Does anyone know if Gigaset is for sale in the USA? Based on my assessment of phones and features, i would like to try the N300IP base along with C610H phones. I can only find the handsets on ebay, no retailers in USA. And I suspect they are using European frequencies.
When a call comes in and is parked using the Park() command it appears that
the billing seconds do not include the time while the caller was parked. Is
there a simple way to correct this with an option or setting in asterisk.
This is not acceptable as we are loosing min when callers park calls. I am using 1.8.x release currently. Thanks
We've deoplyed a number of pure VoIP wireless (wifi & proprietary) phones, but not dect. Is there a simple overview of integrating DECT phones with Asterisk somewhere? I assume the DECT basestation has a multi-account SIP VoIP interface, and the handsets are just plain old dect? Can you push configuration info to individual phones? (Are they individually addressible / configurable through SIP) etc? Thanks
I work for a VoIP provider in Southern California. We are looking
for someone very knowledgeable in Asterisk/VoIP to help work on the following: - Maintenance of current Asterisk servers, updating Asterisk, monitoring
load, and other sysadmin tasks
- Devise and implement scalability strategies so that adding additional
capacity is easy and does not compromise anything about the current system
- Troubleshooting call quality issuses through our network (jitter,
audio dropouts..) Candidates should have the following experience:
- Minimum 3 years working with VoIP/Asterisk
- Have worked in an environment with…
In article <4FECCD0C.email@example.com>,
> On 6/28/2012 3:53 PM, Ernie Dunbar wrote:
> > We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a
> > Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen),
> > and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our
> > PRI to the PSTN and we hope will allow us to failover to other Asterisk
> > servers (ie, Voip2 and Voip3). Voip2 is our current production server,
> > and Voip3…
I have been fighting all night with version 1.8 and have not found a
way to do this with any command or Perl AGI->command. I need to play a
file and wait until the customer presses at least $maxdigits to
return, BUT, the file must continue playing until $maxdigits is
received or $timeout has expired. So far I found impossible to achieve
this functionality. Am I missing something?
I have a bunch of different customers on an Asterisk Box (the PBX).
This Asterisk Box is behind another Asterisk box that provides a PSTN
Up to this point I've been using IAX between the 2 Asterisk boxes, but
I would like to use SIP instead.
After doing some testing I have the following issue. If customer A calls customer B, but the call goes out through the PSTN
and comes back in, the call is rejected at the PBX because it wants
I can guess that this…
I'm currently running Asterisk 10.5.1, compiled from source, and just had someone call saying they couldn't get their voice mail. Looking into the user's voice mail folder, I saw a .lock file.
Removing this file, enabled them to get voice mail.
Is anybody else seeing this? The system is a new install and has only been running for a week with very little traffic (8 person office).