Hi
Running 1.2.26 BRI stuffed. Calls made via PSTN via ISDN interface (Junghanns).
SIP ports mapped through firewall as we often connect from outside, but all SIP
accounts have good passwords.
However our telecoms provider picked up a few suspicious calls to places we do
not normally call at
(why do we find this so surprising in the UK - BT
helpful!).
Paul Redstone
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Hi
Just moved offices in the UK and moved our Asterisk box from old one to new
one. Using idefisk softphones, Junghanns quadbri card for ISDN 2e interfaces.
At both offices we had one standard number and a DDI range, routed with
Asterisk.
We'd set up the configuration so each idefisk set its
Useful discussion on this. There are some other functions in this which need to
be addressed. For example if doing an attended transfer and the recipient phone
number goes to voicemail, you have to wait for the timeout to reconnect to the
original caller - unless someone know differently. There
Hi
Running bristuffed 0.3.0-PRE-1f which is 1.2.1.
Using *2 in features.conf for attended transfer. Works well if someone
answers.
But the following sequence causes issue:
1. Receptionist takes call.
2. *2 then 123 to transfer to extension 123.
3. 123 is busy or does not answer so
Hi
We're using three line SIP phones (X-lite), very nice, with Asterisk 1.2
But we want to prevent either direct incoming calls or calls from other
extensions from ringing if the user is
in another incoming call (i.e incoming into the extension), making an outgoing
call or even checking their
we
use for most users - the hardphone is for reception and as backup in case of
computer failure.
Paul Redstone
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We're using an SC420 and using BRI with a quadbri Junganns card, with IAX
softphones and one hardphone.
Working well except that we sometimes get dropped connections between IAX and
the server with a max retries exceed message, which comes from the chan_iax
driver code. The BRI side of things
Hi
We're having some problems with max retries exceeded errors using IAX2 which
causes dropped calls. Sometimes happens with Firefly softphone, now 1.9.9 (the
current one) but has also happened with a hardphone we use (IAXtel). This is
just for the internal connection between our desktops and
Hi
We've recently set up and are using with success 1.0.7 using a Junghanns
quadbri card to BRI ISDN, and Firefly with IAX2 protocol as softphones Works
very well, however we're getting cases where sometimes the call just drops.
From setting more verbose modes we get a log which is shown
Hi
In the end we found it easy to record our own using this section in
extensions.conf. This also meant that we could add our own company specific
ones in the same voice (not shown here). Basically you get someone to dial the
8NNN1 to record or 8NNN2 to playback. The prompts are shown below
Hi
This works for us:
In IAX.conf
[NNN]
notransfer=yes
secret=**
context=contextname
host=dynamic
type=friend
callerid=John Doe0
mailbox=NNN
qualify=no
language=gb
The mailbox is for the mailbox number.
In extensions.conf we use a macro
[macro-iaxext]
;
; Standard extension
Hi Guys
I'm still wrestling with trying to make IAX2 softphones do attended transfer.
My iax.conf section has:
[XXX]
secret=mysecret
context=mycontext
host=dynamic
type=friend
callerid=Paul Redstone
mailbox=NNN ; notifies if mailbox has something in
qualify=no ; checks for
Hi
I see various discussions on this but cannot get it to work, and is not clear
that anyone resolved this. This seems pretty fundamental so I am missing
something, but I cannot find it anywhere.
# does work for blind transfers - no problem.
But the various * commands given in features.conf
So now that they are done how about you post the files for us? Share the
wealth.
Mark
Will be happy to do so once macro refined a little, but it is rather long
(about 600 lines) and I thought long posts were bad manners.
Otherwise this will be odne by the end of the weekend/
Paul
To add something to a post of a few
days ago on this:
We're just putting in an asterisk
system and wanted to have our own messages.
We're Asterisk and are not yet live but the
following works.
Our PA simply has a list of the
extract from extensions.conf as shown below,
calls a number
Sorry - still forget to clear the HTML flag - thought I'd better post again.
To add something to a post of a few days ago on this:
We're just putting in an asterisk system and wanted to have our own messages.
We're Asterisk and are not yet live but the following works.
Our PA simply has a list
Hi
Newbie asterisk guy here and forgive
this slightly long mail, but I'm stuck on
this for a week.
I'm having major problems getting a
Fritz card to dial out in the UK (or indeed
answer, but I've been concentrating on dialing
out). I'm getting the 0x3301 or 0x3302 error.
My capi.conf
Hi
Repeated e-mail as I forgot to make plain text - sorry. Newbie asterisk guy
here and forgive this slightly long mail, but I'm stuck on this for a week.
I'm having major problems getting a Fritz card to dial out in the UK (or indeed
answer, but I've been concentrating on dialing out). I'm
Hi
Repeated e-mail as I forgot to make plain text - sorry and then forgot subject
line. Newbie asterisk guy here and forgive this slightly long mail, but I'm
stuck on this for a week.
I'm having major problems getting a Fritz card to dial out in the UK (or indeed
answer, but I've been
I'm having major problems
getting a Fritz card to dial out in the UK
(or indeed answer, but I've been
concentrating on dialing out). I'm getting
the 0x3301 or 0x3302 error.Not
a direct answer, but are you _absolutely
sure_ the card works? Ihad the exact
same thing late last year here in
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