Displaying 20 results from an estimated 4000 matches similar to: "call to a particular 800 number never shows answered on Zap channel"
2005 Oct 07
3
call to a particular 800 number never showsanswered on Zap channel
Thanks for the reply. Forgive me for being na?ve, however have jumped in to this asterisk project at work due to some circumstances beyond my control and I don't know a lot about carriers and how this all works. I am figuring it out, but it's a lot of trial by fire.
As far as I know, we only use 1 carrier for our system. We have a PRI from NuVox and we use 7 channels for our asterisk
2005 Oct 11
1
call to a particular 800 numbernevershowsanswered on Zap channel
> Watch the output of 'pri debug span 1' on the Asterisk server while
> placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468)
> might be relevant.
Yes, this is exactly what is happening. Thanks a lot. I am thinking about adding a special case for the IBM 800 number since it is the only one my company is complaining about. Currently I have this in my dialplan:
2005 Oct 11
0
call to a particular 800 number nevershowsanswered on Zap channel
Watch the output of 'pri debug span 1' on the Asterisk server while placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468) might be relevant.
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Andy Goss
> Sent: Monday, October 10, 2005 5:58 AM
> To: Asterisk Users
2005 Oct 11
1
migrated to new ver on voip connection vs1 server voicemail problems
I migrated to a new version of the voip connection vs1 server software
and I am now getting these errors when I try to call my voicemail. Any
thoughts? The files are there, so I don't get it.
Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wav
file 49
Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to open
fd on
2005 Oct 11
5
help with broken voicemail
I can not figure out what the heck is going on. I went back to my old
version and I still get errors when the voicemail system tries to load
any of the greetings, unavail messages, etc. the normal voicemail
prompts work, but any user recording don't work. Leaving a new message
appears to work, but the system wont replay them, it throws errors.
Here is an example of the errors:
Oct 11
2005 Oct 11
2
error message when accessing voicemail
If anyone could tell me what this error is all about, I would be very
grateful.
Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock
path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not
permitted
Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock
path '/var/spool/asterisk/voicemail/default/5933/Old': Operation not
2005 Oct 17
6
initiate call recording from phone.
I am looking for a way to allow a user to record a call simply by
pressing a button or some combination of buttons on their phone. Is
this possible?
I have read the stuff about the monitor command; however, this doesn't
seem to be very interactive for the user.
Thanks,
Andy
--
H. Andy Goss
Network Engineer
Network Advocates Inc.
Main: 502.412.1050
DID: 502.992.5933
Mobile: 502.387.8216
2005 Oct 10
1
customize the pager email
I am running CVS-HEAD-04/12/05-21:44:31 and I am curious if it is
possible to customize the email message sent to the pager email address.
Thanks,
Andy
--
H. Andy Goss
Network Engineer
Network Advocates Inc.
Main: 502.412.1050
DID: 502.992.5933
Mobile: 502.387.8216
agoss@ntad.com
2005 Oct 12
2
Polycom: Button Remapping, HELP!
I need to find a way to have the Polycom phones automatically park
calls. Right now my users hit #70# (I know the last # is optional but
it speeds it up.) to park a call. Personally I think this is easy, but
my users would like one button to do this for them. The Polycom has
buttons we don't need (Transfer & Services), it would be nice if I could
remap one of those buttons to dial
2005 Oct 17
4
Polycom MWI
Hi,
I have lookedaround and don't see this anywhere. Is there a way to
tell the ip500 to not make the aural MWI blips?
2005 Mar 07
2
Strangeness with rsync
Hi all!
I've got a machine setup to be an "RSYNC Server", i.e. running rsync in
daemon mode waiting for connections from various other machines on my
network. This machine is running Debian Sarge and rsync 2.6.3.
For the past several days, I've been getting notices like this in my backup
logs:
============================================================================
=====
2006 Apr 18
12
Formatting data drawn from a DB
Question for all:
Right now i have a Table in a mySQL DB that has a row called
Ingredients. When the data is entered into the DB its enter like so
from a text area:
1 1/2 lbs. beef top sirloin, thinly sliced
1/3 cup white sugar
1/3 cup rice wine vinegar
2 tablespoons frozen OJ concentrate
1 teaspoon salt
1 tablespoon soy sauce
1 cup long grain rice
2 cups water
1/4 cup cornstarch
2 teaspoons
2012 Jul 24
4
Help from DOS Command Prompt
Hi all,
I am new to R.
I downloaded and installed R 2.15.1
I tried typing R.exe --help at the DOS Command line C:\", but I keep
receiving:
[quote]
R.exe is not recognized as an internal or external command
[/quote]
I tried many variations of R.exe --help, but roughly the same response
Any ideas?
thx
w
--
View this message in context:
2005 Jan 18
0
TDM400 - incomming call is answered but if i hang up asterisk never detects it
as the subject states I have a TDM400 that when a call is answered
asterisk runs the dialplan even if i hang up it NEVER detects the hangup
and I am also having a hard time with CID info I don't get that either.
most of our production machines are PRI based and I have little
experience with the TDM400 cards, ANY help would be appreciated.
Thanks in advance!
2008 Jun 19
1
Asterisk + zap + sangoma A104D - how to setup call using particular timeslot
Hi all,
I need to setup call using particular timeslot on one of E1's. I've
looked into
http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels and
it says that:
exten => TestTrakt,1,Dial(ZAP/1-2/517255333)
exten => TestTrakt,2,hangup
should work and force call setup via span 1 (port 1) but when I try
setup call rasterisk says:
-- Executing [TestTrakt at
2006 Feb 24
0
can't dial some particular numbers (providers ?) with asterisk sip / zap channels
I have a strange problem when calling some numbers with asterisk, I get
an hangup for busy condition even if the phone at the other end isn't busy.
I can route the calls via SIP to another carrier and then I have a SIP
code 486
or I can terminate them on digium cards (E1) and I have an Hangup code 17
I know for sure that one of the numbers is hosted by a different
provider than the one
2010 Oct 01
1
Multiple interfaces
Hi,
When is start one vpn i get the following result:
tinc10703003005 Link encap:Ethernet HWaddr C2:F7:7B:75:47:1A
inet addr:192.168.3.20 Bcast:192.168.3.255 Mask:255.255.255.0
inet6 addr: fe80::c0f7:7bff:fe75:471a/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:0 errors:0 dropped:0 overruns:0 frame:0
TX
2005 Feb 22
1
Astersik CVS HEAD + T1 e&m wink + IAX client doesnt detect call answered on Zap channel
Hello,
I've got very annoying behaviour from our asterisk PBX.
We have 12 channels T1 e&m wink start for TDM and using iax softphones
internally (iaxcomm, but tried firefly-thirdparty and discarded for
bad sound quality).
Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card.
In some cases when call is placed from softphone to TDM, system does
not detect call answered on Zap channel and
2010 Jul 20
4
Call not going through and failing because "never answered"
Hi,
I'm trying to use Asterisk to place Automated Voice Calls.
A verbose log from Asterisk CLI taken when I place a call in the spool
directory looks like this:
-- Attempting call on SIP/MTN-NEW/my-number for application
MP3Player(/myfile) (Retry 1)
== Using SIP RTP CoS mark 5
> Channel SIP/MTN-NEW-00000005 was never answered.
[Jul 20 10:52:11] NOTICE[14580]: pbx_spool.c:339
2006 Oct 24
10
Meetme... No channel type registered for 'zap'
When I call meetme:
exten => 1000,1,Answer
exten => 1000,n,Meetme(|||d)
Asterisk is complaing with:
-- Executing Answer("IAX2/xxx.yyy.142.204:4569-2", "") in new stack
-- Executing MeetMe("IAX2/xxx.yyy.142.204:4569-2", "|||d") in new stack
-- Playing 'conf-getconfno' (language 'en')
Warning, flexible rate not