similar to: SJPhone and Asterisk

Hello, I want to buy Grundstream sip phones. Could someone tell me if thay are working with Asterisk ? I want to buy also codec G729. How many channels I should buy if I have 24 ext. Let's say if I buy 20 and there is 20 calls the other calls with use another codec then ? Reagds, -- Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi folks I have been trying to install e-Sword, a free bible software developed with Visual Basic. The app can be found at www.e-sword.net. The installer gives me a 995 Internal Error msg. Attached is a log of the install with -debugmsg +loaddll switches. thanks Philip _____________________________________________________________ The Ng Family Homesite ==> http://www.ng.fm

Hi guys I downloaded and compiled the latest Wine verson and I got the following error msg from the Windoes Installer. The installer wizard fired up properly, asked a few questions and started to copy the application out when it returned an internal error 2352. Below is from the msievent.log. My .wine/config is the standard conf except for minor adjustment to /mnt/floppy and /mnt/cdrom.

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

I am running Arch Linux. Very updated version. When I try to connect to remote servers using OpenSSH I get a hang as show here: $ ssh -v compsci at 10.1.1.12 OpenSSH_7.1p1, OpenSSL 1.0.2d 9 Jul 2015 debug1: Reading configuration data /etc/ssh/ssh_config debug1: Connecting to 10.1.1.12 [10.1.1.12] port 22. debug1: Connection established. debug1: identity file /home/carloabelli/.ssh/id_rsa type 1

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,

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

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 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 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.

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.

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

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 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:

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 =>

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

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'

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 all, I've around 10 people on my network getting this error, Critical transaction failed: Client non-INVITE transaction[trying]: Time out I'm using Asterisk 1.2.5 and Sjphone. Any tips??? 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: