Displaying 20 results from an estimated 3000 matches similar to: "Grandstream BLF and Call-limit"
2008 Feb 11
2
Grandstream GXP2000 Loses Connectivity
I have 20-30 GXP2000's connected to * over a T1 line. Neither end is
NAT'd and there is plenty of bandwidth available over the line. The
GXP's are 1.1.5.15, which is the latest. I have a problem where the
phones keep dropping off of * and I get a "failed to register" message
in the log of *. Sometimes they eventually connect and sometimes, I
have to reboot them to
2007 Nov 08
3
'a' extension
Is there any way to see the called number when a call gets redirected to
the 'a' extension from voicemail? Say x123 calls x456 and it rolls to
voicemail. x123 hits * and gets dumped into the 'a' extension in the
original context. I need some logic in 'a' to do a database lookup
based on the original called number (x456). Any ideas? When I do a
test, it appears
2006 Jan 18
2
1.2 in production w/100+ phones?
Is anybody using 1.2 (or 1.2.1) in a production network using Realtime
(voicemail, sip or extensions) with 100+ SIP phones? If so, what are
your experiences? We've been running 1.0.3 for about a year and it's
been rock-solid. We'd like to upgrade to Realtime and 1.2, but I'm
afraid of killing our stability. Obviously, we'd do it in stages
(upgrade to 1.2, then realtime
2006 Dec 02
2
"Low" beep on voicemail
We've had a few people complain that the "beep" before leaving a
voicemail is not loud enough and too short. Does anybody have a
recorded beep that they can share, that is a little louder and a little
longer? We've had this box in production for 2+ years, so I hate to
mess with the gain on the PRI or anything like that because everything
else works fine.
I know nothing
2008 Jul 15
1
sip prune realtime per issue
I am using realtime on two boxes, one running 1.4.10.1 and one running
1.4.11. Everything works fine except for when I make a database change,
such as a phones password. I change the DB, I prune the peer, I see it
is gone and then I see it show up again in "sip show peer xxxx", but
everything is not being updated. The phone will not register even
though the DB and the phone have
2007 Sep 07
3
Show Callee name on Display
We have users with Cisco 7900 phones running sip. When user A calls
user B, we want user B's name to appear on user A's phone. It shows the
extension they call, but not the internal name of the called user. Is
this possible? We have some people that used to be on an MGCP based
system and they would get the callee's name popup on their phone when
they called someone. I
2007 Aug 10
2
Asterisk Manager to Record Greetings
I am trying to use Asterisk Manager via php to record auto attendant
greetings and I just can't figure out how to do it. I've got the php
page working and I can click to call between two phones. However if I
click to call just a single phone and then try to enable "monitor", when
I pick up the ringing phone, it just hangs up and doesn't record
anything. I'm sure I
2007 Aug 20
4
Realtime Queue Members
Does anybody have realtime queue members working? Not the queues
themselves, just the members. I have realtime working for voicemail and
sippeers, but I can't get queue members to work. Here is what I have:
res_mysql.conf:
[general]
dbhost = 127.0.0.1
dbname = ASTERISK
dbuser = myuser
dbpass = mypass
dbport = 3306
dbsock = /tmp/mysql.sock
queues.conf:
[general]
2007 Dec 06
1
s, CDR and NoCDR in v1.4.10.1
I am running 1.4.10.1. I have a macro that is called from default for a
certain extension (both below). I added NoCDR to s to try and stop
extra CDR records, but I am still getting them. Any idea how to stop them?
extensions.conf:
[macro-STDEXT]
exten =s,1,NoCDR()
exten =s,2,Dial(${ARG1},30,Tt)
exten =s,3,Goto(s-${DIALSTATUS},1)
exten =s-NOANSWER,1,Voicemail(${ARG2}|u)
exten
2007 Oct 26
1
Voicemail Options
I know that you can set it up to where a user hits 0 from their mailbox
and goes to an operator, but can you set up other options as well?
Could I have 0 for an operator and 1 to go to another extension? I know
you can do this by building an AA, but I don't want to have to do that
for every user as there are about 40 people that want this. They won't
all go to the same number.
2007 Aug 22
2
Multiple servers using realtime
I am in the process of setting up several * servers using realtime and
connecting to mysql. I am trying to figure out if I should just use one
database and one set of tables for all of the user data. Or if I should
have separate databases for each * box. The boxes are independent of
each other in that customerA only connects to box A. They will never
fail over to box B or anything like
2006 May 12
2
Voicemail WAV to PDA Problems
Our asterisk server has been up and running for over a year and it works
great. I have emails going to my account as an attachment and I can
listen to them on the desktop and it works fine. I just got a T-Mobile
MDA that runs Windows Pocket (or whatever they call it) and it can check
email. If I have it download the email, it gets the attachment, but it
can't seem to play it (it CAN
2007 Apr 17
1
TM Malaysia E1 PRI signaling
Anyone configured a E1 PRI in Kuala Lampur, Malaysia with TM Malaysia?
What signaling did they provide, framing, formatting?
primary-4ess Lucent 4ESS switch type for the U.S.
primary-5ess Lucent 5ESS switch type for the U.S.
primary-dms100 Northern Telecom DMS-100 switch type for the U.S.
primary-dpnss DPNSS switch type for Europe
primary-net5 NET5 switch type for UK,
2009 Jan 14
2
Set caller ID to anonymous
Hi guys,
I am trying to set the caller ID to 'Anonymous <anonymous>' if the caller is not registered to the asterisk server. But I can't find a solution.
Any ideas?
Regards Philipp
--
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f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a
2006 Jun 12
2
Attended transfer and queue
It seems that Asterisk does not free up agent after attended transfer. The agent stays in 'busy' state for as long as the conversation between the caller and person, to which call was transfered, is active.
Does anyone know any workarounds for this problem?
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2007 Apr 16
2
[OT] Nokia E60 firmware update break SIP
Just a warning for you all that are using Nokia series E phones for SIP
function.
I updated my phones firmware today using the Nokia Updater, and now
the SIP functionality, which previously worked pretty well is
completely broken.
The phone no longer registers with asterisk, although it displays the
little icon as though it has, and it doesn't even seem to try to pass
calls to
2007 Apr 18
2
SIP failover between Sip Providers
Hi all,
lets say I've registered at several Sip-Providers. Provider A offers
best rates but is often too busy to get a line. Sip Provider B is stable
(but more expensive). The asterisk box has a high call volume therefore
problems at provider A will get obvious after a few calls stalled. In
this case astersik shall switch temporarily to provider B but shall test
periodically for selected
2006 Jun 27
5
WebPhone
Hi,
someone know a good webphone, possibily a free one
Thx
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2007 Apr 09
2
Privacy Manager w/ No Recording
Is there a way to use privacy manager without requiring the user to
enter their name? Essentially I am just looking for a way to force the
called user to enter "1" to accept the call. I don't need a name
recording. I want a call to come in, a message to be played, music on
hold, call out to the called party, then enter "1" to accept, "2" to
reject.
Peder
2007 Jan 03
5
Polycom Power Specs
Does anybody happen to know the input power specs for the Polycom IP 500
and IP 600? We've mixed up our power supplies and we've got a whole box
of them and can't figure out which go to the Polycoms. I would rather
not kill the phones by trying random ones....