Re: [Asterisk-Users] IRC Etiquette
My comments on these matters is simple. We (newbies or experienced) still needs to learn from our experiences. Personally, I'm very appreciated when I asked a dumb question, someone replies me with the link to the documentation. Mostly it helps, but again, the documentation is not perfect. I think it's our job to make a perfect documentation so it will help others to understand more. Thanks Isianto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfering with Grandstream Phones
I don't know about new firmware, I'm using 1.0.4.55, but the transfer works fine with gs BT-100 with cvs asterisk (Downloaded yesterday) Where can I get firmware 1.0.4.63? can somebody give me a link? and what improvements from firmware 1.0.4.55? Isianto Interesting! Because a few months ago, maybe 6-9 mths, I had no problem transferring calls with BT-100. Then all of a sudden after a cvs upgrade it stopped working. I haven't tried any newer firmware, though. jeremy - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFAn+GyljK16xgETzkRAklKAJ94s7yZNWi9D1oOLkB2NYFCJewpcgCfcmFu PNPWMqInqTtVRvFNhDcE2Gk= =N6ew -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] grandstream transfer, park and conference
1. Check if Asterisk is always in the media path, i.e. you need the t or T option (or something similar) in your Dial statement. Alternatively you could introduce a canreinvite=no in sip.conf for the GS phones. 2. Check your context setup in extensions.conf and make sure that in call cases your GS phone has the parkedcalls context available Philipp I have an update for this problem, and I discovered strange problems. I can do transfer, call parking nicely now except one thing: * only recognize one dtmf only (for example when I press # on my budgetone, it will say transferring, and put my caller on music, but when I press 234, * only catch 2 (in my budgetone, it will say there's no valid extension .), but if I transfer it to one digit extension first (when the call is received, then I want to do transferring/parking/meetme, I need to transfer the call to extension that has only 1 digit, then it will work perfectly (I can transfer anywhere I want (2/3/4 digits)) Is it a bug? If it is, from budgetone, or *? And how to deal with it? Thanks Isianto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] grandstream transfer, park and conference
What's your extensions.conf and sip.conf for your Grandstreams look like? I'm not in my machine right now, but here's the relevant configs Extensions.conf [ext] Ignorepat=9 Exten=_9XX,1,Dial(zap/1,tTr,20) Exten=_9XX,2,hangup [sip] Include=ext Include=parkedcalls Sip.conf Posrt = 5060 Bindaddr = 0.0.0.0 [Isianto] Type=friend Username=Isianto Secret=xxx Host=dynamic Qualify=50 Context=sip Mailbox=22 Disallow=all Allow=gsm Allow=ulaw Dtmfmode=info [Istiadi] Type=friend Username=istiadi Secret=xxx Host=dynamic Qualify=50 Context=sip Mailbox=23 Disallow=all Allow=gsm Allow=ulaw Dtmfmode=info What are your options in the GS config webpage for: 1) NAT transversal (and are you behind a NAT firewall) Set to no 2) Send Flash event Set to no 3) Send DTMF Dtmf = info (I tried rfcxxx, I forgot) Thanks Isianto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] user password and call waiting
Can you give me an example or point me to the page where account codes are described? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Tuesday, January 20, 2004 1:08 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] user password and call waiting Use account codes. That works ALOT better. If you require passwords then look at app_authenticate. bkw On Tue, 20 Jan 2004, Ing Isianto Istiadi wrote: Dear all, I have a questions: 1. I have 3 FXO connected to 3 analog phones. But I have 5 users using those phone. I want to be able to log who is using the phones and where to. I'd like to use password for each user so that I can keep track who is the caller and for how long. I read the authenticate application, but I think it is for one user only. Forgive my English. Fxo -- phone1 user A use phone1 or phone2 or phone3 after entering Fxo -- phone2 password like 1234, so if A want to call from either phones Fxo -- phone3 A needs to punch 91234xxx The same with user B, B needs to punch 92345xx And so on. But in my logger (either text based or database based), I need to see the caller is A and the rest is the same. Can I do this with *. What is the effective approach? 2. I Use digium hardware (FXO and FXS), * v0.5. Can I activate the caller waiting feature on the fxs's? So if phone 1 is being used, and I called phone 1 from phone 2, phone 1 will get call waiting tone, and from phone 2 will hear the connecting tones? I put callwaiting=yes in Zapata.conf already. But it didn't work.Any help? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] user password and call waiting
Dear all, I have a questions: 1. I have 3 FXO connected to 3 analog phones. But I have 5 users using those phone. I want to be able to log who is using the phones and where to. I'd like to use password for each user so that I can keep track who is the caller and for how long. I read the authenticate application, but I think it is for one user only. Forgive my English. Fxo -- phone1 user A use phone1 or phone2 or phone3 after entering Fxo -- phone2 password like 1234, so if A want to call from either phones Fxo -- phone3 A needs to punch 91234xxx The same with user B, B needs to punch 92345xx And so on. But in my logger (either text based or database based), I need to see the caller is A and the rest is the same. Can I do this with *. What is the effective approach? 2. I Use digium hardware (FXO and FXS), * v0.5. Can I activate the caller waiting feature on the fxs's? So if phone 1 is being used, and I called phone 1 from phone 2, phone 1 will get call waiting tone, and from phone 2 will hear the connecting tones? I put callwaiting=yes in Zapata.conf already. But it didn't work.Any help? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sending voicemail with qmail
Dear all, Is * capable to use qmail as a MTA? If so, how can I set it? Thanks Isianto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: Sending voicemail with qmail and call waiting
Dear all, Is * capable to use qmail as a MTA? If so, how can I set it? I'm using asterisk v0.5, and TDM30B (FXS), Wildcard X100P(FXO), and x-lite(SIP softphone). In zapata.conf, I put already callwaiting=yes. My PSTN doesn't not support the callwaiting feature, so I don't expect the FXO is call waiting enabled. The question is can FXS and SIP support call waiting?? Cause everytime I called my FXS, it always gave me a busy signal (when the fxs port is in use). Thanks Isianto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sending voicemail with qmail
On Thu, 15 Jan 2004 07:14:00 -0500, Andrew Kohlsmith wrote: Is * capable to use qmail as a MTA? If so, how can I set it? It shouldn't be an issue, as qmail has the standard 'sendmail' binary included. Regards, Andrew In My * box, it has a running and working qmail (with sendmail and postfix removed from services). But when my client leaves a voicemail, it says in the log that I need to set my hostname or mydomain in /etc/postfix/main.cf So, If I don't want to use sendmail and postfix at all, Can I do it with pure qmail? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip and x-lite
try this... http://www.fnords.org/~eric/asterisk/ cm Thanks for the Info, and It worked. But I have a couple of questions: 1. There's an echo. How to get rid of the echo? 2. Is there any way to call from x-lite just the extention number? (say that in my extention.conf, I have extention 32 to connect to my fxs card (TDM). If I just call 32, it will time out. The work around that I did is to add the user that I want to call to phone book with the extention like [EMAIL PROTECTED], is there any way to do this? Did I miss something in the configurations? Thanks Isianto I ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip and x-lite
Dear all, Can you give me the configurations for x-lite and sip in *. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] questions
Dear all, I have activated call waiting (but since my pstn doesn't support call waiting, I can't test it with the pstn), and I have 3 fxses. But when I call the extentions (if that extention is already called), then I got the busy tones. Is it possible to use call waiting for fxs phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbe Questions.
Dear all, I'm just finished installing the TDM (2 port) and X100P. I'm using X100P to pstn, and the TDM to the phone. I've loaded the module, and I can also list the card in the /proc/zaptel/ I'm a little confused now. in zapatel.conf, how do I know which channel is which. (TDM or X100P)? Thanks and pardon my English Isianto Istiadi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
