similar to: Outlook Integration

I am going to buy some IP phones from them but I sent them an email couple of weeks ago and got no reply. Has anyone ordered anything from them? Any other places that I can buy from? Sorry if it's a wrong post. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050517/ba256d25/attachment.htm

My router (1605R) currently does not support QoS. Is there any open source software available so that I can set one up before the router? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/38b6e3f2/attachment.htm

Hi, I am thinking of signing up with voice pulse connect to connect to my asterisk server and using it as a regular line. Is it good? Or should I go with vonage or others ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050124/16792f10/attachment.htm

I am thinking of dumping broadvoice so I need some other VoIP providers that have a las vegas DID and a service better than broadvoice. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050131/bf4b1c31/attachment.htm

I finally got my incoming and outgoing to work on Broadvoice with a configuration file that is no where close to the one given by them. Here it Is (sip.conf). For others who have a working config could u please share so that I could compare. Thank You [broadvoice] type=friend username=[number] fromuser=[number] secret=[password] host=sip.broadvoice.com fromdomain=sip.broadvoice.com

I have foloved instructions at this web pages http://www.omniis.com/ntsgr/cms/page.asp?688 and now I'm able to call contacts from Outlook. Now I have few questions. When I place a call, my phone rings before * tries to dial out. Is it posible that * first dials out, and when other side picks up, at that moment that my phone rings? Another question, when I recive a phone call, can that

Is the Broadvoice service up? I just signed up with them and started receiving calls in no time but could not make calls. And after a few minutes I cannot even place calls. register => [number]:[password]@sip.broadvoice.com [broadvoice] type=peer fromuser=[number] host=proxy.lax.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice dtmfmode=inband any help would be

Hi, We are looking towards a 8+ CO line setup (20 extensions) in our office but we do not want an IVR(auto-attendant) feature. All incoming will be answered by a receptionist. I have read the multi-line configuration for cisco 7960 thread in this list but that way I believe we could only display 6 incoming lines. What will happen to the rest? Does the expansion module for the cisco 7960 work

I believe the 911 is a serious issue if one does an asterisk installation in an office. How do you test 911? Won't they arrest you or something for dialing 911 for no reason and talking to one of their agents who could have taken a more important call? On the other hand what an emergency comes up (like someone got seriously injured) and on top of that asterisk crashed all of a sudden

Which would you recommend as far and quality and pricing to connect to asterisk (including DTMF issues)/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050125/bf7bf081/attachment.htm

Is there any way to change the outgoing caller id on BroadVoice I have tried SetCallerID(Name <Number>) but that does not work Thanx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050126/bbd95fcf/attachment.htm

I have a couple of DID's through VP Connect and have been having sound quality issues on incoming calls. During the call, the calling parties voice sometimes sound like it is crackling, in other words it is not very crisp. I would liken it to listening to a radio with a blown speaker. This sound defect comes and goes throughout the call. The other person is always audible but it just isn't

Hi All !!! Somebody can inform me where to get more information about the TAPI (M$) interface of the Asterisk ? Thanks in advance. --- Isamp

Just got a headset for testing asterisk and am using X-Lite. I plugged in the headset into the headset jack and is there any way to configure X-lite to use the headset instead of the speakers? Or will I have to plug the headset in the speaker jack ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to do the following: 1. Call comes in to my * box over IAX (VP Connect DID) 2. Check to see if call should be forwarded to my cell 3. Forward the call to my cell phone and take * out of the media path. I am able to do all of the above except * is not able to natively bridge the call. I am using sixtel and for the call forward portion, but the calls don't connect before sixtel

... I just tried again after removing my hosts file entry (again) and outbound is now working! I had taken it out before, but I think I was getting a different error at the time. Sometimes it seems like asking for help is itself a cure! Thanks anyway! JDC

I did a make webvmail and I get the following error on redhat 9.0 No HTTP directory make: *** [webvmail] Error 1 I have the perl-suidperl rpm installed and apache installed Thanx. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050501/275c1e94/attachment.htm

Hi, I was going to order the T100P but it is replaced by TE110P. On further reading the TE110P does not need an external router (The one that separates the data from pstn lines ?). Has anyone got it configured? And on the wiki it says that the drivers for some distros don't exist yet. Is redhat supported? And if I need to connect a fax machine will the FXO cards on ebay work (I know to

I am running * 1.0.5 and have been having lots of problems with outgoing calls and their sound quality. I am using ULAW for the codec and sixtel for termination. Basically the problem is that portions of the call seem to be lost and replaced with silence. Sometimes I can't hear the person talking othertimes they can't hear me. This situation comes and goes throughout the call. Bandwidth

Funny how a topic will come up that you have never dealt with before, and suddenly it comes up from multiple directions at the same time. I was recently involved in a meeting where TAPI (which I understand only vaguely) was proposed as way to link a custom application to Asterisk for outbound and inbound call processing, much like SugarCRM and probably others are doing. Today I was asked by