dueeseldorf ist nur ein vertriebsbüro:
http://firstwave.ch/kontakt/impressum.html
Markus Wingen
Business Development
[EMAIL PROTECTED]
<>
firstwave SWISS - EMEA Business Relations
Grepperstrasse 23 . CH-6403 Küssnacht am Rigi
Vertriebsbüro Deutschland (Salesoffice)
phone
+49 (0)211 -
Leif Madsen wrote:
Would you mind maybe expanding upon the hardware configuration you are
using and why? I, and I'm sure others, are curious as to what you are
using. I haven't had to roll out any systems yet that require
multiple Digium cards, but I'm sure the information would be quite
useful
We use asterisk coupled with swyxIt via a gnugk (version 2.0.8) and oh323
module: both swyx server and asterisk register on the gnugk. asterisk
receives sip calls from the exterior and routes them to the gk. I've set up
a prefix on swyx so that if I prepend +996 to my phone numerb, the call
gests r
Linus Surguy wrote:
> This isnt actually at all correct, we certainly have a
> Business Highway line in PTP mode with MSN! (Although you are
> right in that this is the default)
Ah, BT, don't you just love them. Speak to three people and get four
answers.
I have just spoken to the ISDN support
[EMAIL PROTECTED] wrote:
> Aug 26 16:53:16 NOTICE[-239408208]: frame.c:120
> ast_smoother_feed: Dropping extra frame of G.729 since we
> already have a VAD frame at the end
I'm also seeing this message streaming in the CLI. I think it
might have something to do with endpoints misbehaving when
t
Consider this dialplan fragment, where the call is being dialed into
[macro-process-routing] over an iax2 channel from another (same build)
Asterisk server:
[macro-process-routing]
; This is the entrypoint of the debug call but is also refered to by
Macro(process-routing) elsewher
Kris Boutilier wrote:
[macro-process-routing]
; This is the entrypoint of the debug call but is also refered to by
Macro(process-routing) elsewhere in the dialplan
; XXX-NNN-6800
exten => _6800,1,Macro(6800-interceptor)
; This is matched when 8 is dialed during m
I've noticed a problem with calls to Hangup when talking to my Norstars over
channelised T-1 E&M trunk lines - it's been present since I started to
fiddle with Asterisk last December and it's still present in 'Asterisk
CVS-HEAD-08/13/04-10:37:13'.
Specifically, when a call is connected to Asterisk
On 26 Aug 2004, at 17:48, Nick Barnes wrote:
Benjamin asked:
I don't have a problem setting this up under Asterisk (that's
the fun part) but what I need is advice on what to ask for
from BT so I don't get the wrong lines / services and so that
it all works smoothly!
OK. You need one of the followin
Alas, the dialplan itself is more complicated than I quoted in the problem
case below: There is an outer context that gathers digits from DID trunks,
pads the patter to comply with the corporate dial plan and then goes into
[macro-process-routing], thus:
[infrom-did]
exten => s,1,DigitTimeout,2
I have just completed a project where I have just implemented running
extensions.conf from mysql.
When the extension is called, the dialplan passes the extension together
with the calling number a to a perl agi script. The perl script does a
couple of sql queries and sets a few variables which tel
I'm having a dialplan problem on one host where trunks get pinned up
flapping between 't' and 'i' states and start eating lots and lots of CPU
(loadavg > 4.00). I haven't been able to pin down the problem reading
through extensions.conf and test calls haven't caught it yet either.
Unfortunatly the
Hi,
For those interested in using MySQL directly from extensions.conf, there's
already a source file floating around for using a MYSQL application to
do SELECT queries.
We're using the MYSQL app a lot in our exensions.conf, but we missed
support for queries that don't return a result like UPDA
On Thu, 26 Aug 2004, Jorge Verastegui G wrote:
> Have the astesrisk and digium people implemented PLC?
No
> Are
> they implmementing it now?
I want to but just haven't got to it yet.
Steve
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://l
When I try to flash my 7960 with SIP I get messages like this in the tftp
server logfile:
Aug 27 02:01:17 home tftpd[32590]: tftpd: trying to get file: P0S3-03-0-00 .bin
and the phone says something similar on the display for a brief moment and
puts a funny char where the space in the filename ab
On Thu, 26 Aug 2004 09:25:55 -0600, Andrew Elchuk
<[EMAIL PROTECTED]> wrote:
> Is a sound card needed in order to playback some of the asterisk sounds
> in /var/lib/asterisk/sounds when dialing out with an X100P? Thanks.
>
No Sound card is requied
___
A
What FXO interface methods are folks using successfully in the UK?
I'm looking for good, known-to-work solutions for commercial use for two
PSTN trunks on an Asterisk box. Here's the options I have, as I see it:
i) Two Digium X100Ps. Pro - cheap (c. £120), CE approved. Con - UK line
impedanc
On Thu, 26 Aug 2004 17:31:46 +0200, Alessio Focardi
<[EMAIL PROTECTED]> wrote:
> Also dialing out works like a charm, the only problem is that calling
> out "asterisk" is displayed on the called phone instead of the sip address of the
> asterisk
> box.
>
In the general section of sip.conf use t
look to your SIPDefault.cnf or SIP.cnf on TFTP server
if you have is correct file name in "image_version:" section
PJ
- Original Message -
From: Tracy R Reed
Newsgroups: gmane.comp.telephony.pbx.asterisk.user
Sent: Friday, August 27, 2004 11:48 AM
Subject: Can't flash 7960: P0S30200
Hello Jason,
Friday, August 27, 2004, 12:18:23 PM, you wrote:
JW> On Thu, 26 Aug 2004 17:31:46 +0200, Alessio Focardi
JW> <[EMAIL PROTECTED]> wrote:
>> Also dialing out works like a charm, the only problem is that calling
>> out "asterisk" is displayed on the called phone instead of the sip addre
On Fri, 27 Aug 2004 09:50:37 +0100, Jon Fautley <[EMAIL PROTECTED]> wrote:
> Heh, good old BT. I've never tested voice over Business Highway, as
> every BT engineer/support/sales person I've spoken to swore blind that
> it wouldn't work - and in BT's eyes, if they say it won't work, it's
> unsuppo
On Fri, 27 Aug 2004 11:15:07 +0100, David Gurr
<[EMAIL PROTECTED]> wrote:
> What FXO interface methods are folks using successfully in the UK?
>
Ditch FXO completely and use a BRI Solution much better quality.
or use Digium TDM400P card with two FXO modules, and apply UK CallerID patches,
In
Well I've just called BT, the confirmed to me that MSNs can only work
with PTMP and DDIs with PTP. As for the seqential MSN issue, they have
assigned me the five I requested - but not sequential. They did explain
to me that they were automatically assigned randomly. I asked them to
escalate my
Kevin Walsh wrote:
[EMAIL PROTECTED] wrote:
On 27 Aug 2004 at 2:33, Kevin Walsh wrote:
There is no packet loss concealment in Asterisk at this time.
Why doesn't asterisk clock to the 1000 interrupts per second instead
of the incoming audio? Were there no interrupts available when it
started? Even
On Friday 27 August 2004 00:57, imail wrote:
> same error :(
> I just cant seem to figure it out, it must be something very obvoius. Can
> someone please point me in the right direction?
[...]
> > > elifeq ($(USE_SIP_MYSQL_FRIENDS),1)
[...]
Looking at the GNU Make manual, there does not seem to b
Michael Manousos [EMAIL PROTECTED] wrote:
> Look at the RTP stack of the receiver. When a packet is received, there
> are two cases:
>
> a) An RTP packet carrying voice frames is received. In that case the
> decoder will play the voice frames.
> b) A CN (Comfort Noise) packet is received. In that
On Thu, Aug 26, 2004 at 07:08:19PM -0600, Marty Mastera wrote:
>
> I thought I would repost this info, since it seems relevant to this
> thread and may have been missed before this thread started...
Marty, I appreciate your repost of this. I had seen it already and have
written to Digium. That
Kevin Walsh wrote:
Michael Manousos [EMAIL PROTECTED] wrote:
Look at the RTP stack of the receiver. When a packet is received, there
are two cases:
a) An RTP packet carrying voice frames is received. In that case the
decoder will play the voice frames.
b) A CN (Comfort Noise) packet is received.
Group
This is strange. When I call my voice mail extension the system does
not pick up my touch tone entries. I have x-lite softphone and a cisco
7960 for my hard phone.
When I call from outside I'm able to check my voice mail without any
problem.
Any help would be great!
__
On Fri, 27 Aug 2004, Michael Manousos wrote:
> I hope that the above issues will start a discussion and result to a
> solution, no just for PLC, but also for the DTX operation.
Yeah - my goal for a reworked jitter buffer includes DTX and PLC. And
other TLAs ;-)
Steve
___
Michael Manousos [EMAIL PROTECTED] wrote:
> Kevin Walsh wrote:
> > Michael Manousos [EMAIL PROTECTED] wrote:
> > > a) The transmitter detected silence and sent nothing but the last CN
> > > packet was lost. According to the above interpretations, the receiver
> > > will try to conseal a packet loss
Hi everybody,
I need a litle help to install Asterisk using ISDN Fritz PCI card on
my linux box fedora 1.
All suggestions with links or samples are welcome.
I would be really pleased for any help :)
Radu,
E-mail: [EMAIL PROTECTED]
___
Asteris
Hi all,
Did anyone manage to make the GotoIf command work with regular expression ?
I would like to make the following thing:
${DTMSeq} contains a menu choice. Only 1, 2 and 3 are allowed menu choices.
If
DTMFSeq contains 1 or 2 or 3 => OK, Goto 4
else
Goto 2
I've tried the f
When I initially signed up with Packet8 and they sent their converter, I
used a X100P card in my Asterisk server so that it could send and
receive calls through Packet8, I suspect the same trick would work for
Vonage.
The benefit is you can then have several phones in the house, or one at
work
Have you considered relocating the hard drive (or Asterisk configs) and
the T100P card to a temporary machine? Even a lower class machine, just
to eliminate the SuperMicro as the possibility?
I'm interested in your research as we will be deploying some low end
$800 1U (very short) SuperMicro se
Been trying to set up a call queue with agent call back without the need for
the agent to have to log in.
Have set up the queue sucessfully.
However I want to remove the requirement for agents to have to log in as
they are on static extensions.
Is there a way of either using extentions in the q
Hi All
I have recently downloaded Asterisk
and was so impressed I thought I would setup a home server and I went out
and got myself a couple of cisco 7940's. (and a sipaura 3000!). thanks
to various posts on this list and the voip-info site I have managed to
get chan_sccp setup and working with
Deon,
When you say "I've tested up to 6 inbound calls at the same time"
with Broadvoice, is this with 6 $19.95 DID numbers that you have
assigned to *?
thanks
Doug
Deon Rodden wrote:
When I initially signed up with Packet8 and they sent their converter,
I used a X100P card in my Asterisk server
Just upgrade from July 12th cvs to last night CVS-HEAD-08/27/04-00:00:09
When call comes in and is sent to a Cisco 7960, I see:
-- Executing Dial("SIP/3008-9a9b", "SIP/3000|15") in new stack
-- Called 3000
Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum retries
Lol. Known issue, I spent an hour working on that problem. The phone's
current firmware is too hold and does not support longer filenames like
that. You have to increment the firmware versions, 2 or 3 firmware
upgrades and you'll be ready to use the latest and greatest.
Try upgrading to P0S3020
No. Just one regular $19.95 residential plan. I've had 6 cell phones
call my DID and my IVR picked up all 6 times. I never got a 7th cell, so
I never tested the limit. But I don't want to abuse my BroadVoice
account so I haven't tried it again. I mainly stick to 1 line, an
occassional 2nd line
Kind of off topic but I know CVS is the "prefered" way of upgrading,
however are there such things as "stable" CVS upgrades? It seems a lot
of the CVS's have a lot of devel bugs in this that I would be scared to
put even near production. Just IMHO. :-)
Matt
-Original Message-
From
Whenever I see the "Maximum retries" message it usually indicated a
communication problem, like one way traffic. Last time I got it, I
traced it to a bad firewall rule, dropped the firewall and it worked,
the time before that when I received it, it was due to a routing error,
the server could g
Hi all,
Did anyone manage to make the GotoIf command work with regular expression ?
I would like to make the following thing:
${DTMSeq} contains a menu choice. Only 1, 2 and 3 are allowed menu choices.
If
DTMFSeq contains 1 or 2 or 3 => OK, Goto 4
else
Goto 2
I've tried the fol
* and the 7960's are on the same wire, no firewall involved whatsoever.
Backing out to July 12th now...
> Whenever I see the "Maximum retries" message it usually indicated a
> communication problem, like one way traffic. Last time I got it, I
> traced it to a bad firewa
ok.. could we add a 'hunt group' to * and roll incoming calls over to
several extensions?
We also signed up with the Broadvoice 'BYOD' $19.95 service just in the
past week
and found the service to work extremely well with Asterisk. I also
updated the * server to 1.0-RC2
before testing it.
Doug
Hi all,
Did anyone manage to make the GotoIf command work with regular expression ?
I would like to make the following thing:
${DTMSeq} contains a menu choice. Only 1, 2 and 3 are allowed menu choices.
If
DTMFSeq contains 1 or 2 or 3 => OK, Goto 4
else
Goto 2
I've tried the follw
After days of searching i've finally figured out that E1 lines in india use
multiple types of signalling from EuroISDN to R2. Digium E1 cards dont work
for R2 type signalling. Can anyone suggest me asterisk compatible E1 cards
which would work on R2. Also if anyone on this list is from INDIA and u
Hi,
I am using a voicetronix OpenLine4. I downloaded a recent asterisk
CVS from voicetronix webpage but have had no luck to reduce echo on
outgoing calls and for it not to crach on incoming calls. I dont
think both problems are related though.
Here is an output of what happens when a new call c
Hi,
did anybody managed to compile h323 channel under Fedora 2? There's only
gcc 3.3 and 3.4. Does h323 from * or opencall work with FC2 and gcc 3.3?
Anybody had similiar problems?
tia
mazek
--
http://www.marcinmazurek.com/ ::: nic-hdl: MM3380-RIPE
GnuPG 6687 E661 98B0 AEE6 DA8B 7F48 AEE4
It's not the card's fault, it's the lack of a software driver fault.
R2 has a country dependent implementation. Some countries even have two
incompatible standards internally.
Em Sex 27 Ago 2004 11:03, Vikram Rangnekar escreveu:
> After days of searching i've finally figured out that E1 lines in
Hi,
there is a feature, which I would like to use with asterisk,
and I assume it exists.
Unfortunately I don't know how to say it in english.
In german it's "einen Ruf heranholen".
It means:
The phone set of my collegue is ringing, and I'm hearing
the ringing.
I know, that my collegue is not at his
I have a T100P card connected to an Adtran and then a T1.
I have added the following configurations to Asterisk...but, when I dial
9 and then a local phone number, it bounces between the dial tone and
silence and the *error* light on the Adtran blinks.
zaptel.conf
span=1,0,0,esf,b8zs
fxsks=1-8
l
In article <[EMAIL PROTECTED]>,
Roger Schreiter <[EMAIL PROTECTED]> wrote:
> Hi,
>
> there is a feature, which I would like to use with asterisk,
> and I assume it exists.
> Unfortunately I don't know how to say it in english.
> In german it's "einen Ruf heranholen".
>
> It means:
> The phone set
On my experience, you should go to SIP whenever possible. 7940/60 on
SIP will do most if not all fuctions.
Try the little chart on support hardware on chan-sccp.sourceforge.net
Lethol
- Original Message -
From: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
Date: Fri, 27 Aug 2004 14:16:11 +010
+++ Marcelo Pacheco [27/08/04 11:06 -0300]:
> It's not the card's fault, it's the lack of a software driver fault.
> R2 has a country dependent implementation. Some countries even have two
> incompatible standards internally.
>
> Em Sex 27 Ago 2004 11:03, Vikram Rangnekar escreveu:
> > After days
What about if I want to call a Free World Dialup number from asterisk
and play a number?
Jason Williams wrote:
On Thu, 26 Aug 2004 09:25:55 -0600, Andrew Elchuk
<[EMAIL PROTECTED]> wrote:
Is a sound card needed in order to playback some of the asterisk sounds
in /var/lib/asterisk/
http://www.voip-info.org/tiki-index.php?page=PBX%20Call%20Pickup
http://www.voip-info.org/tiki-index.php?page=Asterisk%20channels
On Fri, 27 Aug 2004 16:11:46 +0200, Roger Schreiter
<[EMAIL PROTECTED]> wrote:
> Hi,
>
> there is a feature, which I would like to use with asterisk,
> and I assume i
FWIW, I've had Broadvoice running for two or three months now. Very
reliable, good folks in tech-support helped with the initial asterisk
config and getting me SIP credentials. Since my set-up as a home-pbx,
incoming calls ring all my extensions (Sipuras) all the time. If
someone's already talki
First there's Analog R2 and digital R2. I'm concerned with digital R2 (R2D)
only.
R2 is equivalent to the american robbed bit signalling used in the US.
Q.421 is the ITU document number. It costs money to download the
specification. And it is only the generic part of it, it doesn't cover the
co
Hi Rich,
I had to change all my nat=yes to nat=route in the sip.conf.
nat=yes seems to be ignored in today's CVS.
Walter
Message: 5
Date: Fri, 27 Aug 2004 08:45:19 -0600
From: Rich Adamson <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] sip change?
To: Asterisk Users Mailing List - Non-Commercial
Hi all,
I have a need where ACD ringall and ACD roundrobin ring strategies will
be combined. Basically, ring every agent in a specified order, but
whenever it times out and goes to the next agent, I still need the
previous agent(s) to continue to ring.
I would like to develop this extension my
When using the callback feature on agents I notice that when the queue calls
one of the agents and the agent picks up the call they hear nothing until
pressing the # to accept the call.
Only then does my announcement play back to the agent after which the call
is immediately connected.
Is there
Traditional overhead paging systems are a little more
complicated than they first appear. It's not just speakers
and a centralized amplifier. They would have too much cable
loss if done that way. Instead, they use a centralized
power source, and amplifiers at each speaker unit.. The one
I just
-- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1
-- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1
RFC3389: 5 bytes, level 0...
Aug 27 08:32:16 NOTICE[23572]: rtp.c:289 process_rfc3389: RFC3389
support incomplete. Turn off on client if possible
Killed
Whenever I m
Hi,
I am using Asterisk with various brands and models of SIP phones. Especially
the Welltech phones LP201 are particularly nasty with volume and echo. Even
with the input gain (microphone) of the Welltech set to the max, the PSTN
end can hardly hear the SIP user on incoming calls. Ztmonitor als
Interesting... but that was supposed to have been a fix for Uniden
bugs. It shouldn't have negatively impacted 7960's on the same wire.
Must be a broken logic in there somewhere.
Rich
> Hi Rich,
>
> I had to change all my nat=yes to nat=route in the sip.conf.
>
> nat=
After a call is sent to voicemail on an inbound connection from
Broadvoice, the call is hung up in the middle of recording a voice mail
after about 30 or so seconds. I get an error "User hung up". If I answer
the call and not have it go to voicemail, the call will stay connected.
This only seems to
Hi
I am trying to make a call to a Free World Dialup number with the
following call file:
Channel: SIP/[EMAIL PROTECTED]
Callerid: Nagios
MaxRetries: 0
WaitTime: 30
Context: autodialout
Extension: s
Priority: 1
When I put the file in /var/spool/asterisk/outgoing/ directory, the
X-Lite software p
> The way I interact with BroadVoice though isn't officially sanctioned, I
> didn't prefer to use their "Asterisk Only" SIP gateway, in which they
> charge you 3.2 cents a minute (or whatever) when you exceed the first
line.
Where did you get this info? I have been using broadvoice for 2 months n
Hi
(B
(Ba bugfix release is now available for the FWD Assistant
(B...
(B
(Bhttp://www.voip-info.org/tiki-index.php?page=Asterisk+FWD+Assistant
(B
(Bfurther details are on the Wiki.
(B
(Bthanks to everybody who has provided feedback
(B
(Brgds
(Bbenjk
(B
(B--
(BSunrise Telephone System
On Fri, 27 Aug 2004, Andrew Elchuk wrote:
> -- Attempting call on SIP/[EMAIL PROTECTED]:5060 for [EMAIL PROTECTED]:1
> (Retry 1)
> Aug 27 09:20:22 WARNING[1116957488]: chan_sip.c:497 retrans_pkt: Maximum
> retries exceeded on call [EMAIL PROTECTED]
> for seqno 102 (Request)
> Aug 27 09:20:22
Hi Yawl,
After about 6 months of prattting about I've convinced my boss that we
should be installing * into our currently under constuction Data Center in
Somerset NJ. There will be 10 permanent people and DR space for another
50.
My plan is as follows;
ATAComm dual XEON server with quad T1 boar
Yep... It's BroadVoice's problem, not *'s... When * is recording, be it
voicemail or the record() application, * does not transmit a single packet
back to BroadVoice (Confirmed by ethereal and TCPDump) After 30 seconds the
BroadVoice switch will disconnect the call believing that it's a far-end
dis
- Original Message -
From: "Ejay Hire" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[EMAIL PROTECTED]>
Sent: Friday, August 27, 2004 7:53 AM
Subject: RE: [Asterisk-Users] Overhead Paging
>... Instead, they use a centralized
> power source, and ampli
Hi all,
hope this isn't a duplicate - but my first post went AWOL.
Sorry for this cheeky request and one that will probably not meet with
much response but I may as well ask!
I've just recieved my Cisco 7940 and am after upgrading it to the SIP
firmware. I don't (yet) have a support contract and
I am trying to get two * boxes to communicate with eachother.
I have read
http://www.voip-info.org/wiki-Asterisk+-+dual+servers
as well as information on IAX channels, the Dial() command, and the switch
statement in extensions.conf.
But I am having no luck. I have a working * box running with a
Hi all,
a cheeky request and one that will probably not meet with much response
but I may as well ask!
I've just recieved my Cisco 7940 and am after upgrading it to the SIP
firmware. I don't (yet) have a support contract and thus can't download
the firmware image from the Cisco site. I will be
Hi All,
I'd really like to see a show of hands with regard to people's
experience with FXO interfaces. I own a few X100p cards and have had
nothing but problems with them.
I also took part in Sipura's beta program, for the SPA-3000. While it
can be an improvement over the X100p, it presently has
Hi
(B
(BI had asked for some help with the Asterisk Assistants
(B
(Bhttp://www.voip-info.org/tiki-index.php?page=Asterisk+Assistants+for+MacOSX
(B
(Band many have offered assistance with translations which I
(Bam grateful for and like to say thank you again.
(B
(BHowever, there hasn't been
Thanks Don for the help but I found someone with a similar problem and
what they did was remove the audio module with a simple rmmod audio. Now
i get dial tone and everything from my both both my x100p and my s100u.
And can dial out from both.
Thanx again
Dave
>
> - Original Message -
Stupid newbie question that has probably been answered
before … but can a Macro call another Macro ??? is there any rules about
how deep ???
Gary G. Hendershot
Chief Technical Officer
Advanced Digital Technologies
BEGIN:VCARD
VERSION:2.1
N:Hendershot;Gary
FN:Gary Hende
ooohhh I'll take a crack at it! sounds like fun! :)
(B- Original Message -
(BFrom: "Sunrise Ltd" <[EMAIL PROTECTED]>
(BTo: "astusr" <[EMAIL PROTECTED]>
(BSent: Friday, August 27, 2004 8:47 AM
(BSubject: [Asterisk-Users] Are there any graphic designers on this list?
(B
(B
(B> Hi
> >... Instead, they use a centralized
> > power source, and amplifiers at each speaker unit...
>
> I've never seen one like that, That would suck nut bigtime... Especially if
> you had a large building with 100+ speakers, you would have to tune the
> volume on each one... That's not how the TAMB
Deon Rodden wrote:
I've been working with Asterisk for about 2 months now and am doing
well. However I decided to switch platforms from Fedora Core 1, that my
predacessor was using, to Gentoo, for obvious reasons. It just seems
faster and less "bloated" everything I need, nothing I don't.
Anyw
Nevermind. I was a digit off in my zaptel.conf...
the span for my adtran settings is 1,1,0,esf,b8zs instead of the one i
have listed below.
ph...one digit off.
cheers,
Shawn Parker wrote:
I have a T100P card connected to an Adtran and then a T1.
I have added the following configurations to A
Hi,
I have only done some basic testing
with the sipaura 3000 that I bought 2 weeks ago - it appears to work very
well. I have made quite a number of calls - although they have only been
short ones and not had a single echo yet.
No experience yet with any other devices
but would like to hear ab
Does Asterisk need a sound card or
functional Console/dsp to answer inbound DID number from PRI and playback
.gsm files?I can call from any of the SIP extensions on Asterisk and
hear audio from Playback(), MeetMe(), or MOH. The problem I am having
with calls from my PRI is as follows:I have
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 27 August 2004 12:32 am, Brian Capouch wrote:
> I don't know who else may be suffering from this, but the ultra-long
> Reply-to: header seems to break my mail reader.
>
> I have been suffering the zanies for the last week or so--mainly showin
>-Original Message-
>From: Larry Shields [mailto:[EMAIL PROTECTED]
>Sent: Friday, August 27, 2004 12:20 PM
>To: [EMAIL PROTECTED]
>Subject: [Asterisk-Users] No audio on PRI channel answered by
Playback() orMeetMe()
>If I assign the DID to ring extension SIP/2000 and then after time-out
I personally am running a couple of X100p cards, a couple of SPA-3000s
and a T100P. The X100p cards seem mostly flawless, I do have issues if
I am trying to use both at the same time. I suspect it's due to
interrupt sharing, it just hasn't bothered me enough to go fix it. The
other issue wit
> I'd really like to see a show of hands with regard to people's
> experience with FXO interfaces. I own a few X100p cards and have had
> nothing but problems with them.
>
> I also took part in Sipura's beta program, for the SPA-3000. While it
> can be an improvement over the X100p, it presently
I have today reported a bug with the latest channel.c (1.134) that
affects music-on-hold for the first user in a MeetMe room when calling
from a Grandstream BT102. The music is broken up about 5-10 times a
second. It doesn't happen when calling from Firefly. It is also fine
on both clients with 1.1
You should be able to hear the audio - a sound card is not involved.
Try inserting an "answer" command in the dialplan before you try to play
something. Like
Answer
Wait (if you want)
Playback
Hangup
Should work (using the proper dialplan commands)
Regards
Scott Stingel
Scott M. Stingel
Hello,
I’m having some problems with audio on outbound sip to pstn calls from
Asterisk to a Max TNT. When I place the calls it connects but I hear
pulsing / clicking for the first second or two of the call. My service
provider seems to think that the issue may be a result of improper ha
> > OK. You need one of the following:
> >
> > Home Highway
> > Business Highway
> > ISDN2e
> >
> > I can confirm that * works happily with all three - my office lines
> > are (for
> > various reasons, none of which apply any more!) on Business Highway.
> >
>
> Heh, good old BT. I've never tested v
> Well I've just called BT, the confirmed to me that MSNs can only work
> with PTMP and DDIs with PTP. As for the seqential MSN issue, they have
Complete tosh! As I said earlier, we've got it - and have ordered it with
additional lines as well, if you really want it, just argue more, and talk
to a
- Original Message -
From: "Tim Jackson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 27, 2004 7:55 AM
Subject: [Asterisk-Users] xlite Problems
-- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1
-- Attempting native bridge of SIP/1009-3df0 and SIP/
I have an analog Fax machine which I wish to connect to the network and
the Asterisk server. It will connect through a GS Handytone 286
converter and then into the LAN. Is there any information out there on
what I need to write in *sip.conf* and/or *extensions.conf* to make sure
the fax works
Remote Call Pick up feature is very much implemented in asterisk. You
can pick up a call for another extension by dialing *8#
To be able to do that, you need to have the extensions in the same
pickup group, configurable through sip.conf and zapata.conf.
-- sudhir
> -
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