Hello Mason. It's easier than you might think. Here's what we do to
achieve the same effect. Note that, depending on how many digits you store
in your voicemail configs, you may need to change the number of stripped
digits, etc.
We use 3-digit dialing for voicemail, and the mailbox
I'd be curious about this as well. In Asterisk version 1.0.7, it
can't possibly work, unless my C reading skill is completely broken,
because the voicemail app isn't listening for a * but only for a # or a
0. That's also true of /app_voicemail.c/1.203/Thu Mar 10 19:33:15
In looking into this further, it appears that the problem is that
ASterisk is not properly responding to the 401 request that comes back from
BroadVoice. The code is there to do the right thing, and I can say that
ASterisk does the right thing when a 407 response is received from a
Hello. I'm not sure what's going on with the gentleman who is having
trouble receiving inbound calls as of this weekend, but I can say that
while inbound works for me, calling out through BroadVoice doesn't work at
all. SIP traces show that when I send an invite request out to
Hello. You can't have two phones login with the same extension. You
need to assign one phone to 101, and the other to 102. Set the user to 101
on one and 102 on the other.
-Brian
On Feb 11, 8:07am, Juki wrote:
} Subject: [Asterisk-bsd] Asterisk not accepting multiple SIP phone logins
}
Hello folks. I'm not sure if this is the right list for this
question, but I'll start here.
If I'm using a SIP provider and I have an entry in sip.conf that looks
like:
[8315551212]
type = friend
...
dtmfmode = inband
...
When I pick up the phone, call someone through this
Hello. Here is what my extension which uses distinctive ring on a
Cisco 7960 running V6.2 firmware looks like. Note that the distinctive
ring tones are changes in cadence, rather than changes in ringing sounds on
the 7960. Also, if you adjust the ringer volume wile the distinctive ring
[Try this again...]
Hello. Here is what my extension which uses distinctive ring on a
Cisco 7960 running V6.2 firmware looks like. Note that the distinctive
ring tones are changes in cadence, rather than changes in ringing sounds on
the 7960. Also, if you adjust the ringer volume wile
Hello. I've seen this behavior. What happens is that the
Grandstreams forget to continue registering with Asterisk after a while. I
bet when you find this happening, that sip show peers doesn't show ext/ext
ip address for the one that isn't working.
You can work around the
Hello. I have written a small patch to app_voicemail.c which provides
the precise functionality Steve wants. I sent it to this list once, and
got my subscription disabled for my trouble. so, if anyone's interested in
it, it's about a 50 line diff file, which I'd be happy to mail anyone
Hello steve. Here is a patch I wrote for app_voicemail.c which does
exactly as you describe. When the outgoing message is playing, if the
listener hits the * key, they're prompted for a mailbox and password,
whereupon they can check their voicemail as if they were using the internal
One thing that the BSD open source operating system projects do, and
many other projects for that matter, which Asterisk does not seem to do, is
put CVS ID tags in the source files of the package itself. If ID tags were
put into the source files, and even embedded in strings so that
Hello. I just purchased a Grandstream HT-286 from Chagres
Technologies. When I initially set this up, I accidentally mistyped the
new http password to get into the unit, and I cannot now log into the web
server on the device. the user manual has this to say about how to reset
the device
Hello. I'm running an asterisk system where the voicemail box numbers
match the extensions to which they belong. The phone numbers from the PSTN
which access the system are mapped to specific extensions, and if there's
no answer, they forward to their respective mailboxes so callers can
A cisco 1760 router, with a pair of dual FXO cards in it will work
fine. We've been using a couple of these for years, and they're quite
reliable, sound good, and behave themselves with Asterisk, using SIP. Not
the cheapest, perhaps, but a good choice.
If you want to save money,
I don't know if this helps, but I've been running our office IP phone
system on Asterisk, on a NetBSD-1.6.1 system for over a month now, with no
trouble at all. The functionality is limited at the moment, due to the
lack of the features provided by the zaptel drivers, but I hope to
Hello. The Cisco 7905 and 7920 phones are basically the same phone,
with the 7920 having a built-in ethernet switch. Sip and Skinny images
are available for these phones on the Cisco web site if you hav a CCO
account. I believe you select which image you want to run at boot time,
with
, and see
if I can make it go with Asterisk's Skinny module.
-Brian
On Jan 13, 11:54pm, Jan Czmok wrote:
} Subject: Re: [Asterisk-Users] Re: Again: 7920 Cisco IP Phone Skinny SIP
} Brian Buhrow ([EMAIL PROTECTED]) wrote:
}
} Hello. The Cisco 7905 and 7920 phones are basically the same phone
Hello. I think I understand your suggestion, but don't understand how
that's any different than the one I came up with. What I want, is to be
able to define a specific extension, and then have any external SIP phones
register with that extension that want to. It's important that
with asterisk at this
time.
On Thu, 25 Dec 2003, Brian Buhrow wrote:
Hello. I think I understand your suggestion, but don't understand how
that's any different than the one I came up with. What I want, is to be
able to define a specific extension, and then have any external SIP phones
register
Hello. I'm a new Asterisk user, but I'm impressed with the
flexibility and versatility of Asterisk, and am moving quickly to adopt
it's main-line use in our company. Hopefully, you'll be hearing more from
me as the project moves forward.
Right now, though, I have a question about
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