On 01/05/2011 01:51 PM, Tom Rymes wrote:
On 01/05/2011 7:50 AM, Andy Graybeal wrote:
We've got two noisy kitchens that need to talk back and forth.
Andy,
Why, exactly, are you trying to combine an inter-kitchen intercom and
your phone system? Might it make more sense to have a non-
er--Vandal-Resistant-
Substations/SIP-Vandal-Resistant-Substation/
Tilghman,
Thank you for the response. The zenitel.com link looks nice in the picture!
-Andy
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e 12 key keypad for $450.. uhg.
This is great information, thank you for sharing.
-Andy
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On 01/05/2011 07:50 AM, Andy Graybeal wrote:
I'd definitely look into a phone mounted to the wall that has no actual
handset, but merely buttons and a speaker grille. It should probably
additionally be stainless steel, as I suspect it will need a good cleaning
at least daily.
The Po
be no substitute for an old analog wall mount
phone with a really loud ringer (backed by an ATA). That doesn't help
you with intercom though...
j
Jeff, thank you for your insight. Thats the second vote that I
shouldn't be getting a regular phone to act as an intercom in a k
ebsite btw; I like the color scheme.
-Andy
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magined I would find, but I've not found this yet.
Thank you for your response Tilghman.
-Andy
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estion. I have been on the fence on how I should do
this, and your last paragraph succinctly outlines what I've been
thinking and leaning towards. I will follow your direction.
Thank you for your response. I'm good at
er device for the kitchen all-together?
-Andy
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On 01/03/2011 07:53 PM, cjwstudios wrote:
Andy,
The 501 and 320 are EOL. I'd go for the IP335 and a 2626-PWR, since the
2626 can support vlans you can isolate data and voice. Make sure to
spec a UPS on the PoE switch.
CJW,
Awesome. Thanks for the input. For some reason or anot
2883-12883-3445275-427605-427605-3751584-3658873.html
)
It's got 12 PoE ports, it's managed, and it looks like I can pick one up
for under $500.
Any help is appreciated.
-Andy
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ape down the switch if
> needed. You could also cover the phone in "glad wrap" (except the speaker
> of course).
>
Is there a Polycom 501 that is POE and one that isn't? Or all they all POE?
-Andy
--
_
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the handset.
Faceplates.. interesting, a quick search on 'polycom 501 faceplate' or
'polycom 501 stainless steel faceplate' in google doesn't come back very
enthusiastic. Is there such a thing?
Thank you so far for the feedback. It's made me feel more confident and
ex
be cheap to get on ebay though :)
-Andy
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ve been
considering the Cisco 7920 in a holster w/ wired headset.
I'm welcome to any recommendations.
thank you,
-Andy
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s. The
problem is that I need to be able to change the MP3 that is played.
Has anybody managed to solve this problem?
Thanks,
Andy
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difference?
Thanks,
Andy
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asterisk-u
--snip
I am not certain of the reason for rejection but it has to do with the
SDP, it does not seem to be a codec issue, the response is as you have
seen:
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP
192.168.0.14;received=172.28.20.106;branch=z9hG4bK42d2ea03;rport=60017
From: &
ot find my box.
Thanks,
Andy
CST4*CLI> sip set debug on
SIP Debugging enabled
Reliably Transmitting (no NAT) to 192.168.34.1:5060:
OPTIONS sip:192.168.34.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK7a19a314;rport
Max-Forwards: 70
From: "asterisk" ;tag=as613ee548
you
what's being sent back and forth.
On 7/20/2010 9:36 AM, Andy Beak wrote:
Hi,
No that is the correct address. I know it is an internal IP.
We have our machine hosted in racks at our SIP providers data center.
They've patched a new port to our cabinet and linked that to a ga
appears immediately
I don't think it's a timeout issue.
Will reading the source for pbx_spool.c at line 339 give any clues as to
what's happening or will that be a waste of time?
Cheers,
Andy
On 20/07/2010 05:42 PM, Gareth Blades wrote:
If you add qualify=yes to the setting in
1.209 ms 1.196 ms
3 192.168.34.5 (192.168.34.5) 23.270 ms 23.269 ms 23.328 ms
4 * * *
5 * * *
6 * * *^C
Is there a way to test in Asterisk if it is able to reach a particular
IP address? Do you think that there is a routing problem here?
Thanks,
Andy
On 20/07/2010 04:58 PM, Zeesh
.
Thanks for your reply,
Andy
> In your sip.conf, there is no mention of your sip provider's IP
address, username and secret (password). Even if the provider doesn't
have username and secret
> requirements, there should at least be his IP address somewhere in
your sip.conf. Do they
nd have tried shuffling the gsm up above it in case it doesn't
work properly (to no avail).
Can anybody help me on this? My boss is breathing down my neck and I've
never worked with Asterisk before.
Thanks,
Andy
smime.p7s
Description: S/MIME
On 5/5/10, Adrian Marsh wrote:
> Anyone have any experience with a Japanese local VoIP termination
> supplier?
>
>
>
> I've emailed a few companies looking to setup some PSTN to SIP and SIP
> to PSTN termination, but no luck so far.
>
>
>
> Thanks,
>
>
>
> Adrian
>
>
>
>
--
Sent from my mobile d
On 5/5/10, Adrian Marsh wrote:
> Anyone have any experience with a Japanese local VoIP termination
> supplier?
>
>
>
> I've emailed a few companies looking to setup some PSTN to SIP and SIP
> to PSTN termination, but no luck so far.
>
>
>
> Thanks,
>
>
>
> Adrian
>
>
>
>
--
Sent from my mobile d
R_PHONES},dis,3)
exten => s,n,System(/bin/vol_restore)
exten => s,n,Hangup
exten => h,1,System(/bin/vol_restore)
Any suggestions? I am running Asterisk 1.6.2.5.
Thanks,
-Andy
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Hi,
the system() part pointed me in the right direction.. Thanks, going to give
it a test now..
Thanks!
Andy
On 29 March 2010 20:24, Zeeshan Zakaria wrote:
> Hi,
>
> I have done it a few times. Just posted a small blog about it with code.
> Check it at www.ilovetovoip.com/?p
Hi Danny,
Thats excellent, thank you. I have limited knowledge on writing AGI, but I
am always up for a challenge!
Thanks
Andy
On 29 March 2010 14:08, Danny Nicholas wrote:
> The built-in Dial command will not satisfy this requirement (first pickup
> terminates function). You could
hanks!
Andy
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asterisk-users mailing li
Hi.
I have the same opinion as Remco. Seems it is not as convinent as before.
Now each time I download new version I have to visit url
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ and then
choose latest version of them.
On Tue, Dec 1, 2009 at 4:52 PM, Remco Barendse wrote:
>
Hi Thomas,
Hope this will be helpful for you:
http://www.voip-info.org/wiki/view/Asterisk+AGI+php
On Tue, Dec 1, 2009 at 8:46 AM, Thomas Perron wrote:
> I am trying to find an AGI script that runs via PHP and performs the
> send text application.
> Does anyone have any tools or scripts set up
Andy Howell wrote:
> I am unable to dial out over a Wildcard TDM400P. This was working previously,
> so must have
> messed up the config somehow. I'm running Asterisk 1.6.0.15, with FreePBX
> 2.5.2.2.
>
> When I dial, I see:
>
>-- Executing [...@macro-dial
card, channel 1 is my analog phone, 2 my fax, and 4 the POTS line.
More config files etc below. Any ideas?
Thanks,
Andy
/etc/dahdi/system.conf
# Autogenerated by /usr/sbin/dahdi_genconf on Wed Jun 10 22:20:05 2009 -- do
not hand edit
# Dahdi Configuration File
#
# This file is parsed b
o jam the conference ID being used into the
account code or user field CDR fields.
Thanks for any help!
-Andy
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ay
to do it so far.
Thanks.
Andy
On Wed, Sep 2, 2009 at 12:09 AM, Lenz Emilitri wrote:
> Aht i would do is prepare a music on hold that has embedded the
> advertisements ( like one every 20 or 30 seconds) so that the caller hears
> more advertisements as the call progresses; and t
Hi Barry,
I used a "while" loop and Playback() like you suggested. It does the
job. Thank you for the suggestion. I just thought there might be
some built-in function or parameters in queue.conf that can do the
trick.
Thanks.
Andy
On Thu, Aug 27, 2009 at 12:32 PM, Barry L. K
Hi Barry,
Thank you for the hint, but I forgot to mention that we have a few
advertisements, and we want the callers to listen to only one at a
time, and in a round robin or random order. Using Playback() doesn't
seem to serve that purpose. Is there any better way to achieve that?
Thanks.
; ("Hold time")
;periodic-announce = queue-periodic-announce ; ("All reps
busy / wait for next")
;
reportholdtime = no
;
;;memberdelay=1
;; timeoutrestart = no
;
member => Agent/151
member => Agent/152
member => Agent/153
member => Agent/
REGISTER work. At least, changing
that config element and then "sip reload" got my BV peering back.
I'll send a note if I find out anything else. And I'd certainly like
to hear from any other BV users who might have seen a recent change.
Andy Valencia
___
Hi Singh,
Have you tried "soft hangup"?
Andy
On Wed, Mar 25, 2009 at 4:38 PM, Singh Saimbhi wrote:
> Hi,
>
>
>
> I want to send hangup command to the call which was logged in earlier via
> cli. Lets say to '5aec0e7207b24c8e1bdb511a460f7...@callcentric.com
&g
Hi,
Try CLI> soft hangup Local.
Andy
On 8/27/08, Rilawich Ango <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> I have the following queue and members. I found that there is a
> call stuck in the queue so other call can't enter the queue. I want
> to know whether w
that as opposed to waiting for
the extension to become free.
Is there any known way around this..?
My call file looks like this:
Channel: Local/[EMAIL PROTECTED]
MaxRetries: 100
RetryTime: 1
WaitTime: 5
Extension: 666
Archive: yes
Callerid: Callback <666>
Thanks!
Andy
On 21 Aug 2008, at 14:40, Philipp Kempgen wrote:
> Andy Dixon schrieb:
>
>> I am trying to alter the outbound callerID for extensions within a
>> context I have created.
>>
>> I wrote the following:
>>
>> exten => _9.,2,ExecIf($[$["${REALCALLERID
the callerID for (for example) 700 and 701 to be
581557, and any extensions not listed above, it should leave them alone.
If I call from extension 666, I get the correct outbound number (as it
does exist), but the rules above are not being followed.
I have tried to use Set(CALLERID(num)=581500)
Hi,
Why not use MixMonitor(), so you have a single file with the singer
and the music?
Thanks.
Andy
On 5/20/08, Sherwood McGowan <[EMAIL PROTECTED]> wrote:
> Arjan Kroon | Mobillion wrote:
> >
> > Hi,
> >
> >
> >
> > Is it possible top use a form o
ou use for clients you
support.
You will then be able to see the log messages generated on your own
equipment, without needing access to the asterisk box. However, you
will need to log into the asterisk box to make changes as per your
customers' requirements !
Andy
o specify a penalty that is
associated with which queue members receive the call, e.g.
exten => s,1,Answer
exten => s,n,Queue(support|t|||10) <-- penalty 1 gets the call this
time ..
exten => s,n,Queue(support) <--- but somehow specify penalty 5 and
below
.
I have setup call-limit, limitpeer... and everything what was in
documentation but nothing helps.
Can somebody help me?
Thanks a lot!
Andy
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To
to you.
Maybe there is another solution how to do that.
Btw. I am putting this stats in MySQL database.
Andy
Al Baker napsal(a):
> Why would you want a "channel to continue" after the caller has hung up.
> I clearly am missing something here because I can't see what good that
Thank you for your answer.
But the Dial command has a option 'g' which means that after succes will
proceed next priorities in the dialplan. Is there something also for
Queue() because according to manual there is no option for it. So I am
looking for some other solution.
Hello everybody.
Is there a way how to setup asterisk to notify caller's phone?
Example:
I have some numbers and names in asterisk database ( cidname, cidnum),
and I want to display the name of person on my phone ( which has no
addressbook, but can display chars ) which I am calling to be sure t
enable mediaproxy RTP proxy
on my OpenSER box to interoperate correctly
with Asterisk,
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ad ?
Best wishes,
Andy Davidson
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Class 4 soft switch with a full LCR routing engine, reporting system and
analytics engine. It's pretty powerful.
Right now we're using just the SBC component and sending all ingress
traffic to a egress trunk group (pointed to our OpenSER routers) but
we're running a few thousand conc
ought
I'd start investigating now ;)
We have a live Asterisk 1.2 server, and an Asterisk 1.4 server currently
setup for testing.
Regards,
Andy Neillans
Systems Designer
Blueberry Consultants Ltd
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re a Asterisk expert
willing to look at it with me for pay of course?
TIA,
Andy
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I am currently using Viatalk. I signed up with voipjet but can't get the
IAX connection to connect. I am going to give nufone a try and see. Thanks
for the suggestions.
Andy
From: Dovid B [mailto:[EMAIL PROTECTED]
Sent: Sunday, April 08, 2007 12:24 AM
To: [EMAIL PROTECTED]; Ast
s not
allowing me to do this and if so does anyone have any suggestions on some
voip providers that will let me provide the caller id info?
Thanks,
Andy
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the same recording to play to the call screeners?
2. Does anyone have any dial plan examples of this type of set up?
Thanks,
--
Andy Hester
Network Engineer
Architel
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-Original Message-
From: [EMAIL PROTECTED] on behalf of Philipp Kempgen
Sent: Sun 4/1/2007 4:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT
Andy Hester wrote:
> exten => s,n,Set(TIMEOUT(respo
, allow them
to screen the call, connect the call to the first number that accepts the call,
and allow others to reject the call.
Thanks,
Andy
[macro-screen]
exten => s,1,Wait(1)
exten => s,n,Background(csp_ackshort-male)
exten => s,n,Set(TIMEOUT(response=10))
exten => 1,1,NoOp(C
d the community in general wish
to use our hosting.
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Office: (414) 944-0162 x1029
Direct: (414) 944-0190
On Wed, 2007-03-14 at 17:03 -0600, Stephen Bosch wrote:
> Steve Totaro wrote:
> > I think Digium should host a wiki (keeping if
On 1 Feb 2007, at 14:14, Lacy Moore - Aspendora wrote:
> On 2/1/07, Andy Davidson <[EMAIL PROTECTED]> wrote:
> > What I would expect to happen, is that Asterisk would transcode
> > between the ulaw/alaw party, and me, wanting to listen via
g729. Is
> > this wh
D]
Give that a try !
Cheers
-a
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http://www.devonshire.it/ - 0844 704 704 7 - Sheffield, UK
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bug ? Any patches I can try
to see if they work ? Or is it my config which is broken ?
Inbound calls work ok, I guess this is because they are presented as
alaw and asterisk is just passing them through (which of course isn't
what i re
.
cheers
-a
--
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http://www.devonshire.it/ - 0844 704 704 7 - Sheffield, UK
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icemail is ever hitting mailbox 1112).
-a
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http://www.devonshire.it/ - 0844 704 704 7 - Sheffield, UK
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gt;
> You need a timing device on both ends.
>
> Zoa
>
But ztdummy should suffice yes?
Andy
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I set up a trunk and so far calls can be made one way, but not the
other. It is probably just not configured correctly, but I just wanted
to make sure as I can't seem to find any reason at the moment.
Thanks,
Andy
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In the current setup, asterisk is behind a different nat/firewall than
the LAN phones. The phones are using sccp. If the asterisk box is
compromised, it is not on the local LAN. This is what I think he
doesn't want to give up.
Andy
> -Original Message-
> From: [EMAI
up going with
this anyway. Any other feasible ways to accomplish this?
Sorry for the top post... Having to use Outlook for the moment.
Thanks,
Andy Hester
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen
Sent: Friday
ak some things. (Need to be able to record all calls need MWI)
Should I run 2 asterisk boxes connected with maybe TDMoE? Would that
work?
Any suggestions would be greatly appreciated.
Thanks,
Andy Hester
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Hi Steve,
I tried txgain as low as -18 without any problem, but I never tried
anything with decimal points.
Andy
On 12/12/06, Steve Hsieh <[EMAIL PROTECTED]> wrote:
Greetings everyone,
I have a Digium TDM400P card with both an FXO and FXS module to connect to
the phone company an
(hence 4 and half minutes every time)
If this is the case where do I change the ZAP (or is it VM) silence detect setting
Regards
Andy Green
IT Manager
GB eye Ltd
1 Russell St
Kelham Island
Sheffield
S3 8RW
Tel: 0114 252 1611
Fax: 0114 272 9599
mailto:[EMAIL PROTECTED]
http://www.gbeye.com
have checked the manufacturers websites that I know of but don't seem to be able to find anything.
I am not looking for a mobile phone network enabled device as there is no requirement for it to be used away from the local WiFi network
Regards
Andy Green
IT Manager
GB eye Ltd
1 Russell St
K
o see the digium card automatically once zaptel is loaded, rebuilt, fresh installed etc.
Win2000 is asking for a PCI controller driver install, can this be ignored or do i really have to install some win drivers for the card, if so where do I find them?
Regards
Andy Green
IT Manager
GB eye
and why it happened and how to
correct it.
Any suggestions?
Thanks.
Andy
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e, but we have quite a few of them
on hand that we would really like to use.
Any comments/suggestions are greatly appreciated.
Thanks.
Andy
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Hi,
If I just want to briefly test the T1, what is the basic config. I need
to setup?
Thanks!
Andy
Garth van Sittert wrote:
Andy Chung (Power-All) wrote:
Hi all,
I have connected a T1IDA-P to the Digium TE405P. Checked with the
Telco, and confirmed the T1 is up and connected. However, I
Hi all,
I have connected a T1IDA-P to the Digium TE405P. Checked with the Telco,
and confirmed the T1 is up and connected. However, I have no idea how to
test the T1 is really work, because the Asterisk server not yet be
configure. Anyone has the method on how to test the calls through the T1?
Hi Douglas,
Thanks for your advice. So is there any alternatives?
Thanks!
Andy
Douglas Garstang wrote:
That might not be a good idea. If you transfer or forward calls on your phones,
you MUST make sure the transferred or forwarded call goes back to the same
Asterisk box it was handled on
that?
Thanks!
Andy
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on Kapanga Softphone as suggested, and I'll tried it on
Grandstream ATA's later.
Are there anything I'm missing?
Thank you.
Andy
On 8/24/06, Ricardo Carvalho <[EMAIL PROTECTED]> wrote:
Hi,
I've installed Asterisk t38passthrough branch and I'm using one
Gra
o:
Fax machine ---> SIP ATA --LAN--> Asterisk --PRI--> PSTN
Have you tried this? Do you have to disable Echo canneler?
Thanks.
Andy
On 8/15/06, Marco Mouta <[EMAIL PROTECTED]> wrote:
Hi,
>Another question. With latest version of asterisk softwares am I able
>
Hi all,
I just search for the Load Balance and HA solution for the Asterisk
servers. I visited http://www.vovida.org/ and there is a Load Balancer.
Did anyone try that application? If yes, please give comment about it.
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Hi,
Can you give a quick example on how to query an EXTERNAL database?
Thank you.
Andy
On 7/29/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
On Fri, Jul 28, 2006 at 04:08:19PM -0500, shawn bright wrote:
> i would use a dial plan, but we are monitoring about 1200 units in the
> fiel
Check out "ethernet extenders"
http://www.rad-direct.com/App-Ethernet-extender-copper.htm
On Thu, 2006-07-27 at 15:39 -0600, Brian Vincent (C) wrote:
> Two questions:
>
>
>
> 1. We need to run Ethernet out to a really long distance –
> 20,000ft. We have the ability to put a powe
Hi,
I too would like to set a minimum jitterbuffer value, and that seems
to mean that I need to use the old jitterbuffer implementation.
Have you compared the 2 implementations? What are the advantages of
using the new one and what are the disadvantages of using the old one?
Thanks.
Andy
On
try "iax2 show netstats"
On 6/23/06, Dr. Michael J. Chudobiak <[EMAIL PROTECTED]> wrote:
Is it possible to set up some sort of call-quality statistics
reporting/logging for IAX2 calls? Something that can keep track of
dropped packet / jitter trends?
(I know "iax2 show channels" shows this info
Hey all. We have a DS3 circuit with GBLX split off into 7 systems with
a 4 port sangoma card (A104D) in the first 2 systems, and digium T410P
cards in the other 5. GBLX numbers their spans from 0 to 3 instead of
1-4 and we have a NFAS configuration with the d-channel on chan 96. All
of our s
? We have a 1.2.0 with PRI's on it, and 3
others running 1.2.4 and 1.2.7.1. They are all connected to each
other through IAX2.
Please let me know what you need from us to test with you.
Thanks.
Andy
On 6/15/06, Carlos Alperin <[EMAIL PROTECTED]> wrote:
Are you still interest
Went to their site today. Site claims they are still in biz. What is
the story? What really happened to Nufone anyway?
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the right direction. Cheers Andy
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for it and don't seem to be able to work
out how to direct different FXO ports to different * extns.
I am told that my Alcatel is not passing any
info (DID
number etc) down the line
Any help would be greatfully appreciated.
Regards
Andy GreenIT ManagerGB eye Ltd1 Russell
StKelham Isla
gards
Andy GreenIT ManagerGB eye Ltd1 Russell
StKelham IslandSheffieldS3 8RW
Tel: 0114 252 1611Fax: 0114 272 9599
mailto:[EMAIL PROTECTED]http://www.businessgbeye.com
Please read: CHANGE OF COMPANY NAME.
As of 1st January 2006 GB Posters Ltd will be known as GB eye Ltd, please update all record
uggestions. Thanks.
Regards
Andy Tan
--
Andy Tan
[EMAIL PROTECTED]
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or over the web
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this helps. Let us know how it goes.
Andy
On 4/13/06, Gareth Blades <[EMAIL PROTECTED]> wrote:
> Just noticed that I occasionally get these messages:-
>
> Apr 12 09:27:03 WARNING[11390] chan_iax2.c: chan_iax2: ast_sched_runq
> ran 281 scheduled tasks all at once
> Apr
channel/path. That would make off-loading bandwidth utilization
for media impossible. Appreciate any input. Thanks.
Regards
Any Tan
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ding and confidence in it.
References to other systems can be useful also. Hope it helps.
Regards
Andy Tan
On Wed, 12 Apr 2006 11:15:24 +0100, "Joao Pereira"
<[EMAIL PROTECTED]> said:
> Hello to all
> Im looking for a billing tool for Asterisk, that works with PostgreS
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