Displaying 20 results from an estimated 3000 matches similar to: "D-link DPH-80 (SIP) call to asterisk problem"
2004 Jul 22
6
D-Link DPH-80S vs *
List,
The D'Link phones are not reliable at this time. I am trying to get them fix their Firmware to my specifications. It is half done so far. However there are still hurdles. below email is self explanatory. At present if you want to use these phones, you need to buy D'Link's SIP Server and run this as one of your SIP servers in the blend to call to Asterisk.
Seshu Kanuri
"G
2005 Jan 11
1
Dlink DPH-80 DONT work with asterisk
After 20 cups of strong coffee and wasteing most of tonight and obviously
doing lots of googling and emailing many people, i've concluded that dlink
voip phones specifically DPH-80 dosent work with asterisk.
If anyone has had anymore luck with these phones then me please let me know
--
regards
Vikram
2005 Oct 04
2
DPH-140S SIP Phone oddities
Hi, list!
I'm playing on an Asterisk@home installation, since a month or two.
I've had no trouble setting it up 'n running.
I've bought a couple of DLINK's DPH-140S SIP Phones, to use with Asterisk.
>From this phones, I can make & receive calls with no trouble, but, when I
try to use some "interactive" function (eg Directory or Voicemail), the
phone seems
2005 Sep 02
1
Dlink dph-140s/ACT P104SLD
I'm still a learning but I purchased a dph-140s to test with AAH 1.5. I
think this is a rebadged ACT P104SLD which others seem to have working with
*. It seems to be configured and registered similarly to the softphones I've
been using just fine, but it does not receive or send audio (it will send
audio to vm), or perform the loopback test. It seems to signal and receive
calls fine. The
2003 Sep 30
0
Missing ring indications with DPH-100H/chan_h323
hi all
after setting up chan_h323, I don't get any ring indications on my Dlink
DPH-100H phone. Any idea how to debug this?
roy
2009 Jun 20
3
Usb drivers and linux apps
Hello
Situation: I'm moving from a windows XP box to fedora 11. I used winXP to run skype with a dlink dph-50u USB adaptor so that I could talk on skype with my regular phone. Been working fine for months
I'm now on fedora 11, able to run skype linux without problem, even the webcam work! Problem, cannot make the dlink dph-50u to work. After a lot of googling, cannot find anything.
I
2003 May 25
1
iconnecthere problem 481 "Call Leg/Transaction Does Not Exist"
Hi All,
I am trying to use iconnecthere to make outbound calls. I am behind a
linksys router. I keep getting this error
481 "Call Leg/Transaction Does Not Exist". Does anyone have any prior
experience with this problem. Any leads will be much appreciated. Attached
are the conf files and logs
#SIP.CONF
; SIP Configuration for Asterisk
[general]
port = 5060 ; Port
2003 Sep 18
2
Need help with H.323
hi all
I'm trying to setup a dlink dph-100h phone (actually a dph-100m but with
the h.323 software) with asterisk and chan_h323. AFACS, the dph-100h
software can only be configured to point to a gatekeeper. I know I don't
need to do this, but it's a test before I setup my Symbol Netvision
phones (I don't have an access point for them now). First, this is what
I have, and what
2003 May 23
3
iConnectHere - calls dropping out?
Hi all,
This is my first post here - I started with Asterisk a few days ago and have
"fallen in love" - fantastic product. I've only got softphones connected at
the moment - I'll probably order the FXO/FXS cards in about a month (and
then think about getting some hardware SIP phones). Our current phone system
is quite a few years old and isn't growing with us (when a single
2007 Jan 29
3
Pickup() ringing extension and call waiting
Hi All,
I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have
Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would
like to be able to pickup ringing extention from any SIP phone using Pickup()
application.
from my dial plan:
[incoming]
exten => s,1,Dial(SIP/somebody1|60|tTrR)
[internal]
include => outbound-local
include => parkedcalls
2004 Apr 28
2
Asterisk and Iconnecthere pause
Hi, I just got a SPA-2000 in and was finally able to complete my asterisk
setup. I'm making my outgoing calls through iconnecthere from the
asterisk server however I'm running into a problem when placing calls. I
can connect fine but when the person (or answering machine) picks up I
hear them talk for a about half a second then there is a half a second
pause or muted period and then the
2005 Jan 14
1
iconecthere and *
Hi all
I am trying to figuure out how to get iconnecthere incoming calls to work
outbound works fine but incoming goes nowhere but to my iconnecthere vocemail
if I do a sip show registry it shows up as regg'ed
nnn=is my iconnect here number
xxx is my secret
Thank you
Jeremy
[general]
qualify=no
register=NNNNNNNNNNN:XXXX@iconnecthere/NNNNNNNNN
context=default
bind = 0.0.0.0
port=5060
2004 Aug 05
1
iConnectHere and CallerId
Is it possible to send the CallerId to IconnectHere with Asterisk
when making outbound calls?
I read somewhere that it doesn't work.
I have set up everything to send the correct CallerId info to IconnectHere
but I get a "442-887-926267" caller id.
In [globals]
ICONNECT1=1713...(my number)
MYNAME=My Name
I set up the Caller Id in the dialing macro:
[macro-iconnecthere]
exten =>
2004 May 31
1
Failover: iconnecthere to voicepulse
Hi all,
I'm working on a setup for a small office. I'd like to use SIP/iconnecthere
most of the time, because they're cheap. But they only allow a single call.
When the single iconnecthere line is in use, I'd like to use IAX2/voicepulse
instead:
exten => _1NXXNXXXXXX,1,Dial,SIP/BYEXTENSION@iconnect
exten => _1NXXNXXXXXX,2,Dial,IAX2/npI42VoD38@voicepulse/${EXTEN}
Well,
2003 May 27
1
Incoming calls using iconnecthere
Hi All,
I can only seem to get iconnecthere working with incoming calls
intermittently. One minute it seems to work, and the next it doesn't.
I am not aware of anything being changed in the config files. Outgoing
calls work ok all the time.
The Asterisk box is behind NAT so that does complicate things slightly.
However, the Iconnecthere PCPhone client software works perfectly for
2003 Nov 17
1
iconnecthere incoming
Hi guys
I just registered an incoming number with iconnecthere and I'm trying to
set up incoming calls from icconnecthere on my asterisk server. I took a
look at john todds sample sip.conf and extensions.conf file but for some
reason my incoming is still not working. At this point I wish to use
iconnecthere merely for inbound calls. Also my asterisk server is behind
nat. The following
2008 Feb 05
3
wireless VOIP phone recommendations?
I have been using the D-Link DPH-540 wireless VOIP handset, and I really
like this phone. We had tried the UStarcomm phone, but the phone is used in
a noisy environment and the volume wasn't loud enough. The "problem" with
the D-Link phone is the Li-ion battery needs to be replaced and D-Link
doesn't sell a replacement battery and I haven't found any after-market
batteries.
2005 Sep 04
1
FW: Asterisk@home - requesting help on oh323, ISDN BRI and iConnectHere DID
I know almost nothing linux, and don't really know that much about Asterisk
(proper).. but I was 'pulled' by this subject and grabbed an
<mailto:Asterisk@home> Asterisk@home installation CD (version 1.3) and just
went with it. Newbie doesn't quite describe it, I'm a banker.. this simply
goes to show how easy Asterisk is becoming (I certainly hope this direction
was meant
2004 Sep 05
0
iconnect and Asterisk
Hello All,
I have gone thru all the resources I could find on google on asterisk + iconnect and managed to get outgoing calls working. However,
I cannot get incoming calls to work at all. With the sip debug on, I can see that something is happening everytime a call is received
from iconnecthere, but I get an invalid tone on the caller side. The call never rings anywhere on the asterisk. Would
2003 Nov 17
1
SIP calls no longer work
Hello,
I'm having a problem with SIP. More specifically, I
can't make any calls using SIP.
I have had an iConnectHere account and Free World
Dialup account working for quite some time, and now
all of a sudden I can't make any SIP outgoing calls.
PBX*CLI> sip show registry
Host Username Refresh State
192.246.69.223:5060 XXXXX 120 Registered