Displaying 20 results from an estimated 100 matches similar to: "Registering SJPhone with Asterisk"
2003 Nov 18
2
ask problem about softphone--asterisk--softphone, Urgent!!!
Hi, all,
I want to use asterisk SIP as a proxy, and two softphone (Ubiquity SIP Phone) as user agent, like below:
Softphone1<-------------->Asterisk SIP<------------>Softphone2
(User Agent) (Proxy) (User Agent)
155.69.xx.xx 155.69.yy.yy 155.69.zz.zz
zhou mysipproxy.com
2003 Dec 13
2
voice mail - sip:notify message
Hi folks,
To provide MWI, * will send out a sip:notify message to the UA.
The originator of this message is asterisk, as shown below;
NOTIFY sip:1001@www.mysipproxy.com:5065 SIP/2.0
Via: SIP/2.0/UDP 66.121.xxx.yyy:5060;branch=z9hG4bK0466cb21
From: "asterisk" <sip:asterisk@66.121.xxx.yyy>;tag=as0ffb1bdc
<===============
To: <sip:1001@www.mysipproxy.com:5065>
Contact:
2003 Sep 13
2
SJphone DTMF?
Hi. I have sjphone installed on windows and working
except for dtmf. I read the docs for sjphone and it
uses inband dtmf. I configired dtmfmode=inband but it
still does not recognize it. Someone on the lists
said that inband only works using alaw or ulaw but i
tried only allowing that too but still no go. Hmm..
any other ideas? I can't get any other client to work
on windows :-/
I
2003 Aug 26
0
TDM10M && Siemens Euroset 2015
Hi all,
--------
I have installed a TDM400 with one active FXS port (TDM10B) an connected
it to a Siemens Euroset 2015 analogue phone.
I have installed some smom IP phones to the network as well and
configured them as usual (sip.conf). For configuring the TDM10B I have
used FXO signalling in /etc/zaptel.conf and in
/etc/asterisk/zapata.conf. I definded the TDM channel and the Snom
phones to the
2007 Jul 12
0
No subject
CLI>realtime mysql status
Connected to asterisk at 127.0.0.1, port 3306 with username askuser for 1
minutes, 34 seconds.
Thank you very much for your kind attentino. You help is greatly
appreciated.
Thanks,
Mark
------=_NextPart_000_05AD_01C88E72.41751FF0
Content-Type: text/html;
charset="us-ascii"
Content-Transfer-Encoding: quoted-printable
<!DOCTYPE HTML PUBLIC
2009 Jul 20
0
No subject
[2010-03-08 14:03:49 CST] WARNING[26890] loader.c: Error loading module
'chan_dahdi.so': libpri.so.1.4: cannot open shared object file: No such
file or directory
[2010-03-08 14:03:49 CST] WARNING[26890] loader.c: Module
'chan_dahdi.so' could not be loaded.
=20
I am using on CentOS 5.4 64 bit.
Asterisk 1.6.0.25
Asterisk-addons
2006 Mar 30
1
Sjphone looses registry in my Asterisk Server then i need to restart Sjphone PC, why?
Hi all,
I've my Server running well, then sometimes Sjphones looses registry
and it only works well again if i restart the pc running sjphone.
Has any one experience this?
Best regards,
Marco Mouta
2011 Apr 12
0
No subject
r>
<h2>Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010=
)
</h2>With SIP 3.2.X firmware (available on the Polycom download site)=20
and Asterisk 1.6.1, Polycom phones now support a full featured BLF=20
showing statuses of Ringing, Inuse and Online and one touch directed=20
call pickup.
<br>On the asterisk side all that needs to be done is to add a hint
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 Sep 25
0
SJPhone and Asterisk
--- "Keith O'Brien" <keith@voipreviews.com> wrote:
[phone1]
type=friend
username=keith
secret=keith
host=dynamic
qualify=2000
disallow=g729
auth=md5
context=sip
mailbox=9999
callerid="keith@10.1.1.12" <1000>
But the log in SJPhone indicates that the registration is being rejected:
2003-09-25 18:55:34.776 UDP LOCAL->10.1.1.12:5060
REGISTER sip:10.1.1.12
2003 Nov 26
1
Attempting to get SJPhone configured for Asterisk- Help!
I recently setup an Asterisk Server-
I was able to follow a tutorial from http://www.automated.it/guidetoasterisk.htm#_Toc49248752
Until it told me to call another line, let it ring until voice mail picks up.
My problem is the tutorial left out how to configure a SJPhone so that it connects to my asterisk server not directly FWD. I've tried everything I can think of, I must be missing
2003 Dec 21
1
SJphone, Asterisk and DTMF tones ...
Hi,
I am using SJPhone here for testing ivr with Asterisk. I haven't seen any
problem here yet.
I have tried different things for that and finally got it working. I am not
an expert to explain more about that, but here is the general section form
my sip.conf. dont know whether it will help...
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ;
2004 Jan 11
1
New Version of SJPhone
I just installed the new version of SJPhone and it appears that it cannot work with * anymore?
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2004 Jun 11
0
Newbie to SJphone
hi guys,
I installed the SJphone vision 222b on Linux. When I try to dial a
number, SJphone just say "Can not dial phone number in current service
configuration". :( In the options dialog window, I can't see anything is
related to that setting.
Could you tell me how to set the configuration.
Thanks a lot.
2004 Jun 17
0
Re: SJphone registration problem - Help!
Rui,
>Yes, it's just as you said, when I create a new profile, it opens a
>window with the fields(Proxy Domain,Account,Password,CallerID). but it
>doesn't let me to input value to these fields. I don't what's the matter.
Edit the profile, and on the "Initialization" tab and make sure the "Inquired"
box is checked by the fields you listed above.
2004 Aug 10
0
Sjphone Troubles :
Hi,
here is something that is bugging me for some time now...any pointers would
be great.
I am running linux on 1 pc 192.168.x.x and my softphone (Sjphone ) can
connect to it from 192.168.x.y without a problem on port 5060.
However when i run a softphone on the same linux box where i run asterisk it
does not register. I tried even by specifying the host and port in sip.conf
and using the same
2004 Dec 04
2
SJPhone SIP Tab
Hi,
I'm following, http://www.voip-info.org/wiki-Asterisk+phone+sjphone.
However, I cannot find the SIP tab. Would someone please give me a few
pointers? The screen capture can be seen at URL below
http://www.dslreports.com/forum/remark,12022987~mode=flat
Regards,
Norman Zhang
2005 Jan 04
1
Newb howto request: *, Voice Pulse Connect, & SJPhone
I have been picking at Asterisk for about a week, and I think I'm close. I
was hoping for a little guidance to bring this on home.
I want to be able to make outgoing calls from my SJPhone clients using my
VoicePulse Connect account. I have the two requisite items from Voice Pulse,
but I've had no luck successfully integrating the VoicePulse settings into
iax.conf.
My current config:
2005 Feb 23
0
Cant connect to sjphone
Guys.. this is killing me.. I hava a laptop running sjphone and I have 2
dial cmds to connect to that laptop in different places, first, on the main
phones context like this:
exten => 202,1,Dial(SIP/laptop,20,m)
so each phone can call it, and it works great.
Now, I have anyother cmd on a different context, which is the IVR, like
this:
exten =>
2005 Jun 10
1
Request OPTION and 404 Sjphone Xlite
Hi,
I have install asterisk and it works fine.
But when I use Sjphone and I use Ethereal a Client send "Request:OPTIONS
sip:obelix.foo" and Server answer "Status: 404 Not found".
But i can talk with two client and asterisk.
When I use Xlite i don't have this request it's clean.
I don't understand??????????????