date:20070424

Re: [asterisk-users] auto load error in asterisk cli

2007-04-24 Thread Tzafrir Cohen

On Mon, Apr 23, 2007 at 09:33:13PM -0400, Eric Kosten wrote:
 Hello list.  My name is Eric Kosten, and I am new to Linux and asterisk  As
 a new user of asterisk and Linux I an having problems to some that might
 seem small, but these problems are such that I am not sure ware to look!
I managed to take care of some ownership issues, e.g. sip.conf and
 var/log/asterisk were not part of the asterisk user group.  If asterisk
 starts automatically, when I connect to the console the following happens:
 
 WARNING[2683]: db.c:67 dbinit: Unable to open Asterisk database
 
 if I shut asterisk down gracefully and then start it again, things run fine!
 Which database do I need to look for permissions on?  Are these permissions
 user, group or is this a case of this database not belonging to the asterisk
 group?

The Asterisk database is a Berekeley DB file which is normally in
/var/lib/asterisk/astdb .
Asterisk will try to create it if it does not exist.

 
 If autoloaded, I do not connect with vitelity which is my ip provider for
 voip service.
 I have googled the error part:
 
 Unable to open Asterisk database
 
 This is how I came to my conclusion of this possibly being a permissions or
 group membership issue.

Right.

If you still have problems, it wuld also help to state:

* Version of Asterisk you use
* Linux distribution

And in you case, also:  

  ls -al /var/lib/asterisk

-- 
   Tzafrir Cohen   
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[asterisk-users] Request for Configration details

2007-04-24 Thread prasad sathya


Hai all,

Iam a newbie to Asterisk.
I want to configure my Asterisk thru Command Line Interface to connect
two internal extensions and two external numbers and calls should
occur between any of the two numbers. Can anybody kindly send me the
configyration details for

extensions.conf anf sip.conf file.. and if anything else needed to
serve my purpose.
Iam in a great need of this
A reply word helps me a lot
Kindly send ur replies to [EMAIL PROTECTED]

Thanks
Prasad
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Re: [asterisk-users] Purchasing a Sangoma A102 - should I get thehw echo cancellation or not?

2007-04-24 Thread Eric \ManxPower\ Wieling


Rob Townley wrote:

A salesman told me that there are scenarios (analog vs T1 trunk lines)
where echo cancellation will make things worse.  Can anybody clear
that up?


Did the sales person say exactly what is worse than having echo?
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Re: [asterisk-users] Request for Configration details

2007-04-24 Thread Tzafrir Cohen

On Tue, Apr 24, 2007 at 11:44:18AM +0530, prasad sathya wrote:
 Hai all,
 
 Iam a newbie to Asterisk.
 I want to configure my Asterisk thru Command Line Interface to connect
 two internal extensions and two external numbers and calls should
 occur between any of the two numbers. Can anybody kindly send me the
 configyration details for
 
 extensions.conf anf sip.conf file.. and if anything else needed to
 serve my purpose.
 Iam in a great need of this
 A reply word helps me a lot

What version of Asterisk do you use? What type of external numbers?

What phones do you have? What extra hardware do you have?

One place to look at is http://voip-info.org/wiki/view/Asterisk
(one link away from http://voip-info.org/ )

 Kindly send ur replies to [EMAIL PROTECTED]

You asked a question on the mailing list, and thus replies go to the
mailing list. While we want to help you as an Asterisk user (and
hopefully a future contributer), we also want to contribute to the
general knowledge pool [*].

[*] One of the IRC channels I happen to be in right now has the
following in its topic:

  He who asks a question is a fool for a minute; he who doesn't ask is a
  fool for a lifetime -- share the gained knowledge on the Wiki, and
  we'll forget about the minute ;)

   Local value for $WIKI: http://ovip-info.org/

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   Tzafrir Cohen   
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+972-50-7952406   mailto:[EMAIL PROTECTED]   
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Re: [asterisk-users] SIP devices with packet loss tolerance

2007-04-24 Thread Eric \ManxPower\ Wieling

Hoping someone might have experience with poorly-performing net connections and
which devices work best over them.

One of our clients has a number of employees that work from home, and are given a SIP phone to take with them and hook up to their broadband. For the most part, this works fine, but
there are an increasing number where sound quality is poor (chops in and out, generally only noticeable to the listener at the other end, not the employee). Logic suggests it's an upstream
bandwidth issue, so we asked them to try when all other devices were turned off (to cut out the kids using bitTorrent issues), but even with the phone the only device, call quality was still
poor.

Since the connections aren't paid for by the client, we aren't in a position to mandate particular providers or speeds, but in each case, the minimum was a 1mb/256k up ADSL. We asked
the employees to run some speed tests to determine real-world speeds, and in each case upstream was around 220-235k (a little off the official speed but not bad). Certainly way more
than the ~35kbps necessary for a g729 call, even with packet overheads.

PSTN - Asterisk - Internet - SIP Phone.

If the person on the PSTN side is having audio quality problems then the
issue is not with the jitter buffer on the phone. The problem in this
case is the jitter buffer in Asterisk. SIP is a signalling protocol.
Audio is sent using the RTP protocol. In versions of Asterisk before
1.4 there was no RTP jitter buffer in Asterisk.

Lack of an RTP jitter buffer in Asteirsk is why none of my clients have
deployed phones off the corporate network.

If the person on the SIP phone side is having audio problems (not the
case if I read your message correctly) then you have to look at the
jitter buffer settings on the phone.

Remember jitter buffers (and QoS actually) is only applied to and is
only effective for INCOMING traffic.

Yes, applying QoS to the outbound traffic of the internal interface of
your router can give the illusion of limited QoS. This happens because
of the nature of TCP and will do nothing for non-TCP traffic.

Jitter is not the packet latency, but of the VARIANCE in latency. Also,
dejittering audio requires buffering and this buffering adds to the
audio latency. If you had a jitter buffer that could handle 3000ms of
jitter (on a HughesNet satellite connection, for example) your audio
would generally be great, the tradeoff is that you have just added 3
seconds of latency to your audio and in anyone's book that sucks.

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