Re: [asterisk-users] intel vs amd motherboards

[Tzafrir Cohen]

On Sun, Jul 09, 2006 at 05:56:49PM -0400, C F wrote:
> Thanks for that Tzafrir. Why does it ignore the secend CPU?

'show translations' is done by a loop that for each pair of codecs
meassures the time it  takes to convert a relatively short ammount of
data between the two.

Thus each conversion is done by a single CPU.

I'm not saying 'show translations' is anywhere near useless. It is a
standard benchmark that comes with Asterisk and useful as such.
Benchmark can be handy when used right. 

-- 
Tzafrir Cohen  sip:[EMAIL PROTECTED]
icq#16849755   iax:[EMAIL PROTECTED]
+972-50-7952406   
[EMAIL PROTECTED]  http://www.xorcom.com
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Re: [asterisk-users] Re: What's the story with X10*P FXO cards?

[Martin Joseph]

On Jul 9, 2006, at 11:44 PM, Rich Adamson wrote:


Martin Joseph wrote:

On Jul 9, 2006, at 10:51 PM, Vincent Delporte wrote:

Thanks for the info. This little experiment is getting expensive ;-)



Rumor has it that Digium will be announcing some new cards in the near 
future that should also help address the space for small numbers of 
analog pstn lines. Unknown as to what the current status happens to 
be.


PCI card aren't an option for me, as I am using OSX as my Asterisk 
platform,  so that doesn't work.  I am still keeping my eyes open for 
additional devices that might work as an FXO or 2, and have the desired 
characteristics.  The Echo can in the Wellgate 3701a is clearly much 
better then the Grandstream one,  but it still takes some number of 
seconds to train at the beginning of calls, which is kind of funky.


Thanks for the info Rich.
Marty

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[asterisk-users] Urgent Upgrade

[Khaled Chehab]

Dear 

I am in need urgently to upgrade my [EMAIL PROTECTED] from 2.6
to trixbox or 2.8 how can I do that .

 

Regards

 

 

 

 






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Re: [asterisk-users] Re: What's the story with X10*P FXO cards?

[Rich Adamson]

Martin Joseph wrote:


On Jul 9, 2006, at 10:51 PM, Vincent Delporte wrote:

Thanks for the info. This little experiment is getting expensive ;-)

LOL!  I know that feeling... I actually thought I would save money with 
VOIP, what a joke!  Actually I am happy with my setup,  but spent an 
intial $75(us) thinking the HT-488 would provide an FXO for my PSTN 
connection, only to find it doesn't work well enough to actually use for 
that.


I then spent an additional $200(us) for the Wellgate 3701a (1FXS+1FXO) 
which, although horribly documented is a HUGE improvement over the 
Grandstream garbage (HT-488).


So yeah,  it gets expensive quickly.  It's annoying that so many of the 
products are marginal or straight up garbage,  but I guess we are still 
in the infancy of the standards based VOIP business, and many of the 
products are in search of a firmware update, or a decent manual.


Oh well, More work for the stubborn techo-maniacs who refuse to give 
up(like me).


I've gone through the same basic devices over the last three years, but 
have also tested the Mediatrix 1204 box as well.


I've got four analog pstn lines here coming from two different central 
offices, both of which are on relatively long loops with somewhat 
unusual echo characteristics. I have also gone through the spa3000, 
ht488, x100p's, multiple h/w versions of the TDM04b, etc. Also have a 
TDM2400 here for testing, but testing has been held up due to delays in 
being able to run current trunk code on the test machine.


To date, the best devices from a "quality of audio" perspective are the 
Mediatrix 1204 and the Sangoma A200D. The last time I worked with the 
1204 was about two years ago and the box had several compatibility and 
security issues that hopefully have been resolved since then. Its retail 
price is higher then the A200D (in the US). The A200D has been in use 
now for several months and has provided excellent audio (etc), supports 
modem use (fax and POS), and just hasn't failed in any way.


Rumor has it that Digium will be announcing some new cards in the near 
future that should also help address the space for small numbers of 
analog pstn lines. Unknown as to what the current status happens to be.


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[asterisk-users] IVR DTMF

[Khaled Chehab]

Dear 

 

 I want to make billing recharge through receiving digits
from IVR through dtmf and store it on a text file ,

 

How can I do that ?

 

Regards

 






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Re: [asterisk-users] CallerID

[Wilson Pickett]

On 7/10/06, Ryder Brook <[EMAIL PROTECTED]> wrote:

and learning a lot and the stupid mistake was that the telephone that I was
calling from has caller id blocked. Well, the only satisfaction is that I


You should always have a way to test with a call that you know is
working, such as a cell phone if you have one or an associate or
firend. More and more people block CID these days, and as a counter
measure, I've noticed more people filtering their calls with a message
about how they won't accept calls with CID blocked!
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Re: [asterisk-users] intel vs amd motherboards

[olivier.taylor]

ooops,

sorry, you right, forgot to mention it...
It was to be compared with AMD 64.

Olivier

C F a écrit :

Olivier can you please do a cat /proc/cpuinfo and post it here? I
think you have a 64 bit cpu.

On 7/9/06, olivier.taylor <[EMAIL PROTECTED]> wrote:


 Fyi,
 Double Intel Xeon 3Ghz performance below


  g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  
ilbc
g723 - - - - - - - - - 
- -
 gsm - - 2 2 2 2 1 410
2914
ulaw - 2 - 1 2 2 1 410
2914
alaw - 2 1 - 2 2 1 410
2914
g726 - 2 2 2 - 2 1 410
2914
   adpcm - 2 2 2 2 - 1 410
2914
slin - 1 1 1 1 1 - 3 9
2813
   lpc10 - 3 3 3 3 3 2 -11
3015
g729 - 3 3 3 3 3 2 5 -
3015
   speex - 3 3 3 3 3 2 511 
-15
ilbc - 3 3 3 3 3 2 511
30 -


 Olivier


 Tzafrir Cohen a écrit :
 On Sun, Jul 09, 2006 at 05:07:16AM -0400, C F wrote:


 Tzafrir, are you trying to tell me that I can realy do double on the
intel becuase the second CPU will do it?

 In the ideal case you'll get double performance with two CPUs. In
theory.

A case of many concurrent calls is basically something that can be
easily parallelized. So in theory nothing stops you from getting
something closer to double performance. I don't know how close reality
is to that nice theory.

I only remarked that 'show translations' totally ignores the second CPU.




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Re: [asterisk-users] Freeware sip/iax client windows mobile

[Nilesh Londhe]

What are the min cpu requirements for ppciax? Has any one tried ppciax
with Cingular 8125?

On 7/9/06, Administrator TOOTAI <[EMAIL PROTECTED]> wrote:

Attilla De Groot wrote:
> Hi all,
>
>
> I have two pda's and I want to be able to make calls, but I need a
> client for this. The only problem is Windows Mobile 5.0, I can't find
> a freeware client for this, the only one is Sjphone. But this one is
> still beta for windows mobile and it just doesn't work good.
>
> Does anyone have an alternative ?
I'm using ppciax

--
Daniel
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Re: [asterisk-users] Re: What's the story with X10*P FXO cards?

[Martin Joseph]

On Jul 9, 2006, at 10:51 PM, Vincent Delporte wrote:

Thanks for the info. This little experiment is getting expensive ;-)

LOL!  I know that feeling... I actually thought I would save money with 
VOIP, what a joke!  Actually I am happy with my setup,  but spent an 
intial $75(us) thinking the HT-488 would provide an FXO for my PSTN 
connection, only to find it doesn't work well enough to actually use 
for that.


I then spent an additional $200(us) for the Wellgate 3701a (1FXS+1FXO) 
which, although horribly documented is a HUGE improvement over the 
Grandstream garbage (HT-488).


So yeah,  it gets expensive quickly.  It's annoying that so many of the 
products are marginal or straight up garbage,  but I guess we are still 
in the infancy of the standards based VOIP business, and many of the 
products are in search of a firmware update, or a decent manual.


Oh well, More work for the stubborn techo-maniacs who refuse to give 
up(like me).


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[asterisk-users] Re: What's the story with X10*P FXO cards?

[Vincent Delporte]

At 22:36 09/07/2006 -0700, "Michael Graves" <[EMAIL PROTECTED]> wrote:
Skip local FXOs altogether. Setup an account with somone who provides DIDs 
via IP. Call forward your analog line to the IP based number. It will be 
absolutely painless compared to the

troubles of small FXO interfaces.


I'll look into this, although
1. I'm not sure VoIP providers do this here yet
2. While the POTS is very reliable, I can't say the same for ADSL. I'm a 
bit scared to depend on the Net for incoming calls.


At 22:36 09/07/2006 -0700, Rich Adamson <[EMAIL PROTECTED]> wrote:
Some folks have had good luck with the spa3000 (and some not). The TDM400 
card from digium works pretty good, but can be less then acceptable in 
some cases. The Sangoma A200D (with hardware EC) works very well on all 
pstn lines that I've tested, but is rather expensive.


Thanks for the info. This little experiment is getting expensive ;-)

VD.


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[asterisk-users] PRI Random Disconnected

[chan \(Alpha Trilogies Networks\)]

Dear Group,

I am having some problem with PRI, my calls randomly get disconnected
and after I am running Debug, I got the out from CLi screen...

 

Cli messages,

-- Executing Dial("Zap/31-1", "zap/g1/100||rTt") in
new stack

-- Making new call for cr 32809

    -- Requested transfer capability: 0x10 - 3K1AUDIO

> Protocol Discriminator: Q.931 (8)  len=30

> Call Ref: len= 2 (reference 41/0x29) (Originator)

> Message type: SETUP (5)

> [04 03 90 90 a3]

> Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info
transfer capability: 3.1kHz audio (16)

> 
Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)

>  
   Ext: 1  User information layer
1: A-Law (35)

> [18 03 a1 83 81]

> Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0,
Preferred Dchan: 0

>   
ChanSel: Reserved

>  
Ext: 1  Coding: 0   Number Specified   Channel Type: 3

>  
Ext: 1  Channel: 1 ]

> [1e 02 80 83]

> Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU)
standard (0) 0: 0   Location: User (0)

>  
Ext: 1  Progress Description: Calling equipment is non-ISDN. (3) ]

> [6c 03 21 81 20]

> Calling Number (len= 5) [ Ext: 0  TON: National Number
(2)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)

> 
 Presentation: Presentation permitted, user number
passed network screening (1) ' ' ]

> [70 04 c1 31 30 30]

> Called Number (len= 6) [ Ext: 1  TON: Subscriber Number
(4)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '100' ]

    -- Called g1/100

< Protocol Discriminator: Q.931 (8)  len=10

< Call Ref: len= 2 (reference 41/0x29) (Terminator)

< Message type: CALL PROCEEDING (2)

< [18 03 a9 83 81]

< Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0,
Exclusive Dchan: 0

<
   ChanSel: Reserved

<  
Ext: 1  Coding: 0   Number Specified   Channel Type: 3

<  
Ext: 1  Channel: 1 ]

-- Processing IE 24 (cs0, Channel Identification)

    -- Zap/1-1 is proceeding passing it to Zap/31-1

< Protocol Discriminator: Q.931 (8)  len=12

< Call Ref: len= 2 (reference 41/0x29) (Terminator)

< Message type: STATUS (125)

< [08 02 81 e4]

< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0:
0   Location: Private network serving the local user (1)

<      Ext:
1  Cause: Unknown (100), class = Protocol Error (6) ]

< [14 01 09]

< Call State (len= 3) [ Ext: 0  Coding:
CCITT (ITU) standard (0) Call state: Incoming Call Proceeding (9)

-- Processing IE 8 (cs0, Cause)

-- Processing IE 20 (cs0, Call
 State)

< Protocol Discriminator: Q.931 (8)  len=5

< Call Ref: len= 2 (reference 41/0x29) (Terminator)

< Message type: ALERTING (1)

    -- Zap/1-1 is ringing

< Protocol Discriminator: Q.931 (8)  len=12

< Call Ref: len= 2 (reference 41/0x29) (Terminator)

< Message type: CONNECT (7)

< [4c 05 01 80 31 30 30]

< Connected Number (len= 7) [ Ext: 0  TON: Unknown Number Type
(0)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)

<
Ext: 1 Presentation: Presentation permitted, user number not screened (0) '100'
]

-- Processing IE 76 (cs0, Connected Number)

> Protocol Discriminator: Q.931 (8)  len=5

> Call Ref: len= 2 (reference 41/0x29) (Originator)

> Message type: CONNECT ACKNOWLEDGE (15)

    -- Zap/1-1 answered Zap/31-1

    -- Attempting native bridge of Zap/31-1 and Zap/1-1 ß---“Extension 100 pickup the call, and get disconnected”

< Protocol Discriminator: Q.931 (8)  len=9

< Call Ref: len= 2 (reference 41/0x29) (Terminator)

< Message type: DISCONNECT (69)

< [08 02 85 90]

< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0:
0   Location: Private network serving the remote user (5)

< 
Ext: 1  Cause: Unknown (16), class = Normal Event (1) ]

-- Processing IE 8 (cs0, Cause)

    -- Channel 0/1, span 1 got hangup request

 

/…. Zapata.conf file

[channels]

context=from-alcatel

switchtype=national

overlapdial=yes

signalling=pri_net

pridialplan=local

resetinterval=never

facilityenable = yes

priindication=outofband

group=1

usecallerid=yes

hidecallerid=no

threewaycalling=yes

transfer=yes

echocancel=yes

echocancelwhenbridged=yes

echotraining=yes

rxgain=-3.5

txgain=-3.5

busydetect=yes  
;Busydetect I did set to no, and calls does not hangup so I turn to yes.

relaxdtmf=yes

immediate=no

channel=>1-15

channel=>17-31

 

Equipment Topologies Connection:

à
Analog Trunk line PSTN (Alcatel Ompcx Office) < -- > Asterisk (E1 as
PRI_NET) 

Incoming goes into Alcatel and Alcatel Divert it to Asterisk via E1 

Asterisk as a Auto-Attendance System….

 

Asterisk – R1.2.5 / Zaptel 1.2.5 / Libpri 1.2.2 ……….CentOS
4.3 final (Kernel 2.6.9-34.01EL)

 

Appreciate if some one can help.

 

 






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Re: [asterisk-users] Re: What's the story with X10*P FXO cards?

[Rich Adamson]

The Masked Cucumber wrote:

At 12:00 09/07/2006 -0700, Rich Adamson <[EMAIL PROTECTED]> wrote:

Are some of the Ebay ad's misrepresented? Probably.


Thanks a lot for the info on the history of the FXO cards. Obviously, 
the ones I bought aren't the good ones :-)


You're likely to become just about as frustrated with the fxo-sip 
gateway boxes on the market.


Mmm... So, what would you recommend? Getting an exensive FXO/SIP box, or 
a Digium card? The one with a single FXO port sells for about 150E over 
here.


As mentioned previously, recommending a specific device basically 
assumes one understands exactly what type of pstn connection you've got, 
how far you are from the central office, etc. Some folks have had good 
luck with the spa3000 (and some not). The TDM400 card from digium works 
pretty good, but can be less then acceptable in some cases. The Sangoma 
A200D (with hardware EC) works very well on all pstn lines that I've 
tested, but is rather expensive.



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Re: [asterisk-users] Global variables and AGI

[Jay Milk]

Kevin Smith wrote:

Hi everyone,

I know that functions like set_variable and get_variable (using php 
with phpagi) only apply to the channel variable. What I need to do is 
reset a global variable I have in our system. I have a script that is 
going to determine when this will happen, but I just have to make it 
happen. Assuming that I cannot update the variable via the script, it 
is there a way  I can make a call to the system, such as a call file, 
and place it in the context of the dialplan that I need to change the 
variable? If so, is there anything special I need in the call file for 
that to work? Or is there a easier/better way to do this that I 
haven't thought of.


Any suggestions would be helpful. Thanks,
Kevin 
As Timebandit pointed out -- 
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Set

or SetGlobalVar in 1.0.x

If most of the interaction with that variable occurs through agi, you 
might also want to consider storing it outside of Asterisk.  I've stored 
a good number of values in mysql for an asterisk application before.  If 
most of the interaction occurs within the dialplan and/or you're trying 
to avoid agi, you could also use the asterisk database directly with 
DBPut and DBGet.

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[asterisk-users] spandsp and app_*fax.c

[Warrick Zedi]

Greetings,

 

Does anyone know where I can get app_txfax and app_rxfax
that’ll work with spandsp 0.0.3pre22?

 

I’ve tried the source in the snapshots/test-apps
directory. I’ve also tried the files that came with t38bits.tgz. This is
what I get out of make:

 

make[1]: *** No rule to make target
`/usr/local/include/spandsp/alaw_ulaw.h', needed by `app_rxfax.o'.  Stop.

make[1]: Leaving directory `/usr/src/asterisk-1.2.9.1/apps'

make: *** [subdirs] Error 1

 

My real issue is that I can’t get txfax to send a fax
without a core dump using 0.0.2pre26 with Asterisk 1.2.9.1. I’ve tried
many versions of libtiff; 3.6.0, 3.6.1, 3.5.7, 3.8.2 and others - always a core
dump. 

 

I’ve tried the same thing with a brand new trixbox
install – core dump. I did get a step further by commenting lines in t4.c
but I know this is not a real solution. Short of trying to debug this (which
I’m going to try while waiting for responses to this) what should I be
trying to do?

 

Any advice would be very much appreciated. Here’s a backtrace
from gdb if this is at all helpful.

 

 (gdb) frame

#0  0x00930540 in _int_malloc () from
/lib/tls/libc.so.6

 (gdb) bt

#0  0x00930540 in _int_malloc () from
/lib/tls/libc.so.6

#1  0x009320b1 in malloc () from /lib/tls/libc.so.6

#2  0x00932869 in realloc () from /lib/tls/libc.so.6

#3  0x00738f55 in t4_encode_eol (s=0x9f5820) at
t4.c:404

#4  0x0073af7e in t4_encode_row (s=0xb7b163d0,

    bp=0x8c4b348 'ÿ' ...) at t4.c:1570

#5  0x0073b76c in t4_tx_start_page (s=0xb7b163d0) at
t4.c:1854

#6  0x0073ee4e in hdlc_accept (user_data=0xb7b163d0,
ok=1,

    msg=0xb7b16644 "ÿ\023\204ê}",
len=3) at t30.c:1135

#7  0x007328a5 in hdlc_rx_bit (s=0xb7b1661c, new_bit=0)
at hdlc.c:300

#8  0x0072f861 in fsk_rx (s=0xb7b169d8, amp=0x8c1a6fc,
len=160) at fsk.c:344

#9  0x0073f7bd in fax_rx (s=0xb7b163d0, buf=0x8c1a6fc,
len=160) at t30.c:2712

#10 0x00896427 in txfax_exec (chan=0x8bfc770,
data="">

    at app_txfax.c:230

#11 0x0808eac2 in ast_pbx_outgoing_app (

    type=0x8c4ba4c 'ÿ' ..., format=64, data="">

    timeout=2, app=0x8c4bc4c 'ÿ'
...,

    appdata=0x8c4bd4c 'ÿ' ..., reason=0xb7b18438, sync=2,

    cid_num=0x8c4c050 'ÿ' ...,

    cid_name=0x8c4c150 'ÿ' ..., vars=0x0,

    account=0x8c4c250 'ÿ' , locked_channel=0x0) at pbx.c:553

#12 0x002e736d in attempt_thread (data="" at
pbx_spool.c:262

#13 0x00baa371 in start_thread () from
/lib/tls/libpthread.so.0

#14 0x009959be in clone () from /lib/tls/libc.so.6

(gdb)

 

Cheers,
Warrick






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Re: [asterisk-users] CallerID

[Ryder Brook]

Thanks to everyone.I had everything right except that rxgain and txgain were set to 0.I am actually embarassed to say that I spent most of Saturday getting lost and learning a lot and the stupid mistake was that the telephone that I was calling from has caller id blocked. Well, the only satisfaction is that I now know how the whole thing is configured. I also learnt that you have to start with a clean slate for dialplans, and throw away what's left as *.conf files after [EMAIL PROTECTED] installation. Also, avoid deeply nested includes.-bramanRyder Brook PediatricsP.O.Box 608Morrisville, VT 05661 __Do You Yahoo!?Tired of spam?  Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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RE: [asterisk-users] Choppy MOH (Cisco gateway)

[John Sawa]

You will also want to add

no vad 

to your dial-peer config to disable voice activity detection.

I do not think it will resolve your issue, but worth a shot.

-John

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of 
> Bill Gibbs
> Sent: Sunday, July 09, 2006 7:42 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Choppy MOH (Cisco gateway)
> 
> 
> I upgraded one of the boxes to 1.2.9.1 and using native MOH I 
> still get
> it.  I made sure to upgrade zaptel, etc as well.
> 
> I do have something of interest to note...
> Placing the call on hold then taking it off hold and back on the music
> is ok (doing that once it gets choppy) of course this is not practical
> since the person using hold won't know if it's choppy.  It then gets
> choppy again if you wait 15-20 secs.
> 
> I have 2 ways of making outbound calls from all of the boxes, 
> and I did
> the following via 1.2.9.1 and 1.2.4
> 
> 1) Send the outbound call to the Cisco and send out via the PRI (sip
> phone ulaw to Cisco ulaw out the PRI)
> 2) Dial "long distance" to a provider using g729 (Polycom to Asterisk
> ulaw, Asterisk transcoding to g729 to provider)
> 
> If I call from a sip phone OUT to my cell via the long 
> distance provider
> I get no choppiness.   I am not able to get inbound calls from the
> provider so I can only test one way.
> 
> So I then switched talking to my Cisco via g729 (letting asterisk
> transcode ulaw to g729 and also g729 all the way through) and voice is
> fine but MOH is still choppy.  So it must be something with the Cisco
> maybe?  IOS version is 
> Cisco IOS Software, 3600 Software (C3660-IS-M), Version 12.3(14)T6,
> RELEASE SOFTWARE (fc2)
> 
> I have setup for the codecs:
> voice class codec 1
>  codec preference 1 g711ulaw
>  codec preference 2 g729r8
> 
> incoming dial-peer:
> 
> dial-peer voice 1 pots
>  description Match all incoming calls, set DID
>  incoming called-number .T
>  direct-inward-dial
>  forward-digits extra
> 
> dial-peer voice 16 voip
>  description to the asterisk server
>  destination-pattern 
>  voice-class codec 1
>  session protocol sipv2
>  session target ipv4:
>  dtmf-relay sip-notify rtp-nte
> 
> and outbound:
> 
> dial-peer voice 1 pots
>  description Outbound via PRI
>  destination-pattern .T
>  port 1/0:23
>  forward-digits all
> 
> Could this have something to do with the Cisco suppressing the stream
> using silence suppression...I read somewhere that Asterisk 
> relies on Sip
> packets for MOH??? 
> 
> There is not a bandwidth issue, the 3660 and boxes are on the same
> switch VLAN w/ DSCP enabled.
> 
> Bill
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of mike
> Sent: Monday, July 10, 2006 2:51 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Choppy MOH (Cisco gateway)
> 
> i had a similar issue with the first branch of asterisk 1.2 and cheap
> phones (tip-100 from tatung)
> i'll suggest you to upgrade your asterisk box
> are you using bristuff ?
> try bristuff-0.3.0-PRE-1q which comes with asterisk 1.2.9.1
> 
> lemme know
> .mike
> 
> 
> On Sun, 2006-07-09 at 14:49 -0400, Bill Gibbs wrote:
> > Cisco 3660 with a PRI talking SIP to various Asterisk boxes (not
> > connected, separate PBXs)  using ulaw all have issues with music on
> > hold being choppy.  Normal voice and SIP (taking a call 
> from the PRI,
> > placing a call or extension to extension calls) conversations are
> > _perfect_ with no drop outs so it's not a problem with the 
> PRI or the
> > 3660 talking to the Asterisk boxes.  If I call from my 
> Polycom into an
> > extension that immediately starts MusicOnHold it's perfect as well.
> > 
> >  
> > 
> > However, calling into the box via the PRI and being placed 
> on hold the
> > music is choppy.  Also, calling into an extension that spawns
> > MusicOnHold immediately is choppy when it comes in via the Cisco.
> > 
> >  
> > 
> > This happens with mpg123, madplay and I tried using the Asterisk 1.2
> > native mode in musiconhold.conf:
> > 
> >  
> > 
> > [default]
> > 
> > mode => files
> > 
> > directory => /var/lib/asterisk/mohmp3
> > 
> > random => yes
> > 
> >  
> > 
> > Same problem with all 3.
> > 
> >  
> > 
> > Tried converting MP3s to a pcm or ulaw file, same problem 
> (using lame
> > and sox to do the conversions)
> > 
> >  
> > 
> > It seems that this is common issue with no clear resolution.
> > 
> >  
> > 
> > Machines are Pentium 4s 512MB or 1GB RAM.  I would be the 
> only call on
> > the box, no load, etc.
> > 
> > Using ztdummy (or without, same behavior)
> > 
> > Asterisk ver 1.2.4 on all
> > 
> > Normal voice, IVR, play back voicemail, etc are all 100% 
> perfect only
> > on MusicOnHold has this issue
> > 
> > Polycom SIP phones or using X-Lite to test (used to make 
> the call into
> > MusicOnHold or answer the call coming in via the PRI and placing on
> > hold)
> > 
> > Calling in 

RE: [asterisk-users] Choppy MOH (Cisco gateway)

[Bill Gibbs]

I upgraded one of the boxes to 1.2.9.1 and using native MOH I still get
it.  I made sure to upgrade zaptel, etc as well.

I do have something of interest to note...
Placing the call on hold then taking it off hold and back on the music
is ok (doing that once it gets choppy) of course this is not practical
since the person using hold won't know if it's choppy.  It then gets
choppy again if you wait 15-20 secs.

I have 2 ways of making outbound calls from all of the boxes, and I did
the following via 1.2.9.1 and 1.2.4

1) Send the outbound call to the Cisco and send out via the PRI (sip
phone ulaw to Cisco ulaw out the PRI)
2) Dial "long distance" to a provider using g729 (Polycom to Asterisk
ulaw, Asterisk transcoding to g729 to provider)

If I call from a sip phone OUT to my cell via the long distance provider
I get no choppiness.   I am not able to get inbound calls from the
provider so I can only test one way.

So I then switched talking to my Cisco via g729 (letting asterisk
transcode ulaw to g729 and also g729 all the way through) and voice is
fine but MOH is still choppy.  So it must be something with the Cisco
maybe?  IOS version is 
Cisco IOS Software, 3600 Software (C3660-IS-M), Version 12.3(14)T6,
RELEASE SOFTWARE (fc2)

I have setup for the codecs:
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8

incoming dial-peer:

dial-peer voice 1 pots
 description Match all incoming calls, set DID
 incoming called-number .T
 direct-inward-dial
 forward-digits extra

dial-peer voice 16 voip
 description to the asterisk server
 destination-pattern 
 voice-class codec 1
 session protocol sipv2
 session target ipv4:
 dtmf-relay sip-notify rtp-nte

and outbound:

dial-peer voice 1 pots
 description Outbound via PRI
 destination-pattern .T
 port 1/0:23
 forward-digits all

Could this have something to do with the Cisco suppressing the stream
using silence suppression...I read somewhere that Asterisk relies on Sip
packets for MOH??? 

There is not a bandwidth issue, the 3660 and boxes are on the same
switch VLAN w/ DSCP enabled.

Bill

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mike
Sent: Monday, July 10, 2006 2:51 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Choppy MOH (Cisco gateway)

i had a similar issue with the first branch of asterisk 1.2 and cheap
phones (tip-100 from tatung)
i'll suggest you to upgrade your asterisk box
are you using bristuff ?
try bristuff-0.3.0-PRE-1q which comes with asterisk 1.2.9.1

lemme know
.mike


On Sun, 2006-07-09 at 14:49 -0400, Bill Gibbs wrote:
> Cisco 3660 with a PRI talking SIP to various Asterisk boxes (not
> connected, separate PBXs)  using ulaw all have issues with music on
> hold being choppy.  Normal voice and SIP (taking a call from the PRI,
> placing a call or extension to extension calls) conversations are
> _perfect_ with no drop outs so it's not a problem with the PRI or the
> 3660 talking to the Asterisk boxes.  If I call from my Polycom into an
> extension that immediately starts MusicOnHold it's perfect as well.
> 
>  
> 
> However, calling into the box via the PRI and being placed on hold the
> music is choppy.  Also, calling into an extension that spawns
> MusicOnHold immediately is choppy when it comes in via the Cisco.
> 
>  
> 
> This happens with mpg123, madplay and I tried using the Asterisk 1.2
> native mode in musiconhold.conf:
> 
>  
> 
> [default]
> 
> mode => files
> 
> directory => /var/lib/asterisk/mohmp3
> 
> random => yes
> 
>  
> 
> Same problem with all 3.
> 
>  
> 
> Tried converting MP3s to a pcm or ulaw file, same problem (using lame
> and sox to do the conversions)
> 
>  
> 
> It seems that this is common issue with no clear resolution.
> 
>  
> 
> Machines are Pentium 4s 512MB or 1GB RAM.  I would be the only call on
> the box, no load, etc.
> 
> Using ztdummy (or without, same behavior)
> 
> Asterisk ver 1.2.4 on all
> 
> Normal voice, IVR, play back voicemail, etc are all 100% perfect only
> on MusicOnHold has this issue
> 
> Polycom SIP phones or using X-Lite to test (used to make the call into
> MusicOnHold or answer the call coming in via the PRI and placing on
> hold)
> 
> Calling in from landline or cell phone - no difference
> 
>  
> 
> Any ideas?
> 
>  
> 
> Bill
> 
> 
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[asterisk-users] How to transfer other sessions

[PSPunch]

Hi all.


I am looking for a method to transfer a caller
on an existing session to another extension.


For example,

SIP/200 is currently talking with SIP/300.

I want to force user SIP/200 to be transfered to SIP/400.
(without myself being SIP/200. I am just an admin
at the asterisk console)
Then SIP/300 left alone, can be hung up.


Is there any way this can be done from the Asterisk
console or any other Dial plan commands or AGI commands ?


Any advise or pointers I will truly appreciate.
Thank you.

--
David Shimamoto
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RE: [asterisk-users] audio session start delay

[Luca Corti]

On Thu, 2006-07-06 at 23:22 -0300, Fabio wrote:
> are you using SIP reinvite ?

Proably not as I'm using "t" in Dial()s for call transfer.

> post a bit more information (sip.conf)

[general]
context=sip
allowguest=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
domain=mydomain.com
domain=1.2.3.4
allowexternalinvites=no
language=it
relaxdtmf=yes

[authentication]

[as5350] ; My PSTN gateway
type=peer
qualify=200
host=1.2.3.5
fromdomain=1.2.3.5
insecure=very

[ser] ; My SIP proxy
type=peer
qualify=200
host=1.2.3.6
fromdomain=1.2.3.6
insecure=very

[01]; Extension example
callerid=My Name <01>
nat=yes
type=friend
username=01
secret=mypass
host=dynamic
dtmfmode=rfc2833
context=uffici
canreinvite=no
callgroup=1
pickupgroup=1
qualify=no

Thanks

-- 
Luca Corti
PGP Key ID 1F38C091
Adesso dico: "L'usignolo chiuso in gabbia smette di cantare."

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Re: [asterisk-users] NuFone suggests to use Vonage!!!!

[Joe Baptista]

On Sun, 9 Jul 2006, Andrew D Kirch wrote:

> To some extent I see your point and have been on the receiving end of
> one of Jeremy's tirades.
>  I've since decided that NuFone is an interesting study in whether your
> business can survive
> with only clueful customers.

Some people are into S&M I guess.  We have used NuFone.  No problems
during that period.  If you know what you doing it's not bad.

regards
joe
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Re: [asterisk-users] Choppy MOH (Cisco gateway)

[mike]

i had a similar issue with the first branch of asterisk 1.2 and cheap
phones (tip-100 from tatung)
i'll suggest you to upgrade your asterisk box
are you using bristuff ?
try bristuff-0.3.0-PRE-1q which comes with asterisk 1.2.9.1

lemme know
.mike


On Sun, 2006-07-09 at 14:49 -0400, Bill Gibbs wrote:
> Cisco 3660 with a PRI talking SIP to various Asterisk boxes (not
> connected, separate PBXs)  using ulaw all have issues with music on
> hold being choppy.  Normal voice and SIP (taking a call from the PRI,
> placing a call or extension to extension calls) conversations are
> _perfect_ with no drop outs so it’s not a problem with the PRI or the
> 3660 talking to the Asterisk boxes.  If I call from my Polycom into an
> extension that immediately starts MusicOnHold it’s perfect as well.
> 
>  
> 
> However, calling into the box via the PRI and being placed on hold the
> music is choppy.  Also, calling into an extension that spawns
> MusicOnHold immediately is choppy when it comes in via the Cisco.
> 
>  
> 
> This happens with mpg123, madplay and I tried using the Asterisk 1.2
> native mode in musiconhold.conf:
> 
>  
> 
> [default]
> 
> mode => files
> 
> directory => /var/lib/asterisk/mohmp3
> 
> random => yes
> 
>  
> 
> Same problem with all 3.
> 
>  
> 
> Tried converting MP3s to a pcm or ulaw file, same problem (using lame
> and sox to do the conversions)
> 
>  
> 
> It seems that this is common issue with no clear resolution.
> 
>  
> 
> Machines are Pentium 4s 512MB or 1GB RAM.  I would be the only call on
> the box, no load, etc.
> 
> Using ztdummy (or without, same behavior)
> 
> Asterisk ver 1.2.4 on all
> 
> Normal voice, IVR, play back voicemail, etc are all 100% perfect only
> on MusicOnHold has this issue
> 
> Polycom SIP phones or using X-Lite to test (used to make the call into
> MusicOnHold or answer the call coming in via the PRI and placing on
> hold)
> 
> Calling in from landline or cell phone – no difference
> 
>  
> 
> Any ideas?
> 
>  
> 
> Bill
> 
> 
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Re: [asterisk-users] NuFone suggests to use Vonage!!!!

[Andrew D Kirch]

Ronald Wiplinger wrote:

Part of a conversation with NuFone.
It is untrue, that they do not answer, but if than:

Quote:

3. change your attitude towards customers!!



No, if you don't like it, go use Vonage.


End of quote!


I had always problems with these people.

bye

Ronald
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To some extent I see your point and have been on the receiving end of 
one of Jeremy's tirades. 
I've since decided that NuFone is an interesting study in whether your 
business can survive

with only clueful customers.

Andrew
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[asterisk-users] NuFone suggests to use Vonage!!!!

[Ronald Wiplinger]

Part of a conversation with NuFone.
It is untrue, that they do not answer, but if than:

Quote:

3. change your attitude towards customers!!



No, if you don't like it, go use Vonage. 




End of quote!


I had always problems with these people.

bye

Ronald
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Re: [asterisk-users] Re: What's the story with X10*P FXO cards?

[Michael Graves]

Skip local FXOs altogether. Setup an account with somone who provides DIDs via IP. Call forward your analog line to the IP based number. It will be absolutely painless compared to the troubles of small FXO interfaces.



Michael



On Sun, 09 Jul 2006 21:09:40 +0200, The Masked Cucumber wrote:



>At 12:00 09/07/2006 -0700, Rich Adamson <[EMAIL PROTECTED]> wrote:

>>Are some of the Ebay ad's misrepresented? Probably.

>

>Thanks a lot for the info on the history of the FXO cards. Obviously, the 

>ones I bought aren't the good ones :-)

>

>>You're likely to become just about as frustrated with the fxo-sip gateway 

>>boxes on the market.

>

>Mmm... So, what would you recommend? Getting an exensive FXO/SIP box, or a 

>Digium card? The one with a single FXO port sells for about 150E over here.

>

>Cheers

>Vincent.

>

>

>-- 

>No virus found in this outgoing message.

>Checked by AVG Anti-Virus.

>Version: 7.1.394 / Virus Database: 268.9.10/383 - Release Date: 07/07/2006

>

>

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>






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Re: [asterisk-users] 2 Handsets, Same extension

[Michael Graves]

On Sun, 09 Jul 2006 19:17:36 +0100, Thomas Kenyon wrote:

>Is there a Way I can have 2 phones, on the same extension (as in
>Dial(SIP/phone1&SIP/Phone2) ), whereby if one phone is in use, then the
>other one will not be rung if called?

>This is useful to me, in the situation where the phone system has a
>queue and an agent will very often not be at his desk and has access to
>a wireless handset.

>If it helps, both handsets are SIP devices.

I do this throught using Astra 480i CT phones. Each desk phone comes with an 
associated cordless handset. Calls ring on both with only one registration into 
the server. Costs about the same 
as a decent Polycom phones.

Michael



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Re: [asterisk-users] intel vs amd motherboards

[C F]

Thanks for that Tzafrir. Why does it ignore the secend CPU?

BTW, on a side note on this topic, how can one calculate simultaneous
transcoded channels using show transalation?

In the case where it tells me 17 ms for encoding and 4 for decoding,
that gives me 21ms per channel, in what time frame can I squeeze in
how many channels before the calls start becoming  intolerable? In
other words should I aim for a 200ms time frame which means that I
will get around 10 channels? or can I aim for a full second? which
will give me around 50 channels?

Thank You

On 7/9/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

On Sun, Jul 09, 2006 at 05:07:16AM -0400, C F wrote:
> Tzafrir, are you trying to tell me that I can realy do double on the
> intel becuase the second CPU will do it?

In the ideal case you'll get double performance with two CPUs. In
theory.

A case of many concurrent calls is basically something that can be
easily parallelized. So in theory nothing stops you from getting
something closer to double performance. I don't know how close reality
is to that nice theory.

I only remarked that 'show translations' totally ignores the second CPU.

--
Tzafrir Cohen  sip:[EMAIL PROTECTED]
icq#16849755   iax:[EMAIL PROTECTED]
+972-50-7952406
[EMAIL PROTECTED]  http://www.xorcom.com
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Re: [asterisk-users] intel vs amd motherboards

[C F]

Olivier can you please do a cat /proc/cpuinfo and post it here? I
think you have a 64 bit cpu.

On 7/9/06, olivier.taylor <[EMAIL PROTECTED]> wrote:


 Fyi,
 Double Intel Xeon 3Ghz performance below


  g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
g723 - - - - - - - - - - -
 gsm - - 2 2 2 2 1 4102914
ulaw - 2 - 1 2 2 1 4102914
alaw - 2 1 - 2 2 1 4102914
g726 - 2 2 2 - 2 1 4102914
   adpcm - 2 2 2 2 - 1 4102914
slin - 1 1 1 1 1 - 3 92813
   lpc10 - 3 3 3 3 3 2 -113015
g729 - 3 3 3 3 3 2 5 -3015
   speex - 3 3 3 3 3 2 511 -15
ilbc - 3 3 3 3 3 2 51130 -

 Olivier


 Tzafrir Cohen a écrit :
 On Sun, Jul 09, 2006 at 05:07:16AM -0400, C F wrote:


 Tzafrir, are you trying to tell me that I can realy do double on the
intel becuase the second CPU will do it?

 In the ideal case you'll get double performance with two CPUs. In
theory.

A case of many concurrent calls is basically something that can be
easily parallelized. So in theory nothing stops you from getting
something closer to double performance. I don't know how close reality
is to that nice theory.

I only remarked that 'show translations' totally ignores the second CPU.




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Re: [asterisk-users] Global variables and AGI

[Time Bandit]

Hi everyone,

I know that functions like set_variable and get_variable (using php with
phpagi) only apply to the channel variable. What I need to do is reset a
global variable I have in our system. I have a script that is going to
determine when this will happen, but I just have to make it happen.
Assuming that I cannot update the variable via the script, it is there a
way  I can make a call to the system, such as a call file, and place it
in the context of the dialplan that I need to change the variable? If
so, is there anything special I need in the call file for that to work?
Or is there a easier/better way to do this that I haven't thought of.

Reading this : http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Set
you need to use the g option

If you can't use it from phpagi, you could, at worst, set a local
variable from your AGI, then in the dialplan take the value of that
one and apply it to the global one using option g

hth
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[asterisk-users] Global variables and AGI

[Kevin Smith]

Hi everyone,

I know that functions like set_variable and get_variable (using php with 
phpagi) only apply to the channel variable. What I need to do is reset a 
global variable I have in our system. I have a script that is going to 
determine when this will happen, but I just have to make it happen. 
Assuming that I cannot update the variable via the script, it is there a 
way  I can make a call to the system, such as a call file, and place it 
in the context of the dialplan that I need to change the variable? If 
so, is there anything special I need in the call file for that to work? 
Or is there a easier/better way to do this that I haven't thought of.


Any suggestions would be helpful. Thanks,
Kevin
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Re: [asterisk-users] 2 Handsets, Same extension

[Lacy Moore - Aspendora]

It looks like a good starting point would be here:
 
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup
 
On 7/9/06, Thomas Kenyon <[EMAIL PROTECTED]> wrote:
Is there a Way I can have 2 phones, on the same extension (as inDial(SIP/phone1&SIP/Phone2) ), whereby if one phone is in use, then the
other one will not be rung if called?This is useful to me, in the situation where the phone system has aqueue and an agent will very often not be at his desk and has access toa wireless handset.
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Re: [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?

[Martin Joseph]

On Jul 9, 2006, at 12:11 PM, Jay Milk wrote:

Good idea.  Let's write some open-source firmware for Cisco phones.  
What will you be contributing?



Well, it looks like I already contributed the idea didn't I ;~)

I don't have any Cisco phones,  so if you want to send be a couple I'll 
take a look at them.




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Re: [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?

[Jay Milk]

Good idea.  Let's write some open-source firmware for Cisco phones.  
What will you be contributing?


Martin Joseph wrote:




Another side issue... the Cisco phones have no way to "remove" the 
installed firmware. Therefore, there is no way to legally sell a used 
Cisco phone, period.



As a non Cisco user this whole discussion is enough to steer me away 
from there VOIP products for good.  I hate the idea that you need 
special permission from them to use hardware you bought legally.


You bought it you own it, perhaps there should be some sort of open 
sourced firmware project for these phones?


Dunno,  but I hate whoever though up this approach.

Marty


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[asterisk-users] Re: What's the story with X10*P FXO cards?

[The Masked Cucumber]

At 12:00 09/07/2006 -0700, Rich Adamson <[EMAIL PROTECTED]> wrote:

Are some of the Ebay ad's misrepresented? Probably.


Thanks a lot for the info on the history of the FXO cards. Obviously, the 
ones I bought aren't the good ones :-)


You're likely to become just about as frustrated with the fxo-sip gateway 
boxes on the market.


Mmm... So, what would you recommend? Getting an exensive FXO/SIP box, or a 
Digium card? The one with a single FXO port sells for about 150E over here.


Cheers
Vincent.


--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.1.394 / Virus Database: 268.9.10/383 - Release Date: 07/07/2006


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Re: [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?

[Martin Joseph]

Another side issue... the Cisco phones have no way to "remove" the 
installed firmware. Therefore, there is no way to legally sell a used 
Cisco phone, period.



As a non Cisco user this whole discussion is enough to steer me away 
from there VOIP products for good.  I hate the idea that you need 
special permission from them to use hardware you bought legally.


You bought it you own it, perhaps there should be some sort of open 
sourced firmware project for these phones?


Dunno,  but I hate whoever though up this approach.

Marty

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[asterisk-users] Choppy MOH (Cisco gateway)

[Bill Gibbs]

Cisco 3660 with a PRI talking SIP to various Asterisk boxes
(not connected, separate PBXs)  using ulaw all have issues with music on
hold being choppy.  Normal voice and SIP (taking a call from the PRI,
placing a call or extension to extension calls) conversations are _perfect_ with no drop outs so it’s
not a problem with the PRI or the 3660 talking to the Asterisk boxes.  If
I call from my Polycom into an extension that immediately starts MusicOnHold it’s
perfect as well.

 

However, calling into the box via the PRI and being placed
on hold the music is choppy.  Also, calling into an extension that spawns
MusicOnHold immediately is choppy when it comes in via the Cisco.

 

This happens with mpg123, madplay and I tried using the
Asterisk 1.2 native mode in musiconhold.conf:

 

[default]

mode => files

directory => /var/lib/asterisk/mohmp3

random => yes

 

Same problem with all 3.

 

Tried converting MP3s to a pcm or ulaw file, same problem
(using lame and sox to do the conversions)

 

It seems that this is common issue with no clear resolution.

 

Machines are Pentium 4s 512MB or 1GB RAM.  I would be
the only call on the box, no load, etc.

Using ztdummy (or without, same behavior)

Asterisk ver 1.2.4 on all

Normal voice, IVR, play back voicemail, etc are all 100%
perfect only on MusicOnHold has this issue

Polycom SIP phones or using X-Lite to test (used to make the
call into MusicOnHold or answer the call coming in via the PRI and placing on
hold)

Calling in from landline or cell phone – no difference

 

Any ideas?

 

Bill






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[asterisk-users] 2 Handsets, Same extension

[Thomas Kenyon]

Is there a Way I can have 2 phones, on the same extension (as in
Dial(SIP/phone1&SIP/Phone2) ), whereby if one phone is in use, then the
other one will not be rung if called?

This is useful to me, in the situation where the phone system has a
queue and an agent will very often not be at his desk and has access to
a wireless handset.

If it helps, both handsets are SIP devices.

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Re: [asterisk-users] Can one SIP extension be used for two phones?

[alunt2003]

On Sun, Jul 09, 2006 at 11:02:44AM -0400, Time Bandit wrote:
> >Picture this:
> >
> >Exten => 100 #My Phone
> >Exten => 200 #MythPhone
> >
> >Call comes in. Dialplan calls both extensions.
> >MythPhone is an add-on for MythTV,so when i receive a call,the CallerID
> >is flashed up on my TV.
> >I want to add another MythPhone to my other MythTV box upstairs.
> >
> >Do i have to make a third extension and get the dialplan to call all
> >three extensions or can the second MythPhone instance also log into
> >Asterisk using the same extension?
> 
> Each phone needs its extension
> 
> But, you can have the dialplan ring more than one phone and the first
> to pickup the call is the lucky one :)
> 
> like : Dial(SIP/100&SIP/200&SIP/300)
> 
> hth

Yeah, I have the dialplan goto voicemail if unanswered so mythphone
never actually picks up.
I use mythphone purely as a visual indication of who is calling.

Thanks for replying.

Alun.
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Re: [asterisk-users] intel vs amd motherboards

[olivier.taylor]

Fyi,
Double Intel Xeon 3Ghz performance below

 g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex 
ilbc
   g723 - - - - - - - - - -
-
    gsm - - 2 2 2 2 1 4    10    29   
14
   ulaw - 2 - 1 2 2 1 4    10    29   
14
   alaw - 2 1 - 2 2 1 4    10    29   
14
   g726 - 2 2 2 - 2 1 4    10    29   
14
  adpcm - 2 2 2 2 - 1 4    10    29   
14
   slin - 1 1 1 1 1 - 3 9    28   
13
  lpc10 - 3 3 3 3 3 2 -    11    30   
15
   g729 - 3 3 3 3 3 2 5 -    30   
15
  speex - 3 3 3 3 3 2 5    11 -   
15
   ilbc - 3 3 3 3 3 2 5    11    30
-

Olivier


Tzafrir Cohen a écrit :

  On Sun, Jul 09, 2006 at 05:07:16AM -0400, C F wrote:
  
  
Tzafrir, are you trying to tell me that I can realy do double on the
intel becuase the second CPU will do it?

  
  
In the ideal case you'll get double performance with two CPUs. In
theory.

A case of many concurrent calls is basically something that can be
easily parallelized. So in theory nothing stops you from getting
something closer to double performance. I don't know how close reality
is to that nice theory.

I only remarked that 'show translations' totally ignores the second CPU.

  



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Re: [asterisk-users] Can one SIP extension be used for two phones?

[Time Bandit]

Picture this:

Exten => 100 #My Phone
Exten => 200 #MythPhone

Call comes in. Dialplan calls both extensions.
MythPhone is an add-on for MythTV,so when i receive a call,the CallerID
is flashed up on my TV.
I want to add another MythPhone to my other MythTV box upstairs.

Do i have to make a third extension and get the dialplan to call all
three extensions or can the second MythPhone instance also log into
Asterisk using the same extension?


Each phone needs its extension

But, you can have the dialplan ring more than one phone and the first
to pickup the call is the lucky one :)

like : Dial(SIP/100&SIP/200&SIP/300)

hth
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Re: [asterisk-users] Outgoing MSNs and chan_misdn

[Gary Hawkins]

Marco Mouta wrote:
> in your init-misdn.conf (or misdn.conf, not sure now...) you can
> choose the MSNs for your incoming Ports or Outgoing ports,
> msns=3223242,3223243,3223244
> for example.
> Then in your calls,  just set the outgoing callerid for your trunk, to
> one of them. Be aware that as far as i know you must own the MSN you r
> going to set otherwise you are spoofing MSN
> Please give some feedback.

I've got it working now -- thank you!

I notice that you can also use msns=* as well as setting the individual
numbers.  Once I'd entered that into misdn.conf, and used a command of the
form Set(CALLERID(num)=234567) in the dialplan, it now works as I want it to.

Gary H

-- 
Gary Hawkins MBCS <[EMAIL PROTECTED]>
PGP: 0x6D4E5C77 (expires 31 Dec 2006)
Web: http://www.garyhawkins.me.uk
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[asterisk-users] Can one SIP extension be used for two phones?

[alunt2003]

Picture this:

Exten => 100 #My Phone
Exten => 200 #MythPhone

Call comes in. Dialplan calls both extensions.
MythPhone is an add-on for MythTV,so when i receive a call,the CallerID
is flashed up on my TV.
I want to add another MythPhone to my other MythTV box upstairs.

Do i have to make a third extension and get the dialplan to call all
three extensions or can the second MythPhone instance also log into
Asterisk using the same extension?

Thanks, Alun
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[asterisk-users] zap and fax

[Giedrius Augys]

Hi, My situation is : I need to send fax from sip  device attached  fax over zap channel. Using G711, fax send ok, but is it posible to use t.38 protocol. Maybe someone can suggest me what software to use?
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Re: [asterisk-users] Tired of fax calls... :-/

[Thomas Kenyon]

Olivier wrote:
> 2006/7/6, Maxim Vexler <[EMAIL PROTECTED] >:
>
> NVFaxDetect does just that ;)
>
> Do you think NVFaxDetect is reliable ?
> Could you use it along a voicemail (I mean : someone having a single
> extension for voice and fax call, forward all incoming calls to its
> voicemail when leaving the office)
>
> Cheers
>   
If you use NVBackgroundDetect(not-here-greeting) then VoiceMail(sBOX) (s
means no announcement), It appears to work.
You will need a [fax] context to handle the fax.

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Re: [asterisk-users] Phone Ring

[Thomas Kenyon]

Olivier Saulnier wrote:
> Hello,
>
> Do you know where i can download some rings for a PA1688 based Phone?
> All rings on this link are not very nice...:
> http://www.aredfox.com/edownloadsring.htm
>
> Best regards,
>
Look at the technical documentation on the site, iirc the ringfiles are
encoded in  a common codec type that you can sox whatever you want to
generate.

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RE: [asterisk-users] Asterisk with ISDN Fritz PCI card

[Guy Corbaz]

Hi,

Thank you for the suggestion.

I tried to use mISDN first, then CAPI and now I'm trying I4L.

As I'm using Debian, I can not load the FRITZ drivers. I got the source 
from the official site and recompiled it, but there is a strange message in 
the log and the capi drivers are not loaded.


The problem is more linked to drivers that Asterisk. If you have any tips 
to get this up and running, I would be very happy as my search on the 
Internet didn't allowed me to solve that issue.


Bests regards, Guy.

At 11:25 09.07.2006 +1000, you wrote:
What are you using (misdn, capi, something else?) and what problems are 
you having?


I submitted a patch recently to mISDN which should have fixed a problem on 
hangup, if that's the problem you are having then try the latest cvs 
mqueue branch of mISDN.


James

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Guy Corbaz
> Sent: Saturday, 8 July 2006 23:59
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Asterisk with ISDN Fritz PCI card
>
> Dear all,
>
> I'm desperately trying to get Asterisk working with a FRITZ PCI card on
> Debian with kernel 2.6.15.
>
> I'm wondering if anybody has such a working installation.
>
> Thank you for your help, Guy.
>
>
> 
> Guy Corbaz
> ch. du Châtaignier 2
> 1052 Le Mont
> Switzerland
> phone:+41 21 652 26 05
> mobile: +41 79 420 26 06
> e-mail: [EMAIL PROTECTED]
>
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Guy Corbaz
ch. du Châtaignier 2
1052 Le Mont
Switzerland
phone:+41 21 652 26 05
mobile: +41 79 420 26 06
e-mail: [EMAIL PROTECTED] 


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Re: [asterisk-users] intel vs amd motherboards

[Tzafrir Cohen]

On Sun, Jul 09, 2006 at 05:07:16AM -0400, C F wrote:
> Tzafrir, are you trying to tell me that I can realy do double on the
> intel becuase the second CPU will do it?

In the ideal case you'll get double performance with two CPUs. In
theory.

A case of many concurrent calls is basically something that can be
easily parallelized. So in theory nothing stops you from getting
something closer to double performance. I don't know how close reality
is to that nice theory.

I only remarked that 'show translations' totally ignores the second CPU.

-- 
Tzafrir Cohen  sip:[EMAIL PROTECTED]
icq#16849755   iax:[EMAIL PROTECTED]
+972-50-7952406   
[EMAIL PROTECTED]  http://www.xorcom.com
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Re: [asterisk-users] Freeware sip/iax client windows mobile

[Administrator TOOTAI]

Attilla De Groot wrote:

Hi all,


I have two pda's and I want to be able to make calls, but I need a 
client for this. The only problem is Windows Mobile 5.0, I can't find 
a freeware client for this, the only one is Sjphone. But this one is 
still beta for windows mobile and it just doesn't work good.


Does anyone have an alternative ?

I'm using ppciax

--
Daniel
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Re: [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?

[Rich Adamson]

Florian Overkamp wrote:

Michiel van Baak wrote:

If you buy a model without the "spare" in it's name, you
have the license to use them right ?


To use them with a CCM or CCME, yes :-)


How about secondhand phones you get from ebay ?
Is my cisco smartnet account enough to run the phone legally
? It's not a spare model (at least that was not in the deal
description)


My understanding is, if you have any license at all, Cisco will probably 
not bother you. But it is most definitely not the way they intended :-)


The bottom line are the words in the Cisco license agreement that 
essentially says none of their firmware licenses are transferable. A 
purchase from Ebay would be an attempt to transfer the license and 
therefore illegal.


There are some Cisco authorized resellers around that do sell used (or 
in Cisco terms, reconditioned) phones. Some apparently have the 
capability to bundle smartnet contracts with the phones, but they are 
not living up to the actual words in the license.


The flip side of this... how many people would it take for Cisco to 
inventory and enforce their written licenses throughout the world?


Another side issue... the Cisco phones have no way to "remove" the 
installed firmware. Therefore, there is no way to legally sell a used 
Cisco phone, period.



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Re: [asterisk-users] What's the story with X10*P FXO cards?

[Rich Adamson]

It looks like the X101P clones I bought from eBay are dogs, so I'll 
look into buying some FXO-SIP box instead. Hopefully, I won't have the 
same problems with static, or caller ID and call termination not being 
detected.


Still, considering the number of people having similar problems with 
those cards, I was wondering what the problem is. Is it because the 
hardware, no matter what is advertised, is actually not identical from 
card to card so the zaptel driver doesn't work reliably unless they are 
among the few remaining authentic cards made by Digium before it stopped 
manufacturing them? Because they're actually voice softmodems, and 
hence, very sensitive to the  computers in which they're installed 
(voltage on the PCI slot, sharing IRQ's, etc.)? Other reasons?


The X100P-style cards were never manufactured by Digium; they were 
simply analog modem cards that happened to use integrated circuits with 
some voice encoding/decoding functionality. The card was manufactured 
overseas using a chipset from Silicon Labs, and were fairly popular back 
in the pre-broadband analog-modem days. Digium wrote the asterisk 
drivers to take advantage of the functionality within the chipset.


The Silicon Labs chipset used on the card was designed to meet US 
Telephone standards, and several competing analog modem manufactures 
designed their cards using Motorola, Intel, and/or other chipsets. Each 
chipset has its own unique driver requirements in terms of initializing 
the chips, moving data, timing, etc. Also keep in mind the card designs 
were mostly completed back in the days of PCI v1 standards.


Some of the cards sold on Ebay are those that use the Silicon Labs 
chipsets while others are obviously based on the Motorola chipset. The 
two are not interchangeable without modifying the asterisk drivers.


As mentioned, the Silicon Labs chipset used on the card were 
manufactured to US telephone standards (eg, 600 ohm impedance). Other 
countries have different standards (guessing, maybe 30-to-50 different 
telephony standards), and attempting to use the x100p in those 
environments represents electrical mismatches resulting in echo, no 
callerid support, and many other objectionable or non-working conditions 
(depending upon exactly which country you were in).


Are some of the Ebay ad's misrepresented? Probably.

You're likely to become just about as frustrated with the fxo-sip 
gateway boxes on the market. Expensive ones have excellent echo 
cancellation while the cheaper ones are rather poor (or unusable). Some 
will accept incoming pstn calls while others basically only support 
outgoing calls (intended for certain failed conditions). Some are 
targeted for the US market (telephony standards) while others support a 
larger subset. Some work well on long pstn loops while lots of them 
don't work very well at all.


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Re: [asterisk-users] intel vs amd motherboards

[C F]

Tzafrir, are you trying to tell me that I can realy do double on the
intel becuase the second CPU will do it?

On 7/7/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

On Thu, Jul 06, 2006 at 03:32:04PM -0400, C F wrote:
> I have recently build 2 machines, one with an Intel Pentium Dual Core
> CPU, and one SATA HDD, and the other with a single AMD 64 bit CPU, and
> a RAID 1 w 2 SATA HDD. Both costed the same even though the AMD had 2
> HDDs. Here are the show translations from both:
>
> Intel Dual Core machine:
> pbx*CLI> show translation
> Translation times between formats (in milliseconds)
>  Source Format (Rows) Destination Format(Columns)
>
> g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
>   g723 - - - - - - - - - - -
>gsm - - 2 2 2 2 1 517 -17
>   ulaw - 2 - 1 2 2 1 517 -17
>   alaw - 2 1 - 2 2 1 517 -17
>   g726 - 2 2 2 - 2 1 517 -17
>  adpcm - 2 2 2 2 - 1 517 -17
>   slin - 1 1 1 1 1 - 416 -16
>  lpc10 - 3 3 3 3 3 2 -18 -18
>   g729 - 4 4 4 4 4 3 7 - -19
>  speex - - - - - - - - - - -
>   ilbc - 3 3 3 3 3 2 618 - -
>
> AMD 64 bit machine:
> pbx*CLI> show translation
> Translation times between formats (in milliseconds)
>  Source Format (Rows) Destination Format(Columns)
>
> g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
>   g723 - - - - - - - - - - -
>gsm - - 2 2 2 2 1 313 -12
>   ulaw - 3 - 1 2 2 1 313 -12
>   alaw - 3 1 - 2 2 1 313 -12
>   g726 - 3 2 2 - 2 1 313 -12
>  adpcm - 3 2 2 2 - 1 313 -12
>   slin - 2 1 1 1 1 - 212 -11
>  lpc10 - 3 2 2 2 2 1 -13 -12
>   g729 - 4 3 3 3 3 2 4 - -13
>  speex - - - - - - - - - - -
>   ilbc - 4 3 3 3 3 2 414 - -
>
>
> This shows that the AMD 64 bit is worth much more than just the price
> difference.

It shows that the AMD CPU performs better than each of the Intel CPUs
separately: each such translation is inherently a single task eprformed
by a single CPU.

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Re: [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?

[Florian Overkamp]

Michiel van Baak wrote:

If you buy a model without the "spare" in it's name, you
have the license to use them right ?


To use them with a CCM or CCME, yes :-)


How about secondhand phones you get from ebay ?
Is my cisco smartnet account enough to run the phone legally
? It's not a spare model (at least that was not in the deal
description)


My understanding is, if you have any license at all, Cisco will probably 
not bother you. But it is most definitely not the way they intended :-)


Florian
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Re: [asterisk-users] What's the story with X10*P FXO cards?

[trixter aka Bret McDanel]

On Sun, 2006-07-09 at 10:22 +0200, Vincent Delporte wrote:
> Still, considering the number of people having similar problems with those 
> cards, I was wondering what the problem is. Is it because the hardware, no 
> matter what is advertised, is actually not identical from card to card so 
> the zaptel driver doesn't work reliably unless they are among the few 
> remaining authentic cards made by Digium before it stopped manufacturing 
> them? 

digium didnt really make em.  as to the reliability, some of that
depends greatly on what chipset is on the card itself.  There are a
couple different ones that while cost about $5 couldnt be resold for
$100 for good reason.


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[asterisk-users] What's the story with X10*P FXO cards?

[Vincent Delporte]

Hi

	It looks like the X101P clones I bought from eBay are dogs, so I'll look 
into buying some FXO-SIP box instead. Hopefully, I won't have the same 
problems with static, or caller ID and call termination not being detected.


Still, considering the number of people having similar problems with those 
cards, I was wondering what the problem is. Is it because the hardware, no 
matter what is advertised, is actually not identical from card to card so 
the zaptel driver doesn't work reliably unless they are among the few 
remaining authentic cards made by Digium before it stopped manufacturing 
them? Because they're actually voice softmodems, and hence, very sensitive 
to the  computers in which they're installed (voltage on the PCI slot, 
sharing IRQ's, etc.)? Other reasons?


Thank you
VD.


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Re: [asterisk-users] CallerID

[Rich Adamson]

I have 2 POTs line coming into Asterisk. We have callerid feature from 
Verizon on one of the lines.


I am not able to track any CallerID coming in, in the log. I am pretty 
green with asterisk, and it's not clear if I have to activate for 
CallerID in the dialplan. The voicemail keeps saying " call from an 
unknown caller " etc.
Eventually, i would like to pass on the callerID and name to a pager, if 
the call is not picked up, at the extension, after hanging up the call.


Assuming you are trying to use an X100P or Digium TDM analog card, 
you'll probably want to start with zapata.conf entries something like this:

context=inbound-home
usecallerid=yes
signalling=fxs_ks
faxdetect=no
callerid=asreceived
echocancel=yes
usecallerid=yes
hidecallerid=no
echocancelwhenbridged=yes
rxgain=7
txgain=5
channel => 1

The rxgain and txgain values will need to be tweaked to "your" pstn 
lines, and the values will be highly dependent upon exactly how far you 
are from your Verizon central office. Gains that are to high or to low 
can impact how well callerid info is received.


In your extensions.conf, you might want something like this:
[inbound-home]
exten => s,1,NoOp,${CALLERID(all)}
exten => s,2,Dial(${PHONE3}&${PHONE4})

If you start asterisk from the command like (eg, asterisk -rvv), 
you will see the incoming callerid displayed on the command line because 
of the "NoOP" statement shown above.



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Re: [asterisk-users] CallerID

[Wilson Pickett]

On 7/9/06, Ryder Brook <[EMAIL PROTECTED]> wrote:

I have 2 POTs line coming into Asterisk. We have callerid feature from
Verizon on one of the lines.


What interface are the lines connected to?


I am not able to track any CallerID coming in, in the log. I am pretty green
with asterisk, and it's not clear if I have to activate for CallerID in the
dialplan. The voicemail keeps saying " call from an unknown caller " etc.


For zaptel interface, the configuration is in zapata.conf and
/etc/zaptel.conf. Things like usecallerid=yes are done there. As to
your "greenness", everyone starts there. One suggestion is to read
available books that systematically take you through which config
files do what, such as http://asteriskdocs.org

Another is to look at the info  in /usr/src/asterisk/doc/ and the
sample config files which theoretically each contain all available
statements for that file.

Eveyone complains that the existing docs are not accurate. Asterisk
being a moving target, you can only hope to get a basic understanding
reading most of these, but it's well worth the trouble.
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Re: [asterisk-users] Suggesstion Required

[Rich Adamson]

You might want to look around the wiki (www.voip-info.org) as most of 
your questions have answers there.



I want to setup asterisk box to do the following jobs.

1- 100 cuncurent calls
2- 1000 User Registration
3- MySQL Realtim
4- PerlAGI

Here is my question could u please reply it:

1- No RTP only singnaling, Is it possible?
Ans:


Yes, but probably not 100% likely. If any of the 1000 users have their 
equipment behind firewall/nat boxes, you're likely to end up handling 
rtp / transcoding. Also, depending on the exact equipment used by the 
1000 users, not all devices support g729. And, any local calls 
terminating via analog or PRI pstn lines will likely be g711 only.



2- How much RAM?
Ans:


Since memory is cheap, through a couple of gig at it.


3- How much bandhwidth per month with G729
Ans:


Rough answer... about 30 kbps of bandwidth per g729 call. Therefore, 100 
simultaneous calls times 30,000 equals 3,000,000 bps. (That's roughly 
two T1's, or 30% of a 10 meg ethernet, etc.)



4- Proccessor?
Ans:


Any... really doesn't make that big of a difference whether its AMD or 
Intel. Plenty of examples on the wiki.



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