Hi all and thanks for reading.

I am experiencing a frustrating issue with asterisk where on some
calls the volume suddenly drops to inaudible o completely fades away
for a time. If you hold on long enough (20 to 30 seconds) the sound
will come back.

My asterisk server is on a public IP, and basically acts as a VoIP
bridge receiving calls from my customers (all of whom use Grandstream
GXW400X gateways on public IP’s, no NAT) and sending them to different
SIP providers. I am proxying the RTP stream through the server
(canreinvite=no).

I am sure this is an issue with my setup because this happens to
random customers calling to random destinations on any of our VoIP
providers. The symptoms are always the same: voice becomes inaudible
or fades away, most of the time the customer hangs up.

I am running CentOS 5.5 and asterisk 1.6.2.15 on 4 different servers,
all of them Dell PowerEdge.

Any ideas? I do not even know how to start debugging this issue. A SIP
trace obviously shows nothing. Any pointers would be appreciated.

Regards,

Alex

_____

From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Cary Fitch
Sent: Tuesday, March 01, 2011 11:23 AM
To: ‘Asterisk Users Mailing List – Non-Commercial Discussion’
Subject: [asterisk-users] Caller ID

We do not get caller ID (name) on our telco lines.

However we have a few single line extensions with consumer type handsets
that ring at odd hours with “Asterisk” before the phone is picked up, and
“Out of Area” after it is picked up.

I have read that “Asterisk” is what is reported by Asterisk for 0 length
caller ID number.

But since we don’t subscribe to Caller ID Name, I am wondering where the
“Out of Area” is coming from?

Could these be hacking attempts via IP? Perhaps they are doing the caller
ID name? It only happens to a few extensions as far a we know.

TIA for any input or knowledge.

IP Hacking “should not” apply on your Telco lines. I’d start with your CDR
file (/var/log/asterisk/cdr-csv/Master.csv) and go from there.

Hello List

i have asterisk installed in our call centre i have configured the snom
phone 320 and 370 with in sip.conf and dialplan.com and extenssion.com

i have just one question how can i do in order to record all the calls
automatically in our server

Thanks and regards

Hello

I couldn’t find information about what “wink” is in zapata.conf:

www.voip-info.org/wiki/view/Asterisk+config+zapata.conf#TimingParameters

Does someone know what it is, and how it differs from flash?

Thank you.