Displaying 20 results from an estimated 2000 matches similar to: "Multiple voices on 64K channel (was) simple question..."
2004 Jan 06
1
Fw: Pls confirm
----- Original Message -----
From: "Jess Magnaye" <jess@arretni.com>
To: <wipe_out@users.sourceforge.net>
Sent: Tuesday, January 06, 2004 3:19 PM
Subject: Re: [Asterisk-Users] Pls confirm
> Is the format "allow=g723.1" in sip.conf valid?
>
> somehow i cannot get it working to do g723 passthru. also, i've read that
> doing g723 will disable
2004 Jan 05
3
question re voicemail
Hi,
I just setup my * with digium. I started testing voicemail first between atas, and i am not sure why it is not prompting me any when the call is not answered or if busy. i only get continuous ringback and the following message:
asterisk*CLI>
-- Executing Dial("SIP/6882332-1697", "SIP/5104112978|20|tr") in new stack
-- Called 5104112978
--
2003 Dec 23
0
Fw: Fw: Questions and finding
> Thanks for the reply.
>
> 1. My VAD is turned off (00140014), and it didn't help for that cut-off.
I
> am not sure if OutboundProxy has to be configured to have it working fine.
> Or this just happened to me? What is your ATA's software?
>
> 2. I tried dtmfmode=inband on sip.conf, and dtmfmode=rfc2833. None
worked.
> As per ATA, it is by default using rfc2833.
2007 Dec 26
2
No cdr_csv after upgrade from 1.2.x to 1.4.x
After upgrading from 1.2.x to 1.4.x call detail records are not being
written to /var/log/asterisk/cdr-csv/Master.csv
In cdr_manager.conf I have
[general]
Enabled = yes
Apparently there is something else that needs to be configured for call
detail records in 1.4.x. Can someone point me in the right direction?
Don Pobanz
2007 May 27
0
Start recording automatically when
1. RE: Start recording automatically when xferring to
anextension? (Don Pobanz)
Message: 1
Date: Fri, 25 May 2007 11:54:33 -0500
From: "Don Pobanz" <dpobanz@hastingsutilities.com>
Subject: RE: [asterisk-users] Start recording automatically when
xferring to anextension?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
2004 Jan 22
1
simple question...
it just came to my mind, and i haven't done any researches yet if somebody tried this one with asterisk.. :) well just in case somebody or someone on the list aware, i appreciate any advise.
in telco world, there's like 64kbps per channel and voice can be carried on a 16kbps channel. is it possible to configure asterisk to make 4 extensions (ATAs example), to call out using single FXO
2003 Nov 25
2
modem to modem calls through asterisk
Modem connect speeds on calls through * seem to be lower than calls
made through the telephone company lines or our old Rolm PBX. All data
calls have 2 wire analog modems on both ends.
For my set up I have channels of a Zhone channel bank tied to 2 modems.
The Zhone channel bank interfaces my * server with a T400P card.
modem --- Zhone Channel bank - * via T400P card - Zhone channel bank -
2007 Oct 18
1
Limit number of times a call can be forwarded
We have had a few different times when a user has forwarded their phone
to himself. This has overloaded the communications to our operator panel
(FOP). One user should not be able to effect the whole phone system!
Is there a way that the number of times that a call can be forwarded
could be limited like to 10 or even 100? Then even if a user does
something stupid like forwarding their calls to
2004 Jun 14
4
Sipura 2000 not answering em_w calls
I recently purchased a Sipura 2000 and connected a phone to it which is
connected to my asterisk box via sip.
Calls to the Sipura 2000 work fine from another sip device connected
through *, from either an fxo or fxs (via adtran channel bank connected
to a T400P card) port. However, when a call comes in from the phone
company over a T1 with em_w trunks, the phone on the Sipura will ring
but I
2004 Jan 06
4
Pls confirm
Can someone on the list confirm if Asterisk can do g723 or g729? when connecting to provider? or it is only supporting g711?
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2003 Sep 12
2
Voicemail menu structure
There has been discussions about the voicemail menus and some of us
would like to see an overall plan for the voicemail menus.
There are 3 primary ways of arranging the menus. First is a tree
structure, second is a random access structure and the third would be a
hybrid of the two. (Comedian mail is currently a hybrid.)
As was pointed out by Brad Bergman, the ideal would be to have it
2007 Jan 05
4
how to transfer calls when analog phone has no transfer button
When you have a bunch of analog phones that you want to connect to
asterisk, but those analog phones have no transfer button, what are
the options to allow the phones to transfer a call?
--
------------------------------------------------------------
Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507)
2004 Jan 07
4
* crashed
I am just wondering if this is normal. I have my * running for a week now and I'm still testing its interoperability with other voip provider (in sip using codecs other than g711). yesterday, i changed my linux's (RH9). and since the new ip i assigned is located on a different site, i have to shut it down and move it physically. after that, i cannot run my * anymore. i am getting this
2003 Nov 03
5
Red Alarm
Hi list,
Sometimes I receive a Red Alarm in my E1 trunk (E&M immediate start
signaling), and just few seconds after this, all alarms are cleared.
This problem ocurrs many times/day, and if are calls in progress,
these calls just hang-up.
Could it be an asterisk bug? Or may I contact the PSTN provider?
Thanks
Eduardo
2003 Dec 22
3
DID trunks -- equipment requirement
Hi guys,
I posted a somewhat similar question about a month ago and got a
thoughtful resonse from Steven Critchfield, but I've got a quick follow
up question to it.
I'm looking to setup a 16 extension / 10-14 phone line Asterisk install
for a customer who would like to have DID numbers for the extensions,
since they're currently on Centrex and already have the 1-to-1
2004 Jan 07
2
* and Cisco Gateways
Anybody on the list who implemented Cisco ATA + * + Cisco 2600? I cannot get my calls from ATA to terminate to the Cisco gateway via *. I am not sure if it is my hardware problem. I'm getting the following "codec negotiation problem" from Cisco.
23:39:08: Unexpected VoIPCodec Type :g729br8
23:39:08: Unexpected VoIPCodec Type :gsmefr
I appreciate any help I can get. Thanks.
2003 Nov 04
1
asterisk and zplex10b (fwd)
hello all,
I still experience the random off-hook on my fxo cxhannels, i am using a
zplex-10b channel bank. which does not allow me to call out.
The situation still persists...this is what i have in the zapata.conf
[channels]
context=internal
context=incoming
context=default
usecallerid=no
usecallwaiting=no
signalling=fxs_ks
channel=1-8
signalling=fxo_ks
channel=16-24
but i still have the the
2003 Mar 21
8
Help with linejack as a trunk?
I have a linejack and a phone jack in my asterisk server working well
between the SIP phones and the phonejack. what I cannot get to work is
the outbound linejack Phone/phone0 trunk line? how can I get a SIP or
Phone/phone1 phonejack phone to dial 9 then outside number and pickup
Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on
the last digit 2. no outside dial.
2006 Nov 19
4
reduce dialtone volume on zap channel.
Is there a way to reduce the volume of the dial tone on a zap channel? I don't want to reduce the audio volume on calls so txgain in zapata.conf will not work.
I am having problems with asterisk not recognizing the first dialed digit from an analog phone about 8-15% of the time. Once the dialtone goes away, the digits are always recognized.
Any other thoughts on how to solve this are also
2003 Aug 25
2
0 out of voicemail to different secretaries
Is it possible to configure * so that if a caller reaches voicemail for
someone in Engineering, but doesn't want to leave a message they can
press zero (0) and reach the Engineering Secretary or if they are
calling someone in Accounting and reach voicemail, pressing '0' would
reach the Accounting secretary, not the Engineering secretary?
Don Pobanz