Re: [Asterisk-Users] polycom soundpoint ip 300

[Wilson Pickett]

 yes with anybody interesting in  ip phones for
 Asterisk

So you can send them a childish insult? I wrote you off list as
requested, and you wrote back something a 10 year old would say in
school.  What is your point, Harry?
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RE: [Asterisk-Users] RTP session between two end users

[Erdem HAKİ]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
aka ManxPower
Sent: Monday, June 27, 2005 8:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RTP session between two end users

Erdem HAKİ wrote:

 Is it possible that a RTP session between two end users  (so i want to use
 asterisk as a signaling proxy and bypass RTP sessions)?
 
  
 
 I used canreinvite=yes but it didn't work. 
 
 
 Description from asterisk conf. File;
 
 (canreinvite=yes; allow RTP voice traffic to bypass
 Asterisk)


It's sip.conf.  reinvites only work if the codec is the same for the 
two endpoints and Asterisk does NOT have to listen for DTMF (no t or T 
on the dial line, no meetme, etc.)

***
We use same codec and don't use meetme etc...  So what else should i do?

Thanks 

Erdem HAKI
***


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Re: [Asterisk-Users] Passing called number in SIP

[Andres]

However,  it's not really passing the called number per say.  What
it's doing is putting the extension I have in my register statement
into the To field.  I'm assuming the To field is actually being
populated with whatever * set the Contact field to when it
registered.This seems to mean that I need a unique username for
every SIP DID I have if I want to be able to route them to different
context's.

Is there a standard way of handling this issue when you have multiple
SIP DID's ?

Chris

 



Say you have a block of 100 DIDs with Level 3 for example.  You can just 
configure something like this in your incoming context:


exten = _30355597[0-9][0-9],1,Dial(SIP/${EXTEN},30)

(if you want each DID in a different context then you need a line for each)

--
Andres
Network Admin
http://www.telesip.net


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Re: [Asterisk-Users] H323

[Tzafrir Cohen]

On Tue, Jun 28, 2005 at 01:18:18PM +0800, Ronald_Wiplinger wrote:
 Can anybody give me a hint, what I am doing wrong, please?
 
 
 
 
 Asterisk and H.323
 
 1. download all parts
 -
 
 wget 
 http://www.inaccessnetworks.com/asterisk-oh323/download/asterisk-oh323-0.7.1.tar.gz

What version of Asterisk? I suppose you use HEAD, right? 0.7 requires
HEAD.

 wget 
 http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323_1.12.2.tar.gz
 wget 
 http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib_1.5.2.tar.gz

Isn't that an old version of pwlib and oh323?

 wget 
 http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/ohphone_1.4.1.tar.gz
 wget 
 http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz
 wget 
 http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz

WHIW, the debian debs in Sarge are of Asterisk 1.0, oh323 0.66pre3,
openh323 1.1.15 and pwlib 1.8.4 . They build. IIRC I even got some
reports that they work.

 
 and untar it into /usr/src/
 
 
 2. Compiling
 
 
 cd /usr/src/pwlib
 pwlib$ ./configure
 pwlib$ make clean; make opt
 
 cd /usr/src/openh323
 openh323$ patch -p1  
 /usr/src/asterisk-oh323-0.7.1/openh323_1.13.5-make.patch
 
 openh323$ ./configure
 openh323$ make clean; make opt
 
 
 3. Download / update Asterisk
 -
 
 cd /usr/src
 export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
 cvs login- the password is anoncvs.
 
 cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds

This is HEAD. That is: the version whose problems are less known. Be
sure to keep a record of the date you actually downloaded the files from
the CVS, as things may change from day to day.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] polycom soundpoint ip 300

[harry gaillac]

 Wilson Pickett,

I posted this mail for people interested in polycom
ip300 for asterisk.

Harry
--- Wilson Pickett [EMAIL PROTECTED] a écrit :

  yes with anybody interesting in  ip phones for
  Asterisk
 
 So you can send them a childish insult? I wrote you
 off list as
 requested, and you wrote back something a 10 year
 old would say in
 school.  What is your point, Harry?
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[Asterisk-Users] Some phone numbers always busy

[vdasilva]

Hello



I am using Asterisk 1.0.5 with i4l, a HFC
ISDN BRI card, and some Grandstream products. The system works fine except that
some external telephone numbers, when dialed always give a busy tone whereas
other numbers are fine. Ive checked extensions.conf and even tried hard
coding the number in extensions.conf and always get the same result.



Has anyone experienced something similar?



Vicente






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[Asterisk-Users] cheap HFC card on Bristuff vs cheap HFC card on i4l vs Fritz ISDN BRI card on CAPI

[vdasilva]

Hello

I have asterisk running in Red Hat 9 with a cheap HFC card on i4l. I have
choppy sound problems sometimes, and echo problems often. I am using a 2
port Grandstream ATA, Grandstream BT and a Grandstream GPX-2000

I read that changing to BriStuff will fix the echo problems, but have also
read other users say that the only way they solved the echo/choppy sound
problems was using a Fritz ISDN card with the CAPI drivers...

I have tried using bristuff on RH9 but couldn't get my zaptel to compile...
 
Then there is the issue of timing, ztdummy or zaprtcand QoS setup on the
Linux box...

Can anyone who has a 100% working Asterisk implementation using any of the
techniques described above tell me more...

I will happily upgrade to the Fritz card if it will solve all the
problems...

Thanks
Vicente


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stefan
Gofferje
Sent: 07 April 2005 09:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Choppy sounds after transferring to ISDN
clientor after a time

Hi,

Michiel van Baak schrieb:

I had the exact same thing when using a cheap HFC-S card
connected to my outside ISDN line. Replacing the card with
an AVM Fritz!PCI fixed this issue for me.
I tried a lot with the HFC-S card, different archs, SMP,
uniprocessor, nolapic, noapic, dual channel ram setup, 1
dimm only, removed all cards but the HFC-S, 4 different
versions of bristuffed. Nothing solved it.

What do you mean? The choppy sound or the log message? Trouble is, my 
HFC-S is internal ISDN. I can't use a Fritz!PCI for that because it 
isn't capable of NT mode...

Regards,
Stefan

-- 
 (o_   Stefan Gofferje  | Linux Systems Specialist
 //\   Reg'd Linux User #247167 | Network Security Specialist
 V_/_  Linux is like a Wigwam - No gates, no windows, Apache inside

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[Asterisk-Users] Re: H323

[Tony Mountifield]

In article [EMAIL PROTECTED],
Ronald_Wiplinger [EMAIL PROTECTED] wrote:
 Can anybody give me a hint, what I am doing wrong, please?
 
 Asterisk and H.323
 
 1. download all parts
 -
 
 wget 
 http://www.inaccessnetworks.com/asterisk-oh323/download/asterisk-oh323-0.7.1.tar.gz

If you are using HEAD, that's fine. If you are using STABLE, you should
get oh323-0.6.5 instead.

 wget 
 http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323_1.12.2.tar.gz
 wget 
 http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib_1.5.2.tar.gz

You don't need these two - they are old.

 wget 
 http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/ohphone_1.4.1.tar.gz

Don't know anything about this one.

 wget 
 http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz
 wget 
 http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz

These are not just patches, they are the complete source at patchlevel 4.
You use these instead of the old versions mentioned previously.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Passing called number in SIP

[snacktime]

On 6/27/05, Andres [EMAIL PROTECTED] wrote:
 
 
 However,  it's not really passing the called number per say.  What
 it's doing is putting the extension I have in my register statement
 into the To field.  I'm assuming the To field is actually being
 populated with whatever * set the Contact field to when it
 registered.This seems to mean that I need a unique username for
 every SIP DID I have if I want to be able to route them to different
 context's.
 
 Is there a standard way of handling this issue when you have multiple
 SIP DID's ?
 
 Chris
 
 
 
 
 Say you have a block of 100 DIDs with Level 3 for example.  You can just
 configure something like this in your incoming context:
 
 exten = _30355597[0-9][0-9],1,Dial(SIP/${EXTEN},30)
 

I don't get it:)  If I have 100 DID's but only one register statement,
isn't the called number for all 100 going to be the one extension name
I registered as?  Or am I missing something?  At least with the
provider I am testing with the called number is always the extension
in my register statement, regardless of what the DID really is.

For example I have 4 DID's with this one provider.  With the following
register statement they will all come in with the sip user/called
number as 111222:

register = user:[EMAIL PROTECTED]/111222

With this register statement they all come in to sip user/called number s:

register = user:[EMAIL PROTECTED]


What happens if I put a register statement for every DID?  Wouldn't that work?

Chris
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RE: [Asterisk-Users] is teliax down?

[Rick Baranowski]

Looks like the router around them in the Denver area have been having issues
or overloaded. I have been getting choppy calls all day but seems to be
better tonight.

Rick



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Megacz
Sent: Monday, June 27, 2005 6:31 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] is teliax down?


I'm getting really wierd errors from them, like bad packet checksums:

  Jun 27 18:22:56 NOTICE[29051]: rtp.c:435 ast_rtp_read: RTP: Received
packet with bad UDP checksum

  - a

-- 
I didn't see it then, but it turned out that getting fired was the
 best thing that could have ever happened to me. The heaviness of
 being successful was replaced by the lightness of being a beginner
 again, less sure about everything. It freed me to enter one of the
 most creative periods of my life.

  -- Steve Jobs, commencement speech at Stanford, June 2005

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Re: [Asterisk-Users] H323

[Ronald_Wiplinger]

Tzafrir Cohen wrote:


On Tue, Jun 28, 2005 at 01:18:18PM +0800, Ronald_Wiplinger wrote:
 


Can anybody give me a hint, what I am doing wrong, please?




Asterisk and H.323

1. download all parts
-

wget 
http://www.inaccessnetworks.com/asterisk-oh323/download/asterisk-oh323-0.7.1.tar.gz
   



What version of Asterisk? I suppose you use HEAD, right? 0.7 requires
HEAD.

 


It is head!

wget 
http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323_1.12.2.tar.gz
wget 
http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib_1.5.2.tar.gz
   



Isn't that an old version of pwlib and oh323?
 




In the README file is another version, which I cannot find, however on 
the web they  say: this are the files we compiled with


 

wget 
http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/ohphone_1.4.1.tar.gz
wget 
http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz
wget 
http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz
   



WHIW, the debian debs in Sarge are of Asterisk 1.0, oh323 0.66pre3,
openh323 1.1.15 and pwlib 1.8.4 . They build. IIRC I even got some
reports that they work.
 



what is the channel for that right?

It seems that with each try (remove and make it new) I am not anymore 
coming that far as with the try before ;-(



bye

Ronald

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Re: [Asterisk-Users] Dialogic D/300pci-E1 card

[tim panton]

On 28 Jun 2005, at 01:34, Eric Wieling aka ManxPower wrote:

I have no idea.  But since it's NOT the same part number, I would  
assume no.  Perhaps a call to Digium would be in order?


Florin Mandache wrote:



As is on that page :
D/300JCT-1E1 E1 + 30 voice so is compatible ??!??
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric  
Wieling

aka ManxPower
Sent: Monday, June 27, 2005 8:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dialogic D/300pci-E1 card
Florin Mandache wrote:


It is any way to use this card with Asterisk, and if yes, HOW to ?


According to this: http://asterisk.org/index.php?menu=hardware  no.





It isn't - If I remember right, the D/300pci is a 'half-duplex' card.  
It doesn't have enough

dsp resources to do bi-directional audio.

Anyhow, as things currently stand, the dialogic drivers are not free.
Quite honestly, my advice is to buy a new digium card and sell
the Dialogic. Thats what I did and I don't regret it. ( search the
archives for more discussion on this).

Tim.
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Re: [Asterisk-Users] Eicon equipment, BRI Server or PRI?

[Clive]

Hi

As far as I know, only the server versions of Eicon work with 
asterisk (using chan_capi).

There are a few other BRI cards that work with asterisk. Junghanns 
cards seem to work the best from the little I have seen.

Good luck.
Regards
Clive

On 27 Jun 2005 at 23:19, [EMAIL PROTECTED] wrote:

 Hello Everyone,
 I am once again wondering about EICON.  I have had no success with the
 Diva Pro or Diva Pro PCI, so my question is, does anyone use an Eicon
 Server BRI card on Asterisk?  Or would I be better off trying to get a
 split PRI?
 
 Regards,
 Greg
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[Asterisk-Users] HOW TO WRITE ROUTE PATTERNS DIALPLAN

[wassim darwish]

in routes pattern i tried to write pattern to usa
destination and that was 1* it worked well but when i
wanted to specify the number of digits then i tried
1NXXNXX but i didnt work.so i dont know what to
write please help.

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[Asterisk-Users] AVM CAPI INSTALLATION

[sylvain garcia]

I use AVM fritz card A1, I have install with instruction of
Voip-info.or, but it don' work.

capiinfo

Number of Controllers : 1
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.101-02  (49.18)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
   internal controller supported
   DTMF supported
   Supplementary Services supported
   channel allocation supported (leased lines)
B1 protocols support: 0x411f
   64 kbit/s with HDLC framing
   64 kbit/s bit-transparent operation
   V.110 asynconous operation with start/stop byte framing
   V.110 synconous operation with HDLC framing
   T.30 modem for fax group 3
   Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x0b1b
   ISO 7776 (X.75 SLP)
   Transparent
   LAPD with Q.921 for D channel X.25 (SAPI 16)
   T.30 for fax group 3
   ISO 7776 (X.75 SLP) with V.42bis compression
   V.120 asyncronous mode
   V.120 bit-transparent mode
B3 protocols support: 0x80bf
   Transparent
   T.90NL, T.70NL, T.90
   ISO 8208 (X.25 DTE-DTE)
   X.25 DCE
   T.30 for fax group 3
   T.30 for fax group 3 with extensions
   Modem

  0100
  0200
  3900
  1f010040
  1b0b
  bf80
       
  0101 0002   

Supplementary services support: 0x03ff
   Hold / Retrieve
   Terminal Portability
   ECT
   3PTY
   Call Forwarding
   Call Deflection
   MCID
   CCBS

But if I call to asterisk even if asterisk isn't start, it's busy,I dont
unsderstand Help me

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Re: [Asterisk-Users] Re: H323

[Ronald_Wiplinger]

Tony Mountifield wrote:

I corrected according to your suggestion:

wget 
http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz
wget 
http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz
   



These are not just patches, they are the complete source at patchlevel 4.
You use these instead of the old versions mentioned previously.
 


and in asterisk-oh323-0.7.1  I get:


# make
for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done
make[1]: Entering directory `/usr/src/asterisk-oh323-0.7.1/wrapper'
./check_ver /usr/src/pwlib pwlib
./check_ver /usr/src/openh323 openh323
ar rc liboh323wrap_s.a wrapper_misc.o asteriskaudio.o wrapconnection.o 
wrapendpoint.o wrapper.o wrapcaps.o wrapgkserver.o

make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.1/wrapper'
make[1]: Entering directory `/usr/src/asterisk-oh323-0.7.1/asterisk-driver'
gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes 
-Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE 
-I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c

chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory
chan_oh323.c: In function `oh323_exception':
chan_oh323.c:1145: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_indicate':
chan_oh323.c:1326: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_digit':
chan_oh323.c:1388: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_text':
chan_oh323.c:1410: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_call':
chan_oh323.c:1434: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_hangup':
chan_oh323.c:1602: error: dereferencing pointer to incomplete type
chan_oh323.c:1710: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_read':
chan_oh323.c:1738: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_write':
chan_oh323.c:2039: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_answer':
chan_oh323.c:2231: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_fixup':
chan_oh323.c:2275: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `ast_oh323_new':
chan_oh323.c:2507: error: dereferencing pointer to incomplete type
chan_oh323.c:2516: error: dereferencing pointer to incomplete type
chan_oh323.c:2518: error: dereferencing pointer to incomplete type
chan_oh323.c:2525: error: dereferencing pointer to incomplete type
chan_oh323.c:2526: error: dereferencing pointer to incomplete type
chan_oh323.c:2527: error: dereferencing pointer to incomplete type
chan_oh323.c:2528: error: dereferencing pointer to incomplete type
chan_oh323.c:2529: error: dereferencing pointer to incomplete type
chan_oh323.c:2530: error: dereferencing pointer to incomplete type
chan_oh323.c:2531: error: dereferencing pointer to incomplete type
chan_oh323.c:2532: error: dereferencing pointer to incomplete type
chan_oh323.c:2533: error: dereferencing pointer to incomplete type
chan_oh323.c:2534: error: dereferencing pointer to incomplete type
chan_oh323.c:2535: error: dereferencing pointer to incomplete type
chan_oh323.c:2536: error: dereferencing pointer to incomplete type
chan_oh323.c:2537: error: dereferencing pointer to incomplete type
chan_oh323.c:2538: error: dereferencing pointer to incomplete type
chan_oh323.c:2539: error: dereferencing pointer to incomplete type
chan_oh323.c:2540: error: dereferencing pointer to incomplete type
chan_oh323.c:2541: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_request':
chan_oh323.c:2713: error: dereferencing pointer to incomplete type
chan_oh323.c:2715: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_atexit':
chan_oh323.c:4895: warning: passing arg 1 of `ast_channel_unregister' 
from incompatible pointer type

chan_oh323.c: In function `load_module':
chan_oh323.c:5192: warning: passing arg 1 of `ast_channel_register' from 
incompatible pointer type
chan_oh323.c:5192: error: too many arguments to function 
`ast_channel_register'

make[1]: *** [chan_oh323.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.1/asterisk-driver'
make: *** [subdirs_build] Error 1


find / -name channel_pvt.h -printdoes not return anything




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[Asterisk-Users] RE: [Serusers] *** SER - Asterisk

[harry gaillac]

Sorry 

it's asterisk-users@lists.digium.com


--- harry gaillac [EMAIL PROTECTED] a écrit :

 Luca,
 
 you may find help here: 
 
 http://www.cs.colostate.edu/~somlo/CSU-SIP-notes/
 
http://www.asteriskdocs.org/
http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large
 
 ask for help to [EMAIL PROTECTED]
 
 Regards
 harry
 
 --- [EMAIL PROTECTED] [EMAIL PROTECTED] a écrit :
 
  i'm not offended, but i speak a bad english, and
  i've not found documents on ser + asterisk on
  www.asterisk.org...
  if you can help me to found it, give me the
 link...
  
  thanks
  luca
  
  
  -- Initial Header ---
  
  From  : harry gaillac [EMAIL PROTECTED]
  To  : [EMAIL PROTECTED] [EMAIL PROTECTED]
  Cc  : asterisk-users@lists.digium.com
  Date  : Mon, 27 Jun 2005 20:50:27 +0200 (CEST)
  Subject : RE: [Serusers] *** SER - Asterisk
  
  
  
  
  
  
  
   I don't want to offend you but you should have a
  look
   to sems .
  
   You won't find docs to help you at asterisk.org
  
   --- [EMAIL PROTECTED] [EMAIL PROTECTED] a
 écrit
  :
  
hello
   
help me to configure ser + asterisk
how to do the configuration?
   
Luca
   

   
   
  
 


6X velocizzare la tua navigazione a 56k? 6X
 Web
Accelerator di Libero!
Scaricalo su INTERNET GRATIS 6X
  http://www.libero.it

   
   
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 http://mail.iptel.org/mailman/listinfo/serusers
   
  
  
  
  
  
  
  
  
 

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   Appel audio GRATUIT partout dans le monde avec
 le
  nouveau Yahoo! Messenger
   Téléchargez cette version sur
  http://fr.messenger.yahoo.com
   
  
  
  
 


  Navighi a 4 MEGA e i primi 3 mesi sono GRATIS. 
  Scegli Libero Adsl Flat senza limiti su
  http://www.libero.it
  
  
  
 
 
 
   
 
   
   

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[Asterisk-Users] pbx_extension_helper: No application 'agi'

[Tom Fielding]

Hi all,

Sorry for this elementary question (I'm a newbie).

I'm trying to write an agi script (test.agi) and run it when I call
in.  However, I'm getting an error that says application agi isn't
being found. I've put test.agi into agi-bin with permissions 755.

Do I have to compile agi support into Asterisk, or is it built in?  My
test.agi script is php, but not using anything fancy (just sending me
an email) so I didn't install PHP AGI.  Do I have to?

Thanks,
Tom

DEBUG:
Connected to Asterisk 1.0.7 currently running on dev1 (pid = 26799)
Verbosity is at least 10
-- Executing Goto(SIP/4.68.250.152-08129478,
validatenumber|s|1) in new stack
-- Goto (validatenumber,s,1)
Jun 28 01:01:02 WARNING[26800]: pbx.c:1291 pbx_extension_helper: No
application 'agi' for extension (validatenumber, s, 1)
  == Spawn extension (validatenumber, s, 1) exited non-zero on
'SIP/4.68.250.152-08129478'

EXTENSIONS.CONF:
[validatenumber]
exten = s,1,agi(test.agi)
exten = s,2,HangUp
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RE: [Asterisk-Users] cheap HFC card on Bristuff vs cheap HFC card oni4l vs Fritz ISDN BRI card on CAPI

[David Masure]

Hi,

Bristuff works great with HFC card...  your compilation problem may come
from your kernel configuration...

You should check this doc, at least, for the redhat config :
http://www.automated.it/guidetoasterisk.htm

Then, installation of Bristuff works like as charm !

Bye

David Masure


-Message d'origine-
De : vdasilva [mailto:[EMAIL PROTECTED]
Envoyé : mardi 28 juin 2005 09:01
À : 'Asterisk Users Mailing List - Non-Commercial Discussion'
Objet : [Asterisk-Users] cheap HFC card on Bristuff vs cheap HFC card
oni4l vs Fritz ISDN BRI card on CAPI


Hello

I have asterisk running in Red Hat 9 with a cheap HFC card on i4l. I
have
choppy sound problems sometimes, and echo problems often. I am using a 2
port Grandstream ATA, Grandstream BT and a Grandstream GPX-2000

I read that changing to BriStuff will fix the echo problems, but have
also
read other users say that the only way they solved the echo/choppy sound
problems was using a Fritz ISDN card with the CAPI drivers...

I have tried using bristuff on RH9 but couldn't get my zaptel to
compile...
 
Then there is the issue of timing, ztdummy or zaprtcand QoS setup on
the
Linux box...

Can anyone who has a 100% working Asterisk implementation using any of
the
techniques described above tell me more...

I will happily upgrade to the Fritz card if it will solve all the
problems...

Thanks
Vicente


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stefan
Gofferje
Sent: 07 April 2005 09:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Choppy sounds after transferring to ISDN
clientor after a time

Hi,

Michiel van Baak schrieb:

I had the exact same thing when using a cheap HFC-S card
connected to my outside ISDN line. Replacing the card with
an AVM Fritz!PCI fixed this issue for me.
I tried a lot with the HFC-S card, different archs, SMP,
uniprocessor, nolapic, noapic, dual channel ram setup, 1
dimm only, removed all cards but the HFC-S, 4 different
versions of bristuffed. Nothing solved it.

What do you mean? The choppy sound or the log message? Trouble is, my 
HFC-S is internal ISDN. I can't use a Fritz!PCI for that because it 
isn't capable of NT mode...

Regards,
Stefan

-- 
 (o_   Stefan Gofferje  | Linux Systems Specialist
 //\   Reg'd Linux User #247167 | Network Security Specialist
 V_/_  Linux is like a Wigwam - No gates, no windows, Apache inside

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[Asterisk-Users] simultaneus calls?

[Erdem HAKİ]

Hello, 



How can i learn my asterisk how many simulyaneus calls support?



My configuration:  80 GB HDD, 1 GB Ram, P4 2,8 MHz processor,
Fedora Core 3 minimum installation, no digium cards, codecs g729 or gsm, 1Mbit
internet connection.



Thanks for your interest...



Erdem HAKI  [EMAIL PROTECTED]






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Re: [Asterisk-Users] Re: H323

[Ronald_Wiplinger]

Ronald_Wiplinger wrote:


Tony Mountifield wrote:

I corrected according to your suggestion:

wget 
http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz 

wget 
http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz 

  



These are not just patches, they are the complete source at 
patchlevel 4.

You use these instead of the old versions mentioned previously.
 


and in asterisk-oh323-0.7.1  I get:


# make
for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done
make[1]: Entering directory `/usr/src/asterisk-oh323-0.7.1/wrapper'
./check_ver /usr/src/pwlib pwlib
./check_ver /usr/src/openh323 openh323
ar rc liboh323wrap_s.a wrapper_misc.o asteriskaudio.o wrapconnection.o 
wrapendpoint.o wrapper.o wrapcaps.o wrapgkserver.o

make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.1/wrapper'
make[1]: Entering directory 
`/usr/src/asterisk-oh323-0.7.1/asterisk-driver'
gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes 
-Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE 
-I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o 
chan_oh323.c

chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory
chan_oh323.c: In function `oh323_exception':
chan_oh323.c:1145: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_indicate':
chan_oh323.c:1326: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_digit':
chan_oh323.c:1388: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_text':
chan_oh323.c:1410: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_call':
chan_oh323.c:1434: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_hangup':
chan_oh323.c:1602: error: dereferencing pointer to incomplete type
chan_oh323.c:1710: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_read':
chan_oh323.c:1738: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_write':
chan_oh323.c:2039: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_answer':
chan_oh323.c:2231: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_fixup':
chan_oh323.c:2275: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `ast_oh323_new':
chan_oh323.c:2507: error: dereferencing pointer to incomplete type
chan_oh323.c:2516: error: dereferencing pointer to incomplete type
chan_oh323.c:2518: error: dereferencing pointer to incomplete type
chan_oh323.c:2525: error: dereferencing pointer to incomplete type
chan_oh323.c:2526: error: dereferencing pointer to incomplete type
chan_oh323.c:2527: error: dereferencing pointer to incomplete type
chan_oh323.c:2528: error: dereferencing pointer to incomplete type
chan_oh323.c:2529: error: dereferencing pointer to incomplete type
chan_oh323.c:2530: error: dereferencing pointer to incomplete type
chan_oh323.c:2531: error: dereferencing pointer to incomplete type
chan_oh323.c:2532: error: dereferencing pointer to incomplete type
chan_oh323.c:2533: error: dereferencing pointer to incomplete type
chan_oh323.c:2534: error: dereferencing pointer to incomplete type
chan_oh323.c:2535: error: dereferencing pointer to incomplete type
chan_oh323.c:2536: error: dereferencing pointer to incomplete type
chan_oh323.c:2537: error: dereferencing pointer to incomplete type
chan_oh323.c:2538: error: dereferencing pointer to incomplete type
chan_oh323.c:2539: error: dereferencing pointer to incomplete type
chan_oh323.c:2540: error: dereferencing pointer to incomplete type
chan_oh323.c:2541: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_request':
chan_oh323.c:2713: error: dereferencing pointer to incomplete type
chan_oh323.c:2715: error: dereferencing pointer to incomplete type
chan_oh323.c: In function `oh323_atexit':
chan_oh323.c:4895: warning: passing arg 1 of `ast_channel_unregister' 
from incompatible pointer type

chan_oh323.c: In function `load_module':
chan_oh323.c:5192: warning: passing arg 1 of `ast_channel_register' 
from incompatible pointer type
chan_oh323.c:5192: error: too many arguments to function 
`ast_channel_register'

make[1]: *** [chan_oh323.o] Error 1
make[1]: Leaving directory 
`/usr/src/asterisk-oh323-0.7.1/asterisk-driver'

make: *** [subdirs_build] Error 1


find / -name channel_pvt.h -printdoes not return anything



I found a hint to steal it from the current version, but even that 
gives me an error:


make
for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done
make[1]: Entering directory `/usr/src/asterisk-oh323-0.7.1/wrapper'
./check_ver /usr/src/pwlib pwlib
./check_ver /usr/src/openh323 openh323
ar rc liboh323wrap_s.a wrapper_misc.o asteriskaudio.o wrapconnection.o 
wrapendpoint.o wrapper.o wrapcaps.o wrapgkserver.o

make[1]: Leaving directory 

Re: [Asterisk-Users] Re: H323

[Damian Minkov]

try using asterisk-oh323-0.7.2-pre1 it compiles with latest CVS
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[Asterisk-Users] GSM/PSTN Gateway function of DIAX - feedback request

[Dan]

Hi all,

If they are people on this list using this feature, I kindly ask them 
to send me their feedback.
I am interested to support multiple GSM phones and standard voice 
modems.


More, if it is someone with much more experience regardinig direct 
using of bluetooth audio in an

application please send me a mail directly.

Thank you in advance for your support in DIAX development.

Best regards,
Dan


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[Asterisk-Users] Spinlock with ZAPTEL

[Lee Archer]

Title: Spinlock with ZAPTEL






Hi, I'm using Fedora Core 3 and the 1.0.8 version of ZAPTEL. Why do I get spinlocks when I modprobe -r it but the HEAD version unloads fine?

Regards


Lee


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Checked by AVG Anti-Virus.
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Re: [Asterisk-Users] Shoutcast Music On Hold problems?

[Patrick]

On Mon, 2005-06-27 at 22:51 -0700, hank wrote:
  mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/

I haven't tried this myself but if I put www.waixwave.com:8000 in
Firefox I get connection refused. Try another site that actually
streams music. Shoutcast.org should have a nice list.

Regards,
Patrick

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Re: [Asterisk-Users] AVM CAPI INSTALLATION

[Armin Schindler]

On Tue, 28 Jun 2005, sylvain garcia wrote:
 I use AVM fritz card A1, I have install with instruction of
 Voip-info.or, but it don' work.

Which version of Asterisk and chan_capi do you use?
How does your capi.conf look like?

Armin
 
 capiinfo
 
 Number of Controllers : 1
 Controller 1:
 Manufacturer: AVM GmbH
 CAPI Version: 2.0
 Manufacturer Version: 3.101-02  (49.18)
 Serial Number: 101
 BChannels: 2
 Global Options: 0x0039
internal controller supported
DTMF supported
Supplementary Services supported
channel allocation supported (leased lines)
 B1 protocols support: 0x411f
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
V.110 asynconous operation with start/stop byte framing
V.110 synconous operation with HDLC framing
T.30 modem for fax group 3
Modem asyncronous operation with start/stop byte framing
 B2 protocols support: 0x0b1b
ISO 7776 (X.75 SLP)
Transparent
LAPD with Q.921 for D channel X.25 (SAPI 16)
T.30 for fax group 3
ISO 7776 (X.75 SLP) with V.42bis compression
V.120 asyncronous mode
V.120 bit-transparent mode
 B3 protocols support: 0x80bf
Transparent
T.90NL, T.70NL, T.90
ISO 8208 (X.25 DTE-DTE)
X.25 DCE
T.30 for fax group 3
T.30 for fax group 3 with extensions
Modem
 
   0100
   0200
   3900
   1f010040
   1b0b
   bf80
        
   0101 0002   
 
 Supplementary services support: 0x03ff
Hold / Retrieve
Terminal Portability
ECT
3PTY
Call Forwarding
Call Deflection
MCID
CCBS
 
 But if I call to asterisk even if asterisk isn't start, it's busy,I dont
 unsderstand Help me
 
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[Asterisk-Users] E100P configuration

[Musaluke AK]

Hello

I have been buttling to get a Valiant VCL-30 E1 channel bank multiplexer 
talking to asterik's E100P card.

The Spec for the VCL says this:

Coding: hdb3
Signalling: CAS
Timing: can be internal, external sync
Nominal Impedance: 120ohm balanced, 75ohm unbalanced
supports euroisdn

In the zaptel.conf file, I have:
span=1,1,0,cas,hdb3,crc4

The LED on the VCL indicates that I have a Remote Frame Alarm
The E100P also has an alarm on it.

Could anyone point me to something else I may need to configure to get this 
device working with *

Thank you

Anthony
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[Asterisk-Users] HowTo start DIAL by a sillent training as W for modems

[f6hqz-m]

Hi all the list,

I am searching how to insert few seconds of silence just before to send the
DTMF sequence via a FXO WildCard X101P to PSTN.

I remember that Hayes compatible modems knows a special character W that
do a 1 sec pause.
Is it possible to do something like this in DIAL line sequence ?

TIA

Best Regards,
Francois BERGERET,
France.

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Re: [Asterisk-Users] HowTo start DIAL by a sillent training as W for modems

[Doug Lytle]

[EMAIL PROTECTED] wrote:


Hi all the list,

I am searching how to insert few seconds of silence just before to send the
DTMF sequence via a FXO WildCard X101P to PSTN.

I remember that Hayes compatible modems knows a special character W that
do a 1 sec pause.
Is it possible to do something like this in DIAL line sequence ?

 


This should help:

http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial


Doug

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[Asterisk-Users] meetme problem

[Felix Skwarczynski]

Hello,

I seem to have a strange problem, which appeared out of nowhere.
Did anyone see something like this?

2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Illegal seek

2005-06-03 18:44:50 DEBUG[10721] res_agi.c: Zap/65-1 hungup
2005-06-03 18:44:50 DEBUG[10721] channel.c: Avoiding deadlock for 
'Zap/63-1'
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success


Once these messages start showing, I must stop my asterisk (stop now) 
because the load goes sky high.

I'm using an Asterisk CVS-HEAD.

Looking forward for your answer.


Felix

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[Asterisk-Users] meetme problem

[Felix Skwarczynski]

Hello,

I seem to have a strange problem, which appeared out of nowhere.
Did anyone see something like this?

2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Illegal seek
2005-06-03 18:44:50 DEBUG[10721] res_agi.c: Zap/65-1 hungup
2005-06-03 18:44:50 DEBUG[10721] channel.c: Avoiding deadlock for
'Zap/63-1'
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Success

Once these messages start showing, I must stop my asterisk (stop now)
because the load goes sky high.
I'm using an Asterisk CVS-HEAD.

Looking forward for your answer.


Felix


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Re: [Asterisk-Users] AVM CAPI INSTALLATION

[sylvain garcia]

Armin Schindler a crit:

  On Tue, 28 Jun 2005, sylvain garcia wrote:
  
  
I use AVM fritz card A1, I have install with instruction of
Voip-info.or, but it don' work.

  
  
Which version of Asterisk and chan_capi do you use?
How does your capi.conf look like?

Armin
 
  
  
capiinfo

Number of Controllers : 1
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.101-02  (49.18)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
   internal controller supported
   DTMF supported
   Supplementary Services supported
   channel allocation supported (leased lines)
B1 protocols support: 0x411f
   64 kbit/s with HDLC framing
   64 kbit/s bit-transparent operation
   V.110 asynconous operation with start/stop byte framing
   V.110 synconous operation with HDLC framing
   T.30 modem for fax group 3
   Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x0b1b
   ISO 7776 (X.75 SLP)
   Transparent
   LAPD with Q.921 for D channel X.25 (SAPI 16)
   T.30 for fax group 3
   ISO 7776 (X.75 SLP) with V.42bis compression
   V.120 asyncronous mode
   V.120 bit-transparent mode
B3 protocols support: 0x80bf
   Transparent
   T.90NL, T.70NL, T.90
   ISO 8208 (X.25 DTE-DTE)
   X.25 DCE
   T.30 for fax group 3
   T.30 for fax group 3 with extensions
   Modem

  0100
  0200
  3900
  1f010040
  1b0b
  bf80
       
  0101 0002   

Supplementary services support: 0x03ff
   Hold / Retrieve
   Terminal Portability
   ECT
   3PTY
   Call Forwarding
   Call Deflection
   MCID
   CCBS

But if I call to asterisk even if asterisk isn't start, it's busy,I dont
unsderstand Help me

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I use Asterisk 1.0.7 on Dedian Sarge with chan_capi 0.3.5.11

I don't configuure my capi.conf it is like orig;


; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]

; msn=50
; incomingmsn=*
;controller=1
softdtmf=1
accountcode=
context=incomingtest
;echosquelch=1
;echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
;devices=2


;PointToPoint (55512-0)
;for outgoing calls use example 5551212
;and in dialplan you can use callerid like
;exten = _0XXX.,1,StripMSD,1
;exten = _XXX.,2,Dial,CAPI/55512${CALLERIDNUM}:bBYEXTENSION
;
;mode=immediate
;isdnmode=ptp
inal:


my configuration of my dial plan of my internal LAN is correct and
works fine, so I would like test my connection capi in order to receive
incomingcall first and outgoing call in second.

What is the modification to extensions.conf and capi.conf.

Thanks you for your help



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Re: [Asterisk-Users] AVM CAPI INSTALLATION

[Armin Schindler]

On Tue, 28 Jun 2005, sylvain garcia wrote:
 Armin Schindler a écrit :
 
 On Tue, 28 Jun 2005, sylvain garcia wrote:
   
 
 I use AVM fritz card A1, I have install with instruction of
 Voip-info.or, but it don' work.
 
 
 
 Which version of Asterisk and chan_capi do you use?
 How does your capi.conf look like?
 
 I use Asterisk 1.0.7 on Dedian Sarge with chan_capi  0.3.5.11
 
 I don't configuure my capi.conf it is like orig;
 
 
 ; CAPI config
 ;
 ;
 [general]
 nationalprefix=0
 internationalprefix=00
 rxgain=0.8
 txgain=0.8
 
 [interfaces]
 
 ; msn=50
 ; incomingmsn=*
 ;controller=1
 softdtmf=1
 accountcode=
 context=incomingtest
 ;echosquelch=1
 ;echocancel=yes
 ;echotail=64
 ;callgroup=1
 ;deflect=12345678
 ;devices=2

This is just a template. You cannot expect this to work without changing the 
settings to your needs and enviroment.
 
 my configuration of my dial plan of my internal LAN is correct and works
 fine, so I would like test my connection capi in order to receive
 incomingcall first and outgoing call in  second.
 
 What is the modification to extensions.conf and capi.conf.

That depends on what you want to do and which numbers you use.

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[Asterisk-Users] meetme problem

[Felix Skwarczynski]

Hello,

I seem to have a strange problem, which appeared out of nowhere.
Did anyone see something like this?

2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Illegal seek

2005-06-03 18:44:50 DEBUG[10721] res_agi.c: Zap/65-1 hungup
2005-06-03 18:44:50 DEBUG[10721] channel.c: Avoiding deadlock for 
'Zap/63-1'
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success


Once these messages start showing, I must stop my asterisk (stop now) 
because the load goes sky high.

I'm using an Asterisk CVS-HEAD.

Looking forward for your answer.


Felix

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Re: [Asterisk-Users] AVM CAPI INSTALLATION

[sylvain garcia]

Armin Schindler a crit:

  On Tue, 28 Jun 2005, sylvain garcia wrote:
  
  
Armin Schindler a crit :



  On Tue, 28 Jun 2005, sylvain garcia wrote:
 

  
  
I use AVM fritz card A1, I have install with instruction of
Voip-info.or, but it don' work.
   


  
  Which version of Asterisk and chan_capi do you use?
How does your capi.conf look like?
  

I use Asterisk 1.0.7 on Dedian Sarge with chan_capi  0.3.5.11

I don't configuure my capi.conf it is like orig;


; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]

; msn=50
; incomingmsn=*
;controller=1
softdtmf=1
accountcode=
context=incomingtest
;echosquelch=1
;echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
;devices=2

  
  
This is just a template. You cannot expect this to work without changing the 
settings to your needs and enviroment.
 
  
  
my configuration of my dial plan of my internal LAN is correct and works
fine, so I would like test my connection capi in order to receive
incomingcall first and outgoing call in  second.

What is the modification to extensions.conf and capi.conf.

  
  
That depends on what you want to do and which numbers you use.

Armin

ok, thanks,
I would like receive call to sip phone.

In my extension.conf i have:

[localnetwork]
exten = 123,1,Dial(SIP/555,30,Ttr)


I have one number for the external for my AVM CARD: 0572086964

What is the modification for my capi.conf and extension.conf in order
to incoming call of 0572086964 are routed to sip channel 555.
And for outgoinf call please.

Thanks you very much






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Re: [Asterisk-Users] AGI say number but in french

[Arvanitis Kostas]

On Monday 27 June 2005 23:04, David John Walsh wrote:
 Hello,

 does anyone know how to get the say number (say.c) agi application
 to work in french [assuming that I have the French voice files]
 I have looked in the code and about a 1/3 of the way thru there is :
 } else if (!strcasecmp(language, fr) ) {  /* French
 syntax */

 and then further on there is logic for french numbers.

 does anyone know the syntax as looking on the code / google / wiki
 gives me no ideas.

Have you tried setting the channel language to fr (language=fr in the 
device configuration, or SetLanguage(fr) in the extensions.conf file)?

This seems to be all that is needed.
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Re: [Asterisk-Users] SixTel?

[Jason p]

my number with them is ok.
Jason

On 6/28/05, Erik Espinoza [EMAIL PROTECTED] wrote:
 Nope, sixTel has been acting up big time recently. I've put in
 requests for support online. Calling has gone to a message that says
 go to the web site.
 
 This really blows, the prices were good but unless they shape up I'm
 switching providers.
 
 Thanks,
 Erik
 
 On 6/27/05, JD Austin [EMAIL PROTECTED] wrote:
  I was just checking out the dids for all of my fail over providers and
  noticed that neither DID that I have with SixTel work.
  Both pause for a long long time
  The local number gives a recording: 'The number you have dialed is not
  in service or is assigned in a different area code.  Please check your
  number and dial again'.
  The 800 number just rings busy.
  Anyone else having this issue or am I a lone data point?
 
  JD
 
  --
  JD Austin
  Twin Geckos Technology Services LLC
  email: [EMAIL PROTECTED]
  http://www.twingeckos.com
  phone/fax: 480.288.8195
 
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RE: [Asterisk-Users] TDM card and voicemail volume

[Adam Robins]

I was able to raise the volume from inaudible to acceptable by
increasing the RxGain in zapata.conf by 5db.  I'd rather not go the
uncomressed wav route, as it will chew up storage in my email system. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Monday, June 27, 2005 11:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TDM card and voicemail volume


 I saw some conversation about this in the archives, but nothing 
 definitive.
 
 If a call comes in over a CO line via the TDM400P, the Comedian Mail 
 recording volume is so low it's inaudible.  Calls coming in via SIP or

 IAX do not have this problem.
 
 Does anyone have any information on this issue?

Its still a problem. It seems the greater the distance from the Central
Office, the greater the problem (due to the cable loss to the Central
Office plus the problem with the TDM card). 

Part of the problem is there are very few people that understand zaptel,
wctdm, drivers, hardware (chipsets) and transmission engineering.
Actually, there is only one person and he is now very busy doing other
things that are apparently more important.

As someone else mentioned, changing to wav format improves the levels a
little bit, but its certainly not a fix. There are no known work
arounds.


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This transmission is sent in trust, for the sole purpose of delivery to the 
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Re: [Asterisk-Users] AVM CAPI INSTALLATION

[Elmar Haneke]

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8


General config might be ok - as long as national and international 
prefix are 0 and 00




[interfaces]

; msn=50
; incomingmsn=*
;controller=1
softdtmf=1
accountcode=
context=incomingtest
;echosquelch=1
;echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
;devices=2


This should be uncommented (Remove ;) except the echo-lines and deflect.

Elmar




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Re: [Asterisk-Users] AGI say number but in french

[David John Walsh]

it was the second one i needed - thank you.  I only needed the numbers
in french on one   b number and make it that the number could be
dialled from any extension (which is why option a was unsuitable.

thanks again.

On 28/06/05, Arvanitis Kostas [EMAIL PROTECTED] wrote:
 On Monday 27 June 2005 23:04, David John Walsh wrote:
  Hello,
 
  does anyone know how to get the say number (say.c) agi application
  to work in french [assuming that I have the French voice files]
  I have looked in the code and about a 1/3 of the way thru there is :
  } else if (!strcasecmp(language, fr) ) {  /* French
  syntax */
 
  and then further on there is logic for french numbers.
 
  does anyone know the syntax as looking on the code / google / wiki
  gives me no ideas.
 
 Have you tried setting the channel language to fr (language=fr in the
 device configuration, or SetLanguage(fr) in the extensions.conf file)?
 
 This seems to be all that is needed.
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[Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!

[harry gaillac]

To people who have replied to me about Fwd: JE TROUVE
QUE VOUS N'ETES PAS HONETE!,

Have a look at 
http://lists.digium.com/pipermail/asterisk-users/2005-June/114066.html

Harry






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Téléchargez cette version sur http://fr.messenger.yahoo.com
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Re: [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!

[Peter Bowyer]

What's your point?

On 28/06/05, harry gaillac [EMAIL PROTECTED] wrote:
 To people who have replied to me about Fwd: JE TROUVE
 QUE VOUS N'ETES PAS HONETE!,
 
 Have a look at
 http://lists.digium.com/pipermail/asterisk-users/2005-June/114066.html
 
 Harry
 
 
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-- 
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] OT: Good soft-phone on Linux

[Zoltan Szecsei]

Hi Hamish,
Sorry about being a day late... man, these lists are hell to keep up 
with.


You could also try xlite - www.xten.com

Cheers,
Zoltan


Hamish Whittal wrote:


Hi Folks,

I am wanting advise on a good soft-phone on Linux. I have looked at
Gnophone but cannot seem to get it to compile under debian sarge. I am
now looing at sipXphone seem to be picking up that it is not that
stable, but perhaps someone here can advise on what softphone I can use
on Linux.

Thanks in advance,

Hamish
---
| Hamish Whittal| Mobile:   +27 82 803 5533 |
| QED Technologies cc   | landline: +27 21 671 7710 |
| 21 Marne Avenue, Claremont, Cape Town | fax:  +27 21 674 9184 |
|fortune cookie below autogenerated_|
You are confused; but this is your normal state.

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[Asterisk-Users] Using asterisk as Quality Monitoring Platform?

[Aaron Sundman]

Is anyone using Asterisk as a Quality Monitoring Platform for random
recording of inbound calls that come into another ACD?

We use Aspect ACDs for inbound call routing and do all live Quality
Monitoring at the moment.  I have looked at many Quality Recording
systems that run from $30K to $700K.  They all have great
strengths/weaknesses.

The connection to the Aspect ACD could be done via POTS ports or SIP
ports.  The Aspect CTI system could be queried via CRON for choosing
which calls to monitor and when.  Once the calls are recorded (the
tough part) I can set up a PHP based intranet site for hosting,
delivering, and scoring to the supervisors.

Is anyone considering/doing anything similar?  If this discussion has
already taken place please send a URL.  Google searches turn up lots
of good, though non-related, info.

-- 
-Aaron
[EMAIL PROTECTED]
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Re: [Asterisk-Users] is teliax down?

[Mark Musone]

yea, i've been getting kinda crappy calls for the past day too

On 6/28/05, Rick Baranowski [EMAIL PROTECTED] wrote:
 Looks like the router around them in the Denver area have been having issues
 or overloaded. I have been getting choppy calls all day but seems to be
 better tonight.
 
 Rick
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Adam Megacz
 Sent: Monday, June 27, 2005 6:31 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] is teliax down?
 
 
 I'm getting really wierd errors from them, like bad packet checksums:
 
   Jun 27 18:22:56 NOTICE[29051]: rtp.c:435 ast_rtp_read: RTP: Received
 packet with bad UDP checksum
 
   - a
 
 --
 I didn't see it then, but it turned out that getting fired was the
  best thing that could have ever happened to me. The heaviness of
  being successful was replaced by the lightness of being a beginner
  again, less sure about everything. It freed me to enter one of the
  most creative periods of my life.
 
   -- Steve Jobs, commencement speech at Stanford, June 2005
 
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[Asterisk-Users] Using Conferencing and Meetme

[Jean-Marc Salsa]

Hi,

I ve installed recently AAH 1.1
And I was wondering on how to use this conferencing feature ?
I have created extension 200.
and when I try to call 8200, it says that this is not a valid
conference number.
Is there something specific to do ?

Also, when entering MeetMe console,
I cannot see anything. Is that allright ?
meaning that if I have not started any conferencing, then, I shall see
nothing in MeetMe :o)

Thanks for any help !
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Re: [Asterisk-Users] pbx_extension_helper: No application 'agi'

[John Millican]

On Tuesday June 28 2005 4:56 am, Tom Fielding wrote:
 Hi all,

 Sorry for this elementary question (I'm a newbie).

 I'm trying to write an agi script (test.agi) and run it when I call
 in.  However, I'm getting an error that says application agi isn't
 being found. I've put test.agi into agi-bin with permissions 755.

 Do I have to compile agi support into Asterisk, or is it built in?  My
 test.agi script is php, but not using anything fancy (just sending me
 an email) so I didn't install PHP AGI.  Do I have to?

 Thanks,
 Tom

 DEBUG:
 Connected to Asterisk 1.0.7 currently running on dev1 (pid = 26799)
 Verbosity is at least 10
 -- Executing Goto(SIP/4.68.250.152-08129478,
 validatenumber|s|1) in new stack
 -- Goto (validatenumber,s,1)
 Jun 28 01:01:02 WARNING[26800]: pbx.c:1291 pbx_extension_helper: No
 application 'agi' for extension (validatenumber, s, 1)
   == Spawn extension (validatenumber, s, 1) exited non-zero on
 'SIP/4.68.250.152-08129478'

 EXTENSIONS.CONF:
 [validatenumber]
 exten = s,1,agi(test.agi)
 exten = s,2,HangUp

exten = s,1,agi(test.agi) 
should be
exten = s,1,agi,test.agi
If there any arguments to send the script use
exten = s,1,agi,test.agi|args_to_pass

John M
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RE: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread

[Michael Di Martino]

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Monday, June 27, 2005 5:26 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread

On Monday 27 June 2005 15:46, steve szmidt wrote:
 One could probably argue effectively for an Asterisk-Basic list. Or an

 Asterisk-Advanced user list. Something that makes it easier to get 
 started without being overwhelmed by 10,000-15,000 users posts. A 
 place that frequently posted links to the beginner pages on the wiki.

We've effectively argued it to death many many times over the course of
the last few years.  Check the archives -- it's been thought up and
re-thought up and dismissed each and every time.

Basic issue: nobody will want to sit on the newbie list because they'll
end up answering all the same questions over and over since nobody
really seems to want to read for themselves.

It's the same argument that comes around for forums, except that last
time I think we actually witnessed a man lose his mind on the mailing
list.  That was entertaining.  :-)


~

Their would not be so many newbie questions if their was 1. A fully
indexed searchable archive list and 2. Good solid documentation.





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RE: [Asterisk-Users] Livevoip 800 Choppy Audio

[Wiley Siler]

Rich,

You must be a LiveVoip crony.  Regardless, there is nothing wrong with
my post.  Expressing that we should all learn from their mistakes is
hardly an objective.  So whatever your issue, I think you can take your
own advise.

Thanks,
Wiley






-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Monday, June 27, 2005 7:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Livevoip 800 Choppy Audio

Wiley, its about time to grow up and stop this personal BS. We all
certainly
understand your objective.


  From: Wiley Siler [EMAIL PROTECTED]
  Subject: RE: [Asterisk-Users] Livevoip 800 Choppy Audio
  Date: Mon, 27 Jun 2005 08:30:41 -0700 
  To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com


 LiveVoip has been a learning experience for anyone who purchased from
 them.  With any luck, it was a learning experience in what not to do
 for anyone out there that provides similar services.  At least I hope
so
 since it seems obvious that LiveVoip never learned a thing during
their
 interaction with the community.  That is a real shame too considering
 how people will embrace a company.
 
 I like someone else's post survival of the fittest.
 Your damn skippy...
 
 W
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of John
 Kington
 Sent: Sunday, June 26, 2005 7:54 AM
 To: asterisk-users@lists.digium.com
 Subject: RE: [Asterisk-Users] Livevoip 800 Choppy Audio
 
 Wiley, thanks for pointing me to NuFone for tollfree DID. I was
planning
 to 
 report
 on results between LiveVOIP and NuFone. The apparent bankruptsy of
 LiveVOIP
 means that my choppy audio will probably never be resolved. I set up
 both 
 DID to
 go through DISA and I could then use the echo test application.
 Everytime I 
 tested
 LiveVOIP, the audio was choppy. I have not experienced any choppiness
 with
 NuFone but the echo seemed to take longer to get back to me compared
to
 LiveVOIP.
 I now get a message that my call can not be completed when I call the
 LiveVOIP
 DID and I see that I can not register my asterisk to them. I am glad I
 did 
 not have
 big dollars invested in them.
 Regards,
 John
 
 
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---End of Original Message-


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[Asterisk-Users] Asterisk SpanDSP - problems by sending a fax

[Dominik Simon]

Hi group!

Today I tested spandsp with asterisk and started to send a fax, but it dont
work :( Asterisk call and spandsp start sending - here is my console-log:

DIS with final frame tag
In state 10 
Start tx document 
CFR with final frame tag
In state 4 
Start tx page 0 
Start tx page 1 
RTN with final frame tag
In state 14 
free(): invalid pointer 0x80f3c80!
Segmentation fault 
Ouch ... error while writing audio data: : Broken pipe
Warning, flexibel rate not heavily tested!

...and then asterisk stopps :( I tested it on two systems: fedora core 1 and
suse 9.1 - there are no errors by compiling and installing. The problem is
also by installing with the org. suse-rpms of asterisk 1.0.6 .

Is there any prob with a audiolib?

Best regards and thanks for help!
Dominik


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Re: [Asterisk-Users] Using Conferencing and Meetme

[Vamsi Pottangi]

Never uses AAH, but check for two things
if 8200 is mentioned meetme.conf 
and if you have ztdummy initialised ...

~Vamsi

On 6/28/05, Jean-Marc Salsa [EMAIL PROTECTED] wrote:
 Hi,
 
 I ve installed recently AAH 1.1
 And I was wondering on how to use this conferencing feature ?
 I have created extension 200.
 and when I try to call 8200, it says that this is not a valid
 conference number.
 Is there something specific to do ?
 
 Also, when entering MeetMe console,
 I cannot see anything. Is that allright ?
 meaning that if I have not started any conferencing, then, I shall see
 nothing in MeetMe :o)
 
 Thanks for any help !
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RE: [Asterisk-Users] HOW TO WRITE ROUTE PATTERNS DIALPLAN

[Jay Milk]

Try _1NXXNXX -- don't forget the underscore.

 -Original Message-
 From: wassim darwish [mailto:[EMAIL PROTECTED] 
 Sent: Tuesday, June 28, 2005 2:59 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] HOW TO WRITE ROUTE PATTERNS DIALPLAN
 
 
 in routes pattern i tried to write pattern to usa
 destination and that was 1* it worked well but when i
 wanted to specify the number of digits then i tried
 1NXXNXX but i didnt work.so i dont know what to
 write please help.
 
 __
 Do You Yahoo!?
 Tired of spam?  Yahoo! Mail has the best spam protection around 
 http://mail.yahoo.com 
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Re: [Asterisk-Users] Using Conferencing and Meetme

[Vamsi Pottangi]

Sorry, I meant I had never used AAH.


On 6/28/05, Vamsi Pottangi [EMAIL PROTECTED] wrote:
 Never uses AAH, but check for two things
 if 8200 is mentioned meetme.conf
 and if you have ztdummy initialised ...
 
 ~Vamsi
 
 On 6/28/05, Jean-Marc Salsa [EMAIL PROTECTED] wrote:
  Hi,
 
  I ve installed recently AAH 1.1
  And I was wondering on how to use this conferencing feature ?
  I have created extension 200.
  and when I try to call 8200, it says that this is not a valid
  conference number.
  Is there something specific to do ?
 
  Also, when entering MeetMe console,
  I cannot see anything. Is that allright ?
  meaning that if I have not started any conferencing, then, I shall see
  nothing in MeetMe :o)
 
  Thanks for any help !
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[Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]

[John Goerzen]

On 2005-06-28, Rich Adamson [EMAIL PROTECTED] wrote:
 It's probably a $2 decision. Just pick one or two and try them.

 There are a fair number of people on this list (including myself) that 
 stay current with multiple itsp's. Every itsp is going to have a problem
 now and then, so keeping a couple around isn't a bad approach even for
 a home or soho system.

Sound advice.  I have my system configured to just fallback to standard
telco LD if my VOIP provider is unreachable for whatever reason.  Neat
feature of Asterisk even lets me play an alert tone so I know this call
will cost me the princely sum of 6 cents per minute instead of 1.1 :-)


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[Asterisk-Users] help, switch off NOTICE in console

[Anatoly Pugachev]

Hello!
Started to use asterisk.
i'm connecting to it with 'asterisk -r -q' command and everytime  people are
using it, i see following in my asterisk:

Jun 28 18:07:25 NOTICE[20564]: rtp.c:298 process_rfc3389: RFC3389 support 
incomplete.  Turn off on client if possible
Jun 28 18:11:12 NOTICE[20956]: rtp.c:430 ast_rtp_read: RTP: Received packet 
with bad UDP checksum

i tryed 'verbose 0' in CLI, but it doesn't help.
Can someone help me to disable this messages ?
Thanks.

-- 
Anatoly P. Pugachev


pgpsn8D77XCFm.pgp
Description: PGP signature
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RE: [Asterisk-Users] pbx_extension_helper: No application 'agi'

[Jay Milk]

That's incorrect, both versions represent valid syntax.

To the OP -- show us the contents of test.agi

 -Original Message-
 From: John Millican [mailto:[EMAIL PROTECTED] 
 Sent: Tuesday, June 28, 2005 8:23 AM
 To: Tom Fielding; Asterisk Users Mailing List - 
 Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] pbx_extension_helper: No 
 application 'agi'
 
 
 exten = s,1,agi(test.agi) 
 should be
 exten = s,1,agi,test.agi
 If there any arguments to send the script use
 exten = s,1,agi,test.agi|args_to_pass
 
 John M
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[Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]

[John Goerzen]

On 2005-06-28, r00t [EMAIL PROTECTED] wrote:
 I'll second voipjet for outbound only. While many reported problems to

VoipJet bothers me for two reasons.  First, their terms of service are
absolutely insane.  Users are specifically forbidden to place calls
regarding medical or financial matters over VoipJet.  So I couldn't call
my tax preparer or schedule a doctor appointment under their contract.
There are many other insane things about it; see the thread at
http://lists.digium.com/pipermail/asterisk-users/2005-March/094251.html.
One of the little gems is that if you tell anyone you use VoipJet, you
violate your contract.  So all of you that have been praising your
VoipJet service here: prepare to be disconnected! :-)

Secondly, they are not honest about what they are doing.  They clearly
are aiming some services at small self-sufficient end-users, yet they
claim to provide services to commercial carriers only (and their ToS
tries to enforce that.)  Got to love little statements like Emerging
VoIP service providers can make payments through PayPal.

-- John


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RE: [Asterisk-Users] Using Conferencing and Meetme

[Dean Collins]

  Also, when entering MeetMe console,
  I cannot see anything. Is that allright ?
  meaning that if I have not started any conferencing, then, I shall
see
  nothing in MeetMe :o)
You need to type the extension number in the box to see the conference
details

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Re: [Asterisk-Users] Using Conferencing and Meetme

[Jean-Marc Salsa]

I did that ... 8200 ... still ... nothing ... :o)

On 6/28/05, Dean Collins [EMAIL PROTECTED] wrote:
   Also, when entering MeetMe console,
   I cannot see anything. Is that allright ?
   meaning that if I have not started any conferencing, then, I shall
 see
   nothing in MeetMe :o)
 You need to type the extension number in the box to see the conference
 details
 

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RE: [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]

[Wiley Siler]

So far my experience with TOS has been that most of them are pretty odd.

No one wants the liability of a stock trade gone foul or a call to the
doctor that gets disconnected.  Essentially, those things in the TOS are
just a CYA.  They are un-enforced but should someone decide to attempt
to sue based upon a financial loss, the ITSP is covered.  

So, yep.  That is weird but not unexpected.  Heaven knows there have
been worse out there.  I can tell you that in my experience, VoiPJet
(Oh, no I named them! LOL) has been the best dial tone I have gotten for
my LD via VOIP.  NuFone is a close second.

YMMV and you have to be willing to overlook the quirky TOS.

Cheers,
Wiley






-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Goerzen
Sent: Tuesday, June 28, 2005 6:46 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]

On 2005-06-28, r00t [EMAIL PROTECTED] wrote:
 I'll second voipjet for outbound only. While many reported problems to

VoipJet bothers me for two reasons.  First, their terms of service are
absolutely insane.  Users are specifically forbidden to place calls
regarding medical or financial matters over VoipJet.  So I couldn't call
my tax preparer or schedule a doctor appointment under their contract.
There are many other insane things about it; see the thread at
http://lists.digium.com/pipermail/asterisk-users/2005-March/094251.html.
One of the little gems is that if you tell anyone you use VoipJet, you
violate your contract.  So all of you that have been praising your
VoipJet service here: prepare to be disconnected! :-)

Secondly, they are not honest about what they are doing.  They clearly
are aiming some services at small self-sufficient end-users, yet they
claim to provide services to commercial carriers only (and their ToS
tries to enforce that.)  Got to love little statements like Emerging
VoIP service providers can make payments through PayPal.

-- John


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RE: [Asterisk-Users] simultaneus calls?

[Damon Estep]

The 1 m internet connection will be the
limiting factor in your setup, you did not state what type of internet
connection, but given the speed of 1 mbit it must be DSL (or maybe fraction
t/e1).



Is the outbound speed also 1m? Is there
data on the line also? How much? What about voice Qos?



You should start here http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption















From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Erdem HAKI
Sent: Tuesday, June 28, 2005 3:04
AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
simultaneus calls?





Hello, 



How can i learn my asterisk how many simulyaneus calls
support?



My configuration: 80 GB HDD, 1 GB Ram, P4 2,8 MHz
processor, Fedora Core 3 minimum installation, no digium cards, codecs g729 or
gsm, 1Mbit internet connection.



Thanks for your interest...



Erdem HAKI  [EMAIL PROTECTED]








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[Asterisk-Users] Re: Using asterisk as Quality Monitoring Platform?

[Jason Kawakami]

 Is anyone using Asterisk as a Quality Monitoring Platform for random
 recording of inbound calls that come into another ACD?

we have several clients that  we are doing something similar with.  we are
doing full call logging  by tapping the telco lines.

snip
 The connection to the Aspect ACD could be done via POTS ports or SIP
 ports.  The Aspect CTI system could be queried via CRON for choosing
 which calls to monitor and when.  Once the calls are recorded (the
 tough part) I can set up a PHP based intranet site for hosting,
 delivering, and scoring to the supervisors.

on the contrary, the call recording is probably the easy part.  sounds like
you have the tough stuff sorted.

Jason Kawakami
www.optellabs.com
Salt Lake City, UT


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[Asterisk-Users] ClueCon, Vote?

[Brian West]

Ok I have to get a vote of all the people that are going to come to  
Cluecon so we order the beer keg's for the developers board room.


Anyone have any preference? (if you haven't registered for ClueCon  
now is the time to register!)


Choices... choices... choices... I want Red Bull on tap!

/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

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Re: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread

[Andrew Kohlsmith]

On Tuesday 28 June 2005 09:44, Michael Di Martino wrote:
 Their would not be so many newbie questions if their was 1. A fully
 indexed searchable archive list and 2. Good solid documentation.

Alright, you've called my bluff.

http://www.mail-archive.com/asterisk-users%40lists.digium.com/

Is that not fully indexed and searchable enough for you?

http://lists.digium.org is fully indexed, but is not searchable.

Googling for site:lists.digium.com search terms here is fully searchable but 
not indexed.

So now, you have indexed and searchable, indexed, and searchable.  Can we 
please stop complaining about this now?

As for point #2 -- I think everyone is in agreement but it is no small feat 
and there are several people working on it.  Feel free to contribute.

-A.
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[Asterisk-Users] How do you handle NAT?

[Matthew Boehm]

We are interested in how other people are handling NAT problems. We have 
several customers all of which have some sort of firewall/NAT device at 
their location. For simplicity sake, all customers' internal networks 
are 192.168.*.*.


Our asterisk box is on public IP not blocked by any FW/NAT.

I use QUALIFY=yes on all our customers' phones and I feel that sending 
out 80-something keep-alive packets is causing our box to crawl and 
cause bad calls.


Would SER be better in this case? Should I have phones register with SER 
instead of with Asterisk?


Thanks,
Matthew

P.S. Yes, I have read stuff on NAT on the wiki. I'm more interested in 
other real world, working, solutions.


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RE: [Asterisk-Users] simultaneus calls?

[Erdem HAKİ]

Yes it is DSL and outbound speed is aslo
1Mbit, its a dedicated server and we just use to talk. I look at the web
site which you suggested, but i want to learn how many calls supported practically?
Any information do you have?



Thanks



Erdem HAKI  [EMAIL PROTECTED]













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Tuesday, June 28, 2005 5:38
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
simultaneus calls?





The 1 m internet
connection will be the limiting factor in your setup, you did not state what
type of internet connection, but given the speed of 1 mbit it must be DSL (or
maybe fraction t/e1).



Is the outbound speed
also 1m? Is there data on the line also? How much? What about voice Qos?



You should start here http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption















From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erdem HAKI
Sent: Tuesday, June 28, 2005 3:04
AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
simultaneus calls?





Hello, 



How can i learn my asterisk how many simulyaneus calls
support?



My configuration: 80 GB HDD, 1 GB Ram, P4 2,8 MHz
processor, Fedora Core 3 minimum installation, no digium cards, codecs g729 or
gsm, 1Mbit internet connection.



Thanks for your interest...



Erdem HAKI  [EMAIL PROTECTED]








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Re: [Asterisk-Users] Some phone numbers always busy

[Moises Silva]

you say busy because you hear the busy tone, or you say busy because
you see in the console that the $DIALSTATUS is busy???

post the verbose console output please (asterisk
-r)

best regards

On 6/28/05, vdasilva [EMAIL PROTECTED] wrote:
  
  
 
 Hello 
 
   
 
 I am using Asterisk 1.0.5 with i4l, a HFC ISDN BRI card, and some
 Grandstream products. The system works fine except that some external
 telephone numbers, when dialed always give a busy tone whereas other numbers
 are fine. I've checked extensions.conf and even tried hard coding the number
 in extensions.conf and always get the same result. 
 
   
 
 Has anyone experienced something similar? 
 
   
 
 Vicente 
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Re: [Asterisk-Users] LiveVoip is Bankrupt

[steve szmidt]

On Monday 27 June 2005 20:04, Robert Webb wrote:
  I agree with that fact the same questions get posted, but
  that problem is compounded by the fact the archives are not
  really searchable. If the were as lease some users would search.
  The archives need to be fully indexed.

 In a Google search box: site:lists.digium.com What you are searching
 for

The problem many newbies faces is TOO MUCH information. Not being able to see 
the trees because of the forest basically.

It does not matter either if it has been discussed until someone went crazy or 
died. The reason it keeps coming up is because it has not been solved.

I totally agree with why. I sure don't want to be the one babysitting them. 

These posts were simply pointing out what I think, as a former educator, is 
part of the problem. Something which is not that hard to do. And indeed 
during some spare times I may put together something which is a lower 
gradient for those who have a hard time getting it. I sure would like to.

Now it could very well be that many of these people never get anywhere because 
it's just too hard for them. But I know when I started a few years back, that 
a lot of the howto's have a stiff gradient. It skips pieces of information, 
assumes knowledge which is hard to come by and so on. Standard stuff.

I'm not assuming or expecting that anyone is going to act on what I'm saying. 
If it was easy someone would have already implemented it.

But I am saying that I see there are things that CAN be done which will make 
it easier. And if it makes it easier, this list will have less stupid and 
repetitive questions. More people will win using Asterisk and we should all 
win. (Except those who prefer fewer people competed in this arena. And there 
are a few here who are happy it's hard for others to take part of the fruit. 
There always are.)

-- 

Steve Szmidt

They that would give up essential liberty for temporary safety 
deserve neither liberty nor safety.
Benjamin Franklin
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Re: [Asterisk-Users] How do you handle NAT?

[Jean-Michel Hiver]

P.S. Yes, I have read stuff on NAT on the wiki. I'm more interested in 
other real world, working, solutions.


Apparently, Jasomi does pretty good SIP/NAT far end traversal 
solutions. From what I've read on the list, it's meant to be quite good 
- although expensive.


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Re: [Asterisk-Users] How do you handle NAT?

[Ray Van Dolson]

We've been feeling our way along with the NAT stuff (using SIP) as well.

At this point we are fairly small, so the keep-alive packets are not too bad.
What type of user load are you at and what are the specs on your Asterisk box?
I'm concerned we may run into this as well.

We do have the luxury that each Sipura device we use is sitting behind its own
NAT (a customer CPE).  So we can do port-forwarding and in combination with a
STUN server (MyStun), things work quite well.  The only issues left to deal
with are a lingering problem with ip_conntrack entries staying cached because
of the keep alive packets due to qualify=yes after the CPE's IP address
changes.

Curious to hear other's setups as well.  I would *love* to start using the
IAXy instead, but it has a couple shortcomings over the Sipura 2002's we're
using now:

- About $10/more
- Only has one line (apparently two lines is a bit more of a selling point).

Still trying to figure out a good way to make a case for the IAXy though.

Ray

On Tue, Jun 28, 2005 at 09:59:49AM -0500, Matthew Boehm wrote:
 We are interested in how other people are handling NAT problems. We have 
 several customers all of which have some sort of firewall/NAT device at 
 their location. For simplicity sake, all customers' internal networks 
 are 192.168.*.*.
 
 Our asterisk box is on public IP not blocked by any FW/NAT.
 
 I use QUALIFY=yes on all our customers' phones and I feel that sending 
 out 80-something keep-alive packets is causing our box to crawl and 
 cause bad calls.
 
 Would SER be better in this case? Should I have phones register with SER 
 instead of with Asterisk?
 
 Thanks,
 Matthew
 
 P.S. Yes, I have read stuff on NAT on the wiki. I'm more interested in 
 other real world, working, solutions.
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[Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]

[John Goerzen]

On 2005-06-28, Wiley Siler [EMAIL PROTECTED] wrote:
 So far my experience with TOS has been that most of them are pretty odd.

Not THAT odd :-)

 No one wants the liability of a stock trade gone foul or a call to the
 doctor that gets disconnected.  Essentially, those things in the TOS are
 just a CYA.  They are un-enforced but should someone decide to attempt
 to sue based upon a financial loss, the ITSP is covered.  

There is a difference between making users agree not to hold them liable
for these things, and forbidding users from doing these things
altogether.  It is also weird that users are forbidden to tell others
that they use VoipJet.


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Re: [Asterisk-Users] RTP session between two end users

[Eric Wieling aka ManxPower]

Erdem HAKİ wrote:



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
aka ManxPower
Sent: Monday, June 27, 2005 8:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RTP session between two end users

Erdem HAKİ wrote:



Is it possible that a RTP session between two end users  (so i want to use
asterisk as a signaling proxy and bypass RTP sessions)?



I used canreinvite=yes but it didn't work. 



Description from asterisk conf. File;

(canreinvite=yes; allow RTP voice traffic to bypass
Asterisk)




It's sip.conf.  reinvites only work if the codec is the same for the 
two endpoints and Asterisk does NOT have to listen for DTMF (no t or T 
on the dial line, no meetme, etc.)


***
We use same codec and don't use meetme etc...  So what else should i do?


How are you determining if RTP audio is going thru Asterisk? 
Remember, SIP signaling will always go thru Asterisk.


Also do a sip show channels during a call to confirm that the codecs 
are the same.


--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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[Asterisk-Users] MeetMe application in Asterisk V1.07

[monty-asterisk]

Hello list,

I wonder if someone might be able to clear up something for me.

I recently set up asterisk and have now managed to get the MeetMe 
application up and running.



When I dial the extension to access the conference/MeetMe application, the 
only prompt I hear is:You are currently the only person in this 
conference.  When I use a friend's newly installed asterisk, I hear: 
After the tone, say your name and then press the pound key.  We both 
have used virtually the same Meetme configurations.(The FWD 514 extension 
works the same way)  I believe I have all the necessary sounds but am 
really quite stuck here.  Please help! I am using the latest Debian/AMD64 
package and my friend compiled from cvs source I believe.


My settings follow:

extension.conf:

exten = 8800,1,Meetme(8801|aciMps)  ; ext-8800 accesses conf-room  8801

meetme.conf

[rooms]
conf = 8801


Thanks for any help,
  Monty
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RE: [Asterisk-Users] list Searchability

[Wiley Siler]

Great points Steve.  I think the best we can do is all throw the newbies
a bone ounce in a while.  Redirection to the content that is relevant is
enough to get most people on the path.  Like you said, the hardest part
is not seeing the trees for the forest.  

This is the whole teach a man to fish parable.

It is pretty easy to tell someone 
A) How to search and where to look
B) The basics of what Asterisk can do
C) How to be a good list citizen

With those tools, almost anyone can get their start here and beat the
learning curve.   Like it has been pointed out, there is not much else
we can do with so much information in a free flowing format.

Cheers,
Wiley




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of steve
szmidt
Sent: Tuesday, June 28, 2005 8:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt

On Monday 27 June 2005 20:04, Robert Webb wrote:
  I agree with that fact the same questions get posted, but
  that problem is compounded by the fact the archives are not
  really searchable. If the were as lease some users would search.
  The archives need to be fully indexed.

 In a Google search box: site:lists.digium.com What you are searching
 for

The problem many newbies faces is TOO MUCH information. Not being able
to see 
the trees because of the forest basically.

It does not matter either if it has been discussed until someone went
crazy or 
died. The reason it keeps coming up is because it has not been solved.

I totally agree with why. I sure don't want to be the one babysitting
them. 

These posts were simply pointing out what I think, as a former educator,
is 
part of the problem. Something which is not that hard to do. And indeed 
during some spare times I may put together something which is a lower 
gradient for those who have a hard time getting it. I sure would like
to.

Now it could very well be that many of these people never get anywhere
because 
it's just too hard for them. But I know when I started a few years back,
that 
a lot of the howto's have a stiff gradient. It skips pieces of
information, 
assumes knowledge which is hard to come by and so on. Standard stuff.

I'm not assuming or expecting that anyone is going to act on what I'm
saying. 
If it was easy someone would have already implemented it.

But I am saying that I see there are things that CAN be done which will
make 
it easier. And if it makes it easier, this list will have less stupid
and 
repetitive questions. More people will win using Asterisk and we should
all 
win. (Except those who prefer fewer people competed in this arena. And
there 
are a few here who are happy it's hard for others to take part of the
fruit. 
There always are.)

-- 

Steve Szmidt

They that would give up essential liberty for temporary safety 
deserve neither liberty nor safety.
Benjamin Franklin
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Re: [Asterisk-Users] Bad Bad Performance; Max 20 Calls on Quad Proc?

[Matthew Boehm]

Wiley Siler wrote:

What distro and kernel version are you using?
What version of Asterisk?

Thanks,
Wiley


RedHat 9
2.4.20-8smp #1 SMP Tue Dec 28 17:23:01 CST 2004 i686 i686 i386 GNU/Linux

Running CVS-HEAD as of June 27, 2005 around 9PM CST

-Matthew

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[Asterisk-Users] Hangup detection on Panasonic KXTD816

[Hilton Williams]

Hi

I have a Digium TDM400 card with 4 FXO modules 
connected tothe extension ports onaPanasonic KXTD816. 
I'm using [EMAIL PROTECTED] v1.0, which has 
Asterisk 1.07.

There's a problem that Asterisk doesn't detect when 
the line is disconnected on the Panasonic. The Panasonic doesn't provide 
polarity reversal orcurrent drop or anything like that to indicate hangup. 
It just plays the dial tone again.

I've tried several different things, thinking that 
the settings were not correct for South Africa. I've tried adding busydetect=yes and busycount=4, since that worked for 
our analogue Telecom lines (they work the same way as the 
Panasonic).


At the moment, I'm thinking I need to change the 
busy tone in indications.conf. Currently, it is set to 400/500,0/500, 
which are the settings for Alcatel switches in the Western Cape, South 
Africa. The PABX seems to need 400/250,0/250, but that doesn't 
work.

Has anybody tried this before? It's probably 
not specific to South Africa, but Panasonic KXTD PABXs.

Regards
Hilton



  
  

  

  
Datatex Dynamics CC Web site http://www.datatex.co.za/ Email 
  to [EMAIL PROTECTED] Tel 
  +27215924033Fax +27215924077

  

  
The use of the Datatex e-mail facility is not permitted
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of any nature whatsoever. Datatex hereby distances itself
from and accepts no liability in respect of the
unauthorised use of its e-mail facility or the sending of
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purposes. Datatex furthermore disclaims liability for any
unauthorised instruction for which permission was not
granted. Any recipient of an unacceptable communication,
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[Asterisk-Users] TDM400

[Jared Armstrong]

Hi,



I have an TDM400 4 FXO module setup on my dual Celeron server
running asterisk 1.0.2 and I have had to restart the asterisk process multiple
times lately. I was wondering if anyone else has to restart the asterisk
process after storms roll through their area. Before I restart I usually have
sound quality issues and weird pickup problems. But, once I restart the
asterisk process everything returns to normal operation. If this continues I am
going to have to write a CRON script that restarts asterisk once each night,
but I would prefer to avoid this.



Also, the system only has 256 megs of ram, and asterisk is
using 200 megs of this. Could this perhaps be caused by a lack of memory? Or is
this just an issue with the TDM400?



Jared Armstrong








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RE: [Asterisk-Users] list Searchability

[Steve Hanselman]

Might be worth asking the owner of voip-info.org if the mailing list
link can go on the left sidebar permanently?


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: 28 June 2005 16:26
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] list Searchability

Great points Steve.  I think the best we can do is all throw the newbies
a bone ounce in a while.  Redirection to the content that is relevant is
enough to get most people on the path.  Like you said, the hardest part
is not seeing the trees for the forest.

This is the whole teach a man to fish parable.

It is pretty easy to tell someone
A) How to search and where to look
B) The basics of what Asterisk can do
C) How to be a good list citizen

With those tools, almost anyone can get their start here and beat the
learning curve.   Like it has been pointed out, there is not much else
we can do with so much information in a free flowing format.

Cheers,
Wiley




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of steve
szmidt
Sent: Tuesday, June 28, 2005 8:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt

On Monday 27 June 2005 20:04, Robert Webb wrote:
  I agree with that fact the same questions get posted, but
  that problem is compounded by the fact the archives are not
  really searchable. If the were as lease some users would search.
  The archives need to be fully indexed.

 In a Google search box: site:lists.digium.com What you are searching
 for

The problem many newbies faces is TOO MUCH information. Not being able
to see
the trees because of the forest basically.

It does not matter either if it has been discussed until someone went
crazy or
died. The reason it keeps coming up is because it has not been solved.

I totally agree with why. I sure don't want to be the one babysitting
them.

These posts were simply pointing out what I think, as a former educator,
is
part of the problem. Something which is not that hard to do. And indeed
during some spare times I may put together something which is a lower
gradient for those who have a hard time getting it. I sure would like
to.

Now it could very well be that many of these people never get anywhere
because
it's just too hard for them. But I know when I started a few years back,
that
a lot of the howto's have a stiff gradient. It skips pieces of
information,
assumes knowledge which is hard to come by and so on. Standard stuff.

I'm not assuming or expecting that anyone is going to act on what I'm
saying.
If it was easy someone would have already implemented it.

But I am saying that I see there are things that CAN be done which will
make
it easier. And if it makes it easier, this list will have less stupid
and
repetitive questions. More people will win using Asterisk and we should
all
win. (Except those who prefer fewer people competed in this arena. And
there
are a few here who are happy it's hard for others to take part of the
fruit.
There always are.)

--

Steve Szmidt

They that would give up essential liberty for temporary safety
deserve neither liberty nor safety.
Benjamin Franklin
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 183

[Nguyen Trung Tin]

Hello All
I'm using TxFAX and rxFax. this's work when my system connected with PABX. then when i connect my card with PSTN, my system don't work. it's don't send and receive any thing fax document.
Thanks
Please help me___
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Re: [Asterisk-Users] How do you handle NAT?

[Matthew Boehm]

Ray Van Dolson wrote:


What type of user load are you at and what are the specs on your Asterisk box?


	I'm seeing loads on certain Asterisk threads reach in the upper 70% 
periodically. Running a Quad proc P3 500Mhz with RedHat9 on 2.4.20 SMP 
kernel.



We do have the luxury that each Sipura device we use is sitting behind its own
NAT (a customer CPE).  So we can do port-forwarding and in combination with a
STUN server (MyStun), things work quite well.


	The phones that our customers use support STUN. If I turn up a STUN 
server and tell these phones to use that, does that mean I no longer 
need to use QUALIFY?


-Matthew

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[Asterisk-Users] Using asterisk as Quality Monitoring Platform?

[Francois Lambert]

Hi Aaron,

We have 2 products that do what you are looking for. It has the same
features has Nice and Witness including Voice recording, Screen capture,
Evaluation forms and scoring. One of the product is also integrated with
Genesys. The other one is fully based on asterisk.

Let me know, if you would like to get more information

Francois Lambert
COO
Atelka Inc.
Tel. : 514-448-4905 #2200
Cel. : 514-570-4797



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[Asterisk-Users] Unable to connect to remote asterisk

[Jason Greene]

Hello,
 I'm trying to figure out why the asterisk service starts fine, but 
when i try to connect by typing asterisk -r I get:

Unable to connect to remote asterisk

The service is running and lists under ps -ef as:
asterisk -vvvg -c

any help is appreciated
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[Asterisk-Users] Revision I Board TDM04b

[Steve Totaro]

I cannot get this thing to work. Anyone know 
of any tricks?
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RE: [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]

[Wiley Siler]

Touche`.  

Very good points... 

But, hey... As long as it works as promised I am OK to let them make
strange TOS!

W



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Goerzen
Sent: Tuesday, June 28, 2005 8:24 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]

On 2005-06-28, Wiley Siler [EMAIL PROTECTED] wrote:
 So far my experience with TOS has been that most of them are pretty
odd.

Not THAT odd :-)

 No one wants the liability of a stock trade gone foul or a call to the
 doctor that gets disconnected.  Essentially, those things in the TOS
are
 just a CYA.  They are un-enforced but should someone decide to attempt
 to sue based upon a financial loss, the ITSP is covered.  

There is a difference between making users agree not to hold them liable
for these things, and forbidding users from doing these things
altogether.  It is also weird that users are forbidden to tell others
that they use VoipJet.


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[Asterisk-Users] Asterisk dies with Meetme

[João Amaro]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi List

I'm trying to create a conference room using H323 channels.

If i start asterisk normally (service asterisk restart) and connect to
cli using -vvvr options, when a user enters the Conference,
asterisk says You are the only ... and then dies, withou any error
message, nothing at all.
But, if i start asterisk with cli console (-vvc) theres is no
problem and i can create the conference rooms.

Zaptel and ztdummy modules are already loaded.
Anyone with the same problem ?

Asterisk: 1.0.8
Kernel: 2.6.9-5.0.3.ELsmp


Regards, João Amaro


 DUMP 
[EMAIL PROTECTED] ~]# service asterisk restart
[EMAIL PROTECTED] ~]# asterisk -r
(...)
~-- Executing MeetMe(OH323/R47, 1234|ciMps|) in new stack
~  == Parsing '/srv/etc/asterisk/meetme.conf': Found
2005-06-28 16:49:53 WARNING[4555]: channel.c:1913 ast_request: No
channel type registered for 'zap'
2005-06-28 16:49:53 WARNING[4555]: app_meetme.c:227 build_conf: Unable
to open pseudo channel - trying device
~-- Created MeetMe conference 1023 for conference '1234'
~-- Playing 'conf-onlyperson' (language 'en')
Vr-VoIP1*CLI
Disconnected from Asterisk server
Executing last minute cleanups

-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.0 (GNU/Linux)

iD8DBQFCwXQFJUm/Bor63CERAtE7AKCZ/IUG2IK/myBoc8iHsR7uV5PmDgCgzaTp
hwQpYpTzLrp7p72beDciw+Q=
=rJYD
-END PGP SIGNATURE-

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Re: [Asterisk-Users] ASTCC not billing

[Ade Agbero]

I have just installed [EMAIL PROTECTED] version 1.1,I have made a number of successful calls, but the bill cost remains "0".

I have kept everything simple:






Pattern
Comment
Trunks
Connect Fee
Inc. Seconds
Cost per additional minute



DANSAM
0
0
10






Trunk Name
Technology
Peer/Trunk

TEST
SIP
213.45.62.117

Darren Wiebe [EMAIL PROTECTED] wrote:
We are using it on [EMAIL PROTECTED] version 1.1Darren Wiebe[EMAIL PROTECTED]Darren Wiebe wrote: Ade Agbero wrote: I set "notransfer=yes" and on SIP you set "canreinvite=no", but  ASTCC is still not billing.  I formated and reinstalled [EMAIL PROTECTED] and  got the latest CVS of Astcc, but ASTCC is still not billing.  What version of [EMAIL PROTECTED] can be confirmed  working with Astcc. I can try to find that out tomorrow. I don't know off the top of my  head. Did you try the latest stable version of asterisk? That is  what I did to resolve the issue. Darren Wiebe [EMAIL PROTECTED] */Darren Wiebe
 <[EMAIL PROTECTED]>/* wrote: On IAX you set "notransfer=yes" and on SIP you set "canreinvite=no" Darren Juan Luis Moyano wrote: On Sab, 25 de Junio de 2005, 5:20 pm, Darren Wiebe dijo:   Do you have the notransfer and reinvite lines set properly? I had this same problem with ASTCC but found that if I removed asterisk including the source and did a clean reinstall it worked suddenly.  DarrenDarren, how is the proper way of setting notransfer and canreinvite lines on IAX. TIA.___ Asteris
 k-Users
 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users  How much free photo storage do you get? Store your holiday snaps for  FREE with Yahoo! Photos. *Get Yahoo! Photos*    ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Speech driven crm apps

[Dean Collins]

Interesting article, hopefully this might give some people
here some ideas about cool asterisk apps to develop.



Cheers,

Dean








 
  
  Voice being heard in CRM
  
 
 
  
  
  
 
 
  
  By Barney Beal, News
  Editor
  28 Jun 2005 | SearchCRM.com 
  
 

 
  
  
  
 
 
  
  SOUND OFF! Post your comments
  
  
  
  
  
  
  
  
 
 
  
  
  
  
  
  
  
  
  
 





 
  
  
  
  
  
  
  
  
  
 
 
  
  With the acquisition of Convergent Voice and
  the release of eight new voice applications, RightNow Technologies has become
  the first CRM suite provider to make a concerted push into the voice arena. 
  Speech is no longer an interesting
  technology, said Joseph Brown, vice president of Voice Solutions for
  the Bozeman, Mont.-based company. It's
  starting to move from early the adopter stage to an early majority
  market. 
  Brown, the former CEO of Edify, was brought in
  to RightNow to help lead the company's
  foray into the voice technology market. He is joined by David Lanning, the
  president and CEO of Convergent Voice, which RightNow has been working with
  to develop self-service voice applications since 2002. RightNow recently
  completed the acquisition of the Rochester, N.Y.-based company's intellectual property, core development team, accounts
  receivable and contract rights. Terms of the deal were not disclosed. It is
  RightNow's first acquisition. 
  RightNow is making eight new voice
  applications available around its Service module including voice access to
  the knowledge base, a voice incident management system, location finder,
  order status, repair tracking, refund status, password reset and customer
  survey tool. In the future, the company will roll out voice applications tied
  to the Sales and Marketing modules as well, Brown said. 
  We're
  piggy backing and focusing on our strength, Brown said. Our CRM
  environment, our best practices -- we are voice enabling those CRM solutions
  now. What the voice vendors don't
  have is the strength or infrastructure of CRM software. We don't say we sell Web self-service or voice
  self-service--we sell self-service. 
  Despite the emergence of the online channel,
  voice interactions still account for 60 to 90% of all interactions a company
  has with customers, according to RightNow CEO Greg Gianforte. 
  We're
  creating the ability for consumers to do anything they did on the Web on the
  phone, Brown said. 
  The strength of RightNow's
  offering is that it's based on the
  company's existing knowledge base
  technology, said Art Schoeller, senior analyst at the Boston-based Yankee
  Group. 
  I think RightNow is being reasonably
  conservative about this, he said. Here's
  a way they can extend their knowledge base into the voice channel. They're further down the road than building a speech
  recognition or [Interactive Voice Response] from the ground up. 
  Schoeller doesn't
  expect RightNow to compete for customers who are looking for stand alone
  voice applications, but he does see the benefit for customers who have
  already started down RightNow's
  knowledge base approach to CRM. 
  RightNow is offering the voice applications as
  hosted tools. Pricing comes in at $25,000 for the knowledge base and $15,000
  for the application, plus a per minute charge based on volume. 
  It is the start up costs that have delayed the
  adoption of voice technology and RightNow is alleviating that by providing an
  on-demand alternative, Brown said. 
  According to Schoeller, RightNow's voice push should give it a leg up on the other
  CRM suite vendors who may not consider it as important. 
  Voice is still tough enough that when
  people look at it and see what it takes to do productively, [CRM suite
  vendors] realize there are better revenue areas to chase, he said. 
  
 









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Re: [Asterisk-Users] Revision I Board TDM04b

[Mailing List]

need latest zaptel source

_Mobilcomhttp://www.mobilcom.net



  - Original Message - 
  From: 
  Steve Totaro 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Tuesday, June 28, 2005 2:53 
PM
  Subject: [Asterisk-Users] Revision I 
  Board TDM04b
  
  I cannot get this thing to work. Anyone 
  know of any tricks?
  
  

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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 183

[Nguyen Trung Tin]

Hello All
how to install functions allow called record current call by pressed any key to wave file. for examples.
the caller call to asterisk, then press 123 tone to switch to saler person. then two persons conversation, the saler want to save current call. saler may be press any key as *5 or orther key. current context call monitor application.
how to install above function.

Please help me, Thanks
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Re: [Asterisk-Users] pbx_extension_helper: No application 'agi'

[Tom Fielding]

Hi Jay,

It's just the standard test script, agi_test.agi in
/var/lib/asterisk/agi-bin (pasted below).

So am I to assume therefore that I don't have to do anything special
during compilation and installation ('make' and 'make install') to
enable agi support?

Thanks,

Tom

[EMAIL PROTECTED] agi-bin]# cat agi-test.agi 
#!/usr/bin/perl
use strict;

$|=1;

# Setup some variables
my %AGI; my $tests = 0; my $fail = 0; my $pass = 0;

while(STDIN) {
chomp;
last unless length($_);
if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2;
}
}

print STDERR AGI Environment Dump:\n;
foreach my $i (sort keys %AGI) {
print STDERR  -- $i = $AGI{$i}\n;
}

sub checkresult {
my ($res) = @_;
my $retval;
$tests++;
chomp $res;
if ($res =~ /^200/) {
$res =~ /result=(-?\d+)/;
if (!length($1)) {
print STDERR FAIL ($res)\n;
$fail++;
} else {
print STDERR PASS ($1)\n;
$pass++;
}
} else {
print STDERR FAIL (unexpected result '$res')\n;
$fail++;
}
}

print STDERR 1.  Testing 'sendfile'...;
print STREAM FILE beep \\\n;
my $result = STDIN;
checkresult($result);

print STDERR 2.  Testing 'sendtext'...;
print SEND TEXT \hello world\\n;
my $result = STDIN;
checkresult($result);

print STDERR 3.  Testing 'sendimage'...;
print SEND IMAGE asterisk-image\n;
my $result = STDIN;
checkresult($result);

print STDERR 4.  Testing 'saynumber'...;
print SAY NUMBER 192837465 \\\n;
my $result = STDIN;
checkresult($result);

print STDERR 5.  Testing 'waitdtmf'...;
print WAIT FOR DIGIT 1000\n;
my $result = STDIN;
checkresult($result);

print STDERR 6.  Testing 'record'...;
print RECORD FILE testagi gsm 1234 3000\n;
my $result = STDIN;
checkresult($result);

print STDERR 6a.  Testing 'record' playback...;
print STREAM FILE testagi \\\n;
my $result = STDIN;
checkresult($result);

print STDERR == Complete ==\n;
print STDERR $tests tests completed, $pass passed, $fail failed\n;
print STDERR ==\n;



On 6/28/05, Jay Milk [EMAIL PROTECTED] wrote:
 That's incorrect, both versions represent valid syntax.
 
 To the OP -- show us the contents of test.agi
 
  -Original Message-
  From: John Millican [mailto:[EMAIL PROTECTED]
  Sent: Tuesday, June 28, 2005 8:23 AM
  To: Tom Fielding; Asterisk Users Mailing List -
  Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] pbx_extension_helper: No
  application 'agi'
 
 
  exten = s,1,agi(test.agi)
  should be
  exten = s,1,agi,test.agi
  If there any arguments to send the script use
  exten = s,1,agi,test.agi|args_to_pass
 
  John M
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Re: [Asterisk-Users] ASTCC not billing

[Ade Agbero]

Please ignore previous email:

I have just installed [EMAIL PROTECTED] version 1.1,I have made a number of successful calls, but the bill cost remains "0".

I have kept everything simple:







Pattern
Comment
Trunks
Connect Fee
Inc. Seconds
Cost per additional minute

44.*

TEST
0
0
10


1.*

TEST
0
0
10







Trunk Name
Technology
Peer/Trunk

TEST
SIP
213.45.62.117
Darren Wiebe [EMAIL PROTECTED] wrote:
We are using it on [EMAIL PROTECTED] version 1.1Darren Wiebe[EMAIL PROTECTED]Darren Wiebe wrote: Ade Agbero wrote: I set "notransfer=yes" and on SIP you set "canreinvite=no", but  ASTCC is still not billing.  I formated and reinstalled [EMAIL PROTECTED] and  got the latest CVS of Astcc, but ASTCC is still not billing.  What version of [EMAIL PROTECTED] can be confirmed  working with Astcc. I can try to find that out tomorrow. I don't know off the top of my  head. Did you try the latest stable version of asterisk? That is  what I did to resolve the issue. Darren Wiebe [EMAIL PROTECTED] */Darren Wiebe
 <[EMAIL PROTECTED]>/* wrote: On IAX you set "notransfer=yes" and on SIP you set "canreinvite=no" Darren Juan Luis Moyano wrote: On Sab, 25 de Junio de 2005, 5:20 pm, Darren Wiebe dijo:   Do you have the notransfer and reinvite lines set properly? I had this same problem with ASTCC but found that if I removed asterisk including the source and did a clean reinstall it worked suddenly.  DarrenDarren, how is the proper way of setting notransfer and canreinvite lines on IAX. TIA.___ Asteris
 k-Users
 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users  How much free photo storage do you get? Store your holiday snaps for  FREE with Yahoo! Photos. *Get Yahoo! Photos*    ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] pbx_extension_helper: No application 'agi'

[Moises Silva]

i could not follow this message from the beginning, but by the subject
i may have an idea of whats going on.

does the command 'show applications' in asterisk console shows up AGI
in its output?
do you have load = res_agi.so in /etc/asterisk/modules.conf ???

best regards

On 6/28/05, Jay Milk [EMAIL PROTECTED] wrote:
 That's incorrect, both versions represent valid syntax.
 
 To the OP -- show us the contents of test.agi
 
  -Original Message-
  From: John Millican [mailto:[EMAIL PROTECTED]
  Sent: Tuesday, June 28, 2005 8:23 AM
  To: Tom Fielding; Asterisk Users Mailing List -
  Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] pbx_extension_helper: No
  application 'agi'
 
 
  exten = s,1,agi(test.agi)
  should be
  exten = s,1,agi,test.agi
  If there any arguments to send the script use
  exten = s,1,agi,test.agi|args_to_pass
 
  John M
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-- 
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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[Asterisk-Users] Junghanns 4 port BRI problem

[Doug Reid - Stormcorp]

Hi All

I have a Junghanns BRI 4 port installed where only the first channel
of each line is working i.e. channels 1 and 4 work but 2 and 5 don't.

Our config is the same on this box as 15 other similar installations
where all works well. the only error I see is in /var/log/messages:

Jun 28 15:49:31 pbxct kernel: qozap: CRC error for HDLC frame on card 1
(cardID 0) S/T port 2
Jun 28 15:51:27 pbxct kernel: qozap: CRC error for HDLC frame on card 1
(cardID 0) S/T port 2
Jun 28 15:53:09 pbxct kernel: qozap: CRC error for HDLC frame on card 1
(cardID 0) S/T port 2
Jun 28 15:56:48 pbxct kernel: qozap: CRC error for HDLC frame on card 1
(cardID 0) S/T port 2
Jun 28 15:58:06 pbxct kernel: qozap: CRC error for HDLC frame on card 1
(cardID 0) S/T port 2
Jun 28 16:01:01 pbxct kernel: qozap: CRC error for HDLC frame on card 1
(cardID 0) S/T port 2

Can anyone help with this?

Thanks

Doug

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[Asterisk-Users] This weeks Developer meeting

[Brian West]

IAX2/[EMAIL PROTECTED]/996 at 1pm CDT on Thurday the 30th.

If you have any topics that need to be covered please direct them to me.

Thanks,
/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

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RE: [Asterisk-Users] simultaneus calls?

[Bernard Cresencia]

I did a google search on 'voip speed test' - the first
site is very good. Here's the link:
http://www.talkswitch.com/voip/voip_test.php

It will test both your download and upload speeds and
will let you know how many concurrent calls at
different codecs your connection will support. 

Try it a few times and on different times of the day
to get an average.
--- Erdem HAKÝ [EMAIL PROTECTED] wrote:

 Yes it is DSL and outbound speed is aslo 1Mbit, it's
 a dedicated server and
 we just use to talk. I look at the web site which
 you suggested, but i want
 to learn how many calls supported practically? Any
 information do you have?
 
  
 
 Thanks
 
  
 
 Erdem HAKI - [EMAIL PROTECTED]
 
  
 
  
 
   _  
 
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On
 Behalf Of Damon Estep
 Sent: Tuesday, June 28, 2005 5:38 PM
 To: Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: RE: [Asterisk-Users] simultaneus calls?
 
  
 
 The 1 m internet connection will be the limiting
 factor in your setup, you
 did not state what type of internet connection, but
 given the speed of 1
 mbit it must be DSL (or maybe fraction t/e1).
 
  
 
 Is the outbound speed also 1m? Is there data on the
 line also? How much?
 What about voice Qos?
 
  
 
 You should start here

http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption
 
  
 
  
 
   _  
 
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On
 Behalf Of Erdem HAKI
 Sent: Tuesday, June 28, 2005 3:04 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] simultaneus calls?
 
  
 
 Hello, 
 
  
 
 How can i learn my asterisk how many simulyaneus
 calls support?
 
  
 
 My configuration:  80 GB HDD, 1 GB Ram, P4 2,8 MHz
 processor, Fedora Core 3
 minimum installation, no digium cards, codecs g729
 or gsm, 1Mbit internet
 connection.
 
  
 
 Thanks for your interest...
 
  
 
 Erdem HAKI - [EMAIL PROTECTED]
 
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Re: [Asterisk-Users] Hangup detection on Panasonic KXTD816

[Eric Wieling aka ManxPower]

Hilton Williams wrote:


Hi

I have a Digium TDM400 card with 4 FXO modules connected to the extension ports 
on a Panasonic KXTD816.  I'm using [EMAIL PROTECTED] v1.0, which has Asterisk 
1.07.

There's a problem that Asterisk doesn't detect when the line is disconnected on 
the Panasonic.  The Panasonic doesn't provide polarity reversal or current drop 
or anything like that to indicate hangup. It just plays the dial tone again.


Correct.  When I have to interface with a PBX I use FXS ports on 
Asterisk connected to the FXO/CO ports of the PBX.  This seems to 
(mostly) work well, since PBXs tend to be MUCH better at figureing out 
that a line is disconnected than Asterisk is.


--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 181

[Nguyen Trung Tin]

Hello All
How to detect remote called offhook.
i make a context as below
i created call file. copy to /var/spool/asterisk/outgoing.
Channel: vpb/g0/9050718MaxRetries: 1WaitTime: 10Context: ext-calloutExtension: sPriority: 1 
then when i copy to /var/spool/asterisk/outgoing. the asterisk auto call and wait for 10 second, auto playing wave file in context ext-callout, exten s, without remote called not yet offhook.
i want to asterisk don't play, waiting when remote called offhook, the play wave file.

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Re: [Asterisk-Users] DID in Western Canada

[Paul Fielding]

I tried a Calgary DID with Link2Voip, but they never did get it working 
correctly.   My primary complaint with their customer service is that it was 
basically non-existant.   It took 2 weeks before a service guy even 
responded to my problem, we fired a few emails back and forth, and then I 
never, ever, heard from them again.


I ended up just giving up and cancelling my DID.  It wasn't worth the 
hassle


Paul

- Original Message - 
From: Nelson Loyola [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Monday, June 27, 2005 10:11 AM
Subject: [Asterisk-Users] DID in Western Canada



Hello,

I'm having trouble getting finding a company that
provides DID in Western Canada. More specifically in
Edmonton, Alberta.

We have tried getting in contact with Link2Voip and
Calgary Telecom but neither seems to be answering
their phones or email.

I would appreciate it if anyone can point me in the
right direction.

Thank you,
Nelson

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[Asterisk-Users] Correction to Janghanns BRI problem

[Doug Reid - Stormcorp]

Hi all

Correction on my last mail, I found that line 1 both channels work
but on line 2 none work.

I have 2 BRI ISDN lines coming in on port 1 and 2 (4 channels) on a
Junghanns 4 port.

The setup by the Telco on this ISDN is different than our others, they
have 2 lines (4 channels) that are all connected to one telephone number
i.e. 701 5161. The second number should be 701 5162 but this number does
not exist. If we put a Sirrix card in all 4 channels (2 x BRI) work fine
on 701 5161 but when we put a Junghanns in only one line works.

It seems like the second line is not given a B channel from the NTU side
of the Telco.

Error in /var/log/messages:

Jun 28 15:49:31 pbxct kernel: qozap: CRC error for HDLC frame on card 1
(cardID 0) S/T port 2
Jun 28 15:51:27 pbxct kernel: qozap: CRC error for HDLC frame on card 1
(cardID 0) S/T port 2
Jun 28 15:53:09 pbxct kernel: qozap: CRC error for HDLC frame on card 1
(cardID 0) S/T port 2
Jun 28 15:56:48 pbxct kernel: qozap: CRC error for HDLC frame on card 1
(cardID 0) S/T port 2

Please if anyone could suggest a fix here it would be much appreciated.

Thanks

Doug

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Re: [Asterisk-Users] MeetMe application in Asterisk V1.07

[Moises Silva]

i think that the thing that really matters here is wich version of
Asterisk are you using exactly. I dont know wich version the latest
debian package is using, and i dont know wich version from CVS your
friend has compiled.

Also, its needed to show the extensions.conf configuration of your
friend. Specially the options passed to meetme

best regards

On 6/28/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 Hello list,
 
 I wonder if someone might be able to clear up something for me.
 
 I recently set up asterisk and have now managed to get the MeetMe
 application up and running.
 
 
 When I dial the extension to access the conference/MeetMe application, the
 only prompt I hear is:You are currently the only person in this
 conference.  When I use a friend's newly installed asterisk, I hear:
 After the tone, say your name and then press the pound key.  We both
 have used virtually the same Meetme configurations.(The FWD 514 extension
 works the same way)  I believe I have all the necessary sounds but am
 really quite stuck here.  Please help! I am using the latest Debian/AMD64
 package and my friend compiled from cvs source I believe.
 
 My settings follow:
 
 extension.conf:
 
 exten = 8800,1,Meetme(8801|aciMps)  ; ext-8800 accesses conf-room  8801
 
 meetme.conf
 
 [rooms]
 conf = 8801
 
 
 Thanks for any help,
Monty
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