Hi Mark, With regards to number 1, I would say that you need to notify voicenet.ca what number you called, and at what time. Most likely they're routing your calls over several different provides, and some of those routes have DTMF issues.
Alex ___________________________________________ Alex Robar, Technical Support, GearyTech Inc. 3075 Fourteenth Avenue, Unit 3, Markham, Ontario L3R 0G9 Markham: 905-513-8000 x 223 Fax: 905-513-8040 Toronto: 416-226-3614 Toll Free: 888-890-3499 [EMAIL PROTECTED] www.gearytech.com Strategic management of technology for business. -----Original Message----- From: Mark Borg [mailto:[EMAIL PROTECTED] Sent: Friday, May 11, 2007 1:48 PM To: [email protected] Subject: [on-asterisk] Inconsistent DTMF during calls Hi, I am a "new member" trying to get up to speed with asterisk. Got 2 questions: 1. I have a setup at my shop, voip via voicenet.ca; asterisk 1.4 & aastra 480i terminal. No analog lines avail. It seems fine, except, for with some calls, I cannot have a dtmf tone recognised by the called pbx. (like press 0 for operator, etc.) On other pbx's it works OK. 2. I set up a client with two TDM400 & 6 POTS lines (only)& AAstra 9131 terminals. One application is to bridge an inbound call to another line & dial back out to a remote office. HOWEVER, the resulting call is at a very low volume & client is irritated. Changing the line parameters with fxotune just makes a bunch of clipping & echo, and one cannot tell which lines are to be bridged as it depends on the incoming call. Does anyone have insight? Should I (eat the TDM400s and) install sipura hardware? thanks Mark Borg --------------------------------------------------------------------- To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] -- ExchangeDefender Message Security: Click below to verify authenticity http://www.exchangedefender.com/verify.asp?id=l4BHvgin005490&[EMAIL PROTECTED]
