Displaying 20 results from an estimated 10000 matches similar to: "Forwarding Voicemail Crashes Asterisk"
2004 Dec 04
5
Is Gigabit Ethernet necessary?
For an office that is using VoIP phones to connect to Asterisk, is gigabit
ethernet really necessary for the Asterisk box to connect to the switch? I
know that I won't even approach the limits of 100 Mbps, but would gigabit
help with latency / collisions when several calls are underway? The fact
is, anything going outside the office will be over a data T1, so intuition
tells me that 100
2005 Jan 27
3
Linux Bridge + QoS Shaper HOWTO available
I've created a pretty complete HOWTO on creating a Linux Bridge (using
Fedora) to shape LAN <--> WAN traffic. It includes installation
instructions, a script to configure the bridge (which you install as a
service), and 2 scripts to configure the network interfaces using traffic
control.
http://www.burnpc.com/website.nsf/all/3a64a6369757819686256f960068ad75!OpenDocument
If anyone
2006 Feb 14
9
Solution for 1 time blast of 200, 000 recorded calls
Hi,
I'm helping out with a political campaign and would like to use
asterisk to blast out about 200,000 calls with a short message from
the candidate.
Provider:
I'm thinking voipjet may be a good solution?
Hardware setup:
I will have access to several T-1 lines so I would just want to set up
the dialers to limit the number of concurrent calls and so forth.
I found teleyapper on
2005 Jan 18
2
Router Recommendations Please
Hello all,
We've discovered that VoIP (IAX2) + Citrix + Video is pegging the measly
CPU on the Netopia router our ISP provided. We've got 3Mb/3Mb and will
increase to 4/4 next year.
The Netopia simply breaks out our WAN IPs, and we've got a switch hooked up
to it on the inside (Actually I've got a QoS box in-between).
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| Internet |
| on Cat5 |
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2004 Nov 22
6
Linksys RT31P2
Has anyone tried out the Linksys RT31P2 with Asterisk? Seems like a really
great solution for remote users... even supports QoS. Too bad it doesn't
also have VPN functionality built in.
Here's a link to the product:
http://www.linksys.com/products/product.asp?prid=652&scid=29
-Ron
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2004 Jan 01
4
* crash when forward voicemail --Nicolas Gudino
Hey Nicolas,
That did it. I ran that export command you suggested, then launched *,
everything worked fine. I'm still looking for info on what that command
actually does. Can you shed some light please?
Thanks.
JR
-----Original Message-----
From: JR Richardson [mailto:jr.richardson@cox.net]
Sent: Tuesday, December 30, 2003 6:44 PM
To: 'asterisk-users@lists.digium.com'
Subject:
2005 Jan 03
6
QOS / Cisco / Asterisk
We're trying to PQ (Priority Queue) packets on a Cisco using ACL's. What
we're trying to avoid is hardcoding the IP address in the ACL. We were
trying to match by TOS set by Asterisk however it seems we've run into a
snag where the packet TOS tends to get reset somewhere on our network.
Has anyone had this issue? We're running Cisco everywhere inbetween
(even the switches). Is
2004 Dec 16
1
Polycom FX Video Unit - asterisk-oh323
I'm installing an office in a couple of weeks that will have some nice
Polycom FX video units in the conference rooms. I'm thinking that with
asterisk-oh323
http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/#section2
I should hopefully get the ability for phone users to dial an extension
and participate in video conferences, or just simply phone conference with
users in the
2005 Feb 02
1
PRIO / CBQ / HTB queue drop algorithm
Hello all.
I''ve been struggling to QoS VoIP at our site and have a successful
implementation at this point. Basically I had to set aside enough
bandwidth for VoIP by placing all other traffic behind an HTB (multiple
classes and queues behind it). Everything is fine. Here''s the diagram:
-------
| eth |
-------
|
--------
2004 Dec 02
5
drive space for voice mail
Drive space for voice mail
I've looked in the dimensioning information on voip-info.org but can't
find any hard information on the amount of drive space the various codecs
use. Since we would eventually like to support web-based voice mail
retrieval, I'm thinking of the wav format. I've specced out 2x160GB drives
in RAID-1 (software RAID via Linux) for the box. It will be
2003 Dec 30
2
* crash when forward voicemail message [problem solved]
Thanks for all your help Martin,
Guys,
This is a good find and hopefully could help someone else.
I've been having a problem with forwarding voicemail from one mailbox to
another. I ran down the sendmail and soundcard path and came up goose eggs.
With intuitive guidance from Martin Pycko (Digium), I switched from Redhat 9
Kernel linux-2.4.20-8 to Redhat 8 Kernel linux-2.4.18-14 and it
2005 Jan 30
2
PRIO inside HTB - trouble attaching filters correctly?
Hello everyone!
I''m simply trying to put a PRIO inside an HTB (used to throttle). I''ve got
interactive traffic on the network that I want to give priority (VoIP +
Citrix + Video).
I''ve used the filters in a CBQ script fine, but am having trouble
adjusting them to this setup such that they properly assign the traffic.
tc qdisc del root dev $e
tc qdisc add dev $e
2005 Jan 11
5
asterisk-oh323 and outgoing call
Hello.
I'm try to set up asterisk for making outgoing calls with oh323 channel
driver version 0.7.1 with Asterisk CVS-1-01/09/05-01:41:37.
Our provider uses Mera MVTS softswitch and supports only H.323.
We don't use gatekeeper for connection but provider requires SOURCE PHONE
NUMBER for route out calls and I don't know how I can specify this
number.
Call with this string
exten
2009 Jul 30
1
Voicemail Error
Hi All,
I'm trying to test asterisk voicemail on recording my own unavailable
message, busy message or temporary message. I was looking at the console
and saw this message:
app_voicemail store_file Memory map failed
Then i looked at /var/spool/asterisk/.... there were no recorded
greetings. what does the error mean? TIA
Regards
Ron
2004 Nov 15
3
Memory Consumption
Hello,
I use Asterisk 1.0.2 on a RedHat Enterprise Server 3.0 (Kernel 2.4.21)
and i experienced that the memory consumption of the asterisk-process
started by the init.d-script raises continously. Now, after 3 hours of
operation (on our testing-system we have 30 concurrent connections to
another asterisk box using IAX2 and GSM codec) there is already 66MB
allocated. I think this could be ok, but
2009 Sep 16
5
custom voicemail e-mail
Hi All,
I'm trying to use a php script to send voicemail e-mail so i can send
custom e-mail message based on what mailbox.
on my voicemail.conf i have
mailcmd=/var/www/voicemail.php
but when i tried to call an extension and goe to voicemail i'm not
receiving the e-mail.
but when i execute "php /var/www/voicemail.php" on the shell, i can
receive the e-mail.
how would i
2004 Oct 04
5
limited upload speed
HI all,
What is best way to be limited upload speed from LAN users. I read
that it is possible to be done with IMQ interface or with limitation
over gateway interface of router(eth0 in my "scheme"), but i cannot
chose what is preferred way and need from advice.
Please for advise, any example scripts or URL with tutorial are welcome :)
I read couple times Linux Traffic Control.
2004 Oct 04
0
RE: small kernel distro recommendations for QoS box
Why not try something like Pebble linux from
http://www.nycwireless.net/pebble
It is a stripped down Debian install that is aimed at running a wireless
hotspot but it is just Debian and you can install whatever you want. It
fits on a 64MB flash card but if you install almost anything you will
want a 128MB one. It does come with iptables and iproute2 tools as I
recall.
Patrick
2004 Dec 16
1
Dynamically Choose Codec for Bandwidth Management
Is there any way to set Asterisk to choose what codec to allow for a new
call based on current usage? In other words... be able to define a max
number of ulaw calls, then after that only allowing g729? The idea here is
that in general, a T-1 should be enough for our offices to have phone +
citrix + some video (got good QoS in place already). But for usage spikes,
user experience would be kept
2005 Jan 06
1
destroy SIP channel??
I've got a SIP channel that appears to be hung up. It's an extension that
records a .gsm file and fortunately the recording has stopped. I tried zap
destroy channel but I guess that doesn't apply to SIP channels.
Any ideas? I issued a restart when convenient but figure there must be a
better way.
TIA,
-Ron
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