date:20060530





I'm interested too to know about a quad E1 card...

I need to connect it to 2 differents ISDN providers in Europe and to
establish a third connection 
with a Matra PBX.
The server ( IBM XSeries 346 ) has to serve about 60-70 simultaneous
calls ( IVR and max 30 conferences... )

I will also need ( but later, as I think I'll have to write it )
support for videoconferencing over ISDN using different protocols like
h320 or h324m...

What would you recommend ?

Digium TE411P, Sangoma A104D, Eicon Diva Cards ?


Armin Schindler a écrit :

  On Tue, 30 May 2006, olivier.taylor wrote:
  
  
Hi all,

I need your lights :)

There are many hardware provider for E1 cards on the market, what's your
exeperience with E1 and what's the preferred provider for Asterisk out of
Digium?

  
  
I prefer Eicon Diva Server cards, they have good features and are very 
reliable.

Armin
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Re: [Asterisk-Users] I can't call PSTN numbers

2006-05-30 Thread Woodoo People .pGa!

autocreatepeer=yes

[ser_box1]
type=peer
username=ser_box1
insecure=yes
canreinvite=no
context=from-internal
host=ip.address.of.box
nat=yes
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm


 
> It doesn't work for me :-(
> How do you have the peer configuration in asterisk, to connect ot SER?
> 
> >exten => _4XX,1,Dial(SIP/[EMAIL PROTECTED])
> >
> >it works to me (my provider sends me the last 3 digits)
> >
> >> I hava SER with many clients (sipura SPA2100). One of these is an
> >> Asterisk which have others clients (sipuraSPA2100).
> >> I also have a Cisco GW which give me access to the PSTN.
> >> I make calls to all IP phones in my network, but I can't call PSTN
> >> numbers. After I dial, I hear 2 ringbacks but at the same time
> >> Asterisk says:
> >>
> >> Called [EMAIL PROTECTED]
> >> SIP/SER_ip_address-ec75 is circuit-busy
> >> Everyone is  busy/congested at this time (1:0/1/0)

-- 
WoodOO-[P]an[G]alaktikan[A]gent-People <][> http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
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Re: [Asterisk-Users] E1 hardware for asterisk

On Tue, 30 May 2006, Tristan wrote:
> I'm interested too to know about a quad E1 card...
> 
> I need to connect it to 2 differents ISDN providers in Europe and to establish
> a third connection
> with a Matra PBX.
> The server ( IBM XSeries 346 ) has to serve about 60-70 simultaneous calls (
> IVR and max 30 conferences... )
> 
> I will also need ( but later, as I think I'll have to write it ) support for
> videoconferencing over ISDN using different protocols like h320 or h324m...
> 
> What would you recommend ?
> 
> Digium TE411P, Sangoma A104D, Eicon Diva Cards ?

I cannot tell anything about the Digium or Sangoma cards, but the Eicon Diva 
Server Cards are active cards, which means they do the ISDN protocol stuff 
including digital-signal-processing (if needed) on board without using the 
hosts CPU. So in a setup as you described above, I recommend to use the
Eicon cards.

Armin
 
> Armin Schindler a Ã©crit :
> > On Tue, 30 May 2006, olivier.taylor wrote:
> > 
> > > Hi all,
> > > 
> > > I need your lights :)
> > > 
> > > There are many hardware provider for E1 cards on the market, what's
> > > your
> > > exeperience with E1 and what's the preferred provider for Asterisk
> > > out of
> > > Digium?
> > > 
> > 
> > I prefer Eicon Diva Server cards, they have good features and are very
> > reliable.
> > 
> > Armin
> > ___
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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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Re: [Asterisk-Users] Modules for X100P

2006-05-30 Thread Ed


Hans Witvliet wrote:


Some consultants are not very keen on such boards, as ad/da are done in
software instead of hardare.



software DAC, LOL
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Re: [Asterisk-Users] E1 hardware for asterisk

2006-05-30 Thread olivier.taylor





seems to be very good cards, but also, very expensive, isn't it?

Olivier

Armin Schindler a écrit :

  On Tue, 30 May 2006, Tristan wrote:
  
  
I'm interested too to know about a quad E1 card...

I need to connect it to 2 differents ISDN providers in Europe and to establish
a third connection
with a Matra PBX.
The server ( IBM XSeries 346 ) has to serve about 60-70 simultaneous calls (
IVR and max 30 conferences... )

I will also need ( but later, as I think I'll have to write it ) support for
videoconferencing over ISDN using different protocols like h320 or h324m...

What would you recommend ?

Digium TE411P, Sangoma A104D, Eicon Diva Cards ?

  
  
I cannot tell anything about the Digium or Sangoma cards, but the Eicon Diva 
Server Cards are active cards, which means they do the ISDN protocol stuff 
including digital-signal-processing (if needed) on board without using the 
hosts CPU. So in a setup as you described above, I recommend to use the
Eicon cards.

Armin
 
  
  
Armin Schindler a Ã©crit :


  On Tue, 30 May 2006, olivier.taylor wrote:

  
  
Hi all,

I need your lights :)

There are many hardware provider for E1 cards on the market, what's
your
exeperience with E1 and what's the preferred provider for Asterisk
out of
Digium?


  
  I prefer Eicon Diva Server cards, they have good features and are very
reliable.

Armin
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[Asterisk-Users] sip interopability problem

2006-05-30 Thread jorge werth

   Hi, (I tried to send this to the list earlier, it didn't seem to work- my apologies if you see this twice...)I have two asterisk machines side by side both running debian sarge, one running sarge's version of asterisk (1.0.7.dfsg.1-2) and the other running the version of asterisk from www.backports.org (1:1.2.1.dfsg-2bpo1).  I also have a SIP provider who is routing blocks of DID's to both machines.  The sip.conf is nearly identical on both machines (the general section, and the section for the SIP provider in question are exactly the same).  Calls from this SIP provider to the asterisk 1.2 machine do not work, but work fine to the asterisk 1.0 machine.  When these calls are answered on the 1.2 machine neither end can hear the other, then after a few seconds there is a single beep on the phone at my end (no corresponding sound at the remote
 end), and after a few more seconds both ends get the engaged signal.  Looking at the output from tethereal on the 1.2 machine, the remote SIP provider never sends a SIP ACK response, it just keeps sending INVITE's. On the 1.0 machine the remote end sends an ACK response as the call is answered, and sends no further INVITE's, with the call going through correctly.   As a workaround, I am able to receive calls on the 1.0 machine and forward them to the 1.2 machine, but I would like to retire my old asterisk machine.  What could the problem be?  Have I found a bug in asterisk 1.2?  Thanks, Jorge. 
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Re: [Asterisk-Users] mpg123 or asterisk

2006-05-30 Thread Steve Totaro


It has crashed an SGI Altix 350 on a dialy basis.

MBIT Technologies wrote:

Can MAD crash a server like mpg123 can?




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer
Sent: Tuesday, 30 May 2006 5:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] mpg123 or asterisk

Can't you use mpg123 as compiled under x86_32?  I do on a few servers I
have.  I found madplay better process wise than mpg123.

Regards

Lee 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick
Perez
Sent: 29 May 2006 21:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] mpg123 or asterisk

Well, being unable to compile mpg123 under x86_64 i installed lame and
transformed the mp3-->wav-->raw.
and using "files" as the format player.

Are there any good scripts to stress test MoH?
I want to test this machine for 1000 "calls" on hold.

http://www.asteriskguru.com/tutorials/astertest.html
AsterTest is good but i dont have access to another * installation.


On 5/27/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
  
Please let us know your results.  I cannot really test this in 
production system since it is a $16,000/hr call center.  I was using 
madplay but it was crashing and creating zombie processes, I figured 
native was not the way to go since all of the different audio streams.
Mpg123 works perfectly for me under a load of sixty channels, I can 
confirm that for sure.


Thanks,
Steve

Erick Perez wrote:


Interesting.
So, i will have to test then...


On 5/27/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
  
In my very limited testing of native, each channel was receiving a 
different stream (each caller heard something different).  Under a 
high volume of calls, which is going to hurt performance more?  
Transcoding MP3s but sending a single stream or separate streams 
per call under native?


When I say high, I mean 1,000+ calls.

Thanks,
Steve


Erick Perez wrote:


Thanks to all. Native format will be.

On 5/27/06, Matt Riddell (IT) <[EMAIL PROTECTED]> wrote:
  

Vahan Yerkanian wrote:


Erick Perez wrote:
  
should I use mpg123 with asterisk 1.2.7 or should i use the 
native player asterisk has?

the target machine will receive heavy load.


mpg123 was used back when asterisk didn't have native format
  

support. If


you are expecting heavy load, the native format is the way to
  

go. You

might decide not to use mp3 format at all, recompressing your 
MoH
  

files

using sox to the formats you gonna use, such as .al, .ul, 
.gsm, or
  

leave


it at .sln to cut the decoding leg only.
  

Heh, damn this GPRS connection.  In order to pass the time while



  

downloading messages I reply before they are all in, and yet by


the time


I have received all the messages I note that your question has


already


been answered!

:)

--
Cheers,

Matt Riddell
___

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http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) 
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Re: [Asterisk-Users] E1 hardware for asterisk

On Tue, 30 May 2006, olivier.taylor wrote:
> seems to be very good cards, but also, very expensive, isn't it?

these cards are more expensive than passive cards or other cards with less 
features of course. But these cards are really very powerful and when I get 
feedback from users, I always here: "they are worth every cent" and this is 
what I think too.

Armin
 
> Olivier
> 
> Armin Schindler a Ã©critÂ :
> 
>  On Tue, 30 May 2006, Tristan wrote:
>   
> 
>  I'm interested too to know about a quad E1 card...
> 
> I need to connect it to 2 differents ISDN providers in Europe and to establish
> a third connection
> with a Matra PBX.
> The server ( IBM XSeries 346 ) has to serve about 60-70 simultaneous calls (
> IVR and max 30 conferences... )
> 
> I will also need ( but later, as I think I'll have to write it ) support for
> videoconferencing over ISDN using different protocols like h320 or h324m...
> 
> What would you recommend ?
> 
> Digium TE411P, Sangoma A104D, Eicon Diva Cards ?
> 
> 
>  I cannot tell anything about the Digium or Sangoma cards, but the Eicon Diva 
> Server Cards are active cards, which means they do the ISDN protocol stuff 
> including digital-signal-processing (if needed) on board without using the 
> hosts CPU. So in a setup as you described above, I recommend to use the
> Eicon cards.
> 
> Armin
>  
>   
> 
>  Armin Schindler a ÃÂ©crit :
> 
> 
>  On Tue, 30 May 2006, olivier.taylor wrote:
> 
>   
> 
>  Hi all,
> 
> I need your lights :)
> 
> There are many hardware provider for E1 cards on the market, what's
> your
> exeperience with E1 and what's the preferred provider for Asterisk
> out of
> Digium?
> 
> 
> 
>  I prefer Eicon Diva Server cards, they have good features and are very
> reliable.
> 
> Armin
> ___
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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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> 
> 
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Re: [Asterisk-Users] E1 hardware for asterisk





Seems like there is no quad e1 diva server cards...

Does someone knows about digium and sangoma ?

Tristan

olivier.taylor a écrit :

  
seems to be very good cards, but also, very expensive, isn't it?
  
Olivier
  
Armin Schindler a écrit :
  
On Tue, 30 May 2006, Tristan wrote:
  

  I'm interested too to know about a quad E1 card...

I need to connect it to 2 differents ISDN providers in Europe and to establish
a third connection
with a Matra PBX.
The server ( IBM XSeries 346 ) has to serve about 60-70 simultaneous calls (
IVR and max 30 conferences... )

I will also need ( but later, as I think I'll have to write it ) support for
videoconferencing over ISDN using different protocols like h320 or h324m...

What would you recommend ?

Digium TE411P, Sangoma A104D, Eicon Diva Cards ?



I cannot tell anything about the Digium or Sangoma cards, but the Eicon Diva 
Server Cards are active cards, which means they do the ISDN protocol stuff 
including digital-signal-processing (if needed) on board without using the 
hosts CPU. So in a setup as you described above, I recommend to use the
Eicon cards.

Armin
 
  

  Armin Schindler a Ã©crit :

  
On Tue, 30 May 2006, olivier.taylor wrote:

  

  Hi all,

I need your lights :)

There are many hardware provider for E1 cards on the market, what's
your
exeperience with E1 and what's the preferred provider for Asterisk
out of
Digium?



I prefer Eicon Diva Server cards, they have good features and are very
reliable.

Armin
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[Asterisk-Users] I guess my server capacity is ok

2006-05-30 Thread Goke Aruna

can someone overthere help?the server specs are as follows
HP DL380G4 Dual Intel Xeon 3.2GHz processor with 4GB RAM, 
running fedora core 3
asterisk-1.2.5
ss7-0.8.3d.
using sip as advised to receive calls from another gateway in US.
using g729 in transcoding way.

however, I noticed the call hit the 51 active calls which is
102channels, I run "top" to check the system resources usage and i
discovered that the cpu is 100% used. asterisk, sip, ss7  never crashed
throughout.

however, since transcoding takes alot of system resources.. how can I use g729 in passthru mode.

and I guess disabling hyperthreading will save me more system resouces.

I will be glad, if you can give me more info on system management cos i
think with that system, it should able to handle at least five E1's.
I say thank you for finding time to reply my mail.

goksie
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Re: [Asterisk-Users] E1 hardware for asterisk





Sorry, asked the wrong provider, I just looked at the official website,
I was wrong...



Tristan a écrit :

  
Seems like there is no quad e1 diva server cards...
  
Does someone knows about digium and sangoma ?
  
Tristan
  
olivier.taylor a écrit :
  

seems to be very good cards, but also, very expensive, isn't it?

Olivier

Armin Schindler a écrit :

  On Tue, 30 May 2006, Tristan wrote:
  
  
I'm interested too to know about a quad E1 card...

I need to connect it to 2 differents ISDN providers in Europe and to establish
a third connection
with a Matra PBX.
The server ( IBM XSeries 346 ) has to serve about 60-70 simultaneous calls (
IVR and max 30 conferences... )

I will also need ( but later, as I think I'll have to write it ) support for
videoconferencing over ISDN using different protocols like h320 or h324m...

What would you recommend ?

Digium TE411P, Sangoma A104D, Eicon Diva Cards ?

  
  
I cannot tell anything about the Digium or Sangoma cards, but the Eicon Diva 
Server Cards are active cards, which means they do the ISDN protocol stuff 
including digital-signal-processing (if needed) on board without using the 
hosts CPU. So in a setup as you described above, I recommend to use the
Eicon cards.

Armin
 
  
  
Armin Schindler a Ã©crit :


  On Tue, 30 May 2006, olivier.taylor wrote:

  
  
Hi all,

I need your lights :)

There are many hardware provider for E1 cards on the market, what's
your
exeperience with E1 and what's the preferred provider for Asterisk
out of
Digium?


  
  I prefer Eicon Diva Server cards, they have good features and are very
reliable.

Armin
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Re: [Asterisk-Users] sip interopability problem

2006-05-30 Thread Mr shobhit nirala

Hi All My company has vacancy for asterisk professional We are an INDIAN base comapny if any one interested please send mail your resume to [EMAIL PROTECTED] with number subject line --> Asterisk Resume (yrs of exp) DONT MAIL YOUR RESUME ON THIS MAIL ID
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[Asterisk-Users] sIp port numbers

2006-05-30 Thread bails

Hi all I fancied playing with SER and * on the same box. So i thought 
i'd just change the default sip port for * in sip.conf


[general]

port = 5065   ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)

restarted * and now when i issue a


]# netstat -anp |grep 5060
udp0  0 0.0.0.0:50600.0.0.0:*   
9453/asterisk


Its still on port 5060?

asterisk -V

Asterisk 1.2.1


btw  i did a grep -r 5060 * in /etc/asterisk and found only the 1 instance.

Any ideas

Thanks in advance

Bails
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[Asterisk-Users] Asterisk restarting in a minute

2006-05-30 Thread Woodoo People .pGa!

Hi!

Probably any of you meet with the following problem:
asterisk is restarting in a minute (if no active call) if active call,
it says cannot receive a call due to restart in progress.

even if i starting with -c, i have no disconnected, but see the stuff
restarting.

i've tried to recompile, older version, virgin config, etc. same results.
it's happened after a power loss, on a ext3 fs, sitting on a raid1.
astdb was deleted, log is not showing any interesting things.

any ideas please?
-- 
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[EMAIL PROTECTED]@RedHat.users
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Re: [Asterisk-Users] E1 hardware for asterisk

Tristan schrieb:
> Does someone knows about digium and sangoma ?

A lot ;-) What do you want to know?

Christian
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Re: [Asterisk-Users] sIp port numbers

2006-05-30 Thread Thomas Kenyon

bails wrote:
> Hi all I fancied playing with SER and * on the same box. So i thought
> i'd just change the default sip port for * in sip.conf
>
> [general]
>
> port = 5065   ; Port to bind to (SIP is 5060)
> bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
>
> restarted * and now when i issue a
>
>> ]# netstat -anp |grep 5060
>> udp0  0 0.0.0.0:5060   
>> 0.0.0.0:*   9453/asterisk
>
> Its still on port 5060?
>
> asterisk -V
>> Asterisk 1.2.1
>
> btw  i did a grep -r 5060 * in /etc/asterisk and found only the 1
> instance.
>
Which instance is that?
You know you can only have one port declaration in sip.conf. (which may
now be called bindport since 1.2).

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Re: [Asterisk-Users] E1 hardware for asterisk

Tristan schrieb:
> What would you recommend ?
> 
> Digium TE411P, Sangoma A104D, Eicon Diva Cards ?

Ah - I should have read this bevor my last answer. ;-)

I personally prefer the Sangoma E1 cards. The work in almost every PCI
system and the echo cancel - if you really need it - is far better than
the one provided by the Digium cards. Don't know about Eicons echo
cancel as I never used one.

I hope that helped. Feel free to ask me anything you like to know.

Chris
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[Asterisk-Users] Job Opening for asterisk Proff

2006-05-30 Thread Mr shobhit nirala

  Hi All My company has vacancy for asterisk professional We are an INDIAN base comapny if any one interested please send mail your resume to [EMAIL PROTECTED] or [EMAIL PROTECTED] with number subject line --> Asterisk Resume (yrs of
 exp) DONT MAIL YOUR RESUME ON THIS MAIL ID  SHOBHIT NIRALA CONT NO. 9871476403
		Do you Yahoo!? 
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Re: [Asterisk-Users] E1 hardware for asterisk





I need different opinions about these cards to be sure about the one to
buy 
because the server must be up 24/24...

What would you recommand for my needs ?

I need to connect the card to 2 differents ISDN providers in Europe (
EURO-ISDN ) and to connect also with a Matra PBX ( Maybe QSIG ).
I had warnings about clock issues with my current hardware ( Dialogic
stuff )

This server ( an IBM X-Series 346 dual 3Ghz Xeon with 2gb ram ) has to
serve about 60-70 incoming/outgoing PSTN simultaneous calls ( IVR and
max 30-40 conferences... ) and about 10-20 SIP calls to begin...

I will also need support for videoconferencing over ISDN ( but later,
as I think I'll have to write the h320 and H324m stuff ) 

My first choise was a Digium TE411P but Sangoma A104d seems to be quite
good too and now that I found the Diva Server V-4PRI I don't know
anymore the one that would fit my needs the most...

I think that the anti echo module is a must as  I don't want to
overload this critical server, Am I right ?

Christian Victor a écrit :

  Tristan schrieb:
  
  
Does someone knows about digium and sangoma ?

  
  
A lot ;-) What do you want to know?

Christian
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Re: [Asterisk-Users] Problem with IAX2 dialin with portunity

2006-05-30 Thread Matthias Fechner

Hello Joshua,

Joshua Colp wrote:
>> [portunity-out]
>> type=friend
>> host=iax.iaxport.de
>> username=XXX
>> secret=YY
>> context=incoming-portunity
>> notransfer=yes
> Only if the username is specified as portunity-out when the other side dials
> you. Otherwise your Asterisk has no idea what to authenticate them as so it
> takes a guess and in the end settles on guest.

But should not asterisk here see, that the call is comming in from the
host: host=iax.iaxport.de or from the username=iaxXX?
In the SIP configuration I do it this way.

Or need I to define some other parameters in the section
[portunity-out] or easily rename it.

If I get a call, asterisk says the following: (I hope everything is in :) )
___CUT___
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
   Timestamp: 9ms  SCall: 5  DCall: 0 [82.139.223.1:4569]
   VERSION : 2
   CALLED NUMBER   : matthiasfechner
   CODEC_PREFS : (ulaw|alaw|gsm)
   CALLING NUMBER  : [EMAIL PROTECTED]
   CALLING PRESNTN : 1
   CALLING TYPEOFN : 0
   CALLING TRANSIT : 0
   CALLING NAME: Matthias Fechner
   LANGUAGE: de
   USERNAME: iaxX
   FORMAT  : 14
   CAPABILITY  : 63502
   ADSICPE : 0
   DATE TIME   : 2006-05-30  11:50:48

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
   Timestamp: 4ms  SCall: 00084  DCall: 5 [82.139.223.1:4569]
   AUTHMETHODS : 3
   CHALLENGE   : 182242807
   USERNAME: iaxX

Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
AUTHREP
   Timestamp: 00032ms  SCall: 5  DCall: 00084 [82.139.223.1:4569]
   MD5 RESULT  : 872efd005c628f31f74c2b142ca05cb5

Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK
   Timestamp: 00028ms  SCall: 22848  DCall: 4 [192.168.0.151:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACCEPT
   Timestamp: 00022ms  SCall: 00084  DCall: 5 [82.139.223.1:4569]
   FORMAT  : 14

-- Call accepted by 82.139.223.1 (format unknown)
-- Format for call is (gsm|ulaw|alaw)
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK
   Timestamp: 00022ms  SCall: 5  DCall: 00084 [82.139.223.1:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
   Timestamp: 00018ms  SCall: 00049  DCall: 0 [82.139.223.1:4569]
   VERSION : 2
   CALLED NUMBER   : s
   CALLING NUMBER  : [EMAIL PROTECTED]
   CALLING NAME: Matthias Fechner
   LANGUAGE: de
   FORMAT  : 2
   CAPABILITY  : 64798
   ADSICPE : 0
   DATE TIME   : 2006-05-30  11:50:50

Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
   Timestamp: 00018ms  SCall: 6  DCall: 00049 [82.139.223.1:4569]
-- Accepting UNAUTHENTICATED call from 82.139.223.1:
   > requested format = gsm,
   > requested prefs = (),
   > actual format = ulaw,
   > host prefs = (ulaw|alaw|gsm),
   > priority = mine
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACCEPT
   Timestamp: 8ms  SCall: 6  DCall: 00049 [82.139.223.1:4569]
   FORMAT  : 4
___CUT___

Best regards,
Matthias Fechner

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Re: [Asterisk-Users] E1 hardware for asterisk

On Tue, 30 May 2006, Christian Victor wrote:
> Tristan schrieb:
> > What would you recommend ?
> > 
> > Digium TE411P, Sangoma A104D, Eicon Diva Cards ?
> 
> Ah - I should have read this bevor my last answer. ;-)
> 
> I personally prefer the Sangoma E1 cards. The work in almost every PCI

sorry to ask that, but what does "almost every PCI system" mean?

Armin

> system and the echo cancel - if you really need it - is far better than
> the one provided by the Digium cards. Don't know about Eicons echo
> cancel as I never used one.
> 
> I hope that helped. Feel free to ask me anything you like to know.
> 
> Chris
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[Asterisk-Users] problem about asterisk realtime.

2006-05-30 Thread 应芳吴

hi,   Longing for your help. I came into a problem ,Now I want to configure asterisk sip peers from MYSQL database dynamic, flolling the introduction of asterisk realtime,i set the cofiguration of sip users,but I need to cofigure sip peers too.  Where I can find some infomation about cofiguring sip peers? What is the difference of configuration sip peers and configuration sip users?  I also would like to know the responding table fields provided by asterisk for realtime function?  Thanks a lot!!  sharon
		抢注雅虎免费邮箱-3.5G容量，20M附件！ ___
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Re: [ONTP.NET - SPAM] [Asterisk-Users] problem about asterisk realtime.

2006-05-30 Thread Filip Drągowski





http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf

  hi,
   Longing for your help.
 I came into a problem ,Now I want to configure asterisk sip 
peers from MYSQL database dynamic, flolling the introduction of
asterisk 
realtime,i set the cofiguration of sip users,but I need to cofigure sip
  
peers too.
  Where I can find some infomation about cofiguring sip peers? 
What is the difference of configuration sip peers and configuration sip
  
users?
  I also would like to know the responding table fields 
provided by asterisk for realtime function?
  Thanks a lot!!
  sharon
   
  ÇŔ×˘ŃĹť˘ĂâˇŃÓĘĎä-3.5GČÝÁżŁŹ20M¸˝źţŁĄ
  

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Re: [Asterisk-Users] E1 hardware for asterisk

2006-05-30 Thread Wolfgang Zweimueller

Tristan <[EMAIL PROTECTED]> writes:

> This server ( an IBM X-Series 346 dual 3Ghz Xeon with 2gb ram ) has to
> serve about 60-70 incoming/outgoing PSTN simultaneous calls ( IVR and
> max 30-40 conferences... ) and about 10-20 SIP calls to begin...

I tried the Quad-port Digium cards in this special machine and it
crashed the machine! I don't know why it happened but it did it again
and again.

Now I have a Sangoma A104 and it looks O.K. There is an issue with
CPU-load but as far as I can see it's no problem. We will connect this
machine to the telco line the next days.

cu,
Wolfgang
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RE: [Asterisk-Users] E1 hardware for asterisk

2006-05-30 Thread Asterisk

Wolfgang: What kind of CPU-load issue on A104? Could you give me a link or 
something? I also use A104 and it works very good, but recently I noticed a 
behavior which is maybe connected with this issue, so more info would be very 
helpful for me :)

Thanks!

Alex

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolfgang 
Zweimueller
Sent: Tuesday, May 30, 2006 12:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] E1 hardware for asterisk

Tristan <[EMAIL PROTECTED]> writes:

> This server ( an IBM X-Series 346 dual 3Ghz Xeon with 2gb ram ) has to
> serve about 60-70 incoming/outgoing PSTN simultaneous calls ( IVR and
> max 30-40 conferences... ) and about 10-20 SIP calls to begin...

I tried the Quad-port Digium cards in this special machine and it
crashed the machine! I don't know why it happened but it did it again
and again.

Now I have a Sangoma A104 and it looks O.K. There is an issue with
CPU-load but as far as I can see it's no problem. We will connect this
machine to the telco line the next days.

cu,
Wolfgang
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[Asterisk-Users] Re: E1 hardware for asterisk

2006-05-30 Thread Sebastian Kayser

* Christian Victor <[EMAIL PROTECTED]> wrote:
> Tristan schrieb:
> > What would you recommend ?
> > 
> > Digium TE411P, Sangoma A104D, Eicon Diva Cards ?
> 
> Ah - I should have read this bevor my last answer. ;-)
> 
> I personally prefer the Sangoma E1 cards. The work in almost every PCI
> system and the echo cancel - if you really need it - is far better than
> the one provided by the Digium cards. Don't know about Eicons echo
> cancel as I never used one.

Regarding echo cancel. Is there someone with hands-on experience
regarding the echo canceller performance of the Junghanns E1 cards
compared to for example the Sangoma ones?

http://www.junghanns.net/en/singleE1_produkt.html
http://www.junghanns.net/en/doubleE1_produkt.html

- Sebastian
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Re: [Asterisk-Users] E1 hardware for asterisk

2006-05-30 Thread Wolfgang Zweimueller

Hi Alex,

"Asterisk" <[EMAIL PROTECTED]> writes:

> Wolfgang: What kind of CPU-load issue on A104? Could you give me a
> link or something? I also use A104 and it works very good, but
> recently I noticed a behavior which is maybe connected with this
> issue, so more info would be very helpful for me :)

There's a thread regarding this issue in this Mailing list. First
Msg-id is: <[EMAIL PROTECTED]>, Subject
is: "Experience with IBM X346 machines and Sangoma"

This machine is not connected to any line at the moment. So I can not
say if this issue makes a real problem. I can give more info next
week.

What did you observe on your machine?

cu,
Wolfgang
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[Asterisk-Users] Panasonic PBX

2006-05-30 Thread Chris Sutton









The place I currently work at has a Panasonic Key system
with 9 extensions, and no voicemail.  It services 2 PSTN lines.  

 

I am hoping to use Asterisk to host voicemail (I would like
to use the IVR also, but I don’t even know if or how it would work).  

 

Do I need to use a PRI between the two, or is there a simple
solution?  I would like people to be able to answer the phone and transfer the
call to voicemail if the person is not there, or after so many rings, it goes
right to voicemail.  I’m not sure what is needed?  I have seen the
integration How-To but that requires the PRI, and wasn’t sure if that was
the ONLY way to go.  

 

Thanks!

 

Chris

 






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Re: [Asterisk-Users] Panasonic PBX

2006-05-30 Thread Woodoo People .pGa!

> The place I currently work at has a Panasonic Key system with 9 extensions,
> and no voicemail.  It services 2 PSTN lines.  
> 
>  
> 
> I am hoping to use Asterisk to host voicemail (I would like to use the IVR
> also, but I don't even know if or how it would work).  
> 
>  
> 
> Do I need to use a PRI between the two, or is there a simple solution?  I
> would like people to be able to answer the phone and transfer the call to
> voicemail if the person is not there, or after so many rings, it goes right
> to voicemail.  I'm not sure what is needed?  I have seen the integration
> How-To but that requires the PRI, and wasn't sure if that was the ONLY way
> to go.  

i don't think it's the only way. there is no logical difference between
Zap/g0 (includes channel 1-15 of pri) than Zap/g0 (includes bri1-1 and bri1-2
to bri4-1 and bri4-2 if using bristuff) or NTPorts (includes port1,2,3,4 using
mISDN)

keep in mind, the cheapest FXS is a port ATA
but if call-status is a must, and more than 8 channels, go for PRI


-- 
WoodOO-[P]an[G]alaktikan[A]gent-People <][> http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
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RE: [Asterisk-Users] E1 hardware for asterisk

2006-05-30 Thread Asterisk

Thanks a lot Wolfgang!

I use Dell PE2850, so probably this issue does not directly affect my system. 
But I will read thread anyway.

What is happening is that under higher load (60 calls, for example) the 
Asterisk sometimes stops responding in a time manner (AMI messages are delayed, 
etc). Sometimes also D-Channel goes down and up again.  

But I'm not sure if this is Sangoma related problem or Asterisk locking issue. 

Regards!

Alex

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolfgang 
Zweimueller
Sent: Tuesday, May 30, 2006 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] E1 hardware for asterisk

Hi Alex,

"Asterisk" <[EMAIL PROTECTED]> writes:

> Wolfgang: What kind of CPU-load issue on A104? Could you give me a
> link or something? I also use A104 and it works very good, but
> recently I noticed a behavior which is maybe connected with this
> issue, so more info would be very helpful for me :)

There's a thread regarding this issue in this Mailing list. First
Msg-id is: <[EMAIL PROTECTED]>, Subject
is: "Experience with IBM X346 machines and Sangoma"

This machine is not connected to any line at the moment. So I can not
say if this issue makes a real problem. I can give more info next
week.

What did you observe on your machine?

cu,
Wolfgang
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RE: [Asterisk-Users] hint priority and realtime

2006-05-30 Thread Damon Estep

I will give it a try, thank you.

Do you get ringing and on the phone statuses on the subscribed
extensions? What kind of phones are you using?

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jason Bachman
> Sent: Friday, May 26, 2006 2:34 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] hint priority and realtime
> 
> I put my hints in a separate static context, then set the
> subscribecontext in sip.conf to make subscriptions look at that
context
> for hints. Perhaps that would work for you?
> 
> -Jason
> 
> Damon Estep wrote:
> >
> > Can someone shed some light on why the 'hint' feature was
implemented
> > in the 'priority' field that is purely an integer in the rest of the
> > dialplan?
> >
> > There seems to be a conflict with realtime and the hint priority, in
> > order to put in the hints you would have to change the priority
column
> > in the database from int to char and give up some performance (since
> > int indexes better and priority is a parameter in the select)?
> >
> > More importantly, can anyone answer these questions;
> >
> > Can the hint priority by put in mysql realtime?
> >
> > Is there truly an impact to changing the priority datatype to char
or
> > varchar?
> >
> > If it can not be put in realtime, can the hint priority exist in the
> > same context statically, and the numbered portion of the dialplan in
> > realtime? (making it "not so real time")
> >
> > Thanks for any info on this.
> >
> >

> >
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Re: [Asterisk-Users] I can't call PSTN numbers

2006-05-30 Thread Sebastian Milioto

I run ngrep when I call an IP number and when I call a PSTN number,
and the sequece is like that:

For PSTN Numbers:
Sipura ---> Asterisk  (Invite pstn number)
Asterisk--->Sipura(407 Proxy Auth. Required)
Sipura ---> Asterisk (Ack)
Sipura ---> Asterisk (Invite with Proxy Auth.)
Asterisk--->Sipura (100 Trying)
Asterisk--->SER (Invite pstn number)
Asterisk--->Sipura (180 Ringing)
SER> Asterisk (Trying)
SER> Asterisk (404 NOT FOUND)

For IP Numbers:
It is identical to the sequence above, but I get the following instead
"NOT FOUND":

SER>Asterisk (180 Ringing)
SER>Asterisk (200 OK)
Asterisk>SER (Ack)
Asterisk--->Sipura (OK)
Sipura> Asterisk (Ack)

Then the call is established. So.. do you definitely think it is a SER
configuration issue rather than Asterisk configuration issue?

However, when I log in with a Sipura directly into SER, I can get
access to all PSTN numbers. So why not with Asterisk?. I can't find
anything different.

Thanks again for your help

Sebastian

On 5/30/06, Woodoo People .pGa! <[EMAIL PROTECTED]> wrote:

autocreatepeer=yes

[ser_box1]
type=peer
username=ser_box1
insecure=yes
canreinvite=no
context=from-internal
host=ip.address.of.box
nat=yes
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm

> It doesn't work for me :-(
> How do you have the peer configuration in asterisk, to connect ot SER?
>
> >exten => _4XX,1,Dial(SIP/[EMAIL PROTECTED])
> >
> >it works to me (my provider sends me the last 3 digits)
> >
> >> I hava SER with many clients (sipura SPA2100). One of these is an
> >> Asterisk which have others clients (sipuraSPA2100).
> >> I also have a Cisco GW which give me access to the PSTN.
> >> I make calls to all IP phones in my network, but I can't call PSTN
> >> numbers. After I dial, I hear 2 ringbacks but at the same time
> >> Asterisk says:
> >>
> >> Called [EMAIL PROTECTED]
> >> SIP/SER_ip_address-ec75 is circuit-busy
> >> Everyone is  busy/congested at this time (1:0/1/0)

--
WoodOO-[P]an[G]alaktikan[A]gent-People <][> http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
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Re: [Asterisk-Users] Re: E1 hardware for asterisk

> Regarding echo cancel. Is there someone with hands-on experience
> regarding the echo canceller performance of the Junghanns E1 cards
> compared to for example the Sangoma ones?

Well - the Junghanns does the echocancel in software and the Sangoma
A104d does it in hardware.

So on the Sangoma echo cancel does not affect CPU performance while on
the Junghanns it does.

Christian
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[Asterisk-Users] Extensions, devices and dialplan

2006-05-30 Thread Mimmus

Hi,
as already said in others messages on this list, I'm rewriting my dialplan
using AMP/FreePBX as starting point.
I saw that AMP/FreePBX uses the concept of USERS/DEVICES, quite interesting
but not useful to me now. It defines USERS/DEVICES association in AstDB and
then uses dialparties.agi script to dial the right phone. I'd like to drop
this part.
What's the most common method to associate an extension  to SIP/
or IAX/ or Zap/?
Can I use Asterisk variables or is it better to define an AstDB key?
Other suggestions?

Thanks
-- 
Domenico Viggiani

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Re: [Asterisk-Users] E1 hardware for asterisk

Hi Armin,

>> I personally prefer the Sangoma E1 cards. The work in almost every PCI
> 
> sorry to ask that, but what does "almost every PCI system" mean?

First of all compared to the Digium TE4xx the Sangomas work in 3,3V and
5V PCI slots. That means they run in every PCI slot but PCIexpress.

In addition to that the Digiums are said to have problems in some
mainboards. I can not confirm that because we use digiums only in one
specific intel board and there they work good.

Sangoma gives a warranty that their cards work in any board.

Christian
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RE: [Asterisk-Users] ${EXTEN}

2006-05-30 Thread William Piper

Very well... However, if you don't send your dial plan, we don't know what
it is you are trying to do.

Obviously make sure you have set the dtmf to rfc2833 in your sip.conf.

bp

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Akpome
Akpoguma
Sent: Saturday, May 27, 2006 12:09 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] ${EXTEN}


I looked closely to find out what was wrong as I am sure its not a typo. I 
realized that the problem was that dtmf was not working. Therefore asterisk 
wasnt picking up the dialed extensions after the intial dialin. I used a sip

phone and alcatel digital phone

>From: "William Piper" <[EMAIL PROTECTED]>
>Reply-To: Asterisk Users Mailing List - Non-Commercial 
>Discussion
>To: "'Asterisk Users Mailing List - Non-Commercial 
>Discussion'"
>Subject: RE: [Asterisk-Users] ${EXTEN}
>Date: Wed, 24 May 2006 11:01:50 -0400
>
>Send your dial plan where you are having the problem. I agree with Eric, 
>you
>must have something misspelled.
>
>bp

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[Asterisk-Users] Re: Re: E1 hardware for asterisk

2006-05-30 Thread Sebastian Kayser

* Christian Victor <[EMAIL PROTECTED]> wrote:
> > Regarding echo cancel. Is there someone with hands-on experience
> > regarding the echo canceller performance of the Junghanns E1 cards
> > compared to for example the Sangoma ones?
> 
> Well - the Junghanns does the echocancel in software and the Sangoma
> A104d does it in hardware.

Alright, i just had a look at their product lineup. It seems as not only
the A104d but also the low end of their E1 cards (i.e. A101) comes with
this onbard echo canceller (EDAC), right?

> So on the Sangoma echo cancel does not affect CPU performance while on
> the Junghanns it does.

And in terms of quality? Does one of them perform noticable better than
the other.

- Sebastian
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[Asterisk-Users] no extension from ISDN phone with bristuff

2006-05-30 Thread Louis-David Mitterrand

Hello,

I have a Gigaset S44 connected to a quadBRI NT port. Receiving calls 
works phone, however when dialing out from the phone the call is dropped 
to the 's' extension, as if no extension had been dialed:

-- Accepting voice call from '492389990' to 's' on channel 0/2, span 4
-- Executing Directory("Zap/11-1", "default") in new stack
-- Playing 'dir-intro' (language 'fr')
etc...

My zapata.conf contains:

[channels]
language=fr
musiconhold=default
switchtype=euroisdn
priindication=outofband
callerid=asreceived
busydetect=no
callwaiting=yes
callwaitingcallerid=yes
pridialplan=unknown
nationalprefix=0
internationalprefix=00
callgroup=1
pickupgroup=1
hidecallerid=no
usecallerid=yes
echocancel=yes
context=default
;; for TE ports
signalling=bri_cpe_ptmp
group=1
channel=>1-2

channel=>4-5

channel=>7-8

;; for NT ports
signalling=bri_net_ptmp
echocancel=no
pridialplan=local
prilocaldialplan=dynamic
priindication=passthrough
context=international
group=2
channel=>10-11

And currently using "Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1n"

When the phone is connected directly to the telco ISDN plug, outgoing 
calls work fine.

What did I forget?

Thanks,
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Re: [Asterisk-Users] E1 hardware for asterisk

On Tue, 30 May 2006, Christian Victor wrote:
> Hi Armin,
> 
> >> I personally prefer the Sangoma E1 cards. The work in almost every PCI
> > 
> > sorry to ask that, but what does "almost every PCI system" mean?
> 
> First of all compared to the Digium TE4xx the Sangomas work in 3,3V and
> 5V PCI slots. That means they run in every PCI slot but PCIexpress.
> 
> In addition to that the Digiums are said to have problems in some
> mainboards. I can not confirm that because we use digiums only in one
> specific intel board and there they work good.
> 
> Sangoma gives a warranty that their cards work in any board.

Okay, same for the Eicon cards. I was just wondering about the "almost".

Armin

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Re: [Asterisk-Users] Asterisk Inte rnal sip calls I can´t send/recive

2006-05-30 Thread Omar Lopez Limonta


> [entrada]
> exten => s,1,Wait,11
> exten => s,2,Answer
> exten => s,3,Wait,1
> exten => s,4,Dial(SIP/200,60,Ttr)
> exten => s,5,Dial(SIP/201,60,Ttr)
> exten => s,6,Dial(SIP/202,60,Ttr)
> exten => s,7,Dial(SIP/203,60,Ttr)



He probado a añadir esto pero el error persiste , no se si me exprese
bien antes .. lo que quiero decir es que no puedo hacer llamadas entre
telefonos SIP por software en mi LAN.
Es decir la extensión SIP 201 no puede llamar a la 203 y en el log me
salta lo que dije
---
ERROR
--
Verbosity is at least 6
  -- Remote UNIX connection
  -- Executing Dial("SIP/201-979d", "SIP/201|60|t") in new stack
  -- Called 201
May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Critical Request)
== No one is available to answer at this time
  -- Executing VoiceMail("SIP/201-979d", "201") in new stack
  -- Playing 'vm-intro' (language 'es')
== Spawn extension (anurix, 201, 2) exited non-zero on 'SIP/201-979d'
May 29 18:09:34 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call
[EMAIL PROTECTED] for seqno 52991
(Non-critical Response)
-

Sin embargo puedo hacer llamadas desde el telefono SIP software al
exterior sin ningun
problema :S.



--
http://www.tuactualidad.com
IM: pollo.es.pollo en gmail.com
Te lo traigo fresco.
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Re: [Asterisk-Users] Re: Re: E1 hardware for asterisk

>> Well - the Junghanns does the echocancel in software and the Sangoma
>> A104d does it in hardware.
> 
> Alright, i just had a look at their product lineup. It seems as not only
> the A104d but also the low end of their E1 cards (i.e. A101) comes with
> this onbard echo canceller (EDAC), right?

No - the A104d and A108d are Sangomas only E1 cards with hardware echo
cancel.

The analog A200 has (optional) hardware EC too.

>> So on the Sangoma echo cancel does not affect CPU performance while on
>> the Junghanns it does.
> 
> And in terms of quality? Does one of them perform noticable better than
> the other.

Well - I am not sure how the EC in the Jun ghanns driver is implemented.
If you have unlimited CPU power (or the hardware EC is very bad
implemented) it could be of the same quality.

In fact the Sangoma has 128ms echo cancel per channel. As far as I know
neither the Digum harware echo cancel nor the available software
solutions offers this.

Christian
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Re: [Asterisk-Users] Asterisk Inte rnal sip calls I can´t send/recive

2006-05-30 Thread Omar Lopez Limonta


> solo cambia tu extension.conf
>
> [entrada]
> exten => s,1,Wait,11
> exten => s,2,Answer
> exten => s,3,Wait,1
> exten => s,4,Dial(SIP/200,60,Ttr)
> exten => s,5,Dial(SIP/201,60,Ttr)
> exten => s,6,Dial(SIP/202,60,Ttr)
> exten => s,7,Dial(SIP/203,60,Ttr)
>


I try it , but it doesn´t work , i want call to another sip extension
into my lan, i want call to 201 from 203 extension  both are into my
LAN in the same range using SIP Software Phone , when i call to any
extension i get
--
ERROR
--
Verbosity is at least 6
  -- Remote UNIX connection
  -- Executing Dial("SIP/201-979d", "SIP/201|60|t") in new stack
  -- Called 201
May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Critical Request)
== No one is available to answer at this time
  -- Executing VoiceMail("SIP/201-979d", "201") in new stack
  -- Playing 'vm-intro' (language 'es')
== Spawn extension (anurix, 201, 2) exited non-zero on 'SIP/201-979d'
May 29 18:09:34 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call
[EMAIL PROTECTED] for seqno 52991
(Non-critical Response)
-

And voicemail bring me on , and i can stablish SIP conection.
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Re: [Asterisk-Users] Re: Re: E1 hardware for asterisk

On Tue, 30 May 2006, Christian Victor wrote:
> >> Well - the Junghanns does the echocancel in software and the Sangoma
> >> A104d does it in hardware.
> > 
> > Alright, i just had a look at their product lineup. It seems as not only
> > the A104d but also the low end of their E1 cards (i.e. A101) comes with
> > this onbard echo canceller (EDAC), right?
> 
> No - the A104d and A108d are Sangomas only E1 cards with hardware echo
> cancel.
> 
> The analog A200 has (optional) hardware EC too.
> 
> >> So on the Sangoma echo cancel does not affect CPU performance while on
> >> the Junghanns it does.
> > 
> > And in terms of quality? Does one of them perform noticable better than
> > the other.
> 
> Well - I am not sure how the EC in the Jun ghanns driver is implemented.
> If you have unlimited CPU power (or the hardware EC is very bad
> implemented) it could be of the same quality.
> 
> In fact the Sangoma has 128ms echo cancel per channel. As far as I know
> neither the Digum harware echo cancel nor the available software
> solutions offers this.

I thought the Eicon cards were the only ones with 128ms echo-cancel ;-)

Anyway, what time frame do the software implementations use?
Even with enough CPU power, do these software solutions provide as
good results as a DSP on board?

Armin

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Re: [Asterisk-Users] I guess my server capacity is ok

2006-05-30 Thread Lachek Butalek


What process is taking up 100% CPU? Is it Asterisk processes or
something else? Also, is the load spread out over multiple processes,
or do you have one or two processes taking up 90% or more of your
total?

You also have dual CPUs (and hyperthreading, which to FC3 should look
like 4 CPUs if I'm not mistaken) - is the 100% CPU usage across all
two (or four) processors, or is it only CPU1 that peaks at 100%? Have
a look at "Last Used CPU" in top. What load are the other CPUs at?

I don't have personal experience running that large of an
installation, but I imagine your system specs would allow you to
handle more simultaneous calls than 50, even though you're doing some
transcoding.

On 5/30/06, Goke Aruna <[EMAIL PROTECTED]> wrote:

can someone overthere help?

the server specs are as follows
 HP DL380G4 Dual Intel Xeon 3.2GHz processor with 4GB RAM,
 running fedora core 3
 asterisk-1.2.5
 ss7-0.8.3d.
 using sip as advised to receive calls from another gateway in US.
 using g729 in transcoding way.

 however, I noticed the call hit the 51 active calls which is 102channels, I
run "top" to check the system resources usage and i discovered that the cpu
is 100% used. asterisk, sip, ss7  never crashed throughout.

 however, since transcoding takes alot of system resources.. how can I use
g729 in passthru mode.

 and I guess disabling hyperthreading will save me more system resouces.

 I will be glad, if you can give me more info on system management cos i
think with that system, it should able to handle at least five E1's.

I say thank you for finding time to reply my mail.

 goksie

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RE: [Asterisk-Users] I guess my server capacity is ok

2006-05-30 Thread Steve Totaro

G729 is your problem.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
 

> -Original Message-
> From: Lachek Butalek [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, May 30, 2006 10:10 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] I guess my server capacity is ok
> 
> What process is taking up 100% CPU? Is it Asterisk processes or
> something else? Also, is the load spread out over multiple processes,
> or do you have one or two processes taking up 90% or more of your
> total?
> 
> You also have dual CPUs (and hyperthreading, which to FC3 should look
> like 4 CPUs if I'm not mistaken) - is the 100% CPU usage across all
> two (or four) processors, or is it only CPU1 that peaks at 100%? Have
> a look at "Last Used CPU" in top. What load are the other CPUs at?
> 
> I don't have personal experience running that large of an
> installation, but I imagine your system specs would allow you to
> handle more simultaneous calls than 50, even though you're doing some
> transcoding.
> 
> On 5/30/06, Goke Aruna <[EMAIL PROTECTED]> wrote:
> > can someone overthere help?
> >
> > the server specs are as follows
> >  HP DL380G4 Dual Intel Xeon 3.2GHz processor with 4GB RAM,
> >  running fedora core 3
> >  asterisk-1.2.5
> >  ss7-0.8.3d.
> >  using sip as advised to receive calls from another gateway in US.
> >  using g729 in transcoding way.
> >
> >  however, I noticed the call hit the 51 active calls which is
> 102channels, I
> > run "top" to check the system resources usage and i discovered that
the
> cpu
> > is 100% used. asterisk, sip, ss7  never crashed throughout.
> >
> >  however, since transcoding takes alot of system resources.. how can
I
> use
> > g729 in passthru mode.
> >
> >  and I guess disabling hyperthreading will save me more system
resouces.
> >
> >  I will be glad, if you can give me more info on system management
cos i
> > think with that system, it should able to handle at least five E1's.
> >
> > I say thank you for finding time to reply my mail.
> >
> >  goksie
> >
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> >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
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Re: RE : [Asterisk-Users] TDM2400P with echo canceller not working

2006-05-30 Thread Giorgio Incantalupo


Hello Francois,
this is my zaptel.conf:
# 12 red modules (FXO) on tdm2400
fxsks=1-12
fkoks=13-16
defaultzone=it
loadzone=it
This is my zapata.conf:
[channels]
context = outbound_zap
canpark = yes
echocancel = 128
echocancelwhenbridged = yes
faxdetect = incoming
language = us
musiconhold = default
signalling = fxs_ks
callerid = "FGA" <0293500178>
channel => 1

... the same for all FXO channels, haven't specified FXS channels until 
FXO ones work correctly


I haven't seen any led on my echo cancellation module...sorry!

Messages from /var/log/messages:
May 29 09:41:38 pbxtest kernel: VPM: Chip 0: ver 33
May 29 09:41:38 pbxtest kernel: VPM: U-law mode
May 29 09:41:38 pbxtest kernel: VPM: Chip 1: ver 33
May 29 09:41:38 pbxtest kernel: VPM: Chip 2: ver 33
May 29 09:41:38 pbxtest kernel: VPM: Chip 3: ver 33
May 29 09:41:38 pbxtest kernel: VPM: DTMF threshold set to 1250
May 29 09:41:38 pbxtest kernel: VPM: Present and operational

I think it is all right...

Another question for you:
which kind of test or instrument do you use to analyze echo on analog 
lines? I've heard about ztmonitor but I do not know exaclty what it does 
and infos on internet are not many. Or do you use something else? If I 
hade some sort of instrument for measuring echo my work on it could be 
easier instead of changing randomly rxgain, txgain and all the other 
parameters.


TIA


Giorgio Incantalupo

[EMAIL PROTECTED] wrote:

Hello Giorgio,

I am a TDM2400 happy user.  :-)

Could you show your zaptel.conf zapata.conf config files ?
Think to tell us how many modules you have and where they are plugged on the
TDM2400P.
Are the leds on the echocan modules running as a LasVegas casino (scrolling
in a circular pattern) ?
If you have an echocan module aboard and well running you must see it at
Linux boot (syslog):
The 4 echocan chipsets are sequencialy checked and return an hexadecimal
code different of "FF" if ok, just before to tell "VPM:  Present and
operational (Rev X)", if I remember well.

If the VPM is ok, zaptel echocan software is automaticaly disabled for this
zap channels in Asterisk server.
The echotraining is not applicable for this card and generate an error
message on the Asterisk console if enabled in your zapata.conf, but don't
care, this doesn't affect Asterisk and the call itself.

The settings must be done as usual with TDM400 cards.

I hope this can help a little.

Best Regards,
Francois BERGERET,
France.


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Giorgio
Incantalupo
Envoyé : lundi 29 mai 2006 09:33
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] TDM2400P with echo canceller not working


Hi,
I have  a box with Debian Sarge, Asterisk 1.2.1 (and zaptel 1.2.1) and a 
TDM2400P with echo canceller. I installed the card but no echo 
cancellation is being made...seems like the echo canceller module does 
not work, infact the software cancellation is working.


My zapata.conf has echocancel = 128 and echocancelwhenbridged = yes but 
no echotraining parameter which gives a warning.


I found no info about how to use this card and how to correctly set 
zapata.conf, which zaptel version to use, etc...


Does anybody knows how to use this card?

TIA

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--


GIORGIO INCANTALUPO
Tel. +39 02 9350 4780 (104)

FG&A Software
20017 Rho - Via Puccini, 8

E-Mail :
[EMAIL PROTECTED]
Internet:
http://www.fgasoftware.com

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RE: [Asterisk-Users] Re: How to enable call waiting on Sip Phones

2006-05-30 Thread William Piper

Ok, I've been seeing this email floating around for a while now. If you are
using [EMAIL PROTECTED], don't use this, but if you are just using standard 
asterisk with
your own custom dial plan... this should work for you. 

I don't actually use this in my dial plan and haven't tested it but it
should be close enough to get you to what you are trying to accomplish. If
you dial *70, it will activate call waiting, *71 will deactivate it.

[cw]
exten => *70,1,Answer
exten => *70,2,DBput(cw/${CALLERIDNUM}=1)
exten => *70,3,Playback(callwaiting)
exten => *70,4,Playback(activated)
exten => *70,5,Macro(hangupcall)

exten => *71,1,Answer
exten => *71,2,DBdel(cw/${CALLERIDNUM})
exten => *71,3,Playback(callwaiting)
exten => *71,4,Playback(de-activated)
exten => *71,5,Macro(hangupcall)


[inbound-callwaiting]
exten => _.,1,Answer
exten => _.,2,DBget(cw=cw/${EXTEN})
exten => _.,3,dial,SIP/${EXTEN}
exten => _.,4,hangup
exten => _.,103,SetGroup(${EXTEN})
exten => _.,104,GotoIf($["${GROUP_COUNT(${EXTEN})}" <= "1"]?105:206)
exten => _.,105,dial,SIP/${EXTEN}|20
exten => _.,106,voicemail(u${EXTEN})
exten => _.,107,hangup
exten => _.,206,voicemail(b${EXTEN})
exten => _.,208,hangup

Good Luck,

bp

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit
Sent: Monday, May 29, 2006 1:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: How to enable call waiting on Sip Phones

> I gathered that but it has its uses.  Could you then give us soem tips on
> how to get this working.  Call forwarding is a done deal but i cant seem
to
> find any info on call waiting anywhere?  Help needed.  Customer fustrated.
Are you using [EMAIL PROTECTED] ?

If not, are you using AMP (now FreePBX) or you just coded your
dialplan yourself ?

If it is a custom dialplan, please post it.

What SIP phone are you using ?

We need more info to help you.
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RE: [Asterisk-Users] AGI MySql

2006-05-30 Thread William Piper

Why not do: 
exten => s,1,AGI(xyz.agi|${MACRO_EXTEN})

bp

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Akpome
Akpoguma
Sent: Tuesday, May 30, 2006 2:55 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] AGI MySql


I have been able to figure out the first part of my problem.

I wrote my scripts based on wrong assumptions. one of which was that the 
line

exten => s,1,AGI(xyz.agi)

sends an undefined extension value to the script. This is definitely wrong. 
This line actually sends an extension value of "s". Therefore AGI{extension}

in my script can never be undefined as long as the script is being called 
from a dialplan.

This leaves me with the second problem.

thanks for this trouble



>From: "Akpome Akpoguma" <[EMAIL PROTECTED]>
>Reply-To: Asterisk Users Mailing List - Non-Commercial 
>Discussion
>To: asterisk-users@lists.digium.com
>Subject: [Asterisk-Users] AGI MySql
>Date: Mon, 29 May 2006 07:58:40 +
>
>
>The following is my AGI script done in perl
>
>#!/usr/bin/perl
>
>use strict;
>use DBI;
>
>$|=1;
>
>my %AGI;
>
>while() {
>
>   chop;
>   last unless length($_);
>   if (/^agi_(\w+)\:\s+(.*)$/) {
>   $AGI{$1} = $2;
>   }
>   }
>
>my $ext = $AGI{extension};
>
>if (!($ext)) { $ext = 10; }
>
>my $dbh = DBI->connect('dbi:mysql:voiceDb', 'test', 'test', {PrintError=>0,

>RaiseError=>1});
>my $sql = "select filename from contentTable where ext='$ext'" or die 
>$dbh->errstr;
>my $filename = $dbh->selectrow_array($sql);
>$dbh->disconnect;
>
>$filename =~ s/\.wav//i;
>print "STREAM FILE $filename \"\"\n";
>
>exit;
>
>The return value of $filename from the database is supposed to be 
>/var/sounds/scoobie.wav.
>
>There are 2 Problems
>
>1. When I execute this script manually it works well but when I call this 
>script from dialplan I get no return value.
>2. I did print "STREAM FILE /var/sounds/scoobie \"\"\n" and the phone was 
>as silent even though I see no error on the console.
>
>Am clueless as to how to fix this. I need someone's 
>assistance...resposes would be appreciated.
>
>_
>Express yourself instantly with MSN Messenger! Download today it's FREE! 
>http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/
>
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_
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http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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[Asterisk-Users] Hardware requirements for Asterisk

2006-05-30 Thread Luis Uebel

I would like to know what are the minimum hardware requirements for 
Asterisk:

1. Linux kernel 2.4?
2. PC 486 50MHz?
3. Memory 64 Mbytes?

Thanks
Luis


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[Asterisk-Users] LDAP directory app?

2006-05-30 Thread Mimmus

Hi,
is there an Asterisk app (or AGI script) to look up names in a LDAP
directory?

-- 
Domenico Viggiani

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[Asterisk-Users] IAX softphone with RSA support?

2006-05-30 Thread Álvaro Palma

Which (preferible free :-) softphone that supports IAX and RSA 
encryption do you recommend? It seems that IDEFisk doesn't yet.


Thanks a lot for your help.

--
Atly.
Alvaro Palma

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Re: [Asterisk-Users] Hardware requirements for Asterisk

On Tue, 30 May 2006, Luis Uebel wrote:
> I would like to know what are the minimum hardware requirements for Asterisk:
> 1. Linux kernel 2.4?
> 2. PC 486 50MHz?
> 3. Memory 64 Mbytes?

That depends on what you want to do with Asterisk.
The kernel is alright and the 64MB are also enough, but the 50MHz
might be far too slow. But that really depends on what you want to do.

Armin
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Re: [Asterisk-Users] Problem with IAX2 dialin with portunity

2006-05-30 Thread Joshua Colp


Matthias Fechner wrote:

Hello Joshua,

Joshua Colp wrote:

[portunity-out]
type=friend
host=iax.iaxport.de
username=XXX
secret=YY
context=incoming-portunity
notransfer=yes

Only if the username is specified as portunity-out when the other side dials
you. Otherwise your Asterisk has no idea what to authenticate them as so it
takes a guess and in the end settles on guest.


But should not asterisk here see, that the call is comming in from the
host: host=iax.iaxport.de or from the username=iaxXX?
In the SIP configuration I do it this way.

Or need I to define some other parameters in the section
[portunity-out] or easily rename it.

If I get a call, asterisk says the following: (I hope everything is in :) )


The remote side is not sending the username to authenticate as (it's not 
shown in the NEW) so your Asterisk does not know who to authenticate it 
as, and thus chooses guest. As for host based matching - this doesn't 
happen as you might expect. You need to use permit and deny lines to get 
the user entry matched. Check into the sample iax.conf to see how to do 
this.


--
Joshua Colp
Software Developer
Digium
P - 256-428-6066
C - 506-878-0147
[EMAIL PROTECTED]
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[Asterisk-Users] Automon

2006-05-30 Thread Carlos Chavez

Is there a way to control the name and location of the recordings made
with automon?  I need to be able to send the file to the client when
they finish a call.  How can I know which file belongs to the user?

-- 
Carlos Chavez Prats
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001


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[Asterisk-Users] Dumping outbound audio on hold

2006-05-30 Thread Matt


Hi,
I have a setup something like the following:

[callcenter] <--IAX2--> [voip-server] <---IAX2--> [LD provider]

I also have PRIs going in and out of [voip-server].

The problem is with calls going between [callcenter] and [voip-server].
When a call comes in the PRI to [voip-server] and then to
[callcenter], all is fine.
When a call goes out to [voip-server] and then to PRI, all is fine.
When a call goes out to [voip-server] and then to [LD provider], all
is fine, until I put the person on hold.   When I pick them back up,
sometimes outbound audio is dead (the person I called can not hear
me), however inbound audio is fine (I can hear them).

Does anyone have any thoughts on what this issue might be?
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[Asterisk-Users] patch application

2006-05-30 Thread Damon Estep









I have a production server running a CVS Head release dated
8/27, which is pretty much 1.2 minus some last minute additions, 1.2 was
released at the end of august 2005.

 

There is a sip channel patch related to presence and sip
subscriptions that I wish to apply, but since the server has been very stable I
do not want to do a full upgrade to 1.2.7.1.

 

The patch was added to head on 8/29 and released in 1.2.

 

The questions;

 

Is it possible to patch, make, and install only chan_sip.c and
pbx.c without making all of asterisk? I assume the process would be to apply
the diff file, make the specific modules, and copy it to the appropriate
directory. Is this correct? Any pointers appreciated, and specifics really,
really appreciated!

 

The patch in question looks like it impacts pbx.c and
chan_sip.c, but how can I be sure there are not other dependencies?

 

http://bugs.digium.com/view.php?id=3644

 

 

 

 






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Re: [Asterisk-Users] hints/subscriptions accross IAX

Faris Raouf wrote:

> But I need to get an LED to light up on a GS in Location2 when a line on
> the Polycom at Location1 is in use. Is this possible? If so, can anybody
> give me any pointers as to how?

Not at this time, no. There has been talk of building a method for doing
this, but so far there is nothing in Asterisk itself.
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Re: [Asterisk-Users] Memory-leak 1.2.7.1

2006-05-30 Thread Noah Miller


Hi Attilla -


So, this left me only one conclusion. The application with the memory
leak is Asterisk.


I know every situation is different, but I just thought that I'd point
out that I have machines running 1.2.7.1 that I haven't restarted in
months.  Of course, 1.2.7.1 hasn't been out that long, but before that
these boxes were running 1.2.6, 1.2.4, and 1.2.3, and I didn't restart
for any of those updates.  I did a restart when running version 1.2.2,
but that version had a more serious flaw than a memory leak.  All my
machines are running Tao  Linux (a Redhat Enterprise clone).  I'm not
using ztdummy, but I do have zaptel cards.  All machines are either
Xeons or P4's.

Asterisk and zaptel depend on a number of other items including bison,
libnewt, and usb-uhci.  You might want to check and make sure all of
these are working and not leaking anything.

- Noah
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Re: [Asterisk-Users] Compilation issues with s390

Frank Pani wrote:

> make -C gsm lib/libgsm.a
> make[2]: Entering directory `/usr/src/asterisk-1.2/codecs/gsm'
> as   -o src/k6opt.o src/k6opt.s

Sorry, I missed that one when I did the first s390 fix. It's been taken
care of now.
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Re: [Asterisk-Users] patch application

Damon Estep wrote:

> There is a sip channel patch related to presence and sip subscriptions
> that I wish to apply, but since the server has been very stable I do not
> want to do a full upgrade to 1.2.7.1.

There have been hundreds of bugs fixed since 1.2 was released. I think
it would be much better if you just upgraded :)
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Re: [Asterisk-Users] Music on hold problem

2006-05-30 Thread Matt


I am having this issue.. asterisk 1.2.7   using native MOH without
mpg123.   Calls get put on park and EVERYTIME it seems to start the
music off the same (even though I've chosen to do it randomly!)...
then after playing the first song it just goes to dead air.

STEPS TO REPRODUCE:
Xfer --> 70
Caller hears music on hold while allison is telling me "71"
Hang up.
The caller now gets the music *I* was hearing which seems to always be
the same, and then at the end of the song gets dead air!


On 4/15/06, Lacy Moore - Aspendora <[EMAIL PROTECTED]> wrote:

I can see in some cases where you might want to start all callers from the
beginning, but I can also see in some cases where you would not.  This
sounds to me like we need an option.


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[Asterisk-Users] Polycom 501

2006-05-30 Thread Curt Shaffer









Does anyone out there have a sample config they can share
for the Polycom 501? Is it possible to do “sub configs” like you
can with the Aastra 9133i? It could be just me but the boot configs seem a bit
cryptic compared to the aastra. Also do any of you have any comparisons between
these and the Aastra 9133i?

 

Thanks

 

Curt






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Re: [Asterisk-Users] Re: Re: E1 hardware for asterisk

>> In fact the Sangoma has 128ms echo cancel per channel. As far as I know
>> neither the Digum harware echo cancel nor the available software
>> solutions offers this.
> 
> I thought the Eicon cards were the only ones with 128ms echo-cancel ;-)

I am happy that I could enlighten you a bit in this point. ;-)

> Anyway, what time frame do the software implementations use?
> Even with enough CPU power, do these software solutions provide as
> good results as a DSP on board?

To be honest I really don't know a lot about the technical details of
software echo cancellers. My guess would be that they are not as good as
the hardware echo cancellers. And it's more of an esoteric discussion
because usually you don't have unlimited CPU power.

All I can say is that the Sangoma 104d/108d DO provide excellent results
for they have a DSP on board that handles the EC.

Christian

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Re: [Asterisk-Users] E1 hardware for asterisk

Christian Victor schrieb:
> I personally prefer the Sangoma E1 cards. The work in almost every PCI
> system 

Okay guys - I have to add something to this so it will not be misunderstood:

I never had problems with a Sangoma card in any mainboard and just added
the word "almost" to prevent somebody with an exotic (i.e. erroneous)
PCI implementation and a passive 8-fold PCI risercard with a non
standard connector to sue me for what I said. ;-)

>From my experience they work in _every_ PCI slot.

Chris
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[Asterisk-Users] Zaptel and 2.6.9-34.0.1.EL Kernel on CentOS

2006-05-30 Thread David K Parker

Has anyone been able to compile Zaptel after upgrading to 2.6.9-34.0.1.EL kernel? I'm running CentOS and was unable to recompile Zaptel. I reverted back to 2.6.9-22.0.1.EL but even then Zaptel would fail to load. I tried pulling the latest zaptel sources from digium cvs, but they apparently don't have a record in dns for 
cvs.digium.com. I did download zaptel-1.2.5.tar.gz and recompiled then zap would function again with 2.6.9-22.0.1.EL kernel. I tried booting the newer kernel and recompiling zap from the new zap source, but the compile would fail. I'm back to 
2.6.9-22.0.1.EL kernel and zaptel-1.2.5.Thanks,David
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Re: [Asterisk-Users] Asterisk restarting in a minute

2006-05-30 Thread Mojo with Horan & Company, LLC

crontab?  I restart my asterisk nightly with cron but a simple typo 
could make that every minute instead of every day... 



Woodoo People .pGa! wrote:

Hi!

Probably any of you meet with the following problem:
asterisk is restarting in a minute (if no active call) if active call,
it says cannot receive a call due to restart in progress.

even if i starting with -c, i have no disconnected, but see the stuff
restarting.

i've tried to recompile, older version, virgin config, etc. same results.
it's happened after a power loss, on a ext3 fs, sitting on a raid1.
astdb was deleted, log is not showing any interesting things.

any ideas please?


--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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[Asterisk-Users] app_conference sources?

2006-05-30 Thread Henry J. Cobb

The CVS server for app_conference seems to be down.

Can somebody mail me a recent copy of the sources please?

-- 
Henry J. Cobb
http://www.io.com/~hcobb/

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Re: [Asterisk-Users] misdn problem

2006-05-30 Thread asterisk


In my extension.conf I have both

[ext-did]
include => ext-did-custom
exten => 0108680550,1,Set(FROM_DID=0108680550)
exten => 0108680550,n,Set(FAX_RX=disabled)
exten => 0108680550,n,Goto(timeconditions,3,1)
exten => _01086805XX,1,Set(FROM_DID=_01086805XX)
exten => _01086805XX,n,Set(FAX_RX=disabled)
exten => _01086805XX,n,Goto(custom-did-route,${EXTEN},1)


and later:
[custom-did-route]
exten => _01086805XX,1,Set(FROM_DID=${EXTEN})
exten => _01086805XX,2,Macro(exten-vm,${EXTEN:7},${EXTEN:7})
exten => _01086805XX,3,VoiceMail([EMAIL PROTECTED]|u)
; ATTENZIONE ! E' necessario remmare una riga in extensions.conf
;exten => exit-FAILED,n,Hangup()
; in [macro-vm]
;

The point is that the dial plan is correct; if it were not, ALL calls
should be lost,
not about 25% dialing the same number;
Insted, you can see ringing BOTH channel 1 of BOTH ISDN Cards; adding
immediate=yes in misdn.conf
seems to slow down the problem, but not completly solve it.

thanks,

Andrea



   
 Tommaso Calosi
  To 
 Sent by:  Asterisk Users Mailing List -   
 asterisk-users-bo Non-Commercial Discussion   
 [EMAIL PROTECTED]
 m.com  cc 
   
   Subject 
 29/05/2006 10.47  Re: [Asterisk-Users] misdn problem  
   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 <[EMAIL PROTECTED] 
 ists.digium.com>  
   
   




In your extension.conf, in the misdn context you defined in
/etc/asterisk/misdn.conf you have to add something linke the following line

[from-pstn]
exten => 0108680550,1,Dial(SIP/201)

If you don't want to have to write a string for each called extension,
you can put something like. Obviously SIP/201 is the phone you wish to
ring.

[from-pstn]
exten => _.,1,Dial(SIP/201)

[EMAIL PROTECTED] wrote:
> I have two HFC ISDN Cards, configured using mISDN on asterisk svn head
1.2
>
> These two cards are connected to 2 ISDN Lines, receiving calls for 50
> numbers.
>
> Everything is OK on 75 % and bad on 25 %
>
> When is bad, In /var/log/asterisk/full I see
>
> May 26 09:55:28 WARNING[24410] chan_misdn.c: Extension can never match,
so
> disconnecting
> May 26 09:55:28 DEBUG[24410] channel.c: Prodding channel 'mISDN/1-1'
> May 26 09:55:28 DEBUG[24410] channel.c: Scheduling timer at 160 sample
> intervals
> May 26 09:55:28 WARNING[24410] chan_misdn.c: Extension can never match,
so
> disconnecting
> May 26 09:55:28 DEBUG[24410] channel.c: Prodding channel 'mISDN/2-1'
> May 26 09:55:28 DEBUG[24410] channel.c: Scheduling timer at 160 sample
> intervals
>
> on the asterisk console, if  I set misdn set debug 10, I see
>
> P[ 1] handle_frm: frm->addr:42000103 frm->prim:3f082
> P[ 1]  --> lib: NEW_CR Ind with l3id:200ec on this port.
> P[ 1]  --> new_process: New L3Id: 200ec
> P[ 1] handle_frm: frm->addr:42000103 frm->prim:30582
> P[ 1] set_channel: bc->channel:0 channel:1
> P[ 1] lib Got Prim: Addr 42000103 prim 30582 dinfo 200ec
> P[ 1] $$$ find_chan: No channel found for oad:3481303064 dad:0108680550
> P[ 0] $$$ find_chan: No channel found with l3id:200ec
> P[ 1] I IND :SETUP oad:3481303064 dad:0108680550
> P[ 1]  --> mode:TE cause:16 ocause:16 rad:
> P[ 1]  --> facility:FAC_NONE out_facility:FAC_NONE
> P[ 1]  --> info_dad: onumplan:0 dnumplan:2 rnumplan:
> P[ 1]  --> screen:0 --> pres:0
> P[ 1]  --> channel:1 caps:Speech pi:0 keypad:
> P[ 1]  --> urate:0 rate:16 mode:0 user1:0
> P[ 1]  --> pid:336 addr:50010102 l3id:200ec
> P[ 1]  --> b_stid:0 layer_id:50010180
> P[ 1]  --> bc:81681d4 h:0 sh:0
> P[ 1] $$$ find_chan: No channel found for oad:3481303064 dad:0108680550
> P[ 1]  --> Bearer: Speech
> P[ 1]  --> Codec: Alaw
> P[ 0]  --> * NEW CHANNEL dad:0108680550 oad:3481303064
> P[ 1] read_config: Getting Config
> P[ 1] config_jb: Called
> P[ 1]  --> * CallGrp: PickupGrp:
> P[ 1] * Queuing chan 0x842e8b0
> P[ 1] CONTEXT:from-pstn
> P[ 1] T

Re: [Asterisk-Users] Zaptel and 2.6.9-34.0.1.EL Kernel on CentOS

2006-05-30 Thread Marco Mouta

make this :(BE SURE this is the kernel you are booting!!! make ls inside /usr/src/kernels)cd /usr/src/kernels/2.6.9-34.0.1.EL/include/linuxmv spinlock.h spinlock.h.oldwget http://
nerdvittles.com/aah27/spinlock.hshutdown -r nowThen make this command: rebuild_zaptelThis is a well known issue with Centos and ZaptelHope it helps!Best regards,Marco Mouta
ps. give me some feedbackOn 5/30/06, David K Parker <[EMAIL PROTECTED]> wrote:
Has anyone been able to compile Zaptel after upgrading to 2.6.9-34.0.1.EL kernel? I'm running CentOS and was unable to recompile Zaptel. I reverted back to 2.6.9-22.0.1.EL but even then Zaptel would fail to load. I tried pulling the latest zaptel sources from digium cvs, but they apparently don't have a record in dns for 
cvs.digium.com. I did download zaptel-1.2.5.tar.gz and recompiled then zap would function again with 2.6.9-22.0.1.EL
 kernel. I tried booting the newer kernel and recompiling zap from the new zap source, but the compile would fail. I'm back to 
2.6.9-22.0.1.EL kernel and zaptel-1.2.5.Thanks,David

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Re: [Asterisk-Users] Problem with IAX2 dialin with portunity

2006-05-30 Thread Matthias Fechner

Hello Joshua,

* Joshua Colp <[EMAIL PROTECTED]> [30-05-06 12:41]:
> happen as you might expect. You need to use permit and deny lines to get 
> the user entry matched. Check into the sample iax.conf to see how to do 
> this.

thx a lot!
The following entry in iax.conf is doing the trick:

[portunity-in]
type=user
context=incoming-portunity
permit=82.139.223.1/255.255.255.255


Best regards,
Matthias
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Re: [Asterisk-Users] Panasonic PBX

2006-05-30 Thread C F


You should use 2 FXO ports on asterisk to accomplish it, make sure the
ports supports flash, otherwise you will have a hard time transferring
back to the PBX. BTW, panasonic has one of the best documentation for
foreign voicemail integration, just tell the panasonic PBX that you
are using DTMF voicemail, and on what ports (usually 7 & 8)
What Panasonic system is it?

On 5/30/06, Chris Sutton <[EMAIL PROTECTED]> wrote:





The place I currently work at has a Panasonic Key system with 9 extensions,
and no voicemail.  It services 2 PSTN lines.



I am hoping to use Asterisk to host voicemail (I would like to use the IVR
also, but I don't even know if or how it would work).



Do I need to use a PRI between the two, or is there a simple solution?  I
would like people to be able to answer the phone and transfer the call to
voicemail if the person is not there, or after so many rings, it goes right
to voicemail.  I'm not sure what is needed?  I have seen the integration
How-To but that requires the PRI, and wasn't sure if that was the ONLY way
to go.



Thanks!



Chris


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Re: [Asterisk-Users] using a billing system

2006-05-30 Thread Joao Pereira


yes, a2billing.php is in agi-bin:

[EMAIL PROTECTED] locate a2billing.php
/usr/src/a2billing/Chameleon/A2Billing_AGI/a2billing.php
/var/lib/asterisk/agi-bin/a2billing.php

Could be because of the missing pcntl php extension?

[EMAIL PROTECTED] rpm -qa | grep php
php-mysql-4.3.9-3
php-ldap-4.3.9-3
php-odbc-4.3.9-3
php-pgsql-4.3.9-3
php-4.3.9-3
php-pear-4.3.9-3


Thanks
Joao Pereira



Vahan Yerkanian wrote:


exten => _2,1,Answer
exten => _2,2,Wait,2
exten => _2,3,DeadAGI, a2billing.php
exten => _2,4,Wait,2
exten => _2,5,Hangup

I tried it and the call is answered bu Asterisk and never dials the
destination. :(



Yes that's the correct way to launch A2B script. Are you a2billing.php 
is in your agi-bin directory? Also, you can see if the script runs 
without error by executing it from shell(you'll need php cli compiled 
and installed) and keep pressing enter key to see the script output.


Perhaps you have your php binary in the wrong path or a missing php 
extension. Make sure you have pcntl php extension installed too.


HTH,
Vahan

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Re: [Asterisk-Users] Zaptel and 2.6.9-34.0.1.EL Kernel on CentOS

2006-05-30 Thread David K Parker

Yes, that worked like a charm. Thanks!On 5/30/06, Marco Mouta <
[EMAIL PROTECTED]> wrote:
make this :(BE SURE this is the kernel you are booting!!! make ls inside /usr/src/kernels)cd /usr/src/kernels/2.6.9-34.0.1.EL/include/linuxmv spinlock.h spinlock.h.oldwget http://

nerdvittles.com/aah27/spinlock.hshutdown -r nowThen make this command: rebuild_zaptelThis is a well known issue with Centos and ZaptelHope it helps!Best regards,Marco Mouta
ps. give me some feedbackOn 5/30/06, David K Parker <

[EMAIL PROTECTED]> wrote:

Has anyone been able to compile Zaptel after upgrading to 2.6.9-34.0.1.EL kernel? I'm running CentOS and was unable to recompile Zaptel. I reverted back to 2.6.9-22.0.1.EL but even then Zaptel would fail to load. I tried pulling the latest zaptel sources from digium cvs, but they apparently don't have a record in dns for
cvs.digium.com. I did download zaptel-1.2.5.tar.gz and recompiled then zap would function again with 2.6.9-22.0.1.EL

kernel. I tried booting the newer kernel and recompiling zap from the new zap source, but the compile would fail. I'm back to
2.6.9-22.0.1.EL kernel and zaptel-1.2.5.Thanks,David

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[Asterisk-Users] Unicall Protocol Failure

2006-05-30 Thread Anton Krall

Steve Underwood:

Steve, why do some numbers give protocol errors? Ive noticed here in Mexico
that certain numbers when dialed return protocol failure and a busy tone.

Any idea why this happens and why with only certain phone numbers?

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Re: [Asterisk-Users] Asterisk Radius Module

2006-05-30 Thread John Bigelow


Trunk version of Asterisk has a cdr_radius module I believe now.

-John

Oliver Vermeulen wrote:

Hi List,
 
I'm looking for a Asterisk radius module ... Anybody has one ?
 
Thanks,

Oliver
 


*Oliver Vermeulen
*

*_World Venture Group Telecom_* *_
_*

*Corporate Address:**
*Str Avionului Nr 35/bl16J/3
Bucharest, 014333 Romania

Office:   +(40)21-569-4700
Office2: +(40)31-860-0030
Fax:  +(40)31-860-0031
USA DID: +1 (305)722-1457
BE DID:   +(32)9-395-5620
UK DID:   +(44)870-478-8896
SIP : [EMAIL PROTECTED] 
msn: [EMAIL PROTECTED] 
*http://www.wvg-tele.com*

 



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Re: [Asterisk-Users] Compilation issues with s390

2006-05-30 Thread Frank Pani





Did you just make this change recently?   I just tried to download the 1.2
stream with revision 30861 and after trying "make" see the same thing.
Perhaps I need to excercise patients.Thanks again for your help - it'll
be good to say this runs on mainframe once we get it their.

eg:

linuxvm1:/usr/src # svn checkout
http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2
Checked out revision 30861.

linuxvm1:/usr/src # cd asterisk-1.2/
linuxvm1:/usr/src/asterisk-1.2 # make

< ... snip ... >

make[1]: Entering directory `/usr/src/asterisk-1.2/codecs'
make -C gsm lib/libgsm.a
make[2]: Entering directory `/usr/src/asterisk-1.2/codecs/gsm'
as   -o src/k6opt.o src/k6opt.s
src/k6opt.s: Assembler messages:
src/k6opt.s:9: Error: unknown pseudo-op: `.value'
src/k6opt.s:10: Error: unknown pseudo-op: `.value'
src/k6opt.s:11: Error: unknown pseudo-op: `.value'
src/k6opt.s:12: Error: unknown pseudo-op: `.value'
src/k6opt.s:13: Error: unknown pseudo-op: `.value'
src/k6opt.s:14: Error: unknown pseudo-op: `.value'
src/k6opt.s:15: Error: unknown pseudo-op: `.value'
src/k6opt.s:16: Error: unknown pseudo-op: `.value'
src/k6opt.s:17: Error: unknown pseudo-op: `.value'
src/k6opt.s:18: Error: unknown pseudo-op: `.value'
src/k6opt.s:19: Error: unknown pseudo-op: `.value'
src/k6opt.s:20: Error: unknown pseudo-op: `.value'
src/k6opt.s:27: Error: Unrecognized opcode: `pushl'
src/k6opt.s:28: Error: Unrecognized opcode: `movl'
src/k6opt.s:29: Error: Unrecognized opcode: `pushl'



   
 "Kevin P. 
 Fleming"  
 <[EMAIL PROTECTED]  To 
 .com> Asterisk Users Mailing List -   
 Sent by:  Non-Commercial Discussion   
 asterisk-users-bo
 [EMAIL PROTECTED]  cc 
 m.com 
   Subject 
   Re: [Asterisk-Users] Compilation
 05/30/06 01:09 PM issues with s390
   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 <[EMAIL PROTECTED] 
 ists.digium.com>  
   
   




Frank Pani wrote:

> make -C gsm lib/libgsm.a
> make[2]: Entering directory `/usr/src/asterisk-1.2/codecs/gsm'
> as   -o src/k6opt.o src/k6opt.s

Sorry, I missed that one when I did the first s390 fix. It's been taken
care of now.
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[Asterisk-Users] RE: Asterisk-Users Digest, Vol 22, Issue 169

2006-05-30 Thread Luis Uebel

Thanks for fast answer. I would like to hold 4 analog channels or 4 digital 
channels,

voice mail with this system. I read that it's necessary 30 MHz per channel.
Is it true?
In this case, I will need 120MHz.

Luis
I would like to know what are the minimum hardware requirements for 
Asterisk:

1. Linux kernel 2.4?
2. PC 486 50MHz?
3. Memory 64 Mbytes?


That depends on what you want to do with Asterisk.
The kernel is alright and the 64MB are also enough, but the 50MHz
might be far too slow. But that really depends on what you want to do.

Armin



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[Asterisk-Users] Re: Zaptel and 2.6.9-34.0.1.EL Kernel on CentOS

2006-05-30 Thread Steven

The wiki has a lot of info, but the nerdvittles.com site has been indispensable.

-- 
-- 
Steven

http://www.glimasoutheast.org



"Marco Mouta" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
make this :
(BE SURE this is the kernel you are booting!!! make ls inside /usr/src/kernels)

cd /usr/src/kernels/2.6.9-34.0.1.EL/include/linux

mv spinlock.h spinlock.h.old
wget http:// nerdvittles.com/aah27/spinlock.h

shutdown -r now

Then make this command: rebuild_zaptel

This is a well known issue with Centos and Zaptel

Hope it helps!

Best regards,
Marco Mouta

ps. give me some feedback


On 5/30/06, David K Parker <[EMAIL PROTECTED]> wrote:
Has anyone been able to compile Zaptel after upgrading to 2.6.9-34.0.1.EL 
kernel? I'm running CentOS and was unable to recompile 
Zaptel. I reverted back to 2.6.9-22.0.1.EL but even then Zaptel would fail to 
load. I tried pulling the latest zaptel sources from 
digium cvs, but they apparently don't have a record in dns for cvs.digium.com. 
I did download zaptel-1.2.5.tar.gz and recompiled 
then zap would function again with 2.6.9-22.0.1.EL kernel. I tried booting the 
newer kernel and recompiling zap from the new zap 
source, but the compile would fail. I'm back to 2.6.9-22.0.1.EL kernel and 
zaptel-1.2.5.
Thanks,

David


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[Asterisk-Users] CallerID outbound

2006-05-30 Thread George A. Roberts IV




We're using [EMAIL PROTECTED] 2.8 (same thing occurs with 
earlier versions).  We have it set up so that if we don't answer our 
internal SIP phones it does "follow me" to our cell phones.  When Asterisk 
forwards the calls to our cell phones, the Caller ID shows our outbound number, 
not the caller's number.
 
We have 3 queues 
set up (1,2, and 3) and our extensions (801 and 802) are listed as permanently 
logged in to those queues.
 
I'm just wondering 
if there's any way to either 1) have the original caller's CallerID show up when 
calls get forwarded to our cell phones; or ideally 2) be able to override the 
caller ID when calls coming into the queues are forwarded our cell phones so we 
can tell which queue it is.
 
Thoughts?
 
George
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Re: [Asterisk-Users] Automon

2006-05-30 Thread Moises Silva


Is there a way to control the name and location of the recordings made
with automon?


Set the variable TOUCH_MONITOR for arguments to send to the
application Monitor() in either the caller or calle channel. The
caller channel si always checked first. Check the documentation in
voip-info, someone has already taken the time to explain this.

http://www.voip-info.org/wiki-Asterisk+cmd+monitor
http://www.voip-info.org/wiki/view/Asterisk+config+features.conf




How can I know which file belongs to the user?

That will highly depend upon how you handle the "user" concept in your
system. One approach can be simply by creating a database relation
between users and recorded files. So in the moment you start recording
you also insert a register making the relation between the file name
and the user.

More details about how you handle users will be usefull for better advices.

Regards

-- Moises Silva
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
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Re: [Asterisk-Users] Compilation issues with s390