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What is the best asterisk platform to use? What are you guys using? I am looking for something to host either in our data center or at the customer prem where I have the control over the unit and not through a contractor. I dont mind paying a license fee for a front end interface but still would rather not have to pay. Thanks, --Eric -------------- next part -------------- An HTML attachment

Hi all, When moving from a self compiled Asterisk 13.23.1 to Asterisk 13.24.0, has resulted in a MWI clearing delay of around 5 minutes. After listening to a voicemail and deleting it, the Polycom VVX 601's MWI light is left on for around five minutes, before clearing. Installing Asterisk 13.24.1 did not fix this. Moving back to 13.23.1 allows the MWI to clear immediately. I see a note in

...by the phones at home? > > Thanks > Luca Bertoncello > (lucabert at lucabert.de) Try "sip show peer <peername>" for a phone. Then "sip show channels" during an existing call. And "sip show channel <Call-ID>" for more info. Michael http://www.mksolutions.info

...> > Btw: I did all tests with my father in law, since he had time for me > today, but the problem exists an almost all calls, incoming or outgoing, > no matter from/to which network provider... > > Thanks > Luca Bertoncello > (lucabert at lucabert.de) Michael http://www.mksolutions.info

Hi everybody, I'm looking for a solution for the following scenario: ? Asterisk queue ? At peak hours, there will be more callers then queue members/agents, so some callers will spend some time on hold ? Agents should be able to choose which of the on hold calls to answer instead of answering the next one in queue We already have a web interface where agents can see the callers on hold, so

I have a setup with Yealink phones & Asterisk Server (all latest patches). I am using BLF to display the states of other phones. While this works MOST of the time (busy, being called) it does NOT work when a phone is NOT regisstered at all, the yealink phones display a green dot EVEN if a phone is turned off (try explain this to users, they are shaking their heads!!!) I can see on the

Hi, Let say that in a call center, callers are recognized and categorized in 4 priority levels (priority 1 for Very Very Important Personalities, 2 for VIP, and so on) before entering a Queue. How can you make sure a priority 2 caller is answered before priority 3 callers, for instance ? I can think of several solutions but none really pleases me : 1. Have 4 different queues, set penalty value

Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI <--> Asterisk <--> T.38 <--> ATA <--> Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! Aloha, Matt

Am 13.06.2020 um 22:56 schrieb Antony Stone: Hi again, > 2b. Take your Thomson telephone to some other location with Internet access, > let it register to your home Asterisk server, and them make a call to the same > number yet again. I'm sure you can get the Thomson to connect to Asterisk via > some external network, since you say you can do this from your Android phone.

...mmunity.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users Michael http://www.mksolutions.info

Am 22.06.2020 um 17:01 schrieb Telium Technical Support: > I don't know if there was a prior email with more details, but.... > > Latency is as important as speed. Have you checked latency between your device and pop? What about QoS at your location, and does your ITSP support/respect QoS? That's a very good idea... Could you suggest me how can I check it? The Gateway is a

Am 13.06.2020 09:30, schrieb Luca Bertoncello: Hi again (again) I noticed right now another strange detail... I made a call using my mobile phone (connected to the Asterisk). The quality was top... Maybe is the problem in a codec used from our phones at homes? Could someone suggest me how to check the codec used by my mobile phone and the codec used by the phones at home? Thanks Luca

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

Hi, After an upgrade, I discovered yesterday strange things I would like to share here. Basically, I'me comparing platforms: The first one is a 2.6.26 (Debian Lenny) platform, with Asterisk 1.6.1.18, Libpri 1.4.10.2, Dahdi revision 8853 (must be between 2.3 and 2.5, I think). The second one is a 2.6.32 (Debian Squeeze) platform, with Asterisk 10.5.1, Libpri 1.4.12, Dahdi 2.6.1. Both are