Hello everyone, Since I've been working with SIP more and more I've discovered there are still plenty of interop and configuration issues between various pieces of equipment in the real world. I enjoy helping with SIP issues in this forum and others but I thought it would make more sense to aggregate this information in a central location. For instance, earlier today a user had a problem between his Cisco AS5300 and Asterisk 1.2. The solution was fairly technical and not very obvious. I was more than willing to help here but then I thought, wait - what if someone on a Cisco list somewhere has a similar problem? What if I'm not there to read his post and reply? What if he can't find it in the Asterisk archives for some reason? What if he/she never gets the issue worked out? Today I plunked down the $9 for submityoursip.com. My goal is to (eventually) have a site where interop details and implementation quirks between various SIP platforms can be easily searched, discussed, etc. Trying to work with OCS and Asterisk? Need a pointer to a TCP/UDP SIP proxy? Can't figure out how to get your Polycom/Asterisk/Cisco/Snom/Sonus equipment to agree on a codec, method of caller id, or DTMF mode? This wiki should help. I'll be adding some more details, fixing up syntax, etc in the next couple of days but for now I thought I'd get the announcement out to see if anyone would like to help: - Wiki formatting. I don't know anything about MediaWiki. Headings, tables, organization, etc. Help! - SIP devices. Manufacturers, service provider offerings, devices, etc. I've started to make lists to (mostly) empty pages, but this part will never be done! - Debugs/SIP traces. Have a strange interop issue? Post the SIP and we'll (at least I will) take a look at it. Maybe we can figure it out for you and add it to the wiki for everyone. One thing I don't want to do is duplicate effort elsewhere, copy/paste from other sites, etc. If you can link to an external resource, please do! In case you missed it before the address is http://www.submityoursip.com and it's free (of course) and you can sign up for an account if you feel like helping me out... :) Thoughts? Tips? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com
A very good idea. I heartily endorse. Kristian Kielhofner wrote:> Hello everyone, > > Since I've been working with SIP more and more I've discovered there > are still plenty of interop and configuration issues between various > pieces of equipment in the real world. > > I enjoy helping with SIP issues in this forum and others but I > thought it would make more sense to aggregate this information in a > central location. For instance, earlier today a user had a problem > between his Cisco AS5300 and Asterisk 1.2. The solution was fairly > technical and not very obvious. I was more than willing to help here > but then I thought, wait - what if someone on a Cisco list somewhere > has a similar problem? What if I'm not there to read his post and > reply? What if he can't find it in the Asterisk archives for some > reason? What if he/she never gets the issue worked out? > > Today I plunked down the $9 for submityoursip.com. My goal is to > (eventually) have a site where interop details and implementation > quirks between various SIP platforms can be easily searched, > discussed, etc. > > Trying to work with OCS and Asterisk? Need a pointer to a TCP/UDP > SIP proxy? Can't figure out how to get your > Polycom/Asterisk/Cisco/Snom/Sonus equipment to agree on a codec, > method of caller id, or DTMF mode? This wiki should help. > > I'll be adding some more details, fixing up syntax, etc in the next > couple of days but for now I thought I'd get the announcement out to > see if anyone would like to help: > > - Wiki formatting. I don't know anything about MediaWiki. Headings, > tables, organization, etc. Help! > - SIP devices. Manufacturers, service provider offerings, devices, > etc. I've started to make lists to (mostly) empty pages, but this > part will never be done! > - Debugs/SIP traces. Have a strange interop issue? Post the SIP and > we'll (at least I will) take a look at it. Maybe we can figure it out > for you and add it to the wiki for everyone. > > One thing I don't want to do is duplicate effort elsewhere, > copy/paste from other sites, etc. If you can link to an external > resource, please do! > > In case you missed it before the address is > http://www.submityoursip.com and it's free (of course) and you can > sign up for an account if you feel like helping me out... :) > > Thoughts? Tips? >-- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599