Displaying 20 results from an estimated 20000 matches similar to: "iConnectHere broken?"
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
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
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
2003 Mar 03
3
iconnecthere 480 error: is there a workaround?
I am going to have to find a fix for this problem or I'm going to have
to quit using iconnect.
About one call in 10 or so, iconnect's gateway gives me an error
(console output appended below).
So upon receiving the error, which as a 4XX error means, "Fatal,"
asterisk gives up and drops the call. But not iconnect!! The phone at
the other end starts ringing, and rings
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 =>
2003 Apr 20
1
iconnecthere bridging broken on recent CVS?
Trying to figure out what's going on, CVS ident CVS-04/20/03-01:34:54.
I get frequent errors such as this one, which showed up on the CLI
interface within a couple of seconds of a cold start:
WARNING[114696]: File chan_sip.c, Line 393 (retrans_pkt): Maximum
retries exceeded on call 73015f757661435d247414b104964554@192.168.1.10
for seqno 102 (Request)
All calls to iconnecthere terminate
2004 Jul 22
2
NAT + iConnectHere Broken in 1.0RC1
I've been using * CVS code from May of this year and was able to connect
to iConnectHere and receive calls with * being behind NAT. Now that
I've upgraded to 1.0 RC1, this no longer works.
I've tried setting nat=yes in places, externip, et al with no success ..
even though the code I was running from back then worked without that.
Any suggestions?
BTW, I've gotten DTMF from
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 Apr 23
3
Anyone else lose iconnecthere service in recent CVS?
For the past several days I can no longer use iconnecthere with
asterisk. It is broken in BOTH directions; I can neither make nor
receive calls.
On outbound calls I get an immediate error:
-- Got SIP response 400 "Bad or Missing To" back from 213.137.73.140
On incoming calls, the call switches through OK, and for a few seconds I
get audio in both directions, although much
2003 May 11
3
Sound Quality
Hi All,
I've just setup a test Asterisk system that allows incoming/outgoing calls via
an ISDN card (l4i) and incoming/outgoing calls via SIP (iconnecthere). I have
two SIP Softphones (Xten X-Lite) for making and receiving calls.
When receiving an incoming call via the ISDN interface the sound quality is
fine for the Softphone user (i can hear the caller perfectly), but the person
2004 May 02
4
iconnecthere behind NAT, strange deal
I've been to the WIKI and I've searched the archives.
Is anyone on the list successfully using iconnecthere behind NAT?
I was, for over a year, and then I changed my "plan" with them. Now all
my calls get intercepted immediately, "We're sorry, but your account is
temporarily unavailable."
Incoming calls work just fine.
I contacted their so-called
2003 Mar 06
1
More problems with iconnecthere
This may be slight off topic, but perhaps it has relevance:
My iconnecthere account no longer works for "inbound" calls through
NAT using the standard configuration that they provide on their
website. I have sent them a message, but I believe it will be
flushed down the toilet by the first-tier support people.
When I call my iconnect number, it goes directly to voicemail. There
2004 Jun 10
4
incoming DTMF on iConnectHere?
Hi,
Anyone having problems receiving DTMF on incoming iConnectHere
lines? They disappeared for us sometime in the last 12 hours...
And, yes, we've restarted * and rebooted our * machine.
Michael Swan
Neon Software, Inc.
2003 Aug 10
3
Registering SIP with FWD and ICONNECTHERE
Hi! I am new to Asterisk too, I got the similar problem and I would like to
know how to get * to work behind NAT.
When I have the SIP Debug turn on, I got the error 479 from FWD when * try
to register with FWD, it looks like * is using the local IP (192.168.x.x) in
the Contact field. I have put the nat=yes in the [FWD.Pulver.com] content,
but it does not seems to make Asterisk aware the
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 Jul 08
5
Using multiple iconnecthere accounts
Has anybody out there tried to use two different iconnecthere accounts
with Asterisk?
What I want to do is use a second account if the first is busy.
I have tried the following:
exten=>_91NXXNXXXXXX,1,StripMSD,1
exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the
first account
exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is
the second account
But that
2003 Mar 06
2
SIP INVITEs borked with iconnecthere
Symptoms: when calling my iconnect phone number (13033913323 in my
bogus example below) from my cell phone, I can see that the call
makes it to my asterisk server, and my phones even ring once as *
passes the call through during the "180 Ringing" period. However, it
seems that iconnecthere.com cannot see my "100 Trying" and "180
Ringing" messages, as they
2008 Feb 21
3
Voted most stable and easy to use phone?
A while back i had asked about possible replacements for snom 360 phones
that were breaking and causing issues and we all discussed the problems
we had with the 360s and some suggestions were made but the new polycom
phones had just hit the market and not many people were able to comment
on them.
Basically i am looking to get some new phones and in the process get rid
of the countless number of
2010 Mar 30
2
Dropped Calls
I've written about this issue several times, but have not yet found any
solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones
are primarily Snom 300's but I also have a couple of headset phones
connected to Grandstream HT286 SIP adapters. I have 8 offices, each has
it's own asterisk server all running the same versions of asterisk and
Zaptel. Only difference
2004 Jun 24
1
Cisco ATA 186 from iconnecthere, locked?
I wanted to sign up for the pay as you go plan from iconnect
anyway, and see they have the Cisco ATA for $99 and the Grandstream
phone for $39.00
Anyone know if they ship these devices "locked"? I know iconnect
seems pretty friendly about letting any sip device connect.
What sucks is there is no way to contact this company if you're not a
subscriber.. Zip, notta.. No email