date:20040114

At 9:12 PM -0500 1/14/04, Andrew Kohlsmith wrote:
http://www.imagestream.com/PCI_720.html

Regards,
Andrew
Yes, but it's not fully channelized so I suspect it would not do the 
right thing for a Zap driver interface board.  I seem to recall 
prior discussions about it on the list in response to my DS3 question 
a little bit ago...

Now, if you have some time and motivation, look at what would be 
required to port Zap to this ~$3500 board: 
http://www.sbei.net/archive/prs/2003/121703.htm  - they include Linux 
open-source drivers with the card, but I suspect they would only be a 
blueprint for what would ultimately be some extensive additions to 
the Zap architecture, and some minor changes to libpri as well.

JT

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Re: [Asterisk-Users] Re: Proposed solution for exit code priorityjumps

2004-01-14 Thread Adam Hart

Why don't we just go to a script, all this numbering and jumps everywhere
makes it too confusing. Hence why goto's shouldn't be used in programming.

my proposal:

[default]
1:
$ret = EnumLookup(${EXTEN})
if($ret == 2)
dial(blah)
else if($ret == 4)
answer()
Playback(sorry-no-enum-information)
else
dial(Zap/g1/${ENUM})
hangup()
s:
Playback(blah)
hangup()

-Adam

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Re: [Asterisk-Users] Re: Proposed solution for exit code priority jumps

At 8:53 PM -0600 1/14/04, Steven Critchfield wrote:
On Wed, 2004-01-14 at 14:12, Jeremy McNamara wrote:
 John Todd wrote:

  I realize this is a major code shift, since it would require work in
  pretty much every single application.  It's easy for me to talk about
  this since I can't accomplish such a task.  However, without attention
  now, this may never be solved, which would be a pity because it's a
  real crimp in the style of anyone doing more than trivial dialplan
  manipulation - anyone doing cascading dial failovers will attest to that.
 We were talking in the conf the other day about possibly creating a 'F'
 extension for 'failure'.  I'm not sure if this is even remotely close to
  what your talking about or not.

Hmm, F would be good, possibly B also. I could see a good case for the
lead digit for a pattern doing a goto to a context that could define
those for outbound calls.
[normal_extensions]
exten = 9,1,Goto(Extended_out,s,1)
[Extended_out]
exten = s,1,Playtones(dialtone)
exten = F,1,background(Failed)
exten = F,2,hangup
exten = B,1,congestion

exten = _NXX,1,Dial(Zap/g2/${EXTEN})

This example tightly controls the B and F special extensions to a
specific dial in the context.
--
Steven Critchfield [EMAIL PROTECTED]
That's a step in the right direction, I suppose, but we've already 
got applications that break this model (ENUM lookups, as an example, 
have a third return state of +51 on successful tel: lookups.)  As we 
get more sophisticated applications, we start crippling ourselves 
just to stick with the way it's always been.

I see sanity only in one of two paths:

  - eliminate exceptions; reduce possible returns from an application 
to either Success, Fail or Busy, and hand back any other possible 
values by setting some arbitrary string

  - have some method that allows controlled Goto's based on the 
numeric failure code, and then have every application list it's 
possible failure codes in the show application blah summary

JT
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Re: [Asterisk-Users] Static Noise coming from Wildcard FXS: Wildcard TDM400P

2004-01-14 Thread nanog

I figured out what was wrong had something to do with my rj11 cables
being routed next to my power cable but after movin it around the noise went
away.

thanks for the tips once again.
- Original Message - 
From: JR Richardson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 14, 2004 10:48 PM
Subject: RE: [Asterisk-Users] Static Noise coming from Wildcard FXS:
Wildcard TDM400P

I had the same problem, I reseated the daughter card on the board and that
helped but just for a short time.  Eventually the port wouldn't even break
dial-tone. All other ports 2, 34 on the card were fine, it was only port 1
giving a problem.  I swapped daughter boards around between ports but the
problem stayed with port 1.  Maybe this is a sign of a bad run of TDM boards
with some component associated with port one.  I'm in the process of RMA
with Digium for a new card.

What port is the problem you are experiencing?

JR

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, January 14, 2004 6:04 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Static Noise coming from Wildcard FXS: Wildcard
TDM400P

I recently plugged in Phone to my TDM400P Card to test out something I
mostly use sip phones to interface with *. All of sudden I'm getting lot of
line static noise coming of the card is there any settings I should look at
or anything I need to do on the command line at this point I'm open to any
ideas
I'm running
0.7.1 on Redhat 9.0 machine.

Any insight would be greatly appreciated.

-Frankie Gravato
[aolim]:cronparser
[irc]:crontibs

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Re: [Asterisk-Users] * For Call Center

2004-01-14 Thread nanog

sounds like one of those pesky auto dialers the simpsons make fun of.

- Original Message - 
From: empire underground [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 14, 2004 11:08 PM
Subject: [Asterisk-Users] * For Call Center

 Hi Everyone ;)

I have posted something like this before but yeilded no solid help as
of
 yet.
 I am new to * and havent even setup a box for it yet as to I have no clue
 what I should go ahead and buy before wasting a few $k. Im looking to
setup
 * for my office with outbound calling only with some call agents, and also
 remote agents so they can work from home. At this time im not looking to
do
 Voip at all... but that will change in the future. I have a T1 with 12
 anolog lines and the rest for data (768k). I need to know what cards I
 should buy? I would also like to setup the box with 12-16 lines for
outbound
 calling, and im nto trying to do (IVR). What I would like to do is make *
 either a predictive/auto dialer only. I read about a few people doing this
 when searching google but cant find the links anymore :( Aslo someone made
a
 win32 program to log into * and get screen pops of all the info that was
 dialed for that # such as address, name, phone, ect... I dont realy care
if
 I have to write an agi for it in linux because I hate winblows and would
 rather stay far far away from it ;) If anyone can help or point me in the
 right direction it would be much help ;)
 Also I have checked wiki allready... I cant really find anything there for
 this. Also is it even possible for this? I know I would have to write a
agi
 for the screen pops to popup in web browser and rout that info to the
person
 logged in and waiting in that queue, I was thinking about using sql
backend
 for the db and maybe writing agi to import the .csv file? Also I was
 thinking about flying someone down here to Florida if all else fails
(unless
 you already live here) to maybe help setup this type of box, or even
giving
 root access to the box and configuring it? because a commercial  dialer
 costs WAY too much! they want anywhere from $3500-30,000 for dialers...
and
 then even pay another $1,500 for a license per agent that wil be using it!
 talk about getting raped!
 thanx for all you help in advance
 chris

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RE: [Asterisk-Users] Re: Proposed solution for exit code priority jumps

2004-01-14 Thread Adam Goryachev

[EMAIL PROTECTED]  wrote:
 At 8:53 PM -0600 1/14/04, Steven Critchfield wrote:
 On Wed, 2004-01-14 at 14:12, Jeremy McNamara wrote:
  John Todd wrote:
 
   I realize this is a major code shift, since it would require work
in
   pretty much every single application.  It's easy for me to talk
about
   this since I can't accomplish such a task.  However, without
  attention  now, this may never be solved, which would be a pity
  because it's a  real crimp in the style of anyone doing more than
  trivial dialplan  manipulation - anyone doing cascading dial
 failovers will attest to that. 
 
 
  We were talking in the conf the other day about possibly creating a
'F'
  extension for 'failure'.  I'm not sure if this is even
 remotely close to
   what your talking about or not.
 
 Hmm, F would be good, possibly B also. I could see a good case for
the
 lead digit for a pattern doing a goto to a context that could define
 those for outbound calls. 
 
 [normal_extensions]
 exten = 9,1,Goto(Extended_out,s,1)
 
 [Extended_out]
 exten = s,1,Playtones(dialtone)
 
 exten = F,1,background(Failed)
 exten = F,2,hangup
 
 exten = B,1,congestion
 
 exten = _NXX,1,Dial(Zap/g2/${EXTEN})
 
 
 This example tightly controls the B and F special extensions to a
 specific dial in the context. --
 Steven Critchfield [EMAIL PROTECTED]
 
 That's a step in the right direction, I suppose, but we've already
 got applications that break this model (ENUM lookups, as an example,
 have a third return state of +51 on successful tel: lookups.)  As we
 get more sophisticated applications, we start crippling ourselves
 just to stick with the way it's always been.
 
 I see sanity only in one of two paths:
 
- eliminate exceptions; reduce possible returns from an application
 to either Success, Fail or Busy, and hand back any other possible
 values by setting some arbitrary string
 
- have some method that allows controlled Goto's based on the
 numeric failure code, and then have every application list it's
 possible failure codes in the show application blah summary

I wrote an email this morning which is 'awaiting moderator approval' ..
It would be nice if SOMEONE was the moderator, heck, even I would
volunteer to do that... It's damn annoying when you happen to send from
a non-subscribed email address

Anyway, basically what I said is your above option 2, each application
can define what their 'exit codes' are, whether failure or success. In
this way, we basically copy the way bash and most programming languages
are written. (ie, a function/program returns a code to describe what
happened). In bash this happens to be $?. So, pick a variable name, and
use that to return an arbitrary number defined in the show application
output.

Then create a app or something which will allow for case statements.
Maybe like:
Case(checkval,val1=context|ext|prio,val2=prio2,val3)

Regards,
Adam

 --
Adam Goryachev
Website Managers
Ph:  +61 2 9345 4395[EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au

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Re: [Asterisk-Users] linux journal article on asterisk

2004-01-14 Thread Balaji NJL

Hey i stay at Redmond too. Prbly we all can meet.
-B

- Original Message - 
From: Brett Schwarz [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 14, 2004 2:00 PM
Subject: RE: [Asterisk-Users] linux journal article on
asterisk

 arrggh, sorry about the website. It should be back
up
 now...

 Yes, I hang out here, but don't talk much, since I
am
 really busy right now...

 I am *really* close to you, I work in Redmond and
live
 in Bellevue :)

 --- calvis [EMAIL PROTECTED] wrote:
  Thanks for the link.

  This is an interesting article on Asterisk.  I was
  hoping to send him some
  kudos, but his website isn't working at
  http://www.bschwarz.com/.  And I
  just noticed the guy lives near me!

  Does anyone know if he hangs out on the list?

  Charles
  Internet Technology Group, Inc.
  Redmond, WA

  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On
  Behalf Of Tony Kava
  Sent: Wednesday, January 14, 2004 7:56 AM
  To: '[EMAIL PROTECTED]'
  Subject: RE: [Asterisk-Users] linux journal
article
  on asterisk

 For anybody who didn't know there is an
  article on asterisk in 
 February's Linux Journal.

Can you please provide a link to this article?
Franz From: [EMAIL PROTECTED] 

  Here's the link (I believe):

  http://www.linuxjournal.com/article.php?sid=6769

  --
  Tony Kava
  Senior Network Administrator
  Pottawattamie County, Iowa

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[Asterisk-Users] Codec matching weirdness

2004-01-14 Thread Dustin Goodwin

I am experiencing a problem that from list archive it appears others are 
running into. When I dial from Cisco 7960 via the * to Free World Dialup 
destinations that supports G.729 the call fails. The major error from 
the debug log is

Jan 15 00:11:14 NOTICE[22545]: channel.c:1481 ast_set_read_format: 
Unable to find a path from G729A to ULAW
Jan 15 00:11:14 NOTICE[22545]: channel.c:1451 ast_set_write_format: 
Unable to find a path from ULAW to G729A

So I compared the SDP info coming from the 7960, sent out from * and 
returning from the destination system and I have included them below.

Question 1: Why is * sending out SDP info that is different from the SDP 
info contained in original SDP from the phone?
Question 2: Is there a config option to force * to just passthrough the 
codec list sent by the 7960 in the invite?
Question 3: What are SDP codec matching rules for SIP endpoints? How do 
they decide on common codec. Comparing the SDP sent and receive all 
systems claim support for 3 common codecs:
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
Now of course each device specified these 3 codecs in a different order. 
 Under normal circumstances I feel this call should complete why is * 
claiming a codec mismatch?

- Dustin -

From phone
v=0
o=Cisco-SIPUA 5892 12461 IN IP4 192.168.68.12
s=SIP Call
c=IN IP4 192.168.68.12
t=0 0
m=audio 18114 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Sent to remote server by *

v=0
o=root 4205 4205 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 16798 RTP/AVP 4 3 0 8 2 5 10 7 18 110 97 101
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=rtpmap:110 SPEEX/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
 (no NAT) to 192.246.69.223:5060
Received from remote server

v=0
o=root 9755 9756 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 10066 RTP/AVP 18 3 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
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Re: [Asterisk-Users] System Attendent

2004-01-14 Thread Todd Lieberman

I made my own MP3 files for the hold music.  TL

Shad Mortazavi wrote:

Dear All,

I have a number of call queues defined in Asterisk.

I would like to program a system attendant that tells people;

1. Every 60 seconds 'Your call will be answered as soon as possible'
2. Tell the user how many calls are on the queue.
I would then like them put back on hold music.

Does someone have a configuration for this or something similar?

Your help would be greatly appreciated.

Kind Regards

Shad Mortazavi

US Technical Manager
Nexus Management
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Re: [Asterisk-Users] grandstream asterisk configuration

i just saw a UDP blocked message in my gs GUI. ater i rebooted again i got
  MAC Address:00.0B.82.00.3C.13
  Software Version:Program--1.0.3.81Bootloader--1.0.0.7
HTML--1.0.0.18
  detected firewall/NAT type is open Internet


assigning a STUN server also didn't help.

lloked at the voip-info stuff
  a.. use dtmfmode=info in your sip.conf for your Grandstream BudgeTone and
configure the GS accordingly
  b.. make sure to have a username=xxx entry in sip.conf that matches the
phone's name as given in the square brackets
  c.. For most installations, this is needed in the sip.conf user definition
(not in [general]):
disallow=all
allow=ulaw
allow=alaw

and did the same. still didn't work.

what can be done if my nat is actually blocking the udp packets??

chandra


- Original Message -
From: SW [EMAIL PROTECTED]
To: Chandra [EMAIL PROTECTED]; [EMAIL PROTECTED]
Sent: Wednesday, January 14, 2004 10:30 PM
Subject: Re: [Asterisk-Users] grandstream asterisk configuration


 Hi,

 In my experience with GS phones, you need STUN support to make it work
 properly (behind NAT), otherwise you would need lot of trial end error to
 figure out how to do port forwarding. If you have STUN you wouldn't need
to
 touch the Netgear (except for firewalls).

 If you can't run your own stun server (need two public IPs) then use one
of
 many STUN servers out there on public internet.

 For an example enable NAT traversal on your GS phone and point the STUN
 server to one of these STUN servers

 larry.gloo.net or stun01.newkinetics.com.

 Then reboot the GS and see how it discover the NAT (top of the gs web
GUI).
 If it is not a full cone or UDP blocked then you should be fine (Netgear
is
 restricted cone).

 Cheers

 SW


 From: Chandra [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] grandstream asterisk configuration
 Date: Wed, 14 Jan 2004 19:35:48 +0545
 Reply-To: [EMAIL PROTECTED]

 i have forwarded ports 5060 and 5000-5008 the ports used by sip and rtp to
 grandstream from my netgear. rtp.conf uses rtpstart 5000 and rtpend 5008.
i
 have also opened all 5060, 5000-5008 ports in my firewall configuration.
 grandstream uses 5004 port for rtp.

 what am i missing here? please tell me.

 chandra


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Re: [Asterisk-Users] grandstream asterisk configuration

hi i am not talking about * behind NAT. its * outside NAT and GS inside NAT.

chandra

- Original Message -
From: Steve [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 9:50 AM
Subject: Re: [Asterisk-Users] grandstream asterisk configuration

 On Wednesday 14 January 2004 01:52 pm, Mike Machado wrote:
  On Wed, 2004-01-14 at 08:45, SW wrote:
   Hi,

   In my experience with GS phones, you need STUN support to make it work
   properly (behind NAT), otherwise you would need lot of trial end error
to
   figure out how to do port forwarding. If you have STUN you wouldn't
need
   to touch the Netgear (except for firewalls).

 You don't need stun to work with Grandstream.
 My * is behind NAT and so is the GS of course. Two ports are open and
 redirected in the F/W, udp 4569 and 5036.
 I make and receive internal and external calls over both PSTN and the
 Internet.

 GS is configured:
 Software V 1.0.4.30
 Static IP
 SIP Server is Asterisk's IP
 SIP user ID is the extension of GS
 Authenticate ID as user ID
 No pw
 Name is Steve
 Codex are: PCMU, PCMA, PCMU, G729, G726, G728 - note no 723
 723 Rate is 6.3
 Silence Suppression is Yes
 Voice Frames are 2
 IP SoQ is 48
 VLAN 0
 SIP User is NOT phone number
 Dial Plan 202
 SIP register YEs
 Clear Reg oin reboot NO
 Expiration 60
 Early Dial No
 Use # as Dial Key is Yes
 SIP port 5060
 RTP 5004
 Random port is No
 NAT traversal is NO
 keel alive is 20
 TFTP server is 130.94.123.253
 Voice mail ID is 78202
 DTMF is in-audio
 Payload is 101 - this may need to be changed
 NTP time.nist.gov

 Now all my features used to work a few months ago. I then stopped using *
and
 came back a week ago. Updated CVS and now Hold is not working unless I
press
 #(!?) But I can call, receive, transfer and have a working V/M.

 --
 Steve

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  and willing to handle things, or life
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Re: [Asterisk-Users] Re: Proposed solution for exit code priority jumps

2004-01-14 Thread Tilghman Lesher

On Tuesday 13 January 2004 20:48, John Todd wrote:
 At 9:49 PM -0500 11/28/03, John Todd wrote:
 Proposal for Alternate Error Handling Jumping
 
 Why: I have written quite a bit into various extensions.conf files,
 and I've started to find myself getting really, really frustrated
 with the +101 and +51 and +blah format of error handling.  I often
 create very ugly and awkward dialing plans to handle jumps from (as
 an example) multiple Dial statements which directly follow one
 another.  Hardcoding a Goto into each application seems to be a
 method that, as Asterisk matures, should be left behind.
 
 I have whined before about the lack of exit codes from many
 applications (especially Dial) and perhaps there is some middle
 ground.  I have come up with two methods that might make the job of
 the advanced administrator significantly easier, and dialplans more
 compact.  Additionally, logic for handling results of applications
 would be visible in the same configuration line as the application,
 instead of in a long chain of comparisons, or not at all, as is the
 current case.

I have a third solution, which I have discussed with several other
developers, so I'll hash it out here:

We'll define several trappable variables, which can be extended at any
time.  Each application can choose to implement each one in terms of an
error label to goto on any particular condition or to simply return a
generic code and not allow a particular error condition to be handled.

For example, we might define the channel variables TRAP_HANGUP,
TRAP_BUSY, and TRAP_NOANSWER.  Each channel variable would
be defined to a label, which is at least a priority, possibly an
extension and priority, or context, extension, and priority, via the
usual syntax:  [[context|]extension|]priority.

When an application wishes to allow a particular error code to be
caught, it may lookup the appropriate channel variable.  If it exists
and is parsable into a label, then that context, extension, and priority
are set to be next in the dialplan and the application simply returns 0.
If the variable does not exist or is not parsable, then the generic
methods should be used (e.g. priority + 1 for unavailable; priority +
101 for busy).

This will allow for unlimited numbers of possible error or branch
conditions to be caught, without catching undesirable conditions.
There should probably be a generic function that can be called
to do the parsing and extension branching for each appropriate
condition, e.g. ast_app_handle_branch(TRAP_BUSY).

-Tilghman

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[Asterisk-Users] Windows Call Manager : Formerly [Asterisk-Dev] New Bounty

2004-01-14 Thread Christian Hoffmeyer


 I've personally put up a $300 USD bounty on a win32 call manager -
 hopefully a few others will help get the ball rolling :
 http://bugs.digium.com/bug_view_page.php?bug_id=848

Is C# and .NET fine?

This is already nearly done.

I can send you binaries of a single user call manager, and the operator
manager is in the pipe.

Actually, I'll just post these for download.

You have to install the Outlook dialer and the Astring manager separately.

Also, you can point the directory to a local file or a webfile by typing
http://url in the text box.

I'll also include a sample xml file for the directory. This uses the Cisco
directory format so it will integrate with their phones more easily.

Martin Croome is building this.

http://www.yottadot.org/download.php?op=viewsdownloadsid=10

Christian Hoffmeyer
YottaDot Solutions
Huntsville, AL

(w)   256.851.8689
(c)256.655.0321
(iax)  700.859.4508

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[Asterisk-Users] Asterisk in Linux Journal

2004-01-14 Thread Greg Boehnlein

Hello all,
Just got my February copy of Linux Journal, and lo and behold is 
an article on Asterisk. Haven't read through it yet, but a quick glance 
shows that it has some depth to it.

From the Linux Journal website:

Asterisk Open-Source PBX System  by Brett Schwarz 
Integrate land lines and VoIP on your company phone system.

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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Re: [Asterisk-Users] Asterisk 0.7.0

2004-01-14 Thread Adam Hart

 Any link where to directly find the main differences between 0.5.0.
and
 0.7.0??

from ChangeLog off CVS

Asterisk 0.7.0
 -- Removed MP3 format and codec
 -- Can now load and unload SIP,IAX,IAX2,H323 channels without core
 -- Fixed various compiler warnings and clean up source tree
 -- Preliminary AES Support
 -- Fix SIP REINVITE
 -- Outbound SIP registration behind NAT using externip
 -- More CLI documentation and clean up
 -- Pin numbers on MeeMe
 -- Dynamic MeetMe conferences are more consistent with static conferences
 -- Added channel variables ${HANGUPCAUSE}, ${SIPDOMAIN}, ${TIMESTAMP},
${ACCONTCODE}
 -- ODBC support for logging CDRs
 -- Indications for Norway and New Zeland
 -- Major redesign of app_voicemail
 -- Syslog support
 -- Reload logfiles with CLI command 'logger reload' and rotate logs with
logger rotate'
 -- Configurable DEBUG, NOTICE, WARNING, ERROR and ast_verbose messages now
appear on remote console
 -- Properly reaping any zombie processes
 -- Added applications SayUnixTime, SetCDRUserField, HasNewVoicemail,
ZapScan, Random, ResetCDR, NoCDR
 -- Make PRI Hangup Cause available to the dialplan
 -- Verify included contexts in extensions.conf
 -- Add DESTDIR support for building RPMs and packages
 -- Do route lookups on OpenBSD
 -- Add support for building on FreeBSD and OS X
 -- Add support for PostgreSQL in Voicemail
 -- Translate SIP hangup cause to PRI hangup cause where needed
 -- Better support for MOH in IAX2
 -- Fix SIP problem where channels were not removed on BYE
 -- Display codecs by name
 -- Remove MySQL and put PGSql instead for licensing reasons
 -- Better capability matching in SIP
 -- Full IBR4 compliance for chan_zap
 -- More flexible CDR handling
 -- Distinguish between BUSY and FAILURE on outbound calls
 -- Add initial support for SCCP via chan_skinny
 -- Better support for Future Group B signaling

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[Asterisk-Users] Asterisk (outside NAT) + BudgeTone (behind NAT)

I have been really trying to solve the this problem. Has anyone had a
success on this one? I have asterisk setup outside my NAT with public IP and
I am trying to establish a connection from Budgetone behind NAT with private
IP. Everything seems to be working fine. They are registered, call rings
successfully but there is a problem after the caller picks up the phone. IN
CLI i constantly get:

WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error:
Resource temporarily unavailable

i also get some grandstream1 is now too lagged and after sometime i get
grandstream1 is now reacheable messages..

i have this in sip.conf

[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0  ; Address to bind to
;externip = 200.201.202.203 ; Address that we're going to put in SIP
messages if we're behind a NAT
tos=lowdelay
disallow=all; Disallow all codecs
allow=ulaw  ; Allow codecs in order of preference

dtmfmode=info

[grandstream1]
type=friend
host=dynamic
secret=grandstream1
context=outgoing
nat=yes
reinvite=no
canreinvite=no
qualify=200


help.


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Re: [Asterisk-Users] Parking extension not working

2004-01-14 Thread Brian West

UM take that =a - that a is bad

bkw

On Tue, 13 Jan 2004, Sean Garland wrote:

 I have the standard parking.conf but extension 700 doesn't show up in my
 dialplan  Why?  I can dial 701 which tells me that I don't have any
 calls parked there.  700 just gives me invalid extension noise

 Should I have extension 700 defined elsewhere?

 Thanks

 parking.conf
 [general]

 parkext =a 700  ; What ext. to dial to park
 parkpos = 701-705  ; What extensions to park calls
 on
 context = parkedcalls  ; Which context parked calls are
 in
 parkingtime = 300  ; Number of seconds a call can
 be parked f

 *CLI Show dialplan

 [ Context 'parkedcalls' created by 'res_parking' ]
   '701' =  1. ParkedCall(701)
 [res_parking]
   '702' =  1. ParkedCall(702)
 [res_parking]
   '703' =  1. ParkedCall(703)
 [res_parking]
   '704' =  1. ParkedCall(704)
 [res_parking]
   '705' =  1. ParkedCall(705)
 [res_parking]


 Sean Garland
 Siskiyou Technology Consultants

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RE: [Asterisk-Users] Asterisk 0.7.0

2004-01-14 Thread Brian West

Yep that was the problem that was fixed! :)

bkw

On Tue, 13 Jan 2004, Dan Austin wrote:

 Would it be imprudent to ask what is broke with chan_h323?

 I finally managed to get it compiled using 0.7.0, and can
 get calls in and out of * with it, but limited testing shows
 no working RTP, and h.323 debug shows that * thinks the local
 address is 127.0.0.1.

 If that is the concern with chan_h323 in 0.7.0, I'll happily
 wait for the next release.  Otherwise I guess I have more
 digging to do.

 Dan

 -Original Message-
 From: Manuel João S. Costa Amaro [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, January 13, 2004 11:38 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Asterisk 0.7.0


 A Ter, 2004-01-13 às 16:01, Brian West escreveu:
   Why not quickly patch the source an release 0.7.1 if the bug is
   critical?
 
  Give it a few days and I bet we will.  because chan_h323 is broken
  also in 0.7.0 (JerJer :P  but him and I stayed up till 3 am fixing
  it.)
 

  I've tryied it today, for a few minutes. I think there's problem with oh323 also.

 If the problem persists, tomorrow i report.

 Anyone ?

 Cya

 --
 Joao Amaro ([EMAIL PROTECTED])
 Braga, Portugal



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RE: [Asterisk-Users] linux journal article on asterisk

2004-01-14 Thread Franz Edler

 From: [EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 

 For anybody who didn't know there is an article on asterisk in February's
 Linux Journal.

Can you please provide a link to this article?
Franz

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RE: [Asterisk-Users] call parking

2004-01-14 Thread Girish Gopinath

Sean,
Check out this url:
http://www.automated.it/guidetoasterisk.htm

Girish

From: Sean Garland [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] call parking
Date: Tue, 13 Jan 2004 19:41:05 -0800
I am having trouble with call parking...  I am basically using the stock
sample files, but extension 700 doesn't show up in my dialplan.  When I
transfer a call to 700, I get the fast busy like there is extension
700...

HELP!

Sean Garland

_
Free transactions in any ATM across India. 
http://server1.msn.co.in/msnleads/suvidha/dec03.asp?type=hottag Click here.

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Re: [Asterisk-Users] Zultys Zip2 (SIP)

2004-01-14 Thread Terence Parker

Hi,

Thanks for that.

I did try updating the ZIP2 firmware - don't know if it helped or not, 
but I am able to login now by setting absolutely all settings to be the 
desired username - including the 'extension' number, which I just 
entered text for.

This is very stupid though. If all these manufacturers are producing 
things to so-called SIP 'open standard' - why should there be so many 
inconsistencies in how things are done?

Anyways, the important thing is it works now.

Terence


Hello,

I don´t have any Zultys ZIP2 but I have several of Zultys ZIP4x4 and
they are working great with asterisk. And I´m calling in/out without
problem with chan_capi.
Do you have the latest firmware in the ZIP2? They have recently changed
something regarding authentication in the ZIP4x4 firmware.
---JanM---
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Re: [Asterisk-Users] call parking

2004-01-14 Thread Brian West

You can no longer dial 700 directly nor have 700 in your dialplan.  I just
tested this and park does infact work make sure you have T or t on the
dial.

bkw

On Tue, 13 Jan 2004, Sean Garland wrote:

 I am having trouble with call parking...  I am basically using the stock
 sample files, but extension 700 doesn't show up in my dialplan.  When I
 transfer a call to 700, I get the fast busy like there is extension
 700...





 HELP!



 Sean Garland


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[Asterisk-Users] Kernel 2.6 and ztdummy?

2004-01-14 Thread Dan Austin

Title: Kernel 2.6 and ztdummy?






I wanted to solve this myself, but it is time to admit I know nothing about the kernel

Internals and modules. Im starting in the users list, hoping someone has seen

this and knows of a fix, as the developers are clearly busy and this is not critical

for me.

I cannot get ztdummy to load, as it complains about unknown symbols. I search

The kernel headers and all of the zap source and these symbols appear unique

to ztdummy:

fill_td

insert_td_horizontal

uhci_devices

uhci_interrupt

alloc_td

unlink_td

delete_desc

Does anyone have ztdummy loaded on a 2.6 kernel? If it matters, I have all

USB support built into my kernel and not in modules.

Thanks,

Dan

Re: [Asterisk-Users] Best Linux Distribution

2004-01-14 Thread Peer Oliver schmidt

Jose,

Mozilla 1.5 on Gentoo Linux 1.4 has trouble displaying the Asterisk 
pages of the Wiki.  (The irony!)  The text is pushed off the right 
margin of the page.  
The problem is not related to Mozilla 1.5 on Gentoo Linux 1.4, but has 
to do with Mozilla 1.5 on _any_ system. It is a known bug, which keeps 
me from using 1.5 on most of my systems. I have not checked 1.6ß, but 
maybe it is fixed in there. BTW:Firebird 0.7 has the same problem.

 Sometimes clicking back and then forward fixes the problem, but
 not always.
Most of the time Shift-Reload helps.

rgds
pos
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[Asterisk-Users] Why I can not use the conference

Hi All,

The meetme.conf have created as below:
[rooms]
conf = 101
conf = 102

and extensions.conf as below:
exten = _1XX,1,MeetMe,${EXTEN}

why the warning printed when I called 101.
WARNING[27660]: File pbx.c, Line 1051 (pbx_extension_helper): No application 'MeetMe' 
for extension (ipcentrex, 101, 1)

And I found asterisk have not load the meetme.conf when it starts up.

Zhang Peihao
[EMAIL PROTECTED]
2004-01-14



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[Asterisk-Users] Re: Asterisk on FreeBSD 4.9?

2004-01-14 Thread Brian Buhrow

I don't know if this helps, but I've been running our office IP phone
system on Asterisk, on a NetBSD-1.6.1 system for over a month now, with no
trouble at all.  The functionality is limited at the moment, due to the
lack of the features provided by the zaptel drivers, but I hope to remedy
that in the not-too-distant future.
-Brian
Message: 10
Date: Tue, 13 Jan 2004 20:27:07 -0500
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk on FreeBSD 4.9?
Reply-To: [EMAIL PROTECTED]

On Tue, Jan 13, 2004 at 12:24:20PM -0500, Jason T. Nelson wrote:

 love to be able to use Asterisk under FreeBSD. I've browsed the archives
 and perceived what appears to be a slightly hostile attitude towards those
 who ask about Asterisk support of other free operating systems even without
 using Digium hardware. Is this Linux-specific bias intentional or accidental?

I would call it historical. Asterisk was first developed on Linux,
and little attention was paid to portability. This is changing,
though there are still Linuxisms in the code. I would hesitate
to consider it stable yet on anything other than Linux, but
YMMV.

I personally would like to see Asterisk portable to any
*nix with pthreads, and am working to make this happen. As
always help in the form of patches, testing or accounts for
building and testing on less common types of systems are
appreciated.

-w
-- 
/~\  The ASCII Ribbon Campaign
\ /No HTML/RTF in email
 X No Word docs in email
/ \  Respect for open standards
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Re: [Asterisk-Users] Codec problems (SIP)

2004-01-14 Thread Terence Parker

Hi again,

Thanks for your help. Unfortunately that did not seem to solve the problem. After a bit of fiddling around, this is what i've managed to achieve with my asterisk setup so far.


1. With allow=all in sip.conf, nothing seems to work - not even voicemail. The following is sample output:

Executing Ringing(SIP/TerenceParker-1af0, ) in new stack
-- Executing Wait(SIP/TerenceParker-1af0, 2) in new stack
-- Executing VoiceMailMain(SIP/TerenceParker-1af0, ) in new stack
-- Playing 'vm-login' (language 'en')
WARNING[278546]: File app_voicemail.c, Line 2707 (vm_execmain): Couldn't read username
== Spawn extension (sip, 86, 3) exited non-zero on 'SIP/TerenceParker-1af0'

- Why should this happen? Surely with everything enabled, any coded should work!


2. With disallow=all ; allow=alaw ; allow=ulaw ; allow=g729 ; allow=gsm (and i've also tried without some of those and various combinations):

Executing Ringing(SIP/TerenceParker-af02, ) in new stack
-- Executing Wait(SIP/TerenceParker-af02, 2) in new stack
-- Executing VoiceMailMain(SIP/TerenceParker-af02, ) in new stack
-- Playing 'vm-login' (language 'en')
NOTICE[278546]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW
NOTICE[278546]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from GSM to G729A
WARNING[278546]: File chan_sip.c, Line 1182 (sip_write): Asked to transmit frame type 4, while native formats is 256 (read/write = 4/2)
WARNING[278546]: File file.c, Line 521 (ast_readaudio_callback): Failed to write frame
NOTICE[278546]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from ULAW to G729A
WARNING[278546]: File file.c, Line 170 (ast_stopstream): Unable to restore format back to 4
WARNING[278546]: File app_voicemail.c, Line 2707 (vm_execmain): Couldn't read username
== Spawn extension (sip, 86, 3) exited non-zero on 'SIP/TerenceParker-af02'

- I don't understand this as, surely, I have already enabled g729a and ulaw ... how can it complain that it can't transmit in that format, or that it can't find a path?

3. With the default settings (i.e. no allow OR disallow clause) normal IP to IP calls work fine. Calls to voicemail also works fine with no problems. However, PSTN calls through my Voicetronix card or calls routed through FWD fail to work. This is what happens when I dial out with my voicetronix card:

Executing Dial(SIP/TerenceParker-22f3, vpb/1-1/18501) in new stack
Read_channel ##  vpb/1-1: Setting record mode, bridge = 0
--  1-1 requested, got: [vpb/1-1]
--  Calling 1-1/18501 on vpb/1-1 
Read_channel  vpb/1-1 (state=0), res=0, bridge=1
Read_channel  vpb/1-1 (state=0), res=0, bridge=1
Read_channel  vpb/1-1 (state=0), res=0, bridge=1
Read_channel  vpb/1-1 (state=0), res=0, bridge=1
Read_channel  vpb/1-1 (state=0), res=0, bridge=1
--  VPB Calling 1-1/18501 [t=0] on vpb/1-1 returned 0
-- Called 1-1/18501
WARNING[278546]: File chan_sip.c, Line 1182 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4)
WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame
WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame
WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame
-- vpb/1-1 is ringing
WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame
Read_channel  vpb/1-1 (state=0), res=0, bridge=1
Read_channel ##  vpb/1-1: Setting record mode, bridge = 0
WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame
Read_channel  vpb/1-1 (state=5), res=0, bridge=1
WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame
Read_channel  vpb/1-1 (state=5), res=0, bridge=1
--  Event [12=>[00] Loop Drop
] on vpb/1-1
--  vpb/1-1 handle_owned got event: [12=>0]
--  handle_owned: putting frame: [-1=>0], bridge=(nil)
WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame
Read_channel  vpb/1-1 (state=5), res=0, bridge=1
WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame
Read_channel  vpb/1-1 (state=5), res=0, bridge=1
--  Event [102=>[00] Dial End
] on vpb/1-1
--  vpb/1-1 handle_owned got event: [102=>0]
--  handle_owned: putting frame: [4=>4], bridge=(nil)
-- vpb/1-1 answered SIP/TerenceParker-22f3
--  hangup on vpb (vpb/1-1)
Read_channel  vpb/1-1 (state=5), res=0, bridge=1
Read_channel  vpb/1-1 (state=6), res=-1, bridge=1
Read_channel  vpb/1-1 terminating, stopreads=1, owner=yes
--  Hungup on vpb/1-1 complete
== Spawn extension (sip, 918501, 1) exited non-zero on 'SIP/TerenceParker-22f3'

- again, it complains about codecs. So, at the moment, I am utterly confused!

Any help would be gratefully appreciated.

Terence



On 13 Jan 04, at 1:39 AM, Jorge Mendoza wrote:

Try in sip.conf:

disallow=all
allow=alaw
allow=ulaw
allow=gsm

(in that order)
I never tried with FWD

Jorge

[Asterisk-Users] RE: Fax

2004-01-14 Thread Reinhard Max

Hi,

On Tue, 13 Jan 2004 at 21:06, Jason Penton wrote:

 (I have successfully managed to receive faxes thru my isdn card so I
 don't see why I shouldn't be able to send them).

that's interesting, as in my tests it was just the other way around.

I can send faxes through my AVM ISDN card (chan_capi), but when I try
to receive a fax, app_rxfax fails after reporting some carrier
training errors. I've posted the detailed error logs to this list some
weeks ago.

Jason, are you using chan_capi, or chan_modem_i4l to access your ISDN
card?

cu
Reinhard

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[Asterisk-Users] always 4 rings before * answers!?

2004-01-14 Thread Ralf Illing









Hi all,



I am trying to figure out how to configure asterisk to
pick-up immediatly.

I already had a look on the wiki
and google the lists ... I was experimenting around a
lot but still my phone rings exactly 4 times (always) before * answers it.
Anybody has an idea - or maybe sth. I just wrapped?



My Hardware: TDM400P PCI FXS, X100P PCI FXO



extensions.conf

exten =
s,1,Answer

exten =
s,2,setmusiconhold,default

exten =
s,3,responsetimeout,10

exten =
s,4,DigitTimeout,5

exten =
s,5,Goto,language_menu|s|1



---zapata.conf

immediate=yes



Cheers

Ralf

Re: [Asterisk-Users] Why I can not use the conference

2004-01-14 Thread Jeremy McNamara

Zhang Peihao wrote:

Hi All,

The meetme.conf have created as below:
[rooms]
conf = 101
conf = 102

and extensions.conf as below:
exten = _1XX,1,MeetMe,${EXTEN}

why the warning printed when I called 101.
WARNING[27660]: File pbx.c, Line 1051 (pbx_extension_helper): No application 'MeetMe' 
for extension (ipcentrex, 101, 1)

And I found asterisk have not load the meetme.conf when it starts up.
  



You do have a Zaptel device in this box, right?


Jeremy McNamara





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Re: [Asterisk-Users] fw: problem with safe_asterisk

2004-01-14 Thread Matteo Brancaleoni

have you changed the line where is ask
to use another console (default tty9)
?

Il mar, 2004-01-13 alle 23:07, Pat Boyle ha scritto:
 I checked the log files in /var/log/asterisk
 
 There was nothing in there related to these errors. I think the script is
 ending after before asterisk even starts.
 Pat
 
 ---
 
 
 Karsten Wemheuer [EMAIL PROTECTED]
 Tue, 13 Jan 2004 08:34:31 +0100
 
   a.. Previous message: [Asterisk-Users] Fw: problem with safe_asterisk
   b.. Next message: [Asterisk-Users] MeetMe issues?
   c.. Messages sorted by: [ date ] [ thread ] [ subject ] [ author ]
 
 
 
 
 Hi,
 
 
 Pat Boyle wrote:
  I have no problems lauching asterisk from the command line  . . .
   asterisk -c
 
  However, I'm trying to autostart on boot up, so I'm trying safe_asterisk
 
  When I do this, I get:  Asterisk ended with exit status 127.  Asterisk
 died
  with code 127. Aborting.  I've looked through the script but can't find
 what
  the problem is.  I'm running on RedHat Fedora.
 
 Could You please have a look in the logfile. Maybe there are some
 information about the abort. I don't use Fedora but on Debian the log is
 under /var/log/asterisk/messages
 
 HTH,
 
 Karsten
 
 
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-- 
Matteo Brancaleoni
Espia System Administrator
Email : [EMAIL PROTECTED]
Web   : http://www.espia.it
Phone : +39 02 70633354  - ext 201
IAX(2): [EMAIL PROTECTED] - ext 201
Iaxtel: 1-700-56-62458   - ext 201

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Re: [Asterisk-Users] Why I can not use the conference

2004-01-14 Thread Matteo Brancaleoni

meetme requires zaptel

Il mer, 2004-01-14 alle 08:56, Zhang Peihao ha scritto:
 Hi All,
 
 The meetme.conf have created as below:
 [rooms]
 conf = 101
 conf = 102
 
 and extensions.conf as below:
 exten = _1XX,1,MeetMe,${EXTEN}
 
 why the warning printed when I called 101.
 WARNING[27660]: File pbx.c, Line 1051 (pbx_extension_helper): No application 
 'MeetMe' for extension (ipcentrex, 101, 1)
 
 And I found asterisk have not load the meetme.conf when it starts up.
 
 Zhang Peihao
 [EMAIL PROTECTED]
 2004-01-14
 
 
 
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Re: [Asterisk-Users] Re: Again: 7920 Cisco IP Phone Skinny SIP

2004-01-14 Thread Jan Czmok

Siggi Langauf ([EMAIL PROTECTED]) wrote:
 Hi Jan,
 
 first of all: please don't cross-post!
 
 On Tue, 13 Jan 2004, Jan Czmok wrote:
 
 [...]
   SKINNY OffHookMessage
   SKINNY SetSpeakerModeMessage
   SKINNY OnHookMessage
   SKINNY DisplayPromptStatusMessage
   SKINNY DisplayPromptStatusMessage
   SKINNY DisplayPromptStatusMessage
 
 It looks like chan_sccp is doing something at this pont that upsets the
 7920 so that it tries to fall back to SRST mode, before finally
 re-registering.

Okay, might be a reason. but what i saw on the display was:

- Registering to Callmanager
- Registered to Asterisk PBX
- Call ended.

(without hitting any button).

Since i work at an ISP, i probably could setup CME on one of my boxen
and you might be able to connect to it, so that we have packet traces to
follow up.

 That re-registration is rejected by chan_sccp, though, as
 the old connection is not closed, yet. So the 7920 gives up and tries to
 find another CallManager.

Possible reason, looks like that, however i am surprised why the 7920
reboots instead of just looking for another callmanager.

 Right now, Theo as well as Martin Bene are looking at the packet traces,
 so I'm sure the issue will be located and fixed soon.

i am also looking through various sources. I'd like to have the 7920
running !

 
  But if you look at the Support of the 7920 in Callmanager Express, you
  get a file named cmterm_7920.3.3-01-02-021.bin so i was investigating
  further. so i wrote cmterm_7920.3.3-01-02-021 in OS7920.TXT and
  suddenly the Cisco 7920 shows Upgrading Firmware :-)
  Unfortunately for some reason it did not accept the firmware, but it
  still tries to load it.
 
 There should also be a digitally signed version of that file
 (cmterm_7920.*.sbn), which the phone probably requires.

nope. no sbn. according to my cisco source the file is not signed.

 
  Some additional info:
  -
  The 7920 is requesting cmterm_7920.3.3-01-02-021^J.bin
  (so with an Ctrl-J in it), so you have to rename the file.
 
 You'd better remove the trailing Ctrl-J from OS7920.TXT, then (or stop
 using editors that insist on adding one).

i did, i use vi as editor. i was also surprised, but this is coming from
the phone ( i did extensive tethereal and tcpdump watching) :-)

  I also got the information from documents that the 7920 is running in
  7960 emulation mode, so draw your own conclusions in regards of SIP
  possiblity :-)
 
 Nope, there is a statement from Cisco that SIP support for the 7920 is not
 planned, ATM.
 7960 emulation mode refers only to being compatible with a 7960, as long
 as you do _not_ try to upload any firmware. (ie. Skinny-wise)
 However, that compatibility is not quite 100%...

Yep, right. 

  I tried to use some 7960 images, but did not succeed :-(
 
 Of course not, it's totally different hardware.

is it ? the cmterm image is nearly exactly 2 times the 7960 phone, so i
suspect one lower part of the image for the new functions and the rest
for the normal 79xx image. 

 
  Would appreciate some help in this issue :-)
 
 Just sit back and wait!

How can i help ? Just sit back isnt appropate for me :-)

 
 Meanwhile, you can register your 7920 with CallManager Express and connect
 that to asterisk via chan_oh323. (Note: chan_h323 will most likely not
 work, at least if you need two-way audio ;)
 
Hmm. Might be one way to use CME. Will see..

--jan



-- 
Jan Czmok, Network Engineering  Support, Global Access Telecomm, Inc.
Ph.: +49 69 299896-35 - fax: +49 69 299896-40 - sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] always 4 rings before * answers!?

Ralf Illing wrote:

Hi all,

 

I am trying to figure out how to configure asterisk to pick-up immediatly.

I already had a look on the wiki and google the lists ... I was 
experimenting around a lot but still my phone rings exactly 4 times 
(always) before * answers it. Anybody has an idea - or maybe sth. I 
just wrapped?

 

My Hardware: TDM400P PCI FXS, X100P PCI FXO

 

extensions.conf

exten = s,1,Answer

exten = s,2,setmusiconhold,default

exten = s,3,responsetimeout,10

exten = s,4,DigitTimeout,5

exten = s,5,Goto,language_menu|s|1

 

---zapata.conf

immediate=yes

 

Its something you are going to have to live with if you are using analog 
lines.. AFAIK Asterisk picks up the ringing by detecting a number of 
swings on the line.. It can be adjusted but if it is made any lower 
you end up getting lots of phantom calls where Asterisk will start to 
ring you extensions but there is not actually a call coming in..

If you switch to a digital line (BRI or PRI) then there is actually a 
signal sent to asterisk to tell it that a call is about to com in and so 
it will respond immediately..

Later..

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Re: [Asterisk-Users] Re: Voicepulse

2004-01-14 Thread Steven Critchfield

On Tue, 2004-01-13 at 19:42, Chris Albertson wrote:
 --- Steve Sobol [EMAIL PROTECTED] wrote:
  Matt Lawson wrote:
  
   I was just about to write the same thing.  It says busy.  Is is
  REALLY 
   busy or is something else wrong?
   
   This on the heels of switch-1.nufone.net being missing out of DNS.
   
   We have customers that expect their VOIP to work.  Is there anybody
  
   that's reliable?
 
 I've been doing some testing and so far I'm not 100% impressed
 by the VOIP services I've seen.  They provide a good service but
 my local phone company and ATT longdistance service is more
 reliable.

That would be a big DUH! Now the question comes down to choice and
price. 

 But this is not to say _you_ can't built a reliable VOIP based
 system.  Get _two_ providers and set up your dial plan in
 extensions.conf to fail over if one service fails to
 connect to dial via the next one and finally if both fail
 use pstn. your users will see a system the just works.
 
 About Nufone's problem.  I bet they'll start thinking about
 getting a backup DNS service and maybe geographic deversity.
 A company should be able to even stay on the air if there is a
 server room fire using techniques like round robin DNS and
 West cost and East coast servers run by different, unrelated
 hosting companies.  

Of course had you paid attention to the problem you would have been able
to understand that no DNS arrangement would fix having the root servers
modified by a registrar who screwed up. DNS servers don't work if your
whois doesn't point to the proper places. 
-- 
Steven Critchfield [EMAIL PROTECTED]

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[Asterisk-Users] newbie ISDN question

2004-01-14 Thread FastJack

hi everybody, sorry for posting such a stupid question ;)

i've managed to run asterisk* with my AVM fritz2.0 card and a some
VOIP-softphones (SIP, H323). the functions of asterisk* really satisfied me
;)))

now i want to run asterisk* istead of our old PBX. but it would be great to
connect some phones directly to my box. how does a E100P from digium work.
can i connect it to my ISDN-line and my internal phones (ISDN)?

it would look like this:

[PHONE2]
 /
[PC]-[E100P]  - [PHONE1]
 \
 [ISDN-LINE]

thank you for your help!!!
thorsten

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RE: [Asterisk-Users] RE: Fax

2004-01-14 Thread Jason Penton

Hi Reinhard

Hmmm very interesting. 

I am using chan_modem_i4l to access my gazel ISDN PCI cards. I must tell you
though that I have two fax machines the one sends perfectly and the other
fails (sounds similar to your problem wrt training errors). Not quite sure
where to go from here. I am going to listen to the line (as Steve suggested
in an earlier post) and will post my findings.

Good luck 
Jason

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Reinhard Max
 Sent: 14 January 2004 10:30 AM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] RE: Fax
 
 Hi,
 
 On Tue, 13 Jan 2004 at 21:06, Jason Penton wrote:
 
  (I have successfully managed to receive faxes thru my isdn 
 card so I 
  don't see why I shouldn't be able to send them).
 
 that's interesting, as in my tests it was just the other way around.
 
 I can send faxes through my AVM ISDN card (chan_capi), but 
 when I try to receive a fax, app_rxfax fails after reporting 
 some carrier training errors. I've posted the detailed error 
 logs to this list some weeks ago.
 
 Jason, are you using chan_capi, or chan_modem_i4l to access 
 your ISDN card?
 
 cu
   Reinhard
 
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Re: [Asterisk-Users] newbie ISDN question

Hi Thorsten,

the E100P is a PRI ISDN Card (S2M in Germany). You cannot connect
phones to that card.
The quadBRI card has 4 BRI ports that can individually be configured
for TE mode (to connect ISDN lines) or NT mode (to connect ISDN phones).
Please find the details at:

http://www.junghanns.net/asterisk/page17.html

best regards

kapejod
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/

 hi everybody, sorry for posting such a stupid question ;)

 i've managed to run asterisk* with my AVM fritz2.0 card and a some
 VOIP-softphones (SIP, H323). the functions of asterisk* really satisfied
 me ;)))

 now i want to run asterisk* istead of our old PBX. but it would be great
 to connect some phones directly to my box. how does a E100P from digium
 work. can i connect it to my ISDN-line and my internal phones (ISDN)?

 it would look like this:

 [PHONE2]
  /
 [PC]-[E100P]  - [PHONE1]
  \
  [ISDN-LINE]

 thank you for your help!!!
 thorsten

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AW: [Asterisk-Users] newbie ISDN question

2004-01-14 Thread Thomas Haeger

Hi Thorsten,

the E100P is an E1 Card with 30 Channels (PRI), this is not for connecting
Phones directly.
You can youse the TDM10-40B for Analogphones, or you can use the new BRI
Card from kapejod -- http://ns1.jnetdns.de/jn/relaunch/asterisk/page17.html
But the driver is alpha stadium ;-), or you can use VoIP Phones like
Grandstream BudgetTone 100 -- http://www.grandstream.com/y-product.htm


Best regards,

Thomas.

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von FastJack
Gesendet: Mittwoch, 14. Januar 2004 10:22
An: [EMAIL PROTECTED]
Betreff: [Asterisk-Users] newbie ISDN question


hi everybody, sorry for posting such a stupid question ;)

i've managed to run asterisk* with my AVM fritz2.0 card and a some
VOIP-softphones (SIP, H323). the functions of asterisk* really satisfied me
;)))

now i want to run asterisk* istead of our old PBX. but it would be great to
connect some phones directly to my box. how does a E100P from digium work.
can i connect it to my ISDN-line and my internal phones (ISDN)?

it would look like this:

[PHONE2]
 /
[PC]-[E100P]  - [PHONE1]
 \
 [ISDN-LINE]

thank you for your help!!!
thorsten

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Re: [Asterisk-Users] newbie ISDN question

2004-01-14 Thread Peer Oliver schmidt

Hello kapejod,

The quadBRI card has 4 BRI ports that can individually be configured
for TE mode (to connect ISDN lines) or NT mode (to connect ISDN phones).
Please find the details at:
http://www.junghanns.net/asterisk/page17.html
when are you going to release some pricing on the card? It just says 
But me!, but does not show you how... :)

rgds
pos
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Re: [Asterisk-Users] newbie ISDN question

The quadBRI card is EUR 600, excluding VAT.

best regards

kapejod

 Hello kapejod,

 The quadBRI card has 4 BRI ports that can individually be configured
 for TE mode (to connect ISDN lines) or NT mode (to connect ISDN
 phones). Please find the details at:

 http://www.junghanns.net/asterisk/page17.html

 when are you going to release some pricing on the card? It just says
 But me!, but does not show you how... :)

 rgds
 pos

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Re: [Asterisk-Users] newbie ISDN question

Peer Oliver schmidt wrote:

Hello kapejod,

The quadBRI card has 4 BRI ports that can individually be configured
for TE mode (to connect ISDN lines) or NT mode (to connect ISDN phones).
Please find the details at:
http://www.junghanns.net/asterisk/page17.html


when are you going to release some pricing on the card? It just says 
But me!, but does not show you how... :)

rgds
pos
Thats what I was going to ask as well.. so I will.. :)

How much does it cost??

Later..

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Re: [Asterisk-Users] newbie ISDN question

2004-01-14 Thread FastJack

hi klaus-peter,

thank you for your replay. btw: i am using you chan_capi already ;)) it
works great!!!
how many internel phones could be connected to this card?
how stable is the driver (can i use it for a production-system)?

sorry for all that stupid questions - i know linux and ip and pc-hardware
but telephone-technics are all new for me.

how long would delivery of that card take?

thanks (oder besser gesagt: VIELEN DANK ;)  )
thorsten

- Original Message -
From: Klaus-Peter Junghanns [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 14, 2004 11:54 AM
Subject: Re: [Asterisk-Users] newbie ISDN question


 Hi Thorsten,

 the E100P is a PRI ISDN Card (S2M in Germany). You cannot connect
 phones to that card.
 The quadBRI card has 4 BRI ports that can individually be configured
 for TE mode (to connect ISDN lines) or NT mode (to connect ISDN phones).
 Please find the details at:

 http://www.junghanns.net/asterisk/page17.html

 best regards

 kapejod
 --
 Klaus-Peter Junghanns

 CEO, CTO
 Junghanns.NET GmbH
 Breite Straße 13 - 12167 Berlin - Germany
 fon: (de) +49 30 79705390
 fon: (uk) +44 870 1244692
 fax: (de) +49 30 79705391
 iaxtel: 1-700-157-8753
 http://www.Junghanns.NET/asterisk/

  hi everybody, sorry for posting such a stupid question ;)
 
  i've managed to run asterisk* with my AVM fritz2.0 card and a some
  VOIP-softphones (SIP, H323). the functions of asterisk* really satisfied
  me ;)))
 
  now i want to run asterisk* istead of our old PBX. but it would be great
  to connect some phones directly to my box. how does a E100P from digium
  work. can i connect it to my ISDN-line and my internal phones (ISDN)?
 
  it would look like this:
 
  [PHONE2]
   /
  [PC]-[E100P]  - [PHONE1]
   \
   [ISDN-LINE]
 
  thank you for your help!!!
  thorsten
 
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Re: [Asterisk-Users] newbie ISDN question

Thorsten,

theoretically you can connect 8 phones per port, but only 2 can
be used at the same time. We advise to use 2 per port and in
some scenarios 3 might be an option. So you can connect 8 ISDN
phones to the quadBRI card.
The drivers are still released as experimental and have some
bugs. We are planning to be stable in about 2 weeks.

The cards are in stock, so delivery will be fast. We ship with
worldwide with UPS.

best regards

kapejod
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/

 hi klaus-peter,

 thank you for your replay. btw: i am using you chan_capi already ;)) it
 works great!!!
 how many internel phones could be connected to this card?
 how stable is the driver (can i use it for a production-system)?

 sorry for all that stupid questions - i know linux and ip and
 pc-hardware but telephone-technics are all new for me.

 how long would delivery of that card take?

 thanks (oder besser gesagt: VIELEN DANK ;)  )
 thorsten

 - Original Message -
 From: Klaus-Peter Junghanns [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Wednesday, January 14, 2004 11:54 AM
 Subject: Re: [Asterisk-Users] newbie ISDN question


 Hi Thorsten,

 the E100P is a PRI ISDN Card (S2M in Germany). You cannot connect
 phones to that card.
 The quadBRI card has 4 BRI ports that can individually be configured
 for TE mode (to connect ISDN lines) or NT mode (to connect ISDN
 phones). Please find the details at:

 http://www.junghanns.net/asterisk/page17.html

 best regards

 kapejod
 --
 Klaus-Peter Junghanns

 CEO, CTO
 Junghanns.NET GmbH
 Breite Straße 13 - 12167 Berlin - Germany
 fon: (de) +49 30 79705390
 fon: (uk) +44 870 1244692
 fax: (de) +49 30 79705391
 iaxtel: 1-700-157-8753
 http://www.Junghanns.NET/asterisk/

  hi everybody, sorry for posting such a stupid question ;)
 
  i've managed to run asterisk* with my AVM fritz2.0 card and a some
 VOIP-softphones (SIP, H323). the functions of asterisk* really
 satisfied me ;)))
 
  now i want to run asterisk* istead of our old PBX. but it would be
 great to connect some phones directly to my box. how does a E100P
 from digium work. can i connect it to my ISDN-line and my internal
 phones (ISDN)?
 
  it would look like this:
 
  [PHONE2]
   /
  [PC]-[E100P]  - [PHONE1]
   \
   [ISDN-LINE]
 
  thank you for your help!!!
  thorsten
 
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[Asterisk-Users] How to park and pickup a call

Hi All,

How to park and pickup a call? The scenario of park and pickup
described as below.

UserA made a call to UserB, and the call ware connected,
Then UserB parked (or hold) the call, and told UserC to pickup
the call on one line, and then, UserC pressed some keys to 
pickup the call.

Who can tell me what's the Park/Pickup's typical flow in
the Asterisk.  And how to set the sip.conf, extensions.conf 
and parking.conf to implement it.

Zhang Peihao
[EMAIL PROTECTED]
2004-01-14


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Re: [Asterisk-Users] T1 Sync clarification

2004-01-14 Thread Steve Underwood

Stephen Davies wrote:

On Wed, 14 Jan 2004, TC wrote:

 

What are the practical effects with in-correct clock sync
-like to you hear odd buzzing, or dropped voice or gaps of audio ??
   

Old-fart anecdote about this - in the early 80s we had some 1200bps
modems that we used to connect to client sites.  When our phone
company went digital we suddenly started getting a } character at a
regular interval of 10 or 15 seconds.
This turned out to be clock slips in the new digital trunk between the
two exchanges.
So there is one effect of clock slips.

Steve
 

That must have been an FSK modem. Most advanced modems completely loose 
sync on the first sample slip. The sample slip causes a jump in phase, 
and phase is critical to the correct operation of most modems.

Regards,
Steve
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[Asterisk-Users] Asterisk drops calls - E100P

2004-01-14 Thread Daniel Bichara

Hi,

Once a day, * drops all calls (E100P board). Yesterday, I updated * 
version to CVS but I got the problem again today. Monitoring log files, 
I found this messages just before:

Jan 14 09:14:15 WARNING[81926]: File chan_zap.c, Line 5758 
(zt_pri_error): PRI: Read on 25 failed: Unknown error 500
Jan 14 09:14:15 WARNING[81926]: File chan_zap.c, Line 5758 
(zt_pri_error): PRI: Short write: -1/5 (Unknown error 500)
Jan 14 09:14:15 WARNING[81926]: File chan_zap.c, Line 5758 
(zt_pri_error): PRI: Read on 25 failed: Unknown error 500

Few minutes after this, everything becomes fine. Any clue?

Daniel



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Re: [Asterisk-Users] T1 Sync clarification

2004-01-14 Thread Steve Underwood

Rich Adamson wrote:

To complete this rather lengthy topic... what happens if you ignore all of
this and just slap a bunch of systems together with no regard to a master
sync source?  The quality and stability of your network will likely not be
as good as what it could be. If your clocks (in each device) happen to be
running very very close to what is expected, your network might run just
fine. But, if one of the clock's frequency drifts around, it could impact
quality via frame slippage and other unwanted events, and if off by a
large amount could even be the source of failures. (Your milage will vary
directly with the stability of your clocks.)
 

 

What are the practical effects with in-correct clock sync
-like to you hear odd buzzing, or dropped voice or gaps of audio ??
   

As mentioned earlier, it depends entirely upon how far off one clock is
from the clock at the other end of the T1.
If they are off by a little bit, you would see frame slips but probably
not hear any quality differences.
As the slip rate increases (to some unknown value since I've not tried
personally to qualify this), the audio would be infrequently interrupted
from the lost frames. I would expect you to hear it as repetitive clicks 
of some sort that might be construed as noise. The exact noise would again 
depending upon how far off the clocks really were. Each audio channel 
consists of 8,000 voice samples per second (on a normal US T1), so if the 
slip occurred once/second on average and then recovered, one would probably 
not hear 1/8000 second of a hickup.

If the slips were 100/sec average, it's likely the end nodes would have
a hard time recovering from it (best guess), and I would expect noise to
be apparent.
Others that have more experience correlating slip rates to noise levels
might have a better description of the noise vs slip rate.
Rich
 

You would only have a fast slip rate if something is faulty. Anything 
complying with the E1 or T1 specs should never have its clock 50ppm in 
error. Anything coming from the PSTN is essentially bang on, as it comes 
from an atomic clock.

Some people have commented about potentially difference clock rates from 
different providers. In practice that doesn't happen. Providers have a 
rhubidium clock in each exchange. These are so accurate, frame slips 
would be a one a year event. However, phase locking between carriers 
usually ensures even that does not occur. The globe's phone systems are 
pretty much all locked together these days.

The older higher order digital links - 8, 34, 140, and 565M in E1 land, 
and DS3 etc. in T1 land - have a bit stuffing scheme that allows 
individual E1s or T1s to be at slightly different rates. This is called 
PDH - plesiochronous (almost synchronous) digital heirarchy -  and was 
very helpful in moving from a totally analogue network to a mixed 
analogue/digital one. Once the network backbones became 100% digital, 
this became a huge liability. SDH (synchronous digital heirarchy, or 
Sonet) was born to solve this. SDH assumes the entire network is 
perfectly synchronised. Drop and insert is *far* cheaper in a truely 
synchronous stream. SDH is the norm for anything new today.

Regards,
Steve
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Re: [Asterisk-Users] How to park and pickup a call

Zhang Peihao wrote:

Hi All,

How to park and pickup a call? The scenario of park and pickup
described as below.

UserA made a call to UserB, and the call ware connected,
Then UserB parked (or hold) the call, and told UserC to pickup
the call on one line, and then, UserC pressed some keys to 
pickup the call.

Who can tell me what's the Park/Pickup's typical flow in
the Asterisk.  And how to set the sip.conf, extensions.conf 
and parking.conf to implement it.

  

AFAIK call parking does not work with a number of SIP UA's because the
call is transferred into the parking location and then the SIP UA
terminates the call so you will not hear which location the call was
parked to..

You may be able to setup a workaround to the problem using the t
option in your extension configuration and then use the # key to perform
the transfer instead of the SIP transfer key on your phone..

later..

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Re: [Asterisk-Users] How can I get support about Dialogic hardware

2004-01-14 Thread Alastair Maw

On 14/01/04 02:23,  wrote:
 I know that Dialogic hardware is not supported by standard Asterisk, if 
 I want use it ,I must pay for it. But I don't know how to get these 
 pathes and what kind of board is suppoted by Asterisk.

http://www.mail-archive.com/[EMAIL PROTECTED]/msg01563.html

Any Digium people out there - it'd be useful if this information was
available on the asterisk.org web site somewhere.

Regards,

Alastair Maw
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Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

have u had any luck with this?

cm
- Original Message - 
From: Owen Kelso [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 12, 2004 9:51 AM
Subject: Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

 Thanks for all of your responses.

 I confirmed that the phone works perfectly without NAT or through a IPSec
 VPN (yeah, I know, same thing).

 I've concluded that the Netgear router (FVS318) performing the NAT is
 corrupting the outgoing RTP packets.  Traces confirmed that the BudgeTone
 is sending them out with a UDP checksum of 0 but the next hop after the
 Netgear router they are set to a non-zero value (an incorrect one). 
 Asterisk is never even seeing the packets because the kernel is
 recognizing them as corrupt and dropping them, hence the recvfrom()
 Resource temporarily unavailable errors in rtp.c.

 I'm going to write Netgear to see what they have to say about it.  If I
 make any progress I'll post to the list...thanks again, Owen
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RE: [Asterisk-Users] Asterisk on FreeBSD 4.9?

2004-01-14 Thread Troy Settle


I'm about to post on bugs to offer a bounty for work on FreeBSD.  I'm
fairly certain that others will come along to increase that bounty.

Before I do post it, I would like some input on what the requirements
should be.  Here's what I have so far:

 - Must be completed before 6/30/04
 - Support for all Zaptel hardware
 - Commitment of the drivers to both
   4-STABLE and 5-CURRENT/STABLE

I'm not completely conversant on how GPL software can be committed to
the kernel, but I believe it can be done under the contrib/ directory.

I do not want this work to exist as a series of
downloads/checkouts/patches/modules if it can be avoided.  I don't want
to patch my kernel or load modules.  I want to be able to do a cvsup on
/usr/src, add necessary device entries to my kernel config file and
build it.

I'd like to see astersk and libpri installs follow the reccomendations
and requirements found in the FreeBSD hier(1) man page.  Specifically,
it should install completely to /usr/local/.  Preferrably, I'd like to
see a port created for both asterisk and libpri, even just a metaport
that uses CVS to fetch the source and any OS-specific patches.

Any comments before I post the bounty?  I will recommend that those with
suggestions on the requirements and those that offer additional bounties
for this will sit in committee to determine when the requirements of the
bounty have been met.

--
  Troy Settle
  Pulaski Networks
  http://www.psknet.com
  866.477.5638


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 [EMAIL PROTECTED]
 Sent: Tuesday, January 13, 2004 8:27 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Asterisk on FreeBSD 4.9?
 
 
 On Tue, Jan 13, 2004 at 12:24:20PM -0500, Jason T. Nelson wrote:
 
  love to be able to use Asterisk under FreeBSD. I've browsed 
 the archives
  and perceived what appears to be a slightly hostile 
 attitude towards those
  who ask about Asterisk support of other free operating 
 systems even without
  using Digium hardware. Is this Linux-specific bias 
 intentional or accidental?
 
 I would call it historical. Asterisk was first developed on Linux,
 and little attention was paid to portability. This is changing,
 though there are still Linuxisms in the code. I would hesitate
 to consider it stable yet on anything other than Linux, but
 YMMV.
 
 I personally would like to see Asterisk portable to any
 *nix with pthreads, and am working to make this happen. As
 always help in the form of patches, testing or accounts for
 building and testing on less common types of systems are
 appreciated.
 
 -w
 -- 
 /~\  The ASCII Ribbon Campaign
 \ /No HTML/RTF in email
  X No Word docs in email
 / \  Respect for open standards
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[Asterisk-Users] Re: newbie ISDN question

2004-01-14 Thread Reinhard Max

Hi,

On Wed, 14 Jan 2004 at 12:15, Klaus-Peter Junghanns wrote:

 The quadBRI card is EUR 600, excluding VAT.

this looks like a great piece of hardware, but I think it's too
expensive for home users like me who wouldn't really need more than
one or two BRI ports.

So do you have any plans for a singleBRI or doubleBRI version of this
card, or maybe even a variant that comes with a single port
preinstalled and three more ports can be added as needed via
daughterboards like on the TDM400P?

cu
Reinhard

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[Asterisk-Users] grandstream asterisk configuration




hi, I have the following configuration: 
Grandstream -- NAT (Netgear RP614)--Internet--Asterisk(public 
IP) i can register fine and call ringing is working as good. The 
problem is =i cant hear audio both ways and i get this 
error: WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP 
Read error:Resource temporarily unavailable my sip.conf 
file is as follows: [general]port =3D 
5060 
; Port to bind tobindaddr =3D 
0.0.0.0 
; Address to bind to;externip =3D 
200.201.202.203 ; Address that we're going to put in 
=SIPmessages if we're behind a 
NATtos=3Dlowdelaydisallow=3Dall 
; Disallow all 
codecsallow=3Dulaw 
; Allow codecs in order of preference 
dtmfmode=3Dinfo 
[grandstream1]type=3Dfriendhost=3Ddynamicsecret=3Dmysecretcontext=3Doutgoingnat=3Dyesreinvite=3Dnocanreinvite=3Dnoqualify=3D2000 
has anyone done this before? chandra

Re: [Asterisk-Users] grandstream asterisk configuration

2004-01-14 Thread bam

Make sure that udp packets can get from the server back to the grandstream.

At 12:40 14/01/04, you wrote:
 hi,

I have the following configuration:

Grandstream -- NAT (Netgear RP614)--Internet--Asterisk(public IP)

i can register fine and call ringing is working as good. The problem is =
 i cant hear audio both ways and i get this error:
WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error:
 Resource temporarily unavailable
my sip.conf file is as follows:

[general]
 port =3D 5060 ; Port to bind to
 bindaddr =3D 0.0.0.0  ; Address to bind to
 ;externip =3D 200.201.202.203 ; Address that we're going to put in =
 SIP
 messages if we're behind a NAT
 tos=3Dlowdelay
 disallow=3Dall; Disallow all codecs
 allow=3Dulaw  ; Allow codecs in order of preference
dtmfmode=3Dinfo

[grandstream1]
 type=3Dfriend
 host=3Ddynamic
 secret=3Dmysecret
 context=3Doutgoing
 nat=3Dyes
 reinvite=3Dno
 canreinvite=3Dno
 qualify=3D2000
has anyone done this before?

chandra


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Re: [Asterisk-Users] T1 Sync clarification

  What are the practical effects with in-correct clock sync
  -like to you hear odd buzzing, or dropped voice or gaps of audio ??
 
 You may get gaps where frames are discarded, this will be across all timeslots 
 so an individual loss isnt a lot of data, you'll probably get away with the odd 
 one but if you get too many and the T1 realigns it could restart and you could 
 see the whole T1 go down and up..
 
 Not sure how this works in the US with such diversity available but in the UK 
 telcos generally derive sync from another one so most of them are on the same 
 clock source..

It's the same in the US, however in the US there are far more independent
telcos (example, Iowa had the distinction of the most independent telcos 
at 600+ of all states) and many of those do not have an engineering staff 
nor the expertise to address this. Their engineering is typically farmed 
out to either the central office switch vendor or to independent engineering 
firm(s) when needed. Those groups should have addressed it, but in at least 
some cases it was not. 

The US also has some carriers that got into the national and/or international
long distance business with a low budget staff that ran hard but never
documented anything. (I've done some consulting work for two of those and
wouldn't bet a dollar on their attention to detail.)

Rich


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Re: Re: [Asterisk-Users] How to park and pickup a call

Hi WipeOut,

Can you give me a sample? 

Zhang Peihao
2004-01-14

- Original Message - 
From: WipeOut [EMAIL PROTECTED] 
To: asterisk-users [EMAIL PROTECTED] 
Sent: 2004-01-14 11:59:00 
Subject: Re: Re: [Asterisk-Users] How to park and pickup a call

Zhang Peihao wrote:

Hi All,

How to park and pickup a call? The scenario of park and pickup
described as below.

UserA made a call to UserB, and the call ware connected,
Then UserB parked (or hold) the call, and told UserC to pickup
the call on one line, and then, UserC pressed some keys to 
pickup the call.

Who can tell me what's the Park/Pickup's typical flow in
the Asterisk.  And how to set the sip.conf, extensions.conf 
and parking.conf to implement it.

AFAIK call parking does not work with a number of SIP UA's because the
call is transferred into the parking location and then the SIP UA
terminates the call so you will not hear which location the call was
parked to..

You may be able to setup a workaround to the problem using the t
option in your extension configuration and then use the # key to perform
the transfer instead of the SIP transfer key on your phone..

later..

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Re: Re: [Asterisk-Users] How to park and pickup a call

Hi WipeOut,

Can you give me a sample? 

Zhang Peihao
2004-01-14

- Original Message - 
From: WipeOut [EMAIL PROTECTED] 
To: asterisk-users [EMAIL PROTECTED] 
Sent: 2004-01-14 11:59:00 
Subject: Re: Re: [Asterisk-Users] How to park and pickup a call

Zhang Peihao wrote:

Hi All,

How to park and pickup a call? The scenario of park and pickup
described as below.

UserA made a call to UserB, and the call ware connected,
Then UserB parked (or hold) the call, and told UserC to pickup
the call on one line, and then, UserC pressed some keys to 
pickup the call.

Who can tell me what's the Park/Pickup's typical flow in
the Asterisk.  And how to set the sip.conf, extensions.conf 
and parking.conf to implement it.

AFAIK call parking does not work with a number of SIP UA's because the
call is transferred into the parking location and then the SIP UA
terminates the call so you will not hear which location the call was
parked to..

You may be able to setup a workaround to the problem using the t
option in your extension configuration and then use the # key to perform
the transfer instead of the SIP transfer key on your phone..

later..

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Re: [Asterisk-Users] T1 Sync clarification

2004-01-14 Thread Stephen Davies



On Wed, 14 Jan 2004, Steve Underwood wrote:

 That must have been an FSK modem. Most advanced modems completely loose 
 sync on the first sample slip. The sample slip causes a jump in phase, 
 and phase is critical to the correct operation of most modems.

It was V.22.  No error correction or anything new-fangled like that.  
(Not auto dial either).

Steve



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Re: [Asterisk-Users] Re: newbie ISDN question

Hi,

yes, for the home user it's still too expensive. Although it's
really cheap if you compare it to other 4 BRI cards on the market.

Currently i am polishing the driver for the hfc-s pci a chipset,
which i used in numerous el-cheapo ISDN cards (street price around
30 EUR). This will bring zaptel BRI (and even NT mode) to the
home user. :)

best regards

kapejod
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


 Hi,

 On Wed, 14 Jan 2004 at 12:15, Klaus-Peter Junghanns wrote:

 The quadBRI card is EUR 600, excluding VAT.

 this looks like a great piece of hardware, but I think it's too
 expensive for home users like me who wouldn't really need more than one
 or two BRI ports.

 So do you have any plans for a singleBRI or doubleBRI version of this
 card, or maybe even a variant that comes with a single port
 preinstalled and three more ports can be added as needed via
 daughterboards like on the TDM400P?

 cu
   Reinhard

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Re: [Asterisk-Users] T1 Sync clarification

2004-01-14 Thread Steve Underwood

Rich Adamson wrote:

What are the practical effects with in-correct clock sync
-like to you hear odd buzzing, or dropped voice or gaps of audio ??
 

You may get gaps where frames are discarded, this will be across all timeslots 
so an individual loss isnt a lot of data, you'll probably get away with the odd 
one but if you get too many and the T1 realigns it could restart and you could 
see the whole T1 go down and up..

Not sure how this works in the US with such diversity available but in the UK 
telcos generally derive sync from another one so most of them are on the same 
clock source..
   

It's the same in the US, however in the US there are far more independent
telcos (example, Iowa had the distinction of the most independent telcos 
at 600+ of all states) and many of those do not have an engineering staff 
nor the expertise to address this. Their engineering is typically farmed 
out to either the central office switch vendor or to independent engineering 
firm(s) when needed. Those groups should have addressed it, but in at least 
some cases it was not. 

The US also has some carriers that got into the national and/or international
long distance business with a low budget staff that ran hard but never
documented anything. (I've done some consulting work for two of those and
wouldn't bet a dollar on their attention to detail.)
Rich
 

If they have frame slips too often FAX will not work. It would be hard 
for even the most incompetant telco to ignore that. However, their core 
equipment is likely to use a rhubidium clock and keep everything OK, 
even if they done sync to their peers properly.

Regards,
Steve
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Re: [Asterisk-Users] Asterisk on FreeBSD 4.9?

2004-01-14 Thread Olle E. Johansson

Troy Settle wrote:
 - Must be completed before 6/30/04
 - Support for all Zaptel hardware
 - Commitment of the drivers to both
   4-STABLE and 5-CURRENT/STABLE
...Fix the bug in pbx_wilcalu that makes Asterisk/FreeBSD eat 99.99% of the CPU

Maybe add a zap_rtc module for non-zaptel hardware.

Otherwise it works fine on FreeBSD. I'm running several Asterisk/FreeBSD servers, VoIP only.

/O

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Re: [Asterisk-Users] SIP and AGI crash...

2004-01-14 Thread Philipp von Klitzing

Hi!

I'm trying to use the say-ani agi asterisk-perl script and am experiencing
  crashes, I am also experienceing problems with the test-agi scripts shipped
  with asterisk.

 Looks like the AGI script quoted above doesn't detect the hangup
 condition and exit.
 
 Is asterisk exiting on you after the quoted section? From what is quoted
 it doesn't look like a crash, but an over simplistic example app that
 didn't take hangup into account.

Note: There is a bug (in the bug tracker, bug has been closed) known with 
Grandstream phones that shows when using agi-test.agi. So do use a 
different SIP device for testing. See:
http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone

Cheers, Philipp


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Re: [Asterisk-Users] How to park and pickup a call

Zhang Peihao wrote:

Hi WipeOut,

Can you give me a sample? 

Zhang Peihao
2004-01-14
  

In your extensions.conf you can try using the t option on the Dial
app.. so somthing like this..

exten = 1234,1,Dial(SIP/1234,10,t)

Where 10 is the timeout and t is the option to enable the # key to
perform a transfer ( you may want to add T as well)..

From the CLI..

dev02*CLI show application Dial
-= Info about application 'Dial' =-

[-snip-]
't' -- allow the called user transfer the calling user
'T' -- to allow the calling user to transfer the call.
[-snip-]

Remember to include parkedcalls in the appropiate context..

later


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Re: [Asterisk-Users] grandstream asterisk configuration

i have forwarded ports 5060 and 5000-5008 the ports used by sip and rtp to
grandstream from my netgear. rtp.conf uses rtpstart 5000 and rtpend 5008. i
have also opened all 5060, 5000-5008 ports in my firewall configuration.
grandstream uses 5004 port for rtp.

what am i missing here? please tell me.

chandra

- Original Message -
From: bam [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 14, 2004 6:42 PM
Subject: Re: [Asterisk-Users] grandstream asterisk configuration



 Make sure that udp packets can get from the server back to the
grandstream.


 At 12:40 14/01/04, you wrote:
   hi,
 
 I have the following configuration:
 
 Grandstream -- NAT (Netgear RP614)--Internet--Asterisk(public IP)
 
 i can register fine and call ringing is working as good. The problem is =
   i cant hear audio both ways and i get this error:
 
 WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error:
   Resource temporarily unavailable
 
 my sip.conf file is as follows:
 
 [general]
   port =3D 5060 ; Port to bind to
   bindaddr =3D 0.0.0.0  ; Address to bind to
   ;externip =3D 200.201.202.203 ; Address that we're going to put in
=
   SIP
   messages if we're behind a NAT
   tos=3Dlowdelay
   disallow=3Dall; Disallow all codecs
   allow=3Dulaw  ; Allow codecs in order of preference
 
 dtmfmode=3Dinfo
 
 [grandstream1]
   type=3Dfriend
   host=3Ddynamic
   secret=3Dmysecret
   context=3Doutgoing
   nat=3Dyes
   reinvite=3Dno
   canreinvite=3Dno
   qualify=3D2000
 
 has anyone done this before?
 
 chandra


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Re: [Asterisk-Users] hardware requirements of asterisk

 I have been playing with 2 Asterisk boxes for testing purposes, it has 
 been going very well. The 2 boxes are PII celeron 400 (HP Deskpro) with 
 sound cards and lan. I have iax connecting the 2 boxes.
 
 For making cals and testing out recorded message for 1 connection it was 
   working quite well. However, when I stressed it a bit with 2 users 
 making calls, we started to here voice degradation and cracking noises. 
 However, top shows cpu is 94% idle. I am suspecting the network. However 
 it is 100M switch and I have not had any clue. I suppose it should at 
 least be able to handle 10 calls similtaneously for even a small office. 
 So what is the recommended spec for 5 users or 10 users?

Without any other factual detail, best guess is half vs full duplex problem
on one or more of the devices (phones, PC, etc).

Assuming you're using sip phones for testing (and we really don't even know
that for sure) and depending upon exactly what parameters you've applied
for each sip phone definition within asterisk, calls between phones are set 
up by asterisk. Asterisk then instructs the two phones to communicate between 
themselves, and bows out of the audio session. So, if you really are using
two sip phones, then you have a networking problem between those two
devices and not with asterisk.

For anyone to offer suggestions, you really need to provide more facts.


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[Asterisk-Users] TDM switching between Digium TE410P ports

2004-01-14 Thread Jan Baumann

Hi,

I am currently thinking of a new asterisk server equiped with a Digium 
TE410P card and want to

- terminate two E1 PRIs from different carriers with different prefixes
  and possibly clocking (+49 xxx abcde-yy and +49 xxx fghij-yy).
- TDM switch data and modem calls arriving on the PRI trunks for
  ext. 9xx to the third E1 port of the Digium to a Cisco 3640 or AS5300
  access-server for ISDN/modem PPP dialup termination while retaining
  the dialed number.
- use the remaining ext. -100 to -899 for SIP phones, FXS ports, etc.
  with outbound least cost routing between the trunks based on dialed
  number and time of day (should be no problem)
I would very much appreciate to know if someone of you has done these 
kind of things already, especially switching ISDN data and modem calls 
from a CPE-side PRI to a Network-side PRI port on the same Digium 
TE410P. Switchtype for all ports will be NET5/euroISDN.

Many Thanks,
Jan Baumann
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Re: [Asterisk-Users] GrandStream Budgetone Phone DHCP General Observations

2004-01-14 Thread James Sizemore

I'm interested.

TeleSIP wrote:

I'll try to hack a NAT friendly tftp server on monday.
   

Are you still looking for it?  I found one if you need it.  Let me know and
I will post the info.
Andres.

 

--
Nicolas Bougues
Axialys Interactive
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Re: [Asterisk-Users] T1 Sync clarification

At 9:17 PM +0800 1/14/04, Steve Underwood wrote:
[snip]
It's the same in the US, however in the US there are far more independent
telcos (example, Iowa had the distinction of the most independent 
telcos at 600+ of all states) and many of those do not have an 
engineering staff nor the expertise to address this. Their 
engineering is typically farmed out to either the central office 
switch vendor or to independent engineering firm(s) when needed. 
Those groups should have addressed it, but in at least some cases 
it was not.
The US also has some carriers that got into the national and/or international
long distance business with a low budget staff that ran hard but never
documented anything. (I've done some consulting work for two of those and
wouldn't bet a dollar on their attention to detail.)

Rich

If they have frame slips too often FAX will not work. It would be 
hard for even the most incompetant telco to ignore that. However, 
their core equipment is likely to use a rhubidium clock and keep 
everything OK, even if they done sync to their peers properly.

Regards,
Steve
This is getting pretty far off the topic of Asterisk, but I'll 
confirm that several of the small CLECS that I've worked 
for/consulted for do _not_ have their own timing sources in the form 
of a rubidium standard.  These also are carriers that sell PRI's and 
T1 connections out of their switching equipment.  They typically use 
clocking source from one of their interconnect providers, or they 
simply don't know the answer to the question of who provides clock 
in your network?

If they're taking sync off one of their upstreams, this is not so 
bad.  If they simply don't know where they're getting sync, this is 
much worse.  If my experiences have been this poor in what I think 
are fairly dense population/financially wealthy areas, I can only 
imagine what it's like as one moves further away from high-budget 
telephony centers.  A scrupulous tech will fix those problems... but 
there are a dwindling number of scrupulous techs, and an even shorter 
supply of money for rubidium standards or cesium beam timepieces.

In other words: I suspect a great number of Asterisk users, being 
(sometimes) budget conscious,  will run across these types of shady 
clocking situations since the lowest budget carriers often don't have 
the funding to implement the right solutions.

JT
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Re: [Asterisk-Users] Re: Proposed solution for exit code priority jumps

 With that in mind, I'm going to do something I only infrequently do, 
 which is to re-post something in it's entirety and look for comments 
 again instead of just posting the URL.  I'm getting very tired of the 
 current jump-on-error method of priority control and error handling, 
 and I think it's time for something a little more meaningful and 
 robust.  Now that there will soon be the concept of unstable code, 
 I think large ideas like this might see the light of day in the near 
 future.
 
 I realize this is a major code shift, since it would require work in 
 pretty much every single application.  It's easy for me to talk about 
 this since I can't accomplish such a task.  However, without 
 attention now, this may never be solved, which would be a pity 
 because it's a real crimp in the style of anyone doing more than 
 trivial dialplan manipulation - anyone doing cascading dial failovers 
 will attest to that.
 
 Please look at and comment upon method #1 logic and syntax shown 
 below; I think method #2 (alternate method) is a bit too radical.

John,

Absolutely no disrespect intended or implied (to you or anyone)...

I'd vote for method #1, however...

We both know there are probably less then a dozen active developers that
have the knowledge/exerience to address it, and their plates are obviously
very full. The rest of us don't have the skills to participate in the
development even if we could devote the time. Lobbying them directly 
with some form of architectural change document and obtaining their buy-
in might see some results over time.

Since there seems to be more then one topic like this lurking, it would
appear a more formal forum for such proposals would be a Good Thing.
Maybe that could be tied in some how with the movement towards cycling
stable releases.

Rich


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RE: [Asterisk-Users] Asterisk on FreeBSD 4.9?

I'm about to post on bugs to offer a bounty for work on FreeBSD.  I'm
fairly certain that others will come along to increase that bounty.
Before I do post it, I would like some input on what the requirements
should be.  Here's what I have so far:
 - Must be completed before 6/30/04
 - Support for all Zaptel hardware
 - Commitment of the drivers to both
   4-STABLE and 5-CURRENT/STABLE
I'm not completely conversant on how GPL software can be committed to
the kernel, but I believe it can be done under the contrib/ directory.
I do not want this work to exist as a series of
downloads/checkouts/patches/modules if it can be avoided.  I don't want
to patch my kernel or load modules.  I want to be able to do a cvsup on
/usr/src, add necessary device entries to my kernel config file and
build it.
I'd like to see astersk and libpri installs follow the reccomendations
and requirements found in the FreeBSD hier(1) man page.  Specifically,
it should install completely to /usr/local/.  Preferrably, I'd like to
see a port created for both asterisk and libpri, even just a metaport
that uses CVS to fetch the source and any OS-specific patches.
Any comments before I post the bounty?  I will recommend that those with
suggestions on the requirements and those that offer additional bounties
for this will sit in committee to determine when the requirements of the
bounty have been met.
--
  Troy Settle
  Pulaski Networks
  http://www.psknet.com
  866.477.5638
[snip]

Troy -
  While it is not 100% relevant to your requests, I'd like to see 
continued support of NetBSD/OpenBSD in this same vein and added to 
the bounty, since the additional work to get things correctly 
functioning on those two systems seems to be fairly minor while the 
hood is open.  MacOS is a different animal, and (IMHO) lower on the 
must-have list when it comes to Zap device support, though it would 
still be cool.

  If OpenBSD (1st choice) and NetBSD (2nd choice) can be added for 
Zap device support, count me in on the bounty.  Talk to me privately 
if you want to get a dollar figure.  I've had * running on OpenBSD, 
but of course no Zap hardware.  I'd move everything over to OpenBSD 
if it supported Zap, since that's my primary OS for all the platforms 
in my network.  While Linux in it's various flavors is great, it's 
simply not what my network runs, and so my * boxes are the odd man 
out systems, which makes me somewhat uncomfortable from a security 
and management perspective.

  Additionally, if files are to be installed in /usr/local, then I'd 
like to see the configs remain in /etc/asterisk since on my systems 
(and many other people's) the /usr/ directories are for binaries 
only; no configurations or moving parts so those directories can be 
mounted read-only or mounted from a common server if necessary.  I'm 
sure this is what you meant, but I've seen config directories 
unwisely located in /usr/local before, and I wanted to make sure 
everyone is of the same mind where that is concerned.

JT



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[Asterisk-Users] Re: newbie ISDN question

2004-01-14 Thread Reinhard Max

Hi,

On Wed, 14 Jan 2004 at 15:11, Klaus-Peter Junghanns wrote:

 Currently i am polishing the driver for the hfc-s pci a chipset,
 which i used in numerous el-cheapo ISDN cards (street price around
 30 EUR).

ah - that's much closer to the home user's typical budget :)

Is there a list of cards that use this chipset somewhere on the 'net?
I've googled for it, but most pages only talk about cards based on
the HFC-S chipset without listing brand and model names.

 This will bring zaptel BRI (and even NT mode) to the home user. :)

Cool, NT mode is exactly what I am looking for :)

Thanks for the great work you are doing to bring Asterisk to the ISDN
world!

cu
Reinhard

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Re: [Asterisk-Users] Asterisk on FreeBSD 4.9?

2004-01-14 Thread nanog

I'll put up some more $$ along with the others to continue the work on the
FreeBSD part. I'll be able to test this out with my network and well donate
freebsd server on my network for Asterisk Developers to continue the work on
this. Just email me off list for details

-Frankie Gravato - [EMAIL PROTECTED]
Senior Systems and Network Guru
Slingo Inc - www.slingo.com
 First in Fun 


- Original Message - 
From: John Todd [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 14, 2004 9:22 AM
Subject: RE: [Asterisk-Users] Asterisk on FreeBSD 4.9?


 I'm about to post on bugs to offer a bounty for work on FreeBSD.  I'm
 fairly certain that others will come along to increase that bounty.
 
 Before I do post it, I would like some input on what the requirements
 should be.  Here's what I have so far:
 
   - Must be completed before 6/30/04
   - Support for all Zaptel hardware
   - Commitment of the drivers to both
 4-STABLE and 5-CURRENT/STABLE
 
 I'm not completely conversant on how GPL software can be committed to
 the kernel, but I believe it can be done under the contrib/ directory.
 
 I do not want this work to exist as a series of
 downloads/checkouts/patches/modules if it can be avoided.  I don't want
 to patch my kernel or load modules.  I want to be able to do a cvsup on
 /usr/src, add necessary device entries to my kernel config file and
 build it.
 
 I'd like to see astersk and libpri installs follow the reccomendations
 and requirements found in the FreeBSD hier(1) man page.  Specifically,
 it should install completely to /usr/local/.  Preferrably, I'd like to
 see a port created for both asterisk and libpri, even just a metaport
 that uses CVS to fetch the source and any OS-specific patches.
 
 Any comments before I post the bounty?  I will recommend that those with
 suggestions on the requirements and those that offer additional bounties
 for this will sit in committee to determine when the requirements of the
 bounty have been met.
 
 --
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
 
 [snip]

 Troy -
While it is not 100% relevant to your requests, I'd like to see
 continued support of NetBSD/OpenBSD in this same vein and added to
 the bounty, since the additional work to get things correctly
 functioning on those two systems seems to be fairly minor while the
 hood is open.  MacOS is a different animal, and (IMHO) lower on the
 must-have list when it comes to Zap device support, though it would
 still be cool.

If OpenBSD (1st choice) and NetBSD (2nd choice) can be added for
 Zap device support, count me in on the bounty.  Talk to me privately
 if you want to get a dollar figure.  I've had * running on OpenBSD,
 but of course no Zap hardware.  I'd move everything over to OpenBSD
 if it supported Zap, since that's my primary OS for all the platforms
 in my network.  While Linux in it's various flavors is great, it's
 simply not what my network runs, and so my * boxes are the odd man
 out systems, which makes me somewhat uncomfortable from a security
 and management perspective.

Additionally, if files are to be installed in /usr/local, then I'd
 like to see the configs remain in /etc/asterisk since on my systems
 (and many other people's) the /usr/ directories are for binaries
 only; no configurations or moving parts so those directories can be
 mounted read-only or mounted from a common server if necessary.  I'm
 sure this is what you meant, but I've seen config directories
 unwisely located in /usr/local before, and I wanted to make sure
 everyone is of the same mind where that is concerned.

 JT



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[Asterisk-Users] ... H323 - segmentation fault - core dumped

2004-01-14 Thread Jeroen

Hi all,

After having tested the SIP part (successful :-)) we are now  testing 
the H323 part of Asterisk. The H323 channel is up and running (using 
NuFone Network's Open H.323 Channel Driver))

However when dialing to * using ohphone the call can not be set up / 
established

   /H323 debug enabled
   *CLI   == New H.323 Connection created.
   -- Received SETUP message...
   == Setting up Call
  -- Calling party name:  123456
  -- Calling party number:  123456
  -- Called  party name:  111
  -- Called  party number:  111
   Segmentation fault (core dumped)
   Ouch ... _error while writing audio data_: : Broken pipe
   /
Does anybody of you have suggestions of what I am doing wrong?  (is it 
mandatory/recommended to use  asterisk-oh323? - www.inaccessnetworks.com)

Cheers,
Jeroen
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Re: [Asterisk-Users] grandstream asterisk configuration

2004-01-14 Thread Walt Reed

You may need to run Ethereal to make sure packets are REALLY getting
through. 

On Wed, Jan 14, 2004 at 07:35:48PM +0545, Chandra said:
 i have forwarded ports 5060 and 5000-5008 the ports used by sip and rtp to
 grandstream from my netgear. rtp.conf uses rtpstart 5000 and rtpend 5008. i
 have also opened all 5060, 5000-5008 ports in my firewall configuration.
 grandstream uses 5004 port for rtp.
 
 what am i missing here? please tell me.
 
 chandra
 
 - Original Message -
 From: bam [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Wednesday, January 14, 2004 6:42 PM
 Subject: Re: [Asterisk-Users] grandstream asterisk configuration
 
 
 
  Make sure that udp packets can get from the server back to the
 grandstream.
 
 
  At 12:40 14/01/04, you wrote:
hi,
  
  I have the following configuration:
  
  Grandstream -- NAT (Netgear RP614)--Internet--Asterisk(public IP)
  
  i can register fine and call ringing is working as good. The problem is =
i cant hear audio both ways and i get this error:
  
  WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error:
Resource temporarily unavailable
  
  my sip.conf file is as follows:
  
  [general]
port =3D 5060 ; Port to bind to
bindaddr =3D 0.0.0.0  ; Address to bind to
;externip =3D 200.201.202.203 ; Address that we're going to put in
 =
SIP
messages if we're behind a NAT
tos=3Dlowdelay
disallow=3Dall; Disallow all codecs
allow=3Dulaw  ; Allow codecs in order of preference
  
  dtmfmode=3Dinfo
  
  [grandstream1]
type=3Dfriend
host=3Ddynamic
secret=3Dmysecret
context=3Doutgoing
nat=3Dyes
reinvite=3Dno
canreinvite=3Dno
qualify=3D2000
  
  has anyone done this before?
  
  chandra
 
 
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IntelliFAX
Como funciona.
Sua rede fica ligada a internet bem como seus fornecedores.
Você passa o fax normalmente que se encotra ligado em nosso PABX Virtual e seus fornecedores via Internet
usando uma versão light do sistema poderão receber de qualquer canto do mundo este fax em seus faxes.

Basicamente e um servidor que fica ligado a internet e a seu Fax.

Podemos vender pra aquela agencia de turismo !

Abraços vamos conversar

fui.
[]'s

On Wed, 14 Jan 2004 09:13:25 +0800, Steve Underwood wrote: Jason Penton wrote: Hi All I have just a quick question regarding app_txfax for Asterisk. When I send a fax from asterisk to a traditional fax machine connected to asterisk via the digium analog card everything works perfectly. However the same fax machine on the public telephoine network results in errors (looks like some sort of training error). My asterisk box is connected to the pstn using an ISDN card. I don't mind trying to fix this myself but I am puzzled by the different behavior experienced when the fax machine is on the digium card and when it is connected to our public carrier, and therefore have no idea where to start. Would someone (Steve Underwood ;-) )mind at least putting me on the right track so I can address this issue? Thanks in advance Steve Jason I don't know why this would fail. An ISDN card should be properly synchrinised to the PSTN, and uses A-law or u-law. That should be enough to get a good path from the txfax program to the FAX machine. Do you have some kind of codec mismatch in your system? Can you try attaching an analogue phone in parallel with the FAX machine, and listen to the audio? You are probably familiar with how a FAX machine normally sounds, so you can probably recognise the bad distortion a codec issue would cause. The other thing to listen for is clicks. If there is a timing problem, and you have even a single sample slip in the audio stream, a modem will not work. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [Asterisk-Users] Re: newbie ISDN question


 Is there a list of cards that use this chipset somewhere on the 'net?
 I've googled for it, but most pages only talk about cards based on the
 HFC-S chipset without listing brand and model names.

Acer ISDN-Surf, Billion Bipac ISDN, Trust PCI ISDN Modem, D-LINK DMI-128+
to name a few ;-)

regards

kapejod
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
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Re: [Asterisk-Users] Fax

2004-01-14 Thread Carlos Arnt

Só que existe um problema !
O sistema deles * não funciona com o seu !!
O seu e desenvolvido em Windows (Delphi) enquanto que o deles e Linux !!!
O * e totalmente VOIP o seu no momento fala um protocolo louco !!

Me liga então.
Ps. Ao invês de usar sua conta e o forum quer por favor, usar e-mail direto !

Abraços.

[Asterisk-Users] Multiple phonenumbers on one E1 PRI with Digium TE410P ?

2004-01-14 Thread Jan Baumann

Hi,

one short question: Is it possible for the zaptel driver to deal with 
multiple phone numbers on one single E1 PRI line?

I could make my carrier route +49 xxx a-zzz and +49 xxx b-zzz 
and others down one single PRI trunk to our asterisk box terminating in 
a Digium TE410P.

Does the driver handle this and can I put calls coming in all on the 
same physical interface put into different contexts based on the dialed 
prefix?

Thanks and Regards,
Jan Baumann
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RE: [Asterisk-Users] Asterisk on FreeBSD 4.9?

2004-01-14 Thread Troy Settle


John,

I thought you might be interested.  I don't know the particulars about
driver portability between the BSD's, but it seems that at least on x86
hardware, it should be fairly easy.  I'll include those 2 in the bounty.

I'm not sure what hier(1) has on the other BSDs, but in FreeBSD it is
completely acceptable and desirable to have /usr/local/etc/ for local
configurations.  /, /usr are only for the base OS.

Of course, these are simple build-time configuration options to have.  Each
OS (even each linux distro) has it's own heir(1) scheme, perhaps the work to
get a clean and proper installation of asterisk on FreeBSD will prompt the
developers to also have asterisk install itself properly on other platforms
obeying their respective hierarchies.

John,  Do you think you could talk Mark into making some hardware available
for test/development platforms if we end up with a non-digium person
attacking this?

--
  Troy Settle
  Pulaski Networks
  http://www.psknet.com
  540.994.4254 ~ 866.477.5638
  Pulaski Chamber 2002 Small Business Of The Year
  

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of John Todd
 Sent: Wednesday, January 14, 2004 9:22 AM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Asterisk on FreeBSD 4.9?
 
 I'm about to post on bugs to offer a bounty for work on FreeBSD.  I'm
 fairly certain that others will come along to increase that bounty.
 
 Before I do post it, I would like some input on what the requirements
 should be.  Here's what I have so far:
 
   - Must be completed before 6/30/04
   - Support for all Zaptel hardware
   - Commitment of the drivers to both
 4-STABLE and 5-CURRENT/STABLE
 
 I'm not completely conversant on how GPL software can be committed to
 the kernel, but I believe it can be done under the contrib/ 
 directory.
 
 I do not want this work to exist as a series of
 downloads/checkouts/patches/modules if it can be avoided.  I 
 don't want
 to patch my kernel or load modules.  I want to be able to do 
 a cvsup on
 /usr/src, add necessary device entries to my kernel config file and
 build it.
 
 I'd like to see astersk and libpri installs follow the 
 reccomendations
 and requirements found in the FreeBSD hier(1) man page.  
 Specifically,
 it should install completely to /usr/local/.  Preferrably, 
 I'd like to
 see a port created for both asterisk and libpri, even just a metaport
 that uses CVS to fetch the source and any OS-specific patches.
 
 Any comments before I post the bounty?  I will recommend 
 that those with
 suggestions on the requirements and those that offer 
 additional bounties
 for this will sit in committee to determine when the 
 requirements of the
 bounty have been met.
 
 --
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
 
 [snip]
 
 Troy -
While it is not 100% relevant to your requests, I'd like to see 
 continued support of NetBSD/OpenBSD in this same vein and added to 
 the bounty, since the additional work to get things correctly 
 functioning on those two systems seems to be fairly minor while the 
 hood is open.  MacOS is a different animal, and (IMHO) lower on the 
 must-have list when it comes to Zap device support, though it would 
 still be cool.
 
If OpenBSD (1st choice) and NetBSD (2nd choice) can be added for 
 Zap device support, count me in on the bounty.  Talk to me privately 
 if you want to get a dollar figure.  I've had * running on OpenBSD, 
 but of course no Zap hardware.  I'd move everything over to OpenBSD 
 if it supported Zap, since that's my primary OS for all the platforms 
 in my network.  While Linux in it's various flavors is great, it's 
 simply not what my network runs, and so my * boxes are the odd man 
 out systems, which makes me somewhat uncomfortable from a security 
 and management perspective.
 
Additionally, if files are to be installed in /usr/local, then I'd 
 like to see the configs remain in /etc/asterisk since on my systems 
 (and many other people's) the /usr/ directories are for binaries 
 only; no configurations or moving parts so those directories can be 
 mounted read-only or mounted from a common server if necessary.  I'm 
 sure this is what you meant, but I've seen config directories 
 unwisely located in /usr/local before, and I wanted to make sure 
 everyone is of the same mind where that is concerned.
 
 JT
 
 
 
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RE: [Asterisk-Users] linux journal article on asterisk

2004-01-14 Thread firedude

When I get a chance I will zip over to their website and give you the 
absolute url.  I was looking at the hard copy.
AJ


On Wed, 14 Jan 2004, Franz Edler wrote:

  From: [EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 
 
  For anybody who didn't know there is an article on asterisk in February's
  Linux Journal.
 
 Can you please provide a link to this article?
 Franz
 
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RE: [Asterisk-Users] Asterisk on FreeBSD 4.9?

2004-01-14 Thread Jess Kitchen

On Wed, 14 Jan 2004, John Todd wrote:

[snip]

Additionally, if files are to be installed in /usr/local, then I'd
 like to see the configs remain in /etc/asterisk since on my systems
 (and many other people's) the /usr/ directories are for binaries
 only; no configurations or moving parts so those directories can be
 mounted read-only or mounted from a common server if necessary.  I'm
 sure this is what you meant, but I've seen config directories
 unwisely located in /usr/local before, and I wanted to make sure
 everyone is of the same mind where that is concerned.

The existing FreeBSD port stores it's config files under /usr/local/etc
which generally seems to be the 'done thing' (and I thought similarly on
OpenBSD?) - you should be careful that you don't end up with confusion
over this as I'd imagine using the port has been the main way people have
gotten it working so far, perhaps a dialog with the maintainer would be
useful so you're all singing from the same prayer book.

My 2c

Regards,
Jess.

-- 
Jess Kitchen [EMAIL PROTECTED]
http://www.burstfire.net/
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Re: [Asterisk-Users] T1 Sync clarification

 What are the practical effects with in-correct clock sync
 -like to you hear odd buzzing, or dropped voice or gaps of audio ??
  snip
 As mentioned earlier, it depends entirely upon how far off one clock is
 from the clock at the other end of the T1.
  snip
 If they are off by a little bit, you would see frame slips but probably
 not hear any quality differences.
  snip
 You would only have a fast slip rate if something is faulty. Anything 
 complying with the E1 or T1 specs should never have its clock 50ppm in 
 error. Anything coming from the PSTN is essentially bang on, as it comes 
 from an atomic clock.
 snip

So, to summarize and address the original posters questions and stop the
thread from deviating too far off topic... (add to the wiki?)

1. pstn providers worldwide have understood and addressed syncing of digital
   clocks (eg, T1/E1 clocks, not operating system clocks) for years. Its 
   probably safe to assume the majority of pstn providers either sync to
   some common source (eg, atomic clock), or, have internal mechanisms to
   ensure interoperability with all other providers. (Some exceptions do
   exist but their numbers are believed to be very small.)
2. For asterisk purposes, current T1/E1 facilities (regardless of source) 
   carry timing information embedded within the transmit leg (not an optional
   configuration parameter) that is used by the attached device for 
   recover of clock sync.
3. Channel banks typically have only a single T1/E1 uplink, and therefore
   recover clock sync from the T1/E1 receive-side of that link. If a
   specific channel bank model supported two or more uplinks, then the
   manufacturer would provide a user configurable option to select which
   uplink to use for clock sync. 
4. Likewise, since the Digium TE410P (as an example only) supports four
   T1/E1 inputs, a user configurable option is provided to select one
   port for primary clock sync, and alternates (secondaries) should the
   selected primary T1/E1 fail. Users should select the T1/E1 link that
   is closest to the pstn where possible.
5. Asterisk configurations that include multiple T1/E1 links that close
   a wide area loop, for example:
 ast#1 - T1 - ast#2 - T1 - ast#3 - T1 - ast#1
   should not be of concern from a clock sync perspective as each system
   recovers the clock sync from its associated T1 receive leg if that
   receive leg is specified correctly in the TE410P configuration.
   EXCEPT...
   If this same config included a pstn T1 link into ast#2 (as an example)
   then the TE410P should be configured to obtain clock sync from the
   port on which the pstn T1 is connected.
   If that is not configured, then clock sync within the wide area loop
   is 100% dependent upon the accuracy of the TE410P clock (which is not
   a high-accuracy clock), and frame slipage will occur at the T1
   interface to the pstn.
6. If frame slipage does occur, the impact is:
   a. for small slipage: users would not notice
   b. for medium slipage: repetitive clicks are likely to be heard across
  all voice channels in use, and fax machines are likely to be less
  then reliable.
   c. for large slipage: T1/E1 circuits are likely to fail then recover
  intermitently.

General Rules of Thumb:
1. Devices with a single T1/E1 interface will automatically recover clock
   sync from the receive-side of the T1/E1, and users never need to be
   concerned with it.
2. Devices that have multiple T1/E1 interfaces (like the TE410P) need to
   select a clock sync source, and that source should be a T1/E1 port that
   is closest to the pstn (or derived from the pstn) if it exists.

Anyone take exception to any of this before it goes into the wiki?

Rich


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RE: [Asterisk-Users] RE: Fax

2004-01-14 Thread Steven Critchfield

On Wed, 2004-01-14 at 03:37, Jason Penton wrote:
 Hi Reinhard
 
 Hmmm very interesting. 
 
 I am using chan_modem_i4l to access my gazel ISDN PCI cards. I must tell you
 though that I have two fax machines the one sends perfectly and the other
 fails (sounds similar to your problem wrt training errors). Not quite sure
 where to go from here. I am going to listen to the line (as Steve suggested
 in an earlier post) and will post my findings.

I think you will find a common thread wrt echo on chan_modem_i4l. That
method wasn't meant rally for voice traffic and therefore has some
variable delays. In the case of data, the timing isn't critical as a
data app will eat the data at about any speed, in voice it is critical
to have specific timing and even more so in fax or modem audio. If you
can switch to Kapejods CAPI driver then it should be better.

  -Original Message-
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  Reinhard Max
  Sent: 14 January 2004 10:30 AM
  To: [EMAIL PROTECTED]
  Subject: [Asterisk-Users] RE: Fax
  
  Hi,
  
  On Tue, 13 Jan 2004 at 21:06, Jason Penton wrote:
  
   (I have successfully managed to receive faxes thru my isdn 
  card so I 
   don't see why I shouldn't be able to send them).
  
  that's interesting, as in my tests it was just the other way around.
  
  I can send faxes through my AVM ISDN card (chan_capi), but 
  when I try to receive a fax, app_rxfax fails after reporting 
  some carrier training errors. I've posted the detailed error 
  logs to this list some weeks ago.
  
  Jason, are you using chan_capi, or chan_modem_i4l to access 
  your ISDN card?
  
  cu
  Reinhard
  
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-- 
Steven Critchfield [EMAIL PROTECTED]

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[Asterisk-Users] NAT friendly TFTP Server

2004-01-14 Thread TeleSIP




Hello,

For those interested in overcoming the problem with 
some NATs and Firewalls in regards to tftp. I found a nice little tftp 
server here:

http://freshmeat.net/projects/jtftp/?topic_id=87

I tried it and it works great.

Regards,
Andres.

RE: [Asterisk-Users] linux journal article on asterisk

2004-01-14 Thread Tony Kava

   For anybody who didn't know there is an article on asterisk in 
   February's Linux Journal.
  
  Can you please provide a link to this article?
  Franz From: [EMAIL PROTECTED] 

Here's the link (I believe):

http://www.linuxjournal.com/article.php?sid=6769

--
Tony Kava
Senior Network Administrator
Pottawattamie County, Iowa


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Re: [Asterisk-Users] Multiple phonenumbers on one E1 PRI with Digium TE410P ?

Hi Jan,

yes you can:

[zap-in]
exten = _49xxx,1,Goto(contextA)
exten = _49xxx,1,Goto(contextB)

regards

kapejod
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


 Hi,

 one short question: Is it possible for the zaptel driver to deal with
 multiple phone numbers on one single E1 PRI line?

 I could make my carrier route +49 xxx a-zzz and +49 xxx b-zzz
 and others down one single PRI trunk to our asterisk box terminating in
 a Digium TE410P.

 Does the driver handle this and can I put calls coming in all on the
 same physical interface put into different contexts based on the dialed
 prefix?

 Thanks and Regards,
 Jan Baumann

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Re: [Asterisk-Users] Multiple phonenumbers on one E1 PRI with Digium TE410P ?

2004-01-14 Thread Alastair Maw

On 14/01/04 15:09, Jan Baumann wrote:
one short question: Is it possible for the zaptel driver to deal with 
multiple phone numbers on one single E1 PRI line?

I could make my carrier route +49 xxx a-zzz and +49 xxx b-zzz 
and others down one single PRI trunk to our asterisk box terminating in 
a Digium TE410P.

Does the driver handle this and can I put calls coming in all on the 
same physical interface put into different contexts based on the dialed 
prefix?
Yes, it's very easy, all that will work out-of-the-box.
For example:
[default]
exten = _496667XXX,Goto(one,s,1)
exten = _496668XXX,Goto(two,s,1)
[one]
exten = s,1,Playback(hello)
[two]
exten = s,1,Playback(bonjour)
Regards,

Alastair
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RE: [Asterisk-Users] Parking extension not working

2004-01-14 Thread Sean Garland

I have just set the parking extension at 701 and then the range is
702-710 and still I cannot transfer to 701.  Show Dialplan doesn't show
an extension 700, although it shows all the parked location extensions.
If I transfer to 702, I get the message telling me that there is no
parked call there  Still lost!!!

Thanks
Sean 

-Original Message-
From: Girish Gopinath [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, January 13, 2004 11:07 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Parking extension not working

From Andy Powells Getting Started With Asterisk (V 0.1a)
http://www.automated.it/guidetoasterisk.htm

 parking.conf file has this number set at 700. I've changed mine to
701 because I was having an issue with Asterisk - although it would
'see' 
(looking at the console) I had tried to transfer to 700 it appeared not
to believe that I had dialed it. This was essentially due to the 00 in
the 700, changing it to 701 eliminates the problem completely.

Hope it helps...

Girish

From: Sean Garland [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Parking extension not working
Date: Tue, 13 Jan 2004 16:07:56 -0800

I have the standard parking.conf but extension 700 doesn't show up in 
my dialplan  Why?  I can dial 701 which tells me that I don't have 
any calls parked there.  700 just gives me invalid extension noise

Should I have extension 700 defined elsewhere?

Thanks

parking.conf
[general]

parkext =a 700  ; What ext. to dial to park
parkpos = 701-705  ; What extensions to park calls
on
context = parkedcalls  ; Which context parked calls
are
in
parkingtime = 300  ; Number of seconds a call can
be parked f

*CLI Show dialplan

[ Context 'parkedcalls' created by 'res_parking' ]
   '701' =  1. ParkedCall(701)
[res_parking]
   '702' =  1. ParkedCall(702)
[res_parking]
   '703' =  1. ParkedCall(703)
[res_parking]
   '704' =  1. ParkedCall(704)
[res_parking]
   '705' =  1. ParkedCall(705)
[res_parking]


Sean Garland
Siskiyou Technology Consultants

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[Asterisk-Users] Re: failover (was Re: voicepulse)

2004-01-14 Thread Matt Lawson

But this is not to say _you_ can't built a reliable VOIP based
system.  Get _two_ providers and set up your dial plan in
extensions.conf to fail over if one service fails to
connect to dial via the next one and finally if both fail
use pstn. your users will see a system the just works.
Now there's an idea.  

I'm playing with this now, but there's at least 1 case I'm having 
trouble recognizing:

The call connects but then drops due to unauthorized.  It then only 
goes to the h extension and I don't get a chance to try again.  Is 
there anyway to detect this?

I have to cover all of the following cases:

1.  VOIP IP address is not reachable.  Goes to extension n+101 (seems to 
work as expected)

2. VOIP service answers but refuses with call with unauthorized.  It 
just goes to the h extension  Is there any watch to catch this 
failure?  Perhaps put a timer on it and say if the call was less than 5 
seconds or something try the next one?

Yes I am using a correct username and password and getting this today 
(not from Voicepulse, from another provider).  But there's also a 
moderate chance that during our systems' setup a name or password could 
be misspelled so I need to cover this case.

3.  VOIP service connects but reports all busy.  Well this one is hard 
to test.  But I can make the Zap channel busy.  It goes to extension 
n+101 as expected, so I'll have to assume that a busy VOIP service does 
the same thing.

I was trying to determine if the t or h extension would be useful 
for these but I think not.  The timeout has to be set long enough for 
someone to actually answer (20-60 sec or whatever).  The h is always 
visited at the end of the call, whether it was sucessful or not.

Any other cases, or suggestions how to handle case #2?



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RE: [Asterisk-Users] Asterisk on FreeBSD 4.9?