search for: edhorton

...hed by seanbright, tilghman. Tested by pabelanger) * Manager cookies are not compatible with RFC2109. Make that no longer true. (Closes issue #17231. Reported, patched by ecarruda) * With IMAP backend, messages in INBOX were counted twice for MWI (Closes issue #17135. Reported by edhorton. Patched by ebroad, tilghman) * Fix possible segfault when logging (Closes issue #17331. Reported, patched by under. Patched by dvossel) * Fix memory hogging behavior of app_queue (Closes issue #17081. Reported by wliegel. Patched by mmichelson) * Allow type=user SIP endpoints to b...

...hed by seanbright, tilghman. Tested by pabelanger) * Manager cookies are not compatible with RFC2109. Make that no longer true. (Closes issue #17231. Reported, patched by ecarruda) * With IMAP backend, messages in INBOX were counted twice for MWI (Closes issue #17135. Reported by edhorton. Patched by ebroad, tilghman) * Fix possible segfault when logging (Closes issue #17331. Reported, patched by under. Patched by dvossel) * Fix memory hogging behavior of app_queue (Closes issue #17081. Reported by wliegel. Patched by mmichelson) * Allow type=user SIP endpoints to b...

...have calls from soft and ip phones going to asterisk then out to the ta fxo to fxs. Regular pots phones go to a telephone distribution panel with the Centillium bringing service to line one. I'm interested in using the MGCP hopefully to be able to use sip outside of my lan. Edwin Horton edhorton at navcorp.com wrote: I noticed that there is some interest in MGCP slave operation for Asterisk to enable it to work with the AT&T Callvantage offering. I have tried the FXS/FXO connection to Asterisk and the Linksys TA with little success. Dropped calls are the biggest problem, which does...

...Reported, patched by one47. Patched by jpeeler) * Resolve an issue with the Asterisk manager interface leaking memory when disabled. (Reported internally by kmorgan. Patched by russellb) * Support greetingsfolder as documented in voicemail.conf.sample. (Closes issue #17870. Reported by edhorton. Patched by seanbright) * Fix channel redirect out of MeetMe() and other issues with channel softhangup (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb. Patched by russellb) * Fix voicemail sequencing for file based storage. (Closes issue #18498, #18486. Reporte...

...Reported, patched by one47. Patched by jpeeler) * Resolve an issue with the Asterisk manager interface leaking memory when disabled. (Reported internally by kmorgan. Patched by russellb) * Support greetingsfolder as documented in voicemail.conf.sample. (Closes issue #17870. Reported by edhorton. Patched by seanbright) * Fix channel redirect out of MeetMe() and other issues with channel softhangup (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb. Patched by russellb) * Fix voicemail sequencing for file based storage. (Closes issue #18498, #18486. Reporte...

I current have a working Asterisk 1.2.12 server with ODBC voicemail storage, realtime static maps for voicemail, sip and iax configuration files. Realtime extensions, etc. All works great. I have verified that this configuration works on my test server as well. Now I am trying to test the 1.4B3 version on the same test server, and all works well except for ODBC voicemail. I am using the same