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Has anyone considered the idea of splitting the sip registration information in a realtime database from the actual configuration of the peers? I mean, instead of having a table full of the configuration information (i.e. name, regexten, secret, etc) and registration information (i.e. ipaddr, fullcontact, etc), you have separate tables with their own information. This way, you can have separate

asterisk-users-bounces@lists.digium.com wrote: > Vadim Berezniker wrote: > >> That's not a solution, but just a workaround. >> 1.2.1 has a bug where it always uses an empty context when searching >> for a mailbox when using realtime config. >> At around line 546 of apps/app_voicemail.c there is a line that says >> var =

I wrote a connection pooling patch because asterisk is not usable with MSSQL without it. If you're using, or would like to use, MSSQL I recommend you to check it out. http://bugs.digium.com/file_download.php?file_id=8809&type=bug Just so you know, this is a diff against 1.2.1 and it's been only tested with 1.2.1 (because that's what we are using in production). To enable pooling,

I think I have 4 options. 1, Modify chan_sip.c to update a new field in sipusers realtime table with the status of the sip peer/user. Then use agi to dial sip calls. Check the status field if OK then dial the fullcontact from the sip table. If not goto voicemail or where ever else I want the call to go.. The UA would only register to one server, so only one server *should* be writing to the