1 of our customers records all phone calls and needs to be able to be
played back via a searchable web app. I tried ARI but it is very
limited.
Anyone have any ideas?
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asterisk-users mailing
Has this been corrected?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.
Sent: Wednesday, March 07, 2007 11:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Asterisk queue and agents
BJ
] On Behalf Of Hall, Eric
M.
Sent: Tuesday, April 03, 2007 11:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hints not working using SVN-branch-1.4-r59289
Group
I'm having trouble getting hints to work correctly using
SVN-branch-1.4-r59289
I have hints working on several
Just wanted to update the list
I found the problem. In my extensions.conf
I had
exten = 21,hint(SIP/21)
It should be
exten = 21,hint,SIP/21
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M.
Sent: Wednesday, April 04, 2007 1:41 PM
Group
I'm having trouble getting hints to work correctly using
SVN-branch-1.4-r59289
I have hints working on several other systems but I must be missing
something this time around.
VoIPGW*CLI show hints
-= Registered Asterisk Dial Plan Hints =-
[EMAIL
Just wanted to update the group
I updated asterisk to SVN-branch-1.4-r58833M and page no longer crashes
Asterisk. My below example works great.
Thanks!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M.
Sent: Friday, March 02, 2007 3:27 PM
, **2, 2) exited non-zero on
'SIP/36651-b7d1cf48'
VoIP-PBX*CLI
Disconnected from Asterisk server
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M.
Sent: Monday, March 12, 2007 7:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
post into that bug on Mantis a full
DEBUG/VERBOSE log and what it looks like when you do show queues
when one of these agents is on the phone, that'd be real helpful.
Thanks.
On 3/7/07, Hall, Eric M. [EMAIL PROTECTED] wrote:
BJ
Here is the sip.conf file. Hints work great. The only problem
] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.
Sent: Thursday, March 08, 2007 7:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Asterisk queue and agents
Asterisk SVN-branch-1.4-r58243
Voipgw*CLI show agents
56416(Jenifer Henry
the agent channel that the
represents that SIP device?
BJ
On 3/8/07, Hall, Eric M. [EMAIL PROTECTED] wrote:
Sorry
Forgot to tell you I was on exten 56405 called to my cell. I then called
into the Queue with another cell and this is the output.
Also forgot to include the show queue
voipgw*CLI
Have a question for the group
If I have an agent is on the phone outside of the queue should that
person still get queue calls ?
Doing a show agents online I see Available however show hints I see
inuse.
Any ideas
Eric Hall
Vice-president
Amaxx, Inc.
Customized IT Solutions
I think that is already set. Here is my queue.conf
[general]
persistentmembers = yes
autofill = yes
monitor-type = MixMonitor
[support]
musicclass = default
strategy = fewestcalls
timeout = 10
retry = 5
autofill=yes
autopause=yes
setinterfacevar=no
announce-frequency = 90
Not able to get the auto dialer part of asterisk to work with the zap
channel. It works great with the sip channel. Here is the call file and
the CLI output
Call File
Channel: ZAP/G1/6144994925
MaxRetries: 3
RetryTime: 40
WaitTime: 2
Context: amaxx
Extension: 36652
Priority: 1
-users] auto dialer
WaitTime stands for how long to wait until the call is considered NO
ANSWERED
Who can pickup a phone in 2 seconds, if not a robot? Try switch values
between Retrytime and WaitTime.
[]'s
MM
-Original Message-
From: Hall, Eric M. [EMAIL PROTECTED]
To: asterisk-users
Looks like it's a bug
http://bugs.digium.com/view.php?id=9172nbn=3
I have update to Asterisk SVN-branch-1.4-r58243 and will test it tomorrow and
report back to the list.
Eric Hall
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavio Ruiz
Please test with that and report your findings, and if it's still not
working find us on IRC as we'd like to take a further look and see
what might be wrong.
BJ
On 3/7/07, Hall, Eric M. [EMAIL PROTECTED] wrote:
Looks like it's a bug
http://bugs.digium.com/view.php?id=9172nbn=3
I have update
Group
In voicemail.conf I would like to having the following setup per
context not per-mailbox settings
serveremail
userscontext
fromstring
usedirectory
emailbody
pagerfromstring
dialout
sendvoicemail
callback
review
operator
volgain
nextaftercmd
forcename
forcegreetings
D
Not sure why this works
exten = _3665[0-9],1,goto(test|${EXTEN}|1)
but this does not.
exten = _366[50-59],1,goto(test|${EXTEN}|1)
I would like to route 36650 - 36700 to a Context 'test' however I'm only
able to get 10 to work at a time. Any ideas?
Any help would be great!
Group
I'm having some trouble with asterisk and the page cmd.
Any help would be great!
This is what's in my extensions.conf
exten = _**2,1,SIPAddHeader(Call-Info: answer-after=0)
exten = _**2,2,Page(SIP/36651)|d
exten = _**2,3,Hangup
CLI output
-1.4-r57207
Hall, Eric M. wrote:
Group
I'm having some trouble with asterisk and the page cmd.
Any help would be great!
This is what's in my extensions.conf
exten = _**2,1,SIPAddHeader(Call-Info: answer-after=0)
exten = _**2,2,Page(SIP/36651)|d
exten = _**2,3,Hangup
Looks like you
Has anyone got Asterisk IM to work
Using this link
http://www.sipalive.com/dev/asterisk/
And a clean install of asteris 1.4.0-Beta3
I get the following error
Any ideas? I have no idea what the .rej file is telling me so it maybe
easy to see it here but I'm a little out of my strike zone her!
Hello group
I have been asked to get IM via the X-Ten softphone to work with
Asterisk. Anyone have any ideas? I have looked on google and other
places with no luck.
Our system is as followed
Linux CentOS 4.4
Asterisk 1.4.0-beta3
X-Lite v3.0 for Windows
Thanks!
Eric Hall
Kenneth
Thanks for the reply. What I'm looking to do is listed here
http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging
However the patch does not work on the system listed below.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kenneth
Padgett
Group
I have app_swift working on our asterisk server running 1.4-Beta3.
My question is can you read variables with it? Like reading back
callerid number ${CALLERID(number)
Eric Hall
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Great link. After I all you said I get this error loading the module in
asterisk via load app_swift
The 'load' command is deprecated and will be removed in a future
release. Please use 'module load' instead.
[Nov 30 13:54:08] WARNING[7825]: loader.c:362 load_dynamic_module: Error
loading
Fixed my problem!
Note to self... READ EVERYTHING in the instructions!
Again thanks for the information!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M.
Sent: Thursday, November 30, 2006 1:56 PM
To: Asterisk Users Mailing List - Non
Using this link
http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt
This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3
I get the following errors on make install
Any help would be GREAT!
Thanks
[CC] app_cepstral.c - app_cepstral.o
In file
withAsterisk 1.4.0-beta3
Hall, Eric M. wrote:
Using this link
http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt
This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk
1.4.0-beta3
I get the following errors on make install
I'm looking to set up asterisk to call customer 3 days before the app
and remind them we will be out to see them.
I'm looking for any ideas on good ways to do this. Also I think it would
be best to do some type of text to speech however I do not like the
sound of the free one . Any ideas?
Does anyone have a quick howto and a sample to get
whisper paging to work? I'm running sterisk Asterisk 1.4.0-beta2
Thanks for your help!
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To UNSUBSCRIBE or
Here is an output from a 1.4.0-Beta2
voipgw*CLI show channeltypes
TypeDescription Devicestate
Indications Transfer
-- --- ---
---
Agent Call Agent Proxy Channel
I have this phone on my desk. It works very very well!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin
Anderson
Sent: Monday, September 25, 2006 10:25 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Spam? [asterisk-users]
Group
Any known problems
with Asterisk SVN-trunk-r43322 and Asterisk Recording Interface or the vmail.cgi
script?
I'm unable to see
voicemailsvia the web even though the MWI is flashing and if I look in
/var/spool/asterisk/voicemail/default/100/INBOX
I do see msg files
in that folder.
Group
Any
known problems with Asterisk SVN-trunk-r43322 and Asterisk Recording Interface
or the vmail.cgi script?
I'm
unable to see voicemailsvia the web even though the MWI is flashing and
if I look in /var/spool/asterisk/voicemail/default/100/INBOX
I
do see msg files in
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall,
Eric M.
Sent: Wednesday, September 20, 2006 10:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] HINT problems with SVN-trunk-r43322
Group
Looks like
Im unable to get HINTS working with the new SVN-Trunk
State never changed when ringing or on the phone.
Below is my configs (Maybe I missed something)
Thanks for any help you could give!!
##sip.conf##
[general]
callerid=unavailable
context=default
; Default context for
-Commercial Discussion
Subject: Re: [asterisk-users] HINT problems with SVN-trunk-r43322
On Wed, 2006-09-20 at 11:39 -0400, Hall, Eric M. wrote:
I’m unable to get HINTS working with the new SVN-Trunk
State never changed when ringing or on the phone.
Confirmed here, I only noticed because
Group
Looks like the
type=peer
call-limit=2
Works. Now the question is why? The sample I sent is working on a system
build 6 months ago.
Will do some more checking and will report to the list on anything I
find...
Thanks Bradley for this bit of info you gave!!
-Original Message-
I got the config working. Not sure if someone has
pre-recorded sounds for this app or not. Looked all over for them and I'm unable
to locate them.If anyone has sound file they would like to share that would help
me greatly.
Thanks
Sent: Friday, September 15,
2006 5:23 PMTo:
Group
Does anyone
have the FollowMe sound files? Do I need to record them?
Also does anyone
have a working followme.conf file that they would share?
Thanks!
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asterisk-users mailing list
To
Hello
group
I have a customer that has asked me to build an auto dialer that
will call customer a few day before an appt and remind them of the time and date
of the appt.
Does
anyone have any good links for apps that could do this type of auto calling?
They also request that information
I did not get this back from the list so I'm not sure if
thishit the list last week or not so I'm sending it again. Sorry if this
is a duplicate post!
---
Has anyone had problems with a Cisco 7970 running sip image
: [Asterisk-Users] Cisco 7970 problems
Hall, Eric M. wrote:
I did not get this back from the list so I'm not sure if this hit the
list last week or not so I'm sending it again. Sorry if this is a
duplicate post
Has anyone had problems with a Cisco 7970 running sip image
SIP70.8.0-2SR1S hanging up zap channels?
Calls to SIP and IAX
are fine. Just when the call goes out via the zap channels
I have some Cisco
7960 running SIP and they work fine.
Any
ideas?
Thanks-Eric Hall
Group
I have a Cisco 7970 Running the newest SIP image.
I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC
When I get a call the callerid number show something like
[EMAIL PROTECTED] I thought I seen somewhere what that was but I'm
unable to find the correct wording when
to change it to say Eastern instead of
Central
On Fri, 5 May 2006, Hall, Eric M. wrote:
Group
I have a Cisco 7970 Running the newest SIP image.
I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC
When I get a call the callerid number show something like
[EMAIL PROTECTED] I thought I
Aaron
Any idea how to change it from 24hr to 12hr ?
Thanks again!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M.
Sent: Friday, May 05, 2006 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: Spam? Re
I'm having trouble getting callerid name to show up on my phones (Cisco
7960 and a few softphones)
When I look in the CDR database I see the name but not on any phone when
being called.
I'm running
Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC
Any help would be great !
: [Asterisk-Users] CallerID Name problem
Hi,
What protocol for your 7960 phone? SCCP or SIP? You can turn on the SIP
debug on CLI to make sure the callerid and name pass to your phone.
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M
Subject: RE: [Asterisk-Users] CallerID Name problem
How are the calls coming into the PBX. PRI? If so add a Wait(1) before
your try ringing the SIP channel.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Hall, Eric M.
Sent: Monday, May
-Users] CallerID Name problem
You don't need the answer, But you need the wait. CallerID Name comes
over the FACILITY messge many times and it takes a slpit second for it
to come in.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Hall, Eric
M.
Sent
-Commercial Discussion
Subject: Spam? Re: [Asterisk-Users] CallerID Name problem
Do you get caller ID number? If so, WAITing is not going to help, since you
already get the info. If you get caller ID number, then your telco is not
sending the name.
On 5/1/06, Hall, Eric M. [EMAIL PROTECTED] wrote
:
[EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Monday, May
01, 2006 12:37 PM To: Asterisk Users Mailing List - Non-Commercial
Discussion Subject: [Asterisk-Users] CallerID Name
problem I'm having trouble getting callerid name to
show up on my
I'm looking for a good web based voicemail client that can use mysql or
realtime drivers. I can't seem to get vmail.cgi to work with realtime.
Thanks for any help you can give.
___
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: Hall, Eric M. [mailto:[EMAIL PROTECTED]
Sent: Sunday, March 26, 2006 9:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Web based voicemail client
I'm looking for a good web based voicemail client that can use mysql
or
realtime drivers. I can't
will let you let know if this works. There is also a
patch for zaptel but I believe this is for going from 1.3 to 1.4?
Thanks
Hall, Eric M. wrote:
Group
Having trouble installing zaptel. Below is my server specs
Intel Motherboard D101GGC
TE405P
CentOS-4.2-i386
Here is the output trying to do
Group
Having trouble installing zaptel. Below is my server specs
Intel Motherboard D101GGC
TE405P
CentOS-4.2-i386
Here is the output trying to do a 'make'
===
make clean
rm -f torisatool makefw tor2fw.h radfw.h
rm -f ztcfg torisatool makefw
Anyone have any idea what this is talking about.
Here is my zapata.conf
[channels]
switchtype=5ess
signalling=pri_cpe
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
callerid=asreceived
group=1
context=default
musiconhold=default
faxdetect=incoming
channel = 1-23
: [Asterisk-Users] Unknown signalling method 'pri_cpe'
Hall, Eric M. wrote:
[chan_zap.so] = (Zapata Telephony)
Mar 13 20:44:26 ERROR[10829]: chan_zap.c:10598 setup_zap: Unknown
signalling method 'pri_cpe'
Follow the correct order in installing Asterisk as shown on the download
page at http
Group:
I have a customer that is running the following
Asterisk CVS-HEAD dated 2005-08-18
WhitBox Linux respin 2
mysql Ver 11.18 Distrib 3.23.58
Cisco 7960G
We are using the real-time drivers for sip and everything is working
great.
They have a few employees that use the phones from home on
!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Tuesday, February 14, 2006 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Nat, SIP, Realtime problem
Hall, Eric M. wrote
-Commercial Discussion
Subject: Re: [Asterisk-Users] Nat, SIP, Realtime problem
Hall, Eric M. wrote:
Asterisk CVS-HEAD dated 2005-08-18
WhitBox Linux respin 2
mysql Ver 11.18 Distrib 3.23.58
Cisco 7960G
We are using the real-time drivers for sip and everything is working
great.
They have
I'm looking for a ay
to track when an agent logs inand logs out. Best if it could be put in a
mysql db but a text file will be ok for now..
Any help would
be great !
Thanks
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When a user dial voicemail and just hangs up or enters the wrong
password 3 times asterisk will crash.
We are using Cisco 7960G with SIP
My asterisk is CVS-HEAD built on 2005-08-02 23:47:59 UTC
Any help would be great!!!
Thanks
___
Asterisk-Users
] Voicemail crashes asterisk
It was fixed a while ago, download new code. There is a bug in the
tracker on it.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Hall, Eric M.
Sent: Wednesday, August 17, 2005 9:23 AM
To: Asterisk Users Mailing
Has anyone got vmial.cgi to work with realtime drivers?
___
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Looking for a good web app that will show agents that are login to
queue. I tried the operator panel but I'm unable to get the LED to
change color per the doco I have.. It works well for everything else but
no luck on the agent part..
___
Asterisk-Users
Not sure what this is.
When I call my own ext the call will ring for 10 sec and goto the
voicemail. However the phone will keep ringing and I see this on the
asterisk CLI
Jul 27 11:17:48 WARNING[3563]: chan_sip.c:8666 handle_response: Host
'192.168.0.200' does not implement 'PUBLISH'
Have no
-27 at 11:22 -0400, Hall, Eric M. wrote:
Not sure what this is.
When I call my own ext the call will ring for 10 sec and goto the
voicemail. However the phone will keep ringing and I see this on the
asterisk CLI Jul 27 11:17:48 WARNING[3563]: chan_sip.c:8666
handle_response: Host
Has anyone used the latest CVS HEAD and the Quad span T1/E1 5 volts card
from Digium. I'm not able to get it to load with a modprobe. I have a
T100P card and when I install that card it works without any trouble
___
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I'm having trouble
with the latest cvs HEAD (7/22/05) and myWildcard TE405P I just got in
from Digium. I'm not able to get podprobe to work with the release. I get an
error "unable to install" however when I grab the stable it works great but no
realtime drivers for asterisk.
I also tried
Group
This is strange. When I call my voice mail extension the system does
not pick up my touch tone entries. I have x-lite softphone and a cisco
7960 for my hard phone.
When I call from outside I'm able to check my voice mail without any
problem.
Any help would be great!
Group
When I dial a phone number that should go out to the telco my local
phone rings. Does anyone have any hits ?
Thanks
Asterisk Ready.
*CLI -- Called g1/6144196143
Urgent handler
Urgent handler
-- Starting simple switch on 'Zap/2-1'
Urgent handler
Urgent handler
-- Called
useincomingcalleridonzaptransfer=yes
callerid=asreceived
echocancel=yes
echocancelwhenbridged=yes
rxgain=1.0
txgain=1.0
musiconhold=default
Thanks
Eric
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M.
Sent: Monday, August 23, 2004 8:32 AM
My info
Asterisk CVS-HEAD-08/04/04
Redhat 9.0
T100P connected to Telco with 12 Digital trunks WINK start.
I'm able to dial out and able to get calls coming in but my inbound
calls do not display callerid information. Its only shows asterisk
Telco tells me callerid is turned on and working..
Have you tried to run * in debug mode? I have the same problem and I
found that if I run * in debug (asterisk -vgcd) mode MOH works. I
have no idea why but that is the only way I can get MOH to work for me.
Good luck and please report back to the list if you find a fix!
-Original
Has anyone install zaptel-1.0-RC1 on Fedora Core 2?
First thing I found is I need to have a link to 2.6 from 2.6.5
ln -s /usr/src/linux-2.6.5-1.358/ /usr/src/linux-2.6 fixed this problem.
Now I get this.
Install gets this error
make[2]: *** [/root/asterisk/zaptel-1.0-RC1/zaptel.o] Error 1
Looks like the 2.6X stuff is not ready yet..
http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation
-Original Message-
From: Hall, Eric M.
Sent: Wednesday, July 21, 2004 6:15 PM
To: '[EMAIL PROTECTED]'
Subject: Install problems
Has anyone install zaptel-1.0-RC1 on Fedora Core 2
On Wed, 14 Jul 2004, Hall, Eric M. waxed:
FC1
What I don't understand is why it works using the -vgcd but not
when just running asterisk ?
Are there any log messages about the mp3 player not being spawned ?
Like Fork failed or unable to spawn mp3player
?
I am unfamiliar with how FC1 starts
I have been working on the music on hold part for a few hours today and
I found something that just doesn't sound right.
If I just run asterisk via service service asterisk start' everything
work but MOH
If I run it via asterisk -vgcd MOH works...
Any idea what the difference is ?
on hold
On Wed, 14 Jul 2004, Hall, Eric M. waxed:
I have been working on the music on hold part for a few hours today
and I found something that just doesn't sound right.
If I just run asterisk via service service asterisk start' everything
work but MOH If I run it via asterisk -vgcd
Group
Everything is working great with my * server. That's to everyone for all
your help!!!
I have a problem that I can't seem to find a fix for. When I'm on a
call and someone calls in the system never picks up. Also I'm unable to
place calls out if someone is on the phone.
Here is what I have
Hello group
I'm working on getting festival installed and working on my FC1. I ran
into a problem and after searching Google I found this message talking
about a patch for Speech Tools and Festival
http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html
The above site does not have
I had the same problem. What I found is I needed to set register with
proxy to yes in the sip config.
Hope this helps
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben Merrills
Sent: Thursday, July 08, 2004 7:01 AM
To: [EMAIL PROTECTED]
Subject:
I know this is a little off list but I can't think of a better place to
ask this question.
I upgrade the phone to 7.1 and it installed the Universal Application
Loader. Now I'm getting Protocol Application Invalid after it reads tftp
SIP(MAC).cnf
Any ideas?
Again sorry this is off topic
7960
On 08/07/2004 at 08:21 Hall, Eric M. wrote:
I know this is a little off list but I can't think of a better place to
ask this question.
I upgrade the phone to 7.1 and it installed the Universal Application
Loader. Now I'm getting Protocol Application Invalid after it reads
tftp SIP(MAC
an outside phone number as part of
a hunt
Hall, Eric M. wrote:
I'm trying to see if this is even possible.
AFAIK Asterisk has no way of knowing if you do not answer. To Asterisk,
the call is complete and answered when it starts ringing. A PSTN/POTS
call is always going to be the final destination
I search Google to find how to get the message light to flash on my
Cisco 7960 running (Application Load ID POS3-06-3-00) (Boot Load ID
PC03M030) (DSP Load ID PS03AT38)
All I see is about the sip.conf file witch mine has the mailbox= but
still no light. Also the messages button does not work.
I have2 X100P
card and configured everything based on configs here http://www.onlamp.com/onlamp/2004/01/22/examples/config_files.txt
I changed the area
codes to match mine.
When I try to dial
out I get
app_dial.c:554
dial_exec: Unable to create channel of type 'Zap'
A zap show channels
I did as you stated however I get the same error. Here is
my config file. Did I miss something?
Thanks
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wade J.
WepplerSent: Monday, July 05, 2004 10:54 AMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Question
about
Ruuing * in debug I get this
*CLI Jul 5 11:21:02 NOTICE[-1221170256]:
app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' ==
Everyone is busy at this time
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wade J.
WepplerSent: Monday, July 05, 2004 10:54 AMTo:
I'm trying to see if this is even possible.
When you dial ext 2000 I want it to ring my sip phone for 20 sec then
call my cell and let it ring for 10 sec if I do not pick up the call on
my cell I would like it to go back to * and leave a voice message for
me. Here is what I have so far in my
Group
Following the information located on
http://www.asterisk.org/index.php?menu=download
I get the following error installing the zaptel
Any help would be great!!!
Thanks
[EMAIL PROTECTED] zaptel]# make clean; make install
rm -f torisatool makefw tor2fw.h
rm -f zttool
rm -f *.o ztcfg
PM, Hall, Eric M. wrote:
I'm starting to get this to work! Well I got Voice Mail to work!
All calls goes to voice mail without ringing the users phone
(iaxComm).
Here is my iax.conf and my extensions.conf
Any help would be great!!
I don't see anything really obviously wrong here, although I
I'm starting to get this to work! Well I got Voice Mail to work!
All calls goes to voice mail without ringing the users phone (iaxComm).
Here is my iax.conf and my extensions.conf
Any help would be great!!
Thanks
extensions.conf
Description: extensions.conf
iax.conf
Description: iax.conf
I'm trying to use iaxComm and I get the following error.
Apr 1 16:18:04 NOTICE[1142106560]: chan_iax2.c:3393 register_verify: No
registration for peer 'asterisk' (from x.x.x.x)
I'm VERY GREEN with this software so any help on list or off list would
be great
I downloaded iaxComm and get up my iax.conf file and the
extensions.conf. Here is the out but from CLI in iax debug. What did I
forget to do???
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ
Timestamp: 1ms SCall: 10489 DCall: 0 [192.168.50.66:4569]
I have a question for the group.
To get this running do I need any Digium Cards? I understand I will
need them to connect to the public phone system. I'm looking at just
using IP Phones or IP Softphones just to test this app.
Thanks for any help you could give.
2:43 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbie
Hi,
On Wed, 2004-03-31 at 16:00, Hall, Eric M. wrote:
I have a question for the group.
To get this running do I need any Digium Cards? I understand I will
need them to connect to the public phone system. I'm looking
] On Behalf Of Nicolas
Gudino
Sent: Wednesday, March 31, 2004 2:43 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbie
Hi,
On Wed, 2004-03-31 at 16:00, Hall, Eric M. wrote:
I have a question for the group.
To get this running do I need any Digium Cards? I understand I will
need them
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