Re: [asterisk-users] chan_zap.c: Failed to read gains: Invalid argument

[Eric \"ManxPower\" Wieling]

Lee Archer wrote:
I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium TE110P 
card in E1 mode.  I've recently noticed in my logs the following


Jan  5 01:27:11 VERBOSE[22490] logger.c:  [chan_zap.so]Jan  5 01:27:11 
VERBOSE[22490] logger.c:  [chan_zap.so] => (Zapata Telephony w/PRI)


Jan  5 01:27:11 VERBOSE[22490] logger.c:   == Parsing 
'/etc/asterisk/zapata.conf': Jan  5 01:27:11 VERBOSE[22490] logger.c:   
== Parsing '/etc/asterisk/zapata.conf': Found


Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid 
argument
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid 




Which seems to suggest that I've done something wrong with the rx and 
txgain option in /etc/asterisk/zapata.conf.  But these haven't been 
changed in 18 months and still say


; You may also set the default receive and transmit gains (in dB)
;
rxgain=4.0
txgain=0.0

Have I done something wrong or has something changed?


Don't use fractional gains.  i.e. use rxgain=4 and txgain=0

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RE: [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument

[Lee Archer]

Sorry I should have stated that I've tried +x, -x, x.y and x and I still
get the same.

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
"ManxPower" Wieling
Sent: 05 January 2007 08:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] chan_zap.c: Failed to read gains:
Invalidargument

Lee Archer wrote:
> I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium 
> TE110P card in E1 mode.  I've recently noticed in my logs the 
> following
> 
> Jan  5 01:27:11 VERBOSE[22490] logger.c:  [chan_zap.so]Jan  5 01:27:11

> VERBOSE[22490] logger.c:  [chan_zap.so] => (Zapata Telephony w/PRI)
> 
> Jan  5 01:27:11 VERBOSE[22490] logger.c:   == Parsing 
> '/etc/asterisk/zapata.conf': Jan  5 01:27:11 VERBOSE[22490] logger.c:

> == Parsing '/etc/asterisk/zapata.conf': Found
> 
> Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid

> argument Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read 
> gains: Invalid

> 
> Which seems to suggest that I've done something wrong with the rx and 
> txgain option in /etc/asterisk/zapata.conf.  But these haven't been 
> changed in 18 months and still say
> 
> ; You may also set the default receive and transmit gains (in dB) ; 
> rxgain=4.0 txgain=0.0
> 
> Have I done something wrong or has something changed?

Don't use fractional gains.  i.e. use rxgain=4 and txgain=0

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[asterisk-users] Which g729 module for HP DL 360 G3 (Xeon CPU's)?

[Eric Bishop]

I am running a HP DL360 G3 ans want to know the optimal g729 module for it.
There don't seem to be any optimised for Xeon's. I am currently using i686,
but is there a better one to match my Xeon CPU's?



[EMAIL PROTECTED] ~]# cat /proc/cpuinfo
processor   : 0
vendor_id   : GenuineIntel
cpu family  : 15
model   : 2
model name  : Intel(R) Xeon(TM) CPU 2.80GHz
stepping: 7
cpu MHz : 2799.656
cache size  : 512 KB
physical id : 3
siblings: 1
core id : 3
cpu cores   : 1
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: no
fpu : yes
fpu_exception   : yes
cpuid level : 2
wp  : yes
flags   : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov
pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe cid xtpr
bogomips: 5602.71

processor   : 1
vendor_id   : GenuineIntel
cpu family  : 15
model   : 2
model name  : Intel(R) Xeon(TM) CPU 2.80GHz
stepping: 9
cpu MHz : 2799.656
cache size  : 512 KB
physical id : 0
siblings: 1
core id : 0
cpu cores   : 1
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: no
fpu : yes
fpu_exception   : yes
cpuid level : 2
wp  : yes
flags   : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov
pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe cid xtpr
bogomips: 5597.58
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Re: [asterisk-users] chan_zap.c: Failed to read gains: Invalid argument

[Tzafrir Cohen]

On Fri, Jan 05, 2007 at 07:47:15AM -, Lee Archer wrote:
> I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium TE110P
> card in E1 mode.  I've recently noticed in my logs the following
> 
> Jan  5 01:27:11 VERBOSE[22490] logger.c:  [chan_zap.so]Jan  5 01:27:11
> VERBOSE[22490] logger.c:  [chan_zap.so] => (Zapata Telephony w/PRI)
> Jan  5 01:27:11 VERBOSE[22490] logger.c:   == Parsing
> '/etc/asterisk/zapata.conf': Jan  5 01:27:11 VERBOSE[22490] logger.c:
> == Parsing '/etc/asterisk/zapata.conf': Found
> Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
> argument
> Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
> argument

This is a debug message and not even a warning message. I'm not sure
that this is something to worry about.

The code there tries to first read the gains and set the gains based on
them. The return value from the ioctl that sets the gains does not seem
to be checked in several code pathes, though. So it may actually fail
silently.

Do you get the same debug messages on 'reload chan_zap.so' ?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
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RE: [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument

[Lee Archer]

Yes I get the same message after reload chan_zap.so

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: 05 January 2007 08:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] chan_zap.c: Failed to read gains:
Invalidargument

On Fri, Jan 05, 2007 at 07:47:15AM -, Lee Archer wrote:
> I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium 
> TE110P card in E1 mode.  I've recently noticed in my logs the 
> following
> 
> Jan  5 01:27:11 VERBOSE[22490] logger.c:  [chan_zap.so]Jan  5 01:27:11

> VERBOSE[22490] logger.c:  [chan_zap.so] => (Zapata Telephony w/PRI)
> Jan  5 01:27:11 VERBOSE[22490] logger.c:   == Parsing
> '/etc/asterisk/zapata.conf': Jan  5 01:27:11 VERBOSE[22490] logger.c:
> == Parsing '/etc/asterisk/zapata.conf': Found Jan  5 01:27:11 
> DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan  5

> 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid 
> argument

This is a debug message and not even a warning message. I'm not sure
that this is something to worry about.

The code there tries to first read the gains and set the gains based on
them. The return value from the ioctl that sets the gains does not seem
to be checked in several code pathes, though. So it may actually fail
silently.

Do you get the same debug messages on 'reload chan_zap.so' ?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] chan_zap.c: Failed to read gains: Invalid argument

[Tzafrir Cohen]

On Fri, Jan 05, 2007 at 10:53:17AM +0200, Tzafrir Cohen wrote:
> On Fri, Jan 05, 2007 at 07:47:15AM -, Lee Archer wrote:
> > I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium TE110P
> > card in E1 mode.  I've recently noticed in my logs the following
> > 
> > Jan  5 01:27:11 VERBOSE[22490] logger.c:  [chan_zap.so]Jan  5 01:27:11
> > VERBOSE[22490] logger.c:  [chan_zap.so] => (Zapata Telephony w/PRI)
> > Jan  5 01:27:11 VERBOSE[22490] logger.c:   == Parsing
> > '/etc/asterisk/zapata.conf': Jan  5 01:27:11 VERBOSE[22490] logger.c:
> > == Parsing '/etc/asterisk/zapata.conf': Found
> > Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
> > argument
> > Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
> > argument
> 
> This is a debug message and not even a warning message. I'm not sure
> that this is something to worry about.

Sorry, my stupid misreading of the code. If this message was given,
ZT_SETGAINS will not be called.

-- 
   Tzafrir Cohen   
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+972-50-7952406   mailto:[EMAIL PROTECTED]   
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Re: [asterisk-users] Gentoo ebuild for 1.4?

[Benko]

On Wed, 3 Jan 2007 21:29:45 -
"Chris Bagnall" <[EMAIL PROTECTED]> wrote:

> Does anyone know if there's a maintained 1.4 ebuild for Gentoo? Even
> with the ~amd64 keyword, latest in the official Portage repository is
> 1.2.13.

hi!

I would not rely on portage for running asterisk on gentoo imho, it is
not sufficiently maintained currently...

regards
christian
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Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)

[Josué Conti]

Hi All,as good?
Steve Underwood will not work more with channel Unicall for the Asterisk?
It will be discontinued?

Best Regards

Josué

2007/1/4, Moises Silva <[EMAIL PROTECTED]>:


1.2, Zap and Unicall work fine
1.4 Only Zap working, Unicall is broken

On 1/4/07, Erick Perez <[EMAIL PROTECTED]> wrote:
> Question:
> So for people using E1 with R2 or PRI as signaling, what are my
> options in asterisk 1.4 and 1.2?
>
>
> On 1/4/07, Anton Krall <[EMAIL PROTECTED]> wrote:
> > Well Moises, if you do, please drop me a line and I will gladly test
it.
> >
> > I was mentioning digium because AFAIK, the guys at digium are in touch
with
> > the programmers and contributors so I thought maybe they would have an
> > insight on whats going to happen with unicall on 1.4, I mean, somebody
at
> > the source should know right? Many people still use unicall so I
thought
> > somebody would pick up the ball, maybe that's going to be you
hopefuly.
> >
> > Let me know how it goes.
> >
> >
> >
> >
> > |-Original Message-
> > |From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > |[EMAIL PROTECTED] On Behalf Of Moises Silva
> > |Sent: Wednesday, January 03, 2007 5:22 PM
> > |To: Asterisk Users Mailing List - Non-Commercial Discussion
> > |Subject: Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call
> > Announcement...)
> > |
> > |On 1/3/07, Anton Krall <[EMAIL PROTECTED]> wrote:
> > |> And probably wont be as Steve Underwood explained to me that he is
now
> > supporting
> > |openpbx and has stopped support for unicall on asterisk 1.4
> > |>
> > |> Can anybody at digium confirm? Is unicall going to be left out of
1.4?
> > |
> > |This has nothing to do with Digium, it has to do with anybody wanting
> > |to code the version for 1.4, AFAIK Steve never worked for Digium and
> > |Digium never distributed Unicall driver.
> > |
> > |Porting Unicall to 1.4 is in my TODO since 1 month ago, may be this
> > |month I will have the time to give a look at the code and try to make
> > |it work on 1.4, if somebody else cant do it before.
> > |
> > |Regards.
> > |
> > |--
> > |"Su nombre es GNU/Linux, no solamente Linux, mas info en
> > http://www.gnu.org";
> > |___
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> > |To UNSUBSCRIBE or update options visit:
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> >
> >
> >
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>
>
> --
> 
> Erick Perez
> Panama Sistemas
> Integradores de Telefonia IP y Soluciones Para Centros de Datos
> Panama, Republica de Panama
> Cel Panama. +(507) 6694-4780
> 
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[asterisk-users] asterisk 1.4.0 didn't compile chan_zap.so

[Ma Zhiyong]

HI, I use fc6 , the latest stable asterisk 1.4, zaptel 1.4 and libpri 1.4

after I installed zaptel and libpri. I can start zaptel. and my te410p card got 
green lamp. but when I continue to compile and install asterisk, I can't find 
chan_zap.so compiled. 

and in my asterisk cli. I can't 'help zap'. that got nothing.
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Re: [asterisk-users] How big a pipe can IAX2 go?

[Dinesh Nair]

On 01/05/07 06:18 Zoa said the following:


It used to be a problem to have very big iax2 trunks (e.g. > 100 channels).


anyone remember why this was so, and if a bug was opened on this for 1.2 ?

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[asterisk-users] Re: Best inexpensive home office router for VoIP (QoS with maybe PoE)

[Robbie Hughes]

The absolute best results I have had were with m0n0wall (m0n0.ch) which
worked perfectly for me to bounce voip calls over vpns with other traffic
and no user any the wiser. Second after that but with lots of plus points
for value come the draytek routers. A couple of years ago, their firmware
used to be terrible with the boxes crashing every 10 minutes if you tried to
use any of their feautures, but I now have clients on 6 of their 2900
routers, 4 of the 3300v routers and I'm trying out a 2910 (dual wan) for
myself at the moment and they all work perfectly.
The vpn functionality seems to be very robust as well.


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RE: [asterisk-users] POE draw on Aastra 480i

[Watkins, Bradley]

 
> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Allen Casteran
> Sent: Friday, January 05, 2007 12:35 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] POE draw on Aastra 480i
> 
> Anyone know what the POE draw is for the Aastra 480i phones?
> 
> We have switches that will do 15 watts on 12 ports but only 
> do 7.7 watts on all 24 ports. A Cisco 3560 switch will do 
> 15.6 watts on all 24 ports.
> 
> Just trying to find out if we need that much power.
> 
> Can't seem to find any info on the Aastra site.
> 
> Comments?


I can't for sure with the Aastras, but I know a Polycom 601 only draws
about 3.5-4 watts according to the command line of the switches we use
(Nortel 5520).  I can't imagine a 480i uses much more than that.

Regards,
- Brad
The contents of this e-mail are intended for the named addressee only. It 
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Re: [asterisk-users] IAX vs SIP trunks between Asterisk boxes

[Gordon Henderson]

On Thu, 4 Jan 2007, Noah Miller wrote:


Hi Damon -


Can anyone comment on the overhead added when a SIP call comes into one
asterisk box, is routed to another with IAX instead of SIP, and is then 
sent

to the UA from the second box with SIP?

DTMF passthrough issues?


I've got a client with sip phones on several different servers and
IAX links between the servers, so I guess that's pretty similar to
your setup.  I've never bothered to check for overhead since it was
never an issue (all servers are P4 2.8 ghz or Xeon 2.8 ghz, 1GB ram,
with never more than 3-4 calls going through any one of the IAX
links).  I can say that DTMF works fine in this setup.


I'm doing the same on 1GHz processors - CPU usage is virtually nil unless 
there's transcoding going on (about 4% per GSM transcode)


ADSL bandwidth is more of a concern for me in these applications )-:

Gordon
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[asterisk-users] How to build 1.4 with res_crypto.so

[Yann Massard]

Hello,

I am trying to build asterisk 1.4.0 (stable). The problem is "make" does
not build res_crypto.so

I have installed:

gcc
libc6
m4
openssl
zlibc
libkrb5-dev
libncurses5
libncurses5-dev
libssl-dev
zlib1g-dev


I know I will need res_crypto.so for iax, so...

Can anyone tell me how to do?

Thank you very much!

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RE: [asterisk-users] POE draw on Aastra 480i

[shadowym]

 
> 
> Anyone know what the POE draw is for the Aastra 480i phones?
> 
> We have switches that will do 15 watts on 12 ports but only do 7.7 
> watts on all 24 ports. A Cisco 3560 switch will do
> 15.6 watts on all 24 ports.
> 
> Just trying to find out if we need that much power.
> 
> Can't seem to find any info on the Aastra site.
> 
> Comments?

The power brick that comes with the 9133i and 480i is rated for 48v @ .13A
which comes out to 6.24W

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[asterisk-users] Invalid DivertingLegInformation2 component received 0x38

[Andreas Gaufer]

Hi,

I receive data from our ISDN PTP Line (by T-Com, Germany) that seems to be
not processed correctly by Asterisk 1.2.13-BRIstuffed-0.3.0-PRE-1w

pri debug:

1 Length (57) of 0x38 component is too long
1 !! Invalid DivertingLegInformation2 component received 0x38

This only happens on calls that are forwarded to us, the packet in
question seems to contain the RDNIS. I guess this needs some code in
pri_facility.c that I'm unable to write right now.

Should i file a bug-report abut this at http://bugs.digium.com?

Greetings & TIA

Andy
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[asterisk-users] fax transmission

[Vieri Di Paola]

I'm trying to use the txfax application based on
spandsp in Asterisk 1.2. It seems to be working but I
would need a way to reliably check whether the fax has
been completely transferred or not. I'm using a
mail2fax system (as with email2fax and .call files)
but I can't seem to get it working.
If I use "Application" and "Data" in the .call file,
there doesn't seem to be a way to check if the fax has
been correctly transmitted or not (your only option is
to cross your fingers).
If I use a dialplan Context in the .call file and from
there run a custom AGI and try to evaluate the
"result" then it simply doesn't work the way my logic
thinks it should. If someone has a sample AGI script
for txfax (and associated .call file content, if any)
I'd greatly appreciate it.

Ultimately, one may change the txfax source code to
suit one's needs and recompile it into Asterisk but
I'm sure there must be a quicker way.

Any difference with Asterisk 1.4.0?

Regards,

Vieri


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RE: [asterisk-users] Block some number outgoing from joust oneextention

[Mattias Andersson]

Hi!
Exactly what I needed.
It was the 209 part that I did not figure put.
Thanks!
//Mattias


At 03:53 2007-01-05, you wrote:

exten => _9070X./209,1,NoOP,SORRY CHARLIE
exten => _9070X./209,2,Congestion
This would block any call from 209 to 070X as 
long as 9 was your outside digit.


I use the NoOP to help me out with the CLI and debugging :)

Hope this helps

Mark



-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Mattias Andersson

Sent: Thursday, January 04, 2007 5:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Block some number 
outgoing from joust oneextention


Thanks!
I can´t rely figure out how to block for only one extension.
Eg. Extension 209 need to be blocked from making 
calls starting with 070  (eg. 9070).

Some clues did I get bout would it men a new form-internal-blocked dialplan?
Regards
Mattias



On 04/01/07, C F <[EMAIL PROTECTED]> wrote:
The easiest way is thru using contexts.

On 1/3/07, Mattias Andersson 
< [EMAIL PROTECTED]> wrote:

> Hi all!
> I am shore someone have writing about it bout I cant find it.
> I have a extension that I need to block from making expansive mobil calls.
> Everyone else should be aloud to do the calls.
>
> I am shore it is possible to be done sens I had a
> commercial asterisk based PBX that I did that on.
> However I have switch to Trixbox because I need
> some custom functions not supported by the commercial product.
> I would appreciate all help.
> Regards
> Mattias
>
>
>
>
>
>
> 
> Adress:
> Mattias Andersson
> Storskiftesvägen 6
> S-145 60 Norsborg
>
> Mobil: +46-70-799 44 41
> Email: [EMAIL PROTECTED]
> Skype: eskes1
>
>
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--
Mattias Andersson

Storskiftesvägen 6
145 60 Norsborg

m. +46-70-799 44 41
h. +46-8-641 38 97

Email: [EMAIL PROTECTED]


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Adress:
Mattias Andersson
Storskiftesvägen 6
S-145 60 Norsborg

Mobil: +46-70-799 44 41
Email: [EMAIL PROTECTED]
Skype: eskes1 



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Re: [asterisk-users] anyone using metermaid / parked call BLF?

[Dr. Michael J. Chudobiak]

I'm using 1.2.9.1, with the "metermaid" patches to show parking spot 
status on Snom BLF lights.


I see from http://www.asterisk.org/node/97 that the metermaid code has 
changed substantially since 1.2.9.1.


Is anyone successfully using the "new" metermaid functionality in 1.4.x?


> Did anyone get back to you on this?
> Did the Metermaid functionality get written into 1.4?
> I'd love to know if anyone ever replied to you privately.

Jeronimo,

No, I never heard back from anyone. I've cc'd this to asterisk-users - 
maybe someone is familiar with metermaid/1.4 now...


- Mike
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[asterisk-users] Asterisk and IM

[Hall, Eric M.]

Hello group
 I have been asked to get IM via the X-Ten softphone to work with
Asterisk. Anyone have any ideas? I have looked on google and other
places with no luck.
 
Our system is as followed
 
Linux CentOS 4.4
Asterisk 1.4.0-beta3
X-Lite v3.0 for Windows

Thanks!
Eric Hall


 
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[asterisk-users] integrating with Asterisk and OpenSER for Voicemail

[raviprakash sunkara]

Hi Users,

I'm Setting UP the Voicemails  by integrating with  Asterisk and   OpenSER,

After 32  sec or 6 ring, it  has to go the Voicemail server of Asterisk,

In openser.cfg   ... is not hiiting the Asterisk server
. ... any one  help me 


modparam("tm","fr_timer",6)
modparam("tm","fr_inv_timer",24)
modparam("tm","wt_timer",1)
#mrodparam("tm", "ruri_matching", 0)
#modparam("tm", "via1_matching", 0)
modparam("avpops","avp_url","mysql://root:[EMAIL PROTECTED]/openser")
modparam("avpops", "avp_table", "usr_preferences")
modparam("avpops","avp_aliases","inv=i:15")
...
route
{
..
if (loose_route()) {
  t_relay();
   exit;
   };
if(is_method("INVITE") && !has_totag())
   {
   xdbg("user [$ruri] has voicemail redirection
enabled\n");
   # backup R-URI
   avp_write("$ruri","$avp(inv)");
   setflag(2);
   };
..

route(1);
}
route[1] {
if(isflagset(2))
   {
t_on_failure("1");
   };
}
failure_route[1] {
   log("- \n");
   if (t_was_cancelled()) {
   xdbg("transaction was cancelled by UAC\n");
   return;
   }
   # restore initial uri
   avp_pushto("$ruri", "$avp(inv)");
   prefix("9");
   # route to Asterisk Media Server
   rewritehostport("192.168.2.75:5060");
   append_branch();
   t_relay("192.168.2.75:5060");
resetflag(2);
}


.




--
Thanks and Regards
Ravi Prakash Sunkara
[EMAIL PROTECTED]
M:+91 9985077535
O:+91 40 23114549
F:+91 40 40208727
[EMAIL PROTECTED]
www.hyperion-tech.com
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RE: [asterisk-users] no unicall on 1.4

[Anton Krall]

Couldn’t agree with you more Lee.

I think its very difficult for a software company to be able to stay focused
on developing the software while been profitable, that’s why many companies
turn to consulting services (Sun), other develop hardware (IBM and OS/2 :))

Digium has been doing a great job with asterisk but like you said, access to
the code without hassle was a plus for developers, now, well, they are
probably going to turn to some other open source project like openpbx and
probably Digium will end up hiring their own programmers and in time,
asterisk could stop been an open source project and become a commercial only
software.

One thing that’s interesting is how Digium turned to hardware like the TDM
cards for getting money but as of now, Sangoma for example offers a better
product (to me at least due to my experience with Digium hardware, timing
sources, HW compatibility, etc.) so let me ask the awkward question: what is
Digium doing 100% right? HW? No, software, used to, but maybe not now, so?

Don’t get me wrong, I love asterisk and will stick with 1.2 until something
comes along (openpbx goes stable, etc.) but come on guys at Digium, focus...
open source software was about making something by the community for the
community, not getting XXX million USD in VC while losing your best
programmers, the community and the R2MFC market :)
 



|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Lee Howard
|Sent: Thursday, January 04, 2007 10:14 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [asterisk-users] no unicall on 1.4
|
|Anton Krall wrote:
|
|>This is exactly one of the things that Steve and I discussed a bit ago...
|>when did asterisk turn from an open source project with very good
developers
|>into a business that only focuses in $$$?
|>
|>
|
|Well, I think that there can be no doubt that there still are some very
|good developers working on Asterisk, but yes, I do understand what
|you're saying, and I think that we're not the only ones that have
|noticed it. In particular I've noticed how the disclaimer requirement is
|a sore spot, and as well how impossibly difficult it is for Digium
|competitors to get their patches applied to the code base:
|
|http://bugs.digium.com/view.php?id=7742
|
|>That’s why openpbx was born I guess
|>
|>
|
|In part, yes. I think that some of these things are like lead weights to
|the Asterisk development process - I think that Steve Underwood
|appreciated the unfettered CVS commit access to the OpenPBX repository.
|That's a once-in-a-lifetime opportunity that Asterisk may have been able
|to have, itself, possibly. I see Steve's participation in OpenPBX as a
|big selling point (i.e. real T.38 gatewaying and actual spandsp
|integration). However, there's a lot of momentum behind Asterisk, and
|that's compensated somewhat for its lead weights up until now, and
|OpenPBX can't seem to get a public release out.
|
|At Cluecon last year in Chicago anthm told the conference how it was his
|belief that it would be better to start from scratch than to fix up all
|of the problems with Asterisk like OpenPBX is attempting - and thus we
|have FreeSWITCH.
|
|So there are lots of possibilities out there, and I can only think that
|the lead weights in the Asterisk development process will eventually
|lead to more issues than with chan_unicall.
|
|>For example, samba is still free, and people are making a profit from it
by
|>giving out consulting services for deploying samba.. that is a good
working
|>scenario asterisk used to be the same can you spell greedy :)?
|>
|>
|
|Well, when you sell consulting services for deploying Samba your
|business focus is still on the software. If they were selling
|Samba-related hardware or were heavily involved in selling Samba-related
|things like books and tee-shirts, etc., instead of actually working the
|software itself... well, then I think you'd see the same kinds of
|problems that you're frustrated with now. It's all too easy for that
|business activity to become a conflict of interest when it's not
|directly related to the user-experienced software itself.
|
|Lee.
|
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BEGIN:VCARD
VERSION:2.1
X-MS-SIGNATURE:YES
N;LANGUAGE=en-us:Krall;Anton
FN:Anton Krall
ORG:Intruder Consulting
TITLE:A Division of IntruderEnterprises S.A. de C.V.
TEL;WORK;VOICE:+52 (55) 5781-5112 x 201
TEL;WORK;VOICE:+52 (55) 5985-2430 x 201
X-MS-OL-DEFAULT-POSTAL-ADDRESS:0
URL;WORK:http://www.intruder.com.mx
EMAIL;PREF;INTERNET:[EMAIL PROTECTED]
PHOTO;TYPE=JPEG;ENCODING=BASE64:
 /9j/4AAQSkZJRgABAQEAYABgAAD/2wBDAAYEBQYFBAYGBQYHBwYIChAKCgkJChQODwwQFxQY
 GBcUFhYaHSUfGhsjHBYWICwgIyYnKSopGR8tMC0oMCUoKSj/2wBDAQcHBwoIChMKChMoGhYa
 KCgoKCgoKCgoKCgoKCgo

RE: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)

[Anton Krall]

Hi Josue, as of today at least, Steve Underwood is focusing his efforts into
making unicall be the basis for openpbx so will not be devoting more time
into unicall and asterisk.

This could change maybe but that’s what he told me a few days ago.




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josué Conti
Sent: Friday, January 05, 2007 3:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call
Announcement...)

Hi All,as good?
Steve Underwood will not work more with channel Unicall for the Asterisk?
It will be discontinued?
 
Best Regards
 
Josué
 
2007/1/4, Moises Silva <[EMAIL PROTECTED]>: 
1.2, Zap and Unicall work fine
1.4 Only Zap working, Unicall is broken

On 1/4/07, Erick Perez < [EMAIL PROTECTED]> wrote:
> Question:
> So for people using E1 with R2 or PRI as signaling, what are my
> options in asterisk 1.4 and 1.2?
>
>
> On 1/4/07, Anton Krall < [EMAIL PROTECTED]> wrote:
> > Well Moises, if you do, please drop me a line and I will gladly test it.
> >
> > I was mentioning digium because AFAIK, the guys at digium are in touch
with 
> > the programmers and contributors so I thought maybe they would have an
> > insight on whats going to happen with unicall on 1.4, I mean, somebody
at
> > the source should know right? Many people still use unicall so I thought

> > somebody would pick up the ball, maybe that's going to be you hopefuly.
> >
> > Let me know how it goes.
> >
> >
> >
> >
> > |-Original Message- 
> > |From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > |bounces@ lists.digium.com] On Behalf Of Moises Silva
> > |Sent: Wednesday, January 03, 2007 5:22 PM
> > |To: Asterisk Users Mailing List - Non-Commercial Discussion
> > |Subject: Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call
> > Announcement...)
> > |
> > |On 1/3/07, Anton Krall <[EMAIL PROTECTED]> wrote:
> > |> And probably wont be as Steve Underwood explained to me that he is
now 
> > supporting
> > |openpbx and has stopped support for unicall on asterisk 1.4
> > |>
> > |> Can anybody at digium confirm? Is unicall going to be left out of
1.4?
> > | 
> > |This has nothing to do with Digium, it has to do with anybody wanting
> > |to code the version for 1.4, AFAIK Steve never worked for Digium and
> > |Digium never distributed Unicall driver. 
> > |
> > |Porting Unicall to 1.4 is in my TODO since 1 month ago, may be this
> > |month I will have the time to give a look at the code and try to make
> > |it work on 1.4, if somebody else cant do it before. 
> > |
> > |Regards.
> > |
> > |--
> > |"Su nombre es GNU/Linux, no solamente Linux, mas info en
> > http://www.gnu.org";
> > |___
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> > |
> > |asterisk-users mailing list
> > |To UNSUBSCRIBE or update options visit:
> > |   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> 
> --
> 
> Erick Perez
> Panama Sistemas
> Integradores de Telefonia IP y Soluciones Para Centros de Datos
> Panama, Republica de Panama 
> Cel Panama. +(507) 6694-4780
> 
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--
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Re: [asterisk-users] Which is GUI to edit Asterisk IVR logic

[Lee Jenkins]

Olivier wrote:

Hi,

For a 20 users prospective customer, I'm wondering if any GUI would 
allow and end user to edit an Asterisk IVR tree ?


For instance, I'm looking for something allowing to edit interactions like :
"wait up to 20 seconds and say this "to reach sales department, type 1, 
to reach tech support type 2" message"




Trixbox will do that.  Easy.  So will DialplanPro, although Trixbox is 
much more robust right now (and free).  Check them out 
http://www.trixbox.org/.



--

Warm Regards,

Lee

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[asterisk-users] radius

[Khaled]

Please can you  provided me by a radius module name for asterisk,or how to
authorize user and get cdr from radius server.

Regards

 




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Re: [asterisk-users] POE draw on Aastra 480i

[Drew Gibson]

Allen Casteran wrote:


Anyone know what the POE draw is for the Aastra 480i phones?

We have switches that will do 15 watts on 12 ports but only do 7.7 
watts on all 24 ports. A Cisco 3560 switch will do 15.6 watts on all 
24 ports.


Just trying to find out if we need that much power.


According to Aastra tech support, 5 watts (peak) per 480i.
We are testing five phones running on a Linksys SRW208P that will only 
support full 15W on up to 4 of 8 ports. I can power up the switch while 
all phones are connected without any issues.

I would expect your lower power switch will provide ample power.

regards,

Drew

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Re: [asterisk-users] Which g729 module for HP DL 360 G3 (Xeon CPU's)?

[Kevin P. Fleming]

Eric Bishop wrote:
> I am running a HP DL360 G3 ans want to know the optimal g729 module for
> it. There don't seem to be any optimised for Xeon's. I am currently
> using i686, but is there a better one to match my Xeon CPU's?

Xeon is not an architecture, it's a brand name. Various Xeons have been
'prescott', 'nocona', 'pentium4m' and others, and there are newer
architectures that are not yet available as optimization targets in GCC.

The best advice we can give you is to just try the various modules that
seem appropriate for your system and determine which one gives you the
best performance by testing with 'show translation recalc 60' on an idle
system.
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Re: [asterisk-users] over 200 queues, anyone?

[Lenz]

I think we are going to do it if we get big problems with those many  
queues. From what I'm seeing, the biggest problems seem to be related to  
agents, so maybe we can have a try at using straight terminals instead of  
agents.

l.

On Fri, 05 Jan 2007 01:14:08 +0100, Leo Ann Boon <[EMAIL PROTECTED]> wrote:



Why don't you 'invert' the problem? Group the agents into fixed groups  
and put each group in a queue by itself. Each tenant will be assigned a  
group queue. If you have 30 agent in groups of 5, you only need 6 queues  
to handle 200 tenants. Even if you put each agent in a group by herself,  
you're still looking at 30 queues as opposed to 200 queues.


Leo




--
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http://queuemetrics.loway.it
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Re: [asterisk-users] no unicall on 1.4

[Moises Silva]

On 1/5/07, Anton Krall <[EMAIL PROTECTED]> wrote:

when did asterisk turn from an open source project with very good developers
nto a business that only focuses in $$$?

They are not mutually exclusive.


That's why openpbx was born I guess

I dont think so. I think is more because of technical disagreements.


For example, samba is still free, and people are making a profit from it by

When does Asterisk stoped being free, sorry but I missed something? :)


In any case, I (and maybe some other folks) would definitely pay some $$ to
Steve is he would consider supporting unicall for 1.4... I've always
believed that if you make money with something, why not give some to the
good programmers that made it happen...

Of course. What many buissiness people dont get is that programmers
have to eat ;)
And companies like Intruder and Office Connect are selling services
based on software created by people for free, the least think those
companies can do is support the programmers with money, the same thing
you are getting, right?



probably Digium will end up hiring their own programmers...

There are several programmers that work for Digium, but that does not mean
the project stops being open source ( http://en.wikipedia.org/wiki/Open_source )


what is Digium doing 100% right? HW? No, software, used to, but maybe not now, 
so?

Sangoma has a much bigger background in electronics, give Digium a
break, they keep working hard on software, and I dont think they are
going to quit, soon or later the software and hardware will get
better.


come on guys at Digium, focus...
open source software was about making something by the community for the
community, not getting XXX million USD in VC while losing your best
programmers, the community and the R2MFC market :)

Actually you need to read a little more about the open source term,
and dont use it loosely.

I think critics are good, eventually will push Digium to do better
software, but Digium is still a company that needs to make money:
money != evil.

At the end, is open source/freesoftware, if you dont like it, nobody
is stopping you from change it.

Kind Regards

--
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
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[asterisk-users] idle SIP channels problem

[O . Kamal]

I have 2 asterisk servers connected together on internet, when placing one
or two calls, things goes fine, but when placing more calls, i am getting
the below messages on the far end:
Jan  5 17:25:00 ERROR[2679] chan_sip.c: Call from peer 'switch' rejected due
to usage limit of 16
Jan  5 17:25:00 NOTICE[2679] chan_sip.c: Failed to place call for user , too
many calls

I am seeing a lot of idle connection when doing "sip show channels".
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Re: [asterisk-users] Which is GUI to edit Asterisk IVR logic

[Olivier]

By Trixbox, do you mean FreePBX (formely AMP) ?
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Re: [asterisk-users] over 200 queues, anyone?

[BJ Weschke]

On 1/5/07, Lenz <[EMAIL PROTECTED]> wrote:


I think we are going to do it if we get big problems with those many
queues. From what I'm seeing, the biggest problems seem to be related to
agents, so maybe we can have a try at using straight terminals instead of
agents.
l.



Being somewhat familiar with the innards of app_queue, I wouldn't
personally want to set any client of mine up with 200 queues on one
instance of app_queue.

Like another list member mentioned earlier, I think the risk for
deadlock is probably too great. You start adding queue weights into
the mix and I shiver to think of what might happen.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [asterisk-users] Gentoo ebuild for 1.4?

[Sune Kloppenborg Jeppesen]

On Friday 05 January 2007 10:31, Benko wrote:
> On Wed, 3 Jan 2007 21:29:45 -
>
> "Chris Bagnall" <[EMAIL PROTECTED]> wrote:
> > Does anyone know if there's a maintained 1.4 ebuild for Gentoo? Even
> > with the ~amd64 keyword, latest in the official Portage repository is
> > 1.2.13.
Short answer not at the moment.

Long answer is that there is an experimental ebuild in the voip overlay.

emerge layman
layman -a voip

Fix up keywords and emerge. It compiles fine with my useflags, though some 
functionality is still missing (ie. Bristuff). Given some time I'm sure it 
will enter Portage and be maintained (at least security wise).

More info here:

https://bugs.gentoo.org/show_bug.cgi?id=159013

HTH

-- 
Sune Kloppenborg Jeppesen (Jaervosz)


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[asterisk-users] addons 1.4 and cdr_addon_mysql not installed !

[Luca Lafranchi Lists]

Hi,

I have installed asterisk on Ubuntu 6.06 server CD

All required packages has been installed and upgraded

When start "sudo make menuselect" from addons, I can't select all addons
that require mysqlclient (app_addon_sql_mysql, cdr_addon_mysql,
res_config_mysql).

 

If I run "apt-cache search mysqlclient", I find the following installed
packages:

libmysqlclient15-dev - mysql database development files

libmysqlclient15off - mysql database client library

 

with "apt-cache search mysql-client":

mysql-client - mysql database client (current version)

mysql-client-5.0 - mysql database client binaries

 

Any idea ?

 

Thanks

 

Luca

 

 

 

 

 

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Re: [asterisk-users] Asterisk Core Dump in app_queue - Anyone seen?

[BJ Weschke]

On 1/4/07, Douglas Garstang <[EMAIL PROTECTED]> wrote:

Richard,

We have underscores all over the place in our config files, including others in 
queues.conf. I don't think that's the murder weapon.

I think, in general, queues are one of Asterisks biggest features, and also one 
of it's shakiest. The reload, which is run from a script, caused a reload on 3 
servers that are supposed to be redundant, and each crapped it's pants in a 
slightly different manner. The first stopped processing all queue calls (ie 
calls would lockup), the second core dumped, and the third seemed ok until you 
did another 'reload app_queue.so' where it would tell you that the previous 
reload was not finished yet.

Someone made a post yesterday about doing 200 queues on Asterisk. I don't envy 
what he is about to endure.



Doug -

There was some bug fixing done on app_queue post 1.2.9.1 to try to
accomodate some possibly shaky memory management on linked lists that
occurred during a reload. You may want to look at upgrading to latest
1.2.X or backporting those changes and see if the issue still exists.

BJ



--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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[asterisk-users] how to transfer calls when analog phone has no transfer button

[Erick Perez]

When you have a bunch of analog phones that you want to connect to
asterisk, but those analog phones have no transfer button, what are
the options to allow the phones to transfer a call?

--

Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780

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RE: [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument

[Lee Archer]

So anyone else any ideas? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: 05 January 2007 09:30
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] chan_zap.c: Failed to read gains:
Invalidargument

On Fri, Jan 05, 2007 at 10:53:17AM +0200, Tzafrir Cohen wrote:
> On Fri, Jan 05, 2007 at 07:47:15AM -, Lee Archer wrote:
> > I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium 
> > TE110P card in E1 mode.  I've recently noticed in my logs the 
> > following
> > 
> > Jan  5 01:27:11 VERBOSE[22490] logger.c:  [chan_zap.so]Jan  5 
> > 01:27:11 VERBOSE[22490] logger.c:  [chan_zap.so] => (Zapata
Telephony w/PRI)
> > Jan  5 01:27:11 VERBOSE[22490] logger.c:   == Parsing
> > '/etc/asterisk/zapata.conf': Jan  5 01:27:11 VERBOSE[22490]
logger.c:
> > == Parsing '/etc/asterisk/zapata.conf': Found Jan  5 01:27:11 
> > DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan

> > 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid 
> > argument
> 
> This is a debug message and not even a warning message. I'm not sure 
> that this is something to worry about.

Sorry, my stupid misreading of the code. If this message was given,
ZT_SETGAINS will not be called.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Re: POE draw on Aastra 480i

[Allen Casteran]

Drew Gibson wrote:

Allen Casteran wrote:


Anyone know what the POE draw is for the Aastra 480i phones?

We have switches that will do 15 watts on 12 ports but only do 7.7 
watts on all 24 ports. A Cisco 3560 switch will do 15.6 watts on all 
24 ports.


Just trying to find out if we need that much power.


According to Aastra tech support, 5 watts (peak) per 480i.
We are testing five phones running on a Linksys SRW208P that will only 
support full 15W on up to 4 of 8 ports. I can power up the switch while 
all phones are connected without any issues.

I would expect your lower power switch will provide ample power.



Thanks to Drew and everyone else that responded. Your assistance is 
greatly appreciated.


Allen

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RE: [asterisk-users] no unicall on 1.4

[Anton Krall]

I think you are misunderstanding several points here Moises.

I do give Digium a break like you said, that’s why you have options, you can
use digium cards or sangoma cards, it's up to you, I use digium cards from
time to time because I like to support digium in what they are doing.

But from the programmers perspective, if some open source software gets too
controlled by a company, then the synergy of open source is kind of broken,
all projects have rules and what I meant is that Digium right now is too
controlling and that’s why some programmers have decided to leave asterisk
and pursue other projects in which they fell more comfortable and can
provide code in an easier manner.

What I really have never liked is cases like what happened to unicall, there
was no news or announcements, simply , in 1.4 it's not there anymore :) and
you had to find out for yourself so, in one hand we have a lot of code
control and on the other a lack of communication from programmers, to digium
(which controls the code) to us...  this also applies for documentation and
changes, I know there is a change file but to be honest, sometimes it's not
clear enough and you have to go into the code and take a look at what was
changed... but this I can live with, after all, like you said, it is open
source..  

And I just want to say this again, I (and probably some others) am willing
to pay some $$ to Steve Underwood if he would consider porting unicall to
1.4 Steve, please come back! :)
 



|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Moises Silva
|Sent: Friday, January 05, 2007 9:41 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [asterisk-users] no unicall on 1.4
|
|On 1/5/07, Anton Krall <[EMAIL PROTECTED]> wrote:
|> when did asterisk turn from an open source project with very good
developers
|> nto a business that only focuses in $$$?
|They are not mutually exclusive.
|
|> That's why openpbx was born I guess
|I dont think so. I think is more because of technical disagreements.
|
|> For example, samba is still free, and people are making a profit from it
by
|When does Asterisk stoped being free, sorry but I missed something? :)
|
|> In any case, I (and maybe some other folks) would definitely pay some $$
to
|> Steve is he would consider supporting unicall for 1.4... I've always
|> believed that if you make money with something, why not give some to the
|> good programmers that made it happen...
|Of course. What many buissiness people dont get is that programmers
|have to eat ;)
|And companies like Intruder and Office Connect are selling services
|based on software created by people for free, the least think those
|companies can do is support the programmers with money, the same thing
|you are getting, right?
|
|
|> probably Digium will end up hiring their own programmers...
|There are several programmers that work for Digium, but that does not mean
|the project stops being open source (
http://en.wikipedia.org/wiki/Open_source )
|
|> what is Digium doing 100% right? HW? No, software, used to, but maybe not
now,
|so?
|Sangoma has a much bigger background in electronics, give Digium a
|break, they keep working hard on software, and I dont think they are
|going to quit, soon or later the software and hardware will get
|better.
|
|> come on guys at Digium, focus...
|> open source software was about making something by the community for the
|> community, not getting XXX million USD in VC while losing your best
|> programmers, the community and the R2MFC market :)
|Actually you need to read a little more about the open source term,
|and dont use it loosely.
|
|I think critics are good, eventually will push Digium to do better
|software, but Digium is still a company that needs to make money:
|money != evil.
|
|At the end, is open source/freesoftware, if you dont like it, nobody
|is stopping you from change it.
|
|Kind Regards
|
|--
|"Su nombre es GNU/Linux, no solamente Linux, mas info en
http://www.gnu.org";
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BEGIN:VCARD
VERSION:2.1
X-MS-SIGNATURE:YES
N;LANGUAGE=en-us:Krall;Anton
FN:Anton Krall
ORG:Intruder Consulting
TITLE:A Division of IntruderEnterprises S.A. de C.V.
TEL;WORK;VOICE:+52 (55) 5781-5112 x 201
TEL;WORK;VOICE:+52 (55) 5985-2430 x 201
X-MS-OL-DEFAULT-POSTAL-ADDRESS:0
URL;WORK:http://www.intruder.com.mx
EMAIL;PREF;INTERNET:[EMAIL PROTECTED]
PHOTO;TYPE=JPEG;ENCODING=BASE64:
 /9j/4AAQSkZJRgABAQEAYABgAAD/2wBDAAYEBQYFBAYGBQYHBwYIChAKCgkJChQODwwQFxQY
 GBcUFhYaHSUfGhsjHBYWICwgIyYnKSopGR8tMC0oMCUoKSj/2wBDAQcHBwoIChMKChMoGhYa
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 CAAXAEgDASIAAhEBAxEB/8QAHwAAAQUBAQEBAQEAAAECAwQFBgcICQoL/8QAtRAA
 AgEDAwIEAwUFBAQAAAF9AQIDA

Re: [asterisk-users] [Announce] Web-MeetMe 3.0.0 released

[Rob Fugina]

Can you set verbose and debug levels to 10 and run these commands:
*CLI> module unload app_cbmysql.so
*CLI> module load app_cbmysql.so



*CLI> core set verbose 10
Verbosity was 0 and is now 10
*CLI> module unload app_cbmysql.so
Unable to unload resource app_cbmysql.so
Command 'module unload app_cbmysql.so' failed.
*CLI> [Jan  5 11:09:04] WARNING[30610]: loader.c:465 ast_unload_resource:
Firm unload failed for app_cbmysql.so

So I added "noload => app_cbmysql.so" to modules.conf, and load manualy
after restarting asterisk...

*CLI> module load app_cbmysql.so
 == Parsing '/etc/asterisk/cbmysql.conf': Found
*CLI>

This *is* with verbosity set to 10, but this is all I was seeing before...

I suspect a config file issue, but a log of the module loading will

help peg down the problem.



Doesn't look to me like this will be much help, but what do I know...

Rob
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Re: [asterisk-users] how to transfer calls when analog phone has no transfer button

[cb]

On Jan 5, 2007, at 12:02 PM, Erick Perez wrote:


When you have a bunch of analog phones that you want to connect to
asterisk, but those analog phones have no transfer button, what are
the options to allow the phones to transfer a call?


You can setup a dial rule to do transfering based on keypresses. I  
can't give you specifics on how as I'm still using Trixbox which  
handles some of that stuff for you. But in my setup, I was able to  
turn on ## as a lead in for a transfer (ie: ##205 to transfer the  
call to extension 205). This worked fine for softphones.


When I got my Mediatrix box working correctly and connected actual  
analog phones, I planned to do the same, but the Mediatrix seems to  
override something and instead reads a hook flash as the transfer  
key, so on my actual analog phones, I just flash and then dial the  
extension and it transfers.


The ## is still active for my softphones, so I think my flash setup  
on the analogs is something being done by the Mediatrix box itself (I  
think it may be generating whatever a transfer button on an IP phone  
does)


-chris



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[asterisk-users] how to register nokia with Asterisk

[Biju]

hi,
 
i am using nokia e61 . we have an asterisk server 
and i want to use my nokia phone to register with asterisk server .
 
anybody can help me to do this.
 
thanks in advance
 
Biju
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[asterisk-users] ASterisk OOH323c

[Michel]

Hello,

I have asterisk 1.4 with ooh323c addons installed.  (As I am a newbie in 
voip world...my question might be idiot...! ;) Please forgive me!)


I succeed to make H323 call when ooh323c is configured as gateway 
(gatekeeper=DISABLE in ooh323.conf).

When I put  gatekeeper= ip_address, and add an account as follow :

[aaa]
type=friend
username=aaa
password=
host=dynamic
context=test
incominglimit=4
faststart=yes
h245Tunneling=yes

, my H323 softphone can't register. ("sent GRQ"..."gatekeeper not 
responding")



My questions are :


1/ Can ooh323c work as gatekeeper (if yes, even if it is installed on 
the same box as asterisk)?
2/ if yes, Do you know  tutorials for doing this? or Can anyone help me 
or give me some directions?


3/ or do you know tutorials about ooh323c?


Thanks you

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RE: [asterisk-users] HowTO configure voice T1

[Don Pobanz]

> T1s can use many different signalling types. You need to find out
which  
> one is running, what the line encoding is, etc. PRI vs T1 are not the
only  
> distinctions...
> 
> On 1/4/07, Mark Greene <[EMAIL PROTECTED]> wrote:  
>> Alright guys here is my question. What is do I need to set
switchtype, and  
>> signalling to in zapata for a voice T1. This is not a PRI. I cannot
say that  
>> enough. It is NOT, A, PRI. It is just a Voice T1 with 24 voice
channels.  
 
I believe switchtype applies only to ISDN and not to a channelized T1. 
 
>> There is not a D Channel. It runs from one office to another and USED

>> to plug into two opt. 11c but now one end is going to plug into an
asterisk  
>> box. 

 
Our local telco provides E & M with wink so we use 
signalling=em_w
in zapata.conf
 
As others have said, you will need to find out the details for your
setup. 
 
Don Pobanz
 
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RE: [asterisk-users] [Announce] Web-MeetMe 3.0.0 released

[Dan Austin]

> *CLI> core set verbose 10
>  Verbosity was 0 and is now 10
> *CLI> module unload app_cbmysql.so
>  Unable to unload resource app_cbmysql.so
>  Command 'module unload app_cbmysql.so' failed. 
> *CLI> [Jan  5 11:09:04] WARNING[30610]: loader.c:465 
> ast_unload_resource:Firm unload failed for app_cbmysql.so

> So I added "noload => app_cbmysql.so" to modules.conf, and load 
> manualy after restarting asterisk... 

> *CLI> module load app_cbmysql.so
>  == Parsing '/etc/asterisk/cbmysql.conf': Found
> *CLI> 

> This *is* with verbosity set to 10, but this is all I was 
> seeing before...

>> I suspect a config file issue, but a log of the module loading will
>> help peg down the problem.


> Doesn't look to me like this will be much help, but what do I know...
It actually helps quite a bit, along with me taking the time to
fully accept what version you are running (in one ear and out the
other problem)

Trunk has already moved on and code compatible with 1.4, may have
problems on it.  For a sanity check, I wiped out my test system
and rebuilt it with fresh components for 1.4 (libpri, zaptel, asterisk,
asterisk-addons), and I have no issues with unloading and re-loading
the module, and of course the app does what it claims and works as
intended.

So I can either ask that you try 1.4.0, or I will need to setup
a test against trunk.  I'd prefer to wait a bit before coding against
trunk, since it will break again, and likely before not too long.

Dan
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[asterisk-users] Re: [Users] integrating with Asterisk and OpenSER for Voicemail

[Steve Blair]

In the route[1] you do not show a t_relay. Is this just be a typo or is 
the t_relay actually missing from this route block? If it is missing how 
do you initiate the relay which you hope to end up in voicemail?


Daniel-Constantin Mierla wrote:

Hello,

watch the network traffic with "ngrep -d any -qt port 5060" on your 
sip server system and see if the INVITE is sent to asterisk. You can 
plug some xlog("__message__") in your configuration to see how the 
request is processed and if the INVITE hit the failure_route.


Cheers,
Daniel



On 01/05/07 14:57, raviprakash sunkara wrote:


Hi Users,

I'm Setting UP the Voicemails  by integrating with  Asterisk and   
OpenSER,


After 32  sec or 6 ring, it  has to go the Voicemail server of Asterisk,

In openser.cfg   ... is not hiiting the Asterisk server 
. ... any one  help me 



modparam("tm","fr_timer",6)
modparam("tm","fr_inv_timer",24)
modparam("tm","wt_timer",1)
#mrodparam("tm", "ruri_matching", 0)
#modparam("tm", "via1_matching", 0)
modparam("avpops","avp_url","mysql://root:[EMAIL PROTECTED]/openser")
modparam("avpops", "avp_table", "usr_preferences")
modparam("avpops","avp_aliases","inv=i:15")
 ...
route
{
..
 if (loose_route()) {
   t_relay();
exit;
};
if(is_method("INVITE") && !has_totag())
{
xdbg("user [$ruri] has voicemail redirection 
enabled\n");

# backup R-URI
avp_write("$ruri","$avp(inv)");
setflag(2);
};
..

route(1);
}
route[1] {
if(isflagset(2))
{
 t_on_failure("1");
};
}
 failure_route[1] {
log("- \n");
if (t_was_cancelled()) {
xdbg("transaction was cancelled by UAC\n");
return;
}
# restore initial uri
avp_pushto("$ruri", "$avp(inv)");
prefix("9");
# route to Asterisk Media Server
rewritehostport(" 192.168.2.75:5060 
");

append_branch();
t_relay("192.168.2.75:5060 ");
 resetflag(2);
}


.




--
Thanks and Regards
Ravi Prakash Sunkara
[EMAIL PROTECTED] 
M:+91 9985077535
O:+91 40 23114549
F:+91 40 40208727
[EMAIL PROTECTED] 
www.hyperion-tech.com 


___
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http://openser.org/cgi-bin/mailman/listinfo/users
  


___
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Re: [asterisk-users] no unicall on 1.4

[Lee Howard]

Anton Krall wrote:


after all, like you said, it is open
source..

|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Moises Silva

|At the end, is open source/freesoftware, if you dont like it, nobody
|is stopping you from change it.



One of the beauties of the open-source development method is certainly 
the ability for the user to modify the source code independently.


Realize, however, that the essense of that independent code modification 
is also termed as a code fork - particularly if the modified code is 
maintained long-term and especially if it is redistributed.


While I do believe that a light amount of user-experimentation-forking 
is normally to be expected and is healthy (because it generally will 
help in getting increased exposure and, in turn, flushing out bugs and 
such), if you hear the "Don't like it?  Fork." mantra too much or too 
frequently I believe that it is a sign of an unhealthy situation.  I 
hear it all of the time around here - way, way too much, I would think.


I think that a healthier scenario would be where the developers (who 
should actually be listening and care about what users feel, think, and 
want) would work with the users to find possible avenues to solutions... 
and if, indeed, a fork is the best way forward - then to do it in a way 
that is as collaborative as possible.


Lee.

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RE: [asterisk-users] how to transfer calls when analog phone has notransfer button

[Don Pobanz]

> When you have a bunch of analog phones that you want to 
> connect to asterisk, but those analog phones have no 
> transfer button, what are the options to allow the phones 
> to transfer a call?
 
Check out features.conf
You can specify key presses for things such as transfer. 

Don Pobanz
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Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)

[Josué Conti]

Hi Anton, thank's will be its reply.
It would be good for asking for to the Steve that did not abandon this
project (unicall and asterisk-1.4.x), in the versions 1.0.9 and 1.2.x when I
had problems and always I referred it, aiming at the improvement of channel.
It would be a great loss for all. We can make something to help it in the
project? I am the disposal to collaborate in that she will be necessary.

Best Regards

Josué

2007/1/5, Anton Krall <[EMAIL PROTECTED]>:


Hi Josue, as of today at least, Steve Underwood is focusing his efforts
into
making unicall be the basis for openpbx so will not be devoting more time
into unicall and asterisk.

This could change maybe but that's what he told me a few days ago.




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josué Conti
Sent: Friday, January 05, 2007 3:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call
Announcement...)

Hi All,as good?
Steve Underwood will not work more with channel Unicall for the Asterisk?
It will be discontinued?

Best Regards

Josué

2007/1/4, Moises Silva <[EMAIL PROTECTED]>:
1.2, Zap and Unicall work fine
1.4 Only Zap working, Unicall is broken

On 1/4/07, Erick Perez < [EMAIL PROTECTED]> wrote:
> Question:
> So for people using E1 with R2 or PRI as signaling, what are my
> options in asterisk 1.4 and 1.2?
>
>
> On 1/4/07, Anton Krall < [EMAIL PROTECTED]> wrote:
> > Well Moises, if you do, please drop me a line and I will gladly test
it.
> >
> > I was mentioning digium because AFAIK, the guys at digium are in touch
with
> > the programmers and contributors so I thought maybe they would have an
> > insight on whats going to happen with unicall on 1.4, I mean, somebody
at
> > the source should know right? Many people still use unicall so I
thought

> > somebody would pick up the ball, maybe that's going to be you
hopefuly.
> >
> > Let me know how it goes.
> >
> >
> >
> >
> > |-Original Message-
> > |From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > |bounces@ lists.digium.com] On Behalf Of Moises Silva
> > |Sent: Wednesday, January 03, 2007 5:22 PM
> > |To: Asterisk Users Mailing List - Non-Commercial Discussion
> > |Subject: Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call
> > Announcement...)
> > |
> > |On 1/3/07, Anton Krall <[EMAIL PROTECTED]> wrote:
> > |> And probably wont be as Steve Underwood explained to me that he is
now
> > supporting
> > |openpbx and has stopped support for unicall on asterisk 1.4
> > |>
> > |> Can anybody at digium confirm? Is unicall going to be left out of
1.4?
> > |
> > |This has nothing to do with Digium, it has to do with anybody wanting
> > |to code the version for 1.4, AFAIK Steve never worked for Digium and
> > |Digium never distributed Unicall driver.
> > |
> > |Porting Unicall to 1.4 is in my TODO since 1 month ago, may be this
> > |month I will have the time to give a look at the code and try to make
> > |it work on 1.4, if somebody else cant do it before.
> > |
> > |Regards.
> > |
> > |--
> > |"Su nombre es GNU/Linux, no solamente Linux, mas info en
> > http://www.gnu.org";
> > |___
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> > |asterisk-users mailing list
> > |To UNSUBSCRIBE or update options visit:
> > | http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
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> >
>
>
> --
> 
> Erick Perez
> Panama Sistemas
> Integradores de Telefonia IP y Soluciones Para Centros de Datos
> Panama, Republica de Panama
> Cel Panama. +(507) 6694-4780
> 
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--
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http://www.gnu.org";
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Re: [asterisk-users] addons 1.4 and cdr_addon_mysql not installed !

[Tzafrir Cohen]

On Fri, Jan 05, 2007 at 05:44:28PM +0100, Luca Lafranchi Lists wrote:
> Hi,
> 
> I have installed asterisk on Ubuntu 6.06 server CD
> 
> All required packages has been installed and upgraded
> 
> When start "sudo make menuselect" 

As a rule, "make" as a user, "make install" as root. No need for sudo
for anything other than 'make install' and such.

> from addons, I can't select all addons
> that require mysqlclient (app_addon_sql_mysql, cdr_addon_mysql,
> res_config_mysql).
> 
>  
> 
> If I run "apt-cache search mysqlclient", I find the following installed
> packages:
> 
> libmysqlclient15-dev - mysql database development files
> 
> libmysqlclient15off - mysql database client library
> 

You need the -dev one installed (recall that you're building a package.

The relevant build dependencies according to the current Etch package:
  libmysqlclient15-dev asterisk-dev 

(It also requires libsqlite3-dev, but res_sqlite3 has a broken build
process anyway and cannot use the system version of sqlite3)

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
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[asterisk-users] asterisk (FreePBX) and queues

[Felipe Neuwald]

Hi folks,

I'm using a fewestcall queue here, and I'm having the follow problem:

I have 3 static agents in my default queue:
2001
2002
2003

User 2001 and 2002 are logged in, but 2003 are logged out. When someone call
to my default queue, the queue try to ring 2003 (that isn't logged). There
is some way to the queue only ring logged users?

Here is my show queue:

zeus*CLI> show queue 100
100  has 0 calls (max unlimited) in 'fewestcalls' strategy (4s
holdtime), W:0, C:3, A:3, SL:0.0% within 0s
  Members:
 Local/[EMAIL PROTECTED]/n (Unknown) has taken no calls yet
 Local/[EMAIL PROTECTED]/n (Unknown) has taken 1 calls (last was 346
secs ago)
 Local/[EMAIL PROTECTED]/n (Unknown) has taken 2 calls (last was 195
secs ago)
  No Callers

Thank you,

Felipe.
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Re: [asterisk-users] how to transfer calls when analog phone has no transfer button

[Eric \"ManxPower\" Wieling]

Erick Perez wrote:

When you have a bunch of analog phones that you want to connect to
asterisk, but those analog phones have no transfer button, what are
the options to allow the phones to transfer a call?


Press the switch hook for 1 second.  In some parts of the world "FLASH" 
is called "RECALL".

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Re: [asterisk-users] POE draw on Aastra 480i

[Ira]

At 09:35 PM 1/4/2007, you wrote:

Just trying to find out if we need that much power.


I'd guess not, I can tell that my 480iCT is using power because I 
think I can feel warmth, but it sure doesn't feel like there's a 7 
watt light bulb inside.   It claims to need a 48V .13 amp wall wart 
if you power it that way,


Ira 


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[asterisk-users] SIP/TCP?

[Yuan LIU]

I'm still learning some of the basics.  Can someone explain in layman's 
terms what's the difficulty for Asterisk to support SIP/TCP (and even 
RTP/TCP)?



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Re: [asterisk-users] how to transfer calls when analog phone has notransfer button

[Erick Perez]

Don, I suppose that in order for this to work i need canreinvite=no, right?

On 1/5/07, Don Pobanz <[EMAIL PROTECTED]> wrote:

> When you have a bunch of analog phones that you want to
> connect to asterisk, but those analog phones have no
> transfer button, what are the options to allow the phones
> to transfer a call?

Check out features.conf
You can specify key presses for things such as transfer.

Don Pobanz
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--

Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780

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[asterisk-users] asterisk 1.4 debian packages

[Juraj Bednar]

Hello,

are there any (possibly experimental) asterisk debian packages (or at
least a debian/ directory to build our own)?

Previously I used to modify debian/ directory from earlier version,
but 1.4 changed build process, so this is not that easy.


   Thank you,

  Juraj.
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Re: [asterisk-users] Dimensioning a 50 sip phone installation

[Luki]

I was thinking of an HP DL140 with two 250gig sata disks and one
3.8Xeon CPU with 2gig RAM.


Should be plenty if not an overkill. One of our setups: 20 phones, 8
outgoing/incoming SIP trunks, MeetMe conferencing with ztdummy and no
Zap hardware. IVR/voice mail/MOH/Recordings/etc. Runs on a single
PIII-600, 256 MB RAM. CentOS 4.4 with a stock 2.6.9-42 kernel.
Asterisk 1.2.5, in production for 1.5+ years. CPU usage about 2% per
call. Quite reliable (hence not upgraded). This is a g711 only setup
with no transcoding.

--Luki
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Re: [asterisk-users] asterisk 1.4 debian packages

[Tzafrir Cohen]

On Fri, Jan 05, 2007 at 09:16:07PM +0100, Juraj Bednar wrote:
> Hello,
> 
> are there any (possibly experimental) asterisk debian packages (or at
> least a debian/ directory to build our own)?
> 
> Previously I used to modify debian/ directory from earlier version,
> but 1.4 changed build process, so this is not that easy.

The pkg-voip repository has experimental zaptel, libpri and asterisk
packages in the "experimental" branches of each of those packages. They
can be built with svn-buildpackage on etch/sid.

http://svn.debian.org/wsvn/pkg-voip/README?op=file

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
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Re: [asterisk-users] asterisk (FreePBX) and queues

[Alex Robar]

The problem is that you've setup 2003 as a static user. In FreePBX, static
users are ALWAYS in the queue, no matter what.

Take this guy out of the queue as a static agent, and have him login and
logout as he needs. (login via ##* and logout via ##**, where ## is the
number you've given your queue in FreePBX).

Alex

On 1/5/07, Felipe Neuwald <[EMAIL PROTECTED]> wrote:


Hi folks,

I'm using a fewestcall queue here, and I'm having the follow problem:

I have 3 static agents in my default queue:
2001
2002
2003

User 2001 and 2002 are logged in, but 2003 are logged out. When someone
call to my default queue, the queue try to ring 2003 (that isn't logged).
There is some way to the queue only ring logged users?

Here is my show queue:

zeus*CLI> show queue 100
100  has 0 calls (max unlimited) in 'fewestcalls' strategy (4s
holdtime), W:0, C:3, A:3, SL:0.0% within 0s
   Members:
  Local/[EMAIL PROTECTED] /n (Unknown) has taken no calls yet
  Local/[EMAIL PROTECTED]/n (Unknown) has taken 1 calls (last was 346
secs ago)
  Local/[EMAIL PROTECTED]/n (Unknown) has taken 2 calls (last was 195
secs ago)
   No Callers

Thank you,

Felipe.

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--
Alex Robar
[EMAIL PROTECTED]
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[asterisk-users] Has anybody voipstunt working?

[Arik Raffael Funke]

Hi,

I wrote a few days ago about my problem with calls via voipstunt 
stopping ringing after 5-6 rings, subj. "SIP Dial out timeout". Though 
if the remote station picks up before that, everything works flawlessly. 
I am not entirely sure when this phenomenon popped up, but I used the 
this configuration for quite some time before without any hassles. 
Nobody else seems to have this problem. So I guess it nevertheless has 
something to do with my configuration...


Could anybody who has voipstunt working to their satisfaction send me 
their configuration files via PM? [EMAIL PROTECTED]


Thanks a lot!

- Arik


PS: Just a thought: Has something about the sip.conf changed between 
v1.0 and 1.2 that might be related? I am essentially using my sip.conf 
from v1.0. Or in any other configuration file?


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[asterisk-users] Multiple users and a single extension

[Phil Finkler]

Hi all,

 

Quick question.  Is there a way to have multiple people have an
extension, say 900, to their polycom 501 SIP phones on one of the blue
buttons to where when a call comes in, I can have it simul-ring and
folks can pick up the line on their phone?  I'd like to set up a tech
support extension for our techs and have a voice mail box assigned to it
as well.  Right now I just have simulring set up and it rings all of the
techs, but they don't know if it's a tech support call or not because it
just looks like any other incoming call.

 

Thanks in advance,

Phil

 

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Re: [asterisk-users] Multiple users and a single extension

[Gavin Hamill]

On Friday 05 January 2007 21:06, Phil Finkler wrote:
> Hi all,
>
>
>
> Quick question.  Is there a way to have multiple people have an
> extension, say 900, to their polycom 501 SIP phones on one of the blue
> buttons to where when a call comes in, 
 
exten => 900,1,Set(CALLERID(name)=TechSupport)
exten => 900,2,Dial(SIP/101&SIP/102&SIP/103)

will most likely do what you want :)

gdh
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RE: [asterisk-users] how to transfer calls when analog phone hasnotransfer button

[Don Pobanz]

> Erick Perez
> 
> Don, I suppose that in order for this to work i need 
> canreinvite=no, right?
> 

No! It doesn't matter what you have for 'canreinvite' since 
'canreinvite' is a SIP attribute, not an analog phone attribute. 
For analog phones, Asterisk will always be in the call path. :-)

-- 
Don Pobanz
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[asterisk-users] Re: SIP/TCP?

[Steven]

UDP is a stream of packets with no layer 3 receipt acknowledgments.
Great for games and media.
If a packet is dropped or damaged, the receiver just skips it and uses the next 
packet.

TCP is a more controlled transmission of packets with receipt acknowledgements 
sent back after a certain number of packets.
If the ACK is not received by the sender, it resends the packets again.

If you are familiar with serial comm., you can think of TCP as flow control. 
(maybe)

This resending or waiting is bad for the timing of game/media data flow.


TCP is more stable because it is guaranteed delivery.
i.e.: You loose half of a web pages packets, the web server will resend the 
missing parts.

UDP is preferred when timing of the packets is more important than missed 
packets and where missed packet can be accommodated for.




-- 
-- 
Steven

http://www.glimasoutheast.org



"Howard Lowndes" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
>
>
> Yuan LIU wrote:
>> I'm still learning some of the basics.  Can someone explain in layman's 
>> terms what's the difficulty for Asterisk to support 
>> SIP/TCP (and even RTP/TCP)?
>
> Ina a word - ACK
>
>>
>>
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>>
>
> -- 
> Howard.
> LANNet Computing Associates - Your Linux people 
> When you want a computer system that works, just choose Linux;
> When you want a computer system that works, just, choose Microsoft.
> --
> Flatter government, not fatter government; abolish the Australian states.
>
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> 



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RE: [asterisk-users] Best inexpensive home office router for VoIP(QoS with maybe PoE)

[Mike]

You're quite right, I typed before thinking.  Upload is the problem anyways,
since it usually (in homes) uses much more limited bandwidth than
downloading does.
 
No answer to my question though: How do you people handle QoS without
relying on the phones to do that?  I'd like a box that can be purchased and
installed easily (Linksys type of product)
 
Mike  

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: Thursday, January 04, 2007 15:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best inexpensive home office router for
VoIP(QoS with maybe PoE)




Having QoS on your router is valuable to prevent some large download
from buggering your calls though. 


Isn't QoS only useful to prevent large uploads, as download rely on ISP
router prioritizing Voice over Data ?
Cheers



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Re: [asterisk-users] Best inexpensive home office router for VoIP(QoS with maybe PoE)

[Alex Robar]

Hi Mike,

The Linksys WRT54G can do QoS, and I've found it to be a great little
router... I install the DD-WRT open source firmware on mine for additional
features, but the stock firmware works well also.


Alex

On 1/5/07, Mike <[EMAIL PROTECTED]> wrote:


 You're quite right, I typed before thinking.  Upload is the problem
anyways, since it usually (in homes) uses much more limited bandwidth than
downloading does.

No answer to my question though: How do you people handle QoS without
relying on the phones to do that?  I'd like a box that can be purchased and
installed easily (Linksys type of product)

Mike

 --
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Olivier
*Sent:* Thursday, January 04, 2007 15:56
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Best inexpensive home office router for
VoIP(QoS with maybe PoE)


> Having QoS on your router is valuable to prevent some large download
> from buggering your calls though.
>

Isn't QoS only useful to prevent large uploads, as download rely on ISP
router prioritizing Voice over Data ?
Cheers



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--
Alex Robar
[EMAIL PROTECTED]
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RE: [asterisk-users] SIP/TCP?

[James R. Stevens]

TCP is a connection oriented protocol ..as others mentioned, it superiority 
comes because it knows when packets are dropped to resend them. It also has 
mechanisms for flow control etc.. SIP is a connection-less protocol. It uses 
'best effort' transmissions..if u want its delivery guaranteed you must 
encapsulate it.

-Original Message-
From: "Yuan LIU"<[EMAIL PROTECTED]>
Sent: 1/5/07 1:26:38 PM
To: "asterisk-users@lists.digium.com"
Subject: [asterisk-users] SIP/TCP?

I'm still learning some of the basics.  Can someone explain in layman's 
terms what's the difficulty for Asterisk to support SIP/TCP (and even 
RTP/TCP)?


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-- 
This message has been scanned for viruses and
dangerous content by Athens Hyperion Scanner, and is
believed to be clean.


-- 
This message has been scanned for viruses and
dangerous content by Athens Hyperion Scanner, and is
believed to be clean.

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[asterisk-users] Voicemail personalised greetings using DB/IMAP backend?

[Ray Jackson]

Hi all,

I am attempting to build a horizontally scalable Asterisk deployment and 
am getting very close to achieving that goal.  With Asterisk 1.4 I now 
have an IMAP backend for Voicemail messages which is great as users can 
check the same messages either through the voice portal or using 
Webmail.  However, I'm not sure the best way of dealing with 
personalised greetings such as a user's unavailable/busy message etc. 
Despite the IMAP backend these greetings appear to be stored on the 
local file system under /var/spool/asterisk/voicemail/default, which 
means if I build a farm of Asterisk servers - each will have it's own 
spool directory.  My aim is to have *nothing* stored locally at all...


If there a way of storing these greetings in a database table or using 
IMAP?  I saw the ODBC voicemail storage module, but I would prefer to 
stick with a REALTIME/IMAP backend?  If I mount the 
/var/spool/asterisk/voicemail directory remotely using a shared NFS 
mount on a NAS device will this work okay or lead to problems/race 
conditions etc.?  Any advice would be welcome!


Regards,
Ray
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[asterisk-users] Random "unknown" codec format IAX calls

[Max Ochoa]

   I seem to be having a problem that I have narrowed down to a 
disagreement on codec negotiation or codec setup of some kind in an IAX 
peering arrangement. Here's a non-ASCII art version of the setup:


DID origination provider
 via SIP/gsm
 to
Call routing asterisk server
 via IAX/gsm
 to
Client asterisk server
 via SIP/ulaw
 to
Polycom 501 UA

The problem that occurs sporadically (1/10 times) is the call will 
complete and stay active, but there is no audio. There is a channel open 
all the way to the phone, and the codec (gsm) is shown as the format for 
the call for the SIP channel and the IAX channel from the Call routing 
server to the Client asterisk server. However, the Client asterisk 
server shows the call format as "unknown" when a call is open that has 
no audio. The codec was originally forced to to gsm, then forced to 
ulaw, then set for any (allow=all) with the same results. Here's the 
output from the console on calls with these symptoms. IAX debug output 
looked the same for calls that had audio and those that did not, so I'll 
spare posting that.


Asterisk console (verbose):
===
 -- Accepting AUTHENTICATED call from 10.3.0.1 :
  > requested format = gsm,
  > requested prefs = (),
  > actual format = gsm,
  > host prefs = (ulaw|alaw|gsm),
  > priority = mine
   -- Executing Wait("IAX2/customer-8", "0") in new stack
   -- Executing Set("IAX2/customer-8", "_CONTEXTNAME=customer") in new 
stack

   -- Executing Set("IAX2/customer-8", "_VMEXTEN=100") in new stack
   -- Executing Set("IAX2/customer-8", 
"_VOIP_SERVER=customer.voip.domain.net 
") in new stack

   -- Executing Set("IAX2/customer-8", "TIMEOUT(digit)=5") in new stack
   -- Digit timeout set to 5
   -- Executing Set("IAX2/customer-8", "TIMEOUT(response)=6") in new stack
   -- Response timeout set to 6
   -- Executing Dial("IAX2/customer-8", "SIP/100-customer&SIP/101
-customer|25|tr") in new stack
   -- Called 100-customer
   -- Called 101-customer
   -- SIP/100-customer-081940e0 is ringing
   -- SIP/101-customer-081a1c70 is ringing
   -- SIP/100-customer-081940e0 answered IAX2/customer-8

customer*CLI> iax2 show channels
Channel   Peer UsernameID (Lo/Rem)  Seq 
(Tx/Rx)  Lag  Jitter  JitBuf  Format
IAX2/customer-8   10.3.0.1  customer
8/3  00014/00010  00079ms  -0001ms  ms  unknow

1 active IAX channel

customer*CLI> sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold 
Last Message   
10.0.0.103    100-custo  54eb074262d  
00102/0  ulaw  No   Tx: ACK
1 active SIP channel

===

There is a vtun IP tunnel between the Call routing asterisk server and 
the Client asterisk server (the 10.3.0.0/24 subnet.) The 10.0.0.0/24 
subnet is the client's LAN.


Any tips / ideas on what to try next are appreciated.

- Max

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[asterisk-users] DiD for less then $4

[CM Rahman]

Anybody selling DID flat rate for less then $4 with sip or iax incoming? Let me 
know

Thanks

CM

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RE: [asterisk-users] how to transfer calls when analog phone hasnotransfer button

[Doug Crompton]

Well it would be interesting to know what FXS device you are using to
connect the analog phones. I use an SPA-3000 fxo/fxs and with it you could
bypass Asterisk and connect the FXO to FXS or dial directly if it were so
configured, so reinvite would work but wwould probably not be desired but
that is not the question...

I am using the SPA-3000 as both an FXO (connection to telco) and FXS
(connection to my house analog phones) with Asterisk in between. I have
said this before on here but I will say it again. With the SPA-3000 you
cannot have analog phone feature keys, transfer etc. AND still be able to
use DTMF for control outside of the dialplan.

If you want feature key control then you would use rfc2833 DTMF, if you
want to be able to use DTMF incoming or outgoing for control then you must
use inband DTMF. It is either/or.

My choice was to use inband and not have features selected for the analog
phones. To often I would use these phines with banking or on incoming to
control voicemail functions so I wanted that capability.

In that case a hook flash works fine. If you have never done it just flash
the hook for a second (or use the flash key on the phone) and you will get
another dialtone. Then you can call another party, tell them you have a
call to transfer and hangup or click again and bring them in as a
conference.

Doug


On Fri, 5 Jan 2007, Don Pobanz wrote:

> > Erick Perez
> >
> > Don, I suppose that in order for this to work i need
> > canreinvite=no, right?
> >
>
> No! It doesn't matter what you have for 'canreinvite' since
> 'canreinvite' is a SIP attribute, not an analog phone attribute.
> For analog phones, Asterisk will always be in the call path. :-)
>
> --
> Don Pobanz
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"Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety."  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



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[asterisk-users] Re: Best inexpensive home office router for VoIP(QoS with maybe PoE)

[Allen Casteran]

Mike wrote:
You're quite right, I typed before thinking.  Upload is the problem 
anyways, since it usually (in homes) uses much more limited bandwidth 
than downloading does.
 
No answer to my question though: How do you people handle QoS without 
relying on the phones to do that?  I'd like a box that can be purchased 
and installed easily (Linksys type of product)
 


Mike,

Unless your ISP specifically supports QOS on your internet connection 
there is NO QOS beyond your router. Only within your network will the 
QOS be effective. Once the packets go through your router all control is 
lost. :)


This also means that you have little control over the priority of the 
traffic coming through the router's WAN port. The most you could do with 
QOS in this case is to limit outbound traffic from your PC if it would 
interfere with a voice call. The same is not true for the return (ie 
inbound) packets.


Allen.

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RE: [asterisk-users] Voicemail personalised greetings using DB/IMAPbackend?

[Douglas Garstang]

Does this model give you functioning mwi?

> -Original Message-
> From: Ray Jackson [mailto:[EMAIL PROTECTED]
> Sent: Friday, January 05, 2007 3:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Voicemail personalised greetings using
> DB/IMAPbackend?
> 
> 
> Hi all,
> 
> I am attempting to build a horizontally scalable Asterisk 
> deployment and 
> am getting very close to achieving that goal.  With Asterisk 
> 1.4 I now 
> have an IMAP backend for Voicemail messages which is great as 
> users can 
> check the same messages either through the voice portal or using 
> Webmail.  However, I'm not sure the best way of dealing with 
> personalised greetings such as a user's unavailable/busy message etc. 
> Despite the IMAP backend these greetings appear to be stored on the 
> local file system under /var/spool/asterisk/voicemail/default, which 
> means if I build a farm of Asterisk servers - each will have it's own 
> spool directory.  My aim is to have *nothing* stored locally at all...
> 
> If there a way of storing these greetings in a database table 
> or using 
> IMAP?  I saw the ODBC voicemail storage module, but I would prefer to 
> stick with a REALTIME/IMAP backend?  If I mount the 
> /var/spool/asterisk/voicemail directory remotely using a shared NFS 
> mount on a NAS device will this work okay or lead to problems/race 
> conditions etc.?  Any advice would be welcome!
> 
> Regards,
> Ray
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Re: [asterisk-users] Voicemail personalised greetings using DB/IMAP backend?

[Bryan M. Johns]

Ray,

Have you considered using a VM architecture?

Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com


On Jan 5, 2007, at 5:17 PM, Ray Jackson wrote:


Hi all,

I am attempting to build a horizontally scalable Asterisk  
deployment and am getting very close to achieving that goal.  With  
Asterisk 1.4 I now have an IMAP backend for Voicemail messages  
which is great as users can check the same messages either through  
the voice portal or using Webmail.  However, I'm not sure the best  
way of dealing with personalised greetings such as a user's  
unavailable/busy message etc. Despite the IMAP backend these  
greetings appear to be stored on the local file system under /var/ 
spool/asterisk/voicemail/default, which means if I build a farm of  
Asterisk servers - each will have it's own spool directory.  My aim  
is to have *nothing* stored locally at all...


If there a way of storing these greetings in a database table or  
using IMAP?  I saw the ODBC voicemail storage module, but I would  
prefer to stick with a REALTIME/IMAP backend?  If I mount the /var/ 
spool/asterisk/voicemail directory remotely using a shared NFS  
mount on a NAS device will this work okay or lead to problems/race  
conditions etc.?  Any advice would be welcome!


Regards,
Ray
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Re: [asterisk-users] SIP/TCP?

[Tzafrir Cohen]

On Fri, Jan 05, 2007 at 04:11:15PM -0600, James R. Stevens wrote:
> TCP is a connection oriented protocol ..as others mentioned, it 
> superiority comes because it knows when packets are dropped to resend 
> them. It also has mechanisms for flow control etc.. SIP is a 
> connection-less protocol. It uses 'best effort' transmissions..if u 
> want its delivery guaranteed you must encapsulate it.

Actually SIP is all about session management. 
The payload itself is RTP which indeed should be UDP for the reasons you
mention. SIP itself is much less timing sensitive.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Asterisk to a Huawei softX3000 problem has already been solved ！

[Guillermo Salas M.]

On Wed, 2006-12-13 at 21:51 -0300, Josué Conti wrote:
> Kevin, contributes with the list, somebody can have this problem and
> you it can help. The list is here for helping, but also we must
> contribute with it. :)
> Best Regards


I have the same problem.. any one know can I solve it?


Best resgards,


>  
> Josue
>  
> 2006/12/13, kevinho <[EMAIL PROTECTED]>: 
> 
> Asterisk to a Huawei softX3000 problem has already been
> solved ！
> 
> msn:[EMAIL PROTECTED]
> _
> Windows Live Safety Center 为您的计算机提供免费的安全扫描服
> 务。
> http://safety.live.com/site/ZH-CN/default.htm
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-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

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

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting

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Re: [asterisk-users] Voicemail personalised greetings using DB/IMAP backend?

[Kevin P. Fleming]

Ray Jackson wrote:
> If there a way of storing these greetings in a database table or using
> IMAP?

The current implementations of ODBC and IMAP for voicemail use it only
for voicemail, not for greetings. However, there is still work being
done by community members on both methods of storage, so I'd encourage
you to find the issues in the bug tracker where that is happening and
get involved.

For Asterisk 1.6 there is work underway to put in a generalized storage
subsystem that will abstract all this away from app_voicemail, so
storage of greetings in the database/IMAP server will be relatively easy.
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[asterisk-users] Call waiting notification

[Kevin Smith]

Hi everyone,

We are running Polycom 601's. I can't seem to find anything to say one 
way or another on this issue, so I figured I would ask. I have call 
waiting notification working on the phones when a user is on the phone. 
However, is it possible to see the notification on the screen or hear it 
on the line when it is in the "dial" status, IE I just pick the receiver 
off the hook and I am about to dial a number.


Kevin
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[asterisk-users] .call files no longer generating CDR files

[Michael Collins]

I've got a curious one:  all of a sudden my .call files and my manager
API 'Originate' actions are no longer producing a CSV file.  The call
still generates just fine, and Master.csv is updated.  However, I don't
get the usual CSV file in the form of xx.csv where xx=account
number.

I didn't make any changes that I'm aware of.  Is there something to
check?  I'm on 1.2.12, and this machine was working fine just a few days
ago...

Any insights would be much appreciated.

-MC
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Re: [asterisk-users] fax transmission

[Doug Lytle]

Vieri Di Paola wrote:

I'm trying to use the txfax application based on
spandsp in Asterisk 1.2. It seems to be working but I
would need a way to reliably check whether the fax has
been completely transferred or not. I'm using a
  



Then you'd want to use iaxmodem and HylaFAX+.

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] SIP/TCP?

[Yuan LIU]

From: "James R. Stevens" <[EMAIL PROTECTED]>

TCP is a connection oriented protocol ..as others mentioned, it superiority 
comes because it knows when packets are dropped to resend them. It also has 
mechanisms for flow control etc.. SIP is a connection-less protocol. It 
uses 'best effort' transmissions..if u want its delivery guaranteed you 
must encapsulate it.


So I take it that UDP is just a decision due to popular demand; timing 
(jitter) is a frequently cited factor to favour UDP.  Is there any technical 
difficulties in implementing SIP/TCP within Asterisk?


The reason I'm asking is that there are products that support both UDP and 
TCP.  And SIP/TCP, RTP/TCP have their own merits.


Granted, SIP is connectionless.  So is HTTP (well, for its original design 
anyway).  I notice that guaranteed delivery could be a good thing for SIP in 
many situations; there have also been advancements that make timing less an 
issue in RTP/TCP.


Is "switching to" SIP/TCP - RTP/TCP as simple as rewrap messages/streams, or 
is it more involved?


Yuan Liu


-Original Message-
From: "Yuan LIU"<[EMAIL PROTECTED]>
Sent: 1/5/07 1:26:38 PM
To: "asterisk-users@lists.digium.com"
Subject: [asterisk-users] SIP/TCP?

I'm still learning some of the basics.  Can someone explain in layman's
terms what's the difficulty for Asterisk to support SIP/TCP (and even
RTP/TCP)?



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Re: [asterisk-users] Voicemail personalised greetings using DB/IMAPbackend?

[Ray Jackson]

Hi Douglas,

Yes, MWI works fine as each Asterisk server looks after it's own set of 
registered users.  I am simply created a shared backend for voicemail 
storage (IMAP) and MySQL for voicemail configurations.  The missing 
piece is how to store personalised greetings in a shared backend.


Cheers,
Ray

Douglas Garstang wrote:


Does this model give you functioning mwi?

 


-Original Message-
From: Ray Jackson [mailto:[EMAIL PROTECTED]
Sent: Friday, January 05, 2007 3:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voicemail personalised greetings using
DB/IMAPbackend?


Hi all,

I am attempting to build a horizontally scalable Asterisk 
deployment and 
am getting very close to achieving that goal.  With Asterisk 
1.4 I now 
have an IMAP backend for Voicemail messages which is great as 
users can 
check the same messages either through the voice portal or using 
Webmail.  However, I'm not sure the best way of dealing with 
personalised greetings such as a user's unavailable/busy message etc. 
Despite the IMAP backend these greetings appear to be stored on the 
local file system under /var/spool/asterisk/voicemail/default, which 
means if I build a farm of Asterisk servers - each will have it's own 
spool directory.  My aim is to have *nothing* stored locally at all...


If there a way of storing these greetings in a database table 
or using 
IMAP?  I saw the ODBC voicemail storage module, but I would prefer to 
stick with a REALTIME/IMAP backend?  If I mount the 
/var/spool/asterisk/voicemail directory remotely using a shared NFS 
mount on a NAS device will this work okay or lead to problems/race 
conditions etc.?  Any advice would be welcome!


Regards,
Ray
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Re: [asterisk-users] Voicemail personalised greetings using DB/IMAP backend?

[Ray Jackson]

Hi Bryan,

I was trying to avoid creating an architecture dedicated to VM, but have 
Asterisk handle VM in a horizontally scalable way.  I understand there 
are some issues with MWI etc. if you separate out the VM from Asterisk? 
 Could you point me at any good examples of a VM architecture I could 
use as a reference?


Cheers,
Ray

Bryan M. Johns wrote:

Ray,

Have you considered using a VM architecture?

Bryan M. Johns
Partner
*Shelton | Johns Technology Group*
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
*http://www.sheltonjohns.com* 


On Jan 5, 2007, at 5:17 PM, Ray Jackson wrote:


Hi all,

I am attempting to build a horizontally scalable Asterisk deployment 
and am getting very close to achieving that goal.  With Asterisk 1.4 I 
now have an IMAP backend for Voicemail messages which is great as 
users can check the same messages either through the voice portal or 
using Webmail.  However, I'm not sure the best way of dealing with 
personalised greetings such as a user's unavailable/busy message etc. 
Despite the IMAP backend these greetings appear to be stored on the 
local file system under /var/spool/asterisk/voicemail/default, which 
means if I build a farm of Asterisk servers - each will have it's own 
spool directory.  My aim is to have *nothing* stored locally at all...


If there a way of storing these greetings in a database table or using 
IMAP?  I saw the ODBC voicemail storage module, but I would prefer to 
stick with a REALTIME/IMAP backend?  If I mount the 
/var/spool/asterisk/voicemail directory remotely using a shared NFS 
mount on a NAS device will this work okay or lead to problems/race 
conditions etc.?  Any advice would be welcome!


Regards,
Ray
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Re: [asterisk-users] Call waiting notification

[Kevin P. Fleming]

Kevin Smith wrote:
> We are running Polycom 601's. I can't seem to find anything to say one
> way or another on this issue, so I figured I would ask. I have call
> waiting notification working on the phones when a user is on the phone.
> However, is it possible to see the notification on the screen or hear it
> on the line when it is in the "dial" status, IE I just pick the receiver
> off the hook and I am about to dial a number.

Nope.
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Re: [asterisk-users] Voicemail personalised greetings using DB/IMAP backend?

[Ray Jackson]

Hi Kevin,

Thanks for your response.  That answers a few questions I had.  I am 
very happy to get involved in this area if I can help.  Using IMAP and 
REALTIME I have a really nice VM solution with MWI, Webmail access etc. 
and it scales horizontally - I just add a new server into the mix when 
necessary.  Until we get a generlized storage subsystem in place, I may 
look at a 'hack' to get the personalised greetings going... Do you think 
a shared NFS mount is risky for this?  Should I do an rsync periodically 
perhaps to keep greetings on all servers up to date with each other?


Thanks again.
Ray

Kevin P. Fleming wrote:

Ray Jackson wrote:


If there a way of storing these greetings in a database table or using
IMAP?



The current implementations of ODBC and IMAP for voicemail use it only
for voicemail, not for greetings. However, there is still work being
done by community members on both methods of storage, so I'd encourage
you to find the issues in the bug tracker where that is happening and
get involved.

For Asterisk 1.6 there is work underway to put in a generalized storage
subsystem that will abstract all this away from app_voicemail, so
storage of greetings in the database/IMAP server will be relatively easy.
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Re: [asterisk-users] Voicemail personalised greetings using DB/IMAP backend?

[Kevin P. Fleming]

Ray Jackson wrote:
> necessary.  Until we get a generlized storage subsystem in place, I may
> look at a 'hack' to get the personalised greetings going... Do you think
> a shared NFS mount is risky for this?  Should I do an rsync periodically
> perhaps to keep greetings on all servers up to date with each other?

NFS sharing is fine, app_voicemail works quite well over NFS.
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Re: [asterisk-users] Dimensioning a 50 sip phone installation

[Erick Perez]

what if I go with full g711-no transcoding?
remember that I will have an E1 coming in, so my usage can be up to 30
channels at once.
if that is an overkill machine config, and for obvious reasons I cant
use old hardware, what are your suggestions?

thanks,


On 1/5/07, Luki <[EMAIL PROTECTED]> wrote:

> I was thinking of an HP DL140 with two 250gig sata disks and one
> 3.8Xeon CPU with 2gig RAM.

Should be plenty if not an overkill. One of our setups: 20 phones, 8
outgoing/incoming SIP trunks, MeetMe conferencing with ztdummy and no
Zap hardware. IVR/voice mail/MOH/Recordings/etc. Runs on a single
PIII-600, 256 MB RAM. CentOS 4.4 with a stock 2.6.9-42 kernel.
Asterisk 1.2.5, in production for 1.5+ years. CPU usage about 2% per
call. Quite reliable (hence not upgraded). This is a g711 only setup
with no transcoding.

--Luki
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--

Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780

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Re: [asterisk-users] how to transfer calls when analog phone hasnotransfer button

[Erick Perez]

On 1/5/07, Doug Crompton <[EMAIL PROTECTED]> wrote:

Well it would be interesting to know what FXS device you are using to
connect the analog phones. I use an SPA-3000 fxo/fxs and with it you could
bypass Asterisk and connect the FXO to FXS or dial directly if it were so
configured, so reinvite would work but wwould probably not be desired but
that is not the question...


Right, I forgot to mention that.
Plain an simple analog phones will be connected to audiocodes
fxs-to-sip and then the audiocodes talk to asterisk.
im planning *not* to use transcoding and go full g711 ulaw on this one.



I am using the SPA-3000 as both an FXO (connection to telco) and FXS
(connection to my house analog phones) with Asterisk in between. I have
said this before on here but I will say it again. With the SPA-3000 you
cannot have analog phone feature keys, transfer etc. AND still be able to
use DTMF for control outside of the dialplan.

If you want feature key control then you would use rfc2833 DTMF, if you
want to be able to use DTMF incoming or outgoing for control then you must
use inband DTMF. It is either/or.

My choice was to use inband and not have features selected for the analog
phones. To often I would use these phines with banking or on incoming to
control voicemail functions so I wanted that capability.

In that case a hook flash works fine. If you have never done it just flash
the hook for a second (or use the flash key on the phone) and you will get
another dialtone. Then you can call another party, tell them you have a
call to transfer and hangup or click again and bring them in as a
conference.

Doug


On Fri, 5 Jan 2007, Don Pobanz wrote:

> > Erick Perez
> >
> > Don, I suppose that in order for this to work i need
> > canreinvite=no, right?
> >
>
> No! It doesn't matter what you have for 'canreinvite' since
> 'canreinvite' is a SIP attribute, not an analog phone attribute.
> For analog phones, Asterisk will always be in the call path. :-)
>
> --
> Don Pobanz
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"Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety."  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



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--

Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780

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[asterisk-users] RE: [SOLVED] .call files no longer generating CDR files

[Michael Collins]

> I've got a curious one:  all of a sudden my .call files and my manager
> API 'Originate' actions are no longer producing a CSV file.  The call
> still generates just fine, and Master.csv is updated.  However, I
don't
> get the usual CSV file in the form of xx.csv where xx=account
> number.
> 
> I didn't make any changes that I'm aware of.  Is there something to
> check?  I'm on 1.2.12, and this machine was working fine just a few
days
> ago...
> 
> Any insights would be much appreciated.
> 
> -MC

On a hunch, I rebuilt Asterisk and Asterisk-addons from the source and
everything started working again!  Big sigh of relief...

-MC
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Re: [asterisk-users] Voicemail personalised greetings using DB/IMAP backend?

[David Thomas]

On 1/5/07, Ray Jackson <[EMAIL PROTECTED]> wrote:

Hi Kevin,

Thanks for your response.  That answers a few questions I had.  I am
very happy to get involved in this area if I can help.  Using IMAP and
REALTIME I have a really nice VM solution with MWI, Webmail access etc.
and it scales horizontally - I just add a new server into the mix when
necessary.  Until we get a generlized storage subsystem in place, I may
look at a 'hack' to get the personalised greetings going... Do you think
a shared NFS mount is risky for this?  Should I do an rsync periodically
perhaps to keep greetings on all servers up to date with each other?


In the DUNDi * cluster we're designing phones can register with any of
our asterisk boxes. Actually sometimes phones are registered to
multiple boxes. I'm wondering if the new IMAP/MWI would have any
problems with this type setup. Any experiences here?

Regards,
Dave
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Re: [asterisk-users] Dimensioning a 50 sip phone installation

[Leo Ann Boon]

Erick Perez wrote:

what if I go with full g711-no transcoding?
remember that I will have an E1 coming in, so my usage can be up to 30
channels at once.
if that is an overkill machine config, and for obvious reasons I cant
use old hardware, what are your suggestions?
I would suggest you go for a box that has redundant PSU. Most 1U boxes 
can't support redundant PSUs.


IMHO, a 2U industrial PC with a single dual-core Pentium Dxxx 2.8GHz+ 
(or Xeon 3xxx) with hotswap RAID-1 HDD and PSU would be more than 
enough. I generally prefer 2U over 1U, because it's easier to cool and 
there's space to accommodate PCI cards of various sizes.


Leo

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Re: [asterisk-users] Re: SIP/TCP?

[George Pajari]

UDP is preferred for VoIP because by the time a dropped packet is 
detected, the retransmission request is sent to the originator, and the 
replacement packet arrives, it's too late (unless you are running a very 
large jitter buffer which introduces problems of its own).


Conversely if your network has sufficient bandwidth to ensure the 
retransmission happens fast enough, it has sufficient bandwidth to avoid 
dropped packets in the first place.


One of the problems is that some WAN switches/routers discriminate 
against UDP when they become congested because they reason dropping UDP 
packets will cause less of a problem (because they are not 
retransmitted) than dropping TCP packets (which will cause 
retransmissions and not alleviate the congestion).


--
George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)
Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)
Hosted IP PBX Services for SOHO & Small Businesses - www.ip-centrex.ca
 VoIP Service, Equipment, Systems, and Consulting - www.netvoice.ca

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Re: [asterisk-users] Voicemail personalised greetings using DB/IMAP backend?

[Ray Jackson]

David Thomas wrote:


In the DUNDi * cluster we're designing phones can register with any of
our asterisk boxes. Actually sometimes phones are registered to
multiple boxes. I'm wondering if the new IMAP/MWI would have any
problems with this type setup. Any experiences here?


I am using SRV records, RealTime SIP peers and 302 redirects (Transfer) 
and a bit of logic in the dial plans to scale our * cluster.  Phones may 
register against any * server and an inbound call will always find which 
server they are currently registered against.  It seems to work well for 
us.  I am still testing the IMAP voicemail backend, but MWI seems to 
work just fine with this setup... the * server the client was last 
registered against handles MWI normally.  Sorry, I don't use DUNDi but 
I'm sure an IMAP backend for your VM wouldn't have any impact on MWI.  I 
am still in a testing phase though, so my experience with IMAP/MWI is 
limited.


btw. How can a phone register to multiple boxes at the same time?  Do 
you mean you could have multiple registrations that have not yet 
expired?  We only ever care about the latest registration in our setup 
and ignore old/stale registrations...

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Re: [asterisk-users] Asterisk and IM

[Kenneth Padgett]

 I have been asked to get IM via the X-Ten softphone to work with Asterisk.
Anyone have any ideas? I have looked on google and other places with no
luck.

Our system is as followed

Linux CentOS 4.4
Asterisk 1.4.0-beta3
X-Lite v3.0 for Windows


If by IM, you mean the built-in Jabber stuff in v1.4... I am having
trouble with that and CentOS 4.4 myself, can't get the required libs
or some such non-sense.

-Kenneth
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Re: [asterisk-users] Best inexpensive home office router for VoIP (QoS with maybe PoE)

[Kenneth Padgett]

I'm looking for opinions on the "best value" router to use for home offices.
 It should work for a scenario in which there are 3 computers and 2 SIP
phones, handling QoS so that the phones always have higher priority traffic
than the PCs. (and not rely on the phones to do the QoS because some PCs may
not be connected to the phones).


I'm using a Linksys WRTSL54GS and 3rd party firmware with great
results! You won't find QoS features in the default Linksys firmware
though, so if you want something out of the box, this isn't much help.

My main reasons for picking it over the older WRT54G's where:

1) It was (still is?) available in retail stores, whereas the WRT54G's
that run Linux are generally only on ebay these days, they have to be
older versions.

2) it has 32mb of ram, and 8mb of flash.

3) it has USB which is nice to connect storage or network printer too.

If you go with it, make sure it's version 1, the K0 serial number
though. Version 2 has less flash and hasn't been tested. See:

http://wiki.openwrt.org/OpenWrtDocs/Hardware/Linksys/WRTSL54GS
http://wiki.openwrt.org/TableOfHardware

I started with Thibor's firmware (which is based on stock Linksys
code), so no special shell / command line knowledge needed:
http://www.thibor.co.uk/

Eventually moved on to OpenWRT for support of advanced stuff I'm
doing, such as dial on demand PPTP client VPNs to clients, multi-site
VPN with my friends, DNSmasq, QoS, etc.

-Kenneth
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Re: [asterisk-users] [Announce] Web-MeetMe 3.0.0 released

[Rob Fugina]

On 1/5/07, Dan Austin <[EMAIL PROTECTED]> wrote:


Trunk has already moved on and code compatible with 1.4, may have
problems on it.  For a sanity check, I wiped out my test system
and rebuilt it with fresh components for 1.4 (libpri, zaptel, asterisk,
asterisk-addons), and I have no issues with unloading and re-loading
the module, and of course the app does what it claims and works as
intended.

So I can either ask that you try 1.4.0, or I will need to setup
a test against trunk.  I'd prefer to wait a bit before coding against
trunk, since it will break again, and likely before not too long.



I guess I figured that trunk couldn't have gone far from 1.4 yet, so I'll
move to 1.4.  Nothing in particular on trunk I need.  Thanks for your time,
but sorry to have wasted it.

Rob
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Re: [asterisk-users] postgres and asterisk

[chester c young]

use a simple agi - php is easy to do.


--- "O.Kamal" <[EMAIL PROTECTED]> wrote:

> I just need to retrieve a value from a field in a postgres database,
> and
> playback this value when someone dial a specific extension.
> 
> On 1/4/07, Thomas Kenyon <[EMAIL PROTECTED]> wrote:
> >
> > O.Kamal wrote:
> > > I need to retrieve my asterisk to retrieve a values from
> postgresql, i
> > > am looking for some sort of application like *mysql*() app, I
> found one
> > > but it is only available on Suse, is there any way for doing
> this?
> > >
> > > Regards,
> > > O.Youssef
> > >
> > What do you need to do?
> > To get an SQL console with postgres you need to:
> >
> > psql -d  -U 
> >
> > ie:
> >
> > psql -d asterisk -U asterisk
> >
> > The location of psql is different depensing upon distribution but
> > usually it's in either /usr/bin/psql or /usr/local/pgsql/bin/psql.
> >
> > I'm not sure if this is what you want, if you want a pretty GUI
> > front-end then you could look at Pgadmin III (www.pgadmin.org)
> which
> > will run on Windows 2000/XP/2003 or unix/linux running X and
> requires
> > wxWindows and a pile of common libraries.
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Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
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Re: [asterisk-users] Block some number outgoing from joust oneextention

[Marco Mouta]

That's what i told you Mattias.

On 1/5/07, Mattias Andersson <[EMAIL PROTECTED]> wrote:


Hi!
Exactly what I needed.
It was the 209 part that I did not figure put.
Thanks!
//Mattias


At 03:53 2007-01-05, you wrote:
>exten => _9070X./209,1,NoOP,SORRY CHARLIE
>exten => _9070X./209,2,Congestion
>This would block any call from 209 to 070X as
>long as 9 was your outside digit.
>
>I use the NoOP to help me out with the CLI and debugging :)
>
>Hope this helps
>
>Mark
>
>
>
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of Mattias
Andersson
>Sent: Thursday, January 04, 2007 5:12 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [asterisk-users] Block some number
>outgoing from joust oneextention
>
>Thanks!
>I can´t rely figure out how to block for only one extension.
>Eg. Extension 209 need to be blocked from making
>calls starting with 070  (eg. 9070).
>Some clues did I get bout would it men a new form-internal-blocked
dialplan?
>Regards
>Mattias
>
>
>
>On 04/01/07, C F <[EMAIL PROTECTED]> wrote:
>The easiest way is thru using contexts.
>
>On 1/3/07, Mattias Andersson
>< [EMAIL PROTECTED]> wrote:
> > Hi all!
> > I am shore someone have writing about it bout I cant find it.
> > I have a extension that I need to block from making expansive mobil
calls.
> > Everyone else should be aloud to do the calls.
> >
> > I am shore it is possible to be done sens I had a
> > commercial asterisk based PBX that I did that on.
> > However I have switch to Trixbox because I need
> > some custom functions not supported by the commercial product.
> > I would appreciate all help.
> > Regards
> > Mattias
> >
> >
> >
> >
> >
> >
> > 
> > Adress:
> > Mattias Andersson
> > Storskiftesvägen 6
> > S-145 60 Norsborg
> >
> > Mobil: +46-70-799 44 41
> > Email: [EMAIL PROTECTED]
> > Skype: eskes1
> >
> >
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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>
>
>
>--
>Mattias Andersson
>
>Storskiftesvägen 6
>145 60 Norsborg
>
>m. +46-70-799 44 41
>h. +46-8-641 38 97
>
>Email: [EMAIL PROTECTED]
>
>
>--
>No virus found in this outgoing message.
>Checked by AVG Free Edition.
>Version: 7.5.432 / Virus Database: 268.16.4/615
>- Release Date: 1/3/2007 1:34 PM
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Adress:
Mattias Andersson
Storskiftesvägen 6
S-145 60 Norsborg

Mobil: +46-70-799 44 41
Email: [EMAIL PROTECTED]
Skype: eskes1


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RE: Spam? Re: [asterisk-users] Asterisk and IM

[Hall, Eric M.]

Kenneth
 Thanks for the reply. What I'm looking to do is listed here
http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging
However the patch does not work on the system listed below.
 
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kenneth
Padgett
Sent: Friday, January 05, 2007 9:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Spam? Re: [asterisk-users] Asterisk and IM

>  I have been asked to get IM via the X-Ten softphone to work with
Asterisk.
> Anyone have any ideas? I have looked on google and other places with 
> no luck.
>
> Our system is as followed
>
> Linux CentOS 4.4
> Asterisk 1.4.0-beta3
> X-Lite v3.0 for Windows

If by IM, you mean the built-in Jabber stuff in v1.4... I am having
trouble with that and CentOS 4.4 myself, can't get the required libs or
some such non-sense.

-Kenneth
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Re: [asterisk-users] no unicall on 1.4

[Moises Silva]

On 1/5/07, Anton Krall <[EMAIL PROTECTED]> wrote:

I think you are misunderstanding several points here Moises.

May be


I do give Digium a break like you said, that's why you have options

I dont understand this. How is related that you give Digium a break,
with the fact
that I have the option of use sangoma, Digium or clone hardware?


But from the programmers perspective, if some open source software gets too
controlled by a company, then the synergy of open source is kind of broken,

MySQL is open source, but it is mainly ( only ) developed by the
company itself, so please dont contribute to the widely missuse of the
"open source" concept. Open Source does not mean free, Open Source
does not mean anyone is welcomed to contribute code.


all projects have rules and what I meant is that Digium right now is too
controlling and that's why some programmers have decided to leave asterisk

Once again, the rules are not the main problem ( important factor
though ), the main problem are the technical differences.


and pursue other projects in which they fell more comfortable and can
provide code in an easier manner.

Good for them!, more options for everyone.


What I really have never liked is cases like what happened to unicall, there
was no news or announcements, simply , in 1.4 it's not there anymore :)

Anton, honestly, this is the part that "rings" into my head. Digium
NEVER supported formally the development of Unicall, Unicall was NEVER
part of the formal Asterisk release or addons, so it was NEVER there!.
Unicall was only Steves project that happens to work with Asterisk. If
I write a channel driver, as long as I dont sign and fax a disclaimer
to Digium and is accepted, Digium does not have ANY responsibility
about my code, in this case Steve's code. Even AFTER this, please
execute this:

asterisk -vvvr
show warranty

and if you have the free edition as I suppose you will see a BIG

NO WARRANTY

If you want more support BUY ( if you havent done already ) Asterisk
bussiness/enterprise edition.

Kind Regards and Good Look!

Moises

--
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
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