Displaying 20 results from an estimated 10000 matches similar to: "SPA841 3.1.1(a) firmware file"
2004 Dec 17
0
Total newbie here looking to do a VoIPconference call?
Thanks for that. I just got rid of packet 8 and went with 100% asterisk
in my house.
But I use the PAP2-NA and RT31P2 from Linksys for my FXS ports. But
would
like to have an extra FXS laying around just in case..
.o-------------------------------------------------------o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-----Original Message-----
From:
2005 Jun 14
4
488 Not Acceptable Here
I have a whole bunch of remote devices connected to my Asterisk box,
including SPA-3000s, PAP2-NAs and Cisco 7960s. The PAP2-NAs I have only
rolled out recently and I am having a problem that is intermittent and
inconsistent.
It happens to some users but not other users on the same ISP. It happens to
users in 2 different countries where the Internet setup (NAT issues) are
completely different. It
2004 Dec 18
1
One-way audio with SIP client only on certain calls
Hello.
I have an * server set up on a public IP. I have SIP clients at three
different locations, all behind NATs. I have all the SIP users set up
this way:
[user1]
type=friend
username=user1
secret=password1
callerid="User 1"<101>
host=dynamic
qualify=yes
context=outgoing
All three SIP clients are configured to use STUN (using
stun.fwdnet.net:3478).
Furthermore, I have
2005 May 17
1
One * server unavailable when multiple servers connected together
Hello.
I was just brainstorming for a future project and was hoping to get some
creative ideas from the list. If I have multiple * servers at multiple
locations all connected together with a nicely partitioned dialplan (2XX for
office 1, 3XX for office 2, etc.) it's pretty straightforward to link them
all using IAX and allow intra-office transfers.
Further, servers at each location are
2004 Dec 17
2
Total newbie here looking to do a VoIPconfer ence call?
Come to think of it since the DTA310 uses DNS to find the SIP server, you
could setup a DNS cache and override the DNS entry for what packet8 uses
(proxy2-eqix-sjo.packet8.net : 15062, over here) to point to the IP of your
own SIP server? Kind of a hack but it should work as long as it's running
on port 15062. I am very new to this so I don't know if there's a port
standard for SIP
2005 Jun 05
1
Accountcode being ignored?
I have a sip.conf entry for a customer's PBX (IP based authentication) that
reads:
[customer]
type=friend
context=customer
host=x.x.x.x
accountcode=10000
disallow=all
allow=g729
When the customer makes a call to my * server, * recognizes the peer
correctly. However, for some reason, the AccountCode is blank. I have a
NoOp(${ACCOUNTCODE}) and the CLI shows:
-- Executing
2006 Apr 20
4
Announcement System for a Charity
I'm putting together an Asterisk server for a local charity to use as an
announcement system. I've been thinking about how to write the dialplan to
allow different options for different groups' announcements, as well as
mailboxes for the various groups and the charity's administrators. Of
course, this would also need to include an option for the heads of the
different groups to
2005 Sep 03
0
Sipura spa841 problems
Guys.
I just unpacked on of the new spa841 I orderd and I was changing the
ringtone (and listening to the options) when suddently the phone stopped
playing back the tones and now the phone doesn't ring, speaker doesn't work
and no ringtone play can be heard.
Has anybody had this kind of problems?
2005 Aug 03
0
Dead spa841
Hello all,
I had been using a spa841, and all the sudden it's dead. When you power
up, all the lights come on and nothing else.
Has anyone experienced this?
Thanks,
Greg
2006 Mar 02
3
snom 320 MWI light
Hello.
I am using a snom 320 running 5.3.6 with Asterisk 1.2.4. In the sip.conf
entry, I have mailbox=1234@default and vmexten=*98.
The light on the snom 320 turns on when I have voicemail and the retrieve
button dials the correct extensions.
However, the light turns off immediately after making the call to voicemail,
even if I do not check the voicemail.
Any idea on how to get this to behave
2005 Jan 25
3
OT: pinout for"standard"telephoneheadsetrequired.?
> Many thanks Julian.
Are you looking for the pinout for a single plug 2.5mm (cellphone)
headset or a dual plug 3.5mm (computer) headset?
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeel<at>jafferali.net
2005 Jan 06
3
IAX outgoing redundancy
Hello.
I am having an issue where sometimes the cheapest provider for certain
international destinations is not always reliable in completing calls.
However, there is not problem once the call is made (i.e. no lag or echo
or anything). The way I have it set up right now (for example) for Dar
es Salaam, Tanzania is:
exten => _925522XX.,1,Dial(IAX2/livevoip/011${EXTEN:1})
exten =>
2005 May 07
5
Good NAT Pnp Hardphone
Hello All,
I am looking for a sip phone that is capable of automatic nat. The
Cisco ata186 for example works fine for natting with iconnecthere, but
as for asterisk, both my 7960 and polycom ip600 require you to set the
nat ip on the tftp.
Does anyone know a good phone (or ata) that can do this automatically?
For example,
I want to give a phone to my brother, who is going to europe. His ICH
2005 May 11
0
Fw: pinout for"standard"telephoneheadsetrequired.?
I saw these adapters on eBay. 2.5mm stereo jack to modular RJ-9 jack. I
think original site is http://www.ciscoheadsetadapter.com
Mike
>Nabeel,
> I am very interested in what you came up with for a 2.5mm to RJ-10
> adapter.
>
>I played with every combination I could think of but the best I was able to
>come up with had a echo of the far end voice back to the far end.
2005 Jan 11
1
Direct SIP calls to *
Hello.
I have my * server set up and working perfectly. I wanted to allows
calls to sip:nabeel@sip.myserver.net. In sip.conf, I have:
[general]
context=default
Also, in extensions.conf, I have:
[default]
exten => myname,1,Goto(internal,nabeel,1)
However, when I make a call using a "Direct Dial IP" account in X-Lite,
I get the following error in *:
Failed to authenticate
2004 Dec 22
1
Asterisk billing solution
Hello.
I am looking for a simple Asterisk billing solution. I expect about
50-100 users (a mix of IAX and SIP) through 3-5 outgoing providers (all
IAX).
I need something that can handle monthly fees and per call charges
(depending on destination, obviously), and should provide a web
interface for customers and administrators.
Something that can tie in to one of the existing management GUIs
2005 Jan 18
0
Out of 5 Grandstream BudgeTone 101 THREE are
Ronald,
Grandstream products have a one year warrantee. If you don't have any luck
with Pulver, contact us and we can probably get your phones exchanged.
Please don't assume that your experience with Grandstream is typical. We
sell a lot of these phones and the overwhelming majority of the purchasers
are very happy with their units. The quality has improved tremendously over
the last
2004 Dec 30
1
IAXy issues
Hello.
I picked up a couple of IAXy's for testing. Unfortunately, I read the
negative comments only after I bought 'em :(
Regardless, I provisioned one unit using my local Linux computer. Now,
I'm trying to set it up to provision using the remote * server whenever
it tries to register, but it seems I need to know the "service
identifier" for the specific device. I can't
2005 Jan 21
2
Can anyone recoment T1/PRI provider in SouthOntario?
> http://www.mixdown.ca/~andrew/dump/threaded_email.png is what
> a mailing list looks like to most people, and you can see why
> replying to a message, erasing its contents and starting an
> entirely new email about a different topic is frowned upon
> (yours is the highlighted message).
I know this is OT, but can you recommend an email program for Windows
that does something like
2004 Dec 18
5
Q about IAX (and IAXy)
This is somewhat related to my other query on the list regarding NAT
traversal.
I have heard many times that IAX is "NAT-transperant". I am unsure how
it accomplishes this.
I do know that SIP works like this: your SIP device send a request to
the SIP server (usually on port 5060) with whatever command. The SIP
server respends to your device's "apparent" IP and port (this