date:20060505

Marc Scheuffler wrote:
> Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different 
> mobile network providers. Nothing.

There was a bug in various versions of Asterisk when outbound calls were
placed using spool files and then could not detect DTMF from the called
party. Without more details, including the version of Asterisk you are
running, it will be difficult to suggest anything to you other than
upgrading to the latest release.
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Re: [Asterisk-Users] SIP Phones behind dynamic IPs

Chris Bagnall wrote:

> Okay, so assuming I've got to drop the re-registration to a much shorter
> time than the default of every hour, what are the implications of doing so
> (in terms of network traffic, load on the asterisk box, etc.)? What's the
> lowest one can reasonably take it? 10 minutes? 1 minute?

Network traffic is negligible; a few packets each way (REGISTER, 407,
ACK, REGISTER, 200, ACK). There will be some additional load on the
Asterisk server as it has to update the database each time a phone
registers.

The systems I have built all use a 5-minute registration timer and they
seem to work just fine.
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Re: [Asterisk-Users] ISAC support?

Trond G. Andersen wrote:

> Has there been done any work to support ISAC ?

ISAC is a proprietary codec from Global IP Sound. There will not be any
support for it in Asterisk unless GIPS wants to either open-source the
codec (not likely) or allow Digium to license it in the same method as
the G.729 codec is licensed.

In addition, Skype uses ISAC in 16KHz mode, which Asterisk cannot
support yet.
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RE: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones

2006-05-05 Thread Mark Ackroyd


> There was a bug in various versions of Asterisk when outbound calls were
> placed using spool files and then could not detect DTMF from the called
> party. Without more details, including the version of Asterisk you are
> running, it will be difficult to suggest anything to you other than
> upgrading to the latest release.

Works perfectly on mine. Using Digium Digital boards (ZAP -> UK Mobile).
Asterisk version 1.2.1 with a few patches.

Mark




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[Asterisk-Users] DTMF Tones within my Asterisk on all type of Channels

2006-05-05 Thread Stefan Agethen

Hi,

i am using Asterisk 1.2.6 with Debian Kernel 2.6.8-2-386 and up 2 date
Zaptel, Libpri and so on.

My Hardware :

4 x SNOM 320
One Wildcard TDM40B
3 x Allnet 7950
1 x Fritz!Box  7050 - working as a ATA

I use Asterisk in my Business stable since last year, the problem occurs
since 2 or 3 months.

I have detected some DTMF Tones within my conversations and turned on a
DTMF Logfile to get more infos about this.

The LOG reports Tones - they looks like appearing randomly on any type
of channel (mISDN, ZAP, SIP) and i cant find any reason why.

Here is a part of the LOG :

Apr 13 10:18:59 DTMF[5007] channel.c: SIP/30-1352 : *
Apr 13 10:39:52 DTMF[5165] channel.c: SIP/20-5073 : #
[...]
Apr 15 07:50:53 DTMF[14182] channel.c: Zap/1-1 : A
Apr 15 07:51:44 DTMF[14182] channel.c: Zap/1-1 : A
Apr 15 13:04:36 DTMF[14909] channel.c: SIP/50-7aa2 : 4
Apr 16 20:20:11 DTMF[19677] channel.c: mISDN/1-u61 : 9
[...]
Apr 18 07:33:32 DTMF[26583] channel.c: Zap/1-1 : 5
Apr 18 07:38:50 DTMF[26663] channel.c: SIP/50-21c4 : 9
[...]

As you can see, this happens on any type of available channel, the user
can hear the tone and in the baddest case the "*" appears
and the function to transfer in the DIAL is active (tT)..

Can you give me a tip to get this problem solved or locate it ?

Best Greets from Germany,

Stefan
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Re: [Asterisk-Users] Help with IRQ conflict between wct2xxp and eth0

2006-05-05 Thread Giorgio Incantalupo


Hi Phil,
may sound a stupid advice buthave you tried to change PCI slots on 
your server?


Giorgio

Phil Menico wrote:


I have a conflict problem with the eth0 card and wct2xxp digium board. 
The PRI can receive calls but my network connection is gone.


When I "cat /proc/interrupts" I get the following:

1 ..

1 ..

..

..

..

169 0 IO-APIC-level wct2xxp, eth0

..

etc.

even before I "modprobe wct2xxp"

After I "modprobe wct2xxp" and "modprobe wctdm" and again run "cat 
/proc/interrupts"


I then get:

 


..

..

..

..

..

169 118489 IO-APIC-level wct2xxp, eth0

201 118497 IO-APIC-level wctdm

..

etc

 

How can I force the wct2xxp to load on a separate IRQ? I tried moving 
the eth0 to IRQ 10 but could not.


Any ideas?

 
 


Thank you.

_*/Phil Menico/*_

XTEND Communications
171 Madison Avenue, New York, NY 10016
212-951-7632 (Office)
212-951-7683 (Fax)
www.xtend.com

 




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Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

2006-05-05 Thread Mark Coccimiglio





Ok,
   I have to agree here.  IF my simple fax server log/tiff archive is
not enough to satisfy a client that the fax is genuine I would not want
them as my customer.  I don't care how much money they spend.  Business
is business and what I do is what I do.  There has to be at least a
little bit of trust between me and my clients otherwise we spend too
much time bickering about stuff.  Have you ever tried to bill
"b*tch-time"?
I have worked in "High-CYA" positions before and refuse to have that in
my current company.

Aloha,

  Mark C

Scott Gifford wrote:

  
I don't see the advantage to this; the client still has to trust that
all of this is done correctly, and if they don't trust the fax
recipient to put the correct fax in the paper file or keep the correct
TIFF, why would they trust them to do this?

Using a third party to receive and relay the fax, one which is trusted
by both the client and the fax recipient, would solve the problem; the
third party could create a document with the caller information
(ideally from ANI, which is harder to forge), the time, and the
message itself, then digitally sign it.  This might even be an
interesting business plan, for some applications where confirmed
document transmittal is important.

But it's hard for me to imagine this isn't overkill; if a client and a
service provider distrust each other so thoroughly that they have to
communicate through a third party to verify integrity, probably they
just shouldn't do business with each other.

Scott.

  



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Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

2006-05-05 Thread Alejandro Vargas


2006/5/5, Eric ManxPower Wieling <[EMAIL PROTECTED]>:

> whether that makes a difference. Should I switch from
> hostname to IP address in the register string too?

If Asterisk has a DNS lookup failure it will never retry that lookup.


Well... I think it should be very easy to solve for the developpers
what knows exactly where in the code to make the changes...

--
Alejandro Vargas
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Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

2006-05-05 Thread Alejandro Vargas


2006/5/5, Steve Underwood <[EMAIL PROTECTED]>:

In most cases, forensic analysis of the audio from another machine would
easily show it was a fake. It would lack tell-tale fingerprints of the
true path, unless it was done with extreme care. Certainly using exactly
the same model of FAX machine that sent the genuine FAX would be a must.
Not just for the vendor information it sends, but for the fine details
in how its modems behave. To pass of the altered fax as being from the
original sender would require careful control of the DSP.


People must realize that fax is not a secure method to send
information. There are protocols created to solve the legal problems,
that uses digital signs, standard formats, and relies in a 3d party
entity that certificates that the info was sent, recived and red,
logging the exact times. These methods are actually used by many
busines to send price listings, budgets, buy orders, invoices, bank
money transfers, etc. all automatcaly and electronicaly. A stock
system can detect a low count of some products, it sends price-listing
request to the providers and waits for the answers, someone with
authorities chooses the prefered one or request a bettre price. Then
orders the system to buy the products. The provider system receives
the request electronically, prepares the shipping, sends the invoice
electronically, etc. etc. All automatic and all legal.

Faxes are not the right way to do legal and reliable document
transfers. It is only a quick and unreliable way to show a document to
someone instead of dictating it by phone.


--
Alejandro Vargas
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Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

2006-05-05 Thread Steve Underwood


Alejandro Vargas wrote:


2006/5/5, Steve Underwood <[EMAIL PROTECTED]>:


In most cases, forensic analysis of the audio from another machine would
easily show it was a fake. It would lack tell-tale fingerprints of the
true path, unless it was done with extreme care. Certainly using exactly
the same model of FAX machine that sent the genuine FAX would be a must.
Not just for the vendor information it sends, but for the fine details
in how its modems behave. To pass of the altered fax as being from the
original sender would require careful control of the DSP.



People must realize that fax is not a secure method to send
information. There are protocols created to solve the legal problems,
that uses digital signs, standard formats, and relies in a 3d party
entity that certificates that the info was sent, recived and red,
logging the exact times. These methods are actually used by many
busines to send price listings, budgets, buy orders, invoices, bank
money transfers, etc. all automatcaly and electronicaly. A stock
system can detect a low count of some products, it sends price-listing
request to the providers and waits for the answers, someone with
authorities chooses the prefered one or request a bettre price. Then
orders the system to buy the products. The provider system receives
the request electronically, prepares the shipping, sends the invoice
electronically, etc. etc. All automatic and all legal.

Faxes are not the right way to do legal and reliable document
transfers. It is only a quick and unreliable way to show a document to
someone instead of dictating it by phone.


FAXes *are* accepted for many legal things. It is one of the things 
keeping FAX from dying completely. They are completely forgable, yet 
accepted. Before them telex ket going long after its sell by date 
because a telex was acceptable in court. They were far more forgable 
than FAX. An old boss if mine has to show telexes were forged as an 
expert witness. The only reason he could do this is because people are 
sloppy, and didn't get the spacing exactly like a real telex would have.


Steve

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AW: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones

2006-05-05 Thread Marc Scheuffler

Actually I am using Asterisk 1.2.7.1, zaptel 1.2.5, libpri 1.2.2

I ve tried many values for rx/txgain togeher with echocancel and relaxdtmf.

The detection is not working with call file, manager originate and not with the 
dial command to the mobile.
I have no ideas left.

I got it sometimes to work if I use a specific channel (i.e. Dial(ZAP/14/...)
But with the same vaules on a second call there is the DTMF problem again.
Sometimes the DTMF tones are choppy at the other end of the line.

There is also a Warning with outbound calls only:
 -- Called G1/016080...
-- Moving call from channel 2 to channel 31
May  5 12:45:33 WARNING[1403]: chan_zap.c:7744 pri_fixup_principle: Can't fix 
up channel from 2 to 31 because 31 is already in use
May  5 12:45:33 WARNING[1403]: chan_zap.c:9045 pri_dchannel: Unable to move 
channel 31!
-- Zap/31-1 is ringing
-- Zap/31-1 answered Zap/2-1

But everything is ok and working on not moblie connections

My actual settings:

Zaptel.conf
#
# Zaptel Configuration File
#

span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4

#
bchan=1-15
dchan=16
bchan=17-31

bchan=32-46
dchan=47
bchan=48-62

#
loadzone = de
#loadzone = us-old
#loadzone=gr
#loadzone=it
#loadzone=fr
#loadzone=de
#loadzone=uk
#loadzone=fi
#loadzone=jp
#loadzone=sp
#loadzone=no
#loadzone=hu
#loadzone=lt
#loadzone=pl
defaultzone=nl



Zapata.conf

;
; Zapata telephony interface
;
; Configuration file
;


[trunkgroups]

[channels]
;
context=default
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown

;
; sample 1 for Germany
;internationalprefix = 00
;nationalprefix = 0
;localprefix = 
;privateprefix = 
;unknownprefix =
;relaxdtmf=yes
overlapdial=yes
;priindication = outofband
;rxwink=300 ; Atlas seems to use long (250ms) winks
;toneduration=100
;usedistinctiveringdetection=yes
;busydetect=no
;immediate=no
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
;echocancel=yes
echocancelwhenbridged=yes
;rxgain=2.0
;txgain=0.0
;
group=1
callgroup=1
pickupgroup=1
immediate=no
;
;callprogress=yes
;progzone=us
;
;jitterbuffers=4
;

group=1
;switching=euroisdn
signalling=pri_cpe
context=from-pmx
rxgain=2.0
txgain=0.0
relaxdtmf=yes
echocancel=yes
callerid=asreceived
channel=>1-15, 17-31

group=2
;switching=euroisdn
signalling=pri_net
context=to-pmx
callerid=asreceived
channel=>32-46, 48-62






-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Kevin P. Fleming
Gesendet: Freitag, 5. Mai 2006 09:52
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: AW: [Asterisk-Users] DTMF detection when outgoing call 
tomobilephones

Marc Scheuffler wrote:
> Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different 
> mobile network providers. Nothing.

There was a bug in various versions of Asterisk when outbound calls were placed 
using spool files and then could not detect DTMF from the called party. Without 
more details, including the version of Asterisk you are running, it will be 
difficult to suggest anything to you other than upgrading to the latest release.
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 22, Issue 26

2006-05-05 Thread Docksnet


Hi to all.

My asterisk pbx has a tdm400p card with 2 FXO cards on it.
I configured the extensions.conf to send all the call incoming from that 
zap channels to an IVR system.
I see in the asterisk CLI the call incoming and the playback of the 
message custom/myfile but no sound is played on the channel, i cannot 
hear nothing.
If i change the configuration and i send the call to an internal sip 
extensions alla works great. Also works well when i configure a sip 
trunks to answer and playback the ivr message, i can hear it very well.


If I use the zap channel to do an outside call, all works well.

Asterisk varsion is 1.2.7.1

I'm going crazy, please send me some suggestions.
Thanks in advance.
Bye
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Re: [Asterisk-Users] Festival , Cannot hear the words after ","

2006-05-05 Thread Giorgio Incantalupo


Hi John,
are using enclosing in quotation mark your statement? (festival('text to 
speech') )


Giorgio



John Joseph wrote:
Hi 
  I am trying to use festivall with asterisk , I am
using RHEL4 , asterisk1.2.7.1 and festival-1.95-beta 
, I am able to hear the voice form the text file ,

when I dial to the extension, but when I  have ","  in
my text file , it plays only the text upto ","  
  and in the CLI  , the "," is shown as "|"

  I had cut and pasted CLI messages for
reference 


-- Executing Answer("SIP/326-78c7", "") in new
stack
-- Executing Festival("SIP/326-78c7", "Hello |
This is Joseph | How are  U  ") in new stack
  == Parsing '/etc/asterisk/festival.conf': Found
-- Executing Hangup("SIP/326-78c7", "") in new
stack
  == Spawn extension (from-internal, 555, 3) exited
non-zero on 'SIP/326-78c7'
-- Executing Macro("SIP/326-78c7", "hangupcall")
in new stack

 I had followed the link 
http://www.voip-info.org/wiki/view/Asterisk+festival+installation

for the installation
 		Thanks 
 			Joseph John 



Send instant messages to your online friends http://uk.messenger.yahoo.com 
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RE: [Asterisk-Users] Help with IRQ conflict between wct2xxp and eth0

2006-05-05 Thread Kerry Garrison

Go into the BIOS, disable every possible device such as floppy controller,
usb, serial, parallel, etc. If that doesn't work, move card to another slot.


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Giorgio Incantalupo
> Sent: Friday, May 05, 2006 1:39 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Help with IRQ conflict between 
> wct2xxp and eth0
> 
> Hi Phil,
> may sound a stupid advice buthave you tried to change PCI 
> slots on your server?
> 
> Giorgio
> 
> Phil Menico wrote:
> >
> > I have a conflict problem with the eth0 card and wct2xxp 
> digium board. 
> > The PRI can receive calls but my network connection is gone.
> >
> > When I "cat /proc/interrupts" I get the following:
> >
> > 1 ..
> >
> > 1 ..
> >
> > ..
> >
> > ..
> >
> > ..
> >
> > 169 0 IO-APIC-level wct2xxp, eth0
> >
> > ..
> >
> > etc.
> >
> > even before I "modprobe wct2xxp"
> >
> > After I "modprobe wct2xxp" and "modprobe wctdm" and again run "cat 
> > /proc/interrupts"
> >
> > I then get:
> >
> >  
> >
> > ..
> >
> > ..
> >
> > ..
> >
> > ..
> >
> > ..
> >
> > 169 118489 IO-APIC-level wct2xxp, eth0
> >
> > 201 118497 IO-APIC-level wctdm
> >
> > ..
> >
> > etc
> >
> >  
> >
> > How can I force the wct2xxp to load on a separate IRQ? I 
> tried moving 
> > the eth0 to IRQ 10 but could not.
> >
> > Any ideas?
> >
> >  
> >  
> >
> > Thank you.
> >
> > _*/Phil Menico/*_
> >
> > XTEND Communications
> > 171 Madison Avenue, New York, NY 10016
> > 212-951-7632 (Office)
> > 212-951-7683 (Fax)
> > www.xtend.com
> >
> >  
> >
> > 
> --
> > --
> >
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Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?


Tom Engleward wrote:

--- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>
wrote:

If Asterisk has a DNS lookup failure it will

never

retry that lookup.

"Never" meaning until the next "reload" command is
issued, or until the next "restart" command is

issued,

or until the next time the OS reboots, or until

the

next time asterisk and its config files are

deleted

and reinstalled?

Correct.  Perhaps "never automatically" would be a
better choice of words.

Wait, that was a multiple choice question: which
option is the least-drastic sufficiently drastic
option to force asterisk to retry?
If a mere "reload" is sufficient, then the periodic
test-and-reload which I suggested (which Andrew
Kohlsmith correctly said is just a band-aid) would at
least be an effective band-aid for the short term,
until the actual cause of the DNS lookup failure is
found and fixed. A proper fix is preferable, but a
band-aid is better than nothing. 


Any one of those will work.


Of course if using IP
address instead of hostname successfully avoids the
problem, then that's a better band-aid, but only is
practical if the host in question is not subject to
having its IP address changed.


Only if BOTH hosts are on dynamic IPs.  If only one of them is on a 
dynamic IP, then the dynamic one should be registering with the static 
server.  This is why registration was invented.




--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

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Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREA CHABLE?

2006-05-05 Thread Andrew Kohlsmith

On Thursday 04 May 2006 18:44, Colin Anderson wrote:
> Ah, true dat. However, if quality was crappy believe me my users would let
> me know. They are salespeople and wholly intolerant of anything that keeps
> them from yipping on the phone. I also run exactly the same rig at my house
> backhauled to our main Asterisk box to force me to eat my own dog food, and

:-)  I do the *exact* same thing, and for the same reasons.  :-)

> time. The only time I get chop is when I use my cordless to the TDM400:
> Cordless > TDM400 > * > IAX2 > * > PRI. Salespeople keep asking for
> cordless phones, but I keep telling them forget it. Cordless to an FXS to
> IAX over the Internet is just begging for trouble.

I don't have any trouble with my cordless phone to a TDM400, but there could 
be a wealth of factors playing into the chop you're seeing, as I'm sure 
you're aware.  I had a *really* bad digital cordless phone that I actually 
sent to Digium to help them figure out WHY this particular phone hated their 
FXS ports so badly, but I don't think I'd ever heard back from them.  I've 
never had another phone act that way though.  (Panasonic digital cordless, I 
have another one from Panasonic and it works perfectly fine.)

-A.

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Re: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones


Mark Ackroyd wrote:

There was a bug in various versions of Asterisk when outbound calls were
placed using spool files and then could not detect DTMF from the called
party. Without more details, including the version of Asterisk you are
running, it will be difficult to suggest anything to you other than
upgrading to the latest release.


Works perfectly on mine. Using Digium Digital boards (ZAP -> UK Mobile).
Asterisk version 1.2.1 with a few patches.


Also works for me with USA cell phones on at least two different 
Asterisk servers with version 1.2.x



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Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

2006-05-05 Thread Dave Cotton

On Fri, 2006-05-05 at 07:47 -0500, Eric "ManxPower" Wieling wrote:

> Only if BOTH hosts are on dynamic IPs.  If only one of them is on a 
> dynamic IP, then the dynamic one should be registering with the static 
> server.  This is why registration was invented.

Until I got static IPs for all of them I had 3 *s intercommunicating.
Solved the problem by doing a reload from ipup-local when the ip changes
it's handled automatically.

-- 
Dave Cotton <[EMAIL PROTECTED]>

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[Asterisk-Users] Call Hold and Retrieve

2006-05-05 Thread Sean Cook

Our current PBX allows us to put a call on hold and then anyone in the 
building can dial #9XXX and pick up that call.  I know that I can 
replicate a similar function by parking.  But I would really like to 
replicate the existing setup.  Something about having to train people to 
hit more than one button.  Any suggestions?


Sean
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Re: [Asterisk-Users] number that starts with star on PAP2

2006-05-05 Thread Warren Burstein

I wrote:

> In the PAP2's setup there are all of these "Vertical Service Activation
> Codes" that start with star and "Outbound Call Codec Selection Codes",
> also the setup menu is accessed by pressing star four times, could they
> be intefering with dialing numbers that start with a star? And is there
> any way to get *8 and *XXX to dial?

Time Bandit wrote:

> Why I did to mine is modify all the internal "Vertical Service
> Activation Codes" to be "**x" instead of "*x". There is probably a
> better way, but this worked for me.

We tried that, but users report they are still having the same problem 
(the site is located in a different country so I can't check myself).

Philippe Lindheimer wrote:
Yes - that's your problem. You need to porgram the dialpan in the PAP2 
appropriately, disable functions you don't want, etc.

We were trying to dial *100, and there wasn't anything in either of the 
Codes section that started with *1.  Do we have to disable every 
function that starts with a star to get anything to work?  Also, is a 
function disabled by clearing it?

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[Asterisk-Users] Repost: External voicemail and MWI on internal phone

2006-05-05 Thread Reuben Farrelly


Hi,

I didn't get any response to this posting so thought I'd post again in the hope 
that anyone who missed the posting the first time may be able to offer some ideas.


This problem isn't specific to the particular model of phone - I can see from 
sip debug that the local extension itself is not being sent any Messages-Waiting 
headers.


Any alternative strategy would also be welcomed if I'm off on the wrong track 
;-)

Thanks,
Reuben



 Original Message 
Subject: External voicemail and MWI on internal phone
Date: Mon, 17 Apr 2006 10:35:07 +1200
From: Reuben Farrelly <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com

Hi,

I have a Cisco 7941 phone running SIP, and for a variety of reasons [1] have
configured this in the short term to use my local Asterisk server to register
to, and then have my local Asterisk server in turn register to the upstream SIP
proxy at my voice provider.

Voicemail is supplied by my VoIP provider, so I've no real need to set up
voicemail locally.

My provider also runs Asterisk, and correctly sends me down these headers:

Messages-Waiting: yes
Voice-Message: 1/0 (0/0)

However the local phones never see the headers and thus never know that there is
voicemail waiting upstream.

Is there a way to forward on or regenerate the Messages-Waiting and
Voice-Message headers from upstream, to each and every local phone in the same
context so
that any and all phones on the local network see the voicemail present and have
their MWI light up?

Thanks,
Reuben


[1] So far I've been unable to get the 7941 to do authenticate to my
provider, unsure quite why, the phone seems to just ignore WWW-Authenticate
requests from upstream.  A Cisco 7940 and ATA-188 work fine so it may be a bug
or a feature that needs to be configured - dunno.



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[Asterisk-Users] Registering Remote Sipura to Asterisk (both behind firewall)

Can anybody point or provide working configuration how to register
Sipura to Asterisk over the Internet. Both Sipura and Asterisk are
behind firewalls. 

I'll be force to use SIP as that is the only protocol that Sipura is
using. 
Do I need to enter any "STUN Server:" setting in "SIP" tab. 

On Asterisk I think I only need to make changes in sip.conf isn't it? 
What ports do I need to open on my firewall?

-- 
#Joseph
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Re: [Asterisk-Users] number that starts with star on PAP2

2006-05-05 Thread Time Bandit

 > Why I did to mine is modify all the internal "Vertical Service
 > Activation Codes" to be "**x" instead of "*x". There is probably a
 > better way, but this worked for me.

We tried that, but users report they are still having the same problem
(the site is located in a different country so I can't check myself).

Sorry, I don't have my PAP2 under hand, but this is all I did, changed
every *xx to **xx and it worked.

Something that may help you is
http://www.netphonedirectory.com/pap2_dialplan.htm

Philippe Lindheimer wrote:
> Yes - that's your problem. You need to porgram the dialpan in the PAP2
> appropriately, disable functions you don't want, etc.
>
We were trying to dial *100, and there wasn't anything in either of the
Codes section that started with *1.  Do we have to disable every
function that starts with a star to get anything to work?  Also, is a
function disabled by clearing it?

I didn't try that so I don't know. Just make sure that you changed
every single "Vertical Service Activation Codes" to a double *. If you
still can't fix it, let me know and I will get back my PAP2 and try to
help you

hth
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[Asterisk-Users] Dumping queue_log to MySQL









Anyone
have a working solution for this? I played with the demo that came with
QueueMetrics to see how they were doing it and it was working for a bit but now
somehow every night it stopped. Perl and Tail are still running on the server
but the information is not dumping to the MySQL database. I don’t get any
error messages anywhere telling me why it stops. As far as tail and perl are concerned
everything is fine. 

We
will be using this for a call center and need more reliability. Anyone got one
working?

 

Thanks

 

_

 

Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc

 






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[Asterisk-Users] problem g729

2006-05-05 Thread cristian









Hello , 

 

I’m have this problem  before  copy codec in
the /usr/lib/asterisk/modules  before registration …..

 

My asterisk is Asterisk SVN-trunk-r20297 built by xxx@
xxx on a i686 running Linux on 2006-04-20 01:02:07 UTC

 

This erro :

 

codec_g729a.so]May 5 21:39:16 WARNING [6950]:
loader.c:731 __load_resource: misstng mod_data for codec_g729a.so

Segmentation fault

 

Thanks 

 

Cristian Latapiat 






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Re: [Asterisk-Users] problem g729

cristian wrote:

> My asterisk is Asterisk SVN-trunk-r20297 built by xxx@ xxx on a i686 running
> Linux on 2006-04-20 01:02:07 UTC

This has been covered many times on this list already. The G.729 codec
binary is not compatible with current SVN trunk. If you are running SVN
trunk in production, stop doing so now :-)
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Re: [Asterisk-Users] Dumping queue_log to MySQL

2006-05-05 Thread Jon Farmer

We use a Perl script called acd_logger.pl. I can't remember when I got it from now but if you want a copy then mail me directly.RegardsJon Jon FarmerTelford, Shropshire, UK- Original Message From: Kevin Savoy <[EMAIL PROTECTED]>To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, 5 May, 2006 2:57:06 PMSubject: [Asterisk-Users] Dumping queue_log to MySQL   Anyone have a working solution for this? I played with the demo that came with QueueMetrics to see how they were doing it and it was working for a bit but now somehow every night it stopped. Perl and Tail are still running on the server but the information is not dumping to the MySQL database. I don’t get any error messages anywhere telling me why it stops. As far as tail and perl are concerned everything is fine.   We will be using this for a call center and need more reliability. Anyone got one working?     Thanks     _     Kevin Savoy  Business Unit Telecom Analyst  2218 4th Ave W  Williston, ND 58801  Ph: 701-774-4023  Fax: 701-774-2901  http://www.novo1.com  Novo 1 is a service mark of Novo 1, Inc   ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users___
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Re: [Asterisk-Users] FW: NuFone Update: DIDs

2006-05-05 Thread Dovid Bender

i got the same email. They emailed everyone later that
there was an "error".
--- Steve Prior <[EMAIL PROTECTED]> wrote:

> Rich Adamson wrote:
> 
> > 
> > At least outbound calls still work, even though
> they changed IP 
> > addresses (and probably colo locations).
> > 
> > 
> 
> Maybe not so much now.  I just got a disconnect
> notice from nufone
> which states that I have a positive balance in my
> account, but still
> need to add money to bring it up above zero for my
> account to be
> re-enabled.  Somebody just broke the billing code I
> think...
> 
> Steve
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[Asterisk-Users] Call Transfer Disconnect (CT-5)

2006-05-05 Thread Andre Courchesne - Consultant


Hi,

 Anyone has experience in using Call Transfer Disconnect (CT-5) over a 
PRI with Asterisk ?


 Call Transfer Disconnect allows you to transfer a call to a third 
party and disconnect yourself from the communication and also freeing 
your PRI channels.


 Here is a document that explains how it works:
  http://www.callamericacom.com/pdf/ctd_instructions.pdf

 My question is how can I do this with Asterisk, especially with a 
softphone (currently using SJPhone).



Andre Courchesne
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Re: [Asterisk-Users] CARD.XML for MGCP cisco phone

2006-05-05 Thread Nicolas TOUSSAINT


Anyone has a working CARD.XML for cisco
MGCP phones?
 The one on the cisco site is old
and it's not working with the new 
 firmwares.
 
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Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

2006-05-05 Thread Andrew Kohlsmith

On Friday 05 May 2006 08:47, Eric "ManxPower" Wieling wrote:
> Any one of those will work.

I disagree; this is NOT an Asterisk DNS issue.  I specify my hosts by IP and 
this still happens (in fact, it happened this morning).

-A.
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[Asterisk-Users] Realtime, 2 server setup problem?

2006-05-05 Thread Matt Schulte

All, we're running realtime and ast 1.2.7.1 stable. The problem I'm
having is when you register a SIP device on ServerA, works fine, sip
show peer works fine. When you dial the SIP device from ServerB however,
it tries to dial, even does a MySQL query like it should but comes back
saying no route to dest.

When I do a 'sip show peer ..' on ServerB, the peer comes back fine, it
even shows all the proper fields listed in the database (ie, codec, dtmf
mode, etc). However, the IP address field is coming back as
"unspecified".

Now, naturally I ensured server time was acurate with NTP for starters.
This was to avoid the registration appearing as EXPIRED on ServerB.

Sip.conf snippet (same on ServerA and ServerB)

...
rtcachefriends=yes
rtupdate=yes
rtautoclear=yes
rtignoreexpire=yes

So, as you can all see it's most importantly caching the peer and
updating MySQL. And yes if you do a 'realtime load sipusers name
PEERNAME', it shows the ipaddr field with the correct value.

Is there something obvious I am missing? I googled this to death and
cannot find anyone with a similar issue. I remember running into this
issue in the past and quite frankly am unsure if it has *ever* worked.

Thanks in advance,

Matt Schulte



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Re: [Asterisk-Users] Dumping queue_log to MySQL

2006-05-05 Thread BJ Weschke

 On 5/5/06, Jon Farmer <[EMAIL PROTECTED]> wrote:



We use a Perl script called acd_logger.pl. I can't remember when I got it from now but if you want a copy then mail me directly.Regards
Jon
 Jon FarmerTelford, Shropshire, UK 


- Original Message From: Kevin Savoy <
[EMAIL PROTECTED]>To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, 5 May, 2006 2:57:06 PMSubject: [Asterisk-Users] Dumping queue_log to MySQL



Anyone have a working solution for this? I played with the demo that came with QueueMetrics to see how they were doing it and it was working for a bit but now somehow every night it stopped. Perl and Tail are still running on the server but the information is not dumping to the MySQL database. I don't get any error messages anywhere telling me why it stops. As far as tail and perl are concerned everything is fine. 

We will be using this for a call center and need more reliability. Anyone got one working?

 
 We've had a fair amount of success using the script attached with the following mysql schema.
 
 mysql> desc queue_actions;+---+-+--+-+-+---+| Field | Type    | Null | Key | Default | Extra |+---+-+--+-+-+---+
| queueid   | int(11) |  | MUL | 0   |   || uniqueid  | varchar(25) |  | MUL | |   || queuename | varchar(45) |  | MUL | |   || agentname | varchar(45) |  | | |   |
| action    | varchar(25) |  | MUL | |   || alt1  | varchar(10) |  | | |   || alt2  | varchar(10) |  | | |   || userfield | varchar(75) |  | | |   |
| calltype  | int(4)  | YES  | | NULL    |   || nodename  | varchar(25) | YES  | | NULL    |   |+---+-+--+-+-+---+10 rows in set (0.00 sec)
-- Bird's The Word Technologies, Inc.http://www.btwtech.com/ 


insert-queue-info.pl
Description: Binary data
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[Asterisk-Users] asterisk 1.2 & hisax teles 16.3 isa

2006-05-05 Thread Arne Van Theemsche

Hi

I was using asterisk 1.0.7 until now. For my connection with the PSTN I
used a good old isa hisax teles 16.3 isdn card. Now while compiling 1.2
I nothiced that i4l and chan_modem is no longer supported? Is there any
way of getting my good old ISA card to work under 1.2? 
The docs keep saying use misdn, capi, etc from now on. But I don't think my card is CAPI capable

arne

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Re: [Asterisk-Users] Realtime, 2 server setup problem?

Matt Schulte wrote:

> Is there something obvious I am missing? I googled this to death and
> cannot find anyone with a similar issue. I remember running into this
> issue in the past and quite frankly am unsure if it has *ever* worked.

Seriously? This has been discussed many times on this list and the -dev
list.

It is not expected to work at this time. The current version of Asterisk
does not have the ability to share realtime registrations between two
servers (they can share the same table as long as they have separate
lists of peers).

In other words, your last comment above is correct :-)
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RE: [Asterisk-Users] Dumping queue_log to MySQL









http://forums.digium.com/viewtopic.php?t=4073

 

I modified it some, but it integrates with my cms and
provides realtime call stats. Contact me off list if you’ve like to see how it
works.

 



Wes Baehr

Ability Business Computing, Ltd.

Office:  330.882.0455 x25

Cell:  330.882.0455 x35

Fax:  330.882.0455

[EMAIL PROTECTED]

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Savoy
Sent: Friday, May 05, 2006 9:57 AM
To: 'Asterisk
 Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Dumping
queue_log to MySQL



 

Anyone
have a working solution for this? I played with the demo that came with
QueueMetrics to see how they were doing it and it was working for a bit but now
somehow every night it stopped. Perl and Tail are still running on the server
but the information is not dumping to the MySQL database. I don’t get any error
messages anywhere telling me why it stops. As far as tail and perl are
concerned everything is fine. 

We will
be using this for a call center and need more reliability. Anyone got one
working?

 

Thanks

 

_

 

Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave
  W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc

 








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Checked by AVG Free Edition.
Version: 7.1.392 / Virus Database: 268.5.4/332 - Release Date: 5/4/2006
 



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Checked by AVG Free Edition.
Version: 7.1.392 / Virus Database: 268.5.4/332 - Release Date: 5/4/2006
 

  
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Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?


Andrew Kohlsmith wrote:

On Friday 05 May 2006 08:47, Eric "ManxPower" Wieling wrote:

Any one of those will work.


I disagree; this is NOT an Asterisk DNS issue.  I specify my hosts by IP and 
this still happens (in fact, it happened this morning).


Then you are experiencing a different problem.


--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

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[Asterisk-Users] Registering Remote Sipura to Asterisk (both behind firewall)

Can anybody point or provide working configuration how to register
Sipura to Asterisk over the Internet. Both Sipura and Asterisk are
behind firewalls. 

I'll be force to use SIP as that is the only protocol that Sipura is
using. 
Do I need to enter any "STUN Server:" setting in "SIP" tab. 

On Asterisk I think I only need to make changes in sip.conf isn't it? 
What ports do I need to open on my firewall?

-- 
#Joseph
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[Asterisk-Users] Problem on Zap Channel with IVR

2006-05-05 Thread Docksnet


Hi to all.

My asterisk pbx has a tdm400p card with 2 FXO cards on it.
I configured the extensions.conf to send all the call incoming from that 
zap channels to an IVR system.
I see in the asterisk CLI the call incoming and the playback of the 
message custom/myfile but no sound is played on the channel, i cannot 
hear nothing.
If I change the configuration and i send the call to an internal sip 
extensions alla works great. Also works well when i configure a sip 
trunks to answer and playback the ivr message, i can hear it very well.


If I use the zap channel to do an outside call, all works well also.

Asterisk varsion is 1.2.7.1

This is my zapata.conf :

*Code:*

channels]

language=en
context=from-pstn
signalling=fxs_ks
rxwink=300  ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes
busydetect=yes
busycount=4
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no

;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no



and this is what I see in the asterisk cli during the call :

*Code:*

-- Starting simple switch on 'Zap/1-1'
   -- Executing Set("Zap/1-1", "LOOPCOUNT=0") in new stack
   -- Executing Answer("Zap/1-1", "") in new stack
   -- Executing Wait("Zap/1-1", "1") in new stack
   -- Executing Set("Zap/1-1", "TIMEOUT(digit)=10") in new stack
   -- Digit timeout set to 10
   -- Executing Set("Zap/1-1", "TIMEOUT(response)=10") in new stack
   -- Response timeout set to 10
   -- Executing BackGround("Zap/1-1", "custom/benvenut0") in new stack
   -- Playing 'custom/benvenut0' (language 'en')
 == Spawn extension (from-pstn, s, 6) exited non-zero on 'Zap/1-1'
   -- Executing Hangup("Zap/1-1", "") in new stack
 == Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/1-1'
   -- Hungup 'Zap/1-1'



I'm going crazy, please send me some suggestions.
Thanks in advance.
Bye
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[Asterisk-Users] Cisco 7970 running SIP question

2006-05-05 Thread Hall, Eric M.

 
Group
 I have a Cisco 7970 Running the newest SIP image. 
I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC

When I get a call the callerid number show something like
[EMAIL PROTECTED] I thought I seen somewhere what that was but I'm
unable to find the correct wording when searching Google to find that
post again. Can anyone help me out here. How can I remove the asterisk
servers IP from the phone number?


Also I'm unable to get the time zone correct on the phone. It is in UTC
and I'm in EST I see in the file where it looks like it goes but what I
have tried has not worked as of yet. Here is what it looks like

  
   M/D/Y
   EST
  


Thanks again for all your help!!!
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Re: [Asterisk-Users] Cisco 7970 running SIP question

2006-05-05 Thread Aaron Daniel

I don't remember exactly what the reasoning on Cisco's part is of having 
the IP address on there, but it happens on ours too.  It shouldn't cause 
any problems with making outgoing calls from the directory, it's just 
annoying to see it pop up.


As for the date time settings... this is what we have in ours:

  CMLocal
  M/D/YA
  Central Standard/Daylight Time


I'm guessing you should be able to change it to say Eastern instead of 
Central


On Fri, 5 May 2006, Hall, Eric M. wrote:



Group
I have a Cisco 7970 Running the newest SIP image.
I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC

When I get a call the callerid number show something like
[EMAIL PROTECTED] I thought I seen somewhere what that was but I'm
unable to find the correct wording when searching Google to find that
post again. Can anyone help me out here. How can I remove the asterisk
servers IP from the phone number?


Also I'm unable to get the time zone correct on the phone. It is in UTC
and I'm in EST I see in the file where it looks like it goes but what I
have tried has not worked as of yet. Here is what it looks like

 
  M/D/Y
  EST
 


Thanks again for all your help!!!
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--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Re: [Asterisk-Users] Unwanted conference with snom320 and asterisk1.07bristuffed

2006-05-05 Thread Franklin Webb

Title: Messaggio



We use a large number of Snom 320's and we have 
this same problem even with Call join on Xfer set to off.  I had not 
previously linked it decisively to the Snoms, but it sounds like that is likely 
our issue.  We've had to stay at the 4.5 firmware because otherwise we get 
additional incomming calls when our reps put someone on hold to make a 
transfer.
 
If I find any solution I'll be sure to share it 
with the list.
 
Thanks for sharing your experiences,
 
Frank Webb
Assistant Project Leader
Inter Media Marketing Solutions

  - Original Message - 
  From: 
  Alexander 
  Lopez 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Thursday, May 04, 2006 7:35 
AM
  Subject: RE: [Asterisk-Users] Unwanted 
  conference with snom320 and asterisk1.07bristuffed
  
  
  Under Advanced make 
  sure this is set:
   
  Call join on Xfer (2 
  calls): to off 
   
   
   
   
  
  
  
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Tommaso CalosiSent: Thursday, May 04, 2006 4:02 
  AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Unwanted 
  conference with snom320 and asterisk 
  1.07bristuffed
   
  
   I have 
  13 Snom 320 with asterisk 1.07 bristuffed. The problem is that sometimes on 
  random basis, when one customer is placed on hold and another call arrives, 
  the customers are put in conference with each other. This look very strange to 
  me, but I've disabled the confernce button on the snom phones to prevent the 
  human errors, but it still occurs. Investigating I've discovered that 
  a similar problem was fixed with the Snom320 Release 5.2  (http://www.snom.com/snom320_release_notes.html 
  ) It says: fixed unwanted conference bug in offhook/enter during 
  ringback with an incoming call BUT my phones are already running 5.2 
  firmware. Any idea? Am I the only one with this problem? 
  Do you think is the usual  buggy-snom firmware problem? Or it might 
  be an Asterisk problem? 
  
   
  
   
  
   
  
  

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Re: [Asterisk-Users] Dumping queue_log to MySQL

2006-05-05 Thread Matt Roth


Kevin Savoy wrote:

Anyone have a working solution for this? I played with the demo that 
came with QueueMetrics to see how they were doing it and it was 
working for a bit but now somehow every night it stopped. Perl and 
Tail are still running on the server but the information is not 
dumping to the MySQL database. I don’t get any error messages anywhere 
telling me why it stops. As far as tail and perl are concerned 
everything is fine.


We will be using this for a call center and need more reliability. 
Anyone got one working?



Kevin,

We are using QueueMetrics to provide reporting for our inbound call 
center activities. On a heavy day we're handling 13,000+ calls spread 
across roughly 6 to 10 queues. Due to some of the specifics or our 
setup, we are using a custom program to load queue_log data into MySQL.


There may be a little tweaking involved to get things up and running 
(it's rare to find a product that is a *perfect* fit for your business), 
but once you're there I think you'd be happy with QueueMetrics. The 
developers are very responsive to feedback and our floor managers are 
extremely pleased with the reporting.


Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question

2006-05-05 Thread Hall, Eric M.

 Aaron
Yes it is very annoying!
Thanks for the date time settings. That worked GREAT!!!

Thanks
- Eric
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Daniel
Sent: Friday, May 05, 2006 11:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question

I don't remember exactly what the reasoning on Cisco's part is of having
the IP address on there, but it happens on ours too.  It shouldn't cause
any problems with making outgoing calls from the directory, it's just
annoying to see it pop up.

As for the date time settings... this is what we have in ours:

   CMLocal
   M/D/YA
   Central Standard/Daylight Time


I'm guessing you should be able to change it to say Eastern instead of
Central

On Fri, 5 May 2006, Hall, Eric M. wrote:

>
> Group
> I have a Cisco 7970 Running the newest SIP image.
> I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC
>
> When I get a call the callerid number show something like
> [EMAIL PROTECTED] I thought I seen somewhere what that was but I'm 
> unable to find the correct wording when searching Google to find that 
> post again. Can anyone help me out here. How can I remove the asterisk

> servers IP from the phone number?
>
>
> Also I'm unable to get the time zone correct on the phone. It is in 
> UTC and I'm in EST I see in the file where it looks like it goes but 
> what I have tried has not worked as of yet. Here is what it looks like
>
>  
>   M/D/Y
>   EST
>  
>
>
> Thanks again for all your help!!!
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>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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[Asterisk-Users] Music on Hold from Soundcard

2006-05-05 Thread Alex Robar

Hi everyone, Sent this out previously, but it didn't seem to show up. My apologies if this is a duplicate! I've been trying to get MoH to work 
from the line-in on my soundcard, but as of yet have had no success. I found 
this script that should allow for it to happen:
http://www.sineapps.com/news.php?rssid=722


The script, when run as the asterisk 
user, works properly and streams sound to stdin. I can use arecord to record wavs which playback fine. But when Asterisk starts MoH it 
stops it immediately afterwards with no explanation. Has anyone gotten this to 
work? Or does anyone have any ideas on how to debug why MoH stops immediately 
after starting?

Thanks in advance!Alex Robar-- Alex Robar
[EMAIL PROTECTED]

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Re: [Asterisk-Users] number that starts with star on PAP2

2006-05-05 Thread Philippe Lindheimer

You need to modify the dialplan within the PAP2 unit to allow that as a valid number or it won't pass it on. Take a look at the following, it is not specifically for the PAP2 but all the dialplan information should apply:     http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf  From: "Time Bandit" <[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion"Date: Fri, 5 May 2006 09:48:41 -0400Subject: Re: [Asterisk-Users] number that starts with star on PAP2> > Why I did to mine is modify all the internal "Vertical Service> > Activation Codes" to be "**x" instead of "*x". There is probably a> > better way, but this worked for
 me.>> We tried that, but users report they are still having the same problem> (the site is located in a different country so I can't check myself).Sorry, I don't have my PAP2 under hand, but this is all I did, changedevery *xx to **xx and it worked.Something that may help you ishttp://www.netphonedirectory.com/pap2_dialplan.htm>> Philippe Lindheimer wrote:> > Yes - that's your problem. You need to porgram the dialpan in the PAP2> > appropriately, disable functions you don't want, etc.> >> We were trying to dial *100, and there wasn't anything in either of the> Codes section that started with *1. Do we have to disable every> function that starts with a star to get anything to work? Also, is a> function disabled by clearing it?I didn't try that so I don't know. Just make sure that you changedevery single "Vertical Service Activation Codes" to a
 double *. If youstill can't fix it, let me know and I will get back my PAP2 and try tohelp youhth
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[Asterisk-Users] Access to sip.conf username field from dialplan




I have two SIP clients defined in 
sip.conf, as follows:
 
    
[dave]    
username=dave    
type=friend    secret=dave    
host=dynamic    context=local    
accountcode=deskoptional
 
    
[dave-laptop]    
username=dave    
type=friend    secret=dave    
host=dynamic    context=local    
accountcode=deskoptional
 
I want to refer to the username 
field in extensions.conf, but cannot find a function to do it.  I know 
Asterisk can see this value, because both "SIP SHOW PEER dave" and "SIP SHOW 
PEER dave-laptop" display the desired value under "Def. Username".
 
PS: I am new to Asterisk and this 
newsgroup, so a cc to my email address (nntp at deskoptional.com) would be 
appreciated.
 
 
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RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question

2006-05-05 Thread Hall, Eric M.

 
Aaron
 Any idea how to change it from 24hr to 12hr ?

Thanks again!

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M.
Sent: Friday, May 05, 2006 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question

 Aaron
Yes it is very annoying!
Thanks for the date time settings. That worked GREAT!!!

Thanks
- Eric
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Daniel
Sent: Friday, May 05, 2006 11:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question

I don't remember exactly what the reasoning on Cisco's part is of having
the IP address on there, but it happens on ours too.  It shouldn't cause
any problems with making outgoing calls from the directory, it's just
annoying to see it pop up.

As for the date time settings... this is what we have in ours:

   CMLocal
   M/D/YA
   Central Standard/Daylight Time


I'm guessing you should be able to change it to say Eastern instead of
Central

On Fri, 5 May 2006, Hall, Eric M. wrote:

>
> Group
> I have a Cisco 7970 Running the newest SIP image.
> I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC
>
> When I get a call the callerid number show something like
> [EMAIL PROTECTED] I thought I seen somewhere what that was but I'm 
> unable to find the correct wording when searching Google to find that 
> post again. Can anyone help me out here. How can I remove the asterisk

> servers IP from the phone number?
>
>
> Also I'm unable to get the time zone correct on the phone. It is in 
> UTC and I'm in EST I see in the file where it looks like it goes but 
> what I have tried has not worked as of yet. Here is what it looks like
>
>  
>   M/D/Y
>   EST
>  
>
>
> Thanks again for all your help!!!
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question

2006-05-05 Thread Aaron Daniel


The A at the end of the dateTemplate sets that.

Should read M/D/YA instead of M/D/Y.

Aaron

On Fri, 5 May 2006, Hall, Eric M. wrote:



Aaron
Any idea how to change it from 24hr to 12hr ?

Thanks again!

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M.
Sent: Friday, May 05, 2006 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question

Aaron
Yes it is very annoying!
Thanks for the date time settings. That worked GREAT!!!

Thanks
- Eric
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Daniel
Sent: Friday, May 05, 2006 11:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question

I don't remember exactly what the reasoning on Cisco's part is of having
the IP address on there, but it happens on ours too.  It shouldn't cause
any problems with making outgoing calls from the directory, it's just
annoying to see it pop up.

As for the date time settings... this is what we have in ours:

  CMLocal
  M/D/YA
  Central Standard/Daylight Time


I'm guessing you should be able to change it to say Eastern instead of
Central

On Fri, 5 May 2006, Hall, Eric M. wrote:



Group
I have a Cisco 7970 Running the newest SIP image.
I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC

When I get a call the callerid number show something like
[EMAIL PROTECTED] I thought I seen somewhere what that was but I'm
unable to find the correct wording when searching Google to find that
post again. Can anyone help me out here. How can I remove the asterisk



servers IP from the phone number?


Also I'm unable to get the time zone correct on the phone. It is in
UTC and I'm in EST I see in the file where it looks like it goes but
what I have tried has not worked as of yet. Here is what it looks like

 
  M/D/Y
  EST
 


Thanks again for all your help!!!
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--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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RE: [Asterisk-Users] FW: NuFone Update: DIDs

2006-05-05 Thread Steve Jones

I didn't get that message, but I got a "we're sorry for sending out the
bogus messages" message, saying that it was an error..

-Steve

-Original Message-
From: Steve Prior [mailto:[EMAIL PROTECTED] 
Sent: Thursday, May 04, 2006 6:40 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] FW: NuFone Update: DIDs

Rich Adamson wrote:

> 
> At least outbound calls still work, even though they changed IP 
> addresses (and probably colo locations).
> 
> 

Maybe not so much now.  I just got a disconnect notice from nufone
which states that I have a positive balance in my account, but still
need to add money to bring it up above zero for my account to be
re-enabled.  Somebody just broke the billing code I think...

Steve

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RE: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

2006-05-05 Thread Steve Jones

Just as another datapoint, I have a cheap (~$17) walmart cordless phone
at home hooked to my digium dual FXS card, and it works great, with the
possible exception that there is a buzz (I perceive it as ground hum)
for about the first 4 seconds of EVERY call, and slowly it diminishes.
I don't know if there's a component of the echo cancellation that is
fixing it, or if it's some capacitor draining that fixes it, but even my
toughest asterisk critic (my wife) hasn't complained a bit about it.
:-)

-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] 
Sent: Friday, May 05, 2006 8:47 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] why a perfectly fine iax2 host becomes
UNREACHABLE?


I don't have any trouble with my cordless phone to a TDM400, but there
could 
be a wealth of factors playing into the chop you're seeing, as I'm sure 
you're aware.  I had a *really* bad digital cordless phone that I
actually 
sent to Digium to help them figure out WHY this particular phone hated
their 
FXS ports so badly, but I don't think I'd ever heard back from them.
I've 
never had another phone act that way though.  (Panasonic digital
cordless, I 
have another one from Panasonic and it works perfectly fine.)

-A.

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[Asterisk-Users] AASTRA 9133i and PIX Firewall

2006-05-05 Thread Matt


Hi,
Does anyone have experience with the Aastra 9133i and a PIX firewall? 
My problem is that it passes data with no problem at all (that being
audio goes fine)... however my voicemail notifications do not work. 
If I take the phone OUT from behind the PIX firewall, voicemail

notifications work.   What do I need to do (on phone, firewall, or
asterisk) to get the msgwaiting indicator to work?
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Re: [Asterisk-Users] Registering Remote Sipura to Asterisk (both behind firewall)

2006-05-05 Thread Lucas Alvarez

Hi, you have to forward port 5060 udp and all  the others udp ports that 
asterisk uses for RTP, 1 to 2 by default, see rtp.conf.
At the sipura device in the server configuration you have to put the LAN 
ip of your Asterisk box as sip proxy server and firewall's ip as 
out-bound sip proxy.


Joseph wrote:


Can anybody point or provide working configuration how to register
Sipura to Asterisk over the Internet. Both Sipura and Asterisk are
behind firewalls. 


I'll be force to use SIP as that is the only protocol that Sipura is
using. 
Do I need to enter any "STUN Server:" setting in "SIP" tab. 

On Asterisk I think I only need to make changes in sip.conf isn't it? 
What ports do I need to open on my firewall?


 




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Re: [Asterisk-Users] Registering Remote Sipura to Asterisk (both behind firewall)

On Fri, 2006-05-05 at 15:37 -0300, Lucas Alvarez wrote:
> Hi, you have to forward port 5060 udp and all  the others udp ports that 
> asterisk uses for RTP, 1 to 2 by default, see rtp.conf.

Thank you for the hint.  The above part is clear.

> At the sipura device in the server configuration you have to put the LAN 
> ip of your Asterisk box as sip proxy server and firewall's ip as 
> out-bound sip proxy.

Where is the "server configuration" on the Sipura box, is it under "SIP"
tab?
Note: Sipura is running as a stand alone device behind NAT so there is
no Asterisk.
If I had an asterisk on the other end I would use IAX2.

Just to be clear I will be connecting:
Asterisk <--> NAT <--> Internet <--> NAT <--> Sipura-3K (No Asterisk)

I'm force to to this way as FWD IAX is not reliable. 

-- 
#Joseph

> 
> Joseph wrote:
> 
> >Can anybody point or provide working configuration how to register
> >Sipura to Asterisk over the Internet. Both Sipura and Asterisk are
> >behind firewalls. 
> >
> >I'll be force to use SIP as that is the only protocol that Sipura is
> >using. 
> >Do I need to enter any "STUN Server:" setting in "SIP" tab. 
> >
> >On Asterisk I think I only need to make changes in sip.conf isn't it? 
> >What ports do I need to open on my firewall?


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[Asterisk-Users] Call Center Phone with Auto Answer









Can anyone
recommend a phone to use in an inbound call center environment that has an auto
answer feature? We don’t want the agents having to acknowledge the call. The
call should just activate on the headphones. We have tried Grandstream 2000,
Polycom 301, 501 and 601. None of these support it.

 

Thanks

 

_

 

Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc

 






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[Asterisk-Users] Passing SIP Subscriptions???

2006-05-05 Thread Douglas Garstang




Can 
anyone tell me if they know if it's possible to pass/copy sip subscriptions from 
one Asterisk system to another? Can IAX do this? What about regexten? 

 
Doug.
 
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RE: [Asterisk-Users] Call Center Phone with Auto Answer

2006-05-05 Thread Cory Andrews









The Snom 360 spec says it has an auto-answer feature. 
You can download the data sheet here http://www.snom.com/snom360_voip_phone.html

 

 



Cory Andrews

Executive Vice President

++

VoIPSupply.com

PBXSelect.com

++

454 Sonwil
  Drive

Buffalo, NY 14225

voice - 800.398.VoIP X3402

fax - 716.630.1548

e - [EMAIL PROTECTED]

m - 716.907.4059

aim - B2Cory











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Savoy
Sent: Friday, May 05, 2006 3:41 PM
To: 'Asterisk
 Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Call Center
Phone with Auto Answer



 

Can
anyone recommend a phone to use in an inbound call center environment that has
an auto answer feature? We don’t want the agents having to acknowledge
the call. The call should just activate on the headphones. We have tried
Grandstream 2000, Polycom 301, 501 and 601. None of these support it.

 

Thanks

 

_

 

Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave
  W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc

 






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Re: [Asterisk-Users] Call Center Phone with Auto Answer

2006-05-05 Thread Mike Clark


Kevin Savoy wrote:

Can anyone recommend a phone to use in an inbound call center 
environment that has an auto answer feature? We don’t want the agents 
having to acknowledge the call. The call should just activate on the 
headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. 
None of these support it.



My Polycom phones support auto-answer. This link should get you started.

http://www.voip-info.org/wiki-Polycom+auto-answer+config

Mike
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Re: [Asterisk-Users] Call Center Phone with Auto Answer

2006-05-05 Thread Sean Cook


Kevin Savoy wrote:


Can anyone recommend a phone to use in an inbound call center 
environment that has an auto answer feature? We don’t want the agents 
having to acknowledge the call. The call should just activate on the 
headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. 
None of these support it.


Thanks

May be I am not understanding... Why not use agentlogin and have the 
agents always logged in with MOH... they get a beep and they are 
connected.. Change ackcall=no in agents.conf


Then you don't need auto-answer
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RE: [Asterisk-Users] Call Center Phone with Auto Answer

The problem with what is in wiki is that these calls are being sent to a
queue. There is no way to have the queue dial the preceding digit that I can
think of that would trigger this. In the example shown he has an 8 dialed
before the extension. How would I get Asterisk to dial an 8 before sending
the call to the logged in agent in the queue?

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark
Sent: Friday, May 05, 2006 2:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer

Kevin Savoy wrote:

> Can anyone recommend a phone to use in an inbound call center 
> environment that has an auto answer feature? We don't want the agents 
> having to acknowledge the call. The call should just activate on the 
> headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. 
> None of these support it.
>
My Polycom phones support auto-answer. This link should get you started.

http://www.voip-info.org/wiki-Polycom+auto-answer+config

Mike
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RE: [Asterisk-Users] Call Center Phone with Auto Answer

We are using agent login but we don't want MOH on the line at all times as
some of these phones could and probably will be connected in remote
locations. We don't want to stream MOH across frames chewing up bandwidth
when there are no calls to present to that phone. We do have the ackcall=no
in the agents.conf and it seems to have no affect.

Am I missing something here? Appreciate any help

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean Cook
Sent: Friday, May 05, 2006 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer

Kevin Savoy wrote:
>
> Can anyone recommend a phone to use in an inbound call center 
> environment that has an auto answer feature? We don't want the agents 
> having to acknowledge the call. The call should just activate on the 
> headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. 
> None of these support it.
>
> Thanks
>
May be I am not understanding... Why not use agentlogin and have the 
agents always logged in with MOH... they get a beep and they are 
connected.. Change ackcall=no in agents.conf

Then you don't need auto-answer
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Re: [Asterisk-Users] Cisco 7970 running SIP question

2006-05-05 Thread Mailing List

I don't remember exactly what the reasoning on Cisco's part is of having 
the IP address on there, but it happens on ours too.  It shouldn't cause 
any problems with making outgoing calls from the directory, it's just 
annoying to see it pop up.


It's so the phone routes the call to the correct server especially in a 
multiple server environment (ex: dialing a missed call)



_
Mobilcom
http://www.mobilcom.net


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[Asterisk-Users] Code parsing error?




This code executes just fine, and leaves 
the SIP peer's mailbox setting from sip.conf in variable target.
 
    exten => 
1,1,Set(target=${CHANNEL:4}-)    exten => 
1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox})    exten 
=> 1,n,VoiceMailMain(${target})    
 
However, every time it runs I get an error 
in the CLI as follows 
 
    WARNING[5629]: 
pbx.c:1366 ast_func_read: Can't find trailing parenthesis?
 
This happens right after it executes 
the first line of code, then execution continues normally.  I've looked at this until my 
eyes crossed at don't see any unbalanced parens or 
brackets. Perhaps I shouldn't worry since it seems to work, but what's going on 
here?
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[Asterisk-Users] 300 DID's required in Alpine Texas Area code 432

2006-05-05 Thread Bob's Leaky News Service


Have customer who requires 300 DID'd in Alpine Tx 432 if the price is
right. Email with details.
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[Asterisk-Users] Asterisk <--> NAT <--> Internet <--> NAT <--> Sipura-3K (No Asterisk)

So far I have gathered that on my NAT (where asterisk server is) I have
to port forward UDP ports: 5060 and range 1 - 2 to my asterisk
server

But I'm still stuck how to configure Sipura (behind NAT) what sip proxy
and out-bound sip proxy, under which tab to change.

-- 
#Joseph
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[Asterisk-Users] Code parsing error?

2006-05-05 Thread John covici

Did you mean to have the dash inside the braces -- this may be your
problem.

on Friday 05/05/2006 David L. West([EMAIL PROTECTED]) wrote
 > This code executes just fine, and leaves the SIP peer's mailbox setting from 
 > sip.conf in variable target.
 > 
 > exten => 1,1,Set(target=${CHANNEL:4}-)
 > exten => 1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox})
 > exten => 1,n,VoiceMailMain(${target})
 > 
 > However, every time it runs I get an error in the CLI as follows 
 > 
 > WARNING[5629]: pbx.c:1366 ast_func_read: Can't find trailing parenthesis?
 > 
 > This happens right after it executes the first line of code, then execution 
 > continues normally.  I've looked at this until my eyes crossed at don't see 
 > any unbalanced parens or brackets. Perhaps I shouldn't worry since it seems 
 > to work, but what's going on here?
 > 
 > 
 > 
 > 
 > 
 > 
 > 
 > This code executes just fine, and 
 > leaves 
 > the SIP peer's mailbox setting from sip.conf in variable target.
 >  
 >     exten => 
 > 1,1,Set(target=${CHANNEL:4}-)    exten => 
 > 1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox})    
 > exten 
 > => 1,n,VoiceMailMain(${target})    
 >  
 > However, every time it runs I get an 
 > error 
 > in the CLI as follows 
 >  
 >     WARNING[5629]: 
 > pbx.c:1366 ast_func_read: Can't find trailing parenthesis?
 >  
 > This happens right after it 
 > executes 
 > the first line of code, then execution continues normally.  
 >  face=Arial size=2>I've looked at this until 
 > my 
 > eyes crossed at don't see any unbalanced parens or 
 > brackets. Perhaps I shouldn't worry since it seems to work, but what's going 
 > on 
 > here?
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 >http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 [EMAIL PROTECTED]
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[Asterisk-Users] Bandwidth via my Asterisk PBX

2006-05-05 Thread Dakota Burns

Am new to Asterisk - have it up and running & connected to a couple service providers (telasip & teliax).  Nice!
 
Am running our Asterisk PBX on a static DSL connection (~600 kbps up/~4mbps down), and would like to extend VoIP service to 10 non-profits we're working with.  Am I correct in assuming that all calls from each organization would route through our Asterisk server & be passed off to the service provider ( 
i.e. TelaSip, Teliax) thus keeping our bandwidth requirements to a minimum?  Or - are ALL calls that route through our Asterisk PBX consuming our bandwidth through the duration of the call? 
 
Thanks,
Dakota
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[Asterisk-Users] asterisk behind load-balancing switch

2006-05-05 Thread Jack Wei


Hi,


I have 2 SER and 2 Asterisk boxes behind a load-balancing switch.  I 
need Asterisk to initiate the RTP streams to both endpoints.  Can that 
be done?  Right now, Asterisk does it part of the time, so I have to 
create NATs for the Asterisk boxes for the RTP streams to get in.  Thanks.


Jack
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[Asterisk-Users] ODBC Voicemail storage and app_directory

2006-05-05 Thread Gary Reuter


I've checked the wiki, searched the mailing list, Mantis (bug
tracker), and looked through the source code, but found not mention of
this issue:

If using ODBC Storage for voicemail messages, name-playback by
app_directory breaks.
app_directory has no ODBC Storage code and therefore only looks for
the greet.wav file (recording of user's name) in the filesystem,
whereas app_voicemail saves the recording in the database.  (1.2.6
stable)

I'm hopnig I missed something somewhere and someone will point me to a
patch or the bug number in Mantis!
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RE: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREA CHABLE?

2006-05-05 Thread Colin Anderson

>toughest asterisk critic (my wife) 

yes, my wife is worth more in real-world "this sucks" feedback than an
entire office. keeps me on my toes. 
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[Asterisk-Users] Passing Callerid

2006-05-05 Thread William Piper

How do I setup a dial plan that will allow a customer to set their own
callerid?

No matter what I try, the username that I have for them in sip.conf is
passed because I have no callerid setup for them. If I setup a callerid in
sip.conf that is passed & I can change it and it works. 

That is well and fine but I want to send what the customer is passing as
their callerid... what am I doing wrong?

To eliminate the customer being wrong, I tried with a pair of my asterisk
boxes and I have the same problem. I even try setcallerid(1234567890) and it
still doesn't pass.

Could it be because I am having them register?

Thanks,

bp

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[Asterisk-Users] Running Asterisk as non-root


I saw where one should not run Asterisk as root:

http://www.voip-info.org/wiki/index.php?page=Asterisk+non-root

How important is this?

Thanks,  just curious whether it's worth the trouble.

H
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Re: [Asterisk-Users] Running Asterisk as non-root

2006-05-05 Thread Luki


I saw where one should not run Asterisk as root:
How important is this?


It's probably not very important at the moment, however, it's not that
hard to do either. I run Asterisk non-root and in a chrooted
environment -- it keeps all necessary files nicely separated (easily
portable, easy to switch versions), doesn't clog up common
directories. Just make a new directory like /usr/local/asterisk and
use that as the root for the chrooted environment. Chown all /var and
/etc/asterisk files in there to the asterisk user and you're good to
go. The tough part is to get all the shared libraries copies over --
ldd is your friend.

--Luki
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RE: [Asterisk-Users] Running Asterisk as non-root

2006-05-05 Thread Douglas Garstang

If you run Asterisk as non root, you may have problems installing G729 
licenses. The digium registration utility has certain hard coded stuff, and 
doesn't behave well when things aren't installed in the standard location.

Doug.

> -Original Message-
> From: Luki [mailto:[EMAIL PROTECTED]
> Sent: Friday, May 05, 2006 4:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Running Asterisk as non-root
> 
> 
> > I saw where one should not run Asterisk as root:
> > How important is this?
> 
> It's probably not very important at the moment, however, it's not that
> hard to do either. I run Asterisk non-root and in a chrooted
> environment -- it keeps all necessary files nicely separated (easily
> portable, easy to switch versions), doesn't clog up common
> directories. Just make a new directory like /usr/local/asterisk and
> use that as the root for the chrooted environment. Chown all /var and
> /etc/asterisk files in there to the asterisk user and you're good to
> go. The tough part is to get all the shared libraries copies over --
> ldd is your friend.
> 
> --Luki
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
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[Asterisk-Users] Re: Running Asterisk as non-root

Thanks for the swift response Luki.  Have you checked out:

http://www.voip-info.org/wiki/index.php?page=Asterisk+non-root

recently?  Does it seem up to date to you?  It indicates it was
updated in March of this tear, but I did this once and don't want to
go thru all the hassle if the directions are outdated!!

Cheers,
Howard

On 5/5/06, Luki <[EMAIL PROTECTED]> wrote:

> I saw where one should not run Asterisk as root:
> How important is this?

It's probably not very important at the moment, however, it's not that
hard to do either. I run Asterisk non-root and in a chrooted
environment -- it keeps all necessary files nicely separated (easily
portable, easy to switch versions), doesn't clog up common
directories. Just make a new directory like /usr/local/asterisk and
use that as the root for the chrooted environment. Chown all /var and
/etc/asterisk files in there to the asterisk user and you're good to
go. The tough part is to get all the shared libraries copies over --
ldd is your friend.

--Luki
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[Asterisk-Users] Re: Running Asterisk as non-root


Oh boy - thanks.  I use g729...

H

On 5/5/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:

If you run Asterisk as non root, you may have problems installing G729
licenses. The digium registration utility has certain hard coded stuff, and
doesn't behave well when things aren't installed in the standard location.

Doug.

> -Original Message-
> From: Luki [mailto:[EMAIL PROTECTED]
> Sent: Friday, May 05, 2006 4:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Running Asterisk as non-root
>
>
> > I saw where one should not run Asterisk as root:
> > How important is this?
>
> It's probably not very important at the moment, however, it's not that
> hard to do either. I run Asterisk non-root and in a chrooted
> environment -- it keeps all necessary files nicely separated (easily
> portable, easy to switch versions), doesn't clog up common
> directories. Just make a new directory like /usr/local/asterisk and
> use that as the root for the chrooted environment. Chown all /var and
> /etc/asterisk files in there to the asterisk user and you're good to
> go. The tough part is to get all the shared libraries copies over --
> ldd is your friend.
>
> --Luki
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[Asterisk-Users] REGISTER that isn't a register


Anyone come across this message:

WARNING[2203]: chan_sip.c:9633 handle_response_register: Got 200 OK on
REGISTER that isn't a register

I get it from time to time but don't know what to do about it!

Thanks,
h
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[Asterisk-Users] ASTERISK DISA FOR INCOMING DID CALL

2006-05-05 Thread ITN Info - 11-3898-0112









Hi,

 

I am trying to create a
situation where I call the DID number which is 1140636249 and I receive a dial
tone to dial. I d like also to autenticate the number 1130851536. 

I can see that asterisk
decode this number but I dont know how to authenticate this number only. This
is what I am doing

 

Sip.conf

 

[globo]

 

type=friend

username=itn111

fromuser=itn111

secret=123456

insecure=very 

host= globo.net.br 

context=fromttt

fromdomain= globo.net.br 

dtmfmode=rfc2833

disallow=all 

allow=g729

 

register => itn111:[EMAIL PROTECTED]:5060/itn111

 

where itn111 is the LOGIN for
DID and the virtual extension for Extensions.conf file

 

Extensions.conf


 

[fromttt] 

 

exten => itn111,1,Dial(SIP/29650,60,Ttr)


exten => itn111,2,Hangup()

 

This settings above can can
garantee that every call to 1140636249 goes to extension 29650. Do DID part is working ok.

Now I would like to get a
second dial tone when I call 1140636249 for asterisk DISA. 

I d like also to autenticate
the number 1130851536 (caller number) and only this number can receive the call

 

This is what I am trying to
do 

 

exten => itn111,1,Dial(SIP/29650,60,Ttr)


exten => 29650,2,DISA(no-password|brasil)
; I use no-password for this for now and Brasil context

exten => 29650,3,Hangup()

 

or

 

exten => itn111,1,DISA(no-password|brasil)

 

In sip show channels I see 

 

SIP/itn123456-cdfe  (fromgvt   
itn123456    1   ) 
Up DISA  no-password|brasil

 

But there s no dial tone. And
I don t know how to authenticate this number 1130851536. I see that asterisk
collect this number 

 

Can you pls help me to do
this settings ? 

 

 

 

Atenciosamente

 

 

 

Diretoria Comercial - Newton Medina

PABX    11.3085.1536

MSN [EMAIL PROTECTED] 

 

Rua
Augusta 2.212 SL 26 Jardins 01412001

São
Paulo - Brasil 

 

 






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Re: [Asterisk-Users] Code parsing error?

2006-05-05 Thread Ira


At 02:06 PM 5/5/2006, you wrote:

exten => 1,1,Set(target=${CHANNEL:4}-)

However, every time it runs I get an error in the CLI as follows

WARNING[5629]: pbx.c:1366 ast_func_read: Can't find trailing parenthesis?

This happens right after it executes the first line of code, then 
execution continues normally.  I've looked at this until my eyes 
crossed at don't see any unbalanced parens or brackets. Perhaps I 
shouldn't worry since it seems to work, but what's going on here?



I guess I'd have to ask, what's in CHANNEL?

Ira 


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Re: [Asterisk-Users] Re: Running Asterisk as non-root

2006-05-05 Thread Luki


> If you run Asterisk as non root, you may have problems installing G729
> licenses. The digium registration utility has certain hard coded stuff, and
> doesn't behave well when things aren't installed in the standard location.


Good point. However, in the chrooted environment there is no need to
make any changes to any paths or recompile asterisk. Asterisk thinks
it's dealing with /var/lib/whatever while in reality it's accessing
/usr/local/asterisk/var/lib/whatever.

While I do not use g929, I don't think you would have a problem with
the license install as long as you run the installation in the
chrooted environment as well:

chroot /usr/local/asterisk license-installation-script

I don't have time to write up the steps to chroot asterisk, but if
anyone is interested then I will tonight.

--Luki
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Re: [Asterisk-Users] Code parsing error?

2006-05-05 Thread Jon-o Addleman

On Fri, May 05, 2006 at 03:06:43PM -0600, David L. West spake thusly:
> This code executes just fine, and leaves the SIP peer's mailbox setting from 
> sip.conf in variable target.
> 
> exten => 1,1,Set(target=${CHANNEL:4}-)
> exten => 1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox})
  ^^ ^ "  "
There's definitely a ) missing in this line! A good text editor with
bracket matching (I'm using vim now) makes it a lot easier to find this
sort of thing.

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RE: [Asterisk-Users] Call Center Phone with Auto Answer

2006-05-05 Thread Philippe Lindheimer

I don't see any reason you can't use a polycom. You should be able to solve your problem multiple ways. You can simply put the default ring on the Polycom to autoanswer if that is the sole purpose, give it a second extension to be used in the queue that is programmed to autoanswer, as a couple of examples, or design your dialplan such that the appropriate _ALERT_INFO variable is set where the queue is concerned. p From: "Kevin Savoy" <[EMAIL PROTECTED]>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"Date: Fri, 5 May 2006 15:31:41 -0500Subject: RE: [Asterisk-Users] Call Center Phone with Auto AnswerThe problem with what is in wiki is that these calls are being sent to aqueue. There is no way to have the queue dial the
preceding digit that I canthink of that would trigger this. In the example shown he has an 8 dialedbefore the extension. How would I get Asterisk to dial an 8 before sendingthe call to the logged in agent in the queue?Thanks-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of Mike ClarkSent: Friday, May 05, 2006 2:55 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Call Center Phone with Auto AnswerKevin Savoy wrote:> Can anyone recommend a phone to use in an inbound call center > environment that has an auto answer feature? We don't want the agents > having to acknowledge the call. The call should just activate on the > headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. > None of these support it.>My Polycom phones support
auto-answer. This link should get you started.http://www.voip-info.org/wiki-Polycom+auto-answer+configMike___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Silent Attendant


I'd like to set up a "silent attendant".  By this I mean when someone
calls me I'd like for them to hear the comfort ringing tones, but for
the first 5 seconds I'd like to give them the option of pressing 9 to
send the call to an alternate extension;  if they don't press 9, the
call goes to a default extension.

For most callers I just want standard PSTN behaviour, only a select
few will be told about the silent 9 option

I'm stuck tho because it seems I need the Background command for this
to work, but this stops the comfort tones.  Any ideas?

Here's my attendant script, as you can see I currently have Background
commented out so the silent option is non functional.

Thanks,
H

exten => s,1,NoOp(${CONTEXT})

exten => s,n,Set(TIMEOUT(digit)=5)

exten => s,n,Set(TIMEOUT(response)=5)

;Menu prompt disabled for now.  Just let ringing continue for caller
and have it pass thru
;to General Delivery after timeout.
;exten => s,n,Background(AttendantMainMenu)

;exten => s,n,Background(silence/5)

;

exten => 9,1,Dial(Local/[EMAIL PROTECTED])
;

exten => ${EXT_CONFERENCEROOM},1,GoTo(conference-call-menu,s,1)

;

; If they take too long, send to general delivery

exten => t,1,Dial(Local/[EMAIL PROTECTED])
;

; "That's not valid, try again"

exten => i,1,Playback(invalid)

exten => i,2,Goto(Attendant-mainmenu,s,1)

;

exten => h,1,Hangup
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[Asterisk-Users] Info

2006-05-05 Thread Giordano Grandis









Hi all,

anyone could pls explain me what does it means ?

It a part of zaptel.conf file.

 

LBO= Line Build Out 
0: 0 dB (CSU) / 0 - 133 feet (DSX-1)

1: 133 - 266 feet (DSX-1)

2: 266 - 299 feet (DSX-1)

3: 399 - 533 feet (DSX-1)

4: 533 - 655 feet (DSX-1)

5: -7.5 dB (CSU)

6: -15 dB (CSU)

7: -22.5 dB (CSU)

 

Thanks 

 

Giordano








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[Asterisk-Users] How to determine if a device is in a call

2006-05-05 Thread Carl Youngblood


I have gotten intercom working on my office phones (Linksys SPA-942s),
but I have noticed that if someone is in a call, it places the call on
hold and sends the intercom audio to the person holding the phone that
is being paged.  I'd like to add logic to my dialplan that doesn't
send a page to a phone that is currently in a call.  But to do this I
need a function that will tell me if a device is in a call.  Any
suggestions?

Thanks,
Carl
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[Asterisk-Users] CW options not changing

2006-05-05 Thread Kerry Garrison




I have a weird issue 
with a system, running freepbx in devices and users mode (same as dozens of 
systems) but this one when you hit *70 and get "call waiting activated" it is 
not storing the setting in the database. I can manually set CW 900 ENABLED but 
then *71 does not disable it. Any suggestions?
 
 Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com 

 
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Re: [Asterisk-Users] Re: Running Asterisk as non-root

2006-05-05 Thread Ira


At 04:22 PM 5/5/2006, you wrote:

I don't have time to write up the steps to chroot asterisk, but if
anyone is interested then I will tonight.


Personally, I'd be very interested.

Thanks so much, Ira 


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[Asterisk-Users] Re: Code parsing error?

> There's definitely a ) missing in this line! A good text editor with

Thanks( I finally spotted it after fortifying myself with a good Thai 
dinner).  I've been using gEdit, so should probably start shopping for 
something vim-ish. 



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Re: [Asterisk-Users] How to determine if a device is in a call

2006-05-05 Thread Moises Silva


Check in voip-info.org about sip.conf parameter "call-limit". It may help you.

Regards

On 5/5/06, Carl Youngblood <[EMAIL PROTECTED]> wrote:

I have gotten intercom working on my office phones (Linksys SPA-942s),
but I have noticed that if someone is in a call, it places the call on
hold and sends the intercom audio to the person holding the phone that
is being paged.  I'd like to add logic to my dialplan that doesn't
send a page to a phone that is currently in a call.  But to do this I
need a function that will tell me if a device is in a call.  Any
suggestions?

Thanks,
Carl
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RE: [Asterisk-Users] How to determine if a device is in a call

Show application chanisavail

Should be what you need

Wes Baehr
Ability Business Computing, Ltd.
Office: 330.882.0455 x25 Cell: 330.882.0455 x35
[EMAIL PROTECTED]
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carl
Youngblood
Sent: Friday, May 05, 2006 8:35 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] How to determine if a device is in a call

I have gotten intercom working on my office phones (Linksys SPA-942s),
but I have noticed that if someone is in a call, it places the call on
hold and sends the intercom audio to the person holding the phone that
is being paged.  I'd like to add logic to my dialplan that doesn't
send a page to a phone that is currently in a call.  But to do this I
need a function that will tell me if a device is in a call.  Any
suggestions?

Thanks,
Carl
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[Asterisk-Users] Multiple periodic announcements in queues? Possible?

2006-05-05 Thread A.J. Paxson

Hi all!  Curious if there was a way to introduce multiple announcments in
the queues.  Rather than just:

"Thank you for holding.  Your call is important to us"
...wait 60 sec...
[repeat]

Is it possible to:

"Thank you for holding.  Your call is important."
...wait 60 sec...
"Did you know  you can.."
...wait 60 sec...
"We are currently experiencing high call volumes."
...wait 60 sec...
[repeat]

Maybe a:

Periodic-announce = AGI(rotate-msg)
Periodic-announce-frequency = 60

Thanks for any advice!!
~~Aaron

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RE: [Asterisk-Users] Silent Attendant

Simply generate or record a 5-second sample of ringing. Then use
Background() to play that ringing file - if someone presses 9, they will be
routed accordingly, or otherwise sent to your default extension.

Wes Baehr
Ability Business Computing, Ltd.
Office: 330.882.0455 x25 Cell: 330.882.0455 x35
[EMAIL PROTECTED]
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hugolivude
Sent: Friday, May 05, 2006 7:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Silent Attendant

I'd like to set up a "silent attendant".  By this I mean when someone
calls me I'd like for them to hear the comfort ringing tones, but for
the first 5 seconds I'd like to give them the option of pressing 9 to
send the call to an alternate extension;  if they don't press 9, the
call goes to a default extension.

For most callers I just want standard PSTN behaviour, only a select
few will be told about the silent 9 option

I'm stuck tho because it seems I need the Background command for this
to work, but this stops the comfort tones.  Any ideas?

Here's my attendant script, as you can see I currently have Background
commented out so the silent option is non functional.

Thanks,
H

exten => s,1,NoOp(${CONTEXT})

exten => s,n,Set(TIMEOUT(digit)=5)

exten => s,n,Set(TIMEOUT(response)=5)

;Menu prompt disabled for now.  Just let ringing continue for caller
and have it pass thru
;to General Delivery after timeout.
;exten => s,n,Background(AttendantMainMenu)

;exten => s,n,Background(silence/5)

;

exten => 9,1,Dial(Local/[EMAIL PROTECTED])
;

exten => ${EXT_CONFERENCEROOM},1,GoTo(conference-call-menu,s,1)

;

; If they take too long, send to general delivery

exten => t,1,Dial(Local/[EMAIL PROTECTED])
;

; "That's not valid, try again"

exten => i,1,Playback(invalid)

exten => i,2,Goto(Attendant-mainmenu,s,1)

;

exten => h,1,Hangup
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[Asterisk-Users] Re: Code parsing error?




Oh, on the off chance that it's interesting to 
anybody what I'm actually doing here.
 
I have two SIP devices, "dave" and 
"dave-laptop".  I want to have  one voicemail box but be able to check 
it from either one.  In sip.conf, both devices have "mailbox=dave".  
It's this setting that I'm retrieving with the code below and passing into 
VoiceMailMain.

  "David L. West" <[EMAIL PROTECTED]> wrote in 
  message news:[EMAIL PROTECTED]...
  This code executes just fine, and leaves 
  the SIP peer's mailbox setting from sip.conf in variable target.
   
      exten => 
  1,1,Set(target=${CHANNEL:4}-)    exten => 
  1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox})    
  exten => 1,n,VoiceMailMain(${target})    
   
  However, every time it runs I get an 
  error in the CLI as follows 
   
      WARNING[5629]: 
  pbx.c:1366 ast_func_read: Can't find trailing parenthesis?
   
  This happens right after it executes 
  the first line of code, then execution continues normally.  I've looked at this until my 
  eyes crossed at don't see any unbalanced parens or 
  brackets. Perhaps I shouldn't worry since it seems to work, but what's going 
  on here?
  
  

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RE: [Asterisk-Users] Silent Attendant

I changed my mind. This worked for me, using AEL:

Context test {
825 => {
Set(TIMEOUT(response)=5);
Answer();
Ringing();
WaitExten(5);
Goto(menu-main|s|1);
};
};

Or in regular format:

exten => 825,1,Noop
exten => 825,n,Set(TIMEOUT(response)=5)
exten => 825,n,Answer();
exten => 825,n,Ringing();
exten => 825,n,WaitExten(5);
exten => 825,n,Dowhatever


Wes Baehr
Ability Business Computing, Ltd.
Office: 330.882.0455 x25 Cell: 330.882.0455 x35
[EMAIL PROTECTED]
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hugolivude
Sent: Friday, May 05, 2006 7:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Silent Attendant

I'd like to set up a "silent attendant".  By this I mean when someone
calls me I'd like for them to hear the comfort ringing tones, but for
the first 5 seconds I'd like to give them the option of pressing 9 to
send the call to an alternate extension;  if they don't press 9, the
call goes to a default extension.

For most callers I just want standard PSTN behaviour, only a select
few will be told about the silent 9 option

I'm stuck tho because it seems I need the Background command for this
to work, but this stops the comfort tones.  Any ideas?

Here's my attendant script, as you can see I currently have Background
commented out so the silent option is non functional.

Thanks,
H

exten => s,1,NoOp(${CONTEXT})

exten => s,n,Set(TIMEOUT(digit)=5)

exten => s,n,Set(TIMEOUT(response)=5)

;Menu prompt disabled for now.  Just let ringing continue for caller
and have it pass thru
;to General Delivery after timeout.
;exten => s,n,Background(AttendantMainMenu)

;exten => s,n,Background(silence/5)

;

exten => 9,1,Dial(Local/[EMAIL PROTECTED])
;

exten => ${EXT_CONFERENCEROOM},1,GoTo(conference-call-menu,s,1)

;

; If they take too long, send to general delivery

exten => t,1,Dial(Local/[EMAIL PROTECTED])
;

; "That's not valid, try again"

exten => i,1,Playback(invalid)

exten => i,2,Goto(Attendant-mainmenu,s,1)

;

exten => h,1,Hangup
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Re: [Asterisk-Users] Re: Code parsing error?

2006-05-05 Thread A.J. Paxson

Title: Re: [Asterisk-Users] Re: Code parsing error?



On 5/5/06 10:45 PM, "David L. West"  wisely said:

exten =>  1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox})

Well:

Exten => 1,n,Set(target=${SIPPEER(${CUT(target,,1)}):mailbox})

It looks like you didn’t end the parenthesis for SIPPEER().  Maybe?

~~Aaron



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[Asterisk-Users] Re: Re: Code parsing error?

Title: Re: [Asterisk-Users] Re: Code parsing error?




>It looks like you didn¹t end the parenthesis for SIPPEER(). Maybe?
Yup, edited to look like this and it works w/o generating an error:
; Send to the user's vmail box by looking up the "mailbox" setting in 
sip.conf
exten => 1,1,Set(target=${CHANNEL:4}-)
exten => 1,n,Set(target=${CUT(target,,1)}:mailbox})
exten => 1,n,VoiceMailMain(${target}) 
I'm doing some tricky things in the dialplan and sip.conf so that I can have 
sip devices "dave" and "dave-laptop" point to the same voicemail box. Also 
rigging the "local" dialplan context to force calls to "dave" or "dave-laptop" 
to go to "dave", so that if dave-laptop calls somebody who then returns the call 
after dave-laptop has signed off it actually gets routed back to "dave". The 
idea is to abstract devices and users as separate layers rather than rely on the 
assumption that one user = one device. 

  "A.J. Paxson" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]...On 5/5/06 10:45 PM, "David L. West" 
   wisely said:
  
exten 
  => 
   1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox})Well:Exten => 
  1,n,Set(target=${SIPPEER(${CUT(target,,1)}):mailbox})It looks like you 
  didnt end the parenthesis for SIPPEER(). 
   Maybe?~~Aaron 
  
  

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Re: [Asterisk-Users] Silent Attendant