6 dec 2012 kl. 16:54 skrev Danny Nicholas :
> Not sure about this since I use the 10/11 branches and not 1.8, but I think
> you need to use the deprecated call-limit for BLF and the new busylimit for
> the other features you need.
> http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
Hi! I'm helping set up a new Asterisk box. However, since I can't
just take the T1 to play with (and I *will* be making many changes,
e.g., going to Adhearsion), in order to test my dialplan, I'll need to
route calls through the old, Asterisk 1.4 box.
I've never really done this.
What's the
You probably have to do an incremental upgrade to 3.3.1 or something before
you can load 4.x
On Dec 6, 2012 3:10 PM, "Tim Nelson" wrote:
> I have a site with Polycom handsets on all the desks, mostly IP650s, some
> IP550s, and some IP450s as well.
>
> I need to update the firmware on the IP450s.
Tim,
What version are you on? There is a specific upgrade path for pre 3.3.
Dw
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On Dec 6, 2012, at 4:10 PM, "Tim Nelson" wrote:
> I have a site with Polycom handsets on all the desks, mostly IP650s, some
> IP550s, and some IP450s as well.
>
>
What happens if you reinitialize a phone, then do the update? (I keep a
bottle of Ibuprophen on hand just for Polycom issues).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Thursday, December
I have a site with Polycom handsets on all the desks, mostly IP650s, some
IP550s, and some IP450s as well.
I need to update the firmware on the IP450s. However, the firmware simply won't
load.
The latest firmware (4.0.3 Rev F) supports all phones at this site, and was
downloaded from here:
ht
When will packages.asterisk.org be updated with the RPM's?
Thanks
EKG
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
Development Team
Sent: Thursday, December 06, 2012 2:47 PM
To: Asterisk Users Mai
The Asterisk Development Team has announced the release of Asterisk 11.0.2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.0.2 resolves an issue reported by the
community and would have not been possible without y
The Asterisk Development Team has announced the release of Asterisk 10.10.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 10.10.1 resolves an issue reported by the
community and would have not been possible without
The Asterisk Development Team has announced the release of Asterisk 1.8.18.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 1.8.18.1 resolves an issue reported by the
community and would have not been possible witho
On 12/6/2012 12:32 PM, Carlos Alvarez wrote:
We are trying to set up a system where the calls from the queue show a
specific name or number on the phone. The calls would come into one of
a few dozen DID numbers, each one for a specific company. The agent
needs to know which company the call is
The reason I add a new column autoincrement is due to the fact I trust more
mysql about uniquness than asterisk.
Leandro
I am typing from my mobile phone...
Il giorno 06/dic/2012 19:11, "Ron Wheeler"
ha scritto:
> It seems like a safe thing to do.
> You could also ask about the impact of makin
We are trying to set up a system where the calls from the queue show a
specific name or number on the phone. The calls would come into one of a
few dozen DID numbers, each one for a specific company. The agent needs to
know which company the call is for and answer appropriately. I've done a
lot
It seems like a safe thing to do.
You could also ask about the impact of making an existing column a
primary key, in a MySQL forum.
Leandro's solution seems to be a good one as well and does guarantee
uniqueness.
Ron
On 06/12/2012 12:25 PM, Leandro Dardini wrote:
Yes, go for it. However
2012/12/6 Leandro Dardini
> Yes, go for it. However I have added another autoincrement column and
> created the primary key on it.
May I ask why did you choose to add a new column ?
I'm hesitating between both solutions.
For me, your solution has the advantage that, if, for any reason, a
unique
Yes, go for it. However I have added another autoincrement column and
created the primary key on it. On the other columns I need to search I have
created just an index.
Leandro
2012/12/6 Olivier
> Hello,
>
> I need to develop an application that will query (mostly reading) an
> existing MySQL C
Hello,
I need to develop an application that will query (mostly reading) an
existing MySQL CDR database.
This database (named asteriskcdrdb) was created during Freepbx 2.10 install
on my asterisk 1.8 setup.
This database has a single CDR table which is filled by Asterisk.
The tools I'm planning t
We occasionally get a sort of feedback/echo noise on our phones here. (Polycom
IP550 / Asterisk 1.8). It lasts for about a second, and it's described by
users as 'jingle bells'. It happens when people are using the speakerphone,
especially when it's on both ends. It happens on internal calls
Not sure about this since I use the 10/11 branches and not 1.8, but I think
you need to use the deprecated call-limit for BLF and the new busylimit for
the other features you need.
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
From: asterisk-users-boun...@lists.digium.com
[mai
Hello
We have recently upgraded our internal PBX from 1.4 to 1.8. This made the BLF
lamps on our Polycom phones stop working. After a lot of googling and a lot of
testing, I have been unable to find a solution.
I did try to change the call-limit value from 4 to 1, and this actually made
BLF wo
Asterisk wrote:
I followed
https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google to get
incoming and outgoing using google voice working.
However, when calling from google talk client, I see strange behaviour
(describe below):
You'll have to be specific - which google talk client?
On Wednesday 05 December 2012, Paolo De Michele wrote:
> hi all,
>
> I want have an information about ring group in asterisk (1.8.16 - centos
> 6.3)
> I have configured skypeforasterisk for incoming call to one extension
> and it works
> . [stuff deleted] .
> at right time the internal rin
100 extension on a row is not feasible... the queue strategy is the only
possible solution. If you check the queue.conf file you'll find you can
define a "Queue" and add as many members you like. One of the strategy
available is the "Ring all" where all the members in the queue will be
ring. You ca
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