First, thank you to whomever it was that pointed me towards the IPP200. It works great, both under Windows with X-lite and Linux with iaxcomm. It took me about 3 minutes on each to get it configured and working. Second, has anyone tried the VTGO-PC (any version) from ipblue.com? I'd dearly love a softphone with MWI... Tim -->>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> tps@buoy.com >< (631) 399-2910 (888) 924-3728 << >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
Hmmm. I thought it was just that I didn't have X-Lite and Asterisk configured correctly and I've been searching thru docs trying to figure out how to get a MWI working! Does X-Pro have a MWI? Ed -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Tim Sailer Sent: Tuesday, March 02, 2004 9:54 AM To: Asterisk Users Subject: [Asterisk-Users] VTGO-PG and IPP200 First, thank you to whomever it was that pointed me towards the IPP200. It works great, both under Windows with X-lite and Linux with iaxcomm. It took me about 3 minutes on each to get it configured and working. Second, has anyone tried the VTGO-PC (any version) from ipblue.com? I'd dearly love a softphone with MWI... Tim -->>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> tps@buoy.com >< (631) 399-2910 (888) 924-3728 << >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
IAX Phone has MWI and, with a small change in your Asterisk, can give you a count of messages waiting. Plus it has hook-switch integration with the IPP200. Thanks, Steve Steven Sokol Owner/Manager Sokol & Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web: http://www.sokol-associates.com> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Tim Sailer > Sent: Tuesday, March 02, 2004 11:54 AM > To: Asterisk Users > Subject: [Asterisk-Users] VTGO-PG and IPP200 > > First, thank you to whomever it was that pointed me towards the > IPP200. It works great, both under Windows with X-lite and Linux > with iaxcomm. It took me about 3 minutes on each to get it > configured and working. > > Second, has anyone tried the VTGO-PC (any version) from ipblue.com? I'd > dearly love a softphone with MWI... > > Tim > > -- > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< > >> Tim Sailer >< Coastal Internet, Inc. << > >> Network and Systems Operations >< PO Box 726 << > >> http://www.buoy.com >< Moriches, NY 11955 << > >> tps@buoy.com >< (631) 399-2910 (888) 924-3728 << > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Hi, Todd Did you notice that when you made the calls, were the calls indicated as "answered", in both cases? And if so, did the indication "answered" pop up when the calls were actually picked up and answered or right after the call setup was completed -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Todd Wallace Sent: Tuesday, March 02, 2004 4:30 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] H323 calls drop on connect I have something new that is happening to me...When I call from a SIP phone and route out OH323, I get a good clear ringing, connect, then it drops me. If I get a telco recorded message, I hear the complete message. If I get a person that answers, I hear about the first 2 seconds, then it drops me. Any ideas where it look? I feel it is in the OH323 config.. Todd _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004
Right after the call setup was completed... Todd -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of T. Chan Sent: Tuesday, March 02, 2004 4:01 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] H323 calls drop on connect Hi, Todd Did you notice that when you made the calls, were the calls indicated as "answered", in both cases? And if so, did the indication "answered" pop up when the calls were actually picked up and answered or right after the call setup was completed -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Todd Wallace Sent: Tuesday, March 02, 2004 4:30 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] H323 calls drop on connect I have something new that is happening to me...When I call from a SIP phone and route out OH323, I get a good clear ringing, connect, then it drops me. If I get a telco recorded message, I hear the complete message. If I get a person that answers, I hear about the first 2 seconds, then it drops me. Any ideas where it look? I feel it is in the OH323 config.. Todd _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Can you describe your configuration? Any Asterisk log messages? Michael. Todd Wallace wrote:> I have something new that is happening to me...When I call from a SIP phone > and route out OH323, I get a good clear ringing, connect, then it drops me. > If I get a telco recorded message, I hear the complete message. If I get a > person that answers, I hear about the first 2 seconds, then it drops me. > > Any ideas where it look? I feel it is in the OH323 config.. > > > Todd > >
Jason Penton wrote:>Hi all > >Does anyone know where I can get hold of the German 1TR6 ISDN signalling >protocol specification. > >Thanks >Jason > >Is that still used? I thought they were 100% CTR4 these days. Regards, Steve
Hey Steve Apparently so :-(. It is used in our legacy PBX with which I would like to connect my Asterisk box. Cheers Jason> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Steve Underwood > Sent: 03 March 2004 03:56 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] ISDN > > Jason Penton wrote: > > >Hi all > > > >Does anyone know where I can get hold of the German 1TR6 ISDN > >signalling protocol specification. > > > >Thanks > >Jason > > > > > Is that still used? I thought they were 100% CTR4 these days. > > Regards, > Steve > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
Email back from ipblue: Thank you for your inquiry. At the present time we work under SCCP or H.323 protocols. We are developing a SIP phone which will be available sometime in Q2. Regards, Andrew Schecter Vice President of Sales IP blue Software Solutions 15 East 26th Street New York, NY 10010 212.485.1225 (tel) 212.485.1380 (fax) aschecter@ipblue.com -->>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> tps@buoy.com >< (631) 399-2910 (888) 924-3728 << >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<