Re: R: [Asterisk-Users] Problem setting up TDM22B card

[somesh s]

Hi,

Output of scanpci -v

pci bus 0x0003 cardnum 0x00 function 0x00: vendor
0xe159 device 0x0001
 Tiger Jet Network Inc. Intel 537
 CardVendor 0xa159 card 0x0001 (Card unknown)
  STATUS0x0210  COMMAND 0x0107
  CLASS 0x02 0x80 0x00  REVISION 0x00
  BIST  0x00  HEADER 0x00  LATENCY 0x20  CACHE
0x00
  BASE0 0x2001  addr 0x2000  I/O
  BASE1 0xe810  addr 0xe810  MEM
  MAX_LAT   0x80  MIN_GNT 0x01  INT_PIN 0x01  INT_LINE
0x0b
  BYTE_00x01  BYTE_1  0x00  BYTE_2  0x62  BYTE_3 
0xec

Regards,
Somesh S. Shanbhag

--- somesh s <[EMAIL PROTECTED]> wrote:

> Hi,
> 
> I changed the mother board (MB) but it is giving
> still
> the same problem.
>  ouput of dmesg|tail 
> f6 != 58
> f7 != 59
> f8 != 58
> f9 != 59
> fa != 58
> fb != 59
> fc != 58
> fd != 59
> fe != 58
> Freshmaker failed register test
> >>>
> and I have also configured zaptel.conf correctly.
> 
> Whatz next? Can I assume that it is a hardware
> problem?
> 
> Regards,
> Somesh S. Shanbhag
> 
> 
> --- John Novack <[EMAIL PROTECTED]>
> wrote:
> 
> > 
> > 
> > somesh s wrote:
> > 
> > >Hi,
> > >
> > >I didn't get any solution in the mailing list.
> >
>
>[http://asterisk.linkx.net/asteriskusers/200409/msg01167]
> > >
> > >What should be the next step?
> > >
> > >Changing the machine???
> > >Is it machine dependent?...
> > >
> > >Regards,
> > >Somesh S. Shanbhag
> > >
> > >  
> > >
> > Have you talked with Digium support?
> > 
> > Their answer almost always is:
> > 
> > "Try another Motherboard"
> > They won't supply a list that is known to work,
> only
> > ones that are known 
> > NOT to work.
> >  From my limited experience, even if the MB says
> it
> > is PCI 2.2, the TDM 
> > card may or may not work.
> > 
> > If you don't want to change machines, then  use an
> > ATA or two Sipura's 
> > work great.
> > 
> > John Novack
> > 
> > 
> > ___
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> > Asterisk-Users@lists.digium.com
> >
>
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> > To UNSUBSCRIBE or update options visit:
> >   
> >
>
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> > 
> 
> 
> 
>   
>   
>
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>   
>
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> 




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[Asterisk-Users] [Asterisk-User] linux/Asterisk change ip address

[Andrea Cristofanini - Gedam Europe Srl]

Hi list
i have a Asterisk box that use 10 phone with sccp, and some iax2
Every 8 10 hours , my linux machine change ip address and route, and the 
cisco and iax phone cannot see the server ...


What can do that?
there are no other linux box , no any pc that provide DHCP

--
Cheers Andrea

Andrea Cristofanini
Gedam Europe S.r.l.
Gedam Advanced Communication LTD
mobile : +39 3291871756
office : +39 011 5694900
MSN : [EMAIL PROTECTED]
http://www.gedameurope.com
http://www.asterisknews.it
http://freevoip.gedameurope.com

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Re: [Asterisk-Users] Any way to not overwrite sound files on compile?

[gincantalupo]

Hi,
just do NOT type "make samples": this commands writes original sounds in 
your dirs even if your old custom messages are present.
(I'm talking about 1.0.7 * version, maybe in newer * versions this 
command is included inside some install script).



Giorgio.


Matt wrote:


Every time I recompile Asterisk (or upgrade to a new CVS-HEAD,
whatever) asterisk overwrites custom files I have made.  Granted,
these files are named the same as the asterisk default files
(vm-login.gsm, etc) because we had a person here record them to
customize them a bit more for our application.

Short of keeping them somewhere and copying them back every time
(which isn't all that often) I do a re-compile.  Is there some flag or
something to tell Asterisk not to install sound files, or at the very
least not to overwrite ones already existing?
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Re: R: [Asterisk-Users] Problem setting up TDM22B card

[somesh s]

Hi,

I changed the mother board (MB) but it is giving still
the same problem.
 ouput of dmesg|tail 
f6 != 58
f7 != 59
f8 != 58
f9 != 59
fa != 58
fb != 59
fc != 58
fd != 59
fe != 58
Freshmaker failed register test
>>>
and I have also configured zaptel.conf correctly.

Whatz next? Can I assume that it is a hardware
problem?

Regards,
Somesh S. Shanbhag


--- John Novack <[EMAIL PROTECTED]> wrote:

> 
> 
> somesh s wrote:
> 
> >Hi,
> >
> >I didn't get any solution in the mailing list.
>
>[http://asterisk.linkx.net/asteriskusers/200409/msg01167]
> >
> >What should be the next step?
> >
> >Changing the machine???
> >Is it machine dependent?...
> >
> >Regards,
> >Somesh S. Shanbhag
> >
> >  
> >
> Have you talked with Digium support?
> 
> Their answer almost always is:
> 
> "Try another Motherboard"
> They won't supply a list that is known to work, only
> ones that are known 
> NOT to work.
>  From my limited experience, even if the MB says it
> is PCI 2.2, the TDM 
> card may or may not work.
> 
> If you don't want to change machines, then  use an
> ATA or two Sipura's 
> work great.
> 
> John Novack
> 
> 
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> 
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> Asterisk-Users@lists.digium.com
>
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>   
>
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> 





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Re: [Asterisk-Users] files conflict after CVS update

[Michiel van Baak]

On 14:58, Thu 29 Sep 05, Claudio Canseco wrote:
> Hi all,
>  I just updated zaptel and asterisk, but after doing CVS update of asterisk
> (from CVS-HEAD 2005-08-10 17:10:53) I got a file conflict message
> 
> conflicts:
> ast_expr2.h
> ast_expr2f.c
> 
> I ignored them, and then tried to compile asterisk: make clean, make , make
> install. But then the problem shows again, leaving this message:
> 
> ast_expr2f.c:1784: warning: no previous prototype for `ast_yyget_column'
> ast_expr2f.c:1860: warning: no previous prototype for `ast_yyset_column'
> ast_expr2.fl:95: error: conflicting types for `ast_expr'
> include/asterisk/ast_expr.h:26: error: previous declaration of `ast_expr'
> ast_expr2f.c:1259: warning: `yyunput' defined but not used
> make: *** [ast_expr2f.o] Error 1
> 
> Any suggestions about how to deal with this conflicts, and if I can't fix
> that conflict can i go back to my previous CVS version ? and how?
> I hope someone can give me a hint about this.
>  Thanks in advance

Hi,

Remove the 2 files, and update from CVS again :)

-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Why is it drug addicts and computer afficionados are both called users?"

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[Asterisk-Users] Re: NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1

[Lance Grover]

On 7/6/05, Lance Grover <[EMAIL PROTECTED]> wrote:
I am getting:NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 onPrimary D-channel of span 1on my asterisk box and it seems to be causing a poping sound in thephones, I am wondering if anyone can shed some light on this.  I have
scanned the archives and get possibilities ranging form motherboards,to pri, to loaded module problems.  Can someone tell me the best wayto start tracking this down?
Well I finally got this working properly so I just wanted to thank all
those that sent me suggestions.  It seems that this error can come
because of to many modules, or IRQ issues, or whatever but for me it
boiled down to the fact that in my /etc/zaptel.conf I was not using my
PRI as the primary time source, I had it as the secondary and I only
had one PRI attached.  I hate it when it is something little and
easy to fix, especially when it is your fault.  After changing the
config file so that it used the PRI as the primary time source the
error went away and I have not seen it since, it is nice.

Just for those that don't know what I am talking about this is what the offending line said in my zaptel.conf file:
span=1,2,0,esf,b8zs

This is what I changed it to so that now it works:
span=1,1,0,esf,b8zs

it was just that easy.-- Thanks,Lance Grover
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[Asterisk-Users] please help on FreeTDS (writing CDR to MS-SQL or MySQL)

[Andy Kuo]

Hi,
 
I'm trying to install FreeTDS.  I followed the instructions on http://www.voip-info.org/tiki-index.php?page=FreeTDS, but still can't get it to work.

I serched around trying to find instructions on it, and it seems the same info (even wording) appear on all sites I found.
 
I downloaded freetds-0.63, and followed the instructions step by step, and when I try to re-compile Asterisk (#make in /usr/src/asterisk), I got these errors:
 
cdr_tds.c: In function `mssql_connect':cdr_tds.c:429: error: `TDSCONNECTINFO' undeclared (first use in this function)cdr_tds.c:429: error: (Each undeclared identifier is reported only oncecdr_tds.c:429: error: for each function it appears in.)
cdr_tds.c:429: error: `connection' undeclared (first use in this function)cdr_tds.c:474: warning: implicit declaration of function `tds_free_connect'make[1]: *** [cdr_tds.o] Error 1make[1]: Leaving directory `/usr/src/asterisk/cdr'
make: *** [subdirs] Error 1
 
I think I might need to put in info such as the IP, DB name, user, password, etc. of my SQL server, but I'm not sure what the correct format is.
 
Can anyone please give me an example?  Are there anything else I do wrong here?   or Are there different versions of instructions on the topic?
 
Thank you all in advance.
AK
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RE: [Asterisk-Users] OOH323C

[Dan Austin]

Asking which H323 channel is the best turns out to be a deeply
personal issue, at least noting the responses in the past.

I've tried and used all three. Here are my thoughts-

Chan_h323 (the original)-
Did not work in our environment.  Known issues with Cisco's
Call Manager.  Other than the requirements for OpenH323 and
PWLib, it was easy to setup and configure.

Chan_oh323
Worked fine for us.  Has the same dependencies as chan_h323,
also easy to setup and configure.

Chan_h323 (ooh323c based)
This one has been a winner for us.  No dependencies on OpenH323
or PWLib, which while not terrible to build/setup, is extra effort
and can be tricky to match known working versions.
Setup and configuration has been very simple.  If you have configured
the other channels, this one should seem familiar.

A seperate note in favor of the new chan_h323 is the developer support.
I found a couple little bugs that related to our use of Cisco Call 
Manager, and expected little or no interest in getting them resolved.
I had a test version made available to me in just over a day and
complete resolution a few hours later.

Dan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kanishka
Somaratne
Sent: Thursday, September 29, 2005 7:28 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] OOH323C 

hi
has any one used OOH323C i tried this it is installed but do not know
how to 
configure has any one used this, what is the best h323 addon to use with

asterisk 

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Re: [Asterisk-Users] Is this normal?

[Mark Johnson]

This is off list...  I was really concerned about this, too!!  It turns 
out that it is some sort of clean up routine that runs once an hour.  If 
you have calls in progress on channels 3 and 4, those won't show up as 
restarted!!  Good Luck!


Mark

Matthew T. O'Connor wrote:

Hey, I'm up and running fine with 30 Polycom 500s connected to 
Asterisk 1.2Beta on Cent OS 4.1 with a Digium TE110 connected to a PRI 
line.  Nearly every hour, almost on the hour I get this:


Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/1 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/2 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/3 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/4 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/5 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/6 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/7 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/8 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/9 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/10 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/11 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/12 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/13 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/14 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/15 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/16 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/17 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/18 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/19 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/20 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/21 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/22 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/23 
successfully restarted on span 1


Is this normal?

Thanks,

Matthew

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Re: [Asterisk-Users] Is this normal?

[Kevin Bockman]

Matthew T. O'Connor wrote:
Hey, I'm up and running fine with 30 Polycom 500s connected to Asterisk 
1.2Beta on Cent OS 4.1 with a Digium TE110 connected to a PRI line.  
Nearly every hour, almost on the hour I get this:


Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/1 
successfully restarted on span 1


Yes
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[Asterisk-Users] Is this normal?

[Matthew T. O'Connor]

Hey, I'm up and running fine with 30 Polycom 500s connected to Asterisk 
1.2Beta on Cent OS 4.1 with a Digium TE110 connected to a PRI line.  
Nearly every hour, almost on the hour I get this:


Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/1 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/2 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/3 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/4 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/5 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/6 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/7 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/8 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/9 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/10 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/11 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/12 
successfully restarted on span 1
Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/13 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/14 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/15 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/16 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/17 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/18 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/19 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/20 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/21 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/22 
successfully restarted on span 1
Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/23 
successfully restarted on span 1


Is this normal?

Thanks,

Matthew

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[Asterisk-Users] Can't make outside call with SIP softphone

[David H]

Hi,

I am can make local extension to and from SIP X-Lite
softphone, but I can't dial out using X-Lite but local
analog works just fine. Here are my conf files any
idea? 
Thanks,
David

my sip.conf
[general]
bindport=5060   ; UDP Port to bind to
(SIP standard port is 5060)
bindaddr=0.0.0.0; IP address to bind
to (0.0.0.0 binds to all)
allow=all

[3000]
type=friend
allow=all
username=3000
secret=my_passwd
host=dynamic
context=sip
dtmfmode=rfc2833

my extension.conf

[globals]
davidHand=>Zap/1
davidVoicemail=>[EMAIL PROTECTED]
johnHand=>Zap/2
johnVoicemail=>[EMAIL PROTECTED]
davidout=>Zap/3
johnout=>Zap/4

[internal]
exten => 1000,1,Dial(${davidHand},10,r)
exten => 1000,n,Voicemail(u${davidVoicemail})
exten =>
1000,n,Playback(/var/lib/asterisk/sounds/vm-goodbye)
exten => 1000,n,Wait(1)
exten => 1000,n,Hangup()
exten => 1000,102,Voicemail(b${davidVoicemail})
exten => 1000,103,Hangup()

exten => 2000,1,Dial(${johnHand},10,r)
exten => 2000,n,Voicemail(u${johnVoicemail})
exten =>
2000,n,Playback(/var/lib/asterisk/sounds/vm-goodbye)
exten => 2000,n,Wait(1)
exten => 2000,n,Hangup()
exten => 2000,102,Voicemail(b${johnVoicemail})
exten => 2000,103,Hangup()

exten => 3000,1,Dial(SIP/3000,20,tr)
exten => 3000,n, Bye()

exten =>
i,1,Playback(/var/lib/asterisk/sounds/invalid)
exten => i,2,Goto(incoming,s,2)

exten =>
t,1,Playback(/var/lib/asterisk/sounds/vm-goodbye)
exten => t,2,Hangup()

[outgoing]
ignorepat => 9
exten => 9,1,Dial(Zap/3)
exten => 9,n,Congestion()
exten => 9,n,Hangup()

[voicemail]
exten => 2828,1,VoiceMailMain()
exten => 2828,n,Hangup()


[incoming]
exten => s,1,Answer()
exten =>
s,2,Background(/var/lib/asterisk/sounds/vm-enter-num-to-call)
include => internal


[sip]
include => internal

[default]
include => internal
include => outgoing
include => sip




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Re: [Asterisk-Users] SIP Gateway wants T38, Asterisk rejects but media path not established.

[Kevin P. Fleming]

Ray Van Dolson wrote:


Our SIP/PSTN gateway provider seems to think that Asterisk should initiate a
renegotiation to G711 when it sends the 488 message rejecting T38.


This is not correct. The 488 response 'cancels' the INVITE, so no codec 
change was ever actually involved. The gateway should continue sending 
G711 since the other device (Asterisk) did not accept the change.

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Re: [Asterisk-Users] Using Realtime queues and queue members

[Kevin P. Fleming]

William Boehlke wrote:

That's interesting. I haven't been paying attention to it but it has been my
understanding that Business Edition does not support realtime in the current
release. 


Asterisk Business Edition has supported Realtime since its first release.
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[Asterisk-Users] Meetme issue

[niles]

I've noticed that when I use the MeetMe app, it shows the Zap/pseudo  
context

in a fax context.  Anyone know what could cause this??

from my extensions.conf

exten => conf,1,Answer
exten => conf,2,SetMusicOnHold(rachmaninov)
exten => conf,3,Macro(isadmin)
exten => conf,4,GotoIf(${CONFADMIN}?5:7)
exten => conf,5,MeetMe(${CONF},aMXq); admin
exten => conf,6,Hangup
exten => conf,7,MeetMe(${CONF},AMXq); non-admin
exten => conf,8,Hangup


from console, show channels

Zap/pseudo-270871077 [EMAIL PROTECTED]:1 Rsrvd   (None)
IAX2/[EMAIL PROTECTED] [EMAIL PROTECTED]:5  Up  MeetMe(1701|aMXq)

Niles

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Re: [Asterisk-Users] Auto Answer Fax

[Scott Eisert]

I have been just working on the same thing today.

You can start by taking a look at this application for inbound IP faxes:

http://www.voip-info.org/tiki-index.php?page=NVFaxDetect

I can currently detect the fax but can't seem to capture it.

- Scott

On Thursday 29 September 2005 4:52 pm, Rene Nelson wrote:
> Can anyone point me to a good howto or example on how to get * to recognize
> inbound faxes and adjust accordingly? Ideally I would like it to grab the
> fax and email it to me, but I dont know if that is really possible yet or
> not.
>
> Thanks
>
> Neri

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[Asterisk-Users] Voice Prompts, what do you think? Good voice.

[gw]

Repost, first one never made it to the list... 

-Original Message-
From: Gregory Wiktor - ADCom Corp. 
Sent: Thursday, September 29, 2005 3:20 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Voice Prompts, what do you think? Good voice. Should we record
a new prompt-set?

Hello all,
I have someone working for me who has a nice phone voice.  I looked at
some available prompts for asterisk, and found both the free and
commercial ones to be pretty horrible.  The asterisk ones are good, but
I wish I had more to choose from sometimes.

My question is, what do you think, should I bother having her record a
full spectrum of prompts for asterisk?

If you want to hear her voice, the ivr/bri number is 1-914-693-0821

If it is something the community would want, I can spend the time with
her, otherwise I am fine with the regular voices.

If you wanted any custom prompts done, we could do it.  She works for me
anyway so its not a big deal, but for anything extensive I should give
her some money to do it.

This is not a studio setup, but we have a good microphone setup and the
quality even on BRI sounds great. I just need to spend some time
adjusting the audio levels on my bri and in the wav files.

For me, this worked out well, since my own voice I admit sounds very
anal on the phone.

Feel free to call and listen, but the timeout goes to fax.

There is a good chance I will do it, but want some feedback. What would
be especially helpful at this point would be suggestions for new prompts
for asterisk.  This way if I spend like 2 hours doing it with her, at
least I can cover any new prompts that may come up.

I like the Allison prompts, but sometimes they sound too sexy.  April's
voice(my employee) is a bit flirty, but I think it works in a good way.

If I did it I would probably give away a basic prompt set, but charge
$50 or $100 for the extra stuff like pin codes and things that are not
normally used for a personal nature.

Regards,
Gregory Wiktor
[EMAIL PROTECTED]
Web: www.adcomcorp.com
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[Asterisk-Users] SIP Gateway wants T38, Asterisk rejects but media path not established.

[Ray Van Dolson]

Disclaimer: Yes, I know faxing over G711 is unreliable. :-)

We're running Asterisk 1.0.9 which talks to a Audiocodes SIP Gateway.  We're
running Sipura SPA-2002's as ATA's and faxing within our own voice network is
working.  If we try and fax out to the world however, we're running into a
problem.

When the call connects and the modem tones begin to negotiate, our SIP/PSTN
Gateways's SIP server determines it's a fax call and sends an INVITE to our
Asterisk box asking us to negotiate T38.  To this, Asterisk replies with a
488: Not acceptable here message.  There is no SDP payload in this 488
message.  The SIP/PSTN gateway sends an ACK to this.

At this point, nothing happens.  Eventually Asterisk gets bored and sends a
BYE to the SIP/PSTN gateway and the call ends.

Our SIP/PSTN gateway provider seems to think that Asterisk should initiate a
renegotiation to G711 when it sends the 488 message rejecting T38.

Is this how it should work?  I've asked them to look into disabling T38
support for us completely, but it seems like Asterisk and the SIP/PSTN gateway
should be able to negotiate G711, I just don't know what the proper sequence
of events should be.

Any ideas?

Ray
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[Asterisk-Users] TDMoE + kernel badness

[astgroups]

"Badness in local_bh_enable at kernel/softirq.c on 2.6.X"

I'm seeing this on Kernel 2.6.+ implementations, namely Centos 4.1
machines while trying to do TDMoE trunks between two machines. 
2.4 Kernel operates fine on the same hardware

I'm compiling zaptel-1.0.9.2 as per instructions in README.Linux26 +
README.udev

Here are some references where the issue has been reported before but
I've yet to find a documented solution; Any tips?

http://lists.digium.com/pipermail/asterisk-users/2005-February/091867.html

http://bugs.digium.com/view.php?id=5126


Thanks.

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Re: [Asterisk-Users] Meet me conferencing without blind transfers ([EMAIL PROTECTED])

[Mojo with Horan & Company, LLC]

rather than using the blind and attended transfer functions built into 
the phones, try asterisk's features.conf -- we use ** for attended and 
## for blind. (always use blind for meetme conferences.)


as for the difference between attended and blind, here's blind:

Person A calls Person B and asks for Person C.  Person B does a blind 
transfer to Person C's extension.  This disconnects Person B from the 
conversation and puts Person A on hold.  Person C's phone rings, and 
when he picks it up, Person A will be there.


here's attended:

Person A calls Person B and asks for Person C.  Person B does an 
attended transfer to Person C's extension.  This disconnects Person B 
from the conversation and puts Person A on hold.  Person C's phone 
rings, and when he picks it up, Person B will be there.  Person B 
introduces Person A and/or otherwise prepares Person C for the 
conversation.  If Person C does not want to talk to Person A, Person C 
hangs up, and Person B is reconnected to Person A.  If, however, Person 
C _does_ want to talk to Person A, Person _B_ hangs up.  Persons A and C 
 are now connected.


Moj

jennyw wrote:

Hi,

I'm using [EMAIL PROTECTED] and am having trouble using the conference 
bridge that comes built in. We're using Polycom phones.


When we transfer the first person into the conference room (e.g. 8101) , 
they get into the room fine. When we try to transfer a second person 
into the conference room, they get dropped as soon as we finish the 
transfer.  This is using Polycom SoundPoint 301 phones.  Now when we use 
a Polycom SoundPoint 501 phone, we have an option to do a blind transfer 
when we transfer someone (appears the first time you hit the transfer 
button). When we do this, people stay transferred and everything works 
great. However ... we only have one 501 and the rest are 301s. So my 
questions are:


* What's the difference between a blind transfer and a regular transfer?

* Is it possible to do a blind transfer from a Polycom SoundPoint 301?

* Is there another way to get this to work?

Thanks!

Jen

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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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RE: [Asterisk-Users] Using Realtime queues and queue members

[William Boehlke]

That's interesting. I haven't been paying attention to it but it has been my
understanding that Business Edition does not support realtime in the current
release. 

However, if it does have realtime support, we are responsible for dozens of
servers running realtime with MySQL (and Apache, for that matter) on the PBX
without a single instance of instability from either of those applications. 

You don't state what you define as "extremely high call volumes" but if you
expect to be pushing the limit of your server capacity, just add another
one. Hardware is cheap enough.

Good luck!

William Boehlke
Signate



 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Karl
Sent: Thursday, September 29, 2005 3:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Using Realtime queues and queue members

Hello,

We are about to attempt implementing realtime queues using Asterisk Business
Edition and are wondering if anyone has had any experiences or
recommendations /warnings with using this feature.

One issue we're trying to decide on is if we should have a MySQL server
running on the Asterisk server or a remote server to house the
database/tables for the queue settings/members. The concerns with housing
the tables on the Asterisk server are mostly resources; On the remote server
stability (if the remote server crashes or loses network connection will
Asterisk crash?) or the loss of members/queues (if the Asterisk server
remains running but can't connect to the remote server to do queries). We
will be having extremely high call volume going through the Asterisk server.

Any information would be greatly appreciated.

Thank You,

Timothy Karl
InterMedia Marketing Solutions
Programming / Research and Development

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[Asterisk-Users] Meet me conferencing without blind transfers ([EMAIL PROTECTED])

[jennyw]

Hi,

I'm using [EMAIL PROTECTED] and am having trouble using the conference 
bridge that comes built in. We're using Polycom phones.


When we transfer the first person into the conference room (e.g. 8101) , 
they get into the room fine. When we try to transfer a second person 
into the conference room, they get dropped as soon as we finish the 
transfer.  This is using Polycom SoundPoint 301 phones.  Now when we use 
a Polycom SoundPoint 501 phone, we have an option to do a blind transfer 
when we transfer someone (appears the first time you hit the transfer 
button). When we do this, people stay transferred and everything works 
great. However ... we only have one 501 and the rest are 301s. So my 
questions are:


* What's the difference between a blind transfer and a regular transfer?

* Is it possible to do a blind transfer from a Polycom SoundPoint 301?

* Is there another way to get this to work?

Thanks!

Jen

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[Asterisk-Users] Using Realtime queues and queue members

[Tim Karl]

Hello,

We are about to attempt implementing realtime queues using Asterisk
Business Edition and are wondering if anyone has had any experiences or
recommendations /warnings with using this feature.

One issue we're trying to decide on is if we should have a MySQL server
running on the Asterisk server or a remote server to house the
database/tables for the queue settings/members. The concerns with
housing the tables on the Asterisk server are mostly resources; On the
remote server stability (if the remote server crashes or loses network
connection will Asterisk crash?) or the loss of members/queues (if the
Asterisk server remains running but can't connect to the remote server
to do queries). We will be having extremely high call volume going
through the Asterisk server.

Any information would be greatly appreciated.

Thank You,

Timothy Karl
InterMedia Marketing Solutions
Programming / Research and Development

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Re: [Asterisk-Users] Re: * mod core dump help

[Kevin Bockman]

Matt wrote:

hmm, I'm not running in safe mode.

looked at /tmp, no sign of core. Linux is strange, there should be a core
file somewhere... i searched the whole / by *core*.


He means by running safe_asterisk.  You need to run asterisk with -g at 
least to get it to produce a core file.  If you run safe_asterisk, it 
does run with -g plus it will restart Asterisk if it dies.



Kevin
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Mathematicians wanted (was RE: [Asterisk-Users] Best echo canceller?)

[Kris Boutilier]

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Andrew
> Kohlsmith
> Sent: Thursday, September 29, 2005 2:23 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Best echo canceller?
> 
> 
> On Thursday 29 September 2005 17:04, Claudio Canseco wrote:
> >  In your experience what is the best choice for echo canceller ?
> > Which one should work better: STEVE, STEVE2, MARK, MARK2, MARK3, KB1 ?
> 
> KB1 is a refactored MARK2 which seems to work VERY, very well.
> 
{clip}

The refactoring applied to MARK2 to create KB1 was basically intended to make 
the code generally consistent with the Texas Instruments whitepaper referenced 
in the comments at the top of the file. In that document they completely 
outline the operating theory of one particular echo cancellation algorithm and 
completely document an implementation of it in a general purpose TI processor. 
The implementation is also benchmarked and deviations/performance issues 
explored. The MARK2/KB1 implementation is not a 100% complete version of the 
reference code - there are some autotuning elements and perhaps additional 
optimizations suggested by TI that can certainly still be implemented.

I would strongly encourage anyone with a good understanding of mathematics to 
take a look at the whitepaper, compare the KB1 the source and see what can be 
improved on. The echo canceller code itself isn't complicated but the math is 
somewhat of a dark art... Just make sure to add copious plain English 
explanations to any changes you submit so the rest of us can keep up.

:-)
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Re: [Asterisk-Users] Tiny Echo on PSTN via Zaptel

[Shaw Terwilliger]

On Thu, Sep 29, 2005 at 04:22:39PM -0500, Neil Lewis wrote:
> FXOTune is only for TDM cards.  I suggest you try one of the other
> methods listed.

I've tried them all so far.  What I'm hearing sounds a lot like (too loud)
sidetone, but there's just the smallest bit of delay.  It's only noticeable
when I'm bridged to a PSTN line.  If I plug a phone directly into one of
those PSTN lines, there is no problem.

Changing the echo settings has improved it a bit, so it probably is
some sort of echo.

-- 
Shaw Terwilliger <[EMAIL PROTECTED]>
SourceGear LLC


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Re: [Asterisk-Users] Re: * mod core dump help

[Matt]

hmm, I'm not running in safe mode.

looked at /tmp, no sign of core. Linux is strange, there should be a core
file somewhere... i searched the whole / by *core*.

Best Regards

Matt

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__

- Original Message - 
From: "Gustavo A. Gonzalez" <[EMAIL PROTECTED]>
To: 
Sent: Thursday, September 29, 2005 6:04 AM
Subject: [Asterisk-Users] Re: * mod core dump help


> You could  look up at '/tmp' if you are runing * in safe mode ...guess
that
> help you
>
> G.
>
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Re: [Asterisk-Users] Asterisk Echo problems, Urgent, please help,

[Matt]

hi:

We are using 1.0.9 * with sangoma 104 quad card, hooked to 4 E1s. We have no
echo problems at all.

The voice qualities sound and clear, try adjust tx/rxgain a bit. and make
sure your zapata.conf's echocancel param is enabled.

Best Regards

Matt

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

Sent: Thursday, September 29, 2005 6:02 AM
Subject: Re: [Asterisk-Users] Asterisk Echo problems, Urgent, please help,


> What kind of POTS trunks/cards are you using?
>
> --
> Tom
>
> On 9/29/05, Ian Bonham <[EMAIL PROTECTED]> wrote:
> > Hi all,
> >
> > I hope someone can help, as I have an urgent problem.
> >
> > I've got a production Asterisk server thats been deployed, but we are
seeing
> > a strange voice echo problem. There is about a 250ms echo for the users
in
> > the office, and they are hearing their own voice back at them.
> >
> > I'm running the CVS Head code, on RH9.0. This is on a P4 box with 2gb of
> > memory. The client SIP phones are Polycom Soundpoint IP600's, WiFi ZyXel
> > 2000w handsets, and X-Lite (free) PC clients. All see the same problem.
> > There is a bridge into the POTS (BT's SystemX) using a Voicetronix
> > OpenSwitch12 card and the vpbhp driver.
> >
> > The echo occurs on both SIP->POTS calls, and SIP->SIP calls. I've tried
a
> > number of volume adjustments to correct the echo but it is always the
same.
> >
> > If anyone has any ideas I'd really appriciate some help, as this is a
major
> > urgency,
> >
> > Many many thanks,
> >
> > Ian Bonham
> >
> > _
> > FREE pop-up blocking with the new MSN Toolbar - get it now!
> > http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/
> >
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>
>
> --
> Tom
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>

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[Asterisk-Users] files conflict after CVS update

[Claudio Canseco]

Hi all,
 
I just updated zaptel and asterisk, but after doing CVS update of asterisk (from CVS-HEAD 2005-08-10 17:10:53) I got a file conflict message

conflicts:ast_expr2.hast_expr2f.c
I ignored them, and then tried to compile asterisk: make clean, make , make install. But then the problem shows again, leaving this message:

ast_expr2f.c:1784: warning: no previous prototype for `ast_yyget_column'ast_expr2f.c:1860: warning: no previous prototype for `ast_yyset_column'ast_expr2.fl:95: error: conflicting types for `ast_expr'include/asterisk/ast_expr.h:26: error: previous declaration of `ast_expr'
ast_expr2f.c:1259: warning: `yyunput' defined but not usedmake: *** [ast_expr2f.o] Error 1
Any suggestions about how to deal with this conflicts, and if I can't fix that conflict can i go back to my previous CVS version ? and how?
I hope someone can give me a hint about this.
 
Thanks in advance
Regards,
Claudio
 
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[Asterisk-Users] dtmfmode type

[Joseph]

I've noticed that Asterisk only supports the following DTMF modes:
inband, rfc2833, info, auto.

Auto mode is broken. 
Though, Sipura unit have the following options:
inband
AVT
info
auto
Inband + info
AVT + info

What is AVT?
How does for example "inband + info" works, it either/or?

-- 
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Re: [Asterisk-Users] Best echo canceller?

[Andrew Kohlsmith]

On Thursday 29 September 2005 17:04, Claudio Canseco wrote:
>  In your experience what is the best choice for echo canceller ?
> Which one should work better: STEVE, STEVE2, MARK, MARK2, MARK3, KB1 ?

KB1 is a refactored MARK2 which seems to work VERY, very well.

The others are different attempts at different algorithms.  KB1's the new 
default, from MARK2.  Try the others, see if they work better for you.

-A.
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Re: [Asterisk-Users] Best echo canceller?

[Neil Lewis]

KB1 is pretty awsome.  Its relatively new, but in my experience its the
best out.   As a matter of fact, its been made the default echo can for
CVS-Head

NRL

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Re: [Asterisk-Users] Tiny Echo on PSTN via Zaptel

[Neil Lewis]

FXOTune is only for TDM cards.  I suggest you try one of the other
methods listed.

NRL
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Re: [Asterisk-Users] Auto Answer Fax

[Tom Hayden]

You can use SpanDSP.

http://www.voip-info.org/tiki-index.php?page=spandsp

--
Tom

On 9/29/05, Rene Nelson <[EMAIL PROTECTED]> wrote:
> Can anyone point me to a good howto or example on how to get * to recognize
> inbound faxes and adjust accordingly?  Ideally I would like it to grab the
> fax and email it to me, but I dont know if that is really possible yet or
> not.
>
>  Thanks
>
>  Neri
>
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Re: [Asterisk-Users] minor(? ) Grandstream phone issue

[John Novack]

Shawn Rutledge wrote:



- When there is a ring on the landline, asterisk gets the Grandstream to ring.  
But then if I pick up an analog phone directly on the landline to answer the 
call, it doesn't notice very fast, and continues to ring 2 or 3 times.  I'm not 
sure if this is an asterisk issue or something else.

Certainly an Asterisk issue, but not uncommon with many telephone 
systems where an unsupervised telephone is  sharing a line.
Asterisk doesn't look at the DC state of a line  to determine if there 
has been a polarity reversal or a transition to a lower level ( such as 
when you pick up another phone ) so it has no way of knowing other than 
the ring signal disappearing.
I suspect there is some way to shorten that interval. Given the 
different ring cadences around the world it probably should not be too 
short though.


John Novack

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[Asterisk-Users] Best echo canceller?

[Claudio Canseco]

hi!
 
In your experience what is the best choice for echo canceller ?
Which one should work better: STEVE, STEVE2, MARK, MARK2, MARK3, KB1 ?
 
Regards,
Claudio
 
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RE: [Asterisk-Users] Broadvoice inbound issues

[michel vank]

I had the same problem, because the context I used for my FWD service did not contain the DID extensions.
For example, I added the following to my extensions.conf
[fromiaxfwd]include => ext-didwhere "fromiaxfwd" is the context of the the FWD trunk
Also don't forget to setup a DID numbermichel




From:  Rene Nelson <[EMAIL PROTECTED]>Reply-To:  Rene Nelson <[EMAIL PROTECTED]>,Asterisk Users Mailing List - Non-Commercial DiscussionTo:  Asterisk Users Mailing List - Non-Commercial DiscussionSubject:  [Asterisk-Users] Broadvoice inbound issuesDate:  Thu, 29 Sep 2005 14:28:30 -0600MIME-Version:  1.0Received:  from lists.digium.com ([69.16.138.164]) by mc1-f1.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Thu, 29 Sep 2005 13:32:16 -0700Received:  from [69.16.138.164] (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 29D6B2FC8DC;Thu, 29 Sep 2005 15:28:38 -0500 (CDT)Received:  from 
psmtp.com (exprod5mx143.postini.com [64.18.0.98])by lists.digium.com (Postfix) with SMTP id 6C5892FC8CDfor ;Thu, 29 Sep 2005 15:28:22 -0500 (CDT)Received:  from source ([64.233.184.196]) by exprod5mx143.postini.com([64.18.4.10]) with SMTP; Thu, 29 Sep 2005 13:28:30 PDTReceived:  by wproxy.gmail.com with SMTP id 70so41064wrafor ;Thu, 29 Sep 2005 13:28:30 -0700 (PDT)Received:  by 10.54.13.26 with SMTP id 26mr658398wrm;Thu, 29 Sep 2005 13:28:30 -0700 (PDT)Received:  by 10.54.123.16 with HTTP; Thu, 29 Sep 2005 13:28:30 -0700 (PDT)
My SIP seems to be configured correctly as I can dial out and my
minutes show up on my broadvoice bill, but whenever anyone calls my
broadvoice # inbound they just get a busy signal.  I dont get
anything in the logs saying anything came in from broadvoice at all.

Has anyone had this/simmilar problem with inbound from Broadvoice?  Any suggestions?

Thanks

Neri


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[Asterisk-Users] Auto Answer Fax

[Rene Nelson]

Can anyone point me to a good howto or example on how to get * to
recognize inbound faxes and adjust accordingly?  Ideally I would
like it to grab the fax and email it to me, but I dont know if that is
really possible yet or not.

Thanks

Neri
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Re: [Asterisk-Users] Broadvoice inbound issues

[Rene Nelson]

THAT WAS IT!  THANKS A MILLION!  I've been banging my head
for 3 hours on this trying to see what I did wrong from thier
example.  On 9/29/05, Klaus Sonnenleiter <[EMAIL PROTECTED]> wrote:
I've had the same problem. But it was due to a silly mistake: I followed the instructions ontheir web site literally (should have known better ;)). For asterisk as your BYOD device, theytell you all the steps except for one - you need to include an instruction in your dialplan
for the incoming number. So if your number is 123-4567 in the 890 area code, you'd needsomething likeexten => 8901234567,1,Answerexten => 8901234567,2,Dial(Zap/g1,20,20)exten => 8901234567,3,Hangup
in your dialplan. (example for forwarding the call to an analog phone, but works the same fora SIP extension). It's missing from the docs, but after inserting this section, everythingstarted working for me.
HTHKlaus--- Rene Nelson <[EMAIL PROTECTED]> wrote:> My SIP seems to be configured correctly as I can dial out and my minutes> show up on my broadvoice bill, but whenever anyone calls my broadvoice #
> inbound they just get a busy signal. I dont get anything in the logs saying> anything came in from broadvoice at all.>> Has anyone had this/simmilar problem with inbound from Broadvoice? Any
> suggestions?>> Thanks>> Neri> > ___> --Bandwidth and Colocation sponsored by Easynews.com --
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> To UNSUBSCRIBE or update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] minor(? ) Grandstream phone issue

[Shawn Rutledge]

I don't have your problem with not hanging up.  And I do get a dial
tone.  It even stutters when there is voicemail.  I have an older one,
BT-100 I think.

I do have a few other beefs with it though:

- The display backlight times out too fast normally.  I wish it would
stay lit for a few minutes at least, or maybe all the time.  (It
should be configurable.)
- The backlight flashing when there is voicemail is too annoying.  The
message button doesn't light up at all; I wish it would flash, instead
of the display.
- When there is a ring on the landline, asterisk gets the Grandstream
to ring.  But then if I pick up an analog phone directly on the
landline to answer the call, it doesn't notice very fast, and
continues to ring 2 or 3 times.  I'm not sure if this is an asterisk
issue or something else.
- When I use the grandstream to talk on the land line, people tell me
I don't sound very good.  Haven't figured out yet whether this is bad
choice of codec, bad implementation of the mic circuit on the phone,
or something with the interface to the land line.  I suspect the mic
circuit is an area where they may very well have cut corners.

I'm a newbie, and with the complexity of this system I think I'm going
to be a newbie for a while yet.

So is there a similarly affordable ethernet phone that gets everything
right? Overall I think the Grandstream was pretty innovative when it
first came out, but it's too bad the firmware isn't open source,
because most of my beefs could be fixed in software.
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Re: [Asterisk-Users] Broadvoice inbound issues

[Samy Antoun]

Rene,
Try this registration string

1234567890:[EMAIL PROTECTED]/200

Where:
1234567890 = Your BroadVoice number
mypassword = Your BroadVoice password
200 = Your internal extension that will recieve the
incoming calls





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RE: [Asterisk-Users] FWD: '486 Busy here' and 'All Circuits are busynow'

[Hector Elias Menjivar]

Hi:

    Same whit me….can anybody help us

 

    Hector

 

-Mensaje original-
De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Zeeshan
Enviado el: Jueves, 29 de
Septiembre de 2005 01:28 p.m.
Para: 'Asterisk Users Mailing List
- Non-Commercial Discussion'
Asunto: [Asterisk-Users] FWD: '486
Busy here' and 'All Circuits are busynow'

 

Hi,

 

I’ve set up
FreeWorldDialup on my asterisk server but when I dial the service numbers, I
get message ‘486 Busy Here ‘. When I dial any other number, it says 'All
Circuits are busy now'. What is the problem with my settings. I’ve followed all
the instructions step by step.

 

Zeeshan








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Re: [Asterisk-Users] Broadvoice inbound issues

[Klaus Sonnenleiter]

I've had the same problem. But it was due to a silly mistake: I followed the 
instructions on
their web site literally (should have known better ;)). For asterisk as your 
BYOD device, they
tell you all the steps except for one - you need to include an instruction in 
your dialplan
for the incoming number. So if your number is 123-4567 in the 890 area code, 
you'd need
something like

exten => 8901234567,1,Answer
exten => 8901234567,2,Dial(Zap/g1,20,20)
exten => 8901234567,3,Hangup

in your dialplan. (example for forwarding the call to an analog phone, but 
works the same for
a SIP extension). It's missing from the docs, but after inserting this section, 
everything
started working for me.

HTH

Klaus

--- Rene Nelson <[EMAIL PROTECTED]> wrote:

> My SIP seems to be configured correctly as I can dial out and my minutes
> show up on my broadvoice bill, but whenever anyone calls my broadvoice #
> inbound they just get a busy signal. I dont get anything in the logs saying
> anything came in from broadvoice at all.
> 
> Has anyone had this/simmilar problem with inbound from Broadvoice? Any
> suggestions?
> 
> Thanks
> 
> Neri
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Re: [Asterisk-Users] Asterisk for "Man-In-The-Middle" Trunk Side Call Recording?

[Matthew Crocker]

1. Is this possible? Can Asterisk serve as a man-in-the-middle  
between a

traditional PBX and the telco loops?


Yes,  Asterisk can be a PRI client to the telco and a PRI server to  
your internal PBX


2. If it's possible, which of the two possibilities above is  
better? For us,
rack space and power should not be an issue, we're interested in  
what will

be most workable.


I would go for multiple low end servers handling 1 or 2 T1s at a  
time.  You have g.711 coming in and going out so recording in g.711  
won't be too CPU intensive.  23 voice calls at 64kbps would be  
2.944mbps  (23 ^* 64000 * 2) so you'll need a hard drive/filesystem  
that can sustain at least 3mbps writes per T1 in the machine.




3. Once you have the calls recorded, how do you access them? Is it  
possible

to use asterisk's logs to correlate calls recordings with specific
information? We're willing to do some custom programming here if  
need be.


You should be able to tie the cdr information stored in a mysql  
database with the .wav file of the recording.  It is all setup in the  
extensions.conf file so you can choose the recording naming scheme  
there and make it work.  You would probably need to develop a web  
based app to handle the front end stuf.




4. Has anyone done anything like this before? If so, would you be  
interested

in a consulting gig?


I haven't.

One thing to watch out for is the new digium T1 cards that can bridge  
bearer traffic in firmware.  If that happens the voice doesn't leave  
the card and may not be recordable.  I'm not sure how Asterisk would  
handle that.




Thanks in advance for your responses!


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Internet Division
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[Asterisk-Users] FWD via Trunk from DMZ to LAN

[michel vank]

Dear users
 
I have setup 2 asterisk servers, one on the internal LAN and another one in the DMZ. I have configured both servers via [EMAIL PROTECTED] and IAX2 trunks that is all working fine.. EXCEPT! 
 
The server in the DMZ has a trunk to FWD. This trunk is working fine, also the clients on the internal LAN can use this trunk to place calls. I have set up a IAX2 trunk from the internal server to the DMZ server, and some routing rules to to route the calls to FWD. 
 
My problem is, that if somebody calls our FWD number and the call arrives on the DMZ server, I cannot get the call to be routed to a calling group on the internal server.
 
For example: 
FWD clients calls us, asterisks on DMZ answers the call and I want to route the call over the IAX2 trunk to the internal network. 
 
Do you understand what I mean? And how do I do this? (sorry! newbie!)
 
Michel

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[Asterisk-Users] Cisco AS5300 --> [SIP] --> Asterisk - NO AUDIO

[Steve Ducat]

OK, here goes my next problem.

I have puchased a DID which I can connect to via SIP

I have been given the following details:

Username: uka1xx
Password: 1000xx

Server: br.net:5160

My equipment is Asterisk CVS HEAD on Red Hat EL 3.0 (NO NAT)

The other end is a Cisco AS5300 (NO NAT)

I can register with the Cisco with no problem.

When I dial the DID it sends the call to my asterisk server and my
asterisk server sends back the dial tone, no problem.

The problem is when I pick up the phone, no audio.

If I change the dial plan to do a Playback instead of Dial an
extension I can see in the console it answers the call and starts to
play the Playback but no audio.

I can connect direclty to the Cisco AS5300 with sjphone or a budgetone
102 with no problem and get dial tone and full audio both ways but
when I use the asterisk no audio.

I have tried every codec possible. I have installed g729, g723 with no
luck. I have tested both these codecs by forcing my budgetone to use
with no problem so I know the codecs work.

So the problem is when I ask asterisk to register to the Cisco AS5300
as a SIP Client it does everything right except pass the audio.

There is no firewall configured.

I know the Cisco SIP Server works because it works with the softphone
SJPHONE and directly with the Budgetone 102.

I have reinstalled Asterisk so many times.

I have reinstalled g729 & g723 so many times.

The SIP debug output is pasted below.

I have been struggling with this for weeks with no luck.

Any help would be appreciated.

Steven Ducat.


*

<-- SIP read from 203.88.192.42:5160:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP 203.88.192.42:5160;branch=z9hG4bK5aeb.8f7e6027.0
Via: SIP/2.0/UDP  211.147.240.237:5060;rport=57786
From: ;tag=1CA65AC-9C8
To: 
Date: Thu, 29 Sep 2005 20:14:40 GMT
Call-ID: [EMAIL PROTECTED]
Supported: timer,100rel
Min-SE:  1800
Cisco-Guid: 2153363387-811340250-2169109749-53752559
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 5
Remote-Party-ID:
;party=calling;screen=yes;privacy=off
Timestamp: 1128024880
Contact: 
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 432
P-hint: Proxied
P-hint: usrloc applied

v=0
o=CiscoSystemsSIP-GW-UserAgent 5786 3481 IN IP4 211.147.240.237
s=SIP Call
c=IN IP4 211.147.240.237
t=0 0
m=audio 37708 RTP/AVP 18 4 3 8 0 110
c=IN IP4 203.88.192.42
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:110 X-NSE/8000
a=fmtp:110 192-194
a=direction:passive
a=direction:active
a=nortpproxy:yes

--- (24 headers 19 lines)---
Using INVITE request as basis request -
[EMAIL PROTECTED]
Sending to 203.88.192.42 : 5160 (non-NAT)
Found no matching peer or user for '203.88.192.42:5160'
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 110
Peer audio RTP is at port 211.147.240.237:37708
Found description format G729
Found description format G723
Found description format GSM
Found description format PCMA
Found description format PCMU
Found description format X-NSE
Capabilities: us - 0x100 (g729), peer - audio=0x30f
(g723|gsm|ulaw|alaw|g729|speex)/video=0x0 (nothing), combined - 0x100
(g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
Looking for 84104214 in default (domain 70.84.200.204)
list_route: hop: 
Transmitting (no NAT) to 203.88.192.42:5160:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
203.88.192.42:5160;branch=z9hG4bK5aeb.8f7e6027.0;received=203.88.192.42
Via: SIP/2.0/UDP  211.147.240.237:5060;rport=57786
From: ;tag=1CA65AC-9C8
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: 
Content-Length: 0


---
-- Executing Dial("SIP/211.147.240.237-b7116c10", "Local/2001/n")
in new stack
-- Executing Macro("Local/[EMAIL PROTECTED],2",
"oneline|SIP/stevenducat") in new stack
-- Executing Dial("Local/[EMAIL PROTECTED],2",
"SIP/stevenducat|20") in new stack
-- Called 2001/n
We're at 70.84.200.204 port 14922
Answering/Requesting with root capability 0x100 (g729)
12 headers, 8 lines
Reliably Transmitting (NAT) to 83.146.11.93:60073:
INVITE sip:[EMAIL PROTECTED]:18234 SIP/2.0
Via: SIP/2.0/UDP 70.84.200.204:5060;branch=z9hG4bK502b5287;rport
From: "0017911" ;tag=as2c8caf36
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 29 Sep 2005 20:18:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 164

v=0
o=root 14260 14260 IN IP4 70.84.200.204
s=session
c=IN IP4 70.84.200.204
t=0 0
m=audio 149

[Asterisk-Users] Broadvoice inbound issues

[Rene Nelson]

My SIP seems to be configured correctly as I can dial out and my
minutes show up on my broadvoice bill, but whenever anyone calls my
broadvoice # inbound they just get a busy signal.  I dont get
anything in the logs saying anything came in from broadvoice at all.

Has anyone had this/simmilar problem with inbound from Broadvoice?  Any suggestions?

Thanks

Neri
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Re: [Asterisk-Users] minor(? ) Grandstream phone issue

[Casey Boone]

i am having a problem with grandstream as well, mainly in the web 
interface to configure the units. it isnt helped by the fact that unless 
they fixed it today their website is broken on the support page where 
you can download updated firmware.


i have been in contact with their tech support department and have been 
given some beta firmware to test with so hopefully it will resolve the 
web interface issues i have (the interface just stops working after a 
random amount of time, except on one gxp2000 in which it works for about 
15 seconds after the unit starts up, then quits)


i havent had bad voice quality i have to say at least.  that part has 
seemingly worked rather well


Casey Boone

Ade Agbero wrote:
I have had numerous problems with Grandstream HT-386 new and old 
firmware, my convidence in Grandstream is at a very low point right now.
 
I wish you luck,
 
Ade.


*/Bob Weber <[EMAIL PROTECTED]>/* wrote:

I hate to bother the list with this potentially minor issue but
I just wonder if it's a symtom of some other problem.

Every time I make a call the BT-102, with the latest firmware, she just
keeps the LED display lit and the timer counting after hangup.
I check the CLI and the hangup is being executed, I certainly was
concerned
it might be keeping the line open but that doesn't seem to be the case.

I thought that she should go back to the date/time display but that
doesn't
happen. I don't have silence suppresion on and I've gone over the other
configs compared with what's on voip-info many time.
One other thing, there isn't a dial tone when I pick it up. It seems to
'work'; both inbound and outbound.
It's connected to * 1.09 which is registered @broadvoice.

Thanks for any insights bestowed on this noob :)
Bob
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 NEW - crystal clear PC to PC calling worldwide with voicemail 






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RE: [Asterisk-Users] * mod core dump help

[Alberto Risco]

Look in the /tmp dir.  If not, do a find /
-name “core.*”   Look at the date on the core.

 

 

Alberto

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Thursday, September 29, 2005
12:17 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] * mod
core dump help



 



hi all:





 





my written * module crashed core dumppted





 





how do you guys find where the * core dump located at which
folder?





 





Matt







The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.
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[Asterisk-Users] TDM40B - "Unable to play dialtone on channel X" ?

[Derek Conniffe]

Hi all,

Today my TDM40B (a TDM400 with 4 FXS modules) has gone funny - there is 
no dialtone from any port.


When I look at the CLI display in * and pick up a line it says "Sep 29 
20:36:25 WARNING[1093299120]: chan_zap.c:5313 handle_init_event: Unable 
to play dialtone on channel 3" and it does this on, and gives this 
message for, every channel.


Its a bit weird because I have not changed the configuration of asterisk 
at all and cables were not even unplugged.


Does anyone know why this is happening?

thanks,

Derek

--
Derek Conniffe
Rivertower Ltd
DID Number: 01 440 1806 (International: 00 353 1 440 1806)
Ireland: (Freephone) 1800 719 400
Ireland: (Local) 01 440 1800
United Kingdom: 0870 068 2368
International: 00 353 1 440 1800
Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823)
Fax: 01 201 0085 (International: 00 353 1 201 0085)
Email: [EMAIL PROTECTED]
Web: http://www.rivertowerhosting.com

begin:vcard
fn:Derek Conniffe
n:Conniffe;Derek
org:Rivertower Ltd;IT
adr:Dublin 2;;46 Upper Mount Street;Dublin;Dublin;Dublin 2;Ireland
email;internet:[EMAIL PROTECTED]
tel;work:+353 1 201 0146
tel;fax:+353 1 201 0085
tel;cell:+353 86 856 3823
note;quoted-printable:Ireland: (Freephone) 1800 719 400=0D=0A=
	Ireland: (Local) 01 244 9719=0D=0A=
	United Kingdom: 0870 068 2368=0D=0A=
	International: 00 353 1 244 9719=0D=0A=
	
url:http://www.rivertowerhosting.com
version:2.1
end:vcard

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RE: [Asterisk-Users] NAT

[Anders Svensson]

Thanks!!

Problems solved.

Anders

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Samy Antoun
Sent: den 29 september 2005 20:41
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] NAT

Anders,

There are 2 ways to acomplish this:
1. Keep your Asterisk box behinde your router. In this
case you need to do this:
 - Port Forward to Asterisk Box
   UDP 5060
   UDP 1-2

 - sip.conf (Change as needed)
   externip=xxx.xxx.xxx.xxx
   localnet=192.168.1.0/255.255.255.0

 - Extensions
   nat=yes
   qualify=yes

2. Configure your Asterisk box as a router. In this
case you need:
 - Asterisk box must have 2 NIC's
 - A network switch
 - Setup instructions can be found at:
http://samyantoun.50webs.com/asterisk/firewall/firewall.htm

Hope this helps.





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[Asterisk-Users] Yada table in oracle

[Han van Hulst]

I 
just compiled yada and cdr_yada
Everything is 
working only I get an error when asterisk tries to insert a record into the 
database
 
Sep 29 21:05:24 
ERROR[5091]: cdr_yada.c:445 cdr_yada_log: Failed to insert row: ORA-01861: 
literal does not match format string
 
I made to follow 
table in Oracle 
 
SQL> desc 
cdr_yada Name  
Null?    Type - 
 
 ID 
NUMBER(38) CALLDATE   
DATE ACCOUNTCODE    
VARCHAR2(20) SRC    
VARCHAR2(80) DST    
VARCHAR2(80) DCONTEXT   
VARCHAR2(80) CLID   
VARCHAR2(80) CHANNEL    
VARCHAR2(80) DSTCHANNEL 
VARCHAR2(80) LASTAPP    
VARCHAR2(80) LASTDATA   
VARCHAR2(80) DURATION   
NUMBER(38) BILLSEC    
NUMBER(38) DISPOSITION    
VARCHAR2(45) AMAFLAGS   
NUMBER(38) UNIQUEID   
VARCHAR2(32) USERFIELD  
VARCHAR2(255)
 
Who can help me with this problem 
 
Thanks Han
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[Asterisk-Users] Problems using SIPURA and MFC/R2

[Flávio Eduardo de Andrade]

We are using MFC/R2 driver successfully in at least
three places in Brazil.
I have problem with an Asterisk integrated with MFC/R2 with a Siemens Hicom
300. I can get a good audio quality with Grandstream, Polycom, and X-Lite
softfones, but SIPURAS and Linksys get a garbled audio, something like a
“Darth Vader” voice. We have tried everything in Sipura. The SIPURA
2000 and the Linksys work fine calling a URA menu on the Asterisk and can talk
to each other with excellent audio, only SIPURA->PSTN(HICOM) gets garbled.
When I use X-LITE->PSTN it works fine.  The negotiated codecs are ulaw
we’ve forced it in Asterisk and in Sipura. We are using version 0.0.3
pre4 of unicall, The linux version is:

 

Linux [EMAIL PROTECTED] src]# uname -a

Linux asterisk.karsten 2.6.9-5.ELsmp #1 SMP Wed Jan 5 19:30:39 EST
2005 i686 i686 i386 GNU/Linux

 

I did not found significant differences on the sip
negotiation between Sipura and Xlite. The only one is SIPURA offer an NSE
CODEC. On the reply this codec is not negotiated, it uses only g711u and
telephony-event. 

 

Any help would be appreciated. 

 

 

 





Novo telefone em São Paulo: (11) 4062-3535
--
Acesse a nossa Loja Virtual de Voz sobre IP:
http://www.shopvoip.com.br




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RE: [Asterisk-Users] NAT

[Samy Antoun]

> Can externalip be a dns-address?
Yes, as long as you can ping any address in the
Asterisk box (ping yahoo.com)

> How do I configure the Incoming settings in the
> siptrunk? I can call out
> using the trunk but get busy tone when I try to dial
> in. Use AAH
I don't know your provider, but generaly you need to
setup the trunk in AAH so it will be suffixed with
"/YourNumber", you can find an example at:
http://samyantoun.50webs.com/asterisk/athome/setup/trunkbroadvoice.htm

Also, you need to have a DID Route (Inbound Routing in
ver. 2) with your number.




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[Asterisk-Users] FWD: '486 Busy here' and 'All Circuits are busy now'

[Zeeshan]

Hi,

 

I’ve set up FreeWorldDialup on my asterisk server but when I dial
the service numbers, I get message ‘486 Busy Here
‘. When I dial any other number, it says 'All Circuits are busy now'.
What is the problem with my settings. I’ve
followed all the instructions step by step.

 

Zeeshan






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Re: [Asterisk-Users] Unable to send fax using BroadVoice

[Rene Nelson]

So Ariel, in your opinion is there no hope in faxing over the broadvoice network?  
Tom, I'll try your settings and see if it will work for broadvoice.

Thanks to all for the prompt advice.

Neru
On 9/29/05, Tom Vile <[EMAIL PROTECTED]> wrote:
I have finally been successful in send and recieving faxes over my Telasip line with these setttings.



You have to be in as admin and then advanced setting.



On the SIP page change:



RTP Packet Size: 0.010



On the Line Page:



FAX CED Detect Enable:	Yes

FAX CNG Detect Enable:	Yes

FAX Passthru Codec:	  G711u

FAX Codec Symmetric:	No

FAX Passthru Method:	 NSE

FAX Process NSE:	  Yes

Release Unused Codec:  Yes



Click Submit All Changes



Test.  I have had 100% success with this so far.On 9/29/05, Rene Nelson <
[EMAIL PROTECTED]> wrote:
An HP All in One Fax Machine plugged into a Sippura 2000 Analog/Phone Converter.

On 9/29/05, Tom Vile <
[EMAIL PROTECTED]
> wrote:what are you using to send the fax?  ATA and Fax.


On 9/29/05, Rene Nelson <[EMAIL PROTECTED]> wrote:


Has anyone had success sending faxes via a broadvoice byod
account?  Everything 'looks' to go as expected, but then my fax
hangs up and I get a printout with Error 351.  I am wondering if
it is a codec issue or something.

Any help will be great.

Neri


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http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony



www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856



-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205Fax: 518-631-2856


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[Asterisk-Users] Yada table in oracle

[Han van Hulst]

I 
just compiled yada and cdr_yada
Everything is 
working only I get an error when asterisk tries to insert a record into the 
database
 
Sep 29 21:05:24 
ERROR[5091]: cdr_yada.c:445 cdr_yada_log: Failed to insert row: ORA-01861: 
literal does not match format string
 
I made to follow 
table in Oracle 
 
SQL> desc 
cdr_yada Name  
Null?    Type - 
 
 ID 
NUMBER(38) CALLDATE   
DATE ACCOUNTCODE    
VARCHAR2(20) SRC    
VARCHAR2(80) DST    
VARCHAR2(80) DCONTEXT   
VARCHAR2(80) CLID   
VARCHAR2(80) CHANNEL    
VARCHAR2(80) DSTCHANNEL 
VARCHAR2(80) LASTAPP    
VARCHAR2(80) LASTDATA   
VARCHAR2(80) DURATION   
NUMBER(38) BILLSEC    
NUMBER(38) DISPOSITION    
VARCHAR2(45) AMAFLAGS   
NUMBER(38) UNIQUEID   
VARCHAR2(32) USERFIELD  
VARCHAR2(255)
 
Who can help me with this problem 
 
Thanks Han
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[Asterisk-Users] Asterisk for "Man-In-The-Middle" Trunk Side Call Recording?

[Verlin Henderson]

[Cross-posted to the asterisk forums, so if it seems familiar...]

My company is looking to implement 100% recording of inbound and outbound
calls. We're pondering using Asterisk as a trunk-side monitoring system
sitting between our Avaya Definity switch and our mixed bag of 6 T1 and PRI
circuits. I've read with interest Matt Roth's thread on the Asterisk mailing
list, however, since his experience is recording SIP-SIP calls, I'm not sure
how applicable it is to our situation. The use of a RAM disk to buffer the
calls does seem like a great idea, however, I'm concerned that a single
server might not be able to handle the load of 12 T1 ports (6 outbound, 6
connected to the switch). I also understand there might be interrupt issues
with three Digium cards in a single box.

With that in mind, I have two ideas for implementing asterisk as a call
recording solution.

The first solution I call the single-box solution and involves a dual CPU
Xeon server (most likely a Dell PowerEdge 2800, 2850, or similar) with a
large amount of RAM and RAID-1 SCSI setup. We would add three TE411P or
TE410P cards and implement something similar to Matt Roth's setup, but on a
smaller scale.

The second solution involves using either three or six desktop type or 1U
computers with a single CPU, IDE disk, and 512-1GB of RAM. We would record
directly to disk and each box would only serve 1 or 2 outbound T1's (so, 2
or 4 T1's per machine). If we implemented this as a three-box solution, we
would use a single TE411P or TE410P card per server, if implemented using
six boxes, we would use TE210P cards.

My questions about this are:

1. Is this possible? Can Asterisk serve as a man-in-the-middle between a
traditional PBX and the telco loops?

2. If it's possible, which of the two possibilities above is better? For us,
rack space and power should not be an issue, we're interested in what will
be most workable.

3. Once you have the calls recorded, how do you access them? Is it possible
to use asterisk's logs to correlate calls recordings with specific
information? We're willing to do some custom programming here if need be.

4. Has anyone done anything like this before? If so, would you be interested
in a consulting gig?

Thanks in advance for your responses!


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RE: [Asterisk-Users] Polycom Setup Questions

[Jonathan k. Creasy]

Wow...I can be useful...

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew T.
O'Connor
Sent: Thursday, September 29, 2005 2:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom Setup Questions

Matt Gibson wrote:
> Matthew T. O'Connor wrote:
>> I'm not sure I know about 3, where can I read more about the 
>> Polycom's "known issues".  Are you talking about the problems with 
>> type=friend?
>>
>> Thanks for you help.
>>
>> Matt
> Hey Matt,
>
> This is what I was referring to from the wiki:
>
>
http://voip-info.org/tiki-index.php?page=Polycom%20SoundPoint%20IP%20500
>
> look for "phone randomly freezes up"
>
> although, I'm not sure that's your issue anymore. 

Nope not my issue, just to solve the mystery for everyone.  I did fix 
the problem.  Jonathan nailed it on the head when he mentioned the loose

headset cable.  A few of my phone didn't have the headset cable plugged 
all the way (till they clicked), but were in enough to work until they 
slid out just a little.  The two things that threw me off track were 1) 
the person could hear but not talk, I would tend to think that if you 
headset works one way, it should work the other way too, apparently not;

and 2) The headset diagnositc passed, so I guess the cable was back "in"

while I did the test. 

Anyway, I failed to pass the "is it plugged in" question :-)

Thanks for you help.

Matthew

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RE: [Asterisk-Users] NAT

[Anders Svensson]

Hi
Thanks!
2 questions
Can externalip be a dns-address?

How do I configure the Incoming settings in the siptrunk? I can call out
using the trunk but get busy tone when I try to dial in. Use AAH

Thanks

Anders

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Samy Antoun
Sent: den 29 september 2005 20:41
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] NAT

Anders,

There are 2 ways to acomplish this:
1. Keep your Asterisk box behinde your router. In this
case you need to do this:
 - Port Forward to Asterisk Box
   UDP 5060
   UDP 1-2

 - sip.conf (Change as needed)
   externip=xxx.xxx.xxx.xxx
   localnet=192.168.1.0/255.255.255.0

 - Extensions
   nat=yes
   qualify=yes

2. Configure your Asterisk box as a router. In this
case you need:
 - Asterisk box must have 2 NIC's
 - A network switch
 - Setup instructions can be found at:
http://samyantoun.50webs.com/asterisk/firewall/firewall.htm

Hope this helps.





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Re: [Asterisk-Users] Hardware Specifications

[Nathan Pralle]

Dan Journo wrote:
Does anyone know where i can find out how powerful a machine has to be 
to handle a certain amount of call volume?


This is a vastly complex question because there are many factors to 
consider.  How are calls being terminated?  Are they being trunked? 
What codecs do you plan on running?  Will the server be transcoding? 
MOH?  Etc.  There's no clear answer, really.


Success stories and so forth are here:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20hardware%20recommendations

Nathan

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[Asterisk-Users] Asterisk as a Voice Logger alternative to NICE or Witness Systems

[Machen, Matthew T.]

?
Hello 
 
I would like to use Asterisk as a Voice Recorder/Logger.  I have 20+/- Siemens 
digital TDM phones with RJ11 analog recording interfaces on the back of each 
phone (separate from the digital line itself).  I would like to bring the audio 
via copper cable to a Channel bank and convert into a T1 to interface with a 
Digium T1 card in Asterisk.  Also all of the phones will have a TAPI interface 
via USB cable to a PC whereby the PC can send a "start record" and "stop 
record" signal and caller ID information to the server instead of relying on 
VOX (voice activated recording).
 
Phone
Phone T1-ChannelBank---T1 
LINE--Asterisk
Phone
 
 
Question 1:  Can the asterisk T1 line begin audio recording by either VOX or a 
signal from a TAPI device
Question 2:  Has anyone tried this?
 
This would be useful instead of buying a very expensive NICE system or Witness 
system.
 
 
 
Matthew Machen 
Southern Company Network Support 

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Re: [Asterisk-Users] Hardware Specifications

[José Pablo Ezequiel Fernández]

On Thursday 29 September 2005 15:35, Dan Journo wrote:
> Does anyone know where i can find out how powerful a machine has to be to
> handle a certain amount of call volume?
>  Eg, 2Ghz is enough processing power to maintain 100 calls at a time.
> 4Ghz is engouh to process 250 calls etc etc.
Just to show you how hard that wourd be, your mesuring units are extremely 
useless. Messuring processor power using megahertz is like mesurring speed of 
vehicles by messuring horsepower. You'll find big trucks with more 
horse-power that a F1 car, yet, the big truck is slower.
Frecuency of procesor only guarantees the same change in power in the same 
branch of processors, like comparing two P4, or two Athlon XP, but if you 
start comparing AMD vs Intel, that is no longer valid, and let me drop a 
different processor like a Spark, PowerPC, etc, and the comparation is very, 
very useless.
We should start comparing procesors with better mesurring units, like flop 
(floating point operations per second).
Even then, when messuring a system for something like handling calls, bus 
speed may be important, as well as network speed (don't neglect those other 
aspects). If you do transcoding of all the calls, then you'll need a lot of 
processor ;)
-- 
José Pablo Ezequiel Fernández
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Re: [Asterisk-Users] Unable to send fax using BroadVoice

[Ariel Batista]

They do not support faxing over there 
network.  
 
IP faxing has always been a problem. ulaw is the 
only codec you can use to do this with. And for that matter of fact it's hit or 
miss.
 
Ariel

  - Original Message - 
  From: 
  Rene Nelson 
  
  To: asterisk-users@lists.digium.com 
  
  Sent: Thursday, September 29, 2005 2:31 
  PM
  Subject: [Asterisk-Users] Unable to send 
  fax using BroadVoice
  Has anyone had success 
  sending faxes via a broadvoice byod account?  Everything 'looks' to go as 
  expected, but then my fax hangs up and I get a printout with Error 351.  
  I am wondering if it is a codec issue or something.Any help will be 
  great.Neri
  
  

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Re: [Asterisk-Users] minor(? ) Grandstream phone issue

[Ade Agbero]

I have had numerous problems with Grandstream HT-386 new and old firmware, my convidence in Grandstream is at a very low point right now.
 
I wish you luck,
 
Ade.Bob Weber <[EMAIL PROTECTED]> wrote:
I hate to bother the list with this potentially minor issue butI just wonder if it's a symtom of some other problem.Every time I make a call the BT-102, with the latest firmware, she just keeps the LED display lit and the timer counting after hangup.I check the CLI and the hangup is being executed, I certainly was concerned it might be keeping the line open but that doesn't seem to be the case.I thought that she should go back to the date/time display but that doesn't happen. I don't have silence suppresion on and I've gone over the other configs compared with what's on voip-info many time.One other thing, there isn't a dial tone when I pick it up. It seems to 'work'; both inbound and outbound.It's connected to * 1.09 which is registered @broadvoice.Thanks for any insights bestowed on this noob
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Re: [Asterisk-Users] Unable to send fax using BroadVoice

[Tom Vile]

I have finally been successful in send and recieving faxes over my Telasip line with these setttings.



You have to be in as admin and then advanced setting.



On the SIP page change:



RTP Packet Size: 0.010



On the Line Page:



FAX CED Detect Enable:	Yes

FAX CNG Detect Enable:	Yes

FAX Passthru Codec:	  G711u

FAX Codec Symmetric:	No

FAX Passthru Method:	 NSE

FAX Process NSE:	  Yes

Release Unused Codec:  Yes



Click Submit All Changes



Test.  I have had 100% success with this so far.On 9/29/05, Rene Nelson <[EMAIL PROTECTED]> wrote:
An HP All in One Fax Machine plugged into a Sippura 2000 Analog/Phone Converter.
On 9/29/05, Tom Vile <
[EMAIL PROTECTED]
> wrote:what are you using to send the fax?  ATA and Fax.

On 9/29/05, Rene Nelson <[EMAIL PROTECTED]> wrote:


Has anyone had success sending faxes via a broadvoice byod
account?  Everything 'looks' to go as expected, but then my fax
hangs up and I get a printout with Error 351.  I am wondering if
it is a codec issue or something.

Any help will be great.

Neri


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http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony


www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856



-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com
Phone: 518-631-2855 x205Fax: 518-631-2856
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Re: [Asterisk-Users] H323 and Asterisk

[Ade Agbero]

Can you please let me know what version of oh323 you installed and the step by step process of installation.
 
You can post your oh323.conf and extensions.conf, I should be able to help you out with your configuration.
 
Regards,
 
Ade.
Kanishka Somaratne <[EMAIL PROTECTED]> wrote:
hi guysI was working on asterisk and h323 for the past 2 weeksi have the following feedback please let me know if i am wrongh323 implementationI managed to install this it works, but the problem is it accecpts all calls from all ips. there is no way i can let it accecpt calls only from the IPs i give and bill depending on IPoh323 implementationmanaged to install, same as h323 implementation i can't add a list of ips and restrict access, the 729 -> 723.1 codec convertion does not work well, get a robort voiceooh323cinstalleed but do not know how to configure :(woomeralet me know if there's any one who has tried this.what i want to do it accecpt h323 calls and bill depending on the ip address and send the calls via h323 depdning on the gateway IP i addRegardsKansihka
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Re: [Asterisk-Users] Bad FCS nightmare to Nortel SL100 with TE410P

[Gil Kloepfer]

On Tue, Sep 27, 2005 at 01:22:48AM -0500, I wrote:
> I have an * box connected to a Nortel SL100 through a PRI (US) using the
> Digium TE410P (quad-span T1 card).  I don't have access to the SL100 -
> it is handled by another group.
> 
> The span comes up OK (timing, framing fine).  However, as soon as the
> D channel comes up, I get endless "HDLC Bad FCS" errors.

We have resolved the issue - it turned out that there was something
wrong at the SL100 side of the connection.  The bad news is that I
can't get anyone who manages that SL100 to tell me exactly what
the problem actually was.  The best I was able to get was, "It was
some kind of cable or configuration problem."

The only private feedback I got were recommendations to try the
Sangoma cards (it looks like they keep more in the way of statistics).

I also have no idea why the TE410P complained, but the Cisco AS5350
didn't.

Thanks to everyone who provided feedback.

---
Gil Kloepfer
[EMAIL PROTECTED]
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Re: [Asterisk-Users] NAT

[Samy Antoun]

Anders,

There are 2 ways to acomplish this:
1. Keep your Asterisk box behinde your router. In this
case you need to do this:
 - Port Forward to Asterisk Box
   UDP 5060
   UDP 1-2

 - sip.conf (Change as needed)
   externip=xxx.xxx.xxx.xxx
   localnet=192.168.1.0/255.255.255.0

 - Extensions
   nat=yes
   qualify=yes

2. Configure your Asterisk box as a router. In this
case you need:
 - Asterisk box must have 2 NIC's
 - A network switch
 - Setup instructions can be found at:
http://samyantoun.50webs.com/asterisk/firewall/firewall.htm

Hope this helps.





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Re: [Asterisk-Users] Unable to send fax using BroadVoice

[Rene Nelson]

An HP All in One Fax Machine plugged into a Sippura 2000 Analog/Phone Converter.On 9/29/05, Tom Vile <[EMAIL PROTECTED]
> wrote:what are you using to send the fax?  ATA and Fax.
On 9/29/05, Rene Nelson <[EMAIL PROTECTED]> wrote:


Has anyone had success sending faxes via a broadvoice byod
account?  Everything 'looks' to go as expected, but then my fax
hangs up and I get a printout with Error 351.  I am wondering if
it is a codec issue or something.

Any help will be great.

Neri


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[Asterisk-Users] Hardware Specifications

[Dan Journo]

Does anyone know where i can find out how powerful a machine has to be to handle a certain amount of call volume?
 
Eg, 2Ghz is enough processing power to maintain 100 calls at a time.
4Ghz is engouh to process 250 calls etc etc.
 
Thanks
Dan
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Re: [Asterisk-Users] Unable to send fax using BroadVoice

[Tom Vile]

what are you using to send the fax?  ATA and Fax.On 9/29/05, Rene Nelson <[EMAIL PROTECTED]> wrote:
Has anyone had success sending faxes via a broadvoice byod
account?  Everything 'looks' to go as expected, but then my fax
hangs up and I get a printout with Error 351.  I am wondering if
it is a codec issue or something.

Any help will be great.

Neri


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[Asterisk-Users] Unable to send fax using BroadVoice

[Rene Nelson]

Has anyone had success sending faxes via a broadvoice byod
account?  Everything 'looks' to go as expected, but then my fax
hangs up and I get a printout with Error 351.  I am wondering if
it is a codec issue or something.

Any help will be great.

Neri

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Re: [Asterisk-Users] Polycom Setup Questions

[Matthew T. O'Connor]

Matt Gibson wrote:

Matthew T. O'Connor wrote:
I'm not sure I know about 3, where can I read more about the 
Polycom's "known issues".  Are you talking about the problems with 
type=friend?


Thanks for you help.

Matt

Hey Matt,

This is what I was referring to from the wiki:

http://voip-info.org/tiki-index.php?page=Polycom%20SoundPoint%20IP%20500

look for "phone randomly freezes up"

although, I'm not sure that's your issue anymore. 


Nope not my issue, just to solve the mystery for everyone.  I did fix 
the problem.  Jonathan nailed it on the head when he mentioned the loose 
headset cable.  A few of my phone didn't have the headset cable plugged 
all the way (till they clicked), but were in enough to work until they 
slid out just a little.  The two things that threw me off track were 1) 
the person could hear but not talk, I would tend to think that if you 
headset works one way, it should work the other way too, apparently not; 
and 2) The headset diagnositc passed, so I guess the cable was back "in" 
while I did the test. 


Anyway, I failed to pass the "is it plugged in" question :-)

Thanks for you help.

Matthew

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[Asterisk-Users] Re: Voice Encryption

[Scott Huang]

I went over the code. AES128 is the only algorithm that is suppored
today. More importantly there are some concerns on the vulnerability as
discussed in
http://lists.digium.com/pipermail/asterisk-security/2005-August/60.html.
People are using UDP VPNs to satisfy customer requirements.
http://lists.digium.com/pipermail/asterisk-users/2005-August/120293.html

Thanks

-Scott

--

Message: 6
Date: Wed, 28 Sep 2005 19:09:17 -0500
From: Andy Hamilton <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Re: Voice Encryption
To: Asterisk Users Mailing List - Non-Commercial Discussion
        
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1

> Does Asterisk support encryption of voice traffic? I found following wiki
> that describes IAX RSA authentication. I was able to implement the
> public/private key authentication among three Asterisk servers connected
> using IAX protocol. I am not certain if voice traffic can also be encrypted
> among the Asterisk servers. Your help is highly appreciated.

There has been a little discussion of this topic on the
asterisk-security list somewhat recently.
You may want to look at the message archive from August (
http://lists.digium.com/pipermail/asterisk-security/2005-August/thread.html

) as well as the one from September.

Hope this offers a little help; as someone mentioned, not much
documentation is out there.
>From the August thread:

>Basically it's an automatic features (CVS-HEAD only, btw) where anytime
>a pair of IAX peers have encryption set to "yes" (or, as I understand
>it, a mutually-agreed-upon and supported algorithm) then it just happens
>automagically.
(Brian Capouch)

-a
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[Asterisk-Users] Any way to not overwrite sound files on compile?

[Matt]

Every time I recompile Asterisk (or upgrade to a new CVS-HEAD,
whatever) asterisk overwrites custom files I have made.  Granted,
these files are named the same as the asterisk default files
(vm-login.gsm, etc) because we had a person here record them to
customize them a bit more for our application.

Short of keeping them somewhere and copying them back every time
(which isn't all that often) I do a re-compile.  Is there some flag or
something to tell Asterisk not to install sound files, or at the very
least not to overwrite ones already existing?
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RE: [Asterisk-Users] NAT

[Anders Svensson]

Yes that I have done. But don't I have to configure the gateway ip
somewhere?

anders

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug
Sent: den 29 september 2005 19:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] NAT

At 12:14 9/29/2005, Anders Svensson, wrote:

>
>
>Hi!
>
>
>
>Finally I have been able to install AAH and its up and running. I am 
>behind a router and believe I have to configure this somewhere but cant do 
>this with AMP. Can somebody hint a newbie about how to do it

You should be able to do it in AMP.  What extension are
you using?  Extensions -> NAT (change from "never" to "yes")



>Regards
>
>Anders Svensson
>
>
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Re: [Asterisk-Users] Tiny Echo on PSTN via Zaptel

[Shaw Terwilliger]

On Thu, Sep 29, 2005 at 12:10:47PM -0500, Neil Lewis wrote:
> If you are using CVS-Head you can also use the fxotune utility located
> in /usr/src/zaptel to tune
> out echo in the FXO module.  Execute this command: './fxotune -i 4'
> It will automatically configure the FXO modules for echo.

Is fxotune supposed to work with the TE210P?  When I run it, it outputs
the following for all of my FXO (and FXS) ports:

Tuning module 1Skipping non-TDM / non-FXO
Failure!

> You can also try using an different Echo Canceler.  We've recently added
> a new EchoCan to Asterisk:  KB1.  To utilize it, just uncomment its
> define statement in zconfig.h, and comment the other EchoCan out.

I'll give this a shot, then play with the agressive suppressor.
The echo situation is better than it used to be, so I'm making progress.

-- 
Shaw Terwilliger <[EMAIL PROTECTED]>
SourceGear LLC


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Re: [Asterisk-Users] Remotely dialing calls from a polycom phone

[Eric Lawman]

See my above message as to why auto-answer won't work.On 9/29/05, Paul Davidson <[EMAIL PROTECTED]> wrote:
>Message: 7
>Date: Thu, 29 Sep 2005 09:53:27 -0400
>From: Eric Lawman <[EMAIL PROTECTED]>
>Subject: [Asterisk-Users] Remotely dialing calls from a polycom phone
>To: Asterisk-Users@lists.digium.com
>Message-ID: <[EMAIL PROTECTED]>
>Content-Type: text/plain; charset="iso-8859-1"

>I have a Polycom IP600 serving as a receptionist phone. We developed a call
>manager via c/gtk that runs on a touchpad. It allows them to transfer calls,
>transfer to voicemail, page, etc. The problem is this: When paging another
>phone from the touchpad, I have to open a channel to the receptionist >phone.
>This rings the receptionist phone. When she picks up, it then pages the
>desired person. This is fine, except it can be confusing for the
>receptionist. "Why do I have to answer the phone so I can page >someone?"
>What I'm looking for is a little more direct integration between the
>touchpad and the polycom.

>So, the question is, does anyone know of a way to initiate a call on a
>polycom via the remote access port. I was thinking of something similar to
>the way you can reboot one using a NOTIFY message containing >"check-sync".
>I've tried monitoring the sip messages between the phone and asterisk, >but
>so far have only succeeded in making a phone call itself.

>Any suggestions would be appreciated.

>Eric.

First thing that comes to mind here is using auto-answer.  While I
don't have an IP600 to play with (donations accepted), the way I would
envision this is to configure an auto-answer extension on hte Polycom,
and have the touchpad action call the auto-answer extension using a
.call file to Asterisk.  Touch the pad, it drops the .call in,
which goes to a meetme with all the necessary extensions in the paging
group added.  Touch again, drop the call (via the manager API
would be my approach).

All untested- but fairly straightfoward.  If you want to email me directly, I'd be happy to help further.

-Paul Davidson
 PlanCommunications, LLC

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[Asterisk-Users] DTMF tones from PSTN not reaching SIP device

[Bruce Komito]

Greetings, I am PRIs connected to a Cisco 36xx gateway, which in turn
connects to Asterisk via SIP.  The problem I am having is that DTMF tones
originated on the PSTN side are not heard on the SIP device.  On the other
hand, tones originating on the PSTN side are received by Asterisk when
talking to voicemail or an autoattendant.

>From the Cisco debug, I can see the Cisco sending NTE (RFC2833) RTP
packets to Asterisk and it appears that Asterisk is propogating them down
to the SIP device.  However when tones are pressed on the PSTN side, all
that can be heard on the IP phone is silence.  I've tried this on three
different IP phones (Cisco 79xx, ZIP2 and Sipura) and they all behave the
same, leading me to conclude it isn't a phone config problem.  Everything
(Cisco and phones) are configured for dtmfmode=rfc2833.

Anyone got any ideas?

TIA

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815


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Re: [Asterisk-Users] NAT

[Doug]

At 12:14 9/29/2005, Anders Svensson, wrote:




Hi!



Finally I have been able to install AAH and its up and running. I am 
behind a router and believe I have to configure this somewhere but cant do 
this with AMP. Can somebody hint a newbie about how to do it


You should be able to do it in AMP.  What extension are
you using?  Extensions -> NAT (change from "never" to "yes")




Regards

Anders Svensson


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Re: [Asterisk-Users] Cannot figure out why calls from myAsteriskappear to be fr

[Angus Comber]

I use a BT ISDN line.  But calls go through Onetel.

Bizarre, this behaviour has now stopped.  Country code now no longer part of 
CLI seen on my mobile.


Perhaps it is as you say, something my least cost routing company, Onetel 
are doing!


Angus


- Original Message - 
From: "David J Carter" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 


Sent: Thursday, September 29, 2005 5:52 PM
Subject: RE: [Asterisk-Users] Cannot figure out why calls from 
myAsteriskappear to be fr



Do you use BT for you outgoing calls? Or are you using another provider?

I have one customer who uses another provider and there calls come to me
with some strange CLI numbers.

It seems to be they break out where the best rates are at that time.

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ian Bonham
Sent: 29 September 2005 15:59
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Cannot figure out why calls from
myAsteriskappear to be fr

Not sure about the Digium, but I can tell you +34 is Spain, if that helps
you track anything down? I assume you've tested the line with a normal phone

to make sure it's not a telco fault?

Ian




From: "Angus Comber" <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussion
To: 
Subject: [Asterisk-Users] Cannot figure out why calls from my
Asteriskappear to be from country code +34?
Date: Thu, 29 Sep 2005 15:32:39 +0100

Hello

When I dial out from my Asterisk (using Digium analog TDM04B card over pstn



line), calls appear to be from +34

I am in UK which is +44 so cannot work out why seeing +34.

In my zapata.conf I have:

loadzone = uk
defaultzone = uk

I can't find any country specific stuff in any other conf files.

Any ideas how I can correctly set so that calls from my asterisk do not
have +34?

Angus




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Re: [Asterisk-Users] Zap FXO/FXS issues, 1.2.0-beta1

[alan]

> Subject: Re: [Asterisk-Users] Zap FXO/FXS issues, 1.2.0-beta1

Rich Adamson  <[EMAIL PROTECTED]> wrote:

> > We're having issues with the FXO/FXS ports on our Digium TDM cards
> > sporadically. I'm wondering if anyone else has had these problems, or if
> > anyone can provide guidance diagnosing or fixing the issue?
> >
> >
> > The symptoms are that the FXO and FXS ports "stop working", usually
> > after 2-4 weeks of server uptime. When this happens, sending a (SIP)
> > call to an analog phone on an FXS port causes the phone to ring, but
> > when you answer the phone, the off-hook is not detected. Further rings
> > sound like clicks in the off-hook handset. If the handset is hung up,
> > ringing resumes.
>
> The digium analog TDM card (four modules) with early revision
> numbers (rev E/F and before) had a problem where the card fails to
> respond every week (or more). Stopping asterisk, stopping the drivers,
> and reloading the drivers would cause the card to work again.
>
> If that sounds like a match for your problem, contact digium support
> as the card is still under warranty. Replacing it fixed the problem for
> me.


This is our problem, and zttool reports our board revisions as E/F for
both.

Thanks for the pointer! I'll get right on it.

Alan Ferrency
pair Networks, Inc.
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Cannot figure out why calls from myAsteriskappear to be fr

[Angus Comber]

I am seeing this by calling my Nokia mobile phone - using Vodafone in UK. 
If I substitute Asterisk for an Avaya IP Office then just get: 020 8878 
7367 - ie my number but without the country code.  So it must be something 
that the Asterisk is doing.


Angus

- Original Message - 
From: "Ian Bonham" <[EMAIL PROTECTED]>

To: 
Sent: Thursday, September 29, 2005 3:59 PM
Subject: RE: [Asterisk-Users] Cannot figure out why calls from 
myAsteriskappear to be fr



Not sure about the Digium, but I can tell you +34 is Spain, if that helps 
you track anything down? I assume you've tested the line with a normal 
phone to make sure it's not a telco fault?


Ian




From: "Angus Comber" <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussion

To: 
Subject: [Asterisk-Users] Cannot figure out why calls from my 
Asteriskappear to be from country code +34?

Date: Thu, 29 Sep 2005 15:32:39 +0100

Hello

When I dial out from my Asterisk (using Digium analog TDM04B card over 
pstn line), calls appear to be from +34


I am in UK which is +44 so cannot work out why seeing +34.

In my zapata.conf I have:

loadzone = uk
defaultzone = uk

I can't find any country specific stuff in any other conf files.

Any ideas how I can correctly set so that calls from my asterisk do not 
have +34?


Angus




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[Asterisk-Users] NAT

[Anders Svensson]

 

Hi!

 

Finally I have been able to install AAH and its up
and running. I am behind a router and believe I have to configure this
somewhere but cant do this with AMP. Can somebody hint a newbie about how to do
it 

Regards

Anders Svensson 

 






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Re: [Asterisk-Users] Tiny Echo on PSTN via Zaptel

[Neil Lewis]

Tony Nichols wrote:

> I have had problems between the sip/FXO lies and was able to "kill"
> the echo by trying different combinations of the echocancel line to 64
> (I think it has settings in 32 bit increments)
> Just kept trying different ones till it went away. Here is my config:
>
> group=1
> context=line1
> signalling=fxs_ks
> usecallerid=yes
> callerid=asreceived
> echocancel=64
> echocancelwhenbridged=yes
> callgroup=1
> rxgain=1.2
> channel => 1
>
> context=line2
> signalling=fxs_ks
> usecallerid=yes
> callerid=asreceived
> echocancel=64
> echocancelwhenbridged=yes
> callgroup=1
> rxgain=1.2
> channel => 2
> musiconhold=default
> context=line3
> signalling=fxs_ks
> usecallerid=yes
> callerid=asreceived
> echocancel=64
> echocancelwhenbridged=yes
> callgroup=1
> rxgain=1.2
> channel => 3
>
> group=2
> context=line4
> signalling=fxs_ks
> usecallerid=yes
> callerid=asreceived
> echocancel=96
> echocancelwhenbridged=yes
> callgroup=2
> channel => 4
>
> Hope this helps!
>
> t o n y
>
> On 9/28/05, *Shaw Terwilliger* <[EMAIL PROTECTED]
> > wrote:
>
> I'm using Asterisk 1.0.9, a Digium TE210P dual T1 card, and two Rhino
> channel banks (one 12FXO/12FXS, the other 24 FXS).  So it's an analog
> phone on the inside connected to one of the FXS ports, and PSTN line
> connected to one of the FXO ports.
>
> My problem is that as soon as I hear the _first_ ring when I dial out
> through the PSTN line, I hear a tiny echo on the phone (I estimate
> between 20ms and 40ms), which never goes away for this call.  It's
> just
> loud enough to bug the heck out of me when I'm talking (I could
> estimate
> the gain relationship with ztmonitor if it would help).  The sound
> on the recipient end of the connection is perfect.
>
> If I make a call from the phone to the another internal extension
> (another
> phone on an FXS port), there is no echo.  If I call into Comedian
> mail,
> there is no echo.
>
> I've checked all my gains.  The "internal" gains were a bit loud
> to start
> with because of the powered phones, but now they all fall
> comfortably within
> ztmonitor's dynamic range display.  The PSTN line is pretty good
> at tx 0 and
> rx 0, so I left it.  I've tried turning them down, but that didn't
> kill
> the echo.
>
> My zapata.conf includes these lines at the bottom:
>
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=yes
>
> context=companyA-pstn
> txgain=0.0
> rxgain=0.0
> signalling=fxs_ks
> group=1
> channel=1-7
>
> context=companyB-pstn
> txgain=0.0
> rxgain=0.0
> signalling=fxs_ks
> group=2
> channel=11-12
>
> context=internal
> txgain=-12.0
> rxgain=-8.0
> signalling=fxo_ls
> callerid=asreceived
> group=3
> channel=13-48
>
> When the calls are connected, I can use "zap show channel 11" and
> verify
> that the echo cancellation is ON.  But I can still hear one.
> I've also tried echocancelwhenbridged=yes, but it didn't make any
> difference.
>
> --
> Shaw Terwilliger < [EMAIL PROTECTED]
> >
> SourceGear LLC
>
>
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.2.5 (GNU/Linux)
>
> iD8DBQFDOuxyPEbgvbl6u4ERAh/gAKCAu+gAr9TYsMG5TYqozV3ebIvezwCdGIHk
> kftrFLFK4purux/sVIPRhKk=
> =9WG3
> -END PGP SIGNATURE-
>
>
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> 
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>
>
>
> -- 
> A.G. (Tony) Nichols
> I.S. Manager
>
>
>
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>
Echo problems can be remedied in several ways. Echocancel and
echotraining are the primary tools used to remove echo. Experiment with
the echocancel and echotraining values in order to find the right
combination for your echo problem (echocancel: 32, 64, 128, 256;
echotraining: 200, 400, 800). You'll also need to make sure that your
tx/rxgain levels are not set too high. As a rule of thumb, they should
not be set higher that 5, or lower than -5.

If you are using CVS-Head you can also use the fxotune utility located
in /usr/src/zaptel to tune
out echo in the FXO module.  Execute this command: './fxotune -i 4'

Re: [Asterisk-Users] T.38 Faxing -> at astricon ?

[C F]

One cannot and should not compare ISDN/SS7 to VoIP in any way just
because it is digitalized voice. The fact that the transmission is
digital does not change the success of sending faxes. The reason that
VoIP is less reliable than ISDN/SS7 for faxes is becuase of the fact
that an IP network is a packet switched network, while the ISDN/SS7 is
a circuit swithced network. On an IP network there is no gurantee on
any given packet that it will arrive intact and in sequence at the
other end, while this all exists on ISDN/SS7.
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RE: [Asterisk-Users] Cannot figure out why calls from myAsterisk appear to be fr

[David J Carter]

Do you use BT for you outgoing calls? Or are you using another provider?

I have one customer who uses another provider and there calls come to me
with some strange CLI numbers.

It seems to be they break out where the best rates are at that time.

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ian Bonham
Sent: 29 September 2005 15:59
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Cannot figure out why calls from
myAsteriskappear to be fr

Not sure about the Digium, but I can tell you +34 is Spain, if that helps 
you track anything down? I assume you've tested the line with a normal phone

to make sure it's not a telco fault?

Ian



>From: "Angus Comber" <[EMAIL PROTECTED]>
>Reply-To: Asterisk Users Mailing List - Non-Commercial 
>Discussion
>To: 
>Subject: [Asterisk-Users] Cannot figure out why calls from my 
>Asteriskappear to be from country code +34?
>Date: Thu, 29 Sep 2005 15:32:39 +0100
>
>Hello
>
>When I dial out from my Asterisk (using Digium analog TDM04B card over pstn

>line), calls appear to be from +34
>
>I am in UK which is +44 so cannot work out why seeing +34.
>
>In my zapata.conf I have:
>
>loadzone = uk
>defaultzone = uk
>
>I can't find any country specific stuff in any other conf files.
>
>Any ideas how I can correctly set so that calls from my asterisk do not 
>have +34?
>
>Angus
>
>
>
>
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Re: [Asterisk-Users] zttest - 100% ?

[Carlos Antunes]

This might seem a silly question but, what is the true meaning of the numbers zttest spits out?On 9/29/05, Marco Supino <
[EMAIL PROTECTED]> wrote:Hi,My TDM is on its own IRQ, and the x306 has only one full-size PCI slot..
so no playing with it,what results do you get from zttest ? what IRQ is the card on ?Marco.Damian Funnell wrote:> Have you checked that the TDM400P isn't sharing an IRQ with anything
> else?  Don't trust /proc/interrupts - run lspci -v to confirm this.>> We have * running on an x206 and found that the only way to stop the> TDP400P sharing an IRQ with other devices was to juggle cards between
> slots.>> Hope this helps!> Damian.>> FFF Managed Technology Ltd> 60 Cook St> P.O. 6368 Wellesley St> Auckland> t +64 9 356 2911> f +64 9 358 9070
> m +64 21 415 297> w www.fff.co.nz Marco Supino wrote:>>> Hi, I would like to know what type of configuration could get me closer to
>> 100% hits in zttest, when testing a TDM400P with 4 FXO ports, I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh>> CPU, HT is disabled, PCI latency was changed, i still cant get more
>> then 99.975% in the zttest testings, Thanks for any info. Marco.>> ___>> --Bandwidth and Colocation sponsored by 
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Re: [Asterisk-Users] TE205P in loopback?

[Franciraldo Cavalcante Junior]

All the config match. Just to make sure, how did you make your loopback cable? Which pins are conected were?
Thanks in advance,
-f


From: "Steve Totaro" <[EMAIL PROTECTED]>Reply-To: Asterisk Users Mailing List - Non-Commercial DiscussionTo: "Asterisk Users Mailing List - Non-Commercial Discussion"Subject: Re: [Asterisk-Users] TE205P in loopback?Date: Wed, 28 Sep 2005 22:28:02 -0700

It flashes red until the properly made cables are connected and then should turn to solid green.  Again, I suspect your cable is the issue.  Are all the drivers loaded?  ztcfg?  Asterisk running?  One span set to pri_cpe and the other to pri_net?

- Original Message - 
From: Franciraldo Cavalcante Junior 
To: asterisk-users@lists.digium.com 
Sent: Wednesday, September 28, 2005 6:47 PM
Subject: Re: [Asterisk-Users] TE205P in loopback?


The board flashes red without cables and continues flashing red after cables are connected. Should it flash green before? I don't see anything weird in dmegs results.


From: "Steve Totaro" <[EMAIL PROTECTED]>Reply-To: Asterisk Users Mailing List - Non-Commercial DiscussionTo: "Asterisk Users Mailing List - Non-Commercial Discussion"Subject: Re: [Asterisk-Users] TE205P in loopback?Date: Wed, 28 Sep 2005 21:12:08 -0700
If the cables are connected and the lights are still flashing red then I suspect that the cable is the issue.

- Original Message - 
From: Franciraldo Cavalcante Junior 
To: asterisk-users@lists.digium.com 
Sent: Wednesday, September 28, 2005 5:44 PM
Subject: Re: [Asterisk-Users] TE205P in loopback?


I did the cable and configuration as recomended, but I may have another problem. After I run:
modprobe zaptel 
modprobe wct2xxp
I don't see the boards flashing green it's flashing red and in sync, as I implicitly understand from your message . Should it flash green and when correctly cabled and configured, keep green?
How can I debug this?
any help is appreciated.
 
-f


From: "Steve Totaro" <[EMAIL PROTECTED]>Reply-To: Asterisk Users Mailing List - Non-Commercial DiscussionTo: "Asterisk Users Mailing List - Non-Commercial Discussion"Subject: Re: [Asterisk-Users] TE205P in loopback?Date: Wed, 28 Sep 2005 20:12:40 -0700


I just did this on a te410p.  
 
make a t1 crossover, cross pins one with four, and two with five.  Set one span for pri_cpe and the other side for pri_net.  boom, you are done and can start testing.  the lights should turn solid green and you should first see the  D channels come up and then all the B channels, if this happens, you have done it correctly.
 
thanks,
Steve

- Original Message - 
From: Franciraldo Cavalcante Junior 
To: asterisk-users@lists.digium.com 
Sent: Wednesday, September 28, 2005 4:46 PM
Subject: [Asterisk-Users] TE205P in loopback?



I'm a new * user and had just intalled A TE205P line in my Linux machine (running linux kernel 2.6.10). I would like to test the T1 interface but I don't have an ISDN PRI interface available right now. Is there a way to make some sort of loopback test? How to configure it? Do I need the cables to have tx/rx exchanged? 
Any help would be appreciated.
- f
 



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Re: [Asterisk-Users] zttest - 100% ?

[Marco Supino]

Hi,

My TDM is on its own IRQ, and the x306 has only one full-size PCI slot.. 
so no playing with it,


what results do you get from zttest ? what IRQ is the card on ?

Marco.


Damian Funnell wrote:
Have you checked that the TDM400P isn't sharing an IRQ with anything 
else?  Don't trust /proc/interrupts - run lspci -v to confirm this.


We have * running on an x206 and found that the only way to stop the 
TDP400P sharing an IRQ with other devices was to juggle cards between 
slots.


Hope this helps!
Damian.

FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz



Marco Supino wrote:


Hi,

I would like to know what type of configuration could get me closer to 
100% hits in zttest, when testing a TDM400P with 4 FXO ports,


I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh 
CPU, HT is disabled, PCI latency was changed, i still cant get more 
then 99.975% in the zttest testings,


Thanks for any info.

Marco.


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Re: [Asterisk-Users] soft phones for Zaurus PDA

[Leif Madsen]

On 9/29/05, William Suffill <[EMAIL PROTECTED]> wrote:
> Ziaxphone might fit your needs.
> http://www.kauss.org/Stephan/ziaxphone/ Haven't used it
> recently since someone broke the screen on my Zaurus =(

I can vouch for the software. I haven't used it in some time, but it
DID work when I tried it on my 5500.

--
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Astricon 2005, Anaheim, CA, October 12-14
http://www.astricon.net
http://www.oreilly.com/catalog/asterisk
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[Asterisk-Users] minor(? ) Grandstream phone issue

[Bob Weber]

I hate to bother the list with this potentially minor issue but
I just wonder if it's a symtom of some other problem.

Every time I make a call the BT-102, with the latest firmware, she just 
keeps the LED display lit and the timer counting after hangup.
I check the CLI and the hangup is being executed, I certainly was concerned 
it might be keeping the line open but that doesn't seem to be the case.


I thought that she should go back to the date/time display but that doesn't 
happen. I don't have silence suppresion on and I've gone over the other 
configs compared with what's on voip-info many time.
One other thing, there isn't a dial tone when I pick it up.  It seems to 
'work'; both inbound and outbound.

It's connected to * 1.09 which is registered @broadvoice.

Thanks for any insights bestowed on this noob :)
Bob
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Re: [Asterisk-Users] Remotely dialing calls from a polycom phone

[Paul Davidson]

>Message: 7
>Date: Thu, 29 Sep 2005 09:53:27 -0400
>From: Eric Lawman <[EMAIL PROTECTED]>
>Subject: [Asterisk-Users] Remotely dialing calls from a polycom phone
>To: Asterisk-Users@lists.digium.com
>Message-ID: <[EMAIL PROTECTED]>
>Content-Type: text/plain; charset="iso-8859-1"

>I have a Polycom IP600 serving as a receptionist phone. We developed a call
>manager via c/gtk that runs on a touchpad. It allows them to transfer calls,
>transfer to voicemail, page, etc. The problem is this: When paging another
>phone from the touchpad, I have to open a channel to the receptionist >phone.
>This rings the receptionist phone. When she picks up, it then pages the
>desired person. This is fine, except it can be confusing for the
>receptionist. "Why do I have to answer the phone so I can page >someone?"
>What I'm looking for is a little more direct integration between the
>touchpad and the polycom.

>So, the question is, does anyone know of a way to initiate a call on a
>polycom via the remote access port. I was thinking of something similar to
>the way you can reboot one using a NOTIFY message containing >"check-sync".
>I've tried monitoring the sip messages between the phone and asterisk, >but
>so far have only succeeded in making a phone call itself.

>Any suggestions would be appreciated.

>Eric.

First thing that comes to mind here is using auto-answer.  While I
don't have an IP600 to play with (donations accepted), the way I would
envision this is to configure an auto-answer extension on hte Polycom,
and have the touchpad action call the auto-answer extension using a
.call file to Asterisk.  Touch the pad, it drops the .call in,
which goes to a meetme with all the necessary extensions in the paging
group added.  Touch again, drop the call (via the manager API
would be my approach).

All untested- but fairly straightfoward.  If you want to email me directly, I'd be happy to help further.

-Paul Davidson
 PlanCommunications, LLC
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Re: [Asterisk-Users] zttest - 100% ?

[Damian Funnell]

Have you checked that the TDM400P isn't sharing an IRQ with anything 
else?  Don't trust /proc/interrupts - run lspci -v to confirm this.


We have * running on an x206 and found that the only way to stop the 
TDP400P sharing an IRQ with other devices was to juggle cards between slots.


Hope this helps!
Damian.

FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz



Marco Supino wrote:


Hi,

I would like to know what type of configuration could get me closer to 
100% hits in zttest, when testing a TDM400P with 4 FXO ports,


I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh 
CPU, HT is disabled, PCI latency was changed, i still cant get more 
then 99.975% in the zttest testings,


Thanks for any info.

Marco.


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Re: [Asterisk-Users] T.38 Faxing -> at astricon ?

[Craig Guy]

The problem as I see it is that if people start expecting it to work then 
rather than being pleasantly surprised when it does, they will be bitterly 
disappointed when it doesn't.  IMHO analog fax over IP is too flaky to 
encourage the general public to utilise, and any suggestion to the contrary 
is misleading.


Having said that, I have an analog fax connected to an ATA that works 100% 
of the time, however I have my ATA and Asterisk on the same ethernet switch. 
I wouldn't expect to have it work reliably over a WAN or other broadband 
internet connection.


Craig

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


Sent: Thursday, September 29, 2005 10:31 PM
Subject: Re: [Asterisk-Users] T.38 Faxing -> at astricon ?



Roy Sigurd Karlsbakk schrieb:

...
see http://soft-switch.org/foip.html for a brief explaination of why 
this generally doesn't work...



Hi,

maybe one should update this link.

I think, you agree, that VoIP is somewhat similar to ISDN, as it
transports analog audio data in a digitally coded way.

Noone doubts, that ISDN is suitable to transport analog fax.
Finally the PSTN is 99,9% digital (ISDN/SS7), even if some
subscriber lines are still analog.
(Ok, ISDN is a managed network, and thus very high quality.)

Since there are more and more regions in the world, where internet
connectivity quality approaches to ISDN quality, analog faxing
over VoIP becomes reliable and hassle free.

You should have 128kbit in both directions, better 256kbit,
maybe some QoS build in your router (e.g. Linux's iproute2),
and pingtimes below 20ms to the VoIP-provider (PSTN-gateway).

DSL with fastpath or internet by TV cable does provide this
standard imho and become more and more available.


Thus we shouldn't discourage people generally of faxing, even
if there are a lot of trouble reports.

Who can count the success stories with (analog) fax over IP,
which are not posted?

As far as I see, there are more users faxing without observing
quality differences to ISDN than users with problems with fax
over VoIP.

This is, what various partners of ours do report after having
replaced BRI connections by VoIP in some small and middle sized
companies.


Roger.

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RE: [Asterisk-Users] call center software and asterisk

[Sergio Serrano]

www.inconcertCC.com has a solution based on Asterisk.


regards,

srsergio

-Mensaje original-
De: Bartosz Jozwiak [mailto:[EMAIL PROTECTED] 
Enviado el: jueves, 29 de septiembre de 2005 17:17
Para: Asterisk Users Mailing List - Non-Commercial Discussion
CC: Commercial and Business-Oriented Asterisk Discussion
Asunto: [Asterisk-Users] call center software and asterisk

Hi guys,

Need some advise.
Is there some kind of call center software which can "interconnect" with
asterisk?
So, for example, agents can see on their pc's all info about calling client
(based on clid) before they pick up the phone.
And that outbound calls are also "automated".

Commercial solutions more then welcome.

Thx,
Bartosz Jozwiak 

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Re: [Asterisk-Users] Variable in call parking

[Damian Funnell]

Hi Andrew,

Not sure if I understand your question, but this may help - * has the
following settings in features.conf that are related to parking:
parkext =>  ;the extension that users xfer calls to in order to
park them
parkpos => -   ;the extension range that * will use to park
calls.  It will tell the user which extension it has parked each call
on when they are parked.  Dialling this extension retrieves the parkpos.
context => parkedcalls ;context that calls are parked in
parkingtime => xxx ;number of seconds a call will be parked for
before being sent back to the extension that parked it.

Hope this helps,
Damian.

FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz


Andrew Nowrot wrote:

  Hi,

Can anyone tell me if Asterisk sets some variable when doing a call
parking (when someone presses an exten set in features.conf). In can't
find this information on a wiki.

Cheers ;)

Andrew
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Re: [Asterisk-Users] call center software and asterisk

[Nathan Pralle]

AstGUIClient and VICIDIAL seem to be a good tool for the task.  I cannot 
verify, as I have not used them before.


http://astguiclient.sourceforge.net/

Nathan

Bartosz Jozwiak wrote:

Hi guys,

Need some advise.
Is there some kind of call center software which can "interconnect" with 
asterisk?
So, for example, agents can see on their pc's all info about calling 
client (based on clid)

before they pick up the phone.
And that outbound calls are also "automated".

Commercial solutions more then welcome.

Thx,
Bartosz Jozwiak
_


--
-
Nathan E. Pralle
Give the Director a Serpent Deflector
www.nathanpralle.com
-
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Re: [Asterisk-Users] Correction: Asterisk sound files, audio bandwidth, and sound quality

[steve]

On Wed, 28 Sep 2005, Stephen Bosch wrote:

> [EMAIL PROTECTED] wrote:
> >>When I listen to the GSM compressed prompts, I can hear subtle noise
> >>when the person is speaking -- this is irrespective of whether I listen
> >>to the prompts through the TDM-400 on an analogue phone or whether I do
> >>so directly on a workstation. It has to be possible to do better than that.
> >>
> >>It doesn't have to be CD quality sound, but it should be clear. How can
> >>the sound quality be improved within the limitations of an 8 kHz sample
> >>rate? 
> > 
> > I presume you know the answer: augment the .gsm files with 8k sample rate 
> > .wave files?
> 
> If I knew the answer, I wouldn't be asking. There are other ways to
> improve sound quality, like changing the resolution.

I'm sorry, I wasn't trying to be funny.

The recorded prompts supplied with Asterisk are encoded with the .gsm 
codec.  That makes them sound like audio sounds on your GSM cellphone.  
Which is noticably worse than true PCM audio.

Now in the telephone world "best quality" still isn't very good - its ulaw 
or alaw encoded 8kHz audio.  That's frequency response up to 3.5kHz and 
about 12 or 13 bits of dynamic range.

But the fuzzyness you hear on the standard Asterisk prompts is due, I'm 
sure, to the use of gsm compression.

Its not necessary that they are stored in gsm - Asterisk has got an 
elaborate logic for finding the "best" version of the prompt, so you can 
store the same prompt in lots of formats and Asterisk will choose the 
"best".

You can see the formats with "show file formats"

So you need them stored in a non-compressed format - the simplest is 
probably wav format, another possibility is to use sox to put them into 
ulaw directly

Now Digium hasn't made the standard prompts available in a format other 
than gsm.  I don't know why.

For us we recorded the prompts in South African voice and so we have 
those.  You need to either extract the original non-compressed prompts 
from Digium (if they have them), or take it as an opportunity and record 
your own set in Canadian accent.

Hope that helps.

Steve
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