Experience with Eicon Diva PRO 3.0?

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Hello *, is here someone with an experience of the "Eicon Diva PRO 3.0"? I need 2 or 3 cards to connect an "Anlagenanschluß" (I do not know the
name in englisch, but is 2 or more lines with the same numbers) to my
intranet servers (IBM x335/x345 -> requires PCI-X or at least PCI 2.1). Thanks, Greetings and nice Day/Evening
Michelle Konzack

Asterisk Users 3.5 years ago 1 Answer

Troubleshooting one-way audio with H.323 trunk between Asterisk and Avaya IP Office

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I'm attempting to configure an H.323 trunk (using chan_h323) between an Asterisk box and an Avaya IP office. It mostly works. Calls from Polycom SIP devices registered to Asterisk can place calls over the trunk to IP Office extensions and everything works great. However, calling from an IP Office handset to any of the Polycoms results in a one-way call where the Polycom can not hear the Avaya. I think what's happening is somehow, the Polycom is receiving two RTP streams, and one of them is silence. I think this because if I place the call on hold, I will…

Asterisk Users 3.5 years ago 0 Answer

Deadlock detected in asterisk-1.8.9.0 x86_64

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I am having problems with a deadlock in Asterisk 1.8.9.0. The system is a x86_64 machine that is being used as a callcenter. The agents log in via the AgentLogin application, and each Agent/XXXX channel is assigned to one or more queues. A custom separate process generates calls into the queues for the agents to
answer. The calls all go out through a SIP trunk, and all of the agent extensions are SIP. After an hour or so, asterisk deadlocks. Any attempt to run "agent show" or "agent show online" through the console hangs. Also, AMI events seem to…

Asterisk Users 3.5 years ago 0 Answer

Proposed changes to Asterisk release and support cycles

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I've created a page on wiki.asterisk.org outlining some changes we're proposing to make to the Asterisk release and support cycles. As always, before implementing any changes of this type, we'd like to collect some community feedback on the proposal. The page is here: https://wiki.asterisk.org/wiki/x/5ggiAQ Feel free to comment here, or on the page itself if you find any errors or inconsistencies in the page's content.

Asterisk Users 3.5 years ago 5 Answer

Cell Phone as a Queue member

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Use local channel 2012/1/31 Niccolò Belli :
> Hi,
> Is it possible? I tried AddQueueMember(SIP/$TRUNK/number) but it tries to
> call SIP/$TRUNK instead.
>
> Cheers,
> Darkbasic
>
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Asterisk Users 3.5 years ago 4 Answer