Displaying 20 results from an estimated 7000 matches similar to: "Attempting to get SJPhone configured for Asterisk- Help!"
2004 Jun 17
3
SJphone regestration problem - Help!
I am having a problem with SJphone registration, having read the list
and wathced it for a while for similar problems. I just can't seem to
figure out the problem.
I tryed to follow a tutorial from
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone,
but in SJphone (SIP tab), I can't find the following setting.
Use local outbound proxy - checked.
Proxy IP Address:
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 Jun 22
3
asteisk, sip & NAT
hi
My stations are behinds a firewall, the system is windows 2000 & 98, i
use sjphone
asterisk is on the internet gateway where is the firewall Shorewall the
system is linux debian (sid) kernel 2.4.20
j do whaton http://www.automated.it/guidetoasterisk.htm (grateful Andy)
to write my sip.conf but i can't call an external sip user. (an external
user can call me)
i try without asterisk with
2003 Dec 16
2
DIAX-SJPHONE REGISTRATION PROBLEM
I am having a problem with softphone registration, having read the list and watched it for a while for similar problems I just cant seem to figure out the problem. Using SJPHONE or DIAX , I can make outgoing calls but I can't get them to register with asterisk, I have other sip devices registering OK-7940's. From the list and the digium web site this seems to be a straight forward set up
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 Jul 01
2
Today's Message from linphone; update on Khpone and SJPhone and X-Lite
Today's "frustrated programmer" award goes to Linphone, which has the
following debug output:
> (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer!
I get this message when I connect to linphone using a softphone, or when
I try to use linphone to connect to asterisk and listen to an
announcement. I suspect that
2004 Feb 08
1
Registering SJPhone with Asterisk
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 ;
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
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
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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2008 Apr 04
2
SJphone behind NAT/Firewall without sound
Hi.
I need connect some LAN stations with SJphone to an Asterisk Server
published on Internet.
My Lan Clients access to Internet using a small linux firewall/proxy
server. I use the next firewall script. That is a simple script with
default policy ACCEPT, and NAT to share Internet. I can connect to
the asterisk server, authtenticate the users in the server, and dial
to any extension, but
2005 Mar 06
3
SJphone on PDA registering with Asterisk???
I try to setup SJphone on my PDA, but it does not register with Asterisk.
I have setup a sip account on asterisk, ...
Can anybody give me a hint?
bye
Ronald
2003 Jun 20
1
[HS] results testing asterisk with ISDN BRI & look for help to understand configuring SIP with asterisk
configuration
ISDN BRI
card : ISDN Olitec PCI 128 (hisax gazel)
internet connection by ISDN 64kb/s
dynamic IP
nom de domaine registered to : dyndns.org avec ddclient to register IP
par ddclient
asterisk (on internet gateway)
configuration pour ISDN BRI par virtual modems /dev/ttyI* (modem.conf)
logical telephone SIP "SJPHONE" on 2 local stations "windows"
(i don't succeed
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??????????????
2006 Mar 29
1
SJphone Do not send silence - option ? Should be disabled for Asterisk
Hi all,
I would like to hear from you, SjPhone has the option to Do not Send
silence (audio options, advanced), should i use this or remove this
option?
Everything ran well until now, but there was few people on my server,
i'm increasing sip extensions and i want to avoid complains from the
users:)
Best regards,
Marco Mouta
2006 Apr 28
1
Warning: No path to translate with SJPhone
Hi list!
I'm making tests for Asterisk. I've tested with 2 users installing SJphone
and it worked fine, but when I install it over a third user with the
softphone, the phone dial for 2 seconds and a window alert goes out on the
softphone:
Busy
Call rejected: 486 Busy Here
And on my Asterisk server this message:
Apr 28 09:05:37 WARNING[8140]: channel.c:2685 ast_channel_make_compatible:
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:
2004 Jul 23
3
Grandstream Budgetone 101 channels don't disappear on hangup.
Hi there,
I'm having problems with the Grandstream Budgetone 101 on hangup -
"show channels"/"show channels concise" output is still showing the
call's channels as active.
The problem does not exist when I use SJPhone, so I'm assuming it isn't
an Asterisk configuration issue. Has anyone seen this, or better, does
anyone have a fix? :)
Thanks,
David.
--
2004 Jun 25
2
Asterisk & SIP
Good morning all,
I'm setting up Asterisk for the first time with no prior PBX experience.
I'm following Andy Powell's 'Getting Started with Asterisk'
(http://www.automated.it/guidetoasterisk.htm). This is my second time
through that document - as I did something weird the first time and really
upset it somehow - and I wanted to ask a few general questions of the list.