similar to: SJPhone, asterisk and DTMF

Can anyone recommend a client / phone that runs on Windows XP, with either a sound card or some other hardware? Ideally free, but does not have to be. Thanks...

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

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 ;

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:

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

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

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

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

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

I just installed the new version of SJPhone and it appears that it cannot work with * anymore? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040111/25c910bb/attachment.htm

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

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

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

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??????????????

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

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:

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:

I played around tonight for a while trying to place a call to the answering machine at FWD. It didn't work. I sniffed the connection, and it looks like asterisk sends out an invite. The gateway at FWD then sends an Invite back, and then asterisk responds with a "482 Loop Detected" error message. I have attached the output of a sip debug for this session. Contents of the

Just change the install prefix directory in asterisk makefile to something like /usr/local/asterisk, and then you will see all the files needed for asterisk to run. > -----Original Message----- > From: Steven Critchfield [mailto:critch at basesys.com] > Sent: Saturday, 1 March 2003 22:03 > To: asterisk-users at lists.digium.com > Subject: Re: [Asterisk-Users] How could I install