Displaying 20 results from an estimated 4000 matches similar to: "SIP calls no longer work"
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 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 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
2003 Oct 03
4
Iconnect Incomming calls
I have an IconnectHere account with a Inbound number and have setup the sip.conf to register and am recieving the call but When I answer the call it disconnect. I have tried sending the call to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon as I accept the call it disconnects. I believe it may be some type of codec issue but I am not very familiar with that layer.
2003 May 09
4
SIP Confusion
Ok. I am confused. I now have conflicting answers to my question:
Do you need to use a special phone to use SIP? My setup is
X100P and TDM10B.
I would like to connect to iConnectHere, which uses SIP. Has anybody
done this before (using similar equipment to what I have listed above)?
And if it is not possible, could somebody please explain why. I don't
understand
why this wouldn't
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
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 May 15
0
CallerID through iconnecthere not working
I can't get the callerid feature to work when being passed through
iconnecthere.
Is it even possible to specify your own callerid using iconnecthere?
-sip.conf-
...
[iconnect]
type=peer
username=xxxxxxxx
password=xxxx
callerid="Jerky McJerkface" <(555) 867 5309>
host=213.137.73.178
-extensions.conf-
....
exten=>_1NXXNXXXXXX,1,SetCallerId,4168675309
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 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
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 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 Nov 28
2
Deltathree icomming problem
Hi,
I have a deltathree account and I can place calls but I can't receive calls. I use Grandstram sip phones. When I call my deltathree phone # the voicemail is answer :((
I need some help and solutions from the guys who allready are using deltathree. I search on Internet and I try all types of configurations... :(
This is my configurations files:
- sip.conf -
[general]
port = 5060
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 May 24
1
Limiting number of channels or calls
Good afternoon all,
I was wondering if anybody knows of a way to limit the number of calls going
out over an interface (or respond with some sort of 'circuits-busy'
message?)
The reason I ask is my outgoing bandwidth is only 128kbit and if there are
any more than 2 calls going over the internet interface the QoS is reduced
dramatically for all calls.
Failing that, does anybody know if
2003 May 30
2
SIP echo?
I noticed a few other messages posted about this problem, but I couldn't find an answer...
I'm having a problem with SIP echo when calls are received into asterisk via an x100p and bridged with a sip extension (back to the pstn with iconnecthere). the person calling in to asterisk has no echo problems, but the recipient of the pstn call, everything they say, they hear back about 1 second
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 06
1
Behind Firewalls, SonicWalls, etc..
I've searched the archives a bit and have not really come up with
a good answer to my queries.
I have * running on a RH9 box behind a LinkSys NAT box. I can talk
with iConnectHere outbound just fine. I am trying to configure an
inbound Xten softphone from outside. I have that user set as NAT in
sip.conf (seems to help), but I still cannot establish a full session.
I think the problem comes