similar to: RE: Hook/Flash, Hold, Call Waiting, Three Way Calling

Hello All, I have a customer site that is using * for ACD. In comming calls are eventually routed to a support rep via a queue. For new accounts the agent needs to be able to send a flash to the PSTN trunk (a POTS line with 3-way calling enabled), dial the number of an authentication center and then connect all three parties together. The trick is that both the agent and the customer need to be

Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). Is there one (pocket pc softphone) that works? Thanks, Cosmin Prund

Hi, I am trying to get a SIP extension's status without actually making a call. I am using sofia-sip's "options" example utility and the sip clients are SJphone softphones.

Hi All I seem to have a small problem with the music on hold button on SJPhone. I have 2 asterisk installations one from the Rapid distribution and one from the latest CVS. On the rapid dist when I press the music on hold button on my SJPhone I get music on hold. When I do the same I get no music on hold just silence. I create extension like this exten => 1111,1,MusicOnHold(Default),

Hi, I have two softphones connected to an Asterisk "stable". I have two extensions, say 1000 and 2000. When 1000 calls 2000, the call cannot be completed; the softphone (either Diax97a , SJphone, Firefly 1.8) on extension 2000 will ring, but as soon as the call is picked up, extension 2000 will hang up the call. The softphone on 1000 (SIP, SJphone, e.g.) will give a "403

Hello. After doing much web searching and searching archives of this mailing list, I see that my question has been asked at least 6 separate times but no answers have been attached. In a nutshell, is there a way for a SIP phone to easily hook flash a ZAP analog trunk mid-call? (This is important when trying to make use of features on a PSTN analog line such as call waiting, call forwarding,

I have been looking for the answer to this question for a while. Google-ing and reading the archives of Asterisk-Users has not enlightened me. It seems that this question has been asked many times, and many times it has gone unanswered. I have call waiting and three way calling on my PSTN line from Verizon (the local telco). This is connected to a FXO port on a TDM400P. I also have

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",

Hi All, Does any one know of a way to make a three way call from Asterisk using X-Lite. I need the ability to be able to call someone on the PSTN using my IAX provider then bring another person from a local extension into the call if needs be? I believe most three way calling is done using a feature of the phone, and X-Lite doesn't look like it supports this. Can this be

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 , 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

Hi All, I'm currently using SJPhone as a sip client with Asterisk. It works perfectly with a USB Yap Phone but there is was slight problem! If I setup SJphone to use the USB device for audio the ringing tone is also over the USB device. If I'm away from my desk I can't hear the phone ring! Is there an option to use one audio device for calls but another for the ringing sound?

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 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

Hello, I have been working on getting Asterisk up and running with a couple Grandstream hard phones (Budgetone 100s) and a couple softphones. Things work pretty well but we have discovered an odd problem. A calls B and then A can put the call on hold and pick it back up. A calls B and B can put the call on hold but cannot pick it back up. These tests are internal on the hard phones. The

Can PDAs be used as softphones/clients on asterisk? what i wanted to do is to set up 2 PDAs as softphone(client) which allows them to communicate each other through asterisk server(desktop) devices i have: pda compaq model 3680 pda sharp sl5500 access point desktop(asterisk) can i apply my idea on the asterisk? any guides? thanks in advance :) --------------------------------- Do

Hello everybody, I?ve been pulling my hair for a week now over a problem, and I really don?t know where to look anymore. Here?s my setup: There is an Innovaphone IP3000 VoIP gateway on the LAN (10.253.30.254). I can use it to send and receive calls from physical phones attached to it. I have setup Asterisk 1.0.3, with H323 and openH323, and on the same server I also setup GnuGK (10.253.30.1). I

Now here is strange problem i experience. Setup is easy, IAX line out with SIP softphone registered to Asterisk. All work fine except for one client. When using Sjphone the other end can not hear a thing. When using X-pro the opposite happens, local user can not hear a thing. These softphones work fine on other clients on same network. I've also tested several headsets but same outcome. Also

Hi folks last weekend, I tried Windows Messenger first time and was stunned by the little latency it gives. Until now, I've been using softphones on Linux exclusively, like iaxcomm, linphone and sjphone, and they all give me about 1, at times even 2 secs delay. Whereas Messenger really seems to be in the millisec range. Of course, I'm now curious why there is that difference. Clearly,

Can anyone _recommend_ a downloadable OSS softphone that _works_ under Linux and is compatible with Asterisk. So far I have tried kphone and linphone and had problems with both, and I am still waiting to hear back from the X-Lite beta folks. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just