Displaying 20 results from an estimated 20000 matches similar to: "One voicemail -> multiple boxes?"
2006 Jun 16
17
Voicemail with NFS
I have /var/spool/asterisk/voicemail NFS mounted from another server. Everything is fine, until I simulate an NFS server failure, by shutting down the NFS server process.
At this point, Asterisk becomes almost non-responsive. It won't even process a 'sip show peers' command correctly. It displays a few lines of text, pauses for several seconds, and then displays the rest. When a call
2004 Jan 15
3
Voicemail Sequence Bug?
I have a user, running CVS a/o 11/23/03, who has complained about
"phantom" messages showing up days or even weeks after she has deleted them.
So I asked her to let me know when it happened again, and she called a
few minutes ago.
The directory listing below shows a listing of the
/var/spool/asterisk/voicemail/default/XXXX/Old directory, and to my
surprise the messages are indeed
2006 Jan 06
2
SPA-3000 is translating vocal sounds into DTMF
I'm sure there must be a setting I'm missing somewhere, so I thought I
might was well ask here.
Conversations are punctuated by sudden replacement of a given syllable
or so of conversation with a DTMF tone.
I would hope perhaps there's some kind of setting that has to do with
the way it detects inband DTMF? I'm pretty sure it's an artifact of
this particular ATA; my
2010 Oct 30
2
Exceptionally long queue length queuing . . . .
I wonder if anyone out there has a perspective on this. There are a
welter of tickets out there on the matter, most of them closed.
This problem began for me over a year ago, and continues up to the
latest versions I've installed (1.6.2.13).
It happens randomly, and the suggestion on one of the bug tracker
tickets that it is instigated by a small network leg looks to be on
point to me,
2003 Sep 20
2
Oops!!! Current CVS crashes
I don't know whether this ought to go to the bugtracker.
I downloaded the current CVS last night and then again just a few
minutes ago.
In both cases I can crash asterisk very easily by the following method:
1. Call up and leave a voicemail.
2. Log in and listen that I have a new message.
3. Hit "1" to listen, and 'Hasta la vista' asterisk.
I also noticed that the normal
2003 Jul 27
3
Nortel 350
Wondering, since they appear to be plentiful on eBay, whether I could
get a Nortel 350 to use to learn my way around ADSI.
The vendor claims that these are "generic," and looking through the
archives I wonder if that means that they might be unlocked in the sense
that the word is meaningful to asterisk.
Of course I am green as could be on this topic, so this question may
even be a
2006 May 29
4
app_conference DTMFs?
We need to conference together a call center agent, a customer and a third
party IVR and send DTMF tones from the agent to the IVR.
MeetMe has been eating our DTMFs so we'd like to try app_conference.
Has anybody setup such a configuration in app_conference and how did you
configure it?
-HJC
2006 Jun 26
4
Oh oh. Micro$oft just noticed VoIP
It will be interesting to see how many standards get broken, and how
many proprietary hooks get thrown into the pot. The bean counters smell
some money, and their OS franchise is waning:
http://www.nytimes.com/2006/06/26/technology/26soft.html
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
2005 Sep 06
4
Working example of ALERT_INFO with Cisco ATAs?
I am wondering if there are any tricks getting the Cisco ATAs to do
"distinctive rings" via the ALERT_INFO variable?
I have seen some contradictory information in the Wiki, and I tried the
example there. I then sniffed the connection between the server and the
ATA and didn't see the header sent like it is "supposed" to be.
If someone out there has a handle on this and
2005 Mar 18
15
Meetme2 compilation problem
Hi All,
I am trying to compile meetme2 in my asterisk box and getting the
following compilaton error. Please help me to sort it out.
cc -fPIC -c -o app_dial.o app_dial.c
In file included from app_dial.c:14:
/usr/include/asterisk/lock.h: In function `ast_mutex_init':
/usr/include/asterisk/lock.h:317: `PTHREAD_MUTEX_RECURSIVE' undeclared
(first use in this function)
2004 Jan 17
6
Zone Paging
I see a lot of chatter in the archives about intercom and paging, but
has anyone addressed zone paging? Each of the 50 telephones in a large
clinic would be members of one or more paging zones. Someone could then
page Dr. X in zone #1. Would this be possible with analog phones? SIP?
Thanks,
Mike
2006 Jun 27
2
SV: Error in config sample for GoToIf?
Hello
As far as ive understood, you can just write
Exten => s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail)
${AVAILSTATUS} would return 1, and "${AVAILSTATUS}" would return "1"
Jon
-----Oprindelig meddelelse-----
Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Brian Capouch
Sendt: 27. juni 2006 09:10
Til:
2007 Mar 01
5
Asterisk Realtime
Could someone provide some steps for troubleshooting Realtime? I can't see
any signs that it's working. I followed and double-checked a few different
guides around the net, but haven't been able to figure it out.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2003 May 09
2
Configuration for ATA186 behind a NAT?
I wonder if someone out there could loan me a peek at their sip.conf?
I have conflicting advice, for instance, about whether or not to use
"nat=1" and also whether or not the ATA should be registering with the
instance of asterisk it is going to be using to dial out.
Thanks in advance.
B.
2003 Jul 22
2
Cisco 802.11b VoIP phone?
I wonder if anyone could send me a pointer to technical specs and
pricing information.
I got a mail today from an acquaintence that contains what I believe is
some serious misinformation, referring to the 7960 as their new portable
802.11b SIP phone. A quick search of eBay would seem to refute that.
I hope this is an OK question to ask. . .
Thx.
b.
2003 Sep 05
2
VONAGE or IP Dialtone
The Vonage service is offered with a SIP Cisco ATA device for connection
to an analog phone.
Is it possible to connect the Vonage service directly to the Asterisk
PBX bypassing the ATA and FXO card? Are there other services that offer
this capability or something similar to IP dialtone?
Thanks
Kevin
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2003 Oct 06
2
Anyone else use Audacity for prompts?
I am using Audacity to record some voice prompts.
The .wav files I'm producing are of stellar quality. However, once I
turn them into .gsm, they sound buzzy and muffled.
I know that some of this comes with the territory, but I wonder if there
is anyone out there who does this routinely, and who can advise me as to
the MO I could use that results in the highest quality in the resulting
2003 Dec 07
1
Vonage sending Motorola gear now?
I got a call from an ISP friend tonight who said he is getting calls
from people who are getting signed up with Vonage. Instead of sending
them ATA186s, apparently they're receiving something made by Motorola.
They apparently work significantly differently than the Cisco units, and
there have been some problems.
Anybody know anything further?
Thx.
B.
2003 Nov 24
4
One voicemail -> multiple recipients?
The subject pretty much says it all. I have a customer who would like
to have an option where a caller can leave a voicemail in such a fashion
that it would be simultaneously delivered to a set of mailboxes all at
once--the idea is "trouble ticket" type operation where multiple
technicians will *each* get the vm.
He prefers that, if we can do it, to a "shared mailbox"
2006 Apr 07
5
[OT] Centrex Question
I haven't dealt with Centrex for a long time, and one of my customers is
being courted heavily by a Sprint salesperson.
Am I not correct in assuming that each "line" of Centrex corresponds to
an "extension" in the PBX world?
This site has 2 POTS lines and 5 extensions, and they told me that for
the same thing they're paying right now (~$40/POTS line) they will be