similar to: Softphone's

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040728/3306e606/attachment.htm -------------- next part -------------- ? Hi, I'm dialling 1234 in the softphone or Grandstream Phone and without disconnecting the phone i want to dial "10" after dialling "1234". Is it possible to do? Regards shan

Hi all, I'm caught up in a struggle between people how can not make up their mind... Half way implementing a asterisk farm and they come up with another feature they've seen in kamaillo. What he showed me was this: three registered sip users, a) one changes his presence status on his softphone, and all see the status change. b) one calls another, and the third person see the status of

Hi all, I installed the estara softphone and had no problem registering it with asterisk. I could make calls to other hardware SIP phones (Cisco 7960) from the softphone, but I couldn't call the softphone from the Cisco 7960s. The asterisk console gave me an error message saying "unable to create channel" to my softphone. What could be the problem? I searched the archive with no

Hi Friends, Thank you for your quick response. I have successfully implemented Intercom (Dialling within my office LAN) using Asterisk. To implement this, I am using X-Lite Softphone. Now, I want to make calls to US using VoIP Asterisk. I think that there is no need of any external hardware to implement pure VoIP solution. Am I right? I have registered with Vebtel (VoIP IP Telephony

Hi, In the early stages of deciding how to try and develop this environment, I looked at all the protocols that could be used. SIP was chosen just because it seemed to me that it was the most widely used protocol. I believe IAX is a new protocol with a little less documentation and examples. The good thing about this Jain-sip-phone is that it saves a lot of time since many of the important

Hi, I am Chandramouli from India. I have successfully implemented Intercom (Dialling within my office LAN) and Voicemail features using Asterisk. To implement this, I am using X-Lite Softphone. Now, I wish to make calls to US using VoIP Asterisk. I dont think I need any external hardware to implement pure VoIP solution. Here I am sending my configuration file values: Contents of

All, I've fixed my problem getting Asterisk ooh323 channel to stay registered with my Cisco IOPS gatekeeper, now I need to get dialling working. I have the following: [Asterisk with ooh323] ----h323---- [Cisco IOS GK] ----h323---- [Radio system OpenH323] 192.168.1.5 192.168.1.6 192.168.1.7 the Asterisk box has numbers

Hi all, I have the Asterisk set up and 2 softphone (Xlite) set up on two other PC. With the following configuration, I can use one softphone (2000) to call the other one (2001) and/or the voicemail at 2999. Here is my problem: 1. When I dial 9+nxxx-xxxx with one of the softphone to the PSTN via X100P card, I got busy tone. (i.e. I want to use the phone line which is connected to the

I have the very same situation in one of my networks. To solve this you can dial out from the softphone and to move call to the phone you can simply transfer call to the same user (just if you were transferring call to yourself and the other device will ring. While, as you notice, you cannot dial a device, you can surely call your user to tranfer from a device to another. Please note that call

There's a large number of gadgets one can buy that work with Skype through the API. One of the things I'm interested right now is the ability to properly use a mobile phone headset with a SIP/IAX softphone. Is there an softphone that emulates the Skype API? Are there legal implications in writing an softphone that emulates the Skype API? Should I just give up and buy a Siemens DECT

> > I've never seen stats, but it's probably a safe assumption that the > > majority of IP phones are sitting next to a PC and the additional > > expense has been incurred because "people want a phone that looks and > > works like a phone". That's certainly been my experience far > > outweighing any technical issues with quality or reliability

Hi, I'm trying to get a Zyxel P2000W (reportedly also sold as WiSIP by Pulver) to work with an asterisk box. The phone connects nicely to an external VoIP company (sipgate.de reportedly using asterisk themselves) but there is a strange problem with my asterisk: - Incoming calls via ISDN (chan_capi) to the Zyxel work perfectly. - Outgoing calls (capi/sip/iax) via asterisk are dialling out

Hi, (please excuse me for lack of proper jargon usage and the vagueness of description...) i use Asterisk 11.12.1, (well... as included in FreePBX), . . . The softphones are mostly on machines without proper sound hardware (no mics, no speakers/headsets); This is partly because the workforce is quite conservative in what they want to use :) meaning handsets are important; As the handsets have

Hi, (please excuse me for lack of proper jargon usage and the vagueness of description...) i use Asterisk 11.12.1, (well... as included in FreePBX), I have several extensions that can register 2 separate devices (chan_sip) ( FreePBX calls this Devices & Users mode : Users are extension/internal number, devices are the 'SIP Accounts' for the internal 'endpoints' ) (this

I'm looking for a function I can put in my dial plan that tells me if a device is a member of a queue, but I can seem to find one. Basically I want to be able to dial to join a queue and if I'm already on the queue, leave...... exten => 4,1,GotoIf(${is_queue_member(queuename,SIP/${ext})}?leave:join) exten => 4,n(leave),RemoveQueueMember(queuename,SIP/${ext}) exten => 4,n,Hangup

Hi to all, we have a community of people on an * box that use SIP softphones to talk each other. Can you suggest me the quickest and simple way to let someone know who is online without have to call one by one the persons to look if they are present or not?? Something the user list in Microsoft Messenger. I was thinking on some sort of web page that can check the registration of the sip

> > No, this is not the case currently with any of the Cisco SIP software > loads that I am aware of. If you find this to be incorrect, please > let the list know. Cisco has not deployed much of the featureset in > their SCCP phones (such as paging/intercom) into the SIP phones due > to lack of standards/interest/political capital. > > JT Ok, after further

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Am 21.04.15 um 13:38 schrieb akhilesh chand: > Hi Guenther, > > When I executed nmap -p5060 <xx.xx.xx.xx> I got below output. > > [root at ip-172-31-32-117 cel]# nmap -p5060 xx.xx.xx.xx > > Starting Nmap 5.51 ( http://nmap.org ) at 2015-04-21 11:19 UTC Nmap > scan report for

Hi:I 've a question: I'm using asterisk@home; I've seen the dialparties.agi , I want to do this; I've one softphone and I want register it in 2 different Proxy;only X-lite permitted this, all others no; I want have more proxy with others softphone;I run asterisk - R and I've seen when a softphone is registered(when I run it). My question is: Who is the file who makes the

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 >