was to upgrade to SIP 7.5
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
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iQEVAwUBQvupt0tP/KMNOfRbAQJfBAf
works like a charm.
Some versions of Cisco firmware require the tftp server to be specified
with option 150, not option 66 (tftp server name)
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
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using 711u) ...
Thanks a lot ;)
Julien.
Is the 794G configured to use a VLAN ? I had a similar problem with VLAN
and ARP packets. I got around this by not using VLANS but putting the
phone on a separate subnet (with QoS on the phone subnet).
- --
Ron Wellsted
http://www.wellsted.org.uk
found themselves in a similar situation? If so, how did
you work around it?
Is it worth raising a TAC with Cisco?
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
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, Messages URI (set to 8501) or in the SIPDefault.cnf file
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
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--
then lines 5 and 6 appear on the phone console.
Weird!
/rg
There is a known issue with the size of the SIP*.cnf files on some
versions of the firmware. The cure for this is to stripout all comments
from the .cnf files.
HTH
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD: 519961
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playing. Someone have same problem or can help me?
Have you compiled and installed zaptel and loaded ztdummy?
HTH
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
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Simon Morris wrote:
On Fri, 2005-04-22 at 11:13 +0100, Simon Morris wrote:
On Thu, 2005-04-21 at 21:36 +0100, Ron Wellsted wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Morris, Simon wrote:
Hello,
I'd like to program my Cisco phones
?
Regards,
Greg
You could try CDW or Insight, see http://www.voip-info.org/wiki-Cisco
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
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in functionality and build quality.
They are also the best speaker phone for small conferences.
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
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getting into PBX?s and haven?t had much
experience with them.
2.) Registering the phone with more than one server (possibly in
different parts of the world).
3.) Different caller IDs
Personally, I use the extra line buttons as speed dials.
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL
.
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
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iQEVAwUBQmzm+0tP/KMNOfRbAQK8PwgAqRxXp2flCXqTKeavdHbMswURHquzZjYh
DyJeou3WCXsNeTthH7lAi+J8xLQEwjlOva+vW+cUvlEqAzCetGoDLEtsC+HBwCfr
to Asterisk with 7960s. We have had a few
little problems but have not lost a call yet. OK, we have left a few
callers on hold a bit longer than we intended, once or twice ;)
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
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-BEGIN
allow=ulaw
allow=alaw
context=Ipnotic
canreinvite=yes
nat=yes
dtmfmode=rfc2833
But it does'nt work... * try to dial with the port 5060 when i specify to him
to dial on the 5061 one...
Any idea ?
Try
[SIP-OUT]
type=peer
host=10.XX.XX.XX
port=5061
defaultip=10.XX.XX.XX
HTH
- --
Ron
attachment as well as the background process. I need to attach
to the running asterisk in order to do init keys but once I do that,
it seems I cannot just let it go into the background again.
Any suggestions most welcome.
try quit
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED
,NoOp()
exten = _9.,104,SetGroup(voip2)
exten = _9.,105,Dial(${VOIP2}...)
exten = _9.,106,Hangup()
exten = _9.,205,NoOp()
exten = _9.,206,Playtones(congestion)
exten = _9.,207,Congestion()
exten = _9.,208,Hangup()
HTH
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961
the file modifications for udev (see README.udev).
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
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.
Any help on obtaining the updated firmware quickly and painlessly would be
great... :-)
Cheers
M
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
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=autodetect
stripmsd=0
dialtype=tone
mode=immediate
msn=yourMSNhere
group=9
dtmfmode=asterisk
incomingmsn=*
device = /dev/ttyI0
device = /dev/ttyI1
You _WILL_ need to set your MSN (change yourMSNhere to your full MSN,
usually without the area code).
HTH
- --
Ron Wellsted
http://www.wellsted.org.uk
output suggests that zaptel wants the crc-ccitt module to be
loaded.
You need to ensure that your kernel build includes this module and that
it is loaded.
HTH
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961 Gossiptel:9309811
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-BEGIN PGP
? Ive been looking at using a
Juniper Session Border Controller, but not sure if thats gonna do the
trick, and then we also have SER..
Any comments would be great!
Thanks
Ron
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.-Original Message-From: JR Richardson [mailto:[EMAIL PROTECTED]]Sent: Fri 3/10/2006 8:55 PMTo:
[EMAIL PROTECTED]; asterisk-users@lists.digium.comCc: [EMAIL PROTECTED]
Subject: re: [Asterisk-Users] Clustering--Message: 6Date: Fri, 10 Mar 2006 12:22:12 -0700From: Ron
Regarding OSPF, so your saying you have multiple * boxes setup with
same exact config and then just have OSPF fail everthing over to the
new server if it cant get to it? That makes sense, just never of even
thought of doing it that way. Heck, if you want to get real complex
just run BGP and you
from Juniper's site. Have you seen anything on this?
Thanks!
RonOn 3/11/06, Gabriel Afana [EMAIL PROTECTED] wrote:
Hi Ron,
I've been following your
thread. I noticed you mentioned about a Juniper Session Border
Controller. I checked online and read about it, but was unsure exactly how
for it!!!). We're using
OpenSER's send() command to forward registrations from a phone to all Asterisk
systems.
-Original Message-From: Ron McCarthy
[mailto:[EMAIL PROTECTED]]Sent: Sunday, March 12, 2006 1:29
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk
ill be using in the
end :)
Once I get this going, I want to post a entire howto on the wiki.
Thanks!
RonOn 3/12/06, Gabriel Afana [EMAIL PROTECTED] wrote:
Hi Ron,
If the SBC would have served
mearly as a load balancer...I already have one and it didn't work too well.
I have a Foundry
SSL offloading, nothing like packet rewriting though.
So I think we are back to SER or a SBC from someone...
Thanks!
Ron
On 3/12/06, Gabriel Afana [EMAIL PROTECTED] wrote:
On a side note, the ServerIron can do Reverse-Nat
where it will rewrite the source IP to its Virtual IP and when requests
perfectly wherever the call originated.
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 202120
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iQEVAwUBRBa/qUtP/KMNOfRbAQKxGAf8DNDFTudN+rKXVVyhUyAJ2X9Ku9oZYg0F
, I use the archives http://lists.digium.com/mailman/listinfo/
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 202120
FWD:519961
-BEGIN PGP SIGNATURE-
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Anyone have experience with the 3-08-2 release of Cisco's SIP firmware?
Are there any new features in the SIPDefault.cnf?
Thanks,
Ron
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To UNSUBSCRIBE or update
).
Previously I was running 1.0.7 without this problem, I upgraded to fix a
problem with Monitor (the call stopped monitoring when transfered, 1.2.5
has fixed this).
Does any one have any suggestions?
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux
On Wednesday 22 March 2006 00:33, Nathan Alberti wrote:
Here is a dump of the configuration options, you will see there is a
few new, these are also documented on the wiki.
Nathan,
How did you go about obtaining the dump ?
Thanks,
Ron
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wed, 22 Mar 2006, Simone Cittadini wrote:
Ron Wellsted ha scritto:
This is slowly driving me nuts!
I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk
1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing
On Wednesday 22 March 2006 10:01, Nathan Alberti wrote:
Telnet to the phone, login and type show config
Thanks Greg and Nathan!!
Ron
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file look like this:
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone=uk
defaultzone=uk
HTH
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 202120
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-BEGIN PGP SIGNATURE-
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with the entry in your odbc.ini file and does
that use a driver that is in the odbcinst.ini file?
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 202120
FWD:519961
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.2.2 (GNU/Linux)
Comment: Using
- --
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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iQEVAwUBRDpDsktP/KMNOfRbAQJMiAf/Urn4l7OJb5Ki4/1MuxwszUe37bbjTF/Y
at
the TCP header. Also, qualify will send a 'keep-alive' to keep NAT
from losing the association of ports.
HTH,
-Ron
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phones
behind NAT. Otherwise the router doing NAT may flush out the port
mappings relative to your phone. The qualify essentially sends a
keep-alive. We have Polycom IP500s and 501s and this works very well
for them (one sitting right here on my desk).
-Ron
to fully investigate, I
reloaded 8.2 fw which registered straight away.
I did run a quick sip debug ip on the phone and saw asterisk replying to
the register request with 401 errors.
I hope to post more details when I get time.
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N
to fully investigate, I
reloaded 8.2 fw which registered straight away.
I did run a quick sip debug ip on the phone and saw asterisk replying to
the register request with 401 errors.
I hope to post more details when I get time.
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N
,
Ron Arts
NeoNova.nl
Adam Robins wrote:
We have built an Asterisk network using an MPLS-based IP VPN. We have
one location in New Brunswick Canada that consistently gives us major
quality problems, whereas the others are flawless. Quality problems
take the form of static, poor voice tonality
- Network - Listen RTP port )
Thanks
Joao Pereira
AFAIK, the firmware requires the RTP port range to be above 16384. My
7960 with 7.5 SIP firmware is working fine on 18060 - 18078
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
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-BEGIN PGP SIGNATURE
you,
Scott Miller
How about the Cut application (depreciated)
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+cut
or the CUT function
http://www.voip-info.org/tiki-index.php?page=Asterisk+func+cut
depending on which version you are running?
- --
Ron Wellsted
http
up with a solution that will suit
everybody, but at least that would prevent customised files being
overwritten.
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
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Michael Graves wrote:
Hi All,
I'd really like to see a show of hands with regard to people's
experience with FXO interfaces. I own a few X100p cards and have had
nothing but problems with them.
I also took part in Sipura's beta program, for the SPA-3000. While it
can be an improvement over the
to landlines or
landlines to mobiles.
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
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default firmware
of 1.0.1.9 but it shouldn't be even looking at Asterisk yet.
Do I need something more just to browse to these configuration pages in
the device? All 3 units are doing the same thing.
(I did not find anything in the FAQ's or documentation.)
Ron
You were right on. I turned down my security settings in IE and it went
into the config just fine. It's the simple things...
I found the Firefox issue earlier as well.
Thanks for the help, guys.
Ron
Philip Edelbrock wrote:
Ron Bulthuis wrote:
I just purchased a Grandstream gxp-2000
(happen to be booted to the windows side at
the moment) and for some reason Firefox is working fine now.
It was always set to accept cookies. If I get the error again, I will
repost.
Thanks again.
Ron
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, other Finarea sites (1899.com, 18185.co.uk) do use https.
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
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FWD:519961
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Comment: Using GnuPG with Thunderbird - http
7960P0S3-07-5-00/loadInformation7
/Default
This should trigger the upgrade.
HTH
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
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hmm. That's a tad awkward
:-|
- Original Message -
From: Douglas Garstang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, January 19, 2006 5:00 PM
Subject: RE: [Asterisk-Users] Dundi Examples
The
7960s
(SIP 8.2 fw) and HFC BRI ISDN cards last week, I went back to 1.2.7 and
every thing seems fine now.
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 202120
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Version: GnuPG v1.4.2.2 (GNU/Linux
Administrator/
/The Possibility Forge, Inc./
/http://www.possibilityforge.com/
/435.635.0591 x.1004/
You must use | as the separator instead of , in the realtime engine
so stdexten,101,sip/101 should be stdexten|101|sip/101 etc.
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N
problem until
upgrading to 1.2
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
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been discontinued?
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
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Hash: SHA1
Hermann Wecke wrote:
Ron Wellsted wrote:
I have been trying all the major distributors but they are all out of
stock with no dates for new stock to be delivered.
As you are in the UK, why not talking directly to Billion? Maybe they
can help
that they are going to get
harder to find.
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 202120
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iQEVAwUBRNoum0tP
://www.rongage.org/zapata.conf
Thanks for your help!
Ron Gage
Westland, MI
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Atlas.
HELP!
Ron Gage - Westland MI
[EMAIL PROTECTED]
This message was sent using IMP, the Internet Messaging Program.
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asterisk
yesterday and today I am getting D-Channel
errors.
Thanks for your assistance!
Ron
Quoting C F [EMAIL PROTECTED]:
What is the PRI connected to:
What hardware for the T1?
What Motherboard?
On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote:
Hey guys:
I am having a bit of a problem with our PRI under
of queue has not been transmitted
yet.
Ron
On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote:
The PRI is connected at one end to an Adtran Atlas, I believe a 600.
The other end of the PRI is connected to a Digium T100. The two are
seperated
by roughly 10 foot of cat-5.
Motherboard is whatever Dell put
Quoting Bill Gibbs [EMAIL PROTECTED]:
What does the telco say when they test the circuit?
Bill
Bill:
I am having my remote hands check first on the Adtran that is feeding the
Asterisk box, then then go upstream from there.
Thanks for helping me see the obvious path to follow! :)
Ron
it
really helps to have all your ducks in a row when dealing with an ILEC - they
just don't seem to have much of a sense of humor about these things.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ron Gage
Sent: Thursday, August 17, 2006 3:29 PM
help!
Ron
Bill Gibbs wrote:
I know but you could save some time and have it tested while
waiting...they might find a problem and save you a lot of headache. I
can tell you are one of the rare people who actually checks their stuff
before calling anyone but like another posted said, D Channels tend
the victim of a mail loop?). This will require a not
insignificant amount of somebodies time and bandwidth.
Not that I am trying to put you off the idea, but it does need to be
considered very carefully.
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux
I am testing toll free and US DID inbound
as well as A-Z outbound with les.net at the moment. Both the quality and
support are quite good. Ping time to Vancouver is around 80ms.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marcel Eric Loiselle
Sent: Wednesday,
for sipdiscount?
Thanks in advance.
Regards,
Chandra.
See the wiki: http://www.voip-info.org/wiki/view/SipDiscount
and check that you do currently have credit.
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 202120
FWD:519961
to
receive/place a call to tell me if it sounds the way its supposed to or if
there is indeed a problem?
All suggestions/recommendations greatly appreciated.
Much thanks,
-- Ron
[EMAIL PROTECTED]
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Depending on how you connect to the PSTN and what type of call is being
made, you may have access to the ANI II digits. The II digits tell you
what type user/service originated the call from such as: regular phone,
hotel/motel guest phone, pay phone, inmate phone, and various types of
Look here for details on the North American Numbering Plan:
http://www.nanpa.com/reports/reports_npa.html
The report named Non-Geographic NPAs In Service lists the Toll Free and
Premium assignments.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I think this message is saying that it received a 200 OK for a REGISTER
message that Asterisk does not know about (anymore).
Is you system trying to register with an ITSP or other SIP-based system?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
You may need to create (or modify) the tftp file in /etc/xinetd.d. For
example:
service tftp
{
disabled= no
socket_type = dgram
protocol= udp
wait= yes
user= root
:
if (strcasecmp(via, SIP/2.0/UDP)) {
ast_log(LOG_WARNING, Don't know how to respond via
'%s'\n, via);
return -1;
}
Check you log and see if the warning message is there.
Ron
_
From: [EMAIL
Whoops, sorry - it only handles SIP/2.0/UDP; which is what is expected,
but it seems like it only checks for the VIA header for REGISTER, INVITE,
CANCEL, BYE, and SUBSCRIBE requests.
Ron
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andre Kirchner
Sent
Post your IP tables configuration here if it isn't too big.
Ron
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Curt Shaffer
Sent: Saturday, November 18, 2006 5:05 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion
Do your user agents use some services from the server such as DNS?
Ron
-Original Message-
From: Curt Shaffer [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 18, 2006 5:41 PM
To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-
Commercial Discussion'
Subject: RE: [asterisk
(TIMEOUT(digit)=2)
- --
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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iQEVAwUBRWCVKEtP/KMNOfRbAQJnzwf
!!!
Thanks
Ron
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Yeah, doing more testing shows that the speed keys are broken, but dialing
it works!!! Ugg!!!
can you let me know if you get a new firmware? Im going to try and
downgrade...
Thanks!
On 11/22/06, Alban [EMAIL PROTECTED] wrote:
Yes, already.
Waiting now for a new firmware...
Regards,
Alban
:
http://www.snom.com/wiki/index.php/Snom360/Firmware/Release_Notes#6.5.2_beta
http://www.snom.com/wiki/index.php/Snom320/Firmware/Release_Notes#6.5.2_beta
http://www.snom.com/wiki/index.php/Snom300/Firmware/Release_Notes#6.5.2_beta
Regards,
Sven
On Wednesday 22 November 2006 17:56, Ron McCarthy
Hi Guys,
So the new firmware seems to work great, except. if you hit transfer and
then dont hit a key, or dial a extension within literally 2 seconds, the two
calls on hold bridge. As you can imagine, chaos!!!
Is this a firmware problem, or a setting im missing?
Thanks!
On 11/27/06, Jamie
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I have an example here you can look at:
http://www.siplabs.net/wip/sip-redirect.html
Ron
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on it. Any help or
examples would be great on this!
Thanks!
Ron
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Hi Rob,
Well see that would work great if I knew the numbers they would be calling,
but all I know is the source number/phone, i have no clue who they will be
calling. Any ideals now? I wish it was that easy!
Thanks!
On 12/6/06, Rob Schall [EMAIL PROTECTED] wrote:
Ron,
I believe you would
Schall [EMAIL PROTECTED] wrote:
Ron,
By source #, i assume you mean you have something like a SIP phone on
the network with the extension like 4455, and you want that to have a
different caller id when you make outgoing calls, then the rest of the
phones on your network (the rest would show a global
a context just for that
phone you can set a different callerid, then use a default context for
all the other phones.
On 12/6/06, Ron McCarthy [EMAIL PROTECTED] wrote:
Hi List,
Ive got one extension/login that when they call out from that it needs
to
show a different name/number, and then the rest
:
http://lists.digium.com/mailman/listinfo/asterisk-users
I have the same problem with a span from Bell Canada. After time, calls
begin to fail with the same Ring requested ... error message. I found
that if I restart Zaptel and Asterisk, that the problem goes away for a
while.
Ron
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What is needed is a family of astdb manipulation commands:
astdbput family key value
astdbget family [key]
astdbdel family [key]
any others?
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter
= default
include = international
include = national
include = local
include = mobiles
[national_local]
include = default
include = national
include = local
[local_only]
include = default
include = local
Then register your users in the appropriate context.
- --
Ron Wellsted
[EMAIL PROTECTED] http
Hi List,
Has anyone got the record button to work on the Snom's? Im looking to have
it send a email with a attachemnt of what the user records I hope. It looks
like you just point the button to [EMAIL PROTECTED] and just have that extension
record it. Any clue on how to do this, just use
button to record calls perhaps. Let me know if you think of anything!!
Thanks!
Brad
On 1/12/07, Ale [EMAIL PROTECTED] wrote:
Hi,
Ron McCarthy wrote:
Hi List,
Has anyone got the record button to work on the Snom's? Im looking to
have it send a email with a attachemnt of what the user
.
http://www.junghanns.net/downloads/
Cheers,
Steve
On 1/12/07, Ron McCarthy [EMAIL PROTECTED] wrote:
Hi,
This does nothing for me at all! :( I dont have to map the record key to
dial *1 or anything? Also, have you figured a way just to make the
record
button work when not on phone, like a memo
Thanks for your contrib
On Nov 22, 2007 2:56 PM, J. Oquendo [EMAIL PROTECTED] wrote:
So I was bored yesterday and tried solving a few
problems with one stone:
1) Notify me of potential brute forcers (multiple attempts
to register multiple numbers from one address)
2) Notify me of (l)users
Hi List,
What phones support shared line appearance? I would like a phone where we
can place calls on a line and have them picked up at another phone, but we
don't want to use call parking. I want to use this in a multi tenant
environment so I would need multiple lots. Any ideals for me?
Thanks!
The VoIP load includer SER on the router, only difference I am aware of.
On Nov 28, 2007 8:44 AM, Dovid B [EMAIL PROTECTED] wrote:
- Original Message -
From: David Boyd [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Well we need a light on the phone to blink when a call is on hold, but we
want to pick it up from any phone, so its a BLF key/light tied to it. Maybe
you can intergrate that with ques, I guess I need to look into that more,
just have never heard of that being done!
Thanks for the suggestion, Ill
Asterisk 1.4 im guessing? I did not know the Snom's worked with that, Ill
have to check it out then!
Thanks!
Brad
On Nov 28, 2007 9:28 AM, Russell Bryant [EMAIL PROTECTED] wrote:
Ron McCarthy wrote:
What phones support shared line appearance? I would like a phone where
we
can place calls
On Friday 14 December 2007 14:43, Vincent wrote:
OTOH, having to run a separate PC just to handle calls from a single
POST line AND having to install Linux + Asterisk on this thing... It'd
have to be an appliance (which I haven't seen avaiable in this price
range).
Didn't you just define an
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