Displaying 16 results from an estimated 16 matches for "sipton".
Did you mean:
shipton
2003 Apr 02
1
FW: ipDialog Ethernet SIP Phone $199
...003 4:56 PM
To: clay@ctitec.com
Subject: ipDialog Ethernet SIP Phone $199
pad <http://us.st1.yimg.com/store1.yimg.com/Img/trans_1x1.gif>
<http://store1.yimg.com/I/sysconfig_1733_51893>
<http://store1.yimg.com/I/sysconfig_1733_14909> Click to enlarge$199.00
ipDialog SipTone Ethernet Phone
<http://store1.yimg.com/I/sysconfig_1733_0>
The ipDialog SipToneT Ethernet phone is a cost effective Linux based
VoIP telephone designed to provide high quality voice communications
over IP networks.
The best value that we have found.
Check out the specs on this phone be...
2004 Nov 18
3
SipTone II
Anybody used the above phone with asterisk
I have one working ok for calls, but having a problem with voice mail.
Using either the 'Voice mail function key' or dialing 88 (for my system)
just gets me to Call Terminated
Asterisk CLI shows the error message 'unable to get User name'
My Grandstream works ok, asking for User name, then Password
Any ideas ?
--
Clive
Email :
2004 Jul 28
0
SipTone 4 Sale...
Hey Folks,
I'm selling my SipTone on eBay... starting at $100, 17 hours left. It's been
modified (the firmware) so that you are able to telnet into it and possibly
(thanks to cross compiling) run your own software on it.
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=5711945656&ssPageName=STRK:MESE:IT...
2004 May 25
1
SipTone II and Choppy/Stuttering Audio
...m sure some
guru will be able to sort out for me in no time!!
When receiving or making a call about 60 seconds or so into the call we
develop choppy/stutter audio problems. It then seems to clear itself only to
return again, and so the pattern carries on! This has got me stumped!
Our equipment is SipTone II handsets, AVM C2 ISDN Card, Suse Linux 9 and we
are in the UK.
The SipTone II Firmware version is SipTone 1.2.0 rc Z_11
I have tried all codecs on the handset, i.e. g729, g711 ulaw and g711 alaw
(should I have altered something in * as well?)
In sip.conf we have: -
disallow=all
allow=alaw
a...
2004 Jul 07
0
IP Dialog Hangup problem
If receive a call on the IP Dialog SipTone II, and the other end hangs
up first, the siptone immediately enters into the congestion tone. If I
initiate the call from the siptone and the other end hangs up first,
same thing -- congestion.
The same thing happens if we make calls from the analog phones attached
to the Mediatrix 1102.
Th...
2003 Oct 15
4
SIP Telephone Quality/Price
Hi!
I am doing a research about the prices of SIP telephones. If someone can tell me
which one are the cheapest and have an acceptable quality... it will be very
kind.
Best Regards,
Mireia
2004 Apr 28
1
Call forwarding and Caller ID
...ave * prefix the CID External (so that I can
tell that it is a fresh call) or Internal (to tell me that it is a blind
forward). So how do I prefix the CID with say E for external and I for
Internal?
We are using an AVM C2 ISDN card and are in the UK (if any of that helps?)
Our phones are ipDialog SipTone II
Many thanks in advance and kind regards to all.
Nick
2004 Nov 21
4
UK available SIP phone?
Hi,
Anybody here from the UK using Asterisk at home?
I'm looking for a SIP phone which will work with Asterisk and
not leave me broke!
I got one of the Tecom ones from Solwise but it refuses to
login to Asterisk server for some reason. May have to send it back.
What are the other options please?
Thanks
Mike
2004 May 25
0
Question IAX and SIP bound to different IP's on the same * box
...contains malware (Trevor Peirce)
3. RE: Newbie extensions.conf I need to include [SMS] context. (Jay
Milk)
4. Re: Sip Registration Problem (Olle E. Johansson)
5. Using Ser and Asterisk together (=?iso-8859-1?q?Aiden=20Chew?=)
6. RE: 100 analog phones?? HOWTO? (tan@yointernet.com)
7. SipTone II and Choppy/Stuttering Audio (Nick Grindley)
8. RE: Meetme Options (new one) (Ben Merrills)
--__--__--
Message: 1
From: "Gary Ruddock" <garyruddock@hotmail.com>
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Newbie extensions.conf I need to include
[SMS] c...
2009 Sep 23
4
Error When Using Postgresql Schema With Realtime Sip
...an_sip.c: -REALTIME- loading
peer from database to memory. Name: stone. Peer objects: 8
[2009-09-23 11:10:57.3q] VERBOSE[10431] chan_sip.c: -- Registered
SIP 'stone' at 10.215.42.138 port 5060
[2009-09-23 11:10:57.3q] VERBOSE[10431] chan_sip.c: > Saved
useragent "ipDialog SipTone 1.2.0 rc Z_21 UA" for peer stone
[2009-09-23 11:10:57.3q] WARNING[10431] res_config_odbc.c: Key field
'ipaddr' does not exist in table 'foo.sip at asterisk'. Update will fail
[2009-09-23 11:10:57.3q] DEBUG[10431] res_config_odbc.c: Skip: 62; SQL:
UPDATE public.sip SET ipaddr=...
2004 Jun 18
0
SIP error 407 - can't make outgoing calls
I am using a IPDialog siptone II. I can take incoming calls, but when I try
and make an outgoing call I get a SIP 407 error.
Can some kind soul explain to me what I am doing wrong?
Here's what I found in the wiki:
If a proxy does not accept the credentials sent with a request, it SHOULD
return a 407 (Proxy Authentic...
2004 Aug 18
1
Choppiness/Ticking sounds over LAN
Skipped content of type multipart/alternative-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/gif
Size: 145 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040818/d0d64775/attachment.gif
2004 Aug 19
2
residential sip phone
Dear List,
Can anyone recommend a sip phone for residential use? (asterisk home pbx)
Thanks!!!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040819/7b107ebc/attachment.htm
2004 Jul 20
2
No Ringing.
Dear Asterisk Group.
I have two Asterisk servers serving two data/help desk centers, both centers
have a near identical setup.
However, when connected to one of my data centers, I call a user, I can see
on the CLI that the phone is ringing, but I hear no ringing on my SIP soft
phone?
Has anyone had a similar scenario? How as it resolved.
Warm Regards
Shad Mortazavi
2004 Jun 16
4
UIP200
Hi,
We've recently deployed 6 Uniden UIP200 phones (running firmware 4.54).
We've been having some serious problems:
1) All the phones randomly reboot themselves. Typically when trying to
answer or initiate a call.
2) All the phones will disconnect from a calls with the PSTN after 2-3
minutes.
3) The phones are unable to interact with a remote IVR (digit presses
are not received at
2003 Aug 12
12
IP phone recommendation
Hello,
I would like to buy a SIP IP phone, but I don't know wich one to
choose... Can you tell me wich IP Phone is known to work with Asterisk
please.
I've seen the Cisco 7940, but I don't know if it works, and how
expensive is it ?
I'm french, so if you know some french resellers, tell me.
Thanks a lot,
----------------------
Fabrice Tereszkiewicz
Sawadka.org