Displaying 20 results from an estimated 10000 matches similar to: "Changes in SIP code in CVS?"
2003 Apr 22
0
Xten - Free windows SIP client
Same here Michael and the PocketPC version seems unaudible with any codec; early days trying that though.
Simon
-----Original Message-----
>From: "Michael Van Donselaar"<mvand@neb.rr.com>
>Sent: 22/04/03 04:10:24
>To: "asterisk-users@lists.digium.com"<asterisk-users@lists.digium.com>
>Subject: Re: [Asterisk-Users] Xten - Free windows SIP
2003 Feb 22
1
SJPhone, asterisk and DTMF
I'm currently using the SJPhone softphone with asterisk for remote SIP.
When I dial into the voicemail, and attempt to pass the extension, I
"hear" the sounds, but asterisk is not receiving any DTMF signals. If I
use the Estera softphone, asterisk does receive the DTMF signals.
Normally, I'd just say "Use the Estera" softphone to myself, but that's
not an option,
2003 Mar 09
0
ata186's in the UK
Friend of mine in the UK setting up an asterisk box is looking for an
ATA186 source in the UK. Anyone who knows of one, please email me
offlist.
--
William Walsh <william at wxw.org>
Jabber: william at wxw.biz
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
2003 Feb 24
4
Vonage
Ahh Mr Carbuyer ... you should have _specified_ you wanted tires with that new car
We can still help you though, it will just be an extra $$ above the price we quoted you
I understand the concept. I see it in many industries until a company comes along
that cares about it's customers
I still think that digium is the best buy (for the small scale stuff that I'm
interested in anyway) ...
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 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 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
2004 Jun 23
0
connecting to Iconnect here using asterisk
Hi,
I wish to connect several ATA186 Phones to each other, to iconnecthere and
to the PSTN using asterisk.
Please tell the appropriate settings for firewall (ports to open etc.)
sip.conf and extensions.conf(part relevant to iconnect).
Also I would be glad to get a working example of your ATA186 configuration.
I tried searching the mailing lists and several sites but did not find an
answer.
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 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
2005 Jan 25
0
lots of changes in cvs
Hi all,
I''ve been spending a lot of time over the past few weeks on RubyTorrent.
All the changes are in CVS. Highlights are:
- Transfer speeds should be much better, thanks to a better piece
selection algorithm, and being more aggressive about dropping boring
peers and adding new ones,
- Download/upload rate limits now work.
- Many bugs fixed.
- .torrent creation now
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 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 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 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
2003 Apr 17
4
Xten / SIP Phones compared to GnoPhones
I have seen a couple of messages on the Xten and the work done by William
Walsh (Kudos). It is not clear in my mind the advantages of SIP phones
versus using GnoPhones (once we complete the work for the Windows version).
Since I lack the experience with IP SIP phones, can someone, high level,
tell me when it makes sense to use them. Is it complicated to set up on the
Asterisk side?
Thank you.
2003 May 29
0
Would moving asterisk from behind NAT fix iconnecthere problems?
Hi All,
Outbound Iconnecthere calls work without any problem but Inbound
calls are very intermittent. It seemed to work for a week or so
but over the past week 99% of inbound calls are dropped to ICH
voicemail.
Would moving the Asterisk box to a public IP resolve the problem
or is it just an ICH/Asterisk problem?
I am registering against natrelay.deltathree.com. asterisk -vvvc
shows an
2004 Jun 01
0
Unsupported Media error from iConnectHere
I can't talk through iConnectHere. The connection gets made but as soon as
any sound is transmitted the call ends and the Asterisk console shows an
"Unsupported Media" error as follow:
Got SIP response 415 "Unsupported Media" back from 213.137.73.147
My only allowed codecs are alaw and ulaw. My sip.conf looks like:
[iconnect]
type=friend
secret=xxxx
username=yyyyyyy