I've recently upgraded [EMAIL PROTECTED] to CVS HEAD and in addition to losing the
ability to use the MySQL database, I've noticed that my MOH has degraded
significantly. I've tried all sorts of remedies -- removing the x100p
card and loading asterisk without the zaptel drivers and such --
Strange things.
When I run the RxFAX command through an internally dialed extension, I
can *hear* fax tones, meaning, I presume, that the RxFAX application is
running. In fact, doing a show application confirms that. So, I'm
presuming RxFAX application is talking as it should.
However,
Perfect!
Thanks!
B.
Kevin P. Fleming wrote:
Jeffrey Starin wrote:
However, I don't believe a full restore is needed -- I just need to know
the names of the directories under /var/spool/asterisk and re-create
them (I hope!). Can some kind soul give me some direction or tell me
911 Help!
I accidentially deleted all directories under /var/spool/asterisk
I did use the backup facility not too long ago but cannot find the
process for restore.
However, I don't believe a full restore is needed -- I just need to know
the names of the directories under /var/spool/asterisk
I forgot to mention in my previous post that as well as:
/CFIM/2000 :12125553434
showing up in the database, the phone is registered properly and also showing
up in the database show. For some reason it
seems that Asterisk is not reading the database before connecting the call.
Does
I have the standard script for activating call forward and when I do a
database show, I indeed see:
/CFIM/2000 :12125553434
so I presume that means call forwarding is in effect. However, when
anyone dials extension 2000, it rings and no forwarding takes place.
Is there something
Hello, All.
Been working successfully with Asterisk for a while now and have no
complaints. However, I cannot seem to find any information on how to
control the Refresh time period.
I fiddle with the maxexpirey and defaultexpirey values in the general
section of sip.conf but when I do a
I have MeetMe working fine except I have a little confusion over the i
option. Is that supposed to allow for announcing to the conference the
recorded name of the individual who is entering the conference?
If so, it's not working in my verions of Asterisk HEAD. I am using
ztdummy because I
Hi!
I have MeetMe working fine except I thought that if I used the i
option, there would be some kind of announcement made upon a new meetme
person joining a conference. Is that correct? Is it the i option?
Because I'm not hearing an annoucement when a new person joins the
conference.
Tnanks for the reply.
Well, I am only using SIP, I do not have any digium cards or IAX
protocols. The Asterisk box only understand the initial dtmf tones if
FWD is set to inband, anything else the caller cannot even transfer to
the extension that provides DISA. Here is some other information
connection. I'm pretty stumped on this
one.
Steve Maroney wrote:
What about your dialplan ? Make sure the DISA app is going to correct
context that contains the extensions that you want to dial.
Thank you,
Steve Maroney
On Sun, 15 May 2005, Jeffrey Starin wrote:
Tnanks for the reply.
Well, I am only
I have a little confusion about the general settings (other than the
register values) in the SIP
General area. I understand that for examle in a SIP context like [FWD]
or [BROADVOICE]
the entries in those areas are ths settings that take effect in any
communication woth FWD and/or BROADVOICE.
the extensions that you want to dial.
Thank you,
Steve Maroney
On Sun, 15 May 2005, Jeffrey Starin wrote:
Tnanks for the reply.
Well, I am only using SIP, I do not have any digium cards or IAX
protocols. The Asterisk box only understand the initial dtmf tones if
FWD is set to inband, anything
.
Thanks a million!
J.
Johnathan Corgan wrote:
Jeffrey Starin wrote:
I have a little confusion about the general settings (other than the
register values) in the SIP
General area.
[snip]
However, I'm confused as to the purpose of the
general settings -- to what or which connection do they apply? Since
I have a setup which allows users to access my asterisk box via FWD.
That is, a user in say, France can call into a local access number for
FWD, then hit number 7 which dumps them into a DISA request for a
password, which then dumps them into my internal extension so they can
dial out through
respond to the tones.
Any other suggestions?
B..
Andrew Kohlsmith wrote:
On May 14, 2005 11:58 pm, Jeffrey Starin wrote:
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
dtmfmode=inband
It doesn't take a guru. :-)
you can't use inband signaling if you are allowing the possibility
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