Displaying 20 results from an estimated 3000 matches similar to: "Register vs. Host=IPADDR"
2007 Feb 08
3
Automatic Dial, Play message
Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?
For example, you have a database of
FirstName LastName PhoneNumber
Jon
--
***
Forrest Beck
IAXTEL: 17002871718
jonforrest.beck@gmail.com
2007 May 03
3
SIP RealTime Friends
I setup sip realtime. Is it possible to use a type of friend? User
and Peer seem to work fine.
--
***
Forrest Beck
IAXTEL: 17002871718
jonforrest.beck@gmail.com
2007 Feb 25
1
Marks SNMP HowTo
I followed Marks SNMP howto on Voip Magazine and ran into a small
problem... (http://www.voip-magazine.com/content/view/2877/0/1/3/)
When asterisk is running as a non-root user (asterisk) SNMP request
for for the Asterisk MIB tree return nothing. If I quit asterisk and
run it as root, all is fine. Does anyone have a idea what is going
on? I have never used agentX, so I am unsure of what it is
2007 Mar 30
1
Paging
First off, A lot of thanks to this list. I have learned ton from
reading through the posts this past year.
I need some advise.
I have two group of phones connected to a single server.
Group1= SIP/2503&SIP/2504
Group2=SIP/3501&SIP/3502
I'd like to be able to dial an extension and page a certain group of
phones only if ChanIsAvail returns 1.
I am not sure how to go about
2006 May 11
2
Paging and Auto Answer on Grandstream GXP2000
I am looking to setup paging using the auto answer feature on the
Grandstream GXP2000. I am thinking I will follow the method as described
here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
I will setup the 4th account on the phone to auto answer.
Does anyone else have a method that works better? I also looked at the
allpage AGI written on Voip-Info. But it seems
2007 Apr 19
3
Outgoing CallerID
I am not sure of the best way to do this, so I thought I would query the list.
I have about 100 internal extensions ranging from 2000 - 2100. Each
internal extension has a external DID number. For example: 2001 =
5552871620. As you can see the internal externsion and DID don't
match in any way. What would be the best way to set the DID for when
a extension dials out on the PRI? In
2007 Apr 04
1
Polycom
I know this doesn't belong on this list but... I am looking to see if
anyone is using Polycom and knows of a web based software for
creating/managing the cfg files for polycom phones. I see that the
AsteriskNow will add provisioning support for Polycom phones. Since
it is still in beta, I was just looking to see if there was anything
else out there.
Thanks!
--
***
Forrest Beck
IAXTEL:
2006 Jun 28
6
Suggested Phone
We are looking to deploy asterisk at one of our locations that will have
about 50 phones. I have been buying different phones to test there quality
and feature set.
So far we have a
Grandstream 2000
Grandstream HandyTone 488
Cisco 7912
Polycom SoundPoint IP
And we are looking at getting a Linksys SPA-942
Anyone have a favorite?
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2006 May 31
5
SIP Presence
Does anyone have a working implementation of SIP Presence? I have a new
Grandstream GX-2000 phone with the supported hardware and I am not sure how
to setup presence with asterisk.
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2007 May 03
2
zttranscode crashes server
I was just looking to see if anyone else has seen this problem as well.
When asterisk starts up it loads the zttranscode module. The problem
exist when I use the init scripts to stop asterisk and then use the
zaptel init script to unload modules. Since the zaptel init script
didn't load the zttranscode module it will error out when trying to
unload the modules.
I built
2007 Sep 20
1
Paging MEETME_RECORDINGFILE Variable
I am having a weird issue with setting the recording file for the
Page app. Here is some quick background info
I have a macro that pages all my phones:
[macro-pageall]
; Context for paging all devices.
; This will search the sip table in the realtime database
; for all phones that start with a number. That number is
; passed to this macro as ${ARG1}.
;
; ARG1 = The
2006 Oct 31
1
Strange Characters in CLI on TTY9
When I look at TTY9 (using init.d and safe_asterisk to start the
asterisk process), I am getting some strange characters. When a
application is run the and the CLI shows the application executing the
languange almost looks russian...??
Anyone seen this before?
http://picasaweb.google.com/jonforrest.beck/AsteriskCLI
2006 Jun 23
2
Include Text file in Dial Plan
Is there a way to include a search of a text file in the dial plan?
I am trying to think of a good way to keep a sort of Blacklist file that is
checked against before letting a call through. If the callerid is listed in
the file, it will go to Hangup()
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2007 Apr 11
10
Nagios asterisk monitoring
Dear list,
I am trying to configure the nagios plugin called check_sip. I just read the
README file included with the plugin. I follow the readme steps to configure
the plugin, without success. In the nagios web interface I can see (No
output!) In the status information column. If I run the chech_sip plugin
from a linux console, I get
/usr/local/nagios/libexec# ./check_sip -u
2007 Apr 11
5
What is your Backup Strategy?
I was just curious to what your redundancy solution is. I have
considered many options, so I thought I would share and get an idea
for what others are doing. My setup is two different locations with a
10MB WLAN fiber link between the two. Each location has it's own PRI
as well.
I have considered and tested many options this last year or so.
1) Using hearbeat and drbd to monitor the
2007 May 04
4
zaptel compile error
I get the following error when trying to compile zaptel on CentOS 5
kernel 2.6.18-8.1.3.el5
CC [M] /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o
/root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c: In function ?
/root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c:171: error: ? has no
member named ?
make[3]: *** [/root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o] Error 1
make[2]: ***
2007 May 04
2
AsteriskNow!
Does anyone know how to gain access directly to the configuration files in AsteriskNow? I have dual NICs and need to change the binding in the config file. I believe they blocked ssh2 access by default.
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2007 Aug 15
2
Disable MoH for certain phones
Hi,
Is it possible to configure asterisk so it doesn't play MoH from certain
phones?
Regards,
Jan
2007 Oct 06
2
Change verbose level
Hi folks,
How I can change default level in asterisk from 3 level to 7level,
using the script /etc/init.d/asterisk
2008 Apr 14
2
polycom auto answer
I was trying to get my polycom phone to auto answer.
I added this to the dialplan. Used a different phone to call "22"
and the phone rang it did not auto answer.
Did I miss something?
exten => 22,1,SipAddHeader(Call-Info:=\;answer-after=0)
exten => 22,n,SipAddHeader(Alert-Info: Ring Answer)
exten => 22,n,Set(__SIPADDHEADER=Call-Info:\;answer-after=0)
exten =>