* You are viewing the archive for January, 2012

Experience with Eicon Diva PRO 3.0?

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

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

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 hear, occasionally, short bursts of what could be the hold music on the Polycom. Also, when I look at packet captures taken from the Polycom’s port, I see two streams when it’s working (the Polycom calls Avaya) and three when it’s not (Avaya calls Polycom).

I do notice that this “extra” stream has a unique SSRC. If I do the packet captures from the Asterisk box, it looks like the extra stream is generated by Asterisk. Graphically, the RTP streams and their SSRCs I see on a working call (Polycom calls Avaya) looks like this:

Polycom –0x123-> Asterisk –0x123-> IP Office
< -0x456-- <-0x456

The non-working call (Avaya calls Polycom) looks like:

<-0x789--
Polycom –0x123-> Asterisk –0x123-> IP Office
< -0x456-- <-0x456

However, I'm new to Asterisk, and I'm not very familiar with any of these VoIP protocols, so I find myself stuck. Can anyone suggest some troubleshooting steps or material I might read which would help me to determine if my hypothesis is correct, and if so, how I can determine what's responsible for this extra stream?

Console output from a non-working call with sip and h323 debug and trace on follows. Here, 207 is the Avaya handset originating the call, and 216 is the Polycom receiving it. 172.20.20.233 is the IP Office, 172.20.20.205 is Asterisk, and 172.20.32.70 is the Polycom.

asterisk01*CLI>
== New H.323 Connection created.

Deadlock detected in asterisk-1.8.9.0 x86_64

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 stop. However,
the users seem to be still connected, only they do not receive calls anymore (the custom process waits forever for the Originate response). The deadlock is apparently spontaneous – there is no explicit action taken by the administrator that seems to induce
the issue. I will try to make sense of the attached traces, but I hope someone on the list could provide a clue on what to look for.

Backtraces attached to https://issues.asterisk.org/jira/browse/ASTERISK-19285

Proposed changes to Asterisk release and support cycles

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.

Cell Phone as a Queue member

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
>
> –
> _____________________________________________________________________
> — Bandwidth and Colocation Provided by http://www.api-digital.com
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>              http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/mailman/listinfo/asterisk-users

Problem with DTMF in Voicemail main

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
10.1.0 that would affect DTMF detection or app_voicemail itself, but
it’s certainly possible. That’s why we have an issue reporting system,
and it’s also why we produce release candidates to get testing prior to
making official releases.

> I notice that comedian mail has <> instead of [] brackets. Does that
> mean it’s on its way to being deprecated?

I assume you are referring to how app_voicemail (not ‘comedian mail’) is
listed the menuselect tool. Umm… no, those are completely unrelated.
How did you reach that assumption?