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
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…
I am having problems with a deadlock in Asterisk 126.96.36.199.
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…
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.
Use local channel
2012/1/31 Niccolò Belli
> Is it possible? I tried AddQueueMember(SIP/$TRUNK/number) but it tries to
> call SIP/$TRUNK instead.
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On 01/31/2012 12:17 AM, Ira wrote:
> Tonight I tried 4 versions of Asterisk; 10.0.0, 10.0.1, 10.1.0 and trunk.
> On 10.1.0 and trunk, I can't successfully enter the password for any
> mailbox in voicemailmain on my Aastra 480i phones. All four version work
> with a Snom cordless SIP phone. In 10.0.0 and 10.0.1 the Aastra works
> perfectly. So needless to say I'm back to running 10.0.1. The WAF is
> very low for stuff like that. It is quite unlikely that there were any changes between 10.0.1 and
In case a NAT firewall prevents using STUN to open SIP/RTP ports, a
solution is to first connect the phone to the Asterisk server through
a tunnel, and then have data go through the tunnel. Are there hardphones that support OpenVPN? If none, what about SSH? Is this a good alternative to use VoIP with
SIP? If you've tried either or both solutions, I'm interested in any
feedback. Thank you.
I went through the source code and now understand better how dtmfmode=auto
works. In testing I was able to resolve this by setting dtmfmode=auto.
After further testing I will deploy it to production and see if it breaks
anything but I am hoping this will be resolved for the long term. Thanks Bryant
On 30/01/12 17:12, Stuart Elvish wrote:
> Hi all,
> Firstly, apologies if the answer to this question should be obvious.
> I have just started working with SRTP and had a self-signed
> certificate working perfectly. I have now purchased a CA signed
> certificate but can't get it to work properly with Asterisk. I think I
> have a configuration error. No, you've found a bug - I just posted an update about this issue
yesterday, predicting people would get stuck on this issue: http://lists.digium.com/pipermail/asterisk-users/2012-January/269856.html…
I've just come across this issue: https://issues.asterisk.org/jira/browse/ASTERISK-17727 I am strongly in support of TLS and I believe this issue will be a
stumbling block for more and more users - because more and more CAs are
using the intermediate certificate chains For example, the free startssl.com certs are trusted by Android phones
now. I have a UA running on my phone against a SIP proxy with Kamailio.
I have the free cert and the intermediate cert in a single pem file.
It all works. As noted in the bug, there may…