Displaying 20 results from an estimated 1000 matches similar to: "No subject"
2009 Oct 15
2
MWI for multiple voice mail boxes
Hello, all. I have a user who needs to monitor their voice mail box and
the general delivery voice mail box. I defined them in sip.conf as
follows:
[tkeeley](a10f)
mailbox=612 at a10, 610 at a10
However, the MWI does not indicate voice mails for 610 and I keep seeing
this error message:
ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox
610 in context a10
However,
2009 Jun 19
2
IMAP voice mail storage
Hello, all. I am attempting to use IMAP voice mail storage in Asterisk
1.6.1.1 on CentOS 5.3 using Zimbra 5.1.6. I will not be using it as it
has proved terribly unstable - Asterisk segfaults on every voice mail
message although the message is successfully deliver to my email inbox -
but I thought I should report it. Here are the errors from the Asterisk
console:
-- Executing [210 at
2009 Jun 27
1
Multiple parking lots use default park positions
Hello, all. I'm having a deeply frustrating time getting multiple
parking lots to work and am wondering what I am doing wrong. I am using
Asterisk 1.6.1.1. I defined two separate parking lots in features.conf
as follows:
[parkinglot_a100] ; SSI
context => a100-parking
parkpos => 900-920
findslot => next
[parkinglot_a10] ; EBC
context => a10-parking
parkpos => 800-820
2009 Jul 28
0
Call history problems from B2BUA
Hello, all. Alas, another convoluted question. All the simple things
are, well, simple so I suppose we only need to trouble the list with
squirrely problems!
We've noticed a call history problem when using Asterisk where the call
history on the Snom phones (with which we are very pleased) reflects the
number of the PBX extension used by the B2BUA to dial the end point. I
assume the same
2009 Jun 18
2
Incoming SIP and the 's' extension
Hello, all. My apologies up front but I must be brain cramping on
something very simple. I've tried to pare down my configuration to the
absolute minimum for SIP traffic just to understand how it works. My
incoming calls are not finding the "s" extension in my dial-plan. I am
assuming SIP calls can do this. I am using Asterisk 1.6.1.1
sip.conf has nothing but:
[general]
2009 Oct 15
1
Callpickup works for outside calls but not inside calls
Hello, all. I've got a problem where we set up call pickup for a
customer. If the Bob's extension rings and Bob is in Jim's office, Bob
can press the button on his Snom 320 that says "Bob" and pick up his
line. It works great for calls coming in from the outside but does not
work for internal calls. Internal calls generate a
app_directed_pickup.c:204 pickup_exec: No
2009 Jul 03
1
Zimbra IMAP authentication - SOLVED
Hello, everyone. No need to read this message. I'm posting for
documentation for other poor, ignorant slobs like me who are struggling
to pull together the many technologies to make converged networks
happen. Hopefully, this will help save someone else the time I spent.
I started the below email until I realized I had solved multiple parts
of a compound problem but not all at the same time.
2009 Oct 21
1
Incorrect voice mail format on transfer
Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a
multi-tenant environment with IMAP voice mail storage on Zimbra. One of
our clients is having a problem when transferring voice mails from one
mailbox to another (option 8 in the standard voice application menu)
using their Snom 320 and 360 phones.
The end results is the final recipient cannot listen to the voicemail.
We also email
2009 Jun 27
1
Call Parking timeout fails
Hello, all. I'm having a nasty problem with call parking in Asterisk
1.6.1.1 that smells like a bug. When the call returns, it seems to be
returning to a "|" delimited extension and failing. Here is the output
from the console:
[Jun 26 22:20:42] NOTICE[7168]: chan_sip.c:18160 handle_request_invite: Call from 'tkeeley' to extension '56' rejected because extension
2009 Sep 14
1
The "o" dial option
Hello, all. I see there is an "o" option for the Dial() command which
reverts to the previous behavior of using the original callerid
throughout the call - I suppose more specifically, using the callerid
from leg 1 for leg 2 in B2BUA if I understand it correctly.
That seems to be highly desirable behavior; I know we are seeing some
problems with call history and call forwarding because
2009 Aug 27
2
Selective canreinvite in multi-tenant environment
Hello, all. In our multi-tenant environment, we would like to be able
to use the reinvite media redirection within Asterisk for calls within a
tenant but not between tenants. We would like inter-tenant calls to be
fully proxied by the Asterisk server. I think the answer is, "we
can't," but I thought I'd ask anyway.
I'd dearly like to remove the substantial traffic
2009 Jun 21
1
Meetme Talker Optimization
Hello, all. I've been playing with MeetMe and talker optimization
seemed like a great idea. I activated it as follows:
exten => 201,1,MeetMe(100201,cTo)
However, although I can see who is the talker on the CLI
pbx01*CLI> meetme list 100201
User #: 01 1001 Denise Dion-Sullivan Channel: SIP/1001-1e1db7c8 (not talking) 00:00:33
User #: 02 1000 John A. Sullivan III
2009 Aug 03
2
Upgrading from 1.6.1.1 to 1.6.1.2
Hello, all. After reading the README, UPGRADE.txt, and a quick tour
through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2,
one simply compiles and installs over the old installation being careful
to NOT install the sample files? Thanks - John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsullivan at opensourcedevel.com
2009 Jul 24
2
TLS Manager
Hello, all. After many pages of googling and testing in the lab, I'm
still a bit perplexed about how to implement tls protection for the
asterisk manager. manager.conf allows one to specify the cert file but
one normally must also specify the private key file. If I simply enter
the cert file:
sslenable=yes
sslbindport=5038
sslbindaddr=172.x.x.8
sslcert=/etc/pki/tls/certs/pbxc.pem ; path
2009 Aug 26
1
netfilter conntrack mangling canreinvite?
Hello, all. Since implementing an iptables firewall between the
Asterisk PBX and several SIP phones, the Asterisk PBX ability to
"reinvite" has been broken even when the phones are on the same network
(i.e., no firewall between the phones). We've been beating our heads
against the wall thinking it was the complex rule set but it appears the
issue is ip_conntrack_sip.
Before I drop
2009 Jun 02
4
Realtime LDAP passwords
Hello, all. I'm afraid I've been dropped into the deep end even though
I am an Asterisk novice. I've set up a few tiny, tiny systems in the
past and have now been asked to pull together Asterisk, FreePBX,
Kamailio, RTPProxy, and Fedora Directory Server into a VoIP service.
After googling and reading for most of the last 24 hours, I finally have
my head around the components and how
2005 Nov 05
4
Module error in mkinitrd
We are happily running xen on a server which uses a SATA driver (thank
you all you xenists for producing such a great product). However, we
are creating a custom kernel for a domU. When we attempt to run
mkinitrd we get:
No module ata_piix found for kernel 2.6.11.12-xenUIPSec, aborting.
This is the SATA driver. In the past (xen 2.0.5), we simply ran make
ARCH=xen menuconfig and told it to
2009 Jul 03
0
Converged mail box sizes
Just a thought as we explore the brave new world of converged voice and
emails. Voice mail boxes typically hold a very small number of messages
while email folders contain thousands. Do we need to rethink the
traditionally small limits on voice mail boxes when storing in IMAP or
are the messages counted separately? Thanks - John
--
John A. Sullivan III
Open Source Development Corporation
+1
2009 Jun 17
1
Installing LUA
Hello, all. The little bit of reading I've done on lua makes me eager
to give it a try. However, when I try to install it (Asterisk 1.6.1.1
on CentOS 5.3), it is not available in menuselect. I have installed lua
and lua-devel. I've seen very little about it in my Internet searches.
What else must I do so that it installs? Thanks - John
Oh, by the way, I'm having a similar problem
2005 Sep 08
0
Re: Xen on production enviroment
Hi John,
And an excellent stab that was.
I have two network interfaces. eth0 is a 3Com with a
Broadcom Corporation NetXtreme BCM5701 Gigabit Ethernet chip,
and eth1 is a 3Com 3cSOHO100-TX Hurricane chip.
On the gigabit card, I couldn''t test anything, because I could not
find out how to disable/enable/check checksum offloading. Using
ethtool gives: Operation not supported
For the