search for: uc9

I know this has been gone over before but It seems that most of the traffic to this list is the same 10 questions being asked over and over again. What if someone (with some programing skill) wrote a script so that when someone posted to the list, it would search the wiki and google and respond to them. If they where not happy with the response, then they could forward there message back

Hello I saw Ver1.2 is out. Whats new? Thanks for the hard work, David

Hi all I am having some issues with a gs 100 phone. It is on the same network as my * server. There is no firewall. In extentions.conf exten => 5,1,Answer exten => 5,2,MusicOnHold(default) When I dial 5 from the sip phone -- Executing Answer("SIP/mlh-2e75", "") in new stack -- Executing MusicOnHold("SIP/mlh-2e75", "default") in new stack

Is there anyway to support NFTS security on linux. it might mean compiling a new file system into the kernal? I don't know. Is there a how to that anyone can point me to. Sorry for the basic question, Michael

Hi all, I know that this has been passed around before, and I know that it happens about every 3 months or so, but evertime the answers change, so I thought I would pass it around again. A company I work for has 3 incomming lines and 4 phones. They require voicemail and MOH. Their phone systems VM hard drive died today, they were quoted a $2000 to replace it. I started to talk to them about

Hello, I have finished my basic polishing of the Asterisk GUI client I have been writing in Perl/TK and have released a first beta version on sourceforge: http://sourceforge.net/projects/astguiclient/ I am still working on a user manual for the application, but the code works and we have been using the same basic client for the last month here at my company and it is working just fine. I'm

I have spent the morning on this project, still without success. Summary: Yesterday I inadvertently unplugged my Grandstream phone. I might add I did a rebuild of my s/w from CVS at the same time. Since then, the Budgetone seems to talk SIP just fine, but the RTP being sent to it by asterisk "doesn't make any sound." It was suggested I do a factory reset of the phone, which I

I have looked though the archives on this issue and I can not find a final answar on it. So I will bring it up again. I have an x100p card.....if Asterisk picks up a call, and it see if there are fax tones on it, and if so transfer it somewhere. What is a user has picked it up. Can it still transfer it, or would the user have to? Thanks again, Michael

I just got a grandstream SIP phone Here is my sip.conf for the phone [mlh] type=friend insecure=yes username=mlh secret=mlh host=dynamic canreinvite=no The phone as the default config on it. If I use the phone to call a Zap interface (a tdm card) the voice sounds all choppy. If I use the phone to call a x100p card, it does not dial what I dial (no DTMF) I don't know

Does anyone out there use Asterisk with voip(sip or iax) long distance provider? Care to share about your experiances doing this? Michael

I have an asterisk box with no zap hardware in it. I use the ztdummy for music on hold. However, the music is very distorted. I rmmod ztdummy, and the Music sounds great. Michael

We would like to know if there is a way to broadcast (in realtime) a conferance. We hold large phone conferances and would like to know if we could have some of our users listen over a streaming services. Formats we have looked at include: Shoutcast,Real Networks,QuickTime, and dare I say Windows Media player. The issue we have, is that I can not find a way to transfer the stream in realtime

I have an issue getting any incomming calls When the phone rings something picks up and gives it a fast busy. There is no one using Zap/2 it does the same thing with voicemail and voicemail 2 you can see the console output below, I would love any help anyone could shead on this issue, Michael NOTICE[1192484144]: File chan_zap.c, Line 4270 (ss_thread): Got event 2 (Ring/Answered)... --

I just checkout the cvs code for asterisk...... when I use my grandstream phone (that worked on the old code that was about 2 months old) I do not hear anything at all... I get this error: Sep 27 23:20:27 WARNING[1142127920]: File chan_sip.c, Line 444 (retrans_pkt): Maximum retries exceeded on call 0765c89e-9d67-3c0a-b9b9-2e7f3cd1d9ef@192.168.50.248 for seqno 58430 (Response) here is my

I have an X100p card....and it is hard to hear the person on the other end. Should I mess with these values? I have heard both yes and no to this question in the past. If yes, how much louder should I make them? Thanks, MIchael

I have a x100p card that has call waiting on the line comming into it and then into *..... is there any way i can use call waiting on that line? Michael