Hello all,
i would like to have references so i'm giving free help
for any project (commercial or public).
regards,
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To
Trick question I assume?
It was mind numbingly simple on my iPhone...(though none of the voice mail
worked when London a few weeks ago).
- tap voice mail -
- tap speaker (upper right) until it turns blue (is activate)
- tap the message you want to playback
- use assorted controls to delete -
Hello,
Quoting Digium Support:
The TE110P has been discontinued and replaced in our product lineup with
the TE120P, which features many overall improvements and does not suffer
from the HDLC Abort/Bad FCS problems that the TE110P did.
Better switch to TE120P,
On 10/25/07, David Kennedy [EMAIL
really
want to end up with two thread on the one problem :)
Thanks
Dave
On 10/25/07, Matthew Fredrickson [EMAIL PROTECTED] wrote:
Rony Ron wrote:
Hello,
Quoting Digium Support:
The TE110P has been discontinued and replaced in our product lineup
with
the TE120P, which
( but not leave voicemail ) and run the IAX demo at
Digium.
Can anyone tell me what may be running amiss here?
Also, is there a script or makefile target that will fully un-install
asterisk, zaptel, zapata and libpri so that I can try again?
Thanks,
Ron
--
Ronald FoxEmail
: 100x145.
-Dan
The usable resolution for the programmable (logo url) bitmap is 90 x 56.
From:
http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 202120
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.
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iQEVAwUBQ9P59ktP/KMNOfRbAQJjUQf
place a call to my inbound number, the call comes into asterisk, runs
through the steps, then comes in again, and runs through the steps
again. On the calling phone there is silence until the second call
finishes, then I get busy/congestion.
TIA
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[EMAIL PROTECTED] http
Asterisk servers. And I have to reiterate... all was good until the
firmware upgrade.
-Ron
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I'm running 1.6.2.0041 according to my phone.
Which firmware worked for you?
It was the old firmware from when we first got the phones actually.
1.4.x I think. Then I read that they fixed the CID issue and decided
we needed an upgrade. I tried it out on my phone, but didn't really
notice the
I'm running 1.6.2.0041 according to my phone.
Which firmware worked for you?
It was the old firmware from when we first got the phones actually.
1.4.x I think. Then I read that they fixed the CID issue and decided
we needed an upgrade. I tried it out on my phone, but didn't really
I've been running 1.6.4.0064 for the last few weeks..
I've had no problems with it, I haven't done a whole lot of speaker
phone with it yet though.. Once my IP4000 reboots It'll be running it as
well so that will be a good test.
Which bootrom version are you using?
-Ron
It realy is a pain in the *ss.
the problem is just how you explained. when
trying to match the terminating number, there's no SINGLE fixed pattern for the
dialcodes. so how do you know how many digits of the term number to match
against the dialcode? you dont. you have to match the dialcodes
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ron hotmailSent: Friday, January 27, 2006 8:17
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] OT?:
International number parsing
It realy is a pain in the
*ss
One thing I was pondering: you are not, by chance, using the same
sip.cfg between version 1.4.1 and version 1.6.2 are you? The file has
changed significantly between these versions, and certain acoustic
settings that worked with 1.4.1 may not work with 1.6.2 (Not to mention
that ipmid.cfg
need to modify these to fit
your timeserver setup.
-Ron
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Ideally, you should be looking to run at least V6.3 SIP firmware, V7.4
or 7.5 would be much better.
HTH
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don't uplink to another switch where you can
create a non-QoSd bottleneck link.
-Ron
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It does take between 1 and 12 hours for the new settings to take effect.
Dan
I've recently had a problem with codec changes taking affect, but they
were nice enough to on-the-fly move an 800 number to route from one
site to another. It seems there was some kind of cacheing issue, as I
changed
phone calls via VoIP? If so, it is not
recommended to run FAX via VoIP. The two don't mix. FAX is not able to
handle packet loss like VoIP. Also, any codec other than uLaw will not
even come close to working, as the codecs are designed to compress
voice.
HTH,
-Ron
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asterisk manager interface to
monitor calls and that way I can keep the preset concurrent limit.
Any ideas?
TIA!
-Ron
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Ron Senykoff [EMAIL PROTECTED] wrote:
I'm helping out with a political campaign and would like to use asterisk
to blast out about 200,000 calls with a short message from the candidate.
Can you tell me which party this is for, so I can ensure I never vote for
them?
It's a basic GOTV (Get
I just did a quick office poll and everyone agreed if a party candidate
did this to them, they would vote for the candidate's opponent. The office
is rarely unanimous in political matters so this was a pretty interesting
result to me.
I'm pretty sure the feeling is universal.
Like I said
Thanks for all your responses. The reason we would not go through a
provider is that I run Asterisk phone systems, we have access to
bandwidth, and I can do this myself for a fraction of the cost.
Cheers
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Hello all,
Has anyone figured out a way to send email notifications etc. due to
failed IAX2 registration attempts?
Thanks
-Ron
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I would like to kick off an AGI script when an agent answers a call...
thus passing the phone that answered the call, the CID, etc.
Anyone know how I could do this?
TIA
-Ron
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,VoiceMailMain(@Mercury-Network-Emp)
Try adding
dtmfmode=rfc2833
to your sip entry.
Also, check the permissions for the file on your boot server.
HTH,
-Ron
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of having it go
to the data center.
My main concern is the dialplan, I guess if the peer is not local it
would then go out the IAX or SIP gateway to the main * server and then
check in its dial plan/routing table there, correct?
Any help/suggesstion on this would be great!
Thanks
Ron
somemore then, I figure it would be playing a part into this!
Thanks for the help!
RonOn 2/25/06, yusuf [EMAIL PROTECTED] wrote:
Ron McCarthy wrote: Hi List, Im planning on setting up asterisk for a large scale enviorment, with multiple sites. We will be doing quite a bit of inner office calling
maxexcessbuffer=300
minexcessbuffer=60
jittershrinkrate=1
maxjitterinterps=10
resyncthreshold=1500
Thanks,
-Ron
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Also, SATA on a onboard SATA card will eat more CPU then a SCSI system.
Are you running software RAID by chance with your SATA? SCSI or SCSI
Raid will not each CPU near as much since the HBA does all the work and
does tie up the CPU with all its I/O's. We have successfulyl recorded
5+ calls at a
firmware does not allow the softkeys to be programmed :(
Unfortunately you have to make a choice:
SIP firmware - Easy to implement on *, but poor XML support
SCCP firmware - poor/non-trivial asterisk support, great XML support.
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[EMAIL PROTECTED] http://www.wellsted.org.uk
N
I havent got any mails since 2:42 this morning..usually i get at least
the normal 10-15 a hour, if someone gets this can they reply?
Thanks!
Ron
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given us no problems. We did play with IRQs in the BIOS, but not sure if
that was actually needed.
Ron
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regard this as vapourware.
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Description: Digital signature
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On Tuesday 11 March 2008 16:21, [EMAIL PROTECTED]
wrote:
What is the best alternative for getting the IVR and other prompts recorded
for Asterisk.
We decided to record our own. We set up a recording studio, and that has
worked out very well for us.
Let me know if we can help.
Ron
,Hangup
That should capture just about anything that is all digits.
Ron
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typescript.
Then run asterisk -vvvcTn.
Then make a call inbound over the PRI.
Then exit asterisk with stop now.
then exit script with exit.
You should now have a typescript file with all of your asterisk session.
let's see what you get.
Ron
On Sunday 16 March 2008 18:19, broadband Voice wrote:
I tried that and got 14 errors, see below:
Sorry, switch those around, I gave you a zapata.conf, your original
zaptel.conf looks fine.
Ron
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Hello all,
please, is it possible to which party has hangup a call?
if yes, please tell me how ?
thanks,
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5
Ron
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that Raid 5 has deficiencies, and I would
not recommend a Raid 5 set.
With the disk sizes available today (both SATA and SAS), Raid 10 or multiple
Raid 1 sets have many advantages.
Ron
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for yourself, and thoroughly inspect the changes that
go into each new version, to see if they will bite you.
Same holds for cards, hardware and everything you change yourself.
Asterisk can be ready for primetime. But only if you make it
your main source of income.
Ron
/suggestions appreciated
goto bugs.digium.com, Click View results, and enter IAXVAR.
I got 4 hits, one of which is the patch in question:
http://bugs.digium.com/view.php?id=7619
Ron
smime.p7s
Description: S/MIME Cryptographic Signature
to AST_CAUSE_BUSY?
Ron
/Olle
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NeoNova BV
Hi List,
Ive got some of these boxes hooked up to a Digium card running EM wink.
Sometimes these channels go onhook for no reason, or when a person hangs the
phone up they stay on hook. Are their settings on the channel bank or the
card itself I am missing?
If anyone has any help or answers on
in
sip.conf as this will ensure that asterisk is kept in the audio path.
Doing so will allow MixMonitor to work.
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bilal ghayyad schreef:
Dear Jared;
Any web in english?
translate.google.com?
Ron
From where I can buy it?
Regards
Bilal
--
On Thu, 2008-06-12 at 01:23 -0700, bilal ghayyad wrote:
Where did u find a good IAX IP Phone?
I've had good success with my Allnet IP
try pri_cpe instead of pri-cpe
On Thursday 19 June 2008 12:51, Eve-Ellen Cole wrote:
I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1
crossover, and I'm currently stuck. Anyone have any thoughts on what I
can do to get past this?
Asterisk side
Digium TE220B w/
Vitelity provides me with this functionality.
http://www.vitelity.com
Ron
On Thursday 26 June 2008 17:36, Steve Finkelstein wrote:
Hi all,
I was curious if anyone can recommend a company that would work with
small businesses, and capable of using a fallback number (mobile
phone, home
Hi John,
*for the first part:
you can create 3 contexts: internal,external and main
in your internal context you put your internal extension
in the external context you send the send the XXX-XXX- to the
providers trunk
and in the main context you just include the internal context (first)
Hi List,
We are trying to make a click 2 call button, we have a PHP script that
passes the 1st phone number of the 1st leg to a manager script, that then
dials the 1st call, then the 2nd call gets placed inside of Asterisk using a
normal dial command. Problem is, we get no status codes, we cannot
I'm discussing, it is.
Jay,
Why would TBCT not be applicable in a scenario where * is being utilized as a
slave to a main PBX. * might receive a call from the PBX, and then want to
transfer it to another extension on the PBX itself.
Thanks,
Ron
Hello list,
i wanted to setup a small asterisk+ss7 lab this weekend and just installed
asterisk-trunk+ dahdi-complete+libss7+libpri
i had only a sangoma A101 card so i used it and 48h after i'm still
unable to make the card work in that config.
i tried to patch the sangoma drivers thinking that
Hy Guys!
I have Trixbox (2.6.1) set up with 2 analog ph lines going to 2 FXO
ports (2-X100P cards) I also have to deal with Panasonic hardware that
handles the initial calls. I would like Asterisk to serve as an auto
attendant for the first call and as calls come in pass them to the
Panasonic. My
I would add $.02
I found the install on Elastix less than error free. When the ISO cant get
MySQL loaded without errors I worry.
And the documentation (not that trixbox is well documented ) was weak IMHO.
Elvis
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
request to the PBX so Asterisk is
out of the loop?
Thanks,
Ron
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request to the PBX so Asterisk is
out of the loop?
Thanks,
Ron
--
Ron Joffe
Siena Tech, Inc.
3319 Willow Glen Drive
Oak Hill, VA 20171
(919) 928-0404
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in the past. In this case the number of
PRI's entering the PBX far outweigh the number of PRI's in the Asterisk
server, so it is not an option. I tried to simplify the example.
Any other suggestions ?
Ron
--
Ron Joffe
Siena Tech, Inc.
3319 Willow Glen Drive
Oak Hill, VA 20171
(919) 928-0404
standardized such codes ?
Regards
You need the ETSI standards. These include all of these and more are
available from www.etsi.org
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Hi List,
We have noticed on our Snom 360s that under missed/recieved calls the number
is cut off, so you cannot see the entire phone number. Does anyone have a
work around or is this a bug Snom is working on?
Cheers!
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Hi List,
I have a client who is using park heavily, but once we hit the cal button
(in this a hotkey tied to park orbit on the Snom's), we have a 3 second
delay before we here the digit the call is parked on. Is their anyway around
this at all? Does anyone know if we have these same delays if
as well, no luck!!
Asterisk 1.2.13 I am using on both boxes.
Can anyone provide any help on this? I think is rellly weird invites are
failing when im telling * to ignore them basically!!!
Phones are Snom 360's as well.
Thanks!
Ron
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will then remember your settings if possible, if anyone has left you a
voice mail etc.
Is this possible?
Regards,
Paul
This can be done with the Cisco's XML browser, web server scripting,
Asterisk Realtime and some ingenuity. It also requires a carefully
designed dialplan.
- --
Ron Wellsted
love to have three or four cells with the same CID
(all pointing back to my astericks box). It seems damn near impossible hear in
Kalifornia.
Ron Elvis Stephan
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen
Sent: Monday, July 02
of the only telco's get documentation crap)
Does anyone have a suggestion?
Thanks,
MD
You might want to look at the Pirelli Dual Mode DP-L10.
I tested one, and sound quality and stability are much better
than the Nokia E61 or of any other wiFi phone I tested.
Ron
on TV.
Ron Elvis Stephan
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bryan Laird
Sent: Tuesday, July 03, 2007 3:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID Spoofing to be banned
Do your SNOM phones sometimes use answer-after:0, and do
they have keyboard LEDs subscribed to their own extensions?
Do those people hangup calls by puttig down the handset
instead of pressing the X key?
We are seeing hanging channels in this particular case.
Ron
Michael J. Liberatore wrote
, but in this case, we
just want them on hold is all, no dialtone! Any help would be great!
Thanks!
Ron
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Hi list,
Does anyone know any ways to have mutiple parking lots? I've got a pbx
that 2 customers share, both need their own, and then have lights on
the phone flash when they park the call (snom phones). Any ideals I'm
not thinking of?!?
Any help would be great!
Thanks
Ron
In order to use this patch, i have to download the complete version of SVN
asterisk? I highly doubt this will work with the metermaid patch that allows
the call park buttons to work with Snoms. Last time I let anyone share a
PBX!!
Any comments on this would be great!
Thanks
Brad
On 1/26/07,
extension or hunt group.
Do Asterisk and Digium or Sangoma T1/E1 cards know about delimited ANI and
DNIS?
--Ron
On Sun, 18 Feb 2007, Eric ManxPower Wieling wrote:
It will do so automatically if it is working. Asterisk will stuff those
digits into ${EXTEN}, therefore you need an exten = _XXX,1
On Sun, 18 Feb 2007, Matt wrote:
Why would the card care? This would be something you'd take care of in your
dialplan.
Right, the card wouldn't care. So does Asterisk know about how to send
and receive delimited ANI and DNIS through a channelized voice T1?
--Ron
On 2/18/07, Ron Fox
to
an Asterisk box. Can't do that with with PRI and a single T1 because you
only have one control channel.
--Ron
On 2/18/07, Matt [EMAIL PROTECTED] wrote:
Why would the card care? This would be something you'd take care of in
your dialplan.
On 2/18/07, Ron Fox [EMAIL PROTECTED
Hey List,
Asterisk 1.2.13 with Sangoma Card and beta 14 drivers.
I am having problems with deadlock channels and having to kill asterisk, and
then restart it, cannot make calls in or outbound. This has happend about 4
times now, and the system was running fine for a few months fine.
Any
I gues ill look and see what version they are on, its a production system,
so that always scares me!!! But, good ideal!! :)
On 3/8/07, shadowym [EMAIL PROTECTED] wrote:
Ummm.
How about upgrading to production released drivers?
-Original Message-
From: Ron McCarthy [mailto:[EMAIL
Hi List!
Im using (or trying to) use AgentCallBackLogin() to have * find roaming
users, here is a diagram.
Server A (Hq)
Server B(Branch Site) Server C (Branch Site)
All my que users are on Server A, I have Server B/C dial a extension to call
to a SIP 6.x image
in between.
After the first conversion, you should be able to set the password etc
via the SIPDefault.cfg file.
HTH
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Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
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We have one connected.
What's your question ?
On Monday 01 December 2008 13:49, Mark Bergen wrote:
Anyone familiar with getting Asterisk 1.4 and Mitel 3300 to play nice
together?
Mark Bergen
Information Systems Manager
Number TEN Architectural Group
Winnipeg - 204.942.0981
Victoria -
Hi list,
I see their is ExtenSpy(), I want to monitor calls (in and out I hope) from
another phone, all the channels are SIP. ChanSpy() looks like you cannot
give it a context and I want to be able to only monitor certain calls. Any
Ideals on how to do this?
Thanks!
Works like a champ. I have to use the b option as well otherwise it just
goes into a endless beep, sounds good though!
Thanks for the help!
On Wed, Jan 7, 2009 at 4:02 PM, Mark Michelson mmichel...@digium.comwrote:
Ron McCarthy wrote:
Hi list,
I see their is ExtenSpy(), I want to monitor
Hi
the same happened here also with different distros (ubuntu and fedora 9)
each time i run dahdi start the kernel crash.
i was using the dahdi from trunk
regards,
David fire a écrit :
do you have any dahdi card ???
if not edit /etc/dahdi/modules so it dosent load any modules.
David
Hey !
this can drive to heart attacks
randulo a écrit :
Nice one, Olle ! :)
On Wed, Apr 1, 2009 at 9:18 AM, Olle E. Johansson o...@edvina.net wrote:
* NEW CHANNEL DRIVER FOR ASTERISK 1.6 AND VOXSWITCH 3 ADDS AUDIO AND
VIDEO TO MICROBLOGGING!
Great !
thank you very much for your job!
BR,
Matt Florell a écrit :
Hello,
We've released another update to our VICIDIAL/astGUIclient call center
suite: 2.0.5
http://astguiclient.sf.net/
The call center suite client applications run on most modern web
browsers on almost any GUI-capable
the calls disappear from Asterisk (and the people on the calls won't
know the difference). Otherwise, the calls will continue to be bridged
by Asterisk.
Jared,
Is there a debug mode where I can find these specific messages?
Thanks,
Ron
--
Ron Joffe
Siena Tech, Inc.
3319 Willow Glen Drive
Hi @ all,
i like this community,
i don't think that there is any place on this planet from where emails
are not coming directed to this community,
if governments were profiting to each other like the members of this
community do,
there would be no poor on this planet,
there would be no war on
to be connected via NFS due to the fact of the voicemails wanting to be stored on another machine while its primary is down, or is this not even possible?
Any help would be great!
Thanks
Ron
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some routing issue.
Here are mine (with UK regional settings/A-law).
http://www.wellsted.org.uk/spa3102router.html for the router
configuration as bridge and
http://www.wellsted.org.uk/spa3102voice.html for my voice configuration
with UK regionalisation (A-law, UK tones/cadences).
- --
Ron
FXO is coming from the PSTN, FXS is what devices connect to (like a analog phone).If you are using VOIP phone then you dont need the FXS modules, just FXO.On 8/24/06,
joea, j4computers [EMAIL PROTECTED] wrote:
As a complete newcomer to Asterisk, Digium and PBX, I have several questions.But I'll
-BEGIN PGP SIGNED MESSAGE-
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Mindaugas Kuprys wrote:
Hi,
Can anyone suggest good quality IP phone with 2 Ethernet jacks. I wanted
Sipura but they don't have such product.
Go for the Linksys SPA-942. It is what the Sipura SPA-841 evolved into.
- --
Ron Wellsted
[EMAIL
to
report a lack of D-channel.
Can anybody tell me where I am going wrong?
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[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 202120
FWD:519961
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. This is voice mail and ACD
only. It would be nice if I could recycle the Dialogic cards but that's
not a major requirement - I can always go to a pair of Digium TDM cards
if needed.
I appreciate any and all advice!
Thanks!
Ron Gage
Westland, MI
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You need BTs SIN351 at http://www.sinet.bt.com/351v4p5.pdf
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623
Hello list!Is there a way to program one of the buttons on the 501 (Like the services button) to do on the fly call recording? So in the middle of the phonecall you can record the call without have to do a transfer type of setup. Ive looked at the manual but cant seem how to do that, I only see
[EMAIL PROTECTED] wrote:
Hi Ron - Is there a way to program one of the buttons on the 501 (Like the services
button) to do on the fly call recording? So in the middle of the phonecall you can record the call without have to do a transfer type of setup. Ive looked at the manual but cant seem how to do
Im going to get a trial account, .014 to US is not bad at all!Only downside is that g729 is only codec they allow :(On 9/17/06,
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I saw this termination company, www.BuyMin.comthe rates looks good. Has anyone any experience with this company? I use
to be worst on inbound calls (TW - BT).
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Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 202120
FWD:519961
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this in
tftp-config file etc.
Thanks in advance,
Roland
The timeout is set in the dialplan.xml file with the Timeout tag. Like
this:
DIALTEMPLATE
TEMPLATE MATCH=\*1.. Timeout=0 User=Phone/
TEMPLATE MATCH=* Timeout=5 User=Phone/
!-- Anything else --
/DIALTEMPLATE
HTH
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Ron Wellsted
http
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The Netgear FSM7326P switch also supports the Cisco Pre-Standard directly.
We have these powering all our CP7960 phones perfectly.
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Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD: 519961
N 52.567623, W 2.137621
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