Hi,
I have a digium T1 card installed on my Asterisk box. Protocol is PRI.
I am trying to setup so that the box can send and receive faxes. Being
able to receive faxes is a lot more important than being able to send.
I tried spandsp-0.0.2pre25, with proper app_rxfax.c file. But I am not
able to
Using Asterisk agents.
Not recognizing that an agent has made an outgoing call IS THE PROBLEM.
Only workaround I see is to take the agent out of queue on all outgoing
(and direct dialed incoming) calls and put him back in the queue at the
completion of the call. That seems too kloodgy.
Hence the
to an
agent if they are already on a call from the queue, but an incoming
call from another internal extension, or even a DID ought to be able
to get through.
Consider this a feature request?
Tom
On Oct 15, 2005, at 10:04 PM, J Thomas wrote:
One of my friends is facing
One of my friends is facing this problems and I could not find any
solution to that. Hence this post.
In her Asterisk PBX, she has programmed about 10 agents, and strategy is
rrmemory. Everything works fine. When an agent has received an ACD call,
another call is not presented to him as long as
Setting incominglimit = 1 does not really solve the problem as I had
already mentioned. That practically takes away the call waiting and will
block all incoming calls including direct dialed calls. She does not
want that. Moreover, incominglimit is deprecated too.
-- jt
On Sat, 2005-10-15 at
I asked my telco to release caller name on the PRI. Earlier they were
releasing only the phone number.
I still did not see the name, but only the number in caller id. Actually
I now see number twice. When I inquired with them this is the response I
got:
I ran a trace on your TG. I see
I am trying to use TE110p card for Euro ISDN with Ericsson AMS switch. I
consistently get one of the following errors:
PRI got event: HDLC Abort (6) on Primary D-channel of span 1
or
PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1
My zaptel.conf file:
Some of my clients of hosted PBX service want to use it for callback
when they cannot use the ATA.
This is the scenario
1. Asterisk calls Party A at numA.
2. When A picks up the phone, he hears the announcement to enter the
destination number, numB. He enters numB
3. Asterisk Dials numB and
From: Walt Reed [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: Polycom Phones
To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
On
I purchased some 20 Polycom phones (brand new) for a very good price of
around $165 each. Now I am having a nightmare in configuring them. I
pulled the bootrom, SIP and config files from freedomphones.com,
modified them for my need and and started configuring the phones. First
couple of phones
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