similar to: Sjphone Troubles :

Displaying 20 results from an estimated 10000 matches similar to: "Sjphone Troubles :"

2004 Aug 07
2
Asterisk : No Sound Issues
Hi , Thanks greg , for pointing out the valuable resources for reference. I tried SJphone in a windows environment to connect to fwd and it worked fine(including (audio). Now have to do the same thing for linux(red hat 9 ) and hope the nat issue is resolved. Now i would like to connect asterisk to fwd and instead of the SJ phone connecting to fwd directly i would wish to connect through
2004 Aug 07
2
Asterisk : No Sound No Dial
Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",
2004 Aug 06
1
Asterisk Dry Run
Hi everyone, I just installed asterisk on my system with the purpose of rerouting calls on sip channels. I don't think i need any hardware for that. I am using LIPZ4(zultys) and sjphone as softphones. I tried setting up both of them and to call one from the other on the same machine, however could not. I 1-) I could connect sjphone in isolation to freeworld dialup howver i got no sounds
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
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
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:
2004 Aug 10
0
Sound problems ..linux red hat
Hi, I have 2 sound cards on my system a SIS controller with an i_810 module ( generic rockwell itu chipset ) and another recognized as cirrus log ( cs466xx module ) ..this is creative soundblaster 128 pci ... i used sndconfig to configure them and the sound plays on both...cirrus logic is the primary device presently ( red hat fedora core ).. The other apps using sound work well but no
2004 Feb 08
1
Registering SJPhone with Asterisk
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 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 ;
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 Aug 08
1
No Sound and Jungle:
Hi everyone, I am running asterisk on red hat linux 9 box. The sound card is Intel 82801db AC' 97 audio and the module is i810_audio. It runs well with other applications like xmms and the standard tests deliver a sound . I have also tried to record voice and that works well too. 1-)Now when i run asterisk and i dial out an extension to play any sound there is none. The same thing
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
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
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
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.
2005 Jul 08
0
Agent Silent Call Issue (seems like an asterisk bug / SjPhone Bug)
I have an issue with silent calls when an agent gets a call from the queue What happens is - The system dials a call (agent call) - The caller picks up - Asterisk sees the person picked up - Transfered to an agent - Agents phone automaticly picks up (sjphone auto accept on) -The user hears nothing says "Hello, Hello, Hello ???" - Asterisk sees agent as 'Available'
2004 Aug 09
5
Questionaire :
Hi, I have read quitea bit of the available resources and have this idea of asterisk. Would someone kindly answer these briefly 1-) Asterisk does not need a sound card...but if i am to record voice into an extension or dial from CLI ( basically use asterisk itself as a softphone ) then i need a sound card. : Yes/No a-) If yes creative soundblaster pci 128 is my best bet. Yes/No 2-) Which is
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