| Summary: | Voice conferencing system is less than optimal | ||
|---|---|---|---|
| Product: | Community | Reporter: | Jeff McAffer <jeffmcaffer> |
| Component: | CommitterTools | Assignee: | Eclipse Webmaster <webmaster> |
| Status: | RESOLVED FIXED | QA Contact: | |
| Severity: | normal | ||
| Priority: | P3 | CC: | andrea.ross, caniszczyk, david_williams, gunnar, mike.milinkovich, mober.at+eclipse, pascal, wayne.beaton |
| Version: | unspecified | ||
| Target Milestone: | --- | ||
| Hardware: | PC | ||
| OS: | Mac OS X - Carbon (unsup.) | ||
| Whiteboard: | |||
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Description
Jeff McAffer
It took me at least a dozen attempts today to just get past the busy signal using both the 877 and 416 numbers. I have been running into issues similar to those reported by Jeff. I too had trouble, but maybe different? dialed in to "US number", using passcode on wiki page at http://wiki.eclipse.org/Architecture_Council/Meetings/January_13_2011 First time, seemed to connect, but silence for a minute or so. Hung up, redialed, and after a 2-3 minutes heard a message that "the moderator code has already been used, please hang up and dial back in". So, I did ... even though pretty sure I did not use moderator code ... this time, after 5 or 10 minutes, asked who else was on ... there were 5 or 7 of us ... all of us had dialed in on "the US number". One person said they tried the Canadian one, but that it didn't work. Martin (or no other Canadians :) seemed to be on the call by 11:13, so I gave up and dropped off. :( (In reply to comment #3) Oh, and I was not using Skype ... plain 'ol Verizon cell phone. Adding Andrew to this. Our new asterisk facility (http://wiki.eclipse.org/Asterisk) should solve most of this. SkypeOut users should have no trouble calling into our conference system. Quality will depend on a number of variables but most importantly will be the network bandwidth/latency of the Skype user. An option that once existed, Skype to asterisk is no longer for sale: http://www.digium.com/en/products/software/skypeforasterisk.php It provided proprietary software that enabled Skype users to connect to Asterisk directly. Using a SIP client is also a good option. Should we close this bug? Agreed. Please open new bugs if you're still experiencing issues with Asterisk-driven conference calls. |