >> I have been tossing around some ideas about scaling a call center
>> with
>> >> load balancing and redundancy and would like the comunities input,
>> >> thoughts, criticism and anything anyone wants to toss in.
>> >>
>> >> The most evident thing is to start with beefy servers and only run
>> procs
>> >> that are required.  All of the TDM boxes run stripped down
>> versions of
>> >> Linux and Asterisk, they just take the call from the PRIs and 
convert

>> >> them to SIP, everything stays ulaw end to end.
>> >>
>> >> *Shared queues across multiple servers would be ideal*.  I don't
>> think
>> >> it is possible in asterisk, as is.  Maybe DUNDI could be useful
>> but I am
>> >> not up to speed on it enough to really know.
>> >>
>> >> I was toying with a concept of a DB server tracking the number

RE: [asterisk-users] Asterisk Server Down

[Guido Hecken]

> -Ursprüngliche Nachricht-
> Von: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
> Gesendet: Sonntag, 17. September 2006 15:56
> An: asterisk-users@lists.digium.com
> Betreff: Re: [asterisk-users] Asterisk Server Down
> 
> On Sun, Sep 17, 2006 at 03:54:46PM +0200, Guido Hecken wrote:
> > Hi,
> >
> > obviously asterisk doesn't start with the installed(?) start script.
> > Try to start it manually and watch the cli for informations with
> > asterisk -vvvc
> 
> One warning: if your system is normally configured to run as non-root,
> this may cause it to write some fiels as root, and not start properly
> next time you start it with the standard script.
> 
> With the Debian packages, use:
> 
> /etc/init.d/asterisk debug
> 
> Which is normally just a glorified:
> 
>   asterisk -U asterisk -vv

Tzafrir,

you're right, one should proof, under which user asterisk runs...
Besides security reasons, running asterisk as root, doesn't it allow a
higher prioritization of asterisk processes?

Guido

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[asterisk-users] Re: [Users] Integrating the Openser for VoiceMail and PBX with Asterisk, For Account

[Rafael J. Risco G.V.]

try this:
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER
 
On 9/16/06, raviprakash sunkara <[EMAIL PROTECTED]> wrote:

Hi Users,I'm  new to Asterisk programming , I'm in working the Voip Technologies by using the  OpenSER for my call routing process and Radius For AAA.But in Asterisk i need   it for only PBX and VoiceMail,
For Account  I'm using the  Openser + Radius . Main My doubt is  that,    For Call Routing my using the OpenSER. every thing is fine and Good ..        But i need the Voicemail , it forword to the Asterisk Server, that . Is the Openser takes the Accounting part or Asterisk it Take place  
      Who did the Database know the User had a voicemail in his voice mail Box... That Databases i need  How ?Please Help me And Mainly Excuse me in English    
-- 



Thanks and RegardsRavi Prakash Sunkara 

  

M:+91 9985077535O
:+91 40 23114549F:+91 40 40208727 

[EMAIL PROTECTED]
www.hyperion-tech.com

 ___Users mailing listUsers@openser.org
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Re: [asterisk-users] rxfax, spandsp and lack of ecm

[Rosario Pingaro]

I still have problem with the latest implementation of rxfax for 1.2.

In fact it exit with non zero from the macro and it is not going to execute 
the system script the have the fax2mail service.

Has someone the same experience?

Thanks to Steve U.

Rosario

- Original Message - 
From: "Steve Davies" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Thursday, September 14, 2006 5:15 AM
Subject: Re: [asterisk-users] rxfax, spandsp and lack of ecm



On 9/14/06, Steve Underwood <[EMAIL PROTECTED]> wrote:

Steve Davies wrote:

[snip]

>
> This looks pretty good I have to say - The ECM seems as if it may be a
> little intolerant... On a fax machine where I got 100% success in the
> past with 0.0.2, I am now getting "result (60) Disconnected after
> permitted retries." on about every 4th page.
>
> Is the ECM tolerance level tuneable in spandsp, or is this
> hard-defined in the standard? Is it just a matter of changing:
>  #define MAX_MESSAGE_TRIES   3

Your problem probably has nothing to do with tolerance. If an exchange
doesn't succeed after 3 tries, it is unlikely to ever succeed. You are
probably hitting a bug. It is new code. :-)  Can you enable debug with
"|debug" on the command line to rxfax/txfax, and send me the log?


Already done this, and started sending hi-res faxes to make the
problem more evident. This log fails at the end of the second page -
It always seems to allow the communication to get to the end of the
page before failing:

Sep 13 17:14:23 DEBUG[17696]: Launching 'zapEC'
Sep 13 17:14:23 DEBUG[17696]: disabled echo cancellation on channel 4
Sep 13 17:14:23 DEBUG[17696]: Launching 'RxFAX'
Sep 13 17:14:23 DEBUG[17696]: Set channel Zap/4-1 to read format slin
Sep 13 17:14:23 DEBUG[17696]: Set channel Zap/4-1 to write format slin
Sep 13 17:14:25 DEBUG[17696]: FLOW FAX Set rx type 0
Sep 13 17:14:25 DEBUG[17696]: FLOW FAX Set tx type 4
Sep 13 17:14:27 DEBUG[17696]: FLOW FAX Set rx type 4
Sep 13 17:14:27 DEBUG[17696]: FLOW FAX Set tx type 0
Sep 13 17:14:29 DEBUG[17696]: FLOW FAX Set rx type 8
Sep 13 17:14:29 DEBUG[17696]: FLOW FAX Set tx type 0
Sep 13 17:14:30 DEBUG[17696]: FLOW FAX Switching from V.29 + V.21 to V.29
Sep 13 17:14:31 DEBUG[17696]: FLOW FAX Set rx type 0
Sep 13 17:14:31 DEBUG[17696]: FLOW FAX Set tx type 4
Sep 13 17:14:32 DEBUG[17696]: FLOW FAX Set rx type 8
Sep 13 17:14:32 DEBUG[17696]: FLOW FAX Set tx type 0
Sep 13 17:14:33 DEBUG[17696]: FLOW FAX Switching from V.29 + V.21 to V.29
Sep 13 17:17:59 DEBUG[17696]: FLOW FAX Set rx type 4
Sep 13 17:17:59 DEBUG[17696]: FLOW FAX Set tx type 0
Sep 13 17:18:01 DEBUG[17696]:
==
Sep 13 17:18:01 DEBUG[17696]: Pages transferred:  0
Sep 13 17:18:01 DEBUG[17696]: Image size: 1728 x 4529
Sep 13 17:18:01 DEBUG[17696]: Image resolution8037 x 15400
Sep 13 17:18:01 DEBUG[17696]: Transfer Rate:  9600
Sep 13 17:18:01 DEBUG[17696]: Bad rows0
Sep 13 17:18:01 DEBUG[17696]: Longest bad row run 0
Sep 13 17:18:01 DEBUG[17696]: Compression type1
Sep 13 17:18:01 DEBUG[17696]: Image size (bytes)  978264
Sep 13 17:18:01 DEBUG[17696]:
==
Sep 13 17:18:01 DEBUG[17696]: FLOW FAX Set rx type 0
Sep 13 17:18:01 DEBUG[17696]: FLOW FAX Set tx type 4
Sep 13 17:18:02 DEBUG[17696]: FLOW FAX Set rx type 8
Sep 13 17:18:02 DEBUG[17696]: FLOW FAX Set tx type 0
Sep 13 17:18:09 DEBUG[17696]: FLOW FAX Set rx type 0
Sep 13 17:18:09 DEBUG[17696]: FLOW FAX Set tx type 4
Sep 13 17:18:10 DEBUG[17696]: FLOW FAX Set rx type 4
Sep 13 17:18:10 DEBUG[17696]: FLOW FAX Set tx type 0
Sep 13 17:18:14 DEBUG[17696]: FLOW FAX Set rx type 0
Sep 13 17:18:14 DEBUG[17696]: FLOW FAX Set tx type 4
Sep 13 17:18:15 DEBUG[17696]: FLOW FAX Set rx type 4
Sep 13 17:18:15 DEBUG[17696]: FLOW FAX Set tx type 0
Sep 13 17:18:19 DEBUG[17696]: FLOW FAX Set rx type 0
Sep 13 17:18:19 DEBUG[17696]: FLOW FAX Set tx type 4
Sep 13 17:18:20 DEBUG[17696]: FLOW FAX Set rx type 4
Sep 13 17:18:20 DEBUG[17696]: FLOW FAX Set tx type 0
Sep 13 17:18:24 DEBUG[17696]: FLOW FAX Set rx type 0
Sep 13 17:18:24 DEBUG[17696]: FLOW FAX Set tx type 4
Sep 13 17:18:25 DEBUG[17696]: FLOW FAX Set rx type 4
Sep 13 17:18:25 DEBUG[17696]: FLOW FAX Set tx type 0
Sep 13 17:22:16 DEBUG[17696]: FLOW FAX Set rx type 0
Sep 13 17:22:16 DEBUG[17696]: FLOW FAX Set tx type 4
Sep 13 17:22:17 DEBUG[17696]: FLOW FAX Set rx type 0
Sep 13 17:22:17 DEBUG[17696]: FLOW FAX Set tx type 1
Sep 13 17:22:17 DEBUG[17696]:
==
Sep 13 17:22:17 DEBUG[17696]: Fax receive not successful - result (60)
Disconnected after permitted retries.
Sep 13 17:22:17 DEBUG[17696]:
==
Sep 13 17:22:17 DEBUG[17696]: FLOW FAX Set rx type 13
Sep 13 17:22:17 DEBUG[17696]: FLOW FAX FAX exchange complete
Sep 13 17:22:

Re: [asterisk-users] Asterisk Server Down

[Tzafrir Cohen]

On Sun, Sep 17, 2006 at 03:54:46PM +0200, Guido Hecken wrote:
> Hi,
> 
> obviously asterisk doesn't start with the installed(?) start script.
> Try to start it manually and watch the cli for informations with
> asterisk -vvvc

One warning: if your system is normally configured to run as non-root,
this may cause it to write some fiels as root, and not start properly
next time you start it with the standard script.

With the Debian packages, use:

/etc/init.d/asterisk debug

Which is normally just a glorified:

  asterisk -U asterisk -vv

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Re: [asterisk-users] Wrong outgoing port

[Patrick Cervicek]

Master_PE schrieb:


I have changed in sip.conf the bindport (port=) to (bindport=) .

When i try to connect to my provider it doesn't work becourse it  try's 
to connect to port 5060. sip debug says

Retransmitting #5 (NAT) to 62.177.135.42:5060:
REGISTER sip:62.177.135.42 SIP/2.0

trion*CLI> sip show registry
62.177.135.41:38383 31##120 Request Sent

What's wrong. Somehow it looks like a bug.


You changed the port of your Asterisk Server where it should listen to.
When you want to change the port of your provider, you have to use the 
"port" statement or specify the port it in the "register" statement


 register => user[:secret[:[EMAIL PROTECTED]:port][/extension]

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RE: [asterisk-users] Asterisk Server Down

[Guido Hecken]

Hi,

obviously asterisk doesn't start with the installed(?) start script.
Try to start it manually and watch the cli for informations with
asterisk -vvvc
AFAIK a make config in the asterisk source should install the start script
for your system.

Hope it helps...

Guido

Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Gesendet: Sonntag, 17. September 2006 15:27
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] Asterisk Server Down

I rebooted the server on which the Asterisk is hosted on. The * did not come
back up and I get this message when I attempt to use CLI
 
[EMAIL PROTECTED] ~]# asterisk -r
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
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Re: [asterisk-users] Asterisk Server Down

[Patrick Cervicek]

[EMAIL PROTECTED] schrieb:
I rebooted the server on which the Asterisk is hosted on. The * did not 
come back up and I get this message when I attempt to use CLI
 
[EMAIL PROTECTED] ~]# asterisk -r

Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)


Did you configure your server in /etc/rc?.d/, that it sould start after 
reboot?


Tools:
rcconf (debian)
chkconfig (fedora,redhat)

What does /var/log/asterisk say?
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[asterisk-users] Asterisk Server Down

[broadbandvoice]

I rebooted the server on which the Asterisk is hosted on. The * did not come back up and I get this message when I attempt to use CLI
 
[EMAIL PROTECTED] ~]# asterisk -rUnable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)

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Re: [asterisk-users] Starting out

[Brian Rogan]

On Sun, Sep 17, 2006 at 01:25:10PM +0200, Timothy Parez wrote:
> We'll have about 10 internal phones.
> One of the phones should be like a central station, where all other  
> calls can be monitored (if possible)
> and from that phone the user should be able to press a button to take 
> over a call which is rining on another phone.
> 
> Then we need less advanced phones for the rest of us, but we should 
> still be able to  pick up calls that are rining
> on a phone in the same room. (if possible)
> 
> I live in Belgium and we are using ISDN lines.
> If I had to select phones from this page: 
> http://www.voipsolutions.be/index.php/cPath/54_24
> What whould you sugest and why ?

I have used the Aastra 9112i and it seems to be an a very solid, reliable
(not to mention inexpensive) phone.  It also has instructions on how to
configure it with Asterisk right out of the box, which is nice.  I would
recommend it without reservation.

You might also want to look at the Aastra480i for your central station.
One of the neat things that the phone offers is a whole XML based
markup language that allows you to design custom menus which call HTTP
pages and stuff (much like HTML).  This might be just the thing for
being able to take over any current calls from the central phone.

--Brian
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[asterisk-users] Does a "HST Saphir III ML PCI" work with Asterisk?

[Patrick Cervicek]

I am looking for Infos & Tutorials for installing "ISDN Karte PCI HST 
Saphir III ML". Does sombody have infos & links for me?


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[asterisk-users] IAX2 audio problem

[Siqhamo Sifo]

I have a problem when making calls to and from my iax2 client(ifdefisk) I
only get one way audio and even when I make calls to a pstn line I have
tried the ff on iax.conf : allow=all and it does not seem to help.


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[asterisk-users] Starting out

[Timothy Parez]

Hi,

I have to decide on hardware to buy real fast (being rocketed into the 
situation).


We have 1 computer, we'll install hardware from digium in there to 
connect with the ISDN phone lines (2)
It's a normal computer, I have no idea what type of card to take and 
about the 3.3v vs 5v  PCI.


The  idea is the following.
We'll have about 10 internal phones.
One of the phones should be like a central station, where all other  
calls can be monitored (if possible)
and from that phone the user should be able to press a button to take 
over a call which is rining on another phone.


Then we need less advanced phones for the rest of us, but we should 
still be able to  pick up calls that are rining

on a phone in the same room. (if possible)

I live in Belgium and we are using ISDN lines.
If I had to select phones from this page: 
http://www.voipsolutions.be/index.php/cPath/54_24

What whould you sugest and why ?

Also from this page: http://www.asterisk.org/hardware
What would you sugest and why ?

Stuff we need
- Call forwarding (to another internal phone, to a classic phone number)
- Call take over (picking up a phone that is rining somewhere else)
- Menu system (got this working)
- Voicemail (got this working)
- Allowing a employee who's in a hotel somewhere to phone the internal 
numbers using his softphone over the internet
- Allowing that same employee to use his sotphone in order to make 
phonecalls to normal landlines through our server

- Call monitoring/recording (got this working)

I know it's a lot to ask and a lot of it is probably documented 
somewhere (although I couldn't find it in the asterisk manual draft)

but like I said I have very little time to decide

Thank you for any information you might be able to provide.

Tim.
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Re: [asterisk-users] system cmd

[unplug]

Thanks.
Could you tell me some detail how to compile and implement it with asterisk?

On 9/17/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

On Sun, Sep 17, 2006 at 01:13:07PM +0800, unplug wrote:
> How can I use a system cmd to get back the return value in dial plan?
> Say, I want to run a script using system cmd to get the hostname.
> System(hostname)

Reading previous posts helps.

On Sat, Sep 16, 2006 at 01:15:01PM +0200, Christophorus Laube wrote:
> Hi list,
>
> is it possible to call a shell script from * which returns a number or
> a string which can be read to an asterisk variable? Something like
> 'Set(VAR(System(/opt/scripts/something.script)))?
> Does anyone have an idea?

http://pbxfreeware.org/app_backticks.c

--
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+972-50-7952406  jabber:[EMAIL PROTECTED]
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Re: [asterisk-users] Why not g726-32?

[RR]

On 9/16/06, Rich Adamson <[EMAIL PROTECTED]> wrote:

RR wrote:
> All,
>
> is there anyone who uses g726-32 ? If not, then does anyone know why
> don't people use it?

I use g726 on iax links between systems and to teliax.com for LD calls.
Have no idea if its -32 or what though. What ships with asterisk (in
terms of g726) has been working very well for us with the exception of a
period of time where all g726 calls via teliax were not usable. Teliax
had to have had a problem or was playing around as that was the only iax
link that had bad audio.


Thanks Rich for the positive email about g726. Just FYI, (*) supports
only g726-32 AFAIK so that's probably what you've been using. I don't
have the worry of Teliax as I'd probably never be using them or at
least not in the immediate/near future. Like I said, all I want to do
is avoid usage of license fees, save bandwidth, and not stress out my
systems with cpu intensive codecs like g729 and maybe find something
that can still deliver comparable quality.

I'm still confused as to why more people and carriers don't use g726
however. Anyonbe else can shed any light on this?
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[asterisk-users] How does Asterisk determine an incoming SIP Channel name?

[kjcsb]

I have a number of different calls coming in to Asterisk from one SIP proxy. 
All calls are currently allocated the same SIP Channel name but I want them 
to be named differently. Note that Asterisk registers with the SIP Proxy, 
not the other way around.




sip.conf

register=5551234:[EMAIL PROTECTED]/5551234

register=5552345:[EMAIL PROTECTED]/5552345

register=5553456:[EMAIL PROTECTED]/5553456

register=5554567:[EMAIL PROTECTED]/5554567



[5551234  (Accounts)]
username=5551234
type=peer
host=domain.com



[5553456  (Sales)]
username=5553456
type=peer
host=domain.com



[5554567  (Support)]
username=5554567
type=friend
host=domain.com



[5552345]

username=5552345

type=friend

host=domain.com



When a call comes in from the host to 5551234 for example, the channel is 
named SIP/5552345-b7b0b8a8. The same name is given a call from 5553456.




If I remove the [5552345] entry from sip.conf then the channel is named 
SIP/5554567-b7b0b8a8 i.e. the channel is named according to the first 
username for the host starting at the bottom of the sip.conf file.




Could anyone suggest how I can achieve the desired result?



Thanks and regards



Cameron

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Re: [asterisk-users] system cmd

[Tzafrir Cohen]

On Sun, Sep 17, 2006 at 01:13:07PM +0800, unplug wrote:
> How can I use a system cmd to get back the return value in dial plan?
> Say, I want to run a script using system cmd to get the hostname.
> System(hostname)

Reading previous posts helps.

On Sat, Sep 16, 2006 at 01:15:01PM +0200, Christophorus Laube wrote: 
> Hi list, 
> 
> is it possible to call a shell script from * which returns a number or
> a string which can be read to an asterisk variable? Something like
> 'Set(VAR(System(/opt/scripts/something.script)))?
> Does anyone have an idea?

http://pbxfreeware.org/app_backticks.c

-- 
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icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
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