Displaying 20 results from an estimated 1000 matches similar to: "MWI for endpoints not registered at Asterisk"
2004 Dec 07
2
modprobe ztdummy - failed
Hi all,
I have a problem starting the ztdummy. Here is what happens:
[root@asterisk /]# modprobe ztdummy
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'
1 error(s) detected
FATAL: Error running install command for ztdummy
After this, ztdummy is visible with lsmod, but when I try MeetMe, I get
following:
== Parsing
2004 Dec 09
2
MeetMe Features
Hi all,
I had a chance to use some call conferences that had some very neat
functionalities:
- When you call you are first asked for your name
- When someone joins the conference a message "<name> is now joining the
conference." is played.
- When someone leaves the room a message "<name> has left to conference." is
played.
How can I set MeetMe/Asterisk to have
2005 Jan 21
1
Voicemail Synchronization
Hi,
I have stress tested the Asterisk Voicemail.
We have encountered problem with simultaneous calls that are sent to the
same mailbox.
It occurred that several calls were writing to the same file.
It seems that there is a synchronization issue in the Voicemail application.
Did someone else find this issue?
What would be the solution/workaround for it?
Regards,
Stojan Sljivic
2005 Jan 28
4
Ouch ... error while writing audio data: : Broken pipe
Hi,
Can anyone help me with this:
I have downloaded latest stable version of Asterisk using the
asterisk-update.sh script.
Compilation and installation passed well.
When I start Asterisk I get the following error:
[pbx_realtime.so]Jan 28 09:35:08 WARNING[3253]: loader.c:258
ast_load_resource: /usr/lib/asterisk/modules/pbx_realtime.so: undefined
symbol: ast_load_realtime_multientry
Jan 28
2005 Aug 18
8
SNMP for Asterisk
Hi,
Is there a module within the Asterisk standard distribution that provides
SNMP features?
Is there any third party software for that purpose?
Regards,
Stojan Sljivic
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2005 Jun 14
1
Long time to detect hang-up
Hi,
I use Asterisk 1.0.5 and TDM04B.
When an incoming call over ZAP channel hangs-up, it takes 10 seconds until
Asterisk realize that.
How can I shorten the time of hang-up detection?
Regards,
Stojan Sljivic
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2004 May 14
4
sip authentication
Good day all
How do I get my asterisk and sip to use the password.I'm using x-lite.If
I use just the username and no password it still logs on?
Here is my sip.conf entry?
[101]
type=friend
callerid="Test User" <101>
context = test_1 ; Default context for incoming calls
username=101
secret=123456
host=dynamic
dtmfmode=inband ; Choices are inband, rfc2833, or info
2005 Mar 09
6
VoIPJet
Anyone suffering an outage with them right now?
I am getting the following from my box when I try to dial using them....
== No one is available to answer at this time
W
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2008 Apr 20
4
[Bug 15623] New: Support for Vimeo.com
http://bugs.freedesktop.org/show_bug.cgi?id=15623
Summary: Support for Vimeo.com
Product: swfdec
Version: unspecified
Platform: x86 (IA32)
OS/Version: Linux (All)
Status: NEW
Severity: enhancement
Priority: high
Component: plugin
AssignedTo: swfdec at lists.freedesktop.org
ReportedBy:
2005 May 11
3
Grandstream GXP2000 firmware update
I just downloaded the zip file from grandstreams website to upgrade my
gxp2000 firmware from 1.0.0.3 to the latest but seems there are some files
missing on the zip file... Anybody been able to upgrade their firmware?
My website shows this files as missing:
201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin
HTTP/1.0" 200 12737 "-" "Grandstream
2005 Jul 15
8
RE: 2 asterisks connected but trying to bridge
Hey,
For the bridge issue, take a look at 'notransfer=yes' option in your
iax.conf.
It'll force * to stay in the path
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg42262.html
2006 Jun 15
2
MWI not working
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2005 Mar 02
3
cvs stable and 1.0.5
I see that 1.0.5 is out. I thought that if I am tracking cvs v1.0.x I would
always get the newest releases. However, I just did a fresh update and
install from cvs stable and it reports as only being v1.0.3.
Should I just be using the tarballs rather than the cvs -r 1_0? Or maybe my
initial cvs was incorrect?
Thanks!
--
-M
There are 10 kinds of people in this world:
Those who can count in
2005 Mar 21
1
DISA Hangs up after DTMF is sent
Hey, this is happening to anyone who I try this with. We get into the
DISA, then hear the dial tone. Dial 1 then start dialing the number,
and it hangs up. I thought adding a wait time after the DISA may help,
I was wrong. Here is what I have thus far in the DISA extentions.
[DISA]
exten => 7,1,DISA(no-password||"Scheda" <565> 455-1337)
exten => 7,2,Wait(45)
exten =>
2005 May 19
1
ser+asterisk problem
hello
I am using ser with asterisk
asterisk on 5070 (on back end)
ser on 5060 (on front end)
i am getting all requests at asterisk.
i tried by changing asterisk port
bindport=5090
but still getting all requests from sjphone at
asterisk.
can any one tell what is the reason
regrads
Kamran
__________________________________
Yahoo! Mail Mobile
Take Yahoo! Mail with you! Check email on
2005 May 27
1
Temporary unavailable -????
The person on 617 is unavailable --- Why????
*CLI>
-- SIP Seeding peers from Astdb: '617' at 617@192.168.250.107:6990
for 3600
-- Executing Dial("SIP/601-f18a", "SIP/617|60|tr") in new stack
-- Called 617
-- Got SIP response 480 "Temporarily Unavailable" back from
192.168.250.107
-- SIP/617-602e is circuit-busy
*CLI> sip show
2005 Jun 29
2
New Asterisk documentation
Hello,
If asterisk.org can't provide you documentations have
a look here :
http://www.digium.com/index.php?menu=product_detail&category=software&product=ABE
I do hope some people understand my posts.
Regards
Harry
___________________________________________________________________________
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
2005 Jul 09
2
how to edit ring time
i dont how to edit the time for ringing "30000ms" to
"40000ms" when it displayed on console "Nobody picked
up in 30000 ms" and its very short time for ringing .
please if anyone can help me do it please.
____________________________________________________
Sell on Yahoo! Auctions ? no fees. Bid on great items.
http://auctions.yahoo.com/
2005 Sep 27
1
SIP Tandem Inbound only.
Hello,
I have a carrier that is supplying me with DID inbound over SIP to my asterisk
server. Because the CID is different with every call that is coming in the
only way I have to authenticate this carrier is IP based.
In my sip.conf I want to define this user as "type=user", however this can't
work because Asterisk only authenticates users by username, not IP.
I can take
2006 Jan 06
3
bayhamsystems.com experience
Hi all,
Anyone using their services ?
I'm thinking of setting up my servers with their service.
But before starting to mess with my extensions.conf I thought "let's check
the community for their experience".
Thanks,
Michiel van Baak.