Displaying 20 results from an estimated 3000 matches similar to: "User Loses Ability to Make Outgoing Calls"
2009 Jun 24
7
PHP AGI Not Working and Odd Behavior
Hi,
I'm running asterisk 1.4.22 on a debian server.
I have php5 installed and it works correctly command line.
When trying to run a php script via AGI, I get messages such as:
GI Tx >> I>
AGI Rx << #!/usr/bin/php5 -q
AGI Tx >> 510 Invalid or unknown command
The scripts are completely executable and owned by asterisk
-rwxr-xr-x 1 asterisk asterisk
Googling is not helping
2004 Sep 23
1
Alternate MP3 Player
Hi! I am currently working on setting up an Asterisk system, and I was
wondering if anyone has worked on an alternate mp3 player to mpg123.
We have a library of MP3 files that we would like people to be able to select
and play over the phone -- and this will require pause & resume, as well
as fast forward / reverse (jump forward / jump back). It doesn't seem like
mpg123 can do this. Is
2006 Jun 21
0
Re: User Loses Ability to Make Outgoing Call s
If I understand this correctly, this is a user outside your firewall dialing
in to your office over the Internet. Always, inbound calls work, but
sometimes, outbound calls do not work. So if you have replaced the hardware
totally, and you still have the same problem, it could be a routing issue
with an upstream ISP. The way to test for this is to do a traceroute from
her LAN to your office. Then,
2007 Jul 19
8
Blank Voicemails
Hi, we're running Asterisk 1.2.10 and have been randomly being left
blank voicemails with long messages that we can't hear.
I've searched and searched but cannot find a solution.
This is what happens:
Internal Server runs Asterisk 1.2.10 where our mailboxes are
Incoming Server (behind a firewall) runs Asterisk 1.2.13 and calls are
bridged between this server and our internal server.
2006 Jun 22
1
Re: Can I enter an extension to dial whilevoicemail is playing?
The options are not seperated by commas.
exten => s,1,Dial(SIP/50,23,r,d)
should be
exten => s,1,Dial(SIP/50,23,rd)
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of John Klimek
Sent: Thursday, June 22, 2006 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
2008 Oct 29
3
Blank Voicemail.Conf after Password Change
Hi,
For a few weeks now, our asterisk server has been experiencing something
very odd.
From time to time, voicemail.conf would go blank. We finally tracked it
down to happening when someone attempts to change their password.
It seems the file is touched, but not written to, and we're left with a
blank voicemail file.
Permissions seem to be fine:
-rw-rw-r-- 1 asterisk asterisk 12707
2006 Jun 19
1
Can I enter an extension to dial while voicemail is playing?
I have a very, very simple Asterisk setup in my house. I have a
Sipura 3000 with a PSTN line connected and one analog phone connected.
The [incoming] context looks like this:
exten => s,1,Dial(SIP/50,23,r)
exten => s,2,VoiceMail(u50@default)
exten => s,3,Playback(vm-goodbye)
exten => s,4,Hangup
As you can see, when somebody calls in if I don't answer in 23 seconds
then they are
2006 Jun 27
2
Changing standard Voicemail behavior
I am using Trixbox 1.0(Asterisk 1.2.7.1)at a customer site. They whishes
to change the default Voicemail behavior.
Standard behavior
No answer/Busy -> send to Voicemail
Requested behavior
No answer/Busy -> message that if you press 9 you will instead be cent
to reception -> send to Voicemail or Reception if 9 pressed.
I want this to always happen when Voicemail is invoked. How
2007 Jul 19
0
Blank Voicemails/Vonage Problem
Regarding this message, I've actually been told one caller who has
consistently had this problem was using Vonage, but calling from his
Verizon line, it worked. This skewed my survey.
Therefore I do believe it's the same callers having the issue, and in
which case, I think Vonage is to blame.
I found this thread:
2006 May 15
1
Outgoing Calls Not Working all the time
I currently have Asterisk 1.2.7.1 and the Sangoma A200 w/ 6 FXO ports and HW
Echo canceller. I have outgoing calling setup to use a group so that if one
channel is busy it goes to one of the other channels. What's weird is that
when I dial an outside number, sometimes it goes through and other times I
get "You have reached an invalid pager number MCLL327." I have no idea what
that
2007 May 13
1
Sudden appearance of SIP/2.0 401 Unauthorized
Yesterday we moved one of our servers to a new IP. We updated DNS and
various adapters configured to register to that server registered to the
new IP correctly. All seemed to be well.
This evening I discovered that with one exception, all of the adapters
are getting a SIP/2.0 401 Unauthorized message back from asterisk. The
exception is an Innomedia adapter -- Linksys PAP2's and (I
2006 Mar 18
6
ActiveLDAP and variable sub scope object writing
Anyone out there using ActiveLDAP have an idea how I can accomplish
creating an object one level below a known base where we have a variable
item in the middle?
That first sentence doesn''t even make sense to me. Here''s what I want
to do: I have a user class that I use for managing users. Each user
gets a ou called addressbook (which in turn will contain sub-entries,
but
2006 May 05
2
dovecot LDA w/virtual domains and postfix
Hi,
I've been trying to follow the documentation that I am finding, but am running
into trouble getting things set up correctly for postfix + virtual domains
(using ldap) with dovecot LDA. I can get it to work without LDA, but I'm
running into permissions problems when I try to run with LDA. I am wondering
if anyone has any good examples of configuring this.
I basically have a
2006 May 08
3
Most comprehensive management?
I see that Asterisk@home and FreePBX are going along similar lines with
web based management interfaces..
My Asterisk box has one analog phone, one analog line, 3 SIP phones, IAX
inbound numbers, an IAX outbound trunk, IVR menus and voicemail boxes in
different contexts for each of the inbound numbers.. Soon I will be
adding one or more IAXy devices..
Would either Asterisk@home's or
2006 Jan 20
5
Asterisk in SPA9000?
Did Linksys really use Asterisk for the SPA9000 software?
--
Andres
Technical Support
http://www.telesip.net
2006 Mar 05
6
Polycom 501 power over ethernet
When I bought two Polycom 501 SIP phones, I naively thought they were
Power-over-Ethernet (IEEE 802.3af) because they were "powered over
ethernet." Silly me.
Polycom must have some odd voltage or funny way of injecting the
power, because the POE switch I bought for them (Netgear F@510P)
won't power them, though if I use the Polycom-supplied AC adapter and
ethernet power
2006 Mar 16
3
Passing function names from string variables
This may be a more generic Ruby question, so I''m going to ask on the Ruby
forum, but I''m trying to figure out if there is a way to pass in the string
value of a variable as the name of a method.
For example, I would like to do something like:
def sort_obj_by_uid(objects,@attr)
@tmparray = Array.new
@tmphash = Hash.new
for object in @objects
if !
2006 Jun 12
1
MOH too loud
I ripped a rock-and-roll CD for a client's moh. But it's too loud. Is
there a simple way to reduce the gain without having to remix the tracks?
Thanks
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
mike@TelecomMatters.net
www.TelecomMatters.net
2006 May 04
1
SIEVE: Any information at all?
Hi,
I am quickly losing the will to live over the lack of any kind of information
at all about the SIEVE support that is listed in the dovecot features list.
Does anyone have any information at all? I've grepped through the source,
looked through all the configuration files, and googled a fair bit and come
up with nothing at all.
How would a user with maildir enable a sieve script?
2006 Mar 30
9
How is Teliax ?
Hi
I am looking at purchasing some DID lines from Teliax to install it on my
asterisk.
i would like to know some feed back on "Teliax" before i purchase.
suggest me if there are better sevice providers.
thanks
Giridhar Bandi
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