Access-list 100 permit ip host asterisk server any
Class-map match-any voip
Match access-group 100
Policy-map voip
Class voip
Priority 256
Class class-default
Fair-queue
Interface fastethernet 0
Service-policy output voip
Above is what I do to prioritize 256kbit of outbound bandwidth
âechocanâ
make[2]: *** [/usr/src/wanpipe-3.3.16/kdrvtmp/sdla_tdmv.o] Error 1
make[1]: *** [_module_/usr/src/wanpipe-3.3.16/kdrvtmp] Error 2
make[1]: Leaving directory `/usr/src/kernels/2.6.18-128.1.6.el5-x86_64'
Jeremy Mann
This e-mail, facsimile, or letter and any files or attachments transmitted
I'm running some mysql queries on the standard sql logging of calls, and am
interested if anyone has any they'd like to share to get good statistics. I'm
interested in # of calls per day, based on DST. Number of Calls per day based
on SRC, avg duration of calls, etc..
Thanks.
Jeremy Mann
Just FYI:
IP address 89.248.168.176 has been trying to use the recently release SIP
vulnerability in Asterisk to make outbound calls via our box. They are running
a bank account callback scam.
Jeremy Mann
Director of IT
Texas Health Management Group
Direct Line: 817-310-4956
Main Line: 817
to know their line is ringing and not just in use. ?
2. Sort of tied to #1, does anyone have clear dialplan logic and polycom
config information about doing custom ringing per extension on the IP 650 ?
Thanks.
Jeremy Mann
Director of IT
Texas Health Management Group
Direct Line: 817-310-4956
I'm getting the following error over and over on the console:
pbx_dundi.c:2975 dundi_rexmit: Max retries exceeded to host
Any idea how to troubleshoot this?
My network latency is roughly 40-50ms between all hosts in my dundi cloud.
Jeremy Mann
Director of IT
Texas Health Management Group
- source UDP Source port: 4520 Destination port: 4520
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann
Sent: Wednesday, November 05, 2008 8:33 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Dundi Issues
I'm getting the following
-Commercial Discussion
Subject: Re: [asterisk-users] Dundi Issues
Jeremy Mann wrote:
I don't know if it's related, but when doing a packet sniff with
wireshark, I see UDP checksum incorrect messages:
0.058230 source - destination UDP Source port: 4520 Destination port:
4520 [UDP CHECKSUM INCORRECT
= yes
host = dynamic
[1203]
fullname = 1203
secret = 1203
hasvoicemail = yes
mailbox = [EMAIL PROTECTED]
vmsecret = 1234
hassip = yes
hasmanager = no
callwaiting = no
context = from-nortel
subscribecontext = internal
call-limit = 4
dynamic = yes
qualify = yes
host = dynamic
Jeremy Mann
Director
It happens nightly, and I have to reset asterisk to clear it. Zap/Dahdi
channels wont' work until I do.
Jeremy Mann
Director of IT
Texas Health Management Group
Direct Line: 817-310-4956
Main Line: 817-310-4999
Fax: 817-310-4990
Email: [EMAIL PROTECTED]
This e-mail
Does anyone have a recommendation for a headset that plugs into the
Mic/Line-out port on a PC?
Ideally something like the Plantronics SupraPlus. I'd prefer Monaural instead
of stereo, and cheap in price but not in quality.
Thanks for any suggestions...
Jeremy Mann
Director of IT
Texas Health
Does the call-limit directive work on those SIP items defined in users.conf as
it relates to presence and queues?
Jeremy Mann
Director of IT
Texas Health Management Group
Direct Line: 817-310-4956
Main Line: 817-310-4999
Fax: 817-310-4990
Email: [EMAIL PROTECTED
Tried using GROUP()?
When a call comes in or goes out:
Exten = XXX,1,Set(GROUP(bdwi_out_1)=outgoing/incoming);
Exten = XXX,n,GotoIf($[${GROUP_COUNT(outgoing/[EMAIL PROTECTED])}] 1?fail)
Exten = XXX,n,Dial(...)
Exten = XXX(fail),1,Congestion();
Exten = XXX(fail),n,Hangup();
Obviously choose
don't know why it counts the phone as a channel, though.
On Mon, Oct 20, 2008 at 12:14 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
Tried using GROUP()?
When a call comes in or goes out:
Exten = XXX,1,Set(GROUP(bdwi_out_1)=outgoing/incoming);
Exten = XXX,n,GotoIf($[${GROUP_COUNT(outgoing
I have a macro to dial out, similar to yours in that it fails over to Zap/Dahdi
trunks in the event our bandwidth stuff is overloaded.
I run this in a macro, and only set and check groups within that macro. I'm
confused why yours would attach to phones in any way, unless you mean phone
to
Is the IP 650 sidecar compatible with asterisk?
If I pair it with the IP 650 phone, can I have more than 6 lines registered
w/ the server?
Jeremy Mann
Director of IT
Texas Health Management Group
Direct Line: 817-310-4956
Main Line: 817-310-4999
Fax: 817-310-4990
Email: [EMAIL PROTECTED
Can anyone explain parked calls?
I've run so many tests over the last few hours I'm totally confused. Half the
time the call times out and returns back to the user that dialed it, through
the same context it was originated from.
The other half it returns to the park-dial context with a
Forgot to mention, I'm running asterisk 1.4.21.2
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann
Sent: Wednesday, September 17, 2008 2:32 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Parked
to below).
--
context internal {
...
...
t {
jump [EMAIL PROTECTED];
};
includes {
parkedcalls;
};
};
Jeremy Mann
Director of IT
Texas Health Management Group
Direct Line: 817-310-4956
Main Line: 817-310-4999
Fax
-Commercial Discussion
Subject: Re: [asterisk-users] Parked Calls
Jeremy Mann wrote:
Using the default features.conf setup, if I include parkedcalls in my
dialplan, and a call gets parked, and times out, where does the call go?
I can't tell you about AEL, but I have the following:
[park-dial
to another extension/context?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Tuesday, September 16, 2008 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Parked Calls
Jeremy Mann wrote:
Which
context from-pri {
_8505 = {
Wait(1);
Answer();
SetTransferCapability(3K1AUDIO);
Set(GROUP(ZAP)=incoming);
Set(CDR(accountcode)=fax);
Set(CDR(userfield)=bedford);
Does anyone know of a pri splitter device? Something that would take an
incoming PRI, and based on DID send that out one of other multiple PRI ports?
I'm needing to take a single PRI from the telco, and send it to two separate
phone systems(one asterisk) based on DID.
I know I could probably
Of Kevin P. Fleming
Sent: Wednesday, August 27, 2008 8:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI Splitter
Jeremy Mann wrote:
I know I could probably achieve the same thing with a 3 port PRI card in
a server, but I'd like something braindead
Yes, it's an _X. match for local/ld
It actually ended up being oddity with Centos 5.2, I had to upgrade Zaptel to
the newest version and it resolved it, apparently it wasn't passing all the
digits to the line.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
Can anyone help me start to diagnose why a Sangoma A200 wouldn't dial out LD?
Local calls are fine, incoming is fine, just no LD. Bell tech has been on site
and plugged into lines with his test set and was able to dial LD just fine, so
it's not a LEC issue.
No errors in asterisk console,
I have a Sangoma A200DX, and am trying to bridge an FXO channel with FXS for
modem connectivity.
I have Zap/8 as a Fax Machine
Zap/5 is my outside line. When a call rings in on Zap/5 it immediately calls
Zap/8 and bridges the channels. I see it doing a native bridge on the two. I
have echo
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zap Bridged Channels
On Wed, Jul 9, 2008 at 3:28 PM, Jeremy Mann [EMAIL PROTECTED]mailto:[EMAIL
PROTECTED] wrote:
I have a Sangoma A200DX, and am trying to bridge an FXO channel with FXS for
modem connectivity.
I
I need help translating extensions.conf to AEL:
[default]
exten = _X.,1,Set(DID=${EXTEN:6})
exten = _X.,n,Goto(continue,1)
exten = _1X.,1,Set(DID=${EXTEN:7})
exten = _1X.,n,Goto(continue,1)
exten = continue,1,Noop(${DID})
exten = continue,n,Set(GROUP(IAX)=incoming)
exten =
Is there a way to force asterisk to ignore the first ring of a call without
using Wait() ?
When I active *72 call forward on my analog lines from the telco, they always
send a single ring and then do the forwarding. Asterisk picks up essentially a
dead line and rings the phones which gets
this
maybe the way to go.
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP+with+no+passwords
You could also look at the incominglimit and outgoinglimit on IAX peers
On Wed, Apr 23, 2008 at 4:51 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm fairly sure SIP will never work
I have a macro that checks to see if a dundi route exists, if it does it
attempts to dial it. The remote end can set the chan as unavailable, or busy.
If it does the call immediately hangs up instead of returning to the macro for
more processing. Is there a way to force it to return?
Logic
Nevermind, helps when you reload the diaplan at BOTH ends :)
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann
Sent: Thursday, April 24, 2008 9:48 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Macro/Goto Help
I have a macro
, Apr 22, 2008 at 8:23 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
No.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Tuesday, April 22, 2008 6:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
: 192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
== Everyone is busy/congested at this time (1:0/0/1)
What is in the context macro-dundi-lookup?
On Wed, Apr 23, 2008 at 12:47 PM, Jeremy Mann [EMAIL
, Apr 17, 2008 at 7:38 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have it working via IAX, when I try changing everything to SIP I can't
specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Try GROUP()=internal-...
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Friday, April 18, 2008 11:30 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Question on groups
I believe I am close to fixing my problems with my 1.2 to
, in fact if you structure Dundi like
spokes around a small number of master servers, the config gets real
easy.Let me know how it goes.
On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf
Is there a way to specify per user attachment options for voicemail, from
within users.conf?
I know I can enable or disable it globally in voicemail.conf, but I have
certain users that like the attachment feature, and others that don't.
Also, can you enable/disable per user the deletion if
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box is
SIP/[EMAIL PROTECTED]mailto:SIP/[EMAIL PROTECTED]
How can you use
Subject: Re: [asterisk-users] Zap Codec
This is SIP channel you need to limit. Forcing ulaw only, the Polycom will fall
back to ulaw.
Per peer, in your sip.conf:
disallow=all
allow=ulaw
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann
Sent
on your location). You
CANNOT send calls in any other codec over a PSTN line. Generally, if
you want to use G729 then you must buy a G729 license (with a few
exceptions).
Jeremy Mann wrote:
But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only
want ulaw used when SIPPEER-ZAP
in stone.
Tilghman Lesher wrote:
On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote:
But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want
ulaw used when SIPPEER-ZAP is the case.
Set(_SIP_CODEC=ulaw)
Dial(Zap/g0/...)
--
Consulting for Asterisk, Polycom, Sangoma, Digium
a G729
license. No amount of discussion is going to change that.
Jeremy Mann wrote:
Sadly you are correct:
-- Executing [EMAIL PROTECTED]:4] Set(SIP/156-083514c0,
_SIP_CODEC=ulaw) in new stack
-- Executing [EMAIL PROTECTED]:5] NoOp(SIP/156-083514c0, 4) in new
stack
-- Executing
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
Mann
Sent: Tuesday, April 15, 2008 08:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zap Codec
I guess that's my frustration, I don't want it g729, I want it ulaw, I
just wish Zap did codec
Is there a way to force Zap channels to only use ulaw, and not even attempt
g729 negotiation?
My polycom 550 has G729,ulaw as priority, the Zap always fails b/c I'm not
licensed for the codec on the asterisk box.
This e-mail, facsimile, or letter and any files
to give user a prompt before connecting
thecall
I don't entirely remember - I was writing this code from memory.
Have you done any testing?
PaulH
On Tue, 2008-04-01 at 08:47 -0500, Jeremy Mann wrote:
Can I assume after exten=2,1,Playback(thanksfortakingthecall) there's more
logic, or does
,
and set the queue as need the memebrs to accept the calls. (not that I
can remember that option)
PaulH
On Mon, 2008-03-31 at 20:55 -0500, Jeremy Mann wrote:
Please do!
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Paul Hales [EMAIL
Please do!
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Paul Hales [EMAIL
PROTECTED]
Sent: Monday, March 31, 2008 7:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to give user a prompt before
I've got a couple of extensions in users.conf that have both SIP and IAX
access(IAX softphone, SIP hard phone).
I'd like to setup my dial string to check to see which they are actively
registered with, and send the call appropriately.
Right now I have:
Exten =
Is there a way to have a dundi host advertise extensions for another server?
A---B---C
I'd like A to reach C through B. A and C would handle the call, B would just
be the DUNDi intermediary.
Assuming A has 101-199
B has 201-299
And C has 301-399
A sample dundi/extensions/iax
Nevermind, figured it out. I had restrictions on the unsolicited calls in
dundi.conf.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann
Sent: Wednesday, March 12, 2008 12:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] DUNDi
Is there any way that I can have an admin user hit * and then Mute all other
users in a meetme conference? Sort of a moderator function?
I know it can be done with MeetMeAdmin, but as I see it that requires a
separate extension to dial, unless I've got the logic wrong?
If it can be done in a
To you extensions.conf gurus, I'd like some help on having a button/feature to
turn on/off system wide call forwarding.
I need the phone system to forward calls received, after the feature is
activated, to an answering service.
Calls received are on a PRI. I need all DIDs forwarded once the
Perfect! Thanks.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield
Sent: Tuesday, February 19, 2008 11:01 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] MeetMe Admin Functions
In article [EMAIL PROTECTED],
Jeremy Mann
Is it bridging the Zap channels? We have asterisk doing FXO-FXS modem calls
working fine, the key is making sure the channels are bridging and EC is NOT
turning on. If you have anything preventing that the modem calls won't work.
-Original Message-
From: [EMAIL PROTECTED]
Using Asterisk-1.4.17, Zaptel-1.4.8, libpri-1.4.3
Upgraded this morning, now PRI channels are unstable as hell. After about 5
minutes all asterisk commands on the console refuse to respond, attached is the
debug log right before and after the lock-up, IT occurred between 9:18 and
9:20 AM at
Has anyone ever written asterisk logic to Heartbeat remote phone lines?
Something that would dial out and see if a busy tone is encountered and take
some sort of action?
If not, any good ideas on how to do it? Obviously this would involve .call
files.
This
Just curious, if I have my Polycom IP 550 phone VLAN tag 30, will the packets I
send from my PC(on the PC port of the phone) have the same VLAN tag? THe PC is
sending untagged packets.
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that
Are the cordless phones on the 480i CT from Aastra registered independently in
Asterisk? Such that if I have 5 of the cordless phones hooked up, each one is
it's own extension?
This e-mail, facsimile, or letter and any files or attachments transmitted with
it
Do sangoma cards use the standard Zaptel drivers? Or do they have to be
compiled externally like Rhino cards?
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is confidential and privileged. This
On Nov 28, 2007 10:52 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
Do sangoma cards use the standard Zaptel drivers? Or do they have to be
compiled externally like Rhino cards?
Sangoma maintains a patchset that gets applied to the stock zaptel
drivers before compilation. They provide automated
:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail issues in 1.4.11
Jeremy Mann wrote:
Asterisk isn't playing my voicemail greetings even though they are
defined. Below are the relevant configs(from show dialplan) as well as
the level 3
Asterisk isn't playing my voicemail greetings even though they are defined.
Below are the relevant configs(from show dialplan) as well as the level 3
verbose messages asterisk is giving. Also a listing of the directory.
Asterisk just plays the The person at extension... message, not the
Without knowing more, Why fix what isn't broken?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Canfield
Sent: Friday, October 05, 2007 1:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Replace full PRI
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Replace full PRI with SIP/IAX trunks...YES/NO?
Jeremy Mann wrote:
Without knowing more, Why fix what isn't broken?
I should have stated, the PRI is on an existing PBX not asterisk. My
goal was to reuse the existing
Anyone know if Rhino is planning on supporting zaptel 1.4 anytime soon?
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is confidential and privileged. This information
is intended only for the use of
with zaptel 1.4 -- just be sure and get
the latest drivers which are now independent of the zaptel sources.
on Tuesday 10/02/2007 Jeremy Mann([EMAIL PROTECTED]) wrote
Anyone know if Rhino is planning on supporting zaptel 1.4 anytime soon?
This e-mail
Is there a way to tell asterisk, via a sip.conf peer, what IP address to send a
packet out of?
I've got multiple NICs in my box, each with it's own public IP. I need the SIP
messages to originate from any one of the IPs depending on which number was
originally called(and therefore where the
Of Benny Amorsen
Sent: Tuesday, September 25, 2007 1:55 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Multiple Home system with SIP
JM == Jeremy Mann [EMAIL PROTECTED] writes:
I would have answered, but I was prohibited from quoting properly:
JM If you are the intended
And if the Sip provider is sending data from 1 or two fixed hosts?
For instance, they send DID1 to IP A.B.C.D from 1.1.1.1
They send DID2 to IP E.F.G.H from 1.1.1.1
How do you differentiate? Would fromhost= work?
This e-mail, facsimile, or letter and any files
I'm curious if anyone has implemented the following:
Need to setup an on-call queue, that activates after 5PM and de-activates at
8AM, also that activates/deactivates on demand(I'm thinking a feature code
here). The agents need to log in via cell phones, and when calls come in
from outside to
Does G.729 phone - asterisk - G.729 phone work with reinvite turned off?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Watson
Sent: Tuesday, September 18, 2007 1:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I need some extensions logic assistance, I'm trying to dial out one of multiple
SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call
per trunk) and roll over to a second or third depending on that busy status
Here's what I've got for a macro thusfar, but it's not
Asterisk 1.4.11
Sorry, meant to include that
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrea Spadaccini
Sent: Monday, September 10, 2007 10:59 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Failover SIP logic
Ciao Jeremy,
Is the Cisco UC 500 able to integrate with Asterisk? Specifically does it work
via SIP? Just curious, as the Cold Call Cisco sales rep had no clue what SIP
even was, and this device looks interesting.
This e-mail, facsimile, or letter and any files or
I have an issue with receiving inbound calls.
I've got bandwidth.com trunks incoming to my asterisk box, bandwidth sends all
incoming traffic to one of two IP addresses, and requires outbound traffic go
to either of the same two IP addresses.
I've got to use fromuser=DID on outgoing calls so
I have a SIP phone calling via a SIP trunk another asterisk system, that then
sends the call out a ZAP channel.
When I press any of the features defined in features.conf, The end user on the
ZAP side hears the DTMF tones, and none of the features work.
My DTMFmode on the SIP users definition
Is the web GUI for AsteriskNOW able to be loaded on an existing server(that was
installed from ubuntu-server and asterisk loaded from source)?
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is
For 1.4: core set verbose 2
For 1.2: set verbose 2
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Andersen
Sent: Tuesday, August 21, 2007 9:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] CLI Question
1. Yes
2. Yes
3. Yes
Nice sales pitch, sounds like one of those late night get rich now! schemes.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson
Sent: Friday, August 17, 2007 4:35 PM
To: asterisk-users@lists.digium.com
Subject:
Is there a way to limit IAX trunks to a certain number of calls? For instance,
if I'm linking two systems in different regions, can I limit the number of
calls that go across IAX between the systems?
I've got some dialplan logic, but if there's some iax.conf directive to limit
the number of
indeed replace those with Privacy.
Maybe it could be a bug ,
On 8/9/07, Jeremy Mann [EMAIL PROTECTED] wrote:
I have a 2 port T1 card doing PRI passthrough, Span 1 answers from Telco,
Span 2 sends to my existing phone system(Nortel).
My Span1 gets sent to the context from
Should asterisk be intercepting DTMF on a bridged ZAP call? If so, how do I
disable it recognizing #, as it's hanging up my users when they try to enter #.
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information
Is there a way to recognize if someone called our PRI using an 800 number? The
DID is showing my 4 digit primary line, not anything obvious signifying that an
800 number is called?
This e-mail, facsimile, or letter and any files or attachments transmitted with
Does anyone have any tools to process CDR-CSV files into reports? I don't have
anything specific in mind, I'd just like some reporting examples so I don't
have to reinvent the wheel.
This e-mail, facsimile, or letter and any files or attachments transmitted
, it may be a nice starting point for you.
Moj
Alex Balashov wrote:
We at Evariste have a lot of experience writing all sorts of custom CDR
reports and would be happy to write what you need for you--very
inexpensively, guaranteed.
On Mon, 13 Aug 2007, Jeremy Mann wrote:
Does anyone have any
I have a 2 port T1 card doing PRI passthrough, Span 1 answers from Telco, Span
2 sends to my existing phone system(Nortel).
My Span1 gets sent to the context from-pri, detailed here:
[from-pri]
exten = _49XX,1,Set(CALLERID(all)=${CALLERID(all)})
exten = _49XX,2,Dial(Zap/g2/${EXTEN},,twk)
exten
asterisk*CLI show channels
Channel Location State Application(Data)
Zap/3-1 (None) Up Bridged Call(Zap/47-1)
Zap/47-1 [EMAIL PROTECTED] Up Dial(ZAP/g1/2105||TWK)
Zap/25-1 (None) Up Bridged
Is it normal for a PRI to reset the inactive B channels periodically(like once
every hour). I'm seeing on my asterisk console successful restarts, just
curious as this is all new to me.
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I need help on my zaptel.conf and Zapata.conf for a TE207P
I'd like Span 1 to receive a PRI from the phone company(US PRI).
I'd like Span 2 to interface with a Nortel Phone system as a PRI(acting as the
phone company)
Essentially my asterisk box is a man in the middle intercepting calls from
So would the timing be 0?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith
Sent: Tuesday, August 07, 2007 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TE207P Question
As an added note, you
In Zapata.conf, if my PRI is NI-2 configured, do I still use
switchtype=national ?
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it contains information that is confidential and privileged. This information
is intended only for
Does anyone know if X-Ten or SJPhone support multiple cordless handsets for
multiple lines? I have an office with multiple roaming users(nurses) that are
in and out. I'd like to provide them telephones, and my idea is to have a PC
sitting in a corner somewhere running a softphone client.
you would think the telcos would be more interested in selling this to
small/medium businesses that are not ready for a voice pri but it
Since when to the telcos have the consumer's best interest in mind? They can
sell you a PRI at full loop cost with a smaller number of channels in the
ESI Phone systems are supposed to support IP stations via SIP
integration(http://www.esi-estech.com/products/systems/ESICS/), has anyone ever
tried to link Asterisk with one of these?
I'm thinking my asterisk box could be an extension off that phone system, that
would then provide a Dial by
Do you just passthrough from FXO to FXS on the channel bank? Does asterisk do
the passthrough or the channel bank itself?
I ask because we're considering an Adit 600 internally and that's one of my
pending questions.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Jeremy Mann wrote:
Do you just passthrough from FXO to FXS on the channel bank? Does asterisk
do the passthrough or the channel bank itself?
The Adit hooks up to the Asterisk via a T1 cable, so you'd need a Dual PRI card
in your Asterisk box. Our channel bank is on channels 25-48. Asterisk
Can an asterisk box equipped with a Digium T1 card handle Integrated T1
circuits? I have a T1 with 768k data and the remaining channels voice, can the
asterisk box do the Data routing + Voice processing?
It's only going to support 4-5 users(the voice channels won't all be active
obviously).
Here's a silly question, if these are standard POTS you obviously know which
number corresponds to which line, being the case couldn't you tell that ZAP/1
is POTS 555-1234, ZAP/2 is POTS 555-1235, etc etc?
I'm assuming you're trying to identify the inbound number from the telco that
was
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