Hi,

I have a Junghanns QuadBri installed and seems to be detected and working
correctly using the wcb4xxp driver
All ports are configured as TE

The problem is when I connect an ISDN line – the LED turns green and Dahdi
status shows Alarm OK
But when I try to make a call I get “All circuits are busy” cause 27

I have searched and web and tried numerous configs but it doesn’t want to
play.
I can accept incoming calls but there is no Audio which would indicate an
issue with the driver wcb4xxp?

See config files below and dahdi commands – please ignore that RED alarms
there was no ISDN lines plugged in.

Thanks, Joe

Here is CLI from a call

[2012-02-03 08:54:25] VERBOSE[13586] app_dial.c: — Called DAHDI/g0/1901
[2012-02-03 08:54:25] VERBOSE[13586] app_dial.c: — DAHDI/3-1 is proceeding
passing it to SIP/221-00001c80
[2012-02-03 08:54:27] VERBOSE[24518] chan_dahdi.c: — Channel 0/3, span 1
got hangup request, cause 27
[2012-02-03 08:54:27] VERBOSE[13586] app_dial.c: — DAHDI/3-1 is
circuit-busy
[2012-02-03 08:54:27] VERBOSE[13586] chan_dahdi.c: — Hungup ‘DAHDI/3-1′
[2012-02-03 08:54:27] VERBOSE[13586] app_dial.c: == Everyone is
busy/congested at this time (1:0/1/0)

Use jabberd and qmail.

On 2/3/12, bilal ghayyad wrote:
> Hi All;
>
> Any advise for a good collaboration solution (open source)? Chat + Email
> call center.
>
> Regards
> Bilal
>
> –
> _____________________________________________________________________
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Whats asterick?

On Wed, Feb 1, 2012 at 7:48 PM, Josh wrote:
> I am trying to configure Asterick, having the following system setup on
> the Asterick server:
>
> * eth0 faces the external Internet interface, *but* it does not have IP
> address (it has a private one given to it by my ISP’s DHCP server);
> * eth1 faces my internal network (say 10.1.1.0/24);
> * tun0 serves all mobile smartphones and connects to the internal
> network (it has a different ip range, say 10.1.2.0/24) – they are all
> connected via the Internet using OpenVPN;
>
> I would like to configure Asterick for internal calls between ourselves
> (eth1< ->tun0) and I think I have no problem with configuring this part.
> I would also like to use one external VOIP provider to which Asterick
> registers on startup. I think I know how to do that and use the
> “register” option in sip.conf, though I am not sure for the rest of the
> NAT-related entries (see below).
>
> The purpose of registering this external account is so that both the
> smart phones (tun0) and the internal net (eth1) users could use this
> account to make external calls (starting with “0″, i.e “_0[0-9].”
> pattern in extensioins.conf). Obviously, I need these calls to be routed
> properly via the external VOIP account. In addition to that, I would
> also need to receive calls from that external account to a nominated
> internal one (say on extension 20).
>
> Is this achievable?
>
> If so, I am not completely clear on whether I need to explicitly specify
> my public IP address (via externip/externhost) or whether Asterick is
> able to find it without this option? If not, then my plan is to use
> external program to find it and then use a script in Asterick to set it
> up as an environment variable. Would that work? That external IP address
> is going to change, but only in rare circumstances and in such cases I
> have to restart a lot of stuff (including Asterick) on that server (this
> is usually triggered by a monitoring program), so it won’t be a problem
> once it is setup initially. I am also not sure whether to specify
> “nat=yes” or just have “nat=route” only – any ideas?
>
> Is there a comprehensive list of all the options available in sip.conf
> and what they do, because I was unable to find such a list?
>
> If the above is doable, I would also like to add the following 2 features:
>
> 1. Secondary external VOIP account, though I have no idea how to specify
> its port in “register” (it uses port 5065 instead of the standard 5060).
> That account would need to be used on a separate interface (eth2) with a
> different public IP address. Would it be possible to use
> externip/externhost inside that external account section to specify it?
> If this is not possible, then I am thinking of running a separate
> instance of Asterick with the second VOIP account/public IP address set
> up – would that work?
>
> 2. I would like to be able to configure the following work flow: for a
> specific set of (external) calling numbers (including where no Caller ID
> is available):
> a) these callers to be prompted to specify the “reason” for their call;
> b) their response to be temporarily “recorded”/stored (a short message
> of, say no more than 10 seconds long or when they press ‘#’ for that
> recording to stop);
> c) Asterick then rings the nominated number for external VOIP calls
> (extension 20) and play that recorded message back;
> d) then asks for one of four possible outcomes:
> – accept this call (pressing, say 1) in which case the call is connected
> as normal;
> – reject it with a message that that number/person is “unavailable”
> (say, by pressing 0);
> – ask the caller to leave a message by transferring them to a voicemail
> (say by pressing 2); or
> – end the initial call completely with a message that the caller/number
> has been “blacklisted” (say, by pressing the 9 key);
>
> Could this be achieved?
>
> One final question about binding: in order to be able to use both tun0
> and eth1 interfaces so that Asterick serves the calls from both eth1 and
> tun0, do I have to use “bind 0.0.0.0″? Is there an alternative, like
> specifying “bind 10.1.1.1″ for eth1 and then “bind 10.1.2.1″ for the
> tun0 interface – is this possible?
>
> Many thanks in advance!
>
>
> –
> _____________________________________________________________________
> — 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

Hi,

I want to create a system for incoming calls where, under some
circumstances, callers get routed straight to voicemail (or some other
means of recording a message) but if they enter a valid extension number
then the recorded message would be abandoned and they’d be diverted to
the extension number they entered.

I realise this can be done with the voicemail app with operator=yes but
the problem with this is that the caller has to press 0 while the
announcement is being played. If they’re too slow and recording has
started, they’ve missed the opportunity.

So I played around with ConfBridge and a couple of call files, just to
see if I could get it to work. It’s a bit convoluted but the idea is
that the caller gets silently put into a conference, then two call files
make asterisk silently connect to other calls into the same conference,
with one doing the recording and the other using Read() to collect
digits.

If I just had the caller and one of the other calls in the conference
(the one doing Read()) then this worked – Read() managed to read the
DTMF digits and assign them to a variable.

However, when the ‘recording’ call is also in the conference, the ‘read’
call can no longer recognise the DTMF digits. To test, I made the ‘read’
call play a sound before calling Read() and I could hear this being
played so the call was definitely there. However, regardless of the
number of digits I pressed, Read() didn’t notice any of them, even if I
introduced a delay so that the other channels were quiet before the call
to Read().

I realise this might seem a bit like a mad solution but can anyone else
think of a way to get Asterisk to read (and react to) DTMF digits during
a recording?

This is with Asterisk 1.8.7.

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 –0×123-> Asterisk –0×123-> IP Office
< -0x456-- <-0x456

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

<-0x789--
Polycom –0×123-> Asterisk –0×123-> 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.