Yeah, not compiling for me either. Seems to be a problem with pwlib for
me. Anyway, if someone is feeling adventurous and wants to try a
different linux sip client:
http://snapshots.voxgratia.org/#sources
Bruno Hertz wrote:
Kris Edwards [EMAIL PROTECTED] writes:
I've seen some posts
Or: An IAXy by any other name is still an IAXy
http://nextalarm.com/abn.jsp
These folks are selling alarm supervision for us phone-line impaired
folks. What's scary about this is that we probably need to install a
fire/smoke detector with the IAXy ;-)
IAXy users on this list -- who of you
I just installed a new asterisk box with a wctdm with 4 FXO modules. The lines
in the office have terrible static (using standard analog phones) and this
static can obviously be heard through the asterisk box on the sipura sip phones
we installed. This by itself would not be a problem as the
Hola
I have a costumer whit this idea:
I am looking for a solution that will make a call based on SMS request. Can
you solve this problem with Asterisk?
Let me know if you have the solution and what exactly it does.
This is posible???
___
The UIP200 Units look and feel nice too, but there is something not quite
right in their SIP implementation. They will probably work OK, but if they
are like existing phones there will be niggling issues.
It's a shame too, the UIP200 comes very close to being an ideal small
business phone.
I don't have any IAXy's but do have Sipura SPA-2000's most of them get
really hot (what is it with this equipment??).
That doesn't make the IAXy a lesser choice for me, other than that the
price is quite high compared to other ATA's.
Any technical queries aside, I think this is a *very* cool
Is this possible?
I would like to access the web through my broadband at home by dialing
the home network. I think this is possible with a modem and a telephone
line connected to the home network. I was wondering if Asterisk would
allow such a data connection.
Thanks.
Hi, would you check if the entry is removed from the asterisk database? You
do that by issuing this command: database show dnd in the CLI.
Can you set/remove dnd on another channel?
Thank you
Thorben
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
It is possible with an ISDN card, (not with analog line)
Zapras application does that.
http://www.voip-info.org/wiki-Asterisk+cmd+ZapRAS
But it is not good to interface with analog modems, since you need to write
a module to handle the soft modem. As of now nothing like that yet to
asterisk.
On call forwarding just put the set caller id in the exact same way before
the dial statement. This way it will work if * is doing the forwarding
from your dial plan.
It will not work if the SIP phone is doing the forwarding because then the
* dial plan doesn't apply it will just be another
It doesn't do what I want (which is display caller id on the phones, not
the puter) but it looks like a reallly neat app. Too bad it isn't open
source, would be great if it could be hooked up to custom CRM apps.
Thanks for the link!
On Fri, 25 Mar 2005, Henry Devito wrote:
I don't have any IAXy's but do have Sipura SPA-2000's most of
them get really hot (what is it with this equipment??).
My SPA-2000s run warm to the touch, 80-90 degrees at most. That's not
terrible. If yours run hotter, stop calling those college coeds. But
all seriousness aside, my concern
On Sat, 26 Mar 2005, Jay Milk wrote:
I don't have any IAXy's but do have Sipura SPA-2000's most of
them get really hot (what is it with this equipment??).
My SPA-2000s run warm to the touch, 80-90 degrees at most. That's not
terrible. If yours run hotter, stop calling those college coeds. But
U can use the user :
maint with the password : password ..
Eng. Mohamed Farid ,,
Mediterranean Smart Cards Company ,,
Telecommunication Security Administrator ,,
Email : [EMAIL PROTECTED]
Phone : +20 23331439 / +20 2 3331400
Fax : +20 2 7621164
Mobile : +20 0122258350
On Fri, 25 Mar 2005, oi geli wrote:
Is it possible to invite a 3rd party into to
conference? Something like, conference is ongoing,
pressing # would allow to dial the number, if that
number answers, will be automatically added in the
conference.
I did search the mailing list, wiki pages
I tried to found documentation about openloop disconnect on
Asterisk/Zaptel. And up to now, I didn't find anything. Is openloop
disconnect supported by zaptel/wcfxo drivers?
Yes, it works for me and have verified by watching a voltmeter placed
across the pstn line and noting a
Jay Milk wrote:
What a silly thing to say :) Even the cheapest $25 UPS will run DSL,
switch and IAXy for more than the UL-required four hours. I'd be more
concerned about the DSL line being unplugged in the dmarq box outside.
I put a padlock on mine, but that's only an extra five seconds with a
IAXy users on this list -- who of you would trust their home/business
security to an IAXy? Is there a silent majority who loves this product,
or are the over-heating, looking-up issues reported as commonplace as
the frequency of the posts would indicate?
I would not trust the above to voIP
201 - 218 of 218 matches
Mail list logo
