[asterisk-users] automated response

2015-02-11 Thread Joseph Shi
I will not be in the office till Feb 24, but I will be checking my email periodically. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

[asterisk-users] automated response

2014-04-09 Thread Joseph Shi
I will not be in the office from April 10 to 14. I will get back to you when I'm back. Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

[Asterisk-Users] Multiple language problem

2005-07-25 Thread Joseph Shi
There seems to be a problem with call involving parties with different language. For example, if SIP user 101 (with default language en) calls SIP user 102 (with default language fr), SIP user 102 will hear English prompt when he presses the # key for transfer even though his default

[Asterisk-Users] call recording problem

2005-04-28 Thread Joseph Shi
It seems that there areoccasional problems with files generated by soxmix utility. The Asterisk console would show the following message: soxmix: Overriding output size to bytes for compressed data soxmix: help! internal inconsistency - data_written 12156 gsmbytecount 12155. When trying

[Asterisk-Users] Re: Problem with call hold

2005-03-03 Thread Joseph Shi
FYI, I just download the latest stable version from CVS and the problem is gone. - Original Message - From: Joseph Shi To: asterisk-users@lists.digium.com Sent: Monday, February 28, 2005 9:22 PM Subject: Problem with call hold I got a very strange problem with call-hold function

[Asterisk-Users] Problem with call hold

2005-02-28 Thread Joseph Shi
the answer. If somebody can shred some light on the problem, it will be very much appreciated. I'm running the Asterisk stable version at Dec 21, 2004. Thanks ahead. Joseph Shi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

[Asterisk-Users] Bandwdith usage

2004-10-25 Thread Joseph Shi
Does anybody know if the voice actually gets routed through Asterisk for callsbetween SIP devices? I just wonder if calls between SIP devices would take up any bandwidth or CPU at the Asterisk server. Please advise. Thanks, Joseph ___

Re: [Asterisk-Users] Bandwdith usage

2004-10-25 Thread Joseph Shi
[EMAIL PROTECTED] Sent: Monday, October 25, 2004 5:30 AM Subject: RE: [Asterisk-Users] Bandwdith usage Joseph Shi [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) Does anybody know if the voice actually gets routed through Asterisk for calls

[Asterisk-Users] CED tone and answering machine detection

2004-10-11 Thread Joseph Shi
Does anybody know whether Asterisk is capable of detecting fax tone (i.e. CED tone) or answering machine (i.e. long voice) on an outbound call - using Zaptel and/or SIP? It seems that it is only capable of detecting fax tone on an incoming call using Zaptel only. I'm trying to determine if

Re: [Asterisk-Users] VoIP Telephony with Asterisk book

2004-08-29 Thread Joseph Shi
Steve Underwood Wrote: Joseph Shi wrote: Does anyone know if there are any reseller for the book VoIP Telephony with Asterisk in Hong Kong/Asia region? I'm interested in purchasing the book but the shipping charge to Hong Kong is expensive. Thanks. Joseph Just wait for the simplified

[Asterisk-Users] VoIP Telephony with Asterisk book

2004-08-26 Thread Joseph Shi
Does anyone know if there are any reseller for the book "VoIP Telephony with Asterisk" in Hong Kong/Asia region? I'm interested in purchasing the book but the shipping charge to Hong Kong is expensive. Thanks. Joseph ___ Asterisk-Users mailing

[Asterisk-Users] BudgeTone IP phone in Hong Kong

2004-05-26 Thread Joseph Shi
Does anybody know where I can get Grandstream BudgeTone IP phone in Hong Kong or South East Asia region? Please advise. Thanks. Joseph