similar to: Anybody using X-Lite Softphone ? tryed to forwarda call to X-Lite....

Try the snom soft phone! http://snom.com CS > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Dave Chase > Sent: Saturday, February 26, 2005 12:31 PM > To: ich@mateo.ch; Asterisk Users Mailing List - > Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Anybody using

Mateo, Dialing the extension to your softphone is the same as any hardware extension. Exten => 1000,1,Dial,(SIP/1000,20,trf) pretty exten => 1000,2,Macro(vmessage,1000) exten => 1000,3,Hangup Change [mateo01] to [1000] in your sip and you will be saying that ext. 1000 is registered with the specifics you are using. Update the settings in your softphone to register the name and

Hey Guys Im trying to forward a call from the asterisk mainmenue to my second computer with X-Lite installed.. What I've done so far is this: Installed X-lite @my win PC.. X-Lite configuration: Menu | System Settings | SIP Proxy | default Display Name: mateo01 User Name & Authorization User: mateo01 Password: **** Domain/Realm: 192.168.1.** SIP Proxy: 192.168.1.** 192.168.1.**

We're having a strange problem with our receptionist. She runs an xpro softphone and we're using a queue to handle incoming calls. It seems nearly all of the calls that come in through the queue get dropped. At first we thought it might have been human error (clicking the wrong button in xpro or something) or that the person waiting in the queue just gave up and hungup, however it

Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk

Hi, I tested asterisk server with Xpro program, and all the function working well ( like 3 way calling, transfer.... ). But on the VOIP phone, I don't know press which key for 3 way calling function and transfer function... Can anybody teach me ? thanks

Hi everybody, I'm new to this matter and I spent three days in trying to connect one SIP Softphone to an Asterisk Box. I always get error 401 or 403... I don't understand very well settings in Softphone program: con anybody show me how to set up a minimal running system with no public lines or external Proxies? Thank you for your kind help... Ciao Mauro

Hi, I tested asterisk server with Xpro program, and all the function working well ( like 3 way calling, transfer.... ). But on the VOIP phone, I don't know press which key for 3 way calling function and transfer function... Can anybody teach me ? thanks

Hello all, I am trying to complete my dial plan and have come up with an interesting situation. My configuration is set up with 12 xlite SIP clients on SUSE linux workstation. They are calling out via 10 analog lines, TE110P->rhino 24 fxo. It all works and dials out great ... but ... this unit was brought in to handle the "global" office. So the help desk support on the Suse

Hey guys. Im trying to setup Music on Hold. If I transfer a call (with dial) I like to put the call on Music on hold.. Here's what I've tried so far: On my I extensions.conf exten =>1,1,WaitMusicOnHold(30) exten =>1,2,Dial(SIP/mateo,18) exten =>1,3,VoiceMail(1001) I have also added this line to [context].. So it looks like that: ;[context] musiconhold=default Additinaly,

Hello All, Anyone here has experience of accepting a ilbc call and sending it on g711 or g729 I am having problem in VOICE , call goes though but there is no voice. Senario: Call is coming in from Machine A to Machine B, sending to Machine C Machine B is an asterisk box, transcoding it from IBLC to G711 and g729. Problem: Voice is not appearing on the sip user sitting on machine A Already

-----Original Message----- I'm crazy here trying to make X-Pro use ONLY g729, and you're struggling to make it not to use it :)... Can you please indicate what's your config for X-Pro and sip.conf? sip.conf: [12345] type=user username=12345 secret=12345 nat=no host=dynamic reinvite=no canreinvite=no disallow=all allow=g729 allow=g729a allow=g723.1 allow=g726 allow=ulaw allow=alaw

The client is travelling much of the time. Is there some way that he can use Port 80 so that the firewalls that he is behind won't block the connection? Any other hints or suggestions are very welcome!

Am 20.04.15 um 06:31 schrieb akhilesh chand: > Hi Folks, > > I'm trying to register softphone(X-lite) but I'm not able to register > softphone whenever I'm trying to register softphone I got below error Do you have access to the asterisk server? If so, what's in the logs? cheers t.

Check if asterisk is running or not first . If asterisk is running check iptables ( firewall ) might be blocking the connection . You can see listening ports with netstat -upln command On Mon, Apr 20, 2015 at 10:01 AM, akhilesh chand <omakhileshchand at gmail.com> wrote: > Hi Folks, > > I'm trying to register softphone(X-lite) but I'm not able to register >

Am 20.04.15 um 09:43 schrieb akhilesh chand: > Hi Thomas, Hello. > Yes I'm able to access asterisk server but there is no logs capture into > log file related to softphone.If you want more information regarding > configuration means sip.conf and extension.conf I will share. Could you increase the verbose level? # core set verbose 6 # sip set debug on Looking for blocking

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 04/20/2015 12:31 PM, akhilesh chand wrote: > Hi Folks, > > I'm trying to register softphone(X-lite) but I'm not able to > register softphone whenever I'm trying to register softphone I got > below error > > Inline image 1 > > Is there any document/guide line where I will get process to > register

On Mon, Apr 20, 2015 at 1:58 AM, akhilesh chand <omakhileshchand at gmail.com> wrote: > Chain INPUT (policy ACCEPT) > target prot opt source destination > ACCEPT all -- 0.0.0.0/0 0.0.0.0/0 state > RELATED,ESTABLISHED > ACCEPT icmp -- 0.0.0.0/0 0.0.0.0/0 > ACCEPT all -- 0.0.0.0/0 0.0.0.0/0 >

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 04/21/2015 04:58 PM, akhilesh chand wrote: > Hi Guenther, > > What did you recommend to me, I did accordingly but there is no > log showing in asterisk CLI. I'm getting same problem. > > > > Regards Akhilesh Hi Akhilesh, looks like your firewall is blocking it. Have you tried 'nmap -p5060 <ip of your

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 04/24/2015 03:34 PM, akhilesh chand wrote: > Hi Thomas, > > Could you tell how can I change the protocol of corresponding port > means 5060 is configured with tcp protocol I want to configured > with udp. When I execute nmap -p5060 xx.xx.xx.xx I got below > output > > > [root at ip-172-31-32-117 cel]# nmap -p5060