On Apr 22, 2005, at 2:13 AM, C F wrote:
Can you please post your .cnf files?
On 4/21/05, Robert Goodyear [EMAIL PROTECTED] wrote:
Has anyone experienced a problem provisioning lines 5 and 6 of a Cisco
7960 via a SIPx.CNF over TFTP?
I'm going to try Ron Wellsted's suggestion re the .CNF
Using RH 9 with *
Regards,
Umair Bari
David Choo wrote:
We used gentoo internally. I also have * running on CentOS, RHEL.
Best Regards,
==
David Choo
Systems Engineer
Business Technology Division
"Engineered for Changing Businesses"
Espore Corp Pte Ltd
68
On Apr 22, 2005, at 2:54 PM, Jay Milk wrote:
Sveasoft is useless -- use hyperWRT instead.
-Original Message-
From: Gregory Wiktor - ADCom Corp. [mailto:[EMAIL PROTECTED]
Sent: Friday, April 22, 2005 4:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
I am currently working on implementing Hotel Billing in IPSwitchBoard.
The idea is that a receptionist in a hotel can just right click an extension
button and choose Account; IPS will now calculate the call charges made
from that extension and show all calls and charges on a form.
The
--- Weitergeleitete Nachricht / Forwarded Message ---
Date: Fri, 22 Apr 2005 14:12:04 +0200 (MEST)
From: Georg Natsikos [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk transcoding
I would like to learn more over the transcoding function with asterisk. How
On 23/04/05, Michael DiMartino [EMAIL PROTECTED] wrote:
3. Extensions.conf (telx-NY17S)
;Extentions at telx-nyc
exten = _7XXX,1,Dial(IAX2/telx-nyc/${EXTEN})
exten = _7XXX,1,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN})
where username:password is the credientials you need to
I have eight MSN at home. Six are handled by Asterisk. The two remaining
are handled by an external ISDN modem which is connected to my HylaFax
on another machine. Asterisk and ISDN modem are plugged into the same
NT1. That works fine for me...
Works for me too.
We have an old fax machine
This seems to be exactly the application I was looking for :-).
Since I'm working on a project where accounting and billing (http and voip
traffic) is an issue, I'm glad to read that there will be a solution within
a reasonable GUI.
While dealing with squid and the great Squid2MYSQL script - used
HI,all!
I have setted up an Asterisk server on my linux.My sip.conf and extensions.conf configurations like these:
sip.conf
[general]port=5060disallow=allallow=alawallow=ulawallow=gsmnat=yes
dtmfmode=rfc2833canreinvite=no
hi,all!
Can anybody tell me what's the matter? thanks!!!
Do You Yahoo!?
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Hi,all !
My asterisk's console appear some words : "chan_sip.c:7174 handle_request : Failed to authenticate user "top" sip:1002:@10.0.0.1 tag=169447308" .
Can anybody tell me what cause it ? thanks!!!
Do You Yahoo!?
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Asterisk-Users mailing list
My notebook has three USB ports. I would like to use usb-phone(AU-100)
and usb-analogphone-adapter(TJ560B) using 'wcusb','wcfxs' and 'zap/1'
and 'zap/2' on CVS-v1-0-03/05/05 on FC3(2.6.11-1.14_FC3).
I could not make /dev/zap/1, /dev/zap/2 for usb devices. How should
I do? Do I need X100P
Title: Dial While on IVR
While the call is going into the IVR how can I Dial an extension and get immediately connected interrupting the IVR?
Robson
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Exactly what I am looking for also. Because we have multiple phones in one
villa, I would need the ability to group extensions and produce an overall
bill, and I would, of course, need the ability to set the charge rate versus
the cost, i.e., the cost is $.02/min, but we might charge $.50/min
These things are dirt cheap. Are they any good?
MARK.
Iassen Hristov wrote:
Maybe this fits the bill.
http://www.gigafast.com/products/product_detail/EE2400-SS.htm
It retails for less than $100
Message: 9
Date: Fri, 22 Apr 2005 10:42:20 -0700
From: Max Clark [EMAIL PROTECTED]
Subject:
did this. No joy
Ken Godee wrote:
I added the line
exten = 3701,1,Dial(Zap/g1/19173657597)
Unknown Number Plan (0) '19173657597' ]
-- Called g1/19173657597
I know we are moving forward. I didn;t get this last time I tried to
dial.
Try striping the 1 off and dial Dial(Zap/g1/9173657597)
My circuit is from MCI. They tell me to its and ATT switchtype
Andres wrote:
I know we are moving forward. I didn;t get this last time I tried to
dial.
Mark
Why don't you try changing your switchtype to national from 4ess in
your zapata.conf
___
Exactly what I am looking for also. Because we have multiple phones in one
villa, I would need the ability to group extensions and produce an overall
bill, and I would, of course, need the ability to set the charge rate
versus
the cost, i.e., the cost is $.02/min, but we might charge
I've no timer configured, that's it. Thank you for your help.
--
Antoine
Ron Wellsted a écrit :
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Antoine Courouble wrote:
Hi! I'am a new user and have problem with sound on a debian sarge. I
can't play any sound with musiconhold or voicemail. Sounds
Having said that I did it anyway and whadyaknow, it works!!
Thanks very much fellas for your help.
Mark
Mark Phillips wrote:
My circuit is from MCI. They tell me to its and ATT switchtype
Andres wrote:
I know we are moving forward. I didn;t get this last time I tried to
dial.
Mark
Why don't
Also needed is a way to title and logo the print out so it looks like an
invoice. A tempplate would work, and if can use HTML templates that would be
easy to customise. Consider making the data a table that is substituted into
the html template.
Chris Mason
www.anguillaguide.com
-Original
On April 22, 2005 05:35 pm, Mark Phillips wrote:
Ext: 1 Cause: Invalid information element contents
(100), class = Protocol Error (6) ]
-- Processing IE 8 (cs0, Cause)
Your zapata.conf does not match your telco's provisioning of the PRI.
Contact
your switch tech and verify
It can use DNS if the DNS servers are valid. Can you post your SIP.conf?
Didi you configure the phone manually or did you use the cnf files? If you
used cnf files can you post those also?
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When you do a sip show peers from the what IP address does it show for the
841?
- Original Message -
From: Brian Watters [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Saturday, April 23, 2005 12:52 AM
Subject:
Can anyone PLEASE help to get a pa168 ip phone connected to livevoip?
If I set use service it does not work. If I unset it, it works for a
while, then just busy tone.
If I set and unset use service, it will again work for a while.
--
No virus found in this outgoing message.
Checked by AVG
Has anybody success with speed dialing?
If so, I am sure you can help me to get into this club.
tgj wrote:
Hi Ronald,
It seems like you need to put in default as your context. However I think
your problem was that you put the number in CallerID column and The CallerID
in the Name column. I was
When someone teminates a call with my softphone to my
asterisk server i want asterisk to try provider 1
first and if the call does not go through because the
provider is having problems then it will try provider
2 gracefully.
I realize that the provider is having problems
statement is
Hi Guys
I am trying to get the Junghanns card to load on Suse 9.3
and tried to get it running on Fedora Core 3 (latest kernels). I have
heard from a source here in South
Africa that this is about as hard as pulling
teeth. Could someone please confirm this for me and if they do have
The DNS servers are valid. I configured the phone via .cnf files. The
following are the sip.conf and sipMAC.cnf files.
[tycisco]
type=friend
username=username
secret=secret
qualify=200 ; Qualify peer is no more than 200ms away
nat=yes
;insecure=no
host=dynamic
Rich- wouldn't Andrew K's solution work? That seems to
make good sense.
There are no real examples that would address your
points. The
primary reason is that your * can dispatch a call to
a provider
and the provider will accept that handshaking call.
But, if
they are having internal
Thanks Andrew for the great example! Anybody else have
any input?
Tom
--- Andrew Kohlsmith [EMAIL PROTECTED]
wrote:
On April 22, 2005 10:38 pm, Thomas Miller wrote:
When someone teminates a call with my softphone to
m
__
Do You Yahoo!?
Tired
SNIP
#user_info: phone
# SIP Configuration File (stop)
When the phone tries to register, all I get in the Asterisk
console is this:
Apr 23 08:22:29 NOTICE[26568]: chan_sip.c:8804
handle_request_register:
Registration from
'sip:[EMAIL PROTECTED];user=phone'
failed for '24.18.147.95'
Hi,
Have a look at http://www.voip-info.org/wiki-CallingCard+Applications
I recently used this in a hospital for the same concept. Can charge on caller
ID etc. Works really well.
Ties to a MySQL database, so a PHP interface can be coded to view the call
charges etc on a room. It works on a
- Original Message -
From: Thorben Jensen [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Saturday, April 23, 2005 8:11 AM
Subject: [Asterisk-Users] Hotel billing in IPSwitchBoard
I am currently working on
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Robert Webb
Sent: Saturday, April 23, 2005 11:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
List Receiver
Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device
Aye...that was it...
Thanks a billion!
-Original Message-
From: Robert Webb [mailto:[EMAIL PROTECTED] On Behalf Of Robert Webb
Sent: Saturday, April 23, 2005 8:54 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion; List Receiver
Subject: RE:
Works for me too.
We have an old fax machine sitting on the same NT1 as
asterisk. In asterisk I ignored the MNS by setting the line
exten = my_fax_msn,1,wait(30)
Doesn't it work without the wait() in .nl? I just didn't mention the fax
MSNs in my incoming context...
I tried, but my
Hi,
As mentioned before, how about being able to search and replay recordings
from the switchboard. With call records now searchable hopefully it
wouldn't take too much more work to enable. For example, being able to
search on extension by date and time or by cli would be very handy.
Now that makes me very excited. I have implemented a pbx in a datacenter for
a online stock exchange and they want all calls recorded. I am uncertain how
to handle recovery of the calls, though. This would be wonderful.
Chris Mason
www.anguillaguide.com
-Original Message-
From:
Peter thanks for the response.
I put the user name and password in but i still get the same error.
;Extentions at telx-nyc
exten = _70XX,1,Dial(IAX2/telx-nyc:[EMAIL PROTECTED]/${EXTEN})
Apr 23 12:30:35 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected
connect attempt from 192.168.0.251
What
Also needed is a way to title and logo the print out so it looks like an
invoice. A tempplate would work, and if can use HTML templates that would
be
easy to customise. Consider making the data a table that is substituted
into
the html template.
Chris Mason
www.anguillaguide.com
Hi
are you using udev ? If yes, check README.udev in the zaptel directory
On 4/23/05, Terry Wade [EMAIL PROTECTED] wrote:
Hi Guys
I am trying to get the Junghanns card to load on Suse 9.3 and tried to get
it running on Fedora Core 3 (latest kernels). I have heard from a source
On 23/04/05, Michael DiMartino [EMAIL PROTECTED] wrote:
Peter thanks for the response.
I put the user name and password in but i still get the same error.
;Extentions at telx-nyc
exten = _70XX,1,Dial(IAX2/telx-nyc:[EMAIL PROTECTED]/${EXTEN})
Apr 23 12:30:35 NOTICE[147465]:
Taking this idea a little further.
(I apreciate there may be legal issues with this request)
Would it be possible for extensions to be tagged, so that if they make
and / or recive a call the call is automatically recorded each and
every time, at the end of the call the file is closed
I would
Peter Bowyer wrote:
On 23/04/05, Michael DiMartino [EMAIL PROTECTED] wrote:
Peter thanks for the response.
I put the user name and password in but i still get the same error.
;Extentions at telx-nyc
exten = _70XX,1,Dial(IAX2/telx-nyc:[EMAIL PROTECTED]/${EXTEN})
Apr 23 12:30:35 NOTICE[147465]:
Peter Bowyer wrote:
On 23/04/05, Michael DiMartino [EMAIL PROTECTED] wrote:
Peter thanks for the response.
I put the user name and password in but i still get the same error.
;Extentions at telx-nyc
exten = _70XX,1,Dial(IAX2/telx-nyc:[EMAIL PROTECTED]/${EXTEN})
Apr 23 12:30:35 NOTICE[147465]:
Hello all,
Here is our problem ..
IP SIP phones remote ..
They will connect to our IP PBX (Asterisk Server) without issue however no
voice makes it when anyone answers a phone call made by one of these IP
phones.
So this means SIP is working but RTP is not, Here is what I currently have
on
Is there a specific SIP or IAX phone that truly shines above
the rest where it comes to happy compatibility with
Asterisk? I guess Im talking
about feature sets, like early-dial, off hook call announcing, conferencing, echo suppression, etc
etc.
I, like many others, bought a Budgetone
so how do we get this fixed, its happing to my one and only DID as well...
On 4/22/05, Me [EMAIL PROTECTED] wrote:
I had the same problem with another provider whom I got no response from as
usual..
We had 5 or 6 numbers that worked fine and one that just quit sending DTMF.
-
Hi,
I still have a reasonable number of Elmeg CS100 ISDN system phones and also
some Elmeg ISDN Dect sets lying around, which I ultimately would like to
connect to my Asterisk system. Using the basic ISDN functionality using an
NT capable ISDN card is no problem. I would however like to use the
Hello,
Does anyone know what the most affordable 8-port NT-capable ISDN card is
(that is compatible with Asterisk) ?
Thanks,
Taco Scargo
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Ours just started working again..
- Original Message -
From: Justin Richards [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, April 23, 2005 1:14 AM
Subject: Re: [Asterisk-Users] voice pulse connect - no dtmf
On April 23, 2005 12:31 pm, Michael DiMartino wrote:
exten = _70XX,1,Dial(IAX2/telx-nyc:[EMAIL PROTECTED]/${EXTEN})
Apr 23 12:30:35 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected
connect attempt from 192.168.0.251
The extension you're hitting doesn't exist in the context you are being
Peter Bowyer wrote:
On 23/04/05, Michael DiMartino [EMAIL PROTECTED] wrote:
Peter thanks for the response.
I put the user name and password in but i still get the same error.
;Extentions at telx-nyc
exten = _70XX,1,Dial(IAX2/telx-nyc:[EMAIL PROTECTED]/${EXTEN})
Apr 23 12:30:35 NOTICE[147465]:
On April 23, 2005 12:39 pm, Peter Bowyer wrote:
; telx-NY17S - Incoming
[telx-NY17S]
type=peer
secret=telx-NY17S
context=from-telx-NY17S
disallow=all
allow=ulaw
I think your nomenclature's wrong.
when I call an IAX host, I look for a matching type=peer entry in my iax.conf.
When I
(I apreciate there may be legal issues with this request)
The only legal issue is, I believe, you have too announce to the caller
This call may be recorded etc...
You are entitled to use call recording, if in doubt put a big sign up in the
room Calls are recorded
I think it's not an issue
Hi All,
Still having problems :-(
I have an Asterisk 1-0-7 setup on Debian 3.1 (Sparc)
I have severel SIP phones that call between each other and can chat no
probs. I can even call from the SIP phones to the sccp 7920 no
probs
However when I call from the 7290 to any SIP
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Message: 2
Date: Sat, 23 Apr 2005 08:25:29 -0700 (PDT)
From
I'm setting up asterisk and want everything to load on startup. The distro
I'm using is a RHEL4 rebuild (CentOS4).
Because the zaptel init script doesn't work I'm trying to set everything
up from rc.local. However asterisk fails to start with an error that the
zap device isn't loaded.
This is
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Message: 2
Date: Sat, 23 Apr 2005 08:25:29 -0700 (PDT
You need this before wcfxs
/sbin/modprobe zaptel
Regards,
Chris
- Original Message -
From: Remco Barende [EMAIL PROTECTED]
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Saturday, April 23, 2005 12:55 PM
Subject: [Asterisk-Users] ztcfg doesn't do anything from
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Message: 2
Date: Sat, 23 Apr 2005 08:25:29 -0700 (PDT)
From: Thomas Miller [EMAIL PROTECTED]
Subject: Re: [Asterisk
It shows the public IP and not the private IP ..
403/403 67.181.191.99 D N 255.255.255.255 5061
Unmonitored
BRW
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito
Sent: Saturday, April 23, 2005 6:12 AM
To:
IMHO its the Cisco 7960. I have 5 of them littered around mu house. My
wife uses the intercom feature to hunt me down when she has honey do
lists for me. I must get around to breaking that feature ;-}
Chris Coulthurst wrote:
Is there a specific SIP or IAX phone that truly shines above the
Ive tried to replace the gsm,
wav and WAV files in the /var/spool/asterisk/vm/default/201 directory with some
strung-together Allison files, but every time I try, it just plays the default greet.
Is this possible, or is it just that Im
doing something wrong?
Chris Coulthurst
[EMAIL
Hi,
This may be a dumb
question but I know how to provision lines but what is the use for them. Right
now I just have one line provisioned on my cisco 7690
and I get all incoming calls on that line and make calls on that too. Additional
lines may be mean additional extension numbers. But
On Saturday 23 April 2005 19:23, Bob Goddard wrote:
On Saturday 23 April 2005 19:13, Matt Klein wrote:
$4,172.38 USD and I'll programin anything you want for asterisk server.
You are too stupid for the job.
Quoting the 1200-line long Asterisk Digest message in your reply and adding
one
The funniest part is, he thought I was serious. I'd be dumb if I didn't at
least charge $4,172.39 USD for the job.
On Sat, 23 Apr 2005, Gary Stimson wrote:
On Saturday 23 April 2005 19:23, Bob Goddard wrote:
On Saturday 23 April 2005 19:13, Matt Klein wrote:
$4,172.38 USD and I'll programin
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I have to agree, the Cisco 7960 is probably the best (I have yet to try
a 7970/71). Cisco are a pain to deal with (they only want to deal with
large value customers/distributors) and the phone do have some small
quirks/bugs but they are the best in
Depends on the state you are in. In Nebraska there is no law saying you
have to tell someone they are being recorded if you are recording them on a
business line. In Iowa you don't have to tell them , but you have to play a
tone in the background every so many seconds.
- Original Message
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Manjit Riat wrote:
Hi,
This may be a dumb question but I know how to provision lines but
what is the use for them. Right now I just have one line provisioned on
my cisco 7690 and I get all incoming calls on that line and make calls
on that
Also on a side note, previously read
documentation aside, is there a surefire way to concatenate .gsm files together without getting any errors or losing
quality due to re-conversions?
I took several .gsm
files from the Allison pack and cated
them together. When I tried to
Hi,
I was trying to get the solution for the issue with getting dial tone after
dialing 9, in sip phone, but I couldn't get anything. I am using a
Grandstream Budgetone 100. I include ignorepat in the handset context, but
nothing.
Any guideline or help?
Thanks.
Jaime
On Wednesday, July 9,
ignorepat is for Zapata devices. Sip devices sned the number to the
swith AFTER the SIP device feels it has dialed it. I am not a pro on the
GS phones, (never played with them) but I would cheak the documentation
on setting up a 'dialplan'.
I hope this sets you in the right direction.
Alex
In article [EMAIL PROTECTED],
Remco Barende [EMAIL PROTECTED] wrote:
Because the zaptel init script doesn't work I'm trying to set
everything up from rc.local.
You should instead find out why the zaptel init script doesn't work for
you. It works fine for me under Fedora Core 3, with one
At 02:07 PM 4/23/2005, you wrote:
Hi,
This may be a dumb question but I know how to provision lines but what
is the use for them. Right now I just have one line provisioned on my
cisco 7690 and I get all incoming calls on that line and make calls on
that too. Additional lines may be mean
I don't know about other phones but on the Sipura, I set
all the lines to the same extension and incoming calls rollover on to the next
line appearance. Hence, I can hold one and take the next call,switch back
and forth easily, works great.
Anotherreason would be to have more than one
I just got a cisco 7960, a bit tough to get going at first
but it's a great phone. Supports OHVA, and the dialplan is very nice in
that you can have autocompletion based on your plan. for example, if I dial 300,
the phone completes, whereas if i dial a 1+ number there is a
timeout.
For
Grandstream does not support a dialplan. It is supposed to support
Early Dial, but didn't work. I've been told that recent firmware
fixes the early dial bug. I doubt that Early Dial is the solution.
The solution is to buy a good IP Phone. Polycom and SIPura both
support continue dialtone
Chris wrote:
You need this before wcfxs
/sbin/modprobe zaptel
*sigh*
zaptel will automatically load when the card driver loads.
modporbe will also run ztcfg after loading the card driver because (if
you ran make install) /etc/modules.conf tells it to do so.
--
Always do right. This will gratify
*sigh*
I always get an error if I don't.
Regards,
Chris
- Original Message -
From: Eric Wieling aka ManxPower [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, April 23, 2005 7:15 PM
Subject: Re:
I just wrote a simple cgi to have a form generate the number, then the
cgi creates a call file and bingo. Web call.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Elkins
Sent: Friday, April 22, 2005 8:21 AM
To: asterisk-users@lists.digium.com
Try adding a comma to your digitmap where you wish the dialtone to come
back on. Works on a Polycom.
On Apr 23, 2005, at 7:12 PM, Eric Wieling aka ManxPower wrote:
Grandstream does not support a dialplan. It is supposed to support
Early Dial, but didn't work. I've been told that recent
Hello,
I'm having some major problems getting SIP phones to register whenever I put
them behind a Linksys router. The same phones will register behind any other
NAT (I've tried 3 others without problems)
I've been debugging using Ethereal and these are the differences that I
found between
Please make sure you post any solution you find to this issue to the
list I have been frustrated by this as well.
Scott Henderson
Finite Technologies Incorporated
3763 Image Drive, Anchorage, Alaska 99504
Phone:
Is your problem on the same model of Linksys? WRT54G? I haven't had a
chance to try some other Linksys routers so I'm curious.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Henderson
Sent: Saturday, April 23, 2005 7:20 PM
To: Asterisk Users
Couldn't you just assign the OK softkey to send a '#'? Seems that my
local telco (with my 390 phone) sends many macros that are just DTMF
sent from the softkeys.
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED]
The Switch is since 1995 and I get a SX-200 Digital G1005 ENH 672P / F25.0
09-FEB1994 when I look up the software on the switch board so if I am
reading what your telling me then I have to do D4/AMI. So does my zaptel
look correct? Maybe my cableing is off.
Thanks,
-Scott
- Original
I have tried several, dlink doesn't seem to have the same issue and a
more intelligent firewall is not having any problems. We are working
with the Sipura 1001 and 2000 units on this issue.
Scott Henderson
Finite
I've got a 7960 behind a Linksys wireless box and its working just
fine with nat=yes in the sip.conf. Has been for over a year. Not
sure of the model though.
Is your problem on the same model of Linksys? WRT54G? I haven't had a
chance to try some other Linksys routers
The WRT54G work fine...
I have a Sipura 1000 and a Grandstream 286, both nated through a
WRT54G on a single public IP. Worked out of the box -- no special
settings needed. I was even surprised that I did not need to turn on
the NAT handling in the Sipura ATA.
Then I have a WRT54G running as a
I have a whole Asterisk server behind a wtr54gs. We have SPA-2000's
registering from the Internet into it with no problems.
Actually, we don't have it at the moment but did for several months.
Not sure if this helps any or just adds to the confusion.
- Original Message -
From: Rich
Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running
behind my Linksys WTR43GS with no issues. This is at home registering to an
external * box and to vonage.
- Original Message -
From: Luki [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
I'm trying to register BT100s ... (doesn't work)
X-Lite seems to work though
Tomas
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo
Sent: Saturday, April 23, 2005 8:48 PM
To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Have you tried to enable NAT translation on the Grandstream?
On 4/23/05, Tomas Florian [EMAIL PROTECTED] wrote:
I'm trying to register BT100s ... (doesn't work)
X-Lite seems to work though
Tomas
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
When using bristuff I do get an error too if I don't load zaptel first but
not with the tdm driver.
I know that in my modprobe.conf it is specified that ztcfg should be run
after loading the module but why doesn't it?
For some reason ztcfg is only 'accepted' when run from the cli
Thanks!
Jerry,
when you say digitmap, you mean in my extensions.conf file?
Thanks.
Jaime
From: Jerry [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
Yes that's the first thing I tried ... I'm able to make it work (using
different routers than Linksys) in the following ways:
- Set outgoing proxy and no STUN
OR
- No outgoing proxy and set STUN
But once I put it behind Linksys everything registration does not work any
more.
Tomas
Ron Wellsted wrote:
I have to agree, the Cisco 7960 is probably the best (I have yet to try
a 7970/71). Cisco are a pain to deal with (they only want to deal with
large value customers/distributors) and the phone do have some small
quirks/bugs but they are the best in functionality and build
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