Displaying 20 results from an estimated 45 matches for "768003".
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768005
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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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
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 23
1
Grandstream GXP-2000 headset
Hi all
What headset do people use with the GXP-2000? Any recommondations for
or against particular models?
Thanks
Peter
--
Peter Bowyer
Email: peter@bowyer.org
Tel: +44 1296 768003
VoIP: sip:peter@bowyer.org
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
2005 Oct 12
1
MWI for endpoints not registered at Asterisk
Hi,
We have phones registered at another soft switch, and would like to use
Asterisk as a Voicemail system.
Is it possible and how to configure Asterisk to send NOTIFY messages (for
MWI) to the endpoints that are not registered to the Asterisk?
Regards,
Stojan Sljivic
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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.
2013 Feb 07
2
Spatstat - Kest, envelope problem
Hello,
I am writing cause I have a problem when try to create confidence envelopes
in spatstat.
I have a point data representing firms in my study area.
When I try to create confidence envelopes the error message turns up:
[> p <- envelope(point, fun=Kest, nsim=99)
Error in marks.ppp(Y, dfok = FALSE) :
Sorry, not implemented when the marks are a data frame]
Could you write me how can I
2005 Feb 22
3
asterisk -vvvvvvvgrc?
what does the parameter
-vvvvvvvgrc
meanand are there any others as well?
Kindest
Muhammad Muzzamil Luqman
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2005 Jul 27
3
Read Back Caller ID Using Number Announcement in Digital Receptionist
I would like to setup an option in my digital receptionist that callers
can select to hear a read back of their Caller ID. It would be something
like, "the number you are calling from is...". I think I can reuse the
festival script that is built in, but ideally this could be accomplished
without using festival because Allison's voice is so much more pleasant.
I'm just a few
2005 Mar 22
0
RE: [Asterisk-uk] Meet
...touch.
>>
>> If you'd like to be involved, please email me directly or visit the
irc
>> channel.
>>
>> Hope to hear from people soon,
>
> Definitely interested....
>
> Peter
>
> --
> Peter Bowyer
> Email: peter@bowyer.org
> Tel: +44 1296 768003
> VoIP: sip:peter@bowyer.org
All,
You can count myself and my Simonb Clifton (Sales Director, Thorcom
Systems)
in as well.
Regards
Mike
--
Michael J. Tubby, B.Sc. (Hons)
Technical Director
Thorcom Systems Limited
Tel: +44 1905 888 007
_______________________________________________
Asteri...
2005 Jun 02
1
Newbie :Call Forwarding problem
Dear All,
I was trying to enable call forwarding, following the steps of the link
on voip.org regarding this issue it doesn't work and the phone I am
trying to implement on is still ringing. below is my conf in
extensions.conf and the CLI output during the process.
My configuration is :
exten => _*5X.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:2})
exten => _*5X.,2,Hangup
exten =>
2005 Feb 16
3
IAX2: Connection rejected
Hi there,
I am having a problem. It looks like this:
Feb 16 15:01:10 WARNING[11122]: chan_iax2.c:5546 socket_read: Call
rejected by XXX.XXX.XXX.XXX: No authority found
Feb 16 15:01:10 NOTICE[11122]: chan_iax2.c:1375 iax2_destroy: Avoiding
IAX destroy deadlock
-- Hungup 'IAX2/user/1'
Even I have entry in iax.conf for this user as a friend, and * server of
this user is already
2005 May 27
2
CRM integration (was RE: CallerID)
hello everyone,
I just had a thought on this subject why not create a daemon process on
the Client PC That registers its self and What phone the user is
connected. An AGI script could monitor the progress and when answered
could send a push to the registered daemon which would push a link to
the registered daemon on the telephone operators on the desk top. this
would not waist resource as much as