search for: schelin

Displaying 12 results from an estimated 12 matches for "schelin".

2005 Jul 27
19
Full T38 sip Faxing now Available
Hello everybody, for all of you that have searched for a real fax solution, look no further. We now have T38 faxing. Please contact me for more information. Thanks Michael D. Schelin ShellTel 626-814-2354
2005 Jul 18
9
So you all think VoIP sypply is warm and fuzzy
...you for a full refund and express my views to the Voip community. As of now I've herd of nothing but good things about your customer support. I've called and left messages to your support team. I waited 7 days for this unit and have no way to configure it. Email me the CD. Michael D. Schelin Owner Shelltel
2005 Jul 19
1
Re: So you all think VoIP sypply is warm andfuzzy
...t; Why? Because once you get one CD there is no need to keep receiving > them, when ordering in bulk. > > > > So, if any of you plan to order the Mediatrix 2102 from anyone, make > sure you ask for the CD at the time of purchase. > > > > I would like to thank Mr. Schelin for bringing this to our attention > and we will be updating our site to prevent this sort of problem from > occurring again. > > > > If anyone has any further questions about the 2102, or how to order > one with or without the CD, please contact me offlist. > > &g...
2005 May 27
3
G729 vs. gsm
I installed G729 from Diguim and I was expecting the sound quality on my i686 machine to be better than gsm. Compared to gsm, G729 sounds closer and a little robotic. Is this what is supposed to be or am I missing something? I am interested in G729 because the internet in my country is very expensive and I want to save every bit possible. I want to use G729 because it takes less bandwidth for
2005 Jul 01
1
Re: [Asterisk-ss7] Asterisk - ss7
I thought everyone should know this. Jorge, After reading your page in the http://voip-info.org/tiki-index.php?page=Asterisk+SS7 please advise Your U.S. customers that SS7 is not done the same way as in the rest of the world and the requirements are different. The U.S carrier's require 2 redundant links. I know this first hand because we run an SS7 network. CARDOSO Jorge Miguel wrote:
2005 Sep 25
2
change codec based on callerid (sip/iax)
I have been asked if asterisk can change codecs dynamically based on the calling party's caller id. I couldnt find anything, and dont know that this is something that asterisk can do, but it occurs to me that possibly with a reinvite it can be done, however I dont think you can issue those from the dialplan or agi. The only solution I can think of on this is to use something like ser
2005 Apr 20
3
TE110P
Ok I F@%& up. I didn't realize the card is 3.3 volts and my new computer is 5V. Can anyone point me to a PCI to PCI bridge. Any suggestions? Mike
2005 Sep 06
9
civil emergency comms: Asterisk + HAM
The disaster in the Gulf coast and the less than optimal initial response suggests to me that citizens must shoulder more responsibility for emergency management. Communications loss must have played a large role in the failures that occurred. I can't help but wonder if there are fewer ham radio operators today and that if there were more, maybe they could make a difference in future
2005 Jun 24
4
Tellabs Echo Canceller
I am getting ready to experiment with the Tellabs 2752 echo canceller. I have a 255D shelf (and power supply), but am struggling a little on connecting the echo canceller to a PRI. The shelf has 4 25-pair amphenol connectors. The two on the line side are marked "Receive In" and "Send Out". The 2 connectors on the drop side are marked "Send In" and "Receive
2005 Apr 22
4
TE11OP -> Mitel 200Sx??
Hello all. I just received a TE110P and am trying to hook it to my Mitel 200SX has anyone successfully done this? My configuration is as follows. Asterisk -> TE110P ->Kentrox (csu/dsu) -> Mitel T1 Card. All I get is a blinking yellow on my TE110P card and an alarm on my Mitel. T1 card. Any advice would be great. Zaptel.conf span=1,0,1,d4,ami e&m=1-23 dchan=24
2005 May 26
4
YET Another echo issue PRI CARD Any help accepted :-)
Good Day all, I have a Fractional PRI connected to my Asterisk Box via a T100P card. When I initiate a call out to phone number 123-8888 the call sounds great no echo what so ever. If the person at 123-8888 hangs up and calls me right back (same handset on both sides) same trunk line The call always has echo on it. The Asterisk sip extension hears them selves echoing. The remote party
2002 Oct 31
0
Problem with samba2.2.6
Hi, I?m new to this list and I have a little problem I hope someone could help me with. I have a home network running Mendrake Linux 9.0rc3 and 2 clients each running Winxp Pro. The problem is that when I try to copy files to the server the client stops, after some time, to send the files to the server. I get "connection reset by peer" on the server and the client reports that