Hi Nahid,
I think youll want a fax on-ramp and off-ramp on your asterisk boxes instead
of trying to send a fax using VoIP (SIP). I believe it is possible but not
recommended. There are technical reasons for this that you can find online
in many places.
Basically asterisk answers the fax and
to the datasheet:
http://www.digi.com/pdf/prd_mca_datafirequad.pdf
Ive only used asterisk with Cisco SIP gateways so Im not sure if this is
enough information.
thanks again for any help,
cheers,
Mick Hastings
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Hi Clive,
Thank you for your response to my posting.
It looks to me that the intel board is the same as the dialogic board
Can you please tell what that means? I haven't worked with any BRI cards
before so I don't know if it's a good thing or a bad thing.
Is / was the dialogic board
can anybody offer any information as to where I should start to look for
more informaion this topic?
I really appreciate the help.
cheers,
Mick Hastings
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Asterisk-Users mailing list
Hi Nabeel,
I also wrote a siliar script using the same tools, I found I still had a few
problems with it (Im also a terrible programmer) and dont use it anymore.
However, on a windows system you can try the following 2 programs and get
integration with outlook and/or cut and paste dialing.
Hi Eric
I started getting these 'WARNINGS' (WARNINGS are not ERRORS) when I started
using the manager interface (IPSwitchboard / AstWinManager). I sometimes
also get some extra delay when I see these but havent determined if its
related.
If I disconnect my manager program the warnings
Hi all,
I currently have a setup where my users dial in to a dedicated DID that
sends them to VoiceMailMain(). this works fine except for the fact that
nobody can remember the number! (they already have to remember the main
number, their personal number, fax number and mobile number)
What I
Hi Robert,
I just set this up today for dialing international using a calling card
account.
usually we call 0120 982 433
wait for voice prompt
then dial the number
i set it up so the user only has to prefix with 011 then the number like
this:
[brastel]
exten = _011.,1,Dial(SIP/[EMAIL
Hi Folks,
I am running asterisk 1.0.7 with no probs.
Howerver, during my testing stages I used CVS head and was using the
forcegreetings and forcename feature. I am now setting up a production
server and want to use these features on the stable version but its not
there?
Is there a patch I
]
Hi,
Do you have load chan_oss/chan_alsa ? if not then you can't use 'Dial'
app.
Hope this help. :)
Mick Hastings on 2005/1/26 03:31 wrote:
Hi Floks,
snip
*CLI Dial
No such command 'Dial' (type 'help' for help)
*CLI
the same thing for Answer, Hangup, etc
what have I missed?
cheers
Hi Floks,
This is probably really dumb but here goes:
I used to be able to place calls to my SIP phones from the CLI using the
'Dial' command for testing. I have installed asterisk on a new machine and
copied over the .confs and started it up. It all works fine. But when I try
to initiate a
Hi all,
I am wanting to add this patch to my asterisk:
http://bugs.digium.com/bug_view_page.php?bug_id=0002266
but I dont know where to start.
I have checked the wiki and read the patch man page and tried a few things
in a few different directories but no luck. I am really flying blind here,
Hey Olle,
thanks for this info, I have the files but no idea what to do with them, any
pointers would help.
cheers,
Mick
Olle E. Johansson [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Mick Hastings wrote:
Hi Folks,
cheers for all the great info on the list.
I need to create
Hi Folks,
cheers for all the great info on the list.
I need to create a SIP NOTIFY message to reboot my Cisco 7960 phones but I
dont know how.
The admin guide gives an example of the packet (attached), I have tried a
few web searches and found some cool
little programs that generate SIP packets
Hi Norman,
I played with this for ages also. I think there is a small step missing from
the wiki that needs explainantion.
Prior steps in the SJPhone setup:
1/ click on the Options button
2/ go to profiles tab.
3/ click on 'New'
4/ create a new profile called 'asterisk' with profile type
Hi folks,
thanks for your help with my last question re: japanese FXO. It doesnt sound
very compatible so I will use a SIP FXO gateway then.
Untill I find one, im just trying to get my 2 cisco SIP phones talking to my
* server. just as a learning experience for now. heres what I have so far:
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