* You are viewing the archive for May 2nd, 2011

best current version and motherboard/CPU compatibilities

I’ve been away from asterisk for a while since 1.4.16 and only installed
1.6 once to run a test… can someone recommend what the best version to
install is and the recommended CPU/motherboard for an * box these days?
I’m just running about 20 handsets and 4-8 lines with POTS & SIP mix.

I remember there were some issues with bios a while back and a TDM card
was required for timing conferencing, etc… are these requirements
still an issue?

I want to setup another * box and was wondering which CPU/motherboard to
select…
thanks,
daveC

Retrieving sound files from DB as opposed to filesystem

Just from my experience with different DBs, stay away from BLOB data
types as much as possible.

On Mon, May 2, 2011 at 3:15 AM, A E [Gmail] wrote:
> Hello All,
> Probably a silly question, but we’re wondering if people have had any
> experience and have data to demonstrate if the performance of the Asterisk
> system might suffer in terms of latency etc. if we’re to have it retrieve
> sound files from a database using odbc as opposed to storing them locally on
> the filesystem. Note, these are not prompts…these are sound files that are
> being created through a web-app and being stored in the DB as BLOB or
> similar datatype that’s good/efficient to store audio/video files in a DB.
> We need these be made available through the asterisk system to play over the
> phone. Although the DB uses a SAN, the Asterisk System has no connectivity
> to the SAN but is connected on the same physical ethernet switch with a
> multi-Gbps backplane.
> The way the system is being designed, it’s possible for us to end up with
> 000s of these sound files stored in the DB, not to mention several asterisk
> systems in a pool/cluster/farm requesting these files, so using the local
> filesystem might not be scalable or efficient.
> Any advice/comments/suggestions welcome :)
>
>
> –
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ATA refuses to answer a call?

I’m kind of at a loss to diagnose problems like this, yet we get them a lot.

- The ATA (Thomson 784 in this particular case) is logged into the
Asterisk server. ‘sip show peer’ shows their IP address, port, and
useragent.
- The ATA is connected directly to the internet (no NAT, but the sip
configuration has nat=always) and logs in to our server, which is also
directly connected to the internet without any firewalling.
- When people call this extension, the console shows that Asterisk accepts
the call from the DAHDI channel, executes the SIP call, then… nothing.
It either waits until the timeout set in the dialplan is up, then goes to
voicemail (next step), or it sends a ‘hangup cause 102′ to the DAHDI
channel. Conspicuously missing is the console saying “SIP/username is
ringing”.

The following is redacted output from such a call:

sip busy detect

Hi,

I am trying to configure busy detect on sip channel but somehow its not working may be this is my mistake could you please help me to figure out. I have added following options in my sip.conf

[7527]
type=friend
context=from-sip
host=dynamic
dtmfmode=rfc2833
callerid=”Guest” <7527>
mailbox=7527@default
nat=no
qualify=yes
cc_agent_policy=generic
cc_monitor_policy=generic
busylevel=1
limitonpeers=yes
call-limit=1

when 7527 is busy i am getting following error message on CLI. Why i am getting channel status CONGESTION ? instead BUSY ?

[May 2 16:47:31] NOTICE[31757]: chan_sip.c:5658 update_call_counter: Call to peer ’7527′ rejected due to usage limit of 1

Retrieving/Streaming audio/video files from DBusing over AGI

_____

From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of A E [Gmail]
Sent: Monday, May 02, 2011 1:23 PM
To: Asterisk Users Mailing List – Non-Commercial Discussion
Subject: Re: [asterisk-users] Retrieving/Streaming audio/video files from
DBusing over AGI

Just realised that this can better be described another way:

What we’re essentially trying to do is be able to do any one of these

a) stream an audio/video file stored in the DB via AGI into the current
channel so that it plays on the phone

OR

b) Do something like what Realtime Voicemail does, where it gets the file
from the DB, saves as a temp file in the user mailbox directory and then
plays it to the caller but this needs to happen through AGI, something along
the lines of readsql (a la func_odbc) inside of AGI

OR

c) Anything else that’s better than a) and b) above that someone can
suggest.

P.S> I do know about the AGI AddOn of PUT SOUNDFILE and GET SOUNDFILE which
seems to be the only solution we can think of right now, other than of
course having the DB machine exporting the SAN volume as an NFS share for
the Asterisk server to mount, but that sounds like it’ll be bad for
performance?

Thanks again

No takers? :(

[Danny Nicholas]

In your original scenario you were opening yourself to probable latency
issues – I would personally pursue something along the line of option B
where I put the DB data into a temp file and ran a daemon to clear the temp
files hourly or daily as needed. If the delivery worked well across most
LAN’s/WAN’s, some gung-ho developer would have hosed another part of
Asterisk trying to get that “bell and whistle” into the trunk.

asterisk call completion issue

I have call-limit=1 at sip.conf

From: danny@debsinc.com
To: asterisk-users@lists.digium.com
Date: Mon, 2 May 2011 12:20:40 -0500
Subject: Re: [asterisk-users] asterisk call completion issue

From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of satish patel

Sent: Monday, May 02, 2011 12:19
PM

To: asterisk-users

Subject: [asterisk-users] asterisk
call completion issue

Hi All,

I am testing CC feature with asterisk 1.8 but i am having some issue. We have
polycom 501 SIP phone and those are configured with two line with same
extensions. When i am requesting for CC i am not getting call back from
asterisk but it works if i reboot my polycom phone ( In short when phone get
register )

Is this because of two line configured ? or some configuration issue ?

[Danny Nicholas]

I would check call-limit and see what reducing that would do
for you.